diff options
| author | Linus Torvalds <torvalds@linux-foundation.org> | 2011-10-28 14:25:01 -0700 |
|---|---|---|
| committer | Linus Torvalds <torvalds@linux-foundation.org> | 2011-10-28 14:25:01 -0700 |
| commit | 68d99b2c8efcb6ed3807a55569300c53b5f88be5 (patch) | |
| tree | f189c8f2132d3668a2f0e503f5c3f8695b26a1c8 /sound | |
| parent | 0e59e7e7feb5a12938fbf9135147eeda3238c6c4 (diff) | |
| parent | 8128c9f21509f9a8b6da94ac432d845dda458406 (diff) | |
| download | linux-stable-68d99b2c8efcb6ed3807a55569300c53b5f88be5.tar.gz linux-stable-68d99b2c8efcb6ed3807a55569300c53b5f88be5.tar.bz2 linux-stable-68d99b2c8efcb6ed3807a55569300c53b5f88be5.zip | |
Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (549 commits)
ALSA: hda - Fix ADC input-amp handling for Cx20549 codec
ALSA: hda - Keep EAPD turned on for old Conexant chips
ALSA: hda/realtek - Fix missing volume controls with ALC260
ASoC: wm8940: Properly set codec->dapm.bias_level
ALSA: hda - Fix pin-config for ASUS W90V
ALSA: hda - Fix surround/CLFE headphone and speaker pins order
ALSA: hda - Fix typo
ALSA: Update the sound git tree URL
ALSA: HDA: Add new revision for ALC662
ASoC: max98095: Convert codec->hw_write to snd_soc_write
ASoC: keep pointer to resource so it can be freed
ASoC: sgtl5000: Fix wrong mask in some snd_soc_update_bits calls
ASoC: wm8996: Fix wrong mask for setting WM8996_AIF_CLOCKING_2
ASoC: da7210: Add support for line out and DAC
ASoC: da7210: Add support for DAPM
ALSA: hda/realtek - Fix DAC assignments of multiple speakers
ASoC: Use SGTL5000_LINREG_VDDD_MASK instead of hardcoded mask value
ASoC: Set sgtl5000->ldo in ldo_regulator_register
ASoC: wm8996: Use SND_SOC_DAPM_AIF_OUT for AIF2 Capture
ASoC: wm8994: Use SND_SOC_DAPM_AIF_OUT for AIF3 Capture
...
Diffstat (limited to 'sound')
334 files changed, 24703 insertions, 11926 deletions
diff --git a/sound/aoa/codecs/onyx.c b/sound/aoa/codecs/onyx.c index 3687a6cc9881..762af68c8996 100644 --- a/sound/aoa/codecs/onyx.c +++ b/sound/aoa/codecs/onyx.c @@ -1067,7 +1067,6 @@ static int onyx_i2c_probe(struct i2c_client *client, printk(KERN_DEBUG PFX "created and attached onyx instance\n"); return 0; fail: - i2c_set_clientdata(client, NULL); kfree(onyx); return -ENODEV; } @@ -1112,8 +1111,7 @@ static int onyx_i2c_remove(struct i2c_client *client) aoa_codec_unregister(&onyx->codec); of_node_put(onyx->codec.node); - if (onyx->codec_info) - kfree(onyx->codec_info); + kfree(onyx->codec_info); kfree(onyx); return 0; } diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c index d0cead38d5fb..e518d38b1c74 100644 --- a/sound/arm/aaci.c +++ b/sound/arm/aaci.c @@ -443,7 +443,7 @@ static int aaci_pcm_open(struct snd_pcm_substream *substream) mutex_lock(&aaci->irq_lock); if (!aaci->users++) { ret = request_irq(aaci->dev->irq[0], aaci_irq, - IRQF_SHARED | IRQF_DISABLED, DRIVER_NAME, aaci); + IRQF_SHARED, DRIVER_NAME, aaci); if (ret != 0) aaci->users--; } diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c index 88eec3847df2..8ad65352bf91 100644 --- a/sound/arm/pxa2xx-ac97-lib.c +++ b/sound/arm/pxa2xx-ac97-lib.c @@ -359,7 +359,7 @@ int __devinit pxa2xx_ac97_hw_probe(struct platform_device *dev) if (ret) goto err_clk2; - ret = request_irq(IRQ_AC97, pxa2xx_ac97_irq, IRQF_DISABLED, "AC97", NULL); + ret = request_irq(IRQ_AC97, pxa2xx_ac97_irq, 0, "AC97", NULL); if (ret < 0) goto err_irq; diff --git a/sound/core/control.c b/sound/core/control.c index f8c5be464510..978fe1a8e9f0 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -989,7 +989,6 @@ struct user_element { void *tlv_data; /* TLV data */ unsigned long tlv_data_size; /* TLV data size */ void *priv_data; /* private data (like strings for enumerated type) */ - unsigned long priv_data_size; /* size of private data in bytes */ }; static int snd_ctl_elem_user_info(struct snd_kcontrol *kcontrol, @@ -1001,6 +1000,28 @@ static int snd_ctl_elem_user_info(struct snd_kcontrol *kcontrol, return 0; } +static int snd_ctl_elem_user_enum_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct user_element *ue = kcontrol->private_data; + const char *names; + unsigned int item; + + item = uinfo->value.enumerated.item; + + *uinfo = ue->info; + + item = min(item, uinfo->value.enumerated.items - 1); + uinfo->value.enumerated.item = item; + + names = ue->priv_data; + for (; item > 0; --item) + names += strlen(names) + 1; + strcpy(uinfo->value.enumerated.name, names); + + return 0; +} + static int snd_ctl_elem_user_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -1055,11 +1076,46 @@ static int snd_ctl_elem_user_tlv(struct snd_kcontrol *kcontrol, return change; } +static int snd_ctl_elem_init_enum_names(struct user_element *ue) +{ + char *names, *p; + size_t buf_len, name_len; + unsigned int i; + + if (ue->info.value.enumerated.names_length > 64 * 1024) + return -EINVAL; + + names = memdup_user( + (const void __user *)ue->info.value.enumerated.names_ptr, + ue->info.value.enumerated.names_length); + if (IS_ERR(names)) + return PTR_ERR(names); + + /* check that there are enough valid names */ + buf_len = ue->info.value.enumerated.names_length; + p = names; + for (i = 0; i < ue->info.value.enumerated.items; ++i) { + name_len = strnlen(p, buf_len); + if (name_len == 0 || name_len >= 64 || name_len == buf_len) { + kfree(names); + return -EINVAL; + } + p += name_len + 1; + buf_len -= name_len + 1; + } + + ue->priv_data = names; + ue->info.value.enumerated.names_ptr = 0; + + return 0; +} + static void snd_ctl_elem_user_free(struct snd_kcontrol *kcontrol) { struct user_element *ue = kcontrol->private_data; - if (ue->tlv_data) - kfree(ue->tlv_data); + + kfree(ue->tlv_data); + kfree(ue->priv_data); kfree(ue); } @@ -1072,8 +1128,8 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file, long private_size; struct user_element *ue; int idx, err; - - if (card->user_ctl_count >= MAX_USER_CONTROLS) + + if (!replace && card->user_ctl_count >= MAX_USER_CONTROLS) return -ENOMEM; if (info->count < 1) return -EINVAL; @@ -1101,7 +1157,10 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file, memcpy(&kctl.id, &info->id, sizeof(info->id)); kctl.count = info->owner ? info->owner : 1; access |= SNDRV_CTL_ELEM_ACCESS_USER; - kctl.info = snd_ctl_elem_user_info; + if (info->type == SNDRV_CTL_ELEM_TYPE_ENUMERATED) + kctl.info = snd_ctl_elem_user_enum_info; + else + kctl.info = snd_ctl_elem_user_info; if (access & SNDRV_CTL_ELEM_ACCESS_READ) kctl.get = snd_ctl_elem_user_get; if (access & SNDRV_CTL_ELEM_ACCESS_WRITE) @@ -1122,6 +1181,11 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file, if (info->count > 64) return -EINVAL; break; + case SNDRV_CTL_ELEM_TYPE_ENUMERATED: + private_size = sizeof(unsigned int); + if (info->count > 128 || info->value.enumerated.items == 0) + return -EINVAL; + break; case SNDRV_CTL_ELEM_TYPE_BYTES: private_size = sizeof(unsigned char); if (info->count > 512) @@ -1143,9 +1207,17 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file, ue->info.access = 0; ue->elem_data = (char *)ue + sizeof(*ue); ue->elem_data_size = private_size; + if (ue->info.type == SNDRV_CTL_ELEM_TYPE_ENUMERATED) { + err = snd_ctl_elem_init_enum_names(ue); + if (err < 0) { + kfree(ue); + return err; + } + } kctl.private_free = snd_ctl_elem_user_free; _kctl = snd_ctl_new(&kctl, access); if (_kctl == NULL) { + kfree(ue->priv_data); kfree(ue); return -ENOMEM; } diff --git a/sound/core/control_compat.c b/sound/core/control_compat.c index 426874429a5e..2bb95a7a8809 100644 --- a/sound/core/control_compat.c +++ b/sound/core/control_compat.c @@ -83,6 +83,8 @@ struct snd_ctl_elem_info32 { u32 items; u32 item; char name[64]; + u64 names_ptr; + u32 names_length; } enumerated; unsigned char reserved[128]; } value; @@ -372,6 +374,8 @@ static int snd_ctl_elem_add_compat(struct snd_ctl_file *file, &data32->value.enumerated, sizeof(data->value.enumerated))) goto error; + data->value.enumerated.names_ptr = + (uintptr_t)compat_ptr(data->value.enumerated.names_ptr); break; default: break; diff --git a/sound/core/jack.c b/sound/core/jack.c index 53b53e97c896..240a3e13470d 100644 --- a/sound/core/jack.c +++ b/sound/core/jack.c @@ -30,6 +30,7 @@ static int jack_switch_types[] = { SW_LINEOUT_INSERT, SW_JACK_PHYSICAL_INSERT, SW_VIDEOOUT_INSERT, + SW_LINEIN_INSERT, }; static int snd_jack_dev_free(struct snd_device *device) diff --git a/sound/core/oss/mixer_oss.c b/sound/core/oss/mixer_oss.c index d8359cfeca15..1b5e0c49a0ad 100644 --- a/sound/core/oss/mixer_oss.c +++ b/sound/core/oss/mixer_oss.c @@ -499,7 +499,7 @@ static struct snd_kcontrol *snd_mixer_oss_test_id(struct snd_mixer_oss *mixer, c memset(&id, 0, sizeof(id)); id.iface = SNDRV_CTL_ELEM_IFACE_MIXER; - strcpy(id.name, name); + strlcpy(id.name, name, sizeof(id.name)); id.index = index; return snd_ctl_find_id(card, &id); } diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 62e90b862a0d..95d1e789715f 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1399,6 +1399,32 @@ int snd_pcm_hw_constraint_pow2(struct snd_pcm_runtime *runtime, EXPORT_SYMBOL(snd_pcm_hw_constraint_pow2); +static int snd_pcm_hw_rule_noresample_func(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + unsigned int base_rate = (unsigned int)(uintptr_t)rule->private; + struct snd_interval *rate; + + rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + return snd_interval_list(rate, 1, &base_rate, 0); +} + +/** + * snd_pcm_hw_rule_noresample - add a rule to allow disabling hw resampling + * @runtime: PCM runtime instance + * @base_rate: the rate at which the hardware does not resample + */ +int snd_pcm_hw_rule_noresample(struct snd_pcm_runtime *runtime, + unsigned int base_rate) +{ + return snd_pcm_hw_rule_add(runtime, SNDRV_PCM_HW_PARAMS_NORESAMPLE, + SNDRV_PCM_HW_PARAM_RATE, + snd_pcm_hw_rule_noresample_func, + (void *)(uintptr_t)base_rate, + SNDRV_PCM_HW_PARAM_RATE, -1); +} +EXPORT_SYMBOL(snd_pcm_hw_rule_noresample); + static void _snd_pcm_hw_param_any(struct snd_pcm_hw_params *params, snd_pcm_hw_param_t var) { diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index c74e228731ed..d7d2179c0363 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -2058,16 +2058,12 @@ EXPORT_SYMBOL(snd_pcm_open_substream); static int snd_pcm_open_file(struct file *file, struct snd_pcm *pcm, - int stream, - struct snd_pcm_file **rpcm_file) + int stream) { struct snd_pcm_file *pcm_file; struct snd_pcm_substream *substream; int err; - if (rpcm_file) - *rpcm_file = NULL; - err = snd_pcm_open_substream(pcm, stream, file, &substream); if (err < 0) return err; @@ -2083,8 +2079,7 @@ static int snd_pcm_open_file(struct file *file, substream->pcm_release = pcm_release_private; } file->private_data = pcm_file; - if (rpcm_file) - *rpcm_file = pcm_file; + return 0; } @@ -2113,7 +2108,6 @@ static int snd_pcm_capture_open(struct inode *inode, struct file *file) static int snd_pcm_open(struct file *file, struct snd_pcm *pcm, int stream) { int err; - struct snd_pcm_file *pcm_file; wait_queue_t wait; if (pcm == NULL) { @@ -2131,7 +2125,7 @@ static int snd_pcm_open(struct file *file, struct snd_pcm *pcm, int stream) add_wait_queue(&pcm->open_wait, &wait); mutex_lock(&pcm->open_mutex); while (1) { - err = snd_pcm_open_file(file, pcm, stream, &pcm_file); + err = snd_pcm_open_file(file, pcm, stream); if (err >= 0) break; if (err == -EAGAIN) { @@ -3156,8 +3150,8 @@ static const struct vm_operations_struct snd_pcm_vm_ops_data_fault = { /* * mmap the DMA buffer on RAM */ -static int snd_pcm_default_mmap(struct snd_pcm_substream *substream, - struct vm_area_struct *area) +int snd_pcm_lib_default_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *area) { area->vm_flags |= VM_RESERVED; #ifdef ARCH_HAS_DMA_MMAP_COHERENT @@ -3177,6 +3171,7 @@ static int snd_pcm_default_mmap(struct snd_pcm_substream *substream, area->vm_ops = &snd_pcm_vm_ops_data_fault; return 0; } +EXPORT_SYMBOL_GPL(snd_pcm_lib_default_mmap); /* * mmap the DMA buffer on I/O memory area @@ -3242,7 +3237,7 @@ int snd_pcm_mmap_data(struct snd_pcm_substream *substream, struct file *file, if (substream->ops->mmap) err = substream->ops->mmap(substream, area); else - err = snd_pcm_default_mmap(substream, area); + err = snd_pcm_lib_default_mmap(substream, area); if (!err) atomic_inc(&substream->mmap_count); return err; diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index a0da7755fcea..4067f1548949 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -575,7 +575,8 @@ static void loopback_runtime_free(struct snd_pcm_runtime *runtime) static int loopback_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { - return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); + return snd_pcm_lib_alloc_vmalloc_buffer(substream, + params_buffer_bytes(params)); } static int loopback_hw_free(struct snd_pcm_substream *substream) @@ -587,7 +588,7 @@ static int loopback_hw_free(struct snd_pcm_substream *substream) mutex_lock(&dpcm->loopback->cable_lock); cable->valid &= ~(1 << substream->stream); mutex_unlock(&dpcm->loopback->cable_lock); - return snd_pcm_lib_free_pages(substream); + return snd_pcm_lib_free_vmalloc_buffer(substream); } static unsigned int get_cable_index(struct snd_pcm_substream *substream) @@ -740,6 +741,8 @@ static struct snd_pcm_ops loopback_playback_ops = { .prepare = loopback_prepare, .trigger = loopback_trigger, .pointer = loopback_pointer, + .page = snd_pcm_lib_get_vmalloc_page, + .mmap = snd_pcm_lib_mmap_vmalloc, }; static struct snd_pcm_ops loopback_capture_ops = { @@ -751,6 +754,8 @@ static struct snd_pcm_ops loopback_capture_ops = { .prepare = loopback_prepare, .trigger = loopback_trigger, .pointer = loopback_pointer, + .page = snd_pcm_lib_get_vmalloc_page, + .mmap = snd_pcm_lib_mmap_vmalloc, }; static int __devinit loopback_pcm_new(struct loopback *loopback, @@ -771,10 +776,6 @@ static int __devinit loopback_pcm_new(struct loopback *loopback, strcpy(pcm->name, "Loopback PCM"); loopback->pcm[device] = pcm; - - snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_CONTINUOUS, - snd_dma_continuous_data(GFP_KERNEL), - 0, 2 * 1024 * 1024); return 0; } diff --git a/sound/drivers/ml403-ac97cr.c b/sound/drivers/ml403-ac97cr.c index 5cfcb908c430..2c7a7636f472 100644 --- a/sound/drivers/ml403-ac97cr.c +++ b/sound/drivers/ml403-ac97cr.c @@ -1153,7 +1153,7 @@ snd_ml403_ac97cr_create(struct snd_card *card, struct platform_device *pfdev, "0x%x done\n", (unsigned int)ml403_ac97cr->port); /* get irq */ irq = platform_get_irq(pfdev, 0); - if (request_irq(irq, snd_ml403_ac97cr_irq, IRQF_DISABLED, + if (request_irq(irq, snd_ml403_ac97cr_irq, 0, dev_name(&pfdev->dev), (void *)ml403_ac97cr)) { snd_printk(KERN_ERR SND_ML403_AC97CR_DRIVER ": " "unable to grab IRQ %d\n", @@ -1166,7 +1166,7 @@ snd_ml403_ac97cr_create(struct snd_card *card, struct platform_device *pfdev, "request (playback) irq %d done\n", ml403_ac97cr->irq); irq = platform_get_irq(pfdev, 1); - if (request_irq(irq, snd_ml403_ac97cr_irq, IRQF_DISABLED, + if (request_irq(irq, snd_ml403_ac97cr_irq, 0, dev_name(&pfdev->dev), (void *)ml403_ac97cr)) { snd_printk(KERN_ERR SND_ML403_AC97CR_DRIVER ": " "unable to grab IRQ %d\n", diff --git a/sound/drivers/mpu401/mpu401.c b/sound/drivers/mpu401/mpu401.c index 149d05a8202d..1c02852aceea 100644 --- a/sound/drivers/mpu401/mpu401.c +++ b/sound/drivers/mpu401/mpu401.c @@ -86,8 +86,7 @@ static int snd_mpu401_create(int dev, struct snd_card **rcard) } err = snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, port[dev], 0, - irq[dev], irq[dev] >= 0 ? IRQF_DISABLED : 0, - NULL); + irq[dev], NULL); if (err < 0) { printk(KERN_ERR "MPU401 not detected at 0x%lx\n", port[dev]); goto _err; diff --git a/sound/drivers/mpu401/mpu401_uart.c b/sound/drivers/mpu401/mpu401_uart.c index 2af09996a3d0..e91698a634b2 100644 --- a/sound/drivers/mpu401/mpu401_uart.c +++ b/sound/drivers/mpu401/mpu401_uart.c @@ -3,7 +3,7 @@ * Routines for control of MPU-401 in UART mode * * MPU-401 supports UART mode which is not capable generate transmit - * interrupts thus output is done via polling. Also, if irq < 0, then + * interrupts thus output is done via polling. Without interrupt, * input is done also via polling. Do not expect good performance. * * @@ -374,7 +374,7 @@ snd_mpu401_uart_input_trigger(struct snd_rawmidi_substream *substream, int up) /* first time - flush FIFO */ while (max-- > 0) mpu->read(mpu, MPU401D(mpu)); - if (mpu->irq < 0) + if (mpu->info_flags & MPU401_INFO_USE_TIMER) snd_mpu401_uart_add_timer(mpu, 1); } @@ -383,7 +383,7 @@ snd_mpu401_uart_input_trigger(struct snd_rawmidi_substream *substream, int up) snd_mpu401_uart_input_read(mpu); spin_unlock_irqrestore(&mpu->input_lock, flags); } else { - if (mpu->irq < 0) + if (mpu->info_flags & MPU401_INFO_USE_TIMER) snd_mpu401_uart_remove_timer(mpu, 1); clear_bit(MPU401_MODE_BIT_INPUT_TRIGGER, &mpu->mode); } @@ -496,7 +496,7 @@ static struct snd_rawmidi_ops snd_mpu401_uart_input = static void snd_mpu401_uart_free(struct snd_rawmidi *rmidi) { struct snd_mpu401 *mpu = rmidi->private_data; - if (mpu->irq_flags && mpu->irq >= 0) + if (mpu->irq >= 0) free_irq(mpu->irq, (void *) mpu); release_and_free_resource(mpu->res); kfree(mpu); @@ -509,8 +509,7 @@ static void snd_mpu401_uart_free(struct snd_rawmidi *rmidi) * @hardware: the hardware type, MPU401_HW_XXXX * @port: the base address of MPU401 port * @info_flags: bitflags MPU401_INFO_XXX - * @irq: the irq number, -1 if no interrupt for mpu - * @irq_flags: the irq request flags (SA_XXX), 0 if irq was already reserved. + * @irq: the ISA irq number, -1 if not to be allocated * @rrawmidi: the pointer to store the new rawmidi instance * * Creates a new MPU-401 instance. @@ -525,7 +524,7 @@ int snd_mpu401_uart_new(struct snd_card *card, int device, unsigned short hardware, unsigned long port, unsigned int info_flags, - int irq, int irq_flags, + int irq, struct snd_rawmidi ** rrawmidi) { struct snd_mpu401 *mpu; @@ -577,8 +576,8 @@ int snd_mpu401_uart_new(struct snd_card *card, int device, mpu->cport = port + 2; else mpu->cport = port + 1; - if (irq >= 0 && irq_flags) { - if (request_irq(irq, snd_mpu401_uart_interrupt, irq_flags, + if (irq >= 0) { + if (request_irq(irq, snd_mpu401_uart_interrupt, 0, "MPU401 UART", (void *) mpu)) { snd_printk(KERN_ERR "mpu401_uart: " "unable to grab IRQ %d\n", irq); @@ -586,9 +585,10 @@ int snd_mpu401_uart_new(struct snd_card *card, int device, return -EBUSY; } } + if (irq < 0 && !(info_flags & MPU401_INFO_IRQ_HOOK)) + info_flags |= MPU401_INFO_USE_TIMER; mpu->info_flags = info_flags; mpu->irq = irq; - mpu->irq_flags = irq_flags; if (card->shortname[0]) snprintf(rmidi->name, sizeof(rmidi->name), "%s MIDI", card->shortname); diff --git a/sound/drivers/mtpav.c b/sound/drivers/mtpav.c index 5c426df87678..1eef4ccebe4b 100644 --- a/sound/drivers/mtpav.c +++ b/sound/drivers/mtpav.c @@ -589,7 +589,7 @@ static int __devinit snd_mtpav_get_ISA(struct mtpav * mcard) return -EBUSY; } mcard->port = port; - if (request_irq(irq, snd_mtpav_irqh, IRQF_DISABLED, "MOTU MTPAV", mcard)) { + if (request_irq(irq, snd_mtpav_irqh, 0, "MOTU MTPAV", mcard)) { snd_printk(KERN_ERR "MTVAP IRQ %d busy\n", irq); return -EBUSY; } diff --git a/sound/drivers/serial-u16550.c b/sound/drivers/serial-u16550.c index a25fb7b1f441..fc1d822802c3 100644 --- a/sound/drivers/serial-u16550.c +++ b/sound/drivers/serial-u16550.c @@ -816,7 +816,7 @@ static int __devinit snd_uart16550_create(struct snd_card *card, if (irq >= 0 && irq != SNDRV_AUTO_IRQ) { if (request_irq(irq, snd_uart16550_interrupt, - IRQF_DISABLED, "Serial MIDI", uart)) { + 0, "Serial MIDI", uart)) { snd_printk(KERN_WARNING "irq %d busy. Using Polling.\n", irq); } else { diff --git a/sound/firewire/isight.c b/sound/firewire/isight.c index 440030818db7..cd094ecaca3b 100644 --- a/sound/firewire/isight.c +++ b/sound/firewire/isight.c @@ -51,7 +51,6 @@ struct isight { struct fw_unit *unit; struct fw_device *device; u64 audio_base; - struct fw_address_handler iris_handler; struct snd_pcm_substream *pcm; struct mutex mutex; struct iso_packets_buffer buffer; diff --git a/sound/firewire/speakers.c b/sound/firewire/speakers.c index 3fc257da180c..cbe6bb9e53b6 100644 --- a/sound/firewire/speakers.c +++ b/sound/firewire/speakers.c @@ -778,9 +778,10 @@ static int __devexit fwspk_remove(struct device *dev) { struct fwspk *fwspk = dev_get_drvdata(dev); - mutex_lock(&fwspk->mutex); amdtp_out_stream_pcm_abort(&fwspk->stream); snd_card_disconnect(fwspk->card); + + mutex_lock(&fwspk->mutex); fwspk_stop_stream(fwspk); mutex_unlock(&fwspk->mutex); @@ -796,8 +797,8 @@ static void fwspk_bus_reset(struct fw_unit *unit) fcp_bus_reset(fwspk->unit); if (cmp_connection_update(&fwspk->connection) < 0) { - mutex_lock(&fwspk->mutex); amdtp_out_stream_pcm_abort(&fwspk->stream); + mutex_lock(&fwspk->mutex); fwspk_stop_stream(fwspk); mutex_unlock(&fwspk->mutex); return; diff --git a/sound/isa/ad1816a/ad1816a.c b/sound/isa/ad1816a/ad1816a.c index 3cb75bc97699..a87a2b566e19 100644 --- a/sound/isa/ad1816a/ad1816a.c +++ b/sound/isa/ad1816a/ad1816a.c @@ -204,7 +204,7 @@ static int __devinit snd_card_ad1816a_probe(int dev, struct pnp_card_link *pcard if (mpu_port[dev] > 0) { if (snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, - mpu_port[dev], 0, mpu_irq[dev], IRQF_DISABLED, + mpu_port[dev], 0, mpu_irq[dev], NULL) < 0) printk(KERN_ERR PFX "no MPU-401 device at 0x%lx.\n", mpu_port[dev]); } diff --git a/sound/isa/ad1816a/ad1816a_lib.c b/sound/isa/ad1816a/ad1816a_lib.c index 05aef8b97e96..177eed3271bc 100644 --- a/sound/isa/ad1816a/ad1816a_lib.c +++ b/sound/isa/ad1816a/ad1816a_lib.c @@ -595,7 +595,7 @@ int __devinit snd_ad1816a_create(struct snd_card *card, snd_ad1816a_free(chip); return -EBUSY; } - if (request_irq(irq, snd_ad1816a_interrupt, IRQF_DISABLED, "AD1816A", (void *) chip)) { + if (request_irq(irq, snd_ad1816a_interrupt, 0, "AD1816A", (void *) chip)) { snd_printk(KERN_ERR "ad1816a: can't grab IRQ %d\n", irq); snd_ad1816a_free(chip); return -EBUSY; diff --git a/sound/isa/als100.c b/sound/isa/als100.c index 20becc89f6f6..706effd6b3cd 100644 --- a/sound/isa/als100.c +++ b/sound/isa/als100.c @@ -256,7 +256,6 @@ static int __devinit snd_card_als100_probe(int dev, mpu_type, mpu_port[dev], 0, mpu_irq[dev], - mpu_irq[dev] >= 0 ? IRQF_DISABLED : 0, NULL) < 0) snd_printk(KERN_ERR PFX "no MPU-401 device at 0x%lx\n", mpu_port[dev]); } diff --git a/sound/isa/azt2320.c b/sound/isa/azt2320.c index aac8dc15c2fe..b7bdbf307740 100644 --- a/sound/isa/azt2320.c +++ b/sound/isa/azt2320.c @@ -234,8 +234,7 @@ static int __devinit snd_card_azt2320_probe(int dev, if (mpu_port[dev] > 0 && mpu_port[dev] != SNDRV_AUTO_PORT) { if (snd_mpu401_uart_new(card, 0, MPU401_HW_AZT2320, mpu_port[dev], 0, - mpu_irq[dev], IRQF_DISABLED, - NULL) < 0) + mpu_irq[dev], NULL) < 0) snd_printk(KERN_ERR PFX "no MPU-401 device at 0x%lx\n", mpu_port[dev]); } diff --git a/sound/isa/cmi8330.c b/sound/isa/cmi8330.c index fe79a169acb5..dca69f80305f 100644 --- a/sound/isa/cmi8330.c +++ b/sound/isa/cmi8330.c @@ -597,7 +597,7 @@ static int __devinit snd_cmi8330_probe(struct snd_card *card, int dev) if (mpuport[dev] != SNDRV_AUTO_PORT) { if (snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, mpuport[dev], 0, mpuirq[dev], - IRQF_DISABLED, NULL) < 0) + NULL) < 0) printk(KERN_ERR PFX "no MPU-401 device at 0x%lx.\n", mpuport[dev]); } diff --git a/sound/isa/cs423x/cs4231.c b/sound/isa/cs423x/cs4231.c index cb9153e75b82..409fa0ad7843 100644 --- a/sound/isa/cs423x/cs4231.c +++ b/sound/isa/cs423x/cs4231.c @@ -131,7 +131,6 @@ static int __devinit snd_cs4231_probe(struct device *dev, unsigned int n) mpu_irq[n] = -1; if (snd_mpu401_uart_new(card, 0, MPU401_HW_CS4232, mpu_port[n], 0, mpu_irq[n], - mpu_irq[n] >= 0 ? IRQF_DISABLED : 0, NULL) < 0) dev_warn(dev, "MPU401 not detected\n"); } diff --git a/sound/isa/cs423x/cs4236.c b/sound/isa/cs423x/cs4236.c index 999dc1e0fdbd..0dbde461e6c1 100644 --- a/sound/isa/cs423x/cs4236.c +++ b/sound/isa/cs423x/cs4236.c @@ -449,8 +449,7 @@ static int __devinit snd_cs423x_probe(struct snd_card *card, int dev) mpu_irq[dev] = -1; if (snd_mpu401_uart_new(card, 0, MPU401_HW_CS4232, mpu_port[dev], 0, - mpu_irq[dev], - mpu_irq[dev] >= 0 ? IRQF_DISABLED : 0, NULL) < 0) + mpu_irq[dev], NULL) < 0) printk(KERN_WARNING IDENT ": MPU401 not detected\n"); } diff --git a/sound/isa/es1688/es1688.c b/sound/isa/es1688/es1688.c index 0cde8131a575..5493e9e4bcd5 100644 --- a/sound/isa/es1688/es1688.c +++ b/sound/isa/es1688/es1688.c @@ -174,7 +174,7 @@ static int __devinit snd_es1688_probe(struct snd_card *card, unsigned int n) chip->mpu_port > 0) { error = snd_mpu401_uart_new(card, 0, MPU401_HW_ES1688, chip->mpu_port, 0, - mpu_irq[n], IRQF_DISABLED, NULL); + mpu_irq[n], NULL); if (error < 0) return error; } diff --git a/sound/isa/es1688/es1688_lib.c b/sound/isa/es1688/es1688_lib.c index 07676200496a..d3eab6fb0866 100644 --- a/sound/isa/es1688/es1688_lib.c +++ b/sound/isa/es1688/es1688_lib.c @@ -661,7 +661,7 @@ int snd_es1688_create(struct snd_card *card, snd_printk(KERN_ERR "es1688: can't grab port 0x%lx\n", port + 4); return -EBUSY; } - if (request_irq(irq, snd_es1688_interrupt, IRQF_DISABLED, "ES1688", (void *) chip)) { + if (request_irq(irq, snd_es1688_interrupt, 0, "ES1688", (void *) chip)) { snd_printk(KERN_ERR "es1688: can't grab IRQ %d\n", irq); return -EBUSY; } diff --git a/sound/isa/es18xx.c b/sound/isa/es18xx.c index fb4d6b34bbca..bf6ad0bf51c6 100644 --- a/sound/isa/es18xx.c +++ b/sound/isa/es18xx.c @@ -1805,7 +1805,7 @@ static int __devinit snd_es18xx_new_device(struct snd_card *card, return -EBUSY; } - if (request_irq(irq, snd_es18xx_interrupt, IRQF_DISABLED, "ES18xx", + if (request_irq(irq, snd_es18xx_interrupt, 0, "ES18xx", (void *) card)) { snd_es18xx_free(card); snd_printk(KERN_ERR PFX "unable to grap IRQ %d\n", irq); @@ -2160,8 +2160,8 @@ static int __devinit snd_audiodrive_probe(struct snd_card *card, int dev) if (mpu_port[dev] > 0 && mpu_port[dev] != SNDRV_AUTO_PORT) { err = snd_mpu401_uart_new(card, 0, MPU401_HW_ES18XX, - mpu_port[dev], 0, - irq[dev], 0, &chip->rmidi); + mpu_port[dev], MPU401_INFO_IRQ_HOOK, + -1, &chip->rmidi); if (err < 0) return err; } diff --git a/sound/isa/galaxy/galaxy.c b/sound/isa/galaxy/galaxy.c index ee54df082b9c..e51d3244742a 100644 --- a/sound/isa/galaxy/galaxy.c +++ b/sound/isa/galaxy/galaxy.c @@ -585,8 +585,7 @@ static int __devinit snd_galaxy_probe(struct device *dev, unsigned int n) if (mpu_port[n] >= 0) { err = snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, - mpu_port[n], 0, mpu_irq[n], - IRQF_DISABLED, NULL); + mpu_port[n], 0, mpu_irq[n], NULL); if (err < 0) goto error; } diff --git a/sound/isa/gus/gus_main.c b/sound/isa/gus/gus_main.c index 12eb98f2f931..3167e5ac3699 100644 --- a/sound/isa/gus/gus_main.c +++ b/sound/isa/gus/gus_main.c @@ -180,7 +180,7 @@ int snd_gus_create(struct snd_card *card, snd_gus_free(gus); return -EBUSY; } - if (irq >= 0 && request_irq(irq, snd_gus_interrupt, IRQF_DISABLED, "GUS GF1", (void *) gus)) { + if (irq >= 0 && request_irq(irq, snd_gus_interrupt, 0, "GUS GF1", (void *) gus)) { snd_printk(KERN_ERR "gus: can't grab irq %d\n", irq); snd_gus_free(gus); return -EBUSY; diff --git a/sound/isa/gus/gusextreme.c b/sound/isa/gus/gusextreme.c index 008e8e5bfa37..c4733c08b60b 100644 --- a/sound/isa/gus/gusextreme.c +++ b/sound/isa/gus/gusextreme.c @@ -317,8 +317,7 @@ static int __devinit snd_gusextreme_probe(struct device *dev, unsigned int n) if (es1688->mpu_port >= 0x300) { error = snd_mpu401_uart_new(card, 0, MPU401_HW_ES1688, - es1688->mpu_port, 0, - mpu_irq[n], IRQF_DISABLED, NULL); + es1688->mpu_port, 0, mpu_irq[n], NULL); if (error < 0) goto out; } diff --git a/sound/isa/gus/gusmax.c b/sound/isa/gus/gusmax.c index 3e4a58b72913..c43faa057ff6 100644 --- a/sound/isa/gus/gusmax.c +++ b/sound/isa/gus/gusmax.c @@ -291,7 +291,7 @@ static int __devinit snd_gusmax_probe(struct device *pdev, unsigned int dev) goto _err; } - if (request_irq(xirq, snd_gusmax_interrupt, IRQF_DISABLED, "GUS MAX", (void *)maxcard)) { + if (request_irq(xirq, snd_gusmax_interrupt, 0, "GUS MAX", (void *)maxcard)) { snd_printk(KERN_ERR PFX "unable to grab IRQ %d\n", xirq); err = -EBUSY; goto _err; diff --git a/sound/isa/gus/interwave.c b/sound/isa/gus/interwave.c index c7b80e4730fc..5f869a32b48c 100644 --- a/sound/isa/gus/interwave.c +++ b/sound/isa/gus/interwave.c @@ -684,7 +684,7 @@ static int __devinit snd_interwave_probe(struct snd_card *card, int dev) if ((err = snd_gus_initialize(gus)) < 0) return err; - if (request_irq(xirq, snd_interwave_interrupt, IRQF_DISABLED, + if (request_irq(xirq, snd_interwave_interrupt, 0, "InterWave", iwcard)) { snd_printk(KERN_ERR PFX "unable to grab IRQ %d\n", xirq); return -EBUSY; diff --git a/sound/isa/msnd/msnd_pinnacle.c b/sound/isa/msnd/msnd_pinnacle.c index 91d6023a63e5..0961e2cf20ca 100644 --- a/sound/isa/msnd/msnd_pinnacle.c +++ b/sound/isa/msnd/msnd_pinnacle.c @@ -600,7 +600,7 @@ static int __devinit snd_msnd_attach(struct snd_card *card) mpu_io[0], MPU401_MODE_INPUT | MPU401_MODE_OUTPUT, - mpu_irq[0], IRQF_DISABLED, + mpu_irq[0], &chip->rmidi); if (err < 0) { printk(KERN_ERR LOGNAME diff --git a/sound/isa/opl3sa2.c b/sound/isa/opl3sa2.c index 9b915e27b5bd..bbafb0b543ea 100644 --- a/sound/isa/opl3sa2.c +++ b/sound/isa/opl3sa2.c @@ -667,7 +667,7 @@ static int __devinit snd_opl3sa2_probe(struct snd_card *card, int dev) err = snd_opl3sa2_detect(card); if (err < 0) return err; - err = request_irq(xirq, snd_opl3sa2_interrupt, IRQF_DISABLED, + err = request_irq(xirq, snd_opl3sa2_interrupt, 0, "OPL3-SA2", card); if (err) { snd_printk(KERN_ERR PFX "can't grab IRQ %d\n", xirq); @@ -707,8 +707,9 @@ static int __devinit snd_opl3sa2_probe(struct snd_card *card, int dev) } if (midi_port[dev] >= 0x300 && midi_port[dev] < 0x340) { if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_OPL3SA2, - midi_port[dev], 0, - xirq, 0, &chip->rmidi)) < 0) + midi_port[dev], + MPU401_INFO_IRQ_HOOK, -1, + &chip->rmidi)) < 0) return err; } sprintf(card->longname, "%s at 0x%lx, irq %d, dma %d", diff --git a/sound/isa/opti9xx/miro.c b/sound/isa/opti9xx/miro.c index 8c24102d0d93..d94d0f35cb76 100644 --- a/sound/isa/opti9xx/miro.c +++ b/sound/isa/opti9xx/miro.c @@ -1377,8 +1377,7 @@ static int __devinit snd_miro_probe(struct snd_card *card) rmidi = NULL; else { error = snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, - mpu_port, 0, miro->mpu_irq, IRQF_DISABLED, - &rmidi); + mpu_port, 0, miro->mpu_irq, &rmidi); if (error < 0) snd_printk(KERN_WARNING "no MPU-401 device at 0x%lx?\n", mpu_port); diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index c35dc68930dc..6dbbfa76b440 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -892,7 +892,7 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card) #endif #ifdef OPTi93X error = request_irq(irq, snd_opti93x_interrupt, - IRQF_DISABLED, DEV_NAME" - WSS", chip); + 0, DEV_NAME" - WSS", chip); if (error < 0) { snd_printk(KERN_ERR "opti9xx: can't grab IRQ %d\n", irq); return error; @@ -914,7 +914,7 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card) rmidi = NULL; else { error = snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, - mpu_port, 0, mpu_irq, IRQF_DISABLED, &rmidi); + mpu_port, 0, mpu_irq, &rmidi); if (error) snd_printk(KERN_WARNING "no MPU-401 device at 0x%lx?\n", mpu_port); diff --git a/sound/isa/sb/jazz16.c b/sound/isa/sb/jazz16.c index 8ccbcddf08e1..54e3c2c18060 100644 --- a/sound/isa/sb/jazz16.c +++ b/sound/isa/sb/jazz16.c @@ -322,7 +322,6 @@ static int __devinit snd_jazz16_probe(struct device *devptr, unsigned int dev) MPU401_HW_MPU401, mpu_port[dev], 0, mpu_irq[dev], - mpu_irq[dev] >= 0 ? IRQF_DISABLED : 0, NULL) < 0) snd_printk(KERN_ERR "no MPU-401 device at 0x%lx\n", mpu_port[dev]); diff --git a/sound/isa/sb/sb16.c b/sound/isa/sb/sb16.c index 4d1c5a300ff8..237f8bd7fbe4 100644 --- a/sound/isa/sb/sb16.c +++ b/sound/isa/sb/sb16.c @@ -394,8 +394,9 @@ static int __devinit snd_sb16_probe(struct snd_card *card, int dev) if (chip->mpu_port > 0 && chip->mpu_port != SNDRV_AUTO_PORT) { if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_SB, - chip->mpu_port, 0, - xirq, 0, &chip->rmidi)) < 0) + chip->mpu_port, + MPU401_INFO_IRQ_HOOK, -1, + &chip->rmidi)) < 0) return err; chip->rmidi_callback = snd_mpu401_uart_interrupt; } diff --git a/sound/isa/sb/sb_common.c b/sound/isa/sb/sb_common.c index eae6c1c0eff9..d2e19215813e 100644 --- a/sound/isa/sb/sb_common.c +++ b/sound/isa/sb/sb_common.c @@ -240,7 +240,7 @@ int snd_sbdsp_create(struct snd_card *card, if (request_irq(irq, irq_handler, (hardware == SB_HW_ALS4000 || hardware == SB_HW_CS5530) ? - IRQF_SHARED : IRQF_DISABLED, + IRQF_SHARED : 0, "SoundBlaster", (void *) chip)) { snd_printk(KERN_ERR "sb: can't grab irq %d\n", irq); snd_sbdsp_free(chip); diff --git a/sound/isa/sc6000.c b/sound/isa/sc6000.c index 9a8bbf6dd62a..207c161f100c 100644 --- a/sound/isa/sc6000.c +++ b/sound/isa/sc6000.c @@ -658,8 +658,7 @@ static int __devinit snd_sc6000_probe(struct device *devptr, unsigned int dev) if (snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, mpu_port[dev], 0, - mpu_irq[dev], IRQF_DISABLED, - NULL) < 0) + mpu_irq[dev], NULL) < 0) snd_printk(KERN_ERR "no MPU-401 device at 0x%lx ?\n", mpu_port[dev]); } diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c index e2d5d2d3ed96..f2379e102b63 100644 --- a/sound/isa/sscape.c +++ b/sound/isa/sscape.c @@ -825,8 +825,7 @@ static int __devinit create_mpu401(struct snd_card *card, int devnum, int err; err = snd_mpu401_uart_new(card, devnum, MPU401_HW_MPU401, port, - MPU401_INFO_INTEGRATED, irq, IRQF_DISABLED, - &rawmidi); + MPU401_INFO_INTEGRATED, irq, &rawmidi); if (err == 0) { struct snd_mpu401 *mpu = rawmidi->private_data; mpu->open_input = mpu401_open; diff --git a/sound/isa/wavefront/wavefront.c b/sound/isa/wavefront/wavefront.c index 711670e4a425..87142977335a 100644 --- a/sound/isa/wavefront/wavefront.c +++ b/sound/isa/wavefront/wavefront.c @@ -418,7 +418,7 @@ snd_wavefront_probe (struct snd_card *card, int dev) return -EBUSY; } if (request_irq(ics2115_irq[dev], snd_wavefront_ics2115_interrupt, - IRQF_DISABLED, "ICS2115", acard)) { + 0, "ICS2115", acard)) { snd_printk(KERN_ERR "unable to use ICS2115 IRQ %d\n", ics2115_irq[dev]); return -EBUSY; } @@ -449,8 +449,7 @@ snd_wavefront_probe (struct snd_card *card, int dev) if (cs4232_mpu_port[dev] > 0 && cs4232_mpu_port[dev] != SNDRV_AUTO_PORT) { err = snd_mpu401_uart_new(card, midi_dev, MPU401_HW_CS4232, cs4232_mpu_port[dev], 0, - cs4232_mpu_irq[dev], IRQF_DISABLED, - NULL); + cs4232_mpu_irq[dev], NULL); if (err < 0) { snd_printk (KERN_ERR "can't allocate CS4232 MPU-401 device\n"); return err; diff --git a/sound/isa/wss/wss_lib.c b/sound/isa/wss/wss_lib.c index 2a42cc377957..7277c5b7df6c 100644 --- a/sound/isa/wss/wss_lib.c +++ b/sound/isa/wss/wss_lib.c @@ -1833,7 +1833,7 @@ int snd_wss_create(struct snd_card *card, } chip->cport = cport; if (!(hwshare & WSS_HWSHARE_IRQ)) - if (request_irq(irq, snd_wss_interrupt, IRQF_DISABLED, + if (request_irq(irq, snd_wss_interrupt, 0, "WSS", (void *) chip)) { snd_printk(KERN_ERR "wss: can't grab IRQ %d\n", irq); snd_wss_free(chip); diff --git a/sound/mips/Kconfig b/sound/mips/Kconfig index a9823fad85c2..77dd0a13aecc 100644 --- a/sound/mips/Kconfig +++ b/sound/mips/Kconfig @@ -23,12 +23,15 @@ config SND_SGI_HAL2 config SND_AU1X00 - tristate "Au1x00 AC97 Port Driver" + tristate "Au1x00 AC97 Port Driver (DEPRECATED)" depends on SOC_AU1000 || SOC_AU1100 || SOC_AU1500 select SND_PCM select SND_AC97_CODEC help ALSA Sound driver for the Au1x00's AC97 port. + Newer drivers for ASoC are available, please do not use + this driver as it will be removed in the future. + endif # SND_MIPS diff --git a/sound/mips/au1x00.c b/sound/mips/au1x00.c index 446cf9748664..7567ebd71913 100644 --- a/sound/mips/au1x00.c +++ b/sound/mips/au1x00.c @@ -465,13 +465,13 @@ snd_au1000_pcm_new(struct snd_au1000 *au1000) flags = claim_dma_lock(); if ((au1000->stream[PLAYBACK]->dma = request_au1000_dma(DMA_ID_AC97C_TX, - "AC97 TX", au1000_dma_interrupt, IRQF_DISABLED, + "AC97 TX", au1000_dma_interrupt, 0, au1000->stream[PLAYBACK])) < 0) { release_dma_lock(flags); return -EBUSY; } if ((au1000->stream[CAPTURE]->dma = request_au1000_dma(DMA_ID_AC97C_RX, - "AC97 RX", au1000_dma_interrupt, IRQF_DISABLED, + "AC97 RX", au1000_dma_interrupt, 0, au1000->stream[CAPTURE])) < 0){ release_dma_lock(flags); return -EBUSY; diff --git a/sound/oss/sound_timer.c b/sound/oss/sound_timer.c index 48cda6c4c257..8021c85f076d 100644 --- a/sound/oss/sound_timer.c +++ b/sound/oss/sound_timer.c @@ -320,7 +320,7 @@ void sound_timer_init(struct sound_lowlev_timer *t, char *name) n = sound_alloc_timerdev(); if (n == -1) n = 0; /* Overwrite the system timer */ - strcpy(sound_timer.info.name, name); + strlcpy(sound_timer.info.name, name, sizeof(sound_timer.info.name)); sound_timer_devs[n] = &sound_timer; } EXPORT_SYMBOL(sound_timer_init); diff --git a/sound/pci/als4000.c b/sound/pci/als4000.c index a9c1af33f276..04628696eb08 100644 --- a/sound/pci/als4000.c +++ b/sound/pci/als4000.c @@ -931,8 +931,9 @@ static int __devinit snd_card_als4000_probe(struct pci_dev *pci, if ((err = snd_mpu401_uart_new( card, 0, MPU401_HW_ALS4000, iobase + ALS4K_IOB_30_MIDI_DATA, - MPU401_INFO_INTEGRATED, - pci->irq, 0, &chip->rmidi)) < 0) { + MPU401_INFO_INTEGRATED | + MPU401_INFO_IRQ_HOOK, + -1, &chip->rmidi)) < 0) { printk(KERN_ERR "als4000: no MPU-401 device at 0x%lx?\n", iobase + ALS4K_IOB_30_MIDI_DATA); goto out_err; diff --git a/sound/pci/au88x0/au88x0_mpu401.c b/sound/pci/au88x0/au88x0_mpu401.c index 0dc8d259d1ed..e6c6a0febb75 100644 --- a/sound/pci/au88x0/au88x0_mpu401.c +++ b/sound/pci/au88x0/au88x0_mpu401.c @@ -84,7 +84,7 @@ static int __devinit snd_vortex_midi(vortex_t * vortex) #ifdef VORTEX_MPU401_LEGACY if ((temp = snd_mpu401_uart_new(vortex->card, 0, MPU401_HW_MPU401, 0x330, - 0, 0, 0, &rmidi)) != 0) { + MPU401_INFO_IRQ_HOOK, -1, &rmidi)) != 0) { hwwrite(vortex->mmio, VORTEX_CTRL, (hwread(vortex->mmio, VORTEX_CTRL) & ~CTRL_MIDI_PORT) & ~CTRL_MIDI_EN); @@ -94,8 +94,8 @@ static int __devinit snd_vortex_midi(vortex_t * vortex) port = (unsigned long)(vortex->mmio + VORTEX_MIDI_DATA); if ((temp = snd_mpu401_uart_new(vortex->card, 0, MPU401_HW_AUREAL, port, - MPU401_INFO_INTEGRATED | MPU401_INFO_MMIO, - 0, 0, &rmidi)) != 0) { + MPU401_INFO_INTEGRATED | MPU401_INFO_MMIO | + MPU401_INFO_IRQ_HOOK, -1, &rmidi)) != 0) { hwwrite(vortex->mmio, VORTEX_CTRL, (hwread(vortex->mmio, VORTEX_CTRL) & ~CTRL_MIDI_PORT) & ~CTRL_MIDI_EN); diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c index 579fc0dce128..d24fe425e87f 100644 --- a/sound/pci/azt3328.c +++ b/sound/pci/azt3328.c @@ -2652,8 +2652,9 @@ snd_azf3328_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) since our hardware ought to be similar, thus use same ID. */ err = snd_mpu401_uart_new( card, 0, - MPU401_HW_AZT2320, chip->mpu_io, MPU401_INFO_INTEGRATED, - pci->irq, 0, &chip->rmidi + MPU401_HW_AZT2320, chip->mpu_io, + MPU401_INFO_INTEGRATED | MPU401_INFO_IRQ_HOOK, + -1, &chip->rmidi ); if (err < 0) { snd_printk(KERN_ERR "azf3328: no MPU-401 device at 0x%lx?\n", diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c index 9cf99fb7eb9c..da9c73211eca 100644 --- a/sound/pci/cmipci.c +++ b/sound/pci/cmipci.c @@ -3228,8 +3228,9 @@ static int __devinit snd_cmipci_create(struct snd_card *card, struct pci_dev *pc if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_CMIPCI, iomidi, (integrated_midi ? - MPU401_INFO_INTEGRATED : 0), - cm->irq, 0, &cm->rmidi)) < 0) { + MPU401_INFO_INTEGRATED : 0) | + MPU401_INFO_IRQ_HOOK, + -1, &cm->rmidi)) < 0) { printk(KERN_ERR "cmipci: no UART401 device at 0x%lx\n", iomidi); } } diff --git a/sound/pci/ctxfi/ctpcm.c b/sound/pci/ctxfi/ctpcm.c index 457d21189b0d..2c8622617c8c 100644 --- a/sound/pci/ctxfi/ctpcm.c +++ b/sound/pci/ctxfi/ctpcm.c @@ -404,7 +404,7 @@ int ct_alsa_pcm_create(struct ct_atc *atc, int err; int playback_count, capture_count; - playback_count = (IEC958 == device) ? 1 : 8; + playback_count = (IEC958 == device) ? 1 : 256; capture_count = (FRONT == device) ? 1 : 0; err = snd_pcm_new(atc->card, "ctxfi", device, playback_count, capture_count, &pcm); diff --git a/sound/pci/ctxfi/ctsrc.c b/sound/pci/ctxfi/ctsrc.c index c749fa720889..e134b3a5780d 100644 --- a/sound/pci/ctxfi/ctsrc.c +++ b/sound/pci/ctxfi/ctsrc.c @@ -20,7 +20,7 @@ #include "cthardware.h" #include <linux/slab.h> -#define SRC_RESOURCE_NUM 64 +#define SRC_RESOURCE_NUM 256 #define SRCIMP_RESOURCE_NUM 256 static unsigned int conj_mask; diff --git a/sound/pci/ctxfi/ctvmem.h b/sound/pci/ctxfi/ctvmem.h index b23adfca4de6..e6da60eb19ce 100644 --- a/sound/pci/ctxfi/ctvmem.h +++ b/sound/pci/ctxfi/ctvmem.h @@ -18,7 +18,7 @@ #ifndef CTVMEM_H #define CTVMEM_H -#define CT_PTP_NUM 1 /* num of device page table pages */ +#define CT_PTP_NUM 4 /* num of device page table pages */ #include <linux/mutex.h> #include <linux/list.h> diff --git a/sound/pci/emu10k1/emupcm.c b/sound/pci/emu10k1/emupcm.c index 622bace148e3..e22b8e2bbd88 100644 --- a/sound/pci/emu10k1/emupcm.c +++ b/sound/pci/emu10k1/emupcm.c @@ -1146,6 +1146,11 @@ static int snd_emu10k1_playback_open(struct snd_pcm_substream *substream) kfree(epcm); return err; } + err = snd_pcm_hw_rule_noresample(runtime, 48000); + if (err < 0) { + kfree(epcm); + return err; + } mix = &emu->pcm_mixer[substream->number]; for (i = 0; i < 4; i++) mix->send_routing[0][i] = mix->send_routing[1][i] = mix->send_routing[2][i] = i; diff --git a/sound/pci/es1938.c b/sound/pci/es1938.c index 26a5a2f25d4b..718a2643474e 100644 --- a/sound/pci/es1938.c +++ b/sound/pci/es1938.c @@ -1854,8 +1854,9 @@ static int __devinit snd_es1938_probe(struct pci_dev *pci, } } if (snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, - chip->mpu_port, MPU401_INFO_INTEGRATED, - chip->irq, 0, &chip->rmidi) < 0) { + chip->mpu_port, + MPU401_INFO_INTEGRATED | MPU401_INFO_IRQ_HOOK, + -1, &chip->rmidi) < 0) { printk(KERN_ERR "es1938: unable to initialize MPU-401\n"); } else { // this line is vital for MIDI interrupt handling on ess-solo1 diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c index 99ea9320c6b5..407e4abc4356 100644 --- a/sound/pci/es1968.c +++ b/sound/pci/es1968.c @@ -2843,8 +2843,9 @@ static int __devinit snd_es1968_probe(struct pci_dev *pci, if (enable_mpu[dev]) { if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, chip->io_port + ESM_MPU401_PORT, - MPU401_INFO_INTEGRATED, - chip->irq, 0, &chip->rmidi)) < 0) { + MPU401_INFO_INTEGRATED | + MPU401_INFO_IRQ_HOOK, + -1, &chip->rmidi)) < 0) { printk(KERN_WARNING "es1968: skipping MPU-401 MIDI support..\n"); } } diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c index 32b02d906703..136f7232bb7c 100644 --- a/sound/pci/fm801.c +++ b/sound/pci/fm801.c @@ -729,11 +729,14 @@ static struct snd_fm801_tea575x_gpio snd_fm801_tea575x_gpios[] = { { .data = 2, .clk = 0, .wren = 1, .most = 3, .name = "SF64-PCR" }, }; +#define get_tea575x_gpio(chip) \ + (&snd_fm801_tea575x_gpios[((chip)->tea575x_tuner & TUNER_TYPE_MASK) - 1]) + static void snd_fm801_tea575x_set_pins(struct snd_tea575x *tea, u8 pins) { struct fm801 *chip = tea->private_data; unsigned short reg = inw(FM801_REG(chip, GPIO_CTRL)); - struct snd_fm801_tea575x_gpio gpio = snd_fm801_tea575x_gpios[(chip->tea575x_tuner & TUNER_TYPE_MASK) - 1]; + struct snd_fm801_tea575x_gpio gpio = *get_tea575x_gpio(chip); reg &= ~(FM801_GPIO_GP(gpio.data) | FM801_GPIO_GP(gpio.clk) | @@ -751,7 +754,7 @@ static u8 snd_fm801_tea575x_get_pins(struct snd_tea575x *tea) { struct fm801 *chip = tea->private_data; unsigned short reg = inw(FM801_REG(chip, GPIO_CTRL)); - struct snd_fm801_tea575x_gpio gpio = snd_fm801_tea575x_gpios[(chip->tea575x_tuner & TUNER_TYPE_MASK) - 1]; + struct snd_fm801_tea575x_gpio gpio = *get_tea575x_gpio(chip); return (reg & FM801_GPIO_GP(gpio.data)) ? TEA575X_DATA : 0 | (reg & FM801_GPIO_GP(gpio.most)) ? TEA575X_MOST : 0; @@ -761,7 +764,7 @@ static void snd_fm801_tea575x_set_direction(struct snd_tea575x *tea, bool output { struct fm801 *chip = tea->private_data; unsigned short reg = inw(FM801_REG(chip, GPIO_CTRL)); - struct snd_fm801_tea575x_gpio gpio = snd_fm801_tea575x_gpios[(chip->tea575x_tuner & TUNER_TYPE_MASK) - 1]; + struct snd_fm801_tea575x_gpio gpio = *get_tea575x_gpio(chip); /* use GPIO lines and set write enable bit */ reg |= FM801_GPIO_GS(gpio.data) | @@ -1246,7 +1249,7 @@ static int __devinit snd_fm801_create(struct snd_card *card, chip->tea575x_tuner = tea575x_tuner; if (!snd_tea575x_init(&chip->tea)) { snd_printk(KERN_INFO "detected TEA575x radio type %s\n", - snd_fm801_tea575x_gpios[tea575x_tuner - 1].name); + get_tea575x_gpio(chip)->name); break; } } @@ -1256,9 +1259,7 @@ static int __devinit snd_fm801_create(struct snd_card *card, } } if (!(chip->tea575x_tuner & TUNER_DISABLED)) { - strlcpy(chip->tea.card, - snd_fm801_tea575x_gpios[(tea575x_tuner & - TUNER_TYPE_MASK) - 1].name, + strlcpy(chip->tea.card, get_tea575x_gpio(chip)->name, sizeof(chip->tea.card)); } #endif @@ -1311,8 +1312,9 @@ static int __devinit snd_card_fm801_probe(struct pci_dev *pci, } if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_FM801, FM801_REG(chip, MPU401_DATA), - MPU401_INFO_INTEGRATED, - chip->irq, 0, &chip->rmidi)) < 0) { + MPU401_INFO_INTEGRATED | + MPU401_INFO_IRQ_HOOK, + -1, &chip->rmidi)) < 0) { snd_card_free(card); return err; } diff --git a/sound/pci/hda/Makefile b/sound/pci/hda/Makefile index 87365d5ea2a9..f928d6634723 100644 --- a/sound/pci/hda/Makefile +++ b/sound/pci/hda/Makefile @@ -6,6 +6,9 @@ snd-hda-codec-$(CONFIG_PROC_FS) += hda_proc.o snd-hda-codec-$(CONFIG_SND_HDA_HWDEP) += hda_hwdep.o snd-hda-codec-$(CONFIG_SND_HDA_INPUT_BEEP) += hda_beep.o +# for trace-points +CFLAGS_hda_codec.o := -I$(src) + snd-hda-codec-realtek-objs := patch_realtek.o snd-hda-codec-cmedia-objs := patch_cmedia.o snd-hda-codec-analog-objs := patch_analog.o diff --git a/sound/pci/hda/alc260_quirks.c b/sound/pci/hda/alc260_quirks.c index 21ec2cb100b0..3b5170b9700f 100644 --- a/sound/pci/hda/alc260_quirks.c +++ b/sound/pci/hda/alc260_quirks.c @@ -7,9 +7,6 @@ enum { ALC260_AUTO, ALC260_BASIC, - ALC260_HP, - ALC260_HP_DC7600, - ALC260_HP_3013, ALC260_FUJITSU_S702X, ALC260_ACER, ALC260_WILL, @@ -142,8 +139,6 @@ static const struct hda_channel_mode alc260_modes[1] = { /* Mixer combinations * * basic: base_output + input + pc_beep + capture - * HP: base_output + input + capture_alt - * HP_3013: hp_3013 + input + capture * fujitsu: fujitsu + capture * acer: acer + capture */ @@ -170,145 +165,6 @@ static const struct snd_kcontrol_new alc260_input_mixer[] = { { } /* end */ }; -/* update HP, line and mono out pins according to the master switch */ -static void alc260_hp_master_update(struct hda_codec *codec) -{ - update_speakers(codec); -} - -static int alc260_hp_master_sw_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct alc_spec *spec = codec->spec; - *ucontrol->value.integer.value = !spec->master_mute; - return 0; -} - -static int alc260_hp_master_sw_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct alc_spec *spec = codec->spec; - int val = !*ucontrol->value.integer.value; - - if (val == spec->master_mute) - return 0; - spec->master_mute = val; - alc260_hp_master_update(codec); - return 1; -} - -static const struct snd_kcontrol_new alc260_hp_output_mixer[] = { - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Master Playback Switch", - .subdevice = HDA_SUBDEV_NID_FLAG | 0x11, - .info = snd_ctl_boolean_mono_info, - .get = alc260_hp_master_sw_get, - .put = alc260_hp_master_sw_put, - }, - HDA_CODEC_VOLUME("Front Playback Volume", 0x08, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Front Playback Switch", 0x08, 2, HDA_INPUT), - HDA_CODEC_VOLUME("Headphone Playback Volume", 0x09, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Headphone Playback Switch", 0x09, 2, HDA_INPUT), - HDA_CODEC_VOLUME_MONO("Speaker Playback Volume", 0x0a, 1, 0x0, - HDA_OUTPUT), - HDA_BIND_MUTE_MONO("Speaker Playback Switch", 0x0a, 1, 2, HDA_INPUT), - { } /* end */ -}; - -static const struct hda_verb alc260_hp_unsol_verbs[] = { - {0x10, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT}, - {}, -}; - -static void alc260_hp_setup(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - - spec->autocfg.hp_pins[0] = 0x0f; - spec->autocfg.speaker_pins[0] = 0x10; - spec->autocfg.speaker_pins[1] = 0x11; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_PIN; -} - -static const struct snd_kcontrol_new alc260_hp_3013_mixer[] = { - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Master Playback Switch", - .subdevice = HDA_SUBDEV_NID_FLAG | 0x11, - .info = snd_ctl_boolean_mono_info, - .get = alc260_hp_master_sw_get, - .put = alc260_hp_master_sw_put, - }, - HDA_CODEC_VOLUME("Front Playback Volume", 0x09, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Front Playback Switch", 0x10, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Aux-In Playback Volume", 0x07, 0x06, HDA_INPUT), - HDA_CODEC_MUTE("Aux-In Playback Switch", 0x07, 0x06, HDA_INPUT), - HDA_CODEC_VOLUME("Headphone Playback Volume", 0x08, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("Speaker Playback Volume", 0x0a, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_MONO("Speaker Playback Switch", 0x11, 1, 0x0, HDA_OUTPUT), - { } /* end */ -}; - -static void alc260_hp_3013_setup(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - - spec->autocfg.hp_pins[0] = 0x15; - spec->autocfg.speaker_pins[0] = 0x10; - spec->autocfg.speaker_pins[1] = 0x11; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_PIN; -} - -static const struct hda_bind_ctls alc260_dc7600_bind_master_vol = { - .ops = &snd_hda_bind_vol, - .values = { - HDA_COMPOSE_AMP_VAL(0x08, 3, 0, HDA_OUTPUT), - HDA_COMPOSE_AMP_VAL(0x09, 3, 0, HDA_OUTPUT), - HDA_COMPOSE_AMP_VAL(0x0a, 3, 0, HDA_OUTPUT), - 0 - }, -}; - -static const struct hda_bind_ctls alc260_dc7600_bind_switch = { - .ops = &snd_hda_bind_sw, - .values = { - HDA_COMPOSE_AMP_VAL(0x11, 3, 0, HDA_OUTPUT), - HDA_COMPOSE_AMP_VAL(0x15, 3, 0, HDA_OUTPUT), - 0 - }, -}; - -static const struct snd_kcontrol_new alc260_hp_dc7600_mixer[] = { - HDA_BIND_VOL("Master Playback Volume", &alc260_dc7600_bind_master_vol), - HDA_BIND_SW("LineOut Playback Switch", &alc260_dc7600_bind_switch), - HDA_CODEC_MUTE("Speaker Playback Switch", 0x0f, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x10, 0x0, HDA_OUTPUT), - { } /* end */ -}; - -static const struct hda_verb alc260_hp_3013_unsol_verbs[] = { - {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT}, - {}, -}; - -static void alc260_hp_3012_setup(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - - spec->autocfg.hp_pins[0] = 0x10; - spec->autocfg.speaker_pins[0] = 0x0f; - spec->autocfg.speaker_pins[1] = 0x11; - spec->autocfg.speaker_pins[2] = 0x15; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_PIN; -} - /* Fujitsu S702x series laptops. ALC260 pin usage: Mic/Line jack = 0x12, * HP jack = 0x14, CD audio = 0x16, internal speaker = 0x10. */ @@ -480,106 +336,6 @@ static const struct hda_verb alc260_init_verbs[] = { { } }; -#if 0 /* should be identical with alc260_init_verbs? */ -static const struct hda_verb alc260_hp_init_verbs[] = { - /* Headphone and output */ - {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0}, - /* mono output */ - {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, - /* Mic1 (rear panel) pin widget for input and vref at 80% */ - {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24}, - /* Mic2 (front panel) pin widget for input and vref at 80% */ - {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24}, - /* Line In pin widget for input */ - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20}, - /* Line-2 pin widget for output */ - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, - /* CD pin widget for input */ - {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20}, - /* unmute amp left and right */ - {0x04, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000}, - /* set connection select to line in (default select for this ADC) */ - {0x04, AC_VERB_SET_CONNECT_SEL, 0x02}, - /* unmute Line-Out mixer amp left and right (volume = 0) */ - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, - /* mute pin widget amp left and right (no gain on this amp) */ - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000}, - /* unmute HP mixer amp left and right (volume = 0) */ - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, - /* mute pin widget amp left and right (no gain on this amp) */ - {0x10, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000}, - /* Amp Indexes: CD = 0x04, Line In 1 = 0x02, Mic 1 = 0x00 & - * Line In 2 = 0x03 - */ - /* mute analog inputs */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - /* Amp Indexes: DAC = 0x01 & mixer = 0x00 */ - /* Unmute Front out path */ - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))}, - /* Unmute Headphone out path */ - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))}, - /* Unmute Mono out path */ - {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, - {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))}, - { } -}; -#endif - -static const struct hda_verb alc260_hp_3013_init_verbs[] = { - /* Line out and output */ - {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, - /* mono output */ - {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, - /* Mic1 (rear panel) pin widget for input and vref at 80% */ - {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24}, - /* Mic2 (front panel) pin widget for input and vref at 80% */ - {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24}, - /* Line In pin widget for input */ - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20}, - /* Headphone pin widget for output */ - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0}, - /* CD pin widget for input */ - {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20}, - /* unmute amp left and right */ - {0x04, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000}, - /* set connection select to line in (default select for this ADC) */ - {0x04, AC_VERB_SET_CONNECT_SEL, 0x02}, - /* unmute Line-Out mixer amp left and right (volume = 0) */ - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, - /* mute pin widget amp left and right (no gain on this amp) */ - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000}, - /* unmute HP mixer amp left and right (volume = 0) */ - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, - /* mute pin widget amp left and right (no gain on this amp) */ - {0x10, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000}, - /* Amp Indexes: CD = 0x04, Line In 1 = 0x02, Mic 1 = 0x00 & - * Line In 2 = 0x03 - */ - /* mute analog inputs */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - /* Amp Indexes: DAC = 0x01 & mixer = 0x00 */ - /* Unmute Front out path */ - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))}, - /* Unmute Headphone out path */ - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))}, - /* Unmute Mono out path */ - {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, - {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))}, - { } -}; - /* Initialisation sequence for ALC260 as configured in Fujitsu S702x * laptops. ALC260 pin usage: Mic/Line jack = 0x12, HP jack = 0x14, CD * audio = 0x16, internal speaker = 0x10. @@ -1093,9 +849,6 @@ static const struct hda_verb alc260_test_init_verbs[] = { */ static const char * const alc260_models[ALC260_MODEL_LAST] = { [ALC260_BASIC] = "basic", - [ALC260_HP] = "hp", - [ALC260_HP_3013] = "hp-3013", - [ALC260_HP_DC7600] = "hp-dc7600", [ALC260_FUJITSU_S702X] = "fujitsu", [ALC260_ACER] = "acer", [ALC260_WILL] = "will", @@ -1112,15 +865,6 @@ static const struct snd_pci_quirk alc260_cfg_tbl[] = { SND_PCI_QUIRK(0x1025, 0x007f, "Acer", ALC260_WILL), SND_PCI_QUIRK(0x1025, 0x008f, "Acer", ALC260_ACER), SND_PCI_QUIRK(0x1509, 0x4540, "Favorit 100XS", ALC260_FAVORIT100), - SND_PCI_QUIRK(0x103c, 0x2808, "HP d5700", ALC260_HP_3013), - SND_PCI_QUIRK(0x103c, 0x280a, "HP d5750", ALC260_AUTO), /* no quirk */ - SND_PCI_QUIRK(0x103c, 0x3010, "HP", ALC260_HP_3013), - SND_PCI_QUIRK(0x103c, 0x3011, "HP", ALC260_HP_3013), - SND_PCI_QUIRK(0x103c, 0x3012, "HP", ALC260_HP_DC7600), - SND_PCI_QUIRK(0x103c, 0x3013, "HP", ALC260_HP_3013), - SND_PCI_QUIRK(0x103c, 0x3014, "HP", ALC260_HP), - SND_PCI_QUIRK(0x103c, 0x3015, "HP", ALC260_HP), - SND_PCI_QUIRK(0x103c, 0x3016, "HP", ALC260_HP), SND_PCI_QUIRK(0x104d, 0x81bb, "Sony VAIO", ALC260_BASIC), SND_PCI_QUIRK(0x104d, 0x81cc, "Sony VAIO", ALC260_BASIC), SND_PCI_QUIRK(0x104d, 0x81cd, "Sony VAIO", ALC260_BASIC), @@ -1144,54 +888,6 @@ static const struct alc_config_preset alc260_presets[] = { .channel_mode = alc260_modes, .input_mux = &alc260_capture_source, }, - [ALC260_HP] = { - .mixers = { alc260_hp_output_mixer, - alc260_input_mixer }, - .init_verbs = { alc260_init_verbs, - alc260_hp_unsol_verbs }, - .num_dacs = ARRAY_SIZE(alc260_dac_nids), - .dac_nids = alc260_dac_nids, - .num_adc_nids = ARRAY_SIZE(alc260_adc_nids_alt), - .adc_nids = alc260_adc_nids_alt, - .num_channel_mode = ARRAY_SIZE(alc260_modes), - .channel_mode = alc260_modes, - .input_mux = &alc260_capture_source, - .unsol_event = alc_sku_unsol_event, - .setup = alc260_hp_setup, - .init_hook = alc_inithook, - }, - [ALC260_HP_DC7600] = { - .mixers = { alc260_hp_dc7600_mixer, - alc260_input_mixer }, - .init_verbs = { alc260_init_verbs, - alc260_hp_dc7600_verbs }, - .num_dacs = ARRAY_SIZE(alc260_dac_nids), - .dac_nids = alc260_dac_nids, - .num_adc_nids = ARRAY_SIZE(alc260_adc_nids_alt), - .adc_nids = alc260_adc_nids_alt, - .num_channel_mode = ARRAY_SIZE(alc260_modes), - .channel_mode = alc260_modes, - .input_mux = &alc260_capture_source, - .unsol_event = alc_sku_unsol_event, - .setup = alc260_hp_3012_setup, - .init_hook = alc_inithook, - }, - [ALC260_HP_3013] = { - .mixers = { alc260_hp_3013_mixer, - alc260_input_mixer }, - .init_verbs = { alc260_hp_3013_init_verbs, - alc260_hp_3013_unsol_verbs }, - .num_dacs = ARRAY_SIZE(alc260_dac_nids), - .dac_nids = alc260_dac_nids, - .num_adc_nids = ARRAY_SIZE(alc260_adc_nids_alt), - .adc_nids = alc260_adc_nids_alt, - .num_channel_mode = ARRAY_SIZE(alc260_modes), - .channel_mode = alc260_modes, - .input_mux = &alc260_capture_source, - .unsol_event = alc_sku_unsol_event, - .setup = alc260_hp_3013_setup, - .init_hook = alc_inithook, - }, [ALC260_FUJITSU_S702X] = { .mixers = { alc260_fujitsu_mixer }, .init_verbs = { alc260_fujitsu_init_verbs }, diff --git a/sound/pci/hda/alc262_quirks.c b/sound/pci/hda/alc262_quirks.c index 8d2097d77642..7894b2b5aacf 100644 --- a/sound/pci/hda/alc262_quirks.c +++ b/sound/pci/hda/alc262_quirks.c @@ -10,13 +10,7 @@ enum { ALC262_HIPPO, ALC262_HIPPO_1, ALC262_FUJITSU, - ALC262_HP_BPC, - ALC262_HP_BPC_D7000_WL, - ALC262_HP_BPC_D7000_WF, - ALC262_HP_TC_T5735, - ALC262_HP_RP5700, ALC262_BENQ_ED8, - ALC262_SONY_ASSAMD, ALC262_BENQ_T31, ALC262_ULTRA, ALC262_LENOVO_3000, @@ -66,164 +60,31 @@ static const struct snd_kcontrol_new alc262_base_mixer[] = { { } /* end */ }; -/* update HP, line and mono-out pins according to the master switch */ -#define alc262_hp_master_update alc260_hp_master_update +/* bind hp and internal speaker mute (with plug check) as master switch */ -static void alc262_hp_bpc_setup(struct hda_codec *codec) +static int alc262_hippo_master_sw_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct alc_spec *spec = codec->spec; - - spec->autocfg.hp_pins[0] = 0x1b; - spec->autocfg.speaker_pins[0] = 0x16; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_PIN; + *ucontrol->value.integer.value = !spec->master_mute; + return 0; } -static void alc262_hp_wildwest_setup(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - - spec->autocfg.hp_pins[0] = 0x15; - spec->autocfg.speaker_pins[0] = 0x16; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_PIN; -} - -#define alc262_hp_master_sw_get alc260_hp_master_sw_get -#define alc262_hp_master_sw_put alc260_hp_master_sw_put - -#define ALC262_HP_MASTER_SWITCH \ - { \ - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ - .name = "Master Playback Switch", \ - .info = snd_ctl_boolean_mono_info, \ - .get = alc262_hp_master_sw_get, \ - .put = alc262_hp_master_sw_put, \ - }, \ - { \ - .iface = NID_MAPPING, \ - .name = "Master Playback Switch", \ - .private_value = 0x15 | (0x16 << 8) | (0x1b << 16), \ - } - - -static const struct snd_kcontrol_new alc262_HP_BPC_mixer[] = { - ALC262_HP_MASTER_SWITCH, - HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Front Playback Switch", 0x15, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("Speaker Playback Volume", 0x0e, 2, 0x0, - HDA_OUTPUT), - HDA_CODEC_MUTE_MONO("Speaker Playback Switch", 0x16, 2, 0x0, - HDA_OUTPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), - HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT), - HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("AUX IN Playback Volume", 0x0b, 0x06, HDA_INPUT), - HDA_CODEC_MUTE("AUX IN Playback Switch", 0x0b, 0x06, HDA_INPUT), - { } /* end */ -}; - -static const struct snd_kcontrol_new alc262_HP_BPC_WildWest_mixer[] = { - ALC262_HP_MASTER_SWITCH, - HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Front Playback Switch", 0x1b, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0d, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("Speaker Playback Volume", 0x0e, 2, 0x0, - HDA_OUTPUT), - HDA_CODEC_MUTE_MONO("Speaker Playback Switch", 0x16, 2, 0x0, - HDA_OUTPUT), - HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x1a, 0, HDA_INPUT), - HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x01, HDA_INPUT), - HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x01, HDA_INPUT), - HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), - { } /* end */ -}; - -static const struct snd_kcontrol_new alc262_HP_BPC_WildWest_option_mixer[] = { - HDA_CODEC_VOLUME("Rear Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Rear Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Rear Mic Boost Volume", 0x18, 0, HDA_INPUT), - { } /* end */ -}; - -/* mute/unmute internal speaker according to the hp jack and mute state */ -static void alc262_hp_t5735_setup(struct hda_codec *codec) +static int alc262_hippo_master_sw_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct alc_spec *spec = codec->spec; + int val = !*ucontrol->value.integer.value; - spec->autocfg.hp_pins[0] = 0x15; - spec->autocfg.speaker_pins[0] = 0x14; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_PIN; + if (val == spec->master_mute) + return 0; + spec->master_mute = val; + update_outputs(codec); + return 1; } -static const struct snd_kcontrol_new alc262_hp_t5735_mixer[] = { - HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0d, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), - { } /* end */ -}; - -static const struct hda_verb alc262_hp_t5735_verbs[] = { - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - - {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN}, - { } -}; - -static const struct snd_kcontrol_new alc262_hp_rp5700_mixer[] = { - HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0e, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Speaker Playback Switch", 0x16, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x01, HDA_INPUT), - HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x01, HDA_INPUT), - { } /* end */ -}; - -static const struct hda_verb alc262_hp_rp5700_verbs[] = { - {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00}, - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))}, - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x00 << 8))}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x00 << 8))}, - {} -}; - -static const struct hda_input_mux alc262_hp_rp5700_capture_source = { - .num_items = 1, - .items = { - { "Line", 0x1 }, - }, -}; - -/* bind hp and internal speaker mute (with plug check) as master switch */ -#define alc262_hippo_master_update alc262_hp_master_update -#define alc262_hippo_master_sw_get alc262_hp_master_sw_get -#define alc262_hippo_master_sw_put alc262_hp_master_sw_put - #define ALC262_HIPPO_MASTER_SWITCH \ { \ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ @@ -239,6 +100,9 @@ static const struct hda_input_mux alc262_hp_rp5700_capture_source = { (SUBDEV_SPEAKER(0) << 16), \ } +#define alc262_hp_master_sw_get alc262_hippo_master_sw_get +#define alc262_hp_master_sw_put alc262_hippo_master_sw_put + static const struct snd_kcontrol_new alc262_hippo_mixer[] = { ALC262_HIPPO_MASTER_SWITCH, HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x0, HDA_OUTPUT), @@ -279,8 +143,7 @@ static void alc262_hippo_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); } static void alc262_hippo1_setup(struct hda_codec *codec) @@ -289,8 +152,7 @@ static void alc262_hippo1_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x1b; spec->autocfg.speaker_pins[0] = 0x14; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); } @@ -353,8 +215,7 @@ static void alc262_tyan_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x1b; spec->autocfg.speaker_pins[0] = 0x15; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); } @@ -496,8 +357,7 @@ static void alc262_toshiba_s06_setup(struct hda_codec *codec) spec->ext_mic_pin = 0x18; spec->int_mic_pin = 0x12; spec->auto_mic = 1; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_PIN; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_PIN); } /* @@ -571,27 +431,6 @@ static const struct hda_input_mux alc262_fujitsu_capture_source = { }, }; -static const struct hda_input_mux alc262_HP_capture_source = { - .num_items = 5, - .items = { - { "Mic", 0x0 }, - { "Front Mic", 0x1 }, - { "Line", 0x2 }, - { "CD", 0x4 }, - { "AUX IN", 0x6 }, - }, -}; - -static const struct hda_input_mux alc262_HP_D7000_capture_source = { - .num_items = 4, - .items = { - { "Mic", 0x0 }, - { "Front Mic", 0x2 }, - { "Line", 0x1 }, - { "CD", 0x4 }, - }, -}; - static void alc262_fujitsu_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -599,8 +438,7 @@ static void alc262_fujitsu_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.hp_pins[1] = 0x1b; spec->autocfg.speaker_pins[0] = 0x15; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); } /* bind volumes of both NID 0x0c and 0x0d */ @@ -646,8 +484,7 @@ static void alc262_lenovo_3000_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x1b; spec->autocfg.speaker_pins[0] = 0x14; spec->autocfg.speaker_pins[1] = 0x16; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); } static const struct snd_kcontrol_new alc262_lenovo_3000_mixer[] = { @@ -752,8 +589,8 @@ static void alc262_ultra_automute(struct hda_codec *codec) mute = 0; /* auto-mute only when HP is used as HP */ if (!spec->cur_mux[0]) { - spec->jack_present = snd_hda_jack_detect(codec, 0x15); - if (spec->jack_present) + spec->hp_jack_present = snd_hda_jack_detect(codec, 0x15); + if (spec->hp_jack_present) mute = HDA_AMP_MUTE; } /* mute/unmute internal speaker */ @@ -817,206 +654,6 @@ static const struct snd_kcontrol_new alc262_ultra_capture_mixer[] = { { } /* end */ }; -static const struct hda_verb alc262_HP_BPC_init_verbs[] = { - /* - * Unmute ADC0-2 and set the default input to mic-in - */ - {0x07, AC_VERB_SET_CONNECT_SEL, 0x00}, - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x08, AC_VERB_SET_CONNECT_SEL, 0x00}, - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x09, AC_VERB_SET_CONNECT_SEL, 0x00}, - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - - /* Mute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback - * mixer widget - * Note: PASD motherboards uses the Line In 2 as the input for - * front panel mic (mic 2) - */ - /* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */ - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, - - /* - * Set up output mixers (0x0c - 0x0e) - */ - /* set vol=0 to output mixers */ - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - - /* set up input amps for analog loopback */ - /* Amp Indices: DAC = 0, mixer = 1 */ - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - - {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - - {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00}, - {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, - - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20}, - {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24}, - {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24}, - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20}, - {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20}, - - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 }, - {0x18, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 }, - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 }, - {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 }, - {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 }, - {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 }, - - - /* FIXME: use matrix-type input source selection */ - /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 0b, 12 */ - /* Input mixer1: only unmute Mic */ - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x01 << 8))}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x05 << 8))}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x06 << 8))}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x07 << 8))}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x08 << 8))}, - /* Input mixer2 */ - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x01 << 8))}, - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))}, - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))}, - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))}, - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x05 << 8))}, - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x06 << 8))}, - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x07 << 8))}, - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x08 << 8))}, - /* Input mixer3 */ - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x01 << 8))}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x05 << 8))}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x06 << 8))}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x07 << 8))}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x08 << 8))}, - - {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN}, - - { } -}; - -static const struct hda_verb alc262_HP_BPC_WildWest_init_verbs[] = { - /* - * Unmute ADC0-2 and set the default input to mic-in - */ - {0x07, AC_VERB_SET_CONNECT_SEL, 0x00}, - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x08, AC_VERB_SET_CONNECT_SEL, 0x00}, - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x09, AC_VERB_SET_CONNECT_SEL, 0x00}, - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - - /* Mute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback - * mixer widget - * Note: PASD motherboards uses the Line In 2 as the input for front - * panel mic (mic 2) - */ - /* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */ - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, - /* - * Set up output mixers (0x0c - 0x0e) - */ - /* set vol=0 to output mixers */ - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - - /* set up input amps for analog loopback */ - /* Amp Indices: DAC = 0, mixer = 1 */ - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - - - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP }, /* HP */ - {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, /* Mono */ - {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* rear MIC */ - {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, /* Line in */ - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* Front MIC */ - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, /* Line out */ - {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, /* CD in */ - - {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - - {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00}, - {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, - - /* {0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0x7023 }, */ - {0x18, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 }, - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 }, - {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, 0x7023 }, - {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 }, - {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 }, - - /* FIXME: use matrix-type input source selection */ - /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */ - /* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */ - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, /*rear MIC*/ - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))}, /*Line in*/ - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))}, /*F MIC*/ - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x03 << 8))}, /*Front*/ - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x04 << 8))}, /*CD*/ - /* {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x06 << 8))}, */ - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x07 << 8))}, /*HP*/ - /* Input mixer2 */ - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))}, - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))}, - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x03 << 8))}, - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x04 << 8))}, - /* {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x06 << 8))}, */ - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x07 << 8))}, - /* Input mixer3 */ - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x03 << 8))}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x04 << 8))}, - /* {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x06 << 8))}, */ - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x07 << 8))}, - - {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN}, - - { } -}; - static const struct hda_verb alc262_toshiba_rx1_unsol_verbs[] = { {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, /* Front Speaker */ @@ -1042,13 +679,8 @@ static const char * const alc262_models[ALC262_MODEL_LAST] = { [ALC262_HIPPO] = "hippo", [ALC262_HIPPO_1] = "hippo_1", [ALC262_FUJITSU] = "fujitsu", - [ALC262_HP_BPC] = "hp-bpc", - [ALC262_HP_BPC_D7000_WL]= "hp-bpc-d7000", - [ALC262_HP_TC_T5735] = "hp-tc-t5735", - [ALC262_HP_RP5700] = "hp-rp5700", [ALC262_BENQ_ED8] = "benq", [ALC262_BENQ_T31] = "benq-t31", - [ALC262_SONY_ASSAMD] = "sony-assamd", [ALC262_TOSHIBA_S06] = "toshiba-s06", [ALC262_TOSHIBA_RX1] = "toshiba-rx1", [ALC262_ULTRA] = "ultra", @@ -1061,41 +693,6 @@ static const char * const alc262_models[ALC262_MODEL_LAST] = { static const struct snd_pci_quirk alc262_cfg_tbl[] = { SND_PCI_QUIRK(0x1002, 0x437b, "Hippo", ALC262_HIPPO), SND_PCI_QUIRK(0x1033, 0x8895, "NEC Versa S9100", ALC262_NEC), - SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x1200, "HP xw series", - ALC262_HP_BPC), - SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x1300, "HP xw series", - ALC262_HP_BPC), - SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x1500, "HP z series", - ALC262_HP_BPC), - SND_PCI_QUIRK(0x103c, 0x170b, "HP Z200", - ALC262_AUTO), - SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x1700, "HP xw series", - ALC262_HP_BPC), - SND_PCI_QUIRK(0x103c, 0x2800, "HP D7000", ALC262_HP_BPC_D7000_WL), - SND_PCI_QUIRK(0x103c, 0x2801, "HP D7000", ALC262_HP_BPC_D7000_WF), - SND_PCI_QUIRK(0x103c, 0x2802, "HP D7000", ALC262_HP_BPC_D7000_WL), - SND_PCI_QUIRK(0x103c, 0x2803, "HP D7000", ALC262_HP_BPC_D7000_WF), - SND_PCI_QUIRK(0x103c, 0x2804, "HP D7000", ALC262_HP_BPC_D7000_WL), - SND_PCI_QUIRK(0x103c, 0x2805, "HP D7000", ALC262_HP_BPC_D7000_WF), - SND_PCI_QUIRK(0x103c, 0x2806, "HP D7000", ALC262_HP_BPC_D7000_WL), - SND_PCI_QUIRK(0x103c, 0x2807, "HP D7000", ALC262_HP_BPC_D7000_WF), - SND_PCI_QUIRK(0x103c, 0x280c, "HP xw4400", ALC262_HP_BPC), - SND_PCI_QUIRK(0x103c, 0x3014, "HP xw6400", ALC262_HP_BPC), - SND_PCI_QUIRK(0x103c, 0x3015, "HP xw8400", ALC262_HP_BPC), - SND_PCI_QUIRK(0x103c, 0x302f, "HP Thin Client T5735", - ALC262_HP_TC_T5735), - SND_PCI_QUIRK(0x103c, 0x2817, "HP RP5700", ALC262_HP_RP5700), - SND_PCI_QUIRK(0x104d, 0x1f00, "Sony ASSAMD", ALC262_SONY_ASSAMD), - SND_PCI_QUIRK(0x104d, 0x8203, "Sony UX-90", ALC262_HIPPO), - SND_PCI_QUIRK(0x104d, 0x820f, "Sony ASSAMD", ALC262_SONY_ASSAMD), - SND_PCI_QUIRK(0x104d, 0x9016, "Sony VAIO", ALC262_AUTO), /* dig-only */ - SND_PCI_QUIRK(0x104d, 0x9025, "Sony VAIO Z21MN", ALC262_TOSHIBA_S06), - SND_PCI_QUIRK(0x104d, 0x9035, "Sony VAIO VGN-FW170J", ALC262_AUTO), - SND_PCI_QUIRK(0x104d, 0x9047, "Sony VAIO Type G", ALC262_AUTO), -#if 0 /* disable the quirk since model=auto works better in recent versions */ - SND_PCI_QUIRK_MASK(0x104d, 0xff00, 0x9000, "Sony VAIO", - ALC262_SONY_ASSAMD), -#endif SND_PCI_QUIRK(0x1179, 0x0001, "Toshiba dynabook SS RX1", ALC262_TOSHIBA_RX1), SND_PCI_QUIRK(0x1179, 0xff7b, "Toshiba S06", ALC262_TOSHIBA_S06), @@ -1166,68 +763,6 @@ static const struct alc_config_preset alc262_presets[] = { .setup = alc262_fujitsu_setup, .init_hook = alc_inithook, }, - [ALC262_HP_BPC] = { - .mixers = { alc262_HP_BPC_mixer }, - .init_verbs = { alc262_HP_BPC_init_verbs }, - .num_dacs = ARRAY_SIZE(alc262_dac_nids), - .dac_nids = alc262_dac_nids, - .hp_nid = 0x03, - .num_channel_mode = ARRAY_SIZE(alc262_modes), - .channel_mode = alc262_modes, - .input_mux = &alc262_HP_capture_source, - .unsol_event = alc_sku_unsol_event, - .setup = alc262_hp_bpc_setup, - .init_hook = alc_inithook, - }, - [ALC262_HP_BPC_D7000_WF] = { - .mixers = { alc262_HP_BPC_WildWest_mixer }, - .init_verbs = { alc262_HP_BPC_WildWest_init_verbs }, - .num_dacs = ARRAY_SIZE(alc262_dac_nids), - .dac_nids = alc262_dac_nids, - .hp_nid = 0x03, - .num_channel_mode = ARRAY_SIZE(alc262_modes), - .channel_mode = alc262_modes, - .input_mux = &alc262_HP_D7000_capture_source, - .unsol_event = alc_sku_unsol_event, - .setup = alc262_hp_wildwest_setup, - .init_hook = alc_inithook, - }, - [ALC262_HP_BPC_D7000_WL] = { - .mixers = { alc262_HP_BPC_WildWest_mixer, - alc262_HP_BPC_WildWest_option_mixer }, - .init_verbs = { alc262_HP_BPC_WildWest_init_verbs }, - .num_dacs = ARRAY_SIZE(alc262_dac_nids), - .dac_nids = alc262_dac_nids, - .hp_nid = 0x03, - .num_channel_mode = ARRAY_SIZE(alc262_modes), - .channel_mode = alc262_modes, - .input_mux = &alc262_HP_D7000_capture_source, - .unsol_event = alc_sku_unsol_event, - .setup = alc262_hp_wildwest_setup, - .init_hook = alc_inithook, - }, - [ALC262_HP_TC_T5735] = { - .mixers = { alc262_hp_t5735_mixer }, - .init_verbs = { alc262_init_verbs, alc262_hp_t5735_verbs }, - .num_dacs = ARRAY_SIZE(alc262_dac_nids), - .dac_nids = alc262_dac_nids, - .hp_nid = 0x03, - .num_channel_mode = ARRAY_SIZE(alc262_modes), - .channel_mode = alc262_modes, - .input_mux = &alc262_capture_source, - .unsol_event = alc_sku_unsol_event, - .setup = alc262_hp_t5735_setup, - .init_hook = alc_inithook, - }, - [ALC262_HP_RP5700] = { - .mixers = { alc262_hp_rp5700_mixer }, - .init_verbs = { alc262_init_verbs, alc262_hp_rp5700_verbs }, - .num_dacs = ARRAY_SIZE(alc262_dac_nids), - .dac_nids = alc262_dac_nids, - .num_channel_mode = ARRAY_SIZE(alc262_modes), - .channel_mode = alc262_modes, - .input_mux = &alc262_hp_rp5700_capture_source, - }, [ALC262_BENQ_ED8] = { .mixers = { alc262_base_mixer }, .init_verbs = { alc262_init_verbs, alc262_EAPD_verbs }, @@ -1238,19 +773,6 @@ static const struct alc_config_preset alc262_presets[] = { .channel_mode = alc262_modes, .input_mux = &alc262_capture_source, }, - [ALC262_SONY_ASSAMD] = { - .mixers = { alc262_sony_mixer }, - .init_verbs = { alc262_init_verbs, alc262_sony_unsol_verbs}, - .num_dacs = ARRAY_SIZE(alc262_dac_nids), - .dac_nids = alc262_dac_nids, - .hp_nid = 0x02, - .num_channel_mode = ARRAY_SIZE(alc262_modes), - .channel_mode = alc262_modes, - .input_mux = &alc262_capture_source, - .unsol_event = alc_sku_unsol_event, - .setup = alc262_hippo_setup, - .init_hook = alc_inithook, - }, [ALC262_BENQ_T31] = { .mixers = { alc262_benq_t31_mixer }, .init_verbs = { alc262_init_verbs, alc262_benq_t31_EAPD_verbs, diff --git a/sound/pci/hda/alc268_quirks.c b/sound/pci/hda/alc268_quirks.c deleted file mode 100644 index 2e5876ce71fe..000000000000 --- a/sound/pci/hda/alc268_quirks.c +++ /dev/null @@ -1,636 +0,0 @@ -/* - * ALC267/ALC268 quirk models - * included by patch_realtek.c - */ - -/* ALC268 models */ -enum { - ALC268_AUTO, - ALC267_QUANTA_IL1, - ALC268_3ST, - ALC268_TOSHIBA, - ALC268_ACER, - ALC268_ACER_DMIC, - ALC268_ACER_ASPIRE_ONE, - ALC268_DELL, - ALC268_ZEPTO, -#ifdef CONFIG_SND_DEBUG - ALC268_TEST, -#endif - ALC268_MODEL_LAST /* last tag */ -}; - -/* - * ALC268 channel source setting (2 channel) - */ -#define ALC268_DIGOUT_NID ALC880_DIGOUT_NID -#define alc268_modes alc260_modes - -static const hda_nid_t alc268_dac_nids[2] = { - /* front, hp */ - 0x02, 0x03 -}; - -static const hda_nid_t alc268_adc_nids[2] = { - /* ADC0-1 */ - 0x08, 0x07 -}; - -static const hda_nid_t alc268_adc_nids_alt[1] = { - /* ADC0 */ - 0x08 -}; - -static const hda_nid_t alc268_capsrc_nids[2] = { 0x23, 0x24 }; - -static const struct snd_kcontrol_new alc268_base_mixer[] = { - /* output mixer control */ - HDA_CODEC_VOLUME("Front Playback Volume", 0x2, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Headphone Playback Volume", 0x3, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT), - HDA_CODEC_VOLUME("Line In Boost Volume", 0x1a, 0, HDA_INPUT), - { } -}; - -static const struct snd_kcontrol_new alc268_toshiba_mixer[] = { - /* output mixer control */ - HDA_CODEC_VOLUME("Front Playback Volume", 0x2, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Headphone Playback Volume", 0x3, 0x0, HDA_OUTPUT), - ALC262_HIPPO_MASTER_SWITCH, - HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT), - HDA_CODEC_VOLUME("Line In Boost Volume", 0x1a, 0, HDA_INPUT), - { } -}; - -static const struct hda_verb alc268_eapd_verbs[] = { - {0x14, AC_VERB_SET_EAPD_BTLENABLE, 2}, - {0x15, AC_VERB_SET_EAPD_BTLENABLE, 2}, - { } -}; - -/* Toshiba specific */ -static const struct hda_verb alc268_toshiba_verbs[] = { - {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN}, - { } /* end */ -}; - -/* Acer specific */ -/* bind volumes of both NID 0x02 and 0x03 */ -static const struct hda_bind_ctls alc268_acer_bind_master_vol = { - .ops = &snd_hda_bind_vol, - .values = { - HDA_COMPOSE_AMP_VAL(0x02, 3, 0, HDA_OUTPUT), - HDA_COMPOSE_AMP_VAL(0x03, 3, 0, HDA_OUTPUT), - 0 - }, -}; - -static void alc268_acer_setup(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - - spec->autocfg.hp_pins[0] = 0x14; - spec->autocfg.speaker_pins[0] = 0x15; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; -} - -#define alc268_acer_master_sw_get alc262_hp_master_sw_get -#define alc268_acer_master_sw_put alc262_hp_master_sw_put - -static const struct snd_kcontrol_new alc268_acer_aspire_one_mixer[] = { - /* output mixer control */ - HDA_BIND_VOL("Master Playback Volume", &alc268_acer_bind_master_vol), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Master Playback Switch", - .subdevice = HDA_SUBDEV_NID_FLAG | 0x15, - .info = snd_ctl_boolean_mono_info, - .get = alc268_acer_master_sw_get, - .put = alc268_acer_master_sw_put, - }, - HDA_CODEC_VOLUME("Mic Boost Capture Volume", 0x18, 0, HDA_INPUT), - { } -}; - -static const struct snd_kcontrol_new alc268_acer_mixer[] = { - /* output mixer control */ - HDA_BIND_VOL("Master Playback Volume", &alc268_acer_bind_master_vol), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Master Playback Switch", - .subdevice = HDA_SUBDEV_NID_FLAG | 0x14, - .info = snd_ctl_boolean_mono_info, - .get = alc268_acer_master_sw_get, - .put = alc268_acer_master_sw_put, - }, - HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), - HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT), - HDA_CODEC_VOLUME("Line In Boost Volume", 0x1a, 0, HDA_INPUT), - { } -}; - -static const struct snd_kcontrol_new alc268_acer_dmic_mixer[] = { - /* output mixer control */ - HDA_BIND_VOL("Master Playback Volume", &alc268_acer_bind_master_vol), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Master Playback Switch", - .subdevice = HDA_SUBDEV_NID_FLAG | 0x14, - .info = snd_ctl_boolean_mono_info, - .get = alc268_acer_master_sw_get, - .put = alc268_acer_master_sw_put, - }, - HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), - HDA_CODEC_VOLUME("Line In Boost Volume", 0x1a, 0, HDA_INPUT), - { } -}; - -static const struct hda_verb alc268_acer_aspire_one_verbs[] = { - {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN}, - {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT}, - {0x23, AC_VERB_SET_CONNECT_SEL, 0x06}, - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, 0xa017}, - { } -}; - -static const struct hda_verb alc268_acer_verbs[] = { - {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* internal dmic? */ - {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN}, - { } -}; - -/* unsolicited event for HP jack sensing */ -#define alc268_toshiba_setup alc262_hippo_setup - -static void alc268_acer_lc_setup(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - spec->autocfg.hp_pins[0] = 0x15; - spec->autocfg.speaker_pins[0] = 0x14; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; - spec->ext_mic_pin = 0x18; - spec->int_mic_pin = 0x12; - spec->auto_mic = 1; -} - -static const struct snd_kcontrol_new alc268_dell_mixer[] = { - /* output mixer control */ - HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), - HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT), - { } -}; - -static const struct hda_verb alc268_dell_verbs[] = { - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN}, - {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_MIC_EVENT | AC_USRSP_EN}, - { } -}; - -/* mute/unmute internal speaker according to the hp jack and mute state */ -static void alc268_dell_setup(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - - spec->autocfg.hp_pins[0] = 0x15; - spec->autocfg.speaker_pins[0] = 0x14; - spec->ext_mic_pin = 0x18; - spec->int_mic_pin = 0x19; - spec->auto_mic = 1; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_PIN; -} - -static const struct snd_kcontrol_new alc267_quanta_il1_mixer[] = { - HDA_CODEC_VOLUME("Speaker Playback Volume", 0x2, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Headphone Playback Volume", 0x3, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Mic Capture Volume", 0x23, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Mic Capture Switch", 0x23, 2, HDA_OUTPUT), - HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), - HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT), - { } -}; - -static const struct hda_verb alc267_quanta_il1_verbs[] = { - {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN}, - {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_MIC_EVENT | AC_USRSP_EN}, - { } -}; - -static void alc267_quanta_il1_setup(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - spec->autocfg.hp_pins[0] = 0x15; - spec->autocfg.speaker_pins[0] = 0x14; - spec->ext_mic_pin = 0x18; - spec->int_mic_pin = 0x19; - spec->auto_mic = 1; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_PIN; -} - -/* - * generic initialization of ADC, input mixers and output mixers - */ -static const struct hda_verb alc268_base_init_verbs[] = { - /* Unmute DAC0-1 and set vol = 0 */ - {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - - /* - * Set up output mixers (0x0c - 0x0e) - */ - /* set vol=0 to output mixers */ - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x0e, AC_VERB_SET_CONNECT_SEL, 0x00}, - - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0}, - {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, - {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24}, - {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24}, - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20}, - {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20}, - {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20}, - - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - - /* set PCBEEP vol = 0, mute connections */ - {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - - /* Unmute Selector 23h,24h and set the default input to mic-in */ - - {0x23, AC_VERB_SET_CONNECT_SEL, 0x00}, - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x24, AC_VERB_SET_CONNECT_SEL, 0x00}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - - { } -}; - -/* only for model=test */ -#ifdef CONFIG_SND_DEBUG -/* - * generic initialization of ADC, input mixers and output mixers - */ -static const struct hda_verb alc268_volume_init_verbs[] = { - /* set output DAC */ - {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - - {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24}, - {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24}, - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20}, - {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20}, - {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20}, - - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - - {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - { } -}; -#endif /* CONFIG_SND_DEBUG */ - -static const struct snd_kcontrol_new alc268_capture_nosrc_mixer[] = { - HDA_CODEC_VOLUME("Capture Volume", 0x23, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Capture Switch", 0x23, 0x0, HDA_OUTPUT), - { } /* end */ -}; - -static const struct snd_kcontrol_new alc268_capture_alt_mixer[] = { - HDA_CODEC_VOLUME("Capture Volume", 0x23, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Capture Switch", 0x23, 0x0, HDA_OUTPUT), - _DEFINE_CAPSRC(1), - { } /* end */ -}; - -static const struct snd_kcontrol_new alc268_capture_mixer[] = { - HDA_CODEC_VOLUME("Capture Volume", 0x23, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Capture Switch", 0x23, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x24, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x24, 0x0, HDA_OUTPUT), - _DEFINE_CAPSRC(2), - { } /* end */ -}; - -static const struct hda_input_mux alc268_capture_source = { - .num_items = 4, - .items = { - { "Mic", 0x0 }, - { "Front Mic", 0x1 }, - { "Line", 0x2 }, - { "CD", 0x3 }, - }, -}; - -static const struct hda_input_mux alc268_acer_capture_source = { - .num_items = 3, - .items = { - { "Mic", 0x0 }, - { "Internal Mic", 0x1 }, - { "Line", 0x2 }, - }, -}; - -static const struct hda_input_mux alc268_acer_dmic_capture_source = { - .num_items = 3, - .items = { - { "Mic", 0x0 }, - { "Internal Mic", 0x6 }, - { "Line", 0x2 }, - }, -}; - -#ifdef CONFIG_SND_DEBUG -static const struct snd_kcontrol_new alc268_test_mixer[] = { - /* Volume widgets */ - HDA_CODEC_VOLUME("LOUT1 Playback Volume", 0x02, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("LOUT2 Playback Volume", 0x03, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE_MONO("Mono sum Playback Switch", 0x0e, 1, 2, HDA_INPUT), - HDA_BIND_MUTE("LINE-OUT sum Playback Switch", 0x0f, 2, HDA_INPUT), - HDA_BIND_MUTE("HP-OUT sum Playback Switch", 0x10, 2, HDA_INPUT), - HDA_BIND_MUTE("LINE-OUT Playback Switch", 0x14, 2, HDA_OUTPUT), - HDA_BIND_MUTE("HP-OUT Playback Switch", 0x15, 2, HDA_OUTPUT), - HDA_BIND_MUTE("Mono Playback Switch", 0x16, 2, HDA_OUTPUT), - HDA_CODEC_VOLUME("MIC1 Capture Volume", 0x18, 0x0, HDA_INPUT), - HDA_BIND_MUTE("MIC1 Capture Switch", 0x18, 2, HDA_OUTPUT), - HDA_CODEC_VOLUME("MIC2 Capture Volume", 0x19, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("LINE1 Capture Volume", 0x1a, 0x0, HDA_INPUT), - HDA_BIND_MUTE("LINE1 Capture Switch", 0x1a, 2, HDA_OUTPUT), - /* The below appears problematic on some hardwares */ - /*HDA_CODEC_VOLUME("PCBEEP Playback Volume", 0x1d, 0x0, HDA_INPUT),*/ - HDA_CODEC_VOLUME("PCM-IN1 Capture Volume", 0x23, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("PCM-IN1 Capture Switch", 0x23, 2, HDA_OUTPUT), - HDA_CODEC_VOLUME("PCM-IN2 Capture Volume", 0x24, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("PCM-IN2 Capture Switch", 0x24, 2, HDA_OUTPUT), - - /* Modes for retasking pin widgets */ - ALC_PIN_MODE("LINE-OUT pin mode", 0x14, ALC_PIN_DIR_INOUT), - ALC_PIN_MODE("HP-OUT pin mode", 0x15, ALC_PIN_DIR_INOUT), - ALC_PIN_MODE("MIC1 pin mode", 0x18, ALC_PIN_DIR_INOUT), - ALC_PIN_MODE("LINE1 pin mode", 0x1a, ALC_PIN_DIR_INOUT), - - /* Controls for GPIO pins, assuming they are configured as outputs */ - ALC_GPIO_DATA_SWITCH("GPIO pin 0", 0x01, 0x01), - ALC_GPIO_DATA_SWITCH("GPIO pin 1", 0x01, 0x02), - ALC_GPIO_DATA_SWITCH("GPIO pin 2", 0x01, 0x04), - ALC_GPIO_DATA_SWITCH("GPIO pin 3", 0x01, 0x08), - - /* Switches to allow the digital SPDIF output pin to be enabled. - * The ALC268 does not have an SPDIF input. - */ - ALC_SPDIF_CTRL_SWITCH("SPDIF Playback Switch", 0x06, 0x01), - - /* A switch allowing EAPD to be enabled. Some laptops seem to use - * this output to turn on an external amplifier. - */ - ALC_EAPD_CTRL_SWITCH("LINE-OUT EAPD Enable Switch", 0x0f, 0x02), - ALC_EAPD_CTRL_SWITCH("HP-OUT EAPD Enable Switch", 0x10, 0x02), - - { } /* end */ -}; -#endif - -/* - * configuration and preset - */ -static const char * const alc268_models[ALC268_MODEL_LAST] = { - [ALC267_QUANTA_IL1] = "quanta-il1", - [ALC268_3ST] = "3stack", - [ALC268_TOSHIBA] = "toshiba", - [ALC268_ACER] = "acer", - [ALC268_ACER_DMIC] = "acer-dmic", - [ALC268_ACER_ASPIRE_ONE] = "acer-aspire", - [ALC268_DELL] = "dell", - [ALC268_ZEPTO] = "zepto", -#ifdef CONFIG_SND_DEBUG - [ALC268_TEST] = "test", -#endif - [ALC268_AUTO] = "auto", -}; - -static const struct snd_pci_quirk alc268_cfg_tbl[] = { - SND_PCI_QUIRK(0x1025, 0x011e, "Acer Aspire 5720z", ALC268_ACER), - SND_PCI_QUIRK(0x1025, 0x0126, "Acer", ALC268_ACER), - SND_PCI_QUIRK(0x1025, 0x012e, "Acer Aspire 5310", ALC268_ACER), - SND_PCI_QUIRK(0x1025, 0x0130, "Acer Extensa 5210", ALC268_ACER), - SND_PCI_QUIRK(0x1025, 0x0136, "Acer Aspire 5315", ALC268_ACER), - SND_PCI_QUIRK(0x1025, 0x015b, "Acer Aspire One", - ALC268_ACER_ASPIRE_ONE), - SND_PCI_QUIRK(0x1028, 0x0253, "Dell OEM", ALC268_DELL), - SND_PCI_QUIRK(0x1028, 0x02b0, "Dell Inspiron 910", ALC268_AUTO), - SND_PCI_QUIRK_MASK(0x1028, 0xfff0, 0x02b0, - "Dell Inspiron Mini9/Vostro A90", ALC268_DELL), - /* almost compatible with toshiba but with optional digital outs; - * auto-probing seems working fine - */ - SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x3000, "HP TX25xx series", - ALC268_AUTO), - SND_PCI_QUIRK(0x1043, 0x1205, "ASUS W7J", ALC268_3ST), - SND_PCI_QUIRK(0x1170, 0x0040, "ZEPTO", ALC268_ZEPTO), - SND_PCI_QUIRK(0x14c0, 0x0025, "COMPAL IFL90/JFL-92", ALC268_TOSHIBA), - SND_PCI_QUIRK(0x152d, 0x0771, "Quanta IL1", ALC267_QUANTA_IL1), - {} -}; - -/* Toshiba laptops have no unique PCI SSID but only codec SSID */ -static const struct snd_pci_quirk alc268_ssid_cfg_tbl[] = { - SND_PCI_QUIRK(0x1179, 0xff0a, "TOSHIBA X-200", ALC268_AUTO), - SND_PCI_QUIRK(0x1179, 0xff0e, "TOSHIBA X-200 HDMI", ALC268_AUTO), - SND_PCI_QUIRK_MASK(0x1179, 0xff00, 0xff00, "TOSHIBA A/Lx05", - ALC268_TOSHIBA), - {} -}; - -static const struct alc_config_preset alc268_presets[] = { - [ALC267_QUANTA_IL1] = { - .mixers = { alc267_quanta_il1_mixer, alc268_beep_mixer }, - .cap_mixer = alc268_capture_nosrc_mixer, - .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs, - alc267_quanta_il1_verbs }, - .num_dacs = ARRAY_SIZE(alc268_dac_nids), - .dac_nids = alc268_dac_nids, - .num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt), - .adc_nids = alc268_adc_nids_alt, - .hp_nid = 0x03, - .num_channel_mode = ARRAY_SIZE(alc268_modes), - .channel_mode = alc268_modes, - .unsol_event = alc_sku_unsol_event, - .setup = alc267_quanta_il1_setup, - .init_hook = alc_inithook, - }, - [ALC268_3ST] = { - .mixers = { alc268_base_mixer, alc268_beep_mixer }, - .cap_mixer = alc268_capture_alt_mixer, - .init_verbs = { alc268_base_init_verbs }, - .num_dacs = ARRAY_SIZE(alc268_dac_nids), - .dac_nids = alc268_dac_nids, - .num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt), - .adc_nids = alc268_adc_nids_alt, - .capsrc_nids = alc268_capsrc_nids, - .hp_nid = 0x03, - .dig_out_nid = ALC268_DIGOUT_NID, - .num_channel_mode = ARRAY_SIZE(alc268_modes), - .channel_mode = alc268_modes, - .input_mux = &alc268_capture_source, - }, - [ALC268_TOSHIBA] = { - .mixers = { alc268_toshiba_mixer, alc268_beep_mixer }, - .cap_mixer = alc268_capture_alt_mixer, - .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs, - alc268_toshiba_verbs }, - .num_dacs = ARRAY_SIZE(alc268_dac_nids), - .dac_nids = alc268_dac_nids, - .num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt), - .adc_nids = alc268_adc_nids_alt, - .capsrc_nids = alc268_capsrc_nids, - .hp_nid = 0x03, - .num_channel_mode = ARRAY_SIZE(alc268_modes), - .channel_mode = alc268_modes, - .input_mux = &alc268_capture_source, - .unsol_event = alc_sku_unsol_event, - .setup = alc268_toshiba_setup, - .init_hook = alc_inithook, - }, - [ALC268_ACER] = { - .mixers = { alc268_acer_mixer, alc268_beep_mixer }, - .cap_mixer = alc268_capture_alt_mixer, - .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs, - alc268_acer_verbs }, - .num_dacs = ARRAY_SIZE(alc268_dac_nids), - .dac_nids = alc268_dac_nids, - .num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt), - .adc_nids = alc268_adc_nids_alt, - .capsrc_nids = alc268_capsrc_nids, - .hp_nid = 0x02, - .num_channel_mode = ARRAY_SIZE(alc268_modes), - .channel_mode = alc268_modes, - .input_mux = &alc268_acer_capture_source, - .unsol_event = alc_sku_unsol_event, - .setup = alc268_acer_setup, - .init_hook = alc_inithook, - }, - [ALC268_ACER_DMIC] = { - .mixers = { alc268_acer_dmic_mixer, alc268_beep_mixer }, - .cap_mixer = alc268_capture_alt_mixer, - .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs, - alc268_acer_verbs }, - .num_dacs = ARRAY_SIZE(alc268_dac_nids), - .dac_nids = alc268_dac_nids, - .num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt), - .adc_nids = alc268_adc_nids_alt, - .capsrc_nids = alc268_capsrc_nids, - .hp_nid = 0x02, - .num_channel_mode = ARRAY_SIZE(alc268_modes), - .channel_mode = alc268_modes, - .input_mux = &alc268_acer_dmic_capture_source, - .unsol_event = alc_sku_unsol_event, - .setup = alc268_acer_setup, - .init_hook = alc_inithook, - }, - [ALC268_ACER_ASPIRE_ONE] = { - .mixers = { alc268_acer_aspire_one_mixer, alc268_beep_mixer}, - .cap_mixer = alc268_capture_nosrc_mixer, - .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs, - alc268_acer_aspire_one_verbs }, - .num_dacs = ARRAY_SIZE(alc268_dac_nids), - .dac_nids = alc268_dac_nids, - .num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt), - .adc_nids = alc268_adc_nids_alt, - .capsrc_nids = alc268_capsrc_nids, - .hp_nid = 0x03, - .num_channel_mode = ARRAY_SIZE(alc268_modes), - .channel_mode = alc268_modes, - .unsol_event = alc_sku_unsol_event, - .setup = alc268_acer_lc_setup, - .init_hook = alc_inithook, - }, - [ALC268_DELL] = { - .mixers = { alc268_dell_mixer, alc268_beep_mixer}, - .cap_mixer = alc268_capture_nosrc_mixer, - .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs, - alc268_dell_verbs }, - .num_dacs = ARRAY_SIZE(alc268_dac_nids), - .dac_nids = alc268_dac_nids, - .num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt), - .adc_nids = alc268_adc_nids_alt, - .capsrc_nids = alc268_capsrc_nids, - .hp_nid = 0x02, - .num_channel_mode = ARRAY_SIZE(alc268_modes), - .channel_mode = alc268_modes, - .unsol_event = alc_sku_unsol_event, - .setup = alc268_dell_setup, - .init_hook = alc_inithook, - }, - [ALC268_ZEPTO] = { - .mixers = { alc268_base_mixer, alc268_beep_mixer }, - .cap_mixer = alc268_capture_alt_mixer, - .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs, - alc268_toshiba_verbs }, - .num_dacs = ARRAY_SIZE(alc268_dac_nids), - .dac_nids = alc268_dac_nids, - .num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt), - .adc_nids = alc268_adc_nids_alt, - .capsrc_nids = alc268_capsrc_nids, - .hp_nid = 0x03, - .dig_out_nid = ALC268_DIGOUT_NID, - .num_channel_mode = ARRAY_SIZE(alc268_modes), - .channel_mode = alc268_modes, - .input_mux = &alc268_capture_source, - .unsol_event = alc_sku_unsol_event, - .setup = alc268_toshiba_setup, - .init_hook = alc_inithook, - }, -#ifdef CONFIG_SND_DEBUG - [ALC268_TEST] = { - .mixers = { alc268_test_mixer }, - .cap_mixer = alc268_capture_mixer, - .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs, - alc268_volume_init_verbs, - alc268_beep_init_verbs }, - .num_dacs = ARRAY_SIZE(alc268_dac_nids), - .dac_nids = alc268_dac_nids, - .num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt), - .adc_nids = alc268_adc_nids_alt, - .capsrc_nids = alc268_capsrc_nids, - .hp_nid = 0x03, - .dig_out_nid = ALC268_DIGOUT_NID, - .num_channel_mode = ARRAY_SIZE(alc268_modes), - .channel_mode = alc268_modes, - .input_mux = &alc268_capture_source, - }, -#endif -}; - diff --git a/sound/pci/hda/alc269_quirks.c b/sound/pci/hda/alc269_quirks.c deleted file mode 100644 index 5ac0e2162a46..000000000000 --- a/sound/pci/hda/alc269_quirks.c +++ /dev/null @@ -1,674 +0,0 @@ -/* - * ALC269/ALC270/ALC275/ALC276 quirk models - * included by patch_realtek.c - */ - -/* ALC269 models */ -enum { - ALC269_AUTO, - ALC269_BASIC, - ALC269_QUANTA_FL1, - ALC269_AMIC, - ALC269_DMIC, - ALC269VB_AMIC, - ALC269VB_DMIC, - ALC269_FUJITSU, - ALC269_LIFEBOOK, - ALC271_ACER, - ALC269_MODEL_LAST /* last tag */ -}; - -/* - * ALC269 channel source setting (2 channel) - */ -#define ALC269_DIGOUT_NID ALC880_DIGOUT_NID - -#define alc269_dac_nids alc260_dac_nids - -static const hda_nid_t alc269_adc_nids[1] = { - /* ADC1 */ - 0x08, -}; - -static const hda_nid_t alc269_capsrc_nids[1] = { - 0x23, -}; - -static const hda_nid_t alc269vb_adc_nids[1] = { - /* ADC1 */ - 0x09, -}; - -static const hda_nid_t alc269vb_capsrc_nids[1] = { - 0x22, -}; - -#define alc269_modes alc260_modes -#define alc269_capture_source alc880_lg_lw_capture_source - -static const struct snd_kcontrol_new alc269_base_mixer[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), - HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x16, 2, 0x0, HDA_OUTPUT), - { } /* end */ -}; - -static const struct snd_kcontrol_new alc269_quanta_fl1_mixer[] = { - /* output mixer control */ - HDA_BIND_VOL("Master Playback Volume", &alc268_acer_bind_master_vol), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Master Playback Switch", - .subdevice = HDA_SUBDEV_AMP_FLAG, - .info = snd_hda_mixer_amp_switch_info, - .get = snd_hda_mixer_amp_switch_get, - .put = alc268_acer_master_sw_put, - .private_value = HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT), - }, - HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), - HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), - HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x01, HDA_INPUT), - HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT), - { } -}; - -static const struct snd_kcontrol_new alc269_lifebook_mixer[] = { - /* output mixer control */ - HDA_BIND_VOL("Master Playback Volume", &alc268_acer_bind_master_vol), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Master Playback Switch", - .subdevice = HDA_SUBDEV_AMP_FLAG, - .info = snd_hda_mixer_amp_switch_info, - .get = snd_hda_mixer_amp_switch_get, - .put = alc268_acer_master_sw_put, - .private_value = HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT), - }, - HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), - HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), - HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x01, HDA_INPUT), - HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT), - HDA_CODEC_VOLUME("Dock Mic Playback Volume", 0x0b, 0x03, HDA_INPUT), - HDA_CODEC_MUTE("Dock Mic Playback Switch", 0x0b, 0x03, HDA_INPUT), - HDA_CODEC_VOLUME("Dock Mic Boost Volume", 0x1b, 0, HDA_INPUT), - { } -}; - -static const struct snd_kcontrol_new alc269_laptop_mixer[] = { - HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT), - { } /* end */ -}; - -static const struct snd_kcontrol_new alc269vb_laptop_mixer[] = { - HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x21, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT), - { } /* end */ -}; - -static const struct snd_kcontrol_new alc269_asus_mixer[] = { - HDA_CODEC_VOLUME("Master Playback Volume", 0x02, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Master Playback Switch", 0x0c, 0x0, HDA_INPUT), - { } /* end */ -}; - -/* capture mixer elements */ -static const struct snd_kcontrol_new alc269_laptop_analog_capture_mixer[] = { - HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), - HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT), - { } /* end */ -}; - -static const struct snd_kcontrol_new alc269_laptop_digital_capture_mixer[] = { - HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), - { } /* end */ -}; - -static const struct snd_kcontrol_new alc269vb_laptop_analog_capture_mixer[] = { - HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), - HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT), - { } /* end */ -}; - -static const struct snd_kcontrol_new alc269vb_laptop_digital_capture_mixer[] = { - HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), - { } /* end */ -}; - -/* FSC amilo */ -#define alc269_fujitsu_mixer alc269_laptop_mixer - -static const struct hda_verb alc269_quanta_fl1_verbs[] = { - {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, - {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN}, - {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT}, - {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - { } -}; - -static const struct hda_verb alc269_lifebook_verbs[] = { - {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, - {0x1a, AC_VERB_SET_CONNECT_SEL, 0x01}, - {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN}, - {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN}, - {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT}, - {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - { } -}; - -/* toggle speaker-output according to the hp-jack state */ -static void alc269_quanta_fl1_speaker_automute(struct hda_codec *codec) -{ - alc_hp_automute(codec); - - snd_hda_codec_write(codec, 0x20, 0, - AC_VERB_SET_COEF_INDEX, 0x0c); - snd_hda_codec_write(codec, 0x20, 0, - AC_VERB_SET_PROC_COEF, 0x680); - - snd_hda_codec_write(codec, 0x20, 0, - AC_VERB_SET_COEF_INDEX, 0x0c); - snd_hda_codec_write(codec, 0x20, 0, - AC_VERB_SET_PROC_COEF, 0x480); -} - -#define alc269_lifebook_speaker_automute \ - alc269_quanta_fl1_speaker_automute - -static void alc269_lifebook_mic_autoswitch(struct hda_codec *codec) -{ - unsigned int present_laptop; - unsigned int present_dock; - - present_laptop = snd_hda_jack_detect(codec, 0x18); - present_dock = snd_hda_jack_detect(codec, 0x1b); - - /* Laptop mic port overrides dock mic port, design decision */ - if (present_dock) - snd_hda_codec_write(codec, 0x23, 0, - AC_VERB_SET_CONNECT_SEL, 0x3); - if (present_laptop) - snd_hda_codec_write(codec, 0x23, 0, - AC_VERB_SET_CONNECT_SEL, 0x0); - if (!present_dock && !present_laptop) - snd_hda_codec_write(codec, 0x23, 0, - AC_VERB_SET_CONNECT_SEL, 0x1); -} - -static void alc269_quanta_fl1_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - switch (res >> 26) { - case ALC_HP_EVENT: - alc269_quanta_fl1_speaker_automute(codec); - break; - case ALC_MIC_EVENT: - alc_mic_automute(codec); - break; - } -} - -static void alc269_lifebook_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - if ((res >> 26) == ALC_HP_EVENT) - alc269_lifebook_speaker_automute(codec); - if ((res >> 26) == ALC_MIC_EVENT) - alc269_lifebook_mic_autoswitch(codec); -} - -static void alc269_quanta_fl1_setup(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - spec->autocfg.hp_pins[0] = 0x15; - spec->autocfg.speaker_pins[0] = 0x14; - spec->automute_mixer_nid[0] = 0x0c; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_MIXER; - spec->ext_mic_pin = 0x18; - spec->int_mic_pin = 0x19; - spec->auto_mic = 1; -} - -static void alc269_quanta_fl1_init_hook(struct hda_codec *codec) -{ - alc269_quanta_fl1_speaker_automute(codec); - alc_mic_automute(codec); -} - -static void alc269_lifebook_setup(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - spec->autocfg.hp_pins[0] = 0x15; - spec->autocfg.hp_pins[1] = 0x1a; - spec->autocfg.speaker_pins[0] = 0x14; - spec->automute_mixer_nid[0] = 0x0c; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_MIXER; -} - -static void alc269_lifebook_init_hook(struct hda_codec *codec) -{ - alc269_lifebook_speaker_automute(codec); - alc269_lifebook_mic_autoswitch(codec); -} - -static const struct hda_verb alc269_laptop_dmic_init_verbs[] = { - {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, - {0x23, AC_VERB_SET_CONNECT_SEL, 0x05}, - {0x02, AC_VERB_SET_AMP_GAIN_MUTE, 0xb026 }, - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, (0x7019 | (0x00 << 8))}, - {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT}, - {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT}, - {} -}; - -static const struct hda_verb alc269_laptop_amic_init_verbs[] = { - {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, - {0x23, AC_VERB_SET_CONNECT_SEL, 0x01}, - {0x02, AC_VERB_SET_AMP_GAIN_MUTE, 0xb026 }, - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, (0x701b | (0x00 << 8))}, - {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT}, - {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT}, - {} -}; - -static const struct hda_verb alc269vb_laptop_dmic_init_verbs[] = { - {0x21, AC_VERB_SET_CONNECT_SEL, 0x01}, - {0x22, AC_VERB_SET_CONNECT_SEL, 0x06}, - {0x02, AC_VERB_SET_AMP_GAIN_MUTE, 0xb026 }, - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, (0x7019 | (0x00 << 8))}, - {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT}, - {0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT}, - {} -}; - -static const struct hda_verb alc269vb_laptop_amic_init_verbs[] = { - {0x21, AC_VERB_SET_CONNECT_SEL, 0x01}, - {0x22, AC_VERB_SET_CONNECT_SEL, 0x01}, - {0x02, AC_VERB_SET_AMP_GAIN_MUTE, 0xb026 }, - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, (0x7019 | (0x00 << 8))}, - {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT}, - {0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT}, - {} -}; - -static const struct hda_verb alc271_acer_dmic_verbs[] = { - {0x20, AC_VERB_SET_COEF_INDEX, 0x0d}, - {0x20, AC_VERB_SET_PROC_COEF, 0x4000}, - {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x21, AC_VERB_SET_CONNECT_SEL, 0x00}, - {0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT}, - {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT}, - {0x22, AC_VERB_SET_CONNECT_SEL, 6}, - { } -}; - -static void alc269_laptop_amic_setup(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - spec->autocfg.hp_pins[0] = 0x15; - spec->autocfg.speaker_pins[0] = 0x14; - spec->automute_mixer_nid[0] = 0x0c; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_MIXER; - spec->ext_mic_pin = 0x18; - spec->int_mic_pin = 0x19; - spec->auto_mic = 1; -} - -static void alc269_laptop_dmic_setup(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - spec->autocfg.hp_pins[0] = 0x15; - spec->autocfg.speaker_pins[0] = 0x14; - spec->automute_mixer_nid[0] = 0x0c; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_MIXER; - spec->ext_mic_pin = 0x18; - spec->int_mic_pin = 0x12; - spec->auto_mic = 1; -} - -static void alc269vb_laptop_amic_setup(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - spec->autocfg.hp_pins[0] = 0x21; - spec->autocfg.speaker_pins[0] = 0x14; - spec->automute_mixer_nid[0] = 0x0c; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_MIXER; - spec->ext_mic_pin = 0x18; - spec->int_mic_pin = 0x19; - spec->auto_mic = 1; -} - -static void alc269vb_laptop_dmic_setup(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - spec->autocfg.hp_pins[0] = 0x21; - spec->autocfg.speaker_pins[0] = 0x14; - spec->automute_mixer_nid[0] = 0x0c; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_MIXER; - spec->ext_mic_pin = 0x18; - spec->int_mic_pin = 0x12; - spec->auto_mic = 1; -} - -/* - * generic initialization of ADC, input mixers and output mixers - */ -static const struct hda_verb alc269_init_verbs[] = { - /* - * Unmute ADC0 and set the default input to mic-in - */ - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - - /* - * Set up output mixers (0x02 - 0x03) - */ - /* set vol=0 to output mixers */ - {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - - /* set up input amps for analog loopback */ - /* Amp Indices: DAC = 0, mixer = 1 */ - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - - /* FIXME: use Mux-type input source selection */ - /* Mixer elements: 0x18, 19, 1a, 1b, 1d, 0b */ - /* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */ - {0x23, AC_VERB_SET_CONNECT_SEL, 0x00}, - - /* set EAPD */ - {0x14, AC_VERB_SET_EAPD_BTLENABLE, 2}, - { } -}; - -static const struct hda_verb alc269vb_init_verbs[] = { - /* - * Unmute ADC0 and set the default input to mic-in - */ - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - - /* - * Set up output mixers (0x02 - 0x03) - */ - /* set vol=0 to output mixers */ - {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - - /* set up input amps for analog loopback */ - /* Amp Indices: DAC = 0, mixer = 1 */ - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - - /* FIXME: use Mux-type input source selection */ - /* Mixer elements: 0x18, 19, 1a, 1b, 1d, 0b */ - /* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */ - {0x22, AC_VERB_SET_CONNECT_SEL, 0x00}, - - /* set EAPD */ - {0x14, AC_VERB_SET_EAPD_BTLENABLE, 2}, - { } -}; - -/* - * configuration and preset - */ -static const char * const alc269_models[ALC269_MODEL_LAST] = { - [ALC269_BASIC] = "basic", - [ALC269_QUANTA_FL1] = "quanta", - [ALC269_AMIC] = "laptop-amic", - [ALC269_DMIC] = "laptop-dmic", - [ALC269_FUJITSU] = "fujitsu", - [ALC269_LIFEBOOK] = "lifebook", - [ALC269_AUTO] = "auto", -}; - -static const struct snd_pci_quirk alc269_cfg_tbl[] = { - SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_QUANTA_FL1), - SND_PCI_QUIRK(0x1025, 0x047c, "ACER ZGA", ALC271_ACER), - SND_PCI_QUIRK(0x1043, 0x8330, "ASUS Eeepc P703 P900A", - ALC269_AMIC), - SND_PCI_QUIRK(0x1043, 0x1013, "ASUS N61Da", ALC269VB_AMIC), - SND_PCI_QUIRK(0x1043, 0x1113, "ASUS N63Jn", ALC269VB_AMIC), - SND_PCI_QUIRK(0x1043, 0x1143, "ASUS B53f", ALC269VB_AMIC), - SND_PCI_QUIRK(0x1043, 0x1133, "ASUS UJ20ft", ALC269_AMIC), - SND_PCI_QUIRK(0x1043, 0x1183, "ASUS K72DR", ALC269VB_AMIC), - SND_PCI_QUIRK(0x1043, 0x11b3, "ASUS K52DR", ALC269VB_AMIC), - SND_PCI_QUIRK(0x1043, 0x11e3, "ASUS U33Jc", ALC269VB_AMIC), - SND_PCI_QUIRK(0x1043, 0x1273, "ASUS UL80Jt", ALC269VB_AMIC), - SND_PCI_QUIRK(0x1043, 0x1283, "ASUS U53Jc", ALC269_AMIC), - SND_PCI_QUIRK(0x1043, 0x12b3, "ASUS N82JV", ALC269VB_AMIC), - SND_PCI_QUIRK(0x1043, 0x12d3, "ASUS N61Jv", ALC269_AMIC), - SND_PCI_QUIRK(0x1043, 0x13a3, "ASUS UL30Vt", ALC269_AMIC), - SND_PCI_QUIRK(0x1043, 0x1373, "ASUS G73JX", ALC269_AMIC), - SND_PCI_QUIRK(0x1043, 0x1383, "ASUS UJ30Jc", ALC269_AMIC), - SND_PCI_QUIRK(0x1043, 0x13d3, "ASUS N61JA", ALC269_AMIC), - SND_PCI_QUIRK(0x1043, 0x1413, "ASUS UL50", ALC269_AMIC), - SND_PCI_QUIRK(0x1043, 0x1443, "ASUS UL30", ALC269_AMIC), - SND_PCI_QUIRK(0x1043, 0x1453, "ASUS M60Jv", ALC269_AMIC), - SND_PCI_QUIRK(0x1043, 0x1483, "ASUS UL80", ALC269_AMIC), - SND_PCI_QUIRK(0x1043, 0x14f3, "ASUS F83Vf", ALC269_AMIC), - SND_PCI_QUIRK(0x1043, 0x14e3, "ASUS UL20", ALC269_AMIC), - SND_PCI_QUIRK(0x1043, 0x1513, "ASUS UX30", ALC269_AMIC), - SND_PCI_QUIRK(0x1043, 0x1593, "ASUS N51Vn", ALC269_AMIC), - SND_PCI_QUIRK(0x1043, 0x15a3, "ASUS N60Jv", ALC269_AMIC), - SND_PCI_QUIRK(0x1043, 0x15b3, "ASUS N60Dp", ALC269_AMIC), - SND_PCI_QUIRK(0x1043, 0x15c3, "ASUS N70De", ALC269_AMIC), - SND_PCI_QUIRK(0x1043, 0x15e3, "ASUS F83T", ALC269_AMIC), - SND_PCI_QUIRK(0x1043, 0x1643, "ASUS M60J", ALC269_AMIC), - SND_PCI_QUIRK(0x1043, 0x1653, "ASUS U50", ALC269_AMIC), - SND_PCI_QUIRK(0x1043, 0x1693, "ASUS F50N", ALC269_AMIC), - SND_PCI_QUIRK(0x1043, 0x16a3, "ASUS F5Q", ALC269_AMIC), - SND_PCI_QUIRK(0x1043, 0x1723, "ASUS P80", ALC269_AMIC), - SND_PCI_QUIRK(0x1043, 0x1743, "ASUS U80", ALC269_AMIC), - SND_PCI_QUIRK(0x1043, 0x1773, "ASUS U20A", ALC269_AMIC), - SND_PCI_QUIRK(0x1043, 0x1883, "ASUS F81Se", ALC269_AMIC), - SND_PCI_QUIRK(0x104d, 0x9071, "Sony VAIO", ALC269_AUTO), - SND_PCI_QUIRK(0x10cf, 0x1475, "Lifebook ICH9M-based", ALC269_LIFEBOOK), - SND_PCI_QUIRK(0x152d, 0x1778, "Quanta ON1", ALC269_DMIC), - SND_PCI_QUIRK(0x1734, 0x115d, "FSC Amilo", ALC269_FUJITSU), - SND_PCI_QUIRK(0x17aa, 0x3be9, "Quanta Wistron", ALC269_AMIC), - SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_AMIC), - SND_PCI_QUIRK(0x17ff, 0x059a, "Quanta EL3", ALC269_DMIC), - SND_PCI_QUIRK(0x17ff, 0x059b, "Quanta JR1", ALC269_DMIC), - {} -}; - -static const struct alc_config_preset alc269_presets[] = { - [ALC269_BASIC] = { - .mixers = { alc269_base_mixer }, - .init_verbs = { alc269_init_verbs }, - .num_dacs = ARRAY_SIZE(alc269_dac_nids), - .dac_nids = alc269_dac_nids, - .hp_nid = 0x03, - .num_channel_mode = ARRAY_SIZE(alc269_modes), - .channel_mode = alc269_modes, - .input_mux = &alc269_capture_source, - }, - [ALC269_QUANTA_FL1] = { - .mixers = { alc269_quanta_fl1_mixer }, - .init_verbs = { alc269_init_verbs, alc269_quanta_fl1_verbs }, - .num_dacs = ARRAY_SIZE(alc269_dac_nids), - .dac_nids = alc269_dac_nids, - .hp_nid = 0x03, - .num_channel_mode = ARRAY_SIZE(alc269_modes), - .channel_mode = alc269_modes, - .input_mux = &alc269_capture_source, - .unsol_event = alc269_quanta_fl1_unsol_event, - .setup = alc269_quanta_fl1_setup, - .init_hook = alc269_quanta_fl1_init_hook, - }, - [ALC269_AMIC] = { - .mixers = { alc269_laptop_mixer }, - .cap_mixer = alc269_laptop_analog_capture_mixer, - .init_verbs = { alc269_init_verbs, - alc269_laptop_amic_init_verbs }, - .num_dacs = ARRAY_SIZE(alc269_dac_nids), - .dac_nids = alc269_dac_nids, - .hp_nid = 0x03, - .num_channel_mode = ARRAY_SIZE(alc269_modes), - .channel_mode = alc269_modes, - .unsol_event = alc_sku_unsol_event, - .setup = alc269_laptop_amic_setup, - .init_hook = alc_inithook, - }, - [ALC269_DMIC] = { - .mixers = { alc269_laptop_mixer }, - .cap_mixer = alc269_laptop_digital_capture_mixer, - .init_verbs = { alc269_init_verbs, - alc269_laptop_dmic_init_verbs }, - .num_dacs = ARRAY_SIZE(alc269_dac_nids), - .dac_nids = alc269_dac_nids, - .hp_nid = 0x03, - .num_channel_mode = ARRAY_SIZE(alc269_modes), - .channel_mode = alc269_modes, - .unsol_event = alc_sku_unsol_event, - .setup = alc269_laptop_dmic_setup, - .init_hook = alc_inithook, - }, - [ALC269VB_AMIC] = { - .mixers = { alc269vb_laptop_mixer }, - .cap_mixer = alc269vb_laptop_analog_capture_mixer, - .init_verbs = { alc269vb_init_verbs, - alc269vb_laptop_amic_init_verbs }, - .num_dacs = ARRAY_SIZE(alc269_dac_nids), - .dac_nids = alc269_dac_nids, - .hp_nid = 0x03, - .num_channel_mode = ARRAY_SIZE(alc269_modes), - .channel_mode = alc269_modes, - .unsol_event = alc_sku_unsol_event, - .setup = alc269vb_laptop_amic_setup, - .init_hook = alc_inithook, - }, - [ALC269VB_DMIC] = { - .mixers = { alc269vb_laptop_mixer }, - .cap_mixer = alc269vb_laptop_digital_capture_mixer, - .init_verbs = { alc269vb_init_verbs, - alc269vb_laptop_dmic_init_verbs }, - .num_dacs = ARRAY_SIZE(alc269_dac_nids), - .dac_nids = alc269_dac_nids, - .hp_nid = 0x03, - .num_channel_mode = ARRAY_SIZE(alc269_modes), - .channel_mode = alc269_modes, - .unsol_event = alc_sku_unsol_event, - .setup = alc269vb_laptop_dmic_setup, - .init_hook = alc_inithook, - }, - [ALC269_FUJITSU] = { - .mixers = { alc269_fujitsu_mixer }, - .cap_mixer = alc269_laptop_digital_capture_mixer, - .init_verbs = { alc269_init_verbs, - alc269_laptop_dmic_init_verbs }, - .num_dacs = ARRAY_SIZE(alc269_dac_nids), - .dac_nids = alc269_dac_nids, - .hp_nid = 0x03, - .num_channel_mode = ARRAY_SIZE(alc269_modes), - .channel_mode = alc269_modes, - .unsol_event = alc_sku_unsol_event, - .setup = alc269_laptop_dmic_setup, - .init_hook = alc_inithook, - }, - [ALC269_LIFEBOOK] = { - .mixers = { alc269_lifebook_mixer }, - .init_verbs = { alc269_init_verbs, alc269_lifebook_verbs }, - .num_dacs = ARRAY_SIZE(alc269_dac_nids), - .dac_nids = alc269_dac_nids, - .hp_nid = 0x03, - .num_channel_mode = ARRAY_SIZE(alc269_modes), - .channel_mode = alc269_modes, - .input_mux = &alc269_capture_source, - .unsol_event = alc269_lifebook_unsol_event, - .setup = alc269_lifebook_setup, - .init_hook = alc269_lifebook_init_hook, - }, - [ALC271_ACER] = { - .mixers = { alc269_asus_mixer }, - .cap_mixer = alc269vb_laptop_digital_capture_mixer, - .init_verbs = { alc269_init_verbs, alc271_acer_dmic_verbs }, - .num_dacs = ARRAY_SIZE(alc269_dac_nids), - .dac_nids = alc269_dac_nids, - .adc_nids = alc262_dmic_adc_nids, - .num_adc_nids = ARRAY_SIZE(alc262_dmic_adc_nids), - .capsrc_nids = alc262_dmic_capsrc_nids, - .num_channel_mode = ARRAY_SIZE(alc269_modes), - .channel_mode = alc269_modes, - .input_mux = &alc269_capture_source, - .dig_out_nid = ALC880_DIGOUT_NID, - .unsol_event = alc_sku_unsol_event, - .setup = alc269vb_laptop_dmic_setup, - .init_hook = alc_inithook, - }, -}; - diff --git a/sound/pci/hda/alc662_quirks.c b/sound/pci/hda/alc662_quirks.c deleted file mode 100644 index e69a6ea3083a..000000000000 --- a/sound/pci/hda/alc662_quirks.c +++ /dev/null @@ -1,1408 +0,0 @@ -/* - * ALC662/ALC663/ALC665/ALC670 quirk models - * included by patch_realtek.c - */ - -/* ALC662 models */ -enum { - ALC662_AUTO, - ALC662_3ST_2ch_DIG, - ALC662_3ST_6ch_DIG, - ALC662_3ST_6ch, - ALC662_5ST_DIG, - ALC662_LENOVO_101E, - ALC662_ASUS_EEEPC_P701, - ALC662_ASUS_EEEPC_EP20, - ALC663_ASUS_M51VA, - ALC663_ASUS_G71V, - ALC663_ASUS_H13, - ALC663_ASUS_G50V, - ALC662_ECS, - ALC663_ASUS_MODE1, - ALC662_ASUS_MODE2, - ALC663_ASUS_MODE3, - ALC663_ASUS_MODE4, - ALC663_ASUS_MODE5, - ALC663_ASUS_MODE6, - ALC663_ASUS_MODE7, - ALC663_ASUS_MODE8, - ALC272_DELL, - ALC272_DELL_ZM1, - ALC272_SAMSUNG_NC10, - ALC662_MODEL_LAST, -}; - -#define ALC662_DIGOUT_NID 0x06 -#define ALC662_DIGIN_NID 0x0a - -static const hda_nid_t alc662_dac_nids[3] = { - /* front, rear, clfe */ - 0x02, 0x03, 0x04 -}; - -static const hda_nid_t alc272_dac_nids[2] = { - 0x02, 0x03 -}; - -static const hda_nid_t alc662_adc_nids[2] = { - /* ADC1-2 */ - 0x09, 0x08 -}; - -static const hda_nid_t alc272_adc_nids[1] = { - /* ADC1-2 */ - 0x08, -}; - -static const hda_nid_t alc662_capsrc_nids[2] = { 0x22, 0x23 }; -static const hda_nid_t alc272_capsrc_nids[1] = { 0x23 }; - - -/* input MUX */ -/* FIXME: should be a matrix-type input source selection */ -static const struct hda_input_mux alc662_capture_source = { - .num_items = 4, - .items = { - { "Mic", 0x0 }, - { "Front Mic", 0x1 }, - { "Line", 0x2 }, - { "CD", 0x4 }, - }, -}; - -static const struct hda_input_mux alc662_lenovo_101e_capture_source = { - .num_items = 2, - .items = { - { "Mic", 0x1 }, - { "Line", 0x2 }, - }, -}; - -static const struct hda_input_mux alc663_capture_source = { - .num_items = 3, - .items = { - { "Mic", 0x0 }, - { "Front Mic", 0x1 }, - { "Line", 0x2 }, - }, -}; - -#if 0 /* set to 1 for testing other input sources below */ -static const struct hda_input_mux alc272_nc10_capture_source = { - .num_items = 16, - .items = { - { "Autoselect Mic", 0x0 }, - { "Internal Mic", 0x1 }, - { "In-0x02", 0x2 }, - { "In-0x03", 0x3 }, - { "In-0x04", 0x4 }, - { "In-0x05", 0x5 }, - { "In-0x06", 0x6 }, - { "In-0x07", 0x7 }, - { "In-0x08", 0x8 }, - { "In-0x09", 0x9 }, - { "In-0x0a", 0x0a }, - { "In-0x0b", 0x0b }, - { "In-0x0c", 0x0c }, - { "In-0x0d", 0x0d }, - { "In-0x0e", 0x0e }, - { "In-0x0f", 0x0f }, - }, -}; -#endif - -/* - * 2ch mode - */ -static const struct hda_channel_mode alc662_3ST_2ch_modes[1] = { - { 2, NULL } -}; - -/* - * 2ch mode - */ -static const struct hda_verb alc662_3ST_ch2_init[] = { - { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, - { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, - { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, - { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, - { } /* end */ -}; - -/* - * 6ch mode - */ -static const struct hda_verb alc662_3ST_ch6_init[] = { - { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - { 0x18, AC_VERB_SET_CONNECT_SEL, 0x02 }, - { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 }, - { } /* end */ -}; - -static const struct hda_channel_mode alc662_3ST_6ch_modes[2] = { - { 2, alc662_3ST_ch2_init }, - { 6, alc662_3ST_ch6_init }, -}; - -/* - * 2ch mode - */ -static const struct hda_verb alc662_sixstack_ch6_init[] = { - { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 }, - { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 }, - { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { } /* end */ -}; - -/* - * 6ch mode - */ -static const struct hda_verb alc662_sixstack_ch8_init[] = { - { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { } /* end */ -}; - -static const struct hda_channel_mode alc662_5stack_modes[2] = { - { 2, alc662_sixstack_ch6_init }, - { 6, alc662_sixstack_ch8_init }, -}; - -/* Pin assignment: Front=0x14, Rear=0x15, CLFE=0x16, Side=0x17 - * Mic=0x18, Front Mic=0x19, Line-In=0x1a, HP=0x1b - */ - -static const struct snd_kcontrol_new alc662_base_mixer[] = { - /* output mixer control */ - HDA_CODEC_VOLUME("Front Playback Volume", 0x2, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Front Playback Switch", 0x0c, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Surround Playback Volume", 0x3, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Surround Playback Switch", 0x0d, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x04, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x04, 2, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x0e, 1, 0x0, HDA_INPUT), - HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), - - /*Input mixer control */ - HDA_CODEC_VOLUME("CD Playback Volume", 0xb, 0x4, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0xb, 0x4, HDA_INPUT), - HDA_CODEC_VOLUME("Line Playback Volume", 0xb, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("Line Playback Switch", 0xb, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0xb, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0xb, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Playback Volume", 0xb, 0x01, HDA_INPUT), - HDA_CODEC_MUTE("Front Mic Playback Switch", 0xb, 0x01, HDA_INPUT), - { } /* end */ -}; - -static const struct snd_kcontrol_new alc662_3ST_2ch_mixer[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Front Playback Switch", 0x0c, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - { } /* end */ -}; - -static const struct snd_kcontrol_new alc662_3ST_6ch_mixer[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Front Playback Switch", 0x0c, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Surround Playback Volume", 0x03, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Surround Playback Switch", 0x0d, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x04, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x04, 2, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x0e, 1, 0x0, HDA_INPUT), - HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - { } /* end */ -}; - -static const struct snd_kcontrol_new alc662_lenovo_101e_mixer[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Front Playback Switch", 0x02, 2, HDA_INPUT), - HDA_CODEC_VOLUME("Speaker Playback Volume", 0x03, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Speaker Playback Switch", 0x03, 2, HDA_INPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - { } /* end */ -}; - -static const struct snd_kcontrol_new alc662_eeepc_p701_mixer[] = { - HDA_CODEC_VOLUME("Master Playback Volume", 0x02, 0x0, HDA_OUTPUT), - ALC262_HIPPO_MASTER_SWITCH, - - HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - - HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT), - HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - { } /* end */ -}; - -static const struct snd_kcontrol_new alc662_eeepc_ep20_mixer[] = { - ALC262_HIPPO_MASTER_SWITCH, - HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Surround Playback Volume", 0x03, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x04, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x04, 2, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("MuteCtrl Playback Switch", 0x0c, 2, HDA_INPUT), - HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - { } /* end */ -}; - -static const struct hda_bind_ctls alc663_asus_bind_master_vol = { - .ops = &snd_hda_bind_vol, - .values = { - HDA_COMPOSE_AMP_VAL(0x02, 3, 0, HDA_OUTPUT), - HDA_COMPOSE_AMP_VAL(0x03, 3, 0, HDA_OUTPUT), - 0 - }, -}; - -static const struct hda_bind_ctls alc663_asus_one_bind_switch = { - .ops = &snd_hda_bind_sw, - .values = { - HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT), - HDA_COMPOSE_AMP_VAL(0x21, 3, 0, HDA_OUTPUT), - 0 - }, -}; - -static const struct snd_kcontrol_new alc663_m51va_mixer[] = { - HDA_BIND_VOL("Master Playback Volume", &alc663_asus_bind_master_vol), - HDA_BIND_SW("Master Playback Switch", &alc663_asus_one_bind_switch), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - { } /* end */ -}; - -static const struct hda_bind_ctls alc663_asus_tree_bind_switch = { - .ops = &snd_hda_bind_sw, - .values = { - HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT), - HDA_COMPOSE_AMP_VAL(0x15, 3, 0, HDA_OUTPUT), - HDA_COMPOSE_AMP_VAL(0x21, 3, 0, HDA_OUTPUT), - 0 - }, -}; - -static const struct snd_kcontrol_new alc663_two_hp_m1_mixer[] = { - HDA_BIND_VOL("Master Playback Volume", &alc663_asus_bind_master_vol), - HDA_BIND_SW("Master Playback Switch", &alc663_asus_tree_bind_switch), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("F-Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("F-Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - - { } /* end */ -}; - -static const struct hda_bind_ctls alc663_asus_four_bind_switch = { - .ops = &snd_hda_bind_sw, - .values = { - HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT), - HDA_COMPOSE_AMP_VAL(0x15, 3, 0, HDA_OUTPUT), - HDA_COMPOSE_AMP_VAL(0x1b, 3, 0, HDA_OUTPUT), - 0 - }, -}; - -static const struct snd_kcontrol_new alc663_two_hp_m2_mixer[] = { - HDA_BIND_VOL("Master Playback Volume", &alc663_asus_bind_master_vol), - HDA_BIND_SW("Master Playback Switch", &alc663_asus_four_bind_switch), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("F-Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("F-Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - { } /* end */ -}; - -static const struct snd_kcontrol_new alc662_1bjd_mixer[] = { - HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("F-Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("F-Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - { } /* end */ -}; - -static const struct hda_bind_ctls alc663_asus_two_bind_master_vol = { - .ops = &snd_hda_bind_vol, - .values = { - HDA_COMPOSE_AMP_VAL(0x02, 3, 0, HDA_OUTPUT), - HDA_COMPOSE_AMP_VAL(0x04, 3, 0, HDA_OUTPUT), - 0 - }, -}; - -static const struct hda_bind_ctls alc663_asus_two_bind_switch = { - .ops = &snd_hda_bind_sw, - .values = { - HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT), - HDA_COMPOSE_AMP_VAL(0x16, 3, 0, HDA_OUTPUT), - 0 - }, -}; - -static const struct snd_kcontrol_new alc663_asus_21jd_clfe_mixer[] = { - HDA_BIND_VOL("Master Playback Volume", - &alc663_asus_two_bind_master_vol), - HDA_BIND_SW("Master Playback Switch", &alc663_asus_two_bind_switch), - HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x21, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - { } /* end */ -}; - -static const struct snd_kcontrol_new alc663_asus_15jd_clfe_mixer[] = { - HDA_BIND_VOL("Master Playback Volume", &alc663_asus_bind_master_vol), - HDA_BIND_SW("Master Playback Switch", &alc663_asus_two_bind_switch), - HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - { } /* end */ -}; - -static const struct snd_kcontrol_new alc663_g71v_mixer[] = { - HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Front Playback Volume", 0x03, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Front Playback Switch", 0x15, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x21, 0x0, HDA_OUTPUT), - - HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - { } /* end */ -}; - -static const struct snd_kcontrol_new alc663_g50v_mixer[] = { - HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x21, 0x0, HDA_OUTPUT), - - HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), - { } /* end */ -}; - -static const struct hda_bind_ctls alc663_asus_mode7_8_all_bind_switch = { - .ops = &snd_hda_bind_sw, - .values = { - HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT), - HDA_COMPOSE_AMP_VAL(0x15, 3, 0, HDA_OUTPUT), - HDA_COMPOSE_AMP_VAL(0x17, 3, 0, HDA_OUTPUT), - HDA_COMPOSE_AMP_VAL(0x1b, 3, 0, HDA_OUTPUT), - HDA_COMPOSE_AMP_VAL(0x21, 3, 0, HDA_OUTPUT), - 0 - }, -}; - -static const struct hda_bind_ctls alc663_asus_mode7_8_sp_bind_switch = { - .ops = &snd_hda_bind_sw, - .values = { - HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT), - HDA_COMPOSE_AMP_VAL(0x17, 3, 0, HDA_OUTPUT), - 0 - }, -}; - -static const struct snd_kcontrol_new alc663_mode7_mixer[] = { - HDA_BIND_SW("Master Playback Switch", &alc663_asus_mode7_8_all_bind_switch), - HDA_BIND_VOL("Speaker Playback Volume", &alc663_asus_bind_master_vol), - HDA_BIND_SW("Speaker Playback Switch", &alc663_asus_mode7_8_sp_bind_switch), - HDA_CODEC_MUTE("Headphone1 Playback Switch", 0x1b, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Headphone2 Playback Switch", 0x21, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("IntMic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("IntMic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - { } /* end */ -}; - -static const struct snd_kcontrol_new alc663_mode8_mixer[] = { - HDA_BIND_SW("Master Playback Switch", &alc663_asus_mode7_8_all_bind_switch), - HDA_BIND_VOL("Speaker Playback Volume", &alc663_asus_bind_master_vol), - HDA_BIND_SW("Speaker Playback Switch", &alc663_asus_mode7_8_sp_bind_switch), - HDA_CODEC_MUTE("Headphone1 Playback Switch", 0x15, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Headphone2 Playback Switch", 0x21, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - { } /* end */ -}; - - -static const struct snd_kcontrol_new alc662_chmode_mixer[] = { - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Channel Mode", - .info = alc_ch_mode_info, - .get = alc_ch_mode_get, - .put = alc_ch_mode_put, - }, - { } /* end */ -}; - -static const struct hda_verb alc662_init_verbs[] = { - /* ADC: mute amp left and right */ - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x09, AC_VERB_SET_CONNECT_SEL, 0x00}, - - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - - /* Front Pin: output 0 (0x0c) */ - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - - /* Rear Pin: output 1 (0x0d) */ - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - - /* CLFE Pin: output 2 (0x0e) */ - {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - - /* Mic (rear) pin: input vref at 80% */ - {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* Front Mic pin: input vref at 80% */ - {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* Line In pin: input */ - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* Line-2 In: Headphone output (output 0 - 0x0c) */ - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00}, - /* CD pin widget for input */ - {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - - /* FIXME: use matrix-type input source selection */ - /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */ - /* Input mixer */ - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - - { } -}; - -static const struct hda_verb alc662_eapd_init_verbs[] = { - /* always trun on EAPD */ - {0x14, AC_VERB_SET_EAPD_BTLENABLE, 2}, - {0x15, AC_VERB_SET_EAPD_BTLENABLE, 2}, - { } -}; - -static const struct hda_verb alc662_sue_init_verbs[] = { - {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC_FRONT_EVENT}, - {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC_HP_EVENT}, - {} -}; - -static const struct hda_verb alc662_eeepc_sue_init_verbs[] = { - {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT}, - {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT}, - {} -}; - -/* Set Unsolicited Event*/ -static const struct hda_verb alc662_eeepc_ep20_sue_init_verbs[] = { - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT}, - {} -}; - -static const struct hda_verb alc663_m51va_init_verbs[] = { - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x21, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Headphone */ - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(9)}, - {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT}, - {0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT}, - {} -}; - -static const struct hda_verb alc663_21jd_amic_init_verbs[] = { - {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x21, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Headphone */ - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT}, - {0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT}, - {} -}; - -static const struct hda_verb alc662_1bjd_amic_init_verbs[] = { - {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Headphone */ - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT}, - {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT}, - {} -}; - -static const struct hda_verb alc663_15jd_amic_init_verbs[] = { - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Headphone */ - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT}, - {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT}, - {} -}; - -static const struct hda_verb alc663_two_hp_amic_m1_init_verbs[] = { - {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x21, AC_VERB_SET_CONNECT_SEL, 0x0}, /* Headphone */ - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x15, AC_VERB_SET_CONNECT_SEL, 0x0}, /* Headphone */ - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT}, - {0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT}, - {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT}, - {} -}; - -static const struct hda_verb alc663_two_hp_amic_m2_init_verbs[] = { - {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x1b, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Headphone */ - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Headphone */ - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT}, - {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT}, - {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT}, - {} -}; - -static const struct hda_verb alc663_g71v_init_verbs[] = { - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - /* {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, */ - /* {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, */ /* Headphone */ - - {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x21, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Headphone */ - - {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC_FRONT_EVENT}, - {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC_MIC_EVENT}, - {0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC_HP_EVENT}, - {} -}; - -static const struct hda_verb alc663_g50v_init_verbs[] = { - {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x21, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Headphone */ - - {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT}, - {0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT}, - {} -}; - -static const struct hda_verb alc662_ecs_init_verbs[] = { - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, 0x701f}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT}, - {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT}, - {} -}; - -static const struct hda_verb alc272_dell_zm1_init_verbs[] = { - {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x21, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Headphone */ - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(9)}, - {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT}, - {0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT}, - {} -}; - -static const struct hda_verb alc272_dell_init_verbs[] = { - {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x21, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Headphone */ - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(9)}, - {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT}, - {0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT}, - {} -}; - -static const struct hda_verb alc663_mode7_init_verbs[] = { - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x1b, AC_VERB_SET_CONNECT_SEL, 0x01}, - {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x21, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Headphone */ - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(9)}, - {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT}, - {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT}, - {0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT}, - {} -}; - -static const struct hda_verb alc663_mode8_init_verbs[] = { - {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, - {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x21, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Headphone */ - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(9)}, - {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT}, - {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT}, - {0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT}, - {} -}; - -static const struct snd_kcontrol_new alc662_auto_capture_mixer[] = { - HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT), - { } /* end */ -}; - -static const struct snd_kcontrol_new alc272_auto_capture_mixer[] = { - HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), - { } /* end */ -}; - -static void alc662_lenovo_101e_setup(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - - spec->autocfg.hp_pins[0] = 0x1b; - spec->autocfg.line_out_pins[0] = 0x14; - spec->autocfg.speaker_pins[0] = 0x15; - spec->automute = 1; - spec->detect_line = 1; - spec->automute_lines = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; -} - -static void alc662_eeepc_setup(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - - alc262_hippo1_setup(codec); - spec->ext_mic_pin = 0x18; - spec->int_mic_pin = 0x19; - spec->auto_mic = 1; -} - -static void alc662_eeepc_ep20_setup(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - - spec->autocfg.hp_pins[0] = 0x14; - spec->autocfg.speaker_pins[0] = 0x1b; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; -} - -static void alc663_m51va_setup(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - spec->autocfg.hp_pins[0] = 0x21; - spec->autocfg.speaker_pins[0] = 0x14; - spec->automute_mixer_nid[0] = 0x0c; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_MIXER; - spec->ext_mic_pin = 0x18; - spec->int_mic_pin = 0x12; - spec->auto_mic = 1; -} - -/* ***************** Mode1 ******************************/ -static void alc663_mode1_setup(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - spec->autocfg.hp_pins[0] = 0x21; - spec->autocfg.speaker_pins[0] = 0x14; - spec->automute_mixer_nid[0] = 0x0c; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_MIXER; - spec->ext_mic_pin = 0x18; - spec->int_mic_pin = 0x19; - spec->auto_mic = 1; -} - -/* ***************** Mode2 ******************************/ -static void alc662_mode2_setup(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - spec->autocfg.hp_pins[0] = 0x1b; - spec->autocfg.speaker_pins[0] = 0x14; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_PIN; - spec->ext_mic_pin = 0x18; - spec->int_mic_pin = 0x19; - spec->auto_mic = 1; -} - -/* ***************** Mode3 ******************************/ -static void alc663_mode3_setup(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - spec->autocfg.hp_pins[0] = 0x21; - spec->autocfg.hp_pins[0] = 0x15; - spec->autocfg.speaker_pins[0] = 0x14; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_PIN; - spec->ext_mic_pin = 0x18; - spec->int_mic_pin = 0x19; - spec->auto_mic = 1; -} - -/* ***************** Mode4 ******************************/ -static void alc663_mode4_setup(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - spec->autocfg.hp_pins[0] = 0x21; - spec->autocfg.speaker_pins[0] = 0x14; - spec->autocfg.speaker_pins[1] = 0x16; - spec->automute_mixer_nid[0] = 0x0c; - spec->automute_mixer_nid[1] = 0x0e; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_MIXER; - spec->ext_mic_pin = 0x18; - spec->int_mic_pin = 0x19; - spec->auto_mic = 1; -} - -/* ***************** Mode5 ******************************/ -static void alc663_mode5_setup(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - spec->autocfg.hp_pins[0] = 0x15; - spec->autocfg.speaker_pins[0] = 0x14; - spec->autocfg.speaker_pins[1] = 0x16; - spec->automute_mixer_nid[0] = 0x0c; - spec->automute_mixer_nid[1] = 0x0e; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_MIXER; - spec->ext_mic_pin = 0x18; - spec->int_mic_pin = 0x19; - spec->auto_mic = 1; -} - -/* ***************** Mode6 ******************************/ -static void alc663_mode6_setup(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - spec->autocfg.hp_pins[0] = 0x1b; - spec->autocfg.hp_pins[0] = 0x15; - spec->autocfg.speaker_pins[0] = 0x14; - spec->automute_mixer_nid[0] = 0x0c; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_MIXER; - spec->ext_mic_pin = 0x18; - spec->int_mic_pin = 0x19; - spec->auto_mic = 1; -} - -/* ***************** Mode7 ******************************/ -static void alc663_mode7_setup(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - spec->autocfg.hp_pins[0] = 0x1b; - spec->autocfg.hp_pins[0] = 0x21; - spec->autocfg.speaker_pins[0] = 0x14; - spec->autocfg.speaker_pins[0] = 0x17; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_PIN; - spec->ext_mic_pin = 0x18; - spec->int_mic_pin = 0x19; - spec->auto_mic = 1; -} - -/* ***************** Mode8 ******************************/ -static void alc663_mode8_setup(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - spec->autocfg.hp_pins[0] = 0x21; - spec->autocfg.hp_pins[1] = 0x15; - spec->autocfg.speaker_pins[0] = 0x14; - spec->autocfg.speaker_pins[0] = 0x17; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_PIN; - spec->ext_mic_pin = 0x18; - spec->int_mic_pin = 0x12; - spec->auto_mic = 1; -} - -static void alc663_g71v_setup(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - spec->autocfg.hp_pins[0] = 0x21; - spec->autocfg.line_out_pins[0] = 0x15; - spec->autocfg.speaker_pins[0] = 0x14; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; - spec->detect_line = 1; - spec->automute_lines = 1; - spec->ext_mic_pin = 0x18; - spec->int_mic_pin = 0x12; - spec->auto_mic = 1; -} - -#define alc663_g50v_setup alc663_m51va_setup - -static const struct snd_kcontrol_new alc662_ecs_mixer[] = { - HDA_CODEC_VOLUME("Master Playback Volume", 0x02, 0x0, HDA_OUTPUT), - ALC262_HIPPO_MASTER_SWITCH, - - HDA_CODEC_VOLUME("Mic/LineIn Boost Volume", 0x18, 0, HDA_INPUT), - HDA_CODEC_VOLUME("Mic/LineIn Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Mic/LineIn Playback Switch", 0x0b, 0x0, HDA_INPUT), - - HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT), - HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - { } /* end */ -}; - -static const struct snd_kcontrol_new alc272_nc10_mixer[] = { - /* Master Playback automatically created from Speaker and Headphone */ - HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x21, 0x0, HDA_OUTPUT), - - HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), - - HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT), - { } /* end */ -}; - - -/* - * configuration and preset - */ -static const char * const alc662_models[ALC662_MODEL_LAST] = { - [ALC662_3ST_2ch_DIG] = "3stack-dig", - [ALC662_3ST_6ch_DIG] = "3stack-6ch-dig", - [ALC662_3ST_6ch] = "3stack-6ch", - [ALC662_5ST_DIG] = "5stack-dig", - [ALC662_LENOVO_101E] = "lenovo-101e", - [ALC662_ASUS_EEEPC_P701] = "eeepc-p701", - [ALC662_ASUS_EEEPC_EP20] = "eeepc-ep20", - [ALC662_ECS] = "ecs", - [ALC663_ASUS_M51VA] = "m51va", - [ALC663_ASUS_G71V] = "g71v", - [ALC663_ASUS_H13] = "h13", - [ALC663_ASUS_G50V] = "g50v", - [ALC663_ASUS_MODE1] = "asus-mode1", - [ALC662_ASUS_MODE2] = "asus-mode2", - [ALC663_ASUS_MODE3] = "asus-mode3", - [ALC663_ASUS_MODE4] = "asus-mode4", - [ALC663_ASUS_MODE5] = "asus-mode5", - [ALC663_ASUS_MODE6] = "asus-mode6", - [ALC663_ASUS_MODE7] = "asus-mode7", - [ALC663_ASUS_MODE8] = "asus-mode8", - [ALC272_DELL] = "dell", - [ALC272_DELL_ZM1] = "dell-zm1", - [ALC272_SAMSUNG_NC10] = "samsung-nc10", - [ALC662_AUTO] = "auto", -}; - -static const struct snd_pci_quirk alc662_cfg_tbl[] = { - SND_PCI_QUIRK(0x1019, 0x9087, "ECS", ALC662_ECS), - SND_PCI_QUIRK(0x1028, 0x02d6, "DELL", ALC272_DELL), - SND_PCI_QUIRK(0x1028, 0x02f4, "DELL ZM1", ALC272_DELL_ZM1), - SND_PCI_QUIRK(0x1043, 0x1000, "ASUS N50Vm", ALC663_ASUS_MODE1), - SND_PCI_QUIRK(0x1043, 0x1092, "ASUS NB", ALC663_ASUS_MODE3), - SND_PCI_QUIRK(0x1043, 0x1173, "ASUS K73Jn", ALC663_ASUS_MODE1), - SND_PCI_QUIRK(0x1043, 0x11c3, "ASUS M70V", ALC663_ASUS_MODE3), - SND_PCI_QUIRK(0x1043, 0x11d3, "ASUS NB", ALC663_ASUS_MODE1), - SND_PCI_QUIRK(0x1043, 0x11f3, "ASUS NB", ALC662_ASUS_MODE2), - SND_PCI_QUIRK(0x1043, 0x1203, "ASUS NB", ALC663_ASUS_MODE1), - SND_PCI_QUIRK(0x1043, 0x1303, "ASUS G60J", ALC663_ASUS_MODE1), - SND_PCI_QUIRK(0x1043, 0x1333, "ASUS G60Jx", ALC663_ASUS_MODE1), - SND_PCI_QUIRK(0x1043, 0x1339, "ASUS NB", ALC662_ASUS_MODE2), - SND_PCI_QUIRK(0x1043, 0x13e3, "ASUS N71JA", ALC663_ASUS_MODE7), - SND_PCI_QUIRK(0x1043, 0x1463, "ASUS N71", ALC663_ASUS_MODE7), - SND_PCI_QUIRK(0x1043, 0x14d3, "ASUS G72", ALC663_ASUS_MODE8), - SND_PCI_QUIRK(0x1043, 0x1563, "ASUS N90", ALC663_ASUS_MODE3), - SND_PCI_QUIRK(0x1043, 0x15d3, "ASUS N50SF F50SF", ALC663_ASUS_MODE1), - SND_PCI_QUIRK(0x1043, 0x16c3, "ASUS NB", ALC662_ASUS_MODE2), - SND_PCI_QUIRK(0x1043, 0x16f3, "ASUS K40C K50C", ALC662_ASUS_MODE2), - SND_PCI_QUIRK(0x1043, 0x1733, "ASUS N81De", ALC663_ASUS_MODE1), - SND_PCI_QUIRK(0x1043, 0x1753, "ASUS NB", ALC662_ASUS_MODE2), - SND_PCI_QUIRK(0x1043, 0x1763, "ASUS NB", ALC663_ASUS_MODE6), - SND_PCI_QUIRK(0x1043, 0x1765, "ASUS NB", ALC663_ASUS_MODE6), - SND_PCI_QUIRK(0x1043, 0x1783, "ASUS NB", ALC662_ASUS_MODE2), - SND_PCI_QUIRK(0x1043, 0x1793, "ASUS F50GX", ALC663_ASUS_MODE1), - SND_PCI_QUIRK(0x1043, 0x17b3, "ASUS F70SL", ALC663_ASUS_MODE3), - SND_PCI_QUIRK(0x1043, 0x17c3, "ASUS UX20", ALC663_ASUS_M51VA), - SND_PCI_QUIRK(0x1043, 0x17f3, "ASUS X58LE", ALC662_ASUS_MODE2), - SND_PCI_QUIRK(0x1043, 0x1813, "ASUS NB", ALC662_ASUS_MODE2), - SND_PCI_QUIRK(0x1043, 0x1823, "ASUS NB", ALC663_ASUS_MODE5), - SND_PCI_QUIRK(0x1043, 0x1833, "ASUS NB", ALC663_ASUS_MODE6), - SND_PCI_QUIRK(0x1043, 0x1843, "ASUS NB", ALC662_ASUS_MODE2), - SND_PCI_QUIRK(0x1043, 0x1853, "ASUS F50Z", ALC663_ASUS_MODE1), - SND_PCI_QUIRK(0x1043, 0x1864, "ASUS NB", ALC662_ASUS_MODE2), - SND_PCI_QUIRK(0x1043, 0x1876, "ASUS NB", ALC662_ASUS_MODE2), - SND_PCI_QUIRK(0x1043, 0x1878, "ASUS M51VA", ALC663_ASUS_M51VA), - /*SND_PCI_QUIRK(0x1043, 0x1878, "ASUS M50Vr", ALC663_ASUS_MODE1),*/ - SND_PCI_QUIRK(0x1043, 0x1893, "ASUS M50Vm", ALC663_ASUS_MODE3), - SND_PCI_QUIRK(0x1043, 0x1894, "ASUS X55", ALC663_ASUS_MODE3), - SND_PCI_QUIRK(0x1043, 0x18b3, "ASUS N80Vc", ALC663_ASUS_MODE1), - SND_PCI_QUIRK(0x1043, 0x18c3, "ASUS VX5", ALC663_ASUS_MODE1), - SND_PCI_QUIRK(0x1043, 0x18d3, "ASUS N81Te", ALC663_ASUS_MODE1), - SND_PCI_QUIRK(0x1043, 0x18f3, "ASUS N505Tp", ALC663_ASUS_MODE1), - SND_PCI_QUIRK(0x1043, 0x1903, "ASUS F5GL", ALC663_ASUS_MODE1), - SND_PCI_QUIRK(0x1043, 0x1913, "ASUS NB", ALC662_ASUS_MODE2), - SND_PCI_QUIRK(0x1043, 0x1933, "ASUS F80Q", ALC662_ASUS_MODE2), - SND_PCI_QUIRK(0x1043, 0x1943, "ASUS Vx3V", ALC663_ASUS_MODE1), - SND_PCI_QUIRK(0x1043, 0x1953, "ASUS NB", ALC663_ASUS_MODE1), - SND_PCI_QUIRK(0x1043, 0x1963, "ASUS X71C", ALC663_ASUS_MODE3), - SND_PCI_QUIRK(0x1043, 0x1983, "ASUS N5051A", ALC663_ASUS_MODE1), - SND_PCI_QUIRK(0x1043, 0x1993, "ASUS N20", ALC663_ASUS_MODE1), - SND_PCI_QUIRK(0x1043, 0x19a3, "ASUS G50V", ALC663_ASUS_G50V), - /*SND_PCI_QUIRK(0x1043, 0x19a3, "ASUS NB", ALC663_ASUS_MODE1),*/ - SND_PCI_QUIRK(0x1043, 0x19b3, "ASUS F7Z", ALC663_ASUS_MODE1), - SND_PCI_QUIRK(0x1043, 0x19c3, "ASUS F5Z/F6x", ALC662_ASUS_MODE2), - SND_PCI_QUIRK(0x1043, 0x19d3, "ASUS NB", ALC663_ASUS_M51VA), - SND_PCI_QUIRK(0x1043, 0x19e3, "ASUS NB", ALC663_ASUS_MODE1), - SND_PCI_QUIRK(0x1043, 0x19f3, "ASUS NB", ALC663_ASUS_MODE4), - SND_PCI_QUIRK(0x1043, 0x8290, "ASUS P5GC-MX", ALC662_3ST_6ch_DIG), - SND_PCI_QUIRK(0x1043, 0x82a1, "ASUS Eeepc", ALC662_ASUS_EEEPC_P701), - SND_PCI_QUIRK(0x1043, 0x82d1, "ASUS Eeepc EP20", ALC662_ASUS_EEEPC_EP20), - SND_PCI_QUIRK(0x105b, 0x0cd6, "Foxconn", ALC662_ECS), - SND_PCI_QUIRK(0x105b, 0x0d47, "Foxconn 45CMX/45GMX/45CMX-K", - ALC662_3ST_6ch_DIG), - SND_PCI_QUIRK(0x1179, 0xff6e, "Toshiba NB20x", ALC662_AUTO), - SND_PCI_QUIRK(0x144d, 0xca00, "Samsung NC10", ALC272_SAMSUNG_NC10), - SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte 945GCM-S2L", - ALC662_3ST_6ch_DIG), - SND_PCI_QUIRK(0x152d, 0x2304, "Quanta WH1", ALC663_ASUS_H13), - SND_PCI_QUIRK(0x1565, 0x820f, "Biostar TA780G M2+", ALC662_3ST_6ch_DIG), - SND_PCI_QUIRK(0x1631, 0xc10c, "PB RS65", ALC663_ASUS_M51VA), - SND_PCI_QUIRK(0x17aa, 0x101e, "Lenovo", ALC662_LENOVO_101E), - SND_PCI_QUIRK(0x1849, 0x3662, "ASROCK K10N78FullHD-hSLI R3.0", - ALC662_3ST_6ch_DIG), - SND_PCI_QUIRK_MASK(0x1854, 0xf000, 0x2000, "ASUS H13-200x", - ALC663_ASUS_H13), - SND_PCI_QUIRK(0x1991, 0x5628, "Ordissimo EVE", ALC662_LENOVO_101E), - {} -}; - -static const struct alc_config_preset alc662_presets[] = { - [ALC662_3ST_2ch_DIG] = { - .mixers = { alc662_3ST_2ch_mixer }, - .init_verbs = { alc662_init_verbs, alc662_eapd_init_verbs }, - .num_dacs = ARRAY_SIZE(alc662_dac_nids), - .dac_nids = alc662_dac_nids, - .dig_out_nid = ALC662_DIGOUT_NID, - .dig_in_nid = ALC662_DIGIN_NID, - .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes), - .channel_mode = alc662_3ST_2ch_modes, - .input_mux = &alc662_capture_source, - }, - [ALC662_3ST_6ch_DIG] = { - .mixers = { alc662_3ST_6ch_mixer, alc662_chmode_mixer }, - .init_verbs = { alc662_init_verbs, alc662_eapd_init_verbs }, - .num_dacs = ARRAY_SIZE(alc662_dac_nids), - .dac_nids = alc662_dac_nids, - .dig_out_nid = ALC662_DIGOUT_NID, - .dig_in_nid = ALC662_DIGIN_NID, - .num_channel_mode = ARRAY_SIZE(alc662_3ST_6ch_modes), - .channel_mode = alc662_3ST_6ch_modes, - .need_dac_fix = 1, - .input_mux = &alc662_capture_source, - }, - [ALC662_3ST_6ch] = { - .mixers = { alc662_3ST_6ch_mixer, alc662_chmode_mixer }, - .init_verbs = { alc662_init_verbs, alc662_eapd_init_verbs }, - .num_dacs = ARRAY_SIZE(alc662_dac_nids), - .dac_nids = alc662_dac_nids, - .num_channel_mode = ARRAY_SIZE(alc662_3ST_6ch_modes), - .channel_mode = alc662_3ST_6ch_modes, - .need_dac_fix = 1, - .input_mux = &alc662_capture_source, - }, - [ALC662_5ST_DIG] = { - .mixers = { alc662_base_mixer, alc662_chmode_mixer }, - .init_verbs = { alc662_init_verbs, alc662_eapd_init_verbs }, - .num_dacs = ARRAY_SIZE(alc662_dac_nids), - .dac_nids = alc662_dac_nids, - .dig_out_nid = ALC662_DIGOUT_NID, - .dig_in_nid = ALC662_DIGIN_NID, - .num_channel_mode = ARRAY_SIZE(alc662_5stack_modes), - .channel_mode = alc662_5stack_modes, - .input_mux = &alc662_capture_source, - }, - [ALC662_LENOVO_101E] = { - .mixers = { alc662_lenovo_101e_mixer }, - .init_verbs = { alc662_init_verbs, - alc662_eapd_init_verbs, - alc662_sue_init_verbs }, - .num_dacs = ARRAY_SIZE(alc662_dac_nids), - .dac_nids = alc662_dac_nids, - .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes), - .channel_mode = alc662_3ST_2ch_modes, - .input_mux = &alc662_lenovo_101e_capture_source, - .unsol_event = alc_sku_unsol_event, - .setup = alc662_lenovo_101e_setup, - .init_hook = alc_inithook, - }, - [ALC662_ASUS_EEEPC_P701] = { - .mixers = { alc662_eeepc_p701_mixer }, - .init_verbs = { alc662_init_verbs, - alc662_eapd_init_verbs, - alc662_eeepc_sue_init_verbs }, - .num_dacs = ARRAY_SIZE(alc662_dac_nids), - .dac_nids = alc662_dac_nids, - .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes), - .channel_mode = alc662_3ST_2ch_modes, - .unsol_event = alc_sku_unsol_event, - .setup = alc662_eeepc_setup, - .init_hook = alc_inithook, - }, - [ALC662_ASUS_EEEPC_EP20] = { - .mixers = { alc662_eeepc_ep20_mixer, - alc662_chmode_mixer }, - .init_verbs = { alc662_init_verbs, - alc662_eapd_init_verbs, - alc662_eeepc_ep20_sue_init_verbs }, - .num_dacs = ARRAY_SIZE(alc662_dac_nids), - .dac_nids = alc662_dac_nids, - .num_channel_mode = ARRAY_SIZE(alc662_3ST_6ch_modes), - .channel_mode = alc662_3ST_6ch_modes, - .input_mux = &alc662_lenovo_101e_capture_source, - .unsol_event = alc_sku_unsol_event, - .setup = alc662_eeepc_ep20_setup, - .init_hook = alc_inithook, - }, - [ALC662_ECS] = { - .mixers = { alc662_ecs_mixer }, - .init_verbs = { alc662_init_verbs, - alc662_eapd_init_verbs, - alc662_ecs_init_verbs }, - .num_dacs = ARRAY_SIZE(alc662_dac_nids), - .dac_nids = alc662_dac_nids, - .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes), - .channel_mode = alc662_3ST_2ch_modes, - .unsol_event = alc_sku_unsol_event, - .setup = alc662_eeepc_setup, - .init_hook = alc_inithook, - }, - [ALC663_ASUS_M51VA] = { - .mixers = { alc663_m51va_mixer }, - .init_verbs = { alc662_init_verbs, - alc662_eapd_init_verbs, - alc663_m51va_init_verbs }, - .num_dacs = ARRAY_SIZE(alc662_dac_nids), - .dac_nids = alc662_dac_nids, - .dig_out_nid = ALC662_DIGOUT_NID, - .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes), - .channel_mode = alc662_3ST_2ch_modes, - .unsol_event = alc_sku_unsol_event, - .setup = alc663_m51va_setup, - .init_hook = alc_inithook, - }, - [ALC663_ASUS_G71V] = { - .mixers = { alc663_g71v_mixer }, - .init_verbs = { alc662_init_verbs, - alc662_eapd_init_verbs, - alc663_g71v_init_verbs }, - .num_dacs = ARRAY_SIZE(alc662_dac_nids), - .dac_nids = alc662_dac_nids, - .dig_out_nid = ALC662_DIGOUT_NID, - .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes), - .channel_mode = alc662_3ST_2ch_modes, - .unsol_event = alc_sku_unsol_event, - .setup = alc663_g71v_setup, - .init_hook = alc_inithook, - }, - [ALC663_ASUS_H13] = { - .mixers = { alc663_m51va_mixer }, - .init_verbs = { alc662_init_verbs, - alc662_eapd_init_verbs, - alc663_m51va_init_verbs }, - .num_dacs = ARRAY_SIZE(alc662_dac_nids), - .dac_nids = alc662_dac_nids, - .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes), - .channel_mode = alc662_3ST_2ch_modes, - .setup = alc663_m51va_setup, - .unsol_event = alc_sku_unsol_event, - .init_hook = alc_inithook, - }, - [ALC663_ASUS_G50V] = { - .mixers = { alc663_g50v_mixer }, - .init_verbs = { alc662_init_verbs, - alc662_eapd_init_verbs, - alc663_g50v_init_verbs }, - .num_dacs = ARRAY_SIZE(alc662_dac_nids), - .dac_nids = alc662_dac_nids, - .dig_out_nid = ALC662_DIGOUT_NID, - .num_channel_mode = ARRAY_SIZE(alc662_3ST_6ch_modes), - .channel_mode = alc662_3ST_6ch_modes, - .input_mux = &alc663_capture_source, - .unsol_event = alc_sku_unsol_event, - .setup = alc663_g50v_setup, - .init_hook = alc_inithook, - }, - [ALC663_ASUS_MODE1] = { - .mixers = { alc663_m51va_mixer }, - .cap_mixer = alc662_auto_capture_mixer, - .init_verbs = { alc662_init_verbs, - alc662_eapd_init_verbs, - alc663_21jd_amic_init_verbs }, - .num_dacs = ARRAY_SIZE(alc662_dac_nids), - .hp_nid = 0x03, - .dac_nids = alc662_dac_nids, - .dig_out_nid = ALC662_DIGOUT_NID, - .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes), - .channel_mode = alc662_3ST_2ch_modes, - .unsol_event = alc_sku_unsol_event, - .setup = alc663_mode1_setup, - .init_hook = alc_inithook, - }, - [ALC662_ASUS_MODE2] = { - .mixers = { alc662_1bjd_mixer }, - .cap_mixer = alc662_auto_capture_mixer, - .init_verbs = { alc662_init_verbs, - alc662_eapd_init_verbs, - alc662_1bjd_amic_init_verbs }, - .num_dacs = ARRAY_SIZE(alc662_dac_nids), - .dac_nids = alc662_dac_nids, - .dig_out_nid = ALC662_DIGOUT_NID, - .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes), - .channel_mode = alc662_3ST_2ch_modes, - .unsol_event = alc_sku_unsol_event, - .setup = alc662_mode2_setup, - .init_hook = alc_inithook, - }, - [ALC663_ASUS_MODE3] = { - .mixers = { alc663_two_hp_m1_mixer }, - .cap_mixer = alc662_auto_capture_mixer, - .init_verbs = { alc662_init_verbs, - alc662_eapd_init_verbs, - alc663_two_hp_amic_m1_init_verbs }, - .num_dacs = ARRAY_SIZE(alc662_dac_nids), - .hp_nid = 0x03, - .dac_nids = alc662_dac_nids, - .dig_out_nid = ALC662_DIGOUT_NID, - .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes), - .channel_mode = alc662_3ST_2ch_modes, - .unsol_event = alc_sku_unsol_event, - .setup = alc663_mode3_setup, - .init_hook = alc_inithook, - }, - [ALC663_ASUS_MODE4] = { - .mixers = { alc663_asus_21jd_clfe_mixer }, - .cap_mixer = alc662_auto_capture_mixer, - .init_verbs = { alc662_init_verbs, - alc662_eapd_init_verbs, - alc663_21jd_amic_init_verbs}, - .num_dacs = ARRAY_SIZE(alc662_dac_nids), - .hp_nid = 0x03, - .dac_nids = alc662_dac_nids, - .dig_out_nid = ALC662_DIGOUT_NID, - .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes), - .channel_mode = alc662_3ST_2ch_modes, - .unsol_event = alc_sku_unsol_event, - .setup = alc663_mode4_setup, - .init_hook = alc_inithook, - }, - [ALC663_ASUS_MODE5] = { - .mixers = { alc663_asus_15jd_clfe_mixer }, - .cap_mixer = alc662_auto_capture_mixer, - .init_verbs = { alc662_init_verbs, - alc662_eapd_init_verbs, - alc663_15jd_amic_init_verbs }, - .num_dacs = ARRAY_SIZE(alc662_dac_nids), - .hp_nid = 0x03, - .dac_nids = alc662_dac_nids, - .dig_out_nid = ALC662_DIGOUT_NID, - .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes), - .channel_mode = alc662_3ST_2ch_modes, - .unsol_event = alc_sku_unsol_event, - .setup = alc663_mode5_setup, - .init_hook = alc_inithook, - }, - [ALC663_ASUS_MODE6] = { - .mixers = { alc663_two_hp_m2_mixer }, - .cap_mixer = alc662_auto_capture_mixer, - .init_verbs = { alc662_init_verbs, - alc662_eapd_init_verbs, - alc663_two_hp_amic_m2_init_verbs }, - .num_dacs = ARRAY_SIZE(alc662_dac_nids), - .hp_nid = 0x03, - .dac_nids = alc662_dac_nids, - .dig_out_nid = ALC662_DIGOUT_NID, - .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes), - .channel_mode = alc662_3ST_2ch_modes, - .unsol_event = alc_sku_unsol_event, - .setup = alc663_mode6_setup, - .init_hook = alc_inithook, - }, - [ALC663_ASUS_MODE7] = { - .mixers = { alc663_mode7_mixer }, - .cap_mixer = alc662_auto_capture_mixer, - .init_verbs = { alc662_init_verbs, - alc662_eapd_init_verbs, - alc663_mode7_init_verbs }, - .num_dacs = ARRAY_SIZE(alc662_dac_nids), - .hp_nid = 0x03, - .dac_nids = alc662_dac_nids, - .dig_out_nid = ALC662_DIGOUT_NID, - .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes), - .channel_mode = alc662_3ST_2ch_modes, - .unsol_event = alc_sku_unsol_event, - .setup = alc663_mode7_setup, - .init_hook = alc_inithook, - }, - [ALC663_ASUS_MODE8] = { - .mixers = { alc663_mode8_mixer }, - .cap_mixer = alc662_auto_capture_mixer, - .init_verbs = { alc662_init_verbs, - alc662_eapd_init_verbs, - alc663_mode8_init_verbs }, - .num_dacs = ARRAY_SIZE(alc662_dac_nids), - .hp_nid = 0x03, - .dac_nids = alc662_dac_nids, - .dig_out_nid = ALC662_DIGOUT_NID, - .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes), - .channel_mode = alc662_3ST_2ch_modes, - .unsol_event = alc_sku_unsol_event, - .setup = alc663_mode8_setup, - .init_hook = alc_inithook, - }, - [ALC272_DELL] = { - .mixers = { alc663_m51va_mixer }, - .cap_mixer = alc272_auto_capture_mixer, - .init_verbs = { alc662_init_verbs, - alc662_eapd_init_verbs, - alc272_dell_init_verbs }, - .num_dacs = ARRAY_SIZE(alc272_dac_nids), - .dac_nids = alc272_dac_nids, - .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes), - .adc_nids = alc272_adc_nids, - .num_adc_nids = ARRAY_SIZE(alc272_adc_nids), - .capsrc_nids = alc272_capsrc_nids, - .channel_mode = alc662_3ST_2ch_modes, - .unsol_event = alc_sku_unsol_event, - .setup = alc663_m51va_setup, - .init_hook = alc_inithook, - }, - [ALC272_DELL_ZM1] = { - .mixers = { alc663_m51va_mixer }, - .cap_mixer = alc662_auto_capture_mixer, - .init_verbs = { alc662_init_verbs, - alc662_eapd_init_verbs, - alc272_dell_zm1_init_verbs }, - .num_dacs = ARRAY_SIZE(alc272_dac_nids), - .dac_nids = alc272_dac_nids, - .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes), - .adc_nids = alc662_adc_nids, - .num_adc_nids = 1, - .capsrc_nids = alc662_capsrc_nids, - .channel_mode = alc662_3ST_2ch_modes, - .unsol_event = alc_sku_unsol_event, - .setup = alc663_m51va_setup, - .init_hook = alc_inithook, - }, - [ALC272_SAMSUNG_NC10] = { - .mixers = { alc272_nc10_mixer }, - .init_verbs = { alc662_init_verbs, - alc662_eapd_init_verbs, - alc663_21jd_amic_init_verbs }, - .num_dacs = ARRAY_SIZE(alc272_dac_nids), - .dac_nids = alc272_dac_nids, - .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes), - .channel_mode = alc662_3ST_2ch_modes, - /*.input_mux = &alc272_nc10_capture_source,*/ - .unsol_event = alc_sku_unsol_event, - .setup = alc663_mode4_setup, - .init_hook = alc_inithook, - }, -}; - - diff --git a/sound/pci/hda/alc680_quirks.c b/sound/pci/hda/alc680_quirks.c deleted file mode 100644 index 0eeb227c7bc2..000000000000 --- a/sound/pci/hda/alc680_quirks.c +++ /dev/null @@ -1,222 +0,0 @@ -/* - * ALC680 quirk models - * included by patch_realtek.c - */ - -/* ALC680 models */ -enum { - ALC680_AUTO, - ALC680_BASE, - ALC680_MODEL_LAST, -}; - -#define ALC680_DIGIN_NID ALC880_DIGIN_NID -#define ALC680_DIGOUT_NID ALC880_DIGOUT_NID -#define alc680_modes alc260_modes - -static const hda_nid_t alc680_dac_nids[3] = { - /* Lout1, Lout2, hp */ - 0x02, 0x03, 0x04 -}; - -static const hda_nid_t alc680_adc_nids[3] = { - /* ADC0-2 */ - /* DMIC, MIC, Line-in*/ - 0x07, 0x08, 0x09 -}; - -/* - * Analog capture ADC cgange - */ -static hda_nid_t alc680_get_cur_adc(struct hda_codec *codec) -{ - static hda_nid_t pins[] = {0x18, 0x19}; - static hda_nid_t adcs[] = {0x08, 0x09}; - int i; - - for (i = 0; i < ARRAY_SIZE(pins); i++) { - if (!is_jack_detectable(codec, pins[i])) - continue; - if (snd_hda_jack_detect(codec, pins[i])) - return adcs[i]; - } - return 0x07; -} - -static void alc680_rec_autoswitch(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - hda_nid_t nid = alc680_get_cur_adc(codec); - if (spec->cur_adc && nid != spec->cur_adc) { - __snd_hda_codec_cleanup_stream(codec, spec->cur_adc, 1); - spec->cur_adc = nid; - snd_hda_codec_setup_stream(codec, nid, - spec->cur_adc_stream_tag, 0, - spec->cur_adc_format); - } -} - -static int alc680_capture_pcm_prepare(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - unsigned int stream_tag, - unsigned int format, - struct snd_pcm_substream *substream) -{ - struct alc_spec *spec = codec->spec; - hda_nid_t nid = alc680_get_cur_adc(codec); - - spec->cur_adc = nid; - spec->cur_adc_stream_tag = stream_tag; - spec->cur_adc_format = format; - snd_hda_codec_setup_stream(codec, nid, stream_tag, 0, format); - return 0; -} - -static int alc680_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - struct alc_spec *spec = codec->spec; - snd_hda_codec_cleanup_stream(codec, spec->cur_adc); - spec->cur_adc = 0; - return 0; -} - -static const struct hda_pcm_stream alc680_pcm_analog_auto_capture = { - .substreams = 1, /* can be overridden */ - .channels_min = 2, - .channels_max = 2, - /* NID is set in alc_build_pcms */ - .ops = { - .prepare = alc680_capture_pcm_prepare, - .cleanup = alc680_capture_pcm_cleanup - }, -}; - -static const struct snd_kcontrol_new alc680_base_mixer[] = { - /* output mixer control */ - HDA_CODEC_VOLUME("Front Playback Volume", 0x2, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Headphone Playback Volume", 0x4, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x16, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x12, 0, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), - HDA_CODEC_VOLUME("Line In Boost Volume", 0x19, 0, HDA_INPUT), - { } -}; - -static const struct hda_bind_ctls alc680_bind_cap_vol = { - .ops = &snd_hda_bind_vol, - .values = { - HDA_COMPOSE_AMP_VAL(0x07, 3, 0, HDA_INPUT), - HDA_COMPOSE_AMP_VAL(0x08, 3, 0, HDA_INPUT), - HDA_COMPOSE_AMP_VAL(0x09, 3, 0, HDA_INPUT), - 0 - }, -}; - -static const struct hda_bind_ctls alc680_bind_cap_switch = { - .ops = &snd_hda_bind_sw, - .values = { - HDA_COMPOSE_AMP_VAL(0x07, 3, 0, HDA_INPUT), - HDA_COMPOSE_AMP_VAL(0x08, 3, 0, HDA_INPUT), - HDA_COMPOSE_AMP_VAL(0x09, 3, 0, HDA_INPUT), - 0 - }, -}; - -static const struct snd_kcontrol_new alc680_master_capture_mixer[] = { - HDA_BIND_VOL("Capture Volume", &alc680_bind_cap_vol), - HDA_BIND_SW("Capture Switch", &alc680_bind_cap_switch), - { } /* end */ -}; - -/* - * generic initialization of ADC, input mixers and output mixers - */ -static const struct hda_verb alc680_init_verbs[] = { - {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - - {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - - {0x16, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN}, - {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_MIC_EVENT | AC_USRSP_EN}, - {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_MIC_EVENT | AC_USRSP_EN}, - - { } -}; - -/* toggle speaker-output according to the hp-jack state */ -static void alc680_base_setup(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - - spec->autocfg.hp_pins[0] = 0x16; - spec->autocfg.speaker_pins[0] = 0x14; - spec->autocfg.speaker_pins[1] = 0x15; - spec->autocfg.num_inputs = 2; - spec->autocfg.inputs[0].pin = 0x18; - spec->autocfg.inputs[0].type = AUTO_PIN_MIC; - spec->autocfg.inputs[1].pin = 0x19; - spec->autocfg.inputs[1].type = AUTO_PIN_LINE_IN; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; -} - -static void alc680_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - if ((res >> 26) == ALC_HP_EVENT) - alc_hp_automute(codec); - if ((res >> 26) == ALC_MIC_EVENT) - alc680_rec_autoswitch(codec); -} - -static void alc680_inithook(struct hda_codec *codec) -{ - alc_hp_automute(codec); - alc680_rec_autoswitch(codec); -} - -/* - * configuration and preset - */ -static const char * const alc680_models[ALC680_MODEL_LAST] = { - [ALC680_BASE] = "base", - [ALC680_AUTO] = "auto", -}; - -static const struct snd_pci_quirk alc680_cfg_tbl[] = { - SND_PCI_QUIRK(0x1043, 0x12f3, "ASUS NX90", ALC680_BASE), - {} -}; - -static const struct alc_config_preset alc680_presets[] = { - [ALC680_BASE] = { - .mixers = { alc680_base_mixer }, - .cap_mixer = alc680_master_capture_mixer, - .init_verbs = { alc680_init_verbs }, - .num_dacs = ARRAY_SIZE(alc680_dac_nids), - .dac_nids = alc680_dac_nids, - .dig_out_nid = ALC680_DIGOUT_NID, - .num_channel_mode = ARRAY_SIZE(alc680_modes), - .channel_mode = alc680_modes, - .unsol_event = alc680_unsol_event, - .setup = alc680_base_setup, - .init_hook = alc680_inithook, - - }, -}; diff --git a/sound/pci/hda/alc861_quirks.c b/sound/pci/hda/alc861_quirks.c deleted file mode 100644 index d719ec6350eb..000000000000 --- a/sound/pci/hda/alc861_quirks.c +++ /dev/null @@ -1,725 +0,0 @@ -/* - * ALC660/ALC861 quirk models - * included by patch_realtek.c - */ - -/* ALC861 models */ -enum { - ALC861_AUTO, - ALC861_3ST, - ALC660_3ST, - ALC861_3ST_DIG, - ALC861_6ST_DIG, - ALC861_UNIWILL_M31, - ALC861_TOSHIBA, - ALC861_ASUS, - ALC861_ASUS_LAPTOP, - ALC861_MODEL_LAST, -}; - -/* - * ALC861 channel source setting (2/6 channel selection for 3-stack) - */ - -/* - * set the path ways for 2 channel output - * need to set the codec line out and mic 1 pin widgets to inputs - */ -static const struct hda_verb alc861_threestack_ch2_init[] = { - /* set pin widget 1Ah (line in) for input */ - { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 }, - /* set pin widget 18h (mic1/2) for input, for mic also enable - * the vref - */ - { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 }, - - { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb00c }, -#if 0 - { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8)) }, /*mic*/ - { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8)) }, /*line-in*/ -#endif - { } /* end */ -}; -/* - * 6ch mode - * need to set the codec line out and mic 1 pin widgets to outputs - */ -static const struct hda_verb alc861_threestack_ch6_init[] = { - /* set pin widget 1Ah (line in) for output (Back Surround)*/ - { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 }, - /* set pin widget 18h (mic1) for output (CLFE)*/ - { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 }, - - { 0x0c, AC_VERB_SET_CONNECT_SEL, 0x00 }, - { 0x0d, AC_VERB_SET_CONNECT_SEL, 0x00 }, - - { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080 }, -#if 0 - { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x01 << 8)) }, /*mic*/ - { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8)) }, /*line in*/ -#endif - { } /* end */ -}; - -static const struct hda_channel_mode alc861_threestack_modes[2] = { - { 2, alc861_threestack_ch2_init }, - { 6, alc861_threestack_ch6_init }, -}; -/* Set mic1 as input and unmute the mixer */ -static const struct hda_verb alc861_uniwill_m31_ch2_init[] = { - { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 }, - { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x01 << 8)) }, /*mic*/ - { } /* end */ -}; -/* Set mic1 as output and mute mixer */ -static const struct hda_verb alc861_uniwill_m31_ch4_init[] = { - { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 }, - { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8)) }, /*mic*/ - { } /* end */ -}; - -static const struct hda_channel_mode alc861_uniwill_m31_modes[2] = { - { 2, alc861_uniwill_m31_ch2_init }, - { 4, alc861_uniwill_m31_ch4_init }, -}; - -/* Set mic1 and line-in as input and unmute the mixer */ -static const struct hda_verb alc861_asus_ch2_init[] = { - /* set pin widget 1Ah (line in) for input */ - { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 }, - /* set pin widget 18h (mic1/2) for input, for mic also enable - * the vref - */ - { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 }, - - { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb00c }, -#if 0 - { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8)) }, /*mic*/ - { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8)) }, /*line-in*/ -#endif - { } /* end */ -}; -/* Set mic1 nad line-in as output and mute mixer */ -static const struct hda_verb alc861_asus_ch6_init[] = { - /* set pin widget 1Ah (line in) for output (Back Surround)*/ - { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 }, - /* { 0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, */ - /* set pin widget 18h (mic1) for output (CLFE)*/ - { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 }, - /* { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, */ - { 0x0c, AC_VERB_SET_CONNECT_SEL, 0x00 }, - { 0x0d, AC_VERB_SET_CONNECT_SEL, 0x00 }, - - { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080 }, -#if 0 - { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x01 << 8)) }, /*mic*/ - { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8)) }, /*line in*/ -#endif - { } /* end */ -}; - -static const struct hda_channel_mode alc861_asus_modes[2] = { - { 2, alc861_asus_ch2_init }, - { 6, alc861_asus_ch6_init }, -}; - -/* patch-ALC861 */ - -static const struct snd_kcontrol_new alc861_base_mixer[] = { - /* output mixer control */ - HDA_CODEC_MUTE("Front Playback Switch", 0x03, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Surround Playback Switch", 0x06, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x05, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x05, 2, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Side Playback Switch", 0x04, 0x0, HDA_OUTPUT), - - /*Input mixer control */ - /* HDA_CODEC_VOLUME("Input Playback Volume", 0x15, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Input Playback Switch", 0x15, 0x0, HDA_OUTPUT), */ - HDA_CODEC_VOLUME("CD Playback Volume", 0x15, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x15, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Line Playback Volume", 0x15, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("Line Playback Switch", 0x15, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x15, 0x01, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x15, 0x01, HDA_INPUT), - HDA_CODEC_MUTE("Front Mic Playback Switch", 0x10, 0x01, HDA_OUTPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x1a, 0x03, HDA_INPUT), - - { } /* end */ -}; - -static const struct snd_kcontrol_new alc861_3ST_mixer[] = { - /* output mixer control */ - HDA_CODEC_MUTE("Front Playback Switch", 0x03, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Surround Playback Switch", 0x06, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x05, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x05, 2, 0x0, HDA_OUTPUT), - /*HDA_CODEC_MUTE("Side Playback Switch", 0x04, 0x0, HDA_OUTPUT), */ - - /* Input mixer control */ - /* HDA_CODEC_VOLUME("Input Playback Volume", 0x15, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Input Playback Switch", 0x15, 0x0, HDA_OUTPUT), */ - HDA_CODEC_VOLUME("CD Playback Volume", 0x15, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x15, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Line Playback Volume", 0x15, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("Line Playback Switch", 0x15, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x15, 0x01, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x15, 0x01, HDA_INPUT), - HDA_CODEC_MUTE("Front Mic Playback Switch", 0x10, 0x01, HDA_OUTPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x1a, 0x03, HDA_INPUT), - - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Channel Mode", - .info = alc_ch_mode_info, - .get = alc_ch_mode_get, - .put = alc_ch_mode_put, - .private_value = ARRAY_SIZE(alc861_threestack_modes), - }, - { } /* end */ -}; - -static const struct snd_kcontrol_new alc861_toshiba_mixer[] = { - /* output mixer control */ - HDA_CODEC_MUTE("Master Playback Switch", 0x03, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x15, 0x01, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x15, 0x01, HDA_INPUT), - - { } /* end */ -}; - -static const struct snd_kcontrol_new alc861_uniwill_m31_mixer[] = { - /* output mixer control */ - HDA_CODEC_MUTE("Front Playback Switch", 0x03, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Surround Playback Switch", 0x06, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x05, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x05, 2, 0x0, HDA_OUTPUT), - /*HDA_CODEC_MUTE("Side Playback Switch", 0x04, 0x0, HDA_OUTPUT), */ - - /* Input mixer control */ - /* HDA_CODEC_VOLUME("Input Playback Volume", 0x15, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Input Playback Switch", 0x15, 0x0, HDA_OUTPUT), */ - HDA_CODEC_VOLUME("CD Playback Volume", 0x15, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x15, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Line Playback Volume", 0x15, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("Line Playback Switch", 0x15, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x15, 0x01, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x15, 0x01, HDA_INPUT), - HDA_CODEC_MUTE("Front Mic Playback Switch", 0x10, 0x01, HDA_OUTPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x1a, 0x03, HDA_INPUT), - - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Channel Mode", - .info = alc_ch_mode_info, - .get = alc_ch_mode_get, - .put = alc_ch_mode_put, - .private_value = ARRAY_SIZE(alc861_uniwill_m31_modes), - }, - { } /* end */ -}; - -static const struct snd_kcontrol_new alc861_asus_mixer[] = { - /* output mixer control */ - HDA_CODEC_MUTE("Front Playback Switch", 0x03, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Surround Playback Switch", 0x06, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x05, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x05, 2, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Side Playback Switch", 0x04, 0x0, HDA_OUTPUT), - - /* Input mixer control */ - HDA_CODEC_VOLUME("Input Playback Volume", 0x15, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Input Playback Switch", 0x15, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("CD Playback Volume", 0x15, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x15, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Line Playback Volume", 0x15, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("Line Playback Switch", 0x15, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x15, 0x01, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x15, 0x01, HDA_INPUT), - HDA_CODEC_MUTE("Front Mic Playback Switch", 0x10, 0x01, HDA_OUTPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x1a, 0x03, HDA_OUTPUT), - - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Channel Mode", - .info = alc_ch_mode_info, - .get = alc_ch_mode_get, - .put = alc_ch_mode_put, - .private_value = ARRAY_SIZE(alc861_asus_modes), - }, - { } -}; - -/* additional mixer */ -static const struct snd_kcontrol_new alc861_asus_laptop_mixer[] = { - HDA_CODEC_VOLUME("CD Playback Volume", 0x15, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x15, 0x0, HDA_INPUT), - { } -}; - -/* - * generic initialization of ADC, input mixers and output mixers - */ -static const struct hda_verb alc861_base_init_verbs[] = { - /* - * Unmute ADC0 and set the default input to mic-in - */ - /* port-A for surround (rear panel) */ - { 0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 }, - { 0x0e, AC_VERB_SET_CONNECT_SEL, 0x00 }, - /* port-B for mic-in (rear panel) with vref */ - { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 }, - /* port-C for line-in (rear panel) */ - { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 }, - /* port-D for Front */ - { 0x0b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 }, - { 0x0b, AC_VERB_SET_CONNECT_SEL, 0x00 }, - /* port-E for HP out (front panel) */ - { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 }, - /* route front PCM to HP */ - { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x00 }, - /* port-F for mic-in (front panel) with vref */ - { 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 }, - /* port-G for CLFE (rear panel) */ - { 0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 }, - { 0x1f, AC_VERB_SET_CONNECT_SEL, 0x00 }, - /* port-H for side (rear panel) */ - { 0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 }, - { 0x20, AC_VERB_SET_CONNECT_SEL, 0x00 }, - /* CD-in */ - { 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 }, - /* route front mic to ADC1*/ - {0x08, AC_VERB_SET_CONNECT_SEL, 0x00}, - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - - /* Unmute DAC0~3 & spdif out*/ - {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - - /* Unmute Mixer 14 (mic) 1c (Line in)*/ - {0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - - /* Unmute Stereo Mixer 15 */ - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb00c}, /* Output 0~12 step */ - - {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - /* hp used DAC 3 (Front) */ - {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)}, - {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)}, - - { } -}; - -static const struct hda_verb alc861_threestack_init_verbs[] = { - /* - * Unmute ADC0 and set the default input to mic-in - */ - /* port-A for surround (rear panel) */ - { 0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 }, - /* port-B for mic-in (rear panel) with vref */ - { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 }, - /* port-C for line-in (rear panel) */ - { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 }, - /* port-D for Front */ - { 0x0b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 }, - { 0x0b, AC_VERB_SET_CONNECT_SEL, 0x00 }, - /* port-E for HP out (front panel) */ - { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 }, - /* route front PCM to HP */ - { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x00 }, - /* port-F for mic-in (front panel) with vref */ - { 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 }, - /* port-G for CLFE (rear panel) */ - { 0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 }, - /* port-H for side (rear panel) */ - { 0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 }, - /* CD-in */ - { 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 }, - /* route front mic to ADC1*/ - {0x08, AC_VERB_SET_CONNECT_SEL, 0x00}, - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - /* Unmute DAC0~3 & spdif out*/ - {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - - /* Unmute Mixer 14 (mic) 1c (Line in)*/ - {0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - - /* Unmute Stereo Mixer 15 */ - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb00c}, /* Output 0~12 step */ - - {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - /* hp used DAC 3 (Front) */ - {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)}, - {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)}, - { } -}; - -static const struct hda_verb alc861_uniwill_m31_init_verbs[] = { - /* - * Unmute ADC0 and set the default input to mic-in - */ - /* port-A for surround (rear panel) */ - { 0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 }, - /* port-B for mic-in (rear panel) with vref */ - { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 }, - /* port-C for line-in (rear panel) */ - { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 }, - /* port-D for Front */ - { 0x0b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 }, - { 0x0b, AC_VERB_SET_CONNECT_SEL, 0x00 }, - /* port-E for HP out (front panel) */ - /* this has to be set to VREF80 */ - { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 }, - /* route front PCM to HP */ - { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x00 }, - /* port-F for mic-in (front panel) with vref */ - { 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 }, - /* port-G for CLFE (rear panel) */ - { 0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 }, - /* port-H for side (rear panel) */ - { 0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 }, - /* CD-in */ - { 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 }, - /* route front mic to ADC1*/ - {0x08, AC_VERB_SET_CONNECT_SEL, 0x00}, - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - /* Unmute DAC0~3 & spdif out*/ - {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - - /* Unmute Mixer 14 (mic) 1c (Line in)*/ - {0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - - /* Unmute Stereo Mixer 15 */ - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb00c}, /* Output 0~12 step */ - - {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - /* hp used DAC 3 (Front) */ - {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)}, - {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)}, - { } -}; - -static const struct hda_verb alc861_asus_init_verbs[] = { - /* - * Unmute ADC0 and set the default input to mic-in - */ - /* port-A for surround (rear panel) - * according to codec#0 this is the HP jack - */ - { 0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 }, /* was 0x00 */ - /* route front PCM to HP */ - { 0x0e, AC_VERB_SET_CONNECT_SEL, 0x01 }, - /* port-B for mic-in (rear panel) with vref */ - { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 }, - /* port-C for line-in (rear panel) */ - { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 }, - /* port-D for Front */ - { 0x0b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 }, - { 0x0b, AC_VERB_SET_CONNECT_SEL, 0x00 }, - /* port-E for HP out (front panel) */ - /* this has to be set to VREF80 */ - { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 }, - /* route front PCM to HP */ - { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x00 }, - /* port-F for mic-in (front panel) with vref */ - { 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 }, - /* port-G for CLFE (rear panel) */ - { 0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 }, - /* port-H for side (rear panel) */ - { 0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 }, - /* CD-in */ - { 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 }, - /* route front mic to ADC1*/ - {0x08, AC_VERB_SET_CONNECT_SEL, 0x00}, - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - /* Unmute DAC0~3 & spdif out*/ - {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* Unmute Mixer 14 (mic) 1c (Line in)*/ - {0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - - /* Unmute Stereo Mixer 15 */ - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb00c}, /* Output 0~12 step */ - - {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - /* hp used DAC 3 (Front) */ - {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)}, - {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)}, - { } -}; - -/* additional init verbs for ASUS laptops */ -static const struct hda_verb alc861_asus_laptop_init_verbs[] = { - { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x45 }, /* HP-out */ - { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2) }, /* mute line-in */ - { } -}; - -static const struct hda_verb alc861_toshiba_init_verbs[] = { - {0x0f, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT}, - - { } -}; - -/* toggle speaker-output according to the hp-jack state */ -static void alc861_toshiba_automute(struct hda_codec *codec) -{ - unsigned int present = snd_hda_jack_detect(codec, 0x0f); - - snd_hda_codec_amp_stereo(codec, 0x16, HDA_INPUT, 0, - HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); - snd_hda_codec_amp_stereo(codec, 0x1a, HDA_INPUT, 3, - HDA_AMP_MUTE, present ? 0 : HDA_AMP_MUTE); -} - -static void alc861_toshiba_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - if ((res >> 26) == ALC_HP_EVENT) - alc861_toshiba_automute(codec); -} - -#define ALC861_DIGOUT_NID 0x07 - -static const struct hda_channel_mode alc861_8ch_modes[1] = { - { 8, NULL } -}; - -static const hda_nid_t alc861_dac_nids[4] = { - /* front, surround, clfe, side */ - 0x03, 0x06, 0x05, 0x04 -}; - -static const hda_nid_t alc660_dac_nids[3] = { - /* front, clfe, surround */ - 0x03, 0x05, 0x06 -}; - -static const hda_nid_t alc861_adc_nids[1] = { - /* ADC0-2 */ - 0x08, -}; - -static const struct hda_input_mux alc861_capture_source = { - .num_items = 5, - .items = { - { "Mic", 0x0 }, - { "Front Mic", 0x3 }, - { "Line", 0x1 }, - { "CD", 0x4 }, - { "Mixer", 0x5 }, - }, -}; - -/* - * configuration and preset - */ -static const char * const alc861_models[ALC861_MODEL_LAST] = { - [ALC861_3ST] = "3stack", - [ALC660_3ST] = "3stack-660", - [ALC861_3ST_DIG] = "3stack-dig", - [ALC861_6ST_DIG] = "6stack-dig", - [ALC861_UNIWILL_M31] = "uniwill-m31", - [ALC861_TOSHIBA] = "toshiba", - [ALC861_ASUS] = "asus", - [ALC861_ASUS_LAPTOP] = "asus-laptop", - [ALC861_AUTO] = "auto", -}; - -static const struct snd_pci_quirk alc861_cfg_tbl[] = { - SND_PCI_QUIRK(0x1043, 0x1205, "ASUS W7J", ALC861_3ST), - SND_PCI_QUIRK(0x1043, 0x1335, "ASUS F2/3", ALC861_ASUS_LAPTOP), - SND_PCI_QUIRK(0x1043, 0x1338, "ASUS F2/3", ALC861_ASUS_LAPTOP), - SND_PCI_QUIRK(0x1043, 0x1393, "ASUS", ALC861_ASUS), - SND_PCI_QUIRK(0x1043, 0x13d7, "ASUS A9rp", ALC861_ASUS_LAPTOP), - SND_PCI_QUIRK(0x1043, 0x81cb, "ASUS P1-AH2", ALC861_3ST_DIG), - SND_PCI_QUIRK(0x1179, 0xff00, "Toshiba", ALC861_TOSHIBA), - /* FIXME: the entry below breaks Toshiba A100 (model=auto works!) - * Any other models that need this preset? - */ - /* SND_PCI_QUIRK(0x1179, 0xff10, "Toshiba", ALC861_TOSHIBA), */ - SND_PCI_QUIRK(0x1462, 0x7254, "HP dx2200 (MSI MS-7254)", ALC861_3ST), - SND_PCI_QUIRK(0x1462, 0x7297, "HP dx2250 (MSI MS-7297)", ALC861_3ST), - SND_PCI_QUIRK(0x1584, 0x2b01, "Uniwill X40AIx", ALC861_UNIWILL_M31), - SND_PCI_QUIRK(0x1584, 0x9072, "Uniwill m31", ALC861_UNIWILL_M31), - SND_PCI_QUIRK(0x1584, 0x9075, "Airis Praxis N1212", ALC861_ASUS_LAPTOP), - /* FIXME: the below seems conflict */ - /* SND_PCI_QUIRK(0x1584, 0x9075, "Uniwill", ALC861_UNIWILL_M31), */ - SND_PCI_QUIRK(0x1849, 0x0660, "Asrock 939SLI32", ALC660_3ST), - SND_PCI_QUIRK(0x8086, 0xd600, "Intel", ALC861_3ST), - {} -}; - -static const struct alc_config_preset alc861_presets[] = { - [ALC861_3ST] = { - .mixers = { alc861_3ST_mixer }, - .init_verbs = { alc861_threestack_init_verbs }, - .num_dacs = ARRAY_SIZE(alc861_dac_nids), - .dac_nids = alc861_dac_nids, - .num_channel_mode = ARRAY_SIZE(alc861_threestack_modes), - .channel_mode = alc861_threestack_modes, - .need_dac_fix = 1, - .num_adc_nids = ARRAY_SIZE(alc861_adc_nids), - .adc_nids = alc861_adc_nids, - .input_mux = &alc861_capture_source, - }, - [ALC861_3ST_DIG] = { - .mixers = { alc861_base_mixer }, - .init_verbs = { alc861_threestack_init_verbs }, - .num_dacs = ARRAY_SIZE(alc861_dac_nids), - .dac_nids = alc861_dac_nids, - .dig_out_nid = ALC861_DIGOUT_NID, - .num_channel_mode = ARRAY_SIZE(alc861_threestack_modes), - .channel_mode = alc861_threestack_modes, - .need_dac_fix = 1, - .num_adc_nids = ARRAY_SIZE(alc861_adc_nids), - .adc_nids = alc861_adc_nids, - .input_mux = &alc861_capture_source, - }, - [ALC861_6ST_DIG] = { - .mixers = { alc861_base_mixer }, - .init_verbs = { alc861_base_init_verbs }, - .num_dacs = ARRAY_SIZE(alc861_dac_nids), - .dac_nids = alc861_dac_nids, - .dig_out_nid = ALC861_DIGOUT_NID, - .num_channel_mode = ARRAY_SIZE(alc861_8ch_modes), - .channel_mode = alc861_8ch_modes, - .num_adc_nids = ARRAY_SIZE(alc861_adc_nids), - .adc_nids = alc861_adc_nids, - .input_mux = &alc861_capture_source, - }, - [ALC660_3ST] = { - .mixers = { alc861_3ST_mixer }, - .init_verbs = { alc861_threestack_init_verbs }, - .num_dacs = ARRAY_SIZE(alc660_dac_nids), - .dac_nids = alc660_dac_nids, - .num_channel_mode = ARRAY_SIZE(alc861_threestack_modes), - .channel_mode = alc861_threestack_modes, - .need_dac_fix = 1, - .num_adc_nids = ARRAY_SIZE(alc861_adc_nids), - .adc_nids = alc861_adc_nids, - .input_mux = &alc861_capture_source, - }, - [ALC861_UNIWILL_M31] = { - .mixers = { alc861_uniwill_m31_mixer }, - .init_verbs = { alc861_uniwill_m31_init_verbs }, - .num_dacs = ARRAY_SIZE(alc861_dac_nids), - .dac_nids = alc861_dac_nids, - .dig_out_nid = ALC861_DIGOUT_NID, - .num_channel_mode = ARRAY_SIZE(alc861_uniwill_m31_modes), - .channel_mode = alc861_uniwill_m31_modes, - .need_dac_fix = 1, - .num_adc_nids = ARRAY_SIZE(alc861_adc_nids), - .adc_nids = alc861_adc_nids, - .input_mux = &alc861_capture_source, - }, - [ALC861_TOSHIBA] = { - .mixers = { alc861_toshiba_mixer }, - .init_verbs = { alc861_base_init_verbs, - alc861_toshiba_init_verbs }, - .num_dacs = ARRAY_SIZE(alc861_dac_nids), - .dac_nids = alc861_dac_nids, - .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), - .channel_mode = alc883_3ST_2ch_modes, - .num_adc_nids = ARRAY_SIZE(alc861_adc_nids), - .adc_nids = alc861_adc_nids, - .input_mux = &alc861_capture_source, - .unsol_event = alc861_toshiba_unsol_event, - .init_hook = alc861_toshiba_automute, - }, - [ALC861_ASUS] = { - .mixers = { alc861_asus_mixer }, - .init_verbs = { alc861_asus_init_verbs }, - .num_dacs = ARRAY_SIZE(alc861_dac_nids), - .dac_nids = alc861_dac_nids, - .dig_out_nid = ALC861_DIGOUT_NID, - .num_channel_mode = ARRAY_SIZE(alc861_asus_modes), - .channel_mode = alc861_asus_modes, - .need_dac_fix = 1, - .hp_nid = 0x06, - .num_adc_nids = ARRAY_SIZE(alc861_adc_nids), - .adc_nids = alc861_adc_nids, - .input_mux = &alc861_capture_source, - }, - [ALC861_ASUS_LAPTOP] = { - .mixers = { alc861_toshiba_mixer, alc861_asus_laptop_mixer }, - .init_verbs = { alc861_asus_init_verbs, - alc861_asus_laptop_init_verbs }, - .num_dacs = ARRAY_SIZE(alc861_dac_nids), - .dac_nids = alc861_dac_nids, - .dig_out_nid = ALC861_DIGOUT_NID, - .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), - .channel_mode = alc883_3ST_2ch_modes, - .need_dac_fix = 1, - .num_adc_nids = ARRAY_SIZE(alc861_adc_nids), - .adc_nids = alc861_adc_nids, - .input_mux = &alc861_capture_source, - }, -}; - diff --git a/sound/pci/hda/alc861vd_quirks.c b/sound/pci/hda/alc861vd_quirks.c deleted file mode 100644 index 8f28450f41f8..000000000000 --- a/sound/pci/hda/alc861vd_quirks.c +++ /dev/null @@ -1,605 +0,0 @@ -/* - * ALC660-VD/ALC861-VD quirk models - * included by patch_realtek.c - */ - -/* ALC861-VD models */ -enum { - ALC861VD_AUTO, - ALC660VD_3ST, - ALC660VD_3ST_DIG, - ALC660VD_ASUS_V1S, - ALC861VD_3ST, - ALC861VD_3ST_DIG, - ALC861VD_6ST_DIG, - ALC861VD_LENOVO, - ALC861VD_DALLAS, - ALC861VD_HP, - ALC861VD_MODEL_LAST, -}; - -#define ALC861VD_DIGOUT_NID 0x06 - -static const hda_nid_t alc861vd_dac_nids[4] = { - /* front, surr, clfe, side surr */ - 0x02, 0x03, 0x04, 0x05 -}; - -/* dac_nids for ALC660vd are in a different order - according to - * Realtek's driver. - * This should probably result in a different mixer for 6stack models - * of ALC660vd codecs, but for now there is only 3stack mixer - * - and it is the same as in 861vd. - * adc_nids in ALC660vd are (is) the same as in 861vd - */ -static const hda_nid_t alc660vd_dac_nids[3] = { - /* front, rear, clfe, rear_surr */ - 0x02, 0x04, 0x03 -}; - -static const hda_nid_t alc861vd_adc_nids[1] = { - /* ADC0 */ - 0x09, -}; - -static const hda_nid_t alc861vd_capsrc_nids[1] = { 0x22 }; - -/* input MUX */ -/* FIXME: should be a matrix-type input source selection */ -static const struct hda_input_mux alc861vd_capture_source = { - .num_items = 4, - .items = { - { "Mic", 0x0 }, - { "Front Mic", 0x1 }, - { "Line", 0x2 }, - { "CD", 0x4 }, - }, -}; - -static const struct hda_input_mux alc861vd_dallas_capture_source = { - .num_items = 2, - .items = { - { "Mic", 0x0 }, - { "Internal Mic", 0x1 }, - }, -}; - -static const struct hda_input_mux alc861vd_hp_capture_source = { - .num_items = 2, - .items = { - { "Front Mic", 0x0 }, - { "ATAPI Mic", 0x1 }, - }, -}; - -/* - * 2ch mode - */ -static const struct hda_channel_mode alc861vd_3stack_2ch_modes[1] = { - { 2, NULL } -}; - -/* - * 6ch mode - */ -static const struct hda_verb alc861vd_6stack_ch6_init[] = { - { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 }, - { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { } /* end */ -}; - -/* - * 8ch mode - */ -static const struct hda_verb alc861vd_6stack_ch8_init[] = { - { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { } /* end */ -}; - -static const struct hda_channel_mode alc861vd_6stack_modes[2] = { - { 6, alc861vd_6stack_ch6_init }, - { 8, alc861vd_6stack_ch8_init }, -}; - -static const struct snd_kcontrol_new alc861vd_chmode_mixer[] = { - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Channel Mode", - .info = alc_ch_mode_info, - .get = alc_ch_mode_get, - .put = alc_ch_mode_put, - }, - { } /* end */ -}; - -/* Pin assignment: Front=0x14, Rear=0x15, CLFE=0x16, Side=0x17 - * Mic=0x18, Front Mic=0x19, Line-In=0x1a, HP=0x1b - */ -static const struct snd_kcontrol_new alc861vd_6st_mixer[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), - - HDA_CODEC_VOLUME("Surround Playback Volume", 0x03, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT), - - HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x04, 1, 0x0, - HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x04, 2, 0x0, - HDA_OUTPUT), - HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT), - HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT), - - HDA_CODEC_VOLUME("Side Playback Volume", 0x05, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT), - - HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), - - HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - - HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - - HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), - - HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), - - { } /* end */ -}; - -static const struct snd_kcontrol_new alc861vd_3st_mixer[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), - - HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), - - HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - - HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - - HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), - - HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), - - { } /* end */ -}; - -static const struct snd_kcontrol_new alc861vd_lenovo_mixer[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT), - /*HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),*/ - HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT), - - HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), - - HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - - HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - - HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), - - { } /* end */ -}; - -/* Pin assignment: Speaker=0x14, HP = 0x15, - * Mic=0x18, Internal Mic = 0x19, CD = 0x1c, PC Beep = 0x1d - */ -static const struct snd_kcontrol_new alc861vd_dallas_mixer[] = { - HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Speaker Playback Switch", 0x0c, 2, HDA_INPUT), - HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Headphone Playback Switch", 0x0d, 2, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT), - HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - { } /* end */ -}; - -/* Pin assignment: Speaker=0x14, Line-out = 0x15, - * Front Mic=0x18, ATAPI Mic = 0x19, - */ -static const struct snd_kcontrol_new alc861vd_hp_mixer[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), - HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Headphone Playback Switch", 0x0d, 2, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("ATAPI Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("ATAPI Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - - { } /* end */ -}; - -/* - * generic initialization of ADC, input mixers and output mixers - */ -static const struct hda_verb alc861vd_volume_init_verbs[] = { - /* - * Unmute ADC0 and set the default input to mic-in - */ - {0x09, AC_VERB_SET_CONNECT_SEL, 0x00}, - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - - /* Unmute input amps (CD, Line In, Mic 1 & Mic 2) of - * the analog-loopback mixer widget - */ - /* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */ - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - - /* Capture mixer: unmute Mic, F-Mic, Line, CD inputs */ - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)}, - - /* - * Set up output mixers (0x02 - 0x05) - */ - /* set vol=0 to output mixers */ - {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - - /* set up input amps for analog loopback */ - /* Amp Indices: DAC = 0, mixer = 1 */ - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - - { } -}; - -/* - * 3-stack pin configuration: - * front = 0x14, mic/clfe = 0x18, HP = 0x19, line/surr = 0x1a, f-mic = 0x1b - */ -static const struct hda_verb alc861vd_3stack_init_verbs[] = { - /* - * Set pin mode and muting - */ - /* set front pin widgets 0x14 for output */ - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, - - /* Mic (rear) pin: input vref at 80% */ - {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* Front Mic pin: input vref at 80% */ - {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* Line In pin: input */ - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* Line-2 In: Headphone output (output 0 - 0x0c) */ - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00}, - /* CD pin widget for input */ - {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - - { } -}; - -/* - * 6-stack pin configuration: - */ -static const struct hda_verb alc861vd_6stack_init_verbs[] = { - /* - * Set pin mode and muting - */ - /* set front pin widgets 0x14 for output */ - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, - - /* Rear Pin: output 1 (0x0d) */ - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, - /* CLFE Pin: output 2 (0x0e) */ - {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x16, AC_VERB_SET_CONNECT_SEL, 0x02}, - /* Side Pin: output 3 (0x0f) */ - {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x17, AC_VERB_SET_CONNECT_SEL, 0x03}, - - /* Mic (rear) pin: input vref at 80% */ - {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* Front Mic pin: input vref at 80% */ - {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* Line In pin: input */ - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* Line-2 In: Headphone output (output 0 - 0x0c) */ - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00}, - /* CD pin widget for input */ - {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - - { } -}; - -static const struct hda_verb alc861vd_eapd_verbs[] = { - {0x14, AC_VERB_SET_EAPD_BTLENABLE, 2}, - { } -}; - -static const struct hda_verb alc861vd_lenovo_unsol_verbs[] = { - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, - {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT}, - {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT}, - {} -}; - -static void alc861vd_lenovo_setup(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - spec->autocfg.hp_pins[0] = 0x1b; - spec->autocfg.speaker_pins[0] = 0x14; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; -} - -static void alc861vd_lenovo_init_hook(struct hda_codec *codec) -{ - alc_hp_automute(codec); - alc88x_simple_mic_automute(codec); -} - -static void alc861vd_lenovo_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - switch (res >> 26) { - case ALC_MIC_EVENT: - alc88x_simple_mic_automute(codec); - break; - default: - alc_sku_unsol_event(codec, res); - break; - } -} - -static const struct hda_verb alc861vd_dallas_verbs[] = { - {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - - {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50}, - {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50}, - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x09, AC_VERB_SET_CONNECT_SEL, 0x00}, - {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT}, - - { } /* end */ -}; - -/* toggle speaker-output according to the hp-jack state */ -static void alc861vd_dallas_setup(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - - spec->autocfg.hp_pins[0] = 0x15; - spec->autocfg.speaker_pins[0] = 0x14; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; -} - -/* - * configuration and preset - */ -static const char * const alc861vd_models[ALC861VD_MODEL_LAST] = { - [ALC660VD_3ST] = "3stack-660", - [ALC660VD_3ST_DIG] = "3stack-660-digout", - [ALC660VD_ASUS_V1S] = "asus-v1s", - [ALC861VD_3ST] = "3stack", - [ALC861VD_3ST_DIG] = "3stack-digout", - [ALC861VD_6ST_DIG] = "6stack-digout", - [ALC861VD_LENOVO] = "lenovo", - [ALC861VD_DALLAS] = "dallas", - [ALC861VD_HP] = "hp", - [ALC861VD_AUTO] = "auto", -}; - -static const struct snd_pci_quirk alc861vd_cfg_tbl[] = { - SND_PCI_QUIRK(0x1019, 0xa88d, "Realtek ALC660 demo", ALC660VD_3ST), - SND_PCI_QUIRK(0x103c, 0x30bf, "HP TX1000", ALC861VD_HP), - SND_PCI_QUIRK(0x1043, 0x12e2, "Asus z35m", ALC660VD_3ST), - /*SND_PCI_QUIRK(0x1043, 0x1339, "Asus G1", ALC660VD_3ST),*/ /* auto */ - SND_PCI_QUIRK(0x1043, 0x1633, "Asus V1Sn", ALC660VD_ASUS_V1S), - SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS", ALC660VD_3ST_DIG), - SND_PCI_QUIRK(0x10de, 0x03f0, "Realtek ALC660 demo", ALC660VD_3ST), - SND_PCI_QUIRK(0x1179, 0xff00, "Toshiba A135", ALC861VD_LENOVO), - /*SND_PCI_QUIRK(0x1179, 0xff00, "DALLAS", ALC861VD_DALLAS),*/ /*lenovo*/ - SND_PCI_QUIRK(0x1179, 0xff01, "Toshiba A135", ALC861VD_LENOVO), - SND_PCI_QUIRK(0x1179, 0xff03, "Toshiba P205", ALC861VD_LENOVO), - SND_PCI_QUIRK(0x1179, 0xff31, "Toshiba L30-149", ALC861VD_DALLAS), - SND_PCI_QUIRK(0x1565, 0x820d, "Biostar NF61S SE", ALC861VD_6ST_DIG), - SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo", ALC861VD_LENOVO), - SND_PCI_QUIRK(0x1849, 0x0862, "ASRock K8NF6G-VSTA", ALC861VD_6ST_DIG), - {} -}; - -static const struct alc_config_preset alc861vd_presets[] = { - [ALC660VD_3ST] = { - .mixers = { alc861vd_3st_mixer }, - .init_verbs = { alc861vd_volume_init_verbs, - alc861vd_3stack_init_verbs }, - .num_dacs = ARRAY_SIZE(alc660vd_dac_nids), - .dac_nids = alc660vd_dac_nids, - .num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes), - .channel_mode = alc861vd_3stack_2ch_modes, - .input_mux = &alc861vd_capture_source, - }, - [ALC660VD_3ST_DIG] = { - .mixers = { alc861vd_3st_mixer }, - .init_verbs = { alc861vd_volume_init_verbs, - alc861vd_3stack_init_verbs }, - .num_dacs = ARRAY_SIZE(alc660vd_dac_nids), - .dac_nids = alc660vd_dac_nids, - .dig_out_nid = ALC861VD_DIGOUT_NID, - .num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes), - .channel_mode = alc861vd_3stack_2ch_modes, - .input_mux = &alc861vd_capture_source, - }, - [ALC861VD_3ST] = { - .mixers = { alc861vd_3st_mixer }, - .init_verbs = { alc861vd_volume_init_verbs, - alc861vd_3stack_init_verbs }, - .num_dacs = ARRAY_SIZE(alc861vd_dac_nids), - .dac_nids = alc861vd_dac_nids, - .num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes), - .channel_mode = alc861vd_3stack_2ch_modes, - .input_mux = &alc861vd_capture_source, - }, - [ALC861VD_3ST_DIG] = { - .mixers = { alc861vd_3st_mixer }, - .init_verbs = { alc861vd_volume_init_verbs, - alc861vd_3stack_init_verbs }, - .num_dacs = ARRAY_SIZE(alc861vd_dac_nids), - .dac_nids = alc861vd_dac_nids, - .dig_out_nid = ALC861VD_DIGOUT_NID, - .num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes), - .channel_mode = alc861vd_3stack_2ch_modes, - .input_mux = &alc861vd_capture_source, - }, - [ALC861VD_6ST_DIG] = { - .mixers = { alc861vd_6st_mixer, alc861vd_chmode_mixer }, - .init_verbs = { alc861vd_volume_init_verbs, - alc861vd_6stack_init_verbs }, - .num_dacs = ARRAY_SIZE(alc861vd_dac_nids), - .dac_nids = alc861vd_dac_nids, - .dig_out_nid = ALC861VD_DIGOUT_NID, - .num_channel_mode = ARRAY_SIZE(alc861vd_6stack_modes), - .channel_mode = alc861vd_6stack_modes, - .input_mux = &alc861vd_capture_source, - }, - [ALC861VD_LENOVO] = { - .mixers = { alc861vd_lenovo_mixer }, - .init_verbs = { alc861vd_volume_init_verbs, - alc861vd_3stack_init_verbs, - alc861vd_eapd_verbs, - alc861vd_lenovo_unsol_verbs }, - .num_dacs = ARRAY_SIZE(alc660vd_dac_nids), - .dac_nids = alc660vd_dac_nids, - .num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes), - .channel_mode = alc861vd_3stack_2ch_modes, - .input_mux = &alc861vd_capture_source, - .unsol_event = alc861vd_lenovo_unsol_event, - .setup = alc861vd_lenovo_setup, - .init_hook = alc861vd_lenovo_init_hook, - }, - [ALC861VD_DALLAS] = { - .mixers = { alc861vd_dallas_mixer }, - .init_verbs = { alc861vd_dallas_verbs }, - .num_dacs = ARRAY_SIZE(alc861vd_dac_nids), - .dac_nids = alc861vd_dac_nids, - .num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes), - .channel_mode = alc861vd_3stack_2ch_modes, - .input_mux = &alc861vd_dallas_capture_source, - .unsol_event = alc_sku_unsol_event, - .setup = alc861vd_dallas_setup, - .init_hook = alc_hp_automute, - }, - [ALC861VD_HP] = { - .mixers = { alc861vd_hp_mixer }, - .init_verbs = { alc861vd_dallas_verbs, alc861vd_eapd_verbs }, - .num_dacs = ARRAY_SIZE(alc861vd_dac_nids), - .dac_nids = alc861vd_dac_nids, - .dig_out_nid = ALC861VD_DIGOUT_NID, - .num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes), - .channel_mode = alc861vd_3stack_2ch_modes, - .input_mux = &alc861vd_hp_capture_source, - .unsol_event = alc_sku_unsol_event, - .setup = alc861vd_dallas_setup, - .init_hook = alc_hp_automute, - }, - [ALC660VD_ASUS_V1S] = { - .mixers = { alc861vd_lenovo_mixer }, - .init_verbs = { alc861vd_volume_init_verbs, - alc861vd_3stack_init_verbs, - alc861vd_eapd_verbs, - alc861vd_lenovo_unsol_verbs }, - .num_dacs = ARRAY_SIZE(alc660vd_dac_nids), - .dac_nids = alc660vd_dac_nids, - .dig_out_nid = ALC861VD_DIGOUT_NID, - .num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes), - .channel_mode = alc861vd_3stack_2ch_modes, - .input_mux = &alc861vd_capture_source, - .unsol_event = alc861vd_lenovo_unsol_event, - .setup = alc861vd_lenovo_setup, - .init_hook = alc861vd_lenovo_init_hook, - }, -}; - diff --git a/sound/pci/hda/alc880_quirks.c b/sound/pci/hda/alc880_quirks.c index c844d2b59988..bea22edcfd8c 100644 --- a/sound/pci/hda/alc880_quirks.c +++ b/sound/pci/hda/alc880_quirks.c @@ -749,8 +749,7 @@ static void alc880_uniwill_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x16; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); } static void alc880_uniwill_init_hook(struct hda_codec *codec) @@ -781,8 +780,7 @@ static void alc880_uniwill_p53_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x15; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); } static void alc880_uniwill_p53_dcvol_automute(struct hda_codec *codec) @@ -1051,8 +1049,7 @@ static void alc880_lg_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x1b; spec->autocfg.speaker_pins[0] = 0x17; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); } /* @@ -1137,8 +1134,7 @@ static void alc880_lg_lw_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x1b; spec->autocfg.speaker_pins[0] = 0x14; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); } static const struct snd_kcontrol_new alc880_medion_rim_mixer[] = { @@ -1188,7 +1184,7 @@ static void alc880_medion_rim_automute(struct hda_codec *codec) struct alc_spec *spec = codec->spec; alc_hp_automute(codec); /* toggle EAPD */ - if (spec->jack_present) + if (spec->hp_jack_present) snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, 0); else snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, 2); @@ -1210,8 +1206,7 @@ static void alc880_medion_rim_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x1b; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); } #ifdef CONFIG_SND_HDA_POWER_SAVE diff --git a/sound/pci/hda/alc882_quirks.c b/sound/pci/hda/alc882_quirks.c index 617d04723b82..e251514a26a4 100644 --- a/sound/pci/hda/alc882_quirks.c +++ b/sound/pci/hda/alc882_quirks.c @@ -173,8 +173,7 @@ static void alc889_automute_setup(struct hda_codec *codec) spec->autocfg.speaker_pins[2] = 0x17; spec->autocfg.speaker_pins[3] = 0x19; spec->autocfg.speaker_pins[4] = 0x1a; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); } static void alc889_intel_init_hook(struct hda_codec *codec) @@ -191,8 +190,7 @@ static void alc888_fujitsu_xa3530_setup(struct hda_codec *codec) spec->autocfg.hp_pins[1] = 0x1b; /* hp */ spec->autocfg.speaker_pins[0] = 0x14; /* speaker */ spec->autocfg.speaker_pins[1] = 0x15; /* bass */ - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); } /* @@ -475,8 +473,7 @@ static void alc888_acer_aspire_4930g_setup(struct hda_codec *codec) spec->autocfg.speaker_pins[0] = 0x14; spec->autocfg.speaker_pins[1] = 0x16; spec->autocfg.speaker_pins[2] = 0x17; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); } static void alc888_acer_aspire_6530g_setup(struct hda_codec *codec) @@ -487,8 +484,7 @@ static void alc888_acer_aspire_6530g_setup(struct hda_codec *codec) spec->autocfg.speaker_pins[0] = 0x14; spec->autocfg.speaker_pins[1] = 0x16; spec->autocfg.speaker_pins[2] = 0x17; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); } static void alc888_acer_aspire_7730g_setup(struct hda_codec *codec) @@ -499,8 +495,7 @@ static void alc888_acer_aspire_7730g_setup(struct hda_codec *codec) spec->autocfg.speaker_pins[0] = 0x14; spec->autocfg.speaker_pins[1] = 0x16; spec->autocfg.speaker_pins[2] = 0x17; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); } static void alc889_acer_aspire_8930g_setup(struct hda_codec *codec) @@ -511,8 +506,7 @@ static void alc889_acer_aspire_8930g_setup(struct hda_codec *codec) spec->autocfg.speaker_pins[0] = 0x14; spec->autocfg.speaker_pins[1] = 0x16; spec->autocfg.speaker_pins[2] = 0x1b; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); } #define ALC882_DIGOUT_NID 0x06 @@ -1711,8 +1705,7 @@ static void alc885_imac24_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x18; spec->autocfg.speaker_pins[1] = 0x1a; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); } #define alc885_mb5_setup alc885_imac24_setup @@ -1721,12 +1714,11 @@ static void alc885_imac24_setup(struct hda_codec *codec) /* Macbook Air 2,1 */ static void alc885_mba21_setup(struct hda_codec *codec) { - struct alc_spec *spec = codec->spec; + struct alc_spec *spec = codec->spec; - spec->autocfg.hp_pins[0] = 0x14; - spec->autocfg.speaker_pins[0] = 0x18; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + spec->autocfg.hp_pins[0] = 0x14; + spec->autocfg.speaker_pins[0] = 0x18; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); } @@ -1737,8 +1729,7 @@ static void alc885_mbp3_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); } static void alc885_imac91_setup(struct hda_codec *codec) @@ -1748,8 +1739,7 @@ static void alc885_imac91_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x18; spec->autocfg.speaker_pins[1] = 0x1a; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); } static const struct hda_verb alc882_targa_verbs[] = { @@ -1773,7 +1763,7 @@ static void alc882_targa_automute(struct hda_codec *codec) struct alc_spec *spec = codec->spec; alc_hp_automute(codec); snd_hda_codec_write_cache(codec, 1, 0, AC_VERB_SET_GPIO_DATA, - spec->jack_present ? 1 : 3); + spec->hp_jack_present ? 1 : 3); } static void alc882_targa_setup(struct hda_codec *codec) @@ -1782,8 +1772,7 @@ static void alc882_targa_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x1b; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); } static void alc882_targa_unsol_event(struct hda_codec *codec, unsigned int res) @@ -2187,8 +2176,7 @@ static void alc883_medion_wim2160_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x1a; spec->autocfg.speaker_pins[0] = 0x15; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); } static const struct snd_kcontrol_new alc883_acer_aspire_mixer[] = { @@ -2341,8 +2329,7 @@ static void alc883_mitac_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; spec->autocfg.speaker_pins[1] = 0x17; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); } static const struct hda_verb alc883_mitac_verbs[] = { @@ -2507,8 +2494,7 @@ static void alc888_3st_hp_setup(struct hda_codec *codec) spec->autocfg.speaker_pins[0] = 0x14; spec->autocfg.speaker_pins[1] = 0x16; spec->autocfg.speaker_pins[2] = 0x18; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); } static const struct hda_verb alc888_3st_hp_verbs[] = { @@ -2568,8 +2554,7 @@ static void alc888_lenovo_ms7195_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x1b; spec->autocfg.line_out_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x15; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); } /* toggle speaker-output according to the hp-jack state */ @@ -2579,8 +2564,7 @@ static void alc883_lenovo_nb0763_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x15; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); } /* toggle speaker-output according to the hp-jack state */ @@ -2593,8 +2577,7 @@ static void alc883_clevo_m720_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); } static void alc883_clevo_m720_init_hook(struct hda_codec *codec) @@ -2623,8 +2606,7 @@ static void alc883_2ch_fujitsu_pi2515_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x15; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); } static void alc883_haier_w66_setup(struct hda_codec *codec) @@ -2633,8 +2615,7 @@ static void alc883_haier_w66_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x1b; spec->autocfg.speaker_pins[0] = 0x14; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); } static void alc883_lenovo_101e_setup(struct hda_codec *codec) @@ -2644,10 +2625,7 @@ static void alc883_lenovo_101e_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x1b; spec->autocfg.line_out_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x15; - spec->automute = 1; - spec->detect_line = 1; - spec->automute_lines = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); } /* toggle speaker-output according to the hp-jack state */ @@ -2658,8 +2636,7 @@ static void alc883_acer_aspire_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x15; spec->autocfg.speaker_pins[1] = 0x16; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); } static const struct hda_verb alc883_acer_eapd_verbs[] = { @@ -2689,8 +2666,7 @@ static void alc888_6st_dell_setup(struct hda_codec *codec) spec->autocfg.speaker_pins[1] = 0x15; spec->autocfg.speaker_pins[2] = 0x16; spec->autocfg.speaker_pins[3] = 0x17; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); } static void alc888_lenovo_sky_setup(struct hda_codec *codec) @@ -2703,8 +2679,7 @@ static void alc888_lenovo_sky_setup(struct hda_codec *codec) spec->autocfg.speaker_pins[2] = 0x16; spec->autocfg.speaker_pins[3] = 0x17; spec->autocfg.speaker_pins[4] = 0x1a; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); } static void alc883_vaiott_setup(struct hda_codec *codec) @@ -2714,8 +2689,7 @@ static void alc883_vaiott_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; spec->autocfg.speaker_pins[1] = 0x17; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); } static const struct hda_verb alc888_asus_m90v_verbs[] = { @@ -2739,8 +2713,7 @@ static void alc883_mode2_setup(struct hda_codec *codec) spec->ext_mic_pin = 0x18; spec->int_mic_pin = 0x19; spec->auto_mic = 1; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); } static const struct hda_verb alc888_asus_eee1601_verbs[] = { diff --git a/sound/pci/hda/alc_quirks.c b/sound/pci/hda/alc_quirks.c index 2be1129cf458..a18952ed4311 100644 --- a/sound/pci/hda/alc_quirks.c +++ b/sound/pci/hda/alc_quirks.c @@ -453,6 +453,19 @@ static void setup_preset(struct hda_codec *codec, alc_fixup_autocfg_pin_nums(codec); } +static void alc_simple_setup_automute(struct alc_spec *spec, int mode) +{ + int lo_pin = spec->autocfg.line_out_pins[0]; + + if (lo_pin == spec->autocfg.speaker_pins[0] || + lo_pin == spec->autocfg.hp_pins[0]) + lo_pin = 0; + spec->automute_mode = mode; + spec->detect_hp = !!spec->autocfg.hp_pins[0]; + spec->detect_lo = !!lo_pin; + spec->automute_lo = spec->automute_lo_possible = !!lo_pin; + spec->automute_speaker = spec->automute_speaker_possible = !!spec->autocfg.speaker_pins[0]; +} /* auto-toggle front mic */ static void alc88x_simple_mic_automute(struct hda_codec *codec) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index f3aefef37216..1715e8b24ff0 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -34,6 +34,9 @@ #include "hda_beep.h" #include <sound/hda_hwdep.h> +#define CREATE_TRACE_POINTS +#include "hda_trace.h" + /* * vendor / preset table */ @@ -208,15 +211,19 @@ static int codec_exec_verb(struct hda_codec *codec, unsigned int cmd, again: snd_hda_power_up(codec); mutex_lock(&bus->cmd_mutex); + trace_hda_send_cmd(codec, cmd); err = bus->ops.command(bus, cmd); - if (!err && res) + if (!err && res) { *res = bus->ops.get_response(bus, codec->addr); + trace_hda_get_response(codec, *res); + } mutex_unlock(&bus->cmd_mutex); snd_hda_power_down(codec); if (res && *res == -1 && bus->rirb_error) { if (bus->response_reset) { snd_printd("hda_codec: resetting BUS due to " "fatal communication error\n"); + trace_hda_bus_reset(bus); bus->ops.bus_reset(bus); } goto again; @@ -607,6 +614,7 @@ int snd_hda_queue_unsol_event(struct hda_bus *bus, u32 res, u32 res_ex) struct hda_bus_unsolicited *unsol; unsigned int wp; + trace_hda_unsol_event(bus, res, res_ex); unsol = bus->unsol; if (!unsol) return 0; @@ -1483,8 +1491,11 @@ static void really_cleanup_stream(struct hda_codec *codec, struct hda_cvt_setup *q) { hda_nid_t nid = q->nid; - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CHANNEL_STREAMID, 0); - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_STREAM_FORMAT, 0); + if (q->stream_tag || q->channel_id) + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CHANNEL_STREAMID, 0); + if (q->format_id) + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_STREAM_FORMAT, 0 +); memset(q, 0, sizeof(*q)); q->nid = nid; } @@ -1689,6 +1700,29 @@ u32 snd_hda_query_pin_caps(struct hda_codec *codec, hda_nid_t nid) EXPORT_SYMBOL_HDA(snd_hda_query_pin_caps); /** + * snd_hda_override_pin_caps - Override the pin capabilities + * @codec: the CODEC + * @nid: the NID to override + * @caps: the capability bits to set + * + * Override the cached PIN capabilitiy bits value by the given one. + * + * Returns zero if successful or a negative error code. + */ +int snd_hda_override_pin_caps(struct hda_codec *codec, hda_nid_t nid, + unsigned int caps) +{ + struct hda_amp_info *info; + info = get_alloc_amp_hash(codec, HDA_HASH_PINCAP_KEY(nid)); + if (!info) + return -ENOMEM; + info->amp_caps = caps; + info->head.val |= INFO_AMP_CAPS; + return 0; +} +EXPORT_SYMBOL_HDA(snd_hda_override_pin_caps); + +/** * snd_hda_pin_sense - execute pin sense measurement * @codec: the CODEC to sense * @nid: the pin NID to sense @@ -4087,6 +4121,7 @@ static void hda_power_work(struct work_struct *work) return; } + trace_hda_power_down(codec); hda_call_codec_suspend(codec); if (bus->ops.pm_notify) bus->ops.pm_notify(bus); @@ -4125,6 +4160,7 @@ void snd_hda_power_up(struct hda_codec *codec) if (codec->power_on || codec->power_transition) return; + trace_hda_power_up(codec); snd_hda_update_power_acct(codec); codec->power_on = 1; codec->power_jiffies = jiffies; @@ -4537,6 +4573,11 @@ int snd_hda_multi_out_analog_prepare(struct hda_codec *codec, snd_hda_codec_setup_stream(codec, mout->hp_nid, stream_tag, 0, format); /* extra outputs copied from front */ + for (i = 0; i < ARRAY_SIZE(mout->hp_out_nid); i++) + if (!mout->no_share_stream && mout->hp_out_nid[i]) + snd_hda_codec_setup_stream(codec, + mout->hp_out_nid[i], + stream_tag, 0, format); for (i = 0; i < ARRAY_SIZE(mout->extra_out_nid); i++) if (!mout->no_share_stream && mout->extra_out_nid[i]) snd_hda_codec_setup_stream(codec, @@ -4569,6 +4610,10 @@ int snd_hda_multi_out_analog_cleanup(struct hda_codec *codec, snd_hda_codec_cleanup_stream(codec, nids[i]); if (mout->hp_nid) snd_hda_codec_cleanup_stream(codec, mout->hp_nid); + for (i = 0; i < ARRAY_SIZE(mout->hp_out_nid); i++) + if (mout->hp_out_nid[i]) + snd_hda_codec_cleanup_stream(codec, + mout->hp_out_nid[i]); for (i = 0; i < ARRAY_SIZE(mout->extra_out_nid); i++) if (mout->extra_out_nid[i]) snd_hda_codec_cleanup_stream(codec, @@ -4649,6 +4694,27 @@ static void sort_autocfg_input_pins(struct auto_pin_cfg *cfg) } } +/* Reorder the surround channels + * ALSA sequence is front/surr/clfe/side + * HDA sequence is: + * 4-ch: front/surr => OK as it is + * 6-ch: front/clfe/surr + * 8-ch: front/clfe/rear/side|fc + */ +static void reorder_outputs(unsigned int nums, hda_nid_t *pins) +{ + hda_nid_t nid; + + switch (nums) { + case 3: + case 4: + nid = pins[1]; + pins[1] = pins[2]; + pins[2] = nid; + break; + } +} + /* * Parse all pin widgets and store the useful pin nids to cfg * @@ -4666,12 +4732,13 @@ static void sort_autocfg_input_pins(struct auto_pin_cfg *cfg) * The digital input/output pins are assigned to dig_in_pin and dig_out_pin, * respectively. */ -int snd_hda_parse_pin_def_config(struct hda_codec *codec, - struct auto_pin_cfg *cfg, - const hda_nid_t *ignore_nids) +int snd_hda_parse_pin_defcfg(struct hda_codec *codec, + struct auto_pin_cfg *cfg, + const hda_nid_t *ignore_nids, + unsigned int cond_flags) { hda_nid_t nid, end_nid; - short seq, assoc_line_out, assoc_speaker; + short seq, assoc_line_out; short sequences_line_out[ARRAY_SIZE(cfg->line_out_pins)]; short sequences_speaker[ARRAY_SIZE(cfg->speaker_pins)]; short sequences_hp[ARRAY_SIZE(cfg->hp_pins)]; @@ -4682,7 +4749,7 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, memset(sequences_line_out, 0, sizeof(sequences_line_out)); memset(sequences_speaker, 0, sizeof(sequences_speaker)); memset(sequences_hp, 0, sizeof(sequences_hp)); - assoc_line_out = assoc_speaker = 0; + assoc_line_out = 0; end_nid = codec->start_nid + codec->num_nodes; for (nid = codec->start_nid; nid < end_nid; nid++) { @@ -4734,16 +4801,10 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, case AC_JACK_SPEAKER: seq = get_defcfg_sequence(def_conf); assoc = get_defcfg_association(def_conf); - if (!assoc) - continue; - if (!assoc_speaker) - assoc_speaker = assoc; - else if (assoc_speaker != assoc) - continue; if (cfg->speaker_outs >= ARRAY_SIZE(cfg->speaker_pins)) continue; cfg->speaker_pins[cfg->speaker_outs] = nid; - sequences_speaker[cfg->speaker_outs] = seq; + sequences_speaker[cfg->speaker_outs] = (assoc << 4) | seq; cfg->speaker_outs++; break; case AC_JACK_HP_OUT: @@ -4792,7 +4853,8 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, * If no line-out is defined but multiple HPs are found, * some of them might be the real line-outs. */ - if (!cfg->line_outs && cfg->hp_outs > 1) { + if (!cfg->line_outs && cfg->hp_outs > 1 && + !(cond_flags & HDA_PINCFG_NO_HP_FIXUP)) { int i = 0; while (i < cfg->hp_outs) { /* The real HPs should have the sequence 0x0f */ @@ -4829,7 +4891,8 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, * FIX-UP: if no line-outs are detected, try to use speaker or HP pin * as a primary output */ - if (!cfg->line_outs) { + if (!cfg->line_outs && + !(cond_flags & HDA_PINCFG_NO_LO_FIXUP)) { if (cfg->speaker_outs) { cfg->line_outs = cfg->speaker_outs; memcpy(cfg->line_out_pins, cfg->speaker_pins, @@ -4847,21 +4910,9 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, } } - /* Reorder the surround channels - * ALSA sequence is front/surr/clfe/side - * HDA sequence is: - * 4-ch: front/surr => OK as it is - * 6-ch: front/clfe/surr - * 8-ch: front/clfe/rear/side|fc - */ - switch (cfg->line_outs) { - case 3: - case 4: - nid = cfg->line_out_pins[1]; - cfg->line_out_pins[1] = cfg->line_out_pins[2]; - cfg->line_out_pins[2] = nid; - break; - } + reorder_outputs(cfg->line_outs, cfg->line_out_pins); + reorder_outputs(cfg->hp_outs, cfg->hp_pins); + reorder_outputs(cfg->speaker_outs, cfg->speaker_pins); sort_autocfg_input_pins(cfg); @@ -4899,7 +4950,7 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, return 0; } -EXPORT_SYMBOL_HDA(snd_hda_parse_pin_def_config); +EXPORT_SYMBOL_HDA(snd_hda_parse_pin_defcfg); int snd_hda_get_input_pin_attr(unsigned int def_conf) { @@ -5158,30 +5209,6 @@ void snd_array_free(struct snd_array *array) EXPORT_SYMBOL_HDA(snd_array_free); /** - * snd_print_pcm_rates - Print the supported PCM rates to the string buffer - * @pcm: PCM caps bits - * @buf: the string buffer to write - * @buflen: the max buffer length - * - * used by hda_proc.c and hda_eld.c - */ -void snd_print_pcm_rates(int pcm, char *buf, int buflen) -{ - static unsigned int rates[] = { - 8000, 11025, 16000, 22050, 32000, 44100, 48000, 88200, - 96000, 176400, 192000, 384000 - }; - int i, j; - - for (i = 0, j = 0; i < ARRAY_SIZE(rates); i++) - if (pcm & (1 << i)) - j += snprintf(buf + j, buflen - j, " %d", rates[i]); - - buf[j] = '\0'; /* necessary when j == 0 */ -} -EXPORT_SYMBOL_HDA(snd_print_pcm_rates); - -/** * snd_print_pcm_bits - Print the supported PCM fmt bits to the string buffer * @pcm: PCM caps bits * @buf: the string buffer to write @@ -5222,6 +5249,8 @@ static const char *get_jack_default_name(struct hda_codec *codec, hda_nid_t nid, return "Mic"; case SND_JACK_LINEOUT: return "Line-out"; + case SND_JACK_LINEIN: + return "Line-in"; case SND_JACK_HEADSET: return "Headset"; case SND_JACK_VIDEOOUT: diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c index c34f730f4815..1c8ddf547a2d 100644 --- a/sound/pci/hda/hda_eld.c +++ b/sound/pci/hda/hda_eld.c @@ -318,6 +318,11 @@ int snd_hdmi_get_eld(struct hdmi_eld *eld, int size; unsigned char *buf; + /* + * ELD size is initialized to zero in caller function. If no errors and + * ELD is valid, actual eld_size is assigned in hdmi_update_eld() + */ + if (!eld->eld_valid) return -ENOENT; @@ -327,14 +332,13 @@ int snd_hdmi_get_eld(struct hdmi_eld *eld, snd_printd(KERN_INFO "HDMI: ELD buf size is 0, force 128\n"); size = 128; } - if (size < ELD_FIXED_BYTES || size > PAGE_SIZE) { + if (size < ELD_FIXED_BYTES || size > ELD_MAX_SIZE) { snd_printd(KERN_INFO "HDMI: invalid ELD buf size %d\n", size); return -ERANGE; } - buf = kmalloc(size, GFP_KERNEL); - if (!buf) - return -ENOMEM; + /* set ELD buffer */ + buf = eld->eld_buffer; for (i = 0; i < size; i++) { unsigned int val = hdmi_get_eld_data(codec, nid, i); @@ -356,10 +360,31 @@ int snd_hdmi_get_eld(struct hdmi_eld *eld, ret = hdmi_update_eld(eld, buf, size); error: - kfree(buf); return ret; } +/** + * SNDRV_PCM_RATE_* and AC_PAR_PCM values don't match, print correct rates with + * hdmi-specific routine. + */ +static void hdmi_print_pcm_rates(int pcm, char *buf, int buflen) +{ + static unsigned int alsa_rates[] = { + 5512, 8000, 11025, 16000, 22050, 32000, 44100, 48000, 88200, + 96000, 176400, 192000, 384000 + }; + int i, j; + + for (i = 0, j = 0; i < ARRAY_SIZE(alsa_rates); i++) + if (pcm & (1 << i)) + j += snprintf(buf + j, buflen - j, " %d", + alsa_rates[i]); + + buf[j] = '\0'; /* necessary when j == 0 */ +} + +#define SND_PRINT_RATES_ADVISED_BUFSIZE 80 + static void hdmi_show_short_audio_desc(struct cea_sad *a) { char buf[SND_PRINT_RATES_ADVISED_BUFSIZE]; @@ -368,7 +393,7 @@ static void hdmi_show_short_audio_desc(struct cea_sad *a) if (!a->format) return; - snd_print_pcm_rates(a->rates, buf, sizeof(buf)); + hdmi_print_pcm_rates(a->rates, buf, sizeof(buf)); if (a->format == AUDIO_CODING_TYPE_LPCM) snd_print_pcm_bits(a->sample_bits, buf2 + 8, sizeof(buf2) - 8); @@ -427,7 +452,7 @@ static void hdmi_print_sad_info(int i, struct cea_sad *a, i, a->format, cea_audio_coding_type_names[a->format]); snd_iprintf(buffer, "sad%d_channels\t\t%d\n", i, a->channels); - snd_print_pcm_rates(a->rates, buf, sizeof(buf)); + hdmi_print_pcm_rates(a->rates, buf, sizeof(buf)); snd_iprintf(buffer, "sad%d_rates\t\t[0x%x]%s\n", i, a->rates, buf); if (a->format == AUDIO_CODING_TYPE_LPCM) { diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c index bf3ced51e0f8..72e5885007cc 100644 --- a/sound/pci/hda/hda_hwdep.c +++ b/sound/pci/hda/hda_hwdep.c @@ -643,14 +643,14 @@ static inline int strmatch(const char *a, const char *b) static void parse_codec_mode(char *buf, struct hda_bus *bus, struct hda_codec **codecp) { - unsigned int vendorid, subid, caddr; + int vendorid, subid, caddr; struct hda_codec *codec; *codecp = NULL; if (sscanf(buf, "%i %i %i", &vendorid, &subid, &caddr) == 3) { list_for_each_entry(codec, &bus->codec_list, list) { - if (codec->vendor_id == vendorid && - codec->subsystem_id == subid && + if ((vendorid <= 0 || codec->vendor_id == vendorid) && + (subid <= 0 || codec->subsystem_id == subid) && codec->addr == caddr) { *codecp = codec; break; diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 191284a1c0ae..bd7fc99af187 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -34,7 +34,6 @@ * */ -#include <asm/io.h> #include <linux/delay.h> #include <linux/interrupt.h> #include <linux/kernel.h> @@ -46,6 +45,12 @@ #include <linux/pci.h> #include <linux/mutex.h> #include <linux/reboot.h> +#include <linux/io.h> +#ifdef CONFIG_X86 +/* for snoop control */ +#include <asm/pgtable.h> +#include <asm/cacheflush.h> +#endif #include <sound/core.h> #include <sound/initval.h> #include "hda_codec.h" @@ -116,6 +121,22 @@ module_param(power_save_controller, bool, 0644); MODULE_PARM_DESC(power_save_controller, "Reset controller in power save mode."); #endif +static int align_buffer_size = 1; +module_param(align_buffer_size, bool, 0644); +MODULE_PARM_DESC(align_buffer_size, + "Force buffer and period sizes to be multiple of 128 bytes."); + +#ifdef CONFIG_X86 +static bool hda_snoop = true; +module_param_named(snoop, hda_snoop, bool, 0444); +MODULE_PARM_DESC(snoop, "Enable/disable snooping"); +#define azx_snoop(chip) (chip)->snoop +#else +#define hda_snoop true +#define azx_snoop(chip) true +#endif + + MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Intel, ICH6}," "{Intel, ICH6M}," @@ -360,7 +381,7 @@ struct azx_dev { */ unsigned char stream_tag; /* assigned stream */ unsigned char index; /* stream index */ - int device; /* last device number assigned to */ + int assigned_key; /* last device# key assigned to */ unsigned int opened :1; unsigned int running :1; @@ -371,6 +392,7 @@ struct azx_dev { * when link position is not greater than FIFO size */ unsigned int insufficient :1; + unsigned int wc_marked:1; }; /* CORB/RIRB */ @@ -438,6 +460,7 @@ struct azx { unsigned int msi :1; unsigned int irq_pending_warned :1; unsigned int probing :1; /* codec probing phase */ + unsigned int snoop:1; /* for debugging */ unsigned int last_cmd[AZX_MAX_CODECS]; @@ -481,6 +504,7 @@ enum { #define AZX_DCAPS_NO_64BIT (1 << 18) /* No 64bit address */ #define AZX_DCAPS_SYNC_WRITE (1 << 19) /* sync each cmd write */ #define AZX_DCAPS_OLD_SSYNC (1 << 20) /* Old SSYNC reg for ICH */ +#define AZX_DCAPS_BUFSIZE (1 << 21) /* no buffer size alignment */ /* quirks for ATI SB / AMD Hudson */ #define AZX_DCAPS_PRESET_ATI_SB \ @@ -542,6 +566,45 @@ static char *driver_short_names[] __devinitdata = { /* for pcm support */ #define get_azx_dev(substream) (substream->runtime->private_data) +#ifdef CONFIG_X86 +static void __mark_pages_wc(struct azx *chip, void *addr, size_t size, bool on) +{ + if (azx_snoop(chip)) + return; + if (addr && size) { + int pages = (size + PAGE_SIZE - 1) >> PAGE_SHIFT; + if (on) + set_memory_wc((unsigned long)addr, pages); + else + set_memory_wb((unsigned long)addr, pages); + } +} + +static inline void mark_pages_wc(struct azx *chip, struct snd_dma_buffer *buf, + bool on) +{ + __mark_pages_wc(chip, buf->area, buf->bytes, on); +} +static inline void mark_runtime_wc(struct azx *chip, struct azx_dev *azx_dev, + struct snd_pcm_runtime *runtime, bool on) +{ + if (azx_dev->wc_marked != on) { + __mark_pages_wc(chip, runtime->dma_area, runtime->dma_bytes, on); + azx_dev->wc_marked = on; + } +} +#else +/* NOP for other archs */ +static inline void mark_pages_wc(struct azx *chip, struct snd_dma_buffer *buf, + bool on) +{ +} +static inline void mark_runtime_wc(struct azx *chip, struct azx_dev *azx_dev, + struct snd_pcm_runtime *runtime, bool on) +{ +} +#endif + static int azx_acquire_irq(struct azx *chip, int do_disconnect); static int azx_send_cmd(struct hda_bus *bus, unsigned int val); /* @@ -563,6 +626,7 @@ static int azx_alloc_cmd_io(struct azx *chip) snd_printk(KERN_ERR SFX "cannot allocate CORB/RIRB\n"); return err; } + mark_pages_wc(chip, &chip->rb, true); return 0; } @@ -1079,7 +1143,15 @@ static void update_pci_byte(struct pci_dev *pci, unsigned int reg, static void azx_init_pci(struct azx *chip) { - unsigned short snoop; + /* force to non-snoop mode for a new VIA controller when BIOS is set */ + if (chip->snoop && chip->driver_type == AZX_DRIVER_VIA) { + u8 snoop; + pci_read_config_byte(chip->pci, 0x42, &snoop); + if (!(snoop & 0x80) && chip->pci->revision == 0x30) { + chip->snoop = 0; + snd_printdd(SFX "Force to non-snoop mode\n"); + } + } /* Clear bits 0-2 of PCI register TCSEL (at offset 0x44) * TCSEL == Traffic Class Select Register, which sets PCI express QOS @@ -1096,15 +1168,15 @@ static void azx_init_pci(struct azx *chip) * we need to enable snoop. */ if (chip->driver_caps & AZX_DCAPS_ATI_SNOOP) { - snd_printdd(SFX "Enabling ATI snoop\n"); + snd_printdd(SFX "Setting ATI snoop: %d\n", azx_snoop(chip)); update_pci_byte(chip->pci, - ATI_SB450_HDAUDIO_MISC_CNTR2_ADDR, - 0x07, ATI_SB450_HDAUDIO_ENABLE_SNOOP); + ATI_SB450_HDAUDIO_MISC_CNTR2_ADDR, 0x07, + azx_snoop(chip) ? ATI_SB450_HDAUDIO_ENABLE_SNOOP : 0); } /* For NVIDIA HDA, enable snoop */ if (chip->driver_caps & AZX_DCAPS_NVIDIA_SNOOP) { - snd_printdd(SFX "Enabling Nvidia snoop\n"); + snd_printdd(SFX "Setting Nvidia snoop: %d\n", azx_snoop(chip)); update_pci_byte(chip->pci, NVIDIA_HDA_TRANSREG_ADDR, 0x0f, NVIDIA_HDA_ENABLE_COHBITS); @@ -1118,16 +1190,20 @@ static void azx_init_pci(struct azx *chip) /* Enable SCH/PCH snoop if needed */ if (chip->driver_caps & AZX_DCAPS_SCH_SNOOP) { + unsigned short snoop; pci_read_config_word(chip->pci, INTEL_SCH_HDA_DEVC, &snoop); - if (snoop & INTEL_SCH_HDA_DEVC_NOSNOOP) { - pci_write_config_word(chip->pci, INTEL_SCH_HDA_DEVC, - snoop & (~INTEL_SCH_HDA_DEVC_NOSNOOP)); + if ((!azx_snoop(chip) && !(snoop & INTEL_SCH_HDA_DEVC_NOSNOOP)) || + (azx_snoop(chip) && (snoop & INTEL_SCH_HDA_DEVC_NOSNOOP))) { + snoop &= ~INTEL_SCH_HDA_DEVC_NOSNOOP; + if (!azx_snoop(chip)) + snoop |= INTEL_SCH_HDA_DEVC_NOSNOOP; + pci_write_config_word(chip->pci, INTEL_SCH_HDA_DEVC, snoop); pci_read_config_word(chip->pci, INTEL_SCH_HDA_DEVC, &snoop); - snd_printdd(SFX "HDA snoop disabled, enabling ... %s\n", - (snoop & INTEL_SCH_HDA_DEVC_NOSNOOP) - ? "Failed" : "OK"); } + snd_printdd(SFX "SCH snoop: %s\n", + (snoop & INTEL_SCH_HDA_DEVC_NOSNOOP) + ? "Disabled" : "Enabled"); } } @@ -1334,12 +1410,16 @@ static void azx_stream_reset(struct azx *chip, struct azx_dev *azx_dev) */ static int azx_setup_controller(struct azx *chip, struct azx_dev *azx_dev) { + unsigned int val; /* make sure the run bit is zero for SD */ azx_stream_clear(chip, azx_dev); /* program the stream_tag */ - azx_sd_writel(azx_dev, SD_CTL, - (azx_sd_readl(azx_dev, SD_CTL) & ~SD_CTL_STREAM_TAG_MASK)| - (azx_dev->stream_tag << SD_CTL_STREAM_TAG_SHIFT)); + val = azx_sd_readl(azx_dev, SD_CTL); + val = (val & ~SD_CTL_STREAM_TAG_MASK) | + (azx_dev->stream_tag << SD_CTL_STREAM_TAG_SHIFT); + if (!azx_snoop(chip)) + val |= SD_CTL_TRAFFIC_PRIO; + azx_sd_writel(azx_dev, SD_CTL, val); /* program the length of samples in cyclic buffer */ azx_sd_writel(azx_dev, SD_CBL, azx_dev->bufsize); @@ -1533,6 +1613,9 @@ azx_assign_device(struct azx *chip, struct snd_pcm_substream *substream) { int dev, i, nums; struct azx_dev *res = NULL; + /* make a non-zero unique key for the substream */ + int key = (substream->pcm->device << 16) | (substream->number << 2) | + (substream->stream + 1); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { dev = chip->playback_index_offset; @@ -1544,12 +1627,12 @@ azx_assign_device(struct azx *chip, struct snd_pcm_substream *substream) for (i = 0; i < nums; i++, dev++) if (!chip->azx_dev[dev].opened) { res = &chip->azx_dev[dev]; - if (res->device == substream->pcm->device) + if (res->assigned_key == key) break; } if (res) { res->opened = 1; - res->device = substream->pcm->device; + res->assigned_key = key; } return res; } @@ -1599,6 +1682,7 @@ static int azx_pcm_open(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; unsigned long flags; int err; + int buff_step; mutex_lock(&chip->open_mutex); azx_dev = azx_assign_device(chip, substream); @@ -1613,10 +1697,25 @@ static int azx_pcm_open(struct snd_pcm_substream *substream) runtime->hw.rates = hinfo->rates; snd_pcm_limit_hw_rates(runtime); snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); + if (align_buffer_size) + /* constrain buffer sizes to be multiple of 128 + bytes. This is more efficient in terms of memory + access but isn't required by the HDA spec and + prevents users from specifying exact period/buffer + sizes. For example for 44.1kHz, a period size set + to 20ms will be rounded to 19.59ms. */ + buff_step = 128; + else + /* Don't enforce steps on buffer sizes, still need to + be multiple of 4 bytes (HDA spec). Tested on Intel + HDA controllers, may not work on all devices where + option needs to be disabled */ + buff_step = 4; + snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, - 128); + buff_step); snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, - 128); + buff_step); snd_hda_power_up(apcm->codec); err = hinfo->ops.open(hinfo, apcm->codec, substream); if (err < 0) { @@ -1671,19 +1770,30 @@ static int azx_pcm_close(struct snd_pcm_substream *substream) static int azx_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hw_params) { + struct azx_pcm *apcm = snd_pcm_substream_chip(substream); + struct azx *chip = apcm->chip; + struct snd_pcm_runtime *runtime = substream->runtime; struct azx_dev *azx_dev = get_azx_dev(substream); + int ret; + mark_runtime_wc(chip, azx_dev, runtime, false); azx_dev->bufsize = 0; azx_dev->period_bytes = 0; azx_dev->format_val = 0; - return snd_pcm_lib_malloc_pages(substream, + ret = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); + if (ret < 0) + return ret; + mark_runtime_wc(chip, azx_dev, runtime, true); + return ret; } static int azx_pcm_hw_free(struct snd_pcm_substream *substream) { struct azx_pcm *apcm = snd_pcm_substream_chip(substream); struct azx_dev *azx_dev = get_azx_dev(substream); + struct azx *chip = apcm->chip; + struct snd_pcm_runtime *runtime = substream->runtime; struct hda_pcm_stream *hinfo = apcm->hinfo[substream->stream]; /* reset BDL address */ @@ -1696,6 +1806,7 @@ static int azx_pcm_hw_free(struct snd_pcm_substream *substream) snd_hda_codec_cleanup(apcm->codec, hinfo, substream); + mark_runtime_wc(chip, azx_dev, runtime, false); return snd_pcm_lib_free_pages(substream); } @@ -2055,6 +2166,20 @@ static void azx_clear_irq_pending(struct azx *chip) spin_unlock_irq(&chip->reg_lock); } +#ifdef CONFIG_X86 +static int azx_pcm_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *area) +{ + struct azx_pcm *apcm = snd_pcm_substream_chip(substream); + struct azx *chip = apcm->chip; + if (!azx_snoop(chip)) + area->vm_page_prot = pgprot_writecombine(area->vm_page_prot); + return snd_pcm_lib_default_mmap(substream, area); +} +#else +#define azx_pcm_mmap NULL +#endif + static struct snd_pcm_ops azx_pcm_ops = { .open = azx_pcm_open, .close = azx_pcm_close, @@ -2064,6 +2189,7 @@ static struct snd_pcm_ops azx_pcm_ops = { .prepare = azx_pcm_prepare, .trigger = azx_pcm_trigger, .pointer = azx_pcm_pointer, + .mmap = azx_pcm_mmap, .page = snd_pcm_sgbuf_ops_page, }; @@ -2344,13 +2470,19 @@ static int azx_free(struct azx *chip) if (chip->azx_dev) { for (i = 0; i < chip->num_streams; i++) - if (chip->azx_dev[i].bdl.area) + if (chip->azx_dev[i].bdl.area) { + mark_pages_wc(chip, &chip->azx_dev[i].bdl, false); snd_dma_free_pages(&chip->azx_dev[i].bdl); + } } - if (chip->rb.area) + if (chip->rb.area) { + mark_pages_wc(chip, &chip->rb, false); snd_dma_free_pages(&chip->rb); - if (chip->posbuf.area) + } + if (chip->posbuf.area) { + mark_pages_wc(chip, &chip->posbuf, false); snd_dma_free_pages(&chip->posbuf); + } pci_release_regions(chip->pci); pci_disable_device(chip->pci); kfree(chip->azx_dev); @@ -2546,6 +2678,7 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, check_probe_mask(chip, dev); chip->single_cmd = single_cmd; + chip->snoop = hda_snoop; if (bdl_pos_adj[dev] < 0) { switch (chip->driver_type) { @@ -2618,6 +2751,10 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, gcap &= ~ICH6_GCAP_64OK; } + /* disable buffer size rounding to 128-byte multiples if supported */ + if (chip->driver_caps & AZX_DCAPS_BUFSIZE) + align_buffer_size = 0; + /* allow 64bit DMA address if supported by H/W */ if ((gcap & ICH6_GCAP_64OK) && !pci_set_dma_mask(pci, DMA_BIT_MASK(64))) pci_set_consistent_dma_mask(pci, DMA_BIT_MASK(64)); @@ -2669,6 +2806,7 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, snd_printk(KERN_ERR SFX "cannot allocate BDL\n"); goto errout; } + mark_pages_wc(chip, &chip->azx_dev[i].bdl, true); } /* allocate memory for the position buffer */ err = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, @@ -2678,6 +2816,7 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, snd_printk(KERN_ERR SFX "cannot allocate posbuf\n"); goto errout; } + mark_pages_wc(chip, &chip->posbuf, true); /* allocate CORB/RIRB */ err = azx_alloc_cmd_io(chip); if (err < 0) @@ -2819,37 +2958,49 @@ static void __devexit azx_remove(struct pci_dev *pci) static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { /* CPT */ { PCI_DEVICE(0x8086, 0x1c20), - .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP }, + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | + AZX_DCAPS_BUFSIZE }, /* PBG */ { PCI_DEVICE(0x8086, 0x1d20), - .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP }, + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | + AZX_DCAPS_BUFSIZE}, /* Panther Point */ { PCI_DEVICE(0x8086, 0x1e20), - .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP }, + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | + AZX_DCAPS_BUFSIZE}, /* SCH */ { PCI_DEVICE(0x8086, 0x811b), - .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP }, + .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP | + AZX_DCAPS_BUFSIZE}, { PCI_DEVICE(0x8086, 0x2668), - .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC }, /* ICH6 */ + .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC | + AZX_DCAPS_BUFSIZE }, /* ICH6 */ { PCI_DEVICE(0x8086, 0x27d8), - .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC }, /* ICH7 */ + .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC | + AZX_DCAPS_BUFSIZE }, /* ICH7 */ { PCI_DEVICE(0x8086, 0x269a), - .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC }, /* ESB2 */ + .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC | + AZX_DCAPS_BUFSIZE }, /* ESB2 */ { PCI_DEVICE(0x8086, 0x284b), - .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC }, /* ICH8 */ + .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC | + AZX_DCAPS_BUFSIZE }, /* ICH8 */ { PCI_DEVICE(0x8086, 0x293e), - .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC }, /* ICH9 */ + .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC | + AZX_DCAPS_BUFSIZE }, /* ICH9 */ { PCI_DEVICE(0x8086, 0x293f), - .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC }, /* ICH9 */ + .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC | + AZX_DCAPS_BUFSIZE }, /* ICH9 */ { PCI_DEVICE(0x8086, 0x3a3e), - .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC }, /* ICH10 */ + .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC | + AZX_DCAPS_BUFSIZE }, /* ICH10 */ { PCI_DEVICE(0x8086, 0x3a6e), - .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC }, /* ICH10 */ + .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC | + AZX_DCAPS_BUFSIZE }, /* ICH10 */ /* Generic Intel */ { PCI_DEVICE(PCI_VENDOR_ID_INTEL, PCI_ANY_ID), .class = PCI_CLASS_MULTIMEDIA_HD_AUDIO << 8, .class_mask = 0xffffff, - .driver_data = AZX_DRIVER_ICH }, + .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_BUFSIZE }, /* ATI SB 450/600/700/800/900 */ { PCI_DEVICE(0x1002, 0x437b), .driver_data = AZX_DRIVER_ATI | AZX_DCAPS_PRESET_ATI_SB }, diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 2e7ac31afa8d..81e12c0ed0a2 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -267,11 +267,14 @@ int snd_hda_ch_mode_put(struct hda_codec *codec, enum { HDA_FRONT, HDA_REAR, HDA_CLFE, HDA_SIDE }; /* index for dac_nidx */ enum { HDA_DIG_NONE, HDA_DIG_EXCLUSIVE, HDA_DIG_ANALOG_DUP }; /* dig_out_used */ +#define HDA_MAX_OUTS 5 + struct hda_multi_out { int num_dacs; /* # of DACs, must be more than 1 */ const hda_nid_t *dac_nids; /* DAC list */ hda_nid_t hp_nid; /* optional DAC for HP, 0 when not exists */ - hda_nid_t extra_out_nid[3]; /* optional DACs, 0 when not exists */ + hda_nid_t hp_out_nid[HDA_MAX_OUTS]; /* DACs for multiple HPs */ + hda_nid_t extra_out_nid[HDA_MAX_OUTS]; /* other (e.g. speaker) DACs */ hda_nid_t dig_out_nid; /* digital out audio widget */ const hda_nid_t *slave_dig_outs; int max_channels; /* currently supported analog channels */ @@ -333,9 +336,6 @@ int snd_hda_codec_proc_new(struct hda_codec *codec); static inline int snd_hda_codec_proc_new(struct hda_codec *codec) { return 0; } #endif -#define SND_PRINT_RATES_ADVISED_BUFSIZE 80 -void snd_print_pcm_rates(int pcm, char *buf, int buflen); - #define SND_PRINT_BITS_ADVISED_BUFSIZE 16 void snd_print_pcm_bits(int pcm, char *buf, int buflen); @@ -385,7 +385,7 @@ enum { AUTO_PIN_HP_OUT }; -#define AUTO_CFG_MAX_OUTS 5 +#define AUTO_CFG_MAX_OUTS HDA_MAX_OUTS #define AUTO_CFG_MAX_INS 8 struct auto_pin_cfg_item { @@ -443,9 +443,18 @@ struct auto_pin_cfg { #define get_defcfg_device(cfg) \ ((cfg & AC_DEFCFG_DEVICE) >> AC_DEFCFG_DEVICE_SHIFT) -int snd_hda_parse_pin_def_config(struct hda_codec *codec, - struct auto_pin_cfg *cfg, - const hda_nid_t *ignore_nids); +/* bit-flags for snd_hda_parse_pin_def_config() behavior */ +#define HDA_PINCFG_NO_HP_FIXUP (1 << 0) /* no HP-split */ +#define HDA_PINCFG_NO_LO_FIXUP (1 << 1) /* don't take other outs as LO */ + +int snd_hda_parse_pin_defcfg(struct hda_codec *codec, + struct auto_pin_cfg *cfg, + const hda_nid_t *ignore_nids, + unsigned int cond_flags); + +/* older function */ +#define snd_hda_parse_pin_def_config(codec, cfg, ignore) \ + snd_hda_parse_pin_defcfg(codec, cfg, ignore, 0) /* amp values */ #define AMP_IN_MUTE(idx) (0x7080 | ((idx)<<8)) @@ -492,6 +501,8 @@ u32 query_amp_caps(struct hda_codec *codec, hda_nid_t nid, int direction); int snd_hda_override_amp_caps(struct hda_codec *codec, hda_nid_t nid, int dir, unsigned int caps); u32 snd_hda_query_pin_caps(struct hda_codec *codec, hda_nid_t nid); +int snd_hda_override_pin_caps(struct hda_codec *codec, hda_nid_t nid, + unsigned int caps); u32 snd_hda_pin_sense(struct hda_codec *codec, hda_nid_t nid); int snd_hda_jack_detect(struct hda_codec *codec, hda_nid_t nid); @@ -589,7 +600,8 @@ int snd_hda_check_amp_list_power(struct hda_codec *codec, #define get_amp_nid_(pv) ((pv) & 0xffff) #define get_amp_nid(kc) get_amp_nid_((kc)->private_value) #define get_amp_channels(kc) (((kc)->private_value >> 16) & 0x3) -#define get_amp_direction(kc) (((kc)->private_value >> 18) & 0x1) +#define get_amp_direction_(pv) (((pv) >> 18) & 0x1) +#define get_amp_direction(kc) get_amp_direction_((kc)->private_value) #define get_amp_index(kc) (((kc)->private_value >> 19) & 0xf) #define get_amp_offset(kc) (((kc)->private_value >> 23) & 0x3f) #define get_amp_min_mute(kc) (((kc)->private_value >> 29) & 0x1) @@ -607,6 +619,7 @@ struct cea_sad { }; #define ELD_FIXED_BYTES 20 +#define ELD_MAX_SIZE 256 #define ELD_MAX_MNL 16 #define ELD_MAX_SAD 16 @@ -631,6 +644,7 @@ struct hdmi_eld { int spk_alloc; int sad_count; struct cea_sad sad[ELD_MAX_SAD]; + char eld_buffer[ELD_MAX_SIZE]; #ifdef CONFIG_PROC_FS struct snd_info_entry *proc_entry; #endif diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index 2be57b051aa2..2c981b55940b 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -152,12 +152,18 @@ static void print_amp_vals(struct snd_info_buffer *buffer, static void print_pcm_rates(struct snd_info_buffer *buffer, unsigned int pcm) { - char buf[SND_PRINT_RATES_ADVISED_BUFSIZE]; + static unsigned int rates[] = { + 8000, 11025, 16000, 22050, 32000, 44100, 48000, 88200, + 96000, 176400, 192000, 384000 + }; + int i; pcm &= AC_SUPPCM_RATES; snd_iprintf(buffer, " rates [0x%x]:", pcm); - snd_print_pcm_rates(pcm, buf, sizeof(buf)); - snd_iprintf(buffer, "%s\n", buf); + for (i = 0; i < ARRAY_SIZE(rates); i++) + if (pcm & (1 << i)) + snd_iprintf(buffer, " %d", rates[i]); + snd_iprintf(buffer, "\n"); } static void print_pcm_bits(struct snd_info_buffer *buffer, unsigned int pcm) diff --git a/sound/pci/hda/hda_trace.h b/sound/pci/hda/hda_trace.h new file mode 100644 index 000000000000..9884871ddb00 --- /dev/null +++ b/sound/pci/hda/hda_trace.h @@ -0,0 +1,117 @@ +#undef TRACE_SYSTEM +#define TRACE_SYSTEM hda +#define TRACE_INCLUDE_FILE hda_trace + +#if !defined(_TRACE_HDA_H) || defined(TRACE_HEADER_MULTI_READ) +#define _TRACE_HDA_H + +#include <linux/tracepoint.h> + +struct hda_bus; +struct hda_codec; + +DECLARE_EVENT_CLASS(hda_cmd, + + TP_PROTO(struct hda_codec *codec, unsigned int val), + + TP_ARGS(codec, val), + + TP_STRUCT__entry( + __field( unsigned int, card ) + __field( unsigned int, addr ) + __field( unsigned int, val ) + ), + + TP_fast_assign( + __entry->card = (codec)->bus->card->number; + __entry->addr = (codec)->addr; + __entry->val = (val); + ), + + TP_printk("[%d:%d] val=%x", __entry->card, __entry->addr, __entry->val) +); + +DEFINE_EVENT(hda_cmd, hda_send_cmd, + TP_PROTO(struct hda_codec *codec, unsigned int val), + TP_ARGS(codec, val) +); + +DEFINE_EVENT(hda_cmd, hda_get_response, + TP_PROTO(struct hda_codec *codec, unsigned int val), + TP_ARGS(codec, val) +); + +TRACE_EVENT(hda_bus_reset, + + TP_PROTO(struct hda_bus *bus), + + TP_ARGS(bus), + + TP_STRUCT__entry( + __field( unsigned int, card ) + ), + + TP_fast_assign( + __entry->card = (bus)->card->number; + ), + + TP_printk("[%d]", __entry->card) +); + +DECLARE_EVENT_CLASS(hda_power, + + TP_PROTO(struct hda_codec *codec), + + TP_ARGS(codec), + + TP_STRUCT__entry( + __field( unsigned int, card ) + __field( unsigned int, addr ) + ), + + TP_fast_assign( + __entry->card = (codec)->bus->card->number; + __entry->addr = (codec)->addr; + ), + + TP_printk("[%d:%d]", __entry->card, __entry->addr) +); + +DEFINE_EVENT(hda_power, hda_power_down, + TP_PROTO(struct hda_codec *codec), + TP_ARGS(codec) +); + +DEFINE_EVENT(hda_power, hda_power_up, + TP_PROTO(struct hda_codec *codec), + TP_ARGS(codec) +); + +TRACE_EVENT(hda_unsol_event, + + TP_PROTO(struct hda_bus *bus, u32 res, u32 res_ex), + + TP_ARGS(bus, res, res_ex), + + TP_STRUCT__entry( + __field( unsigned int, card ) + __field( u32, res ) + __field( u32, res_ex ) + ), + + TP_fast_assign( + __entry->card = (bus)->card->number; + __entry->res = res; + __entry->res_ex = res_ex; + ), + + TP_printk("[%d] res=%x, res_ex=%x", __entry->card, + __entry->res, __entry->res_ex) +); + +#endif /* _TRACE_HDA_H */ + +/* This part must be outside protection */ +#undef TRACE_INCLUDE_PATH +#define TRACE_INCLUDE_PATH . +#include <trace/define_trace.h> diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 8648917acffb..d8aac588f23b 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -48,6 +48,8 @@ struct ad198x_spec { const hda_nid_t *alt_dac_nid; const struct hda_pcm_stream *stream_analog_alt_playback; + int independent_hp; + int num_active_streams; /* capture */ unsigned int num_adc_nids; @@ -302,6 +304,72 @@ static int ad198x_check_power_status(struct hda_codec *codec, hda_nid_t nid) } #endif +static void activate_ctl(struct hda_codec *codec, const char *name, int active) +{ + struct snd_kcontrol *ctl = snd_hda_find_mixer_ctl(codec, name); + if (ctl) { + ctl->vd[0].access &= ~SNDRV_CTL_ELEM_ACCESS_INACTIVE; + ctl->vd[0].access |= active ? 0 : + SNDRV_CTL_ELEM_ACCESS_INACTIVE; + ctl->vd[0].access &= ~SNDRV_CTL_ELEM_ACCESS_WRITE; + ctl->vd[0].access |= active ? + SNDRV_CTL_ELEM_ACCESS_WRITE : 0; + snd_ctl_notify(codec->bus->card, + SNDRV_CTL_EVENT_MASK_INFO, &ctl->id); + } +} + +static void set_stream_active(struct hda_codec *codec, bool active) +{ + struct ad198x_spec *spec = codec->spec; + if (active) + spec->num_active_streams++; + else + spec->num_active_streams--; + activate_ctl(codec, "Independent HP", spec->num_active_streams == 0); +} + +static int ad1988_independent_hp_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char * const texts[] = { "OFF", "ON", NULL}; + int index; + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = 2; + index = uinfo->value.enumerated.item; + if (index >= 2) + index = 1; + strcpy(uinfo->value.enumerated.name, texts[index]); + return 0; +} + +static int ad1988_independent_hp_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct ad198x_spec *spec = codec->spec; + ucontrol->value.enumerated.item[0] = spec->independent_hp; + return 0; +} + +static int ad1988_independent_hp_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct ad198x_spec *spec = codec->spec; + unsigned int select = ucontrol->value.enumerated.item[0]; + if (spec->independent_hp != select) { + spec->independent_hp = select; + if (spec->independent_hp) + spec->multiout.hp_nid = 0; + else + spec->multiout.hp_nid = spec->alt_dac_nid[0]; + return 1; + } + return 0; +} + /* * Analog playback callbacks */ @@ -310,8 +378,15 @@ static int ad198x_playback_pcm_open(struct hda_pcm_stream *hinfo, struct snd_pcm_substream *substream) { struct ad198x_spec *spec = codec->spec; - return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream, + int err; + set_stream_active(codec, true); + err = snd_hda_multi_out_analog_open(codec, &spec->multiout, substream, hinfo); + if (err < 0) { + set_stream_active(codec, false); + return err; + } + return 0; } static int ad198x_playback_pcm_prepare(struct hda_pcm_stream *hinfo, @@ -333,11 +408,41 @@ static int ad198x_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, return snd_hda_multi_out_analog_cleanup(codec, &spec->multiout); } +static int ad198x_playback_pcm_close(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + set_stream_active(codec, false); + return 0; +} + +static int ad1988_alt_playback_pcm_open(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + struct ad198x_spec *spec = codec->spec; + if (!spec->independent_hp) + return -EBUSY; + set_stream_active(codec, true); + return 0; +} + +static int ad1988_alt_playback_pcm_close(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + set_stream_active(codec, false); + return 0; +} + static const struct hda_pcm_stream ad198x_pcm_analog_alt_playback = { .substreams = 1, .channels_min = 2, .channels_max = 2, - /* NID is set in ad198x_build_pcms */ + .ops = { + .open = ad1988_alt_playback_pcm_open, + .close = ad1988_alt_playback_pcm_close + }, }; /* @@ -402,7 +507,6 @@ static int ad198x_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, return 0; } - /* */ static const struct hda_pcm_stream ad198x_pcm_analog_playback = { @@ -413,7 +517,8 @@ static const struct hda_pcm_stream ad198x_pcm_analog_playback = { .ops = { .open = ad198x_playback_pcm_open, .prepare = ad198x_playback_pcm_prepare, - .cleanup = ad198x_playback_pcm_cleanup + .cleanup = ad198x_playback_pcm_cleanup, + .close = ad198x_playback_pcm_close }, }; @@ -2058,7 +2163,6 @@ static int patch_ad1981(struct hda_codec *codec) enum { AD1988_6STACK, AD1988_6STACK_DIG, - AD1988_6STACK_DIG_FP, AD1988_3STACK, AD1988_3STACK_DIG, AD1988_LAPTOP, @@ -2168,6 +2272,17 @@ static int ad198x_ch_mode_put(struct snd_kcontrol *kcontrol, return err; } +static const struct snd_kcontrol_new ad1988_hp_mixers[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Independent HP", + .info = ad1988_independent_hp_info, + .get = ad1988_independent_hp_get, + .put = ad1988_independent_hp_put, + }, + { } /* end */ +}; + /* 6-stack mode */ static const struct snd_kcontrol_new ad1988_6stack_mixers1[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x04, 0x0, HDA_OUTPUT), @@ -2188,6 +2303,7 @@ static const struct snd_kcontrol_new ad1988_6stack_mixers1_rev2[] = { }; static const struct snd_kcontrol_new ad1988_6stack_mixers2[] = { + HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x29, 2, HDA_INPUT), HDA_BIND_MUTE("Surround Playback Switch", 0x2a, 2, HDA_INPUT), HDA_BIND_MUTE_MONO("Center Playback Switch", 0x27, 1, 2, HDA_INPUT), @@ -2210,13 +2326,6 @@ static const struct snd_kcontrol_new ad1988_6stack_mixers2[] = { HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x39, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Mic Boost Volume", 0x3c, 0x0, HDA_OUTPUT), - - { } /* end */ -}; - -static const struct snd_kcontrol_new ad1988_6stack_fp_mixers[] = { - HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT), - { } /* end */ }; @@ -2238,6 +2347,7 @@ static const struct snd_kcontrol_new ad1988_3stack_mixers1_rev2[] = { }; static const struct snd_kcontrol_new ad1988_3stack_mixers2[] = { + HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x29, 2, HDA_INPUT), HDA_BIND_MUTE("Surround Playback Switch", 0x2c, 2, HDA_INPUT), HDA_BIND_MUTE_MONO("Center Playback Switch", 0x26, 1, 2, HDA_INPUT), @@ -2272,6 +2382,7 @@ static const struct snd_kcontrol_new ad1988_3stack_mixers2[] = { /* laptop mode */ static const struct snd_kcontrol_new ad1988_laptop_mixers[] = { + HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("PCM Playback Volume", 0x04, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("PCM Playback Switch", 0x29, 0x0, HDA_INPUT), HDA_BIND_MUTE("Mono Playback Switch", 0x1e, 2, HDA_INPUT), @@ -2446,7 +2557,7 @@ static const struct hda_verb ad1988_6stack_init_verbs[] = { {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Port-A front headphon path */ - {0x37, AC_VERB_SET_CONNECT_SEL, 0x01}, /* DAC1:04h */ + {0x37, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC0:03h */ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, @@ -2594,7 +2705,7 @@ static const struct hda_verb ad1988_3stack_init_verbs[] = { {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Port-A front headphon path */ - {0x37, AC_VERB_SET_CONNECT_SEL, 0x01}, /* DAC1:04h */ + {0x37, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC0:03h */ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, @@ -2669,7 +2780,7 @@ static const struct hda_verb ad1988_laptop_init_verbs[] = { {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Port-A front headphon path */ - {0x37, AC_VERB_SET_CONNECT_SEL, 0x01}, /* DAC1:04h */ + {0x37, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC0:03h */ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, @@ -2782,11 +2893,11 @@ static inline hda_nid_t ad1988_idx_to_dac(struct hda_codec *codec, int idx) { static const hda_nid_t idx_to_dac[8] = { /* A B C D E F G H */ - 0x04, 0x06, 0x05, 0x04, 0x0a, 0x06, 0x05, 0x0a + 0x03, 0x06, 0x05, 0x04, 0x0a, 0x06, 0x05, 0x0a }; static const hda_nid_t idx_to_dac_rev2[8] = { /* A B C D E F G H */ - 0x04, 0x05, 0x0a, 0x04, 0x06, 0x05, 0x0a, 0x06 + 0x03, 0x05, 0x0a, 0x04, 0x06, 0x05, 0x0a, 0x06 }; if (is_rev2(codec)) return idx_to_dac_rev2[idx]; @@ -3023,8 +3134,8 @@ static void ad1988_auto_set_output_and_unmute(struct hda_codec *codec, snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, pin_type); snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE); switch (nid) { - case 0x11: /* port-A - DAC 04 */ - snd_hda_codec_write(codec, 0x37, 0, AC_VERB_SET_CONNECT_SEL, 0x01); + case 0x11: /* port-A - DAC 03 */ + snd_hda_codec_write(codec, 0x37, 0, AC_VERB_SET_CONNECT_SEL, 0x00); break; case 0x14: /* port-B - DAC 06 */ snd_hda_codec_write(codec, 0x30, 0, AC_VERB_SET_CONNECT_SEL, 0x02); @@ -3150,7 +3261,6 @@ static int ad1988_auto_init(struct hda_codec *codec) static const char * const ad1988_models[AD1988_MODEL_LAST] = { [AD1988_6STACK] = "6stack", [AD1988_6STACK_DIG] = "6stack-dig", - [AD1988_6STACK_DIG_FP] = "6stack-dig-fp", [AD1988_3STACK] = "3stack", [AD1988_3STACK_DIG] = "3stack-dig", [AD1988_LAPTOP] = "laptop", @@ -3208,10 +3318,11 @@ static int patch_ad1988(struct hda_codec *codec) } set_beep_amp(spec, 0x10, 0, HDA_OUTPUT); + if (!spec->multiout.hp_nid) + spec->multiout.hp_nid = ad1988_alt_dac_nid[0]; switch (board_config) { case AD1988_6STACK: case AD1988_6STACK_DIG: - case AD1988_6STACK_DIG_FP: spec->multiout.max_channels = 8; spec->multiout.num_dacs = 4; if (is_rev2(codec)) @@ -3227,19 +3338,7 @@ static int patch_ad1988(struct hda_codec *codec) spec->mixers[1] = ad1988_6stack_mixers2; spec->num_init_verbs = 1; spec->init_verbs[0] = ad1988_6stack_init_verbs; - if (board_config == AD1988_6STACK_DIG_FP) { - spec->num_mixers++; - spec->mixers[2] = ad1988_6stack_fp_mixers; - spec->num_init_verbs++; - spec->init_verbs[1] = ad1988_6stack_fp_init_verbs; - spec->slave_vols = ad1988_6stack_fp_slave_vols; - spec->slave_sws = ad1988_6stack_fp_slave_sws; - spec->alt_dac_nid = ad1988_alt_dac_nid; - spec->stream_analog_alt_playback = - &ad198x_pcm_analog_alt_playback; - } - if ((board_config == AD1988_6STACK_DIG) || - (board_config == AD1988_6STACK_DIG_FP)) { + if (board_config == AD1988_6STACK_DIG) { spec->multiout.dig_out_nid = AD1988_SPDIF_OUT; spec->dig_in_nid = AD1988_SPDIF_IN; } @@ -3282,6 +3381,15 @@ static int patch_ad1988(struct hda_codec *codec) break; } + if (spec->autocfg.hp_pins[0]) { + spec->mixers[spec->num_mixers++] = ad1988_hp_mixers; + spec->slave_vols = ad1988_6stack_fp_slave_vols; + spec->slave_sws = ad1988_6stack_fp_slave_sws; + spec->alt_dac_nid = ad1988_alt_dac_nid; + spec->stream_analog_alt_playback = + &ad198x_pcm_analog_alt_playback; + } + spec->num_adc_nids = ARRAY_SIZE(ad1988_adc_nids); spec->adc_nids = ad1988_adc_nids; spec->capsrc_nids = ad1988_capsrc_nids; diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 76752d8ea733..0c8b5a1993ed 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -136,6 +136,8 @@ struct conexant_spec { unsigned int thinkpad:1; unsigned int hp_laptop:1; unsigned int asus:1; + unsigned int pin_eapd_ctrls:1; + unsigned int single_adc_amp:1; unsigned int adc_switching:1; @@ -1867,39 +1869,6 @@ static const struct hda_verb cxt5051_hp_dv6736_init_verbs[] = { { } /* end */ }; -static const struct hda_verb cxt5051_lenovo_x200_init_verbs[] = { - /* Line in, Mic */ - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03}, - {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03}, - {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03}, - /* SPK */ - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x1a, AC_VERB_SET_CONNECT_SEL, 0x00}, - /* HP, Amp */ - {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x16, AC_VERB_SET_CONNECT_SEL, 0x00}, - /* Docking HP */ - {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x19, AC_VERB_SET_CONNECT_SEL, 0x00}, - /* DAC1 */ - {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* Record selector: Internal mic */ - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x44}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1) | 0x44}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x44}, - /* SPDIF route: PCM */ - {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* needed for W500 Advanced Mini Dock 250410 */ - {0x1c, AC_VERB_SET_CONNECT_SEL, 0x0}, - /* EAPD */ - {0x1a, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */ - {0x16, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CONEXANT_HP_EVENT}, - {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CONEXANT_HP_EVENT}, - { } /* end */ -}; - static const struct hda_verb cxt5051_f700_init_verbs[] = { /* Line in, Mic */ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03}, @@ -1968,7 +1937,6 @@ enum { CXT5051_LAPTOP, /* Laptops w/ EAPD support */ CXT5051_HP, /* no docking */ CXT5051_HP_DV6736, /* HP without mic switch */ - CXT5051_LENOVO_X200, /* Lenovo X200 laptop, also used for Advanced Mini Dock 250410 */ CXT5051_F700, /* HP Compaq Presario F700 */ CXT5051_TOSHIBA, /* Toshiba M300 & co */ CXT5051_IDEAPAD, /* Lenovo IdeaPad Y430 */ @@ -1980,7 +1948,6 @@ static const char *const cxt5051_models[CXT5051_MODELS] = { [CXT5051_LAPTOP] = "laptop", [CXT5051_HP] = "hp", [CXT5051_HP_DV6736] = "hp-dv6736", - [CXT5051_LENOVO_X200] = "lenovo-x200", [CXT5051_F700] = "hp-700", [CXT5051_TOSHIBA] = "toshiba", [CXT5051_IDEAPAD] = "ideapad", @@ -1995,7 +1962,6 @@ static const struct snd_pci_quirk cxt5051_cfg_tbl[] = { SND_PCI_QUIRK(0x14f1, 0x0101, "Conexant Reference board", CXT5051_LAPTOP), SND_PCI_QUIRK(0x14f1, 0x5051, "HP Spartan 1.1", CXT5051_HP), - SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo X200", CXT5051_LENOVO_X200), SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo IdeaPad", CXT5051_IDEAPAD), {} }; @@ -2053,13 +2019,6 @@ static int patch_cxt5051(struct hda_codec *codec) spec->mixers[0] = cxt5051_hp_dv6736_mixers; spec->auto_mic = 0; break; - case CXT5051_LENOVO_X200: - spec->init_verbs[0] = cxt5051_lenovo_x200_init_verbs; - /* Thinkpad X301 does not have S/PDIF wired and no ability - to use a docking station. */ - if (codec->subsystem_id == 0x17aa211f) - spec->multiout.dig_out_nid = 0; - break; case CXT5051_F700: spec->init_verbs[0] = cxt5051_f700_init_verbs; spec->mixers[0] = cxt5051_f700_mixers; @@ -3473,12 +3432,14 @@ static void cx_auto_turn_eapd(struct hda_codec *codec, int num_pins, static void do_automute(struct hda_codec *codec, int num_pins, hda_nid_t *pins, bool on) { + struct conexant_spec *spec = codec->spec; int i; for (i = 0; i < num_pins; i++) snd_hda_codec_write(codec, pins[i], 0, AC_VERB_SET_PIN_WIDGET_CONTROL, on ? PIN_OUT : 0); - cx_auto_turn_eapd(codec, num_pins, pins, on); + if (spec->pin_eapd_ctrls) + cx_auto_turn_eapd(codec, num_pins, pins, on); } static int detect_jacks(struct hda_codec *codec, int num_pins, hda_nid_t *pins) @@ -3503,9 +3464,12 @@ static void cx_auto_update_speakers(struct hda_codec *codec) int on = 1; /* turn on HP EAPD when HP jacks are present */ - if (spec->auto_mute) - on = spec->hp_present; - cx_auto_turn_eapd(codec, cfg->hp_outs, cfg->hp_pins, on); + if (spec->pin_eapd_ctrls) { + if (spec->auto_mute) + on = spec->hp_present; + cx_auto_turn_eapd(codec, cfg->hp_outs, cfg->hp_pins, on); + } + /* mute speakers in auto-mode if HP or LO jacks are plugged */ if (spec->auto_mute) on = !(spec->hp_present || @@ -3932,20 +3896,10 @@ static void cx_auto_parse_beep(struct hda_codec *codec) #define cx_auto_parse_beep(codec) #endif -static bool found_in_nid_list(hda_nid_t nid, const hda_nid_t *list, int nums) -{ - int i; - for (i = 0; i < nums; i++) - if (list[i] == nid) - return true; - return false; -} - -/* parse extra-EAPD that aren't assigned to any pins */ +/* parse EAPDs */ static void cx_auto_parse_eapd(struct hda_codec *codec) { struct conexant_spec *spec = codec->spec; - struct auto_pin_cfg *cfg = &spec->autocfg; hda_nid_t nid, end_nid; end_nid = codec->start_nid + codec->num_nodes; @@ -3954,14 +3908,18 @@ static void cx_auto_parse_eapd(struct hda_codec *codec) continue; if (!(snd_hda_query_pin_caps(codec, nid) & AC_PINCAP_EAPD)) continue; - if (found_in_nid_list(nid, cfg->line_out_pins, cfg->line_outs) || - found_in_nid_list(nid, cfg->hp_pins, cfg->hp_outs) || - found_in_nid_list(nid, cfg->speaker_pins, cfg->speaker_outs)) - continue; spec->eapds[spec->num_eapds++] = nid; if (spec->num_eapds >= ARRAY_SIZE(spec->eapds)) break; } + + /* NOTE: below is a wild guess; if we have more than two EAPDs, + * it's a new chip, where EAPDs are supposed to be associated to + * pins, and we can control EAPD per pin. + * OTOH, if only one or two EAPDs are found, it's an old chip, + * thus it might control over all pins. + */ + spec->pin_eapd_ctrls = spec->num_eapds > 2; } static int cx_auto_parse_auto_config(struct hda_codec *codec) @@ -4067,8 +4025,9 @@ static void cx_auto_init_output(struct hda_codec *codec) } } cx_auto_update_speakers(codec); - /* turn on/off extra EAPDs, too */ - cx_auto_turn_eapd(codec, spec->num_eapds, spec->eapds, true); + /* turn on all EAPDs if no individual EAPD control is available */ + if (!spec->pin_eapd_ctrls) + cx_auto_turn_eapd(codec, spec->num_eapds, spec->eapds, true); } static void cx_auto_init_input(struct hda_codec *codec) @@ -4255,6 +4214,8 @@ static int cx_auto_add_capture_volume(struct hda_codec *codec, hda_nid_t nid, int idx = get_input_connection(codec, adc_nid, nid); if (idx < 0) continue; + if (spec->single_adc_amp) + idx = 0; return cx_auto_add_volume_idx(codec, label, pfx, cidx, adc_nid, HDA_INPUT, idx); } @@ -4295,14 +4256,21 @@ static int cx_auto_build_input_controls(struct hda_codec *codec) struct hda_input_mux *imux = &spec->private_imux; const char *prev_label; int input_conn[HDA_MAX_NUM_INPUTS]; - int i, err, cidx; + int i, j, err, cidx; int multi_connection; + if (!imux->num_items) + return 0; + multi_connection = 0; for (i = 0; i < imux->num_items; i++) { cidx = get_input_connection(codec, spec->imux_info[i].adc, spec->imux_info[i].pin); - input_conn[i] = (spec->imux_info[i].adc << 8) | cidx; + if (cidx < 0) + continue; + input_conn[i] = spec->imux_info[i].adc; + if (!spec->single_adc_amp) + input_conn[i] |= cidx << 8; if (i > 0 && input_conn[i] != input_conn[0]) multi_connection = 1; } @@ -4331,6 +4299,15 @@ static int cx_auto_build_input_controls(struct hda_codec *codec) err = cx_auto_add_capture_volume(codec, nid, "Capture", "", cidx); } else { + bool dup_found = false; + for (j = 0; j < i; j++) { + if (input_conn[j] == input_conn[i]) { + dup_found = true; + break; + } + } + if (dup_found) + continue; err = cx_auto_add_capture_volume(codec, nid, label, " Capture", cidx); } @@ -4394,6 +4371,53 @@ static const struct hda_codec_ops cx_auto_patch_ops = { .reboot_notify = snd_hda_shutup_pins, }; +/* + * pin fix-up + */ +struct cxt_pincfg { + hda_nid_t nid; + u32 val; +}; + +static void apply_pincfg(struct hda_codec *codec, const struct cxt_pincfg *cfg) +{ + for (; cfg->nid; cfg++) + snd_hda_codec_set_pincfg(codec, cfg->nid, cfg->val); + +} + +static void apply_pin_fixup(struct hda_codec *codec, + const struct snd_pci_quirk *quirk, + const struct cxt_pincfg **table) +{ + quirk = snd_pci_quirk_lookup(codec->bus->pci, quirk); + if (quirk) { + snd_printdd(KERN_INFO "hda_codec: applying pincfg for %s\n", + quirk->name); + apply_pincfg(codec, table[quirk->value]); + } +} + +enum { + CXT_PINCFG_LENOVO_X200, +}; + +static const struct cxt_pincfg cxt_pincfg_lenovo_x200[] = { + { 0x16, 0x042140ff }, /* HP (seq# overridden) */ + { 0x17, 0x21a11000 }, /* dock-mic */ + { 0x19, 0x2121103f }, /* dock-HP */ + {} +}; + +static const struct cxt_pincfg *cxt_pincfg_tbl[] = { + [CXT_PINCFG_LENOVO_X200] = cxt_pincfg_lenovo_x200, +}; + +static const struct snd_pci_quirk cxt_fixups[] = { + SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo X200", CXT_PINCFG_LENOVO_X200), + {} +}; + static int patch_conexant_auto(struct hda_codec *codec) { struct conexant_spec *spec; @@ -4407,6 +4431,15 @@ static int patch_conexant_auto(struct hda_codec *codec) return -ENOMEM; codec->spec = spec; codec->pin_amp_workaround = 1; + + switch (codec->vendor_id) { + case 0x14f15045: + spec->single_adc_amp = 1; + break; + } + + apply_pin_fixup(codec, cxt_fixups, cxt_pincfg_tbl); + err = cx_auto_search_adcs(codec); if (err < 0) return err; diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 19cb72db9c38..342540128fb8 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -324,6 +324,66 @@ static int cvt_nid_to_cvt_index(struct hdmi_spec *spec, hda_nid_t cvt_nid) return -EINVAL; } +static int hdmi_eld_ctl_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct hdmi_spec *spec; + int pin_idx; + + spec = codec->spec; + uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES; + + pin_idx = kcontrol->private_value; + uinfo->count = spec->pins[pin_idx].sink_eld.eld_size; + + return 0; +} + +static int hdmi_eld_ctl_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct hdmi_spec *spec; + int pin_idx; + + spec = codec->spec; + pin_idx = kcontrol->private_value; + + memcpy(ucontrol->value.bytes.data, + spec->pins[pin_idx].sink_eld.eld_buffer, ELD_MAX_SIZE); + + return 0; +} + +static struct snd_kcontrol_new eld_bytes_ctl = { + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = "ELD", + .info = hdmi_eld_ctl_info, + .get = hdmi_eld_ctl_get, +}; + +static int hdmi_create_eld_ctl(struct hda_codec *codec, int pin_idx, + int device) +{ + struct snd_kcontrol *kctl; + struct hdmi_spec *spec = codec->spec; + int err; + + kctl = snd_ctl_new1(&eld_bytes_ctl, codec); + if (!kctl) + return -ENOMEM; + kctl->private_value = pin_idx; + kctl->id.device = device; + + err = snd_hda_ctl_add(codec, spec->pins[pin_idx].pin_nid, kctl); + if (err < 0) + return err; + + return 0; +} + #ifdef BE_PARANOID static void hdmi_get_dip_index(struct hda_codec *codec, hda_nid_t pin_nid, int *packet_index, int *byte_index) @@ -967,19 +1027,12 @@ static int hdmi_add_pin(struct hda_codec *codec, hda_nid_t pin_nid) per_pin->pin_nid = pin_nid; - err = snd_hda_input_jack_add(codec, pin_nid, - SND_JACK_VIDEOOUT, NULL); - if (err < 0) - return err; - err = hdmi_read_pin_conn(codec, pin_idx); if (err < 0) return err; spec->num_pins++; - hdmi_present_sense(codec, pin_nid, eld); - return 0; } @@ -1162,6 +1215,25 @@ static int generic_hdmi_build_pcms(struct hda_codec *codec) return 0; } +static int generic_hdmi_build_jack(struct hda_codec *codec, int pin_idx) +{ + int err; + char hdmi_str[32]; + struct hdmi_spec *spec = codec->spec; + struct hdmi_spec_per_pin *per_pin = &spec->pins[pin_idx]; + int pcmdev = spec->pcm_rec[pin_idx].device; + + snprintf(hdmi_str, sizeof(hdmi_str), "HDMI/DP,pcm=%d", pcmdev); + + err = snd_hda_input_jack_add(codec, per_pin->pin_nid, + SND_JACK_VIDEOOUT, pcmdev > 0 ? hdmi_str : NULL); + if (err < 0) + return err; + + hdmi_present_sense(codec, per_pin->pin_nid, &per_pin->sink_eld); + return 0; +} + static int generic_hdmi_build_controls(struct hda_codec *codec) { struct hdmi_spec *spec = codec->spec; @@ -1170,12 +1242,25 @@ static int generic_hdmi_build_controls(struct hda_codec *codec) for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) { struct hdmi_spec_per_pin *per_pin = &spec->pins[pin_idx]; + + err = generic_hdmi_build_jack(codec, pin_idx); + if (err < 0) + return err; + err = snd_hda_create_spdif_out_ctls(codec, per_pin->pin_nid, per_pin->mux_nids[0]); if (err < 0) return err; snd_hda_spdif_ctls_unassign(codec, pin_idx); + + /* add control for ELD Bytes */ + err = hdmi_create_eld_ctl(codec, + pin_idx, + spec->pcm_rec[pin_idx].device); + + if (err < 0) + return err; } return 0; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 7a73621a8909..8f93b97559a5 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -116,6 +116,8 @@ struct alc_spec { const hda_nid_t *capsrc_nids; hda_nid_t dig_in_nid; /* digital-in NID; optional */ hda_nid_t mixer_nid; /* analog-mixer NID */ + DECLARE_BITMAP(vol_ctls, 0x20 << 1); + DECLARE_BITMAP(sw_ctls, 0x20 << 1); /* capture setup for dynamic dual-adc switch */ hda_nid_t cur_adc; @@ -159,23 +161,27 @@ struct alc_spec { void (*power_hook)(struct hda_codec *codec); #endif void (*shutup)(struct hda_codec *codec); + void (*automute_hook)(struct hda_codec *codec); /* for pin sensing */ - unsigned int jack_present: 1; + unsigned int hp_jack_present:1; unsigned int line_jack_present:1; unsigned int master_mute:1; unsigned int auto_mic:1; unsigned int auto_mic_valid_imux:1; /* valid imux for auto-mic */ - unsigned int automute:1; /* HP automute enabled */ - unsigned int detect_line:1; /* Line-out detection enabled */ - unsigned int automute_lines:1; /* automute line-out as well; NOP when automute_hp_lo isn't set */ - unsigned int automute_hp_lo:1; /* both HP and LO available */ + unsigned int automute_speaker:1; /* automute speaker outputs */ + unsigned int automute_lo:1; /* automute LO outputs */ + unsigned int detect_hp:1; /* Headphone detection enabled */ + unsigned int detect_lo:1; /* Line-out detection enabled */ + unsigned int automute_speaker_possible:1; /* there are speakers and either LO or HP */ + unsigned int automute_lo_possible:1; /* there are line outs and HP */ /* other flags */ unsigned int no_analog :1; /* digital I/O only */ unsigned int dyn_adc_switch:1; /* switch ADCs (for ALC275) */ unsigned int single_input_src:1; unsigned int vol_in_capsrc:1; /* use capsrc volume (ADC has no vol) */ + unsigned int parse_flags; /* passed to snd_hda_parse_pin_defcfg() */ /* auto-mute control */ int automute_mode; @@ -193,6 +199,7 @@ struct alc_spec { /* for PLL fix */ hda_nid_t pll_nid; unsigned int pll_coef_idx, pll_coef_bit; + unsigned int coef0; /* fix-up list */ int fixup_id; @@ -202,6 +209,9 @@ struct alc_spec { /* multi-io */ int multi_ios; struct alc_multi_io multi_io[4]; + + /* bind volumes */ + struct snd_array bind_ctls; }; #define ALC_MODEL_AUTO 0 /* common for all chips */ @@ -525,8 +535,8 @@ static void do_automute(struct hda_codec *codec, int num_pins, hda_nid_t *pins, } } -/* Toggle internal speakers muting */ -static void update_speakers(struct hda_codec *codec) +/* Toggle outputs muting */ +static void update_outputs(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; int on; @@ -538,10 +548,10 @@ static void update_speakers(struct hda_codec *codec) do_automute(codec, ARRAY_SIZE(spec->autocfg.hp_pins), spec->autocfg.hp_pins, spec->master_mute, true); - if (!spec->automute) + if (!spec->automute_speaker) on = 0; else - on = spec->jack_present | spec->line_jack_present; + on = spec->hp_jack_present | spec->line_jack_present; on |= spec->master_mute; do_automute(codec, ARRAY_SIZE(spec->autocfg.speaker_pins), spec->autocfg.speaker_pins, on, false); @@ -551,26 +561,35 @@ static void update_speakers(struct hda_codec *codec) if (spec->autocfg.line_out_pins[0] == spec->autocfg.hp_pins[0] || spec->autocfg.line_out_pins[0] == spec->autocfg.speaker_pins[0]) return; - if (!spec->automute || (spec->automute_hp_lo && !spec->automute_lines)) + if (!spec->automute_lo) on = 0; else - on = spec->jack_present; + on = spec->hp_jack_present; on |= spec->master_mute; do_automute(codec, ARRAY_SIZE(spec->autocfg.line_out_pins), spec->autocfg.line_out_pins, on, false); } +static void call_update_outputs(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + if (spec->automute_hook) + spec->automute_hook(codec); + else + update_outputs(codec); +} + /* standard HP-automute helper */ static void alc_hp_automute(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - spec->jack_present = + spec->hp_jack_present = detect_jacks(codec, ARRAY_SIZE(spec->autocfg.hp_pins), spec->autocfg.hp_pins); - if (!spec->automute) + if (!spec->detect_hp || (!spec->automute_speaker && !spec->automute_lo)) return; - update_speakers(codec); + call_update_outputs(codec); } /* standard line-out-automute helper */ @@ -585,9 +604,9 @@ static void alc_line_automute(struct hda_codec *codec) spec->line_jack_present = detect_jacks(codec, ARRAY_SIZE(spec->autocfg.line_out_pins), spec->autocfg.line_out_pins); - if (!spec->automute || !spec->detect_line) + if (!spec->automute_speaker || !spec->detect_lo) return; - update_speakers(codec); + call_update_outputs(codec); } #define get_connection_index(codec, mux, nid) \ @@ -785,7 +804,7 @@ static int alc_automute_mode_info(struct snd_kcontrol *kcontrol, uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; - if (spec->automute_hp_lo) { + if (spec->automute_speaker_possible && spec->automute_lo_possible) { uinfo->value.enumerated.items = 3; texts = texts3; } else { @@ -804,13 +823,12 @@ static int alc_automute_mode_get(struct snd_kcontrol *kcontrol, { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct alc_spec *spec = codec->spec; - unsigned int val; - if (!spec->automute) - val = 0; - else if (!spec->automute_hp_lo || !spec->automute_lines) - val = 1; - else - val = 2; + unsigned int val = 0; + if (spec->automute_speaker) + val++; + if (spec->automute_lo) + val++; + ucontrol->value.enumerated.item[0] = val; return 0; } @@ -823,29 +841,36 @@ static int alc_automute_mode_put(struct snd_kcontrol *kcontrol, switch (ucontrol->value.enumerated.item[0]) { case 0: - if (!spec->automute) + if (!spec->automute_speaker && !spec->automute_lo) return 0; - spec->automute = 0; + spec->automute_speaker = 0; + spec->automute_lo = 0; break; case 1: - if (spec->automute && - (!spec->automute_hp_lo || !spec->automute_lines)) - return 0; - spec->automute = 1; - spec->automute_lines = 0; + if (spec->automute_speaker_possible) { + if (!spec->automute_lo && spec->automute_speaker) + return 0; + spec->automute_speaker = 1; + spec->automute_lo = 0; + } else if (spec->automute_lo_possible) { + if (spec->automute_lo) + return 0; + spec->automute_lo = 1; + } else + return -EINVAL; break; case 2: - if (!spec->automute_hp_lo) + if (!spec->automute_lo_possible || !spec->automute_speaker_possible) return -EINVAL; - if (spec->automute && spec->automute_lines) + if (spec->automute_speaker && spec->automute_lo) return 0; - spec->automute = 1; - spec->automute_lines = 1; + spec->automute_speaker = 1; + spec->automute_lo = 1; break; default: return -EINVAL; } - update_speakers(codec); + call_update_outputs(codec); return 1; } @@ -882,7 +907,7 @@ static int alc_add_automute_mode_enum(struct hda_codec *codec) * Check the availability of HP/line-out auto-mute; * Set up appropriately if really supported */ -static void alc_init_auto_hp(struct hda_codec *codec) +static void alc_init_automute(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; struct auto_pin_cfg *cfg = &spec->autocfg; @@ -897,8 +922,6 @@ static void alc_init_auto_hp(struct hda_codec *codec) present++; if (present < 2) /* need two different output types */ return; - if (present == 3) - spec->automute_hp_lo = 1; /* both HP and LO automute */ if (!cfg->speaker_pins[0] && cfg->line_out_type == AUTO_PIN_SPEAKER_OUT) { @@ -914,6 +937,8 @@ static void alc_init_auto_hp(struct hda_codec *codec) cfg->hp_outs = cfg->line_outs; } + spec->automute_mode = ALC_AUTOMUTE_PIN; + for (i = 0; i < cfg->hp_outs; i++) { hda_nid_t nid = cfg->hp_pins[i]; if (!is_jack_detectable(codec, nid)) @@ -923,28 +948,32 @@ static void alc_init_auto_hp(struct hda_codec *codec) snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT); - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_PIN; - } - if (spec->automute && cfg->line_out_pins[0] && - cfg->speaker_pins[0] && - cfg->line_out_pins[0] != cfg->hp_pins[0] && - cfg->line_out_pins[0] != cfg->speaker_pins[0]) { - for (i = 0; i < cfg->line_outs; i++) { - hda_nid_t nid = cfg->line_out_pins[i]; - if (!is_jack_detectable(codec, nid)) - continue; - snd_printdd("realtek: Enable Line-Out auto-muting " - "on NID 0x%x\n", nid); - snd_hda_codec_write_cache(codec, nid, 0, - AC_VERB_SET_UNSOLICITED_ENABLE, - AC_USRSP_EN | ALC_FRONT_EVENT); - spec->detect_line = 1; + spec->detect_hp = 1; + } + + if (cfg->line_out_type == AUTO_PIN_LINE_OUT && cfg->line_outs) { + if (cfg->speaker_outs) + for (i = 0; i < cfg->line_outs; i++) { + hda_nid_t nid = cfg->line_out_pins[i]; + if (!is_jack_detectable(codec, nid)) + continue; + snd_printdd("realtek: Enable Line-Out " + "auto-muting on NID 0x%x\n", nid); + snd_hda_codec_write_cache(codec, nid, 0, + AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | ALC_FRONT_EVENT); + spec->detect_lo = 1; } - spec->automute_lines = spec->detect_line; + spec->automute_lo_possible = spec->detect_hp; } - if (spec->automute) { + spec->automute_speaker_possible = cfg->speaker_outs && + (spec->detect_hp || spec->detect_lo); + + spec->automute_lo = spec->automute_lo_possible; + spec->automute_speaker = spec->automute_speaker_possible; + + if (spec->automute_speaker_possible || spec->automute_lo_possible) { /* create a control for automute mode */ alc_add_automute_mode_enum(codec); spec->unsol_event = alc_sku_unsol_event; @@ -1145,7 +1174,7 @@ static void alc_init_auto_mic(struct hda_codec *codec) /* check the availabilities of auto-mute and auto-mic switches */ static void alc_auto_check_switches(struct hda_codec *codec) { - alc_init_auto_hp(codec); + alc_init_automute(codec); alc_init_auto_mic(codec); } @@ -1528,6 +1557,15 @@ static void alc_write_coef_idx(struct hda_codec *codec, unsigned int coef_idx, coef_val); } +/* a special bypass for COEF 0; read the cached value at the second time */ +static unsigned int alc_get_coef0(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + if (!spec->coef0) + spec->coef0 = alc_read_coef_idx(codec, 0); + return spec->coef0; +} + /* * Digital I/O handling */ @@ -2368,6 +2406,18 @@ static void alc_free_kctls(struct hda_codec *codec) snd_array_free(&spec->kctls); } +static void alc_free_bind_ctls(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + if (spec->bind_ctls.list) { + struct hda_bind_ctls **ctl = spec->bind_ctls.list; + int i; + for (i = 0; i < spec->bind_ctls.used; i++) + kfree(ctl[i]); + } + snd_array_free(&spec->bind_ctls); +} + static void alc_free(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -2378,6 +2428,7 @@ static void alc_free(struct hda_codec *codec) alc_shutup(codec); snd_hda_input_jack_free(codec); alc_free_kctls(codec); + alc_free_bind_ctls(codec); kfree(spec); snd_hda_detach_beep_device(codec); } @@ -2441,6 +2492,47 @@ static int alc_codec_rename(struct hda_codec *codec, const char *name) } /* + * Rename codecs appropriately from COEF value + */ +struct alc_codec_rename_table { + unsigned int vendor_id; + unsigned short coef_mask; + unsigned short coef_bits; + const char *name; +}; + +static struct alc_codec_rename_table rename_tbl[] = { + { 0x10ec0269, 0xfff0, 0x3010, "ALC277" }, + { 0x10ec0269, 0xf0f0, 0x2010, "ALC259" }, + { 0x10ec0269, 0xf0f0, 0x3010, "ALC258" }, + { 0x10ec0269, 0x00f0, 0x0010, "ALC269VB" }, + { 0x10ec0269, 0xffff, 0xa023, "ALC259" }, + { 0x10ec0269, 0xffff, 0x6023, "ALC281X" }, + { 0x10ec0269, 0x00f0, 0x0020, "ALC269VC" }, + { 0x10ec0887, 0x00f0, 0x0030, "ALC887-VD" }, + { 0x10ec0888, 0x00f0, 0x0030, "ALC888-VD" }, + { 0x10ec0888, 0xf0f0, 0x3020, "ALC886" }, + { 0x10ec0899, 0x2000, 0x2000, "ALC899" }, + { 0x10ec0892, 0xffff, 0x8020, "ALC661" }, + { 0x10ec0892, 0xffff, 0x8011, "ALC661" }, + { 0x10ec0892, 0xffff, 0x4011, "ALC656" }, + { } /* terminator */ +}; + +static int alc_codec_rename_from_preset(struct hda_codec *codec) +{ + const struct alc_codec_rename_table *p; + + for (p = rename_tbl; p->vendor_id; p++) { + if (p->vendor_id != codec->vendor_id) + continue; + if ((alc_get_coef0(codec) & p->coef_mask) == p->coef_bits) + return alc_codec_rename(codec, p->name); + } + return 0; +} + +/* * Automatic parse of I/O pins from the BIOS configuration */ @@ -2448,11 +2540,15 @@ enum { ALC_CTL_WIDGET_VOL, ALC_CTL_WIDGET_MUTE, ALC_CTL_BIND_MUTE, + ALC_CTL_BIND_VOL, + ALC_CTL_BIND_SW, }; static const struct snd_kcontrol_new alc_control_templates[] = { HDA_CODEC_VOLUME(NULL, 0, 0, 0), HDA_CODEC_MUTE(NULL, 0, 0, 0), HDA_BIND_MUTE(NULL, 0, 0, 0), + HDA_BIND_VOL(NULL, 0), + HDA_BIND_SW(NULL, 0), }; /* add dynamic controls */ @@ -2493,13 +2589,14 @@ static int add_control_with_pfx(struct alc_spec *spec, int type, #define __add_pb_sw_ctrl(spec, type, pfx, cidx, val) \ add_control_with_pfx(spec, type, pfx, "Playback", "Switch", cidx, val) +static const char * const channel_name[4] = { + "Front", "Surround", "CLFE", "Side" +}; + static const char *alc_get_line_out_pfx(struct alc_spec *spec, int ch, bool can_be_master, int *index) { struct auto_pin_cfg *cfg = &spec->autocfg; - static const char * const chname[4] = { - "Front", "Surround", NULL /*CLFE*/, "Side" - }; *index = 0; if (cfg->line_outs == 1 && !spec->multi_ios && @@ -2522,7 +2619,10 @@ static const char *alc_get_line_out_pfx(struct alc_spec *spec, int ch, return "PCM"; break; } - return chname[ch]; + if (snd_BUG_ON(ch >= ARRAY_SIZE(channel_name))) + return "PCM"; + + return channel_name[ch]; } /* create input playback/capture controls for the given pin */ @@ -2786,8 +2886,9 @@ static hda_nid_t alc_auto_look_for_dac(struct hda_codec *codec, hda_nid_t pin) if (found_in_nid_list(nid, spec->multiout.dac_nids, spec->multiout.num_dacs)) continue; - if (spec->multiout.hp_nid == nid) - continue; + if (found_in_nid_list(nid, spec->multiout.hp_out_nid, + ARRAY_SIZE(spec->multiout.hp_out_nid))) + continue; if (found_in_nid_list(nid, spec->multiout.extra_out_nid, ARRAY_SIZE(spec->multiout.extra_out_nid))) continue; @@ -2804,6 +2905,29 @@ static hda_nid_t get_dac_if_single(struct hda_codec *codec, hda_nid_t pin) return 0; } +static int alc_auto_fill_extra_dacs(struct hda_codec *codec, int num_outs, + const hda_nid_t *pins, hda_nid_t *dacs) +{ + int i; + + if (num_outs && !dacs[0]) { + dacs[0] = alc_auto_look_for_dac(codec, pins[0]); + if (!dacs[0]) + return 0; + } + + for (i = 1; i < num_outs; i++) + dacs[i] = get_dac_if_single(codec, pins[i]); + for (i = 1; i < num_outs; i++) { + if (!dacs[i]) + dacs[i] = alc_auto_look_for_dac(codec, pins[i]); + } + return 0; +} + +static int alc_auto_fill_multi_ios(struct hda_codec *codec, + unsigned int location); + /* fill in the dac_nids table from the parsed pin configuration */ static int alc_auto_fill_dac_nids(struct hda_codec *codec) { @@ -2815,7 +2939,7 @@ static int alc_auto_fill_dac_nids(struct hda_codec *codec) again: /* set num_dacs once to full for alc_auto_look_for_dac() */ spec->multiout.num_dacs = cfg->line_outs; - spec->multiout.hp_nid = 0; + spec->multiout.hp_out_nid[0] = 0; spec->multiout.extra_out_nid[0] = 0; memset(spec->private_dac_nids, 0, sizeof(spec->private_dac_nids)); spec->multiout.dac_nids = spec->private_dac_nids; @@ -2826,7 +2950,7 @@ static int alc_auto_fill_dac_nids(struct hda_codec *codec) spec->private_dac_nids[i] = get_dac_if_single(codec, cfg->line_out_pins[i]); if (cfg->hp_outs) - spec->multiout.hp_nid = + spec->multiout.hp_out_nid[0] = get_dac_if_single(codec, cfg->hp_pins[0]); if (cfg->speaker_outs) spec->multiout.extra_out_nid[0] = @@ -2858,24 +2982,58 @@ static int alc_auto_fill_dac_nids(struct hda_codec *codec) sizeof(hda_nid_t) * (cfg->line_outs - i - 1)); } - if (cfg->hp_outs && !spec->multiout.hp_nid) - spec->multiout.hp_nid = - alc_auto_look_for_dac(codec, cfg->hp_pins[0]); - if (cfg->speaker_outs && !spec->multiout.extra_out_nid[0]) - spec->multiout.extra_out_nid[0] = - alc_auto_look_for_dac(codec, cfg->speaker_pins[0]); + if (cfg->line_outs == 1 && cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) { + /* try to fill multi-io first */ + unsigned int location, defcfg; + int num_pins; + + defcfg = snd_hda_codec_get_pincfg(codec, cfg->line_out_pins[0]); + location = get_defcfg_location(defcfg); + + num_pins = alc_auto_fill_multi_ios(codec, location); + if (num_pins > 0) { + spec->multi_ios = num_pins; + spec->ext_channel_count = 2; + spec->multiout.num_dacs = num_pins + 1; + } + } + + if (cfg->line_out_type != AUTO_PIN_HP_OUT) + alc_auto_fill_extra_dacs(codec, cfg->hp_outs, cfg->hp_pins, + spec->multiout.hp_out_nid); + if (cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) + alc_auto_fill_extra_dacs(codec, cfg->speaker_outs, cfg->speaker_pins, + spec->multiout.extra_out_nid); return 0; } +static inline unsigned int get_ctl_pos(unsigned int data) +{ + hda_nid_t nid = get_amp_nid_(data); + unsigned int dir = get_amp_direction_(data); + return (nid << 1) | dir; +} + +#define is_ctl_used(bits, data) \ + test_bit(get_ctl_pos(data), bits) +#define mark_ctl_usage(bits, data) \ + set_bit(get_ctl_pos(data), bits) + static int alc_auto_add_vol_ctl(struct hda_codec *codec, const char *pfx, int cidx, hda_nid_t nid, unsigned int chs) { + struct alc_spec *spec = codec->spec; + unsigned int val; if (!nid) return 0; + val = HDA_COMPOSE_AMP_VAL(nid, chs, 0, HDA_OUTPUT); + if (is_ctl_used(spec->vol_ctls, val) && chs != 2) /* exclude LFE */ + return 0; + mark_ctl_usage(spec->vol_ctls, val); return __add_pb_vol_ctrl(codec->spec, ALC_CTL_WIDGET_VOL, pfx, cidx, - HDA_COMPOSE_AMP_VAL(nid, chs, 0, HDA_OUTPUT)); + val); } #define alc_auto_add_stereo_vol(codec, pfx, cidx, nid) \ @@ -2888,6 +3046,7 @@ static int alc_auto_add_sw_ctl(struct hda_codec *codec, const char *pfx, int cidx, hda_nid_t nid, unsigned int chs) { + struct alc_spec *spec = codec->spec; int wid_type; int type; unsigned long val; @@ -2904,6 +3063,9 @@ static int alc_auto_add_sw_ctl(struct hda_codec *codec, type = ALC_CTL_BIND_MUTE; val = HDA_COMPOSE_AMP_VAL(nid, chs, 2, HDA_INPUT); } + if (is_ctl_used(spec->sw_ctls, val) && chs != 2) /* exclude LFE */ + return 0; + mark_ctl_usage(spec->sw_ctls, val); return __add_pb_sw_ctrl(codec->spec, type, pfx, cidx, val); } @@ -2964,7 +3126,7 @@ static int alc_auto_create_multi_out_ctls(struct hda_codec *codec, sw = alc_look_for_out_mute_nid(codec, pin, dac); vol = alc_look_for_out_vol_nid(codec, pin, dac); name = alc_get_line_out_pfx(spec, i, true, &index); - if (!name) { + if (!name || !strcmp(name, "CLFE")) { /* Center/LFE */ err = alc_auto_add_vol_ctl(codec, "Center", 0, vol, 1); if (err < 0) @@ -2990,23 +3152,24 @@ static int alc_auto_create_multi_out_ctls(struct hda_codec *codec, return 0; } -/* add playback controls for speaker and HP outputs */ static int alc_auto_create_extra_out(struct hda_codec *codec, hda_nid_t pin, - hda_nid_t dac, const char *pfx) + hda_nid_t dac, const char *pfx) { struct alc_spec *spec = codec->spec; hda_nid_t sw, vol; int err; - if (!pin) - return 0; if (!dac) { + unsigned int val; /* the corresponding DAC is already occupied */ if (!(get_wcaps(codec, pin) & AC_WCAP_OUT_AMP)) return 0; /* no way */ /* create a switch only */ - return add_pb_sw_ctrl(spec, ALC_CTL_WIDGET_MUTE, pfx, - HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_OUTPUT)); + val = HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_OUTPUT); + if (is_ctl_used(spec->sw_ctls, val)) + return 0; /* already created */ + mark_ctl_usage(spec->sw_ctls, val); + return add_pb_sw_ctrl(spec, ALC_CTL_WIDGET_MUTE, pfx, val); } sw = alc_look_for_out_mute_nid(codec, pin, dac); @@ -3020,20 +3183,112 @@ static int alc_auto_create_extra_out(struct hda_codec *codec, hda_nid_t pin, return 0; } +static struct hda_bind_ctls *new_bind_ctl(struct hda_codec *codec, + unsigned int nums, + struct hda_ctl_ops *ops) +{ + struct alc_spec *spec = codec->spec; + struct hda_bind_ctls **ctlp, *ctl; + snd_array_init(&spec->bind_ctls, sizeof(ctl), 8); + ctlp = snd_array_new(&spec->bind_ctls); + if (!ctlp) + return NULL; + ctl = kzalloc(sizeof(*ctl) + sizeof(long) * (nums + 1), GFP_KERNEL); + *ctlp = ctl; + if (ctl) + ctl->ops = ops; + return ctl; +} + +/* add playback controls for speaker and HP outputs */ +static int alc_auto_create_extra_outs(struct hda_codec *codec, int num_pins, + const hda_nid_t *pins, + const hda_nid_t *dacs, + const char *pfx) +{ + struct alc_spec *spec = codec->spec; + struct hda_bind_ctls *ctl; + char name[32]; + int i, n, err; + + if (!num_pins || !pins[0]) + return 0; + + if (num_pins == 1) { + hda_nid_t dac = *dacs; + if (!dac) + dac = spec->multiout.dac_nids[0]; + return alc_auto_create_extra_out(codec, *pins, dac, pfx); + } + + if (dacs[num_pins - 1]) { + /* OK, we have a multi-output system with individual volumes */ + for (i = 0; i < num_pins; i++) { + snprintf(name, sizeof(name), "%s %s", + pfx, channel_name[i]); + err = alc_auto_create_extra_out(codec, pins[i], dacs[i], + name); + if (err < 0) + return err; + } + return 0; + } + + /* Let's create a bind-controls */ + ctl = new_bind_ctl(codec, num_pins, &snd_hda_bind_sw); + if (!ctl) + return -ENOMEM; + n = 0; + for (i = 0; i < num_pins; i++) { + if (get_wcaps(codec, pins[i]) & AC_WCAP_OUT_AMP) + ctl->values[n++] = + HDA_COMPOSE_AMP_VAL(pins[i], 3, 0, HDA_OUTPUT); + } + if (n) { + snprintf(name, sizeof(name), "%s Playback Switch", pfx); + err = add_control(spec, ALC_CTL_BIND_SW, name, 0, (long)ctl); + if (err < 0) + return err; + } + + ctl = new_bind_ctl(codec, num_pins, &snd_hda_bind_vol); + if (!ctl) + return -ENOMEM; + n = 0; + for (i = 0; i < num_pins; i++) { + hda_nid_t vol; + if (!pins[i] || !dacs[i]) + continue; + vol = alc_look_for_out_vol_nid(codec, pins[i], dacs[i]); + if (vol) + ctl->values[n++] = + HDA_COMPOSE_AMP_VAL(vol, 3, 0, HDA_OUTPUT); + } + if (n) { + snprintf(name, sizeof(name), "%s Playback Volume", pfx); + err = add_control(spec, ALC_CTL_BIND_VOL, name, 0, (long)ctl); + if (err < 0) + return err; + } + return 0; +} + static int alc_auto_create_hp_out(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - return alc_auto_create_extra_out(codec, spec->autocfg.hp_pins[0], - spec->multiout.hp_nid, - "Headphone"); + return alc_auto_create_extra_outs(codec, spec->autocfg.hp_outs, + spec->autocfg.hp_pins, + spec->multiout.hp_out_nid, + "Headphone"); } static int alc_auto_create_speaker_out(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - return alc_auto_create_extra_out(codec, spec->autocfg.speaker_pins[0], - spec->multiout.extra_out_nid[0], - "Speaker"); + return alc_auto_create_extra_outs(codec, spec->autocfg.speaker_outs, + spec->autocfg.speaker_pins, + spec->multiout.extra_out_nid, + "Speaker"); } static void alc_auto_set_output_and_unmute(struct hda_codec *codec, @@ -3090,20 +3345,37 @@ static void alc_auto_init_multi_out(struct hda_codec *codec) static void alc_auto_init_extra_out(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; + int i; hda_nid_t pin, dac; - pin = spec->autocfg.hp_pins[0]; - if (pin) { - dac = spec->multiout.hp_nid; - if (!dac) - dac = spec->multiout.dac_nids[0]; + for (i = 0; i < spec->autocfg.hp_outs; i++) { + if (spec->autocfg.line_out_type == AUTO_PIN_HP_OUT) + break; + pin = spec->autocfg.hp_pins[i]; + if (!pin) + break; + dac = spec->multiout.hp_out_nid[i]; + if (!dac) { + if (i > 0 && spec->multiout.hp_out_nid[0]) + dac = spec->multiout.hp_out_nid[0]; + else + dac = spec->multiout.dac_nids[0]; + } alc_auto_set_output_and_unmute(codec, pin, PIN_HP, dac); } - pin = spec->autocfg.speaker_pins[0]; - if (pin) { - dac = spec->multiout.extra_out_nid[0]; - if (!dac) - dac = spec->multiout.dac_nids[0]; + for (i = 0; i < spec->autocfg.speaker_outs; i++) { + if (spec->autocfg.line_out_type == AUTO_PIN_SPEAKER_OUT) + break; + pin = spec->autocfg.speaker_pins[i]; + if (!pin) + break; + dac = spec->multiout.extra_out_nid[i]; + if (!dac) { + if (i > 0 && spec->multiout.extra_out_nid[0]) + dac = spec->multiout.extra_out_nid[0]; + else + dac = spec->multiout.dac_nids[0]; + } alc_auto_set_output_and_unmute(codec, pin, PIN_OUT, dac); } } @@ -3116,6 +3388,7 @@ static int alc_auto_fill_multi_ios(struct hda_codec *codec, { struct alc_spec *spec = codec->spec; struct auto_pin_cfg *cfg = &spec->autocfg; + hda_nid_t prime_dac = spec->private_dac_nids[0]; int type, i, num_pins = 0; for (type = AUTO_PIN_LINE_IN; type >= AUTO_PIN_MIC; type--) { @@ -3143,8 +3416,13 @@ static int alc_auto_fill_multi_ios(struct hda_codec *codec, } } spec->multiout.num_dacs = 1; - if (num_pins < 2) + if (num_pins < 2) { + /* clear up again */ + memset(spec->private_dac_nids, 0, + sizeof(spec->private_dac_nids)); + spec->private_dac_nids[0] = prime_dac; return 0; + } return num_pins; } @@ -3230,36 +3508,11 @@ static const struct snd_kcontrol_new alc_auto_channel_mode_enum = { .put = alc_auto_ch_mode_put, }; -static int alc_auto_add_multi_channel_mode(struct hda_codec *codec, - int (*fill_dac)(struct hda_codec *)) +static int alc_auto_add_multi_channel_mode(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - struct auto_pin_cfg *cfg = &spec->autocfg; - unsigned int location, defcfg; - int num_pins; - - if (cfg->line_out_type == AUTO_PIN_SPEAKER_OUT && cfg->hp_outs == 1) { - /* use HP as primary out */ - cfg->speaker_outs = cfg->line_outs; - memcpy(cfg->speaker_pins, cfg->line_out_pins, - sizeof(cfg->speaker_pins)); - cfg->line_outs = cfg->hp_outs; - memcpy(cfg->line_out_pins, cfg->hp_pins, sizeof(cfg->hp_pins)); - cfg->hp_outs = 0; - memset(cfg->hp_pins, 0, sizeof(cfg->hp_pins)); - cfg->line_out_type = AUTO_PIN_HP_OUT; - if (fill_dac) - fill_dac(codec); - } - if (cfg->line_outs != 1 || - cfg->line_out_type == AUTO_PIN_SPEAKER_OUT) - return 0; - defcfg = snd_hda_codec_get_pincfg(codec, cfg->line_out_pins[0]); - location = get_defcfg_location(defcfg); - - num_pins = alc_auto_fill_multi_ios(codec, location); - if (num_pins > 0) { + if (spec->multi_ios > 0) { struct snd_kcontrol_new *knew; knew = alc_kcontrol_new(spec); @@ -3269,10 +3522,6 @@ static int alc_auto_add_multi_channel_mode(struct hda_codec *codec, knew->name = kstrdup("Channel Mode", GFP_KERNEL); if (!knew->name) return -ENOMEM; - - spec->multi_ios = num_pins; - spec->ext_channel_count = 2; - spec->multiout.num_dacs = num_pins + 1; } return 0; } @@ -3555,27 +3804,42 @@ static int alc_parse_auto_config(struct hda_codec *codec, const hda_nid_t *ssid_nids) { struct alc_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; int err; - err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, - ignore_nids); + err = snd_hda_parse_pin_defcfg(codec, cfg, ignore_nids, + spec->parse_flags); if (err < 0) return err; - if (!spec->autocfg.line_outs) { - if (spec->autocfg.dig_outs || spec->autocfg.dig_in_pin) { + if (!cfg->line_outs) { + if (cfg->dig_outs || cfg->dig_in_pin) { spec->multiout.max_channels = 2; spec->no_analog = 1; goto dig_only; } return 0; /* can't find valid BIOS pin config */ } + + if (cfg->line_out_type == AUTO_PIN_SPEAKER_OUT && + cfg->line_outs <= cfg->hp_outs) { + /* use HP as primary out */ + cfg->speaker_outs = cfg->line_outs; + memcpy(cfg->speaker_pins, cfg->line_out_pins, + sizeof(cfg->speaker_pins)); + cfg->line_outs = cfg->hp_outs; + memcpy(cfg->line_out_pins, cfg->hp_pins, sizeof(cfg->hp_pins)); + cfg->hp_outs = 0; + memset(cfg->hp_pins, 0, sizeof(cfg->hp_pins)); + cfg->line_out_type = AUTO_PIN_HP_OUT; + } + err = alc_auto_fill_dac_nids(codec); if (err < 0) return err; - err = alc_auto_add_multi_channel_mode(codec, alc_auto_fill_dac_nids); + err = alc_auto_add_multi_channel_mode(codec); if (err < 0) return err; - err = alc_auto_create_multi_out_ctls(codec, &spec->autocfg); + err = alc_auto_create_multi_out_ctls(codec, cfg); if (err < 0) return err; err = alc_auto_create_hp_out(codec); @@ -3678,10 +3942,8 @@ static int patch_alc880(struct hda_codec *codec) if (board_config == ALC_MODEL_AUTO) { /* automatic parse from the BIOS config */ err = alc880_parse_auto_config(codec); - if (err < 0) { - alc_free(codec); - return err; - } + if (err < 0) + goto error; #ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS else if (!err) { printk(KERN_INFO @@ -3706,10 +3968,8 @@ static int patch_alc880(struct hda_codec *codec) if (!spec->no_analog) { err = snd_hda_attach_beep_device(codec, 0x1); - if (err < 0) { - alc_free(codec); - return err; - } + if (err < 0) + goto error; set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); } @@ -3724,6 +3984,10 @@ static int patch_alc880(struct hda_codec *codec) #endif return 0; + + error: + alc_free(codec); + return err; } @@ -3805,10 +4069,8 @@ static int patch_alc260(struct hda_codec *codec) if (board_config == ALC_MODEL_AUTO) { /* automatic parse from the BIOS config */ err = alc260_parse_auto_config(codec); - if (err < 0) { - alc_free(codec); - return err; - } + if (err < 0) + goto error; #ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS else if (!err) { printk(KERN_INFO @@ -3833,10 +4095,8 @@ static int patch_alc260(struct hda_codec *codec) if (!spec->no_analog) { err = snd_hda_attach_beep_device(codec, 0x1); - if (err < 0) { - alc_free(codec); - return err; - } + if (err < 0) + goto error; set_beep_amp(spec, 0x07, 0x05, HDA_INPUT); } @@ -3854,6 +4114,10 @@ static int patch_alc260(struct hda_codec *codec) #endif return 0; + + error: + alc_free(codec); + return err; } @@ -3880,6 +4144,7 @@ enum { PINFIX_LENOVO_Y530, PINFIX_PB_M5210, PINFIX_ACER_ASPIRE_7736, + PINFIX_ASUS_W90V, }; static const struct alc_fixup alc882_fixups[] = { @@ -3911,10 +4176,18 @@ static const struct alc_fixup alc882_fixups[] = { .type = ALC_FIXUP_SKU, .v.sku = ALC_FIXUP_SKU_IGNORE, }, + [PINFIX_ASUS_W90V] = { + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { + { 0x16, 0x99130110 }, /* fix sequence for CLFE */ + { } + } + }, }; static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x0155, "Packard-Bell M5120", PINFIX_PB_M5210), + SND_PCI_QUIRK(0x1043, 0x1873, "ASUS W90V", PINFIX_ASUS_W90V), SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Y530", PINFIX_LENOVO_Y530), SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", PINFIX_ABIT_AW9D_MAX), SND_PCI_QUIRK(0x1025, 0x0296, "Acer Aspire 7736z", PINFIX_ACER_ASPIRE_7736), @@ -3961,6 +4234,10 @@ static int patch_alc882(struct hda_codec *codec) break; } + err = alc_codec_rename_from_preset(codec); + if (err < 0) + goto error; + board_config = alc_board_config(codec, ALC882_MODEL_LAST, alc882_models, alc882_cfg_tbl); @@ -3984,10 +4261,8 @@ static int patch_alc882(struct hda_codec *codec) if (board_config == ALC_MODEL_AUTO) { /* automatic parse from the BIOS config */ err = alc882_parse_auto_config(codec); - if (err < 0) { - alc_free(codec); - return err; - } + if (err < 0) + goto error; #ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS else if (!err) { printk(KERN_INFO @@ -4012,10 +4287,8 @@ static int patch_alc882(struct hda_codec *codec) if (!spec->no_analog && has_cdefine_beep(codec)) { err = snd_hda_attach_beep_device(codec, 0x1); - if (err < 0) { - alc_free(codec); - return err; - } + if (err < 0) + goto error; set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); } @@ -4034,6 +4307,10 @@ static int patch_alc882(struct hda_codec *codec) #endif return 0; + + error: + alc_free(codec); + return err; } @@ -4138,10 +4415,8 @@ static int patch_alc262(struct hda_codec *codec) if (board_config == ALC_MODEL_AUTO) { /* automatic parse from the BIOS config */ err = alc262_parse_auto_config(codec); - if (err < 0) { - alc_free(codec); - return err; - } + if (err < 0) + goto error; #ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS else if (!err) { printk(KERN_INFO @@ -4166,10 +4441,8 @@ static int patch_alc262(struct hda_codec *codec) if (!spec->no_analog && has_cdefine_beep(codec)) { err = snd_hda_attach_beep_device(codec, 0x1); - if (err < 0) { - alc_free(codec); - return err; - } + if (err < 0) + goto error; set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); } @@ -4189,6 +4462,10 @@ static int patch_alc262(struct hda_codec *codec) #endif return 0; + + error: + alc_free(codec); + return err; } /* @@ -4237,14 +4514,9 @@ static int alc268_parse_auto_config(struct hda_codec *codec) /* */ -#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS -#include "alc268_quirks.c" -#endif - static int patch_alc268(struct hda_codec *codec) { struct alc_spec *spec; - int board_config; int i, has_beep, err; spec = kzalloc(sizeof(*spec), GFP_KERNEL); @@ -4255,38 +4527,10 @@ static int patch_alc268(struct hda_codec *codec) /* ALC268 has no aa-loopback mixer */ - board_config = alc_board_config(codec, ALC268_MODEL_LAST, - alc268_models, alc268_cfg_tbl); - - if (board_config < 0) - board_config = alc_board_codec_sid_config(codec, - ALC268_MODEL_LAST, alc268_models, alc268_ssid_cfg_tbl); - - if (board_config < 0) { - printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n", - codec->chip_name); - board_config = ALC_MODEL_AUTO; - } - - if (board_config == ALC_MODEL_AUTO) { - /* automatic parse from the BIOS config */ - err = alc268_parse_auto_config(codec); - if (err < 0) { - alc_free(codec); - return err; - } -#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS - else if (!err) { - printk(KERN_INFO - "hda_codec: Cannot set up configuration " - "from BIOS. Using base mode...\n"); - board_config = ALC268_3ST; - } -#endif - } - - if (board_config != ALC_MODEL_AUTO) - setup_preset(codec, &alc268_presets[board_config]); + /* automatic parse from the BIOS config */ + err = alc268_parse_auto_config(codec); + if (err < 0) + goto error; has_beep = 0; for (i = 0; i < spec->num_mixers; i++) { @@ -4298,10 +4542,8 @@ static int patch_alc268(struct hda_codec *codec) if (has_beep) { err = snd_hda_attach_beep_device(codec, 0x1); - if (err < 0) { - alc_free(codec); - return err; - } + if (err < 0) + goto error; if (!query_amp_caps(codec, 0x1d, HDA_INPUT)) /* override the amp caps for beep generator */ snd_hda_override_amp_caps(codec, 0x1d, HDA_INPUT, @@ -4323,13 +4565,16 @@ static int patch_alc268(struct hda_codec *codec) spec->vmaster_nid = 0x02; codec->patch_ops = alc_patch_ops; - if (board_config == ALC_MODEL_AUTO) - spec->init_hook = alc_auto_init_std; + spec->init_hook = alc_auto_init_std; spec->shutup = alc_eapd_shutup; alc_init_jacks(codec); return 0; + + error: + alc_free(codec); + return err; } /* @@ -4423,9 +4668,9 @@ static void alc269_toggle_power_output(struct hda_codec *codec, int power_up) static void alc269_shutup(struct hda_codec *codec) { - if ((alc_read_coef_idx(codec, 0) & 0x00ff) == 0x017) + if ((alc_get_coef0(codec) & 0x00ff) == 0x017) alc269_toggle_power_output(codec, 0); - if ((alc_read_coef_idx(codec, 0) & 0x00ff) == 0x018) { + if ((alc_get_coef0(codec) & 0x00ff) == 0x018) { alc269_toggle_power_output(codec, 0); msleep(150); } @@ -4434,19 +4679,19 @@ static void alc269_shutup(struct hda_codec *codec) #ifdef CONFIG_PM static int alc269_resume(struct hda_codec *codec) { - if ((alc_read_coef_idx(codec, 0) & 0x00ff) == 0x018) { + if ((alc_get_coef0(codec) & 0x00ff) == 0x018) { alc269_toggle_power_output(codec, 0); msleep(150); } codec->patch_ops.init(codec); - if ((alc_read_coef_idx(codec, 0) & 0x00ff) == 0x017) { + if ((alc_get_coef0(codec) & 0x00ff) == 0x017) { alc269_toggle_power_output(codec, 1); msleep(200); } - if ((alc_read_coef_idx(codec, 0) & 0x00ff) == 0x018) + if ((alc_get_coef0(codec) & 0x00ff) == 0x018) alc269_toggle_power_output(codec, 1); snd_hda_codec_resume_amp(codec); @@ -4515,6 +4760,30 @@ static void alc269_fixup_stereo_dmic(struct hda_codec *codec, alc_write_coef_idx(codec, 0x07, coef | 0x80); } +static void alc269_quanta_automute(struct hda_codec *codec) +{ + update_outputs(codec); + + snd_hda_codec_write(codec, 0x20, 0, + AC_VERB_SET_COEF_INDEX, 0x0c); + snd_hda_codec_write(codec, 0x20, 0, + AC_VERB_SET_PROC_COEF, 0x680); + + snd_hda_codec_write(codec, 0x20, 0, + AC_VERB_SET_COEF_INDEX, 0x0c); + snd_hda_codec_write(codec, 0x20, 0, + AC_VERB_SET_PROC_COEF, 0x480); +} + +static void alc269_fixup_quanta_mute(struct hda_codec *codec, + const struct alc_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + if (action != ALC_FIXUP_ACT_PROBE) + return; + spec->automute_hook = alc269_quanta_automute; +} + enum { ALC269_FIXUP_SONY_VAIO, ALC275_FIXUP_SONY_VAIO_GPIO2, @@ -4526,6 +4795,12 @@ enum { ALC271_FIXUP_DMIC, ALC269_FIXUP_PCM_44K, ALC269_FIXUP_STEREO_DMIC, + ALC269_FIXUP_QUANTA_MUTE, + ALC269_FIXUP_LIFEBOOK, + ALC269_FIXUP_AMIC, + ALC269_FIXUP_DMIC, + ALC269VB_FIXUP_AMIC, + ALC269VB_FIXUP_DMIC, }; static const struct alc_fixup alc269_fixups[] = { @@ -4592,6 +4867,60 @@ static const struct alc_fixup alc269_fixups[] = { .type = ALC_FIXUP_FUNC, .v.func = alc269_fixup_stereo_dmic, }, + [ALC269_FIXUP_QUANTA_MUTE] = { + .type = ALC_FIXUP_FUNC, + .v.func = alc269_fixup_quanta_mute, + }, + [ALC269_FIXUP_LIFEBOOK] = { + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { + { 0x1a, 0x2101103f }, /* dock line-out */ + { 0x1b, 0x23a11040 }, /* dock mic-in */ + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_QUANTA_MUTE + }, + [ALC269_FIXUP_AMIC] = { + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { + { 0x14, 0x99130110 }, /* speaker */ + { 0x15, 0x0121401f }, /* HP out */ + { 0x18, 0x01a19c20 }, /* mic */ + { 0x19, 0x99a3092f }, /* int-mic */ + { } + }, + }, + [ALC269_FIXUP_DMIC] = { + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { + { 0x12, 0x99a3092f }, /* int-mic */ + { 0x14, 0x99130110 }, /* speaker */ + { 0x15, 0x0121401f }, /* HP out */ + { 0x18, 0x01a19c20 }, /* mic */ + { } + }, + }, + [ALC269VB_FIXUP_AMIC] = { + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { + { 0x14, 0x99130110 }, /* speaker */ + { 0x18, 0x01a19c20 }, /* mic */ + { 0x19, 0x99a3092f }, /* int-mic */ + { 0x21, 0x0121401f }, /* HP out */ + { } + }, + }, + [ALC269_FIXUP_DMIC] = { + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { + { 0x12, 0x99a3092f }, /* int-mic */ + { 0x14, 0x99130110 }, /* speaker */ + { 0x18, 0x01a19c20 }, /* mic */ + { 0x21, 0x0121401f }, /* HP out */ + { } + }, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -4607,13 +4936,71 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK_VENDOR(0x104d, "Sony VAIO", ALC269_FIXUP_SONY_VAIO), SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z), SND_PCI_QUIRK_VENDOR(0x1025, "Acer Aspire", ALC271_FIXUP_DMIC), + SND_PCI_QUIRK(0x10cf, 0x1475, "Lifebook", ALC269_FIXUP_LIFEBOOK), SND_PCI_QUIRK(0x17aa, 0x20f2, "Thinkpad SL410/510", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x17aa, 0x215e, "Thinkpad L512", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x17aa, 0x21b8, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x17aa, 0x21ca, "Thinkpad L412", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x17aa, 0x21e9, "Thinkpad Edge 15", ALC269_FIXUP_SKU_IGNORE), + SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_QUANTA_MUTE), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Lenovo Ideapd", ALC269_FIXUP_PCM_44K), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), + +#if 1 + /* Below is a quirk table taken from the old code. + * Basically the device should work as is without the fixup table. + * If BIOS doesn't give a proper info, enable the corresponding + * fixup entry. + */ + SND_PCI_QUIRK(0x1043, 0x8330, "ASUS Eeepc P703 P900A", + ALC269_FIXUP_AMIC), + SND_PCI_QUIRK(0x1043, 0x1013, "ASUS N61Da", ALC269_FIXUP_AMIC), + SND_PCI_QUIRK(0x1043, 0x1113, "ASUS N63Jn", ALC269_FIXUP_AMIC), + SND_PCI_QUIRK(0x1043, 0x1143, "ASUS B53f", ALC269_FIXUP_AMIC), + SND_PCI_QUIRK(0x1043, 0x1133, "ASUS UJ20ft", ALC269_FIXUP_AMIC), + SND_PCI_QUIRK(0x1043, 0x1183, "ASUS K72DR", ALC269_FIXUP_AMIC), + SND_PCI_QUIRK(0x1043, 0x11b3, "ASUS K52DR", ALC269_FIXUP_AMIC), + SND_PCI_QUIRK(0x1043, 0x11e3, "ASUS U33Jc", ALC269_FIXUP_AMIC), + SND_PCI_QUIRK(0x1043, 0x1273, "ASUS UL80Jt", ALC269_FIXUP_AMIC), + SND_PCI_QUIRK(0x1043, 0x1283, "ASUS U53Jc", ALC269_FIXUP_AMIC), + SND_PCI_QUIRK(0x1043, 0x12b3, "ASUS N82JV", ALC269_FIXUP_AMIC), + SND_PCI_QUIRK(0x1043, 0x12d3, "ASUS N61Jv", ALC269_FIXUP_AMIC), + SND_PCI_QUIRK(0x1043, 0x13a3, "ASUS UL30Vt", ALC269_FIXUP_AMIC), + SND_PCI_QUIRK(0x1043, 0x1373, "ASUS G73JX", ALC269_FIXUP_AMIC), + SND_PCI_QUIRK(0x1043, 0x1383, "ASUS UJ30Jc", ALC269_FIXUP_AMIC), + SND_PCI_QUIRK(0x1043, 0x13d3, "ASUS N61JA", ALC269_FIXUP_AMIC), + SND_PCI_QUIRK(0x1043, 0x1413, "ASUS UL50", ALC269_FIXUP_AMIC), + SND_PCI_QUIRK(0x1043, 0x1443, "ASUS UL30", ALC269_FIXUP_AMIC), + SND_PCI_QUIRK(0x1043, 0x1453, "ASUS M60Jv", ALC269_FIXUP_AMIC), + SND_PCI_QUIRK(0x1043, 0x1483, "ASUS UL80", ALC269_FIXUP_AMIC), + SND_PCI_QUIRK(0x1043, 0x14f3, "ASUS F83Vf", ALC269_FIXUP_AMIC), + SND_PCI_QUIRK(0x1043, 0x14e3, "ASUS UL20", ALC269_FIXUP_AMIC), + SND_PCI_QUIRK(0x1043, 0x1513, "ASUS UX30", ALC269_FIXUP_AMIC), + SND_PCI_QUIRK(0x1043, 0x1593, "ASUS N51Vn", ALC269_FIXUP_AMIC), + SND_PCI_QUIRK(0x1043, 0x15a3, "ASUS N60Jv", ALC269_FIXUP_AMIC), + SND_PCI_QUIRK(0x1043, 0x15b3, "ASUS N60Dp", ALC269_FIXUP_AMIC), + SND_PCI_QUIRK(0x1043, 0x15c3, "ASUS N70De", ALC269_FIXUP_AMIC), + SND_PCI_QUIRK(0x1043, 0x15e3, "ASUS F83T", ALC269_FIXUP_AMIC), + SND_PCI_QUIRK(0x1043, 0x1643, "ASUS M60J", ALC269_FIXUP_AMIC), + SND_PCI_QUIRK(0x1043, 0x1653, "ASUS U50", ALC269_FIXUP_AMIC), + SND_PCI_QUIRK(0x1043, 0x1693, "ASUS F50N", ALC269_FIXUP_AMIC), + SND_PCI_QUIRK(0x1043, 0x16a3, "ASUS F5Q", ALC269_FIXUP_AMIC), + SND_PCI_QUIRK(0x1043, 0x1723, "ASUS P80", ALC269_FIXUP_AMIC), + SND_PCI_QUIRK(0x1043, 0x1743, "ASUS U80", ALC269_FIXUP_AMIC), + SND_PCI_QUIRK(0x1043, 0x1773, "ASUS U20A", ALC269_FIXUP_AMIC), + SND_PCI_QUIRK(0x1043, 0x1883, "ASUS F81Se", ALC269_FIXUP_AMIC), + SND_PCI_QUIRK(0x152d, 0x1778, "Quanta ON1", ALC269_FIXUP_DMIC), + SND_PCI_QUIRK(0x17aa, 0x3be9, "Quanta Wistron", ALC269_FIXUP_AMIC), + SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_AMIC), + SND_PCI_QUIRK(0x17ff, 0x059a, "Quanta EL3", ALC269_FIXUP_DMIC), + SND_PCI_QUIRK(0x17ff, 0x059b, "Quanta JR1", ALC269_FIXUP_DMIC), +#endif + {} +}; + +static const struct alc_model_fixup alc269_fixup_models[] = { + {.id = ALC269_FIXUP_AMIC, .name = "laptop-amic"}, + {.id = ALC269_FIXUP_DMIC, .name = "laptop-dmic"}, {} }; @@ -4622,23 +5009,23 @@ static int alc269_fill_coef(struct hda_codec *codec) { int val; - if ((alc_read_coef_idx(codec, 0) & 0x00ff) < 0x015) { + if ((alc_get_coef0(codec) & 0x00ff) < 0x015) { alc_write_coef_idx(codec, 0xf, 0x960b); alc_write_coef_idx(codec, 0xe, 0x8817); } - if ((alc_read_coef_idx(codec, 0) & 0x00ff) == 0x016) { + if ((alc_get_coef0(codec) & 0x00ff) == 0x016) { alc_write_coef_idx(codec, 0xf, 0x960b); alc_write_coef_idx(codec, 0xe, 0x8814); } - if ((alc_read_coef_idx(codec, 0) & 0x00ff) == 0x017) { + if ((alc_get_coef0(codec) & 0x00ff) == 0x017) { val = alc_read_coef_idx(codec, 0x04); /* Power up output pin */ alc_write_coef_idx(codec, 0x04, val | (1<<11)); } - if ((alc_read_coef_idx(codec, 0) & 0x00ff) == 0x018) { + if ((alc_get_coef0(codec) & 0x00ff) == 0x018) { val = alc_read_coef_idx(codec, 0xd); if ((val & 0x0c00) >> 10 != 0x1) { /* Capless ramp up clock control */ @@ -4662,15 +5049,10 @@ static int alc269_fill_coef(struct hda_codec *codec) /* */ -#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS -#include "alc269_quirks.c" -#endif - static int patch_alc269(struct hda_codec *codec) { struct alc_spec *spec; - int board_config, coef; - int err; + int err = 0; spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) @@ -4682,72 +5064,41 @@ static int patch_alc269(struct hda_codec *codec) alc_auto_parse_customize_define(codec); + err = alc_codec_rename_from_preset(codec); + if (err < 0) + goto error; + if (codec->vendor_id == 0x10ec0269) { spec->codec_variant = ALC269_TYPE_ALC269VA; - coef = alc_read_coef_idx(codec, 0); - if ((coef & 0x00f0) == 0x0010) { + switch (alc_get_coef0(codec) & 0x00f0) { + case 0x0010: if (codec->bus->pci->subsystem_vendor == 0x1025 && - spec->cdefine.platform_type == 1) { - alc_codec_rename(codec, "ALC271X"); - } else if ((coef & 0xf000) == 0x2000) { - alc_codec_rename(codec, "ALC259"); - } else if ((coef & 0xf000) == 0x3000) { - alc_codec_rename(codec, "ALC258"); - } else if ((coef & 0xfff0) == 0x3010) { - alc_codec_rename(codec, "ALC277"); - } else { - alc_codec_rename(codec, "ALC269VB"); - } + spec->cdefine.platform_type == 1) + err = alc_codec_rename(codec, "ALC271X"); spec->codec_variant = ALC269_TYPE_ALC269VB; - } else if ((coef & 0x00f0) == 0x0020) { - if (coef == 0xa023) - alc_codec_rename(codec, "ALC259"); - else if (coef == 0x6023) - alc_codec_rename(codec, "ALC281X"); - else if (codec->bus->pci->subsystem_vendor == 0x17aa && - codec->bus->pci->subsystem_device == 0x21f3) - alc_codec_rename(codec, "ALC3202"); - else - alc_codec_rename(codec, "ALC269VC"); + break; + case 0x0020: + if (codec->bus->pci->subsystem_vendor == 0x17aa && + codec->bus->pci->subsystem_device == 0x21f3) + err = alc_codec_rename(codec, "ALC3202"); spec->codec_variant = ALC269_TYPE_ALC269VC; - } else + break; + default: alc_fix_pll_init(codec, 0x20, 0x04, 15); + } + if (err < 0) + goto error; alc269_fill_coef(codec); } - board_config = alc_board_config(codec, ALC269_MODEL_LAST, - alc269_models, alc269_cfg_tbl); - - if (board_config < 0) { - printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n", - codec->chip_name); - board_config = ALC_MODEL_AUTO; - } - - if (board_config == ALC_MODEL_AUTO) { - alc_pick_fixup(codec, NULL, alc269_fixup_tbl, alc269_fixups); - alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE); - } + alc_pick_fixup(codec, alc269_fixup_models, + alc269_fixup_tbl, alc269_fixups); + alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE); - if (board_config == ALC_MODEL_AUTO) { - /* automatic parse from the BIOS config */ - err = alc269_parse_auto_config(codec); - if (err < 0) { - alc_free(codec); - return err; - } -#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS - else if (!err) { - printk(KERN_INFO - "hda_codec: Cannot set up configuration " - "from BIOS. Using base mode...\n"); - board_config = ALC269_BASIC; - } -#endif - } - - if (board_config != ALC_MODEL_AUTO) - setup_preset(codec, &alc269_presets[board_config]); + /* automatic parse from the BIOS config */ + err = alc269_parse_auto_config(codec); + if (err < 0) + goto error; if (!spec->no_analog && !spec->adc_nids) { alc_auto_fill_adc_caps(codec); @@ -4760,10 +5111,8 @@ static int patch_alc269(struct hda_codec *codec) if (!spec->no_analog && has_cdefine_beep(codec)) { err = snd_hda_attach_beep_device(codec, 0x1); - if (err < 0) { - alc_free(codec); - return err; - } + if (err < 0) + goto error; set_beep_amp(spec, 0x0b, 0x04, HDA_INPUT); } @@ -4775,8 +5124,7 @@ static int patch_alc269(struct hda_codec *codec) #ifdef CONFIG_PM codec->patch_ops.resume = alc269_resume; #endif - if (board_config == ALC_MODEL_AUTO) - spec->init_hook = alc_auto_init_std; + spec->init_hook = alc_auto_init_std; spec->shutup = alc269_shutup; alc_init_jacks(codec); @@ -4788,6 +5136,10 @@ static int patch_alc269(struct hda_codec *codec) #endif return 0; + + error: + alc_free(codec); + return err; } /* @@ -4835,14 +5187,9 @@ static const struct snd_pci_quirk alc861_fixup_tbl[] = { /* */ -#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS -#include "alc861_quirks.c" -#endif - static int patch_alc861(struct hda_codec *codec) { struct alc_spec *spec; - int board_config; int err; spec = kzalloc(sizeof(*spec), GFP_KERNEL); @@ -4853,39 +5200,13 @@ static int patch_alc861(struct hda_codec *codec) spec->mixer_nid = 0x15; - board_config = alc_board_config(codec, ALC861_MODEL_LAST, - alc861_models, alc861_cfg_tbl); - - if (board_config < 0) { - printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n", - codec->chip_name); - board_config = ALC_MODEL_AUTO; - } - - if (board_config == ALC_MODEL_AUTO) { - alc_pick_fixup(codec, NULL, alc861_fixup_tbl, alc861_fixups); - alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE); - } - - if (board_config == ALC_MODEL_AUTO) { - /* automatic parse from the BIOS config */ - err = alc861_parse_auto_config(codec); - if (err < 0) { - alc_free(codec); - return err; - } -#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS - else if (!err) { - printk(KERN_INFO - "hda_codec: Cannot set up configuration " - "from BIOS. Using base mode...\n"); - board_config = ALC861_3ST_DIG; - } -#endif - } + alc_pick_fixup(codec, NULL, alc861_fixup_tbl, alc861_fixups); + alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE); - if (board_config != ALC_MODEL_AUTO) - setup_preset(codec, &alc861_presets[board_config]); + /* automatic parse from the BIOS config */ + err = alc861_parse_auto_config(codec); + if (err < 0) + goto error; if (!spec->no_analog && !spec->adc_nids) { alc_auto_fill_adc_caps(codec); @@ -4898,10 +5219,8 @@ static int patch_alc861(struct hda_codec *codec) if (!spec->no_analog) { err = snd_hda_attach_beep_device(codec, 0x23); - if (err < 0) { - alc_free(codec); - return err; - } + if (err < 0) + goto error; set_beep_amp(spec, 0x23, 0, HDA_OUTPUT); } @@ -4910,18 +5229,18 @@ static int patch_alc861(struct hda_codec *codec) alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); codec->patch_ops = alc_patch_ops; - if (board_config == ALC_MODEL_AUTO) { - spec->init_hook = alc_auto_init_std; -#ifdef CONFIG_SND_HDA_POWER_SAVE - spec->power_hook = alc_power_eapd; -#endif - } + spec->init_hook = alc_auto_init_std; #ifdef CONFIG_SND_HDA_POWER_SAVE + spec->power_hook = alc_power_eapd; if (!spec->loopback.amplist) spec->loopback.amplist = alc861_loopbacks; #endif return 0; + + error: + alc_free(codec); + return err; } /* @@ -4943,24 +5262,41 @@ static int alc861vd_parse_auto_config(struct hda_codec *codec) } enum { - ALC660VD_FIX_ASUS_GPIO1 + ALC660VD_FIX_ASUS_GPIO1, + ALC861VD_FIX_DALLAS, }; -/* reset GPIO1 */ +/* exclude VREF80 */ +static void alc861vd_fixup_dallas(struct hda_codec *codec, + const struct alc_fixup *fix, int action) +{ + if (action == ALC_FIXUP_ACT_PRE_PROBE) { + snd_hda_override_pin_caps(codec, 0x18, 0x00001714); + snd_hda_override_pin_caps(codec, 0x19, 0x0000171c); + } +} + static const struct alc_fixup alc861vd_fixups[] = { [ALC660VD_FIX_ASUS_GPIO1] = { .type = ALC_FIXUP_VERBS, .v.verbs = (const struct hda_verb[]) { + /* reset GPIO1 */ {0x01, AC_VERB_SET_GPIO_MASK, 0x03}, {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01}, {0x01, AC_VERB_SET_GPIO_DATA, 0x01}, { } } }, + [ALC861VD_FIX_DALLAS] = { + .type = ALC_FIXUP_FUNC, + .v.func = alc861vd_fixup_dallas, + }, }; static const struct snd_pci_quirk alc861vd_fixup_tbl[] = { + SND_PCI_QUIRK(0x103c, 0x30bf, "HP TX1000", ALC861VD_FIX_DALLAS), SND_PCI_QUIRK(0x1043, 0x1339, "ASUS A7-K", ALC660VD_FIX_ASUS_GPIO1), + SND_PCI_QUIRK(0x1179, 0xff31, "Toshiba L30-149", ALC861VD_FIX_DALLAS), {} }; @@ -4972,14 +5308,10 @@ static const struct hda_verb alc660vd_eapd_verbs[] = { /* */ -#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS -#include "alc861vd_quirks.c" -#endif - static int patch_alc861vd(struct hda_codec *codec) { struct alc_spec *spec; - int err, board_config; + int err; spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) @@ -4989,39 +5321,13 @@ static int patch_alc861vd(struct hda_codec *codec) spec->mixer_nid = 0x0b; - board_config = alc_board_config(codec, ALC861VD_MODEL_LAST, - alc861vd_models, alc861vd_cfg_tbl); + alc_pick_fixup(codec, NULL, alc861vd_fixup_tbl, alc861vd_fixups); + alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE); - if (board_config < 0) { - printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n", - codec->chip_name); - board_config = ALC_MODEL_AUTO; - } - - if (board_config == ALC_MODEL_AUTO) { - alc_pick_fixup(codec, NULL, alc861vd_fixup_tbl, alc861vd_fixups); - alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE); - } - - if (board_config == ALC_MODEL_AUTO) { - /* automatic parse from the BIOS config */ - err = alc861vd_parse_auto_config(codec); - if (err < 0) { - alc_free(codec); - return err; - } -#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS - else if (!err) { - printk(KERN_INFO - "hda_codec: Cannot set up configuration " - "from BIOS. Using base mode...\n"); - board_config = ALC861VD_3ST; - } -#endif - } - - if (board_config != ALC_MODEL_AUTO) - setup_preset(codec, &alc861vd_presets[board_config]); + /* automatic parse from the BIOS config */ + err = alc861vd_parse_auto_config(codec); + if (err < 0) + goto error; if (codec->vendor_id == 0x10ec0660) { /* always turn on EAPD */ @@ -5039,10 +5345,8 @@ static int patch_alc861vd(struct hda_codec *codec) if (!spec->no_analog) { err = snd_hda_attach_beep_device(codec, 0x23); - if (err < 0) { - alc_free(codec); - return err; - } + if (err < 0) + goto error; set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); } @@ -5052,8 +5356,7 @@ static int patch_alc861vd(struct hda_codec *codec) codec->patch_ops = alc_patch_ops; - if (board_config == ALC_MODEL_AUTO) - spec->init_hook = alc_auto_init_std; + spec->init_hook = alc_auto_init_std; spec->shutup = alc_eapd_shutup; #ifdef CONFIG_SND_HDA_POWER_SAVE if (!spec->loopback.amplist) @@ -5061,6 +5364,10 @@ static int patch_alc861vd(struct hda_codec *codec) #endif return 0; + + error: + alc_free(codec); + return err; } /* @@ -5118,6 +5425,14 @@ enum { ALC662_FIXUP_CZC_P10T, ALC662_FIXUP_SKU_IGNORE, ALC662_FIXUP_HP_RP5800, + ALC662_FIXUP_ASUS_MODE1, + ALC662_FIXUP_ASUS_MODE2, + ALC662_FIXUP_ASUS_MODE3, + ALC662_FIXUP_ASUS_MODE4, + ALC662_FIXUP_ASUS_MODE5, + ALC662_FIXUP_ASUS_MODE6, + ALC662_FIXUP_ASUS_MODE7, + ALC662_FIXUP_ASUS_MODE8, }; static const struct alc_fixup alc662_fixups[] = { @@ -5159,37 +5474,204 @@ static const struct alc_fixup alc662_fixups[] = { .chained = true, .chain_id = ALC662_FIXUP_SKU_IGNORE }, + [ALC662_FIXUP_ASUS_MODE1] = { + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { + { 0x14, 0x99130110 }, /* speaker */ + { 0x18, 0x01a19c20 }, /* mic */ + { 0x19, 0x99a3092f }, /* int-mic */ + { 0x21, 0x0121401f }, /* HP out */ + { } + }, + .chained = true, + .chain_id = ALC662_FIXUP_SKU_IGNORE + }, + [ALC662_FIXUP_ASUS_MODE2] = { + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { + { 0x14, 0x99130110 }, /* speaker */ + { 0x18, 0x01a19820 }, /* mic */ + { 0x19, 0x99a3092f }, /* int-mic */ + { 0x1b, 0x0121401f }, /* HP out */ + { } + }, + .chained = true, + .chain_id = ALC662_FIXUP_SKU_IGNORE + }, + [ALC662_FIXUP_ASUS_MODE3] = { + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { + { 0x14, 0x99130110 }, /* speaker */ + { 0x15, 0x0121441f }, /* HP */ + { 0x18, 0x01a19840 }, /* mic */ + { 0x19, 0x99a3094f }, /* int-mic */ + { 0x21, 0x01211420 }, /* HP2 */ + { } + }, + .chained = true, + .chain_id = ALC662_FIXUP_SKU_IGNORE + }, + [ALC662_FIXUP_ASUS_MODE4] = { + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { + { 0x14, 0x99130110 }, /* speaker */ + { 0x16, 0x99130111 }, /* speaker */ + { 0x18, 0x01a19840 }, /* mic */ + { 0x19, 0x99a3094f }, /* int-mic */ + { 0x21, 0x0121441f }, /* HP */ + { } + }, + .chained = true, + .chain_id = ALC662_FIXUP_SKU_IGNORE + }, + [ALC662_FIXUP_ASUS_MODE5] = { + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { + { 0x14, 0x99130110 }, /* speaker */ + { 0x15, 0x0121441f }, /* HP */ + { 0x16, 0x99130111 }, /* speaker */ + { 0x18, 0x01a19840 }, /* mic */ + { 0x19, 0x99a3094f }, /* int-mic */ + { } + }, + .chained = true, + .chain_id = ALC662_FIXUP_SKU_IGNORE + }, + [ALC662_FIXUP_ASUS_MODE6] = { + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { + { 0x14, 0x99130110 }, /* speaker */ + { 0x15, 0x01211420 }, /* HP2 */ + { 0x18, 0x01a19840 }, /* mic */ + { 0x19, 0x99a3094f }, /* int-mic */ + { 0x1b, 0x0121441f }, /* HP */ + { } + }, + .chained = true, + .chain_id = ALC662_FIXUP_SKU_IGNORE + }, + [ALC662_FIXUP_ASUS_MODE7] = { + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { + { 0x14, 0x99130110 }, /* speaker */ + { 0x17, 0x99130111 }, /* speaker */ + { 0x18, 0x01a19840 }, /* mic */ + { 0x19, 0x99a3094f }, /* int-mic */ + { 0x1b, 0x01214020 }, /* HP */ + { 0x21, 0x0121401f }, /* HP */ + { } + }, + .chained = true, + .chain_id = ALC662_FIXUP_SKU_IGNORE + }, + [ALC662_FIXUP_ASUS_MODE8] = { + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { + { 0x14, 0x99130110 }, /* speaker */ + { 0x12, 0x99a30970 }, /* int-mic */ + { 0x15, 0x01214020 }, /* HP */ + { 0x17, 0x99130111 }, /* speaker */ + { 0x18, 0x01a19840 }, /* mic */ + { 0x21, 0x0121401f }, /* HP */ + { } + }, + .chained = true, + .chain_id = ALC662_FIXUP_SKU_IGNORE + }, }; static const struct snd_pci_quirk alc662_fixup_tbl[] = { + SND_PCI_QUIRK(0x1019, 0x9087, "ECS", ALC662_FIXUP_ASUS_MODE2), SND_PCI_QUIRK(0x1025, 0x0308, "Acer Aspire 8942G", ALC662_FIXUP_ASPIRE), SND_PCI_QUIRK(0x1025, 0x031c, "Gateway NV79", ALC662_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x1025, 0x038b, "Acer Aspire 8943G", ALC662_FIXUP_ASPIRE), SND_PCI_QUIRK(0x103c, 0x1632, "HP RP5800", ALC662_FIXUP_HP_RP5800), + SND_PCI_QUIRK(0x105b, 0x0cd6, "Foxconn", ALC662_FIXUP_ASUS_MODE2), SND_PCI_QUIRK(0x144d, 0xc051, "Samsung R720", ALC662_FIXUP_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo Ideapad Y550P", ALC662_FIXUP_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Ideapad Y550", ALC662_FIXUP_IDEAPAD), SND_PCI_QUIRK(0x1b35, 0x2206, "CZC P10T", ALC662_FIXUP_CZC_P10T), + +#if 0 + /* Below is a quirk table taken from the old code. + * Basically the device should work as is without the fixup table. + * If BIOS doesn't give a proper info, enable the corresponding + * fixup entry. + */ + SND_PCI_QUIRK(0x1043, 0x1000, "ASUS N50Vm", ALC662_FIXUP_ASUS_MODE1), + SND_PCI_QUIRK(0x1043, 0x1092, "ASUS NB", ALC662_FIXUP_ASUS_MODE3), + SND_PCI_QUIRK(0x1043, 0x1173, "ASUS K73Jn", ALC662_FIXUP_ASUS_MODE1), + SND_PCI_QUIRK(0x1043, 0x11c3, "ASUS M70V", ALC662_FIXUP_ASUS_MODE3), + SND_PCI_QUIRK(0x1043, 0x11d3, "ASUS NB", ALC662_FIXUP_ASUS_MODE1), + SND_PCI_QUIRK(0x1043, 0x11f3, "ASUS NB", ALC662_FIXUP_ASUS_MODE2), + SND_PCI_QUIRK(0x1043, 0x1203, "ASUS NB", ALC662_FIXUP_ASUS_MODE1), + SND_PCI_QUIRK(0x1043, 0x1303, "ASUS G60J", ALC662_FIXUP_ASUS_MODE1), + SND_PCI_QUIRK(0x1043, 0x1333, "ASUS G60Jx", ALC662_FIXUP_ASUS_MODE1), + SND_PCI_QUIRK(0x1043, 0x1339, "ASUS NB", ALC662_FIXUP_ASUS_MODE2), + SND_PCI_QUIRK(0x1043, 0x13e3, "ASUS N71JA", ALC662_FIXUP_ASUS_MODE7), + SND_PCI_QUIRK(0x1043, 0x1463, "ASUS N71", ALC662_FIXUP_ASUS_MODE7), + SND_PCI_QUIRK(0x1043, 0x14d3, "ASUS G72", ALC662_FIXUP_ASUS_MODE8), + SND_PCI_QUIRK(0x1043, 0x1563, "ASUS N90", ALC662_FIXUP_ASUS_MODE3), + SND_PCI_QUIRK(0x1043, 0x15d3, "ASUS N50SF F50SF", ALC662_FIXUP_ASUS_MODE1), + SND_PCI_QUIRK(0x1043, 0x16c3, "ASUS NB", ALC662_FIXUP_ASUS_MODE2), + SND_PCI_QUIRK(0x1043, 0x16f3, "ASUS K40C K50C", ALC662_FIXUP_ASUS_MODE2), + SND_PCI_QUIRK(0x1043, 0x1733, "ASUS N81De", ALC662_FIXUP_ASUS_MODE1), + SND_PCI_QUIRK(0x1043, 0x1753, "ASUS NB", ALC662_FIXUP_ASUS_MODE2), + SND_PCI_QUIRK(0x1043, 0x1763, "ASUS NB", ALC662_FIXUP_ASUS_MODE6), + SND_PCI_QUIRK(0x1043, 0x1765, "ASUS NB", ALC662_FIXUP_ASUS_MODE6), + SND_PCI_QUIRK(0x1043, 0x1783, "ASUS NB", ALC662_FIXUP_ASUS_MODE2), + SND_PCI_QUIRK(0x1043, 0x1793, "ASUS F50GX", ALC662_FIXUP_ASUS_MODE1), + SND_PCI_QUIRK(0x1043, 0x17b3, "ASUS F70SL", ALC662_FIXUP_ASUS_MODE3), + SND_PCI_QUIRK(0x1043, 0x17f3, "ASUS X58LE", ALC662_FIXUP_ASUS_MODE2), + SND_PCI_QUIRK(0x1043, 0x1813, "ASUS NB", ALC662_FIXUP_ASUS_MODE2), + SND_PCI_QUIRK(0x1043, 0x1823, "ASUS NB", ALC662_FIXUP_ASUS_MODE5), + SND_PCI_QUIRK(0x1043, 0x1833, "ASUS NB", ALC662_FIXUP_ASUS_MODE6), + SND_PCI_QUIRK(0x1043, 0x1843, "ASUS NB", ALC662_FIXUP_ASUS_MODE2), + SND_PCI_QUIRK(0x1043, 0x1853, "ASUS F50Z", ALC662_FIXUP_ASUS_MODE1), + SND_PCI_QUIRK(0x1043, 0x1864, "ASUS NB", ALC662_FIXUP_ASUS_MODE2), + SND_PCI_QUIRK(0x1043, 0x1876, "ASUS NB", ALC662_FIXUP_ASUS_MODE2), + SND_PCI_QUIRK(0x1043, 0x1893, "ASUS M50Vm", ALC662_FIXUP_ASUS_MODE3), + SND_PCI_QUIRK(0x1043, 0x1894, "ASUS X55", ALC662_FIXUP_ASUS_MODE3), + SND_PCI_QUIRK(0x1043, 0x18b3, "ASUS N80Vc", ALC662_FIXUP_ASUS_MODE1), + SND_PCI_QUIRK(0x1043, 0x18c3, "ASUS VX5", ALC662_FIXUP_ASUS_MODE1), + SND_PCI_QUIRK(0x1043, 0x18d3, "ASUS N81Te", ALC662_FIXUP_ASUS_MODE1), + SND_PCI_QUIRK(0x1043, 0x18f3, "ASUS N505Tp", ALC662_FIXUP_ASUS_MODE1), + SND_PCI_QUIRK(0x1043, 0x1903, "ASUS F5GL", ALC662_FIXUP_ASUS_MODE1), + SND_PCI_QUIRK(0x1043, 0x1913, "ASUS NB", ALC662_FIXUP_ASUS_MODE2), + SND_PCI_QUIRK(0x1043, 0x1933, "ASUS F80Q", ALC662_FIXUP_ASUS_MODE2), + SND_PCI_QUIRK(0x1043, 0x1943, "ASUS Vx3V", ALC662_FIXUP_ASUS_MODE1), + SND_PCI_QUIRK(0x1043, 0x1953, "ASUS NB", ALC662_FIXUP_ASUS_MODE1), + SND_PCI_QUIRK(0x1043, 0x1963, "ASUS X71C", ALC662_FIXUP_ASUS_MODE3), + SND_PCI_QUIRK(0x1043, 0x1983, "ASUS N5051A", ALC662_FIXUP_ASUS_MODE1), + SND_PCI_QUIRK(0x1043, 0x1993, "ASUS N20", ALC662_FIXUP_ASUS_MODE1), + SND_PCI_QUIRK(0x1043, 0x19b3, "ASUS F7Z", ALC662_FIXUP_ASUS_MODE1), + SND_PCI_QUIRK(0x1043, 0x19c3, "ASUS F5Z/F6x", ALC662_FIXUP_ASUS_MODE2), + SND_PCI_QUIRK(0x1043, 0x19e3, "ASUS NB", ALC662_FIXUP_ASUS_MODE1), + SND_PCI_QUIRK(0x1043, 0x19f3, "ASUS NB", ALC662_FIXUP_ASUS_MODE4), +#endif {} }; static const struct alc_model_fixup alc662_fixup_models[] = { {.id = ALC272_FIXUP_MARIO, .name = "mario"}, + {.id = ALC662_FIXUP_ASUS_MODE1, .name = "asus-mode1"}, + {.id = ALC662_FIXUP_ASUS_MODE2, .name = "asus-mode2"}, + {.id = ALC662_FIXUP_ASUS_MODE3, .name = "asus-mode3"}, + {.id = ALC662_FIXUP_ASUS_MODE4, .name = "asus-mode4"}, + {.id = ALC662_FIXUP_ASUS_MODE5, .name = "asus-mode5"}, + {.id = ALC662_FIXUP_ASUS_MODE6, .name = "asus-mode6"}, + {.id = ALC662_FIXUP_ASUS_MODE7, .name = "asus-mode7"}, + {.id = ALC662_FIXUP_ASUS_MODE8, .name = "asus-mode8"}, {} }; /* */ -#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS -#include "alc662_quirks.c" -#endif - static int patch_alc662(struct hda_codec *codec) { struct alc_spec *spec; - int err, board_config; - int coef; + int err = 0; spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (!spec) @@ -5199,50 +5681,31 @@ static int patch_alc662(struct hda_codec *codec) spec->mixer_nid = 0x0b; + /* handle multiple HPs as is */ + spec->parse_flags = HDA_PINCFG_NO_HP_FIXUP; + alc_auto_parse_customize_define(codec); alc_fix_pll_init(codec, 0x20, 0x04, 15); - coef = alc_read_coef_idx(codec, 0); - if (coef == 0x8020 || coef == 0x8011) - alc_codec_rename(codec, "ALC661"); - else if (coef & (1 << 14) && - codec->bus->pci->subsystem_vendor == 0x1025 && - spec->cdefine.platform_type == 1) - alc_codec_rename(codec, "ALC272X"); - else if (coef == 0x4011) - alc_codec_rename(codec, "ALC656"); - - board_config = alc_board_config(codec, ALC662_MODEL_LAST, - alc662_models, alc662_cfg_tbl); - if (board_config < 0) { - printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n", - codec->chip_name); - board_config = ALC_MODEL_AUTO; - } + err = alc_codec_rename_from_preset(codec); + if (err < 0) + goto error; - if (board_config == ALC_MODEL_AUTO) { - alc_pick_fixup(codec, alc662_fixup_models, - alc662_fixup_tbl, alc662_fixups); - alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE); - /* automatic parse from the BIOS config */ - err = alc662_parse_auto_config(codec); - if (err < 0) { - alc_free(codec); - return err; - } -#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS - else if (!err) { - printk(KERN_INFO - "hda_codec: Cannot set up configuration " - "from BIOS. Using base mode...\n"); - board_config = ALC662_3ST_2ch_DIG; - } -#endif + if ((alc_get_coef0(codec) & (1 << 14)) && + codec->bus->pci->subsystem_vendor == 0x1025 && + spec->cdefine.platform_type == 1) { + if (alc_codec_rename(codec, "ALC272X") < 0) + goto error; } - if (board_config != ALC_MODEL_AUTO) - setup_preset(codec, &alc662_presets[board_config]); + alc_pick_fixup(codec, alc662_fixup_models, + alc662_fixup_tbl, alc662_fixups); + alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE); + /* automatic parse from the BIOS config */ + err = alc662_parse_auto_config(codec); + if (err < 0) + goto error; if (!spec->no_analog && !spec->adc_nids) { alc_auto_fill_adc_caps(codec); @@ -5255,10 +5718,8 @@ static int patch_alc662(struct hda_codec *codec) if (!spec->no_analog && has_cdefine_beep(codec)) { err = snd_hda_attach_beep_device(codec, 0x1); - if (err < 0) { - alc_free(codec); - return err; - } + if (err < 0) + goto error; switch (codec->vendor_id) { case 0x10ec0662: set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); @@ -5278,8 +5739,7 @@ static int patch_alc662(struct hda_codec *codec) alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); codec->patch_ops = alc_patch_ops; - if (board_config == ALC_MODEL_AUTO) - spec->init_hook = alc_auto_init_std; + spec->init_hook = alc_auto_init_std; spec->shutup = alc_eapd_shutup; alc_init_jacks(codec); @@ -5290,32 +5750,10 @@ static int patch_alc662(struct hda_codec *codec) #endif return 0; -} -static int patch_alc888(struct hda_codec *codec) -{ - if ((alc_read_coef_idx(codec, 0) & 0x00f0)==0x0030){ - kfree(codec->chip_name); - if (codec->vendor_id == 0x10ec0887) - codec->chip_name = kstrdup("ALC887-VD", GFP_KERNEL); - else - codec->chip_name = kstrdup("ALC888-VD", GFP_KERNEL); - if (!codec->chip_name) { - alc_free(codec); - return -ENOMEM; - } - return patch_alc662(codec); - } - return patch_alc882(codec); -} - -static int patch_alc899(struct hda_codec *codec) -{ - if ((alc_read_coef_idx(codec, 0) & 0x2000) != 0x2000) { - kfree(codec->chip_name); - codec->chip_name = kstrdup("ALC898", GFP_KERNEL); - } - return patch_alc882(codec); + error: + alc_free(codec); + return err; } /* @@ -5329,14 +5767,9 @@ static int alc680_parse_auto_config(struct hda_codec *codec) /* */ -#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS -#include "alc680_quirks.c" -#endif - static int patch_alc680(struct hda_codec *codec) { struct alc_spec *spec; - int board_config; int err; spec = kzalloc(sizeof(*spec), GFP_KERNEL); @@ -5347,43 +5780,11 @@ static int patch_alc680(struct hda_codec *codec) /* ALC680 has no aa-loopback mixer */ - board_config = alc_board_config(codec, ALC680_MODEL_LAST, - alc680_models, alc680_cfg_tbl); - - if (board_config < 0) { - printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n", - codec->chip_name); - board_config = ALC_MODEL_AUTO; - } - - if (board_config == ALC_MODEL_AUTO) { - /* automatic parse from the BIOS config */ - err = alc680_parse_auto_config(codec); - if (err < 0) { - alc_free(codec); - return err; - } -#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS - else if (!err) { - printk(KERN_INFO - "hda_codec: Cannot set up configuration " - "from BIOS. Using base mode...\n"); - board_config = ALC680_BASE; - } -#endif - } - - if (board_config != ALC_MODEL_AUTO) { - setup_preset(codec, &alc680_presets[board_config]); -#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS - spec->stream_analog_capture = &alc680_pcm_analog_auto_capture; -#endif - } - - if (!spec->no_analog && !spec->adc_nids) { - alc_auto_fill_adc_caps(codec); - alc_rebuild_imux_for_auto_mic(codec); - alc_remove_invalid_adc_nids(codec); + /* automatic parse from the BIOS config */ + err = alc680_parse_auto_config(codec); + if (err < 0) { + alc_free(codec); + return err; } if (!spec->no_analog && !spec->cap_mixer) @@ -5392,8 +5793,7 @@ static int patch_alc680(struct hda_codec *codec) spec->vmaster_nid = 0x02; codec->patch_ops = alc_patch_ops; - if (board_config == ALC_MODEL_AUTO) - spec->init_hook = alc_auto_init_std; + spec->init_hook = alc_auto_init_std; return 0; } @@ -5421,6 +5821,8 @@ static const struct hda_codec_preset snd_hda_preset_realtek[] = { .patch = patch_alc882 }, { .id = 0x10ec0662, .rev = 0x100101, .name = "ALC662 rev1", .patch = patch_alc662 }, + { .id = 0x10ec0662, .rev = 0x100300, .name = "ALC662 rev3", + .patch = patch_alc662 }, { .id = 0x10ec0663, .name = "ALC663", .patch = patch_alc662 }, { .id = 0x10ec0665, .name = "ALC665", .patch = patch_alc662 }, { .id = 0x10ec0670, .name = "ALC670", .patch = patch_alc662 }, @@ -5433,13 +5835,13 @@ static const struct hda_codec_preset snd_hda_preset_realtek[] = { { .id = 0x10ec0885, .rev = 0x100103, .name = "ALC889A", .patch = patch_alc882 }, { .id = 0x10ec0885, .name = "ALC885", .patch = patch_alc882 }, - { .id = 0x10ec0887, .name = "ALC887", .patch = patch_alc888 }, + { .id = 0x10ec0887, .name = "ALC887", .patch = patch_alc882 }, { .id = 0x10ec0888, .rev = 0x100101, .name = "ALC1200", .patch = patch_alc882 }, - { .id = 0x10ec0888, .name = "ALC888", .patch = patch_alc888 }, + { .id = 0x10ec0888, .name = "ALC888", .patch = patch_alc882 }, { .id = 0x10ec0889, .name = "ALC889", .patch = patch_alc882 }, { .id = 0x10ec0892, .name = "ALC892", .patch = patch_alc662 }, - { .id = 0x10ec0899, .name = "ALC899", .patch = patch_alc899 }, + { .id = 0x10ec0899, .name = "ALC898", .patch = patch_alc882 }, {} /* terminator */ }; diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 987e3cf71a0b..59a52a430f24 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2972,8 +2972,9 @@ static int check_all_dac_nids(struct sigmatel_spec *spec, hda_nid_t nid) static hda_nid_t get_unassigned_dac(struct hda_codec *codec, hda_nid_t nid) { struct sigmatel_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; int j, conn_len; - hda_nid_t conn[HDA_MAX_CONNECTIONS]; + hda_nid_t conn[HDA_MAX_CONNECTIONS], fallback_dac; unsigned int wcaps, wtype; conn_len = snd_hda_get_connections(codec, nid, conn, @@ -3001,10 +3002,21 @@ static hda_nid_t get_unassigned_dac(struct hda_codec *codec, hda_nid_t nid) return conn[j]; } } - /* if all DACs are already assigned, connect to the primary DAC */ + + /* if all DACs are already assigned, connect to the primary DAC, + unless we're assigning a secondary headphone */ + fallback_dac = spec->multiout.dac_nids[0]; + if (spec->multiout.hp_nid) { + for (j = 0; j < cfg->hp_outs; j++) + if (cfg->hp_pins[j] == nid) { + fallback_dac = spec->multiout.hp_nid; + break; + } + } + if (conn_len > 1) { for (j = 0; j < conn_len; j++) { - if (conn[j] == spec->multiout.dac_nids[0]) { + if (conn[j] == fallback_dac) { snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, j); break; @@ -4130,22 +4142,14 @@ static int stac92xx_add_jack(struct hda_codec *codec, #ifdef CONFIG_SND_HDA_INPUT_JACK int def_conf = snd_hda_codec_get_pincfg(codec, nid); int connectivity = get_defcfg_connect(def_conf); - char name[32]; - int err; if (connectivity && connectivity != AC_JACK_PORT_FIXED) return 0; - snprintf(name, sizeof(name), "%s at %s %s Jack", - snd_hda_get_jack_type(def_conf), - snd_hda_get_jack_connectivity(def_conf), - snd_hda_get_jack_location(def_conf)); - - err = snd_hda_input_jack_add(codec, nid, type, name); - if (err < 0) - return err; -#endif /* CONFIG_SND_HDA_INPUT_JACK */ + return snd_hda_input_jack_add(codec, nid, type, NULL); +#else return 0; +#endif /* CONFIG_SND_HDA_INPUT_JACK */ } static int stac_add_event(struct sigmatel_spec *spec, hda_nid_t nid, @@ -5585,9 +5589,7 @@ static void stac92hd8x_fill_auto_spec(struct hda_codec *codec) static int patch_stac92hd83xxx(struct hda_codec *codec) { struct sigmatel_spec *spec; - hda_nid_t conn[STAC92HD83_DAC_COUNT + 1]; int err; - int num_dacs; spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) @@ -5689,22 +5691,6 @@ again: return err; } - /* docking output support */ - num_dacs = snd_hda_get_connections(codec, 0xF, - conn, STAC92HD83_DAC_COUNT + 1) - 1; - /* skip non-DAC connections */ - while (num_dacs >= 0 && - (get_wcaps_type(get_wcaps(codec, conn[num_dacs])) - != AC_WID_AUD_OUT)) - num_dacs--; - /* set port E and F to select the last DAC */ - if (num_dacs >= 0) { - snd_hda_codec_write_cache(codec, 0xE, 0, - AC_VERB_SET_CONNECT_SEL, num_dacs); - snd_hda_codec_write_cache(codec, 0xF, 0, - AC_VERB_SET_CONNECT_SEL, num_dacs); - } - codec->proc_widget_hook = stac92hd_proc_hook; return 0; diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 4ebfbd874c9a..417d62ad3b96 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -1506,39 +1506,49 @@ static int via_build_pcms(struct hda_codec *codec) struct via_spec *spec = codec->spec; struct hda_pcm *info = spec->pcm_rec; - codec->num_pcms = 1; + codec->num_pcms = 0; codec->pcm_info = info; - snprintf(spec->stream_name_analog, sizeof(spec->stream_name_analog), - "%s Analog", codec->chip_name); - info->name = spec->stream_name_analog; + if (spec->multiout.num_dacs || spec->num_adc_nids) { + snprintf(spec->stream_name_analog, + sizeof(spec->stream_name_analog), + "%s Analog", codec->chip_name); + info->name = spec->stream_name_analog; - if (!spec->stream_analog_playback) - spec->stream_analog_playback = &via_pcm_analog_playback; - info->stream[SNDRV_PCM_STREAM_PLAYBACK] = - *spec->stream_analog_playback; - info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = - spec->multiout.dac_nids[0]; - info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max = - spec->multiout.max_channels; + if (spec->multiout.num_dacs) { + if (!spec->stream_analog_playback) + spec->stream_analog_playback = + &via_pcm_analog_playback; + info->stream[SNDRV_PCM_STREAM_PLAYBACK] = + *spec->stream_analog_playback; + info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = + spec->multiout.dac_nids[0]; + info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max = + spec->multiout.max_channels; + } - if (!spec->stream_analog_capture) { - if (spec->dyn_adc_switch) - spec->stream_analog_capture = - &via_pcm_dyn_adc_analog_capture; - else - spec->stream_analog_capture = &via_pcm_analog_capture; + if (!spec->stream_analog_capture) { + if (spec->dyn_adc_switch) + spec->stream_analog_capture = + &via_pcm_dyn_adc_analog_capture; + else + spec->stream_analog_capture = + &via_pcm_analog_capture; + } + if (spec->num_adc_nids) { + info->stream[SNDRV_PCM_STREAM_CAPTURE] = + *spec->stream_analog_capture; + info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = + spec->adc_nids[0]; + if (!spec->dyn_adc_switch) + info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams = + spec->num_adc_nids; + } + codec->num_pcms++; + info++; } - info->stream[SNDRV_PCM_STREAM_CAPTURE] = - *spec->stream_analog_capture; - info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adc_nids[0]; - if (!spec->dyn_adc_switch) - info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams = - spec->num_adc_nids; if (spec->multiout.dig_out_nid || spec->dig_in_nid) { - codec->num_pcms++; - info++; snprintf(spec->stream_name_digital, sizeof(spec->stream_name_digital), "%s Digital", codec->chip_name); @@ -1562,17 +1572,19 @@ static int via_build_pcms(struct hda_codec *codec) info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->dig_in_nid; } + codec->num_pcms++; + info++; } if (spec->hp_dac_nid) { - codec->num_pcms++; - info++; snprintf(spec->stream_name_hp, sizeof(spec->stream_name_hp), "%s HP", codec->chip_name); info->name = spec->stream_name_hp; info->stream[SNDRV_PCM_STREAM_PLAYBACK] = via_pcm_hp_playback; info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->hp_dac_nid; + codec->num_pcms++; + info++; } return 0; } diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index 0ccc0eb75775..8531b983f3af 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -2748,8 +2748,9 @@ static int __devinit snd_ice1712_probe(struct pci_dev *pci, if (!c->no_mpu401) { err = snd_mpu401_uart_new(card, 0, MPU401_HW_ICE1712, ICEREG(ice, MPU1_CTRL), - (c->mpu401_1_info_flags | MPU401_INFO_INTEGRATED), - ice->irq, 0, &ice->rmidi[0]); + c->mpu401_1_info_flags | + MPU401_INFO_INTEGRATED | MPU401_INFO_IRQ_HOOK, + -1, &ice->rmidi[0]); if (err < 0) { snd_card_free(card); return err; @@ -2764,8 +2765,9 @@ static int __devinit snd_ice1712_probe(struct pci_dev *pci, /* 2nd port used */ err = snd_mpu401_uart_new(card, 1, MPU401_HW_ICE1712, ICEREG(ice, MPU2_CTRL), - (c->mpu401_2_info_flags | MPU401_INFO_INTEGRATED), - ice->irq, 0, &ice->rmidi[1]); + c->mpu401_2_info_flags | + MPU401_INFO_INTEGRATED | MPU401_INFO_IRQ_HOOK, + -1, &ice->rmidi[1]); if (err < 0) { snd_card_free(card); diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c index 0378126e6272..2fd4bf2d6653 100644 --- a/sound/pci/maestro3.c +++ b/sound/pci/maestro3.c @@ -2820,8 +2820,8 @@ snd_m3_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) /* TODO enable MIDI IRQ and I/O */ err = snd_mpu401_uart_new(chip->card, 0, MPU401_HW_MPU401, chip->iobase + MPU401_DATA_PORT, - MPU401_INFO_INTEGRATED, - chip->irq, 0, &chip->rmidi); + MPU401_INFO_INTEGRATED | MPU401_INFO_IRQ_HOOK, + -1, &chip->rmidi); if (err < 0) printk(KERN_WARNING "maestro3: no MIDI support.\n"); #endif diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c index 82311fcb86f6..53e5508abcbf 100644 --- a/sound/pci/oxygen/oxygen_lib.c +++ b/sound/pci/oxygen/oxygen_lib.c @@ -678,15 +678,15 @@ int oxygen_pci_probe(struct pci_dev *pci, int index, char *id, goto err_card; if (chip->model.device_config & (MIDI_OUTPUT | MIDI_INPUT)) { - unsigned int info_flags = MPU401_INFO_INTEGRATED; + unsigned int info_flags = + MPU401_INFO_INTEGRATED | MPU401_INFO_IRQ_HOOK; if (chip->model.device_config & MIDI_OUTPUT) info_flags |= MPU401_INFO_OUTPUT; if (chip->model.device_config & MIDI_INPUT) info_flags |= MPU401_INFO_INPUT; err = snd_mpu401_uart_new(card, 0, MPU401_HW_CMIPCI, chip->addr + OXYGEN_MPU401, - info_flags, 0, 0, - &chip->midi); + info_flags, -1, &chip->midi); if (err < 0) goto err_card; } diff --git a/sound/pci/oxygen/xonar_pcm179x.c b/sound/pci/oxygen/xonar_pcm179x.c index 32d096c98f5b..8433aa7c3d75 100644 --- a/sound/pci/oxygen/xonar_pcm179x.c +++ b/sound/pci/oxygen/xonar_pcm179x.c @@ -1074,6 +1074,7 @@ static const struct oxygen_model model_xonar_st = { .device_config = PLAYBACK_0_TO_I2S | PLAYBACK_1_TO_SPDIF | CAPTURE_0_FROM_I2S_2 | + CAPTURE_1_FROM_SPDIF | AC97_FMIC_SWITCH, .dac_channels_pcm = 2, .dac_channels_mixer = 2, diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c index e34ae14908b3..88cc776aa38b 100644 --- a/sound/pci/riptide/riptide.c +++ b/sound/pci/riptide/riptide.c @@ -2109,7 +2109,7 @@ snd_card_riptide_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) val = mpu_port[dev]; pci_write_config_word(chip->pci, PCI_EXT_MPU_Base, val); err = snd_mpu401_uart_new(card, 0, MPU401_HW_RIPTIDE, - val, 0, chip->irq, 0, + val, MPU401_INFO_IRQ_HOOK, -1, &chip->rmidi); if (err < 0) snd_printk(KERN_WARNING diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 493e3946756f..6e2f7ef7ddb1 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -1241,10 +1241,30 @@ static int hdspm_external_sample_rate(struct hdspm *hdspm) return rate; } +/* return latency in samples per period */ +static int hdspm_get_latency(struct hdspm *hdspm) +{ + int n; + + n = hdspm_decode_latency(hdspm->control_register); + + /* Special case for new RME cards with 32 samples period size. + * The three latency bits in the control register + * (HDSP_LatencyMask) encode latency values of 64 samples as + * 0, 128 samples as 1 ... 4096 samples as 6. For old cards, 7 + * denotes 8192 samples, but on new cards like RayDAT or AIO, + * it corresponds to 32 samples. + */ + if ((7 == n) && (RayDAT == hdspm->io_type || AIO == hdspm->io_type)) + n = -1; + + return 1 << (n + 6); +} + /* Latency function */ static inline void hdspm_compute_period_size(struct hdspm *hdspm) { - hdspm->period_bytes = 1 << ((hdspm_decode_latency(hdspm->control_register) + 8)); + hdspm->period_bytes = 4 * hdspm_get_latency(hdspm); } @@ -1303,12 +1323,27 @@ static int hdspm_set_interrupt_interval(struct hdspm *s, unsigned int frames) spin_lock_irq(&s->lock); - frames >>= 7; - n = 0; - while (frames) { - n++; - frames >>= 1; + if (32 == frames) { + /* Special case for new RME cards like RayDAT/AIO which + * support period sizes of 32 samples. Since latency is + * encoded in the three bits of HDSP_LatencyMask, we can only + * have values from 0 .. 7. While 0 still means 64 samples and + * 6 represents 4096 samples on all cards, 7 represents 8192 + * on older cards and 32 samples on new cards. + * + * In other words, period size in samples is calculated by + * 2^(n+6) with n ranging from 0 .. 7. + */ + n = 7; + } else { + frames >>= 7; + n = 0; + while (frames) { + n++; + frames >>= 1; + } } + s->control_register &= ~HDSPM_LatencyMask; s->control_register |= hdspm_encode_latency(n); @@ -4801,8 +4836,7 @@ snd_hdspm_proc_read_madi(struct snd_info_entry * entry, snd_iprintf(buffer, "--- Settings ---\n"); - x = 1 << (6 + hdspm_decode_latency(hdspm->control_register & - HDSPM_LatencyMask)); + x = hdspm_get_latency(hdspm); snd_iprintf(buffer, "Size (Latency): %d samples (2 periods of %lu bytes)\n", @@ -4965,8 +4999,7 @@ snd_hdspm_proc_read_aes32(struct snd_info_entry * entry, snd_iprintf(buffer, "--- Settings ---\n"); - x = 1 << (6 + hdspm_decode_latency(hdspm->control_register & - HDSPM_LatencyMask)); + x = hdspm_get_latency(hdspm); snd_iprintf(buffer, "Size (Latency): %d samples (2 periods of %lu bytes)\n", @@ -5672,19 +5705,6 @@ static int snd_hdspm_prepare(struct snd_pcm_substream *substream) return 0; } -static unsigned int period_sizes_old[] = { - 64, 128, 256, 512, 1024, 2048, 4096 -}; - -static unsigned int period_sizes_new[] = { - 32, 64, 128, 256, 512, 1024, 2048, 4096 -}; - -/* RayDAT and AIO always have a buffer of 16384 samples per channel */ -static unsigned int raydat_aio_buffer_sizes[] = { - 16384 -}; - static struct snd_pcm_hardware snd_hdspm_playback_subinfo = { .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | @@ -5703,8 +5723,8 @@ static struct snd_pcm_hardware snd_hdspm_playback_subinfo = { .channels_max = HDSPM_MAX_CHANNELS, .buffer_bytes_max = HDSPM_CHANNEL_BUFFER_BYTES * HDSPM_MAX_CHANNELS, - .period_bytes_min = (64 * 4), - .period_bytes_max = (4096 * 4) * HDSPM_MAX_CHANNELS, + .period_bytes_min = (32 * 4), + .period_bytes_max = (8192 * 4) * HDSPM_MAX_CHANNELS, .periods_min = 2, .periods_max = 512, .fifo_size = 0 @@ -5728,31 +5748,13 @@ static struct snd_pcm_hardware snd_hdspm_capture_subinfo = { .channels_max = HDSPM_MAX_CHANNELS, .buffer_bytes_max = HDSPM_CHANNEL_BUFFER_BYTES * HDSPM_MAX_CHANNELS, - .period_bytes_min = (64 * 4), - .period_bytes_max = (4096 * 4) * HDSPM_MAX_CHANNELS, + .period_bytes_min = (32 * 4), + .period_bytes_max = (8192 * 4) * HDSPM_MAX_CHANNELS, .periods_min = 2, .periods_max = 512, .fifo_size = 0 }; -static struct snd_pcm_hw_constraint_list hw_constraints_period_sizes_old = { - .count = ARRAY_SIZE(period_sizes_old), - .list = period_sizes_old, - .mask = 0 -}; - -static struct snd_pcm_hw_constraint_list hw_constraints_period_sizes_new = { - .count = ARRAY_SIZE(period_sizes_new), - .list = period_sizes_new, - .mask = 0 -}; - -static struct snd_pcm_hw_constraint_list hw_constraints_raydat_io_buffer = { - .count = ARRAY_SIZE(raydat_aio_buffer_sizes), - .list = raydat_aio_buffer_sizes, - .mask = 0 -}; - static int snd_hdspm_hw_rule_in_channels_rate(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { @@ -5953,26 +5955,29 @@ static int snd_hdspm_playback_open(struct snd_pcm_substream *substream) spin_unlock_irq(&hdspm->lock); snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24); + snd_pcm_hw_constraint_pow2(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE); switch (hdspm->io_type) { case AIO: case RayDAT: - snd_pcm_hw_constraint_list(runtime, 0, - SNDRV_PCM_HW_PARAM_PERIOD_SIZE, - &hw_constraints_period_sizes_new); - snd_pcm_hw_constraint_list(runtime, 0, - SNDRV_PCM_HW_PARAM_BUFFER_SIZE, - &hw_constraints_raydat_io_buffer); - + snd_pcm_hw_constraint_minmax(runtime, + SNDRV_PCM_HW_PARAM_PERIOD_SIZE, + 32, 4096); + /* RayDAT & AIO have a fixed buffer of 16384 samples per channel */ + snd_pcm_hw_constraint_minmax(runtime, + SNDRV_PCM_HW_PARAM_BUFFER_SIZE, + 16384, 16384); break; default: - snd_pcm_hw_constraint_list(runtime, 0, - SNDRV_PCM_HW_PARAM_PERIOD_SIZE, - &hw_constraints_period_sizes_old); + snd_pcm_hw_constraint_minmax(runtime, + SNDRV_PCM_HW_PARAM_PERIOD_SIZE, + 64, 8192); + break; } if (AES32 == hdspm->io_type) { + runtime->hw.rates |= SNDRV_PCM_RATE_KNOT; snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hdspm_hw_constraints_aes32_sample_rates); } else { @@ -6025,24 +6030,28 @@ static int snd_hdspm_capture_open(struct snd_pcm_substream *substream) spin_unlock_irq(&hdspm->lock); snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24); + snd_pcm_hw_constraint_pow2(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE); + switch (hdspm->io_type) { case AIO: case RayDAT: - snd_pcm_hw_constraint_list(runtime, 0, - SNDRV_PCM_HW_PARAM_PERIOD_SIZE, - &hw_constraints_period_sizes_new); - snd_pcm_hw_constraint_list(runtime, 0, - SNDRV_PCM_HW_PARAM_BUFFER_SIZE, - &hw_constraints_raydat_io_buffer); - break; + snd_pcm_hw_constraint_minmax(runtime, + SNDRV_PCM_HW_PARAM_PERIOD_SIZE, + 32, 4096); + snd_pcm_hw_constraint_minmax(runtime, + SNDRV_PCM_HW_PARAM_BUFFER_SIZE, + 16384, 16384); + break; default: - snd_pcm_hw_constraint_list(runtime, 0, - SNDRV_PCM_HW_PARAM_PERIOD_SIZE, - &hw_constraints_period_sizes_old); + snd_pcm_hw_constraint_minmax(runtime, + SNDRV_PCM_HW_PARAM_PERIOD_SIZE, + 64, 8192); + break; } if (AES32 == hdspm->io_type) { + runtime->hw.rates |= SNDRV_PCM_RATE_KNOT; snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hdspm_hw_constraints_aes32_sample_rates); } else { @@ -6088,7 +6097,7 @@ static inline int copy_u32_le(void __user *dest, void __iomem *src) } static int snd_hdspm_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, - unsigned int cmd, unsigned long __user arg) + unsigned int cmd, unsigned long arg) { void __user *argp = (void __user *)arg; struct hdspm *hdspm = hw->private_data; @@ -6213,11 +6222,13 @@ static int snd_hdspm_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, info.line_out = hdspm_line_out(hdspm); info.passthru = 0; spin_unlock_irq(&hdspm->lock); - if (copy_to_user((void __user *) arg, &info, sizeof(info))) + if (copy_to_user(argp, &info, sizeof(info))) return -EFAULT; break; case SNDRV_HDSPM_IOCTL_GET_STATUS: + memset(&status, 0, sizeof(status)); + status.card_type = hdspm->io_type; status.autosync_source = hdspm_autosync_ref(hdspm); @@ -6250,13 +6261,15 @@ static int snd_hdspm_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, break; } - if (copy_to_user((void __user *) arg, &status, sizeof(status))) + if (copy_to_user(argp, &status, sizeof(status))) return -EFAULT; break; case SNDRV_HDSPM_IOCTL_GET_VERSION: + memset(&hdspm_version, 0, sizeof(hdspm_version)); + hdspm_version.card_type = hdspm->io_type; strncpy(hdspm_version.cardname, hdspm->card_name, sizeof(hdspm_version.cardname)); @@ -6267,13 +6280,13 @@ static int snd_hdspm_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, if (hdspm->tco) hdspm_version.addons |= HDSPM_ADDON_TCO; - if (copy_to_user((void __user *) arg, &hdspm_version, + if (copy_to_user(argp, &hdspm_version, sizeof(hdspm_version))) return -EFAULT; break; case SNDRV_HDSPM_IOCTL_GET_MIXER: - if (copy_from_user(&mixer, (void __user *)arg, sizeof(mixer))) + if (copy_from_user(&mixer, argp, sizeof(mixer))) return -EFAULT; if (copy_to_user((void __user *)mixer.mixer, hdspm->mixer, sizeof(struct hdspm_mixer))) diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c index bcf61524a13f..5ffb20b18786 100644 --- a/sound/pci/sis7019.c +++ b/sound/pci/sis7019.c @@ -1234,7 +1234,7 @@ static int sis_resume(struct pci_dev *pci) goto error; } - if (request_irq(pci->irq, sis_interrupt, IRQF_DISABLED|IRQF_SHARED, + if (request_irq(pci->irq, sis_interrupt, IRQF_SHARED, KBUILD_MODNAME, sis)) { printk(KERN_ERR "sis7019: unable to regain IRQ %d\n", pci->irq); goto error; @@ -1340,7 +1340,7 @@ static int __devinit sis_chip_create(struct snd_card *card, if (rc) goto error_out_cleanup; - if (request_irq(pci->irq, sis_interrupt, IRQF_DISABLED|IRQF_SHARED, + if (request_irq(pci->irq, sis_interrupt, IRQF_SHARED, KBUILD_MODNAME, sis)) { printk(KERN_ERR "unable to allocate irq %d\n", sis->irq); goto error_out_cleanup; diff --git a/sound/pci/sonicvibes.c b/sound/pci/sonicvibes.c index 2571a67b389a..c5008166cf1f 100644 --- a/sound/pci/sonicvibes.c +++ b/sound/pci/sonicvibes.c @@ -1493,9 +1493,10 @@ static int __devinit snd_sonic_probe(struct pci_dev *pci, return err; } if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_SONICVIBES, - sonic->midi_port, MPU401_INFO_INTEGRATED, - sonic->irq, 0, - &midi_uart)) < 0) { + sonic->midi_port, + MPU401_INFO_INTEGRATED | + MPU401_INFO_IRQ_HOOK, + -1, &midi_uart)) < 0) { snd_card_free(card); return err; } diff --git a/sound/pci/trident/trident.c b/sound/pci/trident/trident.c index d8a128f6fc02..5e707effdc7c 100644 --- a/sound/pci/trident/trident.c +++ b/sound/pci/trident/trident.c @@ -148,8 +148,9 @@ static int __devinit snd_trident_probe(struct pci_dev *pci, if (trident->device != TRIDENT_DEVICE_ID_SI7018 && (err = snd_mpu401_uart_new(card, 0, MPU401_HW_TRID4DWAVE, trident->midi_port, - MPU401_INFO_INTEGRATED, - trident->irq, 0, &trident->rmidi)) < 0) { + MPU401_INFO_INTEGRATED | + MPU401_INFO_IRQ_HOOK, + -1, &trident->rmidi)) < 0) { snd_card_free(card); return err; } diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index f03fd620a2a0..c3656fffdb50 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -1175,6 +1175,7 @@ static int snd_via82xx_pcm_open(struct via82xx *chip, struct viadev *viadev, struct snd_pcm_runtime *runtime = substream->runtime; int err; struct via_rate_lock *ratep; + bool use_src = false; runtime->hw = snd_via82xx_hw; @@ -1196,6 +1197,7 @@ static int snd_via82xx_pcm_open(struct via82xx *chip, struct viadev *viadev, SNDRV_PCM_RATE_8000_48000); runtime->hw.rate_min = 8000; runtime->hw.rate_max = 48000; + use_src = true; } else if (! ratep->rate) { int idx = viadev->direction ? AC97_RATES_ADC : AC97_RATES_FRONT_DAC; runtime->hw.rates = chip->ac97->rates[idx]; @@ -1212,6 +1214,12 @@ static int snd_via82xx_pcm_open(struct via82xx *chip, struct viadev *viadev, if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0) return err; + if (use_src) { + err = snd_pcm_hw_rule_noresample(runtime, 48000); + if (err < 0) + return err; + } + runtime->private_data = viadev; viadev->substream = substream; @@ -2068,8 +2076,9 @@ static int __devinit snd_via686_init_misc(struct via82xx *chip) pci_write_config_byte(chip->pci, VIA_PNP_CONTROL, legacy_cfg); if (chip->mpu_res) { if (snd_mpu401_uart_new(chip->card, 0, MPU401_HW_VIA686A, - mpu_port, MPU401_INFO_INTEGRATED, - chip->irq, 0, &chip->rmidi) < 0) { + mpu_port, MPU401_INFO_INTEGRATED | + MPU401_INFO_IRQ_HOOK, -1, + &chip->rmidi) < 0) { printk(KERN_WARNING "unable to initialize MPU-401" " at 0x%lx, skipping\n", mpu_port); legacy &= ~VIA_FUNC_ENABLE_MIDI; diff --git a/sound/pci/ymfpci/ymfpci.c b/sound/pci/ymfpci/ymfpci.c index 511d57653124..3253b04da184 100644 --- a/sound/pci/ymfpci/ymfpci.c +++ b/sound/pci/ymfpci/ymfpci.c @@ -305,8 +305,9 @@ static int __devinit snd_card_ymfpci_probe(struct pci_dev *pci, if (chip->mpu_res) { if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_YMFPCI, mpu_port[dev], - MPU401_INFO_INTEGRATED, - pci->irq, 0, &chip->rawmidi)) < 0) { + MPU401_INFO_INTEGRATED | + MPU401_INFO_IRQ_HOOK, + -1, &chip->rawmidi)) < 0) { printk(KERN_WARNING "ymfpci: cannot initialize MPU401 at 0x%lx, skipping...\n", mpu_port[dev]); legacy_ctrl &= ~YMFPCI_LEGACY_MIEN; /* disable MPU401 irq */ pci_write_config_word(pci, PCIR_DSXG_LEGACY, legacy_ctrl); diff --git a/sound/pci/ymfpci/ymfpci_main.c b/sound/pci/ymfpci/ymfpci_main.c index f3260e658b8a..66ea71b2a70d 100644 --- a/sound/pci/ymfpci/ymfpci_main.c +++ b/sound/pci/ymfpci/ymfpci_main.c @@ -897,6 +897,18 @@ static int snd_ymfpci_playback_open_1(struct snd_pcm_substream *substream) struct snd_ymfpci *chip = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; struct snd_ymfpci_pcm *ypcm; + int err; + + runtime->hw = snd_ymfpci_playback; + /* FIXME? True value is 256/48 = 5.33333 ms */ + err = snd_pcm_hw_constraint_minmax(runtime, + SNDRV_PCM_HW_PARAM_PERIOD_TIME, + 5334, UINT_MAX); + if (err < 0) + return err; + err = snd_pcm_hw_rule_noresample(runtime, 48000); + if (err < 0) + return err; ypcm = kzalloc(sizeof(*ypcm), GFP_KERNEL); if (ypcm == NULL) @@ -904,11 +916,8 @@ static int snd_ymfpci_playback_open_1(struct snd_pcm_substream *substream) ypcm->chip = chip; ypcm->type = PLAYBACK_VOICE; ypcm->substream = substream; - runtime->hw = snd_ymfpci_playback; runtime->private_data = ypcm; runtime->private_free = snd_ymfpci_pcm_free_substream; - /* FIXME? True value is 256/48 = 5.33333 ms */ - snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIOD_TIME, 5333, UINT_MAX); return 0; } @@ -1013,6 +1022,18 @@ static int snd_ymfpci_capture_open(struct snd_pcm_substream *substream, struct snd_ymfpci *chip = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; struct snd_ymfpci_pcm *ypcm; + int err; + + runtime->hw = snd_ymfpci_capture; + /* FIXME? True value is 256/48 = 5.33333 ms */ + err = snd_pcm_hw_constraint_minmax(runtime, + SNDRV_PCM_HW_PARAM_PERIOD_TIME, + 5334, UINT_MAX); + if (err < 0) + return err; + err = snd_pcm_hw_rule_noresample(runtime, 48000); + if (err < 0) + return err; ypcm = kzalloc(sizeof(*ypcm), GFP_KERNEL); if (ypcm == NULL) @@ -1022,9 +1043,6 @@ static int snd_ymfpci_capture_open(struct snd_pcm_substream *substream, ypcm->substream = substream; ypcm->capture_bank_number = capture_bank_number; chip->capture_substream[capture_bank_number] = substream; - runtime->hw = snd_ymfpci_capture; - /* FIXME? True value is 256/48 = 5.33333 ms */ - snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIOD_TIME, 5333, UINT_MAX); runtime->private_data = ypcm; runtime->private_free = snd_ymfpci_pcm_free_substream; snd_ymfpci_hw_start(chip); @@ -1615,7 +1633,7 @@ YMFPCI_DOUBLE("ADC Playback Volume", 0, YDSXGR_PRIADCOUTVOL), YMFPCI_DOUBLE("ADC Capture Volume", 0, YDSXGR_PRIADCLOOPVOL), YMFPCI_DOUBLE("ADC Playback Volume", 1, YDSXGR_SECADCOUTVOL), YMFPCI_DOUBLE("ADC Capture Volume", 1, YDSXGR_SECADCLOOPVOL), -YMFPCI_DOUBLE("FM Legacy Volume", 0, YDSXGR_LEGACYOUTVOL), +YMFPCI_DOUBLE("FM Legacy Playback Volume", 0, YDSXGR_LEGACYOUTVOL), YMFPCI_DOUBLE(SNDRV_CTL_NAME_IEC958("AC97 ", PLAYBACK,VOLUME), 0, YDSXGR_ZVOUTVOL), YMFPCI_DOUBLE(SNDRV_CTL_NAME_IEC958("", CAPTURE,VOLUME), 0, YDSXGR_ZVLOOPVOL), YMFPCI_DOUBLE(SNDRV_CTL_NAME_IEC958("AC97 ",PLAYBACK,VOLUME), 1, YDSXGR_SPDIFOUTVOL), diff --git a/sound/ppc/keywest.c b/sound/ppc/keywest.c index 8f064c7ce745..4080becf4cef 100644 --- a/sound/ppc/keywest.c +++ b/sound/ppc/keywest.c @@ -82,7 +82,6 @@ static int keywest_attach_adapter(struct i2c_adapter *adapter) static int keywest_remove(struct i2c_client *client) { - i2c_set_clientdata(client, NULL); if (! keywest_ctx) return 0; if (client == keywest_ctx->client) diff --git a/sound/ppc/snd_ps3.c b/sound/ppc/snd_ps3.c index bc823a547550..775bd95d4be6 100644 --- a/sound/ppc/snd_ps3.c +++ b/sound/ppc/snd_ps3.c @@ -845,7 +845,7 @@ static int __devinit snd_ps3_allocate_irq(void) return ret; } - ret = request_irq(the_card.irq_no, snd_ps3_interrupt, IRQF_DISABLED, + ret = request_irq(the_card.irq_no, snd_ps3_interrupt, 0, SND_PS3_DRIVER_NAME, &the_card); if (ret) { pr_info("%s: request_irq failed (%d)\n", __func__, ret); diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 8224db5f0434..1381db853ef0 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -7,6 +7,8 @@ menuconfig SND_SOC select SND_PCM select AC97_BUS if SND_SOC_AC97_BUS select SND_JACK if INPUT=y || INPUT=SND + select REGMAP_I2C if I2C + select REGMAP_SPI if SPI_MASTER ---help--- If you want ASoC support, you should say Y here and also to the @@ -51,6 +53,7 @@ source "sound/soc/nuc900/Kconfig" source "sound/soc/omap/Kconfig" source "sound/soc/kirkwood/Kconfig" source "sound/soc/mid-x86/Kconfig" +source "sound/soc/mxs/Kconfig" source "sound/soc/pxa/Kconfig" source "sound/soc/samsung/Kconfig" source "sound/soc/s6000/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 4f913876f332..9ea8ac827adc 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -12,6 +12,7 @@ obj-$(CONFIG_SND_SOC) += fsl/ obj-$(CONFIG_SND_SOC) += imx/ obj-$(CONFIG_SND_SOC) += jz4740/ obj-$(CONFIG_SND_SOC) += mid-x86/ +obj-$(CONFIG_SND_SOC) += mxs/ obj-$(CONFIG_SND_SOC) += nuc900/ obj-$(CONFIG_SND_SOC) += omap/ obj-$(CONFIG_SND_SOC) += kirkwood/ diff --git a/sound/soc/atmel/playpaq_wm8510.c b/sound/soc/atmel/playpaq_wm8510.c index 1aac2f4dbcf6..73ae99ad4578 100644 --- a/sound/soc/atmel/playpaq_wm8510.c +++ b/sound/soc/atmel/playpaq_wm8510.c @@ -338,7 +338,6 @@ static int playpaq_wm8510_init(struct snd_soc_pcm_runtime *rtd) /* always connected pins */ snd_soc_dapm_enable_pin(dapm, "Int Mic"); snd_soc_dapm_enable_pin(dapm, "Ext Spk"); - snd_soc_dapm_sync(dapm); @@ -383,14 +382,17 @@ static int __init playpaq_asoc_init(void) _gclk0 = clk_get(NULL, "gclk0"); if (IS_ERR(_gclk0)) { _gclk0 = NULL; + ret = PTR_ERR(_gclk0); goto err_gclk0; } _pll0 = clk_get(NULL, "pll0"); if (IS_ERR(_pll0)) { _pll0 = NULL; + ret = PTR_ERR(_pll0); goto err_pll0; } - if (clk_set_parent(_gclk0, _pll0)) { + ret = clk_set_parent(_gclk0, _pll0); + if (ret) { pr_warning("snd-soc-playpaq: " "Failed to set PLL0 as parent for DAC clock\n"); goto err_set_clk; diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c index bad3aa14d5b3..0377c5451aed 100644 --- a/sound/soc/atmel/sam9g20_wm8731.c +++ b/sound/soc/atmel/sam9g20_wm8731.c @@ -173,8 +173,6 @@ static int at91sam9g20ek_wm8731_init(struct snd_soc_pcm_runtime *rtd) /* always connected */ snd_soc_dapm_enable_pin(dapm, "Ext Spk"); - snd_soc_dapm_sync(dapm); - return 0; } diff --git a/sound/soc/atmel/snd-soc-afeb9260.c b/sound/soc/atmel/snd-soc-afeb9260.c index 5e4d499d8434..d427e9217ce4 100644 --- a/sound/soc/atmel/snd-soc-afeb9260.c +++ b/sound/soc/atmel/snd-soc-afeb9260.c @@ -117,8 +117,6 @@ static int afeb9260_tlv320aic23_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_enable_pin(dapm, "Line In"); snd_soc_dapm_enable_pin(dapm, "Mic Jack"); - snd_soc_dapm_sync(dapm); - return 0; } diff --git a/sound/soc/au1x/Kconfig b/sound/soc/au1x/Kconfig index 4b67140fdec3..6d592546e8fc 100644 --- a/sound/soc/au1x/Kconfig +++ b/sound/soc/au1x/Kconfig @@ -18,10 +18,38 @@ config SND_SOC_AU1XPSC_AC97 select SND_AC97_CODEC select SND_SOC_AC97_BUS +## +## Au1000/1500/1100 DMA + AC97C/I2SC +## +config SND_SOC_AU1XAUDIO + tristate "SoC Audio for Au1000/Au1500/Au1100" + depends on MIPS_ALCHEMY + help + This is a driver set for the AC97 unit and the + old DMA controller as found on the Au1000/Au1500/Au1100 chips. + +config SND_SOC_AU1XAC97C + tristate + select AC97_BUS + select SND_AC97_CODEC + select SND_SOC_AC97_BUS + +config SND_SOC_AU1XI2SC + tristate + ## ## Boards ## +config SND_SOC_DB1000 + tristate "DB1000 Audio support" + depends on SND_SOC_AU1XAUDIO + select SND_SOC_AU1XAC97C + select SND_SOC_AC97_CODEC + help + Select this option to enable AC97 audio on the early DB1x00 series + of boards (DB1000/DB1500/DB1100). + config SND_SOC_DB1200 tristate "DB1200 AC97+I2S audio support" depends on SND_SOC_AU1XPSC diff --git a/sound/soc/au1x/Makefile b/sound/soc/au1x/Makefile index 16873076e8c4..920710514ea0 100644 --- a/sound/soc/au1x/Makefile +++ b/sound/soc/au1x/Makefile @@ -3,11 +3,21 @@ snd-soc-au1xpsc-dbdma-objs := dbdma2.o snd-soc-au1xpsc-i2s-objs := psc-i2s.o snd-soc-au1xpsc-ac97-objs := psc-ac97.o +# Au1000/1500/1100 Audio units +snd-soc-au1x-dma-objs := dma.o +snd-soc-au1x-ac97c-objs := ac97c.o +snd-soc-au1x-i2sc-objs := i2sc.o + obj-$(CONFIG_SND_SOC_AU1XPSC) += snd-soc-au1xpsc-dbdma.o obj-$(CONFIG_SND_SOC_AU1XPSC_I2S) += snd-soc-au1xpsc-i2s.o obj-$(CONFIG_SND_SOC_AU1XPSC_AC97) += snd-soc-au1xpsc-ac97.o +obj-$(CONFIG_SND_SOC_AU1XAUDIO) += snd-soc-au1x-dma.o +obj-$(CONFIG_SND_SOC_AU1XAC97C) += snd-soc-au1x-ac97c.o +obj-$(CONFIG_SND_SOC_AU1XI2SC) += snd-soc-au1x-i2sc.o # Boards +snd-soc-db1000-objs := db1000.o snd-soc-db1200-objs := db1200.o +obj-$(CONFIG_SND_SOC_DB1000) += snd-soc-db1000.o obj-$(CONFIG_SND_SOC_DB1200) += snd-soc-db1200.o diff --git a/sound/soc/au1x/ac97c.c b/sound/soc/au1x/ac97c.c new file mode 100644 index 000000000000..726bd651a105 --- /dev/null +++ b/sound/soc/au1x/ac97c.c @@ -0,0 +1,366 @@ +/* + * Au1000/Au1500/Au1100 AC97C controller driver for ASoC + * + * (c) 2011 Manuel Lauss <manuel.lauss@googlemail.com> + * + * based on the old ALSA driver originally written by + * Charles Eidsness <charles@cooper-street.com> + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/slab.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/mutex.h> +#include <linux/platform_device.h> +#include <linux/suspend.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/initval.h> +#include <sound/soc.h> +#include <asm/mach-au1x00/au1000.h> + +#include "psc.h" + +/* register offsets and bits */ +#define AC97_CONFIG 0x00 +#define AC97_STATUS 0x04 +#define AC97_DATA 0x08 +#define AC97_CMDRESP 0x0c +#define AC97_ENABLE 0x10 + +#define CFG_RC(x) (((x) & 0x3ff) << 13) /* valid rx slots mask */ +#define CFG_XS(x) (((x) & 0x3ff) << 3) /* valid tx slots mask */ +#define CFG_SG (1 << 2) /* sync gate */ +#define CFG_SN (1 << 1) /* sync control */ +#define CFG_RS (1 << 0) /* acrst# control */ +#define STAT_XU (1 << 11) /* tx underflow */ +#define STAT_XO (1 << 10) /* tx overflow */ +#define STAT_RU (1 << 9) /* rx underflow */ +#define STAT_RO (1 << 8) /* rx overflow */ +#define STAT_RD (1 << 7) /* codec ready */ +#define STAT_CP (1 << 6) /* command pending */ +#define STAT_TE (1 << 4) /* tx fifo empty */ +#define STAT_TF (1 << 3) /* tx fifo full */ +#define STAT_RE (1 << 1) /* rx fifo empty */ +#define STAT_RF (1 << 0) /* rx fifo full */ +#define CMD_SET_DATA(x) (((x) & 0xffff) << 16) +#define CMD_GET_DATA(x) ((x) & 0xffff) +#define CMD_READ (1 << 7) +#define CMD_WRITE (0 << 7) +#define CMD_IDX(x) ((x) & 0x7f) +#define EN_D (1 << 1) /* DISable bit */ +#define EN_CE (1 << 0) /* clock enable bit */ + +/* how often to retry failed codec register reads/writes */ +#define AC97_RW_RETRIES 5 + +#define AC97_RATES \ + SNDRV_PCM_RATE_CONTINUOUS + +#define AC97_FMTS \ + (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE) + +/* instance data. There can be only one, MacLeod!!!!, fortunately there IS only + * once AC97C on early Alchemy chips. The newer ones aren't so lucky. + */ +static struct au1xpsc_audio_data *ac97c_workdata; +#define ac97_to_ctx(x) ac97c_workdata + +static inline unsigned long RD(struct au1xpsc_audio_data *ctx, int reg) +{ + return __raw_readl(ctx->mmio + reg); +} + +static inline void WR(struct au1xpsc_audio_data *ctx, int reg, unsigned long v) +{ + __raw_writel(v, ctx->mmio + reg); + wmb(); +} + +static unsigned short au1xac97c_ac97_read(struct snd_ac97 *ac97, + unsigned short r) +{ + struct au1xpsc_audio_data *ctx = ac97_to_ctx(ac97); + unsigned int tmo, retry; + unsigned long data; + + data = ~0; + retry = AC97_RW_RETRIES; + do { + mutex_lock(&ctx->lock); + + tmo = 5; + while ((RD(ctx, AC97_STATUS) & STAT_CP) && tmo--) + udelay(21); /* wait an ac97 frame time */ + if (!tmo) { + pr_debug("ac97rd timeout #1\n"); + goto next; + } + + WR(ctx, AC97_CMDRESP, CMD_IDX(r) | CMD_READ); + + /* stupid errata: data is only valid for 21us, so + * poll, Forrest, poll... + */ + tmo = 0x10000; + while ((RD(ctx, AC97_STATUS) & STAT_CP) && tmo--) + asm volatile ("nop"); + data = RD(ctx, AC97_CMDRESP); + + if (!tmo) + pr_debug("ac97rd timeout #2\n"); + +next: + mutex_unlock(&ctx->lock); + } while (--retry && !tmo); + + pr_debug("AC97RD %04x %04lx %d\n", r, data, retry); + + return retry ? data & 0xffff : 0xffff; +} + +static void au1xac97c_ac97_write(struct snd_ac97 *ac97, unsigned short r, + unsigned short v) +{ + struct au1xpsc_audio_data *ctx = ac97_to_ctx(ac97); + unsigned int tmo, retry; + + retry = AC97_RW_RETRIES; + do { + mutex_lock(&ctx->lock); + + for (tmo = 5; (RD(ctx, AC97_STATUS) & STAT_CP) && tmo; tmo--) + udelay(21); + if (!tmo) { + pr_debug("ac97wr timeout #1\n"); + goto next; + } + + WR(ctx, AC97_CMDRESP, CMD_WRITE | CMD_IDX(r) | CMD_SET_DATA(v)); + + for (tmo = 10; (RD(ctx, AC97_STATUS) & STAT_CP) && tmo; tmo--) + udelay(21); + if (!tmo) + pr_debug("ac97wr timeout #2\n"); +next: + mutex_unlock(&ctx->lock); + } while (--retry && !tmo); + + pr_debug("AC97WR %04x %04x %d\n", r, v, retry); +} + +static void au1xac97c_ac97_warm_reset(struct snd_ac97 *ac97) +{ + struct au1xpsc_audio_data *ctx = ac97_to_ctx(ac97); + + WR(ctx, AC97_CONFIG, ctx->cfg | CFG_SG | CFG_SN); + msleep(20); + WR(ctx, AC97_CONFIG, ctx->cfg | CFG_SG); + WR(ctx, AC97_CONFIG, ctx->cfg); +} + +static void au1xac97c_ac97_cold_reset(struct snd_ac97 *ac97) +{ + struct au1xpsc_audio_data *ctx = ac97_to_ctx(ac97); + int i; + + WR(ctx, AC97_CONFIG, ctx->cfg | CFG_RS); + msleep(500); + WR(ctx, AC97_CONFIG, ctx->cfg); + + /* wait for codec ready */ + i = 50; + while (((RD(ctx, AC97_STATUS) & STAT_RD) == 0) && --i) + msleep(20); + if (!i) + printk(KERN_ERR "ac97c: codec not ready after cold reset\n"); +} + +/* AC97 controller operations */ +struct snd_ac97_bus_ops soc_ac97_ops = { + .read = au1xac97c_ac97_read, + .write = au1xac97c_ac97_write, + .reset = au1xac97c_ac97_cold_reset, + .warm_reset = au1xac97c_ac97_warm_reset, +}; +EXPORT_SYMBOL_GPL(soc_ac97_ops); /* globals be gone! */ + +static int alchemy_ac97c_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct au1xpsc_audio_data *ctx = snd_soc_dai_get_drvdata(dai); + snd_soc_dai_set_dma_data(dai, substream, &ctx->dmaids[0]); + return 0; +} + +static struct snd_soc_dai_ops alchemy_ac97c_ops = { + .startup = alchemy_ac97c_startup, +}; + +static int au1xac97c_dai_probe(struct snd_soc_dai *dai) +{ + return ac97c_workdata ? 0 : -ENODEV; +} + +static struct snd_soc_dai_driver au1xac97c_dai_driver = { + .name = "alchemy-ac97c", + .ac97_control = 1, + .probe = au1xac97c_dai_probe, + .playback = { + .rates = AC97_RATES, + .formats = AC97_FMTS, + .channels_min = 2, + .channels_max = 2, + }, + .capture = { + .rates = AC97_RATES, + .formats = AC97_FMTS, + .channels_min = 2, + .channels_max = 2, + }, + .ops = &alchemy_ac97c_ops, +}; + +static int __devinit au1xac97c_drvprobe(struct platform_device *pdev) +{ + int ret; + struct resource *iores, *dmares; + struct au1xpsc_audio_data *ctx; + + ctx = kzalloc(sizeof(*ctx), GFP_KERNEL); + if (!ctx) + return -ENOMEM; + + mutex_init(&ctx->lock); + + iores = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!iores) { + ret = -ENODEV; + goto out0; + } + + ret = -EBUSY; + if (!request_mem_region(iores->start, resource_size(iores), + pdev->name)) + goto out0; + + ctx->mmio = ioremap_nocache(iores->start, resource_size(iores)); + if (!ctx->mmio) + goto out1; + + dmares = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!dmares) + goto out2; + ctx->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = dmares->start; + + dmares = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!dmares) + goto out2; + ctx->dmaids[SNDRV_PCM_STREAM_CAPTURE] = dmares->start; + + /* switch it on */ + WR(ctx, AC97_ENABLE, EN_D | EN_CE); + WR(ctx, AC97_ENABLE, EN_CE); + + ctx->cfg = CFG_RC(3) | CFG_XS(3); + WR(ctx, AC97_CONFIG, ctx->cfg); + + platform_set_drvdata(pdev, ctx); + + ret = snd_soc_register_dai(&pdev->dev, &au1xac97c_dai_driver); + if (ret) + goto out2; + + ac97c_workdata = ctx; + return 0; + +out2: + iounmap(ctx->mmio); +out1: + release_mem_region(iores->start, resource_size(iores)); +out0: + kfree(ctx); + return ret; +} + +static int __devexit au1xac97c_drvremove(struct platform_device *pdev) +{ + struct au1xpsc_audio_data *ctx = platform_get_drvdata(pdev); + struct resource *r = platform_get_resource(pdev, IORESOURCE_MEM, 0); + + snd_soc_unregister_dai(&pdev->dev); + + WR(ctx, AC97_ENABLE, EN_D); /* clock off, disable */ + + iounmap(ctx->mmio); + release_mem_region(r->start, resource_size(r)); + kfree(ctx); + + ac97c_workdata = NULL; /* MDEV */ + + return 0; +} + +#ifdef CONFIG_PM +static int au1xac97c_drvsuspend(struct device *dev) +{ + struct au1xpsc_audio_data *ctx = dev_get_drvdata(dev); + + WR(ctx, AC97_ENABLE, EN_D); /* clock off, disable */ + + return 0; +} + +static int au1xac97c_drvresume(struct device *dev) +{ + struct au1xpsc_audio_data *ctx = dev_get_drvdata(dev); + + WR(ctx, AC97_ENABLE, EN_D | EN_CE); + WR(ctx, AC97_ENABLE, EN_CE); + WR(ctx, AC97_CONFIG, ctx->cfg); + + return 0; +} + +static const struct dev_pm_ops au1xpscac97_pmops = { + .suspend = au1xac97c_drvsuspend, + .resume = au1xac97c_drvresume, +}; + +#define AU1XPSCAC97_PMOPS (&au1xpscac97_pmops) + +#else + +#define AU1XPSCAC97_PMOPS NULL + +#endif + +static struct platform_driver au1xac97c_driver = { + .driver = { + .name = "alchemy-ac97c", + .owner = THIS_MODULE, + .pm = AU1XPSCAC97_PMOPS, + }, + .probe = au1xac97c_drvprobe, + .remove = __devexit_p(au1xac97c_drvremove), +}; + +static int __init au1xac97c_load(void) +{ + ac97c_workdata = NULL; + return platform_driver_register(&au1xac97c_driver); +} + +static void __exit au1xac97c_unload(void) +{ + platform_driver_unregister(&au1xac97c_driver); +} + +module_init(au1xac97c_load); +module_exit(au1xac97c_unload); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("Au1000/1500/1100 AC97C ASoC driver"); +MODULE_AUTHOR("Manuel Lauss"); diff --git a/sound/soc/au1x/db1000.c b/sound/soc/au1x/db1000.c new file mode 100644 index 000000000000..127477a5e0c7 --- /dev/null +++ b/sound/soc/au1x/db1000.c @@ -0,0 +1,75 @@ +/* + * DB1000/DB1500/DB1100 ASoC audio fabric support code. + * + * (c) 2011 Manuel Lauss <manuel.lauss@googlemail.com> + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/timer.h> +#include <linux/interrupt.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <asm/mach-au1x00/au1000.h> +#include <asm/mach-db1x00/bcsr.h> + +#include "psc.h" + +static struct snd_soc_dai_link db1000_ac97_dai = { + .name = "AC97", + .stream_name = "AC97 HiFi", + .codec_dai_name = "ac97-hifi", + .cpu_dai_name = "alchemy-ac97c", + .platform_name = "alchemy-pcm-dma.0", + .codec_name = "ac97-codec", +}; + +static struct snd_soc_card db1000_ac97 = { + .name = "DB1000_AC97", + .dai_link = &db1000_ac97_dai, + .num_links = 1, +}; + +static int __devinit db1000_audio_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &db1000_ac97; + card->dev = &pdev->dev; + return snd_soc_register_card(card); +} + +static int __devexit db1000_audio_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + snd_soc_unregister_card(card); + return 0; +} + +static struct platform_driver db1000_audio_driver = { + .driver = { + .name = "db1000-audio", + .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, + }, + .probe = db1000_audio_probe, + .remove = __devexit_p(db1000_audio_remove), +}; + +static int __init db1000_audio_load(void) +{ + return platform_driver_register(&db1000_audio_driver); +} + +static void __exit db1000_audio_unload(void) +{ + platform_driver_unregister(&db1000_audio_driver); +} + +module_init(db1000_audio_load); +module_exit(db1000_audio_unload); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("DB1000/DB1500/DB1100 ASoC audio"); +MODULE_AUTHOR("Manuel Lauss"); diff --git a/sound/soc/au1x/db1200.c b/sound/soc/au1x/db1200.c index 1d3e258c9ea8..289312c14b99 100644 --- a/sound/soc/au1x/db1200.c +++ b/sound/soc/au1x/db1200.c @@ -1,7 +1,7 @@ /* * DB1200 ASoC audio fabric support code. * - * (c) 2008-9 Manuel Lauss <manuel.lauss@gmail.com> + * (c) 2008-2011 Manuel Lauss <manuel.lauss@googlemail.com> * */ @@ -21,6 +21,17 @@ #include "../codecs/wm8731.h" #include "psc.h" +static struct platform_device_id db1200_pids[] = { + { + .name = "db1200-ac97", + .driver_data = 0, + }, { + .name = "db1200-i2s", + .driver_data = 1, + }, + {}, +}; + /*------------------------- AC97 PART ---------------------------*/ static struct snd_soc_dai_link db1200_ac97_dai = { @@ -89,36 +100,47 @@ static struct snd_soc_card db1200_i2s_machine = { /*------------------------- COMMON PART ---------------------------*/ -static struct platform_device *db1200_asoc_dev; +static struct snd_soc_card *db1200_cards[] __devinitdata = { + &db1200_ac97_machine, + &db1200_i2s_machine, +}; -static int __init db1200_audio_load(void) +static int __devinit db1200_audio_probe(struct platform_device *pdev) { - int ret; + const struct platform_device_id *pid = platform_get_device_id(pdev); + struct snd_soc_card *card; - ret = -ENOMEM; - db1200_asoc_dev = platform_device_alloc("soc-audio", 1); /* PSC1 */ - if (!db1200_asoc_dev) - goto out; + card = db1200_cards[pid->driver_data]; + card->dev = &pdev->dev; + return snd_soc_register_card(card); +} - /* DB1200 board setup set PSC1MUX to preferred audio device */ - if (bcsr_read(BCSR_RESETS) & BCSR_RESETS_PSC1MUX) - platform_set_drvdata(db1200_asoc_dev, &db1200_i2s_machine); - else - platform_set_drvdata(db1200_asoc_dev, &db1200_ac97_machine); +static int __devexit db1200_audio_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + snd_soc_unregister_card(card); + return 0; +} - ret = platform_device_add(db1200_asoc_dev); +static struct platform_driver db1200_audio_driver = { + .driver = { + .name = "db1200-ac97", + .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, + }, + .id_table = db1200_pids, + .probe = db1200_audio_probe, + .remove = __devexit_p(db1200_audio_remove), +}; - if (ret) { - platform_device_put(db1200_asoc_dev); - db1200_asoc_dev = NULL; - } -out: - return ret; +static int __init db1200_audio_load(void) +{ + return platform_driver_register(&db1200_audio_driver); } static void __exit db1200_audio_unload(void) { - platform_device_unregister(db1200_asoc_dev); + platform_driver_unregister(&db1200_audio_driver); } module_init(db1200_audio_load); diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c index 20bb53a837b1..d7d04e26eee5 100644 --- a/sound/soc/au1x/dbdma2.c +++ b/sound/soc/au1x/dbdma2.c @@ -169,7 +169,7 @@ static int au1x_pcm_dbdma_realloc(struct au1xpsc_audio_dmadata *pcd, au1x_pcm_dbdma_free(pcd); - if (stype == PCM_RX) + if (stype == SNDRV_PCM_STREAM_CAPTURE) pcd->ddma_chan = au1xxx_dbdma_chan_alloc(pcd->ddma_id, DSCR_CMD0_ALWAYS, au1x_pcm_dmarx_cb, (void *)pcd); @@ -198,7 +198,7 @@ static inline struct au1xpsc_audio_dmadata *to_dmadata(struct snd_pcm_substream struct snd_soc_pcm_runtime *rtd = ss->private_data; struct au1xpsc_audio_dmadata *pcd = snd_soc_platform_get_drvdata(rtd->platform); - return &pcd[SUBSTREAM_TYPE(ss)]; + return &pcd[ss->stream]; } static int au1xpsc_pcm_hw_params(struct snd_pcm_substream *substream, @@ -212,7 +212,7 @@ static int au1xpsc_pcm_hw_params(struct snd_pcm_substream *substream, if (ret < 0) goto out; - stype = SUBSTREAM_TYPE(substream); + stype = substream->stream; pcd = to_dmadata(substream); DBG("runtime->dma_area = 0x%08lx dma_addr_t = 0x%08lx dma_size = %d " @@ -255,7 +255,7 @@ static int au1xpsc_pcm_prepare(struct snd_pcm_substream *substream) au1xxx_dbdma_reset(pcd->ddma_chan); - if (SUBSTREAM_TYPE(substream) == PCM_RX) { + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { au1x_pcm_queue_rx(pcd); au1x_pcm_queue_rx(pcd); } else { @@ -293,6 +293,16 @@ au1xpsc_pcm_pointer(struct snd_pcm_substream *substream) static int au1xpsc_pcm_open(struct snd_pcm_substream *substream) { + struct au1xpsc_audio_dmadata *pcd = to_dmadata(substream); + struct snd_soc_pcm_runtime *rtd = substream->private_data; + int stype = substream->stream, *dmaids; + + dmaids = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + if (!dmaids) + return -ENODEV; /* whoa, has ordering changed? */ + + pcd->ddma_id = dmaids[stype]; + snd_soc_set_runtime_hwparams(substream, &au1xpsc_pcm_hardware); return 0; } @@ -340,36 +350,18 @@ struct snd_soc_platform_driver au1xpsc_soc_platform = { static int __devinit au1xpsc_pcm_drvprobe(struct platform_device *pdev) { struct au1xpsc_audio_dmadata *dmadata; - struct resource *r; int ret; dmadata = kzalloc(2 * sizeof(struct au1xpsc_audio_dmadata), GFP_KERNEL); if (!dmadata) return -ENOMEM; - r = platform_get_resource(pdev, IORESOURCE_DMA, 0); - if (!r) { - ret = -ENODEV; - goto out1; - } - dmadata[PCM_TX].ddma_id = r->start; - - /* RX DMA */ - r = platform_get_resource(pdev, IORESOURCE_DMA, 1); - if (!r) { - ret = -ENODEV; - goto out1; - } - dmadata[PCM_RX].ddma_id = r->start; - platform_set_drvdata(pdev, dmadata); ret = snd_soc_register_platform(&pdev->dev, &au1xpsc_soc_platform); - if (!ret) - return ret; + if (ret) + kfree(dmadata); -out1: - kfree(dmadata); return ret; } @@ -405,57 +397,6 @@ static void __exit au1xpsc_audio_dbdma_unload(void) module_init(au1xpsc_audio_dbdma_load); module_exit(au1xpsc_audio_dbdma_unload); - -struct platform_device *au1xpsc_pcm_add(struct platform_device *pdev) -{ - struct resource *res, *r; - struct platform_device *pd; - int id[2]; - int ret; - - r = platform_get_resource(pdev, IORESOURCE_DMA, 0); - if (!r) - return NULL; - id[0] = r->start; - - r = platform_get_resource(pdev, IORESOURCE_DMA, 1); - if (!r) - return NULL; - id[1] = r->start; - - res = kzalloc(sizeof(struct resource) * 2, GFP_KERNEL); - if (!res) - return NULL; - - res[0].start = res[0].end = id[0]; - res[1].start = res[1].end = id[1]; - res[0].flags = res[1].flags = IORESOURCE_DMA; - - pd = platform_device_alloc("au1xpsc-pcm", pdev->id); - if (!pd) - goto out; - - pd->resource = res; - pd->num_resources = 2; - - ret = platform_device_add(pd); - if (!ret) - return pd; - - platform_device_put(pd); -out: - kfree(res); - return NULL; -} -EXPORT_SYMBOL_GPL(au1xpsc_pcm_add); - -void au1xpsc_pcm_destroy(struct platform_device *dmapd) -{ - if (dmapd) - platform_device_unregister(dmapd); -} -EXPORT_SYMBOL_GPL(au1xpsc_pcm_destroy); - MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Au12x0/Au1550 PSC Audio DMA driver"); MODULE_AUTHOR("Manuel Lauss"); diff --git a/sound/soc/au1x/dma.c b/sound/soc/au1x/dma.c new file mode 100644 index 000000000000..177f7137a9c8 --- /dev/null +++ b/sound/soc/au1x/dma.c @@ -0,0 +1,377 @@ +/* + * Au1000/Au1500/Au1100 Audio DMA support. + * + * (c) 2011 Manuel Lauss <manuel.lauss@googlemail.com> + * + * copied almost verbatim from the old ALSA driver, written by + * Charles Eidsness <charles@cooper-street.com> + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/dma-mapping.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <asm/mach-au1x00/au1000.h> +#include <asm/mach-au1x00/au1000_dma.h> + +#include "psc.h" + +#define ALCHEMY_PCM_FMTS \ + (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 | \ + SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | \ + SNDRV_PCM_FMTBIT_U16_LE | SNDRV_PCM_FMTBIT_U16_BE | \ + SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE | \ + SNDRV_PCM_FMTBIT_U32_LE | SNDRV_PCM_FMTBIT_U32_BE | \ + 0) + +struct pcm_period { + u32 start; + u32 relative_end; /* relative to start of buffer */ + struct pcm_period *next; +}; + +struct audio_stream { + struct snd_pcm_substream *substream; + int dma; + struct pcm_period *buffer; + unsigned int period_size; + unsigned int periods; +}; + +struct alchemy_pcm_ctx { + struct audio_stream stream[2]; /* playback & capture */ +}; + +static void au1000_release_dma_link(struct audio_stream *stream) +{ + struct pcm_period *pointer; + struct pcm_period *pointer_next; + + stream->period_size = 0; + stream->periods = 0; + pointer = stream->buffer; + if (!pointer) + return; + do { + pointer_next = pointer->next; + kfree(pointer); + pointer = pointer_next; + } while (pointer != stream->buffer); + stream->buffer = NULL; +} + +static int au1000_setup_dma_link(struct audio_stream *stream, + unsigned int period_bytes, + unsigned int periods) +{ + struct snd_pcm_substream *substream = stream->substream; + struct snd_pcm_runtime *runtime = substream->runtime; + struct pcm_period *pointer; + unsigned long dma_start; + int i; + + dma_start = virt_to_phys(runtime->dma_area); + + if (stream->period_size == period_bytes && + stream->periods == periods) + return 0; /* not changed */ + + au1000_release_dma_link(stream); + + stream->period_size = period_bytes; + stream->periods = periods; + + stream->buffer = kmalloc(sizeof(struct pcm_period), GFP_KERNEL); + if (!stream->buffer) + return -ENOMEM; + pointer = stream->buffer; + for (i = 0; i < periods; i++) { + pointer->start = (u32)(dma_start + (i * period_bytes)); + pointer->relative_end = (u32) (((i+1) * period_bytes) - 0x1); + if (i < periods - 1) { + pointer->next = kmalloc(sizeof(struct pcm_period), + GFP_KERNEL); + if (!pointer->next) { + au1000_release_dma_link(stream); + return -ENOMEM; + } + pointer = pointer->next; + } + } + pointer->next = stream->buffer; + return 0; +} + +static void au1000_dma_stop(struct audio_stream *stream) +{ + if (stream->buffer) + disable_dma(stream->dma); +} + +static void au1000_dma_start(struct audio_stream *stream) +{ + if (!stream->buffer) + return; + + init_dma(stream->dma); + if (get_dma_active_buffer(stream->dma) == 0) { + clear_dma_done0(stream->dma); + set_dma_addr0(stream->dma, stream->buffer->start); + set_dma_count0(stream->dma, stream->period_size >> 1); + set_dma_addr1(stream->dma, stream->buffer->next->start); + set_dma_count1(stream->dma, stream->period_size >> 1); + } else { + clear_dma_done1(stream->dma); + set_dma_addr1(stream->dma, stream->buffer->start); + set_dma_count1(stream->dma, stream->period_size >> 1); + set_dma_addr0(stream->dma, stream->buffer->next->start); + set_dma_count0(stream->dma, stream->period_size >> 1); + } + enable_dma_buffers(stream->dma); + start_dma(stream->dma); +} + +static irqreturn_t au1000_dma_interrupt(int irq, void *ptr) +{ + struct audio_stream *stream = (struct audio_stream *)ptr; + struct snd_pcm_substream *substream = stream->substream; + + switch (get_dma_buffer_done(stream->dma)) { + case DMA_D0: + stream->buffer = stream->buffer->next; + clear_dma_done0(stream->dma); + set_dma_addr0(stream->dma, stream->buffer->next->start); + set_dma_count0(stream->dma, stream->period_size >> 1); + enable_dma_buffer0(stream->dma); + break; + case DMA_D1: + stream->buffer = stream->buffer->next; + clear_dma_done1(stream->dma); + set_dma_addr1(stream->dma, stream->buffer->next->start); + set_dma_count1(stream->dma, stream->period_size >> 1); + enable_dma_buffer1(stream->dma); + break; + case (DMA_D0 | DMA_D1): + pr_debug("DMA %d missed interrupt.\n", stream->dma); + au1000_dma_stop(stream); + au1000_dma_start(stream); + break; + case (~DMA_D0 & ~DMA_D1): + pr_debug("DMA %d empty irq.\n", stream->dma); + } + snd_pcm_period_elapsed(substream); + return IRQ_HANDLED; +} + +static const struct snd_pcm_hardware alchemy_pcm_hardware = { + .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BATCH, + .formats = ALCHEMY_PCM_FMTS, + .rates = SNDRV_PCM_RATE_8000_192000, + .rate_min = SNDRV_PCM_RATE_8000, + .rate_max = SNDRV_PCM_RATE_192000, + .channels_min = 2, + .channels_max = 2, + .period_bytes_min = 1024, + .period_bytes_max = 16 * 1024 - 1, + .periods_min = 4, + .periods_max = 255, + .buffer_bytes_max = 128 * 1024, + .fifo_size = 16, +}; + +static inline struct alchemy_pcm_ctx *ss_to_ctx(struct snd_pcm_substream *ss) +{ + struct snd_soc_pcm_runtime *rtd = ss->private_data; + return snd_soc_platform_get_drvdata(rtd->platform); +} + +static inline struct audio_stream *ss_to_as(struct snd_pcm_substream *ss) +{ + struct alchemy_pcm_ctx *ctx = ss_to_ctx(ss); + return &(ctx->stream[ss->stream]); +} + +static int alchemy_pcm_open(struct snd_pcm_substream *substream) +{ + struct alchemy_pcm_ctx *ctx = ss_to_ctx(substream); + struct snd_soc_pcm_runtime *rtd = substream->private_data; + int *dmaids, s = substream->stream; + char *name; + + dmaids = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + if (!dmaids) + return -ENODEV; /* whoa, has ordering changed? */ + + /* DMA setup */ + name = (s == SNDRV_PCM_STREAM_PLAYBACK) ? "audio-tx" : "audio-rx"; + ctx->stream[s].dma = request_au1000_dma(dmaids[s], name, + au1000_dma_interrupt, 0, + &ctx->stream[s]); + set_dma_mode(ctx->stream[s].dma, + get_dma_mode(ctx->stream[s].dma) & ~DMA_NC); + + ctx->stream[s].substream = substream; + ctx->stream[s].buffer = NULL; + snd_soc_set_runtime_hwparams(substream, &alchemy_pcm_hardware); + + return 0; +} + +static int alchemy_pcm_close(struct snd_pcm_substream *substream) +{ + struct alchemy_pcm_ctx *ctx = ss_to_ctx(substream); + int stype = substream->stream; + + ctx->stream[stype].substream = NULL; + free_au1000_dma(ctx->stream[stype].dma); + + return 0; +} + +static int alchemy_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct audio_stream *stream = ss_to_as(substream); + int err; + + err = snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); + if (err < 0) + return err; + err = au1000_setup_dma_link(stream, + params_period_bytes(hw_params), + params_periods(hw_params)); + if (err) + snd_pcm_lib_free_pages(substream); + + return err; +} + +static int alchemy_pcm_hw_free(struct snd_pcm_substream *substream) +{ + struct audio_stream *stream = ss_to_as(substream); + au1000_release_dma_link(stream); + return snd_pcm_lib_free_pages(substream); +} + +static int alchemy_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct audio_stream *stream = ss_to_as(substream); + int err = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + au1000_dma_start(stream); + break; + case SNDRV_PCM_TRIGGER_STOP: + au1000_dma_stop(stream); + break; + default: + err = -EINVAL; + break; + } + return err; +} + +static snd_pcm_uframes_t alchemy_pcm_pointer(struct snd_pcm_substream *ss) +{ + struct audio_stream *stream = ss_to_as(ss); + long location; + + location = get_dma_residue(stream->dma); + location = stream->buffer->relative_end - location; + if (location == -1) + location = 0; + return bytes_to_frames(ss->runtime, location); +} + +static struct snd_pcm_ops alchemy_pcm_ops = { + .open = alchemy_pcm_open, + .close = alchemy_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = alchemy_pcm_hw_params, + .hw_free = alchemy_pcm_hw_free, + .trigger = alchemy_pcm_trigger, + .pointer = alchemy_pcm_pointer, +}; + +static void alchemy_pcm_free_dma_buffers(struct snd_pcm *pcm) +{ + snd_pcm_lib_preallocate_free_for_all(pcm); +} + +static int alchemy_pcm_new(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_pcm *pcm = rtd->pcm; + + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_CONTINUOUS, + snd_dma_continuous_data(GFP_KERNEL), 65536, (4096 * 1024) - 1); + + return 0; +} + +struct snd_soc_platform_driver alchemy_pcm_soc_platform = { + .ops = &alchemy_pcm_ops, + .pcm_new = alchemy_pcm_new, + .pcm_free = alchemy_pcm_free_dma_buffers, +}; + +static int __devinit alchemy_pcm_drvprobe(struct platform_device *pdev) +{ + struct alchemy_pcm_ctx *ctx; + int ret; + + ctx = kzalloc(sizeof(*ctx), GFP_KERNEL); + if (!ctx) + return -ENOMEM; + + platform_set_drvdata(pdev, ctx); + + ret = snd_soc_register_platform(&pdev->dev, &alchemy_pcm_soc_platform); + if (ret) + kfree(ctx); + + return ret; +} + +static int __devexit alchemy_pcm_drvremove(struct platform_device *pdev) +{ + struct alchemy_pcm_ctx *ctx = platform_get_drvdata(pdev); + + snd_soc_unregister_platform(&pdev->dev); + kfree(ctx); + + return 0; +} + +static struct platform_driver alchemy_pcmdma_driver = { + .driver = { + .name = "alchemy-pcm-dma", + .owner = THIS_MODULE, + }, + .probe = alchemy_pcm_drvprobe, + .remove = __devexit_p(alchemy_pcm_drvremove), +}; + +static int __init alchemy_pcmdma_load(void) +{ + return platform_driver_register(&alchemy_pcmdma_driver); +} + +static void __exit alchemy_pcmdma_unload(void) +{ + platform_driver_unregister(&alchemy_pcmdma_driver); +} + +module_init(alchemy_pcmdma_load); +module_exit(alchemy_pcmdma_unload); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("Au1000/Au1500/Au1100 Audio DMA driver"); +MODULE_AUTHOR("Manuel Lauss"); diff --git a/sound/soc/au1x/i2sc.c b/sound/soc/au1x/i2sc.c new file mode 100644 index 000000000000..6bcf48f5884c --- /dev/null +++ b/sound/soc/au1x/i2sc.c @@ -0,0 +1,349 @@ +/* + * Au1000/Au1500/Au1100 I2S controller driver for ASoC + * + * (c) 2011 Manuel Lauss <manuel.lauss@googlemail.com> + * + * Note: clock supplied to the I2S controller must be 256x samplerate. + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/slab.h> +#include <linux/suspend.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/initval.h> +#include <sound/soc.h> +#include <asm/mach-au1x00/au1000.h> + +#include "psc.h" + +#define I2S_RXTX 0x00 +#define I2S_CFG 0x04 +#define I2S_ENABLE 0x08 + +#define CFG_XU (1 << 25) /* tx underflow */ +#define CFG_XO (1 << 24) +#define CFG_RU (1 << 23) +#define CFG_RO (1 << 22) +#define CFG_TR (1 << 21) +#define CFG_TE (1 << 20) +#define CFG_TF (1 << 19) +#define CFG_RR (1 << 18) +#define CFG_RF (1 << 17) +#define CFG_ICK (1 << 12) /* clock invert */ +#define CFG_PD (1 << 11) /* set to make I2SDIO INPUT */ +#define CFG_LB (1 << 10) /* loopback */ +#define CFG_IC (1 << 9) /* word select invert */ +#define CFG_FM_I2S (0 << 7) /* I2S format */ +#define CFG_FM_LJ (1 << 7) /* left-justified */ +#define CFG_FM_RJ (2 << 7) /* right-justified */ +#define CFG_FM_MASK (3 << 7) +#define CFG_TN (1 << 6) /* tx fifo en */ +#define CFG_RN (1 << 5) /* rx fifo en */ +#define CFG_SZ_8 (0x08) +#define CFG_SZ_16 (0x10) +#define CFG_SZ_18 (0x12) +#define CFG_SZ_20 (0x14) +#define CFG_SZ_24 (0x18) +#define CFG_SZ_MASK (0x1f) +#define EN_D (1 << 1) /* DISable */ +#define EN_CE (1 << 0) /* clock enable */ + +/* only limited by clock generator and board design */ +#define AU1XI2SC_RATES \ + SNDRV_PCM_RATE_CONTINUOUS + +#define AU1XI2SC_FMTS \ + (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 | \ + SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | \ + SNDRV_PCM_FMTBIT_U16_LE | SNDRV_PCM_FMTBIT_U16_BE | \ + SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_U18_3LE | \ + SNDRV_PCM_FMTBIT_S18_3BE | SNDRV_PCM_FMTBIT_U18_3BE | \ + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_U20_3LE | \ + SNDRV_PCM_FMTBIT_S20_3BE | SNDRV_PCM_FMTBIT_U20_3BE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE | \ + SNDRV_PCM_FMTBIT_U24_LE | SNDRV_PCM_FMTBIT_U24_BE | \ + 0) + +static inline unsigned long RD(struct au1xpsc_audio_data *ctx, int reg) +{ + return __raw_readl(ctx->mmio + reg); +} + +static inline void WR(struct au1xpsc_audio_data *ctx, int reg, unsigned long v) +{ + __raw_writel(v, ctx->mmio + reg); + wmb(); +} + +static int au1xi2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) +{ + struct au1xpsc_audio_data *ctx = snd_soc_dai_get_drvdata(cpu_dai); + unsigned long c; + int ret; + + ret = -EINVAL; + c = ctx->cfg; + + c &= ~CFG_FM_MASK; + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + c |= CFG_FM_I2S; + break; + case SND_SOC_DAIFMT_MSB: + c |= CFG_FM_RJ; + break; + case SND_SOC_DAIFMT_LSB: + c |= CFG_FM_LJ; + break; + default: + goto out; + } + + c &= ~(CFG_IC | CFG_ICK); /* IB-IF */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + c |= CFG_IC | CFG_ICK; + break; + case SND_SOC_DAIFMT_NB_IF: + c |= CFG_IC; + break; + case SND_SOC_DAIFMT_IB_NF: + c |= CFG_ICK; + break; + case SND_SOC_DAIFMT_IB_IF: + break; + default: + goto out; + } + + /* I2S controller only supports master */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: /* CODEC slave */ + break; + default: + goto out; + } + + ret = 0; + ctx->cfg = c; +out: + return ret; +} + +static int au1xi2s_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) +{ + struct au1xpsc_audio_data *ctx = snd_soc_dai_get_drvdata(dai); + int stype = SUBSTREAM_TYPE(substream); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + /* power up */ + WR(ctx, I2S_ENABLE, EN_D | EN_CE); + WR(ctx, I2S_ENABLE, EN_CE); + ctx->cfg |= (stype == PCM_TX) ? CFG_TN : CFG_RN; + WR(ctx, I2S_CFG, ctx->cfg); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + ctx->cfg &= ~((stype == PCM_TX) ? CFG_TN : CFG_RN); + WR(ctx, I2S_CFG, ctx->cfg); + WR(ctx, I2S_ENABLE, EN_D); /* power off */ + break; + default: + return -EINVAL; + } + + return 0; +} + +static unsigned long msbits_to_reg(int msbits) +{ + switch (msbits) { + case 8: + return CFG_SZ_8; + case 16: + return CFG_SZ_16; + case 18: + return CFG_SZ_18; + case 20: + return CFG_SZ_20; + case 24: + return CFG_SZ_24; + } + return 0; +} + +static int au1xi2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct au1xpsc_audio_data *ctx = snd_soc_dai_get_drvdata(dai); + unsigned long v; + + v = msbits_to_reg(params->msbits); + if (!v) + return -EINVAL; + + ctx->cfg &= ~CFG_SZ_MASK; + ctx->cfg |= v; + return 0; +} + +static int au1xi2s_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct au1xpsc_audio_data *ctx = snd_soc_dai_get_drvdata(dai); + snd_soc_dai_set_dma_data(dai, substream, &ctx->dmaids[0]); + return 0; +} + +static const struct snd_soc_dai_ops au1xi2s_dai_ops = { + .startup = au1xi2s_startup, + .trigger = au1xi2s_trigger, + .hw_params = au1xi2s_hw_params, + .set_fmt = au1xi2s_set_fmt, +}; + +static struct snd_soc_dai_driver au1xi2s_dai_driver = { + .symmetric_rates = 1, + .playback = { + .rates = AU1XI2SC_RATES, + .formats = AU1XI2SC_FMTS, + .channels_min = 2, + .channels_max = 2, + }, + .capture = { + .rates = AU1XI2SC_RATES, + .formats = AU1XI2SC_FMTS, + .channels_min = 2, + .channels_max = 2, + }, + .ops = &au1xi2s_dai_ops, +}; + +static int __devinit au1xi2s_drvprobe(struct platform_device *pdev) +{ + int ret; + struct resource *iores, *dmares; + struct au1xpsc_audio_data *ctx; + + ctx = kzalloc(sizeof(*ctx), GFP_KERNEL); + if (!ctx) + return -ENOMEM; + + iores = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!iores) { + ret = -ENODEV; + goto out0; + } + + ret = -EBUSY; + if (!request_mem_region(iores->start, resource_size(iores), + pdev->name)) + goto out0; + + ctx->mmio = ioremap_nocache(iores->start, resource_size(iores)); + if (!ctx->mmio) + goto out1; + + dmares = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!dmares) + goto out2; + ctx->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = dmares->start; + + dmares = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!dmares) + goto out2; + ctx->dmaids[SNDRV_PCM_STREAM_CAPTURE] = dmares->start; + + platform_set_drvdata(pdev, ctx); + + ret = snd_soc_register_dai(&pdev->dev, &au1xi2s_dai_driver); + if (ret) + goto out2; + + return 0; + +out2: + iounmap(ctx->mmio); +out1: + release_mem_region(iores->start, resource_size(iores)); +out0: + kfree(ctx); + return ret; +} + +static int __devexit au1xi2s_drvremove(struct platform_device *pdev) +{ + struct au1xpsc_audio_data *ctx = platform_get_drvdata(pdev); + struct resource *r = platform_get_resource(pdev, IORESOURCE_MEM, 0); + + snd_soc_unregister_dai(&pdev->dev); + + WR(ctx, I2S_ENABLE, EN_D); /* clock off, disable */ + + iounmap(ctx->mmio); + release_mem_region(r->start, resource_size(r)); + kfree(ctx); + + return 0; +} + +#ifdef CONFIG_PM +static int au1xi2s_drvsuspend(struct device *dev) +{ + struct au1xpsc_audio_data *ctx = dev_get_drvdata(dev); + + WR(ctx, I2S_ENABLE, EN_D); /* clock off, disable */ + + return 0; +} + +static int au1xi2s_drvresume(struct device *dev) +{ + return 0; +} + +static const struct dev_pm_ops au1xi2sc_pmops = { + .suspend = au1xi2s_drvsuspend, + .resume = au1xi2s_drvresume, +}; + +#define AU1XI2SC_PMOPS (&au1xi2sc_pmops) + +#else + +#define AU1XI2SC_PMOPS NULL + +#endif + +static struct platform_driver au1xi2s_driver = { + .driver = { + .name = "alchemy-i2sc", + .owner = THIS_MODULE, + .pm = AU1XI2SC_PMOPS, + }, + .probe = au1xi2s_drvprobe, + .remove = __devexit_p(au1xi2s_drvremove), +}; + +static int __init au1xi2s_load(void) +{ + return platform_driver_register(&au1xi2s_driver); +} + +static void __exit au1xi2s_unload(void) +{ + platform_driver_unregister(&au1xi2s_driver); +} + +module_init(au1xi2s_load); +module_exit(au1xi2s_unload); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("Au1000/1500/1100 I2S ASoC driver"); +MODULE_AUTHOR("Manuel Lauss"); diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c index d0db66f24a00..0c6acd547141 100644 --- a/sound/soc/au1x/psc-ac97.c +++ b/sound/soc/au1x/psc-ac97.c @@ -41,14 +41,14 @@ (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3BE) #define AC97PCR_START(stype) \ - ((stype) == PCM_TX ? PSC_AC97PCR_TS : PSC_AC97PCR_RS) + ((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_AC97PCR_TS : PSC_AC97PCR_RS) #define AC97PCR_STOP(stype) \ - ((stype) == PCM_TX ? PSC_AC97PCR_TP : PSC_AC97PCR_RP) + ((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_AC97PCR_TP : PSC_AC97PCR_RP) #define AC97PCR_CLRFIFO(stype) \ - ((stype) == PCM_TX ? PSC_AC97PCR_TC : PSC_AC97PCR_RC) + ((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_AC97PCR_TC : PSC_AC97PCR_RC) #define AC97STAT_BUSY(stype) \ - ((stype) == PCM_TX ? PSC_AC97STAT_TB : PSC_AC97STAT_RB) + ((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_AC97STAT_TB : PSC_AC97STAT_RB) /* instance data. There can be only one, MacLeod!!!! */ static struct au1xpsc_audio_data *au1xpsc_ac97_workdata; @@ -215,7 +215,7 @@ static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream, { struct au1xpsc_audio_data *pscdata = snd_soc_dai_get_drvdata(dai); unsigned long r, ro, stat; - int chans, t, stype = SUBSTREAM_TYPE(substream); + int chans, t, stype = substream->stream; chans = params_channels(params); @@ -235,7 +235,7 @@ static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream, r |= PSC_AC97CFG_SET_LEN(params->msbits); /* channels: enable slots for front L/R channel */ - if (stype == PCM_TX) { + if (stype == SNDRV_PCM_STREAM_PLAYBACK) { r &= ~PSC_AC97CFG_TXSLOT_MASK; r |= PSC_AC97CFG_TXSLOT_ENA(3); r |= PSC_AC97CFG_TXSLOT_ENA(4); @@ -294,7 +294,7 @@ static int au1xpsc_ac97_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { struct au1xpsc_audio_data *pscdata = snd_soc_dai_get_drvdata(dai); - int ret, stype = SUBSTREAM_TYPE(substream); + int ret, stype = substream->stream; ret = 0; @@ -324,12 +324,21 @@ static int au1xpsc_ac97_trigger(struct snd_pcm_substream *substream, return ret; } +static int au1xpsc_ac97_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct au1xpsc_audio_data *pscdata = snd_soc_dai_get_drvdata(dai); + snd_soc_dai_set_dma_data(dai, substream, &pscdata->dmaids[0]); + return 0; +} + static int au1xpsc_ac97_probe(struct snd_soc_dai *dai) { return au1xpsc_ac97_workdata ? 0 : -ENODEV; } static struct snd_soc_dai_ops au1xpsc_ac97_dai_ops = { + .startup = au1xpsc_ac97_startup, .trigger = au1xpsc_ac97_trigger, .hw_params = au1xpsc_ac97_hw_params, }; @@ -355,7 +364,7 @@ static const struct snd_soc_dai_driver au1xpsc_ac97_dai_template = { static int __devinit au1xpsc_ac97_drvprobe(struct platform_device *pdev) { int ret; - struct resource *r; + struct resource *iores, *dmares; unsigned long sel; struct au1xpsc_audio_data *wd; @@ -365,20 +374,31 @@ static int __devinit au1xpsc_ac97_drvprobe(struct platform_device *pdev) mutex_init(&wd->lock); - r = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!r) { + iores = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!iores) { ret = -ENODEV; goto out0; } ret = -EBUSY; - if (!request_mem_region(r->start, resource_size(r), pdev->name)) + if (!request_mem_region(iores->start, resource_size(iores), + pdev->name)) goto out0; - wd->mmio = ioremap(r->start, resource_size(r)); + wd->mmio = ioremap(iores->start, resource_size(iores)); if (!wd->mmio) goto out1; + dmares = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!dmares) + goto out2; + wd->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = dmares->start; + + dmares = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!dmares) + goto out2; + wd->dmaids[SNDRV_PCM_STREAM_CAPTURE] = dmares->start; + /* configuration: max dma trigger threshold, enable ac97 */ wd->cfg = PSC_AC97CFG_RT_FIFO8 | PSC_AC97CFG_TT_FIFO8 | PSC_AC97CFG_DE_ENABLE; @@ -401,17 +421,15 @@ static int __devinit au1xpsc_ac97_drvprobe(struct platform_device *pdev) ret = snd_soc_register_dai(&pdev->dev, &wd->dai_drv); if (ret) - goto out1; + goto out2; - wd->dmapd = au1xpsc_pcm_add(pdev); - if (wd->dmapd) { - au1xpsc_ac97_workdata = wd; - return 0; - } + au1xpsc_ac97_workdata = wd; + return 0; - snd_soc_unregister_dai(&pdev->dev); +out2: + iounmap(wd->mmio); out1: - release_mem_region(r->start, resource_size(r)); + release_mem_region(iores->start, resource_size(iores)); out0: kfree(wd); return ret; @@ -422,9 +440,6 @@ static int __devexit au1xpsc_ac97_drvremove(struct platform_device *pdev) struct au1xpsc_audio_data *wd = platform_get_drvdata(pdev); struct resource *r = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (wd->dmapd) - au1xpsc_pcm_destroy(wd->dmapd); - snd_soc_unregister_dai(&pdev->dev); /* disable PSC completely */ diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c index fca091276320..e03c5ce01b30 100644 --- a/sound/soc/au1x/psc-i2s.c +++ b/sound/soc/au1x/psc-i2s.c @@ -42,13 +42,13 @@ (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) #define I2SSTAT_BUSY(stype) \ - ((stype) == PCM_TX ? PSC_I2SSTAT_TB : PSC_I2SSTAT_RB) + ((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_I2SSTAT_TB : PSC_I2SSTAT_RB) #define I2SPCR_START(stype) \ - ((stype) == PCM_TX ? PSC_I2SPCR_TS : PSC_I2SPCR_RS) + ((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_I2SPCR_TS : PSC_I2SPCR_RS) #define I2SPCR_STOP(stype) \ - ((stype) == PCM_TX ? PSC_I2SPCR_TP : PSC_I2SPCR_RP) + ((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_I2SPCR_TP : PSC_I2SPCR_RP) #define I2SPCR_CLRFIFO(stype) \ - ((stype) == PCM_TX ? PSC_I2SPCR_TC : PSC_I2SPCR_RC) + ((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_I2SPCR_TC : PSC_I2SPCR_RC) static int au1xpsc_i2s_set_fmt(struct snd_soc_dai *cpu_dai, @@ -240,7 +240,7 @@ static int au1xpsc_i2s_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { struct au1xpsc_audio_data *pscdata = snd_soc_dai_get_drvdata(dai); - int ret, stype = SUBSTREAM_TYPE(substream); + int ret, stype = substream->stream; switch (cmd) { case SNDRV_PCM_TRIGGER_START: @@ -257,7 +257,16 @@ static int au1xpsc_i2s_trigger(struct snd_pcm_substream *substream, int cmd, return ret; } +static int au1xpsc_i2s_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct au1xpsc_audio_data *pscdata = snd_soc_dai_get_drvdata(dai); + snd_soc_dai_set_dma_data(dai, substream, &pscdata->dmaids[0]); + return 0; +} + static struct snd_soc_dai_ops au1xpsc_i2s_dai_ops = { + .startup = au1xpsc_i2s_startup, .trigger = au1xpsc_i2s_trigger, .hw_params = au1xpsc_i2s_hw_params, .set_fmt = au1xpsc_i2s_set_fmt, @@ -281,7 +290,7 @@ static const struct snd_soc_dai_driver au1xpsc_i2s_dai_template = { static int __devinit au1xpsc_i2s_drvprobe(struct platform_device *pdev) { - struct resource *r; + struct resource *iores, *dmares; unsigned long sel; int ret; struct au1xpsc_audio_data *wd; @@ -290,20 +299,31 @@ static int __devinit au1xpsc_i2s_drvprobe(struct platform_device *pdev) if (!wd) return -ENOMEM; - r = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!r) { + iores = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!iores) { ret = -ENODEV; goto out0; } ret = -EBUSY; - if (!request_mem_region(r->start, resource_size(r), pdev->name)) + if (!request_mem_region(iores->start, resource_size(iores), + pdev->name)) goto out0; - wd->mmio = ioremap(r->start, resource_size(r)); + wd->mmio = ioremap(iores->start, resource_size(iores)); if (!wd->mmio) goto out1; + dmares = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!dmares) + goto out2; + wd->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = dmares->start; + + dmares = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!dmares) + goto out2; + wd->dmaids[SNDRV_PCM_STREAM_CAPTURE] = dmares->start; + /* preserve PSC clock source set up by platform (dev.platform_data * is already occupied by soc layer) */ @@ -330,17 +350,13 @@ static int __devinit au1xpsc_i2s_drvprobe(struct platform_device *pdev) platform_set_drvdata(pdev, wd); ret = snd_soc_register_dai(&pdev->dev, &wd->dai_drv); - if (ret) - goto out1; - - /* finally add the DMA device for this PSC */ - wd->dmapd = au1xpsc_pcm_add(pdev); - if (wd->dmapd) + if (!ret) return 0; - snd_soc_unregister_dai(&pdev->dev); +out2: + iounmap(wd->mmio); out1: - release_mem_region(r->start, resource_size(r)); + release_mem_region(iores->start, resource_size(iores)); out0: kfree(wd); return ret; @@ -351,9 +367,6 @@ static int __devexit au1xpsc_i2s_drvremove(struct platform_device *pdev) struct au1xpsc_audio_data *wd = platform_get_drvdata(pdev); struct resource *r = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (wd->dmapd) - au1xpsc_pcm_destroy(wd->dmapd); - snd_soc_unregister_dai(&pdev->dev); au_writel(0, I2S_CFG(wd)); diff --git a/sound/soc/au1x/psc.h b/sound/soc/au1x/psc.h index b30eadd422a7..b16b2e02e0c9 100644 --- a/sound/soc/au1x/psc.h +++ b/sound/soc/au1x/psc.h @@ -1,7 +1,7 @@ /* - * Au12x0/Au1550 PSC ALSA ASoC audio support. + * Alchemy ALSA ASoC audio support. * - * (c) 2007-2008 MSC Vertriebsges.m.b.H., + * (c) 2007-2011 MSC Vertriebsges.m.b.H., * Manuel Lauss <manuel.lauss@gmail.com> * * This program is free software; you can redistribute it and/or modify @@ -13,10 +13,6 @@ #ifndef _AU1X_PCM_H #define _AU1X_PCM_H -/* DBDMA helpers */ -extern struct platform_device *au1xpsc_pcm_add(struct platform_device *pdev); -extern void au1xpsc_pcm_destroy(struct platform_device *dmapd); - struct au1xpsc_audio_data { void __iomem *mmio; @@ -27,15 +23,9 @@ struct au1xpsc_audio_data { unsigned long pm[2]; struct mutex lock; - struct platform_device *dmapd; + int dmaids[2]; }; -#define PCM_TX 0 -#define PCM_RX 1 - -#define SUBSTREAM_TYPE(substream) \ - ((substream)->stream == SNDRV_PCM_STREAM_PLAYBACK ? PCM_TX : PCM_RX) - /* easy access macros */ #define PSC_CTRL(x) ((unsigned long)((x)->mmio) + PSC_CTRL_OFFSET) #define PSC_SEL(x) ((unsigned long)((x)->mmio) + PSC_SEL_OFFSET) diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig index fe9d548a6837..9f6bc55fc399 100644 --- a/sound/soc/blackfin/Kconfig +++ b/sound/soc/blackfin/Kconfig @@ -27,6 +27,19 @@ config SND_SOC_BFIN_EVAL_ADAU1701 board connected to one of the Blackfin evaluation boards like the BF5XX-STAMP or BF5XX-EZKIT. +config SND_SOC_BFIN_EVAL_ADAU1373 + tristate "Support for the EVAL-ADAU1373 board on Blackfin eval boards" + depends on SND_BF5XX_I2S && I2C + select SND_BF5XX_SOC_I2S + select SND_SOC_ADAU1373 + help + Say Y if you want to add support for the Analog Devices EVAL-ADAU1373 + board connected to one of the Blackfin evaluation boards like the + BF5XX-STAMP or BF5XX-EZKIT. + + Note: This driver assumes that first ADAU1373 DAI is connected to the + first SPORT port on the BF5XX board. + config SND_SOC_BFIN_EVAL_ADAV80X tristate "Support for the EVAL-ADAV80X boards on Blackfin eval boards" depends on SND_BF5XX_I2S && (SPI_MASTER || I2C) diff --git a/sound/soc/blackfin/Makefile b/sound/soc/blackfin/Makefile index 6018bf52a234..1bf86ccaa8de 100644 --- a/sound/soc/blackfin/Makefile +++ b/sound/soc/blackfin/Makefile @@ -21,6 +21,7 @@ snd-ad1980-objs := bf5xx-ad1980.o snd-ssm2602-objs := bf5xx-ssm2602.o snd-ad73311-objs := bf5xx-ad73311.o snd-ad193x-objs := bf5xx-ad193x.o +snd-soc-bfin-eval-adau1373-objs := bfin-eval-adau1373.o snd-soc-bfin-eval-adau1701-objs := bfin-eval-adau1701.o snd-soc-bfin-eval-adav80x-objs := bfin-eval-adav80x.o @@ -29,5 +30,6 @@ obj-$(CONFIG_SND_BF5XX_SOC_AD1980) += snd-ad1980.o obj-$(CONFIG_SND_BF5XX_SOC_SSM2602) += snd-ssm2602.o obj-$(CONFIG_SND_BF5XX_SOC_AD73311) += snd-ad73311.o obj-$(CONFIG_SND_BF5XX_SOC_AD193X) += snd-ad193x.o +obj-$(CONFIG_SND_SOC_BFIN_EVAL_ADAU1373) += snd-soc-bfin-eval-adau1373.o obj-$(CONFIG_SND_SOC_BFIN_EVAL_ADAU1701) += snd-soc-bfin-eval-adau1701.o obj-$(CONFIG_SND_SOC_BFIN_EVAL_ADAV80X) += snd-soc-bfin-eval-adav80x.o diff --git a/sound/soc/blackfin/bf5xx-ac97-pcm.c b/sound/soc/blackfin/bf5xx-ac97-pcm.c index 9e59f680bc19..56815c1d47b3 100644 --- a/sound/soc/blackfin/bf5xx-ac97-pcm.c +++ b/sound/soc/blackfin/bf5xx-ac97-pcm.c @@ -418,7 +418,7 @@ static void bf5xx_pcm_free_dma_buffers(struct snd_pcm *pcm) static u64 bf5xx_pcm_dmamask = DMA_BIT_MASK(32); -int bf5xx_pcm_ac97_new(struct snd_soc_pcm_runtime *rtd) +static int bf5xx_pcm_ac97_new(struct snd_soc_pcm_runtime *rtd) { struct snd_card *card = rtd->card->snd_card; struct snd_soc_dai *dai = rtd->cpu_dai; diff --git a/sound/soc/blackfin/bf5xx-i2s-pcm.c b/sound/soc/blackfin/bf5xx-i2s-pcm.c index 61ddf942fd4d..7565e1576ffa 100644 --- a/sound/soc/blackfin/bf5xx-i2s-pcm.c +++ b/sound/soc/blackfin/bf5xx-i2s-pcm.c @@ -257,7 +257,7 @@ static void bf5xx_pcm_free_dma_buffers(struct snd_pcm *pcm) static u64 bf5xx_pcm_dmamask = DMA_BIT_MASK(32); -int bf5xx_pcm_i2s_new(struct snd_soc_pcm_runtime *rtd) +static int bf5xx_pcm_i2s_new(struct snd_soc_pcm_runtime *rtd) { struct snd_card *card = rtd->card->snd_card; struct snd_soc_dai *dai = rtd->cpu_dai; diff --git a/sound/soc/blackfin/bfin-eval-adau1373.c b/sound/soc/blackfin/bfin-eval-adau1373.c new file mode 100644 index 000000000000..8df2a3b0cb36 --- /dev/null +++ b/sound/soc/blackfin/bfin-eval-adau1373.c @@ -0,0 +1,202 @@ +/* + * Machine driver for EVAL-ADAU1373 on Analog Devices bfin + * evaluation boards. + * + * Copyright 2011 Analog Devices Inc. + * Author: Lars-Peter Clausen <lars@metafoo.de> + * + * Licensed under the GPL-2 or later. + */ + +#include <linux/module.h> +#include <linux/device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/pcm_params.h> + +#include "../codecs/adau1373.h" + +static const struct snd_soc_dapm_widget bfin_eval_adau1373_dapm_widgets[] = { + SND_SOC_DAPM_LINE("Line In1", NULL), + SND_SOC_DAPM_LINE("Line In2", NULL), + SND_SOC_DAPM_LINE("Line In3", NULL), + SND_SOC_DAPM_LINE("Line In4", NULL), + + SND_SOC_DAPM_LINE("Line Out1", NULL), + SND_SOC_DAPM_LINE("Line Out2", NULL), + SND_SOC_DAPM_LINE("Stereo Out", NULL), + SND_SOC_DAPM_HP("Headphone", NULL), + SND_SOC_DAPM_HP("Earpiece", NULL), + SND_SOC_DAPM_SPK("Speaker", NULL), +}; + +static const struct snd_soc_dapm_route bfin_eval_adau1373_dapm_routes[] = { + { "AIN1L", NULL, "Line In1" }, + { "AIN1R", NULL, "Line In1" }, + { "AIN2L", NULL, "Line In2" }, + { "AIN2R", NULL, "Line In2" }, + { "AIN3L", NULL, "Line In3" }, + { "AIN3R", NULL, "Line In3" }, + { "AIN4L", NULL, "Line In4" }, + { "AIN4R", NULL, "Line In4" }, + + /* MICBIAS can be connected via a jumper to the line-in jack, since w + don't know which one is going to be used, just power both. */ + { "Line In1", NULL, "MICBIAS1" }, + { "Line In2", NULL, "MICBIAS1" }, + { "Line In3", NULL, "MICBIAS1" }, + { "Line In4", NULL, "MICBIAS1" }, + { "Line In1", NULL, "MICBIAS2" }, + { "Line In2", NULL, "MICBIAS2" }, + { "Line In3", NULL, "MICBIAS2" }, + { "Line In4", NULL, "MICBIAS2" }, + + { "Line Out1", NULL, "LOUT1L" }, + { "Line Out1", NULL, "LOUT1R" }, + { "Line Out2", NULL, "LOUT2L" }, + { "Line Out2", NULL, "LOUT2R" }, + { "Headphone", NULL, "HPL" }, + { "Headphone", NULL, "HPR" }, + { "Earpiece", NULL, "EP" }, + { "Speaker", NULL, "SPKL" }, + { "Stereo Out", NULL, "SPKR" }, +}; + +static int bfin_eval_adau1373_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret; + int pll_rate; + + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret) + return ret; + + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret) + return ret; + + switch (params_rate(params)) { + case 48000: + case 8000: + case 12000: + case 16000: + case 24000: + case 32000: + pll_rate = 48000 * 1024; + break; + case 44100: + case 7350: + case 11025: + case 14700: + case 22050: + case 29400: + pll_rate = 44100 * 1024; + break; + default: + return -EINVAL; + } + + ret = snd_soc_dai_set_pll(codec_dai, ADAU1373_PLL1, + ADAU1373_PLL_SRC_MCLK1, 12288000, pll_rate); + if (ret) + return ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, ADAU1373_CLK_SRC_PLL1, pll_rate, + SND_SOC_CLOCK_IN); + + return ret; +} + +static int bfin_eval_adau1373_codec_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_dai *codec_dai = rtd->codec_dai; + unsigned int pll_rate = 48000 * 1024; + int ret; + + ret = snd_soc_dai_set_pll(codec_dai, ADAU1373_PLL1, + ADAU1373_PLL_SRC_MCLK1, 12288000, pll_rate); + if (ret) + return ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, ADAU1373_CLK_SRC_PLL1, pll_rate, + SND_SOC_CLOCK_IN); + + return ret; +} +static struct snd_soc_ops bfin_eval_adau1373_ops = { + .hw_params = bfin_eval_adau1373_hw_params, +}; + +static struct snd_soc_dai_link bfin_eval_adau1373_dai = { + .name = "adau1373", + .stream_name = "adau1373", + .cpu_dai_name = "bfin-i2s.0", + .codec_dai_name = "adau1373-aif1", + .platform_name = "bfin-i2s-pcm-audio", + .codec_name = "adau1373.0-001a", + .ops = &bfin_eval_adau1373_ops, + .init = bfin_eval_adau1373_codec_init, +}; + +static struct snd_soc_card bfin_eval_adau1373 = { + .name = "bfin-eval-adau1373", + .dai_link = &bfin_eval_adau1373_dai, + .num_links = 1, + + .dapm_widgets = bfin_eval_adau1373_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(bfin_eval_adau1373_dapm_widgets), + .dapm_routes = bfin_eval_adau1373_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(bfin_eval_adau1373_dapm_routes), +}; + +static int bfin_eval_adau1373_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &bfin_eval_adau1373; + + card->dev = &pdev->dev; + + return snd_soc_register_card(&bfin_eval_adau1373); +} + +static int __devexit bfin_eval_adau1373_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + + snd_soc_unregister_card(card); + + return 0; +} + +static struct platform_driver bfin_eval_adau1373_driver = { + .driver = { + .name = "bfin-eval-adau1373", + .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, + }, + .probe = bfin_eval_adau1373_probe, + .remove = __devexit_p(bfin_eval_adau1373_remove), +}; + +static int __init bfin_eval_adau1373_init(void) +{ + return platform_driver_register(&bfin_eval_adau1373_driver); +} +module_init(bfin_eval_adau1373_init); + +static void __exit bfin_eval_adau1373_exit(void) +{ + platform_driver_unregister(&bfin_eval_adau1373_driver); +} +module_exit(bfin_eval_adau1373_exit); + +MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>"); +MODULE_DESCRIPTION("ALSA SoC bfin adau1373 driver"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:bfin-eval-adau1373"); diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c index 19241576b6b5..5ca122e51183 100644 --- a/sound/soc/codecs/88pm860x-codec.c +++ b/sound/soc/codecs/88pm860x-codec.c @@ -15,6 +15,7 @@ #include <linux/platform_device.h> #include <linux/mfd/88pm860x.h> #include <linux/slab.h> +#include <linux/delay.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -772,11 +773,12 @@ static const struct snd_soc_dapm_widget pm860x_dapm_widgets[] = { SND_SOC_DAPM_AIF_IN("I2S DIN", "I2S Playback", 0, - PM860X_DAC_EN_2, 0, 0), + SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("I2S DIN1", "I2S Playback", 0, - PM860X_DAC_EN_2, 0, 0), + SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("I2S DOUT", "I2S Capture", 0, PM860X_I2S_IFACE_3, 5, 1), + SND_SOC_DAPM_SUPPLY("I2S CLK", PM860X_DAC_EN_2, 0, 0, NULL, 0), SND_SOC_DAPM_MUX("I2S Mic Mux", SND_SOC_NOPM, 0, 0, &i2s_mic_mux), SND_SOC_DAPM_MUX("ADC Left Mux", SND_SOC_NOPM, 0, 0, &adcl_mux), SND_SOC_DAPM_MUX("ADC Right Mux", SND_SOC_NOPM, 0, 0, &adcr_mux), @@ -868,6 +870,11 @@ static const struct snd_soc_dapm_route audio_map[] = { {"Left ADC", NULL, "Left ADC MOD"}, {"Right ADC", NULL, "Right ADC MOD"}, + /* I2S Clock */ + {"I2S DIN", NULL, "I2S CLK"}, + {"I2S DIN1", NULL, "I2S CLK"}, + {"I2S DOUT", NULL, "I2S CLK"}, + /* PCM/AIF1 Inputs */ {"PCM SDO", NULL, "ADC Left Mux"}, {"PCM SDO", NULL, "ADCR EC Mux"}, @@ -1173,6 +1180,9 @@ static int pm860x_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_STANDBY: if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { /* Enable Audio PLL & Audio section */ + data = AUDIO_PLL | AUDIO_SECTION_ON; + pm860x_reg_write(codec->control_data, REG_MISC2, data); + udelay(300); data = AUDIO_PLL | AUDIO_SECTION_RESET | AUDIO_SECTION_ON; pm860x_reg_write(codec->control_data, REG_MISC2, data); diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 665d9240c4ae..4584514d93d4 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -17,6 +17,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AD193X if SND_SOC_I2C_AND_SPI select SND_SOC_AD1980 if SND_SOC_AC97_BUS select SND_SOC_AD73311 + select SND_SOC_ADAU1373 if I2C select SND_SOC_ADAV80X select SND_SOC_ADS117X select SND_SOC_AK4104 if SPI_MASTER @@ -39,6 +40,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX9850 if I2C select SND_SOC_MAX9877 if I2C select SND_SOC_PCM3008 + select SND_SOC_RT5631 if I2C select SND_SOC_SGTL5000 if I2C select SND_SOC_SN95031 if INTEL_SCU_IPC select SND_SOC_SPDIF @@ -47,7 +49,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_STAC9766 if SND_SOC_AC97_BUS select SND_SOC_TLV320AIC23 if I2C select SND_SOC_TLV320AIC26 if SPI_MASTER - select SND_SOC_TVL320AIC32X4 if I2C + select SND_SOC_TLV320AIC32X4 if I2C select SND_SOC_TLV320AIC3X if I2C select SND_SOC_TPA6130A2 if I2C select SND_SOC_TLV320DAC33 if I2C @@ -58,6 +60,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_WL1273 if MFD_WL1273_CORE select SND_SOC_WM1250_EV1 if I2C select SND_SOC_WM2000 if I2C + select SND_SOC_WM5100 if I2C select SND_SOC_WM8350 if MFD_WM8350 select SND_SOC_WM8400 if MFD_WM8400 select SND_SOC_WM8510 if SND_SOC_I2C_AND_SPI @@ -139,6 +142,9 @@ config SND_SOC_ADAU1701 select SIGMA tristate +config SND_SOC_ADAU1373 + tristate + config SND_SOC_ADAV80X tristate @@ -214,6 +220,9 @@ config SND_SOC_MAX9850 config SND_SOC_PCM3008 tristate +config SND_SOC_RT5631 + tristate + #Freescale sgtl5000 codec config SND_SOC_SGTL5000 tristate @@ -240,7 +249,7 @@ config SND_SOC_TLV320AIC26 tristate "TI TLV320AIC26 Codec support" if SND_SOC_OF_SIMPLE depends on SPI -config SND_SOC_TVL320AIC32X4 +config SND_SOC_TLV320AIC32X4 tristate config SND_SOC_TLV320AIC3X @@ -269,6 +278,9 @@ config SND_SOC_WL1273 config SND_SOC_WM1250_EV1 tristate +config SND_SOC_WM5100 + tristate + config SND_SOC_WM8350 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 5119a7e2c1a8..a2c7842e357b 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -5,6 +5,7 @@ snd-soc-ad193x-objs := ad193x.o snd-soc-ad1980-objs := ad1980.o snd-soc-ad73311-objs := ad73311.o snd-soc-adau1701-objs := adau1701.o +snd-soc-adau1373-objs := adau1373.o snd-soc-adav80x-objs := adav80x.o snd-soc-ads117x-objs := ads117x.o snd-soc-ak4104-objs := ak4104.o @@ -25,6 +26,7 @@ snd-soc-max98088-objs := max98088.o snd-soc-max98095-objs := max98095.o snd-soc-max9850-objs := max9850.o snd-soc-pcm3008-objs := pcm3008.o +snd-soc-rt5631-objs := rt5631.o snd-soc-sgtl5000-objs := sgtl5000.o snd-soc-alc5623-objs := alc5623.o snd-soc-sn95031-objs := sn95031.o @@ -43,6 +45,7 @@ snd-soc-uda134x-objs := uda134x.o snd-soc-uda1380-objs := uda1380.o snd-soc-wl1273-objs := wl1273.o snd-soc-wm1250-ev1-objs := wm1250-ev1.o +snd-soc-wm5100-objs := wm5100.o wm5100-tables.o snd-soc-wm8350-objs := wm8350.o snd-soc-wm8400-objs := wm8400.o snd-soc-wm8510-objs := wm8510.o @@ -100,6 +103,7 @@ obj-$(CONFIG_SND_SOC_AD1836) += snd-soc-ad1836.o obj-$(CONFIG_SND_SOC_AD193X) += snd-soc-ad193x.o obj-$(CONFIG_SND_SOC_AD1980) += snd-soc-ad1980.o obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o +obj-$(CONFIG_SND_SOC_ADAU1373) += snd-soc-adau1373.o obj-$(CONFIG_SND_SOC_ADAU1701) += snd-soc-adau1701.o obj-$(CONFIG_SND_SOC_ADAV80X) += snd-soc-adav80x.o obj-$(CONFIG_SND_SOC_ADS117X) += snd-soc-ads117x.o @@ -123,6 +127,7 @@ obj-$(CONFIG_SND_SOC_MAX98088) += snd-soc-max98088.o obj-$(CONFIG_SND_SOC_MAX98095) += snd-soc-max98095.o obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o +obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o obj-$(CONFIG_SND_SOC_SN95031) +=snd-soc-sn95031.o obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif.o @@ -132,7 +137,7 @@ obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o -obj-$(CONFIG_SND_SOC_TVL320AIC32X4) += snd-soc-tlv320aic32x4.o +obj-$(CONFIG_SND_SOC_TLV320AIC32X4) += snd-soc-tlv320aic32x4.o obj-$(CONFIG_SND_SOC_TLV320DAC33) += snd-soc-tlv320dac33.o obj-$(CONFIG_SND_SOC_TWL4030) += snd-soc-twl4030.o obj-$(CONFIG_SND_SOC_TWL6040) += snd-soc-twl6040.o @@ -140,6 +145,7 @@ obj-$(CONFIG_SND_SOC_UDA134X) += snd-soc-uda134x.o obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o obj-$(CONFIG_SND_SOC_WL1273) += snd-soc-wl1273.o obj-$(CONFIG_SND_SOC_WM1250_EV1) += snd-soc-wm1250-ev1.o +obj-$(CONFIG_SND_SOC_WM5100) += snd-soc-wm5100.o obj-$(CONFIG_SND_SOC_WM8350) += snd-soc-wm8350.o obj-$(CONFIG_SND_SOC_WM8400) += snd-soc-wm8400.o obj-$(CONFIG_SND_SOC_WM8510) += snd-soc-wm8510.o diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index eedb6f5e5823..120602130b5c 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -23,7 +23,7 @@ /* codec private data */ struct ad193x_priv { - enum snd_soc_control_type control_type; + struct regmap *regmap; int sysclk; }; @@ -103,12 +103,14 @@ static const struct snd_soc_dapm_route audio_paths[] = { static int ad193x_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; - int reg; - reg = snd_soc_read(codec, AD193X_DAC_CTRL2); - reg = (mute > 0) ? reg | AD193X_DAC_MASTER_MUTE : reg & - (~AD193X_DAC_MASTER_MUTE); - snd_soc_write(codec, AD193X_DAC_CTRL2, reg); + if (mute) + snd_soc_update_bits(codec, AD193X_DAC_CTRL2, + AD193X_DAC_MASTER_MUTE, + AD193X_DAC_MASTER_MUTE); + else + snd_soc_update_bits(codec, AD193X_DAC_CTRL2, + AD193X_DAC_MASTER_MUTE, 0); return 0; } @@ -262,7 +264,7 @@ static int ad193x_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - int word_len = 0, reg = 0, master_rate = 0; + int word_len = 0, master_rate = 0; struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_codec *codec = rtd->codec; @@ -297,18 +299,15 @@ static int ad193x_hw_params(struct snd_pcm_substream *substream, break; } - reg = snd_soc_read(codec, AD193X_PLL_CLK_CTRL0); - reg = (reg & AD193X_PLL_INPUT_MASK) | master_rate; - snd_soc_write(codec, AD193X_PLL_CLK_CTRL0, reg); + snd_soc_update_bits(codec, AD193X_PLL_CLK_CTRL0, + AD193X_PLL_INPUT_MASK, master_rate); - reg = snd_soc_read(codec, AD193X_DAC_CTRL2); - reg = (reg & (~AD193X_DAC_WORD_LEN_MASK)) - | (word_len << AD193X_DAC_WORD_LEN_SHFT); - snd_soc_write(codec, AD193X_DAC_CTRL2, reg); + snd_soc_update_bits(codec, AD193X_DAC_CTRL2, + AD193X_DAC_WORD_LEN_MASK, + word_len << AD193X_DAC_WORD_LEN_SHFT); - reg = snd_soc_read(codec, AD193X_ADC_CTRL1); - reg = (reg & (~AD193X_ADC_WORD_LEN_MASK)) | word_len; - snd_soc_write(codec, AD193X_ADC_CTRL1, reg); + snd_soc_update_bits(codec, AD193X_ADC_CTRL1, + AD193X_ADC_WORD_LEN_MASK, word_len); return 0; } @@ -349,10 +348,8 @@ static int ad193x_probe(struct snd_soc_codec *codec) struct snd_soc_dapm_context *dapm = &codec->dapm; int ret; - if (ad193x->control_type == SND_SOC_I2C) - ret = snd_soc_codec_set_cache_io(codec, 8, 8, ad193x->control_type); - else - ret = snd_soc_codec_set_cache_io(codec, 16, 8, ad193x->control_type); + codec->control_data = ad193x->regmap; + ret = snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "failed to set cache I/O: %d\n", ret); return ret; @@ -388,6 +385,14 @@ static struct snd_soc_codec_driver soc_codec_dev_ad193x = { }; #if defined(CONFIG_SPI_MASTER) + +static const struct regmap_config ad193x_spi_regmap_config = { + .val_bits = 8, + .reg_bits = 16, + .read_flag_mask = 0x09, + .write_flag_mask = 0x08, +}; + static int __devinit ad193x_spi_probe(struct spi_device *spi) { struct ad193x_priv *ad193x; @@ -397,20 +402,36 @@ static int __devinit ad193x_spi_probe(struct spi_device *spi) if (ad193x == NULL) return -ENOMEM; + ad193x->regmap = regmap_init_spi(spi, &ad193x_spi_regmap_config); + if (IS_ERR(ad193x->regmap)) { + ret = PTR_ERR(ad193x->regmap); + goto err_free; + } + spi_set_drvdata(spi, ad193x); - ad193x->control_type = SND_SOC_SPI; ret = snd_soc_register_codec(&spi->dev, &soc_codec_dev_ad193x, &ad193x_dai, 1); if (ret < 0) - kfree(ad193x); + goto err_regmap_exit; + + return 0; + +err_regmap_exit: + regmap_exit(ad193x->regmap); +err_free: + kfree(ad193x); + return ret; } static int __devexit ad193x_spi_remove(struct spi_device *spi) { + struct ad193x_priv *ad193x = spi_get_drvdata(spi); + snd_soc_unregister_codec(&spi->dev); - kfree(spi_get_drvdata(spi)); + regmap_exit(ad193x->regmap); + kfree(ad193x); return 0; } @@ -425,6 +446,12 @@ static struct spi_driver ad193x_spi_driver = { #endif #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + +static const struct regmap_config ad193x_i2c_regmap_config = { + .val_bits = 8, + .reg_bits = 8, +}; + static const struct i2c_device_id ad193x_id[] = { { "ad1936", 0 }, { "ad1937", 0 }, @@ -442,20 +469,35 @@ static int __devinit ad193x_i2c_probe(struct i2c_client *client, if (ad193x == NULL) return -ENOMEM; + ad193x->regmap = regmap_init_i2c(client, &ad193x_i2c_regmap_config); + if (IS_ERR(ad193x->regmap)) { + ret = PTR_ERR(ad193x->regmap); + goto err_free; + } + i2c_set_clientdata(client, ad193x); - ad193x->control_type = SND_SOC_I2C; ret = snd_soc_register_codec(&client->dev, &soc_codec_dev_ad193x, &ad193x_dai, 1); if (ret < 0) - kfree(ad193x); + goto err_regmap_exit; + + return 0; + +err_regmap_exit: + regmap_exit(ad193x->regmap); +err_free: + kfree(ad193x); return ret; } static int __devexit ad193x_i2c_remove(struct i2c_client *client) { + struct ad193x_priv *ad193x = i2c_get_clientdata(client); + snd_soc_unregister_codec(&client->dev); - kfree(i2c_get_clientdata(client)); + regmap_exit(ad193x->regmap); + kfree(ad193x); return 0; } diff --git a/sound/soc/codecs/ad193x.h b/sound/soc/codecs/ad193x.h index cccc2e8e5fbd..1507eaa425a3 100644 --- a/sound/soc/codecs/ad193x.h +++ b/sound/soc/codecs/ad193x.h @@ -9,20 +9,20 @@ #ifndef __AD193X_H__ #define __AD193X_H__ -#define AD193X_PLL_CLK_CTRL0 0x800 +#define AD193X_PLL_CLK_CTRL0 0x00 #define AD193X_PLL_POWERDOWN 0x01 -#define AD193X_PLL_INPUT_MASK (~0x6) +#define AD193X_PLL_INPUT_MASK 0x6 #define AD193X_PLL_INPUT_256 (0 << 1) #define AD193X_PLL_INPUT_384 (1 << 1) #define AD193X_PLL_INPUT_512 (2 << 1) #define AD193X_PLL_INPUT_768 (3 << 1) -#define AD193X_PLL_CLK_CTRL1 0x801 -#define AD193X_DAC_CTRL0 0x802 +#define AD193X_PLL_CLK_CTRL1 0x01 +#define AD193X_DAC_CTRL0 0x02 #define AD193X_DAC_POWERDOWN 0x01 #define AD193X_DAC_SERFMT_MASK 0xC0 #define AD193X_DAC_SERFMT_STEREO (0 << 6) #define AD193X_DAC_SERFMT_TDM (1 << 6) -#define AD193X_DAC_CTRL1 0x803 +#define AD193X_DAC_CTRL1 0x03 #define AD193X_DAC_2_CHANNELS 0 #define AD193X_DAC_4_CHANNELS 1 #define AD193X_DAC_8_CHANNELS 2 @@ -33,11 +33,11 @@ #define AD193X_DAC_BCLK_MASTER (1 << 5) #define AD193X_DAC_LEFT_HIGH (1 << 3) #define AD193X_DAC_BCLK_INV (1 << 7) -#define AD193X_DAC_CTRL2 0x804 +#define AD193X_DAC_CTRL2 0x04 #define AD193X_DAC_WORD_LEN_SHFT 3 #define AD193X_DAC_WORD_LEN_MASK 0x18 #define AD193X_DAC_MASTER_MUTE 1 -#define AD193X_DAC_CHNL_MUTE 0x805 +#define AD193X_DAC_CHNL_MUTE 0x05 #define AD193X_DACL1_MUTE 0 #define AD193X_DACR1_MUTE 1 #define AD193X_DACL2_MUTE 2 @@ -46,28 +46,28 @@ #define AD193X_DACR3_MUTE 5 #define AD193X_DACL4_MUTE 6 #define AD193X_DACR4_MUTE 7 -#define AD193X_DAC_L1_VOL 0x806 -#define AD193X_DAC_R1_VOL 0x807 -#define AD193X_DAC_L2_VOL 0x808 -#define AD193X_DAC_R2_VOL 0x809 -#define AD193X_DAC_L3_VOL 0x80a -#define AD193X_DAC_R3_VOL 0x80b -#define AD193X_DAC_L4_VOL 0x80c -#define AD193X_DAC_R4_VOL 0x80d -#define AD193X_ADC_CTRL0 0x80e +#define AD193X_DAC_L1_VOL 0x06 +#define AD193X_DAC_R1_VOL 0x07 +#define AD193X_DAC_L2_VOL 0x08 +#define AD193X_DAC_R2_VOL 0x09 +#define AD193X_DAC_L3_VOL 0x0a +#define AD193X_DAC_R3_VOL 0x0b +#define AD193X_DAC_L4_VOL 0x0c +#define AD193X_DAC_R4_VOL 0x0d +#define AD193X_ADC_CTRL0 0x0e #define AD193X_ADC_POWERDOWN 0x01 #define AD193X_ADC_HIGHPASS_FILTER 1 #define AD193X_ADCL1_MUTE 2 #define AD193X_ADCR1_MUTE 3 #define AD193X_ADCL2_MUTE 4 #define AD193X_ADCR2_MUTE 5 -#define AD193X_ADC_CTRL1 0x80f +#define AD193X_ADC_CTRL1 0x0f #define AD193X_ADC_SERFMT_MASK 0x60 #define AD193X_ADC_SERFMT_STEREO (0 << 5) #define AD193X_ADC_SERFMT_TDM (1 << 5) #define AD193X_ADC_SERFMT_AUX (2 << 5) #define AD193X_ADC_WORD_LEN_MASK 0x3 -#define AD193X_ADC_CTRL2 0x810 +#define AD193X_ADC_CTRL2 0x10 #define AD193X_ADC_2_CHANNELS 0 #define AD193X_ADC_4_CHANNELS 1 #define AD193X_ADC_8_CHANNELS 2 diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c index 923b364a3e41..e3931cc5e66c 100644 --- a/sound/soc/codecs/ad1980.c +++ b/sound/soc/codecs/ad1980.c @@ -148,7 +148,6 @@ static struct snd_soc_dai_driver ad1980_dai = { .rates = SNDRV_PCM_RATE_48000, .formats = SND_SOC_STD_AC97_FMTS, }, }; -EXPORT_SYMBOL_GPL(ad1980_dai); static int ad1980_reset(struct snd_soc_codec *codec, int try_warm) { @@ -200,18 +199,22 @@ static int ad1980_soc_probe(struct snd_soc_codec *codec) } /* Read out vendor ID to make sure it is ad1980 */ - if (ac97_read(codec, AC97_VENDOR_ID1) != 0x4144) + if (ac97_read(codec, AC97_VENDOR_ID1) != 0x4144) { + ret = -ENODEV; goto reset_err; + } vendor_id2 = ac97_read(codec, AC97_VENDOR_ID2); if (vendor_id2 != 0x5370) { - if (vendor_id2 != 0x5374) + if (vendor_id2 != 0x5374) { + ret = -ENODEV; goto reset_err; - else + } else { printk(KERN_WARNING "ad1980: " "Found AD1981 - only 2/2 IN/OUT Channels " "supported\n"); + } } /* unmute captures and playbacks volume */ diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c new file mode 100644 index 000000000000..1ccf8dd47576 --- /dev/null +++ b/sound/soc/codecs/adau1373.c @@ -0,0 +1,1414 @@ +/* + * Analog Devices ADAU1373 Audio Codec drive + * + * Copyright 2011 Analog Devices Inc. + * Author: Lars-Peter Clausen <lars@metafoo.de> + * + * Licensed under the GPL-2 or later. + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/slab.h> +#include <linux/gcd.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/tlv.h> +#include <sound/soc.h> +#include <sound/adau1373.h> + +#include "adau1373.h" + +struct adau1373_dai { + unsigned int clk_src; + unsigned int sysclk; + bool enable_src; + bool master; +}; + +struct adau1373 { + struct adau1373_dai dais[3]; +}; + +#define ADAU1373_INPUT_MODE 0x00 +#define ADAU1373_AINL_CTRL(x) (0x01 + (x) * 2) +#define ADAU1373_AINR_CTRL(x) (0x02 + (x) * 2) +#define ADAU1373_LLINE_OUT(x) (0x9 + (x) * 2) +#define ADAU1373_RLINE_OUT(x) (0xa + (x) * 2) +#define ADAU1373_LSPK_OUT 0x0d +#define ADAU1373_RSPK_OUT 0x0e +#define ADAU1373_LHP_OUT 0x0f +#define ADAU1373_RHP_OUT 0x10 +#define ADAU1373_ADC_GAIN 0x11 +#define ADAU1373_LADC_MIXER 0x12 +#define ADAU1373_RADC_MIXER 0x13 +#define ADAU1373_LLINE1_MIX 0x14 +#define ADAU1373_RLINE1_MIX 0x15 +#define ADAU1373_LLINE2_MIX 0x16 +#define ADAU1373_RLINE2_MIX 0x17 +#define ADAU1373_LSPK_MIX 0x18 +#define ADAU1373_RSPK_MIX 0x19 +#define ADAU1373_LHP_MIX 0x1a +#define ADAU1373_RHP_MIX 0x1b +#define ADAU1373_EP_MIX 0x1c +#define ADAU1373_HP_CTRL 0x1d +#define ADAU1373_HP_CTRL2 0x1e +#define ADAU1373_LS_CTRL 0x1f +#define ADAU1373_EP_CTRL 0x21 +#define ADAU1373_MICBIAS_CTRL1 0x22 +#define ADAU1373_MICBIAS_CTRL2 0x23 +#define ADAU1373_OUTPUT_CTRL 0x24 +#define ADAU1373_PWDN_CTRL1 0x25 +#define ADAU1373_PWDN_CTRL2 0x26 +#define ADAU1373_PWDN_CTRL3 0x27 +#define ADAU1373_DPLL_CTRL(x) (0x28 + (x) * 7) +#define ADAU1373_PLL_CTRL1(x) (0x29 + (x) * 7) +#define ADAU1373_PLL_CTRL2(x) (0x2a + (x) * 7) +#define ADAU1373_PLL_CTRL3(x) (0x2b + (x) * 7) +#define ADAU1373_PLL_CTRL4(x) (0x2c + (x) * 7) +#define ADAU1373_PLL_CTRL5(x) (0x2d + (x) * 7) +#define ADAU1373_PLL_CTRL6(x) (0x2e + (x) * 7) +#define ADAU1373_PLL_CTRL7(x) (0x2f + (x) * 7) +#define ADAU1373_HEADDECT 0x36 +#define ADAU1373_ADC_DAC_STATUS 0x37 +#define ADAU1373_ADC_CTRL 0x3c +#define ADAU1373_DAI(x) (0x44 + (x)) +#define ADAU1373_CLK_SRC_DIV(x) (0x40 + (x) * 2) +#define ADAU1373_BCLKDIV(x) (0x47 + (x)) +#define ADAU1373_SRC_RATIOA(x) (0x4a + (x) * 2) +#define ADAU1373_SRC_RATIOB(x) (0x4b + (x) * 2) +#define ADAU1373_DEEMP_CTRL 0x50 +#define ADAU1373_SRC_DAI_CTRL(x) (0x51 + (x)) +#define ADAU1373_DIN_MIX_CTRL(x) (0x56 + (x)) +#define ADAU1373_DOUT_MIX_CTRL(x) (0x5b + (x)) +#define ADAU1373_DAI_PBL_VOL(x) (0x62 + (x) * 2) +#define ADAU1373_DAI_PBR_VOL(x) (0x63 + (x) * 2) +#define ADAU1373_DAI_RECL_VOL(x) (0x68 + (x) * 2) +#define ADAU1373_DAI_RECR_VOL(x) (0x69 + (x) * 2) +#define ADAU1373_DAC1_PBL_VOL 0x6e +#define ADAU1373_DAC1_PBR_VOL 0x6f +#define ADAU1373_DAC2_PBL_VOL 0x70 +#define ADAU1373_DAC2_PBR_VOL 0x71 +#define ADAU1373_ADC_RECL_VOL 0x72 +#define ADAU1373_ADC_RECR_VOL 0x73 +#define ADAU1373_DMIC_RECL_VOL 0x74 +#define ADAU1373_DMIC_RECR_VOL 0x75 +#define ADAU1373_VOL_GAIN1 0x76 +#define ADAU1373_VOL_GAIN2 0x77 +#define ADAU1373_VOL_GAIN3 0x78 +#define ADAU1373_HPF_CTRL 0x7d +#define ADAU1373_BASS1 0x7e +#define ADAU1373_BASS2 0x7f +#define ADAU1373_DRC(x) (0x80 + (x) * 0x10) +#define ADAU1373_3D_CTRL1 0xc0 +#define ADAU1373_3D_CTRL2 0xc1 +#define ADAU1373_FDSP_SEL1 0xdc +#define ADAU1373_FDSP_SEL2 0xdd +#define ADAU1373_FDSP_SEL3 0xde +#define ADAU1373_FDSP_SEL4 0xdf +#define ADAU1373_DIGMICCTRL 0xe2 +#define ADAU1373_DIGEN 0xeb +#define ADAU1373_SOFT_RESET 0xff + + +#define ADAU1373_PLL_CTRL6_DPLL_BYPASS BIT(1) +#define ADAU1373_PLL_CTRL6_PLL_EN BIT(0) + +#define ADAU1373_DAI_INVERT_BCLK BIT(7) +#define ADAU1373_DAI_MASTER BIT(6) +#define ADAU1373_DAI_INVERT_LRCLK BIT(4) +#define ADAU1373_DAI_WLEN_16 0x0 +#define ADAU1373_DAI_WLEN_20 0x4 +#define ADAU1373_DAI_WLEN_24 0x8 +#define ADAU1373_DAI_WLEN_32 0xc +#define ADAU1373_DAI_WLEN_MASK 0xc +#define ADAU1373_DAI_FORMAT_RIGHT_J 0x0 +#define ADAU1373_DAI_FORMAT_LEFT_J 0x1 +#define ADAU1373_DAI_FORMAT_I2S 0x2 +#define ADAU1373_DAI_FORMAT_DSP 0x3 + +#define ADAU1373_BCLKDIV_SOURCE BIT(5) +#define ADAU1373_BCLKDIV_32 0x03 +#define ADAU1373_BCLKDIV_64 0x02 +#define ADAU1373_BCLKDIV_128 0x01 +#define ADAU1373_BCLKDIV_256 0x00 + +#define ADAU1373_ADC_CTRL_PEAK_DETECT BIT(0) +#define ADAU1373_ADC_CTRL_RESET BIT(1) +#define ADAU1373_ADC_CTRL_RESET_FORCE BIT(2) + +#define ADAU1373_OUTPUT_CTRL_LDIFF BIT(3) +#define ADAU1373_OUTPUT_CTRL_LNFBEN BIT(2) + +#define ADAU1373_PWDN_CTRL3_PWR_EN BIT(0) + +#define ADAU1373_EP_CTRL_MICBIAS1_OFFSET 4 +#define ADAU1373_EP_CTRL_MICBIAS2_OFFSET 2 + +static const uint8_t adau1373_default_regs[] = { + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 0x00 */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 0x10 */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 0x20 */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x02, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x02, 0x00, 0x00, /* 0x30 */ + 0x00, 0x00, 0x00, 0x80, 0x00, 0x01, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x0a, 0x0a, 0x0a, 0x00, /* 0x40 */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x08, 0x08, 0x08, 0x00, 0x00, 0x00, 0x00, /* 0x50 */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 0x60 */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 0x70 */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x78, 0x18, 0x00, 0x00, 0x00, 0xc0, 0x00, 0x00, /* 0x80 */ + 0x00, 0xc0, 0x88, 0x7a, 0xdf, 0x20, 0x00, 0x00, + 0x78, 0x18, 0x00, 0x00, 0x00, 0xc0, 0x00, 0x00, /* 0x90 */ + 0x00, 0xc0, 0x88, 0x7a, 0xdf, 0x20, 0x00, 0x00, + 0x78, 0x18, 0x00, 0x00, 0x00, 0xc0, 0x00, 0x00, /* 0xa0 */ + 0x00, 0xc0, 0x88, 0x7a, 0xdf, 0x20, 0x00, 0x00, + 0x00, 0x00, 0x00, 0xff, 0xff, 0xff, 0xff, 0xff, /* 0xb0 */ + 0xff, 0xff, 0xff, 0xff, 0xff, 0x1f, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 0xc0 */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 0xd0 */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x02, 0x00, /* 0xe0 */ + 0x00, 0x1f, 0x0f, 0x00, 0x00, +}; + +static const unsigned int adau1373_out_tlv[] = { + TLV_DB_RANGE_HEAD(4), + 0, 7, TLV_DB_SCALE_ITEM(-7900, 400, 1), + 8, 15, TLV_DB_SCALE_ITEM(-4700, 300, 0), + 16, 23, TLV_DB_SCALE_ITEM(-2300, 200, 0), + 24, 31, TLV_DB_SCALE_ITEM(-700, 100, 0), +}; + +static const DECLARE_TLV_DB_MINMAX(adau1373_digital_tlv, -9563, 0); +static const DECLARE_TLV_DB_SCALE(adau1373_in_pga_tlv, -1300, 100, 1); +static const DECLARE_TLV_DB_SCALE(adau1373_ep_tlv, -600, 600, 1); + +static const DECLARE_TLV_DB_SCALE(adau1373_input_boost_tlv, 0, 2000, 0); +static const DECLARE_TLV_DB_SCALE(adau1373_gain_boost_tlv, 0, 600, 0); +static const DECLARE_TLV_DB_SCALE(adau1373_speaker_boost_tlv, 1200, 600, 0); + +static const char *adau1373_fdsp_sel_text[] = { + "None", + "Channel 1", + "Channel 2", + "Channel 3", + "Channel 4", + "Channel 5", +}; + +static const SOC_ENUM_SINGLE_DECL(adau1373_drc1_channel_enum, + ADAU1373_FDSP_SEL1, 4, adau1373_fdsp_sel_text); +static const SOC_ENUM_SINGLE_DECL(adau1373_drc2_channel_enum, + ADAU1373_FDSP_SEL1, 0, adau1373_fdsp_sel_text); +static const SOC_ENUM_SINGLE_DECL(adau1373_drc3_channel_enum, + ADAU1373_FDSP_SEL2, 0, adau1373_fdsp_sel_text); +static const SOC_ENUM_SINGLE_DECL(adau1373_hpf_channel_enum, + ADAU1373_FDSP_SEL3, 0, adau1373_fdsp_sel_text); +static const SOC_ENUM_SINGLE_DECL(adau1373_bass_channel_enum, + ADAU1373_FDSP_SEL4, 4, adau1373_fdsp_sel_text); + +static const char *adau1373_hpf_cutoff_text[] = { + "3.7Hz", "50Hz", "100Hz", "150Hz", "200Hz", "250Hz", "300Hz", "350Hz", + "400Hz", "450Hz", "500Hz", "550Hz", "600Hz", "650Hz", "700Hz", "750Hz", + "800Hz", +}; + +static const SOC_ENUM_SINGLE_DECL(adau1373_hpf_cutoff_enum, + ADAU1373_HPF_CTRL, 3, adau1373_hpf_cutoff_text); + +static const char *adau1373_bass_lpf_cutoff_text[] = { + "801Hz", "1001Hz", +}; + +static const char *adau1373_bass_clip_level_text[] = { + "0.125", "0.250", "0.370", "0.500", "0.625", "0.750", "0.875", +}; + +static const unsigned int adau1373_bass_clip_level_values[] = { + 1, 2, 3, 4, 5, 6, 7, +}; + +static const char *adau1373_bass_hpf_cutoff_text[] = { + "158Hz", "232Hz", "347Hz", "520Hz", +}; + +static const unsigned int adau1373_bass_tlv[] = { + TLV_DB_RANGE_HEAD(4), + 0, 2, TLV_DB_SCALE_ITEM(-600, 600, 1), + 3, 4, TLV_DB_SCALE_ITEM(950, 250, 0), + 5, 7, TLV_DB_SCALE_ITEM(1400, 150, 0), +}; + +static const SOC_ENUM_SINGLE_DECL(adau1373_bass_lpf_cutoff_enum, + ADAU1373_BASS1, 5, adau1373_bass_lpf_cutoff_text); + +static const SOC_VALUE_ENUM_SINGLE_DECL(adau1373_bass_clip_level_enum, + ADAU1373_BASS1, 2, 7, adau1373_bass_clip_level_text, + adau1373_bass_clip_level_values); + +static const SOC_ENUM_SINGLE_DECL(adau1373_bass_hpf_cutoff_enum, + ADAU1373_BASS1, 0, adau1373_bass_hpf_cutoff_text); + +static const char *adau1373_3d_level_text[] = { + "0%", "6.67%", "13.33%", "20%", "26.67%", "33.33%", + "40%", "46.67%", "53.33%", "60%", "66.67%", "73.33%", + "80%", "86.67", "99.33%", "100%" +}; + +static const char *adau1373_3d_cutoff_text[] = { + "No 3D", "0.03125 fs", "0.04583 fs", "0.075 fs", "0.11458 fs", + "0.16875 fs", "0.27083 fs" +}; + +static const SOC_ENUM_SINGLE_DECL(adau1373_3d_level_enum, + ADAU1373_3D_CTRL1, 4, adau1373_3d_level_text); +static const SOC_ENUM_SINGLE_DECL(adau1373_3d_cutoff_enum, + ADAU1373_3D_CTRL1, 0, adau1373_3d_cutoff_text); + +static const unsigned int adau1373_3d_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), + 1, 7, TLV_DB_LINEAR_ITEM(-1800, -120), +}; + +static const char *adau1373_lr_mux_text[] = { + "Mute", + "Right Channel (L+R)", + "Left Channel (L+R)", + "Stereo", +}; + +static const SOC_ENUM_SINGLE_DECL(adau1373_lineout1_lr_mux_enum, + ADAU1373_OUTPUT_CTRL, 4, adau1373_lr_mux_text); +static const SOC_ENUM_SINGLE_DECL(adau1373_lineout2_lr_mux_enum, + ADAU1373_OUTPUT_CTRL, 6, adau1373_lr_mux_text); +static const SOC_ENUM_SINGLE_DECL(adau1373_speaker_lr_mux_enum, + ADAU1373_LS_CTRL, 4, adau1373_lr_mux_text); + +static const struct snd_kcontrol_new adau1373_controls[] = { + SOC_DOUBLE_R_TLV("AIF1 Capture Volume", ADAU1373_DAI_RECL_VOL(0), + ADAU1373_DAI_RECR_VOL(0), 0, 0xff, 1, adau1373_digital_tlv), + SOC_DOUBLE_R_TLV("AIF2 Capture Volume", ADAU1373_DAI_RECL_VOL(1), + ADAU1373_DAI_RECR_VOL(1), 0, 0xff, 1, adau1373_digital_tlv), + SOC_DOUBLE_R_TLV("AIF3 Capture Volume", ADAU1373_DAI_RECL_VOL(2), + ADAU1373_DAI_RECR_VOL(2), 0, 0xff, 1, adau1373_digital_tlv), + + SOC_DOUBLE_R_TLV("ADC Capture Volume", ADAU1373_ADC_RECL_VOL, + ADAU1373_ADC_RECR_VOL, 0, 0xff, 1, adau1373_digital_tlv), + SOC_DOUBLE_R_TLV("DMIC Capture Volume", ADAU1373_DMIC_RECL_VOL, + ADAU1373_DMIC_RECR_VOL, 0, 0xff, 1, adau1373_digital_tlv), + + SOC_DOUBLE_R_TLV("AIF1 Playback Volume", ADAU1373_DAI_PBL_VOL(0), + ADAU1373_DAI_PBR_VOL(0), 0, 0xff, 1, adau1373_digital_tlv), + SOC_DOUBLE_R_TLV("AIF2 Playback Volume", ADAU1373_DAI_PBL_VOL(1), + ADAU1373_DAI_PBR_VOL(1), 0, 0xff, 1, adau1373_digital_tlv), + SOC_DOUBLE_R_TLV("AIF3 Playback Volume", ADAU1373_DAI_PBL_VOL(2), + ADAU1373_DAI_PBR_VOL(2), 0, 0xff, 1, adau1373_digital_tlv), + + SOC_DOUBLE_R_TLV("DAC1 Playback Volume", ADAU1373_DAC1_PBL_VOL, + ADAU1373_DAC1_PBR_VOL, 0, 0xff, 1, adau1373_digital_tlv), + SOC_DOUBLE_R_TLV("DAC2 Playback Volume", ADAU1373_DAC2_PBL_VOL, + ADAU1373_DAC2_PBR_VOL, 0, 0xff, 1, adau1373_digital_tlv), + + SOC_DOUBLE_R_TLV("Lineout1 Playback Volume", ADAU1373_LLINE_OUT(0), + ADAU1373_RLINE_OUT(0), 0, 0x1f, 0, adau1373_out_tlv), + SOC_DOUBLE_R_TLV("Speaker Playback Volume", ADAU1373_LSPK_OUT, + ADAU1373_RSPK_OUT, 0, 0x1f, 0, adau1373_out_tlv), + SOC_DOUBLE_R_TLV("Headphone Playback Volume", ADAU1373_LHP_OUT, + ADAU1373_RHP_OUT, 0, 0x1f, 0, adau1373_out_tlv), + + SOC_DOUBLE_R_TLV("Input 1 Capture Volume", ADAU1373_AINL_CTRL(0), + ADAU1373_AINR_CTRL(0), 0, 0x1f, 0, adau1373_in_pga_tlv), + SOC_DOUBLE_R_TLV("Input 2 Capture Volume", ADAU1373_AINL_CTRL(1), + ADAU1373_AINR_CTRL(1), 0, 0x1f, 0, adau1373_in_pga_tlv), + SOC_DOUBLE_R_TLV("Input 3 Capture Volume", ADAU1373_AINL_CTRL(2), + ADAU1373_AINR_CTRL(2), 0, 0x1f, 0, adau1373_in_pga_tlv), + SOC_DOUBLE_R_TLV("Input 4 Capture Volume", ADAU1373_AINL_CTRL(3), + ADAU1373_AINR_CTRL(3), 0, 0x1f, 0, adau1373_in_pga_tlv), + + SOC_SINGLE_TLV("Earpiece Playback Volume", ADAU1373_EP_CTRL, 0, 3, 0, + adau1373_ep_tlv), + + SOC_DOUBLE_TLV("AIF3 Boost Playback Volume", ADAU1373_VOL_GAIN1, 4, 5, + 1, 0, adau1373_gain_boost_tlv), + SOC_DOUBLE_TLV("AIF2 Boost Playback Volume", ADAU1373_VOL_GAIN1, 2, 3, + 1, 0, adau1373_gain_boost_tlv), + SOC_DOUBLE_TLV("AIF1 Boost Playback Volume", ADAU1373_VOL_GAIN1, 0, 1, + 1, 0, adau1373_gain_boost_tlv), + SOC_DOUBLE_TLV("AIF3 Boost Capture Volume", ADAU1373_VOL_GAIN2, 4, 5, + 1, 0, adau1373_gain_boost_tlv), + SOC_DOUBLE_TLV("AIF2 Boost Capture Volume", ADAU1373_VOL_GAIN2, 2, 3, + 1, 0, adau1373_gain_boost_tlv), + SOC_DOUBLE_TLV("AIF1 Boost Capture Volume", ADAU1373_VOL_GAIN2, 0, 1, + 1, 0, adau1373_gain_boost_tlv), + SOC_DOUBLE_TLV("DMIC Boost Capture Volume", ADAU1373_VOL_GAIN3, 6, 7, + 1, 0, adau1373_gain_boost_tlv), + SOC_DOUBLE_TLV("ADC Boost Capture Volume", ADAU1373_VOL_GAIN3, 4, 5, + 1, 0, adau1373_gain_boost_tlv), + SOC_DOUBLE_TLV("DAC2 Boost Playback Volume", ADAU1373_VOL_GAIN3, 2, 3, + 1, 0, adau1373_gain_boost_tlv), + SOC_DOUBLE_TLV("DAC1 Boost Playback Volume", ADAU1373_VOL_GAIN3, 0, 1, + 1, 0, adau1373_gain_boost_tlv), + + SOC_DOUBLE_TLV("Input 1 Boost Capture Volume", ADAU1373_ADC_GAIN, 0, 4, + 1, 0, adau1373_input_boost_tlv), + SOC_DOUBLE_TLV("Input 2 Boost Capture Volume", ADAU1373_ADC_GAIN, 1, 5, + 1, 0, adau1373_input_boost_tlv), + SOC_DOUBLE_TLV("Input 3 Boost Capture Volume", ADAU1373_ADC_GAIN, 2, 6, + 1, 0, adau1373_input_boost_tlv), + SOC_DOUBLE_TLV("Input 4 Boost Capture Volume", ADAU1373_ADC_GAIN, 3, 7, + 1, 0, adau1373_input_boost_tlv), + + SOC_DOUBLE_TLV("Speaker Boost Playback Volume", ADAU1373_LS_CTRL, 2, 3, + 1, 0, adau1373_speaker_boost_tlv), + + SOC_ENUM("Lineout1 LR Mux", adau1373_lineout1_lr_mux_enum), + SOC_ENUM("Speaker LR Mux", adau1373_speaker_lr_mux_enum), + + SOC_ENUM("HPF Cutoff", adau1373_hpf_cutoff_enum), + SOC_DOUBLE("HPF Switch", ADAU1373_HPF_CTRL, 1, 0, 1, 0), + SOC_ENUM("HPF Channel", adau1373_hpf_channel_enum), + + SOC_ENUM("Bass HPF Cutoff", adau1373_bass_hpf_cutoff_enum), + SOC_VALUE_ENUM("Bass Clip Level Threshold", + adau1373_bass_clip_level_enum), + SOC_ENUM("Bass LPF Cutoff", adau1373_bass_lpf_cutoff_enum), + SOC_DOUBLE("Bass Playback Switch", ADAU1373_BASS2, 0, 1, 1, 0), + SOC_SINGLE_TLV("Bass Playback Volume", ADAU1373_BASS2, 2, 7, 0, + adau1373_bass_tlv), + SOC_ENUM("Bass Channel", adau1373_bass_channel_enum), + + SOC_ENUM("3D Freq", adau1373_3d_cutoff_enum), + SOC_ENUM("3D Level", adau1373_3d_level_enum), + SOC_SINGLE("3D Playback Switch", ADAU1373_3D_CTRL2, 0, 1, 0), + SOC_SINGLE_TLV("3D Playback Volume", ADAU1373_3D_CTRL2, 2, 7, 0, + adau1373_3d_tlv), + SOC_ENUM("3D Channel", adau1373_bass_channel_enum), + + SOC_SINGLE("Zero Cross Switch", ADAU1373_PWDN_CTRL3, 7, 1, 0), +}; + +static const struct snd_kcontrol_new adau1373_lineout2_controls[] = { + SOC_DOUBLE_R_TLV("Lineout2 Playback Volume", ADAU1373_LLINE_OUT(1), + ADAU1373_RLINE_OUT(1), 0, 0x1f, 0, adau1373_out_tlv), + SOC_ENUM("Lineout2 LR Mux", adau1373_lineout2_lr_mux_enum), +}; + +static const struct snd_kcontrol_new adau1373_drc_controls[] = { + SOC_ENUM("DRC1 Channel", adau1373_drc1_channel_enum), + SOC_ENUM("DRC2 Channel", adau1373_drc2_channel_enum), + SOC_ENUM("DRC3 Channel", adau1373_drc3_channel_enum), +}; + +static int adau1373_pll_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + unsigned int pll_id = w->name[3] - '1'; + unsigned int val; + + if (SND_SOC_DAPM_EVENT_ON(event)) + val = ADAU1373_PLL_CTRL6_PLL_EN; + else + val = 0; + + snd_soc_update_bits(codec, ADAU1373_PLL_CTRL6(pll_id), + ADAU1373_PLL_CTRL6_PLL_EN, val); + + if (SND_SOC_DAPM_EVENT_ON(event)) + mdelay(5); + + return 0; +} + +static const char *adau1373_decimator_text[] = { + "ADC", + "DMIC1", +}; + +static const struct soc_enum adau1373_decimator_enum = + SOC_ENUM_SINGLE(0, 0, 2, adau1373_decimator_text); + +static const struct snd_kcontrol_new adau1373_decimator_mux = + SOC_DAPM_ENUM_VIRT("Decimator Mux", adau1373_decimator_enum); + +static const struct snd_kcontrol_new adau1373_left_adc_mixer_controls[] = { + SOC_DAPM_SINGLE("DAC1 Switch", ADAU1373_LADC_MIXER, 4, 1, 0), + SOC_DAPM_SINGLE("Input 4 Switch", ADAU1373_LADC_MIXER, 3, 1, 0), + SOC_DAPM_SINGLE("Input 3 Switch", ADAU1373_LADC_MIXER, 2, 1, 0), + SOC_DAPM_SINGLE("Input 2 Switch", ADAU1373_LADC_MIXER, 1, 1, 0), + SOC_DAPM_SINGLE("Input 1 Switch", ADAU1373_LADC_MIXER, 0, 1, 0), +}; + +static const struct snd_kcontrol_new adau1373_right_adc_mixer_controls[] = { + SOC_DAPM_SINGLE("DAC1 Switch", ADAU1373_RADC_MIXER, 4, 1, 0), + SOC_DAPM_SINGLE("Input 4 Switch", ADAU1373_RADC_MIXER, 3, 1, 0), + SOC_DAPM_SINGLE("Input 3 Switch", ADAU1373_RADC_MIXER, 2, 1, 0), + SOC_DAPM_SINGLE("Input 2 Switch", ADAU1373_RADC_MIXER, 1, 1, 0), + SOC_DAPM_SINGLE("Input 1 Switch", ADAU1373_RADC_MIXER, 0, 1, 0), +}; + +#define DECLARE_ADAU1373_OUTPUT_MIXER_CTRLS(_name, _reg) \ +const struct snd_kcontrol_new _name[] = { \ + SOC_DAPM_SINGLE("Left DAC2 Switch", _reg, 7, 1, 0), \ + SOC_DAPM_SINGLE("Right DAC2 Switch", _reg, 6, 1, 0), \ + SOC_DAPM_SINGLE("Left DAC1 Switch", _reg, 5, 1, 0), \ + SOC_DAPM_SINGLE("Right DAC1 Switch", _reg, 4, 1, 0), \ + SOC_DAPM_SINGLE("Input 4 Bypass Switch", _reg, 3, 1, 0), \ + SOC_DAPM_SINGLE("Input 3 Bypass Switch", _reg, 2, 1, 0), \ + SOC_DAPM_SINGLE("Input 2 Bypass Switch", _reg, 1, 1, 0), \ + SOC_DAPM_SINGLE("Input 1 Bypass Switch", _reg, 0, 1, 0), \ +} + +static DECLARE_ADAU1373_OUTPUT_MIXER_CTRLS(adau1373_left_line1_mixer_controls, + ADAU1373_LLINE1_MIX); +static DECLARE_ADAU1373_OUTPUT_MIXER_CTRLS(adau1373_right_line1_mixer_controls, + ADAU1373_RLINE1_MIX); +static DECLARE_ADAU1373_OUTPUT_MIXER_CTRLS(adau1373_left_line2_mixer_controls, + ADAU1373_LLINE2_MIX); +static DECLARE_ADAU1373_OUTPUT_MIXER_CTRLS(adau1373_right_line2_mixer_controls, + ADAU1373_RLINE2_MIX); +static DECLARE_ADAU1373_OUTPUT_MIXER_CTRLS(adau1373_left_spk_mixer_controls, + ADAU1373_LSPK_MIX); +static DECLARE_ADAU1373_OUTPUT_MIXER_CTRLS(adau1373_right_spk_mixer_controls, + ADAU1373_RSPK_MIX); +static DECLARE_ADAU1373_OUTPUT_MIXER_CTRLS(adau1373_ep_mixer_controls, + ADAU1373_EP_MIX); + +static const struct snd_kcontrol_new adau1373_left_hp_mixer_controls[] = { + SOC_DAPM_SINGLE("Left DAC1 Switch", ADAU1373_LHP_MIX, 5, 1, 0), + SOC_DAPM_SINGLE("Left DAC2 Switch", ADAU1373_LHP_MIX, 4, 1, 0), + SOC_DAPM_SINGLE("Input 4 Bypass Switch", ADAU1373_LHP_MIX, 3, 1, 0), + SOC_DAPM_SINGLE("Input 3 Bypass Switch", ADAU1373_LHP_MIX, 2, 1, 0), + SOC_DAPM_SINGLE("Input 2 Bypass Switch", ADAU1373_LHP_MIX, 1, 1, 0), + SOC_DAPM_SINGLE("Input 1 Bypass Switch", ADAU1373_LHP_MIX, 0, 1, 0), +}; + +static const struct snd_kcontrol_new adau1373_right_hp_mixer_controls[] = { + SOC_DAPM_SINGLE("Right DAC1 Switch", ADAU1373_RHP_MIX, 5, 1, 0), + SOC_DAPM_SINGLE("Right DAC2 Switch", ADAU1373_RHP_MIX, 4, 1, 0), + SOC_DAPM_SINGLE("Input 4 Bypass Switch", ADAU1373_RHP_MIX, 3, 1, 0), + SOC_DAPM_SINGLE("Input 3 Bypass Switch", ADAU1373_RHP_MIX, 2, 1, 0), + SOC_DAPM_SINGLE("Input 2 Bypass Switch", ADAU1373_RHP_MIX, 1, 1, 0), + SOC_DAPM_SINGLE("Input 1 Bypass Switch", ADAU1373_RHP_MIX, 0, 1, 0), +}; + +#define DECLARE_ADAU1373_DSP_CHANNEL_MIXER_CTRLS(_name, _reg) \ +const struct snd_kcontrol_new _name[] = { \ + SOC_DAPM_SINGLE("DMIC2 Swapped Switch", _reg, 6, 1, 0), \ + SOC_DAPM_SINGLE("DMIC2 Switch", _reg, 5, 1, 0), \ + SOC_DAPM_SINGLE("ADC/DMIC1 Swapped Switch", _reg, 4, 1, 0), \ + SOC_DAPM_SINGLE("ADC/DMIC1 Switch", _reg, 3, 1, 0), \ + SOC_DAPM_SINGLE("AIF3 Switch", _reg, 2, 1, 0), \ + SOC_DAPM_SINGLE("AIF2 Switch", _reg, 1, 1, 0), \ + SOC_DAPM_SINGLE("AIF1 Switch", _reg, 0, 1, 0), \ +} + +static DECLARE_ADAU1373_DSP_CHANNEL_MIXER_CTRLS(adau1373_dsp_channel1_mixer_controls, + ADAU1373_DIN_MIX_CTRL(0)); +static DECLARE_ADAU1373_DSP_CHANNEL_MIXER_CTRLS(adau1373_dsp_channel2_mixer_controls, + ADAU1373_DIN_MIX_CTRL(1)); +static DECLARE_ADAU1373_DSP_CHANNEL_MIXER_CTRLS(adau1373_dsp_channel3_mixer_controls, + ADAU1373_DIN_MIX_CTRL(2)); +static DECLARE_ADAU1373_DSP_CHANNEL_MIXER_CTRLS(adau1373_dsp_channel4_mixer_controls, + ADAU1373_DIN_MIX_CTRL(3)); +static DECLARE_ADAU1373_DSP_CHANNEL_MIXER_CTRLS(adau1373_dsp_channel5_mixer_controls, + ADAU1373_DIN_MIX_CTRL(4)); + +#define DECLARE_ADAU1373_DSP_OUTPUT_MIXER_CTRLS(_name, _reg) \ +const struct snd_kcontrol_new _name[] = { \ + SOC_DAPM_SINGLE("DSP Channel5 Switch", _reg, 4, 1, 0), \ + SOC_DAPM_SINGLE("DSP Channel4 Switch", _reg, 3, 1, 0), \ + SOC_DAPM_SINGLE("DSP Channel3 Switch", _reg, 2, 1, 0), \ + SOC_DAPM_SINGLE("DSP Channel2 Switch", _reg, 1, 1, 0), \ + SOC_DAPM_SINGLE("DSP Channel1 Switch", _reg, 0, 1, 0), \ +} + +static DECLARE_ADAU1373_DSP_OUTPUT_MIXER_CTRLS(adau1373_aif1_mixer_controls, + ADAU1373_DOUT_MIX_CTRL(0)); +static DECLARE_ADAU1373_DSP_OUTPUT_MIXER_CTRLS(adau1373_aif2_mixer_controls, + ADAU1373_DOUT_MIX_CTRL(1)); +static DECLARE_ADAU1373_DSP_OUTPUT_MIXER_CTRLS(adau1373_aif3_mixer_controls, + ADAU1373_DOUT_MIX_CTRL(2)); +static DECLARE_ADAU1373_DSP_OUTPUT_MIXER_CTRLS(adau1373_dac1_mixer_controls, + ADAU1373_DOUT_MIX_CTRL(3)); +static DECLARE_ADAU1373_DSP_OUTPUT_MIXER_CTRLS(adau1373_dac2_mixer_controls, + ADAU1373_DOUT_MIX_CTRL(4)); + +static const struct snd_soc_dapm_widget adau1373_dapm_widgets[] = { + /* Datasheet claims Left ADC is bit 6 and Right ADC is bit 7, but that + * doesn't seem to be the case. */ + SND_SOC_DAPM_ADC("Left ADC", NULL, ADAU1373_PWDN_CTRL1, 7, 0), + SND_SOC_DAPM_ADC("Right ADC", NULL, ADAU1373_PWDN_CTRL1, 6, 0), + + SND_SOC_DAPM_ADC("DMIC1", NULL, ADAU1373_DIGMICCTRL, 0, 0), + SND_SOC_DAPM_ADC("DMIC2", NULL, ADAU1373_DIGMICCTRL, 2, 0), + + SND_SOC_DAPM_VIRT_MUX("Decimator Mux", SND_SOC_NOPM, 0, 0, + &adau1373_decimator_mux), + + SND_SOC_DAPM_SUPPLY("MICBIAS2", ADAU1373_PWDN_CTRL1, 5, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MICBIAS1", ADAU1373_PWDN_CTRL1, 4, 0, NULL, 0), + + SND_SOC_DAPM_PGA("IN4PGA", ADAU1373_PWDN_CTRL1, 3, 0, NULL, 0), + SND_SOC_DAPM_PGA("IN3PGA", ADAU1373_PWDN_CTRL1, 2, 0, NULL, 0), + SND_SOC_DAPM_PGA("IN2PGA", ADAU1373_PWDN_CTRL1, 1, 0, NULL, 0), + SND_SOC_DAPM_PGA("IN1PGA", ADAU1373_PWDN_CTRL1, 0, 0, NULL, 0), + + SND_SOC_DAPM_DAC("Left DAC2", NULL, ADAU1373_PWDN_CTRL2, 7, 0), + SND_SOC_DAPM_DAC("Right DAC2", NULL, ADAU1373_PWDN_CTRL2, 6, 0), + SND_SOC_DAPM_DAC("Left DAC1", NULL, ADAU1373_PWDN_CTRL2, 5, 0), + SND_SOC_DAPM_DAC("Right DAC1", NULL, ADAU1373_PWDN_CTRL2, 4, 0), + + SOC_MIXER_ARRAY("Left ADC Mixer", SND_SOC_NOPM, 0, 0, + adau1373_left_adc_mixer_controls), + SOC_MIXER_ARRAY("Right ADC Mixer", SND_SOC_NOPM, 0, 0, + adau1373_right_adc_mixer_controls), + + SOC_MIXER_ARRAY("Left Lineout2 Mixer", ADAU1373_PWDN_CTRL2, 3, 0, + adau1373_left_line2_mixer_controls), + SOC_MIXER_ARRAY("Right Lineout2 Mixer", ADAU1373_PWDN_CTRL2, 2, 0, + adau1373_right_line2_mixer_controls), + SOC_MIXER_ARRAY("Left Lineout1 Mixer", ADAU1373_PWDN_CTRL2, 1, 0, + adau1373_left_line1_mixer_controls), + SOC_MIXER_ARRAY("Right Lineout1 Mixer", ADAU1373_PWDN_CTRL2, 0, 0, + adau1373_right_line1_mixer_controls), + + SOC_MIXER_ARRAY("Earpiece Mixer", ADAU1373_PWDN_CTRL3, 4, 0, + adau1373_ep_mixer_controls), + SOC_MIXER_ARRAY("Left Speaker Mixer", ADAU1373_PWDN_CTRL3, 3, 0, + adau1373_left_spk_mixer_controls), + SOC_MIXER_ARRAY("Right Speaker Mixer", ADAU1373_PWDN_CTRL3, 2, 0, + adau1373_right_spk_mixer_controls), + SOC_MIXER_ARRAY("Left Headphone Mixer", SND_SOC_NOPM, 0, 0, + adau1373_left_hp_mixer_controls), + SOC_MIXER_ARRAY("Right Headphone Mixer", SND_SOC_NOPM, 0, 0, + adau1373_right_hp_mixer_controls), + SND_SOC_DAPM_SUPPLY("Headphone Enable", ADAU1373_PWDN_CTRL3, 1, 0, + NULL, 0), + + SND_SOC_DAPM_SUPPLY("AIF1 CLK", ADAU1373_SRC_DAI_CTRL(0), 0, 0, + NULL, 0), + SND_SOC_DAPM_SUPPLY("AIF2 CLK", ADAU1373_SRC_DAI_CTRL(1), 0, 0, + NULL, 0), + SND_SOC_DAPM_SUPPLY("AIF3 CLK", ADAU1373_SRC_DAI_CTRL(2), 0, 0, + NULL, 0), + SND_SOC_DAPM_SUPPLY("AIF1 IN SRC", ADAU1373_SRC_DAI_CTRL(0), 2, 0, + NULL, 0), + SND_SOC_DAPM_SUPPLY("AIF1 OUT SRC", ADAU1373_SRC_DAI_CTRL(0), 1, 0, + NULL, 0), + SND_SOC_DAPM_SUPPLY("AIF2 IN SRC", ADAU1373_SRC_DAI_CTRL(1), 2, 0, + NULL, 0), + SND_SOC_DAPM_SUPPLY("AIF2 OUT SRC", ADAU1373_SRC_DAI_CTRL(1), 1, 0, + NULL, 0), + SND_SOC_DAPM_SUPPLY("AIF3 IN SRC", ADAU1373_SRC_DAI_CTRL(2), 2, 0, + NULL, 0), + SND_SOC_DAPM_SUPPLY("AIF3 OUT SRC", ADAU1373_SRC_DAI_CTRL(2), 1, 0, + NULL, 0), + + SND_SOC_DAPM_AIF_IN("AIF1 IN", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1 OUT", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIF2 IN", "AIF2 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF2 OUT", "AIF2 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIF3 IN", "AIF3 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF3 OUT", "AIF3 Capture", 0, SND_SOC_NOPM, 0, 0), + + SOC_MIXER_ARRAY("DSP Channel1 Mixer", SND_SOC_NOPM, 0, 0, + adau1373_dsp_channel1_mixer_controls), + SOC_MIXER_ARRAY("DSP Channel2 Mixer", SND_SOC_NOPM, 0, 0, + adau1373_dsp_channel2_mixer_controls), + SOC_MIXER_ARRAY("DSP Channel3 Mixer", SND_SOC_NOPM, 0, 0, + adau1373_dsp_channel3_mixer_controls), + SOC_MIXER_ARRAY("DSP Channel4 Mixer", SND_SOC_NOPM, 0, 0, + adau1373_dsp_channel4_mixer_controls), + SOC_MIXER_ARRAY("DSP Channel5 Mixer", SND_SOC_NOPM, 0, 0, + adau1373_dsp_channel5_mixer_controls), + + SOC_MIXER_ARRAY("AIF1 Mixer", SND_SOC_NOPM, 0, 0, + adau1373_aif1_mixer_controls), + SOC_MIXER_ARRAY("AIF2 Mixer", SND_SOC_NOPM, 0, 0, + adau1373_aif2_mixer_controls), + SOC_MIXER_ARRAY("AIF3 Mixer", SND_SOC_NOPM, 0, 0, + adau1373_aif3_mixer_controls), + SOC_MIXER_ARRAY("DAC1 Mixer", SND_SOC_NOPM, 0, 0, + adau1373_dac1_mixer_controls), + SOC_MIXER_ARRAY("DAC2 Mixer", SND_SOC_NOPM, 0, 0, + adau1373_dac2_mixer_controls), + + SND_SOC_DAPM_SUPPLY("DSP", ADAU1373_DIGEN, 4, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Recording Engine B", ADAU1373_DIGEN, 3, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Recording Engine A", ADAU1373_DIGEN, 2, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Playback Engine B", ADAU1373_DIGEN, 1, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Playback Engine A", ADAU1373_DIGEN, 0, 0, NULL, 0), + + SND_SOC_DAPM_SUPPLY("PLL1", SND_SOC_NOPM, 0, 0, adau1373_pll_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SUPPLY("PLL2", SND_SOC_NOPM, 0, 0, adau1373_pll_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SUPPLY("SYSCLK1", ADAU1373_CLK_SRC_DIV(0), 7, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("SYSCLK2", ADAU1373_CLK_SRC_DIV(1), 7, 0, NULL, 0), + + SND_SOC_DAPM_INPUT("AIN1L"), + SND_SOC_DAPM_INPUT("AIN1R"), + SND_SOC_DAPM_INPUT("AIN2L"), + SND_SOC_DAPM_INPUT("AIN2R"), + SND_SOC_DAPM_INPUT("AIN3L"), + SND_SOC_DAPM_INPUT("AIN3R"), + SND_SOC_DAPM_INPUT("AIN4L"), + SND_SOC_DAPM_INPUT("AIN4R"), + + SND_SOC_DAPM_INPUT("DMIC1DAT"), + SND_SOC_DAPM_INPUT("DMIC2DAT"), + + SND_SOC_DAPM_OUTPUT("LOUT1L"), + SND_SOC_DAPM_OUTPUT("LOUT1R"), + SND_SOC_DAPM_OUTPUT("LOUT2L"), + SND_SOC_DAPM_OUTPUT("LOUT2R"), + SND_SOC_DAPM_OUTPUT("HPL"), + SND_SOC_DAPM_OUTPUT("HPR"), + SND_SOC_DAPM_OUTPUT("SPKL"), + SND_SOC_DAPM_OUTPUT("SPKR"), + SND_SOC_DAPM_OUTPUT("EP"), +}; + +static int adau1373_check_aif_clk(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_codec *codec = source->codec; + struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(codec); + unsigned int dai; + const char *clk; + + dai = sink->name[3] - '1'; + + if (!adau1373->dais[dai].master) + return 0; + + if (adau1373->dais[dai].clk_src == ADAU1373_CLK_SRC_PLL1) + clk = "SYSCLK1"; + else + clk = "SYSCLK2"; + + return strcmp(source->name, clk) == 0; +} + +static int adau1373_check_src(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_codec *codec = source->codec; + struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(codec); + unsigned int dai; + + dai = sink->name[3] - '1'; + + return adau1373->dais[dai].enable_src; +} + +#define DSP_CHANNEL_MIXER_ROUTES(_sink) \ + { _sink, "DMIC2 Swapped Switch", "DMIC2" }, \ + { _sink, "DMIC2 Switch", "DMIC2" }, \ + { _sink, "ADC/DMIC1 Swapped Switch", "Decimator Mux" }, \ + { _sink, "ADC/DMIC1 Switch", "Decimator Mux" }, \ + { _sink, "AIF1 Switch", "AIF1 IN" }, \ + { _sink, "AIF2 Switch", "AIF2 IN" }, \ + { _sink, "AIF3 Switch", "AIF3 IN" } + +#define DSP_OUTPUT_MIXER_ROUTES(_sink) \ + { _sink, "DSP Channel1 Switch", "DSP Channel1 Mixer" }, \ + { _sink, "DSP Channel2 Switch", "DSP Channel2 Mixer" }, \ + { _sink, "DSP Channel3 Switch", "DSP Channel3 Mixer" }, \ + { _sink, "DSP Channel4 Switch", "DSP Channel4 Mixer" }, \ + { _sink, "DSP Channel5 Switch", "DSP Channel5 Mixer" } + +#define LEFT_OUTPUT_MIXER_ROUTES(_sink) \ + { _sink, "Right DAC2 Switch", "Right DAC2" }, \ + { _sink, "Left DAC2 Switch", "Left DAC2" }, \ + { _sink, "Right DAC1 Switch", "Right DAC1" }, \ + { _sink, "Left DAC1 Switch", "Left DAC1" }, \ + { _sink, "Input 1 Bypass Switch", "IN1PGA" }, \ + { _sink, "Input 2 Bypass Switch", "IN2PGA" }, \ + { _sink, "Input 3 Bypass Switch", "IN3PGA" }, \ + { _sink, "Input 4 Bypass Switch", "IN4PGA" } + +#define RIGHT_OUTPUT_MIXER_ROUTES(_sink) \ + { _sink, "Right DAC2 Switch", "Right DAC2" }, \ + { _sink, "Left DAC2 Switch", "Left DAC2" }, \ + { _sink, "Right DAC1 Switch", "Right DAC1" }, \ + { _sink, "Left DAC1 Switch", "Left DAC1" }, \ + { _sink, "Input 1 Bypass Switch", "IN1PGA" }, \ + { _sink, "Input 2 Bypass Switch", "IN2PGA" }, \ + { _sink, "Input 3 Bypass Switch", "IN3PGA" }, \ + { _sink, "Input 4 Bypass Switch", "IN4PGA" } + +static const struct snd_soc_dapm_route adau1373_dapm_routes[] = { + { "Left ADC Mixer", "DAC1 Switch", "Left DAC1" }, + { "Left ADC Mixer", "Input 1 Switch", "IN1PGA" }, + { "Left ADC Mixer", "Input 2 Switch", "IN2PGA" }, + { "Left ADC Mixer", "Input 3 Switch", "IN3PGA" }, + { "Left ADC Mixer", "Input 4 Switch", "IN4PGA" }, + + { "Right ADC Mixer", "DAC1 Switch", "Right DAC1" }, + { "Right ADC Mixer", "Input 1 Switch", "IN1PGA" }, + { "Right ADC Mixer", "Input 2 Switch", "IN2PGA" }, + { "Right ADC Mixer", "Input 3 Switch", "IN3PGA" }, + { "Right ADC Mixer", "Input 4 Switch", "IN4PGA" }, + + { "Left ADC", NULL, "Left ADC Mixer" }, + { "Right ADC", NULL, "Right ADC Mixer" }, + + { "Decimator Mux", "ADC", "Left ADC" }, + { "Decimator Mux", "ADC", "Right ADC" }, + { "Decimator Mux", "DMIC1", "DMIC1" }, + + DSP_CHANNEL_MIXER_ROUTES("DSP Channel1 Mixer"), + DSP_CHANNEL_MIXER_ROUTES("DSP Channel2 Mixer"), + DSP_CHANNEL_MIXER_ROUTES("DSP Channel3 Mixer"), + DSP_CHANNEL_MIXER_ROUTES("DSP Channel4 Mixer"), + DSP_CHANNEL_MIXER_ROUTES("DSP Channel5 Mixer"), + + DSP_OUTPUT_MIXER_ROUTES("AIF1 Mixer"), + DSP_OUTPUT_MIXER_ROUTES("AIF2 Mixer"), + DSP_OUTPUT_MIXER_ROUTES("AIF3 Mixer"), + DSP_OUTPUT_MIXER_ROUTES("DAC1 Mixer"), + DSP_OUTPUT_MIXER_ROUTES("DAC2 Mixer"), + + { "AIF1 OUT", NULL, "AIF1 Mixer" }, + { "AIF2 OUT", NULL, "AIF2 Mixer" }, + { "AIF3 OUT", NULL, "AIF3 Mixer" }, + { "Left DAC1", NULL, "DAC1 Mixer" }, + { "Right DAC1", NULL, "DAC1 Mixer" }, + { "Left DAC2", NULL, "DAC2 Mixer" }, + { "Right DAC2", NULL, "DAC2 Mixer" }, + + LEFT_OUTPUT_MIXER_ROUTES("Left Lineout1 Mixer"), + RIGHT_OUTPUT_MIXER_ROUTES("Right Lineout1 Mixer"), + LEFT_OUTPUT_MIXER_ROUTES("Left Lineout2 Mixer"), + RIGHT_OUTPUT_MIXER_ROUTES("Right Lineout2 Mixer"), + LEFT_OUTPUT_MIXER_ROUTES("Left Speaker Mixer"), + RIGHT_OUTPUT_MIXER_ROUTES("Right Speaker Mixer"), + + { "Left Headphone Mixer", "Left DAC2 Switch", "Left DAC2" }, + { "Left Headphone Mixer", "Left DAC1 Switch", "Left DAC1" }, + { "Left Headphone Mixer", "Input 1 Bypass Switch", "IN1PGA" }, + { "Left Headphone Mixer", "Input 2 Bypass Switch", "IN2PGA" }, + { "Left Headphone Mixer", "Input 3 Bypass Switch", "IN3PGA" }, + { "Left Headphone Mixer", "Input 4 Bypass Switch", "IN4PGA" }, + { "Right Headphone Mixer", "Right DAC2 Switch", "Right DAC2" }, + { "Right Headphone Mixer", "Right DAC1 Switch", "Right DAC1" }, + { "Right Headphone Mixer", "Input 1 Bypass Switch", "IN1PGA" }, + { "Right Headphone Mixer", "Input 2 Bypass Switch", "IN2PGA" }, + { "Right Headphone Mixer", "Input 3 Bypass Switch", "IN3PGA" }, + { "Right Headphone Mixer", "Input 4 Bypass Switch", "IN4PGA" }, + + { "Left Headphone Mixer", NULL, "Headphone Enable" }, + { "Right Headphone Mixer", NULL, "Headphone Enable" }, + + { "Earpiece Mixer", "Right DAC2 Switch", "Right DAC2" }, + { "Earpiece Mixer", "Left DAC2 Switch", "Left DAC2" }, + { "Earpiece Mixer", "Right DAC1 Switch", "Right DAC1" }, + { "Earpiece Mixer", "Left DAC1 Switch", "Left DAC1" }, + { "Earpiece Mixer", "Input 1 Bypass Switch", "IN1PGA" }, + { "Earpiece Mixer", "Input 2 Bypass Switch", "IN2PGA" }, + { "Earpiece Mixer", "Input 3 Bypass Switch", "IN3PGA" }, + { "Earpiece Mixer", "Input 4 Bypass Switch", "IN4PGA" }, + + { "LOUT1L", NULL, "Left Lineout1 Mixer" }, + { "LOUT1R", NULL, "Right Lineout1 Mixer" }, + { "LOUT2L", NULL, "Left Lineout2 Mixer" }, + { "LOUT2R", NULL, "Right Lineout2 Mixer" }, + { "SPKL", NULL, "Left Speaker Mixer" }, + { "SPKR", NULL, "Right Speaker Mixer" }, + { "HPL", NULL, "Left Headphone Mixer" }, + { "HPR", NULL, "Right Headphone Mixer" }, + { "EP", NULL, "Earpiece Mixer" }, + + { "IN1PGA", NULL, "AIN1L" }, + { "IN2PGA", NULL, "AIN2L" }, + { "IN3PGA", NULL, "AIN3L" }, + { "IN4PGA", NULL, "AIN4L" }, + { "IN1PGA", NULL, "AIN1R" }, + { "IN2PGA", NULL, "AIN2R" }, + { "IN3PGA", NULL, "AIN3R" }, + { "IN4PGA", NULL, "AIN4R" }, + + { "SYSCLK1", NULL, "PLL1" }, + { "SYSCLK2", NULL, "PLL2" }, + + { "Left DAC1", NULL, "SYSCLK1" }, + { "Right DAC1", NULL, "SYSCLK1" }, + { "Left DAC2", NULL, "SYSCLK1" }, + { "Right DAC2", NULL, "SYSCLK1" }, + { "Left ADC", NULL, "SYSCLK1" }, + { "Right ADC", NULL, "SYSCLK1" }, + + { "DSP", NULL, "SYSCLK1" }, + + { "AIF1 Mixer", NULL, "DSP" }, + { "AIF2 Mixer", NULL, "DSP" }, + { "AIF3 Mixer", NULL, "DSP" }, + { "DAC1 Mixer", NULL, "DSP" }, + { "DAC2 Mixer", NULL, "DSP" }, + { "DAC1 Mixer", NULL, "Playback Engine A" }, + { "DAC2 Mixer", NULL, "Playback Engine B" }, + { "Left ADC Mixer", NULL, "Recording Engine A" }, + { "Right ADC Mixer", NULL, "Recording Engine A" }, + + { "AIF1 CLK", NULL, "SYSCLK1", adau1373_check_aif_clk }, + { "AIF2 CLK", NULL, "SYSCLK1", adau1373_check_aif_clk }, + { "AIF3 CLK", NULL, "SYSCLK1", adau1373_check_aif_clk }, + { "AIF1 CLK", NULL, "SYSCLK2", adau1373_check_aif_clk }, + { "AIF2 CLK", NULL, "SYSCLK2", adau1373_check_aif_clk }, + { "AIF3 CLK", NULL, "SYSCLK2", adau1373_check_aif_clk }, + + { "AIF1 IN", NULL, "AIF1 CLK" }, + { "AIF1 OUT", NULL, "AIF1 CLK" }, + { "AIF2 IN", NULL, "AIF2 CLK" }, + { "AIF2 OUT", NULL, "AIF2 CLK" }, + { "AIF3 IN", NULL, "AIF3 CLK" }, + { "AIF3 OUT", NULL, "AIF3 CLK" }, + { "AIF1 IN", NULL, "AIF1 IN SRC", adau1373_check_src }, + { "AIF1 OUT", NULL, "AIF1 OUT SRC", adau1373_check_src }, + { "AIF2 IN", NULL, "AIF2 IN SRC", adau1373_check_src }, + { "AIF2 OUT", NULL, "AIF2 OUT SRC", adau1373_check_src }, + { "AIF3 IN", NULL, "AIF3 IN SRC", adau1373_check_src }, + { "AIF3 OUT", NULL, "AIF3 OUT SRC", adau1373_check_src }, + + { "DMIC1", NULL, "DMIC1DAT" }, + { "DMIC1", NULL, "SYSCLK1" }, + { "DMIC1", NULL, "Recording Engine A" }, + { "DMIC2", NULL, "DMIC2DAT" }, + { "DMIC2", NULL, "SYSCLK1" }, + { "DMIC2", NULL, "Recording Engine B" }, +}; + +static int adau1373_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(codec); + struct adau1373_dai *adau1373_dai = &adau1373->dais[dai->id]; + unsigned int div; + unsigned int freq; + unsigned int ctrl; + + freq = adau1373_dai->sysclk; + + if (freq % params_rate(params) != 0) + return -EINVAL; + + switch (freq / params_rate(params)) { + case 1024: /* sysclk / 256 */ + div = 0; + break; + case 1536: /* 2/3 sysclk / 256 */ + div = 1; + break; + case 2048: /* 1/2 sysclk / 256 */ + div = 2; + break; + case 3072: /* 1/3 sysclk / 256 */ + div = 3; + break; + case 4096: /* 1/4 sysclk / 256 */ + div = 4; + break; + case 6144: /* 1/6 sysclk / 256 */ + div = 5; + break; + case 5632: /* 2/11 sysclk / 256 */ + div = 6; + break; + default: + return -EINVAL; + } + + adau1373_dai->enable_src = (div != 0); + + snd_soc_update_bits(codec, ADAU1373_BCLKDIV(dai->id), + ~ADAU1373_BCLKDIV_SOURCE, (div << 2) | ADAU1373_BCLKDIV_64); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + ctrl = ADAU1373_DAI_WLEN_16; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + ctrl = ADAU1373_DAI_WLEN_20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + ctrl = ADAU1373_DAI_WLEN_24; + break; + case SNDRV_PCM_FORMAT_S32_LE: + ctrl = ADAU1373_DAI_WLEN_32; + break; + default: + return -EINVAL; + } + + return snd_soc_update_bits(codec, ADAU1373_DAI(dai->id), + ADAU1373_DAI_WLEN_MASK, ctrl); +} + +static int adau1373_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(codec); + struct adau1373_dai *adau1373_dai = &adau1373->dais[dai->id]; + unsigned int ctrl; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + ctrl = ADAU1373_DAI_MASTER; + adau1373_dai->master = true; + break; + case SND_SOC_DAIFMT_CBS_CFS: + ctrl = 0; + adau1373_dai->master = false; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + ctrl |= ADAU1373_DAI_FORMAT_I2S; + break; + case SND_SOC_DAIFMT_LEFT_J: + ctrl |= ADAU1373_DAI_FORMAT_LEFT_J; + break; + case SND_SOC_DAIFMT_RIGHT_J: + ctrl |= ADAU1373_DAI_FORMAT_RIGHT_J; + break; + case SND_SOC_DAIFMT_DSP_B: + ctrl |= ADAU1373_DAI_FORMAT_DSP; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + ctrl |= ADAU1373_DAI_INVERT_BCLK; + break; + case SND_SOC_DAIFMT_NB_IF: + ctrl |= ADAU1373_DAI_INVERT_LRCLK; + break; + case SND_SOC_DAIFMT_IB_IF: + ctrl |= ADAU1373_DAI_INVERT_LRCLK | ADAU1373_DAI_INVERT_BCLK; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, ADAU1373_DAI(dai->id), + ~ADAU1373_DAI_WLEN_MASK, ctrl); + + return 0; +} + +static int adau1373_set_dai_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(dai->codec); + struct adau1373_dai *adau1373_dai = &adau1373->dais[dai->id]; + + switch (clk_id) { + case ADAU1373_CLK_SRC_PLL1: + case ADAU1373_CLK_SRC_PLL2: + break; + default: + return -EINVAL; + } + + adau1373_dai->sysclk = freq; + adau1373_dai->clk_src = clk_id; + + snd_soc_update_bits(dai->codec, ADAU1373_BCLKDIV(dai->id), + ADAU1373_BCLKDIV_SOURCE, clk_id << 5); + + return 0; +} + +static const struct snd_soc_dai_ops adau1373_dai_ops = { + .hw_params = adau1373_hw_params, + .set_sysclk = adau1373_set_dai_sysclk, + .set_fmt = adau1373_set_dai_fmt, +}; + +#define ADAU1373_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_driver adau1373_dai_driver[] = { + { + .id = 0, + .name = "adau1373-aif1", + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = ADAU1373_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = ADAU1373_FORMATS, + }, + .ops = &adau1373_dai_ops, + .symmetric_rates = 1, + }, + { + .id = 1, + .name = "adau1373-aif2", + .playback = { + .stream_name = "AIF2 Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = ADAU1373_FORMATS, + }, + .capture = { + .stream_name = "AIF2 Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = ADAU1373_FORMATS, + }, + .ops = &adau1373_dai_ops, + .symmetric_rates = 1, + }, + { + .id = 2, + .name = "adau1373-aif3", + .playback = { + .stream_name = "AIF3 Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = ADAU1373_FORMATS, + }, + .capture = { + .stream_name = "AIF3 Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = ADAU1373_FORMATS, + }, + .ops = &adau1373_dai_ops, + .symmetric_rates = 1, + }, +}; + +static int adau1373_set_pll(struct snd_soc_codec *codec, int pll_id, + int source, unsigned int freq_in, unsigned int freq_out) +{ + unsigned int dpll_div = 0; + unsigned int x, r, n, m, i, j, mode; + + switch (pll_id) { + case ADAU1373_PLL1: + case ADAU1373_PLL2: + break; + default: + return -EINVAL; + } + + switch (source) { + case ADAU1373_PLL_SRC_BCLK1: + case ADAU1373_PLL_SRC_BCLK2: + case ADAU1373_PLL_SRC_BCLK3: + case ADAU1373_PLL_SRC_LRCLK1: + case ADAU1373_PLL_SRC_LRCLK2: + case ADAU1373_PLL_SRC_LRCLK3: + case ADAU1373_PLL_SRC_MCLK1: + case ADAU1373_PLL_SRC_MCLK2: + case ADAU1373_PLL_SRC_GPIO1: + case ADAU1373_PLL_SRC_GPIO2: + case ADAU1373_PLL_SRC_GPIO3: + case ADAU1373_PLL_SRC_GPIO4: + break; + default: + return -EINVAL; + } + + if (freq_in < 7813 || freq_in > 27000000) + return -EINVAL; + + if (freq_out < 45158000 || freq_out > 49152000) + return -EINVAL; + + /* APLL input needs to be >= 8Mhz, so in case freq_in is less we use the + * DPLL to get it there. DPLL_out = (DPLL_in / div) * 1024 */ + while (freq_in < 8000000) { + freq_in *= 2; + dpll_div++; + } + + if (freq_out % freq_in != 0) { + /* fout = fin * (r + (n/m)) / x */ + x = DIV_ROUND_UP(freq_in, 13500000); + freq_in /= x; + r = freq_out / freq_in; + i = freq_out % freq_in; + j = gcd(i, freq_in); + n = i / j; + m = freq_in / j; + x--; + mode = 1; + } else { + /* fout = fin / r */ + r = freq_out / freq_in; + n = 0; + m = 0; + x = 0; + mode = 0; + } + + if (r < 2 || r > 8 || x > 3 || m > 0xffff || n > 0xffff) + return -EINVAL; + + if (dpll_div) { + dpll_div = 11 - dpll_div; + snd_soc_update_bits(codec, ADAU1373_PLL_CTRL6(pll_id), + ADAU1373_PLL_CTRL6_DPLL_BYPASS, 0); + } else { + snd_soc_update_bits(codec, ADAU1373_PLL_CTRL6(pll_id), + ADAU1373_PLL_CTRL6_DPLL_BYPASS, + ADAU1373_PLL_CTRL6_DPLL_BYPASS); + } + + snd_soc_write(codec, ADAU1373_DPLL_CTRL(pll_id), + (source << 4) | dpll_div); + snd_soc_write(codec, ADAU1373_PLL_CTRL1(pll_id), (m >> 8) & 0xff); + snd_soc_write(codec, ADAU1373_PLL_CTRL2(pll_id), m & 0xff); + snd_soc_write(codec, ADAU1373_PLL_CTRL3(pll_id), (n >> 8) & 0xff); + snd_soc_write(codec, ADAU1373_PLL_CTRL4(pll_id), n & 0xff); + snd_soc_write(codec, ADAU1373_PLL_CTRL5(pll_id), + (r << 3) | (x << 1) | mode); + + /* Set sysclk to pll_rate / 4 */ + snd_soc_update_bits(codec, ADAU1373_CLK_SRC_DIV(pll_id), 0x3f, 0x09); + + return 0; +} + +static void adau1373_load_drc_settings(struct snd_soc_codec *codec, + unsigned int nr, uint8_t *drc) +{ + unsigned int i; + + for (i = 0; i < ADAU1373_DRC_SIZE; ++i) + snd_soc_write(codec, ADAU1373_DRC(nr) + i, drc[i]); +} + +static bool adau1373_valid_micbias(enum adau1373_micbias_voltage micbias) +{ + switch (micbias) { + case ADAU1373_MICBIAS_2_9V: + case ADAU1373_MICBIAS_2_2V: + case ADAU1373_MICBIAS_2_6V: + case ADAU1373_MICBIAS_1_8V: + return true; + default: + break; + } + return false; +} + +static int adau1373_probe(struct snd_soc_codec *codec) +{ + struct adau1373_platform_data *pdata = codec->dev->platform_data; + bool lineout_differential = false; + unsigned int val; + int ret; + int i; + + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C); + if (ret) { + dev_err(codec->dev, "failed to set cache I/O: %d\n", ret); + return ret; + } + + codec->dapm.idle_bias_off = true; + + if (pdata) { + if (pdata->num_drc > ARRAY_SIZE(pdata->drc_setting)) + return -EINVAL; + + if (!adau1373_valid_micbias(pdata->micbias1) || + !adau1373_valid_micbias(pdata->micbias2)) + return -EINVAL; + + for (i = 0; i < pdata->num_drc; ++i) { + adau1373_load_drc_settings(codec, i, + pdata->drc_setting[i]); + } + + snd_soc_add_controls(codec, adau1373_drc_controls, + pdata->num_drc); + + val = 0; + for (i = 0; i < 4; ++i) { + if (pdata->input_differential[i]) + val |= BIT(i); + } + snd_soc_write(codec, ADAU1373_INPUT_MODE, val); + + val = 0; + if (pdata->lineout_differential) + val |= ADAU1373_OUTPUT_CTRL_LDIFF; + if (pdata->lineout_ground_sense) + val |= ADAU1373_OUTPUT_CTRL_LNFBEN; + snd_soc_write(codec, ADAU1373_OUTPUT_CTRL, val); + + lineout_differential = pdata->lineout_differential; + + snd_soc_write(codec, ADAU1373_EP_CTRL, + (pdata->micbias1 << ADAU1373_EP_CTRL_MICBIAS1_OFFSET) | + (pdata->micbias2 << ADAU1373_EP_CTRL_MICBIAS2_OFFSET)); + } + + if (!lineout_differential) { + snd_soc_add_controls(codec, adau1373_lineout2_controls, + ARRAY_SIZE(adau1373_lineout2_controls)); + } + + snd_soc_write(codec, ADAU1373_ADC_CTRL, + ADAU1373_ADC_CTRL_RESET_FORCE | ADAU1373_ADC_CTRL_PEAK_DETECT); + + return 0; +} + +static int adau1373_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + snd_soc_update_bits(codec, ADAU1373_PWDN_CTRL3, + ADAU1373_PWDN_CTRL3_PWR_EN, ADAU1373_PWDN_CTRL3_PWR_EN); + break; + case SND_SOC_BIAS_OFF: + snd_soc_update_bits(codec, ADAU1373_PWDN_CTRL3, + ADAU1373_PWDN_CTRL3_PWR_EN, 0); + break; + } + codec->dapm.bias_level = level; + return 0; +} + +static int adau1373_remove(struct snd_soc_codec *codec) +{ + adau1373_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int adau1373_suspend(struct snd_soc_codec *codec, pm_message_t state) +{ + return adau1373_set_bias_level(codec, SND_SOC_BIAS_OFF); +} + +static int adau1373_resume(struct snd_soc_codec *codec) +{ + adau1373_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + snd_soc_cache_sync(codec); + + return 0; +} + +static struct snd_soc_codec_driver adau1373_codec_driver = { + .probe = adau1373_probe, + .remove = adau1373_remove, + .suspend = adau1373_suspend, + .resume = adau1373_resume, + .set_bias_level = adau1373_set_bias_level, + .reg_cache_size = ARRAY_SIZE(adau1373_default_regs), + .reg_cache_default = adau1373_default_regs, + .reg_word_size = sizeof(uint8_t), + + .set_pll = adau1373_set_pll, + + .controls = adau1373_controls, + .num_controls = ARRAY_SIZE(adau1373_controls), + .dapm_widgets = adau1373_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(adau1373_dapm_widgets), + .dapm_routes = adau1373_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(adau1373_dapm_routes), +}; + +static int __devinit adau1373_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct adau1373 *adau1373; + int ret; + + adau1373 = kzalloc(sizeof(*adau1373), GFP_KERNEL); + if (!adau1373) + return -ENOMEM; + + dev_set_drvdata(&client->dev, adau1373); + + ret = snd_soc_register_codec(&client->dev, &adau1373_codec_driver, + adau1373_dai_driver, ARRAY_SIZE(adau1373_dai_driver)); + if (ret < 0) + kfree(adau1373); + + return ret; +} + +static int __devexit adau1373_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + kfree(dev_get_drvdata(&client->dev)); + return 0; +} + +static const struct i2c_device_id adau1373_i2c_id[] = { + { "adau1373", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, adau1373_i2c_id); + +static struct i2c_driver adau1373_i2c_driver = { + .driver = { + .name = "adau1373", + .owner = THIS_MODULE, + }, + .probe = adau1373_i2c_probe, + .remove = __devexit_p(adau1373_i2c_remove), + .id_table = adau1373_i2c_id, +}; + +static int __init adau1373_init(void) +{ + return i2c_add_driver(&adau1373_i2c_driver); +} +module_init(adau1373_init); + +static void __exit adau1373_exit(void) +{ + i2c_del_driver(&adau1373_i2c_driver); +} +module_exit(adau1373_exit); + +MODULE_DESCRIPTION("ASoC ADAU1373 driver"); +MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/adau1373.h b/sound/soc/codecs/adau1373.h new file mode 100644 index 000000000000..c6ab5530760c --- /dev/null +++ b/sound/soc/codecs/adau1373.h @@ -0,0 +1,29 @@ +#ifndef __ADAU1373_H__ +#define __ADAU1373_H__ + +enum adau1373_pll_src { + ADAU1373_PLL_SRC_MCLK1 = 0, + ADAU1373_PLL_SRC_BCLK1 = 1, + ADAU1373_PLL_SRC_BCLK2 = 2, + ADAU1373_PLL_SRC_BCLK3 = 3, + ADAU1373_PLL_SRC_LRCLK1 = 4, + ADAU1373_PLL_SRC_LRCLK2 = 5, + ADAU1373_PLL_SRC_LRCLK3 = 6, + ADAU1373_PLL_SRC_GPIO1 = 7, + ADAU1373_PLL_SRC_GPIO2 = 8, + ADAU1373_PLL_SRC_GPIO3 = 9, + ADAU1373_PLL_SRC_GPIO4 = 10, + ADAU1373_PLL_SRC_MCLK2 = 11, +}; + +enum adau1373_pll { + ADAU1373_PLL1 = 0, + ADAU1373_PLL2 = 1, +}; + +enum adau1373_clk_src { + ADAU1373_CLK_SRC_PLL1 = 0, + ADAU1373_CLK_SRC_PLL2 = 1, +}; + +#endif diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index 2758d5fc60d6..8b7e1c50d6e9 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -401,7 +401,7 @@ static int adau1701_digital_mute(struct snd_soc_dai *dai, int mute) } static int adau1701_set_sysclk(struct snd_soc_codec *codec, int clk_id, - unsigned int freq, int dir) + int source, unsigned int freq, int dir) { unsigned int val; @@ -458,6 +458,7 @@ static int adau1701_probe(struct snd_soc_codec *codec) int ret; codec->dapm.idle_bias_off = 1; + codec->control_data = to_i2c_client(codec->dev); ret = adau1701_load_firmware(codec); if (ret) diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c index 300c04b70e71..f9f08948e5e8 100644 --- a/sound/soc/codecs/adav80x.c +++ b/sound/soc/codecs/adav80x.c @@ -523,7 +523,8 @@ static int adav80x_hw_params(struct snd_pcm_substream *substream, } static int adav80x_set_sysclk(struct snd_soc_codec *codec, - int clk_id, unsigned int freq, int dir) + int clk_id, int source, + unsigned int freq, int dir) { struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); diff --git a/sound/soc/codecs/ads117x.h b/sound/soc/codecs/ads117x.h deleted file mode 100644 index 3ce028614002..000000000000 --- a/sound/soc/codecs/ads117x.h +++ /dev/null @@ -1,13 +0,0 @@ -/* - * ads117x.h -- Driver for ads1174/8 ADC chips - * - * Copyright 2009 ShotSpotter Inc. - * Author: Graeme Gregory <gg@slimlogic.co.uk> - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ -extern struct snd_soc_dai_driver ads117x_dai; -extern struct snd_soc_codec_driver soc_codec_dev_ads117x; diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c index cbf0b6d400b8..d3b29dce6ed7 100644 --- a/sound/soc/codecs/ak4104.c +++ b/sound/soc/codecs/ak4104.c @@ -247,7 +247,7 @@ static struct snd_soc_codec_driver soc_codec_device_ak4104 = { .probe = ak4104_probe, .remove = ak4104_remove, .reg_cache_size = AK4104_NUM_REGS, - .reg_word_size = sizeof(u16), + .reg_word_size = sizeof(u8), }; static int ak4104_spi_probe(struct spi_device *spi) diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c index e1a214ee757f..95d782d86e7d 100644 --- a/sound/soc/codecs/ak4535.c +++ b/sound/soc/codecs/ak4535.c @@ -34,74 +34,16 @@ struct ak4535_priv { unsigned int sysclk; enum snd_soc_control_type control_type; - void *control_data; }; /* * ak4535 register cache */ -static const u16 ak4535_reg[AK4535_CACHEREGNUM] = { - 0x0000, 0x0080, 0x0000, 0x0003, - 0x0002, 0x0000, 0x0011, 0x0001, - 0x0000, 0x0040, 0x0036, 0x0010, - 0x0000, 0x0000, 0x0057, 0x0000, -}; - -/* - * read ak4535 register cache - */ -static inline unsigned int ak4535_read_reg_cache(struct snd_soc_codec *codec, - unsigned int reg) -{ - u16 *cache = codec->reg_cache; - if (reg >= AK4535_CACHEREGNUM) - return -1; - return cache[reg]; -} - -/* - * write ak4535 register cache - */ -static inline void ak4535_write_reg_cache(struct snd_soc_codec *codec, - u16 reg, unsigned int value) -{ - u16 *cache = codec->reg_cache; - if (reg >= AK4535_CACHEREGNUM) - return; - cache[reg] = value; -} - -/* - * write to the AK4535 register space - */ -static int ak4535_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - u8 data[2]; - - /* data is - * D15..D8 AK4535 register offset - * D7...D0 register data - */ - data[0] = reg & 0xff; - data[1] = value & 0xff; - - ak4535_write_reg_cache(codec, reg, value); - if (codec->hw_write(codec->control_data, data, 2) == 2) - return 0; - else - return -EIO; -} - -static int ak4535_sync(struct snd_soc_codec *codec) -{ - u16 *cache = codec->reg_cache; - int i, r = 0; - - for (i = 0; i < AK4535_CACHEREGNUM; i++) - r |= ak4535_write(codec, i, cache[i]); - - return r; +static const u8 ak4535_reg[AK4535_CACHEREGNUM] = { + 0x00, 0x80, 0x00, 0x03, + 0x02, 0x00, 0x11, 0x01, + 0x00, 0x40, 0x36, 0x10, + 0x00, 0x00, 0x57, 0x00, }; static const char *ak4535_mono_gain[] = {"+6dB", "-17dB"}; @@ -304,7 +246,7 @@ static int ak4535_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_codec *codec = rtd->codec; struct ak4535_priv *ak4535 = snd_soc_codec_get_drvdata(codec); - u8 mode2 = ak4535_read_reg_cache(codec, AK4535_MODE2) & ~(0x3 << 5); + u8 mode2 = snd_soc_read(codec, AK4535_MODE2) & ~(0x3 << 5); int rate = params_rate(params), fs = 256; if (rate) @@ -323,7 +265,7 @@ static int ak4535_hw_params(struct snd_pcm_substream *substream, } /* set rate */ - ak4535_write(codec, AK4535_MODE2, mode2); + snd_soc_write(codec, AK4535_MODE2, mode2); return 0; } @@ -348,44 +290,37 @@ static int ak4535_set_dai_fmt(struct snd_soc_dai *codec_dai, /* use 32 fs for BCLK to save power */ mode1 |= 0x4; - ak4535_write(codec, AK4535_MODE1, mode1); + snd_soc_write(codec, AK4535_MODE1, mode1); return 0; } static int ak4535_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; - u16 mute_reg = ak4535_read_reg_cache(codec, AK4535_DAC); + u16 mute_reg = snd_soc_read(codec, AK4535_DAC); if (!mute) - ak4535_write(codec, AK4535_DAC, mute_reg & ~0x20); + snd_soc_write(codec, AK4535_DAC, mute_reg & ~0x20); else - ak4535_write(codec, AK4535_DAC, mute_reg | 0x20); + snd_soc_write(codec, AK4535_DAC, mute_reg | 0x20); return 0; } static int ak4535_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - u16 i, mute_reg; - switch (level) { case SND_SOC_BIAS_ON: - mute_reg = ak4535_read_reg_cache(codec, AK4535_DAC); - ak4535_write(codec, AK4535_DAC, mute_reg & ~0x20); + snd_soc_update_bits(codec, AK4535_DAC, 0x20, 0); break; case SND_SOC_BIAS_PREPARE: - mute_reg = ak4535_read_reg_cache(codec, AK4535_DAC); - ak4535_write(codec, AK4535_DAC, mute_reg | 0x20); + snd_soc_update_bits(codec, AK4535_DAC, 0x20, 0x20); break; case SND_SOC_BIAS_STANDBY: - i = ak4535_read_reg_cache(codec, AK4535_PM1); - ak4535_write(codec, AK4535_PM1, i | 0x80); - i = ak4535_read_reg_cache(codec, AK4535_PM2); - ak4535_write(codec, AK4535_PM2, i & (~0x80)); + snd_soc_update_bits(codec, AK4535_PM1, 0x80, 0x80); + snd_soc_update_bits(codec, AK4535_PM2, 0x80, 0); break; case SND_SOC_BIAS_OFF: - i = ak4535_read_reg_cache(codec, AK4535_PM1); - ak4535_write(codec, AK4535_PM1, i & (~0x80)); + snd_soc_update_bits(codec, AK4535_PM1, 0x80, 0); break; } codec->dapm.bias_level = level; @@ -428,7 +363,7 @@ static int ak4535_suspend(struct snd_soc_codec *codec, pm_message_t state) static int ak4535_resume(struct snd_soc_codec *codec) { - ak4535_sync(codec); + snd_soc_cache_sync(codec); ak4535_set_bias_level(codec, SND_SOC_BIAS_STANDBY); return 0; } @@ -436,11 +371,15 @@ static int ak4535_resume(struct snd_soc_codec *codec) static int ak4535_probe(struct snd_soc_codec *codec) { struct ak4535_priv *ak4535 = snd_soc_codec_get_drvdata(codec); + int ret; printk(KERN_INFO "AK4535 Audio Codec %s", AK4535_VERSION); - codec->control_data = ak4535->control_data; - + ret = snd_soc_codec_set_cache_io(codec, 8, 8, ak4535->control_type); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } /* power on device */ ak4535_set_bias_level(codec, SND_SOC_BIAS_STANDBY); @@ -461,8 +400,6 @@ static struct snd_soc_codec_driver soc_codec_dev_ak4535 = { .remove = ak4535_remove, .suspend = ak4535_suspend, .resume = ak4535_resume, - .read = ak4535_read_reg_cache, - .write = ak4535_write, .set_bias_level = ak4535_set_bias_level, .reg_cache_size = ARRAY_SIZE(ak4535_reg), .reg_word_size = sizeof(u8), @@ -485,7 +422,6 @@ static __devinit int ak4535_i2c_probe(struct i2c_client *i2c, return -ENOMEM; i2c_set_clientdata(i2c, ak4535); - ak4535->control_data = i2c; ak4535->control_type = SND_SOC_I2C; ret = snd_soc_register_codec(&i2c->dev, diff --git a/sound/soc/codecs/ak4641.c b/sound/soc/codecs/ak4641.c index 7a64e58cddc4..77838586f358 100644 --- a/sound/soc/codecs/ak4641.c +++ b/sound/soc/codecs/ak4641.c @@ -31,7 +31,6 @@ /* codec private data */ struct ak4641_priv { - struct snd_soc_codec *codec; unsigned int sysclk; int deemph; int playback_fs; @@ -226,7 +225,7 @@ static const struct snd_soc_dapm_widget ak4641_dapm_widgets[] = { SND_SOC_DAPM_PGA("Mono Out 2", AK4641_PM2, 3, 0, NULL, 0), SND_SOC_DAPM_ADC("Voice ADC", "Voice Capture", AK4641_BTIF, 0, 0), - SND_SOC_DAPM_ADC("Voice DAC", "Voice Playback", AK4641_BTIF, 1, 0), + SND_SOC_DAPM_DAC("Voice DAC", "Voice Playback", AK4641_BTIF, 1, 0), SND_SOC_DAPM_MICBIAS("Mic Int Bias", AK4641_MIC, 3, 0), SND_SOC_DAPM_MICBIAS("Mic Ext Bias", AK4641_MIC, 4, 0), diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 65f46047b1cb..d8fc04486abb 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -156,81 +156,22 @@ static const struct snd_kcontrol_new ak4642_snd_controls[] = { struct ak4642_priv { unsigned int sysclk; enum snd_soc_control_type control_type; - void *control_data; }; /* * ak4642 register cache */ -static const u16 ak4642_reg[AK4642_CACHEREGNUM] = { - 0x0000, 0x0000, 0x0001, 0x0000, - 0x0002, 0x0000, 0x0000, 0x0000, - 0x00e1, 0x00e1, 0x0018, 0x0000, - 0x00e1, 0x0018, 0x0011, 0x0008, - 0x0000, 0x0000, 0x0000, 0x0000, - 0x0000, 0x0000, 0x0000, 0x0000, - 0x0000, 0x0000, 0x0000, 0x0000, - 0x0000, 0x0000, 0x0000, 0x0000, - 0x0000, 0x0000, 0x0000, 0x0000, - 0x0000, -}; - -/* - * read ak4642 register cache - */ -static inline unsigned int ak4642_read_reg_cache(struct snd_soc_codec *codec, - unsigned int reg) -{ - u16 *cache = codec->reg_cache; - if (reg >= AK4642_CACHEREGNUM) - return -1; - return cache[reg]; -} - -/* - * write ak4642 register cache - */ -static inline void ak4642_write_reg_cache(struct snd_soc_codec *codec, - u16 reg, unsigned int value) -{ - u16 *cache = codec->reg_cache; - if (reg >= AK4642_CACHEREGNUM) - return; - - cache[reg] = value; -} - -/* - * write to the AK4642 register space - */ -static int ak4642_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - u8 data[2]; - - /* data is - * D15..D8 AK4642 register offset - * D7...D0 register data - */ - data[0] = reg & 0xff; - data[1] = value & 0xff; - - if (codec->hw_write(codec->control_data, data, 2) == 2) { - ak4642_write_reg_cache(codec, reg, value); - return 0; - } else - return -EIO; -} - -static int ak4642_sync(struct snd_soc_codec *codec) -{ - u16 *cache = codec->reg_cache; - int i, r = 0; - - for (i = 0; i < AK4642_CACHEREGNUM; i++) - r |= ak4642_write(codec, i, cache[i]); - - return r; +static const u8 ak4642_reg[AK4642_CACHEREGNUM] = { + 0x00, 0x00, 0x01, 0x00, + 0x02, 0x00, 0x00, 0x00, + 0xe1, 0xe1, 0x18, 0x00, + 0xe1, 0x18, 0x11, 0x08, + 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, + 0x00, }; static int ak4642_dai_startup(struct snd_pcm_substream *substream, @@ -252,8 +193,8 @@ static int ak4642_dai_startup(struct snd_pcm_substream *substream, */ snd_soc_update_bits(codec, MD_CTL4, DACH, DACH); snd_soc_update_bits(codec, MD_CTL3, BST1, BST1); - ak4642_write(codec, L_IVC, 0x91); /* volume */ - ak4642_write(codec, R_IVC, 0x91); /* volume */ + snd_soc_write(codec, L_IVC, 0x91); /* volume */ + snd_soc_write(codec, R_IVC, 0x91); /* volume */ snd_soc_update_bits(codec, PW_MGMT1, PMVCM | PMMIN | PMDAC, PMVCM | PMMIN | PMDAC); snd_soc_update_bits(codec, PW_MGMT2, PMHP_MASK, PMHP); @@ -272,9 +213,9 @@ static int ak4642_dai_startup(struct snd_pcm_substream *substream, * This operation came from example code of * "ASAHI KASEI AK4642" (japanese) manual p94. */ - ak4642_write(codec, SG_SL1, PMMP | MGAIN0); - ak4642_write(codec, TIMER, ZTM(0x3) | WTM(0x3)); - ak4642_write(codec, ALC_CTL1, ALC | LMTH0); + snd_soc_write(codec, SG_SL1, PMMP | MGAIN0); + snd_soc_write(codec, TIMER, ZTM(0x3) | WTM(0x3)); + snd_soc_write(codec, ALC_CTL1, ALC | LMTH0); snd_soc_update_bits(codec, PW_MGMT1, PMVCM | PMADL, PMVCM | PMADL); snd_soc_update_bits(codec, PW_MGMT3, PMADR, PMADR); @@ -462,7 +403,7 @@ static struct snd_soc_dai_driver ak4642_dai = { static int ak4642_resume(struct snd_soc_codec *codec) { - ak4642_sync(codec); + snd_soc_cache_sync(codec); return 0; } @@ -470,11 +411,15 @@ static int ak4642_resume(struct snd_soc_codec *codec) static int ak4642_probe(struct snd_soc_codec *codec) { struct ak4642_priv *ak4642 = snd_soc_codec_get_drvdata(codec); + int ret; dev_info(codec->dev, "AK4642 Audio Codec %s", AK4642_VERSION); - codec->hw_write = (hw_write_t)i2c_master_send; - codec->control_data = ak4642->control_data; + ret = snd_soc_codec_set_cache_io(codec, 8, 8, ak4642->control_type); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } snd_soc_add_controls(codec, ak4642_snd_controls, ARRAY_SIZE(ak4642_snd_controls)); @@ -485,8 +430,6 @@ static int ak4642_probe(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_dev_ak4642 = { .probe = ak4642_probe, .resume = ak4642_resume, - .read = ak4642_read_reg_cache, - .write = ak4642_write, .reg_cache_size = ARRAY_SIZE(ak4642_reg), .reg_word_size = sizeof(u8), .reg_cache_default = ak4642_reg, @@ -504,7 +447,6 @@ static __devinit int ak4642_i2c_probe(struct i2c_client *i2c, return -ENOMEM; i2c_set_clientdata(i2c, ak4642); - ak4642->control_data = i2c; ak4642->control_type = SND_SOC_I2C; ret = snd_soc_register_codec(&i2c->dev, diff --git a/sound/soc/codecs/ak4671.c b/sound/soc/codecs/ak4671.c index 88b29f8c748b..de9ff66d3721 100644 --- a/sound/soc/codecs/ak4671.c +++ b/sound/soc/codecs/ak4671.c @@ -26,7 +26,6 @@ /* codec private data */ struct ak4671_priv { enum snd_soc_control_type control_type; - void *control_data; }; /* ak4671 register cache & default register settings */ @@ -169,18 +168,15 @@ static int ak4671_out2_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_codec *codec = w->codec; - u8 reg; switch (event) { case SND_SOC_DAPM_POST_PMU: - reg = snd_soc_read(codec, AK4671_LOUT2_POWER_MANAGERMENT); - reg |= AK4671_MUTEN; - snd_soc_write(codec, AK4671_LOUT2_POWER_MANAGERMENT, reg); + snd_soc_update_bits(codec, AK4671_LOUT2_POWER_MANAGERMENT, + AK4671_MUTEN, AK4671_MUTEN); break; case SND_SOC_DAPM_PRE_PMD: - reg = snd_soc_read(codec, AK4671_LOUT2_POWER_MANAGERMENT); - reg &= ~AK4671_MUTEN; - snd_soc_write(codec, AK4671_LOUT2_POWER_MANAGERMENT, reg); + snd_soc_update_bits(codec, AK4671_LOUT2_POWER_MANAGERMENT, + AK4671_MUTEN, 0); break; } @@ -576,15 +572,12 @@ static int ak4671_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) static int ak4671_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - u8 reg; - switch (level) { case SND_SOC_BIAS_ON: case SND_SOC_BIAS_PREPARE: case SND_SOC_BIAS_STANDBY: - reg = snd_soc_read(codec, AK4671_AD_DA_POWER_MANAGEMENT); - snd_soc_write(codec, AK4671_AD_DA_POWER_MANAGEMENT, - reg | AK4671_PMVCM); + snd_soc_update_bits(codec, AK4671_AD_DA_POWER_MANAGEMENT, + AK4671_PMVCM, AK4671_PMVCM); break; case SND_SOC_BIAS_OFF: snd_soc_write(codec, AK4671_AD_DA_POWER_MANAGEMENT, 0x00); @@ -629,8 +622,6 @@ static int ak4671_probe(struct snd_soc_codec *codec) struct ak4671_priv *ak4671 = snd_soc_codec_get_drvdata(codec); int ret; - codec->hw_write = (hw_write_t)i2c_master_send; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, ak4671->control_type); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); @@ -675,7 +666,6 @@ static int __devinit ak4671_i2c_probe(struct i2c_client *client, return -ENOMEM; i2c_set_clientdata(client, ak4671); - ak4671->control_data = client; ak4671->control_type = SND_SOC_I2C; ret = snd_soc_register_codec(&client->dev, diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c index eecffb548947..984b14bcb605 100644 --- a/sound/soc/codecs/alc5623.c +++ b/sound/soc/codecs/alc5623.c @@ -40,8 +40,6 @@ MODULE_PARM_DESC(caps_charge, "ALC5623 cap charge time (msecs)"); /* codec private data */ struct alc5623_priv { enum snd_soc_control_type control_type; - void *control_data; - struct mutex mutex; u8 id; unsigned int sysclk; u16 reg_cache[ALC5623_VENDOR_ID2+2]; @@ -55,8 +53,10 @@ static void alc5623_fill_cache(struct snd_soc_codec *codec) u16 *cache = codec->reg_cache; /* not really efficient ... */ + codec->cache_bypass = 1; for (i = 0 ; i < codec->driver->reg_cache_size ; i += step) - cache[i] = codec->hw_read(codec, i); + cache[i] = snd_soc_read(codec, i); + codec->cache_bypass = 0; } static inline int alc5623_reset(struct snd_soc_codec *codec) @@ -1050,9 +1050,7 @@ static int alc5623_i2c_probe(struct i2c_client *client, } i2c_set_clientdata(client, alc5623); - alc5623->control_data = client; alc5623->control_type = SND_SOC_I2C; - mutex_init(&alc5623->mutex); ret = snd_soc_register_codec(&client->dev, &soc_codec_device_alc5623, &alc5623_dai, 1); diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 6cc8678f49f3..f1f237ecec2a 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -128,7 +128,6 @@ static const char *supply_names[] = { /* Private data for the CS4270 */ struct cs4270_private { enum snd_soc_control_type control_type; - void *control_data; unsigned int mclk; /* Input frequency of the MCLK pin */ unsigned int mode; /* The mode (I2S or left-justified) */ unsigned int slave_mode; @@ -262,7 +261,6 @@ static int cs4270_set_dai_fmt(struct snd_soc_dai *codec_dai, { struct snd_soc_codec *codec = codec_dai->codec; struct cs4270_private *cs4270 = snd_soc_codec_get_drvdata(codec); - int ret = 0; /* set DAI format */ switch (format & SND_SOC_DAIFMT_FORMAT_MASK) { @@ -272,7 +270,7 @@ static int cs4270_set_dai_fmt(struct snd_soc_dai *codec_dai, break; default: dev_err(codec->dev, "invalid dai format\n"); - ret = -EINVAL; + return -EINVAL; } /* set master/slave audio interface */ @@ -285,10 +283,11 @@ static int cs4270_set_dai_fmt(struct snd_soc_dai *codec_dai, break; default: /* all other modes are unsupported by the hardware */ - ret = -EINVAL; + dev_err(codec->dev, "Unknown master/slave configuration\n"); + return -EINVAL; } - return ret; + return 0; } /** @@ -490,8 +489,6 @@ static int cs4270_probe(struct snd_soc_codec *codec) struct cs4270_private *cs4270 = snd_soc_codec_get_drvdata(codec); int i, ret; - codec->control_data = cs4270->control_data; - /* Tell ASoC what kind of I/O to use to read the registers. ASoC will * then do the I2C transactions itself. */ @@ -604,7 +601,7 @@ static int cs4270_soc_suspend(struct snd_soc_codec *codec, pm_message_t mesg) static int cs4270_soc_resume(struct snd_soc_codec *codec) { struct cs4270_private *cs4270 = snd_soc_codec_get_drvdata(codec); - struct i2c_client *i2c_client = codec->control_data; + struct i2c_client *i2c_client = to_i2c_client(codec->dev); int reg; regulator_bulk_enable(ARRAY_SIZE(cs4270->supplies), @@ -690,7 +687,6 @@ static int cs4270_i2c_probe(struct i2c_client *i2c_client, } i2c_set_clientdata(i2c_client, cs4270); - cs4270->control_data = i2c_client; cs4270->control_type = SND_SOC_I2C; ret = snd_soc_register_codec(&i2c_client->dev, diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index 083aab96ca80..23d1bd5dadda 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -156,7 +156,6 @@ static const u8 cs4271_dflt_reg[CS4271_NR_REGS] = { struct cs4271_private { /* SND_SOC_I2C or SND_SOC_SPI */ enum snd_soc_control_type bus_type; - void *control_data; unsigned int mclk; bool master; bool deemph; @@ -466,8 +465,6 @@ static int cs4271_probe(struct snd_soc_codec *codec) int ret; int gpio_nreset = -EINVAL; - codec->control_data = cs4271->control_data; - if (cs4271plat && gpio_is_valid(cs4271plat->gpio_nreset)) gpio_nreset = cs4271plat->gpio_nreset; @@ -555,7 +552,6 @@ static int __devinit cs4271_spi_probe(struct spi_device *spi) return -ENOMEM; spi_set_drvdata(spi, cs4271); - cs4271->control_data = spi; cs4271->bus_type = SND_SOC_SPI; return snd_soc_register_codec(&spi->dev, &soc_codec_dev_cs4271, @@ -595,7 +591,6 @@ static int __devinit cs4271_i2c_probe(struct i2c_client *client, return -ENOMEM; i2c_set_clientdata(client, cs4271); - cs4271->control_data = client; cs4271->bus_type = SND_SOC_I2C; return snd_soc_register_codec(&client->dev, &soc_codec_dev_cs4271, diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index 8fb7070108dd..8c3c8205d19e 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -42,7 +42,6 @@ enum master_slave_mode { struct cs42l51_private { enum snd_soc_control_type control_type; - void *control_data; unsigned int mclk; unsigned int audio_mode; /* The mode (I2S or left-justified) */ enum master_slave_mode func; @@ -57,7 +56,7 @@ struct cs42l51_private { static int cs42l51_fill_cache(struct snd_soc_codec *codec) { u8 *cache = codec->reg_cache + 1; - struct i2c_client *i2c_client = codec->control_data; + struct i2c_client *i2c_client = to_i2c_client(codec->dev); s32 length; length = i2c_smbus_read_i2c_block_data(i2c_client, @@ -289,7 +288,6 @@ static int cs42l51_set_dai_fmt(struct snd_soc_dai *codec_dai, { struct snd_soc_codec *codec = codec_dai->codec; struct cs42l51_private *cs42l51 = snd_soc_codec_get_drvdata(codec); - int ret = 0; switch (format & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: @@ -299,7 +297,7 @@ static int cs42l51_set_dai_fmt(struct snd_soc_dai *codec_dai, break; default: dev_err(codec->dev, "invalid DAI format\n"); - ret = -EINVAL; + return -EINVAL; } switch (format & SND_SOC_DAIFMT_MASTER_MASK) { @@ -310,11 +308,11 @@ static int cs42l51_set_dai_fmt(struct snd_soc_dai *codec_dai, cs42l51->func = MODE_SLAVE_AUTO; break; default: - ret = -EINVAL; - break; + dev_err(codec->dev, "Unknown master/slave configuration\n"); + return -EINVAL; } - return ret; + return 0; } struct cs42l51_ratios { @@ -520,8 +518,6 @@ static int cs42l51_probe(struct snd_soc_codec *codec) struct snd_soc_dapm_context *dapm = &codec->dapm; int ret, reg; - codec->control_data = cs42l51->control_data; - ret = cs42l51_fill_cache(codec); if (ret < 0) { dev_err(codec->dev, "failed to fill register cache\n"); @@ -593,7 +589,6 @@ static int cs42l51_i2c_probe(struct i2c_client *i2c_client, } i2c_set_clientdata(i2c_client, cs42l51); - cs42l51->control_data = i2c_client; cs42l51->control_type = SND_SOC_I2C; ret = snd_soc_register_codec(&i2c_client->dev, diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 92fd9d7a9221..0ebcbd534490 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -26,23 +26,41 @@ #include <sound/tlv.h> /* DA7210 register space */ +#define DA7210_CONTROL 0x01 #define DA7210_STATUS 0x02 #define DA7210_STARTUP1 0x03 +#define DA7210_STARTUP2 0x04 +#define DA7210_STARTUP3 0x05 #define DA7210_MIC_L 0x07 #define DA7210_MIC_R 0x08 +#define DA7210_AUX1_L 0x09 +#define DA7210_AUX1_R 0x0A +#define DA7210_AUX2 0x0B +#define DA7210_IN_GAIN 0x0C #define DA7210_INMIX_L 0x0D #define DA7210_INMIX_R 0x0E #define DA7210_ADC_HPF 0x0F #define DA7210_ADC 0x10 +#define DA7210_ADC_EQ1_2 0X11 +#define DA7210_ADC_EQ3_4 0x12 +#define DA7210_ADC_EQ5 0x13 #define DA7210_DAC_HPF 0x14 #define DA7210_DAC_L 0x15 #define DA7210_DAC_R 0x16 #define DA7210_DAC_SEL 0x17 +#define DA7210_SOFTMUTE 0x18 +#define DA7210_DAC_EQ1_2 0x19 +#define DA7210_DAC_EQ3_4 0x1A +#define DA7210_DAC_EQ5 0x1B #define DA7210_OUTMIX_L 0x1C #define DA7210_OUTMIX_R 0x1D +#define DA7210_OUT1_L 0x1E +#define DA7210_OUT1_R 0x1F +#define DA7210_OUT2 0x20 #define DA7210_HP_L_VOL 0x21 #define DA7210_HP_R_VOL 0x22 #define DA7210_HP_CFG 0x23 +#define DA7210_ZERO_CROSS 0x24 #define DA7210_DAI_SRC_SEL 0x25 #define DA7210_DAI_CFG1 0x26 #define DA7210_DAI_CFG3 0x28 @@ -50,6 +68,12 @@ #define DA7210_PLL_DIV2 0x2A #define DA7210_PLL_DIV3 0x2B #define DA7210_PLL 0x2C +#define DA7210_ALC_MAX 0x83 +#define DA7210_ALC_MIN 0x84 +#define DA7210_ALC_NOIS 0x85 +#define DA7210_ALC_ATT 0x86 +#define DA7210_ALC_REL 0x87 +#define DA7210_ALC_DEL 0x88 #define DA7210_A_HID_UNLOCK 0x8A #define DA7210_A_TEST_UNLOCK 0x8B #define DA7210_A_PLL1 0x90 @@ -72,6 +96,7 @@ #define DA7210_IN_R_EN (1 << 7) /* ADC bit fields */ +#define DA7210_ADC_ALC_EN (1 << 0) #define DA7210_ADC_L_EN (1 << 3) #define DA7210_ADC_R_EN (1 << 7) @@ -105,12 +130,17 @@ /* DAI_CFG1 bit fields */ #define DA7210_DAI_WORD_S16_LE (0 << 0) +#define DA7210_DAI_WORD_S20_3LE (1 << 0) #define DA7210_DAI_WORD_S24_LE (2 << 0) +#define DA7210_DAI_WORD_S32_LE (3 << 0) #define DA7210_DAI_FLEN_64BIT (1 << 2) +#define DA7210_DAI_MODE_SLAVE (0 << 7) #define DA7210_DAI_MODE_MASTER (1 << 7) /* DAI_CFG3 bit fields */ #define DA7210_DAI_FORMAT_I2SMODE (0 << 0) +#define DA7210_DAI_FORMAT_LEFT_J (1 << 0) +#define DA7210_DAI_FORMAT_RIGHT_J (2 << 0) #define DA7210_DAI_OE (1 << 3) #define DA7210_DAI_EN (1 << 7) @@ -133,6 +163,43 @@ #define DA7210_PLL_FS_96000 (0xF << 0) #define DA7210_PLL_EN (0x1 << 7) +/* SOFTMUTE bit fields */ +#define DA7210_RAMP_EN (1 << 6) + +/* CONTROL bit fields */ +#define DA7210_NOISE_SUP_EN (1 << 3) + +/* IN_GAIN bit fields */ +#define DA7210_INPGA_L_VOL (0x0F << 0) +#define DA7210_INPGA_R_VOL (0xF0 << 0) + +/* ZERO_CROSS bit fields */ +#define DA7210_AUX1_L_ZC (1 << 0) +#define DA7210_AUX1_R_ZC (1 << 1) +#define DA7210_HP_L_ZC (1 << 6) +#define DA7210_HP_R_ZC (1 << 7) + +/* AUX1_L bit fields */ +#define DA7210_AUX1_L_VOL (0x3F << 0) + +/* AUX1_R bit fields */ +#define DA7210_AUX1_R_VOL (0x3F << 0) + +/* Minimum INPGA and AUX1 volume to enable noise suppression */ +#define DA7210_INPGA_MIN_VOL_NS 0x0A /* 10.5dB */ +#define DA7210_AUX1_MIN_VOL_NS 0x35 /* 6dB */ + +/* OUT1_L bit fields */ +#define DA7210_OUT1_L_EN (1 << 7) + +/* OUT1_R bit fields */ +#define DA7210_OUT1_R_EN (1 << 7) + +/* OUT2 bit fields */ +#define DA7210_OUT2_OUTMIX_R (1 << 5) +#define DA7210_OUT2_OUTMIX_L (1 << 6) +#define DA7210_OUT2_EN (1 << 7) + #define DA7210_VERSION "0.0.1" /* @@ -144,24 +211,351 @@ * mute : 0x10 * reserved : 0x00 - 0x0F * - * ** FIXME ** - * * Reserved area are considered as "mute". - * -> min = -79.5 dB */ -static const DECLARE_TLV_DB_SCALE(hp_out_tlv, -7950, 150, 1); +static const unsigned int hp_out_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0x0, 0x10, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1), + /* -54 dB to +15 dB */ + 0x11, 0x3f, TLV_DB_SCALE_ITEM(-5400, 150, 0), +}; + +static const unsigned int lineout_vol_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0x0, 0x10, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1), + /* -54dB to 15dB */ + 0x11, 0x3f, TLV_DB_SCALE_ITEM(-5400, 150, 0) +}; + +static const unsigned int mono_vol_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0x0, 0x2, TLV_DB_SCALE_ITEM(-1800, 0, 1), + /* -18dB to 6dB */ + 0x3, 0x7, TLV_DB_SCALE_ITEM(-1800, 600, 0) +}; + +static const DECLARE_TLV_DB_SCALE(eq_gain_tlv, -1050, 150, 0); +static const DECLARE_TLV_DB_SCALE(adc_eq_master_gain_tlv, -1800, 600, 1); +static const DECLARE_TLV_DB_SCALE(dac_gain_tlv, -7725, 75, 0); + +/* ADC and DAC high pass filter f0 value */ +static const char const *da7210_hpf_cutoff_txt[] = { + "Fs/8192*pi", "Fs/4096*pi", "Fs/2048*pi", "Fs/1024*pi" +}; + +static const struct soc_enum da7210_dac_hpf_cutoff = + SOC_ENUM_SINGLE(DA7210_DAC_HPF, 0, 4, da7210_hpf_cutoff_txt); + +static const struct soc_enum da7210_adc_hpf_cutoff = + SOC_ENUM_SINGLE(DA7210_ADC_HPF, 0, 4, da7210_hpf_cutoff_txt); + +/* ADC and DAC voice (8kHz) high pass cutoff value */ +static const char const *da7210_vf_cutoff_txt[] = { + "2.5Hz", "25Hz", "50Hz", "100Hz", "150Hz", "200Hz", "300Hz", "400Hz" +}; + +static const struct soc_enum da7210_dac_vf_cutoff = + SOC_ENUM_SINGLE(DA7210_DAC_HPF, 4, 8, da7210_vf_cutoff_txt); + +static const struct soc_enum da7210_adc_vf_cutoff = + SOC_ENUM_SINGLE(DA7210_ADC_HPF, 4, 8, da7210_vf_cutoff_txt); + +static const char *da7210_hp_mode_txt[] = { + "Class H", "Class G" +}; + +static const struct soc_enum da7210_hp_mode_sel = + SOC_ENUM_SINGLE(DA7210_HP_CFG, 0, 2, da7210_hp_mode_txt); + +/* ALC can be enabled only if noise suppression is disabled */ +static int da7210_put_alc_sw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + + if (ucontrol->value.integer.value[0]) { + /* Check if noise suppression is enabled */ + if (snd_soc_read(codec, DA7210_CONTROL) & DA7210_NOISE_SUP_EN) { + dev_dbg(codec->dev, + "Disable noise suppression to enable ALC\n"); + return -EINVAL; + } + } + /* If all conditions are met or we are actually disabling ALC */ + return snd_soc_put_volsw(kcontrol, ucontrol); +} + +/* Noise suppression can be enabled only if following conditions are met + * ALC disabled + * ZC enabled for HP and AUX1 PGA + * INPGA_L_VOL and INPGA_R_VOL >= 10.5 dB + * AUX1_L_VOL and AUX1_R_VOL >= 6 dB + */ +static int da7210_put_noise_sup_sw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + u8 val; + + if (ucontrol->value.integer.value[0]) { + /* Check if ALC is enabled */ + if (snd_soc_read(codec, DA7210_ADC) & DA7210_ADC_ALC_EN) + goto err; + + /* Check ZC for HP and AUX1 PGA */ + if ((snd_soc_read(codec, DA7210_ZERO_CROSS) & + (DA7210_AUX1_L_ZC | DA7210_AUX1_R_ZC | DA7210_HP_L_ZC | + DA7210_HP_R_ZC)) != (DA7210_AUX1_L_ZC | + DA7210_AUX1_R_ZC | DA7210_HP_L_ZC | DA7210_HP_R_ZC)) + goto err; + + /* Check INPGA_L_VOL and INPGA_R_VOL */ + val = snd_soc_read(codec, DA7210_IN_GAIN); + if (((val & DA7210_INPGA_L_VOL) < DA7210_INPGA_MIN_VOL_NS) || + (((val & DA7210_INPGA_R_VOL) >> 4) < + DA7210_INPGA_MIN_VOL_NS)) + goto err; + + /* Check AUX1_L_VOL and AUX1_R_VOL */ + if (((snd_soc_read(codec, DA7210_AUX1_L) & DA7210_AUX1_L_VOL) < + DA7210_AUX1_MIN_VOL_NS) || + ((snd_soc_read(codec, DA7210_AUX1_R) & DA7210_AUX1_R_VOL) < + DA7210_AUX1_MIN_VOL_NS)) + goto err; + } + /* If all conditions are met or we are actually disabling Noise sup */ + return snd_soc_put_volsw(kcontrol, ucontrol); + +err: + return -EINVAL; +} static const struct snd_kcontrol_new da7210_snd_controls[] = { SOC_DOUBLE_R_TLV("HeadPhone Playback Volume", DA7210_HP_L_VOL, DA7210_HP_R_VOL, 0, 0x3F, 0, hp_out_tlv), + SOC_DOUBLE_R_TLV("Digital Playback Volume", + DA7210_DAC_L, DA7210_DAC_R, + 0, 0x77, 1, dac_gain_tlv), + SOC_DOUBLE_R_TLV("Lineout Playback Volume", + DA7210_OUT1_L, DA7210_OUT1_R, + 0, 0x3f, 0, lineout_vol_tlv), + SOC_SINGLE_TLV("Mono Playback Volume", DA7210_OUT2, 0, 0x7, 0, + mono_vol_tlv), + + /* DAC Equalizer controls */ + SOC_SINGLE("DAC EQ Switch", DA7210_DAC_EQ5, 7, 1, 0), + SOC_SINGLE_TLV("DAC EQ1 Volume", DA7210_DAC_EQ1_2, 0, 0xf, 1, + eq_gain_tlv), + SOC_SINGLE_TLV("DAC EQ2 Volume", DA7210_DAC_EQ1_2, 4, 0xf, 1, + eq_gain_tlv), + SOC_SINGLE_TLV("DAC EQ3 Volume", DA7210_DAC_EQ3_4, 0, 0xf, 1, + eq_gain_tlv), + SOC_SINGLE_TLV("DAC EQ4 Volume", DA7210_DAC_EQ3_4, 4, 0xf, 1, + eq_gain_tlv), + SOC_SINGLE_TLV("DAC EQ5 Volume", DA7210_DAC_EQ5, 0, 0xf, 1, + eq_gain_tlv), + + /* ADC Equalizer controls */ + SOC_SINGLE("ADC EQ Switch", DA7210_ADC_EQ5, 7, 1, 0), + SOC_SINGLE_TLV("ADC EQ Master Volume", DA7210_ADC_EQ5, 4, 0x3, + 1, adc_eq_master_gain_tlv), + SOC_SINGLE_TLV("ADC EQ1 Volume", DA7210_ADC_EQ1_2, 0, 0xf, 1, + eq_gain_tlv), + SOC_SINGLE_TLV("ADC EQ2 Volume", DA7210_ADC_EQ1_2, 4, 0xf, 1, + eq_gain_tlv), + SOC_SINGLE_TLV("ADC EQ3 Volume", DA7210_ADC_EQ3_4, 0, 0xf, 1, + eq_gain_tlv), + SOC_SINGLE_TLV("ADC EQ4 Volume", DA7210_ADC_EQ3_4, 4, 0xf, 1, + eq_gain_tlv), + SOC_SINGLE_TLV("ADC EQ5 Volume", DA7210_ADC_EQ5, 0, 0xf, 1, + eq_gain_tlv), + + SOC_SINGLE("DAC HPF Switch", DA7210_DAC_HPF, 3, 1, 0), + SOC_ENUM("DAC HPF Cutoff", da7210_dac_hpf_cutoff), + SOC_SINGLE("DAC Voice Mode Switch", DA7210_DAC_HPF, 7, 1, 0), + SOC_ENUM("DAC Voice Cutoff", da7210_dac_vf_cutoff), + + SOC_SINGLE("ADC HPF Switch", DA7210_ADC_HPF, 3, 1, 0), + SOC_ENUM("ADC HPF Cutoff", da7210_adc_hpf_cutoff), + SOC_SINGLE("ADC Voice Mode Switch", DA7210_ADC_HPF, 7, 1, 0), + SOC_ENUM("ADC Voice Cutoff", da7210_adc_vf_cutoff), + + /* Mute controls */ + SOC_DOUBLE_R("Mic Capture Switch", DA7210_MIC_L, DA7210_MIC_R, 3, 1, 0), + SOC_SINGLE("Aux2 Capture Switch", DA7210_AUX2, 2, 1, 0), + SOC_DOUBLE("ADC Capture Switch", DA7210_ADC, 2, 6, 1, 0), + SOC_SINGLE("Digital Soft Mute Switch", DA7210_SOFTMUTE, 7, 1, 0), + SOC_SINGLE("Digital Soft Mute Rate", DA7210_SOFTMUTE, 0, 0x7, 0), + + /* Zero cross controls */ + SOC_DOUBLE("Aux1 ZC Switch", DA7210_ZERO_CROSS, 0, 1, 1, 0), + SOC_DOUBLE("In PGA ZC Switch", DA7210_ZERO_CROSS, 2, 3, 1, 0), + SOC_DOUBLE("Lineout ZC Switch", DA7210_ZERO_CROSS, 4, 5, 1, 0), + SOC_DOUBLE("Headphone ZC Switch", DA7210_ZERO_CROSS, 6, 7, 1, 0), + + SOC_ENUM("Headphone Class", da7210_hp_mode_sel), + + /* ALC controls */ + SOC_SINGLE_EXT("ALC Enable Switch", DA7210_ADC, 0, 1, 0, + snd_soc_get_volsw, da7210_put_alc_sw), + SOC_SINGLE("ALC Capture Max Volume", DA7210_ALC_MAX, 0, 0x3F, 0), + SOC_SINGLE("ALC Capture Min Volume", DA7210_ALC_MIN, 0, 0x3F, 0), + SOC_SINGLE("ALC Capture Noise Volume", DA7210_ALC_NOIS, 0, 0x3F, 0), + SOC_SINGLE("ALC Capture Attack Rate", DA7210_ALC_ATT, 0, 0xFF, 0), + SOC_SINGLE("ALC Capture Release Rate", DA7210_ALC_REL, 0, 0xFF, 0), + SOC_SINGLE("ALC Capture Release Delay", DA7210_ALC_DEL, 0, 0xFF, 0), + + SOC_SINGLE_EXT("Noise Suppression Enable Switch", DA7210_CONTROL, 3, 1, + 0, snd_soc_get_volsw, da7210_put_noise_sup_sw), +}; + +/* + * DAPM Controls + * + * Current DAPM implementation covers almost all codec components e.g. IOs, + * mixers, PGAs,ADC and DAC. + */ +/* In Mixer Left */ +static const struct snd_kcontrol_new da7210_dapm_inmixl_controls[] = { + SOC_DAPM_SINGLE("Mic Left Switch", DA7210_INMIX_L, 0, 1, 0), + SOC_DAPM_SINGLE("Mic Right Switch", DA7210_INMIX_L, 1, 1, 0), +}; + +/* In Mixer Right */ +static const struct snd_kcontrol_new da7210_dapm_inmixr_controls[] = { + SOC_DAPM_SINGLE("Mic Right Switch", DA7210_INMIX_R, 0, 1, 0), + SOC_DAPM_SINGLE("Mic Left Switch", DA7210_INMIX_R, 1, 1, 0), +}; + +/* Out Mixer Left */ +static const struct snd_kcontrol_new da7210_dapm_outmixl_controls[] = { + SOC_DAPM_SINGLE("DAC Left Switch", DA7210_OUTMIX_L, 4, 1, 0), +}; + +/* Out Mixer Right */ +static const struct snd_kcontrol_new da7210_dapm_outmixr_controls[] = { + SOC_DAPM_SINGLE("DAC Right Switch", DA7210_OUTMIX_R, 4, 1, 0), +}; + +/* Mono Mixer */ +static const struct snd_kcontrol_new da7210_dapm_monomix_controls[] = { + SOC_DAPM_SINGLE("Outmix Right Switch", DA7210_OUT2, 5, 1, 0), + SOC_DAPM_SINGLE("Outmix Left Switch", DA7210_OUT2, 6, 1, 0), +}; + +/* DAPM widgets */ +static const struct snd_soc_dapm_widget da7210_dapm_widgets[] = { + /* Input Side */ + /* Input Lines */ + SND_SOC_DAPM_INPUT("MICL"), + SND_SOC_DAPM_INPUT("MICR"), + + /* Input PGAs */ + SND_SOC_DAPM_PGA("Mic Left", DA7210_STARTUP3, 0, 1, NULL, 0), + SND_SOC_DAPM_PGA("Mic Right", DA7210_STARTUP3, 1, 1, NULL, 0), + + SND_SOC_DAPM_PGA("INPGA Left", DA7210_INMIX_L, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("INPGA Right", DA7210_INMIX_R, 7, 0, NULL, 0), + + /* Input Mixers */ + SND_SOC_DAPM_MIXER("In Mixer Left", SND_SOC_NOPM, 0, 0, + &da7210_dapm_inmixl_controls[0], + ARRAY_SIZE(da7210_dapm_inmixl_controls)), + + SND_SOC_DAPM_MIXER("In Mixer Right", SND_SOC_NOPM, 0, 0, + &da7210_dapm_inmixr_controls[0], + ARRAY_SIZE(da7210_dapm_inmixr_controls)), + + /* ADCs */ + SND_SOC_DAPM_ADC("ADC Left", "Capture", DA7210_STARTUP3, 5, 1), + SND_SOC_DAPM_ADC("ADC Right", "Capture", DA7210_STARTUP3, 6, 1), + + /* Output Side */ + /* DACs */ + SND_SOC_DAPM_DAC("DAC Left", "Playback", DA7210_STARTUP2, 5, 1), + SND_SOC_DAPM_DAC("DAC Right", "Playback", DA7210_STARTUP2, 6, 1), + + /* Output Mixers */ + SND_SOC_DAPM_MIXER("Out Mixer Left", SND_SOC_NOPM, 0, 0, + &da7210_dapm_outmixl_controls[0], + ARRAY_SIZE(da7210_dapm_outmixl_controls)), + + SND_SOC_DAPM_MIXER("Out Mixer Right", SND_SOC_NOPM, 0, 0, + &da7210_dapm_outmixr_controls[0], + ARRAY_SIZE(da7210_dapm_outmixr_controls)), + + SND_SOC_DAPM_MIXER("Mono Mixer", SND_SOC_NOPM, 0, 0, + &da7210_dapm_monomix_controls[0], + ARRAY_SIZE(da7210_dapm_monomix_controls)), + + /* Output PGAs */ + SND_SOC_DAPM_PGA("OUTPGA Left Enable", DA7210_OUTMIX_L, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("OUTPGA Right Enable", DA7210_OUTMIX_R, 7, 0, NULL, 0), + + SND_SOC_DAPM_PGA("Out1 Left", DA7210_STARTUP2, 0, 1, NULL, 0), + SND_SOC_DAPM_PGA("Out1 Right", DA7210_STARTUP2, 1, 1, NULL, 0), + SND_SOC_DAPM_PGA("Out2 Mono", DA7210_STARTUP2, 2, 1, NULL, 0), + SND_SOC_DAPM_PGA("Headphone Left", DA7210_STARTUP2, 3, 1, NULL, 0), + SND_SOC_DAPM_PGA("Headphone Right", DA7210_STARTUP2, 4, 1, NULL, 0), + + /* Output Lines */ + SND_SOC_DAPM_OUTPUT("OUT1L"), + SND_SOC_DAPM_OUTPUT("OUT1R"), + SND_SOC_DAPM_OUTPUT("HPL"), + SND_SOC_DAPM_OUTPUT("HPR"), + SND_SOC_DAPM_OUTPUT("OUT2"), +}; + +/* DAPM audio route definition */ +static const struct snd_soc_dapm_route da7210_audio_map[] = { + /* Dest Connecting Widget source */ + /* Input path */ + {"Mic Left", NULL, "MICL"}, + {"Mic Right", NULL, "MICR"}, + + {"In Mixer Left", "Mic Left Switch", "Mic Left"}, + {"In Mixer Left", "Mic Right Switch", "Mic Right"}, + + {"In Mixer Right", "Mic Right Switch", "Mic Right"}, + {"In Mixer Right", "Mic Left Switch", "Mic Left"}, + + {"INPGA Left", NULL, "In Mixer Left"}, + {"ADC Left", NULL, "INPGA Left"}, + + {"INPGA Right", NULL, "In Mixer Right"}, + {"ADC Right", NULL, "INPGA Right"}, + + /* Output path */ + {"Out Mixer Left", "DAC Left Switch", "DAC Left"}, + {"Out Mixer Right", "DAC Right Switch", "DAC Right"}, + + {"Mono Mixer", "Outmix Right Switch", "Out Mixer Right"}, + {"Mono Mixer", "Outmix Left Switch", "Out Mixer Left"}, + + {"OUTPGA Left Enable", NULL, "Out Mixer Left"}, + {"OUTPGA Right Enable", NULL, "Out Mixer Right"}, + + {"Out1 Left", NULL, "OUTPGA Left Enable"}, + {"OUT1L", NULL, "Out1 Left"}, + + {"Out1 Right", NULL, "OUTPGA Right Enable"}, + {"OUT1R", NULL, "Out1 Right"}, + + {"Headphone Left", NULL, "OUTPGA Left Enable"}, + {"HPL", NULL, "Headphone Left"}, + + {"Headphone Right", NULL, "OUTPGA Right Enable"}, + {"HPR", NULL, "Headphone Right"}, + + {"Out2 Mono", NULL, "Mono Mixer"}, + {"OUT2", NULL, "Out2 Mono"}, }; /* Codec private data */ struct da7210_priv { enum snd_soc_control_type control_type; - void *control_data; }; /* @@ -188,72 +582,15 @@ static const u8 da7210_reg[] = { 0x00, /* R88 */ }; -/* - * Read da7210 register cache - */ -static inline u32 da7210_read_reg_cache(struct snd_soc_codec *codec, u32 reg) -{ - u8 *cache = codec->reg_cache; - BUG_ON(reg >= ARRAY_SIZE(da7210_reg)); - return cache[reg]; -} - -/* - * Write to the da7210 register space - */ -static int da7210_write(struct snd_soc_codec *codec, u32 reg, u32 value) +static int da7210_volatile_register(struct snd_soc_codec *codec, + unsigned int reg) { - u8 *cache = codec->reg_cache; - u8 data[2]; - - BUG_ON(codec->driver->volatile_register); - - data[0] = reg & 0xff; - data[1] = value & 0xff; - - if (reg >= codec->driver->reg_cache_size) - return -EIO; - - if (2 != codec->hw_write(codec->control_data, data, 2)) - return -EIO; - - cache[reg] = value; - return 0; -} - -/* - * Read from the da7210 register space. - */ -static inline u32 da7210_read(struct snd_soc_codec *codec, u32 reg) -{ - if (DA7210_STATUS == reg) - return i2c_smbus_read_byte_data(codec->control_data, reg); - - return da7210_read_reg_cache(codec, reg); -} - -static int da7210_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; - struct snd_soc_codec *codec = dai->codec; - - if (is_play) { - /* Enable Out */ - snd_soc_update_bits(codec, DA7210_OUTMIX_L, 0x1F, 0x10); - snd_soc_update_bits(codec, DA7210_OUTMIX_R, 0x1F, 0x10); - - } else { - /* Volume 7 */ - snd_soc_update_bits(codec, DA7210_MIC_L, 0x7, 0x7); - snd_soc_update_bits(codec, DA7210_MIC_R, 0x7, 0x7); - - /* Enable Mic */ - snd_soc_update_bits(codec, DA7210_INMIX_L, 0x1F, 0x1); - snd_soc_update_bits(codec, DA7210_INMIX_R, 0x1F, 0x1); + switch (reg) { + case DA7210_STATUS: + return 1; + default: + return 0; } - - return 0; } /* @@ -266,93 +603,75 @@ static int da7210_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_codec *codec = rtd->codec; u32 dai_cfg1; - u32 hpf_reg, hpf_mask, hpf_value; u32 fs, bypass; /* set DAI source to Left and Right ADC */ - da7210_write(codec, DA7210_DAI_SRC_SEL, + snd_soc_write(codec, DA7210_DAI_SRC_SEL, DA7210_DAI_OUT_R_SRC | DA7210_DAI_OUT_L_SRC); /* Enable DAI */ - da7210_write(codec, DA7210_DAI_CFG3, DA7210_DAI_OE | DA7210_DAI_EN); + snd_soc_write(codec, DA7210_DAI_CFG3, DA7210_DAI_OE | DA7210_DAI_EN); - dai_cfg1 = 0xFC & da7210_read(codec, DA7210_DAI_CFG1); + dai_cfg1 = 0xFC & snd_soc_read(codec, DA7210_DAI_CFG1); switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: dai_cfg1 |= DA7210_DAI_WORD_S16_LE; break; + case SNDRV_PCM_FORMAT_S20_3LE: + dai_cfg1 |= DA7210_DAI_WORD_S20_3LE; + break; case SNDRV_PCM_FORMAT_S24_LE: dai_cfg1 |= DA7210_DAI_WORD_S24_LE; break; + case SNDRV_PCM_FORMAT_S32_LE: + dai_cfg1 |= DA7210_DAI_WORD_S32_LE; + break; default: return -EINVAL; } - da7210_write(codec, DA7210_DAI_CFG1, dai_cfg1); - - hpf_reg = (SNDRV_PCM_STREAM_PLAYBACK == substream->stream) ? - DA7210_DAC_HPF : DA7210_ADC_HPF; + snd_soc_write(codec, DA7210_DAI_CFG1, dai_cfg1); switch (params_rate(params)) { case 8000: fs = DA7210_PLL_FS_8000; - hpf_mask = DA7210_VOICE_F0_MASK | DA7210_VOICE_EN; - hpf_value = DA7210_VOICE_F0_25 | DA7210_VOICE_EN; bypass = DA7210_PLL_BYP; break; case 11025: fs = DA7210_PLL_FS_11025; - hpf_mask = DA7210_VOICE_F0_MASK | DA7210_VOICE_EN; - hpf_value = DA7210_VOICE_F0_25 | DA7210_VOICE_EN; bypass = 0; break; case 12000: fs = DA7210_PLL_FS_12000; - hpf_mask = DA7210_VOICE_F0_MASK | DA7210_VOICE_EN; - hpf_value = DA7210_VOICE_F0_25 | DA7210_VOICE_EN; bypass = DA7210_PLL_BYP; break; case 16000: fs = DA7210_PLL_FS_16000; - hpf_mask = DA7210_VOICE_F0_MASK | DA7210_VOICE_EN; - hpf_value = DA7210_VOICE_F0_25 | DA7210_VOICE_EN; bypass = DA7210_PLL_BYP; break; case 22050: fs = DA7210_PLL_FS_22050; - hpf_mask = DA7210_VOICE_EN; - hpf_value = 0; bypass = 0; break; case 32000: fs = DA7210_PLL_FS_32000; - hpf_mask = DA7210_VOICE_EN; - hpf_value = 0; bypass = DA7210_PLL_BYP; break; case 44100: fs = DA7210_PLL_FS_44100; - hpf_mask = DA7210_VOICE_EN; - hpf_value = 0; bypass = 0; break; case 48000: fs = DA7210_PLL_FS_48000; - hpf_mask = DA7210_VOICE_EN; - hpf_value = 0; bypass = DA7210_PLL_BYP; break; case 88200: fs = DA7210_PLL_FS_88200; - hpf_mask = DA7210_VOICE_EN; - hpf_value = 0; bypass = 0; break; case 96000: fs = DA7210_PLL_FS_96000; - hpf_mask = DA7210_VOICE_EN; - hpf_value = 0; bypass = DA7210_PLL_BYP; break; default: @@ -362,7 +681,6 @@ static int da7210_hw_params(struct snd_pcm_substream *substream, /* Disable active mode */ snd_soc_update_bits(codec, DA7210_STARTUP1, DA7210_SC_MST_EN, 0); - snd_soc_update_bits(codec, hpf_reg, hpf_mask, hpf_value); snd_soc_update_bits(codec, DA7210_PLL, DA7210_PLL_FS_MASK, fs); snd_soc_update_bits(codec, DA7210_PLL_DIV3, DA7210_PLL_BYP, bypass); @@ -382,13 +700,16 @@ static int da7210_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt) u32 dai_cfg1; u32 dai_cfg3; - dai_cfg1 = 0x7f & da7210_read(codec, DA7210_DAI_CFG1); - dai_cfg3 = 0xfc & da7210_read(codec, DA7210_DAI_CFG3); + dai_cfg1 = 0x7f & snd_soc_read(codec, DA7210_DAI_CFG1); + dai_cfg3 = 0xfc & snd_soc_read(codec, DA7210_DAI_CFG3); switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: dai_cfg1 |= DA7210_DAI_MODE_MASTER; break; + case SND_SOC_DAIFMT_CBS_CFS: + dai_cfg1 |= DA7210_DAI_MODE_SLAVE; + break; default: return -EINVAL; } @@ -401,6 +722,12 @@ static int da7210_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt) case SND_SOC_DAIFMT_I2S: dai_cfg3 |= DA7210_DAI_FORMAT_I2SMODE; break; + case SND_SOC_DAIFMT_LEFT_J: + dai_cfg3 |= DA7210_DAI_FORMAT_LEFT_J; + break; + case SND_SOC_DAIFMT_RIGHT_J: + dai_cfg3 |= DA7210_DAI_FORMAT_RIGHT_J; + break; default: return -EINVAL; } @@ -411,19 +738,32 @@ static int da7210_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt) */ dai_cfg1 |= DA7210_DAI_FLEN_64BIT; - da7210_write(codec, DA7210_DAI_CFG1, dai_cfg1); - da7210_write(codec, DA7210_DAI_CFG3, dai_cfg3); + snd_soc_write(codec, DA7210_DAI_CFG1, dai_cfg1); + snd_soc_write(codec, DA7210_DAI_CFG3, dai_cfg3); + + return 0; +} + +static int da7210_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u8 mute_reg = snd_soc_read(codec, DA7210_DAC_HPF) & 0xFB; + if (mute) + snd_soc_write(codec, DA7210_DAC_HPF, mute_reg | 0x4); + else + snd_soc_write(codec, DA7210_DAC_HPF, mute_reg); return 0; } -#define DA7210_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) +#define DA7210_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) /* DAI operations */ static struct snd_soc_dai_ops da7210_dai_ops = { - .startup = da7210_startup, .hw_params = da7210_hw_params, .set_fmt = da7210_set_dai_fmt, + .digital_mute = da7210_mute, }; static struct snd_soc_dai_driver da7210_dai = { @@ -451,11 +791,15 @@ static struct snd_soc_dai_driver da7210_dai = { static int da7210_probe(struct snd_soc_codec *codec) { struct da7210_priv *da7210 = snd_soc_codec_get_drvdata(codec); + int ret; dev_info(codec->dev, "DA7210 Audio Codec %s\n", DA7210_VERSION); - codec->control_data = da7210->control_data; - codec->hw_write = (hw_write_t)i2c_master_send; + ret = snd_soc_codec_set_cache_io(codec, 8, 8, da7210->control_type); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } /* FIXME * @@ -472,8 +816,8 @@ static int da7210_probe(struct snd_soc_codec *codec) /* * make sure that DA7210 use bypass mode before start up */ - da7210_write(codec, DA7210_STARTUP1, 0); - da7210_write(codec, DA7210_PLL_DIV3, + snd_soc_write(codec, DA7210_STARTUP1, 0); + snd_soc_write(codec, DA7210_PLL_DIV3, DA7210_MCLK_RANGE_10_20_MHZ | DA7210_PLL_BYP); /* @@ -481,36 +825,70 @@ static int da7210_probe(struct snd_soc_codec *codec) */ /* Enable Left & Right MIC PGA and Mic Bias */ - da7210_write(codec, DA7210_MIC_L, DA7210_MIC_L_EN | DA7210_MICBIAS_EN); - da7210_write(codec, DA7210_MIC_R, DA7210_MIC_R_EN); + snd_soc_write(codec, DA7210_MIC_L, DA7210_MIC_L_EN | DA7210_MICBIAS_EN); + snd_soc_write(codec, DA7210_MIC_R, DA7210_MIC_R_EN); /* Enable Left and Right input PGA */ - da7210_write(codec, DA7210_INMIX_L, DA7210_IN_L_EN); - da7210_write(codec, DA7210_INMIX_R, DA7210_IN_R_EN); + snd_soc_write(codec, DA7210_INMIX_L, DA7210_IN_L_EN); + snd_soc_write(codec, DA7210_INMIX_R, DA7210_IN_R_EN); /* Enable Left and Right ADC */ - da7210_write(codec, DA7210_ADC, DA7210_ADC_L_EN | DA7210_ADC_R_EN); + snd_soc_write(codec, DA7210_ADC, DA7210_ADC_L_EN | DA7210_ADC_R_EN); /* * DAC settings */ /* Enable Left and Right DAC */ - da7210_write(codec, DA7210_DAC_SEL, + snd_soc_write(codec, DA7210_DAC_SEL, DA7210_DAC_L_SRC_DAI_L | DA7210_DAC_L_EN | DA7210_DAC_R_SRC_DAI_R | DA7210_DAC_R_EN); /* Enable Left and Right out PGA */ - da7210_write(codec, DA7210_OUTMIX_L, DA7210_OUT_L_EN); - da7210_write(codec, DA7210_OUTMIX_R, DA7210_OUT_R_EN); + snd_soc_write(codec, DA7210_OUTMIX_L, DA7210_OUT_L_EN); + snd_soc_write(codec, DA7210_OUTMIX_R, DA7210_OUT_R_EN); /* Enable Left and Right HeadPhone PGA */ - da7210_write(codec, DA7210_HP_CFG, + snd_soc_write(codec, DA7210_HP_CFG, DA7210_HP_2CAP_MODE | DA7210_HP_SENSE_EN | DA7210_HP_L_EN | DA7210_HP_MODE | DA7210_HP_R_EN); + /* Enable ramp mode for DAC gain update */ + snd_soc_write(codec, DA7210_SOFTMUTE, DA7210_RAMP_EN); + + /* + * For DA7210 codec, there are two ways to enable/disable analog IOs + * and ADC/DAC, + * (1) Using "Enable Bit" of register associated with that IO + * (or ADC/DAC) + * e.g. Mic Left can be enabled using bit 7 of MIC_L(0x7) reg + * + * (2) Using "Standby Bit" of STARTUP2 or STARTUP3 register + * e.g. Mic left can be put to STANDBY using bit 0 of STARTUP3(0x5) + * + * Out of these two methods, the one using STANDBY bits is preferred + * way to enable/disable individual blocks. This is because STANDBY + * registers are part of system controller which allows system power + * up/down in a controlled, pop-free manner. Also, as per application + * note of DA7210, STANDBY register bits are only effective if a + * particular IO (or ADC/DAC) is already enabled using enable/disable + * register bits. Keeping these things in mind, current DAPM + * implementation manipulates only STANDBY bits. + * + * Overall implementation can be outlined as below, + * + * - "Enable bit" of an IO or ADC/DAC is used to enable it in probe() + * - "STANDBY bit" is controlled by DAPM + */ + + /* Enable Line out amplifiers */ + snd_soc_write(codec, DA7210_OUT1_L, DA7210_OUT1_L_EN); + snd_soc_write(codec, DA7210_OUT1_R, DA7210_OUT1_R_EN); + snd_soc_write(codec, DA7210_OUT2, DA7210_OUT2_EN | + DA7210_OUT2_OUTMIX_L | DA7210_OUT2_OUTMIX_R); + /* Diable PLL and bypass it */ - da7210_write(codec, DA7210_PLL, DA7210_PLL_FS_48000); + snd_soc_write(codec, DA7210_PLL, DA7210_PLL_FS_48000); /* * If 48kHz sound came, it use bypass mode, @@ -521,25 +899,22 @@ static int da7210_probe(struct snd_soc_codec *codec) * DA7210_PLL_DIV3 :: DA7210_PLL_BYP bit. * see da7210_hw_params */ - da7210_write(codec, DA7210_PLL_DIV1, 0xE5); /* MCLK = 12.288MHz */ - da7210_write(codec, DA7210_PLL_DIV2, 0x99); - da7210_write(codec, DA7210_PLL_DIV3, 0x0A | + snd_soc_write(codec, DA7210_PLL_DIV1, 0xE5); /* MCLK = 12.288MHz */ + snd_soc_write(codec, DA7210_PLL_DIV2, 0x99); + snd_soc_write(codec, DA7210_PLL_DIV3, 0x0A | DA7210_MCLK_RANGE_10_20_MHZ | DA7210_PLL_BYP); snd_soc_update_bits(codec, DA7210_PLL, DA7210_PLL_EN, DA7210_PLL_EN); /* As suggested by Dialog */ - da7210_write(codec, DA7210_A_HID_UNLOCK, 0x8B); /* unlock */ - da7210_write(codec, DA7210_A_TEST_UNLOCK, 0xB4); - da7210_write(codec, DA7210_A_PLL1, 0x01); - da7210_write(codec, DA7210_A_CP_MODE, 0x7C); - da7210_write(codec, DA7210_A_HID_UNLOCK, 0x00); /* re-lock */ - da7210_write(codec, DA7210_A_TEST_UNLOCK, 0x00); + snd_soc_write(codec, DA7210_A_HID_UNLOCK, 0x8B); /* unlock */ + snd_soc_write(codec, DA7210_A_TEST_UNLOCK, 0xB4); + snd_soc_write(codec, DA7210_A_PLL1, 0x01); + snd_soc_write(codec, DA7210_A_CP_MODE, 0x7C); + snd_soc_write(codec, DA7210_A_HID_UNLOCK, 0x00); /* re-lock */ + snd_soc_write(codec, DA7210_A_TEST_UNLOCK, 0x00); /* Activate all enabled subsystem */ - da7210_write(codec, DA7210_STARTUP1, DA7210_SC_MST_EN); - - snd_soc_add_controls(codec, da7210_snd_controls, - ARRAY_SIZE(da7210_snd_controls)); + snd_soc_write(codec, DA7210_STARTUP1, DA7210_SC_MST_EN); dev_info(codec->dev, "DA7210 Audio Codec %s\n", DA7210_VERSION); @@ -548,11 +923,18 @@ static int da7210_probe(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_dev_da7210 = { .probe = da7210_probe, - .read = da7210_read, - .write = da7210_write, .reg_cache_size = ARRAY_SIZE(da7210_reg), .reg_word_size = sizeof(u8), .reg_cache_default = da7210_reg, + .volatile_register = da7210_volatile_register, + + .controls = da7210_snd_controls, + .num_controls = ARRAY_SIZE(da7210_snd_controls), + + .dapm_widgets = da7210_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(da7210_dapm_widgets), + .dapm_routes = da7210_audio_map, + .num_dapm_routes = ARRAY_SIZE(da7210_audio_map), }; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) @@ -567,7 +949,6 @@ static int __devinit da7210_i2c_probe(struct i2c_client *i2c, return -ENOMEM; i2c_set_clientdata(i2c, da7210); - da7210->control_data = i2c; da7210->control_type = SND_SOC_I2C; ret = snd_soc_register_codec(&i2c->dev, diff --git a/sound/soc/codecs/lm4857.c b/sound/soc/codecs/lm4857.c index 2c2a681da0d7..c387dafc6ab6 100644 --- a/sound/soc/codecs/lm4857.c +++ b/sound/soc/codecs/lm4857.c @@ -3,7 +3,7 @@ * * Copyright 2007 Wolfson Microelectronics PLC. * Author: Graeme Gregory - * graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.com + * graeme.gregory@wolfsonmicro.com * Copyright 2011 Lars-Peter Clausen <lars@metafoo.de> * * This program is free software; you can redistribute it and/or modify it diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index ac65a2d36408..ebbf63c79c34 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -40,7 +40,6 @@ struct max98088_cdata { struct max98088_priv { enum max98088_type devtype; - void *control_data; struct max98088_pdata *pdata; unsigned int sysclk; struct max98088_cdata dai[2]; @@ -1697,13 +1696,19 @@ static struct snd_soc_dai_driver max98088_dai[] = { } }; -static int max98088_get_channel(const char *name) +static const char *eq_mode_name[] = {"EQ1 Mode", "EQ2 Mode"}; + +static int max98088_get_channel(struct snd_soc_codec *codec, const char *name) { - if (strcmp(name, "EQ1 Mode") == 0) - return 0; - if (strcmp(name, "EQ2 Mode") == 0) - return 1; - return -EINVAL; + int i; + + for (i = 0; i < ARRAY_SIZE(eq_mode_name); i++) + if (strcmp(name, eq_mode_name[i]) == 0) + return i; + + /* Shouldn't happen */ + dev_err(codec->dev, "Bad EQ channel name '%s'\n", name); + return -EINVAL; } static void max98088_setup_eq1(struct snd_soc_codec *codec) @@ -1807,10 +1812,13 @@ static int max98088_put_eq_enum(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct max98088_priv *max98088 = snd_soc_codec_get_drvdata(codec); struct max98088_pdata *pdata = max98088->pdata; - int channel = max98088_get_channel(kcontrol->id.name); + int channel = max98088_get_channel(codec, kcontrol->id.name); struct max98088_cdata *cdata; int sel = ucontrol->value.integer.value[0]; + if (channel < 0) + return channel; + cdata = &max98088->dai[channel]; if (sel >= pdata->eq_cfgcnt) @@ -1835,9 +1843,12 @@ static int max98088_get_eq_enum(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct max98088_priv *max98088 = snd_soc_codec_get_drvdata(codec); - int channel = max98088_get_channel(kcontrol->id.name); + int channel = max98088_get_channel(codec, kcontrol->id.name); struct max98088_cdata *cdata; + if (channel < 0) + return channel; + cdata = &max98088->dai[channel]; ucontrol->value.enumerated.item[0] = cdata->eq_sel; return 0; @@ -1852,17 +1863,17 @@ static void max98088_handle_eq_pdata(struct snd_soc_codec *codec) int i, j; const char **t; int ret; - struct snd_kcontrol_new controls[] = { - SOC_ENUM_EXT("EQ1 Mode", + SOC_ENUM_EXT((char *)eq_mode_name[0], max98088->eq_enum, max98088_get_eq_enum, max98088_put_eq_enum), - SOC_ENUM_EXT("EQ2 Mode", + SOC_ENUM_EXT((char *)eq_mode_name[1], max98088->eq_enum, max98088_get_eq_enum, max98088_put_eq_enum), }; + BUILD_BUG_ON(ARRAY_SIZE(controls) != ARRAY_SIZE(eq_mode_name)); cfg = pdata->eq_cfg; cfgcnt = pdata->eq_cfgcnt; @@ -2066,7 +2077,6 @@ static int max98088_i2c_probe(struct i2c_client *i2c, max98088->devtype = id->driver_data; i2c_set_clientdata(i2c, max98088); - max98088->control_data = i2c; max98088->pdata = i2c->dev.platform_data; ret = snd_soc_register_codec(&i2c->dev, diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c index 668434d44303..26d7b089fb9c 100644 --- a/sound/soc/codecs/max98095.c +++ b/sound/soc/codecs/max98095.c @@ -40,7 +40,6 @@ struct max98095_cdata { struct max98095_priv { enum max98095_type devtype; - void *control_data; struct max98095_pdata *pdata; unsigned int sysclk; struct max98095_cdata dai[3]; @@ -618,14 +617,13 @@ static int max98095_volatile(struct snd_soc_codec *codec, unsigned int reg) static int max98095_hw_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { - u8 data[2]; + int ret; - data[0] = reg; - data[1] = value; - if (codec->hw_write(codec->control_data, data, 2) == 2) - return 0; - else - return -EIO; + codec->cache_bypass = 1; + ret = snd_soc_write(codec, reg, value); + codec->cache_bypass = 0; + + return ret ? -EIO : 0; } /* @@ -1992,12 +1990,19 @@ static void max98095_handle_eq_pdata(struct snd_soc_codec *codec) dev_err(codec->dev, "Failed to add EQ control: %d\n", ret); } -static int max98095_get_bq_channel(const char *name) +static const char *bq_mode_name[] = {"Biquad1 Mode", "Biquad2 Mode"}; + +static int max98095_get_bq_channel(struct snd_soc_codec *codec, + const char *name) { - if (strcmp(name, "Biquad1 Mode") == 0) - return 0; - if (strcmp(name, "Biquad2 Mode") == 0) - return 1; + int i; + + for (i = 0; i < ARRAY_SIZE(bq_mode_name); i++) + if (strcmp(name, bq_mode_name[i]) == 0) + return i; + + /* Shouldn't happen */ + dev_err(codec->dev, "Bad biquad channel name '%s'\n", name); return -EINVAL; } @@ -2007,14 +2012,15 @@ static int max98095_put_bq_enum(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec); struct max98095_pdata *pdata = max98095->pdata; - int channel = max98095_get_bq_channel(kcontrol->id.name); + int channel = max98095_get_bq_channel(codec, kcontrol->id.name); struct max98095_cdata *cdata; int sel = ucontrol->value.integer.value[0]; struct max98095_biquad_cfg *coef_set; int fs, best, best_val, i; int regmask, regsave; - BUG_ON(channel > 1); + if (channel < 0) + return channel; if (!pdata || !max98095->bq_textcnt) return 0; @@ -2066,9 +2072,12 @@ static int max98095_get_bq_enum(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec); - int channel = max98095_get_bq_channel(kcontrol->id.name); + int channel = max98095_get_bq_channel(codec, kcontrol->id.name); struct max98095_cdata *cdata; + if (channel < 0) + return channel; + cdata = &max98095->dai[channel]; ucontrol->value.enumerated.item[0] = cdata->bq_sel; @@ -2086,15 +2095,16 @@ static void max98095_handle_bq_pdata(struct snd_soc_codec *codec) int ret; struct snd_kcontrol_new controls[] = { - SOC_ENUM_EXT("Biquad1 Mode", + SOC_ENUM_EXT((char *)bq_mode_name[0], max98095->bq_enum, max98095_get_bq_enum, max98095_put_bq_enum), - SOC_ENUM_EXT("Biquad2 Mode", + SOC_ENUM_EXT((char *)bq_mode_name[1], max98095->bq_enum, max98095_get_bq_enum, max98095_put_bq_enum), }; + BUILD_BUG_ON(ARRAY_SIZE(controls) != ARRAY_SIZE(bq_mode_name)); cfg = pdata->bq_cfg; cfgcnt = pdata->bq_cfgcnt; @@ -2337,7 +2347,6 @@ static int max98095_i2c_probe(struct i2c_client *i2c, max98095->devtype = id->driver_data; i2c_set_clientdata(i2c, max98095); - max98095->control_data = i2c; max98095->pdata = i2c->dev.platform_data; ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_max98095, diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c new file mode 100644 index 000000000000..27a078cbb6eb --- /dev/null +++ b/sound/soc/codecs/rt5631.c @@ -0,0 +1,1773 @@ +/* + * rt5631.c -- RT5631 ALSA Soc Audio driver + * + * Copyright 2011 Realtek Microelectronics + * + * Author: flove <flove@realtek.com> + * + * Based on WM8753.c + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + */ +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/spi/spi.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include "rt5631.h" + +struct rt5631_priv { + int codec_version; + int master; + int sysclk; + int rx_rate; + int bclk_rate; + int dmic_used_flag; +}; + +static const u16 rt5631_reg[RT5631_VENDOR_ID2 + 1] = { + [RT5631_SPK_OUT_VOL] = 0x8888, + [RT5631_HP_OUT_VOL] = 0x8080, + [RT5631_MONO_AXO_1_2_VOL] = 0xa080, + [RT5631_AUX_IN_VOL] = 0x0808, + [RT5631_ADC_REC_MIXER] = 0xf0f0, + [RT5631_VDAC_DIG_VOL] = 0x0010, + [RT5631_OUTMIXER_L_CTRL] = 0xffc0, + [RT5631_OUTMIXER_R_CTRL] = 0xffc0, + [RT5631_AXO1MIXER_CTRL] = 0x88c0, + [RT5631_AXO2MIXER_CTRL] = 0x88c0, + [RT5631_DIG_MIC_CTRL] = 0x3000, + [RT5631_MONO_INPUT_VOL] = 0x8808, + [RT5631_SPK_MIXER_CTRL] = 0xf8f8, + [RT5631_SPK_MONO_OUT_CTRL] = 0xfc00, + [RT5631_SPK_MONO_HP_OUT_CTRL] = 0x4440, + [RT5631_SDP_CTRL] = 0x8000, + [RT5631_MONO_SDP_CTRL] = 0x8000, + [RT5631_STEREO_AD_DA_CLK_CTRL] = 0x2010, + [RT5631_GEN_PUR_CTRL_REG] = 0x0e00, + [RT5631_INT_ST_IRQ_CTRL_2] = 0x071a, + [RT5631_MISC_CTRL] = 0x2040, + [RT5631_DEPOP_FUN_CTRL_2] = 0x8000, + [RT5631_SOFT_VOL_CTRL] = 0x07e0, + [RT5631_ALC_CTRL_1] = 0x0206, + [RT5631_ALC_CTRL_3] = 0x2000, + [RT5631_PSEUDO_SPATL_CTRL] = 0x0553, +}; + +/** + * rt5631_write_index - write index register of 2nd layer + */ +static void rt5631_write_index(struct snd_soc_codec *codec, + unsigned int reg, unsigned int value) +{ + snd_soc_write(codec, RT5631_INDEX_ADD, reg); + snd_soc_write(codec, RT5631_INDEX_DATA, value); +} + +/** + * rt5631_read_index - read index register of 2nd layer + */ +static unsigned int rt5631_read_index(struct snd_soc_codec *codec, + unsigned int reg) +{ + unsigned int value; + + snd_soc_write(codec, RT5631_INDEX_ADD, reg); + value = snd_soc_read(codec, RT5631_INDEX_DATA); + + return value; +} + +static int rt5631_reset(struct snd_soc_codec *codec) +{ + return snd_soc_write(codec, RT5631_RESET, 0); +} + +static int rt5631_volatile_register(struct snd_soc_codec *codec, + unsigned int reg) +{ + switch (reg) { + case RT5631_RESET: + case RT5631_INT_ST_IRQ_CTRL_2: + case RT5631_INDEX_ADD: + case RT5631_INDEX_DATA: + case RT5631_EQ_CTRL: + return 1; + default: + return 0; + } +} + +static int rt5631_readable_register(struct snd_soc_codec *codec, + unsigned int reg) +{ + switch (reg) { + case RT5631_RESET: + case RT5631_SPK_OUT_VOL: + case RT5631_HP_OUT_VOL: + case RT5631_MONO_AXO_1_2_VOL: + case RT5631_AUX_IN_VOL: + case RT5631_STEREO_DAC_VOL_1: + case RT5631_MIC_CTRL_1: + case RT5631_STEREO_DAC_VOL_2: + case RT5631_ADC_CTRL_1: + case RT5631_ADC_REC_MIXER: + case RT5631_ADC_CTRL_2: + case RT5631_VDAC_DIG_VOL: + case RT5631_OUTMIXER_L_CTRL: + case RT5631_OUTMIXER_R_CTRL: + case RT5631_AXO1MIXER_CTRL: + case RT5631_AXO2MIXER_CTRL: + case RT5631_MIC_CTRL_2: + case RT5631_DIG_MIC_CTRL: + case RT5631_MONO_INPUT_VOL: + case RT5631_SPK_MIXER_CTRL: + case RT5631_SPK_MONO_OUT_CTRL: + case RT5631_SPK_MONO_HP_OUT_CTRL: + case RT5631_SDP_CTRL: + case RT5631_MONO_SDP_CTRL: + case RT5631_STEREO_AD_DA_CLK_CTRL: + case RT5631_PWR_MANAG_ADD1: + case RT5631_PWR_MANAG_ADD2: + case RT5631_PWR_MANAG_ADD3: + case RT5631_PWR_MANAG_ADD4: + case RT5631_GEN_PUR_CTRL_REG: + case RT5631_GLOBAL_CLK_CTRL: + case RT5631_PLL_CTRL: + case RT5631_INT_ST_IRQ_CTRL_1: + case RT5631_INT_ST_IRQ_CTRL_2: + case RT5631_GPIO_CTRL: + case RT5631_MISC_CTRL: + case RT5631_DEPOP_FUN_CTRL_1: + case RT5631_DEPOP_FUN_CTRL_2: + case RT5631_JACK_DET_CTRL: + case RT5631_SOFT_VOL_CTRL: + case RT5631_ALC_CTRL_1: + case RT5631_ALC_CTRL_2: + case RT5631_ALC_CTRL_3: + case RT5631_PSEUDO_SPATL_CTRL: + case RT5631_INDEX_ADD: + case RT5631_INDEX_DATA: + case RT5631_EQ_CTRL: + case RT5631_VENDOR_ID: + case RT5631_VENDOR_ID1: + case RT5631_VENDOR_ID2: + return 1; + default: + return 0; + } +} + +static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -4650, 150, 0); +static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -95625, 375, 0); +static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -3450, 150, 0); +/* {0, +20, +24, +30, +35, +40, +44, +50, +52}dB */ +static unsigned int mic_bst_tlv[] = { + TLV_DB_RANGE_HEAD(6), + 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), + 1, 1, TLV_DB_SCALE_ITEM(2000, 0, 0), + 2, 2, TLV_DB_SCALE_ITEM(2400, 0, 0), + 3, 5, TLV_DB_SCALE_ITEM(3000, 500, 0), + 6, 6, TLV_DB_SCALE_ITEM(4400, 0, 0), + 7, 7, TLV_DB_SCALE_ITEM(5000, 0, 0), + 8, 8, TLV_DB_SCALE_ITEM(5200, 0, 0), +}; + +static int rt5631_dmic_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct rt5631_priv *rt5631 = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.integer.value[0] = rt5631->dmic_used_flag; + + return 0; +} + +static int rt5631_dmic_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct rt5631_priv *rt5631 = snd_soc_codec_get_drvdata(codec); + + rt5631->dmic_used_flag = ucontrol->value.integer.value[0]; + return 0; +} + +/* MIC Input Type */ +static const char *rt5631_input_mode[] = { + "Single ended", "Differential"}; + +static const SOC_ENUM_SINGLE_DECL( + rt5631_mic1_mode_enum, RT5631_MIC_CTRL_1, + RT5631_MIC1_DIFF_INPUT_SHIFT, rt5631_input_mode); + +static const SOC_ENUM_SINGLE_DECL( + rt5631_mic2_mode_enum, RT5631_MIC_CTRL_1, + RT5631_MIC2_DIFF_INPUT_SHIFT, rt5631_input_mode); + +/* MONO Input Type */ +static const SOC_ENUM_SINGLE_DECL( + rt5631_monoin_mode_enum, RT5631_MONO_INPUT_VOL, + RT5631_MONO_DIFF_INPUT_SHIFT, rt5631_input_mode); + +/* SPK Ratio Gain Control */ +static const char *rt5631_spk_ratio[] = {"1.00x", "1.09x", "1.27x", "1.44x", + "1.56x", "1.68x", "1.99x", "2.34x"}; + +static const SOC_ENUM_SINGLE_DECL( + rt5631_spk_ratio_enum, RT5631_GEN_PUR_CTRL_REG, + RT5631_SPK_AMP_RATIO_CTRL_SHIFT, rt5631_spk_ratio); + +static const struct snd_kcontrol_new rt5631_snd_controls[] = { + /* MIC */ + SOC_ENUM("MIC1 Mode Control", rt5631_mic1_mode_enum), + SOC_SINGLE_TLV("MIC1 Boost", RT5631_MIC_CTRL_2, + RT5631_MIC1_BOOST_SHIFT, 8, 0, mic_bst_tlv), + SOC_ENUM("MIC2 Mode Control", rt5631_mic2_mode_enum), + SOC_SINGLE_TLV("MIC2 Boost", RT5631_MIC_CTRL_2, + RT5631_MIC2_BOOST_SHIFT, 8, 0, mic_bst_tlv), + /* MONO IN */ + SOC_ENUM("MONOIN Mode Control", rt5631_monoin_mode_enum), + SOC_DOUBLE_TLV("MONOIN_RX Capture Volume", RT5631_MONO_INPUT_VOL, + RT5631_L_VOL_SHIFT, RT5631_R_VOL_SHIFT, + RT5631_VOL_MASK, 1, in_vol_tlv), + /* AXI */ + SOC_DOUBLE_TLV("AXI Capture Volume", RT5631_AUX_IN_VOL, + RT5631_L_VOL_SHIFT, RT5631_R_VOL_SHIFT, + RT5631_VOL_MASK, 1, in_vol_tlv), + /* DAC */ + SOC_DOUBLE_TLV("PCM Playback Volume", RT5631_STEREO_DAC_VOL_2, + RT5631_L_VOL_SHIFT, RT5631_R_VOL_SHIFT, + RT5631_DAC_VOL_MASK, 1, dac_vol_tlv), + SOC_DOUBLE("PCM Playback Switch", RT5631_STEREO_DAC_VOL_1, + RT5631_L_MUTE_SHIFT, RT5631_R_MUTE_SHIFT, 1, 1), + /* AXO */ + SOC_SINGLE("AXO1 Playback Switch", RT5631_MONO_AXO_1_2_VOL, + RT5631_L_MUTE_SHIFT, 1, 1), + SOC_SINGLE("AXO2 Playback Switch", RT5631_MONO_AXO_1_2_VOL, + RT5631_R_VOL_SHIFT, 1, 1), + /* OUTVOL */ + SOC_DOUBLE("OUTVOL Channel Switch", RT5631_SPK_OUT_VOL, + RT5631_L_EN_SHIFT, RT5631_R_EN_SHIFT, 1, 0), + + /* SPK */ + SOC_DOUBLE("Speaker Playback Switch", RT5631_SPK_OUT_VOL, + RT5631_L_MUTE_SHIFT, RT5631_R_MUTE_SHIFT, 1, 1), + SOC_DOUBLE_TLV("Speaker Playback Volume", RT5631_SPK_OUT_VOL, + RT5631_L_VOL_SHIFT, RT5631_R_VOL_SHIFT, 39, 1, out_vol_tlv), + /* MONO OUT */ + SOC_SINGLE("MONO Playback Switch", RT5631_MONO_AXO_1_2_VOL, + RT5631_MUTE_MONO_SHIFT, 1, 1), + /* HP */ + SOC_DOUBLE("HP Playback Switch", RT5631_HP_OUT_VOL, + RT5631_L_MUTE_SHIFT, RT5631_R_MUTE_SHIFT, 1, 1), + SOC_DOUBLE_TLV("HP Playback Volume", RT5631_HP_OUT_VOL, + RT5631_L_VOL_SHIFT, RT5631_R_VOL_SHIFT, + RT5631_VOL_MASK, 1, out_vol_tlv), + /* DMIC */ + SOC_SINGLE_EXT("DMIC Switch", 0, 0, 1, 0, + rt5631_dmic_get, rt5631_dmic_put), + SOC_DOUBLE("DMIC Capture Switch", RT5631_DIG_MIC_CTRL, + RT5631_DMIC_L_CH_MUTE_SHIFT, + RT5631_DMIC_R_CH_MUTE_SHIFT, 1, 1), + + /* SPK Ratio Gain Control */ + SOC_ENUM("SPK Ratio Control", rt5631_spk_ratio_enum), +}; + +static int check_sysclk1_source(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + unsigned int reg; + + reg = snd_soc_read(source->codec, RT5631_GLOBAL_CLK_CTRL); + return reg & RT5631_SYSCLK_SOUR_SEL_PLL; +} + +static int check_dmic_used(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct rt5631_priv *rt5631 = snd_soc_codec_get_drvdata(source->codec); + return rt5631->dmic_used_flag; +} + +static int check_dacl_to_outmixl(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + unsigned int reg; + + reg = snd_soc_read(source->codec, RT5631_OUTMIXER_L_CTRL); + return !(reg & RT5631_M_DAC_L_TO_OUTMIXER_L); +} + +static int check_dacr_to_outmixr(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + unsigned int reg; + + reg = snd_soc_read(source->codec, RT5631_OUTMIXER_R_CTRL); + return !(reg & RT5631_M_DAC_R_TO_OUTMIXER_R); +} + +static int check_dacl_to_spkmixl(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + unsigned int reg; + + reg = snd_soc_read(source->codec, RT5631_SPK_MIXER_CTRL); + return !(reg & RT5631_M_DAC_L_TO_SPKMIXER_L); +} + +static int check_dacr_to_spkmixr(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + unsigned int reg; + + reg = snd_soc_read(source->codec, RT5631_SPK_MIXER_CTRL); + return !(reg & RT5631_M_DAC_R_TO_SPKMIXER_R); +} + +static int check_adcl_select(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + unsigned int reg; + + reg = snd_soc_read(source->codec, RT5631_ADC_REC_MIXER); + return !(reg & RT5631_M_MIC1_TO_RECMIXER_L); +} + +static int check_adcr_select(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + unsigned int reg; + + reg = snd_soc_read(source->codec, RT5631_ADC_REC_MIXER); + return !(reg & RT5631_M_MIC2_TO_RECMIXER_R); +} + +/** + * onebit_depop_power_stage - auto depop in power stage. + * @enable: power on/off + * + * When power on/off headphone, the depop sequence is done by hardware. + */ +static void onebit_depop_power_stage(struct snd_soc_codec *codec, int enable) +{ + unsigned int soft_vol, hp_zc; + + /* enable one-bit depop function */ + snd_soc_update_bits(codec, RT5631_DEPOP_FUN_CTRL_2, + RT5631_EN_ONE_BIT_DEPOP, 0); + + /* keep soft volume and zero crossing setting */ + soft_vol = snd_soc_read(codec, RT5631_SOFT_VOL_CTRL); + snd_soc_write(codec, RT5631_SOFT_VOL_CTRL, 0); + hp_zc = snd_soc_read(codec, RT5631_INT_ST_IRQ_CTRL_2); + snd_soc_write(codec, RT5631_INT_ST_IRQ_CTRL_2, hp_zc & 0xf7ff); + if (enable) { + /* config one-bit depop parameter */ + rt5631_write_index(codec, RT5631_TEST_MODE_CTRL, 0x84c0); + rt5631_write_index(codec, RT5631_SPK_INTL_CTRL, 0x309f); + rt5631_write_index(codec, RT5631_CP_INTL_REG2, 0x6530); + /* power on capless block */ + snd_soc_write(codec, RT5631_DEPOP_FUN_CTRL_2, + RT5631_EN_CAP_FREE_DEPOP); + } else { + /* power off capless block */ + snd_soc_write(codec, RT5631_DEPOP_FUN_CTRL_2, 0); + msleep(100); + } + + /* recover soft volume and zero crossing setting */ + snd_soc_write(codec, RT5631_SOFT_VOL_CTRL, soft_vol); + snd_soc_write(codec, RT5631_INT_ST_IRQ_CTRL_2, hp_zc); +} + +/** + * onebit_depop_mute_stage - auto depop in mute stage. + * @enable: mute/unmute + * + * When mute/unmute headphone, the depop sequence is done by hardware. + */ +static void onebit_depop_mute_stage(struct snd_soc_codec *codec, int enable) +{ + unsigned int soft_vol, hp_zc; + + /* enable one-bit depop function */ + snd_soc_update_bits(codec, RT5631_DEPOP_FUN_CTRL_2, + RT5631_EN_ONE_BIT_DEPOP, 0); + + /* keep soft volume and zero crossing setting */ + soft_vol = snd_soc_read(codec, RT5631_SOFT_VOL_CTRL); + snd_soc_write(codec, RT5631_SOFT_VOL_CTRL, 0); + hp_zc = snd_soc_read(codec, RT5631_INT_ST_IRQ_CTRL_2); + snd_soc_write(codec, RT5631_INT_ST_IRQ_CTRL_2, hp_zc & 0xf7ff); + if (enable) { + schedule_timeout_uninterruptible(msecs_to_jiffies(10)); + /* config one-bit depop parameter */ + rt5631_write_index(codec, RT5631_SPK_INTL_CTRL, 0x307f); + snd_soc_update_bits(codec, RT5631_HP_OUT_VOL, + RT5631_L_MUTE | RT5631_R_MUTE, 0); + msleep(300); + } else { + snd_soc_update_bits(codec, RT5631_HP_OUT_VOL, + RT5631_L_MUTE | RT5631_R_MUTE, + RT5631_L_MUTE | RT5631_R_MUTE); + msleep(100); + } + + /* recover soft volume and zero crossing setting */ + snd_soc_write(codec, RT5631_SOFT_VOL_CTRL, soft_vol); + snd_soc_write(codec, RT5631_INT_ST_IRQ_CTRL_2, hp_zc); +} + +/** + * onebit_depop_power_stage - step by step depop sequence in power stage. + * @enable: power on/off + * + * When power on/off headphone, the depop sequence is done in step by step. + */ +static void depop_seq_power_stage(struct snd_soc_codec *codec, int enable) +{ + unsigned int soft_vol, hp_zc; + + /* depop control by register */ + snd_soc_update_bits(codec, RT5631_DEPOP_FUN_CTRL_2, + RT5631_EN_ONE_BIT_DEPOP, RT5631_EN_ONE_BIT_DEPOP); + + /* keep soft volume and zero crossing setting */ + soft_vol = snd_soc_read(codec, RT5631_SOFT_VOL_CTRL); + snd_soc_write(codec, RT5631_SOFT_VOL_CTRL, 0); + hp_zc = snd_soc_read(codec, RT5631_INT_ST_IRQ_CTRL_2); + snd_soc_write(codec, RT5631_INT_ST_IRQ_CTRL_2, hp_zc & 0xf7ff); + if (enable) { + /* config depop sequence parameter */ + rt5631_write_index(codec, RT5631_SPK_INTL_CTRL, 0x303e); + + /* power on headphone and charge pump */ + snd_soc_update_bits(codec, RT5631_PWR_MANAG_ADD3, + RT5631_PWR_CHARGE_PUMP | RT5631_PWR_HP_L_AMP | + RT5631_PWR_HP_R_AMP, + RT5631_PWR_CHARGE_PUMP | RT5631_PWR_HP_L_AMP | + RT5631_PWR_HP_R_AMP); + + /* power on soft generator and depop mode2 */ + snd_soc_write(codec, RT5631_DEPOP_FUN_CTRL_1, + RT5631_POW_ON_SOFT_GEN | RT5631_EN_DEPOP2_FOR_HP); + msleep(100); + + /* stop depop mode */ + snd_soc_update_bits(codec, RT5631_PWR_MANAG_ADD3, + RT5631_PWR_HP_DEPOP_DIS, RT5631_PWR_HP_DEPOP_DIS); + } else { + /* config depop sequence parameter */ + rt5631_write_index(codec, RT5631_SPK_INTL_CTRL, 0x303F); + snd_soc_write(codec, RT5631_DEPOP_FUN_CTRL_1, + RT5631_POW_ON_SOFT_GEN | RT5631_EN_MUTE_UNMUTE_DEPOP | + RT5631_PD_HPAMP_L_ST_UP | RT5631_PD_HPAMP_R_ST_UP); + msleep(75); + snd_soc_write(codec, RT5631_DEPOP_FUN_CTRL_1, + RT5631_POW_ON_SOFT_GEN | RT5631_PD_HPAMP_L_ST_UP | + RT5631_PD_HPAMP_R_ST_UP); + + /* start depop mode */ + snd_soc_update_bits(codec, RT5631_PWR_MANAG_ADD3, + RT5631_PWR_HP_DEPOP_DIS, 0); + + /* config depop sequence parameter */ + snd_soc_write(codec, RT5631_DEPOP_FUN_CTRL_1, + RT5631_POW_ON_SOFT_GEN | RT5631_EN_DEPOP2_FOR_HP | + RT5631_PD_HPAMP_L_ST_UP | RT5631_PD_HPAMP_R_ST_UP); + msleep(80); + snd_soc_write(codec, RT5631_DEPOP_FUN_CTRL_1, + RT5631_POW_ON_SOFT_GEN); + + /* power down headphone and charge pump */ + snd_soc_update_bits(codec, RT5631_PWR_MANAG_ADD3, + RT5631_PWR_CHARGE_PUMP | RT5631_PWR_HP_L_AMP | + RT5631_PWR_HP_R_AMP, 0); + } + + /* recover soft volume and zero crossing setting */ + snd_soc_write(codec, RT5631_SOFT_VOL_CTRL, soft_vol); + snd_soc_write(codec, RT5631_INT_ST_IRQ_CTRL_2, hp_zc); +} + +/** + * depop_seq_mute_stage - step by step depop sequence in mute stage. + * @enable: mute/unmute + * + * When mute/unmute headphone, the depop sequence is done in step by step. + */ +static void depop_seq_mute_stage(struct snd_soc_codec *codec, int enable) +{ + unsigned int soft_vol, hp_zc; + + /* depop control by register */ + snd_soc_update_bits(codec, RT5631_DEPOP_FUN_CTRL_2, + RT5631_EN_ONE_BIT_DEPOP, RT5631_EN_ONE_BIT_DEPOP); + + /* keep soft volume and zero crossing setting */ + soft_vol = snd_soc_read(codec, RT5631_SOFT_VOL_CTRL); + snd_soc_write(codec, RT5631_SOFT_VOL_CTRL, 0); + hp_zc = snd_soc_read(codec, RT5631_INT_ST_IRQ_CTRL_2); + snd_soc_write(codec, RT5631_INT_ST_IRQ_CTRL_2, hp_zc & 0xf7ff); + if (enable) { + schedule_timeout_uninterruptible(msecs_to_jiffies(10)); + + /* config depop sequence parameter */ + rt5631_write_index(codec, RT5631_SPK_INTL_CTRL, 0x302f); + snd_soc_write(codec, RT5631_DEPOP_FUN_CTRL_1, + RT5631_POW_ON_SOFT_GEN | RT5631_EN_MUTE_UNMUTE_DEPOP | + RT5631_EN_HP_R_M_UN_MUTE_DEPOP | + RT5631_EN_HP_L_M_UN_MUTE_DEPOP); + + snd_soc_update_bits(codec, RT5631_HP_OUT_VOL, + RT5631_L_MUTE | RT5631_R_MUTE, 0); + msleep(160); + } else { + /* config depop sequence parameter */ + rt5631_write_index(codec, RT5631_SPK_INTL_CTRL, 0x302f); + snd_soc_write(codec, RT5631_DEPOP_FUN_CTRL_1, + RT5631_POW_ON_SOFT_GEN | RT5631_EN_MUTE_UNMUTE_DEPOP | + RT5631_EN_HP_R_M_UN_MUTE_DEPOP | + RT5631_EN_HP_L_M_UN_MUTE_DEPOP); + + snd_soc_update_bits(codec, RT5631_HP_OUT_VOL, + RT5631_L_MUTE | RT5631_R_MUTE, + RT5631_L_MUTE | RT5631_R_MUTE); + msleep(150); + } + + /* recover soft volume and zero crossing setting */ + snd_soc_write(codec, RT5631_SOFT_VOL_CTRL, soft_vol); + snd_soc_write(codec, RT5631_INT_ST_IRQ_CTRL_2, hp_zc); +} + +static int hp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt5631_priv *rt5631 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_PRE_PMD: + if (rt5631->codec_version) { + onebit_depop_mute_stage(codec, 0); + onebit_depop_power_stage(codec, 0); + } else { + depop_seq_mute_stage(codec, 0); + depop_seq_power_stage(codec, 0); + } + break; + + case SND_SOC_DAPM_POST_PMU: + if (rt5631->codec_version) { + onebit_depop_power_stage(codec, 1); + onebit_depop_mute_stage(codec, 1); + } else { + depop_seq_power_stage(codec, 1); + depop_seq_mute_stage(codec, 1); + } + break; + + default: + break; + } + + return 0; +} + +static int set_dmic_params(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt5631_priv *rt5631 = snd_soc_codec_get_drvdata(codec); + + switch (rt5631->rx_rate) { + case 44100: + case 48000: + snd_soc_update_bits(codec, RT5631_DIG_MIC_CTRL, + RT5631_DMIC_CLK_CTRL_MASK, + RT5631_DMIC_CLK_CTRL_TO_32FS); + break; + + case 32000: + case 22050: + snd_soc_update_bits(codec, RT5631_DIG_MIC_CTRL, + RT5631_DMIC_CLK_CTRL_MASK, + RT5631_DMIC_CLK_CTRL_TO_64FS); + break; + + case 16000: + case 11025: + case 8000: + snd_soc_update_bits(codec, RT5631_DIG_MIC_CTRL, + RT5631_DMIC_CLK_CTRL_MASK, + RT5631_DMIC_CLK_CTRL_TO_128FS); + break; + + default: + return -EINVAL; + } + + return 0; +} + +static const struct snd_kcontrol_new rt5631_recmixl_mixer_controls[] = { + SOC_DAPM_SINGLE("OUTMIXL Capture Switch", RT5631_ADC_REC_MIXER, + RT5631_M_OUTMIXL_RECMIXL_BIT, 1, 1), + SOC_DAPM_SINGLE("MIC1_BST1 Capture Switch", RT5631_ADC_REC_MIXER, + RT5631_M_MIC1_RECMIXL_BIT, 1, 1), + SOC_DAPM_SINGLE("AXILVOL Capture Switch", RT5631_ADC_REC_MIXER, + RT5631_M_AXIL_RECMIXL_BIT, 1, 1), + SOC_DAPM_SINGLE("MONOIN_RX Capture Switch", RT5631_ADC_REC_MIXER, + RT5631_M_MONO_IN_RECMIXL_BIT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5631_recmixr_mixer_controls[] = { + SOC_DAPM_SINGLE("MONOIN_RX Capture Switch", RT5631_ADC_REC_MIXER, + RT5631_M_MONO_IN_RECMIXR_BIT, 1, 1), + SOC_DAPM_SINGLE("AXIRVOL Capture Switch", RT5631_ADC_REC_MIXER, + RT5631_M_AXIR_RECMIXR_BIT, 1, 1), + SOC_DAPM_SINGLE("MIC2_BST2 Capture Switch", RT5631_ADC_REC_MIXER, + RT5631_M_MIC2_RECMIXR_BIT, 1, 1), + SOC_DAPM_SINGLE("OUTMIXR Capture Switch", RT5631_ADC_REC_MIXER, + RT5631_M_OUTMIXR_RECMIXR_BIT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5631_spkmixl_mixer_controls[] = { + SOC_DAPM_SINGLE("RECMIXL Playback Switch", RT5631_SPK_MIXER_CTRL, + RT5631_M_RECMIXL_SPKMIXL_BIT, 1, 1), + SOC_DAPM_SINGLE("MIC1_P Playback Switch", RT5631_SPK_MIXER_CTRL, + RT5631_M_MIC1P_SPKMIXL_BIT, 1, 1), + SOC_DAPM_SINGLE("DACL Playback Switch", RT5631_SPK_MIXER_CTRL, + RT5631_M_DACL_SPKMIXL_BIT, 1, 1), + SOC_DAPM_SINGLE("OUTMIXL Playback Switch", RT5631_SPK_MIXER_CTRL, + RT5631_M_OUTMIXL_SPKMIXL_BIT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5631_spkmixr_mixer_controls[] = { + SOC_DAPM_SINGLE("OUTMIXR Playback Switch", RT5631_SPK_MIXER_CTRL, + RT5631_M_OUTMIXR_SPKMIXR_BIT, 1, 1), + SOC_DAPM_SINGLE("DACR Playback Switch", RT5631_SPK_MIXER_CTRL, + RT5631_M_DACR_SPKMIXR_BIT, 1, 1), + SOC_DAPM_SINGLE("MIC2_P Playback Switch", RT5631_SPK_MIXER_CTRL, + RT5631_M_MIC2P_SPKMIXR_BIT, 1, 1), + SOC_DAPM_SINGLE("RECMIXR Playback Switch", RT5631_SPK_MIXER_CTRL, + RT5631_M_RECMIXR_SPKMIXR_BIT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5631_outmixl_mixer_controls[] = { + SOC_DAPM_SINGLE("RECMIXL Playback Switch", RT5631_OUTMIXER_L_CTRL, + RT5631_M_RECMIXL_OUTMIXL_BIT, 1, 1), + SOC_DAPM_SINGLE("RECMIXR Playback Switch", RT5631_OUTMIXER_L_CTRL, + RT5631_M_RECMIXR_OUTMIXL_BIT, 1, 1), + SOC_DAPM_SINGLE("DACL Playback Switch", RT5631_OUTMIXER_L_CTRL, + RT5631_M_DACL_OUTMIXL_BIT, 1, 1), + SOC_DAPM_SINGLE("MIC1_BST1 Playback Switch", RT5631_OUTMIXER_L_CTRL, + RT5631_M_MIC1_OUTMIXL_BIT, 1, 1), + SOC_DAPM_SINGLE("MIC2_BST2 Playback Switch", RT5631_OUTMIXER_L_CTRL, + RT5631_M_MIC2_OUTMIXL_BIT, 1, 1), + SOC_DAPM_SINGLE("MONOIN_RXP Playback Switch", RT5631_OUTMIXER_L_CTRL, + RT5631_M_MONO_INP_OUTMIXL_BIT, 1, 1), + SOC_DAPM_SINGLE("AXILVOL Playback Switch", RT5631_OUTMIXER_L_CTRL, + RT5631_M_AXIL_OUTMIXL_BIT, 1, 1), + SOC_DAPM_SINGLE("AXIRVOL Playback Switch", RT5631_OUTMIXER_L_CTRL, + RT5631_M_AXIR_OUTMIXL_BIT, 1, 1), + SOC_DAPM_SINGLE("VDAC Playback Switch", RT5631_OUTMIXER_L_CTRL, + RT5631_M_VDAC_OUTMIXL_BIT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5631_outmixr_mixer_controls[] = { + SOC_DAPM_SINGLE("VDAC Playback Switch", RT5631_OUTMIXER_R_CTRL, + RT5631_M_VDAC_OUTMIXR_BIT, 1, 1), + SOC_DAPM_SINGLE("AXIRVOL Playback Switch", RT5631_OUTMIXER_R_CTRL, + RT5631_M_AXIR_OUTMIXR_BIT, 1, 1), + SOC_DAPM_SINGLE("AXILVOL Playback Switch", RT5631_OUTMIXER_R_CTRL, + RT5631_M_AXIL_OUTMIXR_BIT, 1, 1), + SOC_DAPM_SINGLE("MONOIN_RXN Playback Switch", RT5631_OUTMIXER_R_CTRL, + RT5631_M_MONO_INN_OUTMIXR_BIT, 1, 1), + SOC_DAPM_SINGLE("MIC2_BST2 Playback Switch", RT5631_OUTMIXER_R_CTRL, + RT5631_M_MIC2_OUTMIXR_BIT, 1, 1), + SOC_DAPM_SINGLE("MIC1_BST1 Playback Switch", RT5631_OUTMIXER_R_CTRL, + RT5631_M_MIC1_OUTMIXR_BIT, 1, 1), + SOC_DAPM_SINGLE("DACR Playback Switch", RT5631_OUTMIXER_R_CTRL, + RT5631_M_DACR_OUTMIXR_BIT, 1, 1), + SOC_DAPM_SINGLE("RECMIXR Playback Switch", RT5631_OUTMIXER_R_CTRL, + RT5631_M_RECMIXR_OUTMIXR_BIT, 1, 1), + SOC_DAPM_SINGLE("RECMIXL Playback Switch", RT5631_OUTMIXER_R_CTRL, + RT5631_M_RECMIXL_OUTMIXR_BIT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5631_AXO1MIX_mixer_controls[] = { + SOC_DAPM_SINGLE("MIC1_BST1 Playback Switch", RT5631_AXO1MIXER_CTRL, + RT5631_M_MIC1_AXO1MIX_BIT , 1, 1), + SOC_DAPM_SINGLE("MIC2_BST2 Playback Switch", RT5631_AXO1MIXER_CTRL, + RT5631_M_MIC2_AXO1MIX_BIT, 1, 1), + SOC_DAPM_SINGLE("OUTVOLL Playback Switch", RT5631_AXO1MIXER_CTRL, + RT5631_M_OUTMIXL_AXO1MIX_BIT , 1 , 1), + SOC_DAPM_SINGLE("OUTVOLR Playback Switch", RT5631_AXO1MIXER_CTRL, + RT5631_M_OUTMIXR_AXO1MIX_BIT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5631_AXO2MIX_mixer_controls[] = { + SOC_DAPM_SINGLE("MIC1_BST1 Playback Switch", RT5631_AXO2MIXER_CTRL, + RT5631_M_MIC1_AXO2MIX_BIT, 1, 1), + SOC_DAPM_SINGLE("MIC2_BST2 Playback Switch", RT5631_AXO2MIXER_CTRL, + RT5631_M_MIC2_AXO2MIX_BIT, 1, 1), + SOC_DAPM_SINGLE("OUTVOLL Playback Switch", RT5631_AXO2MIXER_CTRL, + RT5631_M_OUTMIXL_AXO2MIX_BIT, 1, 1), + SOC_DAPM_SINGLE("OUTVOLR Playback Switch", RT5631_AXO2MIXER_CTRL, + RT5631_M_OUTMIXR_AXO2MIX_BIT, 1 , 1), +}; + +static const struct snd_kcontrol_new rt5631_spolmix_mixer_controls[] = { + SOC_DAPM_SINGLE("SPKVOLL Playback Switch", RT5631_SPK_MONO_OUT_CTRL, + RT5631_M_SPKVOLL_SPOLMIX_BIT, 1, 1), + SOC_DAPM_SINGLE("SPKVOLR Playback Switch", RT5631_SPK_MONO_OUT_CTRL, + RT5631_M_SPKVOLR_SPOLMIX_BIT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5631_spormix_mixer_controls[] = { + SOC_DAPM_SINGLE("SPKVOLL Playback Switch", RT5631_SPK_MONO_OUT_CTRL, + RT5631_M_SPKVOLL_SPORMIX_BIT, 1, 1), + SOC_DAPM_SINGLE("SPKVOLR Playback Switch", RT5631_SPK_MONO_OUT_CTRL, + RT5631_M_SPKVOLR_SPORMIX_BIT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5631_monomix_mixer_controls[] = { + SOC_DAPM_SINGLE("OUTVOLL Playback Switch", RT5631_SPK_MONO_OUT_CTRL, + RT5631_M_OUTVOLL_MONOMIX_BIT, 1, 1), + SOC_DAPM_SINGLE("OUTVOLR Playback Switch", RT5631_SPK_MONO_OUT_CTRL, + RT5631_M_OUTVOLR_MONOMIX_BIT, 1, 1), +}; + +/* Left SPK Volume Input */ +static const char *rt5631_spkvoll_sel[] = {"Vmid", "SPKMIXL"}; + +static const SOC_ENUM_SINGLE_DECL( + rt5631_spkvoll_enum, RT5631_SPK_OUT_VOL, + RT5631_L_EN_SHIFT, rt5631_spkvoll_sel); + +static const struct snd_kcontrol_new rt5631_spkvoll_mux_control = + SOC_DAPM_ENUM("Left SPKVOL SRC", rt5631_spkvoll_enum); + +/* Left HP Volume Input */ +static const char *rt5631_hpvoll_sel[] = {"Vmid", "OUTMIXL"}; + +static const SOC_ENUM_SINGLE_DECL( + rt5631_hpvoll_enum, RT5631_HP_OUT_VOL, + RT5631_L_EN_SHIFT, rt5631_hpvoll_sel); + +static const struct snd_kcontrol_new rt5631_hpvoll_mux_control = + SOC_DAPM_ENUM("Left HPVOL SRC", rt5631_hpvoll_enum); + +/* Left Out Volume Input */ +static const char *rt5631_outvoll_sel[] = {"Vmid", "OUTMIXL"}; + +static const SOC_ENUM_SINGLE_DECL( + rt5631_outvoll_enum, RT5631_MONO_AXO_1_2_VOL, + RT5631_L_EN_SHIFT, rt5631_outvoll_sel); + +static const struct snd_kcontrol_new rt5631_outvoll_mux_control = + SOC_DAPM_ENUM("Left OUTVOL SRC", rt5631_outvoll_enum); + +/* Right Out Volume Input */ +static const char *rt5631_outvolr_sel[] = {"Vmid", "OUTMIXR"}; + +static const SOC_ENUM_SINGLE_DECL( + rt5631_outvolr_enum, RT5631_MONO_AXO_1_2_VOL, + RT5631_R_EN_SHIFT, rt5631_outvolr_sel); + +static const struct snd_kcontrol_new rt5631_outvolr_mux_control = + SOC_DAPM_ENUM("Right OUTVOL SRC", rt5631_outvolr_enum); + +/* Right HP Volume Input */ +static const char *rt5631_hpvolr_sel[] = {"Vmid", "OUTMIXR"}; + +static const SOC_ENUM_SINGLE_DECL( + rt5631_hpvolr_enum, RT5631_HP_OUT_VOL, + RT5631_R_EN_SHIFT, rt5631_hpvolr_sel); + +static const struct snd_kcontrol_new rt5631_hpvolr_mux_control = + SOC_DAPM_ENUM("Right HPVOL SRC", rt5631_hpvolr_enum); + +/* Right SPK Volume Input */ +static const char *rt5631_spkvolr_sel[] = {"Vmid", "SPKMIXR"}; + +static const SOC_ENUM_SINGLE_DECL( + rt5631_spkvolr_enum, RT5631_SPK_OUT_VOL, + RT5631_R_EN_SHIFT, rt5631_spkvolr_sel); + +static const struct snd_kcontrol_new rt5631_spkvolr_mux_control = + SOC_DAPM_ENUM("Right SPKVOL SRC", rt5631_spkvolr_enum); + +/* SPO Left Channel Input */ +static const char *rt5631_spol_src_sel[] = { + "SPOLMIX", "MONOIN_RX", "VDAC", "DACL"}; + +static const SOC_ENUM_SINGLE_DECL( + rt5631_spol_src_enum, RT5631_SPK_MONO_HP_OUT_CTRL, + RT5631_SPK_L_MUX_SEL_SHIFT, rt5631_spol_src_sel); + +static const struct snd_kcontrol_new rt5631_spol_mux_control = + SOC_DAPM_ENUM("SPOL SRC", rt5631_spol_src_enum); + +/* SPO Right Channel Input */ +static const char *rt5631_spor_src_sel[] = { + "SPORMIX", "MONOIN_RX", "VDAC", "DACR"}; + +static const SOC_ENUM_SINGLE_DECL( + rt5631_spor_src_enum, RT5631_SPK_MONO_HP_OUT_CTRL, + RT5631_SPK_R_MUX_SEL_SHIFT, rt5631_spor_src_sel); + +static const struct snd_kcontrol_new rt5631_spor_mux_control = + SOC_DAPM_ENUM("SPOR SRC", rt5631_spor_src_enum); + +/* MONO Input */ +static const char *rt5631_mono_src_sel[] = {"MONOMIX", "MONOIN_RX", "VDAC"}; + +static const SOC_ENUM_SINGLE_DECL( + rt5631_mono_src_enum, RT5631_SPK_MONO_HP_OUT_CTRL, + RT5631_MONO_MUX_SEL_SHIFT, rt5631_mono_src_sel); + +static const struct snd_kcontrol_new rt5631_mono_mux_control = + SOC_DAPM_ENUM("MONO SRC", rt5631_mono_src_enum); + +/* Left HPO Input */ +static const char *rt5631_hpl_src_sel[] = {"Left HPVOL", "Left DAC"}; + +static const SOC_ENUM_SINGLE_DECL( + rt5631_hpl_src_enum, RT5631_SPK_MONO_HP_OUT_CTRL, + RT5631_HP_L_MUX_SEL_SHIFT, rt5631_hpl_src_sel); + +static const struct snd_kcontrol_new rt5631_hpl_mux_control = + SOC_DAPM_ENUM("HPL SRC", rt5631_hpl_src_enum); + +/* Right HPO Input */ +static const char *rt5631_hpr_src_sel[] = {"Right HPVOL", "Right DAC"}; + +static const SOC_ENUM_SINGLE_DECL( + rt5631_hpr_src_enum, RT5631_SPK_MONO_HP_OUT_CTRL, + RT5631_HP_R_MUX_SEL_SHIFT, rt5631_hpr_src_sel); + +static const struct snd_kcontrol_new rt5631_hpr_mux_control = + SOC_DAPM_ENUM("HPR SRC", rt5631_hpr_src_enum); + +static const struct snd_soc_dapm_widget rt5631_dapm_widgets[] = { + /* Vmid */ + SND_SOC_DAPM_VMID("Vmid"), + /* PLL1 */ + SND_SOC_DAPM_SUPPLY("PLL1", RT5631_PWR_MANAG_ADD2, + RT5631_PWR_PLL1_BIT, 0, NULL, 0), + + /* Input Side */ + /* Input Lines */ + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_INPUT("MIC2"), + SND_SOC_DAPM_INPUT("AXIL"), + SND_SOC_DAPM_INPUT("AXIR"), + SND_SOC_DAPM_INPUT("MONOIN_RXN"), + SND_SOC_DAPM_INPUT("MONOIN_RXP"), + SND_SOC_DAPM_INPUT("DMIC"), + + /* MICBIAS */ + SND_SOC_DAPM_MICBIAS("MIC Bias1", RT5631_PWR_MANAG_ADD2, + RT5631_PWR_MICBIAS1_VOL_BIT, 0), + SND_SOC_DAPM_MICBIAS("MIC Bias2", RT5631_PWR_MANAG_ADD2, + RT5631_PWR_MICBIAS2_VOL_BIT, 0), + + /* Boost */ + SND_SOC_DAPM_PGA("MIC1 Boost", RT5631_PWR_MANAG_ADD2, + RT5631_PWR_MIC1_BOOT_GAIN_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("MIC2 Boost", RT5631_PWR_MANAG_ADD2, + RT5631_PWR_MIC2_BOOT_GAIN_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("MONOIN_RXP Boost", RT5631_PWR_MANAG_ADD4, + RT5631_PWR_MONO_IN_P_VOL_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("MONOIN_RXN Boost", RT5631_PWR_MANAG_ADD4, + RT5631_PWR_MONO_IN_N_VOL_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("AXIL Boost", RT5631_PWR_MANAG_ADD4, + RT5631_PWR_AXIL_IN_VOL_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("AXIR Boost", RT5631_PWR_MANAG_ADD4, + RT5631_PWR_AXIR_IN_VOL_BIT, 0, NULL, 0), + + /* MONO In */ + SND_SOC_DAPM_MIXER("MONO_IN", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* REC Mixer */ + SND_SOC_DAPM_MIXER("RECMIXL Mixer", RT5631_PWR_MANAG_ADD2, + RT5631_PWR_RECMIXER_L_BIT, 0, + &rt5631_recmixl_mixer_controls[0], + ARRAY_SIZE(rt5631_recmixl_mixer_controls)), + SND_SOC_DAPM_MIXER("RECMIXR Mixer", RT5631_PWR_MANAG_ADD2, + RT5631_PWR_RECMIXER_R_BIT, 0, + &rt5631_recmixr_mixer_controls[0], + ARRAY_SIZE(rt5631_recmixr_mixer_controls)), + /* Because of record duplication for L/R channel, + * L/R ADCs need power up at the same time */ + SND_SOC_DAPM_MIXER("ADC Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* DMIC */ + SND_SOC_DAPM_SUPPLY("DMIC Supply", RT5631_DIG_MIC_CTRL, + RT5631_DMIC_ENA_SHIFT, 0, + set_dmic_params, SND_SOC_DAPM_PRE_PMU), + /* ADC Data Srouce */ + SND_SOC_DAPM_SUPPLY("Left ADC Select", RT5631_INT_ST_IRQ_CTRL_2, + RT5631_ADC_DATA_SEL_MIC1_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Right ADC Select", RT5631_INT_ST_IRQ_CTRL_2, + RT5631_ADC_DATA_SEL_MIC2_SHIFT, 0, NULL, 0), + + /* ADCs */ + SND_SOC_DAPM_ADC("Left ADC", "HIFI Capture", + RT5631_PWR_MANAG_ADD1, RT5631_PWR_ADC_L_CLK_BIT, 0), + SND_SOC_DAPM_ADC("Right ADC", "HIFI Capture", + RT5631_PWR_MANAG_ADD1, RT5631_PWR_ADC_R_CLK_BIT, 0), + + /* DAC and ADC supply power */ + SND_SOC_DAPM_SUPPLY("I2S", RT5631_PWR_MANAG_ADD1, + RT5631_PWR_MAIN_I2S_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC REF", RT5631_PWR_MANAG_ADD1, + RT5631_PWR_DAC_REF_BIT, 0, NULL, 0), + + /* Output Side */ + /* DACs */ + SND_SOC_DAPM_DAC("Left DAC", "HIFI Playback", + RT5631_PWR_MANAG_ADD1, RT5631_PWR_DAC_L_CLK_BIT, 0), + SND_SOC_DAPM_DAC("Right DAC", "HIFI Playback", + RT5631_PWR_MANAG_ADD1, RT5631_PWR_DAC_R_CLK_BIT, 0), + SND_SOC_DAPM_DAC("Voice DAC", "Voice DAC Mono Playback", + SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_PGA("Voice DAC Boost", SND_SOC_NOPM, 0, 0, NULL, 0), + /* DAC supply power */ + SND_SOC_DAPM_SUPPLY("Left DAC To Mixer", RT5631_PWR_MANAG_ADD1, + RT5631_PWR_DAC_L_TO_MIXER_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Right DAC To Mixer", RT5631_PWR_MANAG_ADD1, + RT5631_PWR_DAC_R_TO_MIXER_BIT, 0, NULL, 0), + + /* Left SPK Mixer */ + SND_SOC_DAPM_MIXER("SPKMIXL Mixer", RT5631_PWR_MANAG_ADD2, + RT5631_PWR_SPKMIXER_L_BIT, 0, + &rt5631_spkmixl_mixer_controls[0], + ARRAY_SIZE(rt5631_spkmixl_mixer_controls)), + /* Left Out Mixer */ + SND_SOC_DAPM_MIXER("OUTMIXL Mixer", RT5631_PWR_MANAG_ADD2, + RT5631_PWR_OUTMIXER_L_BIT, 0, + &rt5631_outmixl_mixer_controls[0], + ARRAY_SIZE(rt5631_outmixl_mixer_controls)), + /* Right Out Mixer */ + SND_SOC_DAPM_MIXER("OUTMIXR Mixer", RT5631_PWR_MANAG_ADD2, + RT5631_PWR_OUTMIXER_R_BIT, 0, + &rt5631_outmixr_mixer_controls[0], + ARRAY_SIZE(rt5631_outmixr_mixer_controls)), + /* Right SPK Mixer */ + SND_SOC_DAPM_MIXER("SPKMIXR Mixer", RT5631_PWR_MANAG_ADD2, + RT5631_PWR_SPKMIXER_R_BIT, 0, + &rt5631_spkmixr_mixer_controls[0], + ARRAY_SIZE(rt5631_spkmixr_mixer_controls)), + + /* Volume Mux */ + SND_SOC_DAPM_MUX("Left SPKVOL Mux", RT5631_PWR_MANAG_ADD4, + RT5631_PWR_SPK_L_VOL_BIT, 0, + &rt5631_spkvoll_mux_control), + SND_SOC_DAPM_MUX("Left HPVOL Mux", RT5631_PWR_MANAG_ADD4, + RT5631_PWR_HP_L_OUT_VOL_BIT, 0, + &rt5631_hpvoll_mux_control), + SND_SOC_DAPM_MUX("Left OUTVOL Mux", RT5631_PWR_MANAG_ADD4, + RT5631_PWR_LOUT_VOL_BIT, 0, + &rt5631_outvoll_mux_control), + SND_SOC_DAPM_MUX("Right OUTVOL Mux", RT5631_PWR_MANAG_ADD4, + RT5631_PWR_ROUT_VOL_BIT, 0, + &rt5631_outvolr_mux_control), + SND_SOC_DAPM_MUX("Right HPVOL Mux", RT5631_PWR_MANAG_ADD4, + RT5631_PWR_HP_R_OUT_VOL_BIT, 0, + &rt5631_hpvolr_mux_control), + SND_SOC_DAPM_MUX("Right SPKVOL Mux", RT5631_PWR_MANAG_ADD4, + RT5631_PWR_SPK_R_VOL_BIT, 0, + &rt5631_spkvolr_mux_control), + + /* DAC To HP */ + SND_SOC_DAPM_PGA_S("Left DAC_HP", 0, SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA_S("Right DAC_HP", 0, SND_SOC_NOPM, 0, 0, NULL, 0), + + /* HP Depop */ + SND_SOC_DAPM_PGA_S("HP Depop", 1, SND_SOC_NOPM, 0, 0, + hp_event, SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + + /* AXO1 Mixer */ + SND_SOC_DAPM_MIXER("AXO1MIX Mixer", RT5631_PWR_MANAG_ADD3, + RT5631_PWR_AXO1MIXER_BIT, 0, + &rt5631_AXO1MIX_mixer_controls[0], + ARRAY_SIZE(rt5631_AXO1MIX_mixer_controls)), + /* SPOL Mixer */ + SND_SOC_DAPM_MIXER("SPOLMIX Mixer", SND_SOC_NOPM, 0, 0, + &rt5631_spolmix_mixer_controls[0], + ARRAY_SIZE(rt5631_spolmix_mixer_controls)), + /* MONO Mixer */ + SND_SOC_DAPM_MIXER("MONOMIX Mixer", RT5631_PWR_MANAG_ADD3, + RT5631_PWR_MONOMIXER_BIT, 0, + &rt5631_monomix_mixer_controls[0], + ARRAY_SIZE(rt5631_monomix_mixer_controls)), + /* SPOR Mixer */ + SND_SOC_DAPM_MIXER("SPORMIX Mixer", SND_SOC_NOPM, 0, 0, + &rt5631_spormix_mixer_controls[0], + ARRAY_SIZE(rt5631_spormix_mixer_controls)), + /* AXO2 Mixer */ + SND_SOC_DAPM_MIXER("AXO2MIX Mixer", RT5631_PWR_MANAG_ADD3, + RT5631_PWR_AXO2MIXER_BIT, 0, + &rt5631_AXO2MIX_mixer_controls[0], + ARRAY_SIZE(rt5631_AXO2MIX_mixer_controls)), + + /* Mux */ + SND_SOC_DAPM_MUX("SPOL Mux", SND_SOC_NOPM, 0, 0, + &rt5631_spol_mux_control), + SND_SOC_DAPM_MUX("SPOR Mux", SND_SOC_NOPM, 0, 0, + &rt5631_spor_mux_control), + SND_SOC_DAPM_MUX("MONO Mux", SND_SOC_NOPM, 0, 0, + &rt5631_mono_mux_control), + SND_SOC_DAPM_MUX("HPL Mux", SND_SOC_NOPM, 0, 0, + &rt5631_hpl_mux_control), + SND_SOC_DAPM_MUX("HPR Mux", SND_SOC_NOPM, 0, 0, + &rt5631_hpr_mux_control), + + /* AMP supply */ + SND_SOC_DAPM_SUPPLY("MONO Depop", RT5631_PWR_MANAG_ADD3, + RT5631_PWR_MONO_DEPOP_DIS_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Class D", RT5631_PWR_MANAG_ADD1, + RT5631_PWR_CLASS_D_BIT, 0, NULL, 0), + + /* Output Lines */ + SND_SOC_DAPM_OUTPUT("AUXO1"), + SND_SOC_DAPM_OUTPUT("AUXO2"), + SND_SOC_DAPM_OUTPUT("SPOL"), + SND_SOC_DAPM_OUTPUT("SPOR"), + SND_SOC_DAPM_OUTPUT("HPOL"), + SND_SOC_DAPM_OUTPUT("HPOR"), + SND_SOC_DAPM_OUTPUT("MONO"), +}; + +static const struct snd_soc_dapm_route rt5631_dapm_routes[] = { + {"MIC1 Boost", NULL, "MIC1"}, + {"MIC2 Boost", NULL, "MIC2"}, + {"MONOIN_RXP Boost", NULL, "MONOIN_RXP"}, + {"MONOIN_RXN Boost", NULL, "MONOIN_RXN"}, + {"AXIL Boost", NULL, "AXIL"}, + {"AXIR Boost", NULL, "AXIR"}, + + {"MONO_IN", NULL, "MONOIN_RXP Boost"}, + {"MONO_IN", NULL, "MONOIN_RXN Boost"}, + + {"RECMIXL Mixer", "OUTMIXL Capture Switch", "OUTMIXL Mixer"}, + {"RECMIXL Mixer", "MIC1_BST1 Capture Switch", "MIC1 Boost"}, + {"RECMIXL Mixer", "AXILVOL Capture Switch", "AXIL Boost"}, + {"RECMIXL Mixer", "MONOIN_RX Capture Switch", "MONO_IN"}, + + {"RECMIXR Mixer", "OUTMIXR Capture Switch", "OUTMIXR Mixer"}, + {"RECMIXR Mixer", "MIC2_BST2 Capture Switch", "MIC2 Boost"}, + {"RECMIXR Mixer", "AXIRVOL Capture Switch", "AXIR Boost"}, + {"RECMIXR Mixer", "MONOIN_RX Capture Switch", "MONO_IN"}, + + {"ADC Mixer", NULL, "RECMIXL Mixer"}, + {"ADC Mixer", NULL, "RECMIXR Mixer"}, + + {"Left ADC", NULL, "ADC Mixer"}, + {"Left ADC", NULL, "Left ADC Select", check_adcl_select}, + {"Left ADC", NULL, "PLL1", check_sysclk1_source}, + {"Left ADC", NULL, "I2S"}, + {"Left ADC", NULL, "DAC REF"}, + + {"Right ADC", NULL, "ADC Mixer"}, + {"Right ADC", NULL, "Right ADC Select", check_adcr_select}, + {"Right ADC", NULL, "PLL1", check_sysclk1_source}, + {"Right ADC", NULL, "I2S"}, + {"Right ADC", NULL, "DAC REF"}, + + {"DMIC", NULL, "DMIC Supply", check_dmic_used}, + {"Left ADC", NULL, "DMIC"}, + {"Right ADC", NULL, "DMIC"}, + + {"Left DAC", NULL, "PLL1", check_sysclk1_source}, + {"Left DAC", NULL, "I2S"}, + {"Left DAC", NULL, "DAC REF"}, + {"Right DAC", NULL, "PLL1", check_sysclk1_source}, + {"Right DAC", NULL, "I2S"}, + {"Right DAC", NULL, "DAC REF"}, + + {"Voice DAC Boost", NULL, "Voice DAC"}, + + {"SPKMIXL Mixer", NULL, "Left DAC To Mixer", check_dacl_to_spkmixl}, + {"SPKMIXL Mixer", "RECMIXL Playback Switch", "RECMIXL Mixer"}, + {"SPKMIXL Mixer", "MIC1_P Playback Switch", "MIC1"}, + {"SPKMIXL Mixer", "DACL Playback Switch", "Left DAC"}, + {"SPKMIXL Mixer", "OUTMIXL Playback Switch", "OUTMIXL Mixer"}, + + {"SPKMIXR Mixer", NULL, "Right DAC To Mixer", check_dacr_to_spkmixr}, + {"SPKMIXR Mixer", "OUTMIXR Playback Switch", "OUTMIXR Mixer"}, + {"SPKMIXR Mixer", "DACR Playback Switch", "Right DAC"}, + {"SPKMIXR Mixer", "MIC2_P Playback Switch", "MIC2"}, + {"SPKMIXR Mixer", "RECMIXR Playback Switch", "RECMIXR Mixer"}, + + {"OUTMIXL Mixer", NULL, "Left DAC To Mixer", check_dacl_to_outmixl}, + {"OUTMIXL Mixer", "RECMIXL Playback Switch", "RECMIXL Mixer"}, + {"OUTMIXL Mixer", "RECMIXR Playback Switch", "RECMIXR Mixer"}, + {"OUTMIXL Mixer", "DACL Playback Switch", "Left DAC"}, + {"OUTMIXL Mixer", "MIC1_BST1 Playback Switch", "MIC1 Boost"}, + {"OUTMIXL Mixer", "MIC2_BST2 Playback Switch", "MIC2 Boost"}, + {"OUTMIXL Mixer", "MONOIN_RXP Playback Switch", "MONOIN_RXP Boost"}, + {"OUTMIXL Mixer", "AXILVOL Playback Switch", "AXIL Boost"}, + {"OUTMIXL Mixer", "AXIRVOL Playback Switch", "AXIR Boost"}, + {"OUTMIXL Mixer", "VDAC Playback Switch", "Voice DAC Boost"}, + + {"OUTMIXR Mixer", NULL, "Right DAC To Mixer", check_dacr_to_outmixr}, + {"OUTMIXR Mixer", "RECMIXL Playback Switch", "RECMIXL Mixer"}, + {"OUTMIXR Mixer", "RECMIXR Playback Switch", "RECMIXR Mixer"}, + {"OUTMIXR Mixer", "DACR Playback Switch", "Right DAC"}, + {"OUTMIXR Mixer", "MIC1_BST1 Playback Switch", "MIC1 Boost"}, + {"OUTMIXR Mixer", "MIC2_BST2 Playback Switch", "MIC2 Boost"}, + {"OUTMIXR Mixer", "MONOIN_RXN Playback Switch", "MONOIN_RXN Boost"}, + {"OUTMIXR Mixer", "AXILVOL Playback Switch", "AXIL Boost"}, + {"OUTMIXR Mixer", "AXIRVOL Playback Switch", "AXIR Boost"}, + {"OUTMIXR Mixer", "VDAC Playback Switch", "Voice DAC Boost"}, + + {"Left SPKVOL Mux", "SPKMIXL", "SPKMIXL Mixer"}, + {"Left SPKVOL Mux", "Vmid", "Vmid"}, + {"Left HPVOL Mux", "OUTMIXL", "OUTMIXL Mixer"}, + {"Left HPVOL Mux", "Vmid", "Vmid"}, + {"Left OUTVOL Mux", "OUTMIXL", "OUTMIXL Mixer"}, + {"Left OUTVOL Mux", "Vmid", "Vmid"}, + {"Right OUTVOL Mux", "OUTMIXR", "OUTMIXR Mixer"}, + {"Right OUTVOL Mux", "Vmid", "Vmid"}, + {"Right HPVOL Mux", "OUTMIXR", "OUTMIXR Mixer"}, + {"Right HPVOL Mux", "Vmid", "Vmid"}, + {"Right SPKVOL Mux", "SPKMIXR", "SPKMIXR Mixer"}, + {"Right SPKVOL Mux", "Vmid", "Vmid"}, + + {"AXO1MIX Mixer", "MIC1_BST1 Playback Switch", "MIC1 Boost"}, + {"AXO1MIX Mixer", "OUTVOLL Playback Switch", "Left OUTVOL Mux"}, + {"AXO1MIX Mixer", "OUTVOLR Playback Switch", "Right OUTVOL Mux"}, + {"AXO1MIX Mixer", "MIC2_BST2 Playback Switch", "MIC2 Boost"}, + + {"AXO2MIX Mixer", "MIC1_BST1 Playback Switch", "MIC1 Boost"}, + {"AXO2MIX Mixer", "OUTVOLL Playback Switch", "Left OUTVOL Mux"}, + {"AXO2MIX Mixer", "OUTVOLR Playback Switch", "Right OUTVOL Mux"}, + {"AXO2MIX Mixer", "MIC2_BST2 Playback Switch", "MIC2 Boost"}, + + {"SPOLMIX Mixer", "SPKVOLL Playback Switch", "Left SPKVOL Mux"}, + {"SPOLMIX Mixer", "SPKVOLR Playback Switch", "Right SPKVOL Mux"}, + + {"SPORMIX Mixer", "SPKVOLL Playback Switch", "Left SPKVOL Mux"}, + {"SPORMIX Mixer", "SPKVOLR Playback Switch", "Right SPKVOL Mux"}, + + {"MONOMIX Mixer", "OUTVOLL Playback Switch", "Left OUTVOL Mux"}, + {"MONOMIX Mixer", "OUTVOLR Playback Switch", "Right OUTVOL Mux"}, + + {"SPOL Mux", "SPOLMIX", "SPOLMIX Mixer"}, + {"SPOL Mux", "MONOIN_RX", "MONO_IN"}, + {"SPOL Mux", "VDAC", "Voice DAC Boost"}, + {"SPOL Mux", "DACL", "Left DAC"}, + + {"SPOR Mux", "SPORMIX", "SPORMIX Mixer"}, + {"SPOR Mux", "MONOIN_RX", "MONO_IN"}, + {"SPOR Mux", "VDAC", "Voice DAC Boost"}, + {"SPOR Mux", "DACR", "Right DAC"}, + + {"MONO Mux", "MONOMIX", "MONOMIX Mixer"}, + {"MONO Mux", "MONOIN_RX", "MONO_IN"}, + {"MONO Mux", "VDAC", "Voice DAC Boost"}, + + {"Right DAC_HP", NULL, "Right DAC"}, + {"Left DAC_HP", NULL, "Left DAC"}, + + {"HPL Mux", "Left HPVOL", "Left HPVOL Mux"}, + {"HPL Mux", "Left DAC", "Left DAC_HP"}, + {"HPR Mux", "Right HPVOL", "Right HPVOL Mux"}, + {"HPR Mux", "Right DAC", "Right DAC_HP"}, + + {"HP Depop", NULL, "HPL Mux"}, + {"HP Depop", NULL, "HPR Mux"}, + + {"AUXO1", NULL, "AXO1MIX Mixer"}, + {"AUXO2", NULL, "AXO2MIX Mixer"}, + + {"SPOL", NULL, "Class D"}, + {"SPOL", NULL, "SPOL Mux"}, + {"SPOR", NULL, "Class D"}, + {"SPOR", NULL, "SPOR Mux"}, + + {"HPOL", NULL, "HP Depop"}, + {"HPOR", NULL, "HP Depop"}, + + {"MONO", NULL, "MONO Depop"}, + {"MONO", NULL, "MONO Mux"}, +}; + +struct coeff_clk_div { + u32 mclk; + u32 bclk; + u32 rate; + u16 reg_val; +}; + +/* PLL divisors */ +struct pll_div { + u32 pll_in; + u32 pll_out; + u16 reg_val; +}; + +static const struct pll_div codec_master_pll_div[] = { + {2048000, 8192000, 0x0ea0}, + {3686400, 8192000, 0x4e27}, + {12000000, 8192000, 0x456b}, + {13000000, 8192000, 0x495f}, + {13100000, 8192000, 0x0320}, + {2048000, 11289600, 0xf637}, + {3686400, 11289600, 0x2f22}, + {12000000, 11289600, 0x3e2f}, + {13000000, 11289600, 0x4d5b}, + {13100000, 11289600, 0x363b}, + {2048000, 16384000, 0x1ea0}, + {3686400, 16384000, 0x9e27}, + {12000000, 16384000, 0x452b}, + {13000000, 16384000, 0x542f}, + {13100000, 16384000, 0x03a0}, + {2048000, 16934400, 0xe625}, + {3686400, 16934400, 0x9126}, + {12000000, 16934400, 0x4d2c}, + {13000000, 16934400, 0x742f}, + {13100000, 16934400, 0x3c27}, + {2048000, 22579200, 0x2aa0}, + {3686400, 22579200, 0x2f20}, + {12000000, 22579200, 0x7e2f}, + {13000000, 22579200, 0x742f}, + {13100000, 22579200, 0x3c27}, + {2048000, 24576000, 0x2ea0}, + {3686400, 24576000, 0xee27}, + {12000000, 24576000, 0x2915}, + {13000000, 24576000, 0x772e}, + {13100000, 24576000, 0x0d20}, + {26000000, 24576000, 0x2027}, + {26000000, 22579200, 0x392f}, + {24576000, 22579200, 0x0921}, + {24576000, 24576000, 0x02a0}, +}; + +static const struct pll_div codec_slave_pll_div[] = { + {256000, 2048000, 0x46f0}, + {256000, 4096000, 0x3ea0}, + {352800, 5644800, 0x3ea0}, + {512000, 8192000, 0x3ea0}, + {1024000, 8192000, 0x46f0}, + {705600, 11289600, 0x3ea0}, + {1024000, 16384000, 0x3ea0}, + {1411200, 22579200, 0x3ea0}, + {1536000, 24576000, 0x3ea0}, + {2048000, 16384000, 0x1ea0}, + {2822400, 22579200, 0x1ea0}, + {2822400, 45158400, 0x5ec0}, + {5644800, 45158400, 0x46f0}, + {3072000, 24576000, 0x1ea0}, + {3072000, 49152000, 0x5ec0}, + {6144000, 49152000, 0x46f0}, + {705600, 11289600, 0x3ea0}, + {705600, 8467200, 0x3ab0}, + {24576000, 24576000, 0x02a0}, + {1411200, 11289600, 0x1690}, + {2822400, 11289600, 0x0a90}, + {1536000, 12288000, 0x1690}, + {3072000, 12288000, 0x0a90}, +}; + +static struct coeff_clk_div coeff_div[] = { + /* sysclk is 256fs */ + {2048000, 8000 * 32, 8000, 0x1000}, + {2048000, 8000 * 64, 8000, 0x0000}, + {2822400, 11025 * 32, 11025, 0x1000}, + {2822400, 11025 * 64, 11025, 0x0000}, + {4096000, 16000 * 32, 16000, 0x1000}, + {4096000, 16000 * 64, 16000, 0x0000}, + {5644800, 22050 * 32, 22050, 0x1000}, + {5644800, 22050 * 64, 22050, 0x0000}, + {8192000, 32000 * 32, 32000, 0x1000}, + {8192000, 32000 * 64, 32000, 0x0000}, + {11289600, 44100 * 32, 44100, 0x1000}, + {11289600, 44100 * 64, 44100, 0x0000}, + {12288000, 48000 * 32, 48000, 0x1000}, + {12288000, 48000 * 64, 48000, 0x0000}, + {22579200, 88200 * 32, 88200, 0x1000}, + {22579200, 88200 * 64, 88200, 0x0000}, + {24576000, 96000 * 32, 96000, 0x1000}, + {24576000, 96000 * 64, 96000, 0x0000}, + /* sysclk is 512fs */ + {4096000, 8000 * 32, 8000, 0x3000}, + {4096000, 8000 * 64, 8000, 0x2000}, + {5644800, 11025 * 32, 11025, 0x3000}, + {5644800, 11025 * 64, 11025, 0x2000}, + {8192000, 16000 * 32, 16000, 0x3000}, + {8192000, 16000 * 64, 16000, 0x2000}, + {11289600, 22050 * 32, 22050, 0x3000}, + {11289600, 22050 * 64, 22050, 0x2000}, + {16384000, 32000 * 32, 32000, 0x3000}, + {16384000, 32000 * 64, 32000, 0x2000}, + {22579200, 44100 * 32, 44100, 0x3000}, + {22579200, 44100 * 64, 44100, 0x2000}, + {24576000, 48000 * 32, 48000, 0x3000}, + {24576000, 48000 * 64, 48000, 0x2000}, + {45158400, 88200 * 32, 88200, 0x3000}, + {45158400, 88200 * 64, 88200, 0x2000}, + {49152000, 96000 * 32, 96000, 0x3000}, + {49152000, 96000 * 64, 96000, 0x2000}, + /* sysclk is 24.576Mhz or 22.5792Mhz */ + {24576000, 8000 * 32, 8000, 0x7080}, + {24576000, 8000 * 64, 8000, 0x6080}, + {24576000, 16000 * 32, 16000, 0x5080}, + {24576000, 16000 * 64, 16000, 0x4080}, + {24576000, 24000 * 32, 24000, 0x5000}, + {24576000, 24000 * 64, 24000, 0x4000}, + {24576000, 32000 * 32, 32000, 0x3080}, + {24576000, 32000 * 64, 32000, 0x2080}, + {22579200, 11025 * 32, 11025, 0x7000}, + {22579200, 11025 * 64, 11025, 0x6000}, + {22579200, 22050 * 32, 22050, 0x5000}, + {22579200, 22050 * 64, 22050, 0x4000}, +}; + +static int get_coeff(int mclk, int rate, int timesofbclk) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(coeff_div); i++) { + if (coeff_div[i].mclk == mclk && coeff_div[i].rate == rate && + (coeff_div[i].bclk / coeff_div[i].rate) == timesofbclk) + return i; + } + return -EINVAL; +} + +static int rt5631_hifi_pcm_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + struct rt5631_priv *rt5631 = snd_soc_codec_get_drvdata(codec); + int timesofbclk = 32, coeff; + unsigned int iface = 0; + + dev_dbg(codec->dev, "enter %s\n", __func__); + + rt5631->bclk_rate = snd_soc_params_to_bclk(params); + if (rt5631->bclk_rate < 0) { + dev_err(codec->dev, "Fail to get BCLK rate\n"); + return rt5631->bclk_rate; + } + rt5631->rx_rate = params_rate(params); + + if (rt5631->master) + coeff = get_coeff(rt5631->sysclk, rt5631->rx_rate, + rt5631->bclk_rate / rt5631->rx_rate); + else + coeff = get_coeff(rt5631->sysclk, rt5631->rx_rate, + timesofbclk); + if (coeff < 0) { + dev_err(codec->dev, "Fail to get coeff\n"); + return -EINVAL; + } + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + iface |= RT5631_SDP_I2S_DL_20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + iface |= RT5631_SDP_I2S_DL_24; + break; + case SNDRV_PCM_FORMAT_S8: + iface |= RT5631_SDP_I2S_DL_8; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, RT5631_SDP_CTRL, + RT5631_SDP_I2S_DL_MASK, iface); + snd_soc_write(codec, RT5631_STEREO_AD_DA_CLK_CTRL, + coeff_div[coeff].reg_val); + + return 0; +} + +static int rt5631_hifi_codec_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct rt5631_priv *rt5631 = snd_soc_codec_get_drvdata(codec); + unsigned int iface = 0; + + dev_dbg(codec->dev, "enter %s\n", __func__); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + rt5631->master = 1; + break; + case SND_SOC_DAIFMT_CBS_CFS: + iface |= RT5631_SDP_MODE_SEL_SLAVE; + rt5631->master = 0; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + case SND_SOC_DAIFMT_LEFT_J: + iface |= RT5631_SDP_I2S_DF_LEFT; + break; + case SND_SOC_DAIFMT_DSP_A: + iface |= RT5631_SDP_I2S_DF_PCM_A; + break; + case SND_SOC_DAIFMT_DSP_B: + iface |= RT5631_SDP_I2S_DF_PCM_B; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + iface |= RT5631_SDP_I2S_BCLK_POL_CTRL; + break; + default: + return -EINVAL; + } + + snd_soc_write(codec, RT5631_SDP_CTRL, iface); + + return 0; +} + +static int rt5631_hifi_codec_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct rt5631_priv *rt5631 = snd_soc_codec_get_drvdata(codec); + + dev_dbg(codec->dev, "enter %s, syclk=%d\n", __func__, freq); + + if ((freq >= (256 * 8000)) && (freq <= (512 * 96000))) { + rt5631->sysclk = freq; + return 0; + } + + return -EINVAL; +} + +static int rt5631_codec_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, + int source, unsigned int freq_in, unsigned int freq_out) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct rt5631_priv *rt5631 = snd_soc_codec_get_drvdata(codec); + int i, ret = -EINVAL; + + dev_dbg(codec->dev, "enter %s\n", __func__); + + if (!freq_in || !freq_out) { + dev_dbg(codec->dev, "PLL disabled\n"); + + snd_soc_update_bits(codec, RT5631_GLOBAL_CLK_CTRL, + RT5631_SYSCLK_SOUR_SEL_MASK, + RT5631_SYSCLK_SOUR_SEL_MCLK); + + return 0; + } + + if (rt5631->master) { + for (i = 0; i < ARRAY_SIZE(codec_master_pll_div); i++) + if (freq_in == codec_master_pll_div[i].pll_in && + freq_out == codec_master_pll_div[i].pll_out) { + dev_info(codec->dev, + "change PLL in master mode\n"); + snd_soc_write(codec, RT5631_PLL_CTRL, + codec_master_pll_div[i].reg_val); + schedule_timeout_uninterruptible( + msecs_to_jiffies(20)); + snd_soc_update_bits(codec, + RT5631_GLOBAL_CLK_CTRL, + RT5631_SYSCLK_SOUR_SEL_MASK | + RT5631_PLLCLK_SOUR_SEL_MASK, + RT5631_SYSCLK_SOUR_SEL_PLL | + RT5631_PLLCLK_SOUR_SEL_MCLK); + ret = 0; + break; + } + } else { + for (i = 0; i < ARRAY_SIZE(codec_slave_pll_div); i++) + if (freq_in == codec_slave_pll_div[i].pll_in && + freq_out == codec_slave_pll_div[i].pll_out) { + dev_info(codec->dev, + "change PLL in slave mode\n"); + snd_soc_write(codec, RT5631_PLL_CTRL, + codec_slave_pll_div[i].reg_val); + schedule_timeout_uninterruptible( + msecs_to_jiffies(20)); + snd_soc_update_bits(codec, + RT5631_GLOBAL_CLK_CTRL, + RT5631_SYSCLK_SOUR_SEL_MASK | + RT5631_PLLCLK_SOUR_SEL_MASK, + RT5631_SYSCLK_SOUR_SEL_PLL | + RT5631_PLLCLK_SOUR_SEL_BCLK); + ret = 0; + break; + } + } + + return ret; +} + +static int rt5631_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + snd_soc_update_bits(codec, RT5631_PWR_MANAG_ADD2, + RT5631_PWR_MICBIAS1_VOL | RT5631_PWR_MICBIAS2_VOL, + RT5631_PWR_MICBIAS1_VOL | RT5631_PWR_MICBIAS2_VOL); + break; + + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + snd_soc_update_bits(codec, RT5631_PWR_MANAG_ADD3, + RT5631_PWR_VREF | RT5631_PWR_MAIN_BIAS, + RT5631_PWR_VREF | RT5631_PWR_MAIN_BIAS); + msleep(80); + snd_soc_update_bits(codec, RT5631_PWR_MANAG_ADD3, + RT5631_PWR_FAST_VREF_CTRL, + RT5631_PWR_FAST_VREF_CTRL); + codec->cache_only = false; + snd_soc_cache_sync(codec); + } + break; + + case SND_SOC_BIAS_OFF: + snd_soc_write(codec, RT5631_PWR_MANAG_ADD1, 0x0000); + snd_soc_write(codec, RT5631_PWR_MANAG_ADD2, 0x0000); + snd_soc_write(codec, RT5631_PWR_MANAG_ADD3, 0x0000); + snd_soc_write(codec, RT5631_PWR_MANAG_ADD4, 0x0000); + break; + + default: + break; + } + codec->dapm.bias_level = level; + + return 0; +} + +static int rt5631_probe(struct snd_soc_codec *codec) +{ + struct rt5631_priv *rt5631 = snd_soc_codec_get_drvdata(codec); + unsigned int val; + int ret; + + ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_I2C); + if (ret != 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + val = rt5631_read_index(codec, RT5631_ADDA_MIXER_INTL_REG3); + if (val & 0x0002) + rt5631->codec_version = 1; + else + rt5631->codec_version = 0; + + rt5631_reset(codec); + snd_soc_update_bits(codec, RT5631_PWR_MANAG_ADD3, + RT5631_PWR_VREF | RT5631_PWR_MAIN_BIAS, + RT5631_PWR_VREF | RT5631_PWR_MAIN_BIAS); + msleep(80); + snd_soc_update_bits(codec, RT5631_PWR_MANAG_ADD3, + RT5631_PWR_FAST_VREF_CTRL, RT5631_PWR_FAST_VREF_CTRL); + /* enable HP zero cross */ + snd_soc_write(codec, RT5631_INT_ST_IRQ_CTRL_2, 0x0f18); + /* power off ClassD auto Recovery */ + if (rt5631->codec_version) + snd_soc_update_bits(codec, RT5631_INT_ST_IRQ_CTRL_2, + 0x2000, 0x2000); + else + snd_soc_update_bits(codec, RT5631_INT_ST_IRQ_CTRL_2, + 0x2000, 0); + /* DMIC */ + if (rt5631->dmic_used_flag) { + snd_soc_update_bits(codec, RT5631_GPIO_CTRL, + RT5631_GPIO_PIN_FUN_SEL_MASK | + RT5631_GPIO_DMIC_FUN_SEL_MASK, + RT5631_GPIO_PIN_FUN_SEL_GPIO_DIMC | + RT5631_GPIO_DMIC_FUN_SEL_DIMC); + snd_soc_update_bits(codec, RT5631_DIG_MIC_CTRL, + RT5631_DMIC_L_CH_LATCH_MASK | + RT5631_DMIC_R_CH_LATCH_MASK, + RT5631_DMIC_L_CH_LATCH_FALLING | + RT5631_DMIC_R_CH_LATCH_RISING); + } + + codec->dapm.bias_level = SND_SOC_BIAS_STANDBY; + + return 0; +} + +static int rt5631_remove(struct snd_soc_codec *codec) +{ + rt5631_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +#ifdef CONFIG_PM +static int rt5631_suspend(struct snd_soc_codec *codec, pm_message_t state) +{ + rt5631_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int rt5631_resume(struct snd_soc_codec *codec) +{ + rt5631_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + return 0; +} +#else +#define rt5631_suspend NULL +#define rt5631_resume NULL +#endif + +#define RT5631_STEREO_RATES SNDRV_PCM_RATE_8000_96000 +#define RT5631_FORMAT (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S8) + +static struct snd_soc_dai_ops rt5631_ops = { + .hw_params = rt5631_hifi_pcm_params, + .set_fmt = rt5631_hifi_codec_set_dai_fmt, + .set_sysclk = rt5631_hifi_codec_set_dai_sysclk, + .set_pll = rt5631_codec_set_dai_pll, +}; + +static struct snd_soc_dai_driver rt5631_dai[] = { + { + .name = "rt5631-hifi", + .id = 1, + .playback = { + .stream_name = "HIFI Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT5631_STEREO_RATES, + .formats = RT5631_FORMAT, + }, + .capture = { + .stream_name = "HIFI Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT5631_STEREO_RATES, + .formats = RT5631_FORMAT, + }, + .ops = &rt5631_ops, + }, +}; + +static struct snd_soc_codec_driver soc_codec_dev_rt5631 = { + .probe = rt5631_probe, + .remove = rt5631_remove, + .suspend = rt5631_suspend, + .resume = rt5631_resume, + .set_bias_level = rt5631_set_bias_level, + .reg_cache_size = RT5631_VENDOR_ID2 + 1, + .reg_word_size = sizeof(u16), + .reg_cache_default = rt5631_reg, + .volatile_register = rt5631_volatile_register, + .readable_register = rt5631_readable_register, + .reg_cache_step = 1, + .controls = rt5631_snd_controls, + .num_controls = ARRAY_SIZE(rt5631_snd_controls), + .dapm_widgets = rt5631_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rt5631_dapm_widgets), + .dapm_routes = rt5631_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rt5631_dapm_routes), +}; + +static const struct i2c_device_id rt5631_i2c_id[] = { + { "rt5631", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, rt5631_i2c_id); + +static int rt5631_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct rt5631_priv *rt5631; + int ret; + + rt5631 = kzalloc(sizeof(struct rt5631_priv), GFP_KERNEL); + if (NULL == rt5631) + return -ENOMEM; + + i2c_set_clientdata(i2c, rt5631); + + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5631, + rt5631_dai, ARRAY_SIZE(rt5631_dai)); + if (ret < 0) + kfree(rt5631); + + return ret; +} + +static __devexit int rt5631_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + kfree(i2c_get_clientdata(client)); + return 0; +} + +static struct i2c_driver rt5631_i2c_driver = { + .driver = { + .name = "rt5631", + .owner = THIS_MODULE, + }, + .probe = rt5631_i2c_probe, + .remove = __devexit_p(rt5631_i2c_remove), + .id_table = rt5631_i2c_id, +}; + +static int __init rt5631_modinit(void) +{ + return i2c_add_driver(&rt5631_i2c_driver); +} +module_init(rt5631_modinit); + +static void __exit rt5631_modexit(void) +{ + i2c_del_driver(&rt5631_i2c_driver); +} +module_exit(rt5631_modexit); + +MODULE_DESCRIPTION("ASoC RT5631 driver"); +MODULE_AUTHOR("flove <flove@realtek.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/rt5631.h b/sound/soc/codecs/rt5631.h new file mode 100644 index 000000000000..13401581b0df --- /dev/null +++ b/sound/soc/codecs/rt5631.h @@ -0,0 +1,701 @@ +#ifndef __RTCODEC5631_H__ +#define __RTCODEC5631_H__ + + +#define RT5631_RESET 0x00 +#define RT5631_SPK_OUT_VOL 0x02 +#define RT5631_HP_OUT_VOL 0x04 +#define RT5631_MONO_AXO_1_2_VOL 0x06 +#define RT5631_AUX_IN_VOL 0x0A +#define RT5631_STEREO_DAC_VOL_1 0x0C +#define RT5631_MIC_CTRL_1 0x0E +#define RT5631_STEREO_DAC_VOL_2 0x10 +#define RT5631_ADC_CTRL_1 0x12 +#define RT5631_ADC_REC_MIXER 0x14 +#define RT5631_ADC_CTRL_2 0x16 +#define RT5631_VDAC_DIG_VOL 0x18 +#define RT5631_OUTMIXER_L_CTRL 0x1A +#define RT5631_OUTMIXER_R_CTRL 0x1C +#define RT5631_AXO1MIXER_CTRL 0x1E +#define RT5631_AXO2MIXER_CTRL 0x20 +#define RT5631_MIC_CTRL_2 0x22 +#define RT5631_DIG_MIC_CTRL 0x24 +#define RT5631_MONO_INPUT_VOL 0x26 +#define RT5631_SPK_MIXER_CTRL 0x28 +#define RT5631_SPK_MONO_OUT_CTRL 0x2A +#define RT5631_SPK_MONO_HP_OUT_CTRL 0x2C +#define RT5631_SDP_CTRL 0x34 +#define RT5631_MONO_SDP_CTRL 0x36 +#define RT5631_STEREO_AD_DA_CLK_CTRL 0x38 +#define RT5631_PWR_MANAG_ADD1 0x3A +#define RT5631_PWR_MANAG_ADD2 0x3B +#define RT5631_PWR_MANAG_ADD3 0x3C +#define RT5631_PWR_MANAG_ADD4 0x3E +#define RT5631_GEN_PUR_CTRL_REG 0x40 +#define RT5631_GLOBAL_CLK_CTRL 0x42 +#define RT5631_PLL_CTRL 0x44 +#define RT5631_INT_ST_IRQ_CTRL_1 0x48 +#define RT5631_INT_ST_IRQ_CTRL_2 0x4A +#define RT5631_GPIO_CTRL 0x4C +#define RT5631_MISC_CTRL 0x52 +#define RT5631_DEPOP_FUN_CTRL_1 0x54 +#define RT5631_DEPOP_FUN_CTRL_2 0x56 +#define RT5631_JACK_DET_CTRL 0x5A +#define RT5631_SOFT_VOL_CTRL 0x5C +#define RT5631_ALC_CTRL_1 0x64 +#define RT5631_ALC_CTRL_2 0x65 +#define RT5631_ALC_CTRL_3 0x66 +#define RT5631_PSEUDO_SPATL_CTRL 0x68 +#define RT5631_INDEX_ADD 0x6A +#define RT5631_INDEX_DATA 0x6C +#define RT5631_EQ_CTRL 0x6E +#define RT5631_VENDOR_ID 0x7A +#define RT5631_VENDOR_ID1 0x7C +#define RT5631_VENDOR_ID2 0x7E + +/* Index of Codec Private Register definition */ +#define RT5631_EQ_BW_LOP 0x00 +#define RT5631_EQ_GAIN_LOP 0x01 +#define RT5631_EQ_FC_BP1 0x02 +#define RT5631_EQ_BW_BP1 0x03 +#define RT5631_EQ_GAIN_BP1 0x04 +#define RT5631_EQ_FC_BP2 0x05 +#define RT5631_EQ_BW_BP2 0x06 +#define RT5631_EQ_GAIN_BP2 0x07 +#define RT5631_EQ_FC_BP3 0x08 +#define RT5631_EQ_BW_BP3 0x09 +#define RT5631_EQ_GAIN_BP3 0x0a +#define RT5631_EQ_BW_HIP 0x0b +#define RT5631_EQ_GAIN_HIP 0x0c +#define RT5631_EQ_HPF_A1 0x0d +#define RT5631_EQ_HPF_A2 0x0e +#define RT5631_EQ_HPF_GAIN 0x0f +#define RT5631_EQ_PRE_VOL_CTRL 0x11 +#define RT5631_EQ_POST_VOL_CTRL 0x12 +#define RT5631_TEST_MODE_CTRL 0x39 +#define RT5631_CP_INTL_REG2 0x45 +#define RT5631_ADDA_MIXER_INTL_REG3 0x52 +#define RT5631_SPK_INTL_CTRL 0x56 + + +/* global definition */ +#define RT5631_L_MUTE (0x1 << 15) +#define RT5631_L_MUTE_SHIFT 15 +#define RT5631_L_EN (0x1 << 14) +#define RT5631_L_EN_SHIFT 14 +#define RT5631_R_MUTE (0x1 << 7) +#define RT5631_R_MUTE_SHIFT 7 +#define RT5631_R_EN (0x1 << 6) +#define RT5631_R_EN_SHIFT 6 +#define RT5631_VOL_MASK 0x1f +#define RT5631_L_VOL_SHIFT 8 +#define RT5631_R_VOL_SHIFT 0 + +/* Speaker Output Control(0x02) */ +#define RT5631_SPK_L_VOL_SEL_MASK (0x1 << 14) +#define RT5631_SPK_L_VOL_SEL_VMID (0x0 << 14) +#define RT5631_SPK_L_VOL_SEL_SPKMIX_L (0x1 << 14) +#define RT5631_SPK_R_VOL_SEL_MASK (0x1 << 6) +#define RT5631_SPK_R_VOL_SEL_VMID (0x0 << 6) +#define RT5631_SPK_R_VOL_SEL_SPKMIX_R (0x1 << 6) + +/* Headphone Output Control(0x04) */ +#define RT5631_HP_L_VOL_SEL_MASK (0x1 << 14) +#define RT5631_HP_L_VOL_SEL_VMID (0x0 << 14) +#define RT5631_HP_L_VOL_SEL_OUTMIX_L (0x1 << 14) +#define RT5631_HP_R_VOL_SEL_MASK (0x1 << 6) +#define RT5631_HP_R_VOL_SEL_VMID (0x0 << 6) +#define RT5631_HP_R_VOL_SEL_OUTMIX_R (0x1 << 6) + +/* Output Control for AUXOUT/MONO(0x06) */ +#define RT5631_AUXOUT_1_VOL_SEL_MASK (0x1 << 14) +#define RT5631_AUXOUT_1_VOL_SEL_VMID (0x0 << 14) +#define RT5631_AUXOUT_1_VOL_SEL_OUTMIX_L (0x1 << 14) +#define RT5631_MUTE_MONO (0x1 << 13) +#define RT5631_MUTE_MONO_SHIFT 13 +#define RT5631_AUXOUT_2_VOL_SEL_MASK (0x1 << 6) +#define RT5631_AUXOUT_2_VOL_SEL_VMID (0x0 << 6) +#define RT5631_AUXOUT_2_VOL_SEL_OUTMIX_R (0x1 << 6) + +/* Microphone Input Control 1(0x0E) */ +#define RT5631_MIC1_DIFF_INPUT_CTRL (0x1 << 15) +#define RT5631_MIC1_DIFF_INPUT_SHIFT 15 +#define RT5631_MIC2_DIFF_INPUT_CTRL (0x1 << 7) +#define RT5631_MIC2_DIFF_INPUT_SHIFT 7 + +/* Stereo DAC Digital Volume2(0x10) */ +#define RT5631_DAC_VOL_MASK 0xff + +/* ADC Recording Mixer Control(0x14) */ +#define RT5631_M_OUTMIXER_L_TO_RECMIXER_L (0x1 << 15) +#define RT5631_M_OUTMIXL_RECMIXL_BIT 15 +#define RT5631_M_MIC1_TO_RECMIXER_L (0x1 << 14) +#define RT5631_M_MIC1_RECMIXL_BIT 14 +#define RT5631_M_AXIL_TO_RECMIXER_L (0x1 << 13) +#define RT5631_M_AXIL_RECMIXL_BIT 13 +#define RT5631_M_MONO_IN_TO_RECMIXER_L (0x1 << 12) +#define RT5631_M_MONO_IN_RECMIXL_BIT 12 +#define RT5631_M_OUTMIXER_R_TO_RECMIXER_R (0x1 << 7) +#define RT5631_M_OUTMIXR_RECMIXR_BIT 7 +#define RT5631_M_MIC2_TO_RECMIXER_R (0x1 << 6) +#define RT5631_M_MIC2_RECMIXR_BIT 6 +#define RT5631_M_AXIR_TO_RECMIXER_R (0x1 << 5) +#define RT5631_M_AXIR_RECMIXR_BIT 5 +#define RT5631_M_MONO_IN_TO_RECMIXER_R (0x1 << 4) +#define RT5631_M_MONO_IN_RECMIXR_BIT 4 + +/* Left Output Mixer Control(0x1A) */ +#define RT5631_M_RECMIXER_L_TO_OUTMIXER_L (0x1 << 15) +#define RT5631_M_RECMIXL_OUTMIXL_BIT 15 +#define RT5631_M_RECMIXER_R_TO_OUTMIXER_L (0x1 << 14) +#define RT5631_M_RECMIXR_OUTMIXL_BIT 14 +#define RT5631_M_DAC_L_TO_OUTMIXER_L (0x1 << 13) +#define RT5631_M_DACL_OUTMIXL_BIT 13 +#define RT5631_M_MIC1_TO_OUTMIXER_L (0x1 << 12) +#define RT5631_M_MIC1_OUTMIXL_BIT 12 +#define RT5631_M_MIC2_TO_OUTMIXER_L (0x1 << 11) +#define RT5631_M_MIC2_OUTMIXL_BIT 11 +#define RT5631_M_MONO_IN_P_TO_OUTMIXER_L (0x1 << 10) +#define RT5631_M_MONO_INP_OUTMIXL_BIT 10 +#define RT5631_M_AXIL_TO_OUTMIXER_L (0x1 << 9) +#define RT5631_M_AXIL_OUTMIXL_BIT 9 +#define RT5631_M_AXIR_TO_OUTMIXER_L (0x1 << 8) +#define RT5631_M_AXIR_OUTMIXL_BIT 8 +#define RT5631_M_VDAC_TO_OUTMIXER_L (0x1 << 7) +#define RT5631_M_VDAC_OUTMIXL_BIT 7 + +/* Right Output Mixer Control(0x1C) */ +#define RT5631_M_RECMIXER_L_TO_OUTMIXER_R (0x1 << 15) +#define RT5631_M_RECMIXL_OUTMIXR_BIT 15 +#define RT5631_M_RECMIXER_R_TO_OUTMIXER_R (0x1 << 14) +#define RT5631_M_RECMIXR_OUTMIXR_BIT 14 +#define RT5631_M_DAC_R_TO_OUTMIXER_R (0x1 << 13) +#define RT5631_M_DACR_OUTMIXR_BIT 13 +#define RT5631_M_MIC1_TO_OUTMIXER_R (0x1 << 12) +#define RT5631_M_MIC1_OUTMIXR_BIT 12 +#define RT5631_M_MIC2_TO_OUTMIXER_R (0x1 << 11) +#define RT5631_M_MIC2_OUTMIXR_BIT 11 +#define RT5631_M_MONO_IN_N_TO_OUTMIXER_R (0x1 << 10) +#define RT5631_M_MONO_INN_OUTMIXR_BIT 10 +#define RT5631_M_AXIL_TO_OUTMIXER_R (0x1 << 9) +#define RT5631_M_AXIL_OUTMIXR_BIT 9 +#define RT5631_M_AXIR_TO_OUTMIXER_R (0x1 << 8) +#define RT5631_M_AXIR_OUTMIXR_BIT 8 +#define RT5631_M_VDAC_TO_OUTMIXER_R (0x1 << 7) +#define RT5631_M_VDAC_OUTMIXR_BIT 7 + +/* Lout Mixer Control(0x1E) */ +#define RT5631_M_MIC1_TO_AXO1MIXER (0x1 << 15) +#define RT5631_M_MIC1_AXO1MIX_BIT 15 +#define RT5631_M_MIC2_TO_AXO1MIXER (0x1 << 11) +#define RT5631_M_MIC2_AXO1MIX_BIT 11 +#define RT5631_M_OUTMIXER_L_TO_AXO1MIXER (0x1 << 7) +#define RT5631_M_OUTMIXL_AXO1MIX_BIT 7 +#define RT5631_M_OUTMIXER_R_TO_AXO1MIXER (0x1 << 6) +#define RT5631_M_OUTMIXR_AXO1MIX_BIT 6 + +/* Rout Mixer Control(0x20) */ +#define RT5631_M_MIC1_TO_AXO2MIXER (0x1 << 15) +#define RT5631_M_MIC1_AXO2MIX_BIT 15 +#define RT5631_M_MIC2_TO_AXO2MIXER (0x1 << 11) +#define RT5631_M_MIC2_AXO2MIX_BIT 11 +#define RT5631_M_OUTMIXER_L_TO_AXO2MIXER (0x1 << 7) +#define RT5631_M_OUTMIXL_AXO2MIX_BIT 7 +#define RT5631_M_OUTMIXER_R_TO_AXO2MIXER (0x1 << 6) +#define RT5631_M_OUTMIXR_AXO2MIX_BIT 6 + +/* Micphone Input Control 2(0x22) */ +#define RT5631_MIC_BIAS_90_PRECNET_AVDD 1 +#define RT5631_MIC_BIAS_75_PRECNET_AVDD 2 + +#define RT5631_MIC1_BOOST_CTRL_MASK (0xf << 12) +#define RT5631_MIC1_BOOST_CTRL_BYPASS (0x0 << 12) +#define RT5631_MIC1_BOOST_CTRL_20DB (0x1 << 12) +#define RT5631_MIC1_BOOST_CTRL_24DB (0x2 << 12) +#define RT5631_MIC1_BOOST_CTRL_30DB (0x3 << 12) +#define RT5631_MIC1_BOOST_CTRL_35DB (0x4 << 12) +#define RT5631_MIC1_BOOST_CTRL_40DB (0x5 << 12) +#define RT5631_MIC1_BOOST_CTRL_34DB (0x6 << 12) +#define RT5631_MIC1_BOOST_CTRL_50DB (0x7 << 12) +#define RT5631_MIC1_BOOST_CTRL_52DB (0x8 << 12) +#define RT5631_MIC1_BOOST_SHIFT 12 + +#define RT5631_MIC2_BOOST_CTRL_MASK (0xf << 8) +#define RT5631_MIC2_BOOST_CTRL_BYPASS (0x0 << 8) +#define RT5631_MIC2_BOOST_CTRL_20DB (0x1 << 8) +#define RT5631_MIC2_BOOST_CTRL_24DB (0x2 << 8) +#define RT5631_MIC2_BOOST_CTRL_30DB (0x3 << 8) +#define RT5631_MIC2_BOOST_CTRL_35DB (0x4 << 8) +#define RT5631_MIC2_BOOST_CTRL_40DB (0x5 << 8) +#define RT5631_MIC2_BOOST_CTRL_34DB (0x6 << 8) +#define RT5631_MIC2_BOOST_CTRL_50DB (0x7 << 8) +#define RT5631_MIC2_BOOST_CTRL_52DB (0x8 << 8) +#define RT5631_MIC2_BOOST_SHIFT 8 + +#define RT5631_MICBIAS1_VOLT_CTRL_MASK (0x1 << 7) +#define RT5631_MICBIAS1_VOLT_CTRL_90P (0x0 << 7) +#define RT5631_MICBIAS1_VOLT_CTRL_75P (0x1 << 7) + +#define RT5631_MICBIAS1_S_C_DET_MASK (0x1 << 6) +#define RT5631_MICBIAS1_S_C_DET_DIS (0x0 << 6) +#define RT5631_MICBIAS1_S_C_DET_ENA (0x1 << 6) + +#define RT5631_MICBIAS1_SHORT_CURR_DET_MASK (0x3 << 4) +#define RT5631_MICBIAS1_SHORT_CURR_DET_600UA (0x0 << 4) +#define RT5631_MICBIAS1_SHORT_CURR_DET_1500UA (0x1 << 4) +#define RT5631_MICBIAS1_SHORT_CURR_DET_2000UA (0x2 << 4) + +#define RT5631_MICBIAS2_VOLT_CTRL_MASK (0x1 << 3) +#define RT5631_MICBIAS2_VOLT_CTRL_90P (0x0 << 3) +#define RT5631_MICBIAS2_VOLT_CTRL_75P (0x1 << 3) + +#define RT5631_MICBIAS2_S_C_DET_MASK (0x1 << 2) +#define RT5631_MICBIAS2_S_C_DET_DIS (0x0 << 2) +#define RT5631_MICBIAS2_S_C_DET_ENA (0x1 << 2) + +#define RT5631_MICBIAS2_SHORT_CURR_DET_MASK (0x3) +#define RT5631_MICBIAS2_SHORT_CURR_DET_600UA (0x0) +#define RT5631_MICBIAS2_SHORT_CURR_DET_1500UA (0x1) +#define RT5631_MICBIAS2_SHORT_CURR_DET_2000UA (0x2) + + +/* Digital Microphone Control(0x24) */ +#define RT5631_DMIC_ENA_MASK (0x1 << 15) +#define RT5631_DMIC_ENA_SHIFT 15 +/* DMIC_ENA: DMIC to ADC Digital filter */ +#define RT5631_DMIC_ENA (0x1 << 15) +/* DMIC_DIS: ADC mixer to ADC Digital filter */ +#define RT5631_DMIC_DIS (0x0 << 15) +#define RT5631_DMIC_L_CH_MUTE (0x1 << 13) +#define RT5631_DMIC_L_CH_MUTE_SHIFT 13 +#define RT5631_DMIC_R_CH_MUTE (0x1 << 12) +#define RT5631_DMIC_R_CH_MUTE_SHIFT 12 +#define RT5631_DMIC_L_CH_LATCH_MASK (0x1 << 9) +#define RT5631_DMIC_L_CH_LATCH_RISING (0x1 << 9) +#define RT5631_DMIC_L_CH_LATCH_FALLING (0x0 << 9) +#define RT5631_DMIC_R_CH_LATCH_MASK (0x1 << 8) +#define RT5631_DMIC_R_CH_LATCH_RISING (0x1 << 8) +#define RT5631_DMIC_R_CH_LATCH_FALLING (0x0 << 8) +#define RT5631_DMIC_CLK_CTRL_MASK (0x3 << 4) +#define RT5631_DMIC_CLK_CTRL_TO_128FS (0x0 << 4) +#define RT5631_DMIC_CLK_CTRL_TO_64FS (0x1 << 4) +#define RT5631_DMIC_CLK_CTRL_TO_32FS (0x2 << 4) + +/* Microphone Input Volume(0x26) */ +#define RT5631_MONO_DIFF_INPUT_SHIFT 15 + +/* Speaker Mixer Control(0x28) */ +#define RT5631_M_RECMIXER_L_TO_SPKMIXER_L (0x1 << 15) +#define RT5631_M_RECMIXL_SPKMIXL_BIT 15 +#define RT5631_M_MIC1_P_TO_SPKMIXER_L (0x1 << 14) +#define RT5631_M_MIC1P_SPKMIXL_BIT 14 +#define RT5631_M_DAC_L_TO_SPKMIXER_L (0x1 << 13) +#define RT5631_M_DACL_SPKMIXL_BIT 13 +#define RT5631_M_OUTMIXER_L_TO_SPKMIXER_L (0x1 << 12) +#define RT5631_M_OUTMIXL_SPKMIXL_BIT 12 + +#define RT5631_M_RECMIXER_R_TO_SPKMIXER_R (0x1 << 7) +#define RT5631_M_RECMIXR_SPKMIXR_BIT 7 +#define RT5631_M_MIC2_P_TO_SPKMIXER_R (0x1 << 6) +#define RT5631_M_MIC2P_SPKMIXR_BIT 6 +#define RT5631_M_DAC_R_TO_SPKMIXER_R (0x1 << 5) +#define RT5631_M_DACR_SPKMIXR_BIT 5 +#define RT5631_M_OUTMIXER_R_TO_SPKMIXER_R (0x1 << 4) +#define RT5631_M_OUTMIXR_SPKMIXR_BIT 4 + +/* Speaker/Mono Output Control(0x2A) */ +#define RT5631_M_SPKVOL_L_TO_SPOL_MIXER (0x1 << 15) +#define RT5631_M_SPKVOLL_SPOLMIX_BIT 15 +#define RT5631_M_SPKVOL_R_TO_SPOL_MIXER (0x1 << 14) +#define RT5631_M_SPKVOLR_SPOLMIX_BIT 14 +#define RT5631_M_SPKVOL_L_TO_SPOR_MIXER (0x1 << 13) +#define RT5631_M_SPKVOLL_SPORMIX_BIT 13 +#define RT5631_M_SPKVOL_R_TO_SPOR_MIXER (0x1 << 12) +#define RT5631_M_SPKVOLR_SPORMIX_BIT 12 +#define RT5631_M_OUTVOL_L_TO_MONOMIXER (0x1 << 11) +#define RT5631_M_OUTVOLL_MONOMIX_BIT 11 +#define RT5631_M_OUTVOL_R_TO_MONOMIXER (0x1 << 10) +#define RT5631_M_OUTVOLR_MONOMIX_BIT 10 + +/* Speaker/Mono/HP Output Control(0x2C) */ +#define RT5631_SPK_L_MUX_SEL_MASK (0x3 << 14) +#define RT5631_SPK_L_MUX_SEL_SPKMIXER_L (0x0 << 14) +#define RT5631_SPK_L_MUX_SEL_MONO_IN (0x1 << 14) +#define RT5631_SPK_L_MUX_SEL_DAC_L (0x3 << 14) +#define RT5631_SPK_L_MUX_SEL_SHIFT 14 + +#define RT5631_SPK_R_MUX_SEL_MASK (0x3 << 10) +#define RT5631_SPK_R_MUX_SEL_SPKMIXER_R (0x0 << 10) +#define RT5631_SPK_R_MUX_SEL_MONO_IN (0x1 << 10) +#define RT5631_SPK_R_MUX_SEL_DAC_R (0x3 << 10) +#define RT5631_SPK_R_MUX_SEL_SHIFT 10 + +#define RT5631_MONO_MUX_SEL_MASK (0x3 << 6) +#define RT5631_MONO_MUX_SEL_MONOMIXER (0x0 << 6) +#define RT5631_MONO_MUX_SEL_MONO_IN (0x1 << 6) +#define RT5631_MONO_MUX_SEL_SHIFT 6 + +#define RT5631_HP_L_MUX_SEL_MASK (0x1 << 3) +#define RT5631_HP_L_MUX_SEL_HPVOL_L (0x0 << 3) +#define RT5631_HP_L_MUX_SEL_DAC_L (0x1 << 3) +#define RT5631_HP_L_MUX_SEL_SHIFT 3 + +#define RT5631_HP_R_MUX_SEL_MASK (0x1 << 2) +#define RT5631_HP_R_MUX_SEL_HPVOL_R (0x0 << 2) +#define RT5631_HP_R_MUX_SEL_DAC_R (0x1 << 2) +#define RT5631_HP_R_MUX_SEL_SHIFT 2 + +/* Stereo I2S Serial Data Port Control(0x34) */ +#define RT5631_SDP_MODE_SEL_MASK (0x1 << 15) +#define RT5631_SDP_MODE_SEL_MASTER (0x0 << 15) +#define RT5631_SDP_MODE_SEL_SLAVE (0x1 << 15) + +#define RT5631_SDP_ADC_CPS_SEL_MASK (0x3 << 10) +#define RT5631_SDP_ADC_CPS_SEL_OFF (0x0 << 10) +#define RT5631_SDP_ADC_CPS_SEL_U_LAW (0x1 << 10) +#define RT5631_SDP_ADC_CPS_SEL_A_LAW (0x2 << 10) + +#define RT5631_SDP_DAC_CPS_SEL_MASK (0x3 << 8) +#define RT5631_SDP_DAC_CPS_SEL_OFF (0x0 << 8) +#define RT5631_SDP_DAC_CPS_SEL_U_LAW (0x1 << 8) +#define RT5631_SDP_DAC_CPS_SEL_A_LAW (0x2 << 8) +/* 0:Normal 1:Invert */ +#define RT5631_SDP_I2S_BCLK_POL_CTRL (0x1 << 7) +/* 0:Normal 1:Invert */ +#define RT5631_SDP_DAC_R_INV (0x1 << 6) +/* 0:ADC data appear at left phase of LRCK + * 1:ADC data appear at right phase of LRCK + */ +#define RT5631_SDP_ADC_DATA_L_R_SWAP (0x1 << 5) +/* 0:DAC data appear at left phase of LRCK + * 1:DAC data appear at right phase of LRCK + */ +#define RT5631_SDP_DAC_DATA_L_R_SWAP (0x1 << 4) + +/* Data Length Slection */ +#define RT5631_SDP_I2S_DL_MASK (0x3 << 2) +#define RT5631_SDP_I2S_DL_16 (0x0 << 2) +#define RT5631_SDP_I2S_DL_20 (0x1 << 2) +#define RT5631_SDP_I2S_DL_24 (0x2 << 2) +#define RT5631_SDP_I2S_DL_8 (0x3 << 2) + +/* PCM Data Format Selection */ +#define RT5631_SDP_I2S_DF_MASK (0x3) +#define RT5631_SDP_I2S_DF_I2S (0x0) +#define RT5631_SDP_I2S_DF_LEFT (0x1) +#define RT5631_SDP_I2S_DF_PCM_A (0x2) +#define RT5631_SDP_I2S_DF_PCM_B (0x3) + +/* Stereo AD/DA Clock Control(0x38h) */ +#define RT5631_I2S_PRE_DIV_MASK (0x7 << 13) +#define RT5631_I2S_PRE_DIV_1 (0x0 << 13) +#define RT5631_I2S_PRE_DIV_2 (0x1 << 13) +#define RT5631_I2S_PRE_DIV_4 (0x2 << 13) +#define RT5631_I2S_PRE_DIV_8 (0x3 << 13) +#define RT5631_I2S_PRE_DIV_16 (0x4 << 13) +#define RT5631_I2S_PRE_DIV_32 (0x5 << 13) +/* CLOCK RELATIVE OF BCLK AND LCRK */ +#define RT5631_I2S_LRCK_SEL_N_BCLK_MASK (0x1 << 12) +#define RT5631_I2S_LRCK_SEL_64_BCLK (0x0 << 12) /* 64FS */ +#define RT5631_I2S_LRCK_SEL_32_BCLK (0x1 << 12) /* 32FS */ + +#define RT5631_DAC_OSR_SEL_MASK (0x3 << 10) +#define RT5631_DAC_OSR_SEL_128FS (0x3 << 10) +#define RT5631_DAC_OSR_SEL_64FS (0x3 << 10) +#define RT5631_DAC_OSR_SEL_32FS (0x3 << 10) +#define RT5631_DAC_OSR_SEL_16FS (0x3 << 10) + +#define RT5631_ADC_OSR_SEL_MASK (0x3 << 8) +#define RT5631_ADC_OSR_SEL_128FS (0x3 << 8) +#define RT5631_ADC_OSR_SEL_64FS (0x3 << 8) +#define RT5631_ADC_OSR_SEL_32FS (0x3 << 8) +#define RT5631_ADC_OSR_SEL_16FS (0x3 << 8) + +#define RT5631_ADDA_FILTER_CLK_SEL_256FS (0 << 7) /* 256FS */ +#define RT5631_ADDA_FILTER_CLK_SEL_384FS (1 << 7) /* 384FS */ + +/* Power managment addition 1 (0x3A) */ +#define RT5631_PWR_MAIN_I2S_EN (0x1 << 15) +#define RT5631_PWR_MAIN_I2S_BIT 15 +#define RT5631_PWR_CLASS_D (0x1 << 12) +#define RT5631_PWR_CLASS_D_BIT 12 +#define RT5631_PWR_ADC_L_CLK (0x1 << 11) +#define RT5631_PWR_ADC_L_CLK_BIT 11 +#define RT5631_PWR_ADC_R_CLK (0x1 << 10) +#define RT5631_PWR_ADC_R_CLK_BIT 10 +#define RT5631_PWR_DAC_L_CLK (0x1 << 9) +#define RT5631_PWR_DAC_L_CLK_BIT 9 +#define RT5631_PWR_DAC_R_CLK (0x1 << 8) +#define RT5631_PWR_DAC_R_CLK_BIT 8 +#define RT5631_PWR_DAC_REF (0x1 << 7) +#define RT5631_PWR_DAC_REF_BIT 7 +#define RT5631_PWR_DAC_L_TO_MIXER (0x1 << 6) +#define RT5631_PWR_DAC_L_TO_MIXER_BIT 6 +#define RT5631_PWR_DAC_R_TO_MIXER (0x1 << 5) +#define RT5631_PWR_DAC_R_TO_MIXER_BIT 5 + +/* Power managment addition 2 (0x3B) */ +#define RT5631_PWR_OUTMIXER_L (0x1 << 15) +#define RT5631_PWR_OUTMIXER_L_BIT 15 +#define RT5631_PWR_OUTMIXER_R (0x1 << 14) +#define RT5631_PWR_OUTMIXER_R_BIT 14 +#define RT5631_PWR_SPKMIXER_L (0x1 << 13) +#define RT5631_PWR_SPKMIXER_L_BIT 13 +#define RT5631_PWR_SPKMIXER_R (0x1 << 12) +#define RT5631_PWR_SPKMIXER_R_BIT 12 +#define RT5631_PWR_RECMIXER_L (0x1 << 11) +#define RT5631_PWR_RECMIXER_L_BIT 11 +#define RT5631_PWR_RECMIXER_R (0x1 << 10) +#define RT5631_PWR_RECMIXER_R_BIT 10 +#define RT5631_PWR_MIC1_BOOT_GAIN (0x1 << 5) +#define RT5631_PWR_MIC1_BOOT_GAIN_BIT 5 +#define RT5631_PWR_MIC2_BOOT_GAIN (0x1 << 4) +#define RT5631_PWR_MIC2_BOOT_GAIN_BIT 4 +#define RT5631_PWR_MICBIAS1_VOL (0x1 << 3) +#define RT5631_PWR_MICBIAS1_VOL_BIT 3 +#define RT5631_PWR_MICBIAS2_VOL (0x1 << 2) +#define RT5631_PWR_MICBIAS2_VOL_BIT 2 +#define RT5631_PWR_PLL1 (0x1 << 1) +#define RT5631_PWR_PLL1_BIT 1 +#define RT5631_PWR_PLL2 (0x1 << 0) +#define RT5631_PWR_PLL2_BIT 0 + +/* Power managment addition 3(0x3C) */ +#define RT5631_PWR_VREF (0x1 << 15) +#define RT5631_PWR_VREF_BIT 15 +#define RT5631_PWR_FAST_VREF_CTRL (0x1 << 14) +#define RT5631_PWR_FAST_VREF_CTRL_BIT 14 +#define RT5631_PWR_MAIN_BIAS (0x1 << 13) +#define RT5631_PWR_MAIN_BIAS_BIT 13 +#define RT5631_PWR_AXO1MIXER (0x1 << 11) +#define RT5631_PWR_AXO1MIXER_BIT 11 +#define RT5631_PWR_AXO2MIXER (0x1 << 10) +#define RT5631_PWR_AXO2MIXER_BIT 10 +#define RT5631_PWR_MONOMIXER (0x1 << 9) +#define RT5631_PWR_MONOMIXER_BIT 9 +#define RT5631_PWR_MONO_DEPOP_DIS (0x1 << 8) +#define RT5631_PWR_MONO_DEPOP_DIS_BIT 8 +#define RT5631_PWR_MONO_AMP_EN (0x1 << 7) +#define RT5631_PWR_MONO_AMP_EN_BIT 7 +#define RT5631_PWR_CHARGE_PUMP (0x1 << 4) +#define RT5631_PWR_CHARGE_PUMP_BIT 4 +#define RT5631_PWR_HP_L_AMP (0x1 << 3) +#define RT5631_PWR_HP_L_AMP_BIT 3 +#define RT5631_PWR_HP_R_AMP (0x1 << 2) +#define RT5631_PWR_HP_R_AMP_BIT 2 +#define RT5631_PWR_HP_DEPOP_DIS (0x1 << 1) +#define RT5631_PWR_HP_DEPOP_DIS_BIT 1 +#define RT5631_PWR_HP_AMP_DRIVING (0x1 << 0) +#define RT5631_PWR_HP_AMP_DRIVING_BIT 0 + +/* Power managment addition 4(0x3E) */ +#define RT5631_PWR_SPK_L_VOL (0x1 << 15) +#define RT5631_PWR_SPK_L_VOL_BIT 15 +#define RT5631_PWR_SPK_R_VOL (0x1 << 14) +#define RT5631_PWR_SPK_R_VOL_BIT 14 +#define RT5631_PWR_LOUT_VOL (0x1 << 13) +#define RT5631_PWR_LOUT_VOL_BIT 13 +#define RT5631_PWR_ROUT_VOL (0x1 << 12) +#define RT5631_PWR_ROUT_VOL_BIT 12 +#define RT5631_PWR_HP_L_OUT_VOL (0x1 << 11) +#define RT5631_PWR_HP_L_OUT_VOL_BIT 11 +#define RT5631_PWR_HP_R_OUT_VOL (0x1 << 10) +#define RT5631_PWR_HP_R_OUT_VOL_BIT 10 +#define RT5631_PWR_AXIL_IN_VOL (0x1 << 9) +#define RT5631_PWR_AXIL_IN_VOL_BIT 9 +#define RT5631_PWR_AXIR_IN_VOL (0x1 << 8) +#define RT5631_PWR_AXIR_IN_VOL_BIT 8 +#define RT5631_PWR_MONO_IN_P_VOL (0x1 << 7) +#define RT5631_PWR_MONO_IN_P_VOL_BIT 7 +#define RT5631_PWR_MONO_IN_N_VOL (0x1 << 6) +#define RT5631_PWR_MONO_IN_N_VOL_BIT 6 + +/* General Purpose Control Register(0x40) */ +#define RT5631_SPK_AMP_AUTO_RATIO_EN (0x1 << 15) + +#define RT5631_SPK_AMP_RATIO_CTRL_MASK (0x7 << 12) +#define RT5631_SPK_AMP_RATIO_CTRL_2_34 (0x0 << 12) /* 7.40DB */ +#define RT5631_SPK_AMP_RATIO_CTRL_1_99 (0x1 << 12) /* 5.99DB */ +#define RT5631_SPK_AMP_RATIO_CTRL_1_68 (0x2 << 12) /* 4.50DB */ +#define RT5631_SPK_AMP_RATIO_CTRL_1_56 (0x3 << 12) /* 3.86DB */ +#define RT5631_SPK_AMP_RATIO_CTRL_1_44 (0x4 << 12) /* 3.16DB */ +#define RT5631_SPK_AMP_RATIO_CTRL_1_27 (0x5 << 12) /* 2.10DB */ +#define RT5631_SPK_AMP_RATIO_CTRL_1_09 (0x6 << 12) /* 0.80DB */ +#define RT5631_SPK_AMP_RATIO_CTRL_1_00 (0x7 << 12) /* 0.00DB */ +#define RT5631_SPK_AMP_RATIO_CTRL_SHIFT 12 + +#define RT5631_STEREO_DAC_HI_PASS_FILT_EN (0x1 << 11) +#define RT5631_STEREO_ADC_HI_PASS_FILT_EN (0x1 << 10) +/* Select ADC Wind Filter Clock type */ +#define RT5631_ADC_WIND_FILT_MASK (0x3 << 4) +#define RT5631_ADC_WIND_FILT_8_16_32K (0x0 << 4) /*8/16/32k*/ +#define RT5631_ADC_WIND_FILT_11_22_44K (0x1 << 4) /*11/22/44k*/ +#define RT5631_ADC_WIND_FILT_12_24_48K (0x2 << 4) /*12/24/48k*/ +#define RT5631_ADC_WIND_FILT_EN (0x1 << 3) +/* SelectADC Wind Filter Corner Frequency */ +#define RT5631_ADC_WIND_CNR_FREQ_MASK (0x7 << 0) +#define RT5631_ADC_WIND_CNR_FREQ_82_113_122 (0x0 << 0) /* 82/113/122 Hz */ +#define RT5631_ADC_WIND_CNR_FREQ_102_141_153 (0x1 << 0) /* 102/141/153 Hz */ +#define RT5631_ADC_WIND_CNR_FREQ_131_180_156 (0x2 << 0) /* 131/180/156 Hz */ +#define RT5631_ADC_WIND_CNR_FREQ_163_225_245 (0x3 << 0) /* 163/225/245 Hz */ +#define RT5631_ADC_WIND_CNR_FREQ_204_281_306 (0x4 << 0) /* 204/281/306 Hz */ +#define RT5631_ADC_WIND_CNR_FREQ_261_360_392 (0x5 << 0) /* 261/360/392 Hz */ +#define RT5631_ADC_WIND_CNR_FREQ_327_450_490 (0x6 << 0) /* 327/450/490 Hz */ +#define RT5631_ADC_WIND_CNR_FREQ_408_563_612 (0x7 << 0) /* 408/563/612 Hz */ + +/* Global Clock Control Register(0x42) */ +#define RT5631_SYSCLK_SOUR_SEL_MASK (0x3 << 14) +#define RT5631_SYSCLK_SOUR_SEL_MCLK (0x0 << 14) +#define RT5631_SYSCLK_SOUR_SEL_PLL (0x1 << 14) +#define RT5631_SYSCLK_SOUR_SEL_PLL_TCK (0x2 << 14) + +#define RT5631_PLLCLK_SOUR_SEL_MASK (0x3 << 12) +#define RT5631_PLLCLK_SOUR_SEL_MCLK (0x0 << 12) +#define RT5631_PLLCLK_SOUR_SEL_BCLK (0x1 << 12) +#define RT5631_PLLCLK_SOUR_SEL_VBCLK (0x2 << 12) + +#define RT5631_PLLCLK_PRE_DIV1 (0x0 << 11) +#define RT5631_PLLCLK_PRE_DIV2 (0x1 << 11) + +/* PLL Control(0x44) */ +#define RT5631_PLL_CTRL_M_VAL(m) ((m)&0xf) +#define RT5631_PLL_CTRL_K_VAL(k) (((k)&0x7) << 4) +#define RT5631_PLL_CTRL_N_VAL(n) (((n)&0xff) << 8) + +/* Internal Status and IRQ Control2(0x4A) */ +#define RT5631_ADC_DATA_SEL_MASK (0x3 << 14) +#define RT5631_ADC_DATA_SEL_Disable (0x0 << 14) +#define RT5631_ADC_DATA_SEL_MIC1 (0x1 << 14) +#define RT5631_ADC_DATA_SEL_MIC1_SHIFT 14 +#define RT5631_ADC_DATA_SEL_MIC2 (0x2 << 14) +#define RT5631_ADC_DATA_SEL_MIC2_SHIFT 15 +#define RT5631_ADC_DATA_SEL_STO (0x3 << 14) +#define RT5631_ADC_DATA_SEL_SHIFT 14 + +/* GPIO Pin Configuration(0x4C) */ +#define RT5631_GPIO_PIN_FUN_SEL_MASK (0x1 << 15) +#define RT5631_GPIO_PIN_FUN_SEL_IRQ (0x1 << 15) +#define RT5631_GPIO_PIN_FUN_SEL_GPIO_DIMC (0x0 << 15) + +#define RT5631_GPIO_DMIC_FUN_SEL_MASK (0x1 << 3) +#define RT5631_GPIO_DMIC_FUN_SEL_DIMC (0x1 << 3) +#define RT5631_GPIO_DMIC_FUN_SEL_GPIO (0x0 << 3) + +#define RT5631_GPIO_PIN_CON_MASK (0x1 << 2) +#define RT5631_GPIO_PIN_SET_INPUT (0x0 << 2) +#define RT5631_GPIO_PIN_SET_OUTPUT (0x1 << 2) + +/* De-POP function Control 1(0x54) */ +#define RT5631_POW_ON_SOFT_GEN (0x1 << 15) +#define RT5631_EN_MUTE_UNMUTE_DEPOP (0x1 << 14) +#define RT5631_EN_DEPOP2_FOR_HP (0x1 << 7) +/* Power Down HPAMP_L Starts Up Signal */ +#define RT5631_PD_HPAMP_L_ST_UP (0x1 << 5) +/* Power Down HPAMP_R Starts Up Signal */ +#define RT5631_PD_HPAMP_R_ST_UP (0x1 << 4) +/* Enable left HP mute/unmute depop */ +#define RT5631_EN_HP_L_M_UN_MUTE_DEPOP (0x1 << 1) +/* Enable right HP mute/unmute depop */ +#define RT5631_EN_HP_R_M_UN_MUTE_DEPOP (0x1 << 0) + +/* De-POP Fnction Control(0x56) */ +#define RT5631_EN_ONE_BIT_DEPOP (0x1 << 15) +#define RT5631_EN_CAP_FREE_DEPOP (0x1 << 14) + +/* Jack Detect Control Register(0x5A) */ +#define RT5631_JD_USE_MASK (0x3 << 14) +#define RT5631_JD_USE_JD2 (0x3 << 14) +#define RT5631_JD_USE_JD1 (0x2 << 14) +#define RT5631_JD_USE_GPIO (0x1 << 14) +#define RT5631_JD_OFF (0x0 << 14) +/* JD trigger enable for HP */ +#define RT5631_JD_HP_EN (0x1 << 11) +#define RT5631_JD_HP_TRI_MASK (0x1 << 10) +#define RT5631_JD_HP_TRI_HI (0x1 << 10) +#define RT5631_JD_HP_TRI_LO (0x1 << 10) +/* JD trigger enable for speaker LP/LN */ +#define RT5631_JD_SPK_L_EN (0x1 << 9) +#define RT5631_JD_SPK_L_TRI_MASK (0x1 << 8) +#define RT5631_JD_SPK_L_TRI_HI (0x1 << 8) +#define RT5631_JD_SPK_L_TRI_LO (0x0 << 8) +/* JD trigger enable for speaker RP/RN */ +#define RT5631_JD_SPK_R_EN (0x1 << 7) +#define RT5631_JD_SPK_R_TRI_MASK (0x1 << 6) +#define RT5631_JD_SPK_R_TRI_HI (0x1 << 6) +#define RT5631_JD_SPK_R_TRI_LO (0x0 << 6) +/* JD trigger enable for monoout */ +#define RT5631_JD_MONO_EN (0x1 << 5) +#define RT5631_JD_MONO_TRI_MASK (0x1 << 4) +#define RT5631_JD_MONO_TRI_HI (0x1 << 4) +#define RT5631_JD_MONO_TRI_LO (0x0 << 4) +/* JD trigger enable for Lout */ +#define RT5631_JD_AUX_1_EN (0x1 << 3) +#define RT5631_JD_AUX_1_MASK (0x1 << 2) +#define RT5631_JD_AUX_1_TRI_HI (0x1 << 2) +#define RT5631_JD_AUX_1_TRI_LO (0x0 << 2) +/* JD trigger enable for Rout */ +#define RT5631_JD_AUX_2_EN (0x1 << 1) +#define RT5631_JD_AUX_2_MASK (0x1 << 0) +#define RT5631_JD_AUX_2_TRI_HI (0x1 << 0) +#define RT5631_JD_AUX_2_TRI_LO (0x0 << 0) + +/* ALC CONTROL 1(0x64) */ +#define RT5631_ALC_ATTACK_RATE_MASK (0x1F << 8) +#define RT5631_ALC_RECOVERY_RATE_MASK (0x1F << 0) + +/* ALC CONTROL 2(0x65) */ +/* select Compensation gain for Noise gate function */ +#define RT5631_ALC_COM_NOISE_GATE_MASK (0xF << 0) + +/* ALC CONTROL 3(0x66) */ +#define RT5631_ALC_FUN_MASK (0x3 << 14) +#define RT5631_ALC_FUN_DIS (0x0 << 14) +#define RT5631_ALC_ENA_DAC_PATH (0x1 << 14) +#define RT5631_ALC_ENA_ADC_PATH (0x3 << 14) +#define RT5631_ALC_PARA_UPDATE (0x1 << 13) +#define RT5631_ALC_LIMIT_LEVEL_MASK (0x1F << 8) +#define RT5631_ALC_NOISE_GATE_FUN_MASK (0x1 << 7) +#define RT5631_ALC_NOISE_GATE_FUN_DIS (0x0 << 7) +#define RT5631_ALC_NOISE_GATE_FUN_ENA (0x1 << 7) +/* ALC noise gate hold data function */ +#define RT5631_ALC_NOISE_GATE_H_D_MASK (0x1 << 6) +#define RT5631_ALC_NOISE_GATE_H_D_DIS (0x0 << 6) +#define RT5631_ALC_NOISE_GATE_H_D_ENA (0x1 << 6) + +/* Psedueo Stereo & Spatial Effect Block Control(0x68) */ +#define RT5631_SPATIAL_CTRL_EN (0x1 << 15) +#define RT5631_ALL_PASS_FILTER_EN (0x1 << 14) +#define RT5631_PSEUDO_STEREO_EN (0x1 << 13) +#define RT5631_STEREO_EXPENSION_EN (0x1 << 12) +/* 3D gain parameter */ +#define RT5631_GAIN_3D_PARA_MASK (0x3 << 6) +#define RT5631_GAIN_3D_PARA_1_00 (0x0 << 6) /* 3D gain 1.0 */ +#define RT5631_GAIN_3D_PARA_1_50 (0x1 << 6) /* 3D gain 1.5 */ +#define RT5631_GAIN_3D_PARA_2_00 (0x2 << 6) /* 3D gain 2.0 */ +/* 3D ratio parameter */ +#define RT5631_RATIO_3D_MASK (0x3 << 4) +#define RT5631_RATIO_3D_0_0 (0x0 << 4) /* 3D ratio 0.0 */ +#define RT5631_RATIO_3D_0_66 (0x1 << 4) /* 3D ratio 0.66 */ +#define RT5631_RATIO_3D_1_0 (0x2 << 4) /* 3D ratio 1.0 */ +/* select samplerate for all pass filter */ +#define RT5631_APF_FUN_SLE_MASK (0x3 << 0) +#define RT5631_APF_FUN_SEL_48K (0x3 << 0) +#define RT5631_APF_FUN_SEL_44_1K (0x2 << 0) +#define RT5631_APF_FUN_SEL_32K (0x1 << 0) +#define RT5631_APF_FUN_DIS (0x0 << 0) + +/* EQ CONTROL 1(0x6E) */ +#define RT5631_HW_EQ_PATH_SEL_MASK (0x1 << 15) +#define RT5631_HW_EQ_PATH_SEL_DAC (0x0 << 15) +#define RT5631_HW_EQ_PATH_SEL_ADC (0x1 << 15) +#define RT5631_HW_EQ_UPDATE_CTRL (0x1 << 14) + +#define RT5631_EN_HW_EQ_HPF2 (0x1 << 5) +#define RT5631_EN_HW_EQ_HPF1 (0x1 << 4) +#define RT5631_EN_HW_EQ_BP3 (0x1 << 3) +#define RT5631_EN_HW_EQ_BP2 (0x1 << 2) +#define RT5631_EN_HW_EQ_BP1 (0x1 << 1) +#define RT5631_EN_HW_EQ_LPF (0x1 << 0) + + +#endif /* __RTCODEC5631_H__ */ diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 7e4066e131e6..d15695d1c273 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -20,6 +20,7 @@ #include <linux/regulator/driver.h> #include <linux/regulator/machine.h> #include <linux/regulator/consumer.h> +#include <linux/of_device.h> #include <sound/core.h> #include <sound/tlv.h> #include <sound/pcm.h> @@ -130,16 +131,13 @@ static int mic_bias_event(struct snd_soc_dapm_widget *w, case SND_SOC_DAPM_POST_PMU: /* change mic bias resistor to 4Kohm */ snd_soc_update_bits(w->codec, SGTL5000_CHIP_MIC_CTRL, - SGTL5000_BIAS_R_4k, SGTL5000_BIAS_R_4k); + SGTL5000_BIAS_R_MASK, + SGTL5000_BIAS_R_4k << SGTL5000_BIAS_R_SHIFT); break; case SND_SOC_DAPM_PRE_PMD: - /* - * SGTL5000_BIAS_R_8k as mask to clean the two bits - * of mic bias and output impedance - */ snd_soc_update_bits(w->codec, SGTL5000_CHIP_MIC_CTRL, - SGTL5000_BIAS_R_8k, 0); + SGTL5000_BIAS_R_MASK, 0); break; } return 0; @@ -725,7 +723,9 @@ static int sgtl5000_pcm_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - snd_soc_update_bits(codec, SGTL5000_CHIP_I2S_CTRL, i2s_ctl, i2s_ctl); + snd_soc_update_bits(codec, SGTL5000_CHIP_I2S_CTRL, + SGTL5000_I2S_DLEN_MASK | SGTL5000_I2S_SCLKFREQ_MASK, + i2s_ctl); return 0; } @@ -756,7 +756,7 @@ static int ldo_regulator_enable(struct regulator_dev *dev) /* set voltage to register */ snd_soc_update_bits(codec, SGTL5000_CHIP_LINREG_CTRL, - (0x1 << 4) - 1, reg); + SGTL5000_LINREG_VDDD_MASK, reg); snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, SGTL5000_LINEREG_D_POWERUP, @@ -782,7 +782,7 @@ static int ldo_regulator_disable(struct regulator_dev *dev) /* clear voltage info */ snd_soc_update_bits(codec, SGTL5000_CHIP_LINREG_CTRL, - (0x1 << 4) - 1, 0); + SGTL5000_LINREG_VDDD_MASK, 0); ldo->enabled = 0; @@ -808,6 +808,7 @@ static int ldo_regulator_register(struct snd_soc_codec *codec, int voltage) { struct ldo_regulator *ldo; + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); ldo = kzalloc(sizeof(struct ldo_regulator), GFP_KERNEL); @@ -842,6 +843,7 @@ static int ldo_regulator_register(struct snd_soc_codec *codec, return ret; } + sgtl5000->ldo = ldo; return 0; } @@ -1115,7 +1117,7 @@ static int sgtl5000_set_power_regs(struct snd_soc_codec *codec) /* set voltage to register */ snd_soc_update_bits(codec, SGTL5000_CHIP_LINREG_CTRL, - (0x1 << 4) - 1, 0x8); + SGTL5000_LINREG_VDDD_MASK, 0x8); /* * if vddd linear reg has been enabled, @@ -1146,8 +1148,7 @@ static int sgtl5000_set_power_regs(struct snd_soc_codec *codec) vag = (vag - SGTL5000_ANA_GND_BASE) / SGTL5000_ANA_GND_STP; snd_soc_update_bits(codec, SGTL5000_CHIP_REF_CTRL, - vag << SGTL5000_ANA_GND_SHIFT, - vag << SGTL5000_ANA_GND_SHIFT); + SGTL5000_ANA_GND_MASK, vag << SGTL5000_ANA_GND_SHIFT); /* set line out VAG to vddio / 2, in range (0.8v, 1.675v) */ vag = vddio / 2; @@ -1161,9 +1162,8 @@ static int sgtl5000_set_power_regs(struct snd_soc_codec *codec) SGTL5000_LINE_OUT_GND_STP; snd_soc_update_bits(codec, SGTL5000_CHIP_LINE_OUT_CTRL, - vag << SGTL5000_LINE_OUT_GND_SHIFT | - SGTL5000_LINE_OUT_CURRENT_360u << - SGTL5000_LINE_OUT_CURRENT_SHIFT, + SGTL5000_LINE_OUT_CURRENT_MASK | + SGTL5000_LINE_OUT_GND_MASK, vag << SGTL5000_LINE_OUT_GND_SHIFT | SGTL5000_LINE_OUT_CURRENT_360u << SGTL5000_LINE_OUT_CURRENT_SHIFT); @@ -1436,10 +1436,17 @@ static const struct i2c_device_id sgtl5000_id[] = { MODULE_DEVICE_TABLE(i2c, sgtl5000_id); +static const struct of_device_id sgtl5000_dt_ids[] = { + { .compatible = "fsl,sgtl5000", }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, sgtl5000_dt_ids); + static struct i2c_driver sgtl5000_i2c_driver = { .driver = { .name = "sgtl5000", .owner = THIS_MODULE, + .of_match_table = sgtl5000_dt_ids, }, .probe = sgtl5000_i2c_probe, .remove = __devexit_p(sgtl5000_i2c_remove), diff --git a/sound/soc/codecs/sgtl5000.h b/sound/soc/codecs/sgtl5000.h index eec3ab368f39..8a9f43534b79 100644 --- a/sound/soc/codecs/sgtl5000.h +++ b/sound/soc/codecs/sgtl5000.h @@ -280,7 +280,7 @@ /* * SGTL5000_CHIP_MIC_CTRL */ -#define SGTL5000_BIAS_R_MASK 0x0200 +#define SGTL5000_BIAS_R_MASK 0x0300 #define SGTL5000_BIAS_R_SHIFT 8 #define SGTL5000_BIAS_R_WIDTH 2 #define SGTL5000_BIAS_R_off 0x0 diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c index 84ffdebb8a8b..f681e41fc12e 100644 --- a/sound/soc/codecs/sn95031.c +++ b/sound/soc/codecs/sn95031.c @@ -79,7 +79,7 @@ static void configure_adc(struct snd_soc_codec *sn95031_codec, int val) */ static int find_free_channel(struct snd_soc_codec *sn95031_codec) { - int ret = 0, i, value; + int i, value; /* check whether ADC is enabled */ value = snd_soc_read(sn95031_codec, SN95031_ADC1CNTL1); @@ -91,12 +91,10 @@ static int find_free_channel(struct snd_soc_codec *sn95031_codec) for (i = 0; i < SN95031_ADC_CHANLS_MAX; i++) { value = snd_soc_read(sn95031_codec, SN95031_ADC_CHNL_START_ADDR + i); - if (value & SN95031_STOPBIT_MASK) { - ret = i; + if (value & SN95031_STOPBIT_MASK) break; - } } - return (ret > SN95031_ADC_LOOP_MAX) ? (-EINVAL) : ret; + return (i == SN95031_ADC_CHANLS_MAX) ? (-EINVAL) : i; } /* Initialize the ADC for reading micbias values. Can sleep. */ @@ -104,7 +102,7 @@ static int sn95031_initialize_adc(struct snd_soc_codec *sn95031_codec) { int base_addr, chnl_addr; int value; - static int channel_index; + int channel_index; /* Index of the first channel in which the stop bit is set */ channel_index = find_free_channel(sn95031_codec); @@ -163,7 +161,6 @@ static unsigned int sn95031_get_mic_bias(struct snd_soc_codec *codec) pr_debug("mic bias = %dmV\n", mic_bias); return mic_bias; } -EXPORT_SYMBOL_GPL(sn95031_get_mic_bias); /*end - adc helper functions */ static inline unsigned int sn95031_read(struct snd_soc_codec *codec, @@ -660,7 +657,7 @@ static int sn95031_pcm_spkr_mute(struct snd_soc_dai *dai, int mute) return 0; } -int sn95031_pcm_hw_params(struct snd_pcm_substream *substream, +static int sn95031_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { unsigned int format, rate; @@ -718,7 +715,7 @@ static struct snd_soc_dai_ops sn95031_vib2_dai_ops = { .hw_params = sn95031_pcm_hw_params, }; -struct snd_soc_dai_driver sn95031_dais[] = { +static struct snd_soc_dai_driver sn95031_dais[] = { { .name = "SN95031 Headset", .playback = { @@ -829,7 +826,6 @@ static int sn95031_codec_probe(struct snd_soc_codec *codec) { pr_debug("codec_probe called\n"); - codec->dapm.bias_level = SND_SOC_BIAS_OFF; codec->dapm.idle_bias_off = 1; /* PCM interface config diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 9801cd7cfcb5..3cb3271c5fe2 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -59,6 +59,7 @@ struct ssm2602_priv { struct snd_pcm_substream *slave_substream; enum ssm2602_type type; + unsigned int clk_out_pwr; }; /* @@ -294,7 +295,6 @@ static int ssm2602_startup(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_codec *codec = rtd->codec; struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec); - struct i2c_client *i2c = codec->control_data; struct snd_pcm_runtime *master_runtime; /* The DAI has shared clocks so if we already have a playback or @@ -303,7 +303,7 @@ static int ssm2602_startup(struct snd_pcm_substream *substream, */ if (ssm2602->master_substream) { master_runtime = ssm2602->master_substream->runtime; - dev_dbg(&i2c->dev, "Constraining to %d bits at %dHz\n", + dev_dbg(codec->dev, "Constraining to %d bits at %dHz\n", master_runtime->sample_bits, master_runtime->rate); @@ -343,12 +343,14 @@ static void ssm2602_shutdown(struct snd_pcm_substream *substream, static int ssm2602_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; - u16 mute_reg = snd_soc_read(codec, SSM2602_APDIGI) & ~APDIGI_ENABLE_DAC_MUTE; + if (mute) - snd_soc_write(codec, SSM2602_APDIGI, - mute_reg | APDIGI_ENABLE_DAC_MUTE); + snd_soc_update_bits(codec, SSM2602_APDIGI, + APDIGI_ENABLE_DAC_MUTE, + APDIGI_ENABLE_DAC_MUTE); else - snd_soc_write(codec, SSM2602_APDIGI, mute_reg); + snd_soc_update_bits(codec, SSM2602_APDIGI, + APDIGI_ENABLE_DAC_MUTE, 0); return 0; } @@ -357,16 +359,46 @@ static int ssm2602_set_dai_sysclk(struct snd_soc_dai *codec_dai, { struct snd_soc_codec *codec = codec_dai->codec; struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec); - switch (freq) { - case 11289600: - case 12000000: - case 12288000: - case 16934400: - case 18432000: - ssm2602->sysclk = freq; - return 0; + + if (dir == SND_SOC_CLOCK_IN) { + if (clk_id != SSM2602_SYSCLK) + return -EINVAL; + + switch (freq) { + case 11289600: + case 12000000: + case 12288000: + case 16934400: + case 18432000: + ssm2602->sysclk = freq; + break; + default: + return -EINVAL; + } + } else { + unsigned int mask; + + switch (clk_id) { + case SSM2602_CLK_CLKOUT: + mask = PWR_CLK_OUT_PDN; + break; + case SSM2602_CLK_XTO: + mask = PWR_OSC_PDN; + break; + default: + return -EINVAL; + } + + if (freq == 0) + ssm2602->clk_out_pwr |= mask; + else + ssm2602->clk_out_pwr &= ~mask; + + snd_soc_update_bits(codec, SSM2602_PWR, + PWR_CLK_OUT_PDN | PWR_OSC_PDN, ssm2602->clk_out_pwr); } - return -EINVAL; + + return 0; } static int ssm2602_set_dai_fmt(struct snd_soc_dai *codec_dai, @@ -431,23 +463,27 @@ static int ssm2602_set_dai_fmt(struct snd_soc_dai *codec_dai, static int ssm2602_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - u16 reg = snd_soc_read(codec, SSM2602_PWR); - reg &= ~(PWR_POWER_OFF | PWR_OSC_PDN); + struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec); switch (level) { case SND_SOC_BIAS_ON: - /* vref/mid, osc on, dac unmute */ - snd_soc_write(codec, SSM2602_PWR, reg); + /* vref/mid on, osc and clkout on if enabled */ + snd_soc_update_bits(codec, SSM2602_PWR, + PWR_POWER_OFF | PWR_CLK_OUT_PDN | PWR_OSC_PDN, + ssm2602->clk_out_pwr); break; case SND_SOC_BIAS_PREPARE: break; case SND_SOC_BIAS_STANDBY: /* everything off except vref/vmid, */ - snd_soc_write(codec, SSM2602_PWR, reg | PWR_CLK_OUT_PDN); + snd_soc_update_bits(codec, SSM2602_PWR, + PWR_POWER_OFF | PWR_CLK_OUT_PDN | PWR_OSC_PDN, + PWR_CLK_OUT_PDN | PWR_OSC_PDN); break; case SND_SOC_BIAS_OFF: - /* everything off, dac mute, inactive */ - snd_soc_write(codec, SSM2602_PWR, 0xffff); + /* everything off */ + snd_soc_update_bits(codec, SSM2602_PWR, + PWR_POWER_OFF, PWR_POWER_OFF); break; } @@ -506,12 +542,12 @@ static int ssm2602_resume(struct snd_soc_codec *codec) static int ssm2602_probe(struct snd_soc_codec *codec) { struct snd_soc_dapm_context *dapm = &codec->dapm; - int ret, reg; + int ret; - reg = snd_soc_read(codec, SSM2602_LOUT1V); - snd_soc_write(codec, SSM2602_LOUT1V, reg | LOUT1V_LRHP_BOTH); - reg = snd_soc_read(codec, SSM2602_ROUT1V); - snd_soc_write(codec, SSM2602_ROUT1V, reg | ROUT1V_RLHP_BOTH); + snd_soc_update_bits(codec, SSM2602_LOUT1V, + LOUT1V_LRHP_BOTH, LOUT1V_LRHP_BOTH); + snd_soc_update_bits(codec, SSM2602_ROUT1V, + ROUT1V_RLHP_BOTH, ROUT1V_RLHP_BOTH); ret = snd_soc_add_controls(codec, ssm2602_snd_controls, ARRAY_SIZE(ssm2602_snd_controls)); @@ -544,7 +580,7 @@ static int ssm2604_probe(struct snd_soc_codec *codec) static int ssm260x_probe(struct snd_soc_codec *codec) { struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec); - int ret, reg; + int ret; pr_info("ssm2602 Audio Codec %s", SSM2602_VERSION); @@ -561,10 +597,10 @@ static int ssm260x_probe(struct snd_soc_codec *codec) } /* set the update bits */ - reg = snd_soc_read(codec, SSM2602_LINVOL); - snd_soc_write(codec, SSM2602_LINVOL, reg | LINVOL_LRIN_BOTH); - reg = snd_soc_read(codec, SSM2602_RINVOL); - snd_soc_write(codec, SSM2602_RINVOL, reg | RINVOL_RLIN_BOTH); + snd_soc_update_bits(codec, SSM2602_LINVOL, + LINVOL_LRIN_BOTH, LINVOL_LRIN_BOTH); + snd_soc_update_bits(codec, SSM2602_RINVOL, + RINVOL_RLIN_BOTH, RINVOL_RLIN_BOTH); /*select Line in as default input*/ snd_soc_write(codec, SSM2602_APANA, APANA_SELECT_DAC | APANA_ENABLE_MIC_BOOST); @@ -578,7 +614,12 @@ static int ssm260x_probe(struct snd_soc_codec *codec) break; } - return ret; + if (ret) + return ret; + + ssm2602_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; } /* remove everything here */ diff --git a/sound/soc/codecs/ssm2602.h b/sound/soc/codecs/ssm2602.h index b98c69168036..fbd07d7b73ca 100644 --- a/sound/soc/codecs/ssm2602.h +++ b/sound/soc/codecs/ssm2602.h @@ -116,6 +116,10 @@ #define SSM2602_CACHEREGNUM 10 -#define SSM2602_SYSCLK 0 +enum ssm2602_clk { + SSM2602_SYSCLK, + SSM2602_CLK_CLKOUT, + SSM2602_CLK_XTO +}; #endif diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index fbd7eb9e61ce..bb82408ab8e1 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -524,13 +524,17 @@ static int sta32x_hw_params(struct snd_pcm_substream *substream, rate = params_rate(params); pr_debug("rate: %u\n", rate); for (i = 0; i < ARRAY_SIZE(interpolation_ratios); i++) - if (interpolation_ratios[i].fs == rate) + if (interpolation_ratios[i].fs == rate) { ir = interpolation_ratios[i].ir; + break; + } if (ir < 0) return -EINVAL; for (i = 0; mclk_ratios[ir][i].ratio; i++) - if (mclk_ratios[ir][i].ratio * rate == sta32x->mclk) + if (mclk_ratios[ir][i].ratio * rate == sta32x->mclk) { mcs = mclk_ratios[ir][i].mcs; + break; + } if (mcs < 0) return -EINVAL; @@ -752,25 +756,19 @@ static int sta32x_probe(struct snd_soc_codec *codec) return ret; } - /* read reg reset values into cache */ - for (i = 0; i < STA32X_REGISTER_COUNT; i++) - snd_soc_cache_write(codec, i, sta32x_regs[i]); - - /* preserve reset values of reserved register bits */ - snd_soc_cache_write(codec, STA32X_CONFC, - codec->hw_read(codec, STA32X_CONFC)); - snd_soc_cache_write(codec, STA32X_CONFE, - codec->hw_read(codec, STA32X_CONFE)); - snd_soc_cache_write(codec, STA32X_CONFF, - codec->hw_read(codec, STA32X_CONFF)); - snd_soc_cache_write(codec, STA32X_MMUTE, - codec->hw_read(codec, STA32X_MMUTE)); - snd_soc_cache_write(codec, STA32X_AUTO1, - codec->hw_read(codec, STA32X_AUTO1)); - snd_soc_cache_write(codec, STA32X_AUTO3, - codec->hw_read(codec, STA32X_AUTO3)); - snd_soc_cache_write(codec, STA32X_C3CFG, - codec->hw_read(codec, STA32X_C3CFG)); + /* Chip documentation explicitly requires that the reset values + * of reserved register bits are left untouched. + * Write the register default value to cache for reserved registers, + * so the write to the these registers are suppressed by the cache + * restore code when it skips writes of default registers. + */ + snd_soc_cache_write(codec, STA32X_CONFC, 0xc2); + snd_soc_cache_write(codec, STA32X_CONFE, 0xc2); + snd_soc_cache_write(codec, STA32X_CONFF, 0x5c); + snd_soc_cache_write(codec, STA32X_MMUTE, 0x10); + snd_soc_cache_write(codec, STA32X_AUTO1, 0x60); + snd_soc_cache_write(codec, STA32X_AUTO3, 0x00); + snd_soc_cache_write(codec, STA32X_C3CFG, 0x40); /* FIXME enable thermal warning adjustment and recovery */ snd_soc_update_bits(codec, STA32X_CONFA, @@ -808,6 +806,7 @@ static int sta32x_remove(struct snd_soc_codec *codec) { struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec); + sta32x_set_bias_level(codec, SND_SOC_BIAS_OFF); regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); regulator_bulk_free(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); @@ -832,6 +831,7 @@ static const struct snd_soc_codec_driver sta32x_codec = { .resume = sta32x_resume, .reg_cache_size = STA32X_REGISTER_COUNT, .reg_word_size = sizeof(u8), + .reg_cache_default = sta32x_regs, .volatile_register = sta32x_reg_is_volatile, .set_bias_level = sta32x_set_bias_level, .controls = sta32x_snd_controls, @@ -867,18 +867,8 @@ static __devinit int sta32x_i2c_probe(struct i2c_client *i2c, static __devexit int sta32x_i2c_remove(struct i2c_client *client) { struct sta32x_priv *sta32x = i2c_get_clientdata(client); - struct snd_soc_codec *codec = sta32x->codec; - - if (codec) - sta32x_set_bias_level(codec, SND_SOC_BIAS_OFF); - - regulator_bulk_free(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); - - if (codec) { - snd_soc_unregister_codec(&client->dev); - snd_soc_codec_set_drvdata(codec, NULL); - } + snd_soc_unregister_codec(&client->dev); kfree(sta32x); return 0; } diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index 33bb52f3f683..ab27dbcd1262 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -47,63 +47,6 @@ static const u16 tlv320aic23_reg[] = { 0x0000, 0x0000, 0x0000, 0x0000, /* 12 */ }; -/* - * read tlv320aic23 register cache - */ -static inline unsigned int tlv320aic23_read_reg_cache(struct snd_soc_codec - *codec, unsigned int reg) -{ - u16 *cache = codec->reg_cache; - if (reg >= ARRAY_SIZE(tlv320aic23_reg)) - return -1; - return cache[reg]; -} - -/* - * write tlv320aic23 register cache - */ -static inline void tlv320aic23_write_reg_cache(struct snd_soc_codec *codec, - u8 reg, u16 value) -{ - u16 *cache = codec->reg_cache; - if (reg >= ARRAY_SIZE(tlv320aic23_reg)) - return; - cache[reg] = value; -} - -/* - * write to the tlv320aic23 register space - */ -static int tlv320aic23_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - - u8 data[2]; - - /* TLV320AIC23 has 7 bit address and 9 bits of data - * so we need to switch one data bit into reg and rest - * of data into val - */ - - if (reg > 9 && reg != 15) { - printk(KERN_WARNING "%s Invalid register R%u\n", __func__, reg); - return -1; - } - - data[0] = (reg << 1) | (value >> 8 & 0x01); - data[1] = value & 0xff; - - tlv320aic23_write_reg_cache(codec, reg, value); - - if (codec->hw_write(codec->control_data, data, 2) == 2) - return 0; - - printk(KERN_ERR "%s cannot write %03x to register R%u\n", __func__, - value, reg); - - return -EIO; -} - static const char *rec_src_text[] = { "Line", "Mic" }; static const char *deemph_text[] = {"None", "32Khz", "44.1Khz", "48Khz"}; @@ -139,8 +82,8 @@ static int snd_soc_tlv320aic23_put_volsw(struct snd_kcontrol *kcontrol, */ val = (val >= 4) ? 4 : (3 - val); - reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_ANLG) & (~0x1C0); - tlv320aic23_write(codec, TLV320AIC23_ANLG, reg | (val << 6)); + reg = snd_soc_read(codec, TLV320AIC23_ANLG) & (~0x1C0); + snd_soc_write(codec, TLV320AIC23_ANLG, reg | (val << 6)); return 0; } @@ -151,7 +94,7 @@ static int snd_soc_tlv320aic23_get_volsw(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); u16 val; - val = tlv320aic23_read_reg_cache(codec, TLV320AIC23_ANLG) & (0x1C0); + val = snd_soc_read(codec, TLV320AIC23_ANLG) & (0x1C0); val = val >> 6; val = (val >= 4) ? 4 : (3 - val); ucontrol->value.integer.value[0] = val; @@ -159,15 +102,6 @@ static int snd_soc_tlv320aic23_get_volsw(struct snd_kcontrol *kcontrol, } -#define SOC_TLV320AIC23_SINGLE_TLV(xname, reg, shift, max, invert, tlv_array) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ - .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ - SNDRV_CTL_ELEM_ACCESS_READWRITE,\ - .tlv.p = (tlv_array), \ - .info = snd_soc_info_volsw, .get = snd_soc_tlv320aic23_get_volsw,\ - .put = snd_soc_tlv320aic23_put_volsw, \ - .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) } - static const struct snd_kcontrol_new tlv320aic23_snd_controls[] = { SOC_DOUBLE_R_TLV("Digital Playback Volume", TLV320AIC23_LCHNVOL, TLV320AIC23_RCHNVOL, 0, 127, 0, out_gain_tlv), @@ -178,8 +112,9 @@ static const struct snd_kcontrol_new tlv320aic23_snd_controls[] = { TLV320AIC23_RINVOL, 0, 31, 0, input_gain_tlv), SOC_SINGLE("Mic Input Switch", TLV320AIC23_ANLG, 1, 1, 1), SOC_SINGLE("Mic Booster Switch", TLV320AIC23_ANLG, 0, 1, 0), - SOC_TLV320AIC23_SINGLE_TLV("Sidetone Volume", TLV320AIC23_ANLG, - 6, 4, 0, sidetone_vol_tlv), + SOC_SINGLE_EXT_TLV("Sidetone Volume", TLV320AIC23_ANLG, 6, 4, 0, + snd_soc_tlv320aic23_get_volsw, + snd_soc_tlv320aic23_put_volsw, sidetone_vol_tlv), SOC_ENUM("Playback De-emphasis", tlv320aic23_deemph), }; @@ -240,7 +175,6 @@ static const struct snd_soc_dapm_route tlv320aic23_intercon[] = { /* AIC23 driver data */ struct aic23 { enum snd_soc_control_type control_type; - void *control_data; int mclk; int requested_adc; int requested_dac; @@ -352,7 +286,7 @@ static int find_rate(int mclk, u32 need_adc, u32 need_dac) static void get_current_sample_rates(struct snd_soc_codec *codec, int mclk, u32 *sample_rate_adc, u32 *sample_rate_dac) { - int src = tlv320aic23_read_reg_cache(codec, TLV320AIC23_SRATE); + int src = snd_soc_read(codec, TLV320AIC23_SRATE); int sr = (src >> 2) & 0x0f; int val = (mclk / bosr_usb_divisor_table[src & 3]); int adc = (val * sr_adc_mult_table[sr]) / SR_MULT; @@ -376,7 +310,7 @@ static int set_sample_rate_control(struct snd_soc_codec *codec, int mclk, __func__, sample_rate_adc, sample_rate_dac); return -EINVAL; } - tlv320aic23_write(codec, TLV320AIC23_SRATE, data); + snd_soc_write(codec, TLV320AIC23_SRATE, data); #ifdef DEBUG { u32 adc, dac; @@ -415,9 +349,8 @@ static int tlv320aic23_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - iface_reg = - tlv320aic23_read_reg_cache(codec, - TLV320AIC23_DIGT_FMT) & ~(0x03 << 2); + iface_reg = snd_soc_read(codec, TLV320AIC23_DIGT_FMT) & ~(0x03 << 2); + switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: break; @@ -431,7 +364,7 @@ static int tlv320aic23_hw_params(struct snd_pcm_substream *substream, iface_reg |= (0x03 << 2); break; } - tlv320aic23_write(codec, TLV320AIC23_DIGT_FMT, iface_reg); + snd_soc_write(codec, TLV320AIC23_DIGT_FMT, iface_reg); return 0; } @@ -443,7 +376,7 @@ static int tlv320aic23_pcm_prepare(struct snd_pcm_substream *substream, struct snd_soc_codec *codec = rtd->codec; /* set active */ - tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0001); + snd_soc_write(codec, TLV320AIC23_ACTIVE, 0x0001); return 0; } @@ -458,7 +391,7 @@ static void tlv320aic23_shutdown(struct snd_pcm_substream *substream, /* deactivate */ if (!codec->active) { udelay(50); - tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0); + snd_soc_write(codec, TLV320AIC23_ACTIVE, 0x0); } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) aic23->requested_dac = 0; @@ -471,14 +404,14 @@ static int tlv320aic23_mute(struct snd_soc_dai *dai, int mute) struct snd_soc_codec *codec = dai->codec; u16 reg; - reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_DIGT); + reg = snd_soc_read(codec, TLV320AIC23_DIGT); if (mute) reg |= TLV320AIC23_DACM_MUTE; else reg &= ~TLV320AIC23_DACM_MUTE; - tlv320aic23_write(codec, TLV320AIC23_DIGT, reg); + snd_soc_write(codec, TLV320AIC23_DIGT, reg); return 0; } @@ -489,8 +422,7 @@ static int tlv320aic23_set_dai_fmt(struct snd_soc_dai *codec_dai, struct snd_soc_codec *codec = codec_dai->codec; u16 iface_reg; - iface_reg = - tlv320aic23_read_reg_cache(codec, TLV320AIC23_DIGT_FMT) & (~0x03); + iface_reg = snd_soc_read(codec, TLV320AIC23_DIGT_FMT) & (~0x03); /* set master/slave audio interface */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { @@ -524,7 +456,7 @@ static int tlv320aic23_set_dai_fmt(struct snd_soc_dai *codec_dai, } - tlv320aic23_write(codec, TLV320AIC23_DIGT_FMT, iface_reg); + snd_soc_write(codec, TLV320AIC23_DIGT_FMT, iface_reg); return 0; } @@ -540,26 +472,26 @@ static int tlv320aic23_set_dai_sysclk(struct snd_soc_dai *codec_dai, static int tlv320aic23_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - u16 reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_PWR) & 0xff7f; + u16 reg = snd_soc_read(codec, TLV320AIC23_PWR) & 0xff7f; switch (level) { case SND_SOC_BIAS_ON: /* vref/mid, osc on, dac unmute */ reg &= ~(TLV320AIC23_DEVICE_PWR_OFF | TLV320AIC23_OSC_OFF | \ TLV320AIC23_DAC_OFF); - tlv320aic23_write(codec, TLV320AIC23_PWR, reg); + snd_soc_write(codec, TLV320AIC23_PWR, reg); break; case SND_SOC_BIAS_PREPARE: break; case SND_SOC_BIAS_STANDBY: /* everything off except vref/vmid, */ - tlv320aic23_write(codec, TLV320AIC23_PWR, reg | \ - TLV320AIC23_CLK_OFF); + snd_soc_write(codec, TLV320AIC23_PWR, + reg | TLV320AIC23_CLK_OFF); break; case SND_SOC_BIAS_OFF: /* everything off, dac mute, inactive */ - tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0); - tlv320aic23_write(codec, TLV320AIC23_PWR, 0xffff); + snd_soc_write(codec, TLV320AIC23_ACTIVE, 0x0); + snd_soc_write(codec, TLV320AIC23_PWR, 0xffff); break; } codec->dapm.bias_level = level; @@ -606,13 +538,7 @@ static int tlv320aic23_suspend(struct snd_soc_codec *codec, static int tlv320aic23_resume(struct snd_soc_codec *codec) { - u16 reg; - - /* Sync reg_cache with the hardware */ - for (reg = 0; reg <= TLV320AIC23_ACTIVE; reg++) { - u16 val = tlv320aic23_read_reg_cache(codec, reg); - tlv320aic23_write(codec, reg, val); - } + snd_soc_cache_sync(codec); tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_STANDBY); return 0; @@ -621,46 +547,52 @@ static int tlv320aic23_resume(struct snd_soc_codec *codec) static int tlv320aic23_probe(struct snd_soc_codec *codec) { struct aic23 *aic23 = snd_soc_codec_get_drvdata(codec); - int reg; + int ret; printk(KERN_INFO "AIC23 Audio Codec %s\n", AIC23_VERSION); - codec->control_data = aic23->control_data; - codec->hw_write = (hw_write_t)i2c_master_send; - codec->hw_read = NULL; + + ret = snd_soc_codec_set_cache_io(codec, 7, 9, aic23->control_type); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } /* Reset codec */ - tlv320aic23_write(codec, TLV320AIC23_RESET, 0); + snd_soc_write(codec, TLV320AIC23_RESET, 0); + + /* Write the register default value to cache for reserved registers, + * so the write to the these registers are suppressed by the cache + * restore code when it skips writes of default registers. + */ + snd_soc_cache_write(codec, 0x0A, 0); + snd_soc_cache_write(codec, 0x0B, 0); + snd_soc_cache_write(codec, 0x0C, 0); + snd_soc_cache_write(codec, 0x0D, 0); + snd_soc_cache_write(codec, 0x0E, 0); /* power on device */ tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - tlv320aic23_write(codec, TLV320AIC23_DIGT, TLV320AIC23_DEEMP_44K); + snd_soc_write(codec, TLV320AIC23_DIGT, TLV320AIC23_DEEMP_44K); /* Unmute input */ - reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_LINVOL); - tlv320aic23_write(codec, TLV320AIC23_LINVOL, - (reg & (~TLV320AIC23_LIM_MUTED)) | - (TLV320AIC23_LRS_ENABLED)); + snd_soc_update_bits(codec, TLV320AIC23_LINVOL, + TLV320AIC23_LIM_MUTED, TLV320AIC23_LRS_ENABLED); - reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_RINVOL); - tlv320aic23_write(codec, TLV320AIC23_RINVOL, - (reg & (~TLV320AIC23_LIM_MUTED)) | - TLV320AIC23_LRS_ENABLED); + snd_soc_update_bits(codec, TLV320AIC23_RINVOL, + TLV320AIC23_LIM_MUTED, TLV320AIC23_LRS_ENABLED); - reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_ANLG); - tlv320aic23_write(codec, TLV320AIC23_ANLG, - (reg) & (~TLV320AIC23_BYPASS_ON) & - (~TLV320AIC23_MICM_MUTED)); + snd_soc_update_bits(codec, TLV320AIC23_ANLG, + TLV320AIC23_BYPASS_ON | TLV320AIC23_MICM_MUTED, + 0); /* Default output volume */ - tlv320aic23_write(codec, TLV320AIC23_LCHNVOL, - TLV320AIC23_DEFAULT_OUT_VOL & - TLV320AIC23_OUT_VOL_MASK); - tlv320aic23_write(codec, TLV320AIC23_RCHNVOL, - TLV320AIC23_DEFAULT_OUT_VOL & - TLV320AIC23_OUT_VOL_MASK); + snd_soc_write(codec, TLV320AIC23_LCHNVOL, + TLV320AIC23_DEFAULT_OUT_VOL & TLV320AIC23_OUT_VOL_MASK); + snd_soc_write(codec, TLV320AIC23_RCHNVOL, + TLV320AIC23_DEFAULT_OUT_VOL & TLV320AIC23_OUT_VOL_MASK); - tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x1); + snd_soc_write(codec, TLV320AIC23_ACTIVE, 0x1); snd_soc_add_controls(codec, tlv320aic23_snd_controls, ARRAY_SIZE(tlv320aic23_snd_controls)); @@ -682,8 +614,6 @@ static struct snd_soc_codec_driver soc_codec_dev_tlv320aic23 = { .remove = tlv320aic23_remove, .suspend = tlv320aic23_suspend, .resume = tlv320aic23_resume, - .read = tlv320aic23_read_reg_cache, - .write = tlv320aic23_write, .set_bias_level = tlv320aic23_set_bias_level, .dapm_widgets = tlv320aic23_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(tlv320aic23_dapm_widgets), @@ -710,7 +640,6 @@ static int tlv320aic23_codec_probe(struct i2c_client *i2c, return -ENOMEM; i2c_set_clientdata(i2c, aic23); - aic23->control_data = i2c; aic23->control_type = SND_SOC_I2C; ret = snd_soc_register_codec(&i2c->dev, diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index e93b9d1ae1dd..b21c610051c0 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -528,40 +528,33 @@ static int aic32x4_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec); - u8 value; switch (level) { case SND_SOC_BIAS_ON: if (aic32x4->master) { /* Switch on PLL */ - value = snd_soc_read(codec, AIC32X4_PLLPR); - snd_soc_write(codec, AIC32X4_PLLPR, - (value | AIC32X4_PLLEN)); + snd_soc_update_bits(codec, AIC32X4_PLLPR, + AIC32X4_PLLEN, AIC32X4_PLLEN); /* Switch on NDAC Divider */ - value = snd_soc_read(codec, AIC32X4_NDAC); - snd_soc_write(codec, AIC32X4_NDAC, - value | AIC32X4_NDACEN); + snd_soc_update_bits(codec, AIC32X4_NDAC, + AIC32X4_NDACEN, AIC32X4_NDACEN); /* Switch on MDAC Divider */ - value = snd_soc_read(codec, AIC32X4_MDAC); - snd_soc_write(codec, AIC32X4_MDAC, - value | AIC32X4_MDACEN); + snd_soc_update_bits(codec, AIC32X4_MDAC, + AIC32X4_MDACEN, AIC32X4_MDACEN); /* Switch on NADC Divider */ - value = snd_soc_read(codec, AIC32X4_NADC); - snd_soc_write(codec, AIC32X4_NADC, - value | AIC32X4_MDACEN); + snd_soc_update_bits(codec, AIC32X4_NADC, + AIC32X4_NADCEN, AIC32X4_NADCEN); /* Switch on MADC Divider */ - value = snd_soc_read(codec, AIC32X4_MADC); - snd_soc_write(codec, AIC32X4_MADC, - value | AIC32X4_MDACEN); + snd_soc_update_bits(codec, AIC32X4_MADC, + AIC32X4_MADCEN, AIC32X4_MADCEN); /* Switch on BCLK_N Divider */ - value = snd_soc_read(codec, AIC32X4_BCLKN); - snd_soc_write(codec, AIC32X4_BCLKN, - value | AIC32X4_BCLKEN); + snd_soc_update_bits(codec, AIC32X4_BCLKN, + AIC32X4_BCLKEN, AIC32X4_BCLKEN); } break; case SND_SOC_BIAS_PREPARE: @@ -569,34 +562,28 @@ static int aic32x4_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_STANDBY: if (aic32x4->master) { /* Switch off PLL */ - value = snd_soc_read(codec, AIC32X4_PLLPR); - snd_soc_write(codec, AIC32X4_PLLPR, - (value & ~AIC32X4_PLLEN)); + snd_soc_update_bits(codec, AIC32X4_PLLPR, + AIC32X4_PLLEN, 0); /* Switch off NDAC Divider */ - value = snd_soc_read(codec, AIC32X4_NDAC); - snd_soc_write(codec, AIC32X4_NDAC, - value & ~AIC32X4_NDACEN); + snd_soc_update_bits(codec, AIC32X4_NDAC, + AIC32X4_NDACEN, 0); /* Switch off MDAC Divider */ - value = snd_soc_read(codec, AIC32X4_MDAC); - snd_soc_write(codec, AIC32X4_MDAC, - value & ~AIC32X4_MDACEN); + snd_soc_update_bits(codec, AIC32X4_MDAC, + AIC32X4_MDACEN, 0); /* Switch off NADC Divider */ - value = snd_soc_read(codec, AIC32X4_NADC); - snd_soc_write(codec, AIC32X4_NADC, - value & ~AIC32X4_NDACEN); + snd_soc_update_bits(codec, AIC32X4_NADC, + AIC32X4_NADCEN, 0); /* Switch off MADC Divider */ - value = snd_soc_read(codec, AIC32X4_MADC); - snd_soc_write(codec, AIC32X4_MADC, - value & ~AIC32X4_MDACEN); - value = snd_soc_read(codec, AIC32X4_BCLKN); + snd_soc_update_bits(codec, AIC32X4_MADC, + AIC32X4_MADCEN, 0); /* Switch off BCLK_N Divider */ - snd_soc_write(codec, AIC32X4_BCLKN, - value & ~AIC32X4_BCLKEN); + snd_soc_update_bits(codec, AIC32X4_BCLKN, + AIC32X4_BCLKEN, 0); } break; case SND_SOC_BIAS_OFF: @@ -685,10 +672,10 @@ static int aic32x4_probe(struct snd_soc_codec *codec) } /* Mic PGA routing */ - if (aic32x4->micpga_routing | AIC32X4_MICPGA_ROUTE_LMIC_IN2R_10K) { + if (aic32x4->micpga_routing & AIC32X4_MICPGA_ROUTE_LMIC_IN2R_10K) { snd_soc_write(codec, AIC32X4_LMICPGANIN, AIC32X4_LMICPGANIN_IN2R_10K); } - if (aic32x4->micpga_routing | AIC32X4_MICPGA_ROUTE_RMIC_IN1L_10K) { + if (aic32x4->micpga_routing & AIC32X4_MICPGA_ROUTE_RMIC_IN1L_10K) { snd_soc_write(codec, AIC32X4_RMICPGANIN, AIC32X4_RMICPGANIN_IN1L_10K); } diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 0963c4c7a83f..7a49390bc30d 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -76,7 +76,6 @@ struct aic3x_priv { struct aic3x_disable_nb disable_nb[AIC3X_NUM_SUPPLIES]; enum snd_soc_control_type control_type; struct aic3x_setup_data *setup; - void *control_data; unsigned int sysclk; struct list_head list; int master; @@ -138,7 +137,10 @@ static int aic3x_read(struct snd_soc_codec *codec, unsigned int reg, if (reg >= AIC3X_CACHEREGNUM) return -1; - *value = codec->hw_read(codec, reg); + codec->cache_bypass = 1; + *value = snd_soc_read(codec, reg); + codec->cache_bypass = 0; + cache[reg] = *value; return 0; @@ -198,6 +200,10 @@ static int snd_soc_dapm_put_volsw_aic3x(struct snd_kcontrol *kcontrol, else /* old connection must be powered down */ path->connect = invert ? 1 : 0; + + dapm_mark_dirty(path->source, "tlv320aic3x source"); + dapm_mark_dirty(path->sink, "tlv320aic3x sink"); + break; } @@ -1383,7 +1389,6 @@ static int aic3x_probe(struct snd_soc_codec *codec) int ret, i; INIT_LIST_HEAD(&aic3x->list); - codec->control_data = aic3x->control_data; aic3x->codec = codec; codec->dapm.idle_bias_off = 1; @@ -1495,9 +1500,9 @@ static struct snd_soc_codec_driver soc_codec_dev_aic3x = { */ static const struct i2c_device_id aic3x_i2c_id[] = { - [AIC3X_MODEL_3X] = { "tlv320aic3x", 0 }, - [AIC3X_MODEL_33] = { "tlv320aic33", 0 }, - [AIC3X_MODEL_3007] = { "tlv320aic3007", 0 }, + { "tlv320aic3x", AIC3X_MODEL_3X }, + { "tlv320aic33", AIC3X_MODEL_33 }, + { "tlv320aic3007", AIC3X_MODEL_3007 }, { } }; MODULE_DEVICE_TABLE(i2c, aic3x_i2c_id); @@ -1512,7 +1517,6 @@ static int aic3x_i2c_probe(struct i2c_client *i2c, struct aic3x_pdata *pdata = i2c->dev.platform_data; struct aic3x_priv *aic3x; int ret; - const struct i2c_device_id *tbl; aic3x = kzalloc(sizeof(struct aic3x_priv), GFP_KERNEL); if (aic3x == NULL) { @@ -1520,7 +1524,6 @@ static int aic3x_i2c_probe(struct i2c_client *i2c, return -ENOMEM; } - aic3x->control_data = i2c; aic3x->control_type = SND_SOC_I2C; i2c_set_clientdata(i2c, aic3x); @@ -1531,11 +1534,7 @@ static int aic3x_i2c_probe(struct i2c_client *i2c, aic3x->gpio_reset = -1; } - for (tbl = aic3x_i2c_id; tbl->name[0]; tbl++) { - if (!strcmp(tbl->name, id->name)) - break; - } - aic3x->model = tbl - aic3x_i2c_id; + aic3x->model = id->driver_data; ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_aic3x, &aic3x_dai, 1); diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index faa5e9fb1471..dc8a2b2bdc1c 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -55,13 +55,13 @@ #define BURST_BASEFREQ_HZ 49152000 #define SAMPLES_TO_US(rate, samples) \ - (1000000000 / ((rate * 1000) / samples)) + (1000000000 / (((rate) * 1000) / (samples))) #define US_TO_SAMPLES(rate, us) \ - (rate / (1000000 / (us < 1000000 ? us : 1000000))) + ((rate) / (1000000 / ((us) < 1000000 ? (us) : 1000000))) #define UTHR_FROM_PERIOD_SIZE(samples, playrate, burstrate) \ - ((samples * 5000) / ((burstrate * 5000) / (burstrate - playrate))) + (((samples)*5000) / (((burstrate)*5000) / ((burstrate) - (playrate)))) static void dac33_calculate_times(struct snd_pcm_substream *substream); static int dac33_prepare_chip(struct snd_pcm_substream *substream); @@ -627,18 +627,6 @@ static const struct snd_soc_dapm_route audio_map[] = { {"RIGHT_LO", NULL, "Codec Power"}, }; -static int dac33_add_widgets(struct snd_soc_codec *codec) -{ - struct snd_soc_dapm_context *dapm = &codec->dapm; - - snd_soc_dapm_new_controls(dapm, dac33_dapm_widgets, - ARRAY_SIZE(dac33_dapm_widgets)); - /* set up audio path interconnects */ - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - - return 0; -} - static int dac33_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { @@ -1431,7 +1419,7 @@ static int dac33_soc_probe(struct snd_soc_codec *codec) /* Check if the IRQ number is valid and request it */ if (dac33->irq >= 0) { ret = request_irq(dac33->irq, dac33_interrupt_handler, - IRQF_TRIGGER_RISING | IRQF_DISABLED, + IRQF_TRIGGER_RISING, codec->name, codec); if (ret < 0) { dev_err(codec->dev, "Could not request IRQ%d (%d)\n", @@ -1451,15 +1439,11 @@ static int dac33_soc_probe(struct snd_soc_codec *codec) } } - snd_soc_add_controls(codec, dac33_snd_controls, - ARRAY_SIZE(dac33_snd_controls)); /* Only add the FIFO controls, if we have valid IRQ number */ if (dac33->irq >= 0) snd_soc_add_controls(codec, dac33_mode_snd_controls, ARRAY_SIZE(dac33_mode_snd_controls)); - dac33_add_widgets(codec); - err_power: return ret; } @@ -1502,6 +1486,13 @@ static struct snd_soc_codec_driver soc_codec_dev_tlv320dac33 = { .remove = dac33_soc_remove, .suspend = dac33_soc_suspend, .resume = dac33_soc_resume, + + .controls = dac33_snd_controls, + .num_controls = ARRAY_SIZE(dac33_snd_controls), + .dapm_widgets = dac33_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(dac33_dapm_widgets), + .dapm_routes = audio_map, + .num_dapm_routes = ARRAY_SIZE(audio_map), }; #define DAC33_RATES (SNDRV_PCM_RATE_44100 | \ diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c index 239e0c461068..7eeca79d7387 100644 --- a/sound/soc/codecs/tpa6130a2.c +++ b/sound/soc/codecs/tpa6130a2.c @@ -33,6 +33,11 @@ #include "tpa6130a2.h" +enum tpa_model { + TPA6130A2, + TPA6140A2, +}; + static struct i2c_client *tpa6130a2_client; /* This struct is used to save the context */ @@ -383,7 +388,7 @@ static int __devinit tpa6130a2_probe(struct i2c_client *client, pdata = client->dev.platform_data; data->power_gpio = pdata->power_gpio; - data->id = pdata->id; + data->id = id->driver_data; mutex_init(&data->mutex); @@ -405,7 +410,7 @@ static int __devinit tpa6130a2_probe(struct i2c_client *client, switch (data->id) { default: dev_warn(dev, "Unknown TPA model (%d). Assuming 6130A2\n", - pdata->id); + data->id); case TPA6130A2: regulator = "Vdd"; break; @@ -446,7 +451,6 @@ err_regulator: gpio_free(data->power_gpio); err_gpio: kfree(data); - i2c_set_clientdata(tpa6130a2_client, NULL); tpa6130a2_client = NULL; return ret; @@ -470,7 +474,8 @@ static int __devexit tpa6130a2_remove(struct i2c_client *client) } static const struct i2c_device_id tpa6130a2_id[] = { - { "tpa6130a2", 0 }, + { "tpa6130a2", TPA6130A2 }, + { "tpa6140a2", TPA6140A2 }, { } }; MODULE_DEVICE_TABLE(i2c, tpa6130a2_id); diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 71674bec9604..f798247ac1b2 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -863,34 +863,6 @@ static int digimic_event(struct snd_soc_dapm_widget *w, * Inverting not going to help with these. * Custom volsw and volsw_2r get/put functions to handle these gain bits. */ -#define SOC_DOUBLE_TLV_TWL4030(xname, xreg, shift_left, shift_right, xmax,\ - xinvert, tlv_array) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ - .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ - SNDRV_CTL_ELEM_ACCESS_READWRITE,\ - .tlv.p = (tlv_array), \ - .info = snd_soc_info_volsw, \ - .get = snd_soc_get_volsw_twl4030, \ - .put = snd_soc_put_volsw_twl4030, \ - .private_value = (unsigned long)&(struct soc_mixer_control) \ - {.reg = xreg, .shift = shift_left, .rshift = shift_right,\ - .max = xmax, .invert = xinvert} } -#define SOC_DOUBLE_R_TLV_TWL4030(xname, reg_left, reg_right, xshift, xmax,\ - xinvert, tlv_array) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ - .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ - SNDRV_CTL_ELEM_ACCESS_READWRITE,\ - .tlv.p = (tlv_array), \ - .info = snd_soc_info_volsw_2r, \ - .get = snd_soc_get_volsw_r2_twl4030,\ - .put = snd_soc_put_volsw_r2_twl4030, \ - .private_value = (unsigned long)&(struct soc_mixer_control) \ - {.reg = reg_left, .rreg = reg_right, .shift = xshift, \ - .rshift = xshift, .max = xmax, .invert = xinvert} } -#define SOC_SINGLE_TLV_TWL4030(xname, xreg, xshift, xmax, xinvert, tlv_array) \ - SOC_DOUBLE_TLV_TWL4030(xname, xreg, xshift, xshift, xmax, \ - xinvert, tlv_array) - static int snd_soc_get_volsw_twl4030(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -1197,19 +1169,23 @@ static const struct snd_kcontrol_new twl4030_snd_controls[] = { TWL4030_REG_VDL_APGA_CTL, 1, 1, 0), /* Separate output gain controls */ - SOC_DOUBLE_R_TLV_TWL4030("PreDriv Playback Volume", + SOC_DOUBLE_R_EXT_TLV("PreDriv Playback Volume", TWL4030_REG_PREDL_CTL, TWL4030_REG_PREDR_CTL, - 4, 3, 0, output_tvl), + 4, 3, 0, snd_soc_get_volsw_r2_twl4030, + snd_soc_put_volsw_r2_twl4030, output_tvl), - SOC_DOUBLE_TLV_TWL4030("Headset Playback Volume", - TWL4030_REG_HS_GAIN_SET, 0, 2, 3, 0, output_tvl), + SOC_DOUBLE_EXT_TLV("Headset Playback Volume", + TWL4030_REG_HS_GAIN_SET, 0, 2, 3, 0, snd_soc_get_volsw_twl4030, + snd_soc_put_volsw_twl4030, output_tvl), - SOC_DOUBLE_R_TLV_TWL4030("Carkit Playback Volume", + SOC_DOUBLE_R_EXT_TLV("Carkit Playback Volume", TWL4030_REG_PRECKL_CTL, TWL4030_REG_PRECKR_CTL, - 4, 3, 0, output_tvl), + 4, 3, 0, snd_soc_get_volsw_r2_twl4030, + snd_soc_put_volsw_r2_twl4030, output_tvl), - SOC_SINGLE_TLV_TWL4030("Earpiece Playback Volume", - TWL4030_REG_EAR_CTL, 4, 3, 0, output_ear_tvl), + SOC_SINGLE_EXT_TLV("Earpiece Playback Volume", + TWL4030_REG_EAR_CTL, 4, 3, 0, snd_soc_get_volsw_twl4030, + snd_soc_put_volsw_twl4030, output_ear_tvl), /* Common capture gain controls */ SOC_DOUBLE_R_TLV("TX1 Digital Capture Volume", @@ -1633,17 +1609,6 @@ static const struct snd_soc_dapm_route intercon[] = { }; -static int twl4030_add_widgets(struct snd_soc_codec *codec) -{ - struct snd_soc_dapm_context *dapm = &codec->dapm; - - snd_soc_dapm_new_controls(dapm, twl4030_dapm_widgets, - ARRAY_SIZE(twl4030_dapm_widgets)); - snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon)); - - return 0; -} - static int twl4030_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { @@ -2265,9 +2230,6 @@ static int twl4030_soc_probe(struct snd_soc_codec *codec) twl4030_init_chip(codec); - snd_soc_add_controls(codec, twl4030_snd_controls, - ARRAY_SIZE(twl4030_snd_controls)); - twl4030_add_widgets(codec); return 0; } @@ -2293,6 +2255,13 @@ static struct snd_soc_codec_driver soc_codec_dev_twl4030 = { .reg_cache_size = sizeof(twl4030_reg), .reg_word_size = sizeof(u8), .reg_cache_default = twl4030_reg, + + .controls = twl4030_snd_controls, + .num_controls = ARRAY_SIZE(twl4030_snd_controls), + .dapm_widgets = twl4030_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(twl4030_dapm_widgets), + .dapm_routes = intercon, + .num_dapm_routes = ARRAY_SIZE(intercon), }; static int __devinit twl4030_codec_probe(struct platform_device *pdev) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 443032b3b329..73e11f022ded 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -57,6 +57,13 @@ #define TWL6040_HF_VOL_MASK 0x1F #define TWL6040_HF_VOL_SHIFT 0 +/* Shadow register used by the driver */ +#define TWL6040_REG_SW_SHADOW 0x2F +#define TWL6040_CACHEREGNUM (TWL6040_REG_SW_SHADOW + 1) + +/* TWL6040_REG_SW_SHADOW (0x2F) fields */ +#define TWL6040_EAR_PATH_ENABLE 0x01 + struct twl6040_output { u16 active; u16 left_vol; @@ -65,12 +72,13 @@ struct twl6040_output { u16 right_step; unsigned int step_delay; u16 ramp; - u16 mute; + struct delayed_work work; struct completion ramp_done; }; struct twl6040_jack_data { struct snd_soc_jack *jack; + struct delayed_work work; int report; }; @@ -79,7 +87,6 @@ struct twl6040_data { int plug_irq; int codec_powered; int pll; - int non_lp; int pll_power_mode; int hs_power_mode; int hs_power_mode_locked; @@ -92,104 +99,68 @@ struct twl6040_data { struct twl6040_jack_data hs_jack; struct snd_soc_codec *codec; struct workqueue_struct *workqueue; - struct delayed_work delayed_work; struct mutex mutex; struct twl6040_output headset; struct twl6040_output handsfree; - struct workqueue_struct *hf_workqueue; - struct workqueue_struct *hs_workqueue; - struct delayed_work hs_delayed_work; - struct delayed_work hf_delayed_work; }; /* * twl6040 register cache & default register settings */ static const u8 twl6040_reg[TWL6040_CACHEREGNUM] = { - 0x00, /* not used 0x00 */ - 0x4B, /* TWL6040_ASICID (ro) 0x01 */ - 0x00, /* TWL6040_ASICREV (ro) 0x02 */ - 0x00, /* TWL6040_INTID 0x03 */ - 0x00, /* TWL6040_INTMR 0x04 */ - 0x00, /* TWL6040_NCPCTRL 0x05 */ - 0x00, /* TWL6040_LDOCTL 0x06 */ - 0x60, /* TWL6040_HPPLLCTL 0x07 */ - 0x00, /* TWL6040_LPPLLCTL 0x08 */ - 0x4A, /* TWL6040_LPPLLDIV 0x09 */ - 0x00, /* TWL6040_AMICBCTL 0x0A */ - 0x00, /* TWL6040_DMICBCTL 0x0B */ - 0x18, /* TWL6040_MICLCTL 0x0C - No input selected on Left Mic */ - 0x18, /* TWL6040_MICRCTL 0x0D - No input selected on Right Mic */ - 0x00, /* TWL6040_MICGAIN 0x0E */ - 0x1B, /* TWL6040_LINEGAIN 0x0F */ - 0x00, /* TWL6040_HSLCTL 0x10 */ - 0x00, /* TWL6040_HSRCTL 0x11 */ - 0x00, /* TWL6040_HSGAIN 0x12 */ - 0x00, /* TWL6040_EARCTL 0x13 */ - 0x00, /* TWL6040_HFLCTL 0x14 */ - 0x00, /* TWL6040_HFLGAIN 0x15 */ - 0x00, /* TWL6040_HFRCTL 0x16 */ - 0x00, /* TWL6040_HFRGAIN 0x17 */ - 0x00, /* TWL6040_VIBCTLL 0x18 */ - 0x00, /* TWL6040_VIBDATL 0x19 */ - 0x00, /* TWL6040_VIBCTLR 0x1A */ - 0x00, /* TWL6040_VIBDATR 0x1B */ - 0x00, /* TWL6040_HKCTL1 0x1C */ - 0x00, /* TWL6040_HKCTL2 0x1D */ - 0x00, /* TWL6040_GPOCTL 0x1E */ - 0x00, /* TWL6040_ALB 0x1F */ - 0x00, /* TWL6040_DLB 0x20 */ - 0x00, /* not used 0x21 */ - 0x00, /* not used 0x22 */ - 0x00, /* not used 0x23 */ - 0x00, /* not used 0x24 */ - 0x00, /* not used 0x25 */ - 0x00, /* not used 0x26 */ - 0x00, /* not used 0x27 */ - 0x00, /* TWL6040_TRIM1 0x28 */ - 0x00, /* TWL6040_TRIM2 0x29 */ - 0x00, /* TWL6040_TRIM3 0x2A */ - 0x00, /* TWL6040_HSOTRIM 0x2B */ - 0x00, /* TWL6040_HFOTRIM 0x2C */ - 0x09, /* TWL6040_ACCCTL 0x2D */ - 0x00, /* TWL6040_STATUS (ro) 0x2E */ -}; - -/* - * twl6040 vio/gnd registers: - * registers under vio/gnd supply can be accessed - * before the power-up sequence, after NRESPWRON goes high - */ -static const int twl6040_vio_reg[TWL6040_VIOREGNUM] = { - TWL6040_REG_ASICID, - TWL6040_REG_ASICREV, - TWL6040_REG_INTID, - TWL6040_REG_INTMR, - TWL6040_REG_NCPCTL, - TWL6040_REG_LDOCTL, - TWL6040_REG_AMICBCTL, - TWL6040_REG_DMICBCTL, - TWL6040_REG_HKCTL1, - TWL6040_REG_HKCTL2, - TWL6040_REG_GPOCTL, - TWL6040_REG_TRIM1, - TWL6040_REG_TRIM2, - TWL6040_REG_TRIM3, - TWL6040_REG_HSOTRIM, - TWL6040_REG_HFOTRIM, - TWL6040_REG_ACCCTL, - TWL6040_REG_STATUS, + 0x00, /* not used 0x00 */ + 0x4B, /* REG_ASICID 0x01 (ro) */ + 0x00, /* REG_ASICREV 0x02 (ro) */ + 0x00, /* REG_INTID 0x03 */ + 0x00, /* REG_INTMR 0x04 */ + 0x00, /* REG_NCPCTRL 0x05 */ + 0x00, /* REG_LDOCTL 0x06 */ + 0x60, /* REG_HPPLLCTL 0x07 */ + 0x00, /* REG_LPPLLCTL 0x08 */ + 0x4A, /* REG_LPPLLDIV 0x09 */ + 0x00, /* REG_AMICBCTL 0x0A */ + 0x00, /* REG_DMICBCTL 0x0B */ + 0x00, /* REG_MICLCTL 0x0C */ + 0x00, /* REG_MICRCTL 0x0D */ + 0x00, /* REG_MICGAIN 0x0E */ + 0x1B, /* REG_LINEGAIN 0x0F */ + 0x00, /* REG_HSLCTL 0x10 */ + 0x00, /* REG_HSRCTL 0x11 */ + 0x00, /* REG_HSGAIN 0x12 */ + 0x00, /* REG_EARCTL 0x13 */ + 0x00, /* REG_HFLCTL 0x14 */ + 0x00, /* REG_HFLGAIN 0x15 */ + 0x00, /* REG_HFRCTL 0x16 */ + 0x00, /* REG_HFRGAIN 0x17 */ + 0x00, /* REG_VIBCTLL 0x18 */ + 0x00, /* REG_VIBDATL 0x19 */ + 0x00, /* REG_VIBCTLR 0x1A */ + 0x00, /* REG_VIBDATR 0x1B */ + 0x00, /* REG_HKCTL1 0x1C */ + 0x00, /* REG_HKCTL2 0x1D */ + 0x00, /* REG_GPOCTL 0x1E */ + 0x00, /* REG_ALB 0x1F */ + 0x00, /* REG_DLB 0x20 */ + 0x00, /* not used 0x21 */ + 0x00, /* not used 0x22 */ + 0x00, /* not used 0x23 */ + 0x00, /* not used 0x24 */ + 0x00, /* not used 0x25 */ + 0x00, /* not used 0x26 */ + 0x00, /* not used 0x27 */ + 0x00, /* REG_TRIM1 0x28 */ + 0x00, /* REG_TRIM2 0x29 */ + 0x00, /* REG_TRIM3 0x2A */ + 0x00, /* REG_HSOTRIM 0x2B */ + 0x00, /* REG_HFOTRIM 0x2C */ + 0x09, /* REG_ACCCTL 0x2D */ + 0x00, /* REG_STATUS 0x2E (ro) */ + + 0x00, /* REG_SW_SHADOW 0x2F - Shadow, non HW register */ }; -/* - * twl6040 vdd/vss registers: - * registers under vdd/vss supplies can only be accessed - * after the power-up sequence - */ -static const int twl6040_vdd_reg[TWL6040_VDDREGNUM] = { - TWL6040_REG_HPPLLCTL, - TWL6040_REG_LPPLLCTL, - TWL6040_REG_LPPLLDIV, +/* List of registers to be restored after power up */ +static const int twl6040_restore_list[] = { TWL6040_REG_MICLCTL, TWL6040_REG_MICRCTL, TWL6040_REG_MICGAIN, @@ -202,12 +173,6 @@ static const int twl6040_vdd_reg[TWL6040_VDDREGNUM] = { TWL6040_REG_HFLGAIN, TWL6040_REG_HFRCTL, TWL6040_REG_HFRGAIN, - TWL6040_REG_VIBCTLL, - TWL6040_REG_VIBDATL, - TWL6040_REG_VIBCTLR, - TWL6040_REG_VIBDATR, - TWL6040_REG_ALB, - TWL6040_REG_DLB, }; /* set of rates for each pll: low-power and high-performance */ @@ -275,8 +240,12 @@ static int twl6040_read_reg_volatile(struct snd_soc_codec *codec, if (reg >= TWL6040_CACHEREGNUM) return -EIO; - value = twl6040_reg_read(twl6040, reg); - twl6040_write_reg_cache(codec, reg, value); + if (likely(reg < TWL6040_REG_SW_SHADOW)) { + value = twl6040_reg_read(twl6040, reg); + twl6040_write_reg_cache(codec, reg, value); + } else { + value = twl6040_read_reg_cache(codec, reg); + } return value; } @@ -293,59 +262,51 @@ static int twl6040_write(struct snd_soc_codec *codec, return -EIO; twl6040_write_reg_cache(codec, reg, value); - return twl6040_reg_write(twl6040, reg, value); + if (likely(reg < TWL6040_REG_SW_SHADOW)) + return twl6040_reg_write(twl6040, reg, value); + else + return 0; } -static void twl6040_init_vio_regs(struct snd_soc_codec *codec) +static void twl6040_init_chip(struct snd_soc_codec *codec) { - u8 *cache = codec->reg_cache; - int reg, i; - - for (i = 0; i < TWL6040_VIOREGNUM; i++) { - reg = twl6040_vio_reg[i]; - /* - * skip read-only registers (ASICID, ASICREV, STATUS) - * and registers shared among MFD children - */ - switch (reg) { - case TWL6040_REG_ASICID: - case TWL6040_REG_ASICREV: - case TWL6040_REG_INTID: - case TWL6040_REG_INTMR: - case TWL6040_REG_NCPCTL: - case TWL6040_REG_LDOCTL: - case TWL6040_REG_GPOCTL: - case TWL6040_REG_ACCCTL: - case TWL6040_REG_STATUS: - continue; - default: - break; - } - twl6040_write(codec, reg, cache[reg]); - } + struct twl6040 *twl6040 = codec->control_data; + u8 val; + + /* Update reg_cache: ASICREV, and TRIM values */ + val = twl6040_get_revid(twl6040); + twl6040_write_reg_cache(codec, TWL6040_REG_ASICREV, val); + + twl6040_read_reg_volatile(codec, TWL6040_REG_TRIM1); + twl6040_read_reg_volatile(codec, TWL6040_REG_TRIM2); + twl6040_read_reg_volatile(codec, TWL6040_REG_TRIM3); + twl6040_read_reg_volatile(codec, TWL6040_REG_HSOTRIM); + twl6040_read_reg_volatile(codec, TWL6040_REG_HFOTRIM); + + /* Change chip defaults */ + /* No imput selected for microphone amplifiers */ + twl6040_write_reg_cache(codec, TWL6040_REG_MICLCTL, 0x18); + twl6040_write_reg_cache(codec, TWL6040_REG_MICRCTL, 0x18); + + /* + * We need to lower the default gain values, so the ramp code + * can work correctly for the first playback. + * This reduces the pop noise heard at the first playback. + */ + twl6040_write_reg_cache(codec, TWL6040_REG_HSGAIN, 0xff); + twl6040_write_reg_cache(codec, TWL6040_REG_EARCTL, 0x1e); + twl6040_write_reg_cache(codec, TWL6040_REG_HFLGAIN, 0x1d); + twl6040_write_reg_cache(codec, TWL6040_REG_HFRGAIN, 0x1d); + twl6040_write_reg_cache(codec, TWL6040_REG_LINEGAIN, 0); } -static void twl6040_init_vdd_regs(struct snd_soc_codec *codec) +static void twl6040_restore_regs(struct snd_soc_codec *codec) { u8 *cache = codec->reg_cache; int reg, i; - for (i = 0; i < TWL6040_VDDREGNUM; i++) { - reg = twl6040_vdd_reg[i]; - /* skip vibra and PLL registers */ - switch (reg) { - case TWL6040_REG_VIBCTLL: - case TWL6040_REG_VIBDATL: - case TWL6040_REG_VIBCTLR: - case TWL6040_REG_VIBDATR: - case TWL6040_REG_HPPLLCTL: - case TWL6040_REG_LPPLLCTL: - case TWL6040_REG_LPPLLDIV: - continue; - default: - break; - } - + for (i = 0; i < ARRAY_SIZE(twl6040_restore_list); i++) { + reg = twl6040_restore_list[i]; twl6040_write(codec, reg, cache[reg]); } } @@ -524,18 +485,17 @@ static inline int twl6040_hf_ramp_step(struct snd_soc_codec *codec, static void twl6040_pga_hs_work(struct work_struct *work) { struct twl6040_data *priv = - container_of(work, struct twl6040_data, hs_delayed_work.work); + container_of(work, struct twl6040_data, headset.work.work); struct snd_soc_codec *codec = priv->codec; struct twl6040_output *headset = &priv->headset; - unsigned int delay = headset->step_delay; int i, headset_complete; /* do we need to ramp at all ? */ if (headset->ramp == TWL6040_RAMP_NONE) return; - /* HS PGA volumes have 4 bits of resolution to ramp */ - for (i = 0; i <= 16; i++) { + /* HS PGA gain range: 0x0 - 0xf (0 - 15) */ + for (i = 0; i < 16; i++) { headset_complete = twl6040_hs_ramp_step(codec, headset->left_step, headset->right_step); @@ -544,15 +504,8 @@ static void twl6040_pga_hs_work(struct work_struct *work) if (headset_complete) break; - /* - * TODO: tune: delay is longer over 0dB - * as increases are larger. - */ - if (i >= 8) - schedule_timeout_interruptible(msecs_to_jiffies(delay + - (delay >> 1))); - else - schedule_timeout_interruptible(msecs_to_jiffies(delay)); + schedule_timeout_interruptible( + msecs_to_jiffies(headset->step_delay)); } if (headset->ramp == TWL6040_RAMP_DOWN) { @@ -567,18 +520,18 @@ static void twl6040_pga_hs_work(struct work_struct *work) static void twl6040_pga_hf_work(struct work_struct *work) { struct twl6040_data *priv = - container_of(work, struct twl6040_data, hf_delayed_work.work); + container_of(work, struct twl6040_data, handsfree.work.work); struct snd_soc_codec *codec = priv->codec; struct twl6040_output *handsfree = &priv->handsfree; - unsigned int delay = handsfree->step_delay; int i, handsfree_complete; /* do we need to ramp at all ? */ if (handsfree->ramp == TWL6040_RAMP_NONE) return; - /* HF PGA volumes have 5 bits of resolution to ramp */ - for (i = 0; i <= 32; i++) { + /* + * HF PGA gain range: 0x00 - 0x1d (0 - 29) */ + for (i = 0; i < 30; i++) { handsfree_complete = twl6040_hf_ramp_step(codec, handsfree->left_step, handsfree->right_step); @@ -587,15 +540,8 @@ static void twl6040_pga_hf_work(struct work_struct *work) if (handsfree_complete) break; - /* - * TODO: tune: delay is longer over 0dB - * as increases are larger. - */ - if (i >= 16) - schedule_timeout_interruptible(msecs_to_jiffies(delay + - (delay >> 1))); - else - schedule_timeout_interruptible(msecs_to_jiffies(delay)); + schedule_timeout_interruptible( + msecs_to_jiffies(handsfree->step_delay)); } @@ -607,36 +553,40 @@ static void twl6040_pga_hf_work(struct work_struct *work) handsfree->ramp = TWL6040_RAMP_NONE; } -static int pga_event(struct snd_soc_dapm_widget *w, +static int out_drv_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_codec *codec = w->codec; struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); struct twl6040_output *out; struct delayed_work *work; - struct workqueue_struct *queue; switch (w->shift) { - case 2: - case 3: + case 2: /* Headset output driver */ out = &priv->headset; - work = &priv->hs_delayed_work; - queue = priv->hs_workqueue; + work = &out->work; + /* + * Make sure, that we do not mess up variables for already + * executing work. + */ + cancel_delayed_work_sync(work); + out->left_step = priv->hs_left_step; out->right_step = priv->hs_right_step; out->step_delay = 5; /* 5 ms between volume ramp steps */ break; - case 4: + case 4: /* Handsfree output driver */ out = &priv->handsfree; - work = &priv->hf_delayed_work; - queue = priv->hf_workqueue; + work = &out->work; + /* + * Make sure, that we do not mess up variables for already + * executing work. + */ + cancel_delayed_work_sync(work); + out->left_step = priv->hf_left_step; out->right_step = priv->hf_right_step; out->step_delay = 5; /* 5 ms between volume ramp steps */ - if (SND_SOC_DAPM_EVENT_ON(event)) - priv->non_lp++; - else - priv->non_lp--; break; default: return -1; @@ -648,31 +598,25 @@ static int pga_event(struct snd_soc_dapm_widget *w, break; /* don't use volume ramp for power-up */ + out->ramp = TWL6040_RAMP_UP; out->left_step = out->left_vol; out->right_step = out->right_vol; - if (!delayed_work_pending(work)) { - out->ramp = TWL6040_RAMP_UP; - queue_delayed_work(queue, work, - msecs_to_jiffies(1)); - } + queue_delayed_work(priv->workqueue, work, msecs_to_jiffies(1)); break; case SND_SOC_DAPM_PRE_PMD: if (!out->active) break; - if (!delayed_work_pending(work)) { - /* use volume ramp for power-down */ - out->ramp = TWL6040_RAMP_DOWN; - INIT_COMPLETION(out->ramp_done); + /* use volume ramp for power-down */ + out->ramp = TWL6040_RAMP_DOWN; + INIT_COMPLETION(out->ramp_done); - queue_delayed_work(queue, work, - msecs_to_jiffies(1)); + queue_delayed_work(priv->workqueue, work, msecs_to_jiffies(1)); - wait_for_completion_timeout(&out->ramp_done, - msecs_to_jiffies(2000)); - } + wait_for_completion_timeout(&out->ramp_done, + msecs_to_jiffies(2000)); break; } @@ -683,7 +627,7 @@ static int pga_event(struct snd_soc_dapm_widget *w, static int headset_power_mode(struct snd_soc_codec *codec, int high_perf) { int hslctl, hsrctl; - int mask = TWL6040_HSDRVMODEL | TWL6040_HSDACMODEL; + int mask = TWL6040_HSDRVMODE | TWL6040_HSDACMODE; hslctl = twl6040_read_reg_cache(codec, TWL6040_REG_HSLCTL); hsrctl = twl6040_read_reg_cache(codec, TWL6040_REG_HSRCTL); @@ -705,11 +649,31 @@ static int headset_power_mode(struct snd_soc_codec *codec, int high_perf) static int twl6040_hs_dac_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { + struct snd_soc_codec *codec = w->codec; + u8 hslctl, hsrctl; + + /* + * Workaround for Headset DC offset caused pop noise: + * Both HS DAC need to be turned on (before the HS driver) and off at + * the same time. + */ + hslctl = twl6040_read_reg_cache(codec, TWL6040_REG_HSLCTL); + hsrctl = twl6040_read_reg_cache(codec, TWL6040_REG_HSRCTL); + if (SND_SOC_DAPM_EVENT_ON(event)) { + hslctl |= TWL6040_HSDACENA; + hsrctl |= TWL6040_HSDACENA; + } else { + hslctl &= ~TWL6040_HSDACENA; + hsrctl &= ~TWL6040_HSDACENA; + } + twl6040_write(codec, TWL6040_REG_HSLCTL, hslctl); + twl6040_write(codec, TWL6040_REG_HSRCTL, hsrctl); + msleep(1); return 0; } -static int twl6040_power_mode_event(struct snd_soc_dapm_widget *w, +static int twl6040_ep_drv_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_codec *codec = w->codec; @@ -717,18 +681,12 @@ static int twl6040_power_mode_event(struct snd_soc_dapm_widget *w, int ret = 0; if (SND_SOC_DAPM_EVENT_ON(event)) { - priv->non_lp++; - if (!strcmp(w->name, "Earphone Driver")) { - /* Earphone doesn't support low power mode */ - priv->hs_power_mode_locked = 1; - ret = headset_power_mode(codec, 1); - } + /* Earphone doesn't support low power mode */ + priv->hs_power_mode_locked = 1; + ret = headset_power_mode(codec, 1); } else { - priv->non_lp--; - if (!strcmp(w->name, "Earphone Driver")) { - priv->hs_power_mode_locked = 0; - ret = headset_power_mode(codec, priv->hs_power_mode); - } + priv->hs_power_mode_locked = 0; + ret = headset_power_mode(codec, priv->hs_power_mode); } msleep(1); @@ -770,7 +728,7 @@ EXPORT_SYMBOL_GPL(twl6040_hs_jack_detect); static void twl6040_accessory_work(struct work_struct *work) { struct twl6040_data *priv = container_of(work, - struct twl6040_data, delayed_work.work); + struct twl6040_data, hs_jack.work.work); struct snd_soc_codec *codec = priv->codec; struct twl6040_jack_data *hs_jack = &priv->hs_jack; @@ -781,15 +739,10 @@ static void twl6040_accessory_work(struct work_struct *work) static irqreturn_t twl6040_audio_handler(int irq, void *data) { struct snd_soc_codec *codec = data; - struct twl6040 *twl6040 = codec->control_data; struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); - u8 intid; - - intid = twl6040_reg_read(twl6040, TWL6040_REG_INTID); - if ((intid & TWL6040_PLUGINT) || (intid & TWL6040_UNPLUGINT)) - queue_delayed_work(priv->workqueue, &priv->delayed_work, - msecs_to_jiffies(200)); + queue_delayed_work(priv->workqueue, &priv->hs_jack.work, + msecs_to_jiffies(200)); return IRQ_HANDLED; } @@ -803,25 +756,27 @@ static int twl6040_put_volsw(struct snd_kcontrol *kcontrol, struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; int ret; - unsigned int reg = mc->reg; /* For HS and HF we shadow the values and only actually write * them out when active in order to ensure the amplifier comes on * as quietly as possible. */ - switch (reg) { + switch (mc->reg) { case TWL6040_REG_HSGAIN: out = &twl6040_priv->headset; break; - default: + case TWL6040_REG_HFLGAIN: + out = &twl6040_priv->handsfree; break; + default: + dev_warn(codec->dev, "%s: Unexpected register: 0x%02x\n", + __func__, mc->reg); + return -EINVAL; } - if (out) { - out->left_vol = ucontrol->value.integer.value[0]; - out->right_vol = ucontrol->value.integer.value[1]; - if (!out->active) - return 1; - } + out->left_vol = ucontrol->value.integer.value[0]; + out->right_vol = ucontrol->value.integer.value[1]; + if (!out->active) + return 1; ret = snd_soc_put_volsw(kcontrol, ucontrol); if (ret < 0) @@ -838,112 +793,42 @@ static int twl6040_get_volsw(struct snd_kcontrol *kcontrol, struct twl6040_output *out = &twl6040_priv->headset; struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; - unsigned int reg = mc->reg; - switch (reg) { + switch (mc->reg) { case TWL6040_REG_HSGAIN: out = &twl6040_priv->headset; - ucontrol->value.integer.value[0] = out->left_vol; - ucontrol->value.integer.value[1] = out->right_vol; - return 0; - - default: break; - } - - return snd_soc_get_volsw(kcontrol, ucontrol); -} - -static int twl6040_put_volsw_2r_vu(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - struct twl6040_data *twl6040_priv = snd_soc_codec_get_drvdata(codec); - struct twl6040_output *out = NULL; - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - int ret; - unsigned int reg = mc->reg; - - /* For HS and HF we shadow the values and only actually write - * them out when active in order to ensure the amplifier comes on - * as quietly as possible. */ - switch (reg) { case TWL6040_REG_HFLGAIN: - case TWL6040_REG_HFRGAIN: out = &twl6040_priv->handsfree; break; default: - break; - } - - if (out) { - out->left_vol = ucontrol->value.integer.value[0]; - out->right_vol = ucontrol->value.integer.value[1]; - if (!out->active) - return 1; + dev_warn(codec->dev, "%s: Unexpected register: 0x%02x\n", + __func__, mc->reg); + return -EINVAL; } - ret = snd_soc_put_volsw_2r(kcontrol, ucontrol); - if (ret < 0) - return ret; - - return 1; + ucontrol->value.integer.value[0] = out->left_vol; + ucontrol->value.integer.value[1] = out->right_vol; + return 0; } -static int twl6040_get_volsw_2r(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) +static int twl6040_soc_dapm_put_vibra_enum(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - struct twl6040_data *twl6040_priv = snd_soc_codec_get_drvdata(codec); - struct twl6040_output *out = &twl6040_priv->handsfree; - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - unsigned int reg = mc->reg; - - /* If these are cached registers use the cache */ - switch (reg) { - case TWL6040_REG_HFLGAIN: - case TWL6040_REG_HFRGAIN: - out = &twl6040_priv->handsfree; - ucontrol->value.integer.value[0] = out->left_vol; - ucontrol->value.integer.value[1] = out->right_vol; - return 0; - - default: - break; - } - - return snd_soc_get_volsw_2r(kcontrol, ucontrol); + struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_widget *widget = wlist->widgets[0]; + struct snd_soc_codec *codec = widget->codec; + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned int val; + + /* Do not allow changes while Input/FF efect is running */ + val = twl6040_read_reg_volatile(codec, e->reg); + if (val & TWL6040_VIBENA && !(val & TWL6040_VIBSEL)) + return -EBUSY; + + return snd_soc_dapm_put_enum_double(kcontrol, ucontrol); } -/* double control with volume update */ -#define SOC_TWL6040_DOUBLE_TLV(xname, xreg, shift_left, shift_right, xmax,\ - xinvert, tlv_array)\ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ - .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ - SNDRV_CTL_ELEM_ACCESS_READWRITE,\ - .tlv.p = (tlv_array), \ - .info = snd_soc_info_volsw, .get = twl6040_get_volsw, \ - .put = twl6040_put_volsw, \ - .private_value = (unsigned long)&(struct soc_mixer_control) \ - {.reg = xreg, .shift = shift_left, .rshift = shift_right,\ - .max = xmax, .platform_max = xmax, .invert = xinvert} } - -/* double control with volume update */ -#define SOC_TWL6040_DOUBLE_R_TLV(xname, reg_left, reg_right, xshift, xmax,\ - xinvert, tlv_array)\ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ - .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ - SNDRV_CTL_ELEM_ACCESS_READWRITE | \ - SNDRV_CTL_ELEM_ACCESS_VOLATILE, \ - .tlv.p = (tlv_array), \ - .info = snd_soc_info_volsw_2r, \ - .get = twl6040_get_volsw_2r, .put = twl6040_put_volsw_2r_vu, \ - .private_value = (unsigned long)&(struct soc_mixer_control) \ - {.reg = reg_left, .rreg = reg_right, .shift = xshift, \ - .rshift = xshift, .max = xmax, .invert = xinvert}, } - /* * MICATT volume control: * from -6 to 0 dB in 6 dB steps @@ -1015,6 +900,19 @@ static const struct soc_enum twl6040_hf_enum[] = { twl6040_hf_texts), }; +static const char *twl6040_vibrapath_texts[] = { + "Input FF", "Audio PDM" +}; + +static const struct soc_enum twl6040_vibra_enum[] = { + SOC_ENUM_SINGLE(TWL6040_REG_VIBCTLL, 1, + ARRAY_SIZE(twl6040_vibrapath_texts), + twl6040_vibrapath_texts), + SOC_ENUM_SINGLE(TWL6040_REG_VIBCTLR, 1, + ARRAY_SIZE(twl6040_vibrapath_texts), + twl6040_vibrapath_texts), +}; + static const struct snd_kcontrol_new amicl_control = SOC_DAPM_ENUM("Route", twl6040_enum[0]); @@ -1035,8 +933,25 @@ static const struct snd_kcontrol_new hfl_mux_controls = static const struct snd_kcontrol_new hfr_mux_controls = SOC_DAPM_ENUM("Route", twl6040_hf_enum[1]); -static const struct snd_kcontrol_new ep_driver_switch_controls = - SOC_DAPM_SINGLE("Switch", TWL6040_REG_EARCTL, 0, 1, 0); +static const struct snd_kcontrol_new ep_path_enable_control = + SOC_DAPM_SINGLE("Switch", TWL6040_REG_SW_SHADOW, 0, 1, 0); + +static const struct snd_kcontrol_new auxl_switch_control = + SOC_DAPM_SINGLE("Switch", TWL6040_REG_HFLCTL, 6, 1, 0); + +static const struct snd_kcontrol_new auxr_switch_control = + SOC_DAPM_SINGLE("Switch", TWL6040_REG_HFRCTL, 6, 1, 0); + +/* Vibra playback switches */ +static const struct snd_kcontrol_new vibral_mux_controls = + SOC_DAPM_ENUM_EXT("Route", twl6040_vibra_enum[0], + snd_soc_dapm_get_enum_double, + twl6040_soc_dapm_put_vibra_enum); + +static const struct snd_kcontrol_new vibrar_mux_controls = + SOC_DAPM_ENUM_EXT("Route", twl6040_vibra_enum[1], + snd_soc_dapm_get_enum_double, + twl6040_soc_dapm_put_vibra_enum); /* Headset power mode */ static const char *twl6040_power_mode_texts[] = { @@ -1105,6 +1020,15 @@ int twl6040_get_clk_id(struct snd_soc_codec *codec) } EXPORT_SYMBOL_GPL(twl6040_get_clk_id); +int twl6040_get_trim_value(struct snd_soc_codec *codec, enum twl6040_trim trim) +{ + if (unlikely(trim >= TWL6040_TRIM_INVAL)) + return -EINVAL; + + return twl6040_read_reg_cache(codec, TWL6040_REG_TRIM1 + trim); +} +EXPORT_SYMBOL_GPL(twl6040_get_trim_value); + static const struct snd_kcontrol_new twl6040_snd_controls[] = { /* Capture gains */ SOC_DOUBLE_TLV("Capture Preamplifier Volume", @@ -1117,10 +1041,12 @@ static const struct snd_kcontrol_new twl6040_snd_controls[] = { TWL6040_REG_LINEGAIN, 0, 3, 7, 0, afm_amp_tlv), /* Playback gains */ - SOC_TWL6040_DOUBLE_TLV("Headset Playback Volume", - TWL6040_REG_HSGAIN, 0, 4, 0xF, 1, hs_tlv), - SOC_TWL6040_DOUBLE_R_TLV("Handsfree Playback Volume", - TWL6040_REG_HFLGAIN, TWL6040_REG_HFRGAIN, 0, 0x1D, 1, hf_tlv), + SOC_DOUBLE_EXT_TLV("Headset Playback Volume", + TWL6040_REG_HSGAIN, 0, 4, 0xF, 1, twl6040_get_volsw, + twl6040_put_volsw, hs_tlv), + SOC_DOUBLE_R_EXT_TLV("Handsfree Playback Volume", + TWL6040_REG_HFLGAIN, TWL6040_REG_HFRGAIN, 0, 0x1D, 1, + twl6040_get_volsw, twl6040_put_volsw, hf_tlv), SOC_SINGLE_TLV("Earphone Playback Volume", TWL6040_REG_EARCTL, 1, 0xF, 1, ep_tlv), @@ -1146,6 +1072,10 @@ static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("HFL"), SND_SOC_DAPM_OUTPUT("HFR"), SND_SOC_DAPM_OUTPUT("EP"), + SND_SOC_DAPM_OUTPUT("AUXL"), + SND_SOC_DAPM_OUTPUT("AUXR"), + SND_SOC_DAPM_OUTPUT("VIBRAL"), + SND_SOC_DAPM_OUTPUT("VIBRAR"), /* Analog input muxes for the capture amplifiers */ SND_SOC_DAPM_MUX("Analog Left Capture Route", @@ -1182,59 +1112,76 @@ static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = { TWL6040_REG_DMICBCTL, 4, 0), /* DACs */ - SND_SOC_DAPM_DAC_E("HSDAC Left", "Headset Playback", - TWL6040_REG_HSLCTL, 0, 0, - twl6040_hs_dac_event, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), - SND_SOC_DAPM_DAC_E("HSDAC Right", "Headset Playback", - TWL6040_REG_HSRCTL, 0, 0, - twl6040_hs_dac_event, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), - SND_SOC_DAPM_DAC_E("HFDAC Left", "Handsfree Playback", - TWL6040_REG_HFLCTL, 0, 0, - twl6040_power_mode_event, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), - SND_SOC_DAPM_DAC_E("HFDAC Right", "Handsfree Playback", - TWL6040_REG_HFRCTL, 0, 0, - twl6040_power_mode_event, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), - - SND_SOC_DAPM_MUX("HF Left Playback", + SND_SOC_DAPM_DAC("HSDAC Left", "Headset Playback", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("HSDAC Right", "Headset Playback", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("HFDAC Left", "Handsfree Playback", + TWL6040_REG_HFLCTL, 0, 0), + SND_SOC_DAPM_DAC("HFDAC Right", "Handsfree Playback", + TWL6040_REG_HFRCTL, 0, 0), + /* Virtual DAC for vibra path (DL4 channel) */ + SND_SOC_DAPM_DAC("VIBRA DAC", "Vibra Playback", + SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_MUX("Handsfree Left Playback", SND_SOC_NOPM, 0, 0, &hfl_mux_controls), - SND_SOC_DAPM_MUX("HF Right Playback", + SND_SOC_DAPM_MUX("Handsfree Right Playback", SND_SOC_NOPM, 0, 0, &hfr_mux_controls), /* Analog playback Muxes */ - SND_SOC_DAPM_MUX("HS Left Playback", + SND_SOC_DAPM_MUX("Headset Left Playback", SND_SOC_NOPM, 0, 0, &hsl_mux_controls), - SND_SOC_DAPM_MUX("HS Right Playback", + SND_SOC_DAPM_MUX("Headset Right Playback", SND_SOC_NOPM, 0, 0, &hsr_mux_controls), + SND_SOC_DAPM_MUX("Vibra Left Playback", SND_SOC_NOPM, 0, 0, + &vibral_mux_controls), + SND_SOC_DAPM_MUX("Vibra Right Playback", SND_SOC_NOPM, 0, 0, + &vibrar_mux_controls), + + SND_SOC_DAPM_SWITCH("Earphone Playback", SND_SOC_NOPM, 0, 0, + &ep_path_enable_control), + SND_SOC_DAPM_SWITCH("AUXL Playback", SND_SOC_NOPM, 0, 0, + &auxl_switch_control), + SND_SOC_DAPM_SWITCH("AUXR Playback", SND_SOC_NOPM, 0, 0, + &auxr_switch_control), + /* Analog playback drivers */ - SND_SOC_DAPM_OUT_DRV_E("Handsfree Left Driver", + SND_SOC_DAPM_OUT_DRV_E("HF Left Driver", TWL6040_REG_HFLCTL, 4, 0, NULL, 0, - pga_event, + out_drv_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), - SND_SOC_DAPM_OUT_DRV_E("Handsfree Right Driver", + SND_SOC_DAPM_OUT_DRV_E("HF Right Driver", TWL6040_REG_HFRCTL, 4, 0, NULL, 0, - pga_event, + out_drv_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), - SND_SOC_DAPM_OUT_DRV_E("Headset Left Driver", + SND_SOC_DAPM_OUT_DRV_E("HS Left Driver", TWL6040_REG_HSLCTL, 2, 0, NULL, 0, - pga_event, + out_drv_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), - SND_SOC_DAPM_OUT_DRV_E("Headset Right Driver", + SND_SOC_DAPM_OUT_DRV_E("HS Right Driver", TWL6040_REG_HSRCTL, 2, 0, NULL, 0, - pga_event, + out_drv_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), - SND_SOC_DAPM_SWITCH_E("Earphone Driver", - SND_SOC_NOPM, 0, 0, &ep_driver_switch_controls, - twl6040_power_mode_event, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_OUT_DRV_E("Earphone Driver", + TWL6040_REG_EARCTL, 0, 0, NULL, 0, + twl6040_ep_drv_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_OUT_DRV("Vibra Left Driver", + TWL6040_REG_VIBCTLL, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("Vibra Right Driver", + TWL6040_REG_VIBCTLR, 0, 0, NULL, 0), + + SND_SOC_DAPM_SUPPLY("Vibra Left Control", TWL6040_REG_VIBCTLL, 2, 0, + NULL, 0), + SND_SOC_DAPM_SUPPLY("Vibra Right Control", TWL6040_REG_VIBCTLR, 2, 0, + NULL, 0), + SND_SOC_DAPM_SUPPLY_S("HSDAC Power", 1, SND_SOC_NOPM, 0, 0, + twl6040_hs_dac_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), /* Analog playback PGAs */ - SND_SOC_DAPM_PGA("HFDAC Left PGA", + SND_SOC_DAPM_PGA("HF Left PGA", TWL6040_REG_HFLCTL, 1, 0, NULL, 0), - SND_SOC_DAPM_PGA("HFDAC Right PGA", + SND_SOC_DAPM_PGA("HF Right PGA", TWL6040_REG_HFRCTL, 1, 0, NULL, 0), }; @@ -1256,52 +1203,62 @@ static const struct snd_soc_dapm_route intercon[] = { {"ADC Right", NULL, "MicAmpR"}, /* AFM path */ - {"AFMAmpL", "NULL", "AFML"}, - {"AFMAmpR", "NULL", "AFMR"}, + {"AFMAmpL", NULL, "AFML"}, + {"AFMAmpR", NULL, "AFMR"}, + + {"HSDAC Left", NULL, "HSDAC Power"}, + {"HSDAC Right", NULL, "HSDAC Power"}, - {"HS Left Playback", "HS DAC", "HSDAC Left"}, - {"HS Left Playback", "Line-In amp", "AFMAmpL"}, + {"Headset Left Playback", "HS DAC", "HSDAC Left"}, + {"Headset Left Playback", "Line-In amp", "AFMAmpL"}, - {"HS Right Playback", "HS DAC", "HSDAC Right"}, - {"HS Right Playback", "Line-In amp", "AFMAmpR"}, + {"Headset Right Playback", "HS DAC", "HSDAC Right"}, + {"Headset Right Playback", "Line-In amp", "AFMAmpR"}, - {"Headset Left Driver", "NULL", "HS Left Playback"}, - {"Headset Right Driver", "NULL", "HS Right Playback"}, + {"HS Left Driver", NULL, "Headset Left Playback"}, + {"HS Right Driver", NULL, "Headset Right Playback"}, - {"HSOL", NULL, "Headset Left Driver"}, - {"HSOR", NULL, "Headset Right Driver"}, + {"HSOL", NULL, "HS Left Driver"}, + {"HSOR", NULL, "HS Right Driver"}, /* Earphone playback path */ - {"Earphone Driver", "Switch", "HSDAC Left"}, + {"Earphone Playback", "Switch", "HSDAC Left"}, + {"Earphone Driver", NULL, "Earphone Playback"}, {"EP", NULL, "Earphone Driver"}, - {"HF Left Playback", "HF DAC", "HFDAC Left"}, - {"HF Left Playback", "Line-In amp", "AFMAmpL"}, + {"Handsfree Left Playback", "HF DAC", "HFDAC Left"}, + {"Handsfree Left Playback", "Line-In amp", "AFMAmpL"}, - {"HF Right Playback", "HF DAC", "HFDAC Right"}, - {"HF Right Playback", "Line-In amp", "AFMAmpR"}, + {"Handsfree Right Playback", "HF DAC", "HFDAC Right"}, + {"Handsfree Right Playback", "Line-In amp", "AFMAmpR"}, - {"HFDAC Left PGA", NULL, "HF Left Playback"}, - {"HFDAC Right PGA", NULL, "HF Right Playback"}, + {"HF Left PGA", NULL, "Handsfree Left Playback"}, + {"HF Right PGA", NULL, "Handsfree Right Playback"}, - {"Handsfree Left Driver", "Switch", "HFDAC Left PGA"}, - {"Handsfree Right Driver", "Switch", "HFDAC Right PGA"}, + {"HF Left Driver", NULL, "HF Left PGA"}, + {"HF Right Driver", NULL, "HF Right PGA"}, - {"HFL", NULL, "Handsfree Left Driver"}, - {"HFR", NULL, "Handsfree Right Driver"}, -}; + {"HFL", NULL, "HF Left Driver"}, + {"HFR", NULL, "HF Right Driver"}, -static int twl6040_add_widgets(struct snd_soc_codec *codec) -{ - struct snd_soc_dapm_context *dapm = &codec->dapm; + {"AUXL Playback", "Switch", "HF Left PGA"}, + {"AUXR Playback", "Switch", "HF Right PGA"}, - snd_soc_dapm_new_controls(dapm, twl6040_dapm_widgets, - ARRAY_SIZE(twl6040_dapm_widgets)); - snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon)); - snd_soc_dapm_new_widgets(dapm); + {"AUXL", NULL, "AUXL Playback"}, + {"AUXR", NULL, "AUXR Playback"}, - return 0; -} + /* Vibrator paths */ + {"Vibra Left Playback", "Audio PDM", "VIBRA DAC"}, + {"Vibra Right Playback", "Audio PDM", "VIBRA DAC"}, + + {"Vibra Left Driver", NULL, "Vibra Left Playback"}, + {"Vibra Right Driver", NULL, "Vibra Right Playback"}, + {"Vibra Left Driver", NULL, "Vibra Left Control"}, + {"Vibra Right Driver", NULL, "Vibra Right Control"}, + + {"VIBRAL", NULL, "Vibra Left Driver"}, + {"VIBRAR", NULL, "Vibra Right Driver"}, +}; static int twl6040_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) @@ -1325,8 +1282,7 @@ static int twl6040_set_bias_level(struct snd_soc_codec *codec, priv->codec_powered = 1; - /* initialize vdd/vss registers with reg_cache */ - twl6040_init_vdd_regs(codec); + twl6040_restore_regs(codec); /* Set external boost GPO */ twl6040_write(codec, TWL6040_REG_GPOCTL, 0x02); @@ -1380,13 +1336,6 @@ static int twl6040_hw_params(struct snd_pcm_substream *substream, rate); return -EINVAL; } - /* Capture is not supported with 17.64MHz sysclk */ - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { - dev_err(codec->dev, - "capture mode is not supported at %dHz\n", - rate); - return -EINVAL; - } priv->sysclk = 17640000; break; case 8000: @@ -1419,13 +1368,6 @@ static int twl6040_prepare(struct snd_pcm_substream *substream, return -EINVAL; } - if ((priv->sysclk == 17640000) && priv->non_lp) { - dev_err(codec->dev, - "some enabled paths aren't supported at %dHz\n", - priv->sysclk); - return -EPERM; - } - ret = twl6040_set_pll(twl6040, priv->pll, priv->clk_in, priv->sysclk); if (ret) { dev_err(codec->dev, "Can not set PLL (%d)\n", ret); @@ -1464,11 +1406,11 @@ static struct snd_soc_dai_ops twl6040_dai_ops = { static struct snd_soc_dai_driver twl6040_dai[] = { { - .name = "twl6040-hifi", + .name = "twl6040-legacy", .playback = { .stream_name = "Playback", .channels_min = 1, - .channels_max = 2, + .channels_max = 5, .rates = TWL6040_RATES, .formats = TWL6040_FORMATS, }, @@ -1518,8 +1460,8 @@ static struct snd_soc_dai_driver twl6040_dai[] = { .name = "twl6040-vib", .playback = { .stream_name = "Vibra Playback", - .channels_min = 2, - .channels_max = 2, + .channels_min = 1, + .channels_max = 1, .rates = SNDRV_PCM_RATE_CONTINUOUS, .formats = TWL6040_FORMATS, }, @@ -1562,6 +1504,7 @@ static int twl6040_probe(struct snd_soc_codec *codec) priv->codec = codec; codec->control_data = dev_get_drvdata(codec->dev->parent); + codec->ignore_pmdown_time = 1; if (pdata && pdata->hs_left_step && pdata->hs_right_step) { priv->hs_left_step = pdata->hs_left_step; @@ -1586,33 +1529,21 @@ static int twl6040_probe(struct snd_soc_codec *codec) goto work_err; } - priv->workqueue = create_singlethread_workqueue("twl6040-codec"); + priv->workqueue = alloc_workqueue("twl6040-codec", 0, 0); if (!priv->workqueue) { ret = -ENOMEM; goto work_err; } - INIT_DELAYED_WORK(&priv->delayed_work, twl6040_accessory_work); + INIT_DELAYED_WORK(&priv->hs_jack.work, twl6040_accessory_work); + INIT_DELAYED_WORK(&priv->headset.work, twl6040_pga_hs_work); + INIT_DELAYED_WORK(&priv->handsfree.work, twl6040_pga_hf_work); mutex_init(&priv->mutex); init_completion(&priv->headset.ramp_done); init_completion(&priv->handsfree.ramp_done); - priv->hf_workqueue = create_singlethread_workqueue("twl6040-hf"); - if (priv->hf_workqueue == NULL) { - ret = -ENOMEM; - goto hfwq_err; - } - priv->hs_workqueue = create_singlethread_workqueue("twl6040-hs"); - if (priv->hs_workqueue == NULL) { - ret = -ENOMEM; - goto hswq_err; - } - - INIT_DELAYED_WORK(&priv->hs_delayed_work, twl6040_pga_hs_work); - INIT_DELAYED_WORK(&priv->hf_delayed_work, twl6040_pga_hf_work); - ret = request_threaded_irq(priv->plug_irq, NULL, twl6040_audio_handler, 0, "twl6040_irq_plug", codec); if (ret) { @@ -1620,27 +1551,16 @@ static int twl6040_probe(struct snd_soc_codec *codec) goto plugirq_err; } - /* init vio registers */ - twl6040_init_vio_regs(codec); + twl6040_init_chip(codec); /* power on device */ ret = twl6040_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - if (ret) - goto bias_err; - - snd_soc_add_controls(codec, twl6040_snd_controls, - ARRAY_SIZE(twl6040_snd_controls)); - twl6040_add_widgets(codec); - - return 0; + if (!ret) + return 0; -bias_err: + /* Error path */ free_irq(priv->plug_irq, codec); plugirq_err: - destroy_workqueue(priv->hs_workqueue); -hswq_err: - destroy_workqueue(priv->hf_workqueue); -hfwq_err: destroy_workqueue(priv->workqueue); work_err: kfree(priv); @@ -1654,8 +1574,6 @@ static int twl6040_remove(struct snd_soc_codec *codec) twl6040_set_bias_level(codec, SND_SOC_BIAS_OFF); free_irq(priv->plug_irq, codec); destroy_workqueue(priv->workqueue); - destroy_workqueue(priv->hf_workqueue); - destroy_workqueue(priv->hs_workqueue); kfree(priv); return 0; @@ -1672,6 +1590,13 @@ static struct snd_soc_codec_driver soc_codec_dev_twl6040 = { .reg_cache_size = ARRAY_SIZE(twl6040_reg), .reg_word_size = sizeof(u8), .reg_cache_default = twl6040_reg, + + .controls = twl6040_snd_controls, + .num_controls = ARRAY_SIZE(twl6040_snd_controls), + .dapm_widgets = twl6040_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(twl6040_dapm_widgets), + .dapm_routes = intercon, + .num_dapm_routes = ARRAY_SIZE(intercon), }; static int __devinit twl6040_codec_probe(struct platform_device *pdev) diff --git a/sound/soc/codecs/twl6040.h b/sound/soc/codecs/twl6040.h index d8de67869dd9..a83277bdb851 100644 --- a/sound/soc/codecs/twl6040.h +++ b/sound/soc/codecs/twl6040.h @@ -22,8 +22,21 @@ #ifndef __TWL6040_H__ #define __TWL6040_H__ +enum twl6040_trim { + TWL6040_TRIM_TRIM1 = 0, + TWL6040_TRIM_TRIM2, + TWL6040_TRIM_TRIM3, + TWL6040_TRIM_HSOTRIM, + TWL6040_TRIM_HFOTRIM, + TWL6040_TRIM_INVAL, +}; + +#define TWL6040_HSF_TRIM_LEFT(x) (x & 0x0f) +#define TWL6040_HSF_TRIM_RIGHT(x) ((x >> 4) & 0x0f) + void twl6040_hs_jack_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack, int report); int twl6040_get_clk_id(struct snd_soc_codec *codec); +int twl6040_get_trim_value(struct snd_soc_codec *codec, enum twl6040_trim trim); #endif /* End of __TWL6040_H__ */ diff --git a/sound/soc/codecs/wl1273.c b/sound/soc/codecs/wl1273.c index 5836201834d9..9fa14299cf2c 100644 --- a/sound/soc/codecs/wl1273.c +++ b/sound/soc/codecs/wl1273.c @@ -462,7 +462,6 @@ static int wl1273_probe(struct snd_soc_codec *codec) wl1273->core = *core; snd_soc_codec_set_drvdata(codec, wl1273); - mutex_init(&codec->mutex); r = snd_soc_add_controls(codec, wl1273_controls, ARRAY_SIZE(wl1273_controls)); diff --git a/sound/soc/codecs/wm1250-ev1.c b/sound/soc/codecs/wm1250-ev1.c index bcc208967917..cd0ec0fd1dba 100644 --- a/sound/soc/codecs/wm1250-ev1.c +++ b/sound/soc/codecs/wm1250-ev1.c @@ -12,10 +12,59 @@ #include <linux/init.h> #include <linux/module.h> +#include <linux/slab.h> #include <linux/i2c.h> +#include <linux/gpio.h> #include <sound/soc.h> #include <sound/soc-dapm.h> +#include <sound/wm1250-ev1.h> + +static const char *wm1250_gpio_names[WM1250_EV1_NUM_GPIOS] = { + "WM1250 CLK_ENA", + "WM1250 CLK_SEL0", + "WM1250 CLK_SEL1", + "WM1250 OSR", + "WM1250 MASTER", +}; + +struct wm1250_priv { + struct gpio gpios[WM1250_EV1_NUM_GPIOS]; +}; + +static int wm1250_ev1_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct wm1250_priv *wm1250 = dev_get_drvdata(codec->dev); + int ena; + + if (wm1250) + ena = wm1250->gpios[WM1250_EV1_GPIO_CLK_ENA].gpio; + else + ena = -1; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + + case SND_SOC_BIAS_PREPARE: + break; + + case SND_SOC_BIAS_STANDBY: + if (ena >= 0) + gpio_set_value_cansleep(ena, 1); + break; + + case SND_SOC_BIAS_OFF: + if (ena >= 0) + gpio_set_value_cansleep(ena, 0); + break; + } + + codec->dapm.bias_level = level; + + return 0; +} static const struct snd_soc_dapm_widget wm1250_ev1_dapm_widgets[] = { SND_SOC_DAPM_ADC("ADC", "wm1250-ev1 Capture", SND_SOC_NOPM, 0, 0), @@ -53,18 +102,103 @@ static struct snd_soc_codec_driver soc_codec_dev_wm1250_ev1 = { .num_dapm_widgets = ARRAY_SIZE(wm1250_ev1_dapm_widgets), .dapm_routes = wm1250_ev1_dapm_routes, .num_dapm_routes = ARRAY_SIZE(wm1250_ev1_dapm_routes), + + .set_bias_level = wm1250_ev1_set_bias_level, + .idle_bias_off = true, }; +static int __devinit wm1250_ev1_pdata(struct i2c_client *i2c) +{ + struct wm1250_ev1_pdata *pdata = dev_get_platdata(&i2c->dev); + struct wm1250_priv *wm1250; + int i, ret; + + if (!pdata) + return 0; + + wm1250 = kzalloc(sizeof(*wm1250), GFP_KERNEL); + if (!wm1250) { + dev_err(&i2c->dev, "Unable to allocate private data\n"); + ret = -ENOMEM; + goto err; + } + + for (i = 0; i < ARRAY_SIZE(wm1250->gpios); i++) { + wm1250->gpios[i].gpio = pdata->gpios[i]; + wm1250->gpios[i].label = wm1250_gpio_names[i]; + wm1250->gpios[i].flags = GPIOF_OUT_INIT_LOW; + } + wm1250->gpios[WM1250_EV1_GPIO_CLK_SEL0].flags = GPIOF_OUT_INIT_HIGH; + wm1250->gpios[WM1250_EV1_GPIO_CLK_SEL1].flags = GPIOF_OUT_INIT_HIGH; + + ret = gpio_request_array(wm1250->gpios, ARRAY_SIZE(wm1250->gpios)); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to get GPIOs: %d\n", ret); + goto err_alloc; + } + + dev_set_drvdata(&i2c->dev, wm1250); + + return ret; + +err_alloc: + kfree(wm1250); +err: + return ret; +} + +static void wm1250_ev1_free(struct i2c_client *i2c) +{ + struct wm1250_priv *wm1250 = dev_get_drvdata(&i2c->dev); + + if (wm1250) { + gpio_free_array(wm1250->gpios, ARRAY_SIZE(wm1250->gpios)); + kfree(wm1250); + } +} + static int __devinit wm1250_ev1_probe(struct i2c_client *i2c, - const struct i2c_device_id *id) + const struct i2c_device_id *i2c_id) { - return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm1250_ev1, - &wm1250_ev1_dai, 1); + int id, board, rev, ret; + + dev_set_drvdata(&i2c->dev, NULL); + + board = i2c_smbus_read_byte_data(i2c, 0); + if (board < 0) { + dev_err(&i2c->dev, "Failed to read ID: %d\n", board); + return board; + } + + id = (board & 0xfe) >> 2; + rev = board & 0x3; + + if (id != 1) { + dev_err(&i2c->dev, "Unknown board ID %d\n", id); + return -ENODEV; + } + + dev_info(&i2c->dev, "revision %d\n", rev + 1); + + ret = wm1250_ev1_pdata(i2c); + if (ret != 0) + return ret; + + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm1250_ev1, + &wm1250_ev1_dai, 1); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to register CODEC: %d\n", ret); + wm1250_ev1_free(i2c); + return ret; + } + + return 0; } static int __devexit wm1250_ev1_remove(struct i2c_client *i2c) { snd_soc_unregister_codec(&i2c->dev); + wm1250_ev1_free(i2c); return 0; } diff --git a/sound/soc/codecs/wm5100-tables.c b/sound/soc/codecs/wm5100-tables.c new file mode 100644 index 000000000000..e9ce81a57b85 --- /dev/null +++ b/sound/soc/codecs/wm5100-tables.c @@ -0,0 +1,1531 @@ +/* + * wm5100-tables.c -- WM5100 ALSA SoC Audio driver data + * + * Copyright 2011 Wolfson Microelectronics plc + * + * Author: Mark Brown <broonie@opensource.wolfsonmicro.com> + * + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include "wm5100.h" + +int wm5100_volatile_register(struct snd_soc_codec *codec, unsigned int reg) +{ + switch (reg) { + case WM5100_SOFTWARE_RESET: + case WM5100_DEVICE_REVISION: + case WM5100_FX_CTRL: + case WM5100_INTERRUPT_STATUS_1: + case WM5100_INTERRUPT_STATUS_2: + case WM5100_INTERRUPT_STATUS_3: + case WM5100_INTERRUPT_STATUS_4: + case WM5100_INTERRUPT_RAW_STATUS_2: + case WM5100_INTERRUPT_RAW_STATUS_3: + case WM5100_INTERRUPT_RAW_STATUS_4: + case WM5100_OUTPUT_STATUS_1: + case WM5100_OUTPUT_STATUS_2: + case WM5100_INPUT_ENABLES_STATUS: + case WM5100_MIC_DETECT_3: + return 1; + default: + return 0; + } +} + +int wm5100_readable_register(struct snd_soc_codec *codec, unsigned int reg) +{ + switch (reg) { + case WM5100_SOFTWARE_RESET: + case WM5100_DEVICE_REVISION: + case WM5100_CTRL_IF_1: + case WM5100_TONE_GENERATOR_1: + case WM5100_PWM_DRIVE_1: + case WM5100_PWM_DRIVE_2: + case WM5100_PWM_DRIVE_3: + case WM5100_CLOCKING_1: + case WM5100_CLOCKING_3: + case WM5100_CLOCKING_4: + case WM5100_CLOCKING_5: + case WM5100_CLOCKING_6: + case WM5100_CLOCKING_7: + case WM5100_CLOCKING_8: + case WM5100_ASRC_ENABLE: + case WM5100_ASRC_STATUS: + case WM5100_ASRC_RATE1: + case WM5100_ISRC_1_CTRL_1: + case WM5100_ISRC_1_CTRL_2: + case WM5100_ISRC_2_CTRL1: + case WM5100_ISRC_2_CTRL_2: + case WM5100_FLL1_CONTROL_1: + case WM5100_FLL1_CONTROL_2: + case WM5100_FLL1_CONTROL_3: + case WM5100_FLL1_CONTROL_5: + case WM5100_FLL1_CONTROL_6: + case WM5100_FLL1_EFS_1: + case WM5100_FLL2_CONTROL_1: + case WM5100_FLL2_CONTROL_2: + case WM5100_FLL2_CONTROL_3: + case WM5100_FLL2_CONTROL_5: + case WM5100_FLL2_CONTROL_6: + case WM5100_FLL2_EFS_1: + case WM5100_MIC_CHARGE_PUMP_1: + case WM5100_MIC_CHARGE_PUMP_2: + case WM5100_HP_CHARGE_PUMP_1: + case WM5100_LDO1_CONTROL: + case WM5100_MIC_BIAS_CTRL_1: + case WM5100_MIC_BIAS_CTRL_2: + case WM5100_MIC_BIAS_CTRL_3: + case WM5100_ACCESSORY_DETECT_MODE_1: + case WM5100_HEADPHONE_DETECT_1: + case WM5100_HEADPHONE_DETECT_2: + case WM5100_MIC_DETECT_1: + case WM5100_MIC_DETECT_2: + case WM5100_MIC_DETECT_3: + case WM5100_INPUT_ENABLES: + case WM5100_INPUT_ENABLES_STATUS: + case WM5100_IN1L_CONTROL: + case WM5100_IN1R_CONTROL: + case WM5100_IN2L_CONTROL: + case WM5100_IN2R_CONTROL: + case WM5100_IN3L_CONTROL: + case WM5100_IN3R_CONTROL: + case WM5100_IN4L_CONTROL: + case WM5100_IN4R_CONTROL: + case WM5100_RXANC_SRC: + case WM5100_INPUT_VOLUME_RAMP: + case WM5100_ADC_DIGITAL_VOLUME_1L: + case WM5100_ADC_DIGITAL_VOLUME_1R: + case WM5100_ADC_DIGITAL_VOLUME_2L: + case WM5100_ADC_DIGITAL_VOLUME_2R: + case WM5100_ADC_DIGITAL_VOLUME_3L: + case WM5100_ADC_DIGITAL_VOLUME_3R: + case WM5100_ADC_DIGITAL_VOLUME_4L: + case WM5100_ADC_DIGITAL_VOLUME_4R: + case WM5100_OUTPUT_ENABLES_2: + case WM5100_OUTPUT_STATUS_1: + case WM5100_OUTPUT_STATUS_2: + case WM5100_CHANNEL_ENABLES_1: + case WM5100_OUT_VOLUME_1L: + case WM5100_OUT_VOLUME_1R: + case WM5100_DAC_VOLUME_LIMIT_1L: + case WM5100_DAC_VOLUME_LIMIT_1R: + case WM5100_OUT_VOLUME_2L: + case WM5100_OUT_VOLUME_2R: + case WM5100_DAC_VOLUME_LIMIT_2L: + case WM5100_DAC_VOLUME_LIMIT_2R: + case WM5100_OUT_VOLUME_3L: + case WM5100_OUT_VOLUME_3R: + case WM5100_DAC_VOLUME_LIMIT_3L: + case WM5100_DAC_VOLUME_LIMIT_3R: + case WM5100_OUT_VOLUME_4L: + case WM5100_OUT_VOLUME_4R: + case WM5100_DAC_VOLUME_LIMIT_5L: + case WM5100_DAC_VOLUME_LIMIT_5R: + case WM5100_DAC_VOLUME_LIMIT_6L: + case WM5100_DAC_VOLUME_LIMIT_6R: + case WM5100_DAC_AEC_CONTROL_1: + case WM5100_OUTPUT_VOLUME_RAMP: + case WM5100_DAC_DIGITAL_VOLUME_1L: + case WM5100_DAC_DIGITAL_VOLUME_1R: + case WM5100_DAC_DIGITAL_VOLUME_2L: + case WM5100_DAC_DIGITAL_VOLUME_2R: + case WM5100_DAC_DIGITAL_VOLUME_3L: + case WM5100_DAC_DIGITAL_VOLUME_3R: + case WM5100_DAC_DIGITAL_VOLUME_4L: + case WM5100_DAC_DIGITAL_VOLUME_4R: + case WM5100_DAC_DIGITAL_VOLUME_5L: + case WM5100_DAC_DIGITAL_VOLUME_5R: + case WM5100_DAC_DIGITAL_VOLUME_6L: + case WM5100_DAC_DIGITAL_VOLUME_6R: + case WM5100_PDM_SPK1_CTRL_1: + case WM5100_PDM_SPK1_CTRL_2: + case WM5100_PDM_SPK2_CTRL_1: + case WM5100_PDM_SPK2_CTRL_2: + case WM5100_AUDIO_IF_1_1: + case WM5100_AUDIO_IF_1_2: + case WM5100_AUDIO_IF_1_3: + case WM5100_AUDIO_IF_1_4: + case WM5100_AUDIO_IF_1_5: + case WM5100_AUDIO_IF_1_6: + case WM5100_AUDIO_IF_1_7: + case WM5100_AUDIO_IF_1_8: + case WM5100_AUDIO_IF_1_9: + case WM5100_AUDIO_IF_1_10: + case WM5100_AUDIO_IF_1_11: + case WM5100_AUDIO_IF_1_12: + case WM5100_AUDIO_IF_1_13: + case WM5100_AUDIO_IF_1_14: + case WM5100_AUDIO_IF_1_15: + case WM5100_AUDIO_IF_1_16: + case WM5100_AUDIO_IF_1_17: + case WM5100_AUDIO_IF_1_18: + case WM5100_AUDIO_IF_1_19: + case WM5100_AUDIO_IF_1_20: + case WM5100_AUDIO_IF_1_21: + case WM5100_AUDIO_IF_1_22: + case WM5100_AUDIO_IF_1_23: + case WM5100_AUDIO_IF_1_24: + case WM5100_AUDIO_IF_1_25: + case WM5100_AUDIO_IF_1_26: + case WM5100_AUDIO_IF_1_27: + case WM5100_AUDIO_IF_2_1: + case WM5100_AUDIO_IF_2_2: + case WM5100_AUDIO_IF_2_3: + case WM5100_AUDIO_IF_2_4: + case WM5100_AUDIO_IF_2_5: + case WM5100_AUDIO_IF_2_6: + case WM5100_AUDIO_IF_2_7: + case WM5100_AUDIO_IF_2_8: + case WM5100_AUDIO_IF_2_9: + case WM5100_AUDIO_IF_2_10: + case WM5100_AUDIO_IF_2_11: + case WM5100_AUDIO_IF_2_18: + case WM5100_AUDIO_IF_2_19: + case WM5100_AUDIO_IF_2_26: + case WM5100_AUDIO_IF_2_27: + case WM5100_AUDIO_IF_3_1: + case WM5100_AUDIO_IF_3_2: + case WM5100_AUDIO_IF_3_3: + case WM5100_AUDIO_IF_3_4: + case WM5100_AUDIO_IF_3_5: + case WM5100_AUDIO_IF_3_6: + case WM5100_AUDIO_IF_3_7: + case WM5100_AUDIO_IF_3_8: + case WM5100_AUDIO_IF_3_9: + case WM5100_AUDIO_IF_3_10: + case WM5100_AUDIO_IF_3_11: + case WM5100_AUDIO_IF_3_18: + case WM5100_AUDIO_IF_3_19: + case WM5100_AUDIO_IF_3_26: + case WM5100_AUDIO_IF_3_27: + case WM5100_PWM1MIX_INPUT_1_SOURCE: + case WM5100_PWM1MIX_INPUT_1_VOLUME: + case WM5100_PWM1MIX_INPUT_2_SOURCE: + case WM5100_PWM1MIX_INPUT_2_VOLUME: + case WM5100_PWM1MIX_INPUT_3_SOURCE: + case WM5100_PWM1MIX_INPUT_3_VOLUME: + case WM5100_PWM1MIX_INPUT_4_SOURCE: + case WM5100_PWM1MIX_INPUT_4_VOLUME: + case WM5100_PWM2MIX_INPUT_1_SOURCE: + case WM5100_PWM2MIX_INPUT_1_VOLUME: + case WM5100_PWM2MIX_INPUT_2_SOURCE: + case WM5100_PWM2MIX_INPUT_2_VOLUME: + case WM5100_PWM2MIX_INPUT_3_SOURCE: + case WM5100_PWM2MIX_INPUT_3_VOLUME: + case WM5100_PWM2MIX_INPUT_4_SOURCE: + case WM5100_PWM2MIX_INPUT_4_VOLUME: + case WM5100_OUT1LMIX_INPUT_1_SOURCE: + case WM5100_OUT1LMIX_INPUT_1_VOLUME: + case WM5100_OUT1LMIX_INPUT_2_SOURCE: + case WM5100_OUT1LMIX_INPUT_2_VOLUME: + case WM5100_OUT1LMIX_INPUT_3_SOURCE: + case WM5100_OUT1LMIX_INPUT_3_VOLUME: + case WM5100_OUT1LMIX_INPUT_4_SOURCE: + case WM5100_OUT1LMIX_INPUT_4_VOLUME: + case WM5100_OUT1RMIX_INPUT_1_SOURCE: + case WM5100_OUT1RMIX_INPUT_1_VOLUME: + case WM5100_OUT1RMIX_INPUT_2_SOURCE: + case WM5100_OUT1RMIX_INPUT_2_VOLUME: + case WM5100_OUT1RMIX_INPUT_3_SOURCE: + case WM5100_OUT1RMIX_INPUT_3_VOLUME: + case WM5100_OUT1RMIX_INPUT_4_SOURCE: + case WM5100_OUT1RMIX_INPUT_4_VOLUME: + case WM5100_OUT2LMIX_INPUT_1_SOURCE: + case WM5100_OUT2LMIX_INPUT_1_VOLUME: + case WM5100_OUT2LMIX_INPUT_2_SOURCE: + case WM5100_OUT2LMIX_INPUT_2_VOLUME: + case WM5100_OUT2LMIX_INPUT_3_SOURCE: + case WM5100_OUT2LMIX_INPUT_3_VOLUME: + case WM5100_OUT2LMIX_INPUT_4_SOURCE: + case WM5100_OUT2LMIX_INPUT_4_VOLUME: + case WM5100_OUT2RMIX_INPUT_1_SOURCE: + case WM5100_OUT2RMIX_INPUT_1_VOLUME: + case WM5100_OUT2RMIX_INPUT_2_SOURCE: + case WM5100_OUT2RMIX_INPUT_2_VOLUME: + case WM5100_OUT2RMIX_INPUT_3_SOURCE: + case WM5100_OUT2RMIX_INPUT_3_VOLUME: + case WM5100_OUT2RMIX_INPUT_4_SOURCE: + case WM5100_OUT2RMIX_INPUT_4_VOLUME: + case WM5100_OUT3LMIX_INPUT_1_SOURCE: + case WM5100_OUT3LMIX_INPUT_1_VOLUME: + case WM5100_OUT3LMIX_INPUT_2_SOURCE: + case WM5100_OUT3LMIX_INPUT_2_VOLUME: + case WM5100_OUT3LMIX_INPUT_3_SOURCE: + case WM5100_OUT3LMIX_INPUT_3_VOLUME: + case WM5100_OUT3LMIX_INPUT_4_SOURCE: + case WM5100_OUT3LMIX_INPUT_4_VOLUME: + case WM5100_OUT3RMIX_INPUT_1_SOURCE: + case WM5100_OUT3RMIX_INPUT_1_VOLUME: + case WM5100_OUT3RMIX_INPUT_2_SOURCE: + case WM5100_OUT3RMIX_INPUT_2_VOLUME: + case WM5100_OUT3RMIX_INPUT_3_SOURCE: + case WM5100_OUT3RMIX_INPUT_3_VOLUME: + case WM5100_OUT3RMIX_INPUT_4_SOURCE: + case WM5100_OUT3RMIX_INPUT_4_VOLUME: + case WM5100_OUT4LMIX_INPUT_1_SOURCE: + case WM5100_OUT4LMIX_INPUT_1_VOLUME: + case WM5100_OUT4LMIX_INPUT_2_SOURCE: + case WM5100_OUT4LMIX_INPUT_2_VOLUME: + case WM5100_OUT4LMIX_INPUT_3_SOURCE: + case WM5100_OUT4LMIX_INPUT_3_VOLUME: + case WM5100_OUT4LMIX_INPUT_4_SOURCE: + case WM5100_OUT4LMIX_INPUT_4_VOLUME: + case WM5100_OUT4RMIX_INPUT_1_SOURCE: + case WM5100_OUT4RMIX_INPUT_1_VOLUME: + case WM5100_OUT4RMIX_INPUT_2_SOURCE: + case WM5100_OUT4RMIX_INPUT_2_VOLUME: + case WM5100_OUT4RMIX_INPUT_3_SOURCE: + case WM5100_OUT4RMIX_INPUT_3_VOLUME: + case WM5100_OUT4RMIX_INPUT_4_SOURCE: + case WM5100_OUT4RMIX_INPUT_4_VOLUME: + case WM5100_OUT5LMIX_INPUT_1_SOURCE: + case WM5100_OUT5LMIX_INPUT_1_VOLUME: + case WM5100_OUT5LMIX_INPUT_2_SOURCE: + case WM5100_OUT5LMIX_INPUT_2_VOLUME: + case WM5100_OUT5LMIX_INPUT_3_SOURCE: + case WM5100_OUT5LMIX_INPUT_3_VOLUME: + case WM5100_OUT5LMIX_INPUT_4_SOURCE: + case WM5100_OUT5LMIX_INPUT_4_VOLUME: + case WM5100_OUT5RMIX_INPUT_1_SOURCE: + case WM5100_OUT5RMIX_INPUT_1_VOLUME: + case WM5100_OUT5RMIX_INPUT_2_SOURCE: + case WM5100_OUT5RMIX_INPUT_2_VOLUME: + case WM5100_OUT5RMIX_INPUT_3_SOURCE: + case WM5100_OUT5RMIX_INPUT_3_VOLUME: + case WM5100_OUT5RMIX_INPUT_4_SOURCE: + case WM5100_OUT5RMIX_INPUT_4_VOLUME: + case WM5100_OUT6LMIX_INPUT_1_SOURCE: + case WM5100_OUT6LMIX_INPUT_1_VOLUME: + case WM5100_OUT6LMIX_INPUT_2_SOURCE: + case WM5100_OUT6LMIX_INPUT_2_VOLUME: + case WM5100_OUT6LMIX_INPUT_3_SOURCE: + case WM5100_OUT6LMIX_INPUT_3_VOLUME: + case WM5100_OUT6LMIX_INPUT_4_SOURCE: + case WM5100_OUT6LMIX_INPUT_4_VOLUME: + case WM5100_OUT6RMIX_INPUT_1_SOURCE: + case WM5100_OUT6RMIX_INPUT_1_VOLUME: + case WM5100_OUT6RMIX_INPUT_2_SOURCE: + case WM5100_OUT6RMIX_INPUT_2_VOLUME: + case WM5100_OUT6RMIX_INPUT_3_SOURCE: + case WM5100_OUT6RMIX_INPUT_3_VOLUME: + case WM5100_OUT6RMIX_INPUT_4_SOURCE: + case WM5100_OUT6RMIX_INPUT_4_VOLUME: + case WM5100_AIF1TX1MIX_INPUT_1_SOURCE: + case WM5100_AIF1TX1MIX_INPUT_1_VOLUME: + case WM5100_AIF1TX1MIX_INPUT_2_SOURCE: + case WM5100_AIF1TX1MIX_INPUT_2_VOLUME: + case WM5100_AIF1TX1MIX_INPUT_3_SOURCE: + case WM5100_AIF1TX1MIX_INPUT_3_VOLUME: + case WM5100_AIF1TX1MIX_INPUT_4_SOURCE: + case WM5100_AIF1TX1MIX_INPUT_4_VOLUME: + case WM5100_AIF1TX2MIX_INPUT_1_SOURCE: + case WM5100_AIF1TX2MIX_INPUT_1_VOLUME: + case WM5100_AIF1TX2MIX_INPUT_2_SOURCE: + case WM5100_AIF1TX2MIX_INPUT_2_VOLUME: + case WM5100_AIF1TX2MIX_INPUT_3_SOURCE: + case WM5100_AIF1TX2MIX_INPUT_3_VOLUME: + case WM5100_AIF1TX2MIX_INPUT_4_SOURCE: + case WM5100_AIF1TX2MIX_INPUT_4_VOLUME: + case WM5100_AIF1TX3MIX_INPUT_1_SOURCE: + case WM5100_AIF1TX3MIX_INPUT_1_VOLUME: + case WM5100_AIF1TX3MIX_INPUT_2_SOURCE: + case WM5100_AIF1TX3MIX_INPUT_2_VOLUME: + case WM5100_AIF1TX3MIX_INPUT_3_SOURCE: + case WM5100_AIF1TX3MIX_INPUT_3_VOLUME: + case WM5100_AIF1TX3MIX_INPUT_4_SOURCE: + case WM5100_AIF1TX3MIX_INPUT_4_VOLUME: + case WM5100_AIF1TX4MIX_INPUT_1_SOURCE: + case WM5100_AIF1TX4MIX_INPUT_1_VOLUME: + case WM5100_AIF1TX4MIX_INPUT_2_SOURCE: + case WM5100_AIF1TX4MIX_INPUT_2_VOLUME: + case WM5100_AIF1TX4MIX_INPUT_3_SOURCE: + case WM5100_AIF1TX4MIX_INPUT_3_VOLUME: + case WM5100_AIF1TX4MIX_INPUT_4_SOURCE: + case WM5100_AIF1TX4MIX_INPUT_4_VOLUME: + case WM5100_AIF1TX5MIX_INPUT_1_SOURCE: + case WM5100_AIF1TX5MIX_INPUT_1_VOLUME: + case WM5100_AIF1TX5MIX_INPUT_2_SOURCE: + case WM5100_AIF1TX5MIX_INPUT_2_VOLUME: + case WM5100_AIF1TX5MIX_INPUT_3_SOURCE: + case WM5100_AIF1TX5MIX_INPUT_3_VOLUME: + case WM5100_AIF1TX5MIX_INPUT_4_SOURCE: + case WM5100_AIF1TX5MIX_INPUT_4_VOLUME: + case WM5100_AIF1TX6MIX_INPUT_1_SOURCE: + case WM5100_AIF1TX6MIX_INPUT_1_VOLUME: + case WM5100_AIF1TX6MIX_INPUT_2_SOURCE: + case WM5100_AIF1TX6MIX_INPUT_2_VOLUME: + case WM5100_AIF1TX6MIX_INPUT_3_SOURCE: + case WM5100_AIF1TX6MIX_INPUT_3_VOLUME: + case WM5100_AIF1TX6MIX_INPUT_4_SOURCE: + case WM5100_AIF1TX6MIX_INPUT_4_VOLUME: + case WM5100_AIF1TX7MIX_INPUT_1_SOURCE: + case WM5100_AIF1TX7MIX_INPUT_1_VOLUME: + case WM5100_AIF1TX7MIX_INPUT_2_SOURCE: + case WM5100_AIF1TX7MIX_INPUT_2_VOLUME: + case WM5100_AIF1TX7MIX_INPUT_3_SOURCE: + case WM5100_AIF1TX7MIX_INPUT_3_VOLUME: + case WM5100_AIF1TX7MIX_INPUT_4_SOURCE: + case WM5100_AIF1TX7MIX_INPUT_4_VOLUME: + case WM5100_AIF1TX8MIX_INPUT_1_SOURCE: + case WM5100_AIF1TX8MIX_INPUT_1_VOLUME: + case WM5100_AIF1TX8MIX_INPUT_2_SOURCE: + case WM5100_AIF1TX8MIX_INPUT_2_VOLUME: + case WM5100_AIF1TX8MIX_INPUT_3_SOURCE: + case WM5100_AIF1TX8MIX_INPUT_3_VOLUME: + case WM5100_AIF1TX8MIX_INPUT_4_SOURCE: + case WM5100_AIF1TX8MIX_INPUT_4_VOLUME: + case WM5100_AIF2TX1MIX_INPUT_1_SOURCE: + case WM5100_AIF2TX1MIX_INPUT_1_VOLUME: + case WM5100_AIF2TX1MIX_INPUT_2_SOURCE: + case WM5100_AIF2TX1MIX_INPUT_2_VOLUME: + case WM5100_AIF2TX1MIX_INPUT_3_SOURCE: + case WM5100_AIF2TX1MIX_INPUT_3_VOLUME: + case WM5100_AIF2TX1MIX_INPUT_4_SOURCE: + case WM5100_AIF2TX1MIX_INPUT_4_VOLUME: + case WM5100_AIF2TX2MIX_INPUT_1_SOURCE: + case WM5100_AIF2TX2MIX_INPUT_1_VOLUME: + case WM5100_AIF2TX2MIX_INPUT_2_SOURCE: + case WM5100_AIF2TX2MIX_INPUT_2_VOLUME: + case WM5100_AIF2TX2MIX_INPUT_3_SOURCE: + case WM5100_AIF2TX2MIX_INPUT_3_VOLUME: + case WM5100_AIF2TX2MIX_INPUT_4_SOURCE: + case WM5100_AIF2TX2MIX_INPUT_4_VOLUME: + case WM5100_AIF3TX1MIX_INPUT_1_SOURCE: + case WM5100_AIF3TX1MIX_INPUT_1_VOLUME: + case WM5100_AIF3TX1MIX_INPUT_2_SOURCE: + case WM5100_AIF3TX1MIX_INPUT_2_VOLUME: + case WM5100_AIF3TX1MIX_INPUT_3_SOURCE: + case WM5100_AIF3TX1MIX_INPUT_3_VOLUME: + case WM5100_AIF3TX1MIX_INPUT_4_SOURCE: + case WM5100_AIF3TX1MIX_INPUT_4_VOLUME: + case WM5100_AIF3TX2MIX_INPUT_1_SOURCE: + case WM5100_AIF3TX2MIX_INPUT_1_VOLUME: + case WM5100_AIF3TX2MIX_INPUT_2_SOURCE: + case WM5100_AIF3TX2MIX_INPUT_2_VOLUME: + case WM5100_AIF3TX2MIX_INPUT_3_SOURCE: + case WM5100_AIF3TX2MIX_INPUT_3_VOLUME: + case WM5100_AIF3TX2MIX_INPUT_4_SOURCE: + case WM5100_AIF3TX2MIX_INPUT_4_VOLUME: + case WM5100_EQ1MIX_INPUT_1_SOURCE: + case WM5100_EQ1MIX_INPUT_1_VOLUME: + case WM5100_EQ1MIX_INPUT_2_SOURCE: + case WM5100_EQ1MIX_INPUT_2_VOLUME: + case WM5100_EQ1MIX_INPUT_3_SOURCE: + case WM5100_EQ1MIX_INPUT_3_VOLUME: + case WM5100_EQ1MIX_INPUT_4_SOURCE: + case WM5100_EQ1MIX_INPUT_4_VOLUME: + case WM5100_EQ2MIX_INPUT_1_SOURCE: + case WM5100_EQ2MIX_INPUT_1_VOLUME: + case WM5100_EQ2MIX_INPUT_2_SOURCE: + case WM5100_EQ2MIX_INPUT_2_VOLUME: + case WM5100_EQ2MIX_INPUT_3_SOURCE: + case WM5100_EQ2MIX_INPUT_3_VOLUME: + case WM5100_EQ2MIX_INPUT_4_SOURCE: + case WM5100_EQ2MIX_INPUT_4_VOLUME: + case WM5100_EQ3MIX_INPUT_1_SOURCE: + case WM5100_EQ3MIX_INPUT_1_VOLUME: + case WM5100_EQ3MIX_INPUT_2_SOURCE: + case WM5100_EQ3MIX_INPUT_2_VOLUME: + case WM5100_EQ3MIX_INPUT_3_SOURCE: + case WM5100_EQ3MIX_INPUT_3_VOLUME: + case WM5100_EQ3MIX_INPUT_4_SOURCE: + case WM5100_EQ3MIX_INPUT_4_VOLUME: + case WM5100_EQ4MIX_INPUT_1_SOURCE: + case WM5100_EQ4MIX_INPUT_1_VOLUME: + case WM5100_EQ4MIX_INPUT_2_SOURCE: + case WM5100_EQ4MIX_INPUT_2_VOLUME: + case WM5100_EQ4MIX_INPUT_3_SOURCE: + case WM5100_EQ4MIX_INPUT_3_VOLUME: + case WM5100_EQ4MIX_INPUT_4_SOURCE: + case WM5100_EQ4MIX_INPUT_4_VOLUME: + case WM5100_DRC1LMIX_INPUT_1_SOURCE: + case WM5100_DRC1LMIX_INPUT_1_VOLUME: + case WM5100_DRC1LMIX_INPUT_2_SOURCE: + case WM5100_DRC1LMIX_INPUT_2_VOLUME: + case WM5100_DRC1LMIX_INPUT_3_SOURCE: + case WM5100_DRC1LMIX_INPUT_3_VOLUME: + case WM5100_DRC1LMIX_INPUT_4_SOURCE: + case WM5100_DRC1LMIX_INPUT_4_VOLUME: + case WM5100_DRC1RMIX_INPUT_1_SOURCE: + case WM5100_DRC1RMIX_INPUT_1_VOLUME: + case WM5100_DRC1RMIX_INPUT_2_SOURCE: + case WM5100_DRC1RMIX_INPUT_2_VOLUME: + case WM5100_DRC1RMIX_INPUT_3_SOURCE: + case WM5100_DRC1RMIX_INPUT_3_VOLUME: + case WM5100_DRC1RMIX_INPUT_4_SOURCE: + case WM5100_DRC1RMIX_INPUT_4_VOLUME: + case WM5100_HPLP1MIX_INPUT_1_SOURCE: + case WM5100_HPLP1MIX_INPUT_1_VOLUME: + case WM5100_HPLP1MIX_INPUT_2_SOURCE: + case WM5100_HPLP1MIX_INPUT_2_VOLUME: + case WM5100_HPLP1MIX_INPUT_3_SOURCE: + case WM5100_HPLP1MIX_INPUT_3_VOLUME: + case WM5100_HPLP1MIX_INPUT_4_SOURCE: + case WM5100_HPLP1MIX_INPUT_4_VOLUME: + case WM5100_HPLP2MIX_INPUT_1_SOURCE: + case WM5100_HPLP2MIX_INPUT_1_VOLUME: + case WM5100_HPLP2MIX_INPUT_2_SOURCE: + case WM5100_HPLP2MIX_INPUT_2_VOLUME: + case WM5100_HPLP2MIX_INPUT_3_SOURCE: + case WM5100_HPLP2MIX_INPUT_3_VOLUME: + case WM5100_HPLP2MIX_INPUT_4_SOURCE: + case WM5100_HPLP2MIX_INPUT_4_VOLUME: + case WM5100_HPLP3MIX_INPUT_1_SOURCE: + case WM5100_HPLP3MIX_INPUT_1_VOLUME: + case WM5100_HPLP3MIX_INPUT_2_SOURCE: + case WM5100_HPLP3MIX_INPUT_2_VOLUME: + case WM5100_HPLP3MIX_INPUT_3_SOURCE: + case WM5100_HPLP3MIX_INPUT_3_VOLUME: + case WM5100_HPLP3MIX_INPUT_4_SOURCE: + case WM5100_HPLP3MIX_INPUT_4_VOLUME: + case WM5100_HPLP4MIX_INPUT_1_SOURCE: + case WM5100_HPLP4MIX_INPUT_1_VOLUME: + case WM5100_HPLP4MIX_INPUT_2_SOURCE: + case WM5100_HPLP4MIX_INPUT_2_VOLUME: + case WM5100_HPLP4MIX_INPUT_3_SOURCE: + case WM5100_HPLP4MIX_INPUT_3_VOLUME: + case WM5100_HPLP4MIX_INPUT_4_SOURCE: + case WM5100_HPLP4MIX_INPUT_4_VOLUME: + case WM5100_DSP1LMIX_INPUT_1_SOURCE: + case WM5100_DSP1LMIX_INPUT_1_VOLUME: + case WM5100_DSP1LMIX_INPUT_2_SOURCE: + case WM5100_DSP1LMIX_INPUT_2_VOLUME: + case WM5100_DSP1LMIX_INPUT_3_SOURCE: + case WM5100_DSP1LMIX_INPUT_3_VOLUME: + case WM5100_DSP1LMIX_INPUT_4_SOURCE: + case WM5100_DSP1LMIX_INPUT_4_VOLUME: + case WM5100_DSP1RMIX_INPUT_1_SOURCE: + case WM5100_DSP1RMIX_INPUT_1_VOLUME: + case WM5100_DSP1RMIX_INPUT_2_SOURCE: + case WM5100_DSP1RMIX_INPUT_2_VOLUME: + case WM5100_DSP1RMIX_INPUT_3_SOURCE: + case WM5100_DSP1RMIX_INPUT_3_VOLUME: + case WM5100_DSP1RMIX_INPUT_4_SOURCE: + case WM5100_DSP1RMIX_INPUT_4_VOLUME: + case WM5100_DSP1AUX1MIX_INPUT_1_SOURCE: + case WM5100_DSP1AUX2MIX_INPUT_1_SOURCE: + case WM5100_DSP1AUX3MIX_INPUT_1_SOURCE: + case WM5100_DSP1AUX4MIX_INPUT_1_SOURCE: + case WM5100_DSP1AUX5MIX_INPUT_1_SOURCE: + case WM5100_DSP1AUX6MIX_INPUT_1_SOURCE: + case WM5100_DSP2LMIX_INPUT_1_SOURCE: + case WM5100_DSP2LMIX_INPUT_1_VOLUME: + case WM5100_DSP2LMIX_INPUT_2_SOURCE: + case WM5100_DSP2LMIX_INPUT_2_VOLUME: + case WM5100_DSP2LMIX_INPUT_3_SOURCE: + case WM5100_DSP2LMIX_INPUT_3_VOLUME: + case WM5100_DSP2LMIX_INPUT_4_SOURCE: + case WM5100_DSP2LMIX_INPUT_4_VOLUME: + case WM5100_DSP2RMIX_INPUT_1_SOURCE: + case WM5100_DSP2RMIX_INPUT_1_VOLUME: + case WM5100_DSP2RMIX_INPUT_2_SOURCE: + case WM5100_DSP2RMIX_INPUT_2_VOLUME: + case WM5100_DSP2RMIX_INPUT_3_SOURCE: + case WM5100_DSP2RMIX_INPUT_3_VOLUME: + case WM5100_DSP2RMIX_INPUT_4_SOURCE: + case WM5100_DSP2RMIX_INPUT_4_VOLUME: + case WM5100_DSP2AUX1MIX_INPUT_1_SOURCE: + case WM5100_DSP2AUX2MIX_INPUT_1_SOURCE: + case WM5100_DSP2AUX3MIX_INPUT_1_SOURCE: + case WM5100_DSP2AUX4MIX_INPUT_1_SOURCE: + case WM5100_DSP2AUX5MIX_INPUT_1_SOURCE: + case WM5100_DSP2AUX6MIX_INPUT_1_SOURCE: + case WM5100_DSP3LMIX_INPUT_1_SOURCE: + case WM5100_DSP3LMIX_INPUT_1_VOLUME: + case WM5100_DSP3LMIX_INPUT_2_SOURCE: + case WM5100_DSP3LMIX_INPUT_2_VOLUME: + case WM5100_DSP3LMIX_INPUT_3_SOURCE: + case WM5100_DSP3LMIX_INPUT_3_VOLUME: + case WM5100_DSP3LMIX_INPUT_4_SOURCE: + case WM5100_DSP3LMIX_INPUT_4_VOLUME: + case WM5100_DSP3RMIX_INPUT_1_SOURCE: + case WM5100_DSP3RMIX_INPUT_1_VOLUME: + case WM5100_DSP3RMIX_INPUT_2_SOURCE: + case WM5100_DSP3RMIX_INPUT_2_VOLUME: + case WM5100_DSP3RMIX_INPUT_3_SOURCE: + case WM5100_DSP3RMIX_INPUT_3_VOLUME: + case WM5100_DSP3RMIX_INPUT_4_SOURCE: + case WM5100_DSP3RMIX_INPUT_4_VOLUME: + case WM5100_DSP3AUX1MIX_INPUT_1_SOURCE: + case WM5100_DSP3AUX2MIX_INPUT_1_SOURCE: + case WM5100_DSP3AUX3MIX_INPUT_1_SOURCE: + case WM5100_DSP3AUX4MIX_INPUT_1_SOURCE: + case WM5100_DSP3AUX5MIX_INPUT_1_SOURCE: + case WM5100_DSP3AUX6MIX_INPUT_1_SOURCE: + case WM5100_ASRC1LMIX_INPUT_1_SOURCE: + case WM5100_ASRC1RMIX_INPUT_1_SOURCE: + case WM5100_ASRC2LMIX_INPUT_1_SOURCE: + case WM5100_ASRC2RMIX_INPUT_1_SOURCE: + case WM5100_ISRC1DEC1MIX_INPUT_1_SOURCE: + case WM5100_ISRC1DEC2MIX_INPUT_1_SOURCE: + case WM5100_ISRC1DEC3MIX_INPUT_1_SOURCE: + case WM5100_ISRC1DEC4MIX_INPUT_1_SOURCE: + case WM5100_ISRC1INT1MIX_INPUT_1_SOURCE: + case WM5100_ISRC1INT2MIX_INPUT_1_SOURCE: + case WM5100_ISRC1INT3MIX_INPUT_1_SOURCE: + case WM5100_ISRC1INT4MIX_INPUT_1_SOURCE: + case WM5100_ISRC2DEC1MIX_INPUT_1_SOURCE: + case WM5100_ISRC2DEC2MIX_INPUT_1_SOURCE: + case WM5100_ISRC2DEC3MIX_INPUT_1_SOURCE: + case WM5100_ISRC2DEC4MIX_INPUT_1_SOURCE: + case WM5100_ISRC2INT1MIX_INPUT_1_SOURCE: + case WM5100_ISRC2INT2MIX_INPUT_1_SOURCE: + case WM5100_ISRC2INT3MIX_INPUT_1_SOURCE: + case WM5100_ISRC2INT4MIX_INPUT_1_SOURCE: + case WM5100_GPIO_CTRL_1: + case WM5100_GPIO_CTRL_2: + case WM5100_GPIO_CTRL_3: + case WM5100_GPIO_CTRL_4: + case WM5100_GPIO_CTRL_5: + case WM5100_GPIO_CTRL_6: + case WM5100_MISC_PAD_CTRL_1: + case WM5100_MISC_PAD_CTRL_2: + case WM5100_MISC_PAD_CTRL_3: + case WM5100_MISC_PAD_CTRL_4: + case WM5100_MISC_PAD_CTRL_5: + case WM5100_MISC_GPIO_1: + case WM5100_INTERRUPT_STATUS_1: + case WM5100_INTERRUPT_STATUS_2: + case WM5100_INTERRUPT_STATUS_3: + case WM5100_INTERRUPT_STATUS_4: + case WM5100_INTERRUPT_RAW_STATUS_2: + case WM5100_INTERRUPT_RAW_STATUS_3: + case WM5100_INTERRUPT_RAW_STATUS_4: + case WM5100_INTERRUPT_STATUS_1_MASK: + case WM5100_INTERRUPT_STATUS_2_MASK: + case WM5100_INTERRUPT_STATUS_3_MASK: + case WM5100_INTERRUPT_STATUS_4_MASK: + case WM5100_INTERRUPT_CONTROL: + case WM5100_IRQ_DEBOUNCE_1: + case WM5100_IRQ_DEBOUNCE_2: + case WM5100_FX_CTRL: + case WM5100_EQ1_1: + case WM5100_EQ1_2: + case WM5100_EQ1_3: + case WM5100_EQ1_4: + case WM5100_EQ1_5: + case WM5100_EQ1_6: + case WM5100_EQ1_7: + case WM5100_EQ1_8: + case WM5100_EQ1_9: + case WM5100_EQ1_10: + case WM5100_EQ1_11: + case WM5100_EQ1_12: + case WM5100_EQ1_13: + case WM5100_EQ1_14: + case WM5100_EQ1_15: + case WM5100_EQ1_16: + case WM5100_EQ1_17: + case WM5100_EQ1_18: + case WM5100_EQ1_19: + case WM5100_EQ1_20: + case WM5100_EQ2_1: + case WM5100_EQ2_2: + case WM5100_EQ2_3: + case WM5100_EQ2_4: + case WM5100_EQ2_5: + case WM5100_EQ2_6: + case WM5100_EQ2_7: + case WM5100_EQ2_8: + case WM5100_EQ2_9: + case WM5100_EQ2_10: + case WM5100_EQ2_11: + case WM5100_EQ2_12: + case WM5100_EQ2_13: + case WM5100_EQ2_14: + case WM5100_EQ2_15: + case WM5100_EQ2_16: + case WM5100_EQ2_17: + case WM5100_EQ2_18: + case WM5100_EQ2_19: + case WM5100_EQ2_20: + case WM5100_EQ3_1: + case WM5100_EQ3_2: + case WM5100_EQ3_3: + case WM5100_EQ3_4: + case WM5100_EQ3_5: + case WM5100_EQ3_6: + case WM5100_EQ3_7: + case WM5100_EQ3_8: + case WM5100_EQ3_9: + case WM5100_EQ3_10: + case WM5100_EQ3_11: + case WM5100_EQ3_12: + case WM5100_EQ3_13: + case WM5100_EQ3_14: + case WM5100_EQ3_15: + case WM5100_EQ3_16: + case WM5100_EQ3_17: + case WM5100_EQ3_18: + case WM5100_EQ3_19: + case WM5100_EQ3_20: + case WM5100_EQ4_1: + case WM5100_EQ4_2: + case WM5100_EQ4_3: + case WM5100_EQ4_4: + case WM5100_EQ4_5: + case WM5100_EQ4_6: + case WM5100_EQ4_7: + case WM5100_EQ4_8: + case WM5100_EQ4_9: + case WM5100_EQ4_10: + case WM5100_EQ4_11: + case WM5100_EQ4_12: + case WM5100_EQ4_13: + case WM5100_EQ4_14: + case WM5100_EQ4_15: + case WM5100_EQ4_16: + case WM5100_EQ4_17: + case WM5100_EQ4_18: + case WM5100_EQ4_19: + case WM5100_EQ4_20: + case WM5100_DRC1_CTRL1: + case WM5100_DRC1_CTRL2: + case WM5100_DRC1_CTRL3: + case WM5100_DRC1_CTRL4: + case WM5100_DRC1_CTRL5: + case WM5100_HPLPF1_1: + case WM5100_HPLPF1_2: + case WM5100_HPLPF2_1: + case WM5100_HPLPF2_2: + case WM5100_HPLPF3_1: + case WM5100_HPLPF3_2: + case WM5100_HPLPF4_1: + case WM5100_HPLPF4_2: + case WM5100_DSP1_DM_0: + case WM5100_DSP1_DM_1: + case WM5100_DSP1_DM_2: + case WM5100_DSP1_DM_3: + case WM5100_DSP1_DM_508: + case WM5100_DSP1_DM_509: + case WM5100_DSP1_DM_510: + case WM5100_DSP1_DM_511: + case WM5100_DSP1_PM_0: + case WM5100_DSP1_PM_1: + case WM5100_DSP1_PM_2: + case WM5100_DSP1_PM_3: + case WM5100_DSP1_PM_4: + case WM5100_DSP1_PM_5: + case WM5100_DSP1_PM_1530: + case WM5100_DSP1_PM_1531: + case WM5100_DSP1_PM_1532: + case WM5100_DSP1_PM_1533: + case WM5100_DSP1_PM_1534: + case WM5100_DSP1_PM_1535: + case WM5100_DSP1_ZM_0: + case WM5100_DSP1_ZM_1: + case WM5100_DSP1_ZM_2: + case WM5100_DSP1_ZM_3: + case WM5100_DSP1_ZM_2044: + case WM5100_DSP1_ZM_2045: + case WM5100_DSP1_ZM_2046: + case WM5100_DSP1_ZM_2047: + case WM5100_DSP2_DM_0: + case WM5100_DSP2_DM_1: + case WM5100_DSP2_DM_2: + case WM5100_DSP2_DM_3: + case WM5100_DSP2_DM_508: + case WM5100_DSP2_DM_509: + case WM5100_DSP2_DM_510: + case WM5100_DSP2_DM_511: + case WM5100_DSP2_PM_0: + case WM5100_DSP2_PM_1: + case WM5100_DSP2_PM_2: + case WM5100_DSP2_PM_3: + case WM5100_DSP2_PM_4: + case WM5100_DSP2_PM_5: + case WM5100_DSP2_PM_1530: + case WM5100_DSP2_PM_1531: + case WM5100_DSP2_PM_1532: + case WM5100_DSP2_PM_1533: + case WM5100_DSP2_PM_1534: + case WM5100_DSP2_PM_1535: + case WM5100_DSP2_ZM_0: + case WM5100_DSP2_ZM_1: + case WM5100_DSP2_ZM_2: + case WM5100_DSP2_ZM_3: + case WM5100_DSP2_ZM_2044: + case WM5100_DSP2_ZM_2045: + case WM5100_DSP2_ZM_2046: + case WM5100_DSP2_ZM_2047: + case WM5100_DSP3_DM_0: + case WM5100_DSP3_DM_1: + case WM5100_DSP3_DM_2: + case WM5100_DSP3_DM_3: + case WM5100_DSP3_DM_508: + case WM5100_DSP3_DM_509: + case WM5100_DSP3_DM_510: + case WM5100_DSP3_DM_511: + case WM5100_DSP3_PM_0: + case WM5100_DSP3_PM_1: + case WM5100_DSP3_PM_2: + case WM5100_DSP3_PM_3: + case WM5100_DSP3_PM_4: + case WM5100_DSP3_PM_5: + case WM5100_DSP3_PM_1530: + case WM5100_DSP3_PM_1531: + case WM5100_DSP3_PM_1532: + case WM5100_DSP3_PM_1533: + case WM5100_DSP3_PM_1534: + case WM5100_DSP3_PM_1535: + case WM5100_DSP3_ZM_0: + case WM5100_DSP3_ZM_1: + case WM5100_DSP3_ZM_2: + case WM5100_DSP3_ZM_3: + case WM5100_DSP3_ZM_2044: + case WM5100_DSP3_ZM_2045: + case WM5100_DSP3_ZM_2046: + case WM5100_DSP3_ZM_2047: + return 1; + default: + return 0; + } +} + +u16 wm5100_reg_defaults[WM5100_MAX_REGISTER + 1] = { + [0x0000] = 0x0000, /* R0 - software reset */ + [0x0001] = 0x0000, /* R1 - Device Revision */ + [0x0010] = 0x0801, /* R16 - Ctrl IF 1 */ + [0x0020] = 0x0000, /* R32 - Tone Generator 1 */ + [0x0030] = 0x0000, /* R48 - PWM Drive 1 */ + [0x0031] = 0x0100, /* R49 - PWM Drive 2 */ + [0x0032] = 0x0100, /* R50 - PWM Drive 3 */ + [0x0100] = 0x0002, /* R256 - Clocking 1 */ + [0x0101] = 0x0000, /* R257 - Clocking 3 */ + [0x0102] = 0x0011, /* R258 - Clocking 4 */ + [0x0103] = 0x0011, /* R259 - Clocking 5 */ + [0x0104] = 0x0011, /* R260 - Clocking 6 */ + [0x0107] = 0x0000, /* R263 - Clocking 7 */ + [0x0108] = 0x0000, /* R264 - Clocking 8 */ + [0x0120] = 0x0000, /* R288 - ASRC_ENABLE */ + [0x0121] = 0x0000, /* R289 - ASRC_STATUS */ + [0x0122] = 0x0000, /* R290 - ASRC_RATE1 */ + [0x0141] = 0x8000, /* R321 - ISRC 1 CTRL 1 */ + [0x0142] = 0x0000, /* R322 - ISRC 1 CTRL 2 */ + [0x0143] = 0x8000, /* R323 - ISRC 2 CTRL1 */ + [0x0144] = 0x0000, /* R324 - ISRC 2 CTRL 2 */ + [0x0182] = 0x0000, /* R386 - FLL1 Control 1 */ + [0x0183] = 0x0000, /* R387 - FLL1 Control 2 */ + [0x0184] = 0x0000, /* R388 - FLL1 Control 3 */ + [0x0186] = 0x0177, /* R390 - FLL1 Control 5 */ + [0x0187] = 0x0001, /* R391 - FLL1 Control 6 */ + [0x0188] = 0x0000, /* R392 - FLL1 EFS 1 */ + [0x01A2] = 0x0000, /* R418 - FLL2 Control 1 */ + [0x01A3] = 0x0000, /* R419 - FLL2 Control 2 */ + [0x01A4] = 0x0000, /* R420 - FLL2 Control 3 */ + [0x01A6] = 0x0177, /* R422 - FLL2 Control 5 */ + [0x01A7] = 0x0001, /* R423 - FLL2 Control 6 */ + [0x01A8] = 0x0000, /* R424 - FLL2 EFS 1 */ + [0x0200] = 0x0020, /* R512 - Mic Charge Pump 1 */ + [0x0201] = 0xB084, /* R513 - Mic Charge Pump 2 */ + [0x0202] = 0xBBDE, /* R514 - HP Charge Pump 1 */ + [0x0211] = 0x20D4, /* R529 - LDO1 Control */ + [0x0215] = 0x0062, /* R533 - Mic Bias Ctrl 1 */ + [0x0216] = 0x0062, /* R534 - Mic Bias Ctrl 2 */ + [0x0217] = 0x0062, /* R535 - Mic Bias Ctrl 3 */ + [0x0280] = 0x0004, /* R640 - Accessory Detect Mode 1 */ + [0x0288] = 0x0020, /* R648 - Headphone Detect 1 */ + [0x0289] = 0x0000, /* R649 - Headphone Detect 2 */ + [0x0290] = 0x1100, /* R656 - Mic Detect 1 */ + [0x0291] = 0x009F, /* R657 - Mic Detect 2 */ + [0x0292] = 0x0000, /* R658 - Mic Detect 3 */ + [0x0301] = 0x0000, /* R769 - Input Enables */ + [0x0302] = 0x0000, /* R770 - Input Enables Status */ + [0x0310] = 0x2280, /* R784 - Status */ + [0x0311] = 0x0080, /* R785 - IN1R Control */ + [0x0312] = 0x2280, /* R786 - IN2L Control */ + [0x0313] = 0x0080, /* R787 - IN2R Control */ + [0x0314] = 0x2280, /* R788 - IN3L Control */ + [0x0315] = 0x0080, /* R789 - IN3R Control */ + [0x0316] = 0x2280, /* R790 - IN4L Control */ + [0x0317] = 0x0080, /* R791 - IN4R Control */ + [0x0318] = 0x0000, /* R792 - RXANC_SRC */ + [0x0319] = 0x0022, /* R793 - Input Volume Ramp */ + [0x0320] = 0x0180, /* R800 - ADC Digital Volume 1L */ + [0x0321] = 0x0180, /* R801 - ADC Digital Volume 1R */ + [0x0322] = 0x0180, /* R802 - ADC Digital Volume 2L */ + [0x0323] = 0x0180, /* R803 - ADC Digital Volume 2R */ + [0x0324] = 0x0180, /* R804 - ADC Digital Volume 3L */ + [0x0325] = 0x0180, /* R805 - ADC Digital Volume 3R */ + [0x0326] = 0x0180, /* R806 - ADC Digital Volume 4L */ + [0x0327] = 0x0180, /* R807 - ADC Digital Volume 4R */ + [0x0401] = 0x0000, /* R1025 - Output Enables 2 */ + [0x0402] = 0x0000, /* R1026 - Output Status 1 */ + [0x0403] = 0x0000, /* R1027 - Output Status 2 */ + [0x0408] = 0x0000, /* R1032 - Channel Enables 1 */ + [0x0410] = 0x0080, /* R1040 - Out Volume 1L */ + [0x0411] = 0x0080, /* R1041 - Out Volume 1R */ + [0x0412] = 0x0080, /* R1042 - DAC Volume Limit 1L */ + [0x0413] = 0x0080, /* R1043 - DAC Volume Limit 1R */ + [0x0414] = 0x0080, /* R1044 - Out Volume 2L */ + [0x0415] = 0x0080, /* R1045 - Out Volume 2R */ + [0x0416] = 0x0080, /* R1046 - DAC Volume Limit 2L */ + [0x0417] = 0x0080, /* R1047 - DAC Volume Limit 2R */ + [0x0418] = 0x0080, /* R1048 - Out Volume 3L */ + [0x0419] = 0x0080, /* R1049 - Out Volume 3R */ + [0x041A] = 0x0080, /* R1050 - DAC Volume Limit 3L */ + [0x041B] = 0x0080, /* R1051 - DAC Volume Limit 3R */ + [0x041C] = 0x0080, /* R1052 - Out Volume 4L */ + [0x041D] = 0x0080, /* R1053 - Out Volume 4R */ + [0x041E] = 0x0080, /* R1054 - DAC Volume Limit 5L */ + [0x041F] = 0x0080, /* R1055 - DAC Volume Limit 5R */ + [0x0420] = 0x0080, /* R1056 - DAC Volume Limit 6L */ + [0x0421] = 0x0080, /* R1057 - DAC Volume Limit 6R */ + [0x0440] = 0x0000, /* R1088 - DAC AEC Control 1 */ + [0x0441] = 0x0022, /* R1089 - Output Volume Ramp */ + [0x0480] = 0x0180, /* R1152 - DAC Digital Volume 1L */ + [0x0481] = 0x0180, /* R1153 - DAC Digital Volume 1R */ + [0x0482] = 0x0180, /* R1154 - DAC Digital Volume 2L */ + [0x0483] = 0x0180, /* R1155 - DAC Digital Volume 2R */ + [0x0484] = 0x0180, /* R1156 - DAC Digital Volume 3L */ + [0x0485] = 0x0180, /* R1157 - DAC Digital Volume 3R */ + [0x0486] = 0x0180, /* R1158 - DAC Digital Volume 4L */ + [0x0487] = 0x0180, /* R1159 - DAC Digital Volume 4R */ + [0x0488] = 0x0180, /* R1160 - DAC Digital Volume 5L */ + [0x0489] = 0x0180, /* R1161 - DAC Digital Volume 5R */ + [0x048A] = 0x0180, /* R1162 - DAC Digital Volume 6L */ + [0x048B] = 0x0180, /* R1163 - DAC Digital Volume 6R */ + [0x04C0] = 0x0069, /* R1216 - PDM SPK1 CTRL 1 */ + [0x04C1] = 0x0000, /* R1217 - PDM SPK1 CTRL 2 */ + [0x04C2] = 0x0069, /* R1218 - PDM SPK2 CTRL 1 */ + [0x04C3] = 0x0000, /* R1219 - PDM SPK2 CTRL 2 */ + [0x0500] = 0x000C, /* R1280 - Audio IF 1_1 */ + [0x0501] = 0x0008, /* R1281 - Audio IF 1_2 */ + [0x0502] = 0x0000, /* R1282 - Audio IF 1_3 */ + [0x0503] = 0x0000, /* R1283 - Audio IF 1_4 */ + [0x0504] = 0x0000, /* R1284 - Audio IF 1_5 */ + [0x0505] = 0x0300, /* R1285 - Audio IF 1_6 */ + [0x0506] = 0x0300, /* R1286 - Audio IF 1_7 */ + [0x0507] = 0x1820, /* R1287 - Audio IF 1_8 */ + [0x0508] = 0x1820, /* R1288 - Audio IF 1_9 */ + [0x0509] = 0x0000, /* R1289 - Audio IF 1_10 */ + [0x050A] = 0x0001, /* R1290 - Audio IF 1_11 */ + [0x050B] = 0x0002, /* R1291 - Audio IF 1_12 */ + [0x050C] = 0x0003, /* R1292 - Audio IF 1_13 */ + [0x050D] = 0x0004, /* R1293 - Audio IF 1_14 */ + [0x050E] = 0x0005, /* R1294 - Audio IF 1_15 */ + [0x050F] = 0x0006, /* R1295 - Audio IF 1_16 */ + [0x0510] = 0x0007, /* R1296 - Audio IF 1_17 */ + [0x0511] = 0x0000, /* R1297 - Audio IF 1_18 */ + [0x0512] = 0x0001, /* R1298 - Audio IF 1_19 */ + [0x0513] = 0x0002, /* R1299 - Audio IF 1_20 */ + [0x0514] = 0x0003, /* R1300 - Audio IF 1_21 */ + [0x0515] = 0x0004, /* R1301 - Audio IF 1_22 */ + [0x0516] = 0x0005, /* R1302 - Audio IF 1_23 */ + [0x0517] = 0x0006, /* R1303 - Audio IF 1_24 */ + [0x0518] = 0x0007, /* R1304 - Audio IF 1_25 */ + [0x0519] = 0x0000, /* R1305 - Audio IF 1_26 */ + [0x051A] = 0x0000, /* R1306 - Audio IF 1_27 */ + [0x0540] = 0x000C, /* R1344 - Audio IF 2_1 */ + [0x0541] = 0x0008, /* R1345 - Audio IF 2_2 */ + [0x0542] = 0x0000, /* R1346 - Audio IF 2_3 */ + [0x0543] = 0x0000, /* R1347 - Audio IF 2_4 */ + [0x0544] = 0x0000, /* R1348 - Audio IF 2_5 */ + [0x0545] = 0x0300, /* R1349 - Audio IF 2_6 */ + [0x0546] = 0x0300, /* R1350 - Audio IF 2_7 */ + [0x0547] = 0x1820, /* R1351 - Audio IF 2_8 */ + [0x0548] = 0x1820, /* R1352 - Audio IF 2_9 */ + [0x0549] = 0x0000, /* R1353 - Audio IF 2_10 */ + [0x054A] = 0x0001, /* R1354 - Audio IF 2_11 */ + [0x0551] = 0x0000, /* R1361 - Audio IF 2_18 */ + [0x0552] = 0x0001, /* R1362 - Audio IF 2_19 */ + [0x0559] = 0x0000, /* R1369 - Audio IF 2_26 */ + [0x055A] = 0x0000, /* R1370 - Audio IF 2_27 */ + [0x0580] = 0x000C, /* R1408 - Audio IF 3_1 */ + [0x0581] = 0x0008, /* R1409 - Audio IF 3_2 */ + [0x0582] = 0x0000, /* R1410 - Audio IF 3_3 */ + [0x0583] = 0x0000, /* R1411 - Audio IF 3_4 */ + [0x0584] = 0x0000, /* R1412 - Audio IF 3_5 */ + [0x0585] = 0x0300, /* R1413 - Audio IF 3_6 */ + [0x0586] = 0x0300, /* R1414 - Audio IF 3_7 */ + [0x0587] = 0x1820, /* R1415 - Audio IF 3_8 */ + [0x0588] = 0x1820, /* R1416 - Audio IF 3_9 */ + [0x0589] = 0x0000, /* R1417 - Audio IF 3_10 */ + [0x058A] = 0x0001, /* R1418 - Audio IF 3_11 */ + [0x0591] = 0x0000, /* R1425 - Audio IF 3_18 */ + [0x0592] = 0x0001, /* R1426 - Audio IF 3_19 */ + [0x0599] = 0x0000, /* R1433 - Audio IF 3_26 */ + [0x059A] = 0x0000, /* R1434 - Audio IF 3_27 */ + [0x0640] = 0x0000, /* R1600 - PWM1MIX Input 1 Source */ + [0x0641] = 0x0080, /* R1601 - PWM1MIX Input 1 Volume */ + [0x0642] = 0x0000, /* R1602 - PWM1MIX Input 2 Source */ + [0x0643] = 0x0080, /* R1603 - PWM1MIX Input 2 Volume */ + [0x0644] = 0x0000, /* R1604 - PWM1MIX Input 3 Source */ + [0x0645] = 0x0080, /* R1605 - PWM1MIX Input 3 Volume */ + [0x0646] = 0x0000, /* R1606 - PWM1MIX Input 4 Source */ + [0x0647] = 0x0080, /* R1607 - PWM1MIX Input 4 Volume */ + [0x0648] = 0x0000, /* R1608 - PWM2MIX Input 1 Source */ + [0x0649] = 0x0080, /* R1609 - PWM2MIX Input 1 Volume */ + [0x064A] = 0x0000, /* R1610 - PWM2MIX Input 2 Source */ + [0x064B] = 0x0080, /* R1611 - PWM2MIX Input 2 Volume */ + [0x064C] = 0x0000, /* R1612 - PWM2MIX Input 3 Source */ + [0x064D] = 0x0080, /* R1613 - PWM2MIX Input 3 Volume */ + [0x064E] = 0x0000, /* R1614 - PWM2MIX Input 4 Source */ + [0x064F] = 0x0080, /* R1615 - PWM2MIX Input 4 Volume */ + [0x0680] = 0x0000, /* R1664 - OUT1LMIX Input 1 Source */ + [0x0681] = 0x0080, /* R1665 - OUT1LMIX Input 1 Volume */ + [0x0682] = 0x0000, /* R1666 - OUT1LMIX Input 2 Source */ + [0x0683] = 0x0080, /* R1667 - OUT1LMIX Input 2 Volume */ + [0x0684] = 0x0000, /* R1668 - OUT1LMIX Input 3 Source */ + [0x0685] = 0x0080, /* R1669 - OUT1LMIX Input 3 Volume */ + [0x0686] = 0x0000, /* R1670 - OUT1LMIX Input 4 Source */ + [0x0687] = 0x0080, /* R1671 - OUT1LMIX Input 4 Volume */ + [0x0688] = 0x0000, /* R1672 - OUT1RMIX Input 1 Source */ + [0x0689] = 0x0080, /* R1673 - OUT1RMIX Input 1 Volume */ + [0x068A] = 0x0000, /* R1674 - OUT1RMIX Input 2 Source */ + [0x068B] = 0x0080, /* R1675 - OUT1RMIX Input 2 Volume */ + [0x068C] = 0x0000, /* R1676 - OUT1RMIX Input 3 Source */ + [0x068D] = 0x0080, /* R1677 - OUT1RMIX Input 3 Volume */ + [0x068E] = 0x0000, /* R1678 - OUT1RMIX Input 4 Source */ + [0x068F] = 0x0080, /* R1679 - OUT1RMIX Input 4 Volume */ + [0x0690] = 0x0000, /* R1680 - OUT2LMIX Input 1 Source */ + [0x0691] = 0x0080, /* R1681 - OUT2LMIX Input 1 Volume */ + [0x0692] = 0x0000, /* R1682 - OUT2LMIX Input 2 Source */ + [0x0693] = 0x0080, /* R1683 - OUT2LMIX Input 2 Volume */ + [0x0694] = 0x0000, /* R1684 - OUT2LMIX Input 3 Source */ + [0x0695] = 0x0080, /* R1685 - OUT2LMIX Input 3 Volume */ + [0x0696] = 0x0000, /* R1686 - OUT2LMIX Input 4 Source */ + [0x0697] = 0x0080, /* R1687 - OUT2LMIX Input 4 Volume */ + [0x0698] = 0x0000, /* R1688 - OUT2RMIX Input 1 Source */ + [0x0699] = 0x0080, /* R1689 - OUT2RMIX Input 1 Volume */ + [0x069A] = 0x0000, /* R1690 - OUT2RMIX Input 2 Source */ + [0x069B] = 0x0080, /* R1691 - OUT2RMIX Input 2 Volume */ + [0x069C] = 0x0000, /* R1692 - OUT2RMIX Input 3 Source */ + [0x069D] = 0x0080, /* R1693 - OUT2RMIX Input 3 Volume */ + [0x069E] = 0x0000, /* R1694 - OUT2RMIX Input 4 Source */ + [0x069F] = 0x0080, /* R1695 - OUT2RMIX Input 4 Volume */ + [0x06A0] = 0x0000, /* R1696 - OUT3LMIX Input 1 Source */ + [0x06A1] = 0x0080, /* R1697 - OUT3LMIX Input 1 Volume */ + [0x06A2] = 0x0000, /* R1698 - OUT3LMIX Input 2 Source */ + [0x06A3] = 0x0080, /* R1699 - OUT3LMIX Input 2 Volume */ + [0x06A4] = 0x0000, /* R1700 - OUT3LMIX Input 3 Source */ + [0x06A5] = 0x0080, /* R1701 - OUT3LMIX Input 3 Volume */ + [0x06A6] = 0x0000, /* R1702 - OUT3LMIX Input 4 Source */ + [0x06A7] = 0x0080, /* R1703 - OUT3LMIX Input 4 Volume */ + [0x06A8] = 0x0000, /* R1704 - OUT3RMIX Input 1 Source */ + [0x06A9] = 0x0080, /* R1705 - OUT3RMIX Input 1 Volume */ + [0x06AA] = 0x0000, /* R1706 - OUT3RMIX Input 2 Source */ + [0x06AB] = 0x0080, /* R1707 - OUT3RMIX Input 2 Volume */ + [0x06AC] = 0x0000, /* R1708 - OUT3RMIX Input 3 Source */ + [0x06AD] = 0x0080, /* R1709 - OUT3RMIX Input 3 Volume */ + [0x06AE] = 0x0000, /* R1710 - OUT3RMIX Input 4 Source */ + [0x06AF] = 0x0080, /* R1711 - OUT3RMIX Input 4 Volume */ + [0x06B0] = 0x0000, /* R1712 - OUT4LMIX Input 1 Source */ + [0x06B1] = 0x0080, /* R1713 - OUT4LMIX Input 1 Volume */ + [0x06B2] = 0x0000, /* R1714 - OUT4LMIX Input 2 Source */ + [0x06B3] = 0x0080, /* R1715 - OUT4LMIX Input 2 Volume */ + [0x06B4] = 0x0000, /* R1716 - OUT4LMIX Input 3 Source */ + [0x06B5] = 0x0080, /* R1717 - OUT4LMIX Input 3 Volume */ + [0x06B6] = 0x0000, /* R1718 - OUT4LMIX Input 4 Source */ + [0x06B7] = 0x0080, /* R1719 - OUT4LMIX Input 4 Volume */ + [0x06B8] = 0x0000, /* R1720 - OUT4RMIX Input 1 Source */ + [0x06B9] = 0x0080, /* R1721 - OUT4RMIX Input 1 Volume */ + [0x06BA] = 0x0000, /* R1722 - OUT4RMIX Input 2 Source */ + [0x06BB] = 0x0080, /* R1723 - OUT4RMIX Input 2 Volume */ + [0x06BC] = 0x0000, /* R1724 - OUT4RMIX Input 3 Source */ + [0x06BD] = 0x0080, /* R1725 - OUT4RMIX Input 3 Volume */ + [0x06BE] = 0x0000, /* R1726 - OUT4RMIX Input 4 Source */ + [0x06BF] = 0x0080, /* R1727 - OUT4RMIX Input 4 Volume */ + [0x06C0] = 0x0000, /* R1728 - OUT5LMIX Input 1 Source */ + [0x06C1] = 0x0080, /* R1729 - OUT5LMIX Input 1 Volume */ + [0x06C2] = 0x0000, /* R1730 - OUT5LMIX Input 2 Source */ + [0x06C3] = 0x0080, /* R1731 - OUT5LMIX Input 2 Volume */ + [0x06C4] = 0x0000, /* R1732 - OUT5LMIX Input 3 Source */ + [0x06C5] = 0x0080, /* R1733 - OUT5LMIX Input 3 Volume */ + [0x06C6] = 0x0000, /* R1734 - OUT5LMIX Input 4 Source */ + [0x06C7] = 0x0080, /* R1735 - OUT5LMIX Input 4 Volume */ + [0x06C8] = 0x0000, /* R1736 - OUT5RMIX Input 1 Source */ + [0x06C9] = 0x0080, /* R1737 - OUT5RMIX Input 1 Volume */ + [0x06CA] = 0x0000, /* R1738 - OUT5RMIX Input 2 Source */ + [0x06CB] = 0x0080, /* R1739 - OUT5RMIX Input 2 Volume */ + [0x06CC] = 0x0000, /* R1740 - OUT5RMIX Input 3 Source */ + [0x06CD] = 0x0080, /* R1741 - OUT5RMIX Input 3 Volume */ + [0x06CE] = 0x0000, /* R1742 - OUT5RMIX Input 4 Source */ + [0x06CF] = 0x0080, /* R1743 - OUT5RMIX Input 4 Volume */ + [0x06D0] = 0x0000, /* R1744 - OUT6LMIX Input 1 Source */ + [0x06D1] = 0x0080, /* R1745 - OUT6LMIX Input 1 Volume */ + [0x06D2] = 0x0000, /* R1746 - OUT6LMIX Input 2 Source */ + [0x06D3] = 0x0080, /* R1747 - OUT6LMIX Input 2 Volume */ + [0x06D4] = 0x0000, /* R1748 - OUT6LMIX Input 3 Source */ + [0x06D5] = 0x0080, /* R1749 - OUT6LMIX Input 3 Volume */ + [0x06D6] = 0x0000, /* R1750 - OUT6LMIX Input 4 Source */ + [0x06D7] = 0x0080, /* R1751 - OUT6LMIX Input 4 Volume */ + [0x06D8] = 0x0000, /* R1752 - OUT6RMIX Input 1 Source */ + [0x06D9] = 0x0080, /* R1753 - OUT6RMIX Input 1 Volume */ + [0x06DA] = 0x0000, /* R1754 - OUT6RMIX Input 2 Source */ + [0x06DB] = 0x0080, /* R1755 - OUT6RMIX Input 2 Volume */ + [0x06DC] = 0x0000, /* R1756 - OUT6RMIX Input 3 Source */ + [0x06DD] = 0x0080, /* R1757 - OUT6RMIX Input 3 Volume */ + [0x06DE] = 0x0000, /* R1758 - OUT6RMIX Input 4 Source */ + [0x06DF] = 0x0080, /* R1759 - OUT6RMIX Input 4 Volume */ + [0x0700] = 0x0000, /* R1792 - AIF1TX1MIX Input 1 Source */ + [0x0701] = 0x0080, /* R1793 - AIF1TX1MIX Input 1 Volume */ + [0x0702] = 0x0000, /* R1794 - AIF1TX1MIX Input 2 Source */ + [0x0703] = 0x0080, /* R1795 - AIF1TX1MIX Input 2 Volume */ + [0x0704] = 0x0000, /* R1796 - AIF1TX1MIX Input 3 Source */ + [0x0705] = 0x0080, /* R1797 - AIF1TX1MIX Input 3 Volume */ + [0x0706] = 0x0000, /* R1798 - AIF1TX1MIX Input 4 Source */ + [0x0707] = 0x0080, /* R1799 - AIF1TX1MIX Input 4 Volume */ + [0x0708] = 0x0000, /* R1800 - AIF1TX2MIX Input 1 Source */ + [0x0709] = 0x0080, /* R1801 - AIF1TX2MIX Input 1 Volume */ + [0x070A] = 0x0000, /* R1802 - AIF1TX2MIX Input 2 Source */ + [0x070B] = 0x0080, /* R1803 - AIF1TX2MIX Input 2 Volume */ + [0x070C] = 0x0000, /* R1804 - AIF1TX2MIX Input 3 Source */ + [0x070D] = 0x0080, /* R1805 - AIF1TX2MIX Input 3 Volume */ + [0x070E] = 0x0000, /* R1806 - AIF1TX2MIX Input 4 Source */ + [0x070F] = 0x0080, /* R1807 - AIF1TX2MIX Input 4 Volume */ + [0x0710] = 0x0000, /* R1808 - AIF1TX3MIX Input 1 Source */ + [0x0711] = 0x0080, /* R1809 - AIF1TX3MIX Input 1 Volume */ + [0x0712] = 0x0000, /* R1810 - AIF1TX3MIX Input 2 Source */ + [0x0713] = 0x0080, /* R1811 - AIF1TX3MIX Input 2 Volume */ + [0x0714] = 0x0000, /* R1812 - AIF1TX3MIX Input 3 Source */ + [0x0715] = 0x0080, /* R1813 - AIF1TX3MIX Input 3 Volume */ + [0x0716] = 0x0000, /* R1814 - AIF1TX3MIX Input 4 Source */ + [0x0717] = 0x0080, /* R1815 - AIF1TX3MIX Input 4 Volume */ + [0x0718] = 0x0000, /* R1816 - AIF1TX4MIX Input 1 Source */ + [0x0719] = 0x0080, /* R1817 - AIF1TX4MIX Input 1 Volume */ + [0x071A] = 0x0000, /* R1818 - AIF1TX4MIX Input 2 Source */ + [0x071B] = 0x0080, /* R1819 - AIF1TX4MIX Input 2 Volume */ + [0x071C] = 0x0000, /* R1820 - AIF1TX4MIX Input 3 Source */ + [0x071D] = 0x0080, /* R1821 - AIF1TX4MIX Input 3 Volume */ + [0x071E] = 0x0000, /* R1822 - AIF1TX4MIX Input 4 Source */ + [0x071F] = 0x0080, /* R1823 - AIF1TX4MIX Input 4 Volume */ + [0x0720] = 0x0000, /* R1824 - AIF1TX5MIX Input 1 Source */ + [0x0721] = 0x0080, /* R1825 - AIF1TX5MIX Input 1 Volume */ + [0x0722] = 0x0000, /* R1826 - AIF1TX5MIX Input 2 Source */ + [0x0723] = 0x0080, /* R1827 - AIF1TX5MIX Input 2 Volume */ + [0x0724] = 0x0000, /* R1828 - AIF1TX5MIX Input 3 Source */ + [0x0725] = 0x0080, /* R1829 - AIF1TX5MIX Input 3 Volume */ + [0x0726] = 0x0000, /* R1830 - AIF1TX5MIX Input 4 Source */ + [0x0727] = 0x0080, /* R1831 - AIF1TX5MIX Input 4 Volume */ + [0x0728] = 0x0000, /* R1832 - AIF1TX6MIX Input 1 Source */ + [0x0729] = 0x0080, /* R1833 - AIF1TX6MIX Input 1 Volume */ + [0x072A] = 0x0000, /* R1834 - AIF1TX6MIX Input 2 Source */ + [0x072B] = 0x0080, /* R1835 - AIF1TX6MIX Input 2 Volume */ + [0x072C] = 0x0000, /* R1836 - AIF1TX6MIX Input 3 Source */ + [0x072D] = 0x0080, /* R1837 - AIF1TX6MIX Input 3 Volume */ + [0x072E] = 0x0000, /* R1838 - AIF1TX6MIX Input 4 Source */ + [0x072F] = 0x0080, /* R1839 - AIF1TX6MIX Input 4 Volume */ + [0x0730] = 0x0000, /* R1840 - AIF1TX7MIX Input 1 Source */ + [0x0731] = 0x0080, /* R1841 - AIF1TX7MIX Input 1 Volume */ + [0x0732] = 0x0000, /* R1842 - AIF1TX7MIX Input 2 Source */ + [0x0733] = 0x0080, /* R1843 - AIF1TX7MIX Input 2 Volume */ + [0x0734] = 0x0000, /* R1844 - AIF1TX7MIX Input 3 Source */ + [0x0735] = 0x0080, /* R1845 - AIF1TX7MIX Input 3 Volume */ + [0x0736] = 0x0000, /* R1846 - AIF1TX7MIX Input 4 Source */ + [0x0737] = 0x0080, /* R1847 - AIF1TX7MIX Input 4 Volume */ + [0x0738] = 0x0000, /* R1848 - AIF1TX8MIX Input 1 Source */ + [0x0739] = 0x0080, /* R1849 - AIF1TX8MIX Input 1 Volume */ + [0x073A] = 0x0000, /* R1850 - AIF1TX8MIX Input 2 Source */ + [0x073B] = 0x0080, /* R1851 - AIF1TX8MIX Input 2 Volume */ + [0x073C] = 0x0000, /* R1852 - AIF1TX8MIX Input 3 Source */ + [0x073D] = 0x0080, /* R1853 - AIF1TX8MIX Input 3 Volume */ + [0x073E] = 0x0000, /* R1854 - AIF1TX8MIX Input 4 Source */ + [0x073F] = 0x0080, /* R1855 - AIF1TX8MIX Input 4 Volume */ + [0x0740] = 0x0000, /* R1856 - AIF2TX1MIX Input 1 Source */ + [0x0741] = 0x0080, /* R1857 - AIF2TX1MIX Input 1 Volume */ + [0x0742] = 0x0000, /* R1858 - AIF2TX1MIX Input 2 Source */ + [0x0743] = 0x0080, /* R1859 - AIF2TX1MIX Input 2 Volume */ + [0x0744] = 0x0000, /* R1860 - AIF2TX1MIX Input 3 Source */ + [0x0745] = 0x0080, /* R1861 - AIF2TX1MIX Input 3 Volume */ + [0x0746] = 0x0000, /* R1862 - AIF2TX1MIX Input 4 Source */ + [0x0747] = 0x0080, /* R1863 - AIF2TX1MIX Input 4 Volume */ + [0x0748] = 0x0000, /* R1864 - AIF2TX2MIX Input 1 Source */ + [0x0749] = 0x0080, /* R1865 - AIF2TX2MIX Input 1 Volume */ + [0x074A] = 0x0000, /* R1866 - AIF2TX2MIX Input 2 Source */ + [0x074B] = 0x0080, /* R1867 - AIF2TX2MIX Input 2 Volume */ + [0x074C] = 0x0000, /* R1868 - AIF2TX2MIX Input 3 Source */ + [0x074D] = 0x0080, /* R1869 - AIF2TX2MIX Input 3 Volume */ + [0x074E] = 0x0000, /* R1870 - AIF2TX2MIX Input 4 Source */ + [0x074F] = 0x0080, /* R1871 - AIF2TX2MIX Input 4 Volume */ + [0x0780] = 0x0000, /* R1920 - AIF3TX1MIX Input 1 Source */ + [0x0781] = 0x0080, /* R1921 - AIF3TX1MIX Input 1 Volume */ + [0x0782] = 0x0000, /* R1922 - AIF3TX1MIX Input 2 Source */ + [0x0783] = 0x0080, /* R1923 - AIF3TX1MIX Input 2 Volume */ + [0x0784] = 0x0000, /* R1924 - AIF3TX1MIX Input 3 Source */ + [0x0785] = 0x0080, /* R1925 - AIF3TX1MIX Input 3 Volume */ + [0x0786] = 0x0000, /* R1926 - AIF3TX1MIX Input 4 Source */ + [0x0787] = 0x0080, /* R1927 - AIF3TX1MIX Input 4 Volume */ + [0x0788] = 0x0000, /* R1928 - AIF3TX2MIX Input 1 Source */ + [0x0789] = 0x0080, /* R1929 - AIF3TX2MIX Input 1 Volume */ + [0x078A] = 0x0000, /* R1930 - AIF3TX2MIX Input 2 Source */ + [0x078B] = 0x0080, /* R1931 - AIF3TX2MIX Input 2 Volume */ + [0x078C] = 0x0000, /* R1932 - AIF3TX2MIX Input 3 Source */ + [0x078D] = 0x0080, /* R1933 - AIF3TX2MIX Input 3 Volume */ + [0x078E] = 0x0000, /* R1934 - AIF3TX2MIX Input 4 Source */ + [0x078F] = 0x0080, /* R1935 - AIF3TX2MIX Input 4 Volume */ + [0x0880] = 0x0000, /* R2176 - EQ1MIX Input 1 Source */ + [0x0881] = 0x0080, /* R2177 - EQ1MIX Input 1 Volume */ + [0x0882] = 0x0000, /* R2178 - EQ1MIX Input 2 Source */ + [0x0883] = 0x0080, /* R2179 - EQ1MIX Input 2 Volume */ + [0x0884] = 0x0000, /* R2180 - EQ1MIX Input 3 Source */ + [0x0885] = 0x0080, /* R2181 - EQ1MIX Input 3 Volume */ + [0x0886] = 0x0000, /* R2182 - EQ1MIX Input 4 Source */ + [0x0887] = 0x0080, /* R2183 - EQ1MIX Input 4 Volume */ + [0x0888] = 0x0000, /* R2184 - EQ2MIX Input 1 Source */ + [0x0889] = 0x0080, /* R2185 - EQ2MIX Input 1 Volume */ + [0x088A] = 0x0000, /* R2186 - EQ2MIX Input 2 Source */ + [0x088B] = 0x0080, /* R2187 - EQ2MIX Input 2 Volume */ + [0x088C] = 0x0000, /* R2188 - EQ2MIX Input 3 Source */ + [0x088D] = 0x0080, /* R2189 - EQ2MIX Input 3 Volume */ + [0x088E] = 0x0000, /* R2190 - EQ2MIX Input 4 Source */ + [0x088F] = 0x0080, /* R2191 - EQ2MIX Input 4 Volume */ + [0x0890] = 0x0000, /* R2192 - EQ3MIX Input 1 Source */ + [0x0891] = 0x0080, /* R2193 - EQ3MIX Input 1 Volume */ + [0x0892] = 0x0000, /* R2194 - EQ3MIX Input 2 Source */ + [0x0893] = 0x0080, /* R2195 - EQ3MIX Input 2 Volume */ + [0x0894] = 0x0000, /* R2196 - EQ3MIX Input 3 Source */ + [0x0895] = 0x0080, /* R2197 - EQ3MIX Input 3 Volume */ + [0x0896] = 0x0000, /* R2198 - EQ3MIX Input 4 Source */ + [0x0897] = 0x0080, /* R2199 - EQ3MIX Input 4 Volume */ + [0x0898] = 0x0000, /* R2200 - EQ4MIX Input 1 Source */ + [0x0899] = 0x0080, /* R2201 - EQ4MIX Input 1 Volume */ + [0x089A] = 0x0000, /* R2202 - EQ4MIX Input 2 Source */ + [0x089B] = 0x0080, /* R2203 - EQ4MIX Input 2 Volume */ + [0x089C] = 0x0000, /* R2204 - EQ4MIX Input 3 Source */ + [0x089D] = 0x0080, /* R2205 - EQ4MIX Input 3 Volume */ + [0x089E] = 0x0000, /* R2206 - EQ4MIX Input 4 Source */ + [0x089F] = 0x0080, /* R2207 - EQ4MIX Input 4 Volume */ + [0x08C0] = 0x0000, /* R2240 - DRC1LMIX Input 1 Source */ + [0x08C1] = 0x0080, /* R2241 - DRC1LMIX Input 1 Volume */ + [0x08C2] = 0x0000, /* R2242 - DRC1LMIX Input 2 Source */ + [0x08C3] = 0x0080, /* R2243 - DRC1LMIX Input 2 Volume */ + [0x08C4] = 0x0000, /* R2244 - DRC1LMIX Input 3 Source */ + [0x08C5] = 0x0080, /* R2245 - DRC1LMIX Input 3 Volume */ + [0x08C6] = 0x0000, /* R2246 - DRC1LMIX Input 4 Source */ + [0x08C7] = 0x0080, /* R2247 - DRC1LMIX Input 4 Volume */ + [0x08C8] = 0x0000, /* R2248 - DRC1RMIX Input 1 Source */ + [0x08C9] = 0x0080, /* R2249 - DRC1RMIX Input 1 Volume */ + [0x08CA] = 0x0000, /* R2250 - DRC1RMIX Input 2 Source */ + [0x08CB] = 0x0080, /* R2251 - DRC1RMIX Input 2 Volume */ + [0x08CC] = 0x0000, /* R2252 - DRC1RMIX Input 3 Source */ + [0x08CD] = 0x0080, /* R2253 - DRC1RMIX Input 3 Volume */ + [0x08CE] = 0x0000, /* R2254 - DRC1RMIX Input 4 Source */ + [0x08CF] = 0x0080, /* R2255 - DRC1RMIX Input 4 Volume */ + [0x0900] = 0x0000, /* R2304 - HPLP1MIX Input 1 Source */ + [0x0901] = 0x0080, /* R2305 - HPLP1MIX Input 1 Volume */ + [0x0902] = 0x0000, /* R2306 - HPLP1MIX Input 2 Source */ + [0x0903] = 0x0080, /* R2307 - HPLP1MIX Input 2 Volume */ + [0x0904] = 0x0000, /* R2308 - HPLP1MIX Input 3 Source */ + [0x0905] = 0x0080, /* R2309 - HPLP1MIX Input 3 Volume */ + [0x0906] = 0x0000, /* R2310 - HPLP1MIX Input 4 Source */ + [0x0907] = 0x0080, /* R2311 - HPLP1MIX Input 4 Volume */ + [0x0908] = 0x0000, /* R2312 - HPLP2MIX Input 1 Source */ + [0x0909] = 0x0080, /* R2313 - HPLP2MIX Input 1 Volume */ + [0x090A] = 0x0000, /* R2314 - HPLP2MIX Input 2 Source */ + [0x090B] = 0x0080, /* R2315 - HPLP2MIX Input 2 Volume */ + [0x090C] = 0x0000, /* R2316 - HPLP2MIX Input 3 Source */ + [0x090D] = 0x0080, /* R2317 - HPLP2MIX Input 3 Volume */ + [0x090E] = 0x0000, /* R2318 - HPLP2MIX Input 4 Source */ + [0x090F] = 0x0080, /* R2319 - HPLP2MIX Input 4 Volume */ + [0x0910] = 0x0000, /* R2320 - HPLP3MIX Input 1 Source */ + [0x0911] = 0x0080, /* R2321 - HPLP3MIX Input 1 Volume */ + [0x0912] = 0x0000, /* R2322 - HPLP3MIX Input 2 Source */ + [0x0913] = 0x0080, /* R2323 - HPLP3MIX Input 2 Volume */ + [0x0914] = 0x0000, /* R2324 - HPLP3MIX Input 3 Source */ + [0x0915] = 0x0080, /* R2325 - HPLP3MIX Input 3 Volume */ + [0x0916] = 0x0000, /* R2326 - HPLP3MIX Input 4 Source */ + [0x0917] = 0x0080, /* R2327 - HPLP3MIX Input 4 Volume */ + [0x0918] = 0x0000, /* R2328 - HPLP4MIX Input 1 Source */ + [0x0919] = 0x0080, /* R2329 - HPLP4MIX Input 1 Volume */ + [0x091A] = 0x0000, /* R2330 - HPLP4MIX Input 2 Source */ + [0x091B] = 0x0080, /* R2331 - HPLP4MIX Input 2 Volume */ + [0x091C] = 0x0000, /* R2332 - HPLP4MIX Input 3 Source */ + [0x091D] = 0x0080, /* R2333 - HPLP4MIX Input 3 Volume */ + [0x091E] = 0x0000, /* R2334 - HPLP4MIX Input 4 Source */ + [0x091F] = 0x0080, /* R2335 - HPLP4MIX Input 4 Volume */ + [0x0940] = 0x0000, /* R2368 - DSP1LMIX Input 1 Source */ + [0x0941] = 0x0080, /* R2369 - DSP1LMIX Input 1 Volume */ + [0x0942] = 0x0000, /* R2370 - DSP1LMIX Input 2 Source */ + [0x0943] = 0x0080, /* R2371 - DSP1LMIX Input 2 Volume */ + [0x0944] = 0x0000, /* R2372 - DSP1LMIX Input 3 Source */ + [0x0945] = 0x0080, /* R2373 - DSP1LMIX Input 3 Volume */ + [0x0946] = 0x0000, /* R2374 - DSP1LMIX Input 4 Source */ + [0x0947] = 0x0080, /* R2375 - DSP1LMIX Input 4 Volume */ + [0x0948] = 0x0000, /* R2376 - DSP1RMIX Input 1 Source */ + [0x0949] = 0x0080, /* R2377 - DSP1RMIX Input 1 Volume */ + [0x094A] = 0x0000, /* R2378 - DSP1RMIX Input 2 Source */ + [0x094B] = 0x0080, /* R2379 - DSP1RMIX Input 2 Volume */ + [0x094C] = 0x0000, /* R2380 - DSP1RMIX Input 3 Source */ + [0x094D] = 0x0080, /* R2381 - DSP1RMIX Input 3 Volume */ + [0x094E] = 0x0000, /* R2382 - DSP1RMIX Input 4 Source */ + [0x094F] = 0x0080, /* R2383 - DSP1RMIX Input 4 Volume */ + [0x0950] = 0x0000, /* R2384 - DSP1AUX1MIX Input 1 Source */ + [0x0958] = 0x0000, /* R2392 - DSP1AUX2MIX Input 1 Source */ + [0x0960] = 0x0000, /* R2400 - DSP1AUX3MIX Input 1 Source */ + [0x0968] = 0x0000, /* R2408 - DSP1AUX4MIX Input 1 Source */ + [0x0970] = 0x0000, /* R2416 - DSP1AUX5MIX Input 1 Source */ + [0x0978] = 0x0000, /* R2424 - DSP1AUX6MIX Input 1 Source */ + [0x0980] = 0x0000, /* R2432 - DSP2LMIX Input 1 Source */ + [0x0981] = 0x0080, /* R2433 - DSP2LMIX Input 1 Volume */ + [0x0982] = 0x0000, /* R2434 - DSP2LMIX Input 2 Source */ + [0x0983] = 0x0080, /* R2435 - DSP2LMIX Input 2 Volume */ + [0x0984] = 0x0000, /* R2436 - DSP2LMIX Input 3 Source */ + [0x0985] = 0x0080, /* R2437 - DSP2LMIX Input 3 Volume */ + [0x0986] = 0x0000, /* R2438 - DSP2LMIX Input 4 Source */ + [0x0987] = 0x0080, /* R2439 - DSP2LMIX Input 4 Volume */ + [0x0988] = 0x0000, /* R2440 - DSP2RMIX Input 1 Source */ + [0x0989] = 0x0080, /* R2441 - DSP2RMIX Input 1 Volume */ + [0x098A] = 0x0000, /* R2442 - DSP2RMIX Input 2 Source */ + [0x098B] = 0x0080, /* R2443 - DSP2RMIX Input 2 Volume */ + [0x098C] = 0x0000, /* R2444 - DSP2RMIX Input 3 Source */ + [0x098D] = 0x0080, /* R2445 - DSP2RMIX Input 3 Volume */ + [0x098E] = 0x0000, /* R2446 - DSP2RMIX Input 4 Source */ + [0x098F] = 0x0080, /* R2447 - DSP2RMIX Input 4 Volume */ + [0x0990] = 0x0000, /* R2448 - DSP2AUX1MIX Input 1 Source */ + [0x0998] = 0x0000, /* R2456 - DSP2AUX2MIX Input 1 Source */ + [0x09A0] = 0x0000, /* R2464 - DSP2AUX3MIX Input 1 Source */ + [0x09A8] = 0x0000, /* R2472 - DSP2AUX4MIX Input 1 Source */ + [0x09B0] = 0x0000, /* R2480 - DSP2AUX5MIX Input 1 Source */ + [0x09B8] = 0x0000, /* R2488 - DSP2AUX6MIX Input 1 Source */ + [0x09C0] = 0x0000, /* R2496 - DSP3LMIX Input 1 Source */ + [0x09C1] = 0x0080, /* R2497 - DSP3LMIX Input 1 Volume */ + [0x09C2] = 0x0000, /* R2498 - DSP3LMIX Input 2 Source */ + [0x09C3] = 0x0080, /* R2499 - DSP3LMIX Input 2 Volume */ + [0x09C4] = 0x0000, /* R2500 - DSP3LMIX Input 3 Source */ + [0x09C5] = 0x0080, /* R2501 - DSP3LMIX Input 3 Volume */ + [0x09C6] = 0x0000, /* R2502 - DSP3LMIX Input 4 Source */ + [0x09C7] = 0x0080, /* R2503 - DSP3LMIX Input 4 Volume */ + [0x09C8] = 0x0000, /* R2504 - DSP3RMIX Input 1 Source */ + [0x09C9] = 0x0080, /* R2505 - DSP3RMIX Input 1 Volume */ + [0x09CA] = 0x0000, /* R2506 - DSP3RMIX Input 2 Source */ + [0x09CB] = 0x0080, /* R2507 - DSP3RMIX Input 2 Volume */ + [0x09CC] = 0x0000, /* R2508 - DSP3RMIX Input 3 Source */ + [0x09CD] = 0x0080, /* R2509 - DSP3RMIX Input 3 Volume */ + [0x09CE] = 0x0000, /* R2510 - DSP3RMIX Input 4 Source */ + [0x09CF] = 0x0080, /* R2511 - DSP3RMIX Input 4 Volume */ + [0x09D0] = 0x0000, /* R2512 - DSP3AUX1MIX Input 1 Source */ + [0x09D8] = 0x0000, /* R2520 - DSP3AUX2MIX Input 1 Source */ + [0x09E0] = 0x0000, /* R2528 - DSP3AUX3MIX Input 1 Source */ + [0x09E8] = 0x0000, /* R2536 - DSP3AUX4MIX Input 1 Source */ + [0x09F0] = 0x0000, /* R2544 - DSP3AUX5MIX Input 1 Source */ + [0x09F8] = 0x0000, /* R2552 - DSP3AUX6MIX Input 1 Source */ + [0x0A80] = 0x0000, /* R2688 - ASRC1LMIX Input 1 Source */ + [0x0A88] = 0x0000, /* R2696 - ASRC1RMIX Input 1 Source */ + [0x0A90] = 0x0000, /* R2704 - ASRC2LMIX Input 1 Source */ + [0x0A98] = 0x0000, /* R2712 - ASRC2RMIX Input 1 Source */ + [0x0B00] = 0x0000, /* R2816 - ISRC1DEC1MIX Input 1 Source */ + [0x0B08] = 0x0000, /* R2824 - ISRC1DEC2MIX Input 1 Source */ + [0x0B10] = 0x0000, /* R2832 - ISRC1DEC3MIX Input 1 Source */ + [0x0B18] = 0x0000, /* R2840 - ISRC1DEC4MIX Input 1 Source */ + [0x0B20] = 0x0000, /* R2848 - ISRC1INT1MIX Input 1 Source */ + [0x0B28] = 0x0000, /* R2856 - ISRC1INT2MIX Input 1 Source */ + [0x0B30] = 0x0000, /* R2864 - ISRC1INT3MIX Input 1 Source */ + [0x0B38] = 0x0000, /* R2872 - ISRC1INT4MIX Input 1 Source */ + [0x0B40] = 0x0000, /* R2880 - ISRC2DEC1MIX Input 1 Source */ + [0x0B48] = 0x0000, /* R2888 - ISRC2DEC2MIX Input 1 Source */ + [0x0B50] = 0x0000, /* R2896 - ISRC2DEC3MIX Input 1 Source */ + [0x0B58] = 0x0000, /* R2904 - ISRC2DEC4MIX Input 1 Source */ + [0x0B60] = 0x0000, /* R2912 - ISRC2INT1MIX Input 1 Source */ + [0x0B68] = 0x0000, /* R2920 - ISRC2INT2MIX Input 1 Source */ + [0x0B70] = 0x0000, /* R2928 - ISRC2INT3MIX Input 1 Source */ + [0x0B78] = 0x0000, /* R2936 - ISRC2INT4MIX Input 1 Source */ + [0x0C00] = 0xA001, /* R3072 - GPIO CTRL 1 */ + [0x0C01] = 0xA001, /* R3073 - GPIO CTRL 2 */ + [0x0C02] = 0xA001, /* R3074 - GPIO CTRL 3 */ + [0x0C03] = 0xA001, /* R3075 - GPIO CTRL 4 */ + [0x0C04] = 0xA001, /* R3076 - GPIO CTRL 5 */ + [0x0C05] = 0xA001, /* R3077 - GPIO CTRL 6 */ + [0x0C23] = 0x4003, /* R3107 - Misc Pad Ctrl 1 */ + [0x0C24] = 0x0000, /* R3108 - Misc Pad Ctrl 2 */ + [0x0C25] = 0x0000, /* R3109 - Misc Pad Ctrl 3 */ + [0x0C26] = 0x0000, /* R3110 - Misc Pad Ctrl 4 */ + [0x0C27] = 0x0000, /* R3111 - Misc Pad Ctrl 5 */ + [0x0C28] = 0x0000, /* R3112 - Misc GPIO 1 */ + [0x0D00] = 0x0000, /* R3328 - Interrupt Status 1 */ + [0x0D01] = 0x0000, /* R3329 - Interrupt Status 2 */ + [0x0D02] = 0x0000, /* R3330 - Interrupt Status 3 */ + [0x0D03] = 0x0000, /* R3331 - Interrupt Status 4 */ + [0x0D04] = 0x0000, /* R3332 - Interrupt Raw Status 2 */ + [0x0D05] = 0x0000, /* R3333 - Interrupt Raw Status 3 */ + [0x0D06] = 0x0000, /* R3334 - Interrupt Raw Status 4 */ + [0x0D07] = 0xFFFF, /* R3335 - Interrupt Status 1 Mask */ + [0x0D08] = 0xFFFF, /* R3336 - Interrupt Status 2 Mask */ + [0x0D09] = 0xFFFF, /* R3337 - Interrupt Status 3 Mask */ + [0x0D0A] = 0xFFFF, /* R3338 - Interrupt Status 4 Mask */ + [0x0D1F] = 0x0000, /* R3359 - Interrupt Control */ + [0x0D20] = 0xFFFF, /* R3360 - IRQ Debounce 1 */ + [0x0D21] = 0xFFFF, /* R3361 - IRQ Debounce 2 */ + [0x0E00] = 0x0000, /* R3584 - FX_Ctrl */ + [0x0E10] = 0x6318, /* R3600 - EQ1_1 */ + [0x0E11] = 0x6300, /* R3601 - EQ1_2 */ + [0x0E12] = 0x0FC8, /* R3602 - EQ1_3 */ + [0x0E13] = 0x03FE, /* R3603 - EQ1_4 */ + [0x0E14] = 0x00E0, /* R3604 - EQ1_5 */ + [0x0E15] = 0x1EC4, /* R3605 - EQ1_6 */ + [0x0E16] = 0xF136, /* R3606 - EQ1_7 */ + [0x0E17] = 0x0409, /* R3607 - EQ1_8 */ + [0x0E18] = 0x04CC, /* R3608 - EQ1_9 */ + [0x0E19] = 0x1C9B, /* R3609 - EQ1_10 */ + [0x0E1A] = 0xF337, /* R3610 - EQ1_11 */ + [0x0E1B] = 0x040B, /* R3611 - EQ1_12 */ + [0x0E1C] = 0x0CBB, /* R3612 - EQ1_13 */ + [0x0E1D] = 0x16F8, /* R3613 - EQ1_14 */ + [0x0E1E] = 0xF7D9, /* R3614 - EQ1_15 */ + [0x0E1F] = 0x040A, /* R3615 - EQ1_16 */ + [0x0E20] = 0x1F14, /* R3616 - EQ1_17 */ + [0x0E21] = 0x058C, /* R3617 - EQ1_18 */ + [0x0E22] = 0x0563, /* R3618 - EQ1_19 */ + [0x0E23] = 0x4000, /* R3619 - EQ1_20 */ + [0x0E26] = 0x6318, /* R3622 - EQ2_1 */ + [0x0E27] = 0x6300, /* R3623 - EQ2_2 */ + [0x0E28] = 0x0FC8, /* R3624 - EQ2_3 */ + [0x0E29] = 0x03FE, /* R3625 - EQ2_4 */ + [0x0E2A] = 0x00E0, /* R3626 - EQ2_5 */ + [0x0E2B] = 0x1EC4, /* R3627 - EQ2_6 */ + [0x0E2C] = 0xF136, /* R3628 - EQ2_7 */ + [0x0E2D] = 0x0409, /* R3629 - EQ2_8 */ + [0x0E2E] = 0x04CC, /* R3630 - EQ2_9 */ + [0x0E2F] = 0x1C9B, /* R3631 - EQ2_10 */ + [0x0E30] = 0xF337, /* R3632 - EQ2_11 */ + [0x0E31] = 0x040B, /* R3633 - EQ2_12 */ + [0x0E32] = 0x0CBB, /* R3634 - EQ2_13 */ + [0x0E33] = 0x16F8, /* R3635 - EQ2_14 */ + [0x0E34] = 0xF7D9, /* R3636 - EQ2_15 */ + [0x0E35] = 0x040A, /* R3637 - EQ2_16 */ + [0x0E36] = 0x1F14, /* R3638 - EQ2_17 */ + [0x0E37] = 0x058C, /* R3639 - EQ2_18 */ + [0x0E38] = 0x0563, /* R3640 - EQ2_19 */ + [0x0E39] = 0x4000, /* R3641 - EQ2_20 */ + [0x0E3C] = 0x6318, /* R3644 - EQ3_1 */ + [0x0E3D] = 0x6300, /* R3645 - EQ3_2 */ + [0x0E3E] = 0x0FC8, /* R3646 - EQ3_3 */ + [0x0E3F] = 0x03FE, /* R3647 - EQ3_4 */ + [0x0E40] = 0x00E0, /* R3648 - EQ3_5 */ + [0x0E41] = 0x1EC4, /* R3649 - EQ3_6 */ + [0x0E42] = 0xF136, /* R3650 - EQ3_7 */ + [0x0E43] = 0x0409, /* R3651 - EQ3_8 */ + [0x0E44] = 0x04CC, /* R3652 - EQ3_9 */ + [0x0E45] = 0x1C9B, /* R3653 - EQ3_10 */ + [0x0E46] = 0xF337, /* R3654 - EQ3_11 */ + [0x0E47] = 0x040B, /* R3655 - EQ3_12 */ + [0x0E48] = 0x0CBB, /* R3656 - EQ3_13 */ + [0x0E49] = 0x16F8, /* R3657 - EQ3_14 */ + [0x0E4A] = 0xF7D9, /* R3658 - EQ3_15 */ + [0x0E4B] = 0x040A, /* R3659 - EQ3_16 */ + [0x0E4C] = 0x1F14, /* R3660 - EQ3_17 */ + [0x0E4D] = 0x058C, /* R3661 - EQ3_18 */ + [0x0E4E] = 0x0563, /* R3662 - EQ3_19 */ + [0x0E4F] = 0x4000, /* R3663 - EQ3_20 */ + [0x0E52] = 0x6318, /* R3666 - EQ4_1 */ + [0x0E53] = 0x6300, /* R3667 - EQ4_2 */ + [0x0E54] = 0x0FC8, /* R3668 - EQ4_3 */ + [0x0E55] = 0x03FE, /* R3669 - EQ4_4 */ + [0x0E56] = 0x00E0, /* R3670 - EQ4_5 */ + [0x0E57] = 0x1EC4, /* R3671 - EQ4_6 */ + [0x0E58] = 0xF136, /* R3672 - EQ4_7 */ + [0x0E59] = 0x0409, /* R3673 - EQ4_8 */ + [0x0E5A] = 0x04CC, /* R3674 - EQ4_9 */ + [0x0E5B] = 0x1C9B, /* R3675 - EQ4_10 */ + [0x0E5C] = 0xF337, /* R3676 - EQ4_11 */ + [0x0E5D] = 0x040B, /* R3677 - EQ4_12 */ + [0x0E5E] = 0x0CBB, /* R3678 - EQ4_13 */ + [0x0E5F] = 0x16F8, /* R3679 - EQ4_14 */ + [0x0E60] = 0xF7D9, /* R3680 - EQ4_15 */ + [0x0E61] = 0x040A, /* R3681 - EQ4_16 */ + [0x0E62] = 0x1F14, /* R3682 - EQ4_17 */ + [0x0E63] = 0x058C, /* R3683 - EQ4_18 */ + [0x0E64] = 0x0563, /* R3684 - EQ4_19 */ + [0x0E65] = 0x4000, /* R3685 - EQ4_20 */ + [0x0E80] = 0x0018, /* R3712 - DRC1 ctrl1 */ + [0x0E81] = 0x0933, /* R3713 - DRC1 ctrl2 */ + [0x0E82] = 0x0018, /* R3714 - DRC1 ctrl3 */ + [0x0E83] = 0x0000, /* R3715 - DRC1 ctrl4 */ + [0x0E84] = 0x0000, /* R3716 - DRC1 ctrl5 */ + [0x0EC0] = 0x0000, /* R3776 - HPLPF1_1 */ + [0x0EC1] = 0x0000, /* R3777 - HPLPF1_2 */ + [0x0EC4] = 0x0000, /* R3780 - HPLPF2_1 */ + [0x0EC5] = 0x0000, /* R3781 - HPLPF2_2 */ + [0x0EC8] = 0x0000, /* R3784 - HPLPF3_1 */ + [0x0EC9] = 0x0000, /* R3785 - HPLPF3_2 */ + [0x0ECC] = 0x0000, /* R3788 - HPLPF4_1 */ + [0x0ECD] = 0x0000, /* R3789 - HPLPF4_2 */ + [0x4000] = 0x0000, /* R16384 - DSP1 DM 0 */ + [0x4001] = 0x0000, /* R16385 - DSP1 DM 1 */ + [0x4002] = 0x0000, /* R16386 - DSP1 DM 2 */ + [0x4003] = 0x0000, /* R16387 - DSP1 DM 3 */ + [0x41FC] = 0x0000, /* R16892 - DSP1 DM 508 */ + [0x41FD] = 0x0000, /* R16893 - DSP1 DM 509 */ + [0x41FE] = 0x0000, /* R16894 - DSP1 DM 510 */ + [0x41FF] = 0x0000, /* R16895 - DSP1 DM 511 */ + [0x4800] = 0x0000, /* R18432 - DSP1 PM 0 */ + [0x4801] = 0x0000, /* R18433 - DSP1 PM 1 */ + [0x4802] = 0x0000, /* R18434 - DSP1 PM 2 */ + [0x4803] = 0x0000, /* R18435 - DSP1 PM 3 */ + [0x4804] = 0x0000, /* R18436 - DSP1 PM 4 */ + [0x4805] = 0x0000, /* R18437 - DSP1 PM 5 */ + [0x4DFA] = 0x0000, /* R19962 - DSP1 PM 1530 */ + [0x4DFB] = 0x0000, /* R19963 - DSP1 PM 1531 */ + [0x4DFC] = 0x0000, /* R19964 - DSP1 PM 1532 */ + [0x4DFD] = 0x0000, /* R19965 - DSP1 PM 1533 */ + [0x4DFE] = 0x0000, /* R19966 - DSP1 PM 1534 */ + [0x4DFF] = 0x0000, /* R19967 - DSP1 PM 1535 */ + [0x5000] = 0x0000, /* R20480 - DSP1 ZM 0 */ + [0x5001] = 0x0000, /* R20481 - DSP1 ZM 1 */ + [0x5002] = 0x0000, /* R20482 - DSP1 ZM 2 */ + [0x5003] = 0x0000, /* R20483 - DSP1 ZM 3 */ + [0x57FC] = 0x0000, /* R22524 - DSP1 ZM 2044 */ + [0x57FD] = 0x0000, /* R22525 - DSP1 ZM 2045 */ + [0x57FE] = 0x0000, /* R22526 - DSP1 ZM 2046 */ + [0x57FF] = 0x0000, /* R22527 - DSP1 ZM 2047 */ + [0x6000] = 0x0000, /* R24576 - DSP2 DM 0 */ + [0x6001] = 0x0000, /* R24577 - DSP2 DM 1 */ + [0x6002] = 0x0000, /* R24578 - DSP2 DM 2 */ + [0x6003] = 0x0000, /* R24579 - DSP2 DM 3 */ + [0x61FC] = 0x0000, /* R25084 - DSP2 DM 508 */ + [0x61FD] = 0x0000, /* R25085 - DSP2 DM 509 */ + [0x61FE] = 0x0000, /* R25086 - DSP2 DM 510 */ + [0x61FF] = 0x0000, /* R25087 - DSP2 DM 511 */ + [0x6800] = 0x0000, /* R26624 - DSP2 PM 0 */ + [0x6801] = 0x0000, /* R26625 - DSP2 PM 1 */ + [0x6802] = 0x0000, /* R26626 - DSP2 PM 2 */ + [0x6803] = 0x0000, /* R26627 - DSP2 PM 3 */ + [0x6804] = 0x0000, /* R26628 - DSP2 PM 4 */ + [0x6805] = 0x0000, /* R26629 - DSP2 PM 5 */ + [0x6DFA] = 0x0000, /* R28154 - DSP2 PM 1530 */ + [0x6DFB] = 0x0000, /* R28155 - DSP2 PM 1531 */ + [0x6DFC] = 0x0000, /* R28156 - DSP2 PM 1532 */ + [0x6DFD] = 0x0000, /* R28157 - DSP2 PM 1533 */ + [0x6DFE] = 0x0000, /* R28158 - DSP2 PM 1534 */ + [0x6DFF] = 0x0000, /* R28159 - DSP2 PM 1535 */ + [0x7000] = 0x0000, /* R28672 - DSP2 ZM 0 */ + [0x7001] = 0x0000, /* R28673 - DSP2 ZM 1 */ + [0x7002] = 0x0000, /* R28674 - DSP2 ZM 2 */ + [0x7003] = 0x0000, /* R28675 - DSP2 ZM 3 */ + [0x77FC] = 0x0000, /* R30716 - DSP2 ZM 2044 */ + [0x77FD] = 0x0000, /* R30717 - DSP2 ZM 2045 */ + [0x77FE] = 0x0000, /* R30718 - DSP2 ZM 2046 */ + [0x77FF] = 0x0000, /* R30719 - DSP2 ZM 2047 */ + [0x8000] = 0x0000, /* R32768 - DSP3 DM 0 */ + [0x8001] = 0x0000, /* R32769 - DSP3 DM 1 */ + [0x8002] = 0x0000, /* R32770 - DSP3 DM 2 */ + [0x8003] = 0x0000, /* R32771 - DSP3 DM 3 */ + [0x81FC] = 0x0000, /* R33276 - DSP3 DM 508 */ + [0x81FD] = 0x0000, /* R33277 - DSP3 DM 509 */ + [0x81FE] = 0x0000, /* R33278 - DSP3 DM 510 */ + [0x81FF] = 0x0000, /* R33279 - DSP3 DM 511 */ + [0x8800] = 0x0000, /* R34816 - DSP3 PM 0 */ + [0x8801] = 0x0000, /* R34817 - DSP3 PM 1 */ + [0x8802] = 0x0000, /* R34818 - DSP3 PM 2 */ + [0x8803] = 0x0000, /* R34819 - DSP3 PM 3 */ + [0x8804] = 0x0000, /* R34820 - DSP3 PM 4 */ + [0x8805] = 0x0000, /* R34821 - DSP3 PM 5 */ + [0x8DFA] = 0x0000, /* R36346 - DSP3 PM 1530 */ + [0x8DFB] = 0x0000, /* R36347 - DSP3 PM 1531 */ + [0x8DFC] = 0x0000, /* R36348 - DSP3 PM 1532 */ + [0x8DFD] = 0x0000, /* R36349 - DSP3 PM 1533 */ + [0x8DFE] = 0x0000, /* R36350 - DSP3 PM 1534 */ + [0x8DFF] = 0x0000, /* R36351 - DSP3 PM 1535 */ + [0x9000] = 0x0000, /* R36864 - DSP3 ZM 0 */ + [0x9001] = 0x0000, /* R36865 - DSP3 ZM 1 */ + [0x9002] = 0x0000, /* R36866 - DSP3 ZM 2 */ + [0x9003] = 0x0000, /* R36867 - DSP3 ZM 3 */ + [0x97FC] = 0x0000, /* R38908 - DSP3 ZM 2044 */ + [0x97FD] = 0x0000, /* R38909 - DSP3 ZM 2045 */ + [0x97FE] = 0x0000, /* R38910 - DSP3 ZM 2046 */ + [0x97FF] = 0x0000 /* R38911 - DSP3 ZM 2047 */ +}; diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c new file mode 100644 index 000000000000..5d88c99aaea6 --- /dev/null +++ b/sound/soc/codecs/wm5100.c @@ -0,0 +1,2809 @@ +/* + * wm5100.c -- WM5100 ALSA SoC Audio driver + * + * Copyright 2011 Wolfson Microelectronics plc + * + * Author: Mark Brown <broonie@opensource.wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/gcd.h> +#include <linux/gpio.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/regulator/consumer.h> +#include <linux/regulator/fixed.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/jack.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <sound/wm5100.h> + +#include "wm5100.h" + +#define WM5100_NUM_CORE_SUPPLIES 2 +static const char *wm5100_core_supply_names[WM5100_NUM_CORE_SUPPLIES] = { + "DBVDD1", + "LDOVDD", /* If DCVDD is supplied externally specify as LDOVDD */ +}; + +#define WM5100_AIFS 3 +#define WM5100_SYNC_SRS 3 + +struct wm5100_fll { + int fref; + int fout; + int src; + struct completion lock; +}; + +/* codec private data */ +struct wm5100_priv { + struct snd_soc_codec *codec; + + struct regulator_bulk_data core_supplies[WM5100_NUM_CORE_SUPPLIES]; + struct regulator *cpvdd; + struct regulator *dbvdd2; + struct regulator *dbvdd3; + + int rev; + + int sysclk; + int asyncclk; + + bool aif_async[WM5100_AIFS]; + bool aif_symmetric[WM5100_AIFS]; + int sr_ref[WM5100_SYNC_SRS]; + + bool out_ena[2]; + + struct snd_soc_jack *jack; + bool jack_detecting; + bool jack_mic; + int jack_mode; + + struct wm5100_fll fll[2]; + + struct wm5100_pdata pdata; + +#ifdef CONFIG_GPIOLIB + struct gpio_chip gpio_chip; +#endif +}; + +static int wm5100_sr_code[] = { + 0, + 12000, + 24000, + 48000, + 96000, + 192000, + 384000, + 768000, + 0, + 11025, + 22050, + 44100, + 88200, + 176400, + 352800, + 705600, + 4000, + 8000, + 16000, + 32000, + 64000, + 128000, + 256000, + 512000, +}; + +static int wm5100_sr_regs[WM5100_SYNC_SRS] = { + WM5100_CLOCKING_4, + WM5100_CLOCKING_5, + WM5100_CLOCKING_6, +}; + +static int wm5100_alloc_sr(struct snd_soc_codec *codec, int rate) +{ + struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec); + int sr_code, sr_free, i; + + for (i = 0; i < ARRAY_SIZE(wm5100_sr_code); i++) + if (wm5100_sr_code[i] == rate) + break; + if (i == ARRAY_SIZE(wm5100_sr_code)) { + dev_err(codec->dev, "Unsupported sample rate: %dHz\n", rate); + return -EINVAL; + } + sr_code = i; + + if ((wm5100->sysclk % rate) == 0) { + /* Is this rate already in use? */ + sr_free = -1; + for (i = 0; i < ARRAY_SIZE(wm5100_sr_regs); i++) { + if (!wm5100->sr_ref[i] && sr_free == -1) { + sr_free = i; + continue; + } + if ((snd_soc_read(codec, wm5100_sr_regs[i]) & + WM5100_SAMPLE_RATE_1_MASK) == sr_code) + break; + } + + if (i < ARRAY_SIZE(wm5100_sr_regs)) { + wm5100->sr_ref[i]++; + dev_dbg(codec->dev, "SR %dHz, slot %d, ref %d\n", + rate, i, wm5100->sr_ref[i]); + return i; + } + + if (sr_free == -1) { + dev_err(codec->dev, "All SR slots already in use\n"); + return -EBUSY; + } + + dev_dbg(codec->dev, "Allocating SR slot %d for %dHz\n", + sr_free, rate); + wm5100->sr_ref[sr_free]++; + snd_soc_update_bits(codec, wm5100_sr_regs[sr_free], + WM5100_SAMPLE_RATE_1_MASK, + sr_code); + + return sr_free; + + } else { + dev_err(codec->dev, + "SR %dHz incompatible with %dHz SYSCLK and %dHz ASYNCCLK\n", + rate, wm5100->sysclk, wm5100->asyncclk); + return -EINVAL; + } +} + +static void wm5100_free_sr(struct snd_soc_codec *codec, int rate) +{ + struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec); + int i, sr_code; + + for (i = 0; i < ARRAY_SIZE(wm5100_sr_code); i++) + if (wm5100_sr_code[i] == rate) + break; + if (i == ARRAY_SIZE(wm5100_sr_code)) { + dev_err(codec->dev, "Unsupported sample rate: %dHz\n", rate); + return; + } + sr_code = wm5100_sr_code[i]; + + for (i = 0; i < ARRAY_SIZE(wm5100_sr_regs); i++) { + if (!wm5100->sr_ref[i]) + continue; + + if ((snd_soc_read(codec, wm5100_sr_regs[i]) & + WM5100_SAMPLE_RATE_1_MASK) == sr_code) + break; + } + if (i < ARRAY_SIZE(wm5100_sr_regs)) { + wm5100->sr_ref[i]--; + dev_dbg(codec->dev, "Dereference SR %dHz, count now %d\n", + rate, wm5100->sr_ref[i]); + } else { + dev_warn(codec->dev, "Freeing unreferenced sample rate %dHz\n", + rate); + } +} + +static int wm5100_reset(struct snd_soc_codec *codec) +{ + struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec); + + if (wm5100->pdata.reset) { + gpio_set_value_cansleep(wm5100->pdata.reset, 0); + gpio_set_value_cansleep(wm5100->pdata.reset, 1); + + return 0; + } else { + return snd_soc_write(codec, WM5100_SOFTWARE_RESET, 0); + } +} + +static DECLARE_TLV_DB_SCALE(in_tlv, -6300, 100, 0); +static DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); +static DECLARE_TLV_DB_SCALE(mixer_tlv, -3200, 100, 0); +static DECLARE_TLV_DB_SCALE(out_tlv, -6400, 100, 0); +static DECLARE_TLV_DB_SCALE(digital_tlv, -6400, 50, 0); + +static const char *wm5100_mixer_texts[] = { + "None", + "Tone Generator 1", + "Tone Generator 2", + "AEC loopback", + "IN1L", + "IN1R", + "IN2L", + "IN2R", + "IN3L", + "IN3R", + "IN4L", + "IN4R", + "AIF1RX1", + "AIF1RX2", + "AIF1RX3", + "AIF1RX4", + "AIF1RX5", + "AIF1RX6", + "AIF1RX7", + "AIF1RX8", + "AIF2RX1", + "AIF2RX2", + "AIF3RX1", + "AIF3RX2", + "EQ1", + "EQ2", + "EQ3", + "EQ4", + "DRC1L", + "DRC1R", + "LHPF1", + "LHPF2", + "LHPF3", + "LHPF4", + "DSP1.1", + "DSP1.2", + "DSP1.3", + "DSP1.4", + "DSP1.5", + "DSP1.6", + "DSP2.1", + "DSP2.2", + "DSP2.3", + "DSP2.4", + "DSP2.5", + "DSP2.6", + "DSP3.1", + "DSP3.2", + "DSP3.3", + "DSP3.4", + "DSP3.5", + "DSP3.6", + "ASRC1L", + "ASRC1R", + "ASRC2L", + "ASRC2R", + "ISRC1INT1", + "ISRC1INT2", + "ISRC1INT3", + "ISRC1INT4", + "ISRC2INT1", + "ISRC2INT2", + "ISRC2INT3", + "ISRC2INT4", + "ISRC1DEC1", + "ISRC1DEC2", + "ISRC1DEC3", + "ISRC1DEC4", + "ISRC2DEC1", + "ISRC2DEC2", + "ISRC2DEC3", + "ISRC2DEC4", +}; + +static int wm5100_mixer_values[] = { + 0x00, + 0x04, /* Tone */ + 0x05, + 0x08, /* AEC */ + 0x10, /* Input */ + 0x11, + 0x12, + 0x13, + 0x14, + 0x15, + 0x16, + 0x17, + 0x20, /* AIF */ + 0x21, + 0x22, + 0x23, + 0x24, + 0x25, + 0x26, + 0x27, + 0x28, + 0x29, + 0x30, /* AIF3 - check */ + 0x31, + 0x50, /* EQ */ + 0x51, + 0x52, + 0x53, + 0x54, + 0x58, /* DRC */ + 0x59, + 0x60, /* LHPF1 */ + 0x61, /* LHPF2 */ + 0x62, /* LHPF3 */ + 0x63, /* LHPF4 */ + 0x68, /* DSP1 */ + 0x69, + 0x6a, + 0x6b, + 0x6c, + 0x6d, + 0x70, /* DSP2 */ + 0x71, + 0x72, + 0x73, + 0x74, + 0x75, + 0x78, /* DSP3 */ + 0x79, + 0x7a, + 0x7b, + 0x7c, + 0x7d, + 0x90, /* ASRC1 */ + 0x91, + 0x92, /* ASRC2 */ + 0x93, + 0xa0, /* ISRC1DEC1 */ + 0xa1, + 0xa2, + 0xa3, + 0xa4, /* ISRC1INT1 */ + 0xa5, + 0xa6, + 0xa7, + 0xa8, /* ISRC2DEC1 */ + 0xa9, + 0xaa, + 0xab, + 0xac, /* ISRC2INT1 */ + 0xad, + 0xae, + 0xaf, +}; + +#define WM5100_MIXER_CONTROLS(name, base) \ + SOC_SINGLE_TLV(name " Input 1 Volume", base + 1 , \ + WM5100_MIXER_VOL_SHIFT, 80, 0, mixer_tlv), \ + SOC_SINGLE_TLV(name " Input 2 Volume", base + 3 , \ + WM5100_MIXER_VOL_SHIFT, 80, 0, mixer_tlv), \ + SOC_SINGLE_TLV(name " Input 3 Volume", base + 5 , \ + WM5100_MIXER_VOL_SHIFT, 80, 0, mixer_tlv), \ + SOC_SINGLE_TLV(name " Input 4 Volume", base + 7 , \ + WM5100_MIXER_VOL_SHIFT, 80, 0, mixer_tlv) + +#define WM5100_MUX_ENUM_DECL(name, reg) \ + SOC_VALUE_ENUM_SINGLE_DECL(name, reg, 0, 0xff, \ + wm5100_mixer_texts, wm5100_mixer_values) + +#define WM5100_MUX_CTL_DECL(name) \ + const struct snd_kcontrol_new name##_mux = \ + SOC_DAPM_VALUE_ENUM("Route", name##_enum) + +#define WM5100_MIXER_ENUMS(name, base_reg) \ + static WM5100_MUX_ENUM_DECL(name##_in1_enum, base_reg); \ + static WM5100_MUX_ENUM_DECL(name##_in2_enum, base_reg + 2); \ + static WM5100_MUX_ENUM_DECL(name##_in3_enum, base_reg + 4); \ + static WM5100_MUX_ENUM_DECL(name##_in4_enum, base_reg + 6); \ + static WM5100_MUX_CTL_DECL(name##_in1); \ + static WM5100_MUX_CTL_DECL(name##_in2); \ + static WM5100_MUX_CTL_DECL(name##_in3); \ + static WM5100_MUX_CTL_DECL(name##_in4) + +WM5100_MIXER_ENUMS(HPOUT1L, WM5100_OUT1LMIX_INPUT_1_SOURCE); +WM5100_MIXER_ENUMS(HPOUT1R, WM5100_OUT1RMIX_INPUT_1_SOURCE); +WM5100_MIXER_ENUMS(HPOUT2L, WM5100_OUT2LMIX_INPUT_1_SOURCE); +WM5100_MIXER_ENUMS(HPOUT2R, WM5100_OUT2RMIX_INPUT_1_SOURCE); +WM5100_MIXER_ENUMS(HPOUT3L, WM5100_OUT3LMIX_INPUT_1_SOURCE); +WM5100_MIXER_ENUMS(HPOUT3R, WM5100_OUT3RMIX_INPUT_1_SOURCE); + +WM5100_MIXER_ENUMS(SPKOUTL, WM5100_OUT4LMIX_INPUT_1_SOURCE); +WM5100_MIXER_ENUMS(SPKOUTR, WM5100_OUT4RMIX_INPUT_1_SOURCE); +WM5100_MIXER_ENUMS(SPKDAT1L, WM5100_OUT5LMIX_INPUT_1_SOURCE); +WM5100_MIXER_ENUMS(SPKDAT1R, WM5100_OUT5RMIX_INPUT_1_SOURCE); +WM5100_MIXER_ENUMS(SPKDAT2L, WM5100_OUT6LMIX_INPUT_1_SOURCE); +WM5100_MIXER_ENUMS(SPKDAT2R, WM5100_OUT6RMIX_INPUT_1_SOURCE); + +WM5100_MIXER_ENUMS(PWM1, WM5100_PWM1MIX_INPUT_1_SOURCE); +WM5100_MIXER_ENUMS(PWM2, WM5100_PWM1MIX_INPUT_1_SOURCE); + +WM5100_MIXER_ENUMS(AIF1TX1, WM5100_AIF1TX1MIX_INPUT_1_SOURCE); +WM5100_MIXER_ENUMS(AIF1TX2, WM5100_AIF1TX2MIX_INPUT_1_SOURCE); +WM5100_MIXER_ENUMS(AIF1TX3, WM5100_AIF1TX3MIX_INPUT_1_SOURCE); +WM5100_MIXER_ENUMS(AIF1TX4, WM5100_AIF1TX4MIX_INPUT_1_SOURCE); +WM5100_MIXER_ENUMS(AIF1TX5, WM5100_AIF1TX5MIX_INPUT_1_SOURCE); +WM5100_MIXER_ENUMS(AIF1TX6, WM5100_AIF1TX6MIX_INPUT_1_SOURCE); +WM5100_MIXER_ENUMS(AIF1TX7, WM5100_AIF1TX7MIX_INPUT_1_SOURCE); +WM5100_MIXER_ENUMS(AIF1TX8, WM5100_AIF1TX8MIX_INPUT_1_SOURCE); + +WM5100_MIXER_ENUMS(AIF2TX1, WM5100_AIF2TX1MIX_INPUT_1_SOURCE); +WM5100_MIXER_ENUMS(AIF2TX2, WM5100_AIF2TX2MIX_INPUT_1_SOURCE); + +WM5100_MIXER_ENUMS(AIF3TX1, WM5100_AIF1TX1MIX_INPUT_1_SOURCE); +WM5100_MIXER_ENUMS(AIF3TX2, WM5100_AIF1TX2MIX_INPUT_1_SOURCE); + +WM5100_MIXER_ENUMS(EQ1, WM5100_EQ1MIX_INPUT_1_SOURCE); +WM5100_MIXER_ENUMS(EQ2, WM5100_EQ2MIX_INPUT_1_SOURCE); +WM5100_MIXER_ENUMS(EQ3, WM5100_EQ3MIX_INPUT_1_SOURCE); +WM5100_MIXER_ENUMS(EQ4, WM5100_EQ4MIX_INPUT_1_SOURCE); + +WM5100_MIXER_ENUMS(DRC1L, WM5100_DRC1LMIX_INPUT_1_SOURCE); +WM5100_MIXER_ENUMS(DRC1R, WM5100_DRC1RMIX_INPUT_1_SOURCE); + +WM5100_MIXER_ENUMS(LHPF1, WM5100_HPLP1MIX_INPUT_1_SOURCE); +WM5100_MIXER_ENUMS(LHPF2, WM5100_HPLP2MIX_INPUT_1_SOURCE); +WM5100_MIXER_ENUMS(LHPF3, WM5100_HPLP3MIX_INPUT_1_SOURCE); +WM5100_MIXER_ENUMS(LHPF4, WM5100_HPLP4MIX_INPUT_1_SOURCE); + +#define WM5100_MUX(name, ctrl) \ + SND_SOC_DAPM_VALUE_MUX(name, SND_SOC_NOPM, 0, 0, ctrl) + +#define WM5100_MIXER_WIDGETS(name, name_str) \ + WM5100_MUX(name_str " Input 1", &name##_in1_mux), \ + WM5100_MUX(name_str " Input 2", &name##_in2_mux), \ + WM5100_MUX(name_str " Input 3", &name##_in3_mux), \ + WM5100_MUX(name_str " Input 4", &name##_in4_mux), \ + SND_SOC_DAPM_MIXER(name_str " Mixer", SND_SOC_NOPM, 0, 0, NULL, 0) + +#define WM5100_MIXER_INPUT_ROUTES(name) \ + { name, "Tone Generator 1", "Tone Generator 1" }, \ + { name, "Tone Generator 2", "Tone Generator 2" }, \ + { name, "IN1L", "IN1L PGA" }, \ + { name, "IN1R", "IN1R PGA" }, \ + { name, "IN2L", "IN2L PGA" }, \ + { name, "IN2R", "IN2R PGA" }, \ + { name, "IN3L", "IN3L PGA" }, \ + { name, "IN3R", "IN3R PGA" }, \ + { name, "IN4L", "IN4L PGA" }, \ + { name, "IN4R", "IN4R PGA" }, \ + { name, "AIF1RX1", "AIF1RX1" }, \ + { name, "AIF1RX2", "AIF1RX2" }, \ + { name, "AIF1RX3", "AIF1RX3" }, \ + { name, "AIF1RX4", "AIF1RX4" }, \ + { name, "AIF1RX5", "AIF1RX5" }, \ + { name, "AIF1RX6", "AIF1RX6" }, \ + { name, "AIF1RX7", "AIF1RX7" }, \ + { name, "AIF1RX8", "AIF1RX8" }, \ + { name, "AIF2RX1", "AIF2RX1" }, \ + { name, "AIF2RX2", "AIF2RX2" }, \ + { name, "AIF3RX1", "AIF3RX1" }, \ + { name, "AIF3RX2", "AIF3RX2" }, \ + { name, "EQ1", "EQ1" }, \ + { name, "EQ2", "EQ2" }, \ + { name, "EQ3", "EQ3" }, \ + { name, "EQ4", "EQ4" }, \ + { name, "DRC1L", "DRC1L" }, \ + { name, "DRC1R", "DRC1R" }, \ + { name, "LHPF1", "LHPF1" }, \ + { name, "LHPF2", "LHPF2" }, \ + { name, "LHPF3", "LHPF3" }, \ + { name, "LHPF4", "LHPF4" } + +#define WM5100_MIXER_ROUTES(widget, name) \ + { widget, NULL, name " Mixer" }, \ + { name " Mixer", NULL, name " Input 1" }, \ + { name " Mixer", NULL, name " Input 2" }, \ + { name " Mixer", NULL, name " Input 3" }, \ + { name " Mixer", NULL, name " Input 4" }, \ + WM5100_MIXER_INPUT_ROUTES(name " Input 1"), \ + WM5100_MIXER_INPUT_ROUTES(name " Input 2"), \ + WM5100_MIXER_INPUT_ROUTES(name " Input 3"), \ + WM5100_MIXER_INPUT_ROUTES(name " Input 4") + +static const char *wm5100_lhpf_mode_text[] = { + "Low-pass", "High-pass" +}; + +static const struct soc_enum wm5100_lhpf1_mode = + SOC_ENUM_SINGLE(WM5100_HPLPF1_1, WM5100_LHPF1_MODE_SHIFT, 2, + wm5100_lhpf_mode_text); + +static const struct soc_enum wm5100_lhpf2_mode = + SOC_ENUM_SINGLE(WM5100_HPLPF2_1, WM5100_LHPF2_MODE_SHIFT, 2, + wm5100_lhpf_mode_text); + +static const struct soc_enum wm5100_lhpf3_mode = + SOC_ENUM_SINGLE(WM5100_HPLPF3_1, WM5100_LHPF3_MODE_SHIFT, 2, + wm5100_lhpf_mode_text); + +static const struct soc_enum wm5100_lhpf4_mode = + SOC_ENUM_SINGLE(WM5100_HPLPF4_1, WM5100_LHPF4_MODE_SHIFT, 2, + wm5100_lhpf_mode_text); + +static const struct snd_kcontrol_new wm5100_snd_controls[] = { +SOC_SINGLE("IN1 High Performance Switch", WM5100_IN1L_CONTROL, + WM5100_IN1_OSR_SHIFT, 1, 0), +SOC_SINGLE("IN2 High Performance Switch", WM5100_IN2L_CONTROL, + WM5100_IN2_OSR_SHIFT, 1, 0), +SOC_SINGLE("IN3 High Performance Switch", WM5100_IN3L_CONTROL, + WM5100_IN3_OSR_SHIFT, 1, 0), +SOC_SINGLE("IN4 High Performance Switch", WM5100_IN4L_CONTROL, + WM5100_IN4_OSR_SHIFT, 1, 0), + +/* Only applicable for analogue inputs */ +SOC_DOUBLE_R_TLV("IN1 Volume", WM5100_IN1L_CONTROL, WM5100_IN1R_CONTROL, + WM5100_IN1L_PGA_VOL_SHIFT, 94, 0, in_tlv), +SOC_DOUBLE_R_TLV("IN2 Volume", WM5100_IN2L_CONTROL, WM5100_IN2R_CONTROL, + WM5100_IN2L_PGA_VOL_SHIFT, 94, 0, in_tlv), +SOC_DOUBLE_R_TLV("IN3 Volume", WM5100_IN3L_CONTROL, WM5100_IN3R_CONTROL, + WM5100_IN3L_PGA_VOL_SHIFT, 94, 0, in_tlv), +SOC_DOUBLE_R_TLV("IN4 Volume", WM5100_IN4L_CONTROL, WM5100_IN4R_CONTROL, + WM5100_IN4L_PGA_VOL_SHIFT, 94, 0, in_tlv), + +SOC_DOUBLE_R_TLV("IN1 Digital Volume", WM5100_ADC_DIGITAL_VOLUME_1L, + WM5100_ADC_DIGITAL_VOLUME_1R, WM5100_IN1L_VOL_SHIFT, 191, + 0, digital_tlv), +SOC_DOUBLE_R_TLV("IN2 Digital Volume", WM5100_ADC_DIGITAL_VOLUME_2L, + WM5100_ADC_DIGITAL_VOLUME_2R, WM5100_IN2L_VOL_SHIFT, 191, + 0, digital_tlv), +SOC_DOUBLE_R_TLV("IN3 Digital Volume", WM5100_ADC_DIGITAL_VOLUME_3L, + WM5100_ADC_DIGITAL_VOLUME_3R, WM5100_IN3L_VOL_SHIFT, 191, + 0, digital_tlv), +SOC_DOUBLE_R_TLV("IN4 Digital Volume", WM5100_ADC_DIGITAL_VOLUME_4L, + WM5100_ADC_DIGITAL_VOLUME_4R, WM5100_IN4L_VOL_SHIFT, 191, + 0, digital_tlv), + +SOC_DOUBLE_R("IN1 Switch", WM5100_ADC_DIGITAL_VOLUME_1L, + WM5100_ADC_DIGITAL_VOLUME_1R, WM5100_IN1L_MUTE_SHIFT, 1, 1), +SOC_DOUBLE_R("IN2 Switch", WM5100_ADC_DIGITAL_VOLUME_2L, + WM5100_ADC_DIGITAL_VOLUME_2R, WM5100_IN2L_MUTE_SHIFT, 1, 1), +SOC_DOUBLE_R("IN3 Switch", WM5100_ADC_DIGITAL_VOLUME_3L, + WM5100_ADC_DIGITAL_VOLUME_3R, WM5100_IN3L_MUTE_SHIFT, 1, 1), +SOC_DOUBLE_R("IN4 Switch", WM5100_ADC_DIGITAL_VOLUME_4L, + WM5100_ADC_DIGITAL_VOLUME_4R, WM5100_IN4L_MUTE_SHIFT, 1, 1), + +SOC_SINGLE("HPOUT1 High Performance Switch", WM5100_OUT_VOLUME_1L, + WM5100_OUT1_OSR_SHIFT, 1, 0), +SOC_SINGLE("HPOUT2 High Performance Switch", WM5100_OUT_VOLUME_2L, + WM5100_OUT2_OSR_SHIFT, 1, 0), +SOC_SINGLE("HPOUT3 High Performance Switch", WM5100_OUT_VOLUME_3L, + WM5100_OUT3_OSR_SHIFT, 1, 0), +SOC_SINGLE("SPKOUT High Performance Switch", WM5100_OUT_VOLUME_4L, + WM5100_OUT4_OSR_SHIFT, 1, 0), +SOC_SINGLE("SPKDAT1 High Performance Switch", WM5100_DAC_VOLUME_LIMIT_5L, + WM5100_OUT5_OSR_SHIFT, 1, 0), +SOC_SINGLE("SPKDAT2 High Performance Switch", WM5100_DAC_VOLUME_LIMIT_6L, + WM5100_OUT6_OSR_SHIFT, 1, 0), + +SOC_DOUBLE_R_TLV("HPOUT1 Digital Volume", WM5100_DAC_DIGITAL_VOLUME_1L, + WM5100_DAC_DIGITAL_VOLUME_1R, WM5100_OUT1L_VOL_SHIFT, 159, 0, + digital_tlv), +SOC_DOUBLE_R_TLV("HPOUT2 Digital Volume", WM5100_DAC_DIGITAL_VOLUME_2L, + WM5100_DAC_DIGITAL_VOLUME_2R, WM5100_OUT2L_VOL_SHIFT, 159, 0, + digital_tlv), +SOC_DOUBLE_R_TLV("HPOUT3 Digital Volume", WM5100_DAC_DIGITAL_VOLUME_3L, + WM5100_DAC_DIGITAL_VOLUME_3R, WM5100_OUT3L_VOL_SHIFT, 159, 0, + digital_tlv), +SOC_DOUBLE_R_TLV("SPKOUT Digital Volume", WM5100_DAC_DIGITAL_VOLUME_4L, + WM5100_DAC_DIGITAL_VOLUME_4R, WM5100_OUT4L_VOL_SHIFT, 159, 0, + digital_tlv), +SOC_DOUBLE_R_TLV("SPKDAT1 Digital Volume", WM5100_DAC_DIGITAL_VOLUME_5L, + WM5100_DAC_DIGITAL_VOLUME_5R, WM5100_OUT5L_VOL_SHIFT, 159, 0, + digital_tlv), +SOC_DOUBLE_R_TLV("SPKDAT2 Digital Volume", WM5100_DAC_DIGITAL_VOLUME_6L, + WM5100_DAC_DIGITAL_VOLUME_6R, WM5100_OUT6L_VOL_SHIFT, 159, 0, + digital_tlv), + +SOC_DOUBLE_R("HPOUT1 Digital Switch", WM5100_DAC_DIGITAL_VOLUME_1L, + WM5100_DAC_DIGITAL_VOLUME_1R, WM5100_OUT1L_MUTE_SHIFT, 1, 1), +SOC_DOUBLE_R("HPOUT2 Digital Switch", WM5100_DAC_DIGITAL_VOLUME_2L, + WM5100_DAC_DIGITAL_VOLUME_2R, WM5100_OUT2L_MUTE_SHIFT, 1, 1), +SOC_DOUBLE_R("HPOUT3 Digital Switch", WM5100_DAC_DIGITAL_VOLUME_3L, + WM5100_DAC_DIGITAL_VOLUME_3R, WM5100_OUT3L_MUTE_SHIFT, 1, 1), +SOC_DOUBLE_R("SPKOUT Digital Switch", WM5100_DAC_DIGITAL_VOLUME_4L, + WM5100_DAC_DIGITAL_VOLUME_4R, WM5100_OUT4L_MUTE_SHIFT, 1, 1), +SOC_DOUBLE_R("SPKDAT1 Digital Switch", WM5100_DAC_DIGITAL_VOLUME_5L, + WM5100_DAC_DIGITAL_VOLUME_5R, WM5100_OUT5L_MUTE_SHIFT, 1, 1), +SOC_DOUBLE_R("SPKDAT2 Digital Switch", WM5100_DAC_DIGITAL_VOLUME_6L, + WM5100_DAC_DIGITAL_VOLUME_6R, WM5100_OUT6L_MUTE_SHIFT, 1, 1), + +/* FIXME: Only valid from -12dB to 0dB (52-64) */ +SOC_DOUBLE_R_TLV("HPOUT1 Volume", WM5100_OUT_VOLUME_1L, WM5100_OUT_VOLUME_1R, + WM5100_OUT1L_PGA_VOL_SHIFT, 64, 0, out_tlv), +SOC_DOUBLE_R_TLV("HPOUT2 Volume", WM5100_OUT_VOLUME_2L, WM5100_OUT_VOLUME_2R, + WM5100_OUT2L_PGA_VOL_SHIFT, 64, 0, out_tlv), +SOC_DOUBLE_R_TLV("HPOUT3 Volume", WM5100_OUT_VOLUME_3L, WM5100_OUT_VOLUME_3R, + WM5100_OUT2L_PGA_VOL_SHIFT, 64, 0, out_tlv), + +SOC_DOUBLE("SPKDAT1 Switch", WM5100_PDM_SPK1_CTRL_1, WM5100_SPK1L_MUTE_SHIFT, + WM5100_SPK1R_MUTE_SHIFT, 1, 1), +SOC_DOUBLE("SPKDAT2 Switch", WM5100_PDM_SPK2_CTRL_1, WM5100_SPK2L_MUTE_SHIFT, + WM5100_SPK2R_MUTE_SHIFT, 1, 1), + +SOC_SINGLE_TLV("EQ1 Band 1 Volume", WM5100_EQ1_1, WM5100_EQ1_B1_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ1 Band 2 Volume", WM5100_EQ1_1, WM5100_EQ1_B2_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ1 Band 3 Volume", WM5100_EQ1_1, WM5100_EQ1_B3_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ1 Band 4 Volume", WM5100_EQ1_2, WM5100_EQ1_B4_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ1 Band 5 Volume", WM5100_EQ1_2, WM5100_EQ1_B5_GAIN_SHIFT, + 24, 0, eq_tlv), + +SOC_SINGLE_TLV("EQ2 Band 1 Volume", WM5100_EQ2_1, WM5100_EQ2_B1_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ2 Band 2 Volume", WM5100_EQ2_1, WM5100_EQ2_B2_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ2 Band 3 Volume", WM5100_EQ2_1, WM5100_EQ2_B3_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ2 Band 4 Volume", WM5100_EQ2_2, WM5100_EQ2_B4_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ2 Band 5 Volume", WM5100_EQ2_2, WM5100_EQ2_B5_GAIN_SHIFT, + 24, 0, eq_tlv), + +SOC_SINGLE_TLV("EQ3 Band 1 Volume", WM5100_EQ1_1, WM5100_EQ3_B1_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ3 Band 2 Volume", WM5100_EQ3_1, WM5100_EQ3_B2_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ3 Band 3 Volume", WM5100_EQ3_1, WM5100_EQ3_B3_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ3 Band 4 Volume", WM5100_EQ3_2, WM5100_EQ3_B4_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ3 Band 5 Volume", WM5100_EQ3_2, WM5100_EQ3_B5_GAIN_SHIFT, + 24, 0, eq_tlv), + +SOC_SINGLE_TLV("EQ4 Band 1 Volume", WM5100_EQ4_1, WM5100_EQ4_B1_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ4 Band 2 Volume", WM5100_EQ4_1, WM5100_EQ4_B2_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ4 Band 3 Volume", WM5100_EQ4_1, WM5100_EQ4_B3_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ4 Band 4 Volume", WM5100_EQ4_2, WM5100_EQ4_B4_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ4 Band 5 Volume", WM5100_EQ4_2, WM5100_EQ4_B5_GAIN_SHIFT, + 24, 0, eq_tlv), + +SOC_ENUM("LHPF1 Mode", wm5100_lhpf1_mode), +SOC_ENUM("LHPF2 Mode", wm5100_lhpf2_mode), +SOC_ENUM("LHPF3 Mode", wm5100_lhpf3_mode), +SOC_ENUM("LHPF4 Mode", wm5100_lhpf4_mode), + +WM5100_MIXER_CONTROLS("HPOUT1L", WM5100_OUT1LMIX_INPUT_1_SOURCE), +WM5100_MIXER_CONTROLS("HPOUT1R", WM5100_OUT1RMIX_INPUT_1_SOURCE), +WM5100_MIXER_CONTROLS("HPOUT2L", WM5100_OUT2LMIX_INPUT_1_SOURCE), +WM5100_MIXER_CONTROLS("HPOUT2R", WM5100_OUT2RMIX_INPUT_1_SOURCE), +WM5100_MIXER_CONTROLS("HPOUT3L", WM5100_OUT3LMIX_INPUT_1_SOURCE), +WM5100_MIXER_CONTROLS("HPOUT3R", WM5100_OUT3RMIX_INPUT_1_SOURCE), + +WM5100_MIXER_CONTROLS("SPKOUTL", WM5100_OUT4LMIX_INPUT_1_SOURCE), +WM5100_MIXER_CONTROLS("SPKOUTR", WM5100_OUT4RMIX_INPUT_1_SOURCE), +WM5100_MIXER_CONTROLS("SPKDAT1L", WM5100_OUT5LMIX_INPUT_1_SOURCE), +WM5100_MIXER_CONTROLS("SPKDAT1R", WM5100_OUT5RMIX_INPUT_1_SOURCE), +WM5100_MIXER_CONTROLS("SPKDAT2L", WM5100_OUT6LMIX_INPUT_1_SOURCE), +WM5100_MIXER_CONTROLS("SPKDAT2R", WM5100_OUT6RMIX_INPUT_1_SOURCE), + +WM5100_MIXER_CONTROLS("PWM1", WM5100_PWM1MIX_INPUT_1_SOURCE), +WM5100_MIXER_CONTROLS("PWM2", WM5100_PWM2MIX_INPUT_1_SOURCE), + +WM5100_MIXER_CONTROLS("AIF1TX1", WM5100_AIF1TX1MIX_INPUT_1_SOURCE), +WM5100_MIXER_CONTROLS("AIF1TX2", WM5100_AIF1TX2MIX_INPUT_1_SOURCE), +WM5100_MIXER_CONTROLS("AIF1TX3", WM5100_AIF1TX3MIX_INPUT_1_SOURCE), +WM5100_MIXER_CONTROLS("AIF1TX4", WM5100_AIF1TX4MIX_INPUT_1_SOURCE), +WM5100_MIXER_CONTROLS("AIF1TX5", WM5100_AIF1TX5MIX_INPUT_1_SOURCE), +WM5100_MIXER_CONTROLS("AIF1TX6", WM5100_AIF1TX6MIX_INPUT_1_SOURCE), +WM5100_MIXER_CONTROLS("AIF1TX7", WM5100_AIF1TX7MIX_INPUT_1_SOURCE), +WM5100_MIXER_CONTROLS("AIF1TX8", WM5100_AIF1TX8MIX_INPUT_1_SOURCE), + +WM5100_MIXER_CONTROLS("AIF2TX1", WM5100_AIF2TX1MIX_INPUT_1_SOURCE), +WM5100_MIXER_CONTROLS("AIF2TX2", WM5100_AIF2TX2MIX_INPUT_1_SOURCE), + +WM5100_MIXER_CONTROLS("AIF3TX1", WM5100_AIF3TX1MIX_INPUT_1_SOURCE), +WM5100_MIXER_CONTROLS("AIF3TX2", WM5100_AIF3TX2MIX_INPUT_1_SOURCE), + +WM5100_MIXER_CONTROLS("EQ1", WM5100_EQ1MIX_INPUT_1_SOURCE), +WM5100_MIXER_CONTROLS("EQ2", WM5100_EQ2MIX_INPUT_1_SOURCE), +WM5100_MIXER_CONTROLS("EQ3", WM5100_EQ3MIX_INPUT_1_SOURCE), +WM5100_MIXER_CONTROLS("EQ4", WM5100_EQ4MIX_INPUT_1_SOURCE), + +WM5100_MIXER_CONTROLS("DRC1L", WM5100_DRC1LMIX_INPUT_1_SOURCE), +WM5100_MIXER_CONTROLS("DRC1R", WM5100_DRC1RMIX_INPUT_1_SOURCE), + +WM5100_MIXER_CONTROLS("LHPF1", WM5100_HPLP1MIX_INPUT_1_SOURCE), +WM5100_MIXER_CONTROLS("LHPF2", WM5100_HPLP2MIX_INPUT_1_SOURCE), +WM5100_MIXER_CONTROLS("LHPF3", WM5100_HPLP3MIX_INPUT_1_SOURCE), +WM5100_MIXER_CONTROLS("LHPF4", WM5100_HPLP4MIX_INPUT_1_SOURCE), +}; + +static void wm5100_seq_notifier(struct snd_soc_dapm_context *dapm, + enum snd_soc_dapm_type event, int subseq) +{ + struct snd_soc_codec *codec = container_of(dapm, + struct snd_soc_codec, dapm); + struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec); + u16 val, expect, i; + + /* Wait for the outputs to flag themselves as enabled */ + if (wm5100->out_ena[0]) { + expect = snd_soc_read(codec, WM5100_CHANNEL_ENABLES_1); + for (i = 0; i < 200; i++) { + val = snd_soc_read(codec, WM5100_OUTPUT_STATUS_1); + if (val == expect) { + wm5100->out_ena[0] = false; + break; + } + } + if (i == 200) { + dev_err(codec->dev, "Timeout waiting for OUTPUT1 %x\n", + expect); + } + } + + if (wm5100->out_ena[1]) { + expect = snd_soc_read(codec, WM5100_OUTPUT_ENABLES_2); + for (i = 0; i < 200; i++) { + val = snd_soc_read(codec, WM5100_OUTPUT_STATUS_2); + if (val == expect) { + wm5100->out_ena[1] = false; + break; + } + } + if (i == 200) { + dev_err(codec->dev, "Timeout waiting for OUTPUT2 %x\n", + expect); + } + } +} + +static int wm5100_out_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event) +{ + struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(w->codec); + + switch (w->reg) { + case WM5100_CHANNEL_ENABLES_1: + wm5100->out_ena[0] = true; + break; + case WM5100_OUTPUT_ENABLES_2: + wm5100->out_ena[0] = true; + break; + default: + break; + } + + return 0; +} + +static int wm5100_cp_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event) +{ + struct snd_soc_codec *codec = w->codec; + struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec); + int ret; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + ret = regulator_enable(wm5100->cpvdd); + if (ret != 0) { + dev_err(codec->dev, "Failed to enable CPVDD: %d\n", + ret); + return ret; + } + return ret; + + case SND_SOC_DAPM_POST_PMD: + ret = regulator_disable_deferred(wm5100->cpvdd, 20); + if (ret != 0) { + dev_err(codec->dev, "Failed to disable CPVDD: %d\n", + ret); + return ret; + } + return ret; + + default: + BUG(); + return 0; + } +} + +static int wm5100_dbvdd_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event) +{ + struct snd_soc_codec *codec = w->codec; + struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec); + struct regulator *regulator; + int ret; + + switch (w->shift) { + case 2: + regulator = wm5100->dbvdd2; + break; + case 3: + regulator = wm5100->dbvdd3; + break; + default: + BUG(); + return 0; + } + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + ret = regulator_enable(regulator); + if (ret != 0) { + dev_err(codec->dev, "Failed to enable DBVDD%d: %d\n", + w->shift, ret); + return ret; + } + return ret; + + case SND_SOC_DAPM_POST_PMD: + ret = regulator_disable(regulator); + if (ret != 0) { + dev_err(codec->dev, "Failed to enable DBVDD%d: %d\n", + w->shift, ret); + return ret; + } + return ret; + + default: + BUG(); + return 0; + } +} + +static void wm5100_log_status3(struct snd_soc_codec *codec, int val) +{ + if (val & WM5100_SPK_SHUTDOWN_WARN_EINT) + dev_crit(codec->dev, "Speaker shutdown warning\n"); + if (val & WM5100_SPK_SHUTDOWN_EINT) + dev_crit(codec->dev, "Speaker shutdown\n"); + if (val & WM5100_CLKGEN_ERR_EINT) + dev_crit(codec->dev, "SYSCLK underclocked\n"); + if (val & WM5100_CLKGEN_ERR_ASYNC_EINT) + dev_crit(codec->dev, "ASYNCCLK underclocked\n"); +} + +static void wm5100_log_status4(struct snd_soc_codec *codec, int val) +{ + if (val & WM5100_AIF3_ERR_EINT) + dev_err(codec->dev, "AIF3 configuration error\n"); + if (val & WM5100_AIF2_ERR_EINT) + dev_err(codec->dev, "AIF2 configuration error\n"); + if (val & WM5100_AIF1_ERR_EINT) + dev_err(codec->dev, "AIF1 configuration error\n"); + if (val & WM5100_CTRLIF_ERR_EINT) + dev_err(codec->dev, "Control interface error\n"); + if (val & WM5100_ISRC2_UNDERCLOCKED_EINT) + dev_err(codec->dev, "ISRC2 underclocked\n"); + if (val & WM5100_ISRC1_UNDERCLOCKED_EINT) + dev_err(codec->dev, "ISRC1 underclocked\n"); + if (val & WM5100_FX_UNDERCLOCKED_EINT) + dev_err(codec->dev, "FX underclocked\n"); + if (val & WM5100_AIF3_UNDERCLOCKED_EINT) + dev_err(codec->dev, "AIF3 underclocked\n"); + if (val & WM5100_AIF2_UNDERCLOCKED_EINT) + dev_err(codec->dev, "AIF2 underclocked\n"); + if (val & WM5100_AIF1_UNDERCLOCKED_EINT) + dev_err(codec->dev, "AIF1 underclocked\n"); + if (val & WM5100_ASRC_UNDERCLOCKED_EINT) + dev_err(codec->dev, "ASRC underclocked\n"); + if (val & WM5100_DAC_UNDERCLOCKED_EINT) + dev_err(codec->dev, "DAC underclocked\n"); + if (val & WM5100_ADC_UNDERCLOCKED_EINT) + dev_err(codec->dev, "ADC underclocked\n"); + if (val & WM5100_MIXER_UNDERCLOCKED_EINT) + dev_err(codec->dev, "Mixer underclocked\n"); +} + +static int wm5100_post_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event) +{ + struct snd_soc_codec *codec = w->codec; + int ret; + + ret = snd_soc_read(codec, WM5100_INTERRUPT_RAW_STATUS_3); + ret &= WM5100_SPK_SHUTDOWN_WARN_STS | + WM5100_SPK_SHUTDOWN_STS | WM5100_CLKGEN_ERR_STS | + WM5100_CLKGEN_ERR_ASYNC_STS; + wm5100_log_status3(codec, ret); + + ret = snd_soc_read(codec, WM5100_INTERRUPT_RAW_STATUS_4); + wm5100_log_status4(codec, ret); + + return 0; +} + +static const struct snd_soc_dapm_widget wm5100_dapm_widgets[] = { +SND_SOC_DAPM_SUPPLY("SYSCLK", WM5100_CLOCKING_3, WM5100_SYSCLK_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_SUPPLY("ASYNCCLK", WM5100_CLOCKING_6, WM5100_ASYNC_CLK_ENA_SHIFT, + 0, NULL, 0), + +SND_SOC_DAPM_SUPPLY("CP1", WM5100_HP_CHARGE_PUMP_1, WM5100_CP1_ENA_SHIFT, 0, + wm5100_cp_ev, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +SND_SOC_DAPM_SUPPLY("CP2", WM5100_MIC_CHARGE_PUMP_1, WM5100_CP2_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_SUPPLY("CP2 Active", WM5100_MIC_CHARGE_PUMP_1, + WM5100_CP2_BYPASS_SHIFT, 1, wm5100_cp_ev, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +SND_SOC_DAPM_SUPPLY("DBVDD2", SND_SOC_NOPM, 2, 0, wm5100_dbvdd_ev, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +SND_SOC_DAPM_SUPPLY("DBVDD3", SND_SOC_NOPM, 3, 0, wm5100_dbvdd_ev, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + +SND_SOC_DAPM_SUPPLY("MICBIAS1", WM5100_MIC_BIAS_CTRL_1, WM5100_MICB1_ENA_SHIFT, + 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("MICBIAS2", WM5100_MIC_BIAS_CTRL_2, WM5100_MICB2_ENA_SHIFT, + 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("MICBIAS3", WM5100_MIC_BIAS_CTRL_3, WM5100_MICB3_ENA_SHIFT, + 0, NULL, 0), + +SND_SOC_DAPM_INPUT("IN1L"), +SND_SOC_DAPM_INPUT("IN1R"), +SND_SOC_DAPM_INPUT("IN2L"), +SND_SOC_DAPM_INPUT("IN2R"), +SND_SOC_DAPM_INPUT("IN3L"), +SND_SOC_DAPM_INPUT("IN3R"), +SND_SOC_DAPM_INPUT("IN4L"), +SND_SOC_DAPM_INPUT("IN4R"), +SND_SOC_DAPM_INPUT("TONE"), + +SND_SOC_DAPM_PGA_E("IN1L PGA", WM5100_INPUT_ENABLES, WM5100_IN1L_ENA_SHIFT, 0, + NULL, 0, wm5100_out_ev, SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("IN1R PGA", WM5100_INPUT_ENABLES, WM5100_IN1R_ENA_SHIFT, 0, + NULL, 0, wm5100_out_ev, SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("IN2L PGA", WM5100_INPUT_ENABLES, WM5100_IN2L_ENA_SHIFT, 0, + NULL, 0, wm5100_out_ev, SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("IN2R PGA", WM5100_INPUT_ENABLES, WM5100_IN2R_ENA_SHIFT, 0, + NULL, 0, wm5100_out_ev, SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("IN3L PGA", WM5100_INPUT_ENABLES, WM5100_IN3L_ENA_SHIFT, 0, + NULL, 0, wm5100_out_ev, SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("IN3R PGA", WM5100_INPUT_ENABLES, WM5100_IN3R_ENA_SHIFT, 0, + NULL, 0, wm5100_out_ev, SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("IN4L PGA", WM5100_INPUT_ENABLES, WM5100_IN4L_ENA_SHIFT, 0, + NULL, 0, wm5100_out_ev, SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("IN4R PGA", WM5100_INPUT_ENABLES, WM5100_IN4R_ENA_SHIFT, 0, + NULL, 0, wm5100_out_ev, SND_SOC_DAPM_POST_PMU), + +SND_SOC_DAPM_PGA("Tone Generator 1", WM5100_TONE_GENERATOR_1, + WM5100_TONE1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("Tone Generator 2", WM5100_TONE_GENERATOR_1, + WM5100_TONE2_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_AIF_IN("AIF1RX1", "AIF1 Playback", 0, + WM5100_AUDIO_IF_1_27, WM5100_AIF1RX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX2", "AIF1 Playback", 1, + WM5100_AUDIO_IF_1_27, WM5100_AIF1RX2_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX3", "AIF1 Playback", 2, + WM5100_AUDIO_IF_1_27, WM5100_AIF1RX3_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX4", "AIF1 Playback", 3, + WM5100_AUDIO_IF_1_27, WM5100_AIF1RX4_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX5", "AIF1 Playback", 4, + WM5100_AUDIO_IF_1_27, WM5100_AIF1RX5_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX6", "AIF1 Playback", 5, + WM5100_AUDIO_IF_1_27, WM5100_AIF1RX6_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX7", "AIF1 Playback", 6, + WM5100_AUDIO_IF_1_27, WM5100_AIF1RX7_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX8", "AIF1 Playback", 7, + WM5100_AUDIO_IF_1_27, WM5100_AIF1RX8_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_IN("AIF2RX1", "AIF2 Playback", 0, + WM5100_AUDIO_IF_2_27, WM5100_AIF2RX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF2RX2", "AIF2 Playback", 1, + WM5100_AUDIO_IF_2_27, WM5100_AIF2RX2_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_IN("AIF3RX1", "AIF3 Playback", 0, + WM5100_AUDIO_IF_3_27, WM5100_AIF3RX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF3RX2", "AIF3 Playback", 1, + WM5100_AUDIO_IF_3_27, WM5100_AIF3RX2_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_OUT("AIF1TX1", "AIF1 Capture", 0, + WM5100_AUDIO_IF_1_26, WM5100_AIF1TX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX2", "AIF1 Capture", 1, + WM5100_AUDIO_IF_1_26, WM5100_AIF1TX2_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX3", "AIF1 Capture", 2, + WM5100_AUDIO_IF_1_26, WM5100_AIF1TX3_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX4", "AIF1 Capture", 3, + WM5100_AUDIO_IF_1_26, WM5100_AIF1TX4_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX5", "AIF1 Capture", 4, + WM5100_AUDIO_IF_1_26, WM5100_AIF1TX5_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX6", "AIF1 Capture", 5, + WM5100_AUDIO_IF_1_26, WM5100_AIF1TX6_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX7", "AIF1 Capture", 6, + WM5100_AUDIO_IF_1_26, WM5100_AIF1TX7_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX8", "AIF1 Capture", 7, + WM5100_AUDIO_IF_1_26, WM5100_AIF1TX8_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_OUT("AIF2TX1", "AIF2 Capture", 0, + WM5100_AUDIO_IF_2_26, WM5100_AIF2TX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF2TX2", "AIF2 Capture", 1, + WM5100_AUDIO_IF_2_26, WM5100_AIF2TX2_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_OUT("AIF3TX1", "AIF3 Capture", 0, + WM5100_AUDIO_IF_3_26, WM5100_AIF3TX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF3TX2", "AIF3 Capture", 1, + WM5100_AUDIO_IF_3_26, WM5100_AIF3TX2_ENA_SHIFT, 0), + +SND_SOC_DAPM_PGA_E("OUT6L", WM5100_OUTPUT_ENABLES_2, WM5100_OUT6L_ENA_SHIFT, 0, + NULL, 0, wm5100_out_ev, SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT6R", WM5100_OUTPUT_ENABLES_2, WM5100_OUT6R_ENA_SHIFT, 0, + NULL, 0, wm5100_out_ev, SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT5L", WM5100_OUTPUT_ENABLES_2, WM5100_OUT5L_ENA_SHIFT, 0, + NULL, 0, wm5100_out_ev, SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT5R", WM5100_OUTPUT_ENABLES_2, WM5100_OUT5R_ENA_SHIFT, 0, + NULL, 0, wm5100_out_ev, SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT4L", WM5100_OUTPUT_ENABLES_2, WM5100_OUT4L_ENA_SHIFT, 0, + NULL, 0, wm5100_out_ev, SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT4R", WM5100_OUTPUT_ENABLES_2, WM5100_OUT4R_ENA_SHIFT, 0, + NULL, 0, wm5100_out_ev, SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT3L", WM5100_CHANNEL_ENABLES_1, WM5100_HP3L_ENA_SHIFT, 0, + NULL, 0, wm5100_out_ev, SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT3R", WM5100_CHANNEL_ENABLES_1, WM5100_HP3R_ENA_SHIFT, 0, + NULL, 0, wm5100_out_ev, SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT2L", WM5100_CHANNEL_ENABLES_1, WM5100_HP2L_ENA_SHIFT, 0, + NULL, 0, wm5100_out_ev, SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT2R", WM5100_CHANNEL_ENABLES_1, WM5100_HP2R_ENA_SHIFT, 0, + NULL, 0, wm5100_out_ev, SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT1L", WM5100_CHANNEL_ENABLES_1, WM5100_HP1L_ENA_SHIFT, 0, + NULL, 0, wm5100_out_ev, SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT1R", WM5100_CHANNEL_ENABLES_1, WM5100_HP1R_ENA_SHIFT, 0, + NULL, 0, wm5100_out_ev, SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("PWM1 Driver", WM5100_PWM_DRIVE_1, WM5100_PWM1_ENA_SHIFT, 0, + NULL, 0, wm5100_out_ev, SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("PWM2 Driver", WM5100_PWM_DRIVE_1, WM5100_PWM2_ENA_SHIFT, 0, + NULL, 0, wm5100_out_ev, SND_SOC_DAPM_POST_PMU), + +SND_SOC_DAPM_PGA("EQ1", WM5100_EQ1_1, WM5100_EQ1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("EQ2", WM5100_EQ2_1, WM5100_EQ2_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("EQ3", WM5100_EQ3_1, WM5100_EQ3_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("EQ4", WM5100_EQ4_1, WM5100_EQ4_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("DRC1L", WM5100_DRC1_CTRL1, WM5100_DRCL_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("DRC1R", WM5100_DRC1_CTRL1, WM5100_DRCR_ENA_SHIFT, 0, + NULL, 0), + +SND_SOC_DAPM_PGA("LHPF1", WM5100_HPLPF1_1, WM5100_LHPF1_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("LHPF2", WM5100_HPLPF2_1, WM5100_LHPF2_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("LHPF3", WM5100_HPLPF3_1, WM5100_LHPF3_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("LHPF4", WM5100_HPLPF4_1, WM5100_LHPF4_ENA_SHIFT, 0, + NULL, 0), + +WM5100_MIXER_WIDGETS(EQ1, "EQ1"), +WM5100_MIXER_WIDGETS(EQ2, "EQ2"), +WM5100_MIXER_WIDGETS(EQ3, "EQ3"), +WM5100_MIXER_WIDGETS(EQ4, "EQ4"), + +WM5100_MIXER_WIDGETS(DRC1L, "DRC1L"), +WM5100_MIXER_WIDGETS(DRC1R, "DRC1R"), + +WM5100_MIXER_WIDGETS(LHPF1, "LHPF1"), +WM5100_MIXER_WIDGETS(LHPF2, "LHPF2"), +WM5100_MIXER_WIDGETS(LHPF3, "LHPF3"), +WM5100_MIXER_WIDGETS(LHPF4, "LHPF4"), + +WM5100_MIXER_WIDGETS(AIF1TX1, "AIF1TX1"), +WM5100_MIXER_WIDGETS(AIF1TX2, "AIF1TX2"), +WM5100_MIXER_WIDGETS(AIF1TX3, "AIF1TX3"), +WM5100_MIXER_WIDGETS(AIF1TX4, "AIF1TX4"), +WM5100_MIXER_WIDGETS(AIF1TX5, "AIF1TX5"), +WM5100_MIXER_WIDGETS(AIF1TX6, "AIF1TX6"), +WM5100_MIXER_WIDGETS(AIF1TX7, "AIF1TX7"), +WM5100_MIXER_WIDGETS(AIF1TX8, "AIF1TX8"), + +WM5100_MIXER_WIDGETS(AIF2TX1, "AIF2TX1"), +WM5100_MIXER_WIDGETS(AIF2TX2, "AIF2TX2"), + +WM5100_MIXER_WIDGETS(AIF3TX1, "AIF3TX1"), +WM5100_MIXER_WIDGETS(AIF3TX2, "AIF3TX2"), + +WM5100_MIXER_WIDGETS(HPOUT1L, "HPOUT1L"), +WM5100_MIXER_WIDGETS(HPOUT1R, "HPOUT1R"), +WM5100_MIXER_WIDGETS(HPOUT2L, "HPOUT2L"), +WM5100_MIXER_WIDGETS(HPOUT2R, "HPOUT2R"), +WM5100_MIXER_WIDGETS(HPOUT3L, "HPOUT3L"), +WM5100_MIXER_WIDGETS(HPOUT3R, "HPOUT3R"), + +WM5100_MIXER_WIDGETS(SPKOUTL, "SPKOUTL"), +WM5100_MIXER_WIDGETS(SPKOUTR, "SPKOUTR"), +WM5100_MIXER_WIDGETS(SPKDAT1L, "SPKDAT1L"), +WM5100_MIXER_WIDGETS(SPKDAT1R, "SPKDAT1R"), +WM5100_MIXER_WIDGETS(SPKDAT2L, "SPKDAT2L"), +WM5100_MIXER_WIDGETS(SPKDAT2R, "SPKDAT2R"), + +WM5100_MIXER_WIDGETS(PWM1, "PWM1"), +WM5100_MIXER_WIDGETS(PWM2, "PWM2"), + +SND_SOC_DAPM_OUTPUT("HPOUT1L"), +SND_SOC_DAPM_OUTPUT("HPOUT1R"), +SND_SOC_DAPM_OUTPUT("HPOUT2L"), +SND_SOC_DAPM_OUTPUT("HPOUT2R"), +SND_SOC_DAPM_OUTPUT("HPOUT3L"), +SND_SOC_DAPM_OUTPUT("HPOUT3R"), +SND_SOC_DAPM_OUTPUT("SPKOUTL"), +SND_SOC_DAPM_OUTPUT("SPKOUTR"), +SND_SOC_DAPM_OUTPUT("SPKDAT1"), +SND_SOC_DAPM_OUTPUT("SPKDAT2"), +SND_SOC_DAPM_OUTPUT("PWM1"), +SND_SOC_DAPM_OUTPUT("PWM2"), +}; + +/* We register a _POST event if we don't have IRQ support so we can + * look at the error status from the CODEC - if we've got the IRQ + * hooked up then we will get prompted to look by an interrupt. + */ +static const struct snd_soc_dapm_widget wm5100_dapm_widgets_noirq[] = { +SND_SOC_DAPM_POST("Post", wm5100_post_ev), +}; + +static const struct snd_soc_dapm_route wm5100_dapm_routes[] = { + { "IN1L", NULL, "SYSCLK" }, + { "IN1R", NULL, "SYSCLK" }, + { "IN2L", NULL, "SYSCLK" }, + { "IN2R", NULL, "SYSCLK" }, + { "IN3L", NULL, "SYSCLK" }, + { "IN3R", NULL, "SYSCLK" }, + { "IN4L", NULL, "SYSCLK" }, + { "IN4R", NULL, "SYSCLK" }, + + { "OUT1L", NULL, "SYSCLK" }, + { "OUT1R", NULL, "SYSCLK" }, + { "OUT2L", NULL, "SYSCLK" }, + { "OUT2R", NULL, "SYSCLK" }, + { "OUT3L", NULL, "SYSCLK" }, + { "OUT3R", NULL, "SYSCLK" }, + { "OUT4L", NULL, "SYSCLK" }, + { "OUT4R", NULL, "SYSCLK" }, + { "OUT5L", NULL, "SYSCLK" }, + { "OUT5R", NULL, "SYSCLK" }, + { "OUT6L", NULL, "SYSCLK" }, + { "OUT6R", NULL, "SYSCLK" }, + + { "AIF1RX1", NULL, "SYSCLK" }, + { "AIF1RX2", NULL, "SYSCLK" }, + { "AIF1RX3", NULL, "SYSCLK" }, + { "AIF1RX4", NULL, "SYSCLK" }, + { "AIF1RX5", NULL, "SYSCLK" }, + { "AIF1RX6", NULL, "SYSCLK" }, + { "AIF1RX7", NULL, "SYSCLK" }, + { "AIF1RX8", NULL, "SYSCLK" }, + + { "AIF2RX1", NULL, "SYSCLK" }, + { "AIF2RX1", NULL, "DBVDD2" }, + { "AIF2RX2", NULL, "SYSCLK" }, + { "AIF2RX2", NULL, "DBVDD2" }, + + { "AIF3RX1", NULL, "SYSCLK" }, + { "AIF3RX1", NULL, "DBVDD3" }, + { "AIF3RX2", NULL, "SYSCLK" }, + { "AIF3RX2", NULL, "DBVDD3" }, + + { "AIF1TX1", NULL, "SYSCLK" }, + { "AIF1TX2", NULL, "SYSCLK" }, + { "AIF1TX3", NULL, "SYSCLK" }, + { "AIF1TX4", NULL, "SYSCLK" }, + { "AIF1TX5", NULL, "SYSCLK" }, + { "AIF1TX6", NULL, "SYSCLK" }, + { "AIF1TX7", NULL, "SYSCLK" }, + { "AIF1TX8", NULL, "SYSCLK" }, + + { "AIF2TX1", NULL, "SYSCLK" }, + { "AIF2TX1", NULL, "DBVDD2" }, + { "AIF2TX2", NULL, "SYSCLK" }, + { "AIF2TX2", NULL, "DBVDD2" }, + + { "AIF3TX1", NULL, "SYSCLK" }, + { "AIF3TX1", NULL, "DBVDD3" }, + { "AIF3TX2", NULL, "SYSCLK" }, + { "AIF3TX2", NULL, "DBVDD3" }, + + { "MICBIAS1", NULL, "CP2" }, + { "MICBIAS2", NULL, "CP2" }, + { "MICBIAS3", NULL, "CP2" }, + + { "IN1L PGA", NULL, "CP2" }, + { "IN1R PGA", NULL, "CP2" }, + { "IN2L PGA", NULL, "CP2" }, + { "IN2R PGA", NULL, "CP2" }, + { "IN3L PGA", NULL, "CP2" }, + { "IN3R PGA", NULL, "CP2" }, + { "IN4L PGA", NULL, "CP2" }, + { "IN4R PGA", NULL, "CP2" }, + + { "IN1L PGA", NULL, "CP2 Active" }, + { "IN1R PGA", NULL, "CP2 Active" }, + { "IN2L PGA", NULL, "CP2 Active" }, + { "IN2R PGA", NULL, "CP2 Active" }, + { "IN3L PGA", NULL, "CP2 Active" }, + { "IN3R PGA", NULL, "CP2 Active" }, + { "IN4L PGA", NULL, "CP2 Active" }, + { "IN4R PGA", NULL, "CP2 Active" }, + + { "OUT1L", NULL, "CP1" }, + { "OUT1R", NULL, "CP1" }, + { "OUT2L", NULL, "CP1" }, + { "OUT2R", NULL, "CP1" }, + { "OUT3L", NULL, "CP1" }, + { "OUT3R", NULL, "CP1" }, + + { "Tone Generator 1", NULL, "TONE" }, + { "Tone Generator 2", NULL, "TONE" }, + + { "IN1L PGA", NULL, "IN1L" }, + { "IN1R PGA", NULL, "IN1R" }, + { "IN2L PGA", NULL, "IN2L" }, + { "IN2R PGA", NULL, "IN2R" }, + { "IN3L PGA", NULL, "IN3L" }, + { "IN3R PGA", NULL, "IN3R" }, + { "IN4L PGA", NULL, "IN4L" }, + { "IN4R PGA", NULL, "IN4R" }, + + WM5100_MIXER_ROUTES("OUT1L", "HPOUT1L"), + WM5100_MIXER_ROUTES("OUT1R", "HPOUT1R"), + WM5100_MIXER_ROUTES("OUT2L", "HPOUT2L"), + WM5100_MIXER_ROUTES("OUT2R", "HPOUT2R"), + WM5100_MIXER_ROUTES("OUT3L", "HPOUT3L"), + WM5100_MIXER_ROUTES("OUT3R", "HPOUT3R"), + + WM5100_MIXER_ROUTES("OUT4L", "SPKOUTL"), + WM5100_MIXER_ROUTES("OUT4R", "SPKOUTR"), + WM5100_MIXER_ROUTES("OUT5L", "SPKDAT1L"), + WM5100_MIXER_ROUTES("OUT5R", "SPKDAT1R"), + WM5100_MIXER_ROUTES("OUT6L", "SPKDAT2L"), + WM5100_MIXER_ROUTES("OUT6R", "SPKDAT2R"), + + WM5100_MIXER_ROUTES("PWM1 Driver", "PWM1"), + WM5100_MIXER_ROUTES("PWM2 Driver", "PWM2"), + + WM5100_MIXER_ROUTES("AIF1TX1", "AIF1TX1"), + WM5100_MIXER_ROUTES("AIF1TX2", "AIF1TX2"), + WM5100_MIXER_ROUTES("AIF1TX3", "AIF1TX3"), + WM5100_MIXER_ROUTES("AIF1TX4", "AIF1TX4"), + WM5100_MIXER_ROUTES("AIF1TX5", "AIF1TX5"), + WM5100_MIXER_ROUTES("AIF1TX6", "AIF1TX6"), + WM5100_MIXER_ROUTES("AIF1TX7", "AIF1TX7"), + WM5100_MIXER_ROUTES("AIF1TX8", "AIF1TX8"), + + WM5100_MIXER_ROUTES("AIF2TX1", "AIF2TX1"), + WM5100_MIXER_ROUTES("AIF2TX2", "AIF2TX2"), + + WM5100_MIXER_ROUTES("AIF3TX1", "AIF3TX1"), + WM5100_MIXER_ROUTES("AIF3TX2", "AIF3TX2"), + + WM5100_MIXER_ROUTES("EQ1", "EQ1"), + WM5100_MIXER_ROUTES("EQ2", "EQ2"), + WM5100_MIXER_ROUTES("EQ3", "EQ3"), + WM5100_MIXER_ROUTES("EQ4", "EQ4"), + + WM5100_MIXER_ROUTES("DRC1L", "DRC1L"), + WM5100_MIXER_ROUTES("DRC1R", "DRC1R"), + + WM5100_MIXER_ROUTES("LHPF1", "LHPF1"), + WM5100_MIXER_ROUTES("LHPF2", "LHPF2"), + WM5100_MIXER_ROUTES("LHPF3", "LHPF3"), + WM5100_MIXER_ROUTES("LHPF4", "LHPF4"), + + { "HPOUT1L", NULL, "OUT1L" }, + { "HPOUT1R", NULL, "OUT1R" }, + { "HPOUT2L", NULL, "OUT2L" }, + { "HPOUT2R", NULL, "OUT2R" }, + { "HPOUT3L", NULL, "OUT3L" }, + { "HPOUT3R", NULL, "OUT3R" }, + { "SPKOUTL", NULL, "OUT4L" }, + { "SPKOUTR", NULL, "OUT4R" }, + { "SPKDAT1", NULL, "OUT5L" }, + { "SPKDAT1", NULL, "OUT5R" }, + { "SPKDAT2", NULL, "OUT6L" }, + { "SPKDAT2", NULL, "OUT6R" }, + { "PWM1", NULL, "PWM1 Driver" }, + { "PWM2", NULL, "PWM2 Driver" }, +}; + +static struct { + int reg; + int val; +} wm5100_reva_patches[] = { + { WM5100_AUDIO_IF_1_10, 0 }, + { WM5100_AUDIO_IF_1_11, 1 }, + { WM5100_AUDIO_IF_1_12, 2 }, + { WM5100_AUDIO_IF_1_13, 3 }, + { WM5100_AUDIO_IF_1_14, 4 }, + { WM5100_AUDIO_IF_1_15, 5 }, + { WM5100_AUDIO_IF_1_16, 6 }, + { WM5100_AUDIO_IF_1_17, 7 }, + + { WM5100_AUDIO_IF_1_18, 0 }, + { WM5100_AUDIO_IF_1_19, 1 }, + { WM5100_AUDIO_IF_1_20, 2 }, + { WM5100_AUDIO_IF_1_21, 3 }, + { WM5100_AUDIO_IF_1_22, 4 }, + { WM5100_AUDIO_IF_1_23, 5 }, + { WM5100_AUDIO_IF_1_24, 6 }, + { WM5100_AUDIO_IF_1_25, 7 }, + + { WM5100_AUDIO_IF_2_10, 0 }, + { WM5100_AUDIO_IF_2_11, 1 }, + + { WM5100_AUDIO_IF_2_18, 0 }, + { WM5100_AUDIO_IF_2_19, 1 }, + + { WM5100_AUDIO_IF_3_10, 0 }, + { WM5100_AUDIO_IF_3_11, 1 }, + + { WM5100_AUDIO_IF_3_18, 0 }, + { WM5100_AUDIO_IF_3_19, 1 }, +}; + +static int wm5100_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec); + int ret, i; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + + case SND_SOC_BIAS_PREPARE: + break; + + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + ret = regulator_bulk_enable(ARRAY_SIZE(wm5100->core_supplies), + wm5100->core_supplies); + if (ret != 0) { + dev_err(codec->dev, + "Failed to enable supplies: %d\n", + ret); + return ret; + } + + if (wm5100->pdata.ldo_ena) { + gpio_set_value_cansleep(wm5100->pdata.ldo_ena, + 1); + msleep(2); + } + + codec->cache_only = false; + + switch (wm5100->rev) { + case 0: + snd_soc_write(codec, 0x11, 0x3); + snd_soc_write(codec, 0x203, 0xc); + snd_soc_write(codec, 0x206, 0); + snd_soc_write(codec, 0x207, 0xf0); + snd_soc_write(codec, 0x208, 0x3c); + snd_soc_write(codec, 0x209, 0); + snd_soc_write(codec, 0x211, 0x20d8); + snd_soc_write(codec, 0x11, 0); + + for (i = 0; + i < ARRAY_SIZE(wm5100_reva_patches); + i++) + snd_soc_write(codec, + wm5100_reva_patches[i].reg, + wm5100_reva_patches[i].val); + break; + default: + break; + } + + snd_soc_cache_sync(codec); + } + break; + + case SND_SOC_BIAS_OFF: + if (wm5100->pdata.ldo_ena) + gpio_set_value_cansleep(wm5100->pdata.ldo_ena, 0); + regulator_bulk_disable(ARRAY_SIZE(wm5100->core_supplies), + wm5100->core_supplies); + break; + } + codec->dapm.bias_level = level; + + return 0; +} + +static int wm5100_dai_to_base(struct snd_soc_dai *dai) +{ + switch (dai->id) { + case 0: + return WM5100_AUDIO_IF_1_1 - 1; + case 1: + return WM5100_AUDIO_IF_2_1 - 1; + case 2: + return WM5100_AUDIO_IF_3_1 - 1; + default: + BUG(); + return -EINVAL; + } +} + +static int wm5100_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + int lrclk, bclk, mask, base; + + base = wm5100_dai_to_base(dai); + if (base < 0) + return base; + + lrclk = 0; + bclk = 0; + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + mask = 0; + break; + case SND_SOC_DAIFMT_DSP_B: + mask = 1; + break; + case SND_SOC_DAIFMT_I2S: + mask = 2; + break; + case SND_SOC_DAIFMT_LEFT_J: + mask = 3; + break; + default: + dev_err(codec->dev, "Unsupported DAI format %d\n", + fmt & SND_SOC_DAIFMT_FORMAT_MASK); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + break; + case SND_SOC_DAIFMT_CBS_CFM: + lrclk |= WM5100_AIF1TX_LRCLK_MSTR; + break; + case SND_SOC_DAIFMT_CBM_CFS: + bclk |= WM5100_AIF1_BCLK_MSTR; + break; + case SND_SOC_DAIFMT_CBM_CFM: + lrclk |= WM5100_AIF1TX_LRCLK_MSTR; + bclk |= WM5100_AIF1_BCLK_MSTR; + break; + default: + dev_err(codec->dev, "Unsupported master mode %d\n", + fmt & SND_SOC_DAIFMT_MASTER_MASK); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + bclk |= WM5100_AIF1_BCLK_INV; + lrclk |= WM5100_AIF1TX_LRCLK_INV; + break; + case SND_SOC_DAIFMT_IB_NF: + bclk |= WM5100_AIF1_BCLK_INV; + break; + case SND_SOC_DAIFMT_NB_IF: + lrclk |= WM5100_AIF1TX_LRCLK_INV; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, base + 1, WM5100_AIF1_BCLK_MSTR | + WM5100_AIF1_BCLK_INV, bclk); + snd_soc_update_bits(codec, base + 2, WM5100_AIF1TX_LRCLK_MSTR | + WM5100_AIF1TX_LRCLK_INV, lrclk); + snd_soc_update_bits(codec, base + 3, WM5100_AIF1TX_LRCLK_MSTR | + WM5100_AIF1TX_LRCLK_INV, lrclk); + snd_soc_update_bits(codec, base + 5, WM5100_AIF1_FMT_MASK, mask); + + return 0; +} + +#define WM5100_NUM_BCLK_RATES 19 + +static int wm5100_bclk_rates_dat[WM5100_NUM_BCLK_RATES] = { + 32000, + 48000, + 64000, + 96000, + 128000, + 192000, + 256000, + 384000, + 512000, + 768000, + 1024000, + 1536000, + 2048000, + 3072000, + 4096000, + 6144000, + 8192000, + 12288000, + 24576000, +}; + +static int wm5100_bclk_rates_cd[WM5100_NUM_BCLK_RATES] = { + 29400, + 44100, + 58800, + 88200, + 117600, + 176400, + 235200, + 352800, + 470400, + 705600, + 940800, + 1411200, + 1881600, + 2882400, + 3763200, + 5644800, + 7526400, + 11289600, + 22579600, +}; + +static int wm5100_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec); + bool async = wm5100->aif_async[dai->id]; + int i, base, bclk, aif_rate, lrclk, wl, fl, sr; + int *bclk_rates; + + base = wm5100_dai_to_base(dai); + if (base < 0) + return base; + + /* Data sizes if not using TDM */ + wl = snd_pcm_format_width(params_format(params)); + if (wl < 0) + return wl; + fl = snd_soc_params_to_frame_size(params); + if (fl < 0) + return fl; + + dev_dbg(codec->dev, "Word length %d bits, frame length %d bits\n", + wl, fl); + + /* Target BCLK rate */ + bclk = snd_soc_params_to_bclk(params); + if (bclk < 0) + return bclk; + + /* Root for BCLK depends on SYS/ASYNCCLK */ + if (!async) { + aif_rate = wm5100->sysclk; + sr = wm5100_alloc_sr(codec, params_rate(params)); + if (sr < 0) + return sr; + } else { + /* If we're in ASYNCCLK set the ASYNC sample rate */ + aif_rate = wm5100->asyncclk; + sr = 3; + + for (i = 0; i < ARRAY_SIZE(wm5100_sr_code); i++) + if (params_rate(params) == wm5100_sr_code[i]) + break; + if (i == ARRAY_SIZE(wm5100_sr_code)) { + dev_err(codec->dev, "Invalid rate %dHzn", + params_rate(params)); + return -EINVAL; + } + + /* TODO: We should really check for symmetry */ + snd_soc_update_bits(codec, WM5100_CLOCKING_8, + WM5100_ASYNC_SAMPLE_RATE_MASK, i); + } + + if (!aif_rate) { + dev_err(codec->dev, "%s has no rate set\n", + async ? "ASYNCCLK" : "SYSCLK"); + return -EINVAL; + } + + dev_dbg(codec->dev, "Target BCLK is %dHz, using %dHz %s\n", + bclk, aif_rate, async ? "ASYNCCLK" : "SYSCLK"); + + if (aif_rate % 4000) + bclk_rates = wm5100_bclk_rates_cd; + else + bclk_rates = wm5100_bclk_rates_dat; + + for (i = 0; i < WM5100_NUM_BCLK_RATES; i++) + if (bclk_rates[i] >= bclk && (bclk_rates[i] % bclk == 0)) + break; + if (i == WM5100_NUM_BCLK_RATES) { + dev_err(codec->dev, + "No valid BCLK for %dHz found from %dHz %s\n", + bclk, aif_rate, async ? "ASYNCCLK" : "SYSCLK"); + return -EINVAL; + } + + bclk = i; + dev_dbg(codec->dev, "Setting %dHz BCLK\n", bclk_rates[bclk]); + snd_soc_update_bits(codec, base + 1, WM5100_AIF1_BCLK_FREQ_MASK, bclk); + + lrclk = bclk_rates[bclk] / params_rate(params); + dev_dbg(codec->dev, "Setting %dHz LRCLK\n", bclk_rates[bclk] / lrclk); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK || + wm5100->aif_symmetric[dai->id]) + snd_soc_update_bits(codec, base + 7, + WM5100_AIF1RX_BCPF_MASK, lrclk); + else + snd_soc_update_bits(codec, base + 6, + WM5100_AIF1TX_BCPF_MASK, lrclk); + + i = (wl << WM5100_AIF1TX_WL_SHIFT) | fl; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + snd_soc_update_bits(codec, base + 9, + WM5100_AIF1RX_WL_MASK | + WM5100_AIF1RX_SLOT_LEN_MASK, i); + else + snd_soc_update_bits(codec, base + 8, + WM5100_AIF1TX_WL_MASK | + WM5100_AIF1TX_SLOT_LEN_MASK, i); + + snd_soc_update_bits(codec, base + 4, WM5100_AIF1_RATE_MASK, sr); + + return 0; +} + +static struct snd_soc_dai_ops wm5100_dai_ops = { + .set_fmt = wm5100_set_fmt, + .hw_params = wm5100_hw_params, +}; + +static int wm5100_set_sysclk(struct snd_soc_codec *codec, int clk_id, + int source, unsigned int freq, int dir) +{ + struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec); + int *rate_store; + int fval, audio_rate, ret, reg; + + switch (clk_id) { + case WM5100_CLK_SYSCLK: + reg = WM5100_CLOCKING_3; + rate_store = &wm5100->sysclk; + break; + case WM5100_CLK_ASYNCCLK: + reg = WM5100_CLOCKING_7; + rate_store = &wm5100->asyncclk; + break; + case WM5100_CLK_32KHZ: + /* The 32kHz clock is slightly different to the others */ + switch (source) { + case WM5100_CLKSRC_MCLK1: + case WM5100_CLKSRC_MCLK2: + case WM5100_CLKSRC_SYSCLK: + snd_soc_update_bits(codec, WM5100_CLOCKING_1, + WM5100_CLK_32K_SRC_MASK, + source); + break; + default: + return -EINVAL; + } + return 0; + + case WM5100_CLK_AIF1: + case WM5100_CLK_AIF2: + case WM5100_CLK_AIF3: + /* Not real clocks, record which clock domain they're in */ + switch (source) { + case WM5100_CLKSRC_SYSCLK: + wm5100->aif_async[clk_id - 1] = false; + break; + case WM5100_CLKSRC_ASYNCCLK: + wm5100->aif_async[clk_id - 1] = true; + break; + default: + dev_err(codec->dev, "Invalid source %d\n", source); + return -EINVAL; + } + return 0; + + case WM5100_CLK_OPCLK: + switch (freq) { + case 5644800: + case 6144000: + snd_soc_update_bits(codec, WM5100_MISC_GPIO_1, + WM5100_OPCLK_SEL_MASK, 0); + break; + case 11289600: + case 12288000: + snd_soc_update_bits(codec, WM5100_MISC_GPIO_1, + WM5100_OPCLK_SEL_MASK, 0); + break; + case 22579200: + case 24576000: + snd_soc_update_bits(codec, WM5100_MISC_GPIO_1, + WM5100_OPCLK_SEL_MASK, 0); + break; + default: + dev_err(codec->dev, "Unsupported OPCLK %dHz\n", + freq); + return -EINVAL; + } + return 0; + + default: + dev_err(codec->dev, "Unknown clock %d\n", clk_id); + return -EINVAL; + } + + switch (source) { + case WM5100_CLKSRC_SYSCLK: + case WM5100_CLKSRC_ASYNCCLK: + dev_err(codec->dev, "Invalid source %d\n", source); + return -EINVAL; + } + + switch (freq) { + case 5644800: + case 6144000: + fval = 0; + break; + case 11289600: + case 12288000: + fval = 1; + break; + case 22579200: + case 24576000: + fval = 2; + break; + default: + dev_err(codec->dev, "Invalid clock rate: %d\n", freq); + return -EINVAL; + } + + switch (freq) { + case 5644800: + case 11289600: + case 22579200: + audio_rate = 44100; + break; + + case 6144000: + case 12288000: + case 24576000: + audio_rate = 48000; + break; + + default: + BUG(); + audio_rate = 0; + break; + } + + /* TODO: Check if MCLKs are in use and enable/disable pulls to + * match. + */ + + snd_soc_update_bits(codec, reg, WM5100_SYSCLK_FREQ_MASK | + WM5100_SYSCLK_SRC_MASK, + fval << WM5100_SYSCLK_FREQ_SHIFT | source); + + /* If this is SYSCLK then configure the clock rate for the + * internal audio functions to the natural sample rate for + * this clock rate. + */ + if (clk_id == WM5100_CLK_SYSCLK) { + dev_dbg(codec->dev, "Setting primary audio rate to %dHz", + audio_rate); + if (0 && *rate_store) + wm5100_free_sr(codec, audio_rate); + ret = wm5100_alloc_sr(codec, audio_rate); + if (ret != 0) + dev_warn(codec->dev, "Primary audio slot is %d\n", + ret); + } + + *rate_store = freq; + + return 0; +} + +struct _fll_div { + u16 fll_fratio; + u16 fll_outdiv; + u16 fll_refclk_div; + u16 n; + u16 theta; + u16 lambda; +}; + +static struct { + unsigned int min; + unsigned int max; + u16 fll_fratio; + int ratio; +} fll_fratios[] = { + { 0, 64000, 4, 16 }, + { 64000, 128000, 3, 8 }, + { 128000, 256000, 2, 4 }, + { 256000, 1000000, 1, 2 }, + { 1000000, 13500000, 0, 1 }, +}; + +static int fll_factors(struct _fll_div *fll_div, unsigned int Fref, + unsigned int Fout) +{ + unsigned int target; + unsigned int div; + unsigned int fratio, gcd_fll; + int i; + + /* Fref must be <=13.5MHz */ + div = 1; + fll_div->fll_refclk_div = 0; + while ((Fref / div) > 13500000) { + div *= 2; + fll_div->fll_refclk_div++; + + if (div > 8) { + pr_err("Can't scale %dMHz input down to <=13.5MHz\n", + Fref); + return -EINVAL; + } + } + + pr_debug("FLL Fref=%u Fout=%u\n", Fref, Fout); + + /* Apply the division for our remaining calculations */ + Fref /= div; + + /* Fvco should be 90-100MHz; don't check the upper bound */ + div = 2; + while (Fout * div < 90000000) { + div++; + if (div > 64) { + pr_err("Unable to find FLL_OUTDIV for Fout=%uHz\n", + Fout); + return -EINVAL; + } + } + target = Fout * div; + fll_div->fll_outdiv = div - 1; + + pr_debug("FLL Fvco=%dHz\n", target); + + /* Find an appropraite FLL_FRATIO and factor it out of the target */ + for (i = 0; i < ARRAY_SIZE(fll_fratios); i++) { + if (fll_fratios[i].min <= Fref && Fref <= fll_fratios[i].max) { + fll_div->fll_fratio = fll_fratios[i].fll_fratio; + fratio = fll_fratios[i].ratio; + break; + } + } + if (i == ARRAY_SIZE(fll_fratios)) { + pr_err("Unable to find FLL_FRATIO for Fref=%uHz\n", Fref); + return -EINVAL; + } + + fll_div->n = target / (fratio * Fref); + + if (target % Fref == 0) { + fll_div->theta = 0; + fll_div->lambda = 0; + } else { + gcd_fll = gcd(target, fratio * Fref); + + fll_div->theta = (target - (fll_div->n * fratio * Fref)) + / gcd_fll; + fll_div->lambda = (fratio * Fref) / gcd_fll; + } + + pr_debug("FLL N=%x THETA=%x LAMBDA=%x\n", + fll_div->n, fll_div->theta, fll_div->lambda); + pr_debug("FLL_FRATIO=%x(%d) FLL_OUTDIV=%x FLL_REFCLK_DIV=%x\n", + fll_div->fll_fratio, fratio, fll_div->fll_outdiv, + fll_div->fll_refclk_div); + + return 0; +} + +static int wm5100_set_fll(struct snd_soc_codec *codec, int fll_id, int source, + unsigned int Fref, unsigned int Fout) +{ + struct i2c_client *i2c = to_i2c_client(codec->dev); + struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec); + struct _fll_div factors; + struct wm5100_fll *fll; + int ret, base, lock, i, timeout; + + switch (fll_id) { + case WM5100_FLL1: + fll = &wm5100->fll[0]; + base = WM5100_FLL1_CONTROL_1 - 1; + lock = WM5100_FLL1_LOCK_STS; + break; + case WM5100_FLL2: + fll = &wm5100->fll[1]; + base = WM5100_FLL2_CONTROL_2 - 1; + lock = WM5100_FLL2_LOCK_STS; + break; + default: + dev_err(codec->dev, "Unknown FLL %d\n",fll_id); + return -EINVAL; + } + + if (!Fout) { + dev_dbg(codec->dev, "FLL%d disabled", fll_id); + fll->fout = 0; + snd_soc_update_bits(codec, base + 1, WM5100_FLL1_ENA, 0); + return 0; + } + + switch (source) { + case WM5100_FLL_SRC_MCLK1: + case WM5100_FLL_SRC_MCLK2: + case WM5100_FLL_SRC_FLL1: + case WM5100_FLL_SRC_FLL2: + case WM5100_FLL_SRC_AIF1BCLK: + case WM5100_FLL_SRC_AIF2BCLK: + case WM5100_FLL_SRC_AIF3BCLK: + break; + default: + dev_err(codec->dev, "Invalid FLL source %d\n", source); + return -EINVAL; + } + + ret = fll_factors(&factors, Fref, Fout); + if (ret < 0) + return ret; + + /* Disable the FLL while we reconfigure */ + snd_soc_update_bits(codec, base + 1, WM5100_FLL1_ENA, 0); + + snd_soc_update_bits(codec, base + 2, + WM5100_FLL1_OUTDIV_MASK | WM5100_FLL1_FRATIO_MASK, + (factors.fll_outdiv << WM5100_FLL1_OUTDIV_SHIFT) | + factors.fll_fratio); + snd_soc_update_bits(codec, base + 3, WM5100_FLL1_THETA_MASK, + factors.theta); + snd_soc_update_bits(codec, base + 5, WM5100_FLL1_N_MASK, factors.n); + snd_soc_update_bits(codec, base + 6, + WM5100_FLL1_REFCLK_DIV_MASK | + WM5100_FLL1_REFCLK_SRC_MASK, + (factors.fll_refclk_div + << WM5100_FLL1_REFCLK_DIV_SHIFT) | source); + snd_soc_update_bits(codec, base + 7, WM5100_FLL1_LAMBDA_MASK, + factors.lambda); + + /* Clear any pending completions */ + try_wait_for_completion(&fll->lock); + + snd_soc_update_bits(codec, base + 1, WM5100_FLL1_ENA, WM5100_FLL1_ENA); + + if (i2c->irq) + timeout = 2; + else + timeout = 50; + + /* Poll for the lock; will use interrupt when we can test */ + for (i = 0; i < timeout; i++) { + if (i2c->irq) { + ret = wait_for_completion_timeout(&fll->lock, + msecs_to_jiffies(25)); + if (ret > 0) + break; + } else { + msleep(1); + } + + ret = snd_soc_read(codec, + WM5100_INTERRUPT_RAW_STATUS_3); + if (ret < 0) { + dev_err(codec->dev, + "Failed to read FLL status: %d\n", + ret); + continue; + } + if (ret & lock) + break; + } + if (i == timeout) { + dev_err(codec->dev, "FLL%d lock timed out\n", fll_id); + return -ETIMEDOUT; + } + + fll->src = source; + fll->fref = Fref; + fll->fout = Fout; + + dev_dbg(codec->dev, "FLL%d running %dHz->%dHz\n", fll_id, + Fref, Fout); + + return 0; +} + +/* Actually go much higher */ +#define WM5100_RATES SNDRV_PCM_RATE_8000_192000 + +#define WM5100_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_driver wm5100_dai[] = { + { + .name = "wm5100-aif1", + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 2, + .channels_max = 2, + .rates = WM5100_RATES, + .formats = WM5100_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 2, + .channels_max = 2, + .rates = WM5100_RATES, + .formats = WM5100_FORMATS, + }, + .ops = &wm5100_dai_ops, + }, + { + .name = "wm5100-aif2", + .id = 1, + .playback = { + .stream_name = "AIF2 Playback", + .channels_min = 2, + .channels_max = 2, + .rates = WM5100_RATES, + .formats = WM5100_FORMATS, + }, + .capture = { + .stream_name = "AIF2 Capture", + .channels_min = 2, + .channels_max = 2, + .rates = WM5100_RATES, + .formats = WM5100_FORMATS, + }, + .ops = &wm5100_dai_ops, + }, + { + .name = "wm5100-aif3", + .id = 2, + .playback = { + .stream_name = "AIF3 Playback", + .channels_min = 2, + .channels_max = 2, + .rates = WM5100_RATES, + .formats = WM5100_FORMATS, + }, + .capture = { + .stream_name = "AIF3 Capture", + .channels_min = 2, + .channels_max = 2, + .rates = WM5100_RATES, + .formats = WM5100_FORMATS, + }, + .ops = &wm5100_dai_ops, + }, +}; + +static int wm5100_dig_vu[] = { + WM5100_ADC_DIGITAL_VOLUME_1L, + WM5100_ADC_DIGITAL_VOLUME_1R, + WM5100_ADC_DIGITAL_VOLUME_2L, + WM5100_ADC_DIGITAL_VOLUME_2R, + WM5100_ADC_DIGITAL_VOLUME_3L, + WM5100_ADC_DIGITAL_VOLUME_3R, + WM5100_ADC_DIGITAL_VOLUME_4L, + WM5100_ADC_DIGITAL_VOLUME_4R, + + WM5100_DAC_DIGITAL_VOLUME_1L, + WM5100_DAC_DIGITAL_VOLUME_1R, + WM5100_DAC_DIGITAL_VOLUME_2L, + WM5100_DAC_DIGITAL_VOLUME_2R, + WM5100_DAC_DIGITAL_VOLUME_3L, + WM5100_DAC_DIGITAL_VOLUME_3R, + WM5100_DAC_DIGITAL_VOLUME_4L, + WM5100_DAC_DIGITAL_VOLUME_4R, + WM5100_DAC_DIGITAL_VOLUME_5L, + WM5100_DAC_DIGITAL_VOLUME_5R, + WM5100_DAC_DIGITAL_VOLUME_6L, + WM5100_DAC_DIGITAL_VOLUME_6R, +}; + +static void wm5100_set_detect_mode(struct snd_soc_codec *codec, int the_mode) +{ + struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec); + struct wm5100_jack_mode *mode = &wm5100->pdata.jack_modes[the_mode]; + + BUG_ON(the_mode >= ARRAY_SIZE(wm5100->pdata.jack_modes)); + + gpio_set_value_cansleep(wm5100->pdata.hp_pol, mode->hp_pol); + snd_soc_update_bits(codec, WM5100_ACCESSORY_DETECT_MODE_1, + WM5100_ACCDET_BIAS_SRC_MASK | + WM5100_ACCDET_SRC, + (mode->bias << WM5100_ACCDET_BIAS_SRC_SHIFT) | + mode->micd_src << WM5100_ACCDET_SRC_SHIFT); + snd_soc_update_bits(codec, WM5100_MISC_CONTROL, + WM5100_HPCOM_SRC, + mode->micd_src << WM5100_HPCOM_SRC_SHIFT); + + wm5100->jack_mode = the_mode; + + dev_dbg(codec->dev, "Set microphone polarity to %d\n", + wm5100->jack_mode); +} + +static void wm5100_micd_irq(struct snd_soc_codec *codec) +{ + struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec); + int val; + + val = snd_soc_read(codec, WM5100_MIC_DETECT_3); + + dev_dbg(codec->dev, "Microphone event: %x\n", val); + + if (!(val & WM5100_ACCDET_VALID)) { + dev_warn(codec->dev, "Microphone detection state invalid\n"); + return; + } + + /* No accessory, reset everything and report removal */ + if (!(val & WM5100_ACCDET_STS)) { + dev_dbg(codec->dev, "Jack removal detected\n"); + wm5100->jack_mic = false; + wm5100->jack_detecting = true; + snd_soc_jack_report(wm5100->jack, 0, + SND_JACK_LINEOUT | SND_JACK_HEADSET | + SND_JACK_BTN_0); + + snd_soc_update_bits(codec, WM5100_MIC_DETECT_1, + WM5100_ACCDET_RATE_MASK, + WM5100_ACCDET_RATE_MASK); + return; + } + + /* If the measurement is very high we've got a microphone, + * either we just detected one or if we already reported then + * we've got a button release event. + */ + if (val & 0x400) { + if (wm5100->jack_detecting) { + dev_dbg(codec->dev, "Microphone detected\n"); + wm5100->jack_mic = true; + snd_soc_jack_report(wm5100->jack, + SND_JACK_HEADSET, + SND_JACK_HEADSET | SND_JACK_BTN_0); + + /* Increase poll rate to give better responsiveness + * for buttons */ + snd_soc_update_bits(codec, WM5100_MIC_DETECT_1, + WM5100_ACCDET_RATE_MASK, + 5 << WM5100_ACCDET_RATE_SHIFT); + } else { + dev_dbg(codec->dev, "Mic button up\n"); + snd_soc_jack_report(wm5100->jack, 0, SND_JACK_BTN_0); + } + + return; + } + + /* If we detected a lower impedence during initial startup + * then we probably have the wrong polarity, flip it. Don't + * do this for the lowest impedences to speed up detection of + * plain headphones. + */ + if (wm5100->jack_detecting && (val & 0x3f8)) { + wm5100_set_detect_mode(codec, !wm5100->jack_mode); + + return; + } + + /* Don't distinguish between buttons, just report any low + * impedence as BTN_0. + */ + if (val & 0x3fc) { + if (wm5100->jack_mic) { + dev_dbg(codec->dev, "Mic button detected\n"); + snd_soc_jack_report(wm5100->jack, SND_JACK_BTN_0, + SND_JACK_BTN_0); + } else if (wm5100->jack_detecting) { + dev_dbg(codec->dev, "Headphone detected\n"); + snd_soc_jack_report(wm5100->jack, SND_JACK_HEADPHONE, + SND_JACK_HEADPHONE); + + /* Increase the detection rate a bit for + * responsiveness. + */ + snd_soc_update_bits(codec, WM5100_MIC_DETECT_1, + WM5100_ACCDET_RATE_MASK, + 7 << WM5100_ACCDET_RATE_SHIFT); + } + } +} + +int wm5100_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack) +{ + struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec); + + if (jack) { + wm5100->jack = jack; + wm5100->jack_detecting = true; + + wm5100_set_detect_mode(codec, 0); + + /* Slowest detection rate, gives debounce for initial + * detection */ + snd_soc_update_bits(codec, WM5100_MIC_DETECT_1, + WM5100_ACCDET_BIAS_STARTTIME_MASK | + WM5100_ACCDET_RATE_MASK, + (7 << WM5100_ACCDET_BIAS_STARTTIME_SHIFT) | + WM5100_ACCDET_RATE_MASK); + + /* We need the charge pump to power MICBIAS */ + snd_soc_dapm_force_enable_pin(&codec->dapm, "CP2"); + snd_soc_dapm_force_enable_pin(&codec->dapm, "SYSCLK"); + snd_soc_dapm_sync(&codec->dapm); + + /* We start off just enabling microphone detection - even a + * plain headphone will trigger detection. + */ + snd_soc_update_bits(codec, WM5100_MIC_DETECT_1, + WM5100_ACCDET_ENA, WM5100_ACCDET_ENA); + + snd_soc_update_bits(codec, WM5100_INTERRUPT_STATUS_3_MASK, + WM5100_IM_ACCDET_EINT, 0); + } else { + snd_soc_update_bits(codec, WM5100_INTERRUPT_STATUS_3_MASK, + WM5100_IM_HPDET_EINT | + WM5100_IM_ACCDET_EINT, + WM5100_IM_HPDET_EINT | + WM5100_IM_ACCDET_EINT); + snd_soc_update_bits(codec, WM5100_MIC_DETECT_1, + WM5100_ACCDET_ENA, 0); + wm5100->jack = NULL; + } + + return 0; +} + +static irqreturn_t wm5100_irq(int irq, void *data) +{ + struct snd_soc_codec *codec = data; + struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec); + irqreturn_t status = IRQ_NONE; + int irq_val; + + irq_val = snd_soc_read(codec, WM5100_INTERRUPT_STATUS_3); + if (irq_val < 0) { + dev_err(codec->dev, "Failed to read IRQ status 3: %d\n", + irq_val); + irq_val = 0; + } + irq_val &= ~snd_soc_read(codec, WM5100_INTERRUPT_STATUS_3_MASK); + + snd_soc_write(codec, WM5100_INTERRUPT_STATUS_3, irq_val); + + if (irq_val) + status = IRQ_HANDLED; + + wm5100_log_status3(codec, irq_val); + + if (irq_val & WM5100_FLL1_LOCK_EINT) { + dev_dbg(codec->dev, "FLL1 locked\n"); + complete(&wm5100->fll[0].lock); + } + if (irq_val & WM5100_FLL2_LOCK_EINT) { + dev_dbg(codec->dev, "FLL2 locked\n"); + complete(&wm5100->fll[1].lock); + } + + if (irq_val & WM5100_ACCDET_EINT) + wm5100_micd_irq(codec); + + irq_val = snd_soc_read(codec, WM5100_INTERRUPT_STATUS_4); + if (irq_val < 0) { + dev_err(codec->dev, "Failed to read IRQ status 4: %d\n", + irq_val); + irq_val = 0; + } + irq_val &= ~snd_soc_read(codec, WM5100_INTERRUPT_STATUS_4_MASK); + + if (irq_val) + status = IRQ_HANDLED; + + snd_soc_write(codec, WM5100_INTERRUPT_STATUS_4, irq_val); + + wm5100_log_status4(codec, irq_val); + + return status; +} + +static irqreturn_t wm5100_edge_irq(int irq, void *data) +{ + irqreturn_t ret = IRQ_NONE; + irqreturn_t val; + + do { + val = wm5100_irq(irq, data); + if (val != IRQ_NONE) + ret = val; + } while (val != IRQ_NONE); + + return ret; +} + +#ifdef CONFIG_GPIOLIB +static inline struct wm5100_priv *gpio_to_wm5100(struct gpio_chip *chip) +{ + return container_of(chip, struct wm5100_priv, gpio_chip); +} + +static void wm5100_gpio_set(struct gpio_chip *chip, unsigned offset, int value) +{ + struct wm5100_priv *wm5100 = gpio_to_wm5100(chip); + struct snd_soc_codec *codec = wm5100->codec; + + snd_soc_update_bits(codec, WM5100_GPIO_CTRL_1 + offset, + WM5100_GP1_LVL, !!value << WM5100_GP1_LVL_SHIFT); +} + +static int wm5100_gpio_direction_out(struct gpio_chip *chip, + unsigned offset, int value) +{ + struct wm5100_priv *wm5100 = gpio_to_wm5100(chip); + struct snd_soc_codec *codec = wm5100->codec; + int val; + + val = (1 << WM5100_GP1_FN_SHIFT) | (!!value << WM5100_GP1_LVL_SHIFT); + + return snd_soc_update_bits(codec, WM5100_GPIO_CTRL_1 + offset, + WM5100_GP1_FN_MASK | WM5100_GP1_DIR | + WM5100_GP1_LVL, val); +} + +static int wm5100_gpio_get(struct gpio_chip *chip, unsigned offset) +{ + struct wm5100_priv *wm5100 = gpio_to_wm5100(chip); + struct snd_soc_codec *codec = wm5100->codec; + int ret; + + ret = snd_soc_read(codec, WM5100_GPIO_CTRL_1 + offset); + if (ret < 0) + return ret; + + return (ret & WM5100_GP1_LVL) != 0; +} + +static int wm5100_gpio_direction_in(struct gpio_chip *chip, unsigned offset) +{ + struct wm5100_priv *wm5100 = gpio_to_wm5100(chip); + struct snd_soc_codec *codec = wm5100->codec; + + return snd_soc_update_bits(codec, WM5100_GPIO_CTRL_1 + offset, + WM5100_GP1_FN_MASK | WM5100_GP1_DIR, + (1 << WM5100_GP1_FN_SHIFT) | + (1 << WM5100_GP1_DIR_SHIFT)); +} + +static struct gpio_chip wm5100_template_chip = { + .label = "wm5100", + .owner = THIS_MODULE, + .direction_output = wm5100_gpio_direction_out, + .set = wm5100_gpio_set, + .direction_input = wm5100_gpio_direction_in, + .get = wm5100_gpio_get, + .can_sleep = 1, +}; + +static void wm5100_init_gpio(struct snd_soc_codec *codec) +{ + struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec); + int ret; + + wm5100->gpio_chip = wm5100_template_chip; + wm5100->gpio_chip.ngpio = 6; + wm5100->gpio_chip.dev = codec->dev; + + if (wm5100->pdata.gpio_base) + wm5100->gpio_chip.base = wm5100->pdata.gpio_base; + else + wm5100->gpio_chip.base = -1; + + ret = gpiochip_add(&wm5100->gpio_chip); + if (ret != 0) + dev_err(codec->dev, "Failed to add GPIOs: %d\n", ret); +} + +static void wm5100_free_gpio(struct snd_soc_codec *codec) +{ + struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec); + int ret; + + ret = gpiochip_remove(&wm5100->gpio_chip); + if (ret != 0) + dev_err(codec->dev, "Failed to remove GPIOs: %d\n", ret); +} +#else +static void wm5100_init_gpio(struct snd_soc_codec *codec) +{ +} + +static void wm5100_free_gpio(struct snd_soc_codec *codec) +{ +} +#endif + +static int wm5100_probe(struct snd_soc_codec *codec) +{ + struct i2c_client *i2c = to_i2c_client(codec->dev); + struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec); + int ret, i, irq_flags; + + wm5100->codec = codec; + + ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_I2C); + if (ret != 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + for (i = 0; i < ARRAY_SIZE(wm5100->core_supplies); i++) + wm5100->core_supplies[i].supply = wm5100_core_supply_names[i]; + + ret = regulator_bulk_get(&i2c->dev, ARRAY_SIZE(wm5100->core_supplies), + wm5100->core_supplies); + if (ret != 0) { + dev_err(codec->dev, "Failed to request core supplies: %d\n", + ret); + return ret; + } + + wm5100->cpvdd = regulator_get(&i2c->dev, "CPVDD"); + if (IS_ERR(wm5100->cpvdd)) { + ret = PTR_ERR(wm5100->cpvdd); + dev_err(&i2c->dev, "Failed to get CPVDD: %d\n", ret); + goto err_core; + } + + wm5100->dbvdd2 = regulator_get(&i2c->dev, "DBVDD2"); + if (IS_ERR(wm5100->dbvdd2)) { + ret = PTR_ERR(wm5100->dbvdd2); + dev_err(&i2c->dev, "Failed to get DBVDD2: %d\n", ret); + goto err_cpvdd; + } + + wm5100->dbvdd3 = regulator_get(&i2c->dev, "DBVDD3"); + if (IS_ERR(wm5100->dbvdd3)) { + ret = PTR_ERR(wm5100->dbvdd3); + dev_err(&i2c->dev, "Failed to get DBVDD2: %d\n", ret); + goto err_dbvdd2; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(wm5100->core_supplies), + wm5100->core_supplies); + if (ret != 0) { + dev_err(codec->dev, "Failed to enable core supplies: %d\n", + ret); + goto err_dbvdd3; + } + + if (wm5100->pdata.ldo_ena) { + ret = gpio_request_one(wm5100->pdata.ldo_ena, + GPIOF_OUT_INIT_HIGH, "WM5100 LDOENA"); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to request LDOENA %d: %d\n", + wm5100->pdata.ldo_ena, ret); + goto err_enable; + } + msleep(2); + } + + if (wm5100->pdata.reset) { + ret = gpio_request_one(wm5100->pdata.reset, + GPIOF_OUT_INIT_HIGH, "WM5100 /RESET"); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to request /RESET %d: %d\n", + wm5100->pdata.reset, ret); + goto err_ldo; + } + } + + ret = snd_soc_read(codec, WM5100_SOFTWARE_RESET); + if (ret < 0) { + dev_err(codec->dev, "Failed to read ID register\n"); + goto err_reset; + } + switch (ret) { + case 0x8997: + case 0x5100: + break; + + default: + dev_err(codec->dev, "Device is not a WM5100, ID is %x\n", ret); + ret = -EINVAL; + goto err_reset; + } + + ret = snd_soc_read(codec, WM5100_DEVICE_REVISION); + if (ret < 0) { + dev_err(codec->dev, "Failed to read revision register\n"); + goto err_reset; + } + wm5100->rev = ret & WM5100_DEVICE_REVISION_MASK; + + dev_info(codec->dev, "revision %c\n", wm5100->rev + 'A'); + + ret = wm5100_reset(codec); + if (ret < 0) { + dev_err(codec->dev, "Failed to issue reset\n"); + goto err_reset; + } + + codec->cache_only = true; + + wm5100_init_gpio(codec); + + for (i = 0; i < ARRAY_SIZE(wm5100_dig_vu); i++) + snd_soc_update_bits(codec, wm5100_dig_vu[i], WM5100_OUT_VU, + WM5100_OUT_VU); + + for (i = 0; i < ARRAY_SIZE(wm5100->pdata.in_mode); i++) { + snd_soc_update_bits(codec, WM5100_IN1L_CONTROL, + WM5100_IN1_MODE_MASK | + WM5100_IN1_DMIC_SUP_MASK, + (wm5100->pdata.in_mode[i] << + WM5100_IN1_MODE_SHIFT) | + (wm5100->pdata.dmic_sup[i] << + WM5100_IN1_DMIC_SUP_SHIFT)); + } + + for (i = 0; i < ARRAY_SIZE(wm5100->pdata.gpio_defaults); i++) { + if (!wm5100->pdata.gpio_defaults[i]) + continue; + + snd_soc_write(codec, WM5100_GPIO_CTRL_1 + i, + wm5100->pdata.gpio_defaults[i]); + } + + /* Don't debounce interrupts to support use of SYSCLK only */ + snd_soc_write(codec, WM5100_IRQ_DEBOUNCE_1, 0); + snd_soc_write(codec, WM5100_IRQ_DEBOUNCE_2, 0); + + /* TODO: check if we're symmetric */ + + if (i2c->irq) { + if (wm5100->pdata.irq_flags) + irq_flags = wm5100->pdata.irq_flags; + else + irq_flags = IRQF_TRIGGER_LOW; + + irq_flags |= IRQF_ONESHOT; + + if (irq_flags & (IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING)) + ret = request_threaded_irq(i2c->irq, NULL, + wm5100_edge_irq, + irq_flags, "wm5100", codec); + else + ret = request_threaded_irq(i2c->irq, NULL, wm5100_irq, + irq_flags, "wm5100", codec); + + if (ret != 0) { + dev_err(codec->dev, "Failed to request IRQ %d: %d\n", + i2c->irq, ret); + } else { + /* Enable default interrupts */ + snd_soc_update_bits(codec, + WM5100_INTERRUPT_STATUS_3_MASK, + WM5100_IM_SPK_SHUTDOWN_WARN_EINT | + WM5100_IM_SPK_SHUTDOWN_EINT | + WM5100_IM_ASRC2_LOCK_EINT | + WM5100_IM_ASRC1_LOCK_EINT | + WM5100_IM_FLL2_LOCK_EINT | + WM5100_IM_FLL1_LOCK_EINT | + WM5100_CLKGEN_ERR_EINT | + WM5100_CLKGEN_ERR_ASYNC_EINT, 0); + + snd_soc_update_bits(codec, + WM5100_INTERRUPT_STATUS_4_MASK, + WM5100_AIF3_ERR_EINT | + WM5100_AIF2_ERR_EINT | + WM5100_AIF1_ERR_EINT | + WM5100_CTRLIF_ERR_EINT | + WM5100_ISRC2_UNDERCLOCKED_EINT | + WM5100_ISRC1_UNDERCLOCKED_EINT | + WM5100_FX_UNDERCLOCKED_EINT | + WM5100_AIF3_UNDERCLOCKED_EINT | + WM5100_AIF2_UNDERCLOCKED_EINT | + WM5100_AIF1_UNDERCLOCKED_EINT | + WM5100_ASRC_UNDERCLOCKED_EINT | + WM5100_DAC_UNDERCLOCKED_EINT | + WM5100_ADC_UNDERCLOCKED_EINT | + WM5100_MIXER_UNDERCLOCKED_EINT, 0); + } + } else { + snd_soc_dapm_new_controls(&codec->dapm, + wm5100_dapm_widgets_noirq, + ARRAY_SIZE(wm5100_dapm_widgets_noirq)); + } + + if (wm5100->pdata.hp_pol) { + ret = gpio_request_one(wm5100->pdata.hp_pol, + GPIOF_OUT_INIT_HIGH, "WM5100 HP_POL"); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to request HP_POL %d: %d\n", + wm5100->pdata.hp_pol, ret); + goto err_gpio; + } + } + + /* We'll get woken up again when the system has something useful + * for us to do. + */ + if (wm5100->pdata.ldo_ena) + gpio_set_value_cansleep(wm5100->pdata.ldo_ena, 0); + regulator_bulk_disable(ARRAY_SIZE(wm5100->core_supplies), + wm5100->core_supplies); + + return 0; + +err_gpio: + if (i2c->irq) + free_irq(i2c->irq, codec); + wm5100_free_gpio(codec); +err_reset: + if (wm5100->pdata.reset) { + gpio_set_value_cansleep(wm5100->pdata.reset, 1); + gpio_free(wm5100->pdata.reset); + } +err_ldo: + if (wm5100->pdata.ldo_ena) { + gpio_set_value_cansleep(wm5100->pdata.ldo_ena, 0); + gpio_free(wm5100->pdata.ldo_ena); + } +err_enable: + regulator_bulk_disable(ARRAY_SIZE(wm5100->core_supplies), + wm5100->core_supplies); +err_dbvdd3: + regulator_put(wm5100->dbvdd3); +err_dbvdd2: + regulator_put(wm5100->dbvdd2); +err_cpvdd: + regulator_put(wm5100->cpvdd); +err_core: + regulator_bulk_free(ARRAY_SIZE(wm5100->core_supplies), + wm5100->core_supplies); + + return ret; +} + +static int wm5100_remove(struct snd_soc_codec *codec) +{ + struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec); + struct i2c_client *i2c = to_i2c_client(codec->dev); + + wm5100_set_bias_level(codec, SND_SOC_BIAS_OFF); + if (wm5100->pdata.hp_pol) { + gpio_free(wm5100->pdata.hp_pol); + } + if (i2c->irq) + free_irq(i2c->irq, codec); + wm5100_free_gpio(codec); + if (wm5100->pdata.reset) { + gpio_set_value_cansleep(wm5100->pdata.reset, 1); + gpio_free(wm5100->pdata.reset); + } + if (wm5100->pdata.ldo_ena) { + gpio_set_value_cansleep(wm5100->pdata.ldo_ena, 0); + gpio_free(wm5100->pdata.ldo_ena); + } + regulator_put(wm5100->dbvdd3); + regulator_put(wm5100->dbvdd2); + regulator_put(wm5100->cpvdd); + regulator_bulk_free(ARRAY_SIZE(wm5100->core_supplies), + wm5100->core_supplies); + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_wm5100 = { + .probe = wm5100_probe, + .remove = wm5100_remove, + + .set_sysclk = wm5100_set_sysclk, + .set_pll = wm5100_set_fll, + .set_bias_level = wm5100_set_bias_level, + .idle_bias_off = 1, + + .seq_notifier = wm5100_seq_notifier, + .controls = wm5100_snd_controls, + .num_controls = ARRAY_SIZE(wm5100_snd_controls), + .dapm_widgets = wm5100_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm5100_dapm_widgets), + .dapm_routes = wm5100_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(wm5100_dapm_routes), + + .reg_cache_size = ARRAY_SIZE(wm5100_reg_defaults), + .reg_word_size = sizeof(u16), + .compress_type = SND_SOC_RBTREE_COMPRESSION, + .reg_cache_default = wm5100_reg_defaults, + + .volatile_register = wm5100_volatile_register, + .readable_register = wm5100_readable_register, +}; + +static __devinit int wm5100_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct wm5100_pdata *pdata = dev_get_platdata(&i2c->dev); + struct wm5100_priv *wm5100; + int ret, i; + + wm5100 = kzalloc(sizeof(struct wm5100_priv), GFP_KERNEL); + if (wm5100 == NULL) + return -ENOMEM; + + for (i = 0; i < ARRAY_SIZE(wm5100->fll); i++) + init_completion(&wm5100->fll[i].lock); + + if (pdata) + wm5100->pdata = *pdata; + + i2c_set_clientdata(i2c, wm5100); + + ret = snd_soc_register_codec(&i2c->dev, + &soc_codec_dev_wm5100, wm5100_dai, + ARRAY_SIZE(wm5100_dai)); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to register WM5100: %d\n", ret); + kfree(wm5100); + } + + return ret; +} + +static __devexit int wm5100_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + kfree(i2c_get_clientdata(client)); + return 0; +} + +static const struct i2c_device_id wm5100_i2c_id[] = { + { "wm5100", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, wm5100_i2c_id); + +static struct i2c_driver wm5100_i2c_driver = { + .driver = { + .name = "wm5100", + .owner = THIS_MODULE, + }, + .probe = wm5100_i2c_probe, + .remove = __devexit_p(wm5100_i2c_remove), + .id_table = wm5100_i2c_id, +}; + +static int __init wm5100_modinit(void) +{ + return i2c_add_driver(&wm5100_i2c_driver); +} +module_init(wm5100_modinit); + +static void __exit wm5100_exit(void) +{ + i2c_del_driver(&wm5100_i2c_driver); +} +module_exit(wm5100_exit); + +MODULE_DESCRIPTION("ASoC WM5100 driver"); +MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm5100.h b/sound/soc/codecs/wm5100.h new file mode 100644 index 000000000000..970759636bdc --- /dev/null +++ b/sound/soc/codecs/wm5100.h @@ -0,0 +1,5155 @@ +/* + * wm5100.h -- WM5100 ALSA SoC Audio driver + * + * Copyright 2011 Wolfson Microelectronics plc + * + * Author: Mark Brown <broonie@opensource.wolfsonmicro.com> + * + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef WM5100_ASOC_H +#define WM5100_ASOC_H + +#include <sound/soc.h> + +int wm5100_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack); + +#define WM5100_CLK_AIF1 1 +#define WM5100_CLK_AIF2 2 +#define WM5100_CLK_AIF3 3 +#define WM5100_CLK_SYSCLK 4 +#define WM5100_CLK_ASYNCCLK 5 +#define WM5100_CLK_32KHZ 6 +#define WM5100_CLK_OPCLK 7 + +#define WM5100_CLKSRC_MCLK1 0 +#define WM5100_CLKSRC_MCLK2 1 +#define WM5100_CLKSRC_SYSCLK 2 +#define WM5100_CLKSRC_FLL1 4 +#define WM5100_CLKSRC_FLL2 5 +#define WM5100_CLKSRC_AIF1BCLK 8 +#define WM5100_CLKSRC_AIF2BCLK 9 +#define WM5100_CLKSRC_AIF3BCLK 10 +#define WM5100_CLKSRC_ASYNCCLK 0x100 + +#define WM5100_FLL1 1 +#define WM5100_FLL2 2 + +#define WM5100_FLL_SRC_MCLK1 0x0 +#define WM5100_FLL_SRC_MCLK2 0x1 +#define WM5100_FLL_SRC_FLL1 0x4 +#define WM5100_FLL_SRC_FLL2 0x5 +#define WM5100_FLL_SRC_AIF1BCLK 0x8 +#define WM5100_FLL_SRC_AIF2BCLK 0x9 +#define WM5100_FLL_SRC_AIF3BCLK 0xa + +/* + * Register values. + */ +#define WM5100_SOFTWARE_RESET 0x00 +#define WM5100_DEVICE_REVISION 0x01 +#define WM5100_CTRL_IF_1 0x10 +#define WM5100_TONE_GENERATOR_1 0x20 +#define WM5100_PWM_DRIVE_1 0x30 +#define WM5100_PWM_DRIVE_2 0x31 +#define WM5100_PWM_DRIVE_3 0x32 +#define WM5100_CLOCKING_1 0x100 +#define WM5100_CLOCKING_3 0x101 +#define WM5100_CLOCKING_4 0x102 +#define WM5100_CLOCKING_5 0x103 +#define WM5100_CLOCKING_6 0x104 +#define WM5100_CLOCKING_7 0x107 +#define WM5100_CLOCKING_8 0x108 +#define WM5100_ASRC_ENABLE 0x120 +#define WM5100_ASRC_STATUS 0x121 +#define WM5100_ASRC_RATE1 0x122 +#define WM5100_ISRC_1_CTRL_1 0x141 +#define WM5100_ISRC_1_CTRL_2 0x142 +#define WM5100_ISRC_2_CTRL1 0x143 +#define WM5100_ISRC_2_CTRL_2 0x144 +#define WM5100_FLL1_CONTROL_1 0x182 +#define WM5100_FLL1_CONTROL_2 0x183 +#define WM5100_FLL1_CONTROL_3 0x184 +#define WM5100_FLL1_CONTROL_5 0x186 +#define WM5100_FLL1_CONTROL_6 0x187 +#define WM5100_FLL1_EFS_1 0x188 +#define WM5100_FLL2_CONTROL_1 0x1A2 +#define WM5100_FLL2_CONTROL_2 0x1A3 +#define WM5100_FLL2_CONTROL_3 0x1A4 +#define WM5100_FLL2_CONTROL_5 0x1A6 +#define WM5100_FLL2_CONTROL_6 0x1A7 +#define WM5100_FLL2_EFS_1 0x1A8 +#define WM5100_MIC_CHARGE_PUMP_1 0x200 +#define WM5100_MIC_CHARGE_PUMP_2 0x201 +#define WM5100_HP_CHARGE_PUMP_1 0x202 +#define WM5100_LDO1_CONTROL 0x211 +#define WM5100_MIC_BIAS_CTRL_1 0x215 +#define WM5100_MIC_BIAS_CTRL_2 0x216 +#define WM5100_MIC_BIAS_CTRL_3 0x217 +#define WM5100_ACCESSORY_DETECT_MODE_1 0x280 +#define WM5100_HEADPHONE_DETECT_1 0x288 +#define WM5100_HEADPHONE_DETECT_2 0x289 +#define WM5100_MIC_DETECT_1 0x290 +#define WM5100_MIC_DETECT_2 0x291 +#define WM5100_MIC_DETECT_3 0x292 +#define WM5100_MISC_CONTROL 0x2BB +#define WM5100_INPUT_ENABLES 0x301 +#define WM5100_INPUT_ENABLES_STATUS 0x302 +#define WM5100_IN1L_CONTROL 0x310 +#define WM5100_IN1R_CONTROL 0x311 +#define WM5100_IN2L_CONTROL 0x312 +#define WM5100_IN2R_CONTROL 0x313 +#define WM5100_IN3L_CONTROL 0x314 +#define WM5100_IN3R_CONTROL 0x315 +#define WM5100_IN4L_CONTROL 0x316 +#define WM5100_IN4R_CONTROL 0x317 +#define WM5100_RXANC_SRC 0x318 +#define WM5100_INPUT_VOLUME_RAMP 0x319 +#define WM5100_ADC_DIGITAL_VOLUME_1L 0x320 +#define WM5100_ADC_DIGITAL_VOLUME_1R 0x321 +#define WM5100_ADC_DIGITAL_VOLUME_2L 0x322 +#define WM5100_ADC_DIGITAL_VOLUME_2R 0x323 +#define WM5100_ADC_DIGITAL_VOLUME_3L 0x324 +#define WM5100_ADC_DIGITAL_VOLUME_3R 0x325 +#define WM5100_ADC_DIGITAL_VOLUME_4L 0x326 +#define WM5100_ADC_DIGITAL_VOLUME_4R 0x327 +#define WM5100_OUTPUT_ENABLES_2 0x401 +#define WM5100_OUTPUT_STATUS_1 0x402 +#define WM5100_OUTPUT_STATUS_2 0x403 +#define WM5100_CHANNEL_ENABLES_1 0x408 +#define WM5100_OUT_VOLUME_1L 0x410 +#define WM5100_OUT_VOLUME_1R 0x411 +#define WM5100_DAC_VOLUME_LIMIT_1L 0x412 +#define WM5100_DAC_VOLUME_LIMIT_1R 0x413 +#define WM5100_OUT_VOLUME_2L 0x414 +#define WM5100_OUT_VOLUME_2R 0x415 +#define WM5100_DAC_VOLUME_LIMIT_2L 0x416 +#define WM5100_DAC_VOLUME_LIMIT_2R 0x417 +#define WM5100_OUT_VOLUME_3L 0x418 +#define WM5100_OUT_VOLUME_3R 0x419 +#define WM5100_DAC_VOLUME_LIMIT_3L 0x41A +#define WM5100_DAC_VOLUME_LIMIT_3R 0x41B +#define WM5100_OUT_VOLUME_4L 0x41C +#define WM5100_OUT_VOLUME_4R 0x41D +#define WM5100_DAC_VOLUME_LIMIT_5L 0x41E +#define WM5100_DAC_VOLUME_LIMIT_5R 0x41F +#define WM5100_DAC_VOLUME_LIMIT_6L 0x420 +#define WM5100_DAC_VOLUME_LIMIT_6R 0x421 +#define WM5100_DAC_AEC_CONTROL_1 0x440 +#define WM5100_OUTPUT_VOLUME_RAMP 0x441 +#define WM5100_DAC_DIGITAL_VOLUME_1L 0x480 +#define WM5100_DAC_DIGITAL_VOLUME_1R 0x481 +#define WM5100_DAC_DIGITAL_VOLUME_2L 0x482 +#define WM5100_DAC_DIGITAL_VOLUME_2R 0x483 +#define WM5100_DAC_DIGITAL_VOLUME_3L 0x484 +#define WM5100_DAC_DIGITAL_VOLUME_3R 0x485 +#define WM5100_DAC_DIGITAL_VOLUME_4L 0x486 +#define WM5100_DAC_DIGITAL_VOLUME_4R 0x487 +#define WM5100_DAC_DIGITAL_VOLUME_5L 0x488 +#define WM5100_DAC_DIGITAL_VOLUME_5R 0x489 +#define WM5100_DAC_DIGITAL_VOLUME_6L 0x48A +#define WM5100_DAC_DIGITAL_VOLUME_6R 0x48B +#define WM5100_PDM_SPK1_CTRL_1 0x4C0 +#define WM5100_PDM_SPK1_CTRL_2 0x4C1 +#define WM5100_PDM_SPK2_CTRL_1 0x4C2 +#define WM5100_PDM_SPK2_CTRL_2 0x4C3 +#define WM5100_AUDIO_IF_1_1 0x500 +#define WM5100_AUDIO_IF_1_2 0x501 +#define WM5100_AUDIO_IF_1_3 0x502 +#define WM5100_AUDIO_IF_1_4 0x503 +#define WM5100_AUDIO_IF_1_5 0x504 +#define WM5100_AUDIO_IF_1_6 0x505 +#define WM5100_AUDIO_IF_1_7 0x506 +#define WM5100_AUDIO_IF_1_8 0x507 +#define WM5100_AUDIO_IF_1_9 0x508 +#define WM5100_AUDIO_IF_1_10 0x509 +#define WM5100_AUDIO_IF_1_11 0x50A +#define WM5100_AUDIO_IF_1_12 0x50B +#define WM5100_AUDIO_IF_1_13 0x50C +#define WM5100_AUDIO_IF_1_14 0x50D +#define WM5100_AUDIO_IF_1_15 0x50E +#define WM5100_AUDIO_IF_1_16 0x50F +#define WM5100_AUDIO_IF_1_17 0x510 +#define WM5100_AUDIO_IF_1_18 0x511 +#define WM5100_AUDIO_IF_1_19 0x512 +#define WM5100_AUDIO_IF_1_20 0x513 +#define WM5100_AUDIO_IF_1_21 0x514 +#define WM5100_AUDIO_IF_1_22 0x515 +#define WM5100_AUDIO_IF_1_23 0x516 +#define WM5100_AUDIO_IF_1_24 0x517 +#define WM5100_AUDIO_IF_1_25 0x518 +#define WM5100_AUDIO_IF_1_26 0x519 +#define WM5100_AUDIO_IF_1_27 0x51A +#define WM5100_AUDIO_IF_2_1 0x540 +#define WM5100_AUDIO_IF_2_2 0x541 +#define WM5100_AUDIO_IF_2_3 0x542 +#define WM5100_AUDIO_IF_2_4 0x543 +#define WM5100_AUDIO_IF_2_5 0x544 +#define WM5100_AUDIO_IF_2_6 0x545 +#define WM5100_AUDIO_IF_2_7 0x546 +#define WM5100_AUDIO_IF_2_8 0x547 +#define WM5100_AUDIO_IF_2_9 0x548 +#define WM5100_AUDIO_IF_2_10 0x549 +#define WM5100_AUDIO_IF_2_11 0x54A +#define WM5100_AUDIO_IF_2_18 0x551 +#define WM5100_AUDIO_IF_2_19 0x552 +#define WM5100_AUDIO_IF_2_26 0x559 +#define WM5100_AUDIO_IF_2_27 0x55A +#define WM5100_AUDIO_IF_3_1 0x580 +#define WM5100_AUDIO_IF_3_2 0x581 +#define WM5100_AUDIO_IF_3_3 0x582 +#define WM5100_AUDIO_IF_3_4 0x583 +#define WM5100_AUDIO_IF_3_5 0x584 +#define WM5100_AUDIO_IF_3_6 0x585 +#define WM5100_AUDIO_IF_3_7 0x586 +#define WM5100_AUDIO_IF_3_8 0x587 +#define WM5100_AUDIO_IF_3_9 0x588 +#define WM5100_AUDIO_IF_3_10 0x589 +#define WM5100_AUDIO_IF_3_11 0x58A +#define WM5100_AUDIO_IF_3_18 0x591 +#define WM5100_AUDIO_IF_3_19 0x592 +#define WM5100_AUDIO_IF_3_26 0x599 +#define WM5100_AUDIO_IF_3_27 0x59A +#define WM5100_PWM1MIX_INPUT_1_SOURCE 0x640 +#define WM5100_PWM1MIX_INPUT_1_VOLUME 0x641 +#define WM5100_PWM1MIX_INPUT_2_SOURCE 0x642 +#define WM5100_PWM1MIX_INPUT_2_VOLUME 0x643 +#define WM5100_PWM1MIX_INPUT_3_SOURCE 0x644 +#define WM5100_PWM1MIX_INPUT_3_VOLUME 0x645 +#define WM5100_PWM1MIX_INPUT_4_SOURCE 0x646 +#define WM5100_PWM1MIX_INPUT_4_VOLUME 0x647 +#define WM5100_PWM2MIX_INPUT_1_SOURCE 0x648 +#define WM5100_PWM2MIX_INPUT_1_VOLUME 0x649 +#define WM5100_PWM2MIX_INPUT_2_SOURCE 0x64A +#define WM5100_PWM2MIX_INPUT_2_VOLUME 0x64B +#define WM5100_PWM2MIX_INPUT_3_SOURCE 0x64C +#define WM5100_PWM2MIX_INPUT_3_VOLUME 0x64D +#define WM5100_PWM2MIX_INPUT_4_SOURCE 0x64E +#define WM5100_PWM2MIX_INPUT_4_VOLUME 0x64F +#define WM5100_OUT1LMIX_INPUT_1_SOURCE 0x680 +#define WM5100_OUT1LMIX_INPUT_1_VOLUME 0x681 +#define WM5100_OUT1LMIX_INPUT_2_SOURCE 0x682 +#define WM5100_OUT1LMIX_INPUT_2_VOLUME 0x683 +#define WM5100_OUT1LMIX_INPUT_3_SOURCE 0x684 +#define WM5100_OUT1LMIX_INPUT_3_VOLUME 0x685 +#define WM5100_OUT1LMIX_INPUT_4_SOURCE 0x686 +#define WM5100_OUT1LMIX_INPUT_4_VOLUME 0x687 +#define WM5100_OUT1RMIX_INPUT_1_SOURCE 0x688 +#define WM5100_OUT1RMIX_INPUT_1_VOLUME 0x689 +#define WM5100_OUT1RMIX_INPUT_2_SOURCE 0x68A +#define WM5100_OUT1RMIX_INPUT_2_VOLUME 0x68B +#define WM5100_OUT1RMIX_INPUT_3_SOURCE 0x68C +#define WM5100_OUT1RMIX_INPUT_3_VOLUME 0x68D +#define WM5100_OUT1RMIX_INPUT_4_SOURCE 0x68E +#define WM5100_OUT1RMIX_INPUT_4_VOLUME 0x68F +#define WM5100_OUT2LMIX_INPUT_1_SOURCE 0x690 +#define WM5100_OUT2LMIX_INPUT_1_VOLUME 0x691 +#define WM5100_OUT2LMIX_INPUT_2_SOURCE 0x692 +#define WM5100_OUT2LMIX_INPUT_2_VOLUME 0x693 +#define WM5100_OUT2LMIX_INPUT_3_SOURCE 0x694 +#define WM5100_OUT2LMIX_INPUT_3_VOLUME 0x695 +#define WM5100_OUT2LMIX_INPUT_4_SOURCE 0x696 +#define WM5100_OUT2LMIX_INPUT_4_VOLUME 0x697 +#define WM5100_OUT2RMIX_INPUT_1_SOURCE 0x698 +#define WM5100_OUT2RMIX_INPUT_1_VOLUME 0x699 +#define WM5100_OUT2RMIX_INPUT_2_SOURCE 0x69A +#define WM5100_OUT2RMIX_INPUT_2_VOLUME 0x69B +#define WM5100_OUT2RMIX_INPUT_3_SOURCE 0x69C +#define WM5100_OUT2RMIX_INPUT_3_VOLUME 0x69D +#define WM5100_OUT2RMIX_INPUT_4_SOURCE 0x69E +#define WM5100_OUT2RMIX_INPUT_4_VOLUME 0x69F +#define WM5100_OUT3LMIX_INPUT_1_SOURCE 0x6A0 +#define WM5100_OUT3LMIX_INPUT_1_VOLUME 0x6A1 +#define WM5100_OUT3LMIX_INPUT_2_SOURCE 0x6A2 +#define WM5100_OUT3LMIX_INPUT_2_VOLUME 0x6A3 +#define WM5100_OUT3LMIX_INPUT_3_SOURCE 0x6A4 +#define WM5100_OUT3LMIX_INPUT_3_VOLUME 0x6A5 +#define WM5100_OUT3LMIX_INPUT_4_SOURCE 0x6A6 +#define WM5100_OUT3LMIX_INPUT_4_VOLUME 0x6A7 +#define WM5100_OUT3RMIX_INPUT_1_SOURCE 0x6A8 +#define WM5100_OUT3RMIX_INPUT_1_VOLUME 0x6A9 +#define WM5100_OUT3RMIX_INPUT_2_SOURCE 0x6AA +#define WM5100_OUT3RMIX_INPUT_2_VOLUME 0x6AB +#define WM5100_OUT3RMIX_INPUT_3_SOURCE 0x6AC +#define WM5100_OUT3RMIX_INPUT_3_VOLUME 0x6AD +#define WM5100_OUT3RMIX_INPUT_4_SOURCE 0x6AE +#define WM5100_OUT3RMIX_INPUT_4_VOLUME 0x6AF +#define WM5100_OUT4LMIX_INPUT_1_SOURCE 0x6B0 +#define WM5100_OUT4LMIX_INPUT_1_VOLUME 0x6B1 +#define WM5100_OUT4LMIX_INPUT_2_SOURCE 0x6B2 +#define WM5100_OUT4LMIX_INPUT_2_VOLUME 0x6B3 +#define WM5100_OUT4LMIX_INPUT_3_SOURCE 0x6B4 +#define WM5100_OUT4LMIX_INPUT_3_VOLUME 0x6B5 +#define WM5100_OUT4LMIX_INPUT_4_SOURCE 0x6B6 +#define WM5100_OUT4LMIX_INPUT_4_VOLUME 0x6B7 +#define WM5100_OUT4RMIX_INPUT_1_SOURCE 0x6B8 +#define WM5100_OUT4RMIX_INPUT_1_VOLUME 0x6B9 +#define WM5100_OUT4RMIX_INPUT_2_SOURCE 0x6BA +#define WM5100_OUT4RMIX_INPUT_2_VOLUME 0x6BB +#define WM5100_OUT4RMIX_INPUT_3_SOURCE 0x6BC +#define WM5100_OUT4RMIX_INPUT_3_VOLUME 0x6BD +#define WM5100_OUT4RMIX_INPUT_4_SOURCE 0x6BE +#define WM5100_OUT4RMIX_INPUT_4_VOLUME 0x6BF +#define WM5100_OUT5LMIX_INPUT_1_SOURCE 0x6C0 +#define WM5100_OUT5LMIX_INPUT_1_VOLUME 0x6C1 +#define WM5100_OUT5LMIX_INPUT_2_SOURCE 0x6C2 +#define WM5100_OUT5LMIX_INPUT_2_VOLUME 0x6C3 +#define WM5100_OUT5LMIX_INPUT_3_SOURCE 0x6C4 +#define WM5100_OUT5LMIX_INPUT_3_VOLUME 0x6C5 +#define WM5100_OUT5LMIX_INPUT_4_SOURCE 0x6C6 +#define WM5100_OUT5LMIX_INPUT_4_VOLUME 0x6C7 +#define WM5100_OUT5RMIX_INPUT_1_SOURCE 0x6C8 +#define WM5100_OUT5RMIX_INPUT_1_VOLUME 0x6C9 +#define WM5100_OUT5RMIX_INPUT_2_SOURCE 0x6CA +#define WM5100_OUT5RMIX_INPUT_2_VOLUME 0x6CB +#define WM5100_OUT5RMIX_INPUT_3_SOURCE 0x6CC +#define WM5100_OUT5RMIX_INPUT_3_VOLUME 0x6CD +#define WM5100_OUT5RMIX_INPUT_4_SOURCE 0x6CE +#define WM5100_OUT5RMIX_INPUT_4_VOLUME 0x6CF +#define WM5100_OUT6LMIX_INPUT_1_SOURCE 0x6D0 +#define WM5100_OUT6LMIX_INPUT_1_VOLUME 0x6D1 +#define WM5100_OUT6LMIX_INPUT_2_SOURCE 0x6D2 +#define WM5100_OUT6LMIX_INPUT_2_VOLUME 0x6D3 +#define WM5100_OUT6LMIX_INPUT_3_SOURCE 0x6D4 +#define WM5100_OUT6LMIX_INPUT_3_VOLUME 0x6D5 +#define WM5100_OUT6LMIX_INPUT_4_SOURCE 0x6D6 +#define WM5100_OUT6LMIX_INPUT_4_VOLUME 0x6D7 +#define WM5100_OUT6RMIX_INPUT_1_SOURCE 0x6D8 +#define WM5100_OUT6RMIX_INPUT_1_VOLUME 0x6D9 +#define WM5100_OUT6RMIX_INPUT_2_SOURCE 0x6DA +#define WM5100_OUT6RMIX_INPUT_2_VOLUME 0x6DB +#define WM5100_OUT6RMIX_INPUT_3_SOURCE 0x6DC +#define WM5100_OUT6RMIX_INPUT_3_VOLUME 0x6DD +#define WM5100_OUT6RMIX_INPUT_4_SOURCE 0x6DE +#define WM5100_OUT6RMIX_INPUT_4_VOLUME 0x6DF +#define WM5100_AIF1TX1MIX_INPUT_1_SOURCE 0x700 +#define WM5100_AIF1TX1MIX_INPUT_1_VOLUME 0x701 +#define WM5100_AIF1TX1MIX_INPUT_2_SOURCE 0x702 +#define WM5100_AIF1TX1MIX_INPUT_2_VOLUME 0x703 +#define WM5100_AIF1TX1MIX_INPUT_3_SOURCE 0x704 +#define WM5100_AIF1TX1MIX_INPUT_3_VOLUME 0x705 +#define WM5100_AIF1TX1MIX_INPUT_4_SOURCE 0x706 +#define WM5100_AIF1TX1MIX_INPUT_4_VOLUME 0x707 +#define WM5100_AIF1TX2MIX_INPUT_1_SOURCE 0x708 +#define WM5100_AIF1TX2MIX_INPUT_1_VOLUME 0x709 +#define WM5100_AIF1TX2MIX_INPUT_2_SOURCE 0x70A +#define WM5100_AIF1TX2MIX_INPUT_2_VOLUME 0x70B +#define WM5100_AIF1TX2MIX_INPUT_3_SOURCE 0x70C +#define WM5100_AIF1TX2MIX_INPUT_3_VOLUME 0x70D +#define WM5100_AIF1TX2MIX_INPUT_4_SOURCE 0x70E +#define WM5100_AIF1TX2MIX_INPUT_4_VOLUME 0x70F +#define WM5100_AIF1TX3MIX_INPUT_1_SOURCE 0x710 +#define WM5100_AIF1TX3MIX_INPUT_1_VOLUME 0x711 +#define WM5100_AIF1TX3MIX_INPUT_2_SOURCE 0x712 +#define WM5100_AIF1TX3MIX_INPUT_2_VOLUME 0x713 +#define WM5100_AIF1TX3MIX_INPUT_3_SOURCE 0x714 +#define WM5100_AIF1TX3MIX_INPUT_3_VOLUME 0x715 +#define WM5100_AIF1TX3MIX_INPUT_4_SOURCE 0x716 +#define WM5100_AIF1TX3MIX_INPUT_4_VOLUME 0x717 +#define WM5100_AIF1TX4MIX_INPUT_1_SOURCE 0x718 +#define WM5100_AIF1TX4MIX_INPUT_1_VOLUME 0x719 +#define WM5100_AIF1TX4MIX_INPUT_2_SOURCE 0x71A +#define WM5100_AIF1TX4MIX_INPUT_2_VOLUME 0x71B +#define WM5100_AIF1TX4MIX_INPUT_3_SOURCE 0x71C +#define WM5100_AIF1TX4MIX_INPUT_3_VOLUME 0x71D +#define WM5100_AIF1TX4MIX_INPUT_4_SOURCE 0x71E +#define WM5100_AIF1TX4MIX_INPUT_4_VOLUME 0x71F +#define WM5100_AIF1TX5MIX_INPUT_1_SOURCE 0x720 +#define WM5100_AIF1TX5MIX_INPUT_1_VOLUME 0x721 +#define WM5100_AIF1TX5MIX_INPUT_2_SOURCE 0x722 +#define WM5100_AIF1TX5MIX_INPUT_2_VOLUME 0x723 +#define WM5100_AIF1TX5MIX_INPUT_3_SOURCE 0x724 +#define WM5100_AIF1TX5MIX_INPUT_3_VOLUME 0x725 +#define WM5100_AIF1TX5MIX_INPUT_4_SOURCE 0x726 +#define WM5100_AIF1TX5MIX_INPUT_4_VOLUME 0x727 +#define WM5100_AIF1TX6MIX_INPUT_1_SOURCE 0x728 +#define WM5100_AIF1TX6MIX_INPUT_1_VOLUME 0x729 +#define WM5100_AIF1TX6MIX_INPUT_2_SOURCE 0x72A +#define WM5100_AIF1TX6MIX_INPUT_2_VOLUME 0x72B +#define WM5100_AIF1TX6MIX_INPUT_3_SOURCE 0x72C +#define WM5100_AIF1TX6MIX_INPUT_3_VOLUME 0x72D +#define WM5100_AIF1TX6MIX_INPUT_4_SOURCE 0x72E +#define WM5100_AIF1TX6MIX_INPUT_4_VOLUME 0x72F +#define WM5100_AIF1TX7MIX_INPUT_1_SOURCE 0x730 +#define WM5100_AIF1TX7MIX_INPUT_1_VOLUME 0x731 +#define WM5100_AIF1TX7MIX_INPUT_2_SOURCE 0x732 +#define WM5100_AIF1TX7MIX_INPUT_2_VOLUME 0x733 +#define WM5100_AIF1TX7MIX_INPUT_3_SOURCE 0x734 +#define WM5100_AIF1TX7MIX_INPUT_3_VOLUME 0x735 +#define WM5100_AIF1TX7MIX_INPUT_4_SOURCE 0x736 +#define WM5100_AIF1TX7MIX_INPUT_4_VOLUME 0x737 +#define WM5100_AIF1TX8MIX_INPUT_1_SOURCE 0x738 +#define WM5100_AIF1TX8MIX_INPUT_1_VOLUME 0x739 +#define WM5100_AIF1TX8MIX_INPUT_2_SOURCE 0x73A +#define WM5100_AIF1TX8MIX_INPUT_2_VOLUME 0x73B +#define WM5100_AIF1TX8MIX_INPUT_3_SOURCE 0x73C +#define WM5100_AIF1TX8MIX_INPUT_3_VOLUME 0x73D +#define WM5100_AIF1TX8MIX_INPUT_4_SOURCE 0x73E +#define WM5100_AIF1TX8MIX_INPUT_4_VOLUME 0x73F +#define WM5100_AIF2TX1MIX_INPUT_1_SOURCE 0x740 +#define WM5100_AIF2TX1MIX_INPUT_1_VOLUME 0x741 +#define WM5100_AIF2TX1MIX_INPUT_2_SOURCE 0x742 +#define WM5100_AIF2TX1MIX_INPUT_2_VOLUME 0x743 +#define WM5100_AIF2TX1MIX_INPUT_3_SOURCE 0x744 +#define WM5100_AIF2TX1MIX_INPUT_3_VOLUME 0x745 +#define WM5100_AIF2TX1MIX_INPUT_4_SOURCE 0x746 +#define WM5100_AIF2TX1MIX_INPUT_4_VOLUME 0x747 +#define WM5100_AIF2TX2MIX_INPUT_1_SOURCE 0x748 +#define WM5100_AIF2TX2MIX_INPUT_1_VOLUME 0x749 +#define WM5100_AIF2TX2MIX_INPUT_2_SOURCE 0x74A +#define WM5100_AIF2TX2MIX_INPUT_2_VOLUME 0x74B +#define WM5100_AIF2TX2MIX_INPUT_3_SOURCE 0x74C +#define WM5100_AIF2TX2MIX_INPUT_3_VOLUME 0x74D +#define WM5100_AIF2TX2MIX_INPUT_4_SOURCE 0x74E +#define WM5100_AIF2TX2MIX_INPUT_4_VOLUME 0x74F +#define WM5100_AIF3TX1MIX_INPUT_1_SOURCE 0x780 +#define WM5100_AIF3TX1MIX_INPUT_1_VOLUME 0x781 +#define WM5100_AIF3TX1MIX_INPUT_2_SOURCE 0x782 +#define WM5100_AIF3TX1MIX_INPUT_2_VOLUME 0x783 +#define WM5100_AIF3TX1MIX_INPUT_3_SOURCE 0x784 +#define WM5100_AIF3TX1MIX_INPUT_3_VOLUME 0x785 +#define WM5100_AIF3TX1MIX_INPUT_4_SOURCE 0x786 +#define WM5100_AIF3TX1MIX_INPUT_4_VOLUME 0x787 +#define WM5100_AIF3TX2MIX_INPUT_1_SOURCE 0x788 +#define WM5100_AIF3TX2MIX_INPUT_1_VOLUME 0x789 +#define WM5100_AIF3TX2MIX_INPUT_2_SOURCE 0x78A +#define WM5100_AIF3TX2MIX_INPUT_2_VOLUME 0x78B +#define WM5100_AIF3TX2MIX_INPUT_3_SOURCE 0x78C +#define WM5100_AIF3TX2MIX_INPUT_3_VOLUME 0x78D +#define WM5100_AIF3TX2MIX_INPUT_4_SOURCE 0x78E +#define WM5100_AIF3TX2MIX_INPUT_4_VOLUME 0x78F +#define WM5100_EQ1MIX_INPUT_1_SOURCE 0x880 +#define WM5100_EQ1MIX_INPUT_1_VOLUME 0x881 +#define WM5100_EQ1MIX_INPUT_2_SOURCE 0x882 +#define WM5100_EQ1MIX_INPUT_2_VOLUME 0x883 +#define WM5100_EQ1MIX_INPUT_3_SOURCE 0x884 +#define WM5100_EQ1MIX_INPUT_3_VOLUME 0x885 +#define WM5100_EQ1MIX_INPUT_4_SOURCE 0x886 +#define WM5100_EQ1MIX_INPUT_4_VOLUME 0x887 +#define WM5100_EQ2MIX_INPUT_1_SOURCE 0x888 +#define WM5100_EQ2MIX_INPUT_1_VOLUME 0x889 +#define WM5100_EQ2MIX_INPUT_2_SOURCE 0x88A +#define WM5100_EQ2MIX_INPUT_2_VOLUME 0x88B +#define WM5100_EQ2MIX_INPUT_3_SOURCE 0x88C +#define WM5100_EQ2MIX_INPUT_3_VOLUME 0x88D +#define WM5100_EQ2MIX_INPUT_4_SOURCE 0x88E +#define WM5100_EQ2MIX_INPUT_4_VOLUME 0x88F +#define WM5100_EQ3MIX_INPUT_1_SOURCE 0x890 +#define WM5100_EQ3MIX_INPUT_1_VOLUME 0x891 +#define WM5100_EQ3MIX_INPUT_2_SOURCE 0x892 +#define WM5100_EQ3MIX_INPUT_2_VOLUME 0x893 +#define WM5100_EQ3MIX_INPUT_3_SOURCE 0x894 +#define WM5100_EQ3MIX_INPUT_3_VOLUME 0x895 +#define WM5100_EQ3MIX_INPUT_4_SOURCE 0x896 +#define WM5100_EQ3MIX_INPUT_4_VOLUME 0x897 +#define WM5100_EQ4MIX_INPUT_1_SOURCE 0x898 +#define WM5100_EQ4MIX_INPUT_1_VOLUME 0x899 +#define WM5100_EQ4MIX_INPUT_2_SOURCE 0x89A +#define WM5100_EQ4MIX_INPUT_2_VOLUME 0x89B +#define WM5100_EQ4MIX_INPUT_3_SOURCE 0x89C +#define WM5100_EQ4MIX_INPUT_3_VOLUME 0x89D +#define WM5100_EQ4MIX_INPUT_4_SOURCE 0x89E +#define WM5100_EQ4MIX_INPUT_4_VOLUME 0x89F +#define WM5100_DRC1LMIX_INPUT_1_SOURCE 0x8C0 +#define WM5100_DRC1LMIX_INPUT_1_VOLUME 0x8C1 +#define WM5100_DRC1LMIX_INPUT_2_SOURCE 0x8C2 +#define WM5100_DRC1LMIX_INPUT_2_VOLUME 0x8C3 +#define WM5100_DRC1LMIX_INPUT_3_SOURCE 0x8C4 +#define WM5100_DRC1LMIX_INPUT_3_VOLUME 0x8C5 +#define WM5100_DRC1LMIX_INPUT_4_SOURCE 0x8C6 +#define WM5100_DRC1LMIX_INPUT_4_VOLUME 0x8C7 +#define WM5100_DRC1RMIX_INPUT_1_SOURCE 0x8C8 +#define WM5100_DRC1RMIX_INPUT_1_VOLUME 0x8C9 +#define WM5100_DRC1RMIX_INPUT_2_SOURCE 0x8CA +#define WM5100_DRC1RMIX_INPUT_2_VOLUME 0x8CB +#define WM5100_DRC1RMIX_INPUT_3_SOURCE 0x8CC +#define WM5100_DRC1RMIX_INPUT_3_VOLUME 0x8CD +#define WM5100_DRC1RMIX_INPUT_4_SOURCE 0x8CE +#define WM5100_DRC1RMIX_INPUT_4_VOLUME 0x8CF +#define WM5100_HPLP1MIX_INPUT_1_SOURCE 0x900 +#define WM5100_HPLP1MIX_INPUT_1_VOLUME 0x901 +#define WM5100_HPLP1MIX_INPUT_2_SOURCE 0x902 +#define WM5100_HPLP1MIX_INPUT_2_VOLUME 0x903 +#define WM5100_HPLP1MIX_INPUT_3_SOURCE 0x904 +#define WM5100_HPLP1MIX_INPUT_3_VOLUME 0x905 +#define WM5100_HPLP1MIX_INPUT_4_SOURCE 0x906 +#define WM5100_HPLP1MIX_INPUT_4_VOLUME 0x907 +#define WM5100_HPLP2MIX_INPUT_1_SOURCE 0x908 +#define WM5100_HPLP2MIX_INPUT_1_VOLUME 0x909 +#define WM5100_HPLP2MIX_INPUT_2_SOURCE 0x90A +#define WM5100_HPLP2MIX_INPUT_2_VOLUME 0x90B +#define WM5100_HPLP2MIX_INPUT_3_SOURCE 0x90C +#define WM5100_HPLP2MIX_INPUT_3_VOLUME 0x90D +#define WM5100_HPLP2MIX_INPUT_4_SOURCE 0x90E +#define WM5100_HPLP2MIX_INPUT_4_VOLUME 0x90F +#define WM5100_HPLP3MIX_INPUT_1_SOURCE 0x910 +#define WM5100_HPLP3MIX_INPUT_1_VOLUME 0x911 +#define WM5100_HPLP3MIX_INPUT_2_SOURCE 0x912 +#define WM5100_HPLP3MIX_INPUT_2_VOLUME 0x913 +#define WM5100_HPLP3MIX_INPUT_3_SOURCE 0x914 +#define WM5100_HPLP3MIX_INPUT_3_VOLUME 0x915 +#define WM5100_HPLP3MIX_INPUT_4_SOURCE 0x916 +#define WM5100_HPLP3MIX_INPUT_4_VOLUME 0x917 +#define WM5100_HPLP4MIX_INPUT_1_SOURCE 0x918 +#define WM5100_HPLP4MIX_INPUT_1_VOLUME 0x919 +#define WM5100_HPLP4MIX_INPUT_2_SOURCE 0x91A +#define WM5100_HPLP4MIX_INPUT_2_VOLUME 0x91B +#define WM5100_HPLP4MIX_INPUT_3_SOURCE 0x91C +#define WM5100_HPLP4MIX_INPUT_3_VOLUME 0x91D +#define WM5100_HPLP4MIX_INPUT_4_SOURCE 0x91E +#define WM5100_HPLP4MIX_INPUT_4_VOLUME 0x91F +#define WM5100_DSP1LMIX_INPUT_1_SOURCE 0x940 +#define WM5100_DSP1LMIX_INPUT_1_VOLUME 0x941 +#define WM5100_DSP1LMIX_INPUT_2_SOURCE 0x942 +#define WM5100_DSP1LMIX_INPUT_2_VOLUME 0x943 +#define WM5100_DSP1LMIX_INPUT_3_SOURCE 0x944 +#define WM5100_DSP1LMIX_INPUT_3_VOLUME 0x945 +#define WM5100_DSP1LMIX_INPUT_4_SOURCE 0x946 +#define WM5100_DSP1LMIX_INPUT_4_VOLUME 0x947 +#define WM5100_DSP1RMIX_INPUT_1_SOURCE 0x948 +#define WM5100_DSP1RMIX_INPUT_1_VOLUME 0x949 +#define WM5100_DSP1RMIX_INPUT_2_SOURCE 0x94A +#define WM5100_DSP1RMIX_INPUT_2_VOLUME 0x94B +#define WM5100_DSP1RMIX_INPUT_3_SOURCE 0x94C +#define WM5100_DSP1RMIX_INPUT_3_VOLUME 0x94D +#define WM5100_DSP1RMIX_INPUT_4_SOURCE 0x94E +#define WM5100_DSP1RMIX_INPUT_4_VOLUME 0x94F +#define WM5100_DSP1AUX1MIX_INPUT_1_SOURCE 0x950 +#define WM5100_DSP1AUX2MIX_INPUT_1_SOURCE 0x958 +#define WM5100_DSP1AUX3MIX_INPUT_1_SOURCE 0x960 +#define WM5100_DSP1AUX4MIX_INPUT_1_SOURCE 0x968 +#define WM5100_DSP1AUX5MIX_INPUT_1_SOURCE 0x970 +#define WM5100_DSP1AUX6MIX_INPUT_1_SOURCE 0x978 +#define WM5100_DSP2LMIX_INPUT_1_SOURCE 0x980 +#define WM5100_DSP2LMIX_INPUT_1_VOLUME 0x981 +#define WM5100_DSP2LMIX_INPUT_2_SOURCE 0x982 +#define WM5100_DSP2LMIX_INPUT_2_VOLUME 0x983 +#define WM5100_DSP2LMIX_INPUT_3_SOURCE 0x984 +#define WM5100_DSP2LMIX_INPUT_3_VOLUME 0x985 +#define WM5100_DSP2LMIX_INPUT_4_SOURCE 0x986 +#define WM5100_DSP2LMIX_INPUT_4_VOLUME 0x987 +#define WM5100_DSP2RMIX_INPUT_1_SOURCE 0x988 +#define WM5100_DSP2RMIX_INPUT_1_VOLUME 0x989 +#define WM5100_DSP2RMIX_INPUT_2_SOURCE 0x98A +#define WM5100_DSP2RMIX_INPUT_2_VOLUME 0x98B +#define WM5100_DSP2RMIX_INPUT_3_SOURCE 0x98C +#define WM5100_DSP2RMIX_INPUT_3_VOLUME 0x98D +#define WM5100_DSP2RMIX_INPUT_4_SOURCE 0x98E +#define WM5100_DSP2RMIX_INPUT_4_VOLUME 0x98F +#define WM5100_DSP2AUX1MIX_INPUT_1_SOURCE 0x990 +#define WM5100_DSP2AUX2MIX_INPUT_1_SOURCE 0x998 +#define WM5100_DSP2AUX3MIX_INPUT_1_SOURCE 0x9A0 +#define WM5100_DSP2AUX4MIX_INPUT_1_SOURCE 0x9A8 +#define WM5100_DSP2AUX5MIX_INPUT_1_SOURCE 0x9B0 +#define WM5100_DSP2AUX6MIX_INPUT_1_SOURCE 0x9B8 +#define WM5100_DSP3LMIX_INPUT_1_SOURCE 0x9C0 +#define WM5100_DSP3LMIX_INPUT_1_VOLUME 0x9C1 +#define WM5100_DSP3LMIX_INPUT_2_SOURCE 0x9C2 +#define WM5100_DSP3LMIX_INPUT_2_VOLUME 0x9C3 +#define WM5100_DSP3LMIX_INPUT_3_SOURCE 0x9C4 +#define WM5100_DSP3LMIX_INPUT_3_VOLUME 0x9C5 +#define WM5100_DSP3LMIX_INPUT_4_SOURCE 0x9C6 +#define WM5100_DSP3LMIX_INPUT_4_VOLUME 0x9C7 +#define WM5100_DSP3RMIX_INPUT_1_SOURCE 0x9C8 +#define WM5100_DSP3RMIX_INPUT_1_VOLUME 0x9C9 +#define WM5100_DSP3RMIX_INPUT_2_SOURCE 0x9CA +#define WM5100_DSP3RMIX_INPUT_2_VOLUME 0x9CB +#define WM5100_DSP3RMIX_INPUT_3_SOURCE 0x9CC +#define WM5100_DSP3RMIX_INPUT_3_VOLUME 0x9CD +#define WM5100_DSP3RMIX_INPUT_4_SOURCE 0x9CE +#define WM5100_DSP3RMIX_INPUT_4_VOLUME 0x9CF +#define WM5100_DSP3AUX1MIX_INPUT_1_SOURCE 0x9D0 +#define WM5100_DSP3AUX2MIX_INPUT_1_SOURCE 0x9D8 +#define WM5100_DSP3AUX3MIX_INPUT_1_SOURCE 0x9E0 +#define WM5100_DSP3AUX4MIX_INPUT_1_SOURCE 0x9E8 +#define WM5100_DSP3AUX5MIX_INPUT_1_SOURCE 0x9F0 +#define WM5100_DSP3AUX6MIX_INPUT_1_SOURCE 0x9F8 +#define WM5100_ASRC1LMIX_INPUT_1_SOURCE 0xA80 +#define WM5100_ASRC1RMIX_INPUT_1_SOURCE 0xA88 +#define WM5100_ASRC2LMIX_INPUT_1_SOURCE 0xA90 +#define WM5100_ASRC2RMIX_INPUT_1_SOURCE 0xA98 +#define WM5100_ISRC1DEC1MIX_INPUT_1_SOURCE 0xB00 +#define WM5100_ISRC1DEC2MIX_INPUT_1_SOURCE 0xB08 +#define WM5100_ISRC1DEC3MIX_INPUT_1_SOURCE 0xB10 +#define WM5100_ISRC1DEC4MIX_INPUT_1_SOURCE 0xB18 +#define WM5100_ISRC1INT1MIX_INPUT_1_SOURCE 0xB20 +#define WM5100_ISRC1INT2MIX_INPUT_1_SOURCE 0xB28 +#define WM5100_ISRC1INT3MIX_INPUT_1_SOURCE 0xB30 +#define WM5100_ISRC1INT4MIX_INPUT_1_SOURCE 0xB38 +#define WM5100_ISRC2DEC1MIX_INPUT_1_SOURCE 0xB40 +#define WM5100_ISRC2DEC2MIX_INPUT_1_SOURCE 0xB48 +#define WM5100_ISRC2DEC3MIX_INPUT_1_SOURCE 0xB50 +#define WM5100_ISRC2DEC4MIX_INPUT_1_SOURCE 0xB58 +#define WM5100_ISRC2INT1MIX_INPUT_1_SOURCE 0xB60 +#define WM5100_ISRC2INT2MIX_INPUT_1_SOURCE 0xB68 +#define WM5100_ISRC2INT3MIX_INPUT_1_SOURCE 0xB70 +#define WM5100_ISRC2INT4MIX_INPUT_1_SOURCE 0xB78 +#define WM5100_GPIO_CTRL_1 0xC00 +#define WM5100_GPIO_CTRL_2 0xC01 +#define WM5100_GPIO_CTRL_3 0xC02 +#define WM5100_GPIO_CTRL_4 0xC03 +#define WM5100_GPIO_CTRL_5 0xC04 +#define WM5100_GPIO_CTRL_6 0xC05 +#define WM5100_MISC_PAD_CTRL_1 0xC23 +#define WM5100_MISC_PAD_CTRL_2 0xC24 +#define WM5100_MISC_PAD_CTRL_3 0xC25 +#define WM5100_MISC_PAD_CTRL_4 0xC26 +#define WM5100_MISC_PAD_CTRL_5 0xC27 +#define WM5100_MISC_GPIO_1 0xC28 +#define WM5100_INTERRUPT_STATUS_1 0xD00 +#define WM5100_INTERRUPT_STATUS_2 0xD01 +#define WM5100_INTERRUPT_STATUS_3 0xD02 +#define WM5100_INTERRUPT_STATUS_4 0xD03 +#define WM5100_INTERRUPT_RAW_STATUS_2 0xD04 +#define WM5100_INTERRUPT_RAW_STATUS_3 0xD05 +#define WM5100_INTERRUPT_RAW_STATUS_4 0xD06 +#define WM5100_INTERRUPT_STATUS_1_MASK 0xD07 +#define WM5100_INTERRUPT_STATUS_2_MASK 0xD08 +#define WM5100_INTERRUPT_STATUS_3_MASK 0xD09 +#define WM5100_INTERRUPT_STATUS_4_MASK 0xD0A +#define WM5100_INTERRUPT_CONTROL 0xD1F +#define WM5100_IRQ_DEBOUNCE_1 0xD20 +#define WM5100_IRQ_DEBOUNCE_2 0xD21 +#define WM5100_FX_CTRL 0xE00 +#define WM5100_EQ1_1 0xE10 +#define WM5100_EQ1_2 0xE11 +#define WM5100_EQ1_3 0xE12 +#define WM5100_EQ1_4 0xE13 +#define WM5100_EQ1_5 0xE14 +#define WM5100_EQ1_6 0xE15 +#define WM5100_EQ1_7 0xE16 +#define WM5100_EQ1_8 0xE17 +#define WM5100_EQ1_9 0xE18 +#define WM5100_EQ1_10 0xE19 +#define WM5100_EQ1_11 0xE1A +#define WM5100_EQ1_12 0xE1B +#define WM5100_EQ1_13 0xE1C +#define WM5100_EQ1_14 0xE1D +#define WM5100_EQ1_15 0xE1E +#define WM5100_EQ1_16 0xE1F +#define WM5100_EQ1_17 0xE20 +#define WM5100_EQ1_18 0xE21 +#define WM5100_EQ1_19 0xE22 +#define WM5100_EQ1_20 0xE23 +#define WM5100_EQ2_1 0xE26 +#define WM5100_EQ2_2 0xE27 +#define WM5100_EQ2_3 0xE28 +#define WM5100_EQ2_4 0xE29 +#define WM5100_EQ2_5 0xE2A +#define WM5100_EQ2_6 0xE2B +#define WM5100_EQ2_7 0xE2C +#define WM5100_EQ2_8 0xE2D +#define WM5100_EQ2_9 0xE2E +#define WM5100_EQ2_10 0xE2F +#define WM5100_EQ2_11 0xE30 +#define WM5100_EQ2_12 0xE31 +#define WM5100_EQ2_13 0xE32 +#define WM5100_EQ2_14 0xE33 +#define WM5100_EQ2_15 0xE34 +#define WM5100_EQ2_16 0xE35 +#define WM5100_EQ2_17 0xE36 +#define WM5100_EQ2_18 0xE37 +#define WM5100_EQ2_19 0xE38 +#define WM5100_EQ2_20 0xE39 +#define WM5100_EQ3_1 0xE3C +#define WM5100_EQ3_2 0xE3D +#define WM5100_EQ3_3 0xE3E +#define WM5100_EQ3_4 0xE3F +#define WM5100_EQ3_5 0xE40 +#define WM5100_EQ3_6 0xE41 +#define WM5100_EQ3_7 0xE42 +#define WM5100_EQ3_8 0xE43 +#define WM5100_EQ3_9 0xE44 +#define WM5100_EQ3_10 0xE45 +#define WM5100_EQ3_11 0xE46 +#define WM5100_EQ3_12 0xE47 +#define WM5100_EQ3_13 0xE48 +#define WM5100_EQ3_14 0xE49 +#define WM5100_EQ3_15 0xE4A +#define WM5100_EQ3_16 0xE4B +#define WM5100_EQ3_17 0xE4C +#define WM5100_EQ3_18 0xE4D +#define WM5100_EQ3_19 0xE4E +#define WM5100_EQ3_20 0xE4F +#define WM5100_EQ4_1 0xE52 +#define WM5100_EQ4_2 0xE53 +#define WM5100_EQ4_3 0xE54 +#define WM5100_EQ4_4 0xE55 +#define WM5100_EQ4_5 0xE56 +#define WM5100_EQ4_6 0xE57 +#define WM5100_EQ4_7 0xE58 +#define WM5100_EQ4_8 0xE59 +#define WM5100_EQ4_9 0xE5A +#define WM5100_EQ4_10 0xE5B +#define WM5100_EQ4_11 0xE5C +#define WM5100_EQ4_12 0xE5D +#define WM5100_EQ4_13 0xE5E +#define WM5100_EQ4_14 0xE5F +#define WM5100_EQ4_15 0xE60 +#define WM5100_EQ4_16 0xE61 +#define WM5100_EQ4_17 0xE62 +#define WM5100_EQ4_18 0xE63 +#define WM5100_EQ4_19 0xE64 +#define WM5100_EQ4_20 0xE65 +#define WM5100_DRC1_CTRL1 0xE80 +#define WM5100_DRC1_CTRL2 0xE81 +#define WM5100_DRC1_CTRL3 0xE82 +#define WM5100_DRC1_CTRL4 0xE83 +#define WM5100_DRC1_CTRL5 0xE84 +#define WM5100_HPLPF1_1 0xEC0 +#define WM5100_HPLPF1_2 0xEC1 +#define WM5100_HPLPF2_1 0xEC4 +#define WM5100_HPLPF2_2 0xEC5 +#define WM5100_HPLPF3_1 0xEC8 +#define WM5100_HPLPF3_2 0xEC9 +#define WM5100_HPLPF4_1 0xECC +#define WM5100_HPLPF4_2 0xECD +#define WM5100_DSP1_DM_0 0x4000 +#define WM5100_DSP1_DM_1 0x4001 +#define WM5100_DSP1_DM_2 0x4002 +#define WM5100_DSP1_DM_3 0x4003 +#define WM5100_DSP1_DM_508 0x41FC +#define WM5100_DSP1_DM_509 0x41FD +#define WM5100_DSP1_DM_510 0x41FE +#define WM5100_DSP1_DM_511 0x41FF +#define WM5100_DSP1_PM_0 0x4800 +#define WM5100_DSP1_PM_1 0x4801 +#define WM5100_DSP1_PM_2 0x4802 +#define WM5100_DSP1_PM_3 0x4803 +#define WM5100_DSP1_PM_4 0x4804 +#define WM5100_DSP1_PM_5 0x4805 +#define WM5100_DSP1_PM_1530 0x4DFA +#define WM5100_DSP1_PM_1531 0x4DFB +#define WM5100_DSP1_PM_1532 0x4DFC +#define WM5100_DSP1_PM_1533 0x4DFD +#define WM5100_DSP1_PM_1534 0x4DFE +#define WM5100_DSP1_PM_1535 0x4DFF +#define WM5100_DSP1_ZM_0 0x5000 +#define WM5100_DSP1_ZM_1 0x5001 +#define WM5100_DSP1_ZM_2 0x5002 +#define WM5100_DSP1_ZM_3 0x5003 +#define WM5100_DSP1_ZM_2044 0x57FC +#define WM5100_DSP1_ZM_2045 0x57FD +#define WM5100_DSP1_ZM_2046 0x57FE +#define WM5100_DSP1_ZM_2047 0x57FF +#define WM5100_DSP2_DM_0 0x6000 +#define WM5100_DSP2_DM_1 0x6001 +#define WM5100_DSP2_DM_2 0x6002 +#define WM5100_DSP2_DM_3 0x6003 +#define WM5100_DSP2_DM_508 0x61FC +#define WM5100_DSP2_DM_509 0x61FD +#define WM5100_DSP2_DM_510 0x61FE +#define WM5100_DSP2_DM_511 0x61FF +#define WM5100_DSP2_PM_0 0x6800 +#define WM5100_DSP2_PM_1 0x6801 +#define WM5100_DSP2_PM_2 0x6802 +#define WM5100_DSP2_PM_3 0x6803 +#define WM5100_DSP2_PM_4 0x6804 +#define WM5100_DSP2_PM_5 0x6805 +#define WM5100_DSP2_PM_1530 0x6DFA +#define WM5100_DSP2_PM_1531 0x6DFB +#define WM5100_DSP2_PM_1532 0x6DFC +#define WM5100_DSP2_PM_1533 0x6DFD +#define WM5100_DSP2_PM_1534 0x6DFE +#define WM5100_DSP2_PM_1535 0x6DFF +#define WM5100_DSP2_ZM_0 0x7000 +#define WM5100_DSP2_ZM_1 0x7001 +#define WM5100_DSP2_ZM_2 0x7002 +#define WM5100_DSP2_ZM_3 0x7003 +#define WM5100_DSP2_ZM_2044 0x77FC +#define WM5100_DSP2_ZM_2045 0x77FD +#define WM5100_DSP2_ZM_2046 0x77FE +#define WM5100_DSP2_ZM_2047 0x77FF +#define WM5100_DSP3_DM_0 0x8000 +#define WM5100_DSP3_DM_1 0x8001 +#define WM5100_DSP3_DM_2 0x8002 +#define WM5100_DSP3_DM_3 0x8003 +#define WM5100_DSP3_DM_508 0x81FC +#define WM5100_DSP3_DM_509 0x81FD +#define WM5100_DSP3_DM_510 0x81FE +#define WM5100_DSP3_DM_511 0x81FF +#define WM5100_DSP3_PM_0 0x8800 +#define WM5100_DSP3_PM_1 0x8801 +#define WM5100_DSP3_PM_2 0x8802 +#define WM5100_DSP3_PM_3 0x8803 +#define WM5100_DSP3_PM_4 0x8804 +#define WM5100_DSP3_PM_5 0x8805 +#define WM5100_DSP3_PM_1530 0x8DFA +#define WM5100_DSP3_PM_1531 0x8DFB +#define WM5100_DSP3_PM_1532 0x8DFC +#define WM5100_DSP3_PM_1533 0x8DFD +#define WM5100_DSP3_PM_1534 0x8DFE +#define WM5100_DSP3_PM_1535 0x8DFF +#define WM5100_DSP3_ZM_0 0x9000 +#define WM5100_DSP3_ZM_1 0x9001 +#define WM5100_DSP3_ZM_2 0x9002 +#define WM5100_DSP3_ZM_3 0x9003 +#define WM5100_DSP3_ZM_2044 0x97FC +#define WM5100_DSP3_ZM_2045 0x97FD +#define WM5100_DSP3_ZM_2046 0x97FE +#define WM5100_DSP3_ZM_2047 0x97FF + +#define WM5100_REGISTER_COUNT 1435 +#define WM5100_MAX_REGISTER 0x97FF + +/* + * Field Definitions. + */ + +/* + * R0 (0x00) - software reset + */ +#define WM5100_SW_RST_DEV_ID1_MASK 0xFFFF /* SW_RST_DEV_ID1 - [15:0] */ +#define WM5100_SW_RST_DEV_ID1_SHIFT 0 /* SW_RST_DEV_ID1 - [15:0] */ +#define WM5100_SW_RST_DEV_ID1_WIDTH 16 /* SW_RST_DEV_ID1 - [15:0] */ + +/* + * R1 (0x01) - Device Revision + */ +#define WM5100_DEVICE_REVISION_MASK 0x000F /* DEVICE_REVISION - [3:0] */ +#define WM5100_DEVICE_REVISION_SHIFT 0 /* DEVICE_REVISION - [3:0] */ +#define WM5100_DEVICE_REVISION_WIDTH 4 /* DEVICE_REVISION - [3:0] */ + +/* + * R16 (0x10) - Ctrl IF 1 + */ +#define WM5100_AUTO_INC 0x0001 /* AUTO_INC */ +#define WM5100_AUTO_INC_MASK 0x0001 /* AUTO_INC */ +#define WM5100_AUTO_INC_SHIFT 0 /* AUTO_INC */ +#define WM5100_AUTO_INC_WIDTH 1 /* AUTO_INC */ + +/* + * R32 (0x20) - Tone Generator 1 + */ +#define WM5100_TONE_RATE_MASK 0x3000 /* TONE_RATE - [13:12] */ +#define WM5100_TONE_RATE_SHIFT 12 /* TONE_RATE - [13:12] */ +#define WM5100_TONE_RATE_WIDTH 2 /* TONE_RATE - [13:12] */ +#define WM5100_TONE_OFFSET_MASK 0x0300 /* TONE_OFFSET - [9:8] */ +#define WM5100_TONE_OFFSET_SHIFT 8 /* TONE_OFFSET - [9:8] */ +#define WM5100_TONE_OFFSET_WIDTH 2 /* TONE_OFFSET - [9:8] */ +#define WM5100_TONE2_ENA 0x0002 /* TONE2_ENA */ +#define WM5100_TONE2_ENA_MASK 0x0002 /* TONE2_ENA */ +#define WM5100_TONE2_ENA_SHIFT 1 /* TONE2_ENA */ +#define WM5100_TONE2_ENA_WIDTH 1 /* TONE2_ENA */ +#define WM5100_TONE1_ENA 0x0001 /* TONE1_ENA */ +#define WM5100_TONE1_ENA_MASK 0x0001 /* TONE1_ENA */ +#define WM5100_TONE1_ENA_SHIFT 0 /* TONE1_ENA */ +#define WM5100_TONE1_ENA_WIDTH 1 /* TONE1_ENA */ + +/* + * R48 (0x30) - PWM Drive 1 + */ +#define WM5100_PWM_RATE_MASK 0x3000 /* PWM_RATE - [13:12] */ +#define WM5100_PWM_RATE_SHIFT 12 /* PWM_RATE - [13:12] */ +#define WM5100_PWM_RATE_WIDTH 2 /* PWM_RATE - [13:12] */ +#define WM5100_PWM_CLK_SEL_MASK 0x0300 /* PWM_CLK_SEL - [9:8] */ +#define WM5100_PWM_CLK_SEL_SHIFT 8 /* PWM_CLK_SEL - [9:8] */ +#define WM5100_PWM_CLK_SEL_WIDTH 2 /* PWM_CLK_SEL - [9:8] */ +#define WM5100_PWM2_OVD 0x0020 /* PWM2_OVD */ +#define WM5100_PWM2_OVD_MASK 0x0020 /* PWM2_OVD */ +#define WM5100_PWM2_OVD_SHIFT 5 /* PWM2_OVD */ +#define WM5100_PWM2_OVD_WIDTH 1 /* PWM2_OVD */ +#define WM5100_PWM1_OVD 0x0010 /* PWM1_OVD */ +#define WM5100_PWM1_OVD_MASK 0x0010 /* PWM1_OVD */ +#define WM5100_PWM1_OVD_SHIFT 4 /* PWM1_OVD */ +#define WM5100_PWM1_OVD_WIDTH 1 /* PWM1_OVD */ +#define WM5100_PWM2_ENA 0x0002 /* PWM2_ENA */ +#define WM5100_PWM2_ENA_MASK 0x0002 /* PWM2_ENA */ +#define WM5100_PWM2_ENA_SHIFT 1 /* PWM2_ENA */ +#define WM5100_PWM2_ENA_WIDTH 1 /* PWM2_ENA */ +#define WM5100_PWM1_ENA 0x0001 /* PWM1_ENA */ +#define WM5100_PWM1_ENA_MASK 0x0001 /* PWM1_ENA */ +#define WM5100_PWM1_ENA_SHIFT 0 /* PWM1_ENA */ +#define WM5100_PWM1_ENA_WIDTH 1 /* PWM1_ENA */ + +/* + * R49 (0x31) - PWM Drive 2 + */ +#define WM5100_PWM1_LVL_MASK 0x03FF /* PWM1_LVL - [9:0] */ +#define WM5100_PWM1_LVL_SHIFT 0 /* PWM1_LVL - [9:0] */ +#define WM5100_PWM1_LVL_WIDTH 10 /* PWM1_LVL - [9:0] */ + +/* + * R50 (0x32) - PWM Drive 3 + */ +#define WM5100_PWM2_LVL_MASK 0x03FF /* PWM2_LVL - [9:0] */ +#define WM5100_PWM2_LVL_SHIFT 0 /* PWM2_LVL - [9:0] */ +#define WM5100_PWM2_LVL_WIDTH 10 /* PWM2_LVL - [9:0] */ + +/* + * R256 (0x100) - Clocking 1 + */ +#define WM5100_CLK_32K_SRC_MASK 0x000F /* CLK_32K_SRC - [3:0] */ +#define WM5100_CLK_32K_SRC_SHIFT 0 /* CLK_32K_SRC - [3:0] */ +#define WM5100_CLK_32K_SRC_WIDTH 4 /* CLK_32K_SRC - [3:0] */ + +/* + * R257 (0x101) - Clocking 3 + */ +#define WM5100_SYSCLK_FREQ_MASK 0x0700 /* SYSCLK_FREQ - [10:8] */ +#define WM5100_SYSCLK_FREQ_SHIFT 8 /* SYSCLK_FREQ - [10:8] */ +#define WM5100_SYSCLK_FREQ_WIDTH 3 /* SYSCLK_FREQ - [10:8] */ +#define WM5100_SYSCLK_ENA 0x0040 /* SYSCLK_ENA */ +#define WM5100_SYSCLK_ENA_MASK 0x0040 /* SYSCLK_ENA */ +#define WM5100_SYSCLK_ENA_SHIFT 6 /* SYSCLK_ENA */ +#define WM5100_SYSCLK_ENA_WIDTH 1 /* SYSCLK_ENA */ +#define WM5100_SYSCLK_SRC_MASK 0x000F /* SYSCLK_SRC - [3:0] */ +#define WM5100_SYSCLK_SRC_SHIFT 0 /* SYSCLK_SRC - [3:0] */ +#define WM5100_SYSCLK_SRC_WIDTH 4 /* SYSCLK_SRC - [3:0] */ + +/* + * R258 (0x102) - Clocking 4 + */ +#define WM5100_SAMPLE_RATE_1_MASK 0x001F /* SAMPLE_RATE_1 - [4:0] */ +#define WM5100_SAMPLE_RATE_1_SHIFT 0 /* SAMPLE_RATE_1 - [4:0] */ +#define WM5100_SAMPLE_RATE_1_WIDTH 5 /* SAMPLE_RATE_1 - [4:0] */ + +/* + * R259 (0x103) - Clocking 5 + */ +#define WM5100_SAMPLE_RATE_2_MASK 0x001F /* SAMPLE_RATE_2 - [4:0] */ +#define WM5100_SAMPLE_RATE_2_SHIFT 0 /* SAMPLE_RATE_2 - [4:0] */ +#define WM5100_SAMPLE_RATE_2_WIDTH 5 /* SAMPLE_RATE_2 - [4:0] */ + +/* + * R260 (0x104) - Clocking 6 + */ +#define WM5100_SAMPLE_RATE_3_MASK 0x001F /* SAMPLE_RATE_3 - [4:0] */ +#define WM5100_SAMPLE_RATE_3_SHIFT 0 /* SAMPLE_RATE_3 - [4:0] */ +#define WM5100_SAMPLE_RATE_3_WIDTH 5 /* SAMPLE_RATE_3 - [4:0] */ + +/* + * R263 (0x107) - Clocking 7 + */ +#define WM5100_ASYNC_CLK_FREQ_MASK 0x0700 /* ASYNC_CLK_FREQ - [10:8] */ +#define WM5100_ASYNC_CLK_FREQ_SHIFT 8 /* ASYNC_CLK_FREQ - [10:8] */ +#define WM5100_ASYNC_CLK_FREQ_WIDTH 3 /* ASYNC_CLK_FREQ - [10:8] */ +#define WM5100_ASYNC_CLK_ENA 0x0040 /* ASYNC_CLK_ENA */ +#define WM5100_ASYNC_CLK_ENA_MASK 0x0040 /* ASYNC_CLK_ENA */ +#define WM5100_ASYNC_CLK_ENA_SHIFT 6 /* ASYNC_CLK_ENA */ +#define WM5100_ASYNC_CLK_ENA_WIDTH 1 /* ASYNC_CLK_ENA */ +#define WM5100_ASYNC_CLK_SRC_MASK 0x000F /* ASYNC_CLK_SRC - [3:0] */ +#define WM5100_ASYNC_CLK_SRC_SHIFT 0 /* ASYNC_CLK_SRC - [3:0] */ +#define WM5100_ASYNC_CLK_SRC_WIDTH 4 /* ASYNC_CLK_SRC - [3:0] */ + +/* + * R264 (0x108) - Clocking 8 + */ +#define WM5100_ASYNC_SAMPLE_RATE_MASK 0x001F /* ASYNC_SAMPLE_RATE - [4:0] */ +#define WM5100_ASYNC_SAMPLE_RATE_SHIFT 0 /* ASYNC_SAMPLE_RATE - [4:0] */ +#define WM5100_ASYNC_SAMPLE_RATE_WIDTH 5 /* ASYNC_SAMPLE_RATE - [4:0] */ + +/* + * R288 (0x120) - ASRC_ENABLE + */ +#define WM5100_ASRC2L_ENA 0x0008 /* ASRC2L_ENA */ +#define WM5100_ASRC2L_ENA_MASK 0x0008 /* ASRC2L_ENA */ +#define WM5100_ASRC2L_ENA_SHIFT 3 /* ASRC2L_ENA */ +#define WM5100_ASRC2L_ENA_WIDTH 1 /* ASRC2L_ENA */ +#define WM5100_ASRC2R_ENA 0x0004 /* ASRC2R_ENA */ +#define WM5100_ASRC2R_ENA_MASK 0x0004 /* ASRC2R_ENA */ +#define WM5100_ASRC2R_ENA_SHIFT 2 /* ASRC2R_ENA */ +#define WM5100_ASRC2R_ENA_WIDTH 1 /* ASRC2R_ENA */ +#define WM5100_ASRC1L_ENA 0x0002 /* ASRC1L_ENA */ +#define WM5100_ASRC1L_ENA_MASK 0x0002 /* ASRC1L_ENA */ +#define WM5100_ASRC1L_ENA_SHIFT 1 /* ASRC1L_ENA */ +#define WM5100_ASRC1L_ENA_WIDTH 1 /* ASRC1L_ENA */ +#define WM5100_ASRC1R_ENA 0x0001 /* ASRC1R_ENA */ +#define WM5100_ASRC1R_ENA_MASK 0x0001 /* ASRC1R_ENA */ +#define WM5100_ASRC1R_ENA_SHIFT 0 /* ASRC1R_ENA */ +#define WM5100_ASRC1R_ENA_WIDTH 1 /* ASRC1R_ENA */ + +/* + * R289 (0x121) - ASRC_STATUS + */ +#define WM5100_ASRC2L_ENA_STS 0x0008 /* ASRC2L_ENA_STS */ +#define WM5100_ASRC2L_ENA_STS_MASK 0x0008 /* ASRC2L_ENA_STS */ +#define WM5100_ASRC2L_ENA_STS_SHIFT 3 /* ASRC2L_ENA_STS */ +#define WM5100_ASRC2L_ENA_STS_WIDTH 1 /* ASRC2L_ENA_STS */ +#define WM5100_ASRC2R_ENA_STS 0x0004 /* ASRC2R_ENA_STS */ +#define WM5100_ASRC2R_ENA_STS_MASK 0x0004 /* ASRC2R_ENA_STS */ +#define WM5100_ASRC2R_ENA_STS_SHIFT 2 /* ASRC2R_ENA_STS */ +#define WM5100_ASRC2R_ENA_STS_WIDTH 1 /* ASRC2R_ENA_STS */ +#define WM5100_ASRC1L_ENA_STS 0x0002 /* ASRC1L_ENA_STS */ +#define WM5100_ASRC1L_ENA_STS_MASK 0x0002 /* ASRC1L_ENA_STS */ +#define WM5100_ASRC1L_ENA_STS_SHIFT 1 /* ASRC1L_ENA_STS */ +#define WM5100_ASRC1L_ENA_STS_WIDTH 1 /* ASRC1L_ENA_STS */ +#define WM5100_ASRC1R_ENA_STS 0x0001 /* ASRC1R_ENA_STS */ +#define WM5100_ASRC1R_ENA_STS_MASK 0x0001 /* ASRC1R_ENA_STS */ +#define WM5100_ASRC1R_ENA_STS_SHIFT 0 /* ASRC1R_ENA_STS */ +#define WM5100_ASRC1R_ENA_STS_WIDTH 1 /* ASRC1R_ENA_STS */ + +/* + * R290 (0x122) - ASRC_RATE1 + */ +#define WM5100_ASRC_RATE1_MASK 0x0006 /* ASRC_RATE1 - [2:1] */ +#define WM5100_ASRC_RATE1_SHIFT 1 /* ASRC_RATE1 - [2:1] */ +#define WM5100_ASRC_RATE1_WIDTH 2 /* ASRC_RATE1 - [2:1] */ + +/* + * R321 (0x141) - ISRC 1 CTRL 1 + */ +#define WM5100_ISRC1_DFS_ENA 0x2000 /* ISRC1_DFS_ENA */ +#define WM5100_ISRC1_DFS_ENA_MASK 0x2000 /* ISRC1_DFS_ENA */ +#define WM5100_ISRC1_DFS_ENA_SHIFT 13 /* ISRC1_DFS_ENA */ +#define WM5100_ISRC1_DFS_ENA_WIDTH 1 /* ISRC1_DFS_ENA */ +#define WM5100_ISRC1_CLK_SEL_MASK 0x0300 /* ISRC1_CLK_SEL - [9:8] */ +#define WM5100_ISRC1_CLK_SEL_SHIFT 8 /* ISRC1_CLK_SEL - [9:8] */ +#define WM5100_ISRC1_CLK_SEL_WIDTH 2 /* ISRC1_CLK_SEL - [9:8] */ +#define WM5100_ISRC1_FSH_MASK 0x000C /* ISRC1_FSH - [3:2] */ +#define WM5100_ISRC1_FSH_SHIFT 2 /* ISRC1_FSH - [3:2] */ +#define WM5100_ISRC1_FSH_WIDTH 2 /* ISRC1_FSH - [3:2] */ +#define WM5100_ISRC1_FSL_MASK 0x0003 /* ISRC1_FSL - [1:0] */ +#define WM5100_ISRC1_FSL_SHIFT 0 /* ISRC1_FSL - [1:0] */ +#define WM5100_ISRC1_FSL_WIDTH 2 /* ISRC1_FSL - [1:0] */ + +/* + * R322 (0x142) - ISRC 1 CTRL 2 + */ +#define WM5100_ISRC1_INT1_ENA 0x8000 /* ISRC1_INT1_ENA */ +#define WM5100_ISRC1_INT1_ENA_MASK 0x8000 /* ISRC1_INT1_ENA */ +#define WM5100_ISRC1_INT1_ENA_SHIFT 15 /* ISRC1_INT1_ENA */ +#define WM5100_ISRC1_INT1_ENA_WIDTH 1 /* ISRC1_INT1_ENA */ +#define WM5100_ISRC1_INT2_ENA 0x4000 /* ISRC1_INT2_ENA */ +#define WM5100_ISRC1_INT2_ENA_MASK 0x4000 /* ISRC1_INT2_ENA */ +#define WM5100_ISRC1_INT2_ENA_SHIFT 14 /* ISRC1_INT2_ENA */ +#define WM5100_ISRC1_INT2_ENA_WIDTH 1 /* ISRC1_INT2_ENA */ +#define WM5100_ISRC1_INT3_ENA 0x2000 /* ISRC1_INT3_ENA */ +#define WM5100_ISRC1_INT3_ENA_MASK 0x2000 /* ISRC1_INT3_ENA */ +#define WM5100_ISRC1_INT3_ENA_SHIFT 13 /* ISRC1_INT3_ENA */ +#define WM5100_ISRC1_INT3_ENA_WIDTH 1 /* ISRC1_INT3_ENA */ +#define WM5100_ISRC1_INT4_ENA 0x1000 /* ISRC1_INT4_ENA */ +#define WM5100_ISRC1_INT4_ENA_MASK 0x1000 /* ISRC1_INT4_ENA */ +#define WM5100_ISRC1_INT4_ENA_SHIFT 12 /* ISRC1_INT4_ENA */ +#define WM5100_ISRC1_INT4_ENA_WIDTH 1 /* ISRC1_INT4_ENA */ +#define WM5100_ISRC1_DEC1_ENA 0x0200 /* ISRC1_DEC1_ENA */ +#define WM5100_ISRC1_DEC1_ENA_MASK 0x0200 /* ISRC1_DEC1_ENA */ +#define WM5100_ISRC1_DEC1_ENA_SHIFT 9 /* ISRC1_DEC1_ENA */ +#define WM5100_ISRC1_DEC1_ENA_WIDTH 1 /* ISRC1_DEC1_ENA */ +#define WM5100_ISRC1_DEC2_ENA 0x0100 /* ISRC1_DEC2_ENA */ +#define WM5100_ISRC1_DEC2_ENA_MASK 0x0100 /* ISRC1_DEC2_ENA */ +#define WM5100_ISRC1_DEC2_ENA_SHIFT 8 /* ISRC1_DEC2_ENA */ +#define WM5100_ISRC1_DEC2_ENA_WIDTH 1 /* ISRC1_DEC2_ENA */ +#define WM5100_ISRC1_DEC3_ENA 0x0080 /* ISRC1_DEC3_ENA */ +#define WM5100_ISRC1_DEC3_ENA_MASK 0x0080 /* ISRC1_DEC3_ENA */ +#define WM5100_ISRC1_DEC3_ENA_SHIFT 7 /* ISRC1_DEC3_ENA */ +#define WM5100_ISRC1_DEC3_ENA_WIDTH 1 /* ISRC1_DEC3_ENA */ +#define WM5100_ISRC1_DEC4_ENA 0x0040 /* ISRC1_DEC4_ENA */ +#define WM5100_ISRC1_DEC4_ENA_MASK 0x0040 /* ISRC1_DEC4_ENA */ +#define WM5100_ISRC1_DEC4_ENA_SHIFT 6 /* ISRC1_DEC4_ENA */ +#define WM5100_ISRC1_DEC4_ENA_WIDTH 1 /* ISRC1_DEC4_ENA */ +#define WM5100_ISRC1_NOTCH_ENA 0x0001 /* ISRC1_NOTCH_ENA */ +#define WM5100_ISRC1_NOTCH_ENA_MASK 0x0001 /* ISRC1_NOTCH_ENA */ +#define WM5100_ISRC1_NOTCH_ENA_SHIFT 0 /* ISRC1_NOTCH_ENA */ +#define WM5100_ISRC1_NOTCH_ENA_WIDTH 1 /* ISRC1_NOTCH_ENA */ + +/* + * R323 (0x143) - ISRC 2 CTRL1 + */ +#define WM5100_ISRC2_DFS_ENA 0x2000 /* ISRC2_DFS_ENA */ +#define WM5100_ISRC2_DFS_ENA_MASK 0x2000 /* ISRC2_DFS_ENA */ +#define WM5100_ISRC2_DFS_ENA_SHIFT 13 /* ISRC2_DFS_ENA */ +#define WM5100_ISRC2_DFS_ENA_WIDTH 1 /* ISRC2_DFS_ENA */ +#define WM5100_ISRC2_CLK_SEL_MASK 0x0300 /* ISRC2_CLK_SEL - [9:8] */ +#define WM5100_ISRC2_CLK_SEL_SHIFT 8 /* ISRC2_CLK_SEL - [9:8] */ +#define WM5100_ISRC2_CLK_SEL_WIDTH 2 /* ISRC2_CLK_SEL - [9:8] */ +#define WM5100_ISRC2_FSH_MASK 0x000C /* ISRC2_FSH - [3:2] */ +#define WM5100_ISRC2_FSH_SHIFT 2 /* ISRC2_FSH - [3:2] */ +#define WM5100_ISRC2_FSH_WIDTH 2 /* ISRC2_FSH - [3:2] */ +#define WM5100_ISRC2_FSL_MASK 0x0003 /* ISRC2_FSL - [1:0] */ +#define WM5100_ISRC2_FSL_SHIFT 0 /* ISRC2_FSL - [1:0] */ +#define WM5100_ISRC2_FSL_WIDTH 2 /* ISRC2_FSL - [1:0] */ + +/* + * R324 (0x144) - ISRC 2 CTRL 2 + */ +#define WM5100_ISRC2_INT1_ENA 0x8000 /* ISRC2_INT1_ENA */ +#define WM5100_ISRC2_INT1_ENA_MASK 0x8000 /* ISRC2_INT1_ENA */ +#define WM5100_ISRC2_INT1_ENA_SHIFT 15 /* ISRC2_INT1_ENA */ +#define WM5100_ISRC2_INT1_ENA_WIDTH 1 /* ISRC2_INT1_ENA */ +#define WM5100_ISRC2_INT2_ENA 0x4000 /* ISRC2_INT2_ENA */ +#define WM5100_ISRC2_INT2_ENA_MASK 0x4000 /* ISRC2_INT2_ENA */ +#define WM5100_ISRC2_INT2_ENA_SHIFT 14 /* ISRC2_INT2_ENA */ +#define WM5100_ISRC2_INT2_ENA_WIDTH 1 /* ISRC2_INT2_ENA */ +#define WM5100_ISRC2_INT3_ENA 0x2000 /* ISRC2_INT3_ENA */ +#define WM5100_ISRC2_INT3_ENA_MASK 0x2000 /* ISRC2_INT3_ENA */ +#define WM5100_ISRC2_INT3_ENA_SHIFT 13 /* ISRC2_INT3_ENA */ +#define WM5100_ISRC2_INT3_ENA_WIDTH 1 /* ISRC2_INT3_ENA */ +#define WM5100_ISRC2_INT4_ENA 0x1000 /* ISRC2_INT4_ENA */ +#define WM5100_ISRC2_INT4_ENA_MASK 0x1000 /* ISRC2_INT4_ENA */ +#define WM5100_ISRC2_INT4_ENA_SHIFT 12 /* ISRC2_INT4_ENA */ +#define WM5100_ISRC2_INT4_ENA_WIDTH 1 /* ISRC2_INT4_ENA */ +#define WM5100_ISRC2_DEC1_ENA 0x0200 /* ISRC2_DEC1_ENA */ +#define WM5100_ISRC2_DEC1_ENA_MASK 0x0200 /* ISRC2_DEC1_ENA */ +#define WM5100_ISRC2_DEC1_ENA_SHIFT 9 /* ISRC2_DEC1_ENA */ +#define WM5100_ISRC2_DEC1_ENA_WIDTH 1 /* ISRC2_DEC1_ENA */ +#define WM5100_ISRC2_DEC2_ENA 0x0100 /* ISRC2_DEC2_ENA */ +#define WM5100_ISRC2_DEC2_ENA_MASK 0x0100 /* ISRC2_DEC2_ENA */ +#define WM5100_ISRC2_DEC2_ENA_SHIFT 8 /* ISRC2_DEC2_ENA */ +#define WM5100_ISRC2_DEC2_ENA_WIDTH 1 /* ISRC2_DEC2_ENA */ +#define WM5100_ISRC2_DEC3_ENA 0x0080 /* ISRC2_DEC3_ENA */ +#define WM5100_ISRC2_DEC3_ENA_MASK 0x0080 /* ISRC2_DEC3_ENA */ +#define WM5100_ISRC2_DEC3_ENA_SHIFT 7 /* ISRC2_DEC3_ENA */ +#define WM5100_ISRC2_DEC3_ENA_WIDTH 1 /* ISRC2_DEC3_ENA */ +#define WM5100_ISRC2_DEC4_ENA 0x0040 /* ISRC2_DEC4_ENA */ +#define WM5100_ISRC2_DEC4_ENA_MASK 0x0040 /* ISRC2_DEC4_ENA */ +#define WM5100_ISRC2_DEC4_ENA_SHIFT 6 /* ISRC2_DEC4_ENA */ +#define WM5100_ISRC2_DEC4_ENA_WIDTH 1 /* ISRC2_DEC4_ENA */ +#define WM5100_ISRC2_NOTCH_ENA 0x0001 /* ISRC2_NOTCH_ENA */ +#define WM5100_ISRC2_NOTCH_ENA_MASK 0x0001 /* ISRC2_NOTCH_ENA */ +#define WM5100_ISRC2_NOTCH_ENA_SHIFT 0 /* ISRC2_NOTCH_ENA */ +#define WM5100_ISRC2_NOTCH_ENA_WIDTH 1 /* ISRC2_NOTCH_ENA */ + +/* + * R386 (0x182) - FLL1 Control 1 + */ +#define WM5100_FLL1_ENA 0x0001 /* FLL1_ENA */ +#define WM5100_FLL1_ENA_MASK 0x0001 /* FLL1_ENA */ +#define WM5100_FLL1_ENA_SHIFT 0 /* FLL1_ENA */ +#define WM5100_FLL1_ENA_WIDTH 1 /* FLL1_ENA */ + +/* + * R387 (0x183) - FLL1 Control 2 + */ +#define WM5100_FLL1_OUTDIV_MASK 0x3F00 /* FLL1_OUTDIV - [13:8] */ +#define WM5100_FLL1_OUTDIV_SHIFT 8 /* FLL1_OUTDIV - [13:8] */ +#define WM5100_FLL1_OUTDIV_WIDTH 6 /* FLL1_OUTDIV - [13:8] */ +#define WM5100_FLL1_FRATIO_MASK 0x0007 /* FLL1_FRATIO - [2:0] */ +#define WM5100_FLL1_FRATIO_SHIFT 0 /* FLL1_FRATIO - [2:0] */ +#define WM5100_FLL1_FRATIO_WIDTH 3 /* FLL1_FRATIO - [2:0] */ + +/* + * R388 (0x184) - FLL1 Control 3 + */ +#define WM5100_FLL1_THETA_MASK 0xFFFF /* FLL1_THETA - [15:0] */ +#define WM5100_FLL1_THETA_SHIFT 0 /* FLL1_THETA - [15:0] */ +#define WM5100_FLL1_THETA_WIDTH 16 /* FLL1_THETA - [15:0] */ + +/* + * R390 (0x186) - FLL1 Control 5 + */ +#define WM5100_FLL1_N_MASK 0x03FF /* FLL1_N - [9:0] */ +#define WM5100_FLL1_N_SHIFT 0 /* FLL1_N - [9:0] */ +#define WM5100_FLL1_N_WIDTH 10 /* FLL1_N - [9:0] */ + +/* + * R391 (0x187) - FLL1 Control 6 + */ +#define WM5100_FLL1_REFCLK_DIV_MASK 0x00C0 /* FLL1_REFCLK_DIV - [7:6] */ +#define WM5100_FLL1_REFCLK_DIV_SHIFT 6 /* FLL1_REFCLK_DIV - [7:6] */ +#define WM5100_FLL1_REFCLK_DIV_WIDTH 2 /* FLL1_REFCLK_DIV - [7:6] */ +#define WM5100_FLL1_REFCLK_SRC_MASK 0x000F /* FLL1_REFCLK_SRC - [3:0] */ +#define WM5100_FLL1_REFCLK_SRC_SHIFT 0 /* FLL1_REFCLK_SRC - [3:0] */ +#define WM5100_FLL1_REFCLK_SRC_WIDTH 4 /* FLL1_REFCLK_SRC - [3:0] */ + +/* + * R392 (0x188) - FLL1 EFS 1 + */ +#define WM5100_FLL1_LAMBDA_MASK 0xFFFF /* FLL1_LAMBDA - [15:0] */ +#define WM5100_FLL1_LAMBDA_SHIFT 0 /* FLL1_LAMBDA - [15:0] */ +#define WM5100_FLL1_LAMBDA_WIDTH 16 /* FLL1_LAMBDA - [15:0] */ + +/* + * R418 (0x1A2) - FLL2 Control 1 + */ +#define WM5100_FLL2_ENA 0x0001 /* FLL2_ENA */ +#define WM5100_FLL2_ENA_MASK 0x0001 /* FLL2_ENA */ +#define WM5100_FLL2_ENA_SHIFT 0 /* FLL2_ENA */ +#define WM5100_FLL2_ENA_WIDTH 1 /* FLL2_ENA */ + +/* + * R419 (0x1A3) - FLL2 Control 2 + */ +#define WM5100_FLL2_OUTDIV_MASK 0x3F00 /* FLL2_OUTDIV - [13:8] */ +#define WM5100_FLL2_OUTDIV_SHIFT 8 /* FLL2_OUTDIV - [13:8] */ +#define WM5100_FLL2_OUTDIV_WIDTH 6 /* FLL2_OUTDIV - [13:8] */ +#define WM5100_FLL2_FRATIO_MASK 0x0007 /* FLL2_FRATIO - [2:0] */ +#define WM5100_FLL2_FRATIO_SHIFT 0 /* FLL2_FRATIO - [2:0] */ +#define WM5100_FLL2_FRATIO_WIDTH 3 /* FLL2_FRATIO - [2:0] */ + +/* + * R420 (0x1A4) - FLL2 Control 3 + */ +#define WM5100_FLL2_THETA_MASK 0xFFFF /* FLL2_THETA - [15:0] */ +#define WM5100_FLL2_THETA_SHIFT 0 /* FLL2_THETA - [15:0] */ +#define WM5100_FLL2_THETA_WIDTH 16 /* FLL2_THETA - [15:0] */ + +/* + * R422 (0x1A6) - FLL2 Control 5 + */ +#define WM5100_FLL2_N_MASK 0x03FF /* FLL2_N - [9:0] */ +#define WM5100_FLL2_N_SHIFT 0 /* FLL2_N - [9:0] */ +#define WM5100_FLL2_N_WIDTH 10 /* FLL2_N - [9:0] */ + +/* + * R423 (0x1A7) - FLL2 Control 6 + */ +#define WM5100_FLL2_REFCLK_DIV_MASK 0x00C0 /* FLL2_REFCLK_DIV - [7:6] */ +#define WM5100_FLL2_REFCLK_DIV_SHIFT 6 /* FLL2_REFCLK_DIV - [7:6] */ +#define WM5100_FLL2_REFCLK_DIV_WIDTH 2 /* FLL2_REFCLK_DIV - [7:6] */ +#define WM5100_FLL2_REFCLK_SRC_MASK 0x000F /* FLL2_REFCLK_SRC - [3:0] */ +#define WM5100_FLL2_REFCLK_SRC_SHIFT 0 /* FLL2_REFCLK_SRC - [3:0] */ +#define WM5100_FLL2_REFCLK_SRC_WIDTH 4 /* FLL2_REFCLK_SRC - [3:0] */ + +/* + * R424 (0x1A8) - FLL2 EFS 1 + */ +#define WM5100_FLL2_LAMBDA_MASK 0xFFFF /* FLL2_LAMBDA - [15:0] */ +#define WM5100_FLL2_LAMBDA_SHIFT 0 /* FLL2_LAMBDA - [15:0] */ +#define WM5100_FLL2_LAMBDA_WIDTH 16 /* FLL2_LAMBDA - [15:0] */ + +/* + * R512 (0x200) - Mic Charge Pump 1 + */ +#define WM5100_CP2_BYPASS 0x0020 /* CP2_BYPASS */ +#define WM5100_CP2_BYPASS_MASK 0x0020 /* CP2_BYPASS */ +#define WM5100_CP2_BYPASS_SHIFT 5 /* CP2_BYPASS */ +#define WM5100_CP2_BYPASS_WIDTH 1 /* CP2_BYPASS */ +#define WM5100_CP2_ENA 0x0001 /* CP2_ENA */ +#define WM5100_CP2_ENA_MASK 0x0001 /* CP2_ENA */ +#define WM5100_CP2_ENA_SHIFT 0 /* CP2_ENA */ +#define WM5100_CP2_ENA_WIDTH 1 /* CP2_ENA */ + +/* + * R513 (0x201) - Mic Charge Pump 2 + */ +#define WM5100_LDO2_VSEL_MASK 0xF800 /* LDO2_VSEL - [15:11] */ +#define WM5100_LDO2_VSEL_SHIFT 11 /* LDO2_VSEL - [15:11] */ +#define WM5100_LDO2_VSEL_WIDTH 5 /* LDO2_VSEL - [15:11] */ + +/* + * R514 (0x202) - HP Charge Pump 1 + */ +#define WM5100_CP1_ENA 0x0001 /* CP1_ENA */ +#define WM5100_CP1_ENA_MASK 0x0001 /* CP1_ENA */ +#define WM5100_CP1_ENA_SHIFT 0 /* CP1_ENA */ +#define WM5100_CP1_ENA_WIDTH 1 /* CP1_ENA */ + +/* + * R529 (0x211) - LDO1 Control + */ +#define WM5100_LDO1_BYPASS 0x0002 /* LDO1_BYPASS */ +#define WM5100_LDO1_BYPASS_MASK 0x0002 /* LDO1_BYPASS */ +#define WM5100_LDO1_BYPASS_SHIFT 1 /* LDO1_BYPASS */ +#define WM5100_LDO1_BYPASS_WIDTH 1 /* LDO1_BYPASS */ + +/* + * R533 (0x215) - Mic Bias Ctrl 1 + */ +#define WM5100_MICB1_DISCH 0x0040 /* MICB1_DISCH */ +#define WM5100_MICB1_DISCH_MASK 0x0040 /* MICB1_DISCH */ +#define WM5100_MICB1_DISCH_SHIFT 6 /* MICB1_DISCH */ +#define WM5100_MICB1_DISCH_WIDTH 1 /* MICB1_DISCH */ +#define WM5100_MICB1_RATE 0x0020 /* MICB1_RATE */ +#define WM5100_MICB1_RATE_MASK 0x0020 /* MICB1_RATE */ +#define WM5100_MICB1_RATE_SHIFT 5 /* MICB1_RATE */ +#define WM5100_MICB1_RATE_WIDTH 1 /* MICB1_RATE */ +#define WM5100_MICB1_LVL_MASK 0x001C /* MICB1_LVL - [4:2] */ +#define WM5100_MICB1_LVL_SHIFT 2 /* MICB1_LVL - [4:2] */ +#define WM5100_MICB1_LVL_WIDTH 3 /* MICB1_LVL - [4:2] */ +#define WM5100_MICB1_BYPASS 0x0002 /* MICB1_BYPASS */ +#define WM5100_MICB1_BYPASS_MASK 0x0002 /* MICB1_BYPASS */ +#define WM5100_MICB1_BYPASS_SHIFT 1 /* MICB1_BYPASS */ +#define WM5100_MICB1_BYPASS_WIDTH 1 /* MICB1_BYPASS */ +#define WM5100_MICB1_ENA 0x0001 /* MICB1_ENA */ +#define WM5100_MICB1_ENA_MASK 0x0001 /* MICB1_ENA */ +#define WM5100_MICB1_ENA_SHIFT 0 /* MICB1_ENA */ +#define WM5100_MICB1_ENA_WIDTH 1 /* MICB1_ENA */ + +/* + * R534 (0x216) - Mic Bias Ctrl 2 + */ +#define WM5100_MICB2_DISCH 0x0040 /* MICB2_DISCH */ +#define WM5100_MICB2_DISCH_MASK 0x0040 /* MICB2_DISCH */ +#define WM5100_MICB2_DISCH_SHIFT 6 /* MICB2_DISCH */ +#define WM5100_MICB2_DISCH_WIDTH 1 /* MICB2_DISCH */ +#define WM5100_MICB2_RATE 0x0020 /* MICB2_RATE */ +#define WM5100_MICB2_RATE_MASK 0x0020 /* MICB2_RATE */ +#define WM5100_MICB2_RATE_SHIFT 5 /* MICB2_RATE */ +#define WM5100_MICB2_RATE_WIDTH 1 /* MICB2_RATE */ +#define WM5100_MICB2_LVL_MASK 0x001C /* MICB2_LVL - [4:2] */ +#define WM5100_MICB2_LVL_SHIFT 2 /* MICB2_LVL - [4:2] */ +#define WM5100_MICB2_LVL_WIDTH 3 /* MICB2_LVL - [4:2] */ +#define WM5100_MICB2_BYPASS 0x0002 /* MICB2_BYPASS */ +#define WM5100_MICB2_BYPASS_MASK 0x0002 /* MICB2_BYPASS */ +#define WM5100_MICB2_BYPASS_SHIFT 1 /* MICB2_BYPASS */ +#define WM5100_MICB2_BYPASS_WIDTH 1 /* MICB2_BYPASS */ +#define WM5100_MICB2_ENA 0x0001 /* MICB2_ENA */ +#define WM5100_MICB2_ENA_MASK 0x0001 /* MICB2_ENA */ +#define WM5100_MICB2_ENA_SHIFT 0 /* MICB2_ENA */ +#define WM5100_MICB2_ENA_WIDTH 1 /* MICB2_ENA */ + +/* + * R535 (0x217) - Mic Bias Ctrl 3 + */ +#define WM5100_MICB3_DISCH 0x0040 /* MICB3_DISCH */ +#define WM5100_MICB3_DISCH_MASK 0x0040 /* MICB3_DISCH */ +#define WM5100_MICB3_DISCH_SHIFT 6 /* MICB3_DISCH */ +#define WM5100_MICB3_DISCH_WIDTH 1 /* MICB3_DISCH */ +#define WM5100_MICB3_RATE 0x0020 /* MICB3_RATE */ +#define WM5100_MICB3_RATE_MASK 0x0020 /* MICB3_RATE */ +#define WM5100_MICB3_RATE_SHIFT 5 /* MICB3_RATE */ +#define WM5100_MICB3_RATE_WIDTH 1 /* MICB3_RATE */ +#define WM5100_MICB3_LVL_MASK 0x001C /* MICB3_LVL - [4:2] */ +#define WM5100_MICB3_LVL_SHIFT 2 /* MICB3_LVL - [4:2] */ +#define WM5100_MICB3_LVL_WIDTH 3 /* MICB3_LVL - [4:2] */ +#define WM5100_MICB3_BYPASS 0x0002 /* MICB3_BYPASS */ +#define WM5100_MICB3_BYPASS_MASK 0x0002 /* MICB3_BYPASS */ +#define WM5100_MICB3_BYPASS_SHIFT 1 /* MICB3_BYPASS */ +#define WM5100_MICB3_BYPASS_WIDTH 1 /* MICB3_BYPASS */ +#define WM5100_MICB3_ENA 0x0001 /* MICB3_ENA */ +#define WM5100_MICB3_ENA_MASK 0x0001 /* MICB3_ENA */ +#define WM5100_MICB3_ENA_SHIFT 0 /* MICB3_ENA */ +#define WM5100_MICB3_ENA_WIDTH 1 /* MICB3_ENA */ + +/* + * R640 (0x280) - Accessory Detect Mode 1 + */ +#define WM5100_ACCDET_BIAS_SRC_MASK 0xC000 /* ACCDET_BIAS_SRC - [15:14] */ +#define WM5100_ACCDET_BIAS_SRC_SHIFT 14 /* ACCDET_BIAS_SRC - [15:14] */ +#define WM5100_ACCDET_BIAS_SRC_WIDTH 2 /* ACCDET_BIAS_SRC - [15:14] */ +#define WM5100_ACCDET_SRC 0x2000 /* ACCDET_SRC */ +#define WM5100_ACCDET_SRC_MASK 0x2000 /* ACCDET_SRC */ +#define WM5100_ACCDET_SRC_SHIFT 13 /* ACCDET_SRC */ +#define WM5100_ACCDET_SRC_WIDTH 1 /* ACCDET_SRC */ +#define WM5100_ACCDET_MODE_MASK 0x0003 /* ACCDET_MODE - [1:0] */ +#define WM5100_ACCDET_MODE_SHIFT 0 /* ACCDET_MODE - [1:0] */ +#define WM5100_ACCDET_MODE_WIDTH 2 /* ACCDET_MODE - [1:0] */ + +/* + * R648 (0x288) - Headphone Detect 1 + */ +#define WM5100_HP_HOLDTIME_MASK 0x00E0 /* HP_HOLDTIME - [7:5] */ +#define WM5100_HP_HOLDTIME_SHIFT 5 /* HP_HOLDTIME - [7:5] */ +#define WM5100_HP_HOLDTIME_WIDTH 3 /* HP_HOLDTIME - [7:5] */ +#define WM5100_HP_CLK_DIV_MASK 0x0018 /* HP_CLK_DIV - [4:3] */ +#define WM5100_HP_CLK_DIV_SHIFT 3 /* HP_CLK_DIV - [4:3] */ +#define WM5100_HP_CLK_DIV_WIDTH 2 /* HP_CLK_DIV - [4:3] */ +#define WM5100_HP_STEP_SIZE 0x0002 /* HP_STEP_SIZE */ +#define WM5100_HP_STEP_SIZE_MASK 0x0002 /* HP_STEP_SIZE */ +#define WM5100_HP_STEP_SIZE_SHIFT 1 /* HP_STEP_SIZE */ +#define WM5100_HP_STEP_SIZE_WIDTH 1 /* HP_STEP_SIZE */ +#define WM5100_HP_POLL 0x0001 /* HP_POLL */ +#define WM5100_HP_POLL_MASK 0x0001 /* HP_POLL */ +#define WM5100_HP_POLL_SHIFT 0 /* HP_POLL */ +#define WM5100_HP_POLL_WIDTH 1 /* HP_POLL */ + +/* + * R649 (0x289) - Headphone Detect 2 + */ +#define WM5100_HP_DONE 0x0080 /* HP_DONE */ +#define WM5100_HP_DONE_MASK 0x0080 /* HP_DONE */ +#define WM5100_HP_DONE_SHIFT 7 /* HP_DONE */ +#define WM5100_HP_DONE_WIDTH 1 /* HP_DONE */ +#define WM5100_HP_LVL_MASK 0x007F /* HP_LVL - [6:0] */ +#define WM5100_HP_LVL_SHIFT 0 /* HP_LVL - [6:0] */ +#define WM5100_HP_LVL_WIDTH 7 /* HP_LVL - [6:0] */ + +/* + * R656 (0x290) - Mic Detect 1 + */ +#define WM5100_ACCDET_BIAS_STARTTIME_MASK 0xF000 /* ACCDET_BIAS_STARTTIME - [15:12] */ +#define WM5100_ACCDET_BIAS_STARTTIME_SHIFT 12 /* ACCDET_BIAS_STARTTIME - [15:12] */ +#define WM5100_ACCDET_BIAS_STARTTIME_WIDTH 4 /* ACCDET_BIAS_STARTTIME - [15:12] */ +#define WM5100_ACCDET_RATE_MASK 0x0F00 /* ACCDET_RATE - [11:8] */ +#define WM5100_ACCDET_RATE_SHIFT 8 /* ACCDET_RATE - [11:8] */ +#define WM5100_ACCDET_RATE_WIDTH 4 /* ACCDET_RATE - [11:8] */ +#define WM5100_ACCDET_DBTIME 0x0002 /* ACCDET_DBTIME */ +#define WM5100_ACCDET_DBTIME_MASK 0x0002 /* ACCDET_DBTIME */ +#define WM5100_ACCDET_DBTIME_SHIFT 1 /* ACCDET_DBTIME */ +#define WM5100_ACCDET_DBTIME_WIDTH 1 /* ACCDET_DBTIME */ +#define WM5100_ACCDET_ENA 0x0001 /* ACCDET_ENA */ +#define WM5100_ACCDET_ENA_MASK 0x0001 /* ACCDET_ENA */ +#define WM5100_ACCDET_ENA_SHIFT 0 /* ACCDET_ENA */ +#define WM5100_ACCDET_ENA_WIDTH 1 /* ACCDET_ENA */ + +/* + * R657 (0x291) - Mic Detect 2 + */ +#define WM5100_ACCDET_LVL_SEL_MASK 0x00FF /* ACCDET_LVL_SEL - [7:0] */ +#define WM5100_ACCDET_LVL_SEL_SHIFT 0 /* ACCDET_LVL_SEL - [7:0] */ +#define WM5100_ACCDET_LVL_SEL_WIDTH 8 /* ACCDET_LVL_SEL - [7:0] */ + +/* + * R658 (0x292) - Mic Detect 3 + */ +#define WM5100_ACCDET_LVL_MASK 0x07FC /* ACCDET_LVL - [10:2] */ +#define WM5100_ACCDET_LVL_SHIFT 2 /* ACCDET_LVL - [10:2] */ +#define WM5100_ACCDET_LVL_WIDTH 9 /* ACCDET_LVL - [10:2] */ +#define WM5100_ACCDET_VALID 0x0002 /* ACCDET_VALID */ +#define WM5100_ACCDET_VALID_MASK 0x0002 /* ACCDET_VALID */ +#define WM5100_ACCDET_VALID_SHIFT 1 /* ACCDET_VALID */ +#define WM5100_ACCDET_VALID_WIDTH 1 /* ACCDET_VALID */ +#define WM5100_ACCDET_STS 0x0001 /* ACCDET_STS */ +#define WM5100_ACCDET_STS_MASK 0x0001 /* ACCDET_STS */ +#define WM5100_ACCDET_STS_SHIFT 0 /* ACCDET_STS */ +#define WM5100_ACCDET_STS_WIDTH 1 /* ACCDET_STS */ + +/* + * R699 (0x2BB) - Misc Control + */ +#define WM5100_HPCOM_SRC 0x200 /* HPCOM_SRC */ +#define WM5100_HPCOM_SRC_SHIFT 9 /* HPCOM_SRC */ + +/* + * R769 (0x301) - Input Enables + */ +#define WM5100_IN4L_ENA 0x0080 /* IN4L_ENA */ +#define WM5100_IN4L_ENA_MASK 0x0080 /* IN4L_ENA */ +#define WM5100_IN4L_ENA_SHIFT 7 /* IN4L_ENA */ +#define WM5100_IN4L_ENA_WIDTH 1 /* IN4L_ENA */ +#define WM5100_IN4R_ENA 0x0040 /* IN4R_ENA */ +#define WM5100_IN4R_ENA_MASK 0x0040 /* IN4R_ENA */ +#define WM5100_IN4R_ENA_SHIFT 6 /* IN4R_ENA */ +#define WM5100_IN4R_ENA_WIDTH 1 /* IN4R_ENA */ +#define WM5100_IN3L_ENA 0x0020 /* IN3L_ENA */ +#define WM5100_IN3L_ENA_MASK 0x0020 /* IN3L_ENA */ +#define WM5100_IN3L_ENA_SHIFT 5 /* IN3L_ENA */ +#define WM5100_IN3L_ENA_WIDTH 1 /* IN3L_ENA */ +#define WM5100_IN3R_ENA 0x0010 /* IN3R_ENA */ +#define WM5100_IN3R_ENA_MASK 0x0010 /* IN3R_ENA */ +#define WM5100_IN3R_ENA_SHIFT 4 /* IN3R_ENA */ +#define WM5100_IN3R_ENA_WIDTH 1 /* IN3R_ENA */ +#define WM5100_IN2L_ENA 0x0008 /* IN2L_ENA */ +#define WM5100_IN2L_ENA_MASK 0x0008 /* IN2L_ENA */ +#define WM5100_IN2L_ENA_SHIFT 3 /* IN2L_ENA */ +#define WM5100_IN2L_ENA_WIDTH 1 /* IN2L_ENA */ +#define WM5100_IN2R_ENA 0x0004 /* IN2R_ENA */ +#define WM5100_IN2R_ENA_MASK 0x0004 /* IN2R_ENA */ +#define WM5100_IN2R_ENA_SHIFT 2 /* IN2R_ENA */ +#define WM5100_IN2R_ENA_WIDTH 1 /* IN2R_ENA */ +#define WM5100_IN1L_ENA 0x0002 /* IN1L_ENA */ +#define WM5100_IN1L_ENA_MASK 0x0002 /* IN1L_ENA */ +#define WM5100_IN1L_ENA_SHIFT 1 /* IN1L_ENA */ +#define WM5100_IN1L_ENA_WIDTH 1 /* IN1L_ENA */ +#define WM5100_IN1R_ENA 0x0001 /* IN1R_ENA */ +#define WM5100_IN1R_ENA_MASK 0x0001 /* IN1R_ENA */ +#define WM5100_IN1R_ENA_SHIFT 0 /* IN1R_ENA */ +#define WM5100_IN1R_ENA_WIDTH 1 /* IN1R_ENA */ + +/* + * R770 (0x302) - Input Enables Status + */ +#define WM5100_IN4L_ENA_STS 0x0080 /* IN4L_ENA_STS */ +#define WM5100_IN4L_ENA_STS_MASK 0x0080 /* IN4L_ENA_STS */ +#define WM5100_IN4L_ENA_STS_SHIFT 7 /* IN4L_ENA_STS */ +#define WM5100_IN4L_ENA_STS_WIDTH 1 /* IN4L_ENA_STS */ +#define WM5100_IN4R_ENA_STS 0x0040 /* IN4R_ENA_STS */ +#define WM5100_IN4R_ENA_STS_MASK 0x0040 /* IN4R_ENA_STS */ +#define WM5100_IN4R_ENA_STS_SHIFT 6 /* IN4R_ENA_STS */ +#define WM5100_IN4R_ENA_STS_WIDTH 1 /* IN4R_ENA_STS */ +#define WM5100_IN3L_ENA_STS 0x0020 /* IN3L_ENA_STS */ +#define WM5100_IN3L_ENA_STS_MASK 0x0020 /* IN3L_ENA_STS */ +#define WM5100_IN3L_ENA_STS_SHIFT 5 /* IN3L_ENA_STS */ +#define WM5100_IN3L_ENA_STS_WIDTH 1 /* IN3L_ENA_STS */ +#define WM5100_IN3R_ENA_STS 0x0010 /* IN3R_ENA_STS */ +#define WM5100_IN3R_ENA_STS_MASK 0x0010 /* IN3R_ENA_STS */ +#define WM5100_IN3R_ENA_STS_SHIFT 4 /* IN3R_ENA_STS */ +#define WM5100_IN3R_ENA_STS_WIDTH 1 /* IN3R_ENA_STS */ +#define WM5100_IN2L_ENA_STS 0x0008 /* IN2L_ENA_STS */ +#define WM5100_IN2L_ENA_STS_MASK 0x0008 /* IN2L_ENA_STS */ +#define WM5100_IN2L_ENA_STS_SHIFT 3 /* IN2L_ENA_STS */ +#define WM5100_IN2L_ENA_STS_WIDTH 1 /* IN2L_ENA_STS */ +#define WM5100_IN2R_ENA_STS 0x0004 /* IN2R_ENA_STS */ +#define WM5100_IN2R_ENA_STS_MASK 0x0004 /* IN2R_ENA_STS */ +#define WM5100_IN2R_ENA_STS_SHIFT 2 /* IN2R_ENA_STS */ +#define WM5100_IN2R_ENA_STS_WIDTH 1 /* IN2R_ENA_STS */ +#define WM5100_IN1L_ENA_STS 0x0002 /* IN1L_ENA_STS */ +#define WM5100_IN1L_ENA_STS_MASK 0x0002 /* IN1L_ENA_STS */ +#define WM5100_IN1L_ENA_STS_SHIFT 1 /* IN1L_ENA_STS */ +#define WM5100_IN1L_ENA_STS_WIDTH 1 /* IN1L_ENA_STS */ +#define WM5100_IN1R_ENA_STS 0x0001 /* IN1R_ENA_STS */ +#define WM5100_IN1R_ENA_STS_MASK 0x0001 /* IN1R_ENA_STS */ +#define WM5100_IN1R_ENA_STS_SHIFT 0 /* IN1R_ENA_STS */ +#define WM5100_IN1R_ENA_STS_WIDTH 1 /* IN1R_ENA_STS */ + +/* + * R784 (0x310) - IN1L Control + */ +#define WM5100_IN_RATE_MASK 0xC000 /* IN_RATE - [15:14] */ +#define WM5100_IN_RATE_SHIFT 14 /* IN_RATE - [15:14] */ +#define WM5100_IN_RATE_WIDTH 2 /* IN_RATE - [15:14] */ +#define WM5100_IN1_OSR 0x2000 /* IN1_OSR */ +#define WM5100_IN1_OSR_MASK 0x2000 /* IN1_OSR */ +#define WM5100_IN1_OSR_SHIFT 13 /* IN1_OSR */ +#define WM5100_IN1_OSR_WIDTH 1 /* IN1_OSR */ +#define WM5100_IN1_DMIC_SUP_MASK 0x1800 /* IN1_DMIC_SUP - [12:11] */ +#define WM5100_IN1_DMIC_SUP_SHIFT 11 /* IN1_DMIC_SUP - [12:11] */ +#define WM5100_IN1_DMIC_SUP_WIDTH 2 /* IN1_DMIC_SUP - [12:11] */ +#define WM5100_IN1_MODE_MASK 0x0600 /* IN1_MODE - [10:9] */ +#define WM5100_IN1_MODE_SHIFT 9 /* IN1_MODE - [10:9] */ +#define WM5100_IN1_MODE_WIDTH 2 /* IN1_MODE - [10:9] */ +#define WM5100_IN1L_PGA_VOL_MASK 0x00FE /* IN1L_PGA_VOL - [7:1] */ +#define WM5100_IN1L_PGA_VOL_SHIFT 1 /* IN1L_PGA_VOL - [7:1] */ +#define WM5100_IN1L_PGA_VOL_WIDTH 7 /* IN1L_PGA_VOL - [7:1] */ + +/* + * R785 (0x311) - IN1R Control + */ +#define WM5100_IN1R_PGA_VOL_MASK 0x00FE /* IN1R_PGA_VOL - [7:1] */ +#define WM5100_IN1R_PGA_VOL_SHIFT 1 /* IN1R_PGA_VOL - [7:1] */ +#define WM5100_IN1R_PGA_VOL_WIDTH 7 /* IN1R_PGA_VOL - [7:1] */ + +/* + * R786 (0x312) - IN2L Control + */ +#define WM5100_IN2_OSR 0x2000 /* IN2_OSR */ +#define WM5100_IN2_OSR_MASK 0x2000 /* IN2_OSR */ +#define WM5100_IN2_OSR_SHIFT 13 /* IN2_OSR */ +#define WM5100_IN2_OSR_WIDTH 1 /* IN2_OSR */ +#define WM5100_IN2_DMIC_SUP_MASK 0x1800 /* IN2_DMIC_SUP - [12:11] */ +#define WM5100_IN2_DMIC_SUP_SHIFT 11 /* IN2_DMIC_SUP - [12:11] */ +#define WM5100_IN2_DMIC_SUP_WIDTH 2 /* IN2_DMIC_SUP - [12:11] */ +#define WM5100_IN2_MODE_MASK 0x0600 /* IN2_MODE - [10:9] */ +#define WM5100_IN2_MODE_SHIFT 9 /* IN2_MODE - [10:9] */ +#define WM5100_IN2_MODE_WIDTH 2 /* IN2_MODE - [10:9] */ +#define WM5100_IN2L_PGA_VOL_MASK 0x00FE /* IN2L_PGA_VOL - [7:1] */ +#define WM5100_IN2L_PGA_VOL_SHIFT 1 /* IN2L_PGA_VOL - [7:1] */ +#define WM5100_IN2L_PGA_VOL_WIDTH 7 /* IN2L_PGA_VOL - [7:1] */ + +/* + * R787 (0x313) - IN2R Control + */ +#define WM5100_IN2R_PGA_VOL_MASK 0x00FE /* IN2R_PGA_VOL - [7:1] */ +#define WM5100_IN2R_PGA_VOL_SHIFT 1 /* IN2R_PGA_VOL - [7:1] */ +#define WM5100_IN2R_PGA_VOL_WIDTH 7 /* IN2R_PGA_VOL - [7:1] */ + +/* + * R788 (0x314) - IN3L Control + */ +#define WM5100_IN3_OSR 0x2000 /* IN3_OSR */ +#define WM5100_IN3_OSR_MASK 0x2000 /* IN3_OSR */ +#define WM5100_IN3_OSR_SHIFT 13 /* IN3_OSR */ +#define WM5100_IN3_OSR_WIDTH 1 /* IN3_OSR */ +#define WM5100_IN3_DMIC_SUP_MASK 0x1800 /* IN3_DMIC_SUP - [12:11] */ +#define WM5100_IN3_DMIC_SUP_SHIFT 11 /* IN3_DMIC_SUP - [12:11] */ +#define WM5100_IN3_DMIC_SUP_WIDTH 2 /* IN3_DMIC_SUP - [12:11] */ +#define WM5100_IN3_MODE_MASK 0x0600 /* IN3_MODE - [10:9] */ +#define WM5100_IN3_MODE_SHIFT 9 /* IN3_MODE - [10:9] */ +#define WM5100_IN3_MODE_WIDTH 2 /* IN3_MODE - [10:9] */ +#define WM5100_IN3L_PGA_VOL_MASK 0x00FE /* IN3L_PGA_VOL - [7:1] */ +#define WM5100_IN3L_PGA_VOL_SHIFT 1 /* IN3L_PGA_VOL - [7:1] */ +#define WM5100_IN3L_PGA_VOL_WIDTH 7 /* IN3L_PGA_VOL - [7:1] */ + +/* + * R789 (0x315) - IN3R Control + */ +#define WM5100_IN3R_PGA_VOL_MASK 0x00FE /* IN3R_PGA_VOL - [7:1] */ +#define WM5100_IN3R_PGA_VOL_SHIFT 1 /* IN3R_PGA_VOL - [7:1] */ +#define WM5100_IN3R_PGA_VOL_WIDTH 7 /* IN3R_PGA_VOL - [7:1] */ + +/* + * R790 (0x316) - IN4L Control + */ +#define WM5100_IN4_OSR 0x2000 /* IN4_OSR */ +#define WM5100_IN4_OSR_MASK 0x2000 /* IN4_OSR */ +#define WM5100_IN4_OSR_SHIFT 13 /* IN4_OSR */ +#define WM5100_IN4_OSR_WIDTH 1 /* IN4_OSR */ +#define WM5100_IN4_DMIC_SUP_MASK 0x1800 /* IN4_DMIC_SUP - [12:11] */ +#define WM5100_IN4_DMIC_SUP_SHIFT 11 /* IN4_DMIC_SUP - [12:11] */ +#define WM5100_IN4_DMIC_SUP_WIDTH 2 /* IN4_DMIC_SUP - [12:11] */ +#define WM5100_IN4_MODE_MASK 0x0600 /* IN4_MODE - [10:9] */ +#define WM5100_IN4_MODE_SHIFT 9 /* IN4_MODE - [10:9] */ +#define WM5100_IN4_MODE_WIDTH 2 /* IN4_MODE - [10:9] */ +#define WM5100_IN4L_PGA_VOL_MASK 0x00FE /* IN4L_PGA_VOL - [7:1] */ +#define WM5100_IN4L_PGA_VOL_SHIFT 1 /* IN4L_PGA_VOL - [7:1] */ +#define WM5100_IN4L_PGA_VOL_WIDTH 7 /* IN4L_PGA_VOL - [7:1] */ + +/* + * R791 (0x317) - IN4R Control + */ +#define WM5100_IN4R_PGA_VOL_MASK 0x00FE /* IN4R_PGA_VOL - [7:1] */ +#define WM5100_IN4R_PGA_VOL_SHIFT 1 /* IN4R_PGA_VOL - [7:1] */ +#define WM5100_IN4R_PGA_VOL_WIDTH 7 /* IN4R_PGA_VOL - [7:1] */ + +/* + * R792 (0x318) - RXANC_SRC + */ +#define WM5100_IN_RXANC_SEL_MASK 0x0007 /* IN_RXANC_SEL - [2:0] */ +#define WM5100_IN_RXANC_SEL_SHIFT 0 /* IN_RXANC_SEL - [2:0] */ +#define WM5100_IN_RXANC_SEL_WIDTH 3 /* IN_RXANC_SEL - [2:0] */ + +/* + * R793 (0x319) - Input Volume Ramp + */ +#define WM5100_IN_VD_RAMP_MASK 0x0070 /* IN_VD_RAMP - [6:4] */ +#define WM5100_IN_VD_RAMP_SHIFT 4 /* IN_VD_RAMP - [6:4] */ +#define WM5100_IN_VD_RAMP_WIDTH 3 /* IN_VD_RAMP - [6:4] */ +#define WM5100_IN_VI_RAMP_MASK 0x0007 /* IN_VI_RAMP - [2:0] */ +#define WM5100_IN_VI_RAMP_SHIFT 0 /* IN_VI_RAMP - [2:0] */ +#define WM5100_IN_VI_RAMP_WIDTH 3 /* IN_VI_RAMP - [2:0] */ + +/* + * R800 (0x320) - ADC Digital Volume 1L + */ +#define WM5100_IN_VU 0x0200 /* IN_VU */ +#define WM5100_IN_VU_MASK 0x0200 /* IN_VU */ +#define WM5100_IN_VU_SHIFT 9 /* IN_VU */ +#define WM5100_IN_VU_WIDTH 1 /* IN_VU */ +#define WM5100_IN1L_MUTE 0x0100 /* IN1L_MUTE */ +#define WM5100_IN1L_MUTE_MASK 0x0100 /* IN1L_MUTE */ +#define WM5100_IN1L_MUTE_SHIFT 8 /* IN1L_MUTE */ +#define WM5100_IN1L_MUTE_WIDTH 1 /* IN1L_MUTE */ +#define WM5100_IN1L_VOL_MASK 0x00FF /* IN1L_VOL - [7:0] */ +#define WM5100_IN1L_VOL_SHIFT 0 /* IN1L_VOL - [7:0] */ +#define WM5100_IN1L_VOL_WIDTH 8 /* IN1L_VOL - [7:0] */ + +/* + * R801 (0x321) - ADC Digital Volume 1R + */ +#define WM5100_IN_VU 0x0200 /* IN_VU */ +#define WM5100_IN_VU_MASK 0x0200 /* IN_VU */ +#define WM5100_IN_VU_SHIFT 9 /* IN_VU */ +#define WM5100_IN_VU_WIDTH 1 /* IN_VU */ +#define WM5100_IN1R_MUTE 0x0100 /* IN1R_MUTE */ +#define WM5100_IN1R_MUTE_MASK 0x0100 /* IN1R_MUTE */ +#define WM5100_IN1R_MUTE_SHIFT 8 /* IN1R_MUTE */ +#define WM5100_IN1R_MUTE_WIDTH 1 /* IN1R_MUTE */ +#define WM5100_IN1R_VOL_MASK 0x00FF /* IN1R_VOL - [7:0] */ +#define WM5100_IN1R_VOL_SHIFT 0 /* IN1R_VOL - [7:0] */ +#define WM5100_IN1R_VOL_WIDTH 8 /* IN1R_VOL - [7:0] */ + +/* + * R802 (0x322) - ADC Digital Volume 2L + */ +#define WM5100_IN_VU 0x0200 /* IN_VU */ +#define WM5100_IN_VU_MASK 0x0200 /* IN_VU */ +#define WM5100_IN_VU_SHIFT 9 /* IN_VU */ +#define WM5100_IN_VU_WIDTH 1 /* IN_VU */ +#define WM5100_IN2L_MUTE 0x0100 /* IN2L_MUTE */ +#define WM5100_IN2L_MUTE_MASK 0x0100 /* IN2L_MUTE */ +#define WM5100_IN2L_MUTE_SHIFT 8 /* IN2L_MUTE */ +#define WM5100_IN2L_MUTE_WIDTH 1 /* IN2L_MUTE */ +#define WM5100_IN2L_VOL_MASK 0x00FF /* IN2L_VOL - [7:0] */ +#define WM5100_IN2L_VOL_SHIFT 0 /* IN2L_VOL - [7:0] */ +#define WM5100_IN2L_VOL_WIDTH 8 /* IN2L_VOL - [7:0] */ + +/* + * R803 (0x323) - ADC Digital Volume 2R + */ +#define WM5100_IN_VU 0x0200 /* IN_VU */ +#define WM5100_IN_VU_MASK 0x0200 /* IN_VU */ +#define WM5100_IN_VU_SHIFT 9 /* IN_VU */ +#define WM5100_IN_VU_WIDTH 1 /* IN_VU */ +#define WM5100_IN2R_MUTE 0x0100 /* IN2R_MUTE */ +#define WM5100_IN2R_MUTE_MASK 0x0100 /* IN2R_MUTE */ +#define WM5100_IN2R_MUTE_SHIFT 8 /* IN2R_MUTE */ +#define WM5100_IN2R_MUTE_WIDTH 1 /* IN2R_MUTE */ +#define WM5100_IN2R_VOL_MASK 0x00FF /* IN2R_VOL - [7:0] */ +#define WM5100_IN2R_VOL_SHIFT 0 /* IN2R_VOL - [7:0] */ +#define WM5100_IN2R_VOL_WIDTH 8 /* IN2R_VOL - [7:0] */ + +/* + * R804 (0x324) - ADC Digital Volume 3L + */ +#define WM5100_IN_VU 0x0200 /* IN_VU */ +#define WM5100_IN_VU_MASK 0x0200 /* IN_VU */ +#define WM5100_IN_VU_SHIFT 9 /* IN_VU */ +#define WM5100_IN_VU_WIDTH 1 /* IN_VU */ +#define WM5100_IN3L_MUTE 0x0100 /* IN3L_MUTE */ +#define WM5100_IN3L_MUTE_MASK 0x0100 /* IN3L_MUTE */ +#define WM5100_IN3L_MUTE_SHIFT 8 /* IN3L_MUTE */ +#define WM5100_IN3L_MUTE_WIDTH 1 /* IN3L_MUTE */ +#define WM5100_IN3L_VOL_MASK 0x00FF /* IN3L_VOL - [7:0] */ +#define WM5100_IN3L_VOL_SHIFT 0 /* IN3L_VOL - [7:0] */ +#define WM5100_IN3L_VOL_WIDTH 8 /* IN3L_VOL - [7:0] */ + +/* + * R805 (0x325) - ADC Digital Volume 3R + */ +#define WM5100_IN_VU 0x0200 /* IN_VU */ +#define WM5100_IN_VU_MASK 0x0200 /* IN_VU */ +#define WM5100_IN_VU_SHIFT 9 /* IN_VU */ +#define WM5100_IN_VU_WIDTH 1 /* IN_VU */ +#define WM5100_IN3R_MUTE 0x0100 /* IN3R_MUTE */ +#define WM5100_IN3R_MUTE_MASK 0x0100 /* IN3R_MUTE */ +#define WM5100_IN3R_MUTE_SHIFT 8 /* IN3R_MUTE */ +#define WM5100_IN3R_MUTE_WIDTH 1 /* IN3R_MUTE */ +#define WM5100_IN3R_VOL_MASK 0x00FF /* IN3R_VOL - [7:0] */ +#define WM5100_IN3R_VOL_SHIFT 0 /* IN3R_VOL - [7:0] */ +#define WM5100_IN3R_VOL_WIDTH 8 /* IN3R_VOL - [7:0] */ + +/* + * R806 (0x326) - ADC Digital Volume 4L + */ +#define WM5100_IN_VU 0x0200 /* IN_VU */ +#define WM5100_IN_VU_MASK 0x0200 /* IN_VU */ +#define WM5100_IN_VU_SHIFT 9 /* IN_VU */ +#define WM5100_IN_VU_WIDTH 1 /* IN_VU */ +#define WM5100_IN4L_MUTE 0x0100 /* IN4L_MUTE */ +#define WM5100_IN4L_MUTE_MASK 0x0100 /* IN4L_MUTE */ +#define WM5100_IN4L_MUTE_SHIFT 8 /* IN4L_MUTE */ +#define WM5100_IN4L_MUTE_WIDTH 1 /* IN4L_MUTE */ +#define WM5100_IN4L_VOL_MASK 0x00FF /* IN4L_VOL - [7:0] */ +#define WM5100_IN4L_VOL_SHIFT 0 /* IN4L_VOL - [7:0] */ +#define WM5100_IN4L_VOL_WIDTH 8 /* IN4L_VOL - [7:0] */ + +/* + * R807 (0x327) - ADC Digital Volume 4R + */ +#define WM5100_IN_VU 0x0200 /* IN_VU */ +#define WM5100_IN_VU_MASK 0x0200 /* IN_VU */ +#define WM5100_IN_VU_SHIFT 9 /* IN_VU */ +#define WM5100_IN_VU_WIDTH 1 /* IN_VU */ +#define WM5100_IN4R_MUTE 0x0100 /* IN4R_MUTE */ +#define WM5100_IN4R_MUTE_MASK 0x0100 /* IN4R_MUTE */ +#define WM5100_IN4R_MUTE_SHIFT 8 /* IN4R_MUTE */ +#define WM5100_IN4R_MUTE_WIDTH 1 /* IN4R_MUTE */ +#define WM5100_IN4R_VOL_MASK 0x00FF /* IN4R_VOL - [7:0] */ +#define WM5100_IN4R_VOL_SHIFT 0 /* IN4R_VOL - [7:0] */ +#define WM5100_IN4R_VOL_WIDTH 8 /* IN4R_VOL - [7:0] */ + +/* + * R1025 (0x401) - Output Enables 2 + */ +#define WM5100_OUT6L_ENA 0x0800 /* OUT6L_ENA */ +#define WM5100_OUT6L_ENA_MASK 0x0800 /* OUT6L_ENA */ +#define WM5100_OUT6L_ENA_SHIFT 11 /* OUT6L_ENA */ +#define WM5100_OUT6L_ENA_WIDTH 1 /* OUT6L_ENA */ +#define WM5100_OUT6R_ENA 0x0400 /* OUT6R_ENA */ +#define WM5100_OUT6R_ENA_MASK 0x0400 /* OUT6R_ENA */ +#define WM5100_OUT6R_ENA_SHIFT 10 /* OUT6R_ENA */ +#define WM5100_OUT6R_ENA_WIDTH 1 /* OUT6R_ENA */ +#define WM5100_OUT5L_ENA 0x0200 /* OUT5L_ENA */ +#define WM5100_OUT5L_ENA_MASK 0x0200 /* OUT5L_ENA */ +#define WM5100_OUT5L_ENA_SHIFT 9 /* OUT5L_ENA */ +#define WM5100_OUT5L_ENA_WIDTH 1 /* OUT5L_ENA */ +#define WM5100_OUT5R_ENA 0x0100 /* OUT5R_ENA */ +#define WM5100_OUT5R_ENA_MASK 0x0100 /* OUT5R_ENA */ +#define WM5100_OUT5R_ENA_SHIFT 8 /* OUT5R_ENA */ +#define WM5100_OUT5R_ENA_WIDTH 1 /* OUT5R_ENA */ +#define WM5100_OUT4L_ENA 0x0080 /* OUT4L_ENA */ +#define WM5100_OUT4L_ENA_MASK 0x0080 /* OUT4L_ENA */ +#define WM5100_OUT4L_ENA_SHIFT 7 /* OUT4L_ENA */ +#define WM5100_OUT4L_ENA_WIDTH 1 /* OUT4L_ENA */ +#define WM5100_OUT4R_ENA 0x0040 /* OUT4R_ENA */ +#define WM5100_OUT4R_ENA_MASK 0x0040 /* OUT4R_ENA */ +#define WM5100_OUT4R_ENA_SHIFT 6 /* OUT4R_ENA */ +#define WM5100_OUT4R_ENA_WIDTH 1 /* OUT4R_ENA */ + +/* + * R1026 (0x402) - Output Status 1 + */ +#define WM5100_OUT3L_ENA_STS 0x0020 /* OUT3L_ENA_STS */ +#define WM5100_OUT3L_ENA_STS_MASK 0x0020 /* OUT3L_ENA_STS */ +#define WM5100_OUT3L_ENA_STS_SHIFT 5 /* OUT3L_ENA_STS */ +#define WM5100_OUT3L_ENA_STS_WIDTH 1 /* OUT3L_ENA_STS */ +#define WM5100_OUT3R_ENA_STS 0x0010 /* OUT3R_ENA_STS */ +#define WM5100_OUT3R_ENA_STS_MASK 0x0010 /* OUT3R_ENA_STS */ +#define WM5100_OUT3R_ENA_STS_SHIFT 4 /* OUT3R_ENA_STS */ +#define WM5100_OUT3R_ENA_STS_WIDTH 1 /* OUT3R_ENA_STS */ +#define WM5100_OUT2L_ENA_STS 0x0008 /* OUT2L_ENA_STS */ +#define WM5100_OUT2L_ENA_STS_MASK 0x0008 /* OUT2L_ENA_STS */ +#define WM5100_OUT2L_ENA_STS_SHIFT 3 /* OUT2L_ENA_STS */ +#define WM5100_OUT2L_ENA_STS_WIDTH 1 /* OUT2L_ENA_STS */ +#define WM5100_OUT2R_ENA_STS 0x0004 /* OUT2R_ENA_STS */ +#define WM5100_OUT2R_ENA_STS_MASK 0x0004 /* OUT2R_ENA_STS */ +#define WM5100_OUT2R_ENA_STS_SHIFT 2 /* OUT2R_ENA_STS */ +#define WM5100_OUT2R_ENA_STS_WIDTH 1 /* OUT2R_ENA_STS */ +#define WM5100_OUT1L_ENA_STS 0x0002 /* OUT1L_ENA_STS */ +#define WM5100_OUT1L_ENA_STS_MASK 0x0002 /* OUT1L_ENA_STS */ +#define WM5100_OUT1L_ENA_STS_SHIFT 1 /* OUT1L_ENA_STS */ +#define WM5100_OUT1L_ENA_STS_WIDTH 1 /* OUT1L_ENA_STS */ +#define WM5100_OUT1R_ENA_STS 0x0001 /* OUT1R_ENA_STS */ +#define WM5100_OUT1R_ENA_STS_MASK 0x0001 /* OUT1R_ENA_STS */ +#define WM5100_OUT1R_ENA_STS_SHIFT 0 /* OUT1R_ENA_STS */ +#define WM5100_OUT1R_ENA_STS_WIDTH 1 /* OUT1R_ENA_STS */ + +/* + * R1027 (0x403) - Output Status 2 + */ +#define WM5100_OUT6L_ENA_STS 0x0800 /* OUT6L_ENA_STS */ +#define WM5100_OUT6L_ENA_STS_MASK 0x0800 /* OUT6L_ENA_STS */ +#define WM5100_OUT6L_ENA_STS_SHIFT 11 /* OUT6L_ENA_STS */ +#define WM5100_OUT6L_ENA_STS_WIDTH 1 /* OUT6L_ENA_STS */ +#define WM5100_OUT6R_ENA_STS 0x0400 /* OUT6R_ENA_STS */ +#define WM5100_OUT6R_ENA_STS_MASK 0x0400 /* OUT6R_ENA_STS */ +#define WM5100_OUT6R_ENA_STS_SHIFT 10 /* OUT6R_ENA_STS */ +#define WM5100_OUT6R_ENA_STS_WIDTH 1 /* OUT6R_ENA_STS */ +#define WM5100_OUT5L_ENA_STS 0x0200 /* OUT5L_ENA_STS */ +#define WM5100_OUT5L_ENA_STS_MASK 0x0200 /* OUT5L_ENA_STS */ +#define WM5100_OUT5L_ENA_STS_SHIFT 9 /* OUT5L_ENA_STS */ +#define WM5100_OUT5L_ENA_STS_WIDTH 1 /* OUT5L_ENA_STS */ +#define WM5100_OUT5R_ENA_STS 0x0100 /* OUT5R_ENA_STS */ +#define WM5100_OUT5R_ENA_STS_MASK 0x0100 /* OUT5R_ENA_STS */ +#define WM5100_OUT5R_ENA_STS_SHIFT 8 /* OUT5R_ENA_STS */ +#define WM5100_OUT5R_ENA_STS_WIDTH 1 /* OUT5R_ENA_STS */ +#define WM5100_OUT4L_ENA_STS 0x0080 /* OUT4L_ENA_STS */ +#define WM5100_OUT4L_ENA_STS_MASK 0x0080 /* OUT4L_ENA_STS */ +#define WM5100_OUT4L_ENA_STS_SHIFT 7 /* OUT4L_ENA_STS */ +#define WM5100_OUT4L_ENA_STS_WIDTH 1 /* OUT4L_ENA_STS */ +#define WM5100_OUT4R_ENA_STS 0x0040 /* OUT4R_ENA_STS */ +#define WM5100_OUT4R_ENA_STS_MASK 0x0040 /* OUT4R_ENA_STS */ +#define WM5100_OUT4R_ENA_STS_SHIFT 6 /* OUT4R_ENA_STS */ +#define WM5100_OUT4R_ENA_STS_WIDTH 1 /* OUT4R_ENA_STS */ + +/* + * R1032 (0x408) - Channel Enables 1 + */ +#define WM5100_HP3L_ENA 0x0020 /* HP3L_ENA */ +#define WM5100_HP3L_ENA_MASK 0x0020 /* HP3L_ENA */ +#define WM5100_HP3L_ENA_SHIFT 5 /* HP3L_ENA */ +#define WM5100_HP3L_ENA_WIDTH 1 /* HP3L_ENA */ +#define WM5100_HP3R_ENA 0x0010 /* HP3R_ENA */ +#define WM5100_HP3R_ENA_MASK 0x0010 /* HP3R_ENA */ +#define WM5100_HP3R_ENA_SHIFT 4 /* HP3R_ENA */ +#define WM5100_HP3R_ENA_WIDTH 1 /* HP3R_ENA */ +#define WM5100_HP2L_ENA 0x0008 /* HP2L_ENA */ +#define WM5100_HP2L_ENA_MASK 0x0008 /* HP2L_ENA */ +#define WM5100_HP2L_ENA_SHIFT 3 /* HP2L_ENA */ +#define WM5100_HP2L_ENA_WIDTH 1 /* HP2L_ENA */ +#define WM5100_HP2R_ENA 0x0004 /* HP2R_ENA */ +#define WM5100_HP2R_ENA_MASK 0x0004 /* HP2R_ENA */ +#define WM5100_HP2R_ENA_SHIFT 2 /* HP2R_ENA */ +#define WM5100_HP2R_ENA_WIDTH 1 /* HP2R_ENA */ +#define WM5100_HP1L_ENA 0x0002 /* HP1L_ENA */ +#define WM5100_HP1L_ENA_MASK 0x0002 /* HP1L_ENA */ +#define WM5100_HP1L_ENA_SHIFT 1 /* HP1L_ENA */ +#define WM5100_HP1L_ENA_WIDTH 1 /* HP1L_ENA */ +#define WM5100_HP1R_ENA 0x0001 /* HP1R_ENA */ +#define WM5100_HP1R_ENA_MASK 0x0001 /* HP1R_ENA */ +#define WM5100_HP1R_ENA_SHIFT 0 /* HP1R_ENA */ +#define WM5100_HP1R_ENA_WIDTH 1 /* HP1R_ENA */ + +/* + * R1040 (0x410) - Out Volume 1L + */ +#define WM5100_OUT_RATE_MASK 0xC000 /* OUT_RATE - [15:14] */ +#define WM5100_OUT_RATE_SHIFT 14 /* OUT_RATE - [15:14] */ +#define WM5100_OUT_RATE_WIDTH 2 /* OUT_RATE - [15:14] */ +#define WM5100_OUT1_OSR 0x2000 /* OUT1_OSR */ +#define WM5100_OUT1_OSR_MASK 0x2000 /* OUT1_OSR */ +#define WM5100_OUT1_OSR_SHIFT 13 /* OUT1_OSR */ +#define WM5100_OUT1_OSR_WIDTH 1 /* OUT1_OSR */ +#define WM5100_OUT1_MONO 0x1000 /* OUT1_MONO */ +#define WM5100_OUT1_MONO_MASK 0x1000 /* OUT1_MONO */ +#define WM5100_OUT1_MONO_SHIFT 12 /* OUT1_MONO */ +#define WM5100_OUT1_MONO_WIDTH 1 /* OUT1_MONO */ +#define WM5100_OUT1L_ANC_SRC 0x0800 /* OUT1L_ANC_SRC */ +#define WM5100_OUT1L_ANC_SRC_MASK 0x0800 /* OUT1L_ANC_SRC */ +#define WM5100_OUT1L_ANC_SRC_SHIFT 11 /* OUT1L_ANC_SRC */ +#define WM5100_OUT1L_ANC_SRC_WIDTH 1 /* OUT1L_ANC_SRC */ +#define WM5100_OUT1L_PGA_VOL_MASK 0x00FE /* OUT1L_PGA_VOL - [7:1] */ +#define WM5100_OUT1L_PGA_VOL_SHIFT 1 /* OUT1L_PGA_VOL - [7:1] */ +#define WM5100_OUT1L_PGA_VOL_WIDTH 7 /* OUT1L_PGA_VOL - [7:1] */ + +/* + * R1041 (0x411) - Out Volume 1R + */ +#define WM5100_OUT1R_ANC_SRC 0x0800 /* OUT1R_ANC_SRC */ +#define WM5100_OUT1R_ANC_SRC_MASK 0x0800 /* OUT1R_ANC_SRC */ +#define WM5100_OUT1R_ANC_SRC_SHIFT 11 /* OUT1R_ANC_SRC */ +#define WM5100_OUT1R_ANC_SRC_WIDTH 1 /* OUT1R_ANC_SRC */ +#define WM5100_OUT1R_PGA_VOL_MASK 0x00FE /* OUT1R_PGA_VOL - [7:1] */ +#define WM5100_OUT1R_PGA_VOL_SHIFT 1 /* OUT1R_PGA_VOL - [7:1] */ +#define WM5100_OUT1R_PGA_VOL_WIDTH 7 /* OUT1R_PGA_VOL - [7:1] */ + +/* + * R1042 (0x412) - DAC Volume Limit 1L + */ +#define WM5100_OUT1L_VOL_LIM_MASK 0x00FF /* OUT1L_VOL_LIM - [7:0] */ +#define WM5100_OUT1L_VOL_LIM_SHIFT 0 /* OUT1L_VOL_LIM - [7:0] */ +#define WM5100_OUT1L_VOL_LIM_WIDTH 8 /* OUT1L_VOL_LIM - [7:0] */ + +/* + * R1043 (0x413) - DAC Volume Limit 1R + */ +#define WM5100_OUT1R_VOL_LIM_MASK 0x00FF /* OUT1R_VOL_LIM - [7:0] */ +#define WM5100_OUT1R_VOL_LIM_SHIFT 0 /* OUT1R_VOL_LIM - [7:0] */ +#define WM5100_OUT1R_VOL_LIM_WIDTH 8 /* OUT1R_VOL_LIM - [7:0] */ + +/* + * R1044 (0x414) - Out Volume 2L + */ +#define WM5100_OUT2_OSR 0x2000 /* OUT2_OSR */ +#define WM5100_OUT2_OSR_MASK 0x2000 /* OUT2_OSR */ +#define WM5100_OUT2_OSR_SHIFT 13 /* OUT2_OSR */ +#define WM5100_OUT2_OSR_WIDTH 1 /* OUT2_OSR */ +#define WM5100_OUT2_MONO 0x1000 /* OUT2_MONO */ +#define WM5100_OUT2_MONO_MASK 0x1000 /* OUT2_MONO */ +#define WM5100_OUT2_MONO_SHIFT 12 /* OUT2_MONO */ +#define WM5100_OUT2_MONO_WIDTH 1 /* OUT2_MONO */ +#define WM5100_OUT2L_ANC_SRC 0x0800 /* OUT2L_ANC_SRC */ +#define WM5100_OUT2L_ANC_SRC_MASK 0x0800 /* OUT2L_ANC_SRC */ +#define WM5100_OUT2L_ANC_SRC_SHIFT 11 /* OUT2L_ANC_SRC */ +#define WM5100_OUT2L_ANC_SRC_WIDTH 1 /* OUT2L_ANC_SRC */ +#define WM5100_OUT2L_PGA_VOL_MASK 0x00FE /* OUT2L_PGA_VOL - [7:1] */ +#define WM5100_OUT2L_PGA_VOL_SHIFT 1 /* OUT2L_PGA_VOL - [7:1] */ +#define WM5100_OUT2L_PGA_VOL_WIDTH 7 /* OUT2L_PGA_VOL - [7:1] */ + +/* + * R1045 (0x415) - Out Volume 2R + */ +#define WM5100_OUT2R_ANC_SRC 0x0800 /* OUT2R_ANC_SRC */ +#define WM5100_OUT2R_ANC_SRC_MASK 0x0800 /* OUT2R_ANC_SRC */ +#define WM5100_OUT2R_ANC_SRC_SHIFT 11 /* OUT2R_ANC_SRC */ +#define WM5100_OUT2R_ANC_SRC_WIDTH 1 /* OUT2R_ANC_SRC */ +#define WM5100_OUT2R_PGA_VOL_MASK 0x00FE /* OUT2R_PGA_VOL - [7:1] */ +#define WM5100_OUT2R_PGA_VOL_SHIFT 1 /* OUT2R_PGA_VOL - [7:1] */ +#define WM5100_OUT2R_PGA_VOL_WIDTH 7 /* OUT2R_PGA_VOL - [7:1] */ + +/* + * R1046 (0x416) - DAC Volume Limit 2L + */ +#define WM5100_OUT2L_VOL_LIM_MASK 0x00FF /* OUT2L_VOL_LIM - [7:0] */ +#define WM5100_OUT2L_VOL_LIM_SHIFT 0 /* OUT2L_VOL_LIM - [7:0] */ +#define WM5100_OUT2L_VOL_LIM_WIDTH 8 /* OUT2L_VOL_LIM - [7:0] */ + +/* + * R1047 (0x417) - DAC Volume Limit 2R + */ +#define WM5100_OUT2R_VOL_LIM_MASK 0x00FF /* OUT2R_VOL_LIM - [7:0] */ +#define WM5100_OUT2R_VOL_LIM_SHIFT 0 /* OUT2R_VOL_LIM - [7:0] */ +#define WM5100_OUT2R_VOL_LIM_WIDTH 8 /* OUT2R_VOL_LIM - [7:0] */ + +/* + * R1048 (0x418) - Out Volume 3L + */ +#define WM5100_OUT3_OSR 0x2000 /* OUT3_OSR */ +#define WM5100_OUT3_OSR_MASK 0x2000 /* OUT3_OSR */ +#define WM5100_OUT3_OSR_SHIFT 13 /* OUT3_OSR */ +#define WM5100_OUT3_OSR_WIDTH 1 /* OUT3_OSR */ +#define WM5100_OUT3_MONO 0x1000 /* OUT3_MONO */ +#define WM5100_OUT3_MONO_MASK 0x1000 /* OUT3_MONO */ +#define WM5100_OUT3_MONO_SHIFT 12 /* OUT3_MONO */ +#define WM5100_OUT3_MONO_WIDTH 1 /* OUT3_MONO */ +#define WM5100_OUT3L_ANC_SRC 0x0800 /* OUT3L_ANC_SRC */ +#define WM5100_OUT3L_ANC_SRC_MASK 0x0800 /* OUT3L_ANC_SRC */ +#define WM5100_OUT3L_ANC_SRC_SHIFT 11 /* OUT3L_ANC_SRC */ +#define WM5100_OUT3L_ANC_SRC_WIDTH 1 /* OUT3L_ANC_SRC */ +#define WM5100_OUT3L_PGA_VOL_MASK 0x00FE /* OUT3L_PGA_VOL - [7:1] */ +#define WM5100_OUT3L_PGA_VOL_SHIFT 1 /* OUT3L_PGA_VOL - [7:1] */ +#define WM5100_OUT3L_PGA_VOL_WIDTH 7 /* OUT3L_PGA_VOL - [7:1] */ + +/* + * R1049 (0x419) - Out Volume 3R + */ +#define WM5100_OUT3R_ANC_SRC 0x0800 /* OUT3R_ANC_SRC */ +#define WM5100_OUT3R_ANC_SRC_MASK 0x0800 /* OUT3R_ANC_SRC */ +#define WM5100_OUT3R_ANC_SRC_SHIFT 11 /* OUT3R_ANC_SRC */ +#define WM5100_OUT3R_ANC_SRC_WIDTH 1 /* OUT3R_ANC_SRC */ +#define WM5100_OUT3R_PGA_VOL_MASK 0x00FE /* OUT3R_PGA_VOL - [7:1] */ +#define WM5100_OUT3R_PGA_VOL_SHIFT 1 /* OUT3R_PGA_VOL - [7:1] */ +#define WM5100_OUT3R_PGA_VOL_WIDTH 7 /* OUT3R_PGA_VOL - [7:1] */ + +/* + * R1050 (0x41A) - DAC Volume Limit 3L + */ +#define WM5100_OUT3L_VOL_LIM_MASK 0x00FF /* OUT3L_VOL_LIM - [7:0] */ +#define WM5100_OUT3L_VOL_LIM_SHIFT 0 /* OUT3L_VOL_LIM - [7:0] */ +#define WM5100_OUT3L_VOL_LIM_WIDTH 8 /* OUT3L_VOL_LIM - [7:0] */ + +/* + * R1051 (0x41B) - DAC Volume Limit 3R + */ +#define WM5100_OUT3R_VOL_LIM_MASK 0x00FF /* OUT3R_VOL_LIM - [7:0] */ +#define WM5100_OUT3R_VOL_LIM_SHIFT 0 /* OUT3R_VOL_LIM - [7:0] */ +#define WM5100_OUT3R_VOL_LIM_WIDTH 8 /* OUT3R_VOL_LIM - [7:0] */ + +/* + * R1052 (0x41C) - Out Volume 4L + */ +#define WM5100_OUT4_OSR 0x2000 /* OUT4_OSR */ +#define WM5100_OUT4_OSR_MASK 0x2000 /* OUT4_OSR */ +#define WM5100_OUT4_OSR_SHIFT 13 /* OUT4_OSR */ +#define WM5100_OUT4_OSR_WIDTH 1 /* OUT4_OSR */ +#define WM5100_OUT4L_ANC_SRC 0x0800 /* OUT4L_ANC_SRC */ +#define WM5100_OUT4L_ANC_SRC_MASK 0x0800 /* OUT4L_ANC_SRC */ +#define WM5100_OUT4L_ANC_SRC_SHIFT 11 /* OUT4L_ANC_SRC */ +#define WM5100_OUT4L_ANC_SRC_WIDTH 1 /* OUT4L_ANC_SRC */ +#define WM5100_OUT4L_VOL_LIM_MASK 0x00FF /* OUT4L_VOL_LIM - [7:0] */ +#define WM5100_OUT4L_VOL_LIM_SHIFT 0 /* OUT4L_VOL_LIM - [7:0] */ +#define WM5100_OUT4L_VOL_LIM_WIDTH 8 /* OUT4L_VOL_LIM - [7:0] */ + +/* + * R1053 (0x41D) - Out Volume 4R + */ +#define WM5100_OUT4R_ANC_SRC 0x0800 /* OUT4R_ANC_SRC */ +#define WM5100_OUT4R_ANC_SRC_MASK 0x0800 /* OUT4R_ANC_SRC */ +#define WM5100_OUT4R_ANC_SRC_SHIFT 11 /* OUT4R_ANC_SRC */ +#define WM5100_OUT4R_ANC_SRC_WIDTH 1 /* OUT4R_ANC_SRC */ +#define WM5100_OUT4R_VOL_LIM_MASK 0x00FF /* OUT4R_VOL_LIM - [7:0] */ +#define WM5100_OUT4R_VOL_LIM_SHIFT 0 /* OUT4R_VOL_LIM - [7:0] */ +#define WM5100_OUT4R_VOL_LIM_WIDTH 8 /* OUT4R_VOL_LIM - [7:0] */ + +/* + * R1054 (0x41E) - DAC Volume Limit 5L + */ +#define WM5100_OUT5_OSR 0x2000 /* OUT5_OSR */ +#define WM5100_OUT5_OSR_MASK 0x2000 /* OUT5_OSR */ +#define WM5100_OUT5_OSR_SHIFT 13 /* OUT5_OSR */ +#define WM5100_OUT5_OSR_WIDTH 1 /* OUT5_OSR */ +#define WM5100_OUT5L_ANC_SRC 0x0800 /* OUT5L_ANC_SRC */ +#define WM5100_OUT5L_ANC_SRC_MASK 0x0800 /* OUT5L_ANC_SRC */ +#define WM5100_OUT5L_ANC_SRC_SHIFT 11 /* OUT5L_ANC_SRC */ +#define WM5100_OUT5L_ANC_SRC_WIDTH 1 /* OUT5L_ANC_SRC */ +#define WM5100_OUT5L_VOL_LIM_MASK 0x00FF /* OUT5L_VOL_LIM - [7:0] */ +#define WM5100_OUT5L_VOL_LIM_SHIFT 0 /* OUT5L_VOL_LIM - [7:0] */ +#define WM5100_OUT5L_VOL_LIM_WIDTH 8 /* OUT5L_VOL_LIM - [7:0] */ + +/* + * R1055 (0x41F) - DAC Volume Limit 5R + */ +#define WM5100_OUT5R_ANC_SRC 0x0800 /* OUT5R_ANC_SRC */ +#define WM5100_OUT5R_ANC_SRC_MASK 0x0800 /* OUT5R_ANC_SRC */ +#define WM5100_OUT5R_ANC_SRC_SHIFT 11 /* OUT5R_ANC_SRC */ +#define WM5100_OUT5R_ANC_SRC_WIDTH 1 /* OUT5R_ANC_SRC */ +#define WM5100_OUT5R_VOL_LIM_MASK 0x00FF /* OUT5R_VOL_LIM - [7:0] */ +#define WM5100_OUT5R_VOL_LIM_SHIFT 0 /* OUT5R_VOL_LIM - [7:0] */ +#define WM5100_OUT5R_VOL_LIM_WIDTH 8 /* OUT5R_VOL_LIM - [7:0] */ + +/* + * R1056 (0x420) - DAC Volume Limit 6L + */ +#define WM5100_OUT6_OSR 0x2000 /* OUT6_OSR */ +#define WM5100_OUT6_OSR_MASK 0x2000 /* OUT6_OSR */ +#define WM5100_OUT6_OSR_SHIFT 13 /* OUT6_OSR */ +#define WM5100_OUT6_OSR_WIDTH 1 /* OUT6_OSR */ +#define WM5100_OUT6L_ANC_SRC 0x0800 /* OUT6L_ANC_SRC */ +#define WM5100_OUT6L_ANC_SRC_MASK 0x0800 /* OUT6L_ANC_SRC */ +#define WM5100_OUT6L_ANC_SRC_SHIFT 11 /* OUT6L_ANC_SRC */ +#define WM5100_OUT6L_ANC_SRC_WIDTH 1 /* OUT6L_ANC_SRC */ +#define WM5100_OUT6L_VOL_LIM_MASK 0x00FF /* OUT6L_VOL_LIM - [7:0] */ +#define WM5100_OUT6L_VOL_LIM_SHIFT 0 /* OUT6L_VOL_LIM - [7:0] */ +#define WM5100_OUT6L_VOL_LIM_WIDTH 8 /* OUT6L_VOL_LIM - [7:0] */ + +/* + * R1057 (0x421) - DAC Volume Limit 6R + */ +#define WM5100_OUT6R_ANC_SRC 0x0800 /* OUT6R_ANC_SRC */ +#define WM5100_OUT6R_ANC_SRC_MASK 0x0800 /* OUT6R_ANC_SRC */ +#define WM5100_OUT6R_ANC_SRC_SHIFT 11 /* OUT6R_ANC_SRC */ +#define WM5100_OUT6R_ANC_SRC_WIDTH 1 /* OUT6R_ANC_SRC */ +#define WM5100_OUT6R_VOL_LIM_MASK 0x00FF /* OUT6R_VOL_LIM - [7:0] */ +#define WM5100_OUT6R_VOL_LIM_SHIFT 0 /* OUT6R_VOL_LIM - [7:0] */ +#define WM5100_OUT6R_VOL_LIM_WIDTH 8 /* OUT6R_VOL_LIM - [7:0] */ + +/* + * R1088 (0x440) - DAC AEC Control 1 + */ +#define WM5100_AEC_LOOPBACK_SRC_MASK 0x003C /* AEC_LOOPBACK_SRC - [5:2] */ +#define WM5100_AEC_LOOPBACK_SRC_SHIFT 2 /* AEC_LOOPBACK_SRC - [5:2] */ +#define WM5100_AEC_LOOPBACK_SRC_WIDTH 4 /* AEC_LOOPBACK_SRC - [5:2] */ +#define WM5100_AEC_ENA_STS 0x0002 /* AEC_ENA_STS */ +#define WM5100_AEC_ENA_STS_MASK 0x0002 /* AEC_ENA_STS */ +#define WM5100_AEC_ENA_STS_SHIFT 1 /* AEC_ENA_STS */ +#define WM5100_AEC_ENA_STS_WIDTH 1 /* AEC_ENA_STS */ +#define WM5100_AEC_LOOPBACK_ENA 0x0001 /* AEC_LOOPBACK_ENA */ +#define WM5100_AEC_LOOPBACK_ENA_MASK 0x0001 /* AEC_LOOPBACK_ENA */ +#define WM5100_AEC_LOOPBACK_ENA_SHIFT 0 /* AEC_LOOPBACK_ENA */ +#define WM5100_AEC_LOOPBACK_ENA_WIDTH 1 /* AEC_LOOPBACK_ENA */ + +/* + * R1089 (0x441) - Output Volume Ramp + */ +#define WM5100_OUT_VD_RAMP_MASK 0x0070 /* OUT_VD_RAMP - [6:4] */ +#define WM5100_OUT_VD_RAMP_SHIFT 4 /* OUT_VD_RAMP - [6:4] */ +#define WM5100_OUT_VD_RAMP_WIDTH 3 /* OUT_VD_RAMP - [6:4] */ +#define WM5100_OUT_VI_RAMP_MASK 0x0007 /* OUT_VI_RAMP - [2:0] */ +#define WM5100_OUT_VI_RAMP_SHIFT 0 /* OUT_VI_RAMP - [2:0] */ +#define WM5100_OUT_VI_RAMP_WIDTH 3 /* OUT_VI_RAMP - [2:0] */ + +/* + * R1152 (0x480) - DAC Digital Volume 1L + */ +#define WM5100_OUT_VU 0x0200 /* OUT_VU */ +#define WM5100_OUT_VU_MASK 0x0200 /* OUT_VU */ +#define WM5100_OUT_VU_SHIFT 9 /* OUT_VU */ +#define WM5100_OUT_VU_WIDTH 1 /* OUT_VU */ +#define WM5100_OUT1L_MUTE 0x0100 /* OUT1L_MUTE */ +#define WM5100_OUT1L_MUTE_MASK 0x0100 /* OUT1L_MUTE */ +#define WM5100_OUT1L_MUTE_SHIFT 8 /* OUT1L_MUTE */ +#define WM5100_OUT1L_MUTE_WIDTH 1 /* OUT1L_MUTE */ +#define WM5100_OUT1L_VOL_MASK 0x00FF /* OUT1L_VOL - [7:0] */ +#define WM5100_OUT1L_VOL_SHIFT 0 /* OUT1L_VOL - [7:0] */ +#define WM5100_OUT1L_VOL_WIDTH 8 /* OUT1L_VOL - [7:0] */ + +/* + * R1153 (0x481) - DAC Digital Volume 1R + */ +#define WM5100_OUT_VU 0x0200 /* OUT_VU */ +#define WM5100_OUT_VU_MASK 0x0200 /* OUT_VU */ +#define WM5100_OUT_VU_SHIFT 9 /* OUT_VU */ +#define WM5100_OUT_VU_WIDTH 1 /* OUT_VU */ +#define WM5100_OUT1R_MUTE 0x0100 /* OUT1R_MUTE */ +#define WM5100_OUT1R_MUTE_MASK 0x0100 /* OUT1R_MUTE */ +#define WM5100_OUT1R_MUTE_SHIFT 8 /* OUT1R_MUTE */ +#define WM5100_OUT1R_MUTE_WIDTH 1 /* OUT1R_MUTE */ +#define WM5100_OUT1R_VOL_MASK 0x00FF /* OUT1R_VOL - [7:0] */ +#define WM5100_OUT1R_VOL_SHIFT 0 /* OUT1R_VOL - [7:0] */ +#define WM5100_OUT1R_VOL_WIDTH 8 /* OUT1R_VOL - [7:0] */ + +/* + * R1154 (0x482) - DAC Digital Volume 2L + */ +#define WM5100_OUT_VU 0x0200 /* OUT_VU */ +#define WM5100_OUT_VU_MASK 0x0200 /* OUT_VU */ +#define WM5100_OUT_VU_SHIFT 9 /* OUT_VU */ +#define WM5100_OUT_VU_WIDTH 1 /* OUT_VU */ +#define WM5100_OUT2L_MUTE 0x0100 /* OUT2L_MUTE */ +#define WM5100_OUT2L_MUTE_MASK 0x0100 /* OUT2L_MUTE */ +#define WM5100_OUT2L_MUTE_SHIFT 8 /* OUT2L_MUTE */ +#define WM5100_OUT2L_MUTE_WIDTH 1 /* OUT2L_MUTE */ +#define WM5100_OUT2L_VOL_MASK 0x00FF /* OUT2L_VOL - [7:0] */ +#define WM5100_OUT2L_VOL_SHIFT 0 /* OUT2L_VOL - [7:0] */ +#define WM5100_OUT2L_VOL_WIDTH 8 /* OUT2L_VOL - [7:0] */ + +/* + * R1155 (0x483) - DAC Digital Volume 2R + */ +#define WM5100_OUT_VU 0x0200 /* OUT_VU */ +#define WM5100_OUT_VU_MASK 0x0200 /* OUT_VU */ +#define WM5100_OUT_VU_SHIFT 9 /* OUT_VU */ +#define WM5100_OUT_VU_WIDTH 1 /* OUT_VU */ +#define WM5100_OUT2R_MUTE 0x0100 /* OUT2R_MUTE */ +#define WM5100_OUT2R_MUTE_MASK 0x0100 /* OUT2R_MUTE */ +#define WM5100_OUT2R_MUTE_SHIFT 8 /* OUT2R_MUTE */ +#define WM5100_OUT2R_MUTE_WIDTH 1 /* OUT2R_MUTE */ +#define WM5100_OUT2R_VOL_MASK 0x00FF /* OUT2R_VOL - [7:0] */ +#define WM5100_OUT2R_VOL_SHIFT 0 /* OUT2R_VOL - [7:0] */ +#define WM5100_OUT2R_VOL_WIDTH 8 /* OUT2R_VOL - [7:0] */ + +/* + * R1156 (0x484) - DAC Digital Volume 3L + */ +#define WM5100_OUT_VU 0x0200 /* OUT_VU */ +#define WM5100_OUT_VU_MASK 0x0200 /* OUT_VU */ +#define WM5100_OUT_VU_SHIFT 9 /* OUT_VU */ +#define WM5100_OUT_VU_WIDTH 1 /* OUT_VU */ +#define WM5100_OUT3L_MUTE 0x0100 /* OUT3L_MUTE */ +#define WM5100_OUT3L_MUTE_MASK 0x0100 /* OUT3L_MUTE */ +#define WM5100_OUT3L_MUTE_SHIFT 8 /* OUT3L_MUTE */ +#define WM5100_OUT3L_MUTE_WIDTH 1 /* OUT3L_MUTE */ +#define WM5100_OUT3L_VOL_MASK 0x00FF /* OUT3L_VOL - [7:0] */ +#define WM5100_OUT3L_VOL_SHIFT 0 /* OUT3L_VOL - [7:0] */ +#define WM5100_OUT3L_VOL_WIDTH 8 /* OUT3L_VOL - [7:0] */ + +/* + * R1157 (0x485) - DAC Digital Volume 3R + */ +#define WM5100_OUT_VU 0x0200 /* OUT_VU */ +#define WM5100_OUT_VU_MASK 0x0200 /* OUT_VU */ +#define WM5100_OUT_VU_SHIFT 9 /* OUT_VU */ +#define WM5100_OUT_VU_WIDTH 1 /* OUT_VU */ +#define WM5100_OUT3R_MUTE 0x0100 /* OUT3R_MUTE */ +#define WM5100_OUT3R_MUTE_MASK 0x0100 /* OUT3R_MUTE */ +#define WM5100_OUT3R_MUTE_SHIFT 8 /* OUT3R_MUTE */ +#define WM5100_OUT3R_MUTE_WIDTH 1 /* OUT3R_MUTE */ +#define WM5100_OUT3R_VOL_MASK 0x00FF /* OUT3R_VOL - [7:0] */ +#define WM5100_OUT3R_VOL_SHIFT 0 /* OUT3R_VOL - [7:0] */ +#define WM5100_OUT3R_VOL_WIDTH 8 /* OUT3R_VOL - [7:0] */ + +/* + * R1158 (0x486) - DAC Digital Volume 4L + */ +#define WM5100_OUT_VU 0x0200 /* OUT_VU */ +#define WM5100_OUT_VU_MASK 0x0200 /* OUT_VU */ +#define WM5100_OUT_VU_SHIFT 9 /* OUT_VU */ +#define WM5100_OUT_VU_WIDTH 1 /* OUT_VU */ +#define WM5100_OUT4L_MUTE 0x0100 /* OUT4L_MUTE */ +#define WM5100_OUT4L_MUTE_MASK 0x0100 /* OUT4L_MUTE */ +#define WM5100_OUT4L_MUTE_SHIFT 8 /* OUT4L_MUTE */ +#define WM5100_OUT4L_MUTE_WIDTH 1 /* OUT4L_MUTE */ +#define WM5100_OUT4L_VOL_MASK 0x00FF /* OUT4L_VOL - [7:0] */ +#define WM5100_OUT4L_VOL_SHIFT 0 /* OUT4L_VOL - [7:0] */ +#define WM5100_OUT4L_VOL_WIDTH 8 /* OUT4L_VOL - [7:0] */ + +/* + * R1159 (0x487) - DAC Digital Volume 4R + */ +#define WM5100_OUT_VU 0x0200 /* OUT_VU */ +#define WM5100_OUT_VU_MASK 0x0200 /* OUT_VU */ +#define WM5100_OUT_VU_SHIFT 9 /* OUT_VU */ +#define WM5100_OUT_VU_WIDTH 1 /* OUT_VU */ +#define WM5100_OUT4R_MUTE 0x0100 /* OUT4R_MUTE */ +#define WM5100_OUT4R_MUTE_MASK 0x0100 /* OUT4R_MUTE */ +#define WM5100_OUT4R_MUTE_SHIFT 8 /* OUT4R_MUTE */ +#define WM5100_OUT4R_MUTE_WIDTH 1 /* OUT4R_MUTE */ +#define WM5100_OUT4R_VOL_MASK 0x00FF /* OUT4R_VOL - [7:0] */ +#define WM5100_OUT4R_VOL_SHIFT 0 /* OUT4R_VOL - [7:0] */ +#define WM5100_OUT4R_VOL_WIDTH 8 /* OUT4R_VOL - [7:0] */ + +/* + * R1160 (0x488) - DAC Digital Volume 5L + */ +#define WM5100_OUT_VU 0x0200 /* OUT_VU */ +#define WM5100_OUT_VU_MASK 0x0200 /* OUT_VU */ +#define WM5100_OUT_VU_SHIFT 9 /* OUT_VU */ +#define WM5100_OUT_VU_WIDTH 1 /* OUT_VU */ +#define WM5100_OUT5L_MUTE 0x0100 /* OUT5L_MUTE */ +#define WM5100_OUT5L_MUTE_MASK 0x0100 /* OUT5L_MUTE */ +#define WM5100_OUT5L_MUTE_SHIFT 8 /* OUT5L_MUTE */ +#define WM5100_OUT5L_MUTE_WIDTH 1 /* OUT5L_MUTE */ +#define WM5100_OUT5L_VOL_MASK 0x00FF /* OUT5L_VOL - [7:0] */ +#define WM5100_OUT5L_VOL_SHIFT 0 /* OUT5L_VOL - [7:0] */ +#define WM5100_OUT5L_VOL_WIDTH 8 /* OUT5L_VOL - [7:0] */ + +/* + * R1161 (0x489) - DAC Digital Volume 5R + */ +#define WM5100_OUT_VU 0x0200 /* OUT_VU */ +#define WM5100_OUT_VU_MASK 0x0200 /* OUT_VU */ +#define WM5100_OUT_VU_SHIFT 9 /* OUT_VU */ +#define WM5100_OUT_VU_WIDTH 1 /* OUT_VU */ +#define WM5100_OUT5R_MUTE 0x0100 /* OUT5R_MUTE */ +#define WM5100_OUT5R_MUTE_MASK 0x0100 /* OUT5R_MUTE */ +#define WM5100_OUT5R_MUTE_SHIFT 8 /* OUT5R_MUTE */ +#define WM5100_OUT5R_MUTE_WIDTH 1 /* OUT5R_MUTE */ +#define WM5100_OUT5R_VOL_MASK 0x00FF /* OUT5R_VOL - [7:0] */ +#define WM5100_OUT5R_VOL_SHIFT 0 /* OUT5R_VOL - [7:0] */ +#define WM5100_OUT5R_VOL_WIDTH 8 /* OUT5R_VOL - [7:0] */ + +/* + * R1162 (0x48A) - DAC Digital Volume 6L + */ +#define WM5100_OUT_VU 0x0200 /* OUT_VU */ +#define WM5100_OUT_VU_MASK 0x0200 /* OUT_VU */ +#define WM5100_OUT_VU_SHIFT 9 /* OUT_VU */ +#define WM5100_OUT_VU_WIDTH 1 /* OUT_VU */ +#define WM5100_OUT6L_MUTE 0x0100 /* OUT6L_MUTE */ +#define WM5100_OUT6L_MUTE_MASK 0x0100 /* OUT6L_MUTE */ +#define WM5100_OUT6L_MUTE_SHIFT 8 /* OUT6L_MUTE */ +#define WM5100_OUT6L_MUTE_WIDTH 1 /* OUT6L_MUTE */ +#define WM5100_OUT6L_VOL_MASK 0x00FF /* OUT6L_VOL - [7:0] */ +#define WM5100_OUT6L_VOL_SHIFT 0 /* OUT6L_VOL - [7:0] */ +#define WM5100_OUT6L_VOL_WIDTH 8 /* OUT6L_VOL - [7:0] */ + +/* + * R1163 (0x48B) - DAC Digital Volume 6R + */ +#define WM5100_OUT_VU 0x0200 /* OUT_VU */ +#define WM5100_OUT_VU_MASK 0x0200 /* OUT_VU */ +#define WM5100_OUT_VU_SHIFT 9 /* OUT_VU */ +#define WM5100_OUT_VU_WIDTH 1 /* OUT_VU */ +#define WM5100_OUT6R_MUTE 0x0100 /* OUT6R_MUTE */ +#define WM5100_OUT6R_MUTE_MASK 0x0100 /* OUT6R_MUTE */ +#define WM5100_OUT6R_MUTE_SHIFT 8 /* OUT6R_MUTE */ +#define WM5100_OUT6R_MUTE_WIDTH 1 /* OUT6R_MUTE */ +#define WM5100_OUT6R_VOL_MASK 0x00FF /* OUT6R_VOL - [7:0] */ +#define WM5100_OUT6R_VOL_SHIFT 0 /* OUT6R_VOL - [7:0] */ +#define WM5100_OUT6R_VOL_WIDTH 8 /* OUT6R_VOL - [7:0] */ + +/* + * R1216 (0x4C0) - PDM SPK1 CTRL 1 + */ +#define WM5100_SPK1R_MUTE 0x2000 /* SPK1R_MUTE */ +#define WM5100_SPK1R_MUTE_MASK 0x2000 /* SPK1R_MUTE */ +#define WM5100_SPK1R_MUTE_SHIFT 13 /* SPK1R_MUTE */ +#define WM5100_SPK1R_MUTE_WIDTH 1 /* SPK1R_MUTE */ +#define WM5100_SPK1L_MUTE 0x1000 /* SPK1L_MUTE */ +#define WM5100_SPK1L_MUTE_MASK 0x1000 /* SPK1L_MUTE */ +#define WM5100_SPK1L_MUTE_SHIFT 12 /* SPK1L_MUTE */ +#define WM5100_SPK1L_MUTE_WIDTH 1 /* SPK1L_MUTE */ +#define WM5100_SPK1_MUTE_ENDIAN 0x0100 /* SPK1_MUTE_ENDIAN */ +#define WM5100_SPK1_MUTE_ENDIAN_MASK 0x0100 /* SPK1_MUTE_ENDIAN */ +#define WM5100_SPK1_MUTE_ENDIAN_SHIFT 8 /* SPK1_MUTE_ENDIAN */ +#define WM5100_SPK1_MUTE_ENDIAN_WIDTH 1 /* SPK1_MUTE_ENDIAN */ +#define WM5100_SPK1_MUTE_SEQ1_MASK 0x00FF /* SPK1_MUTE_SEQ1 - [7:0] */ +#define WM5100_SPK1_MUTE_SEQ1_SHIFT 0 /* SPK1_MUTE_SEQ1 - [7:0] */ +#define WM5100_SPK1_MUTE_SEQ1_WIDTH 8 /* SPK1_MUTE_SEQ1 - [7:0] */ + +/* + * R1217 (0x4C1) - PDM SPK1 CTRL 2 + */ +#define WM5100_SPK1_FMT 0x0001 /* SPK1_FMT */ +#define WM5100_SPK1_FMT_MASK 0x0001 /* SPK1_FMT */ +#define WM5100_SPK1_FMT_SHIFT 0 /* SPK1_FMT */ +#define WM5100_SPK1_FMT_WIDTH 1 /* SPK1_FMT */ + +/* + * R1218 (0x4C2) - PDM SPK2 CTRL 1 + */ +#define WM5100_SPK2R_MUTE 0x2000 /* SPK2R_MUTE */ +#define WM5100_SPK2R_MUTE_MASK 0x2000 /* SPK2R_MUTE */ +#define WM5100_SPK2R_MUTE_SHIFT 13 /* SPK2R_MUTE */ +#define WM5100_SPK2R_MUTE_WIDTH 1 /* SPK2R_MUTE */ +#define WM5100_SPK2L_MUTE 0x1000 /* SPK2L_MUTE */ +#define WM5100_SPK2L_MUTE_MASK 0x1000 /* SPK2L_MUTE */ +#define WM5100_SPK2L_MUTE_SHIFT 12 /* SPK2L_MUTE */ +#define WM5100_SPK2L_MUTE_WIDTH 1 /* SPK2L_MUTE */ +#define WM5100_SPK2_MUTE_ENDIAN 0x0100 /* SPK2_MUTE_ENDIAN */ +#define WM5100_SPK2_MUTE_ENDIAN_MASK 0x0100 /* SPK2_MUTE_ENDIAN */ +#define WM5100_SPK2_MUTE_ENDIAN_SHIFT 8 /* SPK2_MUTE_ENDIAN */ +#define WM5100_SPK2_MUTE_ENDIAN_WIDTH 1 /* SPK2_MUTE_ENDIAN */ +#define WM5100_SPK2_MUTE_SEQ1_MASK 0x00FF /* SPK2_MUTE_SEQ1 - [7:0] */ +#define WM5100_SPK2_MUTE_SEQ1_SHIFT 0 /* SPK2_MUTE_SEQ1 - [7:0] */ +#define WM5100_SPK2_MUTE_SEQ1_WIDTH 8 /* SPK2_MUTE_SEQ1 - [7:0] */ + +/* + * R1219 (0x4C3) - PDM SPK2 CTRL 2 + */ +#define WM5100_SPK2_FMT 0x0001 /* SPK2_FMT */ +#define WM5100_SPK2_FMT_MASK 0x0001 /* SPK2_FMT */ +#define WM5100_SPK2_FMT_SHIFT 0 /* SPK2_FMT */ +#define WM5100_SPK2_FMT_WIDTH 1 /* SPK2_FMT */ + +/* + * R1280 (0x500) - Audio IF 1_1 + */ +#define WM5100_AIF1_BCLK_INV 0x0080 /* AIF1_BCLK_INV */ +#define WM5100_AIF1_BCLK_INV_MASK 0x0080 /* AIF1_BCLK_INV */ +#define WM5100_AIF1_BCLK_INV_SHIFT 7 /* AIF1_BCLK_INV */ +#define WM5100_AIF1_BCLK_INV_WIDTH 1 /* AIF1_BCLK_INV */ +#define WM5100_AIF1_BCLK_FRC 0x0040 /* AIF1_BCLK_FRC */ +#define WM5100_AIF1_BCLK_FRC_MASK 0x0040 /* AIF1_BCLK_FRC */ +#define WM5100_AIF1_BCLK_FRC_SHIFT 6 /* AIF1_BCLK_FRC */ +#define WM5100_AIF1_BCLK_FRC_WIDTH 1 /* AIF1_BCLK_FRC */ +#define WM5100_AIF1_BCLK_MSTR 0x0020 /* AIF1_BCLK_MSTR */ +#define WM5100_AIF1_BCLK_MSTR_MASK 0x0020 /* AIF1_BCLK_MSTR */ +#define WM5100_AIF1_BCLK_MSTR_SHIFT 5 /* AIF1_BCLK_MSTR */ +#define WM5100_AIF1_BCLK_MSTR_WIDTH 1 /* AIF1_BCLK_MSTR */ +#define WM5100_AIF1_BCLK_FREQ_MASK 0x001F /* AIF1_BCLK_FREQ - [4:0] */ +#define WM5100_AIF1_BCLK_FREQ_SHIFT 0 /* AIF1_BCLK_FREQ - [4:0] */ +#define WM5100_AIF1_BCLK_FREQ_WIDTH 5 /* AIF1_BCLK_FREQ - [4:0] */ + +/* + * R1281 (0x501) - Audio IF 1_2 + */ +#define WM5100_AIF1TX_DAT_TRI 0x0020 /* AIF1TX_DAT_TRI */ +#define WM5100_AIF1TX_DAT_TRI_MASK 0x0020 /* AIF1TX_DAT_TRI */ +#define WM5100_AIF1TX_DAT_TRI_SHIFT 5 /* AIF1TX_DAT_TRI */ +#define WM5100_AIF1TX_DAT_TRI_WIDTH 1 /* AIF1TX_DAT_TRI */ +#define WM5100_AIF1TX_LRCLK_SRC 0x0008 /* AIF1TX_LRCLK_SRC */ +#define WM5100_AIF1TX_LRCLK_SRC_MASK 0x0008 /* AIF1TX_LRCLK_SRC */ +#define WM5100_AIF1TX_LRCLK_SRC_SHIFT 3 /* AIF1TX_LRCLK_SRC */ +#define WM5100_AIF1TX_LRCLK_SRC_WIDTH 1 /* AIF1TX_LRCLK_SRC */ +#define WM5100_AIF1TX_LRCLK_INV 0x0004 /* AIF1TX_LRCLK_INV */ +#define WM5100_AIF1TX_LRCLK_INV_MASK 0x0004 /* AIF1TX_LRCLK_INV */ +#define WM5100_AIF1TX_LRCLK_INV_SHIFT 2 /* AIF1TX_LRCLK_INV */ +#define WM5100_AIF1TX_LRCLK_INV_WIDTH 1 /* AIF1TX_LRCLK_INV */ +#define WM5100_AIF1TX_LRCLK_FRC 0x0002 /* AIF1TX_LRCLK_FRC */ +#define WM5100_AIF1TX_LRCLK_FRC_MASK 0x0002 /* AIF1TX_LRCLK_FRC */ +#define WM5100_AIF1TX_LRCLK_FRC_SHIFT 1 /* AIF1TX_LRCLK_FRC */ +#define WM5100_AIF1TX_LRCLK_FRC_WIDTH 1 /* AIF1TX_LRCLK_FRC */ +#define WM5100_AIF1TX_LRCLK_MSTR 0x0001 /* AIF1TX_LRCLK_MSTR */ +#define WM5100_AIF1TX_LRCLK_MSTR_MASK 0x0001 /* AIF1TX_LRCLK_MSTR */ +#define WM5100_AIF1TX_LRCLK_MSTR_SHIFT 0 /* AIF1TX_LRCLK_MSTR */ +#define WM5100_AIF1TX_LRCLK_MSTR_WIDTH 1 /* AIF1TX_LRCLK_MSTR */ + +/* + * R1282 (0x502) - Audio IF 1_3 + */ +#define WM5100_AIF1RX_LRCLK_INV 0x0004 /* AIF1RX_LRCLK_INV */ +#define WM5100_AIF1RX_LRCLK_INV_MASK 0x0004 /* AIF1RX_LRCLK_INV */ +#define WM5100_AIF1RX_LRCLK_INV_SHIFT 2 /* AIF1RX_LRCLK_INV */ +#define WM5100_AIF1RX_LRCLK_INV_WIDTH 1 /* AIF1RX_LRCLK_INV */ +#define WM5100_AIF1RX_LRCLK_FRC 0x0002 /* AIF1RX_LRCLK_FRC */ +#define WM5100_AIF1RX_LRCLK_FRC_MASK 0x0002 /* AIF1RX_LRCLK_FRC */ +#define WM5100_AIF1RX_LRCLK_FRC_SHIFT 1 /* AIF1RX_LRCLK_FRC */ +#define WM5100_AIF1RX_LRCLK_FRC_WIDTH 1 /* AIF1RX_LRCLK_FRC */ +#define WM5100_AIF1RX_LRCLK_MSTR 0x0001 /* AIF1RX_LRCLK_MSTR */ +#define WM5100_AIF1RX_LRCLK_MSTR_MASK 0x0001 /* AIF1RX_LRCLK_MSTR */ +#define WM5100_AIF1RX_LRCLK_MSTR_SHIFT 0 /* AIF1RX_LRCLK_MSTR */ +#define WM5100_AIF1RX_LRCLK_MSTR_WIDTH 1 /* AIF1RX_LRCLK_MSTR */ + +/* + * R1283 (0x503) - Audio IF 1_4 + */ +#define WM5100_AIF1_TRI 0x0040 /* AIF1_TRI */ +#define WM5100_AIF1_TRI_MASK 0x0040 /* AIF1_TRI */ +#define WM5100_AIF1_TRI_SHIFT 6 /* AIF1_TRI */ +#define WM5100_AIF1_TRI_WIDTH 1 /* AIF1_TRI */ +#define WM5100_AIF1_RATE_MASK 0x0003 /* AIF1_RATE - [1:0] */ +#define WM5100_AIF1_RATE_SHIFT 0 /* AIF1_RATE - [1:0] */ +#define WM5100_AIF1_RATE_WIDTH 2 /* AIF1_RATE - [1:0] */ + +/* + * R1284 (0x504) - Audio IF 1_5 + */ +#define WM5100_AIF1_FMT_MASK 0x0007 /* AIF1_FMT - [2:0] */ +#define WM5100_AIF1_FMT_SHIFT 0 /* AIF1_FMT - [2:0] */ +#define WM5100_AIF1_FMT_WIDTH 3 /* AIF1_FMT - [2:0] */ + +/* + * R1285 (0x505) - Audio IF 1_6 + */ +#define WM5100_AIF1TX_BCPF_MASK 0x1FFF /* AIF1TX_BCPF - [12:0] */ +#define WM5100_AIF1TX_BCPF_SHIFT 0 /* AIF1TX_BCPF - [12:0] */ +#define WM5100_AIF1TX_BCPF_WIDTH 13 /* AIF1TX_BCPF - [12:0] */ + +/* + * R1286 (0x506) - Audio IF 1_7 + */ +#define WM5100_AIF1RX_BCPF_MASK 0x1FFF /* AIF1RX_BCPF - [12:0] */ +#define WM5100_AIF1RX_BCPF_SHIFT 0 /* AIF1RX_BCPF - [12:0] */ +#define WM5100_AIF1RX_BCPF_WIDTH 13 /* AIF1RX_BCPF - [12:0] */ + +/* + * R1287 (0x507) - Audio IF 1_8 + */ +#define WM5100_AIF1TX_WL_MASK 0x3F00 /* AIF1TX_WL - [13:8] */ +#define WM5100_AIF1TX_WL_SHIFT 8 /* AIF1TX_WL - [13:8] */ +#define WM5100_AIF1TX_WL_WIDTH 6 /* AIF1TX_WL - [13:8] */ +#define WM5100_AIF1TX_SLOT_LEN_MASK 0x00FF /* AIF1TX_SLOT_LEN - [7:0] */ +#define WM5100_AIF1TX_SLOT_LEN_SHIFT 0 /* AIF1TX_SLOT_LEN - [7:0] */ +#define WM5100_AIF1TX_SLOT_LEN_WIDTH 8 /* AIF1TX_SLOT_LEN - [7:0] */ + +/* + * R1288 (0x508) - Audio IF 1_9 + */ +#define WM5100_AIF1RX_WL_MASK 0x3F00 /* AIF1RX_WL - [13:8] */ +#define WM5100_AIF1RX_WL_SHIFT 8 /* AIF1RX_WL - [13:8] */ +#define WM5100_AIF1RX_WL_WIDTH 6 /* AIF1RX_WL - [13:8] */ +#define WM5100_AIF1RX_SLOT_LEN_MASK 0x00FF /* AIF1RX_SLOT_LEN - [7:0] */ +#define WM5100_AIF1RX_SLOT_LEN_SHIFT 0 /* AIF1RX_SLOT_LEN - [7:0] */ +#define WM5100_AIF1RX_SLOT_LEN_WIDTH 8 /* AIF1RX_SLOT_LEN - [7:0] */ + +/* + * R1289 (0x509) - Audio IF 1_10 + */ +#define WM5100_AIF1TX1_SLOT_MASK 0x003F /* AIF1TX1_SLOT - [5:0] */ +#define WM5100_AIF1TX1_SLOT_SHIFT 0 /* AIF1TX1_SLOT - [5:0] */ +#define WM5100_AIF1TX1_SLOT_WIDTH 6 /* AIF1TX1_SLOT - [5:0] */ + +/* + * R1290 (0x50A) - Audio IF 1_11 + */ +#define WM5100_AIF1TX2_SLOT_MASK 0x003F /* AIF1TX2_SLOT - [5:0] */ +#define WM5100_AIF1TX2_SLOT_SHIFT 0 /* AIF1TX2_SLOT - [5:0] */ +#define WM5100_AIF1TX2_SLOT_WIDTH 6 /* AIF1TX2_SLOT - [5:0] */ + +/* + * R1291 (0x50B) - Audio IF 1_12 + */ +#define WM5100_AIF1TX3_SLOT_MASK 0x003F /* AIF1TX3_SLOT - [5:0] */ +#define WM5100_AIF1TX3_SLOT_SHIFT 0 /* AIF1TX3_SLOT - [5:0] */ +#define WM5100_AIF1TX3_SLOT_WIDTH 6 /* AIF1TX3_SLOT - [5:0] */ + +/* + * R1292 (0x50C) - Audio IF 1_13 + */ +#define WM5100_AIF1TX4_SLOT_MASK 0x003F /* AIF1TX4_SLOT - [5:0] */ +#define WM5100_AIF1TX4_SLOT_SHIFT 0 /* AIF1TX4_SLOT - [5:0] */ +#define WM5100_AIF1TX4_SLOT_WIDTH 6 /* AIF1TX4_SLOT - [5:0] */ + +/* + * R1293 (0x50D) - Audio IF 1_14 + */ +#define WM5100_AIF1TX5_SLOT_MASK 0x003F /* AIF1TX5_SLOT - [5:0] */ +#define WM5100_AIF1TX5_SLOT_SHIFT 0 /* AIF1TX5_SLOT - [5:0] */ +#define WM5100_AIF1TX5_SLOT_WIDTH 6 /* AIF1TX5_SLOT - [5:0] */ + +/* + * R1294 (0x50E) - Audio IF 1_15 + */ +#define WM5100_AIF1TX6_SLOT_MASK 0x003F /* AIF1TX6_SLOT - [5:0] */ +#define WM5100_AIF1TX6_SLOT_SHIFT 0 /* AIF1TX6_SLOT - [5:0] */ +#define WM5100_AIF1TX6_SLOT_WIDTH 6 /* AIF1TX6_SLOT - [5:0] */ + +/* + * R1295 (0x50F) - Audio IF 1_16 + */ +#define WM5100_AIF1TX7_SLOT_MASK 0x003F /* AIF1TX7_SLOT - [5:0] */ +#define WM5100_AIF1TX7_SLOT_SHIFT 0 /* AIF1TX7_SLOT - [5:0] */ +#define WM5100_AIF1TX7_SLOT_WIDTH 6 /* AIF1TX7_SLOT - [5:0] */ + +/* + * R1296 (0x510) - Audio IF 1_17 + */ +#define WM5100_AIF1TX8_SLOT_MASK 0x003F /* AIF1TX8_SLOT - [5:0] */ +#define WM5100_AIF1TX8_SLOT_SHIFT 0 /* AIF1TX8_SLOT - [5:0] */ +#define WM5100_AIF1TX8_SLOT_WIDTH 6 /* AIF1TX8_SLOT - [5:0] */ + +/* + * R1297 (0x511) - Audio IF 1_18 + */ +#define WM5100_AIF1RX1_SLOT_MASK 0x003F /* AIF1RX1_SLOT - [5:0] */ +#define WM5100_AIF1RX1_SLOT_SHIFT 0 /* AIF1RX1_SLOT - [5:0] */ +#define WM5100_AIF1RX1_SLOT_WIDTH 6 /* AIF1RX1_SLOT - [5:0] */ + +/* + * R1298 (0x512) - Audio IF 1_19 + */ +#define WM5100_AIF1RX2_SLOT_MASK 0x003F /* AIF1RX2_SLOT - [5:0] */ +#define WM5100_AIF1RX2_SLOT_SHIFT 0 /* AIF1RX2_SLOT - [5:0] */ +#define WM5100_AIF1RX2_SLOT_WIDTH 6 /* AIF1RX2_SLOT - [5:0] */ + +/* + * R1299 (0x513) - Audio IF 1_20 + */ +#define WM5100_AIF1RX3_SLOT_MASK 0x003F /* AIF1RX3_SLOT - [5:0] */ +#define WM5100_AIF1RX3_SLOT_SHIFT 0 /* AIF1RX3_SLOT - [5:0] */ +#define WM5100_AIF1RX3_SLOT_WIDTH 6 /* AIF1RX3_SLOT - [5:0] */ + +/* + * R1300 (0x514) - Audio IF 1_21 + */ +#define WM5100_AIF1RX4_SLOT_MASK 0x003F /* AIF1RX4_SLOT - [5:0] */ +#define WM5100_AIF1RX4_SLOT_SHIFT 0 /* AIF1RX4_SLOT - [5:0] */ +#define WM5100_AIF1RX4_SLOT_WIDTH 6 /* AIF1RX4_SLOT - [5:0] */ + +/* + * R1301 (0x515) - Audio IF 1_22 + */ +#define WM5100_AIF1RX5_SLOT_MASK 0x003F /* AIF1RX5_SLOT - [5:0] */ +#define WM5100_AIF1RX5_SLOT_SHIFT 0 /* AIF1RX5_SLOT - [5:0] */ +#define WM5100_AIF1RX5_SLOT_WIDTH 6 /* AIF1RX5_SLOT - [5:0] */ + +/* + * R1302 (0x516) - Audio IF 1_23 + */ +#define WM5100_AIF1RX6_SLOT_MASK 0x003F /* AIF1RX6_SLOT - [5:0] */ +#define WM5100_AIF1RX6_SLOT_SHIFT 0 /* AIF1RX6_SLOT - [5:0] */ +#define WM5100_AIF1RX6_SLOT_WIDTH 6 /* AIF1RX6_SLOT - [5:0] */ + +/* + * R1303 (0x517) - Audio IF 1_24 + */ +#define WM5100_AIF1RX7_SLOT_MASK 0x003F /* AIF1RX7_SLOT - [5:0] */ +#define WM5100_AIF1RX7_SLOT_SHIFT 0 /* AIF1RX7_SLOT - [5:0] */ +#define WM5100_AIF1RX7_SLOT_WIDTH 6 /* AIF1RX7_SLOT - [5:0] */ + +/* + * R1304 (0x518) - Audio IF 1_25 + */ +#define WM5100_AIF1RX8_SLOT_MASK 0x003F /* AIF1RX8_SLOT - [5:0] */ +#define WM5100_AIF1RX8_SLOT_SHIFT 0 /* AIF1RX8_SLOT - [5:0] */ +#define WM5100_AIF1RX8_SLOT_WIDTH 6 /* AIF1RX8_SLOT - [5:0] */ + +/* + * R1305 (0x519) - Audio IF 1_26 + */ +#define WM5100_AIF1TX8_ENA 0x0080 /* AIF1TX8_ENA */ +#define WM5100_AIF1TX8_ENA_MASK 0x0080 /* AIF1TX8_ENA */ +#define WM5100_AIF1TX8_ENA_SHIFT 7 /* AIF1TX8_ENA */ +#define WM5100_AIF1TX8_ENA_WIDTH 1 /* AIF1TX8_ENA */ +#define WM5100_AIF1TX7_ENA 0x0040 /* AIF1TX7_ENA */ +#define WM5100_AIF1TX7_ENA_MASK 0x0040 /* AIF1TX7_ENA */ +#define WM5100_AIF1TX7_ENA_SHIFT 6 /* AIF1TX7_ENA */ +#define WM5100_AIF1TX7_ENA_WIDTH 1 /* AIF1TX7_ENA */ +#define WM5100_AIF1TX6_ENA 0x0020 /* AIF1TX6_ENA */ +#define WM5100_AIF1TX6_ENA_MASK 0x0020 /* AIF1TX6_ENA */ +#define WM5100_AIF1TX6_ENA_SHIFT 5 /* AIF1TX6_ENA */ +#define WM5100_AIF1TX6_ENA_WIDTH 1 /* AIF1TX6_ENA */ +#define WM5100_AIF1TX5_ENA 0x0010 /* AIF1TX5_ENA */ +#define WM5100_AIF1TX5_ENA_MASK 0x0010 /* AIF1TX5_ENA */ +#define WM5100_AIF1TX5_ENA_SHIFT 4 /* AIF1TX5_ENA */ +#define WM5100_AIF1TX5_ENA_WIDTH 1 /* AIF1TX5_ENA */ +#define WM5100_AIF1TX4_ENA 0x0008 /* AIF1TX4_ENA */ +#define WM5100_AIF1TX4_ENA_MASK 0x0008 /* AIF1TX4_ENA */ +#define WM5100_AIF1TX4_ENA_SHIFT 3 /* AIF1TX4_ENA */ +#define WM5100_AIF1TX4_ENA_WIDTH 1 /* AIF1TX4_ENA */ +#define WM5100_AIF1TX3_ENA 0x0004 /* AIF1TX3_ENA */ +#define WM5100_AIF1TX3_ENA_MASK 0x0004 /* AIF1TX3_ENA */ +#define WM5100_AIF1TX3_ENA_SHIFT 2 /* AIF1TX3_ENA */ +#define WM5100_AIF1TX3_ENA_WIDTH 1 /* AIF1TX3_ENA */ +#define WM5100_AIF1TX2_ENA 0x0002 /* AIF1TX2_ENA */ +#define WM5100_AIF1TX2_ENA_MASK 0x0002 /* AIF1TX2_ENA */ +#define WM5100_AIF1TX2_ENA_SHIFT 1 /* AIF1TX2_ENA */ +#define WM5100_AIF1TX2_ENA_WIDTH 1 /* AIF1TX2_ENA */ +#define WM5100_AIF1TX1_ENA 0x0001 /* AIF1TX1_ENA */ +#define WM5100_AIF1TX1_ENA_MASK 0x0001 /* AIF1TX1_ENA */ +#define WM5100_AIF1TX1_ENA_SHIFT 0 /* AIF1TX1_ENA */ +#define WM5100_AIF1TX1_ENA_WIDTH 1 /* AIF1TX1_ENA */ + +/* + * R1306 (0x51A) - Audio IF 1_27 + */ +#define WM5100_AIF1RX8_ENA 0x0080 /* AIF1RX8_ENA */ +#define WM5100_AIF1RX8_ENA_MASK 0x0080 /* AIF1RX8_ENA */ +#define WM5100_AIF1RX8_ENA_SHIFT 7 /* AIF1RX8_ENA */ +#define WM5100_AIF1RX8_ENA_WIDTH 1 /* AIF1RX8_ENA */ +#define WM5100_AIF1RX7_ENA 0x0040 /* AIF1RX7_ENA */ +#define WM5100_AIF1RX7_ENA_MASK 0x0040 /* AIF1RX7_ENA */ +#define WM5100_AIF1RX7_ENA_SHIFT 6 /* AIF1RX7_ENA */ +#define WM5100_AIF1RX7_ENA_WIDTH 1 /* AIF1RX7_ENA */ +#define WM5100_AIF1RX6_ENA 0x0020 /* AIF1RX6_ENA */ +#define WM5100_AIF1RX6_ENA_MASK 0x0020 /* AIF1RX6_ENA */ +#define WM5100_AIF1RX6_ENA_SHIFT 5 /* AIF1RX6_ENA */ +#define WM5100_AIF1RX6_ENA_WIDTH 1 /* AIF1RX6_ENA */ +#define WM5100_AIF1RX5_ENA 0x0010 /* AIF1RX5_ENA */ +#define WM5100_AIF1RX5_ENA_MASK 0x0010 /* AIF1RX5_ENA */ +#define WM5100_AIF1RX5_ENA_SHIFT 4 /* AIF1RX5_ENA */ +#define WM5100_AIF1RX5_ENA_WIDTH 1 /* AIF1RX5_ENA */ +#define WM5100_AIF1RX4_ENA 0x0008 /* AIF1RX4_ENA */ +#define WM5100_AIF1RX4_ENA_MASK 0x0008 /* AIF1RX4_ENA */ +#define WM5100_AIF1RX4_ENA_SHIFT 3 /* AIF1RX4_ENA */ +#define WM5100_AIF1RX4_ENA_WIDTH 1 /* AIF1RX4_ENA */ +#define WM5100_AIF1RX3_ENA 0x0004 /* AIF1RX3_ENA */ +#define WM5100_AIF1RX3_ENA_MASK 0x0004 /* AIF1RX3_ENA */ +#define WM5100_AIF1RX3_ENA_SHIFT 2 /* AIF1RX3_ENA */ +#define WM5100_AIF1RX3_ENA_WIDTH 1 /* AIF1RX3_ENA */ +#define WM5100_AIF1RX2_ENA 0x0002 /* AIF1RX2_ENA */ +#define WM5100_AIF1RX2_ENA_MASK 0x0002 /* AIF1RX2_ENA */ +#define WM5100_AIF1RX2_ENA_SHIFT 1 /* AIF1RX2_ENA */ +#define WM5100_AIF1RX2_ENA_WIDTH 1 /* AIF1RX2_ENA */ +#define WM5100_AIF1RX1_ENA 0x0001 /* AIF1RX1_ENA */ +#define WM5100_AIF1RX1_ENA_MASK 0x0001 /* AIF1RX1_ENA */ +#define WM5100_AIF1RX1_ENA_SHIFT 0 /* AIF1RX1_ENA */ +#define WM5100_AIF1RX1_ENA_WIDTH 1 /* AIF1RX1_ENA */ + +/* + * R1344 (0x540) - Audio IF 2_1 + */ +#define WM5100_AIF2_BCLK_INV 0x0080 /* AIF2_BCLK_INV */ +#define WM5100_AIF2_BCLK_INV_MASK 0x0080 /* AIF2_BCLK_INV */ +#define WM5100_AIF2_BCLK_INV_SHIFT 7 /* AIF2_BCLK_INV */ +#define WM5100_AIF2_BCLK_INV_WIDTH 1 /* AIF2_BCLK_INV */ +#define WM5100_AIF2_BCLK_FRC 0x0040 /* AIF2_BCLK_FRC */ +#define WM5100_AIF2_BCLK_FRC_MASK 0x0040 /* AIF2_BCLK_FRC */ +#define WM5100_AIF2_BCLK_FRC_SHIFT 6 /* AIF2_BCLK_FRC */ +#define WM5100_AIF2_BCLK_FRC_WIDTH 1 /* AIF2_BCLK_FRC */ +#define WM5100_AIF2_BCLK_MSTR 0x0020 /* AIF2_BCLK_MSTR */ +#define WM5100_AIF2_BCLK_MSTR_MASK 0x0020 /* AIF2_BCLK_MSTR */ +#define WM5100_AIF2_BCLK_MSTR_SHIFT 5 /* AIF2_BCLK_MSTR */ +#define WM5100_AIF2_BCLK_MSTR_WIDTH 1 /* AIF2_BCLK_MSTR */ +#define WM5100_AIF2_BCLK_FREQ_MASK 0x001F /* AIF2_BCLK_FREQ - [4:0] */ +#define WM5100_AIF2_BCLK_FREQ_SHIFT 0 /* AIF2_BCLK_FREQ - [4:0] */ +#define WM5100_AIF2_BCLK_FREQ_WIDTH 5 /* AIF2_BCLK_FREQ - [4:0] */ + +/* + * R1345 (0x541) - Audio IF 2_2 + */ +#define WM5100_AIF2TX_DAT_TRI 0x0020 /* AIF2TX_DAT_TRI */ +#define WM5100_AIF2TX_DAT_TRI_MASK 0x0020 /* AIF2TX_DAT_TRI */ +#define WM5100_AIF2TX_DAT_TRI_SHIFT 5 /* AIF2TX_DAT_TRI */ +#define WM5100_AIF2TX_DAT_TRI_WIDTH 1 /* AIF2TX_DAT_TRI */ +#define WM5100_AIF2TX_LRCLK_SRC 0x0008 /* AIF2TX_LRCLK_SRC */ +#define WM5100_AIF2TX_LRCLK_SRC_MASK 0x0008 /* AIF2TX_LRCLK_SRC */ +#define WM5100_AIF2TX_LRCLK_SRC_SHIFT 3 /* AIF2TX_LRCLK_SRC */ +#define WM5100_AIF2TX_LRCLK_SRC_WIDTH 1 /* AIF2TX_LRCLK_SRC */ +#define WM5100_AIF2TX_LRCLK_INV 0x0004 /* AIF2TX_LRCLK_INV */ +#define WM5100_AIF2TX_LRCLK_INV_MASK 0x0004 /* AIF2TX_LRCLK_INV */ +#define WM5100_AIF2TX_LRCLK_INV_SHIFT 2 /* AIF2TX_LRCLK_INV */ +#define WM5100_AIF2TX_LRCLK_INV_WIDTH 1 /* AIF2TX_LRCLK_INV */ +#define WM5100_AIF2TX_LRCLK_FRC 0x0002 /* AIF2TX_LRCLK_FRC */ +#define WM5100_AIF2TX_LRCLK_FRC_MASK 0x0002 /* AIF2TX_LRCLK_FRC */ +#define WM5100_AIF2TX_LRCLK_FRC_SHIFT 1 /* AIF2TX_LRCLK_FRC */ +#define WM5100_AIF2TX_LRCLK_FRC_WIDTH 1 /* AIF2TX_LRCLK_FRC */ +#define WM5100_AIF2TX_LRCLK_MSTR 0x0001 /* AIF2TX_LRCLK_MSTR */ +#define WM5100_AIF2TX_LRCLK_MSTR_MASK 0x0001 /* AIF2TX_LRCLK_MSTR */ +#define WM5100_AIF2TX_LRCLK_MSTR_SHIFT 0 /* AIF2TX_LRCLK_MSTR */ +#define WM5100_AIF2TX_LRCLK_MSTR_WIDTH 1 /* AIF2TX_LRCLK_MSTR */ + +/* + * R1346 (0x542) - Audio IF 2_3 + */ +#define WM5100_AIF2RX_LRCLK_INV 0x0004 /* AIF2RX_LRCLK_INV */ +#define WM5100_AIF2RX_LRCLK_INV_MASK 0x0004 /* AIF2RX_LRCLK_INV */ +#define WM5100_AIF2RX_LRCLK_INV_SHIFT 2 /* AIF2RX_LRCLK_INV */ +#define WM5100_AIF2RX_LRCLK_INV_WIDTH 1 /* AIF2RX_LRCLK_INV */ +#define WM5100_AIF2RX_LRCLK_FRC 0x0002 /* AIF2RX_LRCLK_FRC */ +#define WM5100_AIF2RX_LRCLK_FRC_MASK 0x0002 /* AIF2RX_LRCLK_FRC */ +#define WM5100_AIF2RX_LRCLK_FRC_SHIFT 1 /* AIF2RX_LRCLK_FRC */ +#define WM5100_AIF2RX_LRCLK_FRC_WIDTH 1 /* AIF2RX_LRCLK_FRC */ +#define WM5100_AIF2RX_LRCLK_MSTR 0x0001 /* AIF2RX_LRCLK_MSTR */ +#define WM5100_AIF2RX_LRCLK_MSTR_MASK 0x0001 /* AIF2RX_LRCLK_MSTR */ +#define WM5100_AIF2RX_LRCLK_MSTR_SHIFT 0 /* AIF2RX_LRCLK_MSTR */ +#define WM5100_AIF2RX_LRCLK_MSTR_WIDTH 1 /* AIF2RX_LRCLK_MSTR */ + +/* + * R1347 (0x543) - Audio IF 2_4 + */ +#define WM5100_AIF2_TRI 0x0040 /* AIF2_TRI */ +#define WM5100_AIF2_TRI_MASK 0x0040 /* AIF2_TRI */ +#define WM5100_AIF2_TRI_SHIFT 6 /* AIF2_TRI */ +#define WM5100_AIF2_TRI_WIDTH 1 /* AIF2_TRI */ +#define WM5100_AIF2_RATE_MASK 0x0003 /* AIF2_RATE - [1:0] */ +#define WM5100_AIF2_RATE_SHIFT 0 /* AIF2_RATE - [1:0] */ +#define WM5100_AIF2_RATE_WIDTH 2 /* AIF2_RATE - [1:0] */ + +/* + * R1348 (0x544) - Audio IF 2_5 + */ +#define WM5100_AIF2_FMT_MASK 0x0007 /* AIF2_FMT - [2:0] */ +#define WM5100_AIF2_FMT_SHIFT 0 /* AIF2_FMT - [2:0] */ +#define WM5100_AIF2_FMT_WIDTH 3 /* AIF2_FMT - [2:0] */ + +/* + * R1349 (0x545) - Audio IF 2_6 + */ +#define WM5100_AIF2TX_BCPF_MASK 0x1FFF /* AIF2TX_BCPF - [12:0] */ +#define WM5100_AIF2TX_BCPF_SHIFT 0 /* AIF2TX_BCPF - [12:0] */ +#define WM5100_AIF2TX_BCPF_WIDTH 13 /* AIF2TX_BCPF - [12:0] */ + +/* + * R1350 (0x546) - Audio IF 2_7 + */ +#define WM5100_AIF2RX_BCPF_MASK 0x1FFF /* AIF2RX_BCPF - [12:0] */ +#define WM5100_AIF2RX_BCPF_SHIFT 0 /* AIF2RX_BCPF - [12:0] */ +#define WM5100_AIF2RX_BCPF_WIDTH 13 /* AIF2RX_BCPF - [12:0] */ + +/* + * R1351 (0x547) - Audio IF 2_8 + */ +#define WM5100_AIF2TX_WL_MASK 0x3F00 /* AIF2TX_WL - [13:8] */ +#define WM5100_AIF2TX_WL_SHIFT 8 /* AIF2TX_WL - [13:8] */ +#define WM5100_AIF2TX_WL_WIDTH 6 /* AIF2TX_WL - [13:8] */ +#define WM5100_AIF2TX_SLOT_LEN_MASK 0x00FF /* AIF2TX_SLOT_LEN - [7:0] */ +#define WM5100_AIF2TX_SLOT_LEN_SHIFT 0 /* AIF2TX_SLOT_LEN - [7:0] */ +#define WM5100_AIF2TX_SLOT_LEN_WIDTH 8 /* AIF2TX_SLOT_LEN - [7:0] */ + +/* + * R1352 (0x548) - Audio IF 2_9 + */ +#define WM5100_AIF2RX_WL_MASK 0x3F00 /* AIF2RX_WL - [13:8] */ +#define WM5100_AIF2RX_WL_SHIFT 8 /* AIF2RX_WL - [13:8] */ +#define WM5100_AIF2RX_WL_WIDTH 6 /* AIF2RX_WL - [13:8] */ +#define WM5100_AIF2RX_SLOT_LEN_MASK 0x00FF /* AIF2RX_SLOT_LEN - [7:0] */ +#define WM5100_AIF2RX_SLOT_LEN_SHIFT 0 /* AIF2RX_SLOT_LEN - [7:0] */ +#define WM5100_AIF2RX_SLOT_LEN_WIDTH 8 /* AIF2RX_SLOT_LEN - [7:0] */ + +/* + * R1353 (0x549) - Audio IF 2_10 + */ +#define WM5100_AIF2TX1_SLOT_MASK 0x003F /* AIF2TX1_SLOT - [5:0] */ +#define WM5100_AIF2TX1_SLOT_SHIFT 0 /* AIF2TX1_SLOT - [5:0] */ +#define WM5100_AIF2TX1_SLOT_WIDTH 6 /* AIF2TX1_SLOT - [5:0] */ + +/* + * R1354 (0x54A) - Audio IF 2_11 + */ +#define WM5100_AIF2TX2_SLOT_MASK 0x003F /* AIF2TX2_SLOT - [5:0] */ +#define WM5100_AIF2TX2_SLOT_SHIFT 0 /* AIF2TX2_SLOT - [5:0] */ +#define WM5100_AIF2TX2_SLOT_WIDTH 6 /* AIF2TX2_SLOT - [5:0] */ + +/* + * R1361 (0x551) - Audio IF 2_18 + */ +#define WM5100_AIF2RX1_SLOT_MASK 0x003F /* AIF2RX1_SLOT - [5:0] */ +#define WM5100_AIF2RX1_SLOT_SHIFT 0 /* AIF2RX1_SLOT - [5:0] */ +#define WM5100_AIF2RX1_SLOT_WIDTH 6 /* AIF2RX1_SLOT - [5:0] */ + +/* + * R1362 (0x552) - Audio IF 2_19 + */ +#define WM5100_AIF2RX2_SLOT_MASK 0x003F /* AIF2RX2_SLOT - [5:0] */ +#define WM5100_AIF2RX2_SLOT_SHIFT 0 /* AIF2RX2_SLOT - [5:0] */ +#define WM5100_AIF2RX2_SLOT_WIDTH 6 /* AIF2RX2_SLOT - [5:0] */ + +/* + * R1369 (0x559) - Audio IF 2_26 + */ +#define WM5100_AIF2TX2_ENA 0x0002 /* AIF2TX2_ENA */ +#define WM5100_AIF2TX2_ENA_MASK 0x0002 /* AIF2TX2_ENA */ +#define WM5100_AIF2TX2_ENA_SHIFT 1 /* AIF2TX2_ENA */ +#define WM5100_AIF2TX2_ENA_WIDTH 1 /* AIF2TX2_ENA */ +#define WM5100_AIF2TX1_ENA 0x0001 /* AIF2TX1_ENA */ +#define WM5100_AIF2TX1_ENA_MASK 0x0001 /* AIF2TX1_ENA */ +#define WM5100_AIF2TX1_ENA_SHIFT 0 /* AIF2TX1_ENA */ +#define WM5100_AIF2TX1_ENA_WIDTH 1 /* AIF2TX1_ENA */ + +/* + * R1370 (0x55A) - Audio IF 2_27 + */ +#define WM5100_AIF2RX2_ENA 0x0002 /* AIF2RX2_ENA */ +#define WM5100_AIF2RX2_ENA_MASK 0x0002 /* AIF2RX2_ENA */ +#define WM5100_AIF2RX2_ENA_SHIFT 1 /* AIF2RX2_ENA */ +#define WM5100_AIF2RX2_ENA_WIDTH 1 /* AIF2RX2_ENA */ +#define WM5100_AIF2RX1_ENA 0x0001 /* AIF2RX1_ENA */ +#define WM5100_AIF2RX1_ENA_MASK 0x0001 /* AIF2RX1_ENA */ +#define WM5100_AIF2RX1_ENA_SHIFT 0 /* AIF2RX1_ENA */ +#define WM5100_AIF2RX1_ENA_WIDTH 1 /* AIF2RX1_ENA */ + +/* + * R1408 (0x580) - Audio IF 3_1 + */ +#define WM5100_AIF3_BCLK_INV 0x0080 /* AIF3_BCLK_INV */ +#define WM5100_AIF3_BCLK_INV_MASK 0x0080 /* AIF3_BCLK_INV */ +#define WM5100_AIF3_BCLK_INV_SHIFT 7 /* AIF3_BCLK_INV */ +#define WM5100_AIF3_BCLK_INV_WIDTH 1 /* AIF3_BCLK_INV */ +#define WM5100_AIF3_BCLK_FRC 0x0040 /* AIF3_BCLK_FRC */ +#define WM5100_AIF3_BCLK_FRC_MASK 0x0040 /* AIF3_BCLK_FRC */ +#define WM5100_AIF3_BCLK_FRC_SHIFT 6 /* AIF3_BCLK_FRC */ +#define WM5100_AIF3_BCLK_FRC_WIDTH 1 /* AIF3_BCLK_FRC */ +#define WM5100_AIF3_BCLK_MSTR 0x0020 /* AIF3_BCLK_MSTR */ +#define WM5100_AIF3_BCLK_MSTR_MASK 0x0020 /* AIF3_BCLK_MSTR */ +#define WM5100_AIF3_BCLK_MSTR_SHIFT 5 /* AIF3_BCLK_MSTR */ +#define WM5100_AIF3_BCLK_MSTR_WIDTH 1 /* AIF3_BCLK_MSTR */ +#define WM5100_AIF3_BCLK_FREQ_MASK 0x001F /* AIF3_BCLK_FREQ - [4:0] */ +#define WM5100_AIF3_BCLK_FREQ_SHIFT 0 /* AIF3_BCLK_FREQ - [4:0] */ +#define WM5100_AIF3_BCLK_FREQ_WIDTH 5 /* AIF3_BCLK_FREQ - [4:0] */ + +/* + * R1409 (0x581) - Audio IF 3_2 + */ +#define WM5100_AIF3TX_DAT_TRI 0x0020 /* AIF3TX_DAT_TRI */ +#define WM5100_AIF3TX_DAT_TRI_MASK 0x0020 /* AIF3TX_DAT_TRI */ +#define WM5100_AIF3TX_DAT_TRI_SHIFT 5 /* AIF3TX_DAT_TRI */ +#define WM5100_AIF3TX_DAT_TRI_WIDTH 1 /* AIF3TX_DAT_TRI */ +#define WM5100_AIF3TX_LRCLK_SRC 0x0008 /* AIF3TX_LRCLK_SRC */ +#define WM5100_AIF3TX_LRCLK_SRC_MASK 0x0008 /* AIF3TX_LRCLK_SRC */ +#define WM5100_AIF3TX_LRCLK_SRC_SHIFT 3 /* AIF3TX_LRCLK_SRC */ +#define WM5100_AIF3TX_LRCLK_SRC_WIDTH 1 /* AIF3TX_LRCLK_SRC */ +#define WM5100_AIF3TX_LRCLK_INV 0x0004 /* AIF3TX_LRCLK_INV */ +#define WM5100_AIF3TX_LRCLK_INV_MASK 0x0004 /* AIF3TX_LRCLK_INV */ +#define WM5100_AIF3TX_LRCLK_INV_SHIFT 2 /* AIF3TX_LRCLK_INV */ +#define WM5100_AIF3TX_LRCLK_INV_WIDTH 1 /* AIF3TX_LRCLK_INV */ +#define WM5100_AIF3TX_LRCLK_FRC 0x0002 /* AIF3TX_LRCLK_FRC */ +#define WM5100_AIF3TX_LRCLK_FRC_MASK 0x0002 /* AIF3TX_LRCLK_FRC */ +#define WM5100_AIF3TX_LRCLK_FRC_SHIFT 1 /* AIF3TX_LRCLK_FRC */ +#define WM5100_AIF3TX_LRCLK_FRC_WIDTH 1 /* AIF3TX_LRCLK_FRC */ +#define WM5100_AIF3TX_LRCLK_MSTR 0x0001 /* AIF3TX_LRCLK_MSTR */ +#define WM5100_AIF3TX_LRCLK_MSTR_MASK 0x0001 /* AIF3TX_LRCLK_MSTR */ +#define WM5100_AIF3TX_LRCLK_MSTR_SHIFT 0 /* AIF3TX_LRCLK_MSTR */ +#define WM5100_AIF3TX_LRCLK_MSTR_WIDTH 1 /* AIF3TX_LRCLK_MSTR */ + +/* + * R1410 (0x582) - Audio IF 3_3 + */ +#define WM5100_AIF3RX_LRCLK_INV 0x0004 /* AIF3RX_LRCLK_INV */ +#define WM5100_AIF3RX_LRCLK_INV_MASK 0x0004 /* AIF3RX_LRCLK_INV */ +#define WM5100_AIF3RX_LRCLK_INV_SHIFT 2 /* AIF3RX_LRCLK_INV */ +#define WM5100_AIF3RX_LRCLK_INV_WIDTH 1 /* AIF3RX_LRCLK_INV */ +#define WM5100_AIF3RX_LRCLK_FRC 0x0002 /* AIF3RX_LRCLK_FRC */ +#define WM5100_AIF3RX_LRCLK_FRC_MASK 0x0002 /* AIF3RX_LRCLK_FRC */ +#define WM5100_AIF3RX_LRCLK_FRC_SHIFT 1 /* AIF3RX_LRCLK_FRC */ +#define WM5100_AIF3RX_LRCLK_FRC_WIDTH 1 /* AIF3RX_LRCLK_FRC */ +#define WM5100_AIF3RX_LRCLK_MSTR 0x0001 /* AIF3RX_LRCLK_MSTR */ +#define WM5100_AIF3RX_LRCLK_MSTR_MASK 0x0001 /* AIF3RX_LRCLK_MSTR */ +#define WM5100_AIF3RX_LRCLK_MSTR_SHIFT 0 /* AIF3RX_LRCLK_MSTR */ +#define WM5100_AIF3RX_LRCLK_MSTR_WIDTH 1 /* AIF3RX_LRCLK_MSTR */ + +/* + * R1411 (0x583) - Audio IF 3_4 + */ +#define WM5100_AIF3_TRI 0x0040 /* AIF3_TRI */ +#define WM5100_AIF3_TRI_MASK 0x0040 /* AIF3_TRI */ +#define WM5100_AIF3_TRI_SHIFT 6 /* AIF3_TRI */ +#define WM5100_AIF3_TRI_WIDTH 1 /* AIF3_TRI */ +#define WM5100_AIF3_RATE_MASK 0x0003 /* AIF3_RATE - [1:0] */ +#define WM5100_AIF3_RATE_SHIFT 0 /* AIF3_RATE - [1:0] */ +#define WM5100_AIF3_RATE_WIDTH 2 /* AIF3_RATE - [1:0] */ + +/* + * R1412 (0x584) - Audio IF 3_5 + */ +#define WM5100_AIF3_FMT_MASK 0x0007 /* AIF3_FMT - [2:0] */ +#define WM5100_AIF3_FMT_SHIFT 0 /* AIF3_FMT - [2:0] */ +#define WM5100_AIF3_FMT_WIDTH 3 /* AIF3_FMT - [2:0] */ + +/* + * R1413 (0x585) - Audio IF 3_6 + */ +#define WM5100_AIF3TX_BCPF_MASK 0x1FFF /* AIF3TX_BCPF - [12:0] */ +#define WM5100_AIF3TX_BCPF_SHIFT 0 /* AIF3TX_BCPF - [12:0] */ +#define WM5100_AIF3TX_BCPF_WIDTH 13 /* AIF3TX_BCPF - [12:0] */ + +/* + * R1414 (0x586) - Audio IF 3_7 + */ +#define WM5100_AIF3RX_BCPF_MASK 0x1FFF /* AIF3RX_BCPF - [12:0] */ +#define WM5100_AIF3RX_BCPF_SHIFT 0 /* AIF3RX_BCPF - [12:0] */ +#define WM5100_AIF3RX_BCPF_WIDTH 13 /* AIF3RX_BCPF - [12:0] */ + +/* + * R1415 (0x587) - Audio IF 3_8 + */ +#define WM5100_AIF3TX_WL_MASK 0x3F00 /* AIF3TX_WL - [13:8] */ +#define WM5100_AIF3TX_WL_SHIFT 8 /* AIF3TX_WL - [13:8] */ +#define WM5100_AIF3TX_WL_WIDTH 6 /* AIF3TX_WL - [13:8] */ +#define WM5100_AIF3TX_SLOT_LEN_MASK 0x00FF /* AIF3TX_SLOT_LEN - [7:0] */ +#define WM5100_AIF3TX_SLOT_LEN_SHIFT 0 /* AIF3TX_SLOT_LEN - [7:0] */ +#define WM5100_AIF3TX_SLOT_LEN_WIDTH 8 /* AIF3TX_SLOT_LEN - [7:0] */ + +/* + * R1416 (0x588) - Audio IF 3_9 + */ +#define WM5100_AIF3RX_WL_MASK 0x3F00 /* AIF3RX_WL - [13:8] */ +#define WM5100_AIF3RX_WL_SHIFT 8 /* AIF3RX_WL - [13:8] */ +#define WM5100_AIF3RX_WL_WIDTH 6 /* AIF3RX_WL - [13:8] */ +#define WM5100_AIF3RX_SLOT_LEN_MASK 0x00FF /* AIF3RX_SLOT_LEN - [7:0] */ +#define WM5100_AIF3RX_SLOT_LEN_SHIFT 0 /* AIF3RX_SLOT_LEN - [7:0] */ +#define WM5100_AIF3RX_SLOT_LEN_WIDTH 8 /* AIF3RX_SLOT_LEN - [7:0] */ + +/* + * R1417 (0x589) - Audio IF 3_10 + */ +#define WM5100_AIF3TX1_SLOT_MASK 0x003F /* AIF3TX1_SLOT - [5:0] */ +#define WM5100_AIF3TX1_SLOT_SHIFT 0 /* AIF3TX1_SLOT - [5:0] */ +#define WM5100_AIF3TX1_SLOT_WIDTH 6 /* AIF3TX1_SLOT - [5:0] */ + +/* + * R1418 (0x58A) - Audio IF 3_11 + */ +#define WM5100_AIF3TX2_SLOT_MASK 0x003F /* AIF3TX2_SLOT - [5:0] */ +#define WM5100_AIF3TX2_SLOT_SHIFT 0 /* AIF3TX2_SLOT - [5:0] */ +#define WM5100_AIF3TX2_SLOT_WIDTH 6 /* AIF3TX2_SLOT - [5:0] */ + +/* + * R1425 (0x591) - Audio IF 3_18 + */ +#define WM5100_AIF3RX1_SLOT_MASK 0x003F /* AIF3RX1_SLOT - [5:0] */ +#define WM5100_AIF3RX1_SLOT_SHIFT 0 /* AIF3RX1_SLOT - [5:0] */ +#define WM5100_AIF3RX1_SLOT_WIDTH 6 /* AIF3RX1_SLOT - [5:0] */ + +/* + * R1426 (0x592) - Audio IF 3_19 + */ +#define WM5100_AIF3RX2_SLOT_MASK 0x003F /* AIF3RX2_SLOT - [5:0] */ +#define WM5100_AIF3RX2_SLOT_SHIFT 0 /* AIF3RX2_SLOT - [5:0] */ +#define WM5100_AIF3RX2_SLOT_WIDTH 6 /* AIF3RX2_SLOT - [5:0] */ + +/* + * R1433 (0x599) - Audio IF 3_26 + */ +#define WM5100_AIF3TX2_ENA 0x0002 /* AIF3TX2_ENA */ +#define WM5100_AIF3TX2_ENA_MASK 0x0002 /* AIF3TX2_ENA */ +#define WM5100_AIF3TX2_ENA_SHIFT 1 /* AIF3TX2_ENA */ +#define WM5100_AIF3TX2_ENA_WIDTH 1 /* AIF3TX2_ENA */ +#define WM5100_AIF3TX1_ENA 0x0001 /* AIF3TX1_ENA */ +#define WM5100_AIF3TX1_ENA_MASK 0x0001 /* AIF3TX1_ENA */ +#define WM5100_AIF3TX1_ENA_SHIFT 0 /* AIF3TX1_ENA */ +#define WM5100_AIF3TX1_ENA_WIDTH 1 /* AIF3TX1_ENA */ + +/* + * R1434 (0x59A) - Audio IF 3_27 + */ +#define WM5100_AIF3RX2_ENA 0x0002 /* AIF3RX2_ENA */ +#define WM5100_AIF3RX2_ENA_MASK 0x0002 /* AIF3RX2_ENA */ +#define WM5100_AIF3RX2_ENA_SHIFT 1 /* AIF3RX2_ENA */ +#define WM5100_AIF3RX2_ENA_WIDTH 1 /* AIF3RX2_ENA */ +#define WM5100_AIF3RX1_ENA 0x0001 /* AIF3RX1_ENA */ +#define WM5100_AIF3RX1_ENA_MASK 0x0001 /* AIF3RX1_ENA */ +#define WM5100_AIF3RX1_ENA_SHIFT 0 /* AIF3RX1_ENA */ +#define WM5100_AIF3RX1_ENA_WIDTH 1 /* AIF3RX1_ENA */ + +#define WM5100_MIXER_VOL_MASK 0x00FE /* MIXER_VOL - [7:1] */ +#define WM5100_MIXER_VOL_SHIFT 1 /* MIXER_VOL - [7:1] */ +#define WM5100_MIXER_VOL_WIDTH 7 /* MIXER_VOL - [7:1] */ + +/* + * R3072 (0xC00) - GPIO CTRL 1 + */ +#define WM5100_GP1_DIR 0x8000 /* GP1_DIR */ +#define WM5100_GP1_DIR_MASK 0x8000 /* GP1_DIR */ +#define WM5100_GP1_DIR_SHIFT 15 /* GP1_DIR */ +#define WM5100_GP1_DIR_WIDTH 1 /* GP1_DIR */ +#define WM5100_GP1_PU 0x4000 /* GP1_PU */ +#define WM5100_GP1_PU_MASK 0x4000 /* GP1_PU */ +#define WM5100_GP1_PU_SHIFT 14 /* GP1_PU */ +#define WM5100_GP1_PU_WIDTH 1 /* GP1_PU */ +#define WM5100_GP1_PD 0x2000 /* GP1_PD */ +#define WM5100_GP1_PD_MASK 0x2000 /* GP1_PD */ +#define WM5100_GP1_PD_SHIFT 13 /* GP1_PD */ +#define WM5100_GP1_PD_WIDTH 1 /* GP1_PD */ +#define WM5100_GP1_POL 0x0400 /* GP1_POL */ +#define WM5100_GP1_POL_MASK 0x0400 /* GP1_POL */ +#define WM5100_GP1_POL_SHIFT 10 /* GP1_POL */ +#define WM5100_GP1_POL_WIDTH 1 /* GP1_POL */ +#define WM5100_GP1_OP_CFG 0x0200 /* GP1_OP_CFG */ +#define WM5100_GP1_OP_CFG_MASK 0x0200 /* GP1_OP_CFG */ +#define WM5100_GP1_OP_CFG_SHIFT 9 /* GP1_OP_CFG */ +#define WM5100_GP1_OP_CFG_WIDTH 1 /* GP1_OP_CFG */ +#define WM5100_GP1_DB 0x0100 /* GP1_DB */ +#define WM5100_GP1_DB_MASK 0x0100 /* GP1_DB */ +#define WM5100_GP1_DB_SHIFT 8 /* GP1_DB */ +#define WM5100_GP1_DB_WIDTH 1 /* GP1_DB */ +#define WM5100_GP1_LVL 0x0040 /* GP1_LVL */ +#define WM5100_GP1_LVL_MASK 0x0040 /* GP1_LVL */ +#define WM5100_GP1_LVL_SHIFT 6 /* GP1_LVL */ +#define WM5100_GP1_LVL_WIDTH 1 /* GP1_LVL */ +#define WM5100_GP1_FN_MASK 0x003F /* GP1_FN - [5:0] */ +#define WM5100_GP1_FN_SHIFT 0 /* GP1_FN - [5:0] */ +#define WM5100_GP1_FN_WIDTH 6 /* GP1_FN - [5:0] */ + +/* + * R3073 (0xC01) - GPIO CTRL 2 + */ +#define WM5100_GP2_DIR 0x8000 /* GP2_DIR */ +#define WM5100_GP2_DIR_MASK 0x8000 /* GP2_DIR */ +#define WM5100_GP2_DIR_SHIFT 15 /* GP2_DIR */ +#define WM5100_GP2_DIR_WIDTH 1 /* GP2_DIR */ +#define WM5100_GP2_PU 0x4000 /* GP2_PU */ +#define WM5100_GP2_PU_MASK 0x4000 /* GP2_PU */ +#define WM5100_GP2_PU_SHIFT 14 /* GP2_PU */ +#define WM5100_GP2_PU_WIDTH 1 /* GP2_PU */ +#define WM5100_GP2_PD 0x2000 /* GP2_PD */ +#define WM5100_GP2_PD_MASK 0x2000 /* GP2_PD */ +#define WM5100_GP2_PD_SHIFT 13 /* GP2_PD */ +#define WM5100_GP2_PD_WIDTH 1 /* GP2_PD */ +#define WM5100_GP2_POL 0x0400 /* GP2_POL */ +#define WM5100_GP2_POL_MASK 0x0400 /* GP2_POL */ +#define WM5100_GP2_POL_SHIFT 10 /* GP2_POL */ +#define WM5100_GP2_POL_WIDTH 1 /* GP2_POL */ +#define WM5100_GP2_OP_CFG 0x0200 /* GP2_OP_CFG */ +#define WM5100_GP2_OP_CFG_MASK 0x0200 /* GP2_OP_CFG */ +#define WM5100_GP2_OP_CFG_SHIFT 9 /* GP2_OP_CFG */ +#define WM5100_GP2_OP_CFG_WIDTH 1 /* GP2_OP_CFG */ +#define WM5100_GP2_DB 0x0100 /* GP2_DB */ +#define WM5100_GP2_DB_MASK 0x0100 /* GP2_DB */ +#define WM5100_GP2_DB_SHIFT 8 /* GP2_DB */ +#define WM5100_GP2_DB_WIDTH 1 /* GP2_DB */ +#define WM5100_GP2_LVL 0x0040 /* GP2_LVL */ +#define WM5100_GP2_LVL_MASK 0x0040 /* GP2_LVL */ +#define WM5100_GP2_LVL_SHIFT 6 /* GP2_LVL */ +#define WM5100_GP2_LVL_WIDTH 1 /* GP2_LVL */ +#define WM5100_GP2_FN_MASK 0x003F /* GP2_FN - [5:0] */ +#define WM5100_GP2_FN_SHIFT 0 /* GP2_FN - [5:0] */ +#define WM5100_GP2_FN_WIDTH 6 /* GP2_FN - [5:0] */ + +/* + * R3074 (0xC02) - GPIO CTRL 3 + */ +#define WM5100_GP3_DIR 0x8000 /* GP3_DIR */ +#define WM5100_GP3_DIR_MASK 0x8000 /* GP3_DIR */ +#define WM5100_GP3_DIR_SHIFT 15 /* GP3_DIR */ +#define WM5100_GP3_DIR_WIDTH 1 /* GP3_DIR */ +#define WM5100_GP3_PU 0x4000 /* GP3_PU */ +#define WM5100_GP3_PU_MASK 0x4000 /* GP3_PU */ +#define WM5100_GP3_PU_SHIFT 14 /* GP3_PU */ +#define WM5100_GP3_PU_WIDTH 1 /* GP3_PU */ +#define WM5100_GP3_PD 0x2000 /* GP3_PD */ +#define WM5100_GP3_PD_MASK 0x2000 /* GP3_PD */ +#define WM5100_GP3_PD_SHIFT 13 /* GP3_PD */ +#define WM5100_GP3_PD_WIDTH 1 /* GP3_PD */ +#define WM5100_GP3_POL 0x0400 /* GP3_POL */ +#define WM5100_GP3_POL_MASK 0x0400 /* GP3_POL */ +#define WM5100_GP3_POL_SHIFT 10 /* GP3_POL */ +#define WM5100_GP3_POL_WIDTH 1 /* GP3_POL */ +#define WM5100_GP3_OP_CFG 0x0200 /* GP3_OP_CFG */ +#define WM5100_GP3_OP_CFG_MASK 0x0200 /* GP3_OP_CFG */ +#define WM5100_GP3_OP_CFG_SHIFT 9 /* GP3_OP_CFG */ +#define WM5100_GP3_OP_CFG_WIDTH 1 /* GP3_OP_CFG */ +#define WM5100_GP3_DB 0x0100 /* GP3_DB */ +#define WM5100_GP3_DB_MASK 0x0100 /* GP3_DB */ +#define WM5100_GP3_DB_SHIFT 8 /* GP3_DB */ +#define WM5100_GP3_DB_WIDTH 1 /* GP3_DB */ +#define WM5100_GP3_LVL 0x0040 /* GP3_LVL */ +#define WM5100_GP3_LVL_MASK 0x0040 /* GP3_LVL */ +#define WM5100_GP3_LVL_SHIFT 6 /* GP3_LVL */ +#define WM5100_GP3_LVL_WIDTH 1 /* GP3_LVL */ +#define WM5100_GP3_FN_MASK 0x003F /* GP3_FN - [5:0] */ +#define WM5100_GP3_FN_SHIFT 0 /* GP3_FN - [5:0] */ +#define WM5100_GP3_FN_WIDTH 6 /* GP3_FN - [5:0] */ + +/* + * R3075 (0xC03) - GPIO CTRL 4 + */ +#define WM5100_GP4_DIR 0x8000 /* GP4_DIR */ +#define WM5100_GP4_DIR_MASK 0x8000 /* GP4_DIR */ +#define WM5100_GP4_DIR_SHIFT 15 /* GP4_DIR */ +#define WM5100_GP4_DIR_WIDTH 1 /* GP4_DIR */ +#define WM5100_GP4_PU 0x4000 /* GP4_PU */ +#define WM5100_GP4_PU_MASK 0x4000 /* GP4_PU */ +#define WM5100_GP4_PU_SHIFT 14 /* GP4_PU */ +#define WM5100_GP4_PU_WIDTH 1 /* GP4_PU */ +#define WM5100_GP4_PD 0x2000 /* GP4_PD */ +#define WM5100_GP4_PD_MASK 0x2000 /* GP4_PD */ +#define WM5100_GP4_PD_SHIFT 13 /* GP4_PD */ +#define WM5100_GP4_PD_WIDTH 1 /* GP4_PD */ +#define WM5100_GP4_POL 0x0400 /* GP4_POL */ +#define WM5100_GP4_POL_MASK 0x0400 /* GP4_POL */ +#define WM5100_GP4_POL_SHIFT 10 /* GP4_POL */ +#define WM5100_GP4_POL_WIDTH 1 /* GP4_POL */ +#define WM5100_GP4_OP_CFG 0x0200 /* GP4_OP_CFG */ +#define WM5100_GP4_OP_CFG_MASK 0x0200 /* GP4_OP_CFG */ +#define WM5100_GP4_OP_CFG_SHIFT 9 /* GP4_OP_CFG */ +#define WM5100_GP4_OP_CFG_WIDTH 1 /* GP4_OP_CFG */ +#define WM5100_GP4_DB 0x0100 /* GP4_DB */ +#define WM5100_GP4_DB_MASK 0x0100 /* GP4_DB */ +#define WM5100_GP4_DB_SHIFT 8 /* GP4_DB */ +#define WM5100_GP4_DB_WIDTH 1 /* GP4_DB */ +#define WM5100_GP4_LVL 0x0040 /* GP4_LVL */ +#define WM5100_GP4_LVL_MASK 0x0040 /* GP4_LVL */ +#define WM5100_GP4_LVL_SHIFT 6 /* GP4_LVL */ +#define WM5100_GP4_LVL_WIDTH 1 /* GP4_LVL */ +#define WM5100_GP4_FN_MASK 0x003F /* GP4_FN - [5:0] */ +#define WM5100_GP4_FN_SHIFT 0 /* GP4_FN - [5:0] */ +#define WM5100_GP4_FN_WIDTH 6 /* GP4_FN - [5:0] */ + +/* + * R3076 (0xC04) - GPIO CTRL 5 + */ +#define WM5100_GP5_DIR 0x8000 /* GP5_DIR */ +#define WM5100_GP5_DIR_MASK 0x8000 /* GP5_DIR */ +#define WM5100_GP5_DIR_SHIFT 15 /* GP5_DIR */ +#define WM5100_GP5_DIR_WIDTH 1 /* GP5_DIR */ +#define WM5100_GP5_PU 0x4000 /* GP5_PU */ +#define WM5100_GP5_PU_MASK 0x4000 /* GP5_PU */ +#define WM5100_GP5_PU_SHIFT 14 /* GP5_PU */ +#define WM5100_GP5_PU_WIDTH 1 /* GP5_PU */ +#define WM5100_GP5_PD 0x2000 /* GP5_PD */ +#define WM5100_GP5_PD_MASK 0x2000 /* GP5_PD */ +#define WM5100_GP5_PD_SHIFT 13 /* GP5_PD */ +#define WM5100_GP5_PD_WIDTH 1 /* GP5_PD */ +#define WM5100_GP5_POL 0x0400 /* GP5_POL */ +#define WM5100_GP5_POL_MASK 0x0400 /* GP5_POL */ +#define WM5100_GP5_POL_SHIFT 10 /* GP5_POL */ +#define WM5100_GP5_POL_WIDTH 1 /* GP5_POL */ +#define WM5100_GP5_OP_CFG 0x0200 /* GP5_OP_CFG */ +#define WM5100_GP5_OP_CFG_MASK 0x0200 /* GP5_OP_CFG */ +#define WM5100_GP5_OP_CFG_SHIFT 9 /* GP5_OP_CFG */ +#define WM5100_GP5_OP_CFG_WIDTH 1 /* GP5_OP_CFG */ +#define WM5100_GP5_DB 0x0100 /* GP5_DB */ +#define WM5100_GP5_DB_MASK 0x0100 /* GP5_DB */ +#define WM5100_GP5_DB_SHIFT 8 /* GP5_DB */ +#define WM5100_GP5_DB_WIDTH 1 /* GP5_DB */ +#define WM5100_GP5_LVL 0x0040 /* GP5_LVL */ +#define WM5100_GP5_LVL_MASK 0x0040 /* GP5_LVL */ +#define WM5100_GP5_LVL_SHIFT 6 /* GP5_LVL */ +#define WM5100_GP5_LVL_WIDTH 1 /* GP5_LVL */ +#define WM5100_GP5_FN_MASK 0x003F /* GP5_FN - [5:0] */ +#define WM5100_GP5_FN_SHIFT 0 /* GP5_FN - [5:0] */ +#define WM5100_GP5_FN_WIDTH 6 /* GP5_FN - [5:0] */ + +/* + * R3077 (0xC05) - GPIO CTRL 6 + */ +#define WM5100_GP6_DIR 0x8000 /* GP6_DIR */ +#define WM5100_GP6_DIR_MASK 0x8000 /* GP6_DIR */ +#define WM5100_GP6_DIR_SHIFT 15 /* GP6_DIR */ +#define WM5100_GP6_DIR_WIDTH 1 /* GP6_DIR */ +#define WM5100_GP6_PU 0x4000 /* GP6_PU */ +#define WM5100_GP6_PU_MASK 0x4000 /* GP6_PU */ +#define WM5100_GP6_PU_SHIFT 14 /* GP6_PU */ +#define WM5100_GP6_PU_WIDTH 1 /* GP6_PU */ +#define WM5100_GP6_PD 0x2000 /* GP6_PD */ +#define WM5100_GP6_PD_MASK 0x2000 /* GP6_PD */ +#define WM5100_GP6_PD_SHIFT 13 /* GP6_PD */ +#define WM5100_GP6_PD_WIDTH 1 /* GP6_PD */ +#define WM5100_GP6_POL 0x0400 /* GP6_POL */ +#define WM5100_GP6_POL_MASK 0x0400 /* GP6_POL */ +#define WM5100_GP6_POL_SHIFT 10 /* GP6_POL */ +#define WM5100_GP6_POL_WIDTH 1 /* GP6_POL */ +#define WM5100_GP6_OP_CFG 0x0200 /* GP6_OP_CFG */ +#define WM5100_GP6_OP_CFG_MASK 0x0200 /* GP6_OP_CFG */ +#define WM5100_GP6_OP_CFG_SHIFT 9 /* GP6_OP_CFG */ +#define WM5100_GP6_OP_CFG_WIDTH 1 /* GP6_OP_CFG */ +#define WM5100_GP6_DB 0x0100 /* GP6_DB */ +#define WM5100_GP6_DB_MASK 0x0100 /* GP6_DB */ +#define WM5100_GP6_DB_SHIFT 8 /* GP6_DB */ +#define WM5100_GP6_DB_WIDTH 1 /* GP6_DB */ +#define WM5100_GP6_LVL 0x0040 /* GP6_LVL */ +#define WM5100_GP6_LVL_MASK 0x0040 /* GP6_LVL */ +#define WM5100_GP6_LVL_SHIFT 6 /* GP6_LVL */ +#define WM5100_GP6_LVL_WIDTH 1 /* GP6_LVL */ +#define WM5100_GP6_FN_MASK 0x003F /* GP6_FN - [5:0] */ +#define WM5100_GP6_FN_SHIFT 0 /* GP6_FN - [5:0] */ +#define WM5100_GP6_FN_WIDTH 6 /* GP6_FN - [5:0] */ + +/* + * R3107 (0xC23) - Misc Pad Ctrl 1 + */ +#define WM5100_LDO1ENA_PD 0x8000 /* LDO1ENA_PD */ +#define WM5100_LDO1ENA_PD_MASK 0x8000 /* LDO1ENA_PD */ +#define WM5100_LDO1ENA_PD_SHIFT 15 /* LDO1ENA_PD */ +#define WM5100_LDO1ENA_PD_WIDTH 1 /* LDO1ENA_PD */ +#define WM5100_MCLK2_PD 0x2000 /* MCLK2_PD */ +#define WM5100_MCLK2_PD_MASK 0x2000 /* MCLK2_PD */ +#define WM5100_MCLK2_PD_SHIFT 13 /* MCLK2_PD */ +#define WM5100_MCLK2_PD_WIDTH 1 /* MCLK2_PD */ +#define WM5100_MCLK1_PD 0x1000 /* MCLK1_PD */ +#define WM5100_MCLK1_PD_MASK 0x1000 /* MCLK1_PD */ +#define WM5100_MCLK1_PD_SHIFT 12 /* MCLK1_PD */ +#define WM5100_MCLK1_PD_WIDTH 1 /* MCLK1_PD */ +#define WM5100_RESET_PU 0x0002 /* RESET_PU */ +#define WM5100_RESET_PU_MASK 0x0002 /* RESET_PU */ +#define WM5100_RESET_PU_SHIFT 1 /* RESET_PU */ +#define WM5100_RESET_PU_WIDTH 1 /* RESET_PU */ +#define WM5100_ADDR_PD 0x0001 /* ADDR_PD */ +#define WM5100_ADDR_PD_MASK 0x0001 /* ADDR_PD */ +#define WM5100_ADDR_PD_SHIFT 0 /* ADDR_PD */ +#define WM5100_ADDR_PD_WIDTH 1 /* ADDR_PD */ + +/* + * R3108 (0xC24) - Misc Pad Ctrl 2 + */ +#define WM5100_DMICDAT4_PD 0x0008 /* DMICDAT4_PD */ +#define WM5100_DMICDAT4_PD_MASK 0x0008 /* DMICDAT4_PD */ +#define WM5100_DMICDAT4_PD_SHIFT 3 /* DMICDAT4_PD */ +#define WM5100_DMICDAT4_PD_WIDTH 1 /* DMICDAT4_PD */ +#define WM5100_DMICDAT3_PD 0x0004 /* DMICDAT3_PD */ +#define WM5100_DMICDAT3_PD_MASK 0x0004 /* DMICDAT3_PD */ +#define WM5100_DMICDAT3_PD_SHIFT 2 /* DMICDAT3_PD */ +#define WM5100_DMICDAT3_PD_WIDTH 1 /* DMICDAT3_PD */ +#define WM5100_DMICDAT2_PD 0x0002 /* DMICDAT2_PD */ +#define WM5100_DMICDAT2_PD_MASK 0x0002 /* DMICDAT2_PD */ +#define WM5100_DMICDAT2_PD_SHIFT 1 /* DMICDAT2_PD */ +#define WM5100_DMICDAT2_PD_WIDTH 1 /* DMICDAT2_PD */ +#define WM5100_DMICDAT1_PD 0x0001 /* DMICDAT1_PD */ +#define WM5100_DMICDAT1_PD_MASK 0x0001 /* DMICDAT1_PD */ +#define WM5100_DMICDAT1_PD_SHIFT 0 /* DMICDAT1_PD */ +#define WM5100_DMICDAT1_PD_WIDTH 1 /* DMICDAT1_PD */ + +/* + * R3109 (0xC25) - Misc Pad Ctrl 3 + */ +#define WM5100_AIF1RXLRCLK_PU 0x0020 /* AIF1RXLRCLK_PU */ +#define WM5100_AIF1RXLRCLK_PU_MASK 0x0020 /* AIF1RXLRCLK_PU */ +#define WM5100_AIF1RXLRCLK_PU_SHIFT 5 /* AIF1RXLRCLK_PU */ +#define WM5100_AIF1RXLRCLK_PU_WIDTH 1 /* AIF1RXLRCLK_PU */ +#define WM5100_AIF1RXLRCLK_PD 0x0010 /* AIF1RXLRCLK_PD */ +#define WM5100_AIF1RXLRCLK_PD_MASK 0x0010 /* AIF1RXLRCLK_PD */ +#define WM5100_AIF1RXLRCLK_PD_SHIFT 4 /* AIF1RXLRCLK_PD */ +#define WM5100_AIF1RXLRCLK_PD_WIDTH 1 /* AIF1RXLRCLK_PD */ +#define WM5100_AIF1BCLK_PU 0x0008 /* AIF1BCLK_PU */ +#define WM5100_AIF1BCLK_PU_MASK 0x0008 /* AIF1BCLK_PU */ +#define WM5100_AIF1BCLK_PU_SHIFT 3 /* AIF1BCLK_PU */ +#define WM5100_AIF1BCLK_PU_WIDTH 1 /* AIF1BCLK_PU */ +#define WM5100_AIF1BCLK_PD 0x0004 /* AIF1BCLK_PD */ +#define WM5100_AIF1BCLK_PD_MASK 0x0004 /* AIF1BCLK_PD */ +#define WM5100_AIF1BCLK_PD_SHIFT 2 /* AIF1BCLK_PD */ +#define WM5100_AIF1BCLK_PD_WIDTH 1 /* AIF1BCLK_PD */ +#define WM5100_AIF1RXDAT_PU 0x0002 /* AIF1RXDAT_PU */ +#define WM5100_AIF1RXDAT_PU_MASK 0x0002 /* AIF1RXDAT_PU */ +#define WM5100_AIF1RXDAT_PU_SHIFT 1 /* AIF1RXDAT_PU */ +#define WM5100_AIF1RXDAT_PU_WIDTH 1 /* AIF1RXDAT_PU */ +#define WM5100_AIF1RXDAT_PD 0x0001 /* AIF1RXDAT_PD */ +#define WM5100_AIF1RXDAT_PD_MASK 0x0001 /* AIF1RXDAT_PD */ +#define WM5100_AIF1RXDAT_PD_SHIFT 0 /* AIF1RXDAT_PD */ +#define WM5100_AIF1RXDAT_PD_WIDTH 1 /* AIF1RXDAT_PD */ + +/* + * R3110 (0xC26) - Misc Pad Ctrl 4 + */ +#define WM5100_AIF2RXLRCLK_PU 0x0020 /* AIF2RXLRCLK_PU */ +#define WM5100_AIF2RXLRCLK_PU_MASK 0x0020 /* AIF2RXLRCLK_PU */ +#define WM5100_AIF2RXLRCLK_PU_SHIFT 5 /* AIF2RXLRCLK_PU */ +#define WM5100_AIF2RXLRCLK_PU_WIDTH 1 /* AIF2RXLRCLK_PU */ +#define WM5100_AIF2RXLRCLK_PD 0x0010 /* AIF2RXLRCLK_PD */ +#define WM5100_AIF2RXLRCLK_PD_MASK 0x0010 /* AIF2RXLRCLK_PD */ +#define WM5100_AIF2RXLRCLK_PD_SHIFT 4 /* AIF2RXLRCLK_PD */ +#define WM5100_AIF2RXLRCLK_PD_WIDTH 1 /* AIF2RXLRCLK_PD */ +#define WM5100_AIF2BCLK_PU 0x0008 /* AIF2BCLK_PU */ +#define WM5100_AIF2BCLK_PU_MASK 0x0008 /* AIF2BCLK_PU */ +#define WM5100_AIF2BCLK_PU_SHIFT 3 /* AIF2BCLK_PU */ +#define WM5100_AIF2BCLK_PU_WIDTH 1 /* AIF2BCLK_PU */ +#define WM5100_AIF2BCLK_PD 0x0004 /* AIF2BCLK_PD */ +#define WM5100_AIF2BCLK_PD_MASK 0x0004 /* AIF2BCLK_PD */ +#define WM5100_AIF2BCLK_PD_SHIFT 2 /* AIF2BCLK_PD */ +#define WM5100_AIF2BCLK_PD_WIDTH 1 /* AIF2BCLK_PD */ +#define WM5100_AIF2RXDAT_PU 0x0002 /* AIF2RXDAT_PU */ +#define WM5100_AIF2RXDAT_PU_MASK 0x0002 /* AIF2RXDAT_PU */ +#define WM5100_AIF2RXDAT_PU_SHIFT 1 /* AIF2RXDAT_PU */ +#define WM5100_AIF2RXDAT_PU_WIDTH 1 /* AIF2RXDAT_PU */ +#define WM5100_AIF2RXDAT_PD 0x0001 /* AIF2RXDAT_PD */ +#define WM5100_AIF2RXDAT_PD_MASK 0x0001 /* AIF2RXDAT_PD */ +#define WM5100_AIF2RXDAT_PD_SHIFT 0 /* AIF2RXDAT_PD */ +#define WM5100_AIF2RXDAT_PD_WIDTH 1 /* AIF2RXDAT_PD */ + +/* + * R3111 (0xC27) - Misc Pad Ctrl 5 + */ +#define WM5100_AIF3RXLRCLK_PU 0x0020 /* AIF3RXLRCLK_PU */ +#define WM5100_AIF3RXLRCLK_PU_MASK 0x0020 /* AIF3RXLRCLK_PU */ +#define WM5100_AIF3RXLRCLK_PU_SHIFT 5 /* AIF3RXLRCLK_PU */ +#define WM5100_AIF3RXLRCLK_PU_WIDTH 1 /* AIF3RXLRCLK_PU */ +#define WM5100_AIF3RXLRCLK_PD 0x0010 /* AIF3RXLRCLK_PD */ +#define WM5100_AIF3RXLRCLK_PD_MASK 0x0010 /* AIF3RXLRCLK_PD */ +#define WM5100_AIF3RXLRCLK_PD_SHIFT 4 /* AIF3RXLRCLK_PD */ +#define WM5100_AIF3RXLRCLK_PD_WIDTH 1 /* AIF3RXLRCLK_PD */ +#define WM5100_AIF3BCLK_PU 0x0008 /* AIF3BCLK_PU */ +#define WM5100_AIF3BCLK_PU_MASK 0x0008 /* AIF3BCLK_PU */ +#define WM5100_AIF3BCLK_PU_SHIFT 3 /* AIF3BCLK_PU */ +#define WM5100_AIF3BCLK_PU_WIDTH 1 /* AIF3BCLK_PU */ +#define WM5100_AIF3BCLK_PD 0x0004 /* AIF3BCLK_PD */ +#define WM5100_AIF3BCLK_PD_MASK 0x0004 /* AIF3BCLK_PD */ +#define WM5100_AIF3BCLK_PD_SHIFT 2 /* AIF3BCLK_PD */ +#define WM5100_AIF3BCLK_PD_WIDTH 1 /* AIF3BCLK_PD */ +#define WM5100_AIF3RXDAT_PU 0x0002 /* AIF3RXDAT_PU */ +#define WM5100_AIF3RXDAT_PU_MASK 0x0002 /* AIF3RXDAT_PU */ +#define WM5100_AIF3RXDAT_PU_SHIFT 1 /* AIF3RXDAT_PU */ +#define WM5100_AIF3RXDAT_PU_WIDTH 1 /* AIF3RXDAT_PU */ +#define WM5100_AIF3RXDAT_PD 0x0001 /* AIF3RXDAT_PD */ +#define WM5100_AIF3RXDAT_PD_MASK 0x0001 /* AIF3RXDAT_PD */ +#define WM5100_AIF3RXDAT_PD_SHIFT 0 /* AIF3RXDAT_PD */ +#define WM5100_AIF3RXDAT_PD_WIDTH 1 /* AIF3RXDAT_PD */ + +/* + * R3112 (0xC28) - Misc GPIO 1 + */ +#define WM5100_OPCLK_SEL_MASK 0x0003 /* OPCLK_SEL - [1:0] */ +#define WM5100_OPCLK_SEL_SHIFT 0 /* OPCLK_SEL - [1:0] */ +#define WM5100_OPCLK_SEL_WIDTH 2 /* OPCLK_SEL - [1:0] */ + +/* + * R3328 (0xD00) - Interrupt Status 1 + */ +#define WM5100_GP6_EINT 0x0020 /* GP6_EINT */ +#define WM5100_GP6_EINT_MASK 0x0020 /* GP6_EINT */ +#define WM5100_GP6_EINT_SHIFT 5 /* GP6_EINT */ +#define WM5100_GP6_EINT_WIDTH 1 /* GP6_EINT */ +#define WM5100_GP5_EINT 0x0010 /* GP5_EINT */ +#define WM5100_GP5_EINT_MASK 0x0010 /* GP5_EINT */ +#define WM5100_GP5_EINT_SHIFT 4 /* GP5_EINT */ +#define WM5100_GP5_EINT_WIDTH 1 /* GP5_EINT */ +#define WM5100_GP4_EINT 0x0008 /* GP4_EINT */ +#define WM5100_GP4_EINT_MASK 0x0008 /* GP4_EINT */ +#define WM5100_GP4_EINT_SHIFT 3 /* GP4_EINT */ +#define WM5100_GP4_EINT_WIDTH 1 /* GP4_EINT */ +#define WM5100_GP3_EINT 0x0004 /* GP3_EINT */ +#define WM5100_GP3_EINT_MASK 0x0004 /* GP3_EINT */ +#define WM5100_GP3_EINT_SHIFT 2 /* GP3_EINT */ +#define WM5100_GP3_EINT_WIDTH 1 /* GP3_EINT */ +#define WM5100_GP2_EINT 0x0002 /* GP2_EINT */ +#define WM5100_GP2_EINT_MASK 0x0002 /* GP2_EINT */ +#define WM5100_GP2_EINT_SHIFT 1 /* GP2_EINT */ +#define WM5100_GP2_EINT_WIDTH 1 /* GP2_EINT */ +#define WM5100_GP1_EINT 0x0001 /* GP1_EINT */ +#define WM5100_GP1_EINT_MASK 0x0001 /* GP1_EINT */ +#define WM5100_GP1_EINT_SHIFT 0 /* GP1_EINT */ +#define WM5100_GP1_EINT_WIDTH 1 /* GP1_EINT */ + +/* + * R3329 (0xD01) - Interrupt Status 2 + */ +#define WM5100_DSP_IRQ6_EINT 0x0020 /* DSP_IRQ6_EINT */ +#define WM5100_DSP_IRQ6_EINT_MASK 0x0020 /* DSP_IRQ6_EINT */ +#define WM5100_DSP_IRQ6_EINT_SHIFT 5 /* DSP_IRQ6_EINT */ +#define WM5100_DSP_IRQ6_EINT_WIDTH 1 /* DSP_IRQ6_EINT */ +#define WM5100_DSP_IRQ5_EINT 0x0010 /* DSP_IRQ5_EINT */ +#define WM5100_DSP_IRQ5_EINT_MASK 0x0010 /* DSP_IRQ5_EINT */ +#define WM5100_DSP_IRQ5_EINT_SHIFT 4 /* DSP_IRQ5_EINT */ +#define WM5100_DSP_IRQ5_EINT_WIDTH 1 /* DSP_IRQ5_EINT */ +#define WM5100_DSP_IRQ4_EINT 0x0008 /* DSP_IRQ4_EINT */ +#define WM5100_DSP_IRQ4_EINT_MASK 0x0008 /* DSP_IRQ4_EINT */ +#define WM5100_DSP_IRQ4_EINT_SHIFT 3 /* DSP_IRQ4_EINT */ +#define WM5100_DSP_IRQ4_EINT_WIDTH 1 /* DSP_IRQ4_EINT */ +#define WM5100_DSP_IRQ3_EINT 0x0004 /* DSP_IRQ3_EINT */ +#define WM5100_DSP_IRQ3_EINT_MASK 0x0004 /* DSP_IRQ3_EINT */ +#define WM5100_DSP_IRQ3_EINT_SHIFT 2 /* DSP_IRQ3_EINT */ +#define WM5100_DSP_IRQ3_EINT_WIDTH 1 /* DSP_IRQ3_EINT */ +#define WM5100_DSP_IRQ2_EINT 0x0002 /* DSP_IRQ2_EINT */ +#define WM5100_DSP_IRQ2_EINT_MASK 0x0002 /* DSP_IRQ2_EINT */ +#define WM5100_DSP_IRQ2_EINT_SHIFT 1 /* DSP_IRQ2_EINT */ +#define WM5100_DSP_IRQ2_EINT_WIDTH 1 /* DSP_IRQ2_EINT */ +#define WM5100_DSP_IRQ1_EINT 0x0001 /* DSP_IRQ1_EINT */ +#define WM5100_DSP_IRQ1_EINT_MASK 0x0001 /* DSP_IRQ1_EINT */ +#define WM5100_DSP_IRQ1_EINT_SHIFT 0 /* DSP_IRQ1_EINT */ +#define WM5100_DSP_IRQ1_EINT_WIDTH 1 /* DSP_IRQ1_EINT */ + +/* + * R3330 (0xD02) - Interrupt Status 3 + */ +#define WM5100_SPK_SHUTDOWN_WARN_EINT 0x8000 /* SPK_SHUTDOWN_WARN_EINT */ +#define WM5100_SPK_SHUTDOWN_WARN_EINT_MASK 0x8000 /* SPK_SHUTDOWN_WARN_EINT */ +#define WM5100_SPK_SHUTDOWN_WARN_EINT_SHIFT 15 /* SPK_SHUTDOWN_WARN_EINT */ +#define WM5100_SPK_SHUTDOWN_WARN_EINT_WIDTH 1 /* SPK_SHUTDOWN_WARN_EINT */ +#define WM5100_SPK_SHUTDOWN_EINT 0x4000 /* SPK_SHUTDOWN_EINT */ +#define WM5100_SPK_SHUTDOWN_EINT_MASK 0x4000 /* SPK_SHUTDOWN_EINT */ +#define WM5100_SPK_SHUTDOWN_EINT_SHIFT 14 /* SPK_SHUTDOWN_EINT */ +#define WM5100_SPK_SHUTDOWN_EINT_WIDTH 1 /* SPK_SHUTDOWN_EINT */ +#define WM5100_HPDET_EINT 0x2000 /* HPDET_EINT */ +#define WM5100_HPDET_EINT_MASK 0x2000 /* HPDET_EINT */ +#define WM5100_HPDET_EINT_SHIFT 13 /* HPDET_EINT */ +#define WM5100_HPDET_EINT_WIDTH 1 /* HPDET_EINT */ +#define WM5100_ACCDET_EINT 0x1000 /* ACCDET_EINT */ +#define WM5100_ACCDET_EINT_MASK 0x1000 /* ACCDET_EINT */ +#define WM5100_ACCDET_EINT_SHIFT 12 /* ACCDET_EINT */ +#define WM5100_ACCDET_EINT_WIDTH 1 /* ACCDET_EINT */ +#define WM5100_DRC_SIG_DET_EINT 0x0200 /* DRC_SIG_DET_EINT */ +#define WM5100_DRC_SIG_DET_EINT_MASK 0x0200 /* DRC_SIG_DET_EINT */ +#define WM5100_DRC_SIG_DET_EINT_SHIFT 9 /* DRC_SIG_DET_EINT */ +#define WM5100_DRC_SIG_DET_EINT_WIDTH 1 /* DRC_SIG_DET_EINT */ +#define WM5100_ASRC2_LOCK_EINT 0x0100 /* ASRC2_LOCK_EINT */ +#define WM5100_ASRC2_LOCK_EINT_MASK 0x0100 /* ASRC2_LOCK_EINT */ +#define WM5100_ASRC2_LOCK_EINT_SHIFT 8 /* ASRC2_LOCK_EINT */ +#define WM5100_ASRC2_LOCK_EINT_WIDTH 1 /* ASRC2_LOCK_EINT */ +#define WM5100_ASRC1_LOCK_EINT 0x0080 /* ASRC1_LOCK_EINT */ +#define WM5100_ASRC1_LOCK_EINT_MASK 0x0080 /* ASRC1_LOCK_EINT */ +#define WM5100_ASRC1_LOCK_EINT_SHIFT 7 /* ASRC1_LOCK_EINT */ +#define WM5100_ASRC1_LOCK_EINT_WIDTH 1 /* ASRC1_LOCK_EINT */ +#define WM5100_FLL2_LOCK_EINT 0x0008 /* FLL2_LOCK_EINT */ +#define WM5100_FLL2_LOCK_EINT_MASK 0x0008 /* FLL2_LOCK_EINT */ +#define WM5100_FLL2_LOCK_EINT_SHIFT 3 /* FLL2_LOCK_EINT */ +#define WM5100_FLL2_LOCK_EINT_WIDTH 1 /* FLL2_LOCK_EINT */ +#define WM5100_FLL1_LOCK_EINT 0x0004 /* FLL1_LOCK_EINT */ +#define WM5100_FLL1_LOCK_EINT_MASK 0x0004 /* FLL1_LOCK_EINT */ +#define WM5100_FLL1_LOCK_EINT_SHIFT 2 /* FLL1_LOCK_EINT */ +#define WM5100_FLL1_LOCK_EINT_WIDTH 1 /* FLL1_LOCK_EINT */ +#define WM5100_CLKGEN_ERR_EINT 0x0002 /* CLKGEN_ERR_EINT */ +#define WM5100_CLKGEN_ERR_EINT_MASK 0x0002 /* CLKGEN_ERR_EINT */ +#define WM5100_CLKGEN_ERR_EINT_SHIFT 1 /* CLKGEN_ERR_EINT */ +#define WM5100_CLKGEN_ERR_EINT_WIDTH 1 /* CLKGEN_ERR_EINT */ +#define WM5100_CLKGEN_ERR_ASYNC_EINT 0x0001 /* CLKGEN_ERR_ASYNC_EINT */ +#define WM5100_CLKGEN_ERR_ASYNC_EINT_MASK 0x0001 /* CLKGEN_ERR_ASYNC_EINT */ +#define WM5100_CLKGEN_ERR_ASYNC_EINT_SHIFT 0 /* CLKGEN_ERR_ASYNC_EINT */ +#define WM5100_CLKGEN_ERR_ASYNC_EINT_WIDTH 1 /* CLKGEN_ERR_ASYNC_EINT */ + +/* + * R3331 (0xD03) - Interrupt Status 4 + */ +#define WM5100_AIF3_ERR_EINT 0x2000 /* AIF3_ERR_EINT */ +#define WM5100_AIF3_ERR_EINT_MASK 0x2000 /* AIF3_ERR_EINT */ +#define WM5100_AIF3_ERR_EINT_SHIFT 13 /* AIF3_ERR_EINT */ +#define WM5100_AIF3_ERR_EINT_WIDTH 1 /* AIF3_ERR_EINT */ +#define WM5100_AIF2_ERR_EINT 0x1000 /* AIF2_ERR_EINT */ +#define WM5100_AIF2_ERR_EINT_MASK 0x1000 /* AIF2_ERR_EINT */ +#define WM5100_AIF2_ERR_EINT_SHIFT 12 /* AIF2_ERR_EINT */ +#define WM5100_AIF2_ERR_EINT_WIDTH 1 /* AIF2_ERR_EINT */ +#define WM5100_AIF1_ERR_EINT 0x0800 /* AIF1_ERR_EINT */ +#define WM5100_AIF1_ERR_EINT_MASK 0x0800 /* AIF1_ERR_EINT */ +#define WM5100_AIF1_ERR_EINT_SHIFT 11 /* AIF1_ERR_EINT */ +#define WM5100_AIF1_ERR_EINT_WIDTH 1 /* AIF1_ERR_EINT */ +#define WM5100_CTRLIF_ERR_EINT 0x0400 /* CTRLIF_ERR_EINT */ +#define WM5100_CTRLIF_ERR_EINT_MASK 0x0400 /* CTRLIF_ERR_EINT */ +#define WM5100_CTRLIF_ERR_EINT_SHIFT 10 /* CTRLIF_ERR_EINT */ +#define WM5100_CTRLIF_ERR_EINT_WIDTH 1 /* CTRLIF_ERR_EINT */ +#define WM5100_ISRC2_UNDERCLOCKED_EINT 0x0200 /* ISRC2_UNDERCLOCKED_EINT */ +#define WM5100_ISRC2_UNDERCLOCKED_EINT_MASK 0x0200 /* ISRC2_UNDERCLOCKED_EINT */ +#define WM5100_ISRC2_UNDERCLOCKED_EINT_SHIFT 9 /* ISRC2_UNDERCLOCKED_EINT */ +#define WM5100_ISRC2_UNDERCLOCKED_EINT_WIDTH 1 /* ISRC2_UNDERCLOCKED_EINT */ +#define WM5100_ISRC1_UNDERCLOCKED_EINT 0x0100 /* ISRC1_UNDERCLOCKED_EINT */ +#define WM5100_ISRC1_UNDERCLOCKED_EINT_MASK 0x0100 /* ISRC1_UNDERCLOCKED_EINT */ +#define WM5100_ISRC1_UNDERCLOCKED_EINT_SHIFT 8 /* ISRC1_UNDERCLOCKED_EINT */ +#define WM5100_ISRC1_UNDERCLOCKED_EINT_WIDTH 1 /* ISRC1_UNDERCLOCKED_EINT */ +#define WM5100_FX_UNDERCLOCKED_EINT 0x0080 /* FX_UNDERCLOCKED_EINT */ +#define WM5100_FX_UNDERCLOCKED_EINT_MASK 0x0080 /* FX_UNDERCLOCKED_EINT */ +#define WM5100_FX_UNDERCLOCKED_EINT_SHIFT 7 /* FX_UNDERCLOCKED_EINT */ +#define WM5100_FX_UNDERCLOCKED_EINT_WIDTH 1 /* FX_UNDERCLOCKED_EINT */ +#define WM5100_AIF3_UNDERCLOCKED_EINT 0x0040 /* AIF3_UNDERCLOCKED_EINT */ +#define WM5100_AIF3_UNDERCLOCKED_EINT_MASK 0x0040 /* AIF3_UNDERCLOCKED_EINT */ +#define WM5100_AIF3_UNDERCLOCKED_EINT_SHIFT 6 /* AIF3_UNDERCLOCKED_EINT */ +#define WM5100_AIF3_UNDERCLOCKED_EINT_WIDTH 1 /* AIF3_UNDERCLOCKED_EINT */ +#define WM5100_AIF2_UNDERCLOCKED_EINT 0x0020 /* AIF2_UNDERCLOCKED_EINT */ +#define WM5100_AIF2_UNDERCLOCKED_EINT_MASK 0x0020 /* AIF2_UNDERCLOCKED_EINT */ +#define WM5100_AIF2_UNDERCLOCKED_EINT_SHIFT 5 /* AIF2_UNDERCLOCKED_EINT */ +#define WM5100_AIF2_UNDERCLOCKED_EINT_WIDTH 1 /* AIF2_UNDERCLOCKED_EINT */ +#define WM5100_AIF1_UNDERCLOCKED_EINT 0x0010 /* AIF1_UNDERCLOCKED_EINT */ +#define WM5100_AIF1_UNDERCLOCKED_EINT_MASK 0x0010 /* AIF1_UNDERCLOCKED_EINT */ +#define WM5100_AIF1_UNDERCLOCKED_EINT_SHIFT 4 /* AIF1_UNDERCLOCKED_EINT */ +#define WM5100_AIF1_UNDERCLOCKED_EINT_WIDTH 1 /* AIF1_UNDERCLOCKED_EINT */ +#define WM5100_ASRC_UNDERCLOCKED_EINT 0x0008 /* ASRC_UNDERCLOCKED_EINT */ +#define WM5100_ASRC_UNDERCLOCKED_EINT_MASK 0x0008 /* ASRC_UNDERCLOCKED_EINT */ +#define WM5100_ASRC_UNDERCLOCKED_EINT_SHIFT 3 /* ASRC_UNDERCLOCKED_EINT */ +#define WM5100_ASRC_UNDERCLOCKED_EINT_WIDTH 1 /* ASRC_UNDERCLOCKED_EINT */ +#define WM5100_DAC_UNDERCLOCKED_EINT 0x0004 /* DAC_UNDERCLOCKED_EINT */ +#define WM5100_DAC_UNDERCLOCKED_EINT_MASK 0x0004 /* DAC_UNDERCLOCKED_EINT */ +#define WM5100_DAC_UNDERCLOCKED_EINT_SHIFT 2 /* DAC_UNDERCLOCKED_EINT */ +#define WM5100_DAC_UNDERCLOCKED_EINT_WIDTH 1 /* DAC_UNDERCLOCKED_EINT */ +#define WM5100_ADC_UNDERCLOCKED_EINT 0x0002 /* ADC_UNDERCLOCKED_EINT */ +#define WM5100_ADC_UNDERCLOCKED_EINT_MASK 0x0002 /* ADC_UNDERCLOCKED_EINT */ +#define WM5100_ADC_UNDERCLOCKED_EINT_SHIFT 1 /* ADC_UNDERCLOCKED_EINT */ +#define WM5100_ADC_UNDERCLOCKED_EINT_WIDTH 1 /* ADC_UNDERCLOCKED_EINT */ +#define WM5100_MIXER_UNDERCLOCKED_EINT 0x0001 /* MIXER_UNDERCLOCKED_EINT */ +#define WM5100_MIXER_UNDERCLOCKED_EINT_MASK 0x0001 /* MIXER_UNDERCLOCKED_EINT */ +#define WM5100_MIXER_UNDERCLOCKED_EINT_SHIFT 0 /* MIXER_UNDERCLOCKED_EINT */ +#define WM5100_MIXER_UNDERCLOCKED_EINT_WIDTH 1 /* MIXER_UNDERCLOCKED_EINT */ + +/* + * R3332 (0xD04) - Interrupt Raw Status 2 + */ +#define WM5100_DSP_IRQ6_STS 0x0020 /* DSP_IRQ6_STS */ +#define WM5100_DSP_IRQ6_STS_MASK 0x0020 /* DSP_IRQ6_STS */ +#define WM5100_DSP_IRQ6_STS_SHIFT 5 /* DSP_IRQ6_STS */ +#define WM5100_DSP_IRQ6_STS_WIDTH 1 /* DSP_IRQ6_STS */ +#define WM5100_DSP_IRQ5_STS 0x0010 /* DSP_IRQ5_STS */ +#define WM5100_DSP_IRQ5_STS_MASK 0x0010 /* DSP_IRQ5_STS */ +#define WM5100_DSP_IRQ5_STS_SHIFT 4 /* DSP_IRQ5_STS */ +#define WM5100_DSP_IRQ5_STS_WIDTH 1 /* DSP_IRQ5_STS */ +#define WM5100_DSP_IRQ4_STS 0x0008 /* DSP_IRQ4_STS */ +#define WM5100_DSP_IRQ4_STS_MASK 0x0008 /* DSP_IRQ4_STS */ +#define WM5100_DSP_IRQ4_STS_SHIFT 3 /* DSP_IRQ4_STS */ +#define WM5100_DSP_IRQ4_STS_WIDTH 1 /* DSP_IRQ4_STS */ +#define WM5100_DSP_IRQ3_STS 0x0004 /* DSP_IRQ3_STS */ +#define WM5100_DSP_IRQ3_STS_MASK 0x0004 /* DSP_IRQ3_STS */ +#define WM5100_DSP_IRQ3_STS_SHIFT 2 /* DSP_IRQ3_STS */ +#define WM5100_DSP_IRQ3_STS_WIDTH 1 /* DSP_IRQ3_STS */ +#define WM5100_DSP_IRQ2_STS 0x0002 /* DSP_IRQ2_STS */ +#define WM5100_DSP_IRQ2_STS_MASK 0x0002 /* DSP_IRQ2_STS */ +#define WM5100_DSP_IRQ2_STS_SHIFT 1 /* DSP_IRQ2_STS */ +#define WM5100_DSP_IRQ2_STS_WIDTH 1 /* DSP_IRQ2_STS */ +#define WM5100_DSP_IRQ1_STS 0x0001 /* DSP_IRQ1_STS */ +#define WM5100_DSP_IRQ1_STS_MASK 0x0001 /* DSP_IRQ1_STS */ +#define WM5100_DSP_IRQ1_STS_SHIFT 0 /* DSP_IRQ1_STS */ +#define WM5100_DSP_IRQ1_STS_WIDTH 1 /* DSP_IRQ1_STS */ + +/* + * R3333 (0xD05) - Interrupt Raw Status 3 + */ +#define WM5100_SPK_SHUTDOWN_WARN_STS 0x8000 /* SPK_SHUTDOWN_WARN_STS */ +#define WM5100_SPK_SHUTDOWN_WARN_STS_MASK 0x8000 /* SPK_SHUTDOWN_WARN_STS */ +#define WM5100_SPK_SHUTDOWN_WARN_STS_SHIFT 15 /* SPK_SHUTDOWN_WARN_STS */ +#define WM5100_SPK_SHUTDOWN_WARN_STS_WIDTH 1 /* SPK_SHUTDOWN_WARN_STS */ +#define WM5100_SPK_SHUTDOWN_STS 0x4000 /* SPK_SHUTDOWN_STS */ +#define WM5100_SPK_SHUTDOWN_STS_MASK 0x4000 /* SPK_SHUTDOWN_STS */ +#define WM5100_SPK_SHUTDOWN_STS_SHIFT 14 /* SPK_SHUTDOWN_STS */ +#define WM5100_SPK_SHUTDOWN_STS_WIDTH 1 /* SPK_SHUTDOWN_STS */ +#define WM5100_HPDET_STS 0x2000 /* HPDET_STS */ +#define WM5100_HPDET_STS_MASK 0x2000 /* HPDET_STS */ +#define WM5100_HPDET_STS_SHIFT 13 /* HPDET_STS */ +#define WM5100_HPDET_STS_WIDTH 1 /* HPDET_STS */ +#define WM5100_DRC_SID_DET_STS 0x0200 /* DRC_SID_DET_STS */ +#define WM5100_DRC_SID_DET_STS_MASK 0x0200 /* DRC_SID_DET_STS */ +#define WM5100_DRC_SID_DET_STS_SHIFT 9 /* DRC_SID_DET_STS */ +#define WM5100_DRC_SID_DET_STS_WIDTH 1 /* DRC_SID_DET_STS */ +#define WM5100_ASRC2_LOCK_STS 0x0100 /* ASRC2_LOCK_STS */ +#define WM5100_ASRC2_LOCK_STS_MASK 0x0100 /* ASRC2_LOCK_STS */ +#define WM5100_ASRC2_LOCK_STS_SHIFT 8 /* ASRC2_LOCK_STS */ +#define WM5100_ASRC2_LOCK_STS_WIDTH 1 /* ASRC2_LOCK_STS */ +#define WM5100_ASRC1_LOCK_STS 0x0080 /* ASRC1_LOCK_STS */ +#define WM5100_ASRC1_LOCK_STS_MASK 0x0080 /* ASRC1_LOCK_STS */ +#define WM5100_ASRC1_LOCK_STS_SHIFT 7 /* ASRC1_LOCK_STS */ +#define WM5100_ASRC1_LOCK_STS_WIDTH 1 /* ASRC1_LOCK_STS */ +#define WM5100_FLL2_LOCK_STS 0x0008 /* FLL2_LOCK_STS */ +#define WM5100_FLL2_LOCK_STS_MASK 0x0008 /* FLL2_LOCK_STS */ +#define WM5100_FLL2_LOCK_STS_SHIFT 3 /* FLL2_LOCK_STS */ +#define WM5100_FLL2_LOCK_STS_WIDTH 1 /* FLL2_LOCK_STS */ +#define WM5100_FLL1_LOCK_STS 0x0004 /* FLL1_LOCK_STS */ +#define WM5100_FLL1_LOCK_STS_MASK 0x0004 /* FLL1_LOCK_STS */ +#define WM5100_FLL1_LOCK_STS_SHIFT 2 /* FLL1_LOCK_STS */ +#define WM5100_FLL1_LOCK_STS_WIDTH 1 /* FLL1_LOCK_STS */ +#define WM5100_CLKGEN_ERR_STS 0x0002 /* CLKGEN_ERR_STS */ +#define WM5100_CLKGEN_ERR_STS_MASK 0x0002 /* CLKGEN_ERR_STS */ +#define WM5100_CLKGEN_ERR_STS_SHIFT 1 /* CLKGEN_ERR_STS */ +#define WM5100_CLKGEN_ERR_STS_WIDTH 1 /* CLKGEN_ERR_STS */ +#define WM5100_CLKGEN_ERR_ASYNC_STS 0x0001 /* CLKGEN_ERR_ASYNC_STS */ +#define WM5100_CLKGEN_ERR_ASYNC_STS_MASK 0x0001 /* CLKGEN_ERR_ASYNC_STS */ +#define WM5100_CLKGEN_ERR_ASYNC_STS_SHIFT 0 /* CLKGEN_ERR_ASYNC_STS */ +#define WM5100_CLKGEN_ERR_ASYNC_STS_WIDTH 1 /* CLKGEN_ERR_ASYNC_STS */ + +/* + * R3334 (0xD06) - Interrupt Raw Status 4 + */ +#define WM5100_AIF3_ERR_STS 0x2000 /* AIF3_ERR_STS */ +#define WM5100_AIF3_ERR_STS_MASK 0x2000 /* AIF3_ERR_STS */ +#define WM5100_AIF3_ERR_STS_SHIFT 13 /* AIF3_ERR_STS */ +#define WM5100_AIF3_ERR_STS_WIDTH 1 /* AIF3_ERR_STS */ +#define WM5100_AIF2_ERR_STS 0x1000 /* AIF2_ERR_STS */ +#define WM5100_AIF2_ERR_STS_MASK 0x1000 /* AIF2_ERR_STS */ +#define WM5100_AIF2_ERR_STS_SHIFT 12 /* AIF2_ERR_STS */ +#define WM5100_AIF2_ERR_STS_WIDTH 1 /* AIF2_ERR_STS */ +#define WM5100_AIF1_ERR_STS 0x0800 /* AIF1_ERR_STS */ +#define WM5100_AIF1_ERR_STS_MASK 0x0800 /* AIF1_ERR_STS */ +#define WM5100_AIF1_ERR_STS_SHIFT 11 /* AIF1_ERR_STS */ +#define WM5100_AIF1_ERR_STS_WIDTH 1 /* AIF1_ERR_STS */ +#define WM5100_CTRLIF_ERR_STS 0x0400 /* CTRLIF_ERR_STS */ +#define WM5100_CTRLIF_ERR_STS_MASK 0x0400 /* CTRLIF_ERR_STS */ +#define WM5100_CTRLIF_ERR_STS_SHIFT 10 /* CTRLIF_ERR_STS */ +#define WM5100_CTRLIF_ERR_STS_WIDTH 1 /* CTRLIF_ERR_STS */ +#define WM5100_ISRC2_UNDERCLOCKED_STS 0x0200 /* ISRC2_UNDERCLOCKED_STS */ +#define WM5100_ISRC2_UNDERCLOCKED_STS_MASK 0x0200 /* ISRC2_UNDERCLOCKED_STS */ +#define WM5100_ISRC2_UNDERCLOCKED_STS_SHIFT 9 /* ISRC2_UNDERCLOCKED_STS */ +#define WM5100_ISRC2_UNDERCLOCKED_STS_WIDTH 1 /* ISRC2_UNDERCLOCKED_STS */ +#define WM5100_ISRC1_UNDERCLOCKED_STS 0x0100 /* ISRC1_UNDERCLOCKED_STS */ +#define WM5100_ISRC1_UNDERCLOCKED_STS_MASK 0x0100 /* ISRC1_UNDERCLOCKED_STS */ +#define WM5100_ISRC1_UNDERCLOCKED_STS_SHIFT 8 /* ISRC1_UNDERCLOCKED_STS */ +#define WM5100_ISRC1_UNDERCLOCKED_STS_WIDTH 1 /* ISRC1_UNDERCLOCKED_STS */ +#define WM5100_FX_UNDERCLOCKED_STS 0x0080 /* FX_UNDERCLOCKED_STS */ +#define WM5100_FX_UNDERCLOCKED_STS_MASK 0x0080 /* FX_UNDERCLOCKED_STS */ +#define WM5100_FX_UNDERCLOCKED_STS_SHIFT 7 /* FX_UNDERCLOCKED_STS */ +#define WM5100_FX_UNDERCLOCKED_STS_WIDTH 1 /* FX_UNDERCLOCKED_STS */ +#define WM5100_AIF3_UNDERCLOCKED_STS 0x0040 /* AIF3_UNDERCLOCKED_STS */ +#define WM5100_AIF3_UNDERCLOCKED_STS_MASK 0x0040 /* AIF3_UNDERCLOCKED_STS */ +#define WM5100_AIF3_UNDERCLOCKED_STS_SHIFT 6 /* AIF3_UNDERCLOCKED_STS */ +#define WM5100_AIF3_UNDERCLOCKED_STS_WIDTH 1 /* AIF3_UNDERCLOCKED_STS */ +#define WM5100_AIF2_UNDERCLOCKED_STS 0x0020 /* AIF2_UNDERCLOCKED_STS */ +#define WM5100_AIF2_UNDERCLOCKED_STS_MASK 0x0020 /* AIF2_UNDERCLOCKED_STS */ +#define WM5100_AIF2_UNDERCLOCKED_STS_SHIFT 5 /* AIF2_UNDERCLOCKED_STS */ +#define WM5100_AIF2_UNDERCLOCKED_STS_WIDTH 1 /* AIF2_UNDERCLOCKED_STS */ +#define WM5100_AIF1_UNDERCLOCKED_STS 0x0010 /* AIF1_UNDERCLOCKED_STS */ +#define WM5100_AIF1_UNDERCLOCKED_STS_MASK 0x0010 /* AIF1_UNDERCLOCKED_STS */ +#define WM5100_AIF1_UNDERCLOCKED_STS_SHIFT 4 /* AIF1_UNDERCLOCKED_STS */ +#define WM5100_AIF1_UNDERCLOCKED_STS_WIDTH 1 /* AIF1_UNDERCLOCKED_STS */ +#define WM5100_ASRC_UNDERCLOCKED_STS 0x0008 /* ASRC_UNDERCLOCKED_STS */ +#define WM5100_ASRC_UNDERCLOCKED_STS_MASK 0x0008 /* ASRC_UNDERCLOCKED_STS */ +#define WM5100_ASRC_UNDERCLOCKED_STS_SHIFT 3 /* ASRC_UNDERCLOCKED_STS */ +#define WM5100_ASRC_UNDERCLOCKED_STS_WIDTH 1 /* ASRC_UNDERCLOCKED_STS */ +#define WM5100_DAC_UNDERCLOCKED_STS 0x0004 /* DAC_UNDERCLOCKED_STS */ +#define WM5100_DAC_UNDERCLOCKED_STS_MASK 0x0004 /* DAC_UNDERCLOCKED_STS */ +#define WM5100_DAC_UNDERCLOCKED_STS_SHIFT 2 /* DAC_UNDERCLOCKED_STS */ +#define WM5100_DAC_UNDERCLOCKED_STS_WIDTH 1 /* DAC_UNDERCLOCKED_STS */ +#define WM5100_ADC_UNDERCLOCKED_STS 0x0002 /* ADC_UNDERCLOCKED_STS */ +#define WM5100_ADC_UNDERCLOCKED_STS_MASK 0x0002 /* ADC_UNDERCLOCKED_STS */ +#define WM5100_ADC_UNDERCLOCKED_STS_SHIFT 1 /* ADC_UNDERCLOCKED_STS */ +#define WM5100_ADC_UNDERCLOCKED_STS_WIDTH 1 /* ADC_UNDERCLOCKED_STS */ +#define WM5100_MIXER_UNDERCLOCKED_STS 0x0001 /* MIXER_UNDERCLOCKED_STS */ +#define WM5100_MIXER_UNDERCLOCKED_STS_MASK 0x0001 /* MIXER_UNDERCLOCKED_STS */ +#define WM5100_MIXER_UNDERCLOCKED_STS_SHIFT 0 /* MIXER_UNDERCLOCKED_STS */ +#define WM5100_MIXER_UNDERCLOCKED_STS_WIDTH 1 /* MIXER_UNDERCLOCKED_STS */ + +/* + * R3335 (0xD07) - Interrupt Status 1 Mask + */ +#define WM5100_IM_GP6_EINT 0x0020 /* IM_GP6_EINT */ +#define WM5100_IM_GP6_EINT_MASK 0x0020 /* IM_GP6_EINT */ +#define WM5100_IM_GP6_EINT_SHIFT 5 /* IM_GP6_EINT */ +#define WM5100_IM_GP6_EINT_WIDTH 1 /* IM_GP6_EINT */ +#define WM5100_IM_GP5_EINT 0x0010 /* IM_GP5_EINT */ +#define WM5100_IM_GP5_EINT_MASK 0x0010 /* IM_GP5_EINT */ +#define WM5100_IM_GP5_EINT_SHIFT 4 /* IM_GP5_EINT */ +#define WM5100_IM_GP5_EINT_WIDTH 1 /* IM_GP5_EINT */ +#define WM5100_IM_GP4_EINT 0x0008 /* IM_GP4_EINT */ +#define WM5100_IM_GP4_EINT_MASK 0x0008 /* IM_GP4_EINT */ +#define WM5100_IM_GP4_EINT_SHIFT 3 /* IM_GP4_EINT */ +#define WM5100_IM_GP4_EINT_WIDTH 1 /* IM_GP4_EINT */ +#define WM5100_IM_GP3_EINT 0x0004 /* IM_GP3_EINT */ +#define WM5100_IM_GP3_EINT_MASK 0x0004 /* IM_GP3_EINT */ +#define WM5100_IM_GP3_EINT_SHIFT 2 /* IM_GP3_EINT */ +#define WM5100_IM_GP3_EINT_WIDTH 1 /* IM_GP3_EINT */ +#define WM5100_IM_GP2_EINT 0x0002 /* IM_GP2_EINT */ +#define WM5100_IM_GP2_EINT_MASK 0x0002 /* IM_GP2_EINT */ +#define WM5100_IM_GP2_EINT_SHIFT 1 /* IM_GP2_EINT */ +#define WM5100_IM_GP2_EINT_WIDTH 1 /* IM_GP2_EINT */ +#define WM5100_IM_GP1_EINT 0x0001 /* IM_GP1_EINT */ +#define WM5100_IM_GP1_EINT_MASK 0x0001 /* IM_GP1_EINT */ +#define WM5100_IM_GP1_EINT_SHIFT 0 /* IM_GP1_EINT */ +#define WM5100_IM_GP1_EINT_WIDTH 1 /* IM_GP1_EINT */ + +/* + * R3336 (0xD08) - Interrupt Status 2 Mask + */ +#define WM5100_IM_DSP_IRQ6_EINT 0x0020 /* IM_DSP_IRQ6_EINT */ +#define WM5100_IM_DSP_IRQ6_EINT_MASK 0x0020 /* IM_DSP_IRQ6_EINT */ +#define WM5100_IM_DSP_IRQ6_EINT_SHIFT 5 /* IM_DSP_IRQ6_EINT */ +#define WM5100_IM_DSP_IRQ6_EINT_WIDTH 1 /* IM_DSP_IRQ6_EINT */ +#define WM5100_IM_DSP_IRQ5_EINT 0x0010 /* IM_DSP_IRQ5_EINT */ +#define WM5100_IM_DSP_IRQ5_EINT_MASK 0x0010 /* IM_DSP_IRQ5_EINT */ +#define WM5100_IM_DSP_IRQ5_EINT_SHIFT 4 /* IM_DSP_IRQ5_EINT */ +#define WM5100_IM_DSP_IRQ5_EINT_WIDTH 1 /* IM_DSP_IRQ5_EINT */ +#define WM5100_IM_DSP_IRQ4_EINT 0x0008 /* IM_DSP_IRQ4_EINT */ +#define WM5100_IM_DSP_IRQ4_EINT_MASK 0x0008 /* IM_DSP_IRQ4_EINT */ +#define WM5100_IM_DSP_IRQ4_EINT_SHIFT 3 /* IM_DSP_IRQ4_EINT */ +#define WM5100_IM_DSP_IRQ4_EINT_WIDTH 1 /* IM_DSP_IRQ4_EINT */ +#define WM5100_IM_DSP_IRQ3_EINT 0x0004 /* IM_DSP_IRQ3_EINT */ +#define WM5100_IM_DSP_IRQ3_EINT_MASK 0x0004 /* IM_DSP_IRQ3_EINT */ +#define WM5100_IM_DSP_IRQ3_EINT_SHIFT 2 /* IM_DSP_IRQ3_EINT */ +#define WM5100_IM_DSP_IRQ3_EINT_WIDTH 1 /* IM_DSP_IRQ3_EINT */ +#define WM5100_IM_DSP_IRQ2_EINT 0x0002 /* IM_DSP_IRQ2_EINT */ +#define WM5100_IM_DSP_IRQ2_EINT_MASK 0x0002 /* IM_DSP_IRQ2_EINT */ +#define WM5100_IM_DSP_IRQ2_EINT_SHIFT 1 /* IM_DSP_IRQ2_EINT */ +#define WM5100_IM_DSP_IRQ2_EINT_WIDTH 1 /* IM_DSP_IRQ2_EINT */ +#define WM5100_IM_DSP_IRQ1_EINT 0x0001 /* IM_DSP_IRQ1_EINT */ +#define WM5100_IM_DSP_IRQ1_EINT_MASK 0x0001 /* IM_DSP_IRQ1_EINT */ +#define WM5100_IM_DSP_IRQ1_EINT_SHIFT 0 /* IM_DSP_IRQ1_EINT */ +#define WM5100_IM_DSP_IRQ1_EINT_WIDTH 1 /* IM_DSP_IRQ1_EINT */ + +/* + * R3337 (0xD09) - Interrupt Status 3 Mask + */ +#define WM5100_IM_SPK_SHUTDOWN_WARN_EINT 0x8000 /* IM_SPK_SHUTDOWN_WARN_EINT */ +#define WM5100_IM_SPK_SHUTDOWN_WARN_EINT_MASK 0x8000 /* IM_SPK_SHUTDOWN_WARN_EINT */ +#define WM5100_IM_SPK_SHUTDOWN_WARN_EINT_SHIFT 15 /* IM_SPK_SHUTDOWN_WARN_EINT */ +#define WM5100_IM_SPK_SHUTDOWN_WARN_EINT_WIDTH 1 /* IM_SPK_SHUTDOWN_WARN_EINT */ +#define WM5100_IM_SPK_SHUTDOWN_EINT 0x4000 /* IM_SPK_SHUTDOWN_EINT */ +#define WM5100_IM_SPK_SHUTDOWN_EINT_MASK 0x4000 /* IM_SPK_SHUTDOWN_EINT */ +#define WM5100_IM_SPK_SHUTDOWN_EINT_SHIFT 14 /* IM_SPK_SHUTDOWN_EINT */ +#define WM5100_IM_SPK_SHUTDOWN_EINT_WIDTH 1 /* IM_SPK_SHUTDOWN_EINT */ +#define WM5100_IM_HPDET_EINT 0x2000 /* IM_HPDET_EINT */ +#define WM5100_IM_HPDET_EINT_MASK 0x2000 /* IM_HPDET_EINT */ +#define WM5100_IM_HPDET_EINT_SHIFT 13 /* IM_HPDET_EINT */ +#define WM5100_IM_HPDET_EINT_WIDTH 1 /* IM_HPDET_EINT */ +#define WM5100_IM_ACCDET_EINT 0x1000 /* IM_ACCDET_EINT */ +#define WM5100_IM_ACCDET_EINT_MASK 0x1000 /* IM_ACCDET_EINT */ +#define WM5100_IM_ACCDET_EINT_SHIFT 12 /* IM_ACCDET_EINT */ +#define WM5100_IM_ACCDET_EINT_WIDTH 1 /* IM_ACCDET_EINT */ +#define WM5100_IM_DRC_SIG_DET_EINT 0x0200 /* IM_DRC_SIG_DET_EINT */ +#define WM5100_IM_DRC_SIG_DET_EINT_MASK 0x0200 /* IM_DRC_SIG_DET_EINT */ +#define WM5100_IM_DRC_SIG_DET_EINT_SHIFT 9 /* IM_DRC_SIG_DET_EINT */ +#define WM5100_IM_DRC_SIG_DET_EINT_WIDTH 1 /* IM_DRC_SIG_DET_EINT */ +#define WM5100_IM_ASRC2_LOCK_EINT 0x0100 /* IM_ASRC2_LOCK_EINT */ +#define WM5100_IM_ASRC2_LOCK_EINT_MASK 0x0100 /* IM_ASRC2_LOCK_EINT */ +#define WM5100_IM_ASRC2_LOCK_EINT_SHIFT 8 /* IM_ASRC2_LOCK_EINT */ +#define WM5100_IM_ASRC2_LOCK_EINT_WIDTH 1 /* IM_ASRC2_LOCK_EINT */ +#define WM5100_IM_ASRC1_LOCK_EINT 0x0080 /* IM_ASRC1_LOCK_EINT */ +#define WM5100_IM_ASRC1_LOCK_EINT_MASK 0x0080 /* IM_ASRC1_LOCK_EINT */ +#define WM5100_IM_ASRC1_LOCK_EINT_SHIFT 7 /* IM_ASRC1_LOCK_EINT */ +#define WM5100_IM_ASRC1_LOCK_EINT_WIDTH 1 /* IM_ASRC1_LOCK_EINT */ +#define WM5100_IM_FLL2_LOCK_EINT 0x0008 /* IM_FLL2_LOCK_EINT */ +#define WM5100_IM_FLL2_LOCK_EINT_MASK 0x0008 /* IM_FLL2_LOCK_EINT */ +#define WM5100_IM_FLL2_LOCK_EINT_SHIFT 3 /* IM_FLL2_LOCK_EINT */ +#define WM5100_IM_FLL2_LOCK_EINT_WIDTH 1 /* IM_FLL2_LOCK_EINT */ +#define WM5100_IM_FLL1_LOCK_EINT 0x0004 /* IM_FLL1_LOCK_EINT */ +#define WM5100_IM_FLL1_LOCK_EINT_MASK 0x0004 /* IM_FLL1_LOCK_EINT */ +#define WM5100_IM_FLL1_LOCK_EINT_SHIFT 2 /* IM_FLL1_LOCK_EINT */ +#define WM5100_IM_FLL1_LOCK_EINT_WIDTH 1 /* IM_FLL1_LOCK_EINT */ +#define WM5100_IM_CLKGEN_ERR_EINT 0x0002 /* IM_CLKGEN_ERR_EINT */ +#define WM5100_IM_CLKGEN_ERR_EINT_MASK 0x0002 /* IM_CLKGEN_ERR_EINT */ +#define WM5100_IM_CLKGEN_ERR_EINT_SHIFT 1 /* IM_CLKGEN_ERR_EINT */ +#define WM5100_IM_CLKGEN_ERR_EINT_WIDTH 1 /* IM_CLKGEN_ERR_EINT */ +#define WM5100_IM_CLKGEN_ERR_ASYNC_EINT 0x0001 /* IM_CLKGEN_ERR_ASYNC_EINT */ +#define WM5100_IM_CLKGEN_ERR_ASYNC_EINT_MASK 0x0001 /* IM_CLKGEN_ERR_ASYNC_EINT */ +#define WM5100_IM_CLKGEN_ERR_ASYNC_EINT_SHIFT 0 /* IM_CLKGEN_ERR_ASYNC_EINT */ +#define WM5100_IM_CLKGEN_ERR_ASYNC_EINT_WIDTH 1 /* IM_CLKGEN_ERR_ASYNC_EINT */ + +/* + * R3338 (0xD0A) - Interrupt Status 4 Mask + */ +#define WM5100_IM_AIF3_ERR_EINT 0x2000 /* IM_AIF3_ERR_EINT */ +#define WM5100_IM_AIF3_ERR_EINT_MASK 0x2000 /* IM_AIF3_ERR_EINT */ +#define WM5100_IM_AIF3_ERR_EINT_SHIFT 13 /* IM_AIF3_ERR_EINT */ +#define WM5100_IM_AIF3_ERR_EINT_WIDTH 1 /* IM_AIF3_ERR_EINT */ +#define WM5100_IM_AIF2_ERR_EINT 0x1000 /* IM_AIF2_ERR_EINT */ +#define WM5100_IM_AIF2_ERR_EINT_MASK 0x1000 /* IM_AIF2_ERR_EINT */ +#define WM5100_IM_AIF2_ERR_EINT_SHIFT 12 /* IM_AIF2_ERR_EINT */ +#define WM5100_IM_AIF2_ERR_EINT_WIDTH 1 /* IM_AIF2_ERR_EINT */ +#define WM5100_IM_AIF1_ERR_EINT 0x0800 /* IM_AIF1_ERR_EINT */ +#define WM5100_IM_AIF1_ERR_EINT_MASK 0x0800 /* IM_AIF1_ERR_EINT */ +#define WM5100_IM_AIF1_ERR_EINT_SHIFT 11 /* IM_AIF1_ERR_EINT */ +#define WM5100_IM_AIF1_ERR_EINT_WIDTH 1 /* IM_AIF1_ERR_EINT */ +#define WM5100_IM_CTRLIF_ERR_EINT 0x0400 /* IM_CTRLIF_ERR_EINT */ +#define WM5100_IM_CTRLIF_ERR_EINT_MASK 0x0400 /* IM_CTRLIF_ERR_EINT */ +#define WM5100_IM_CTRLIF_ERR_EINT_SHIFT 10 /* IM_CTRLIF_ERR_EINT */ +#define WM5100_IM_CTRLIF_ERR_EINT_WIDTH 1 /* IM_CTRLIF_ERR_EINT */ +#define WM5100_IM_ISRC2_UNDERCLOCKED_EINT 0x0200 /* IM_ISRC2_UNDERCLOCKED_EINT */ +#define WM5100_IM_ISRC2_UNDERCLOCKED_EINT_MASK 0x0200 /* IM_ISRC2_UNDERCLOCKED_EINT */ +#define WM5100_IM_ISRC2_UNDERCLOCKED_EINT_SHIFT 9 /* IM_ISRC2_UNDERCLOCKED_EINT */ +#define WM5100_IM_ISRC2_UNDERCLOCKED_EINT_WIDTH 1 /* IM_ISRC2_UNDERCLOCKED_EINT */ +#define WM5100_IM_ISRC1_UNDERCLOCKED_EINT 0x0100 /* IM_ISRC1_UNDERCLOCKED_EINT */ +#define WM5100_IM_ISRC1_UNDERCLOCKED_EINT_MASK 0x0100 /* IM_ISRC1_UNDERCLOCKED_EINT */ +#define WM5100_IM_ISRC1_UNDERCLOCKED_EINT_SHIFT 8 /* IM_ISRC1_UNDERCLOCKED_EINT */ +#define WM5100_IM_ISRC1_UNDERCLOCKED_EINT_WIDTH 1 /* IM_ISRC1_UNDERCLOCKED_EINT */ +#define WM5100_IM_FX_UNDERCLOCKED_EINT 0x0080 /* IM_FX_UNDERCLOCKED_EINT */ +#define WM5100_IM_FX_UNDERCLOCKED_EINT_MASK 0x0080 /* IM_FX_UNDERCLOCKED_EINT */ +#define WM5100_IM_FX_UNDERCLOCKED_EINT_SHIFT 7 /* IM_FX_UNDERCLOCKED_EINT */ +#define WM5100_IM_FX_UNDERCLOCKED_EINT_WIDTH 1 /* IM_FX_UNDERCLOCKED_EINT */ +#define WM5100_IM_AIF3_UNDERCLOCKED_EINT 0x0040 /* IM_AIF3_UNDERCLOCKED_EINT */ +#define WM5100_IM_AIF3_UNDERCLOCKED_EINT_MASK 0x0040 /* IM_AIF3_UNDERCLOCKED_EINT */ +#define WM5100_IM_AIF3_UNDERCLOCKED_EINT_SHIFT 6 /* IM_AIF3_UNDERCLOCKED_EINT */ +#define WM5100_IM_AIF3_UNDERCLOCKED_EINT_WIDTH 1 /* IM_AIF3_UNDERCLOCKED_EINT */ +#define WM5100_IM_AIF2_UNDERCLOCKED_EINT 0x0020 /* IM_AIF2_UNDERCLOCKED_EINT */ +#define WM5100_IM_AIF2_UNDERCLOCKED_EINT_MASK 0x0020 /* IM_AIF2_UNDERCLOCKED_EINT */ +#define WM5100_IM_AIF2_UNDERCLOCKED_EINT_SHIFT 5 /* IM_AIF2_UNDERCLOCKED_EINT */ +#define WM5100_IM_AIF2_UNDERCLOCKED_EINT_WIDTH 1 /* IM_AIF2_UNDERCLOCKED_EINT */ +#define WM5100_IM_AIF1_UNDERCLOCKED_EINT 0x0010 /* IM_AIF1_UNDERCLOCKED_EINT */ +#define WM5100_IM_AIF1_UNDERCLOCKED_EINT_MASK 0x0010 /* IM_AIF1_UNDERCLOCKED_EINT */ +#define WM5100_IM_AIF1_UNDERCLOCKED_EINT_SHIFT 4 /* IM_AIF1_UNDERCLOCKED_EINT */ +#define WM5100_IM_AIF1_UNDERCLOCKED_EINT_WIDTH 1 /* IM_AIF1_UNDERCLOCKED_EINT */ +#define WM5100_IM_ASRC_UNDERCLOCKED_EINT 0x0008 /* IM_ASRC_UNDERCLOCKED_EINT */ +#define WM5100_IM_ASRC_UNDERCLOCKED_EINT_MASK 0x0008 /* IM_ASRC_UNDERCLOCKED_EINT */ +#define WM5100_IM_ASRC_UNDERCLOCKED_EINT_SHIFT 3 /* IM_ASRC_UNDERCLOCKED_EINT */ +#define WM5100_IM_ASRC_UNDERCLOCKED_EINT_WIDTH 1 /* IM_ASRC_UNDERCLOCKED_EINT */ +#define WM5100_IM_DAC_UNDERCLOCKED_EINT 0x0004 /* IM_DAC_UNDERCLOCKED_EINT */ +#define WM5100_IM_DAC_UNDERCLOCKED_EINT_MASK 0x0004 /* IM_DAC_UNDERCLOCKED_EINT */ +#define WM5100_IM_DAC_UNDERCLOCKED_EINT_SHIFT 2 /* IM_DAC_UNDERCLOCKED_EINT */ +#define WM5100_IM_DAC_UNDERCLOCKED_EINT_WIDTH 1 /* IM_DAC_UNDERCLOCKED_EINT */ +#define WM5100_IM_ADC_UNDERCLOCKED_EINT 0x0002 /* IM_ADC_UNDERCLOCKED_EINT */ +#define WM5100_IM_ADC_UNDERCLOCKED_EINT_MASK 0x0002 /* IM_ADC_UNDERCLOCKED_EINT */ +#define WM5100_IM_ADC_UNDERCLOCKED_EINT_SHIFT 1 /* IM_ADC_UNDERCLOCKED_EINT */ +#define WM5100_IM_ADC_UNDERCLOCKED_EINT_WIDTH 1 /* IM_ADC_UNDERCLOCKED_EINT */ +#define WM5100_IM_MIXER_UNDERCLOCKED_EINT 0x0001 /* IM_MIXER_UNDERCLOCKED_EINT */ +#define WM5100_IM_MIXER_UNDERCLOCKED_EINT_MASK 0x0001 /* IM_MIXER_UNDERCLOCKED_EINT */ +#define WM5100_IM_MIXER_UNDERCLOCKED_EINT_SHIFT 0 /* IM_MIXER_UNDERCLOCKED_EINT */ +#define WM5100_IM_MIXER_UNDERCLOCKED_EINT_WIDTH 1 /* IM_MIXER_UNDERCLOCKED_EINT */ + +/* + * R3359 (0xD1F) - Interrupt Control + */ +#define WM5100_IM_IRQ 0x0001 /* IM_IRQ */ +#define WM5100_IM_IRQ_MASK 0x0001 /* IM_IRQ */ +#define WM5100_IM_IRQ_SHIFT 0 /* IM_IRQ */ +#define WM5100_IM_IRQ_WIDTH 1 /* IM_IRQ */ + +/* + * R3360 (0xD20) - IRQ Debounce 1 + */ +#define WM5100_SPK_SHUTDOWN_WARN_DB 0x0200 /* SPK_SHUTDOWN_WARN_DB */ +#define WM5100_SPK_SHUTDOWN_WARN_DB_MASK 0x0200 /* SPK_SHUTDOWN_WARN_DB */ +#define WM5100_SPK_SHUTDOWN_WARN_DB_SHIFT 9 /* SPK_SHUTDOWN_WARN_DB */ +#define WM5100_SPK_SHUTDOWN_WARN_DB_WIDTH 1 /* SPK_SHUTDOWN_WARN_DB */ +#define WM5100_SPK_SHUTDOWN_DB 0x0100 /* SPK_SHUTDOWN_DB */ +#define WM5100_SPK_SHUTDOWN_DB_MASK 0x0100 /* SPK_SHUTDOWN_DB */ +#define WM5100_SPK_SHUTDOWN_DB_SHIFT 8 /* SPK_SHUTDOWN_DB */ +#define WM5100_SPK_SHUTDOWN_DB_WIDTH 1 /* SPK_SHUTDOWN_DB */ +#define WM5100_FLL1_LOCK_IRQ_DB 0x0008 /* FLL1_LOCK_IRQ_DB */ +#define WM5100_FLL1_LOCK_IRQ_DB_MASK 0x0008 /* FLL1_LOCK_IRQ_DB */ +#define WM5100_FLL1_LOCK_IRQ_DB_SHIFT 3 /* FLL1_LOCK_IRQ_DB */ +#define WM5100_FLL1_LOCK_IRQ_DB_WIDTH 1 /* FLL1_LOCK_IRQ_DB */ +#define WM5100_FLL2_LOCK_IRQ_DB 0x0004 /* FLL2_LOCK_IRQ_DB */ +#define WM5100_FLL2_LOCK_IRQ_DB_MASK 0x0004 /* FLL2_LOCK_IRQ_DB */ +#define WM5100_FLL2_LOCK_IRQ_DB_SHIFT 2 /* FLL2_LOCK_IRQ_DB */ +#define WM5100_FLL2_LOCK_IRQ_DB_WIDTH 1 /* FLL2_LOCK_IRQ_DB */ +#define WM5100_CLKGEN_ERR_IRQ_DB 0x0002 /* CLKGEN_ERR_IRQ_DB */ +#define WM5100_CLKGEN_ERR_IRQ_DB_MASK 0x0002 /* CLKGEN_ERR_IRQ_DB */ +#define WM5100_CLKGEN_ERR_IRQ_DB_SHIFT 1 /* CLKGEN_ERR_IRQ_DB */ +#define WM5100_CLKGEN_ERR_IRQ_DB_WIDTH 1 /* CLKGEN_ERR_IRQ_DB */ +#define WM5100_CLKGEN_ERR_ASYNC_IRQ_DB 0x0001 /* CLKGEN_ERR_ASYNC_IRQ_DB */ +#define WM5100_CLKGEN_ERR_ASYNC_IRQ_DB_MASK 0x0001 /* CLKGEN_ERR_ASYNC_IRQ_DB */ +#define WM5100_CLKGEN_ERR_ASYNC_IRQ_DB_SHIFT 0 /* CLKGEN_ERR_ASYNC_IRQ_DB */ +#define WM5100_CLKGEN_ERR_ASYNC_IRQ_DB_WIDTH 1 /* CLKGEN_ERR_ASYNC_IRQ_DB */ + +/* + * R3361 (0xD21) - IRQ Debounce 2 + */ +#define WM5100_AIF_ERR_DB 0x0001 /* AIF_ERR_DB */ +#define WM5100_AIF_ERR_DB_MASK 0x0001 /* AIF_ERR_DB */ +#define WM5100_AIF_ERR_DB_SHIFT 0 /* AIF_ERR_DB */ +#define WM5100_AIF_ERR_DB_WIDTH 1 /* AIF_ERR_DB */ + +/* + * R3584 (0xE00) - FX_Ctrl + */ +#define WM5100_FX_STS_MASK 0xFFC0 /* FX_STS - [15:6] */ +#define WM5100_FX_STS_SHIFT 6 /* FX_STS - [15:6] */ +#define WM5100_FX_STS_WIDTH 10 /* FX_STS - [15:6] */ +#define WM5100_FX_RATE_MASK 0x0003 /* FX_RATE - [1:0] */ +#define WM5100_FX_RATE_SHIFT 0 /* FX_RATE - [1:0] */ +#define WM5100_FX_RATE_WIDTH 2 /* FX_RATE - [1:0] */ + +/* + * R3600 (0xE10) - EQ1_1 + */ +#define WM5100_EQ1_B1_GAIN_MASK 0xF800 /* EQ1_B1_GAIN - [15:11] */ +#define WM5100_EQ1_B1_GAIN_SHIFT 11 /* EQ1_B1_GAIN - [15:11] */ +#define WM5100_EQ1_B1_GAIN_WIDTH 5 /* EQ1_B1_GAIN - [15:11] */ +#define WM5100_EQ1_B2_GAIN_MASK 0x07C0 /* EQ1_B2_GAIN - [10:6] */ +#define WM5100_EQ1_B2_GAIN_SHIFT 6 /* EQ1_B2_GAIN - [10:6] */ +#define WM5100_EQ1_B2_GAIN_WIDTH 5 /* EQ1_B2_GAIN - [10:6] */ +#define WM5100_EQ1_B3_GAIN_MASK 0x003E /* EQ1_B3_GAIN - [5:1] */ +#define WM5100_EQ1_B3_GAIN_SHIFT 1 /* EQ1_B3_GAIN - [5:1] */ +#define WM5100_EQ1_B3_GAIN_WIDTH 5 /* EQ1_B3_GAIN - [5:1] */ +#define WM5100_EQ1_ENA 0x0001 /* EQ1_ENA */ +#define WM5100_EQ1_ENA_MASK 0x0001 /* EQ1_ENA */ +#define WM5100_EQ1_ENA_SHIFT 0 /* EQ1_ENA */ +#define WM5100_EQ1_ENA_WIDTH 1 /* EQ1_ENA */ + +/* + * R3601 (0xE11) - EQ1_2 + */ +#define WM5100_EQ1_B4_GAIN_MASK 0xF800 /* EQ1_B4_GAIN - [15:11] */ +#define WM5100_EQ1_B4_GAIN_SHIFT 11 /* EQ1_B4_GAIN - [15:11] */ +#define WM5100_EQ1_B4_GAIN_WIDTH 5 /* EQ1_B4_GAIN - [15:11] */ +#define WM5100_EQ1_B5_GAIN_MASK 0x07C0 /* EQ1_B5_GAIN - [10:6] */ +#define WM5100_EQ1_B5_GAIN_SHIFT 6 /* EQ1_B5_GAIN - [10:6] */ +#define WM5100_EQ1_B5_GAIN_WIDTH 5 /* EQ1_B5_GAIN - [10:6] */ + +/* + * R3602 (0xE12) - EQ1_3 + */ +#define WM5100_EQ1_B1_A_MASK 0xFFFF /* EQ1_B1_A - [15:0] */ +#define WM5100_EQ1_B1_A_SHIFT 0 /* EQ1_B1_A - [15:0] */ +#define WM5100_EQ1_B1_A_WIDTH 16 /* EQ1_B1_A - [15:0] */ + +/* + * R3603 (0xE13) - EQ1_4 + */ +#define WM5100_EQ1_B1_B_MASK 0xFFFF /* EQ1_B1_B - [15:0] */ +#define WM5100_EQ1_B1_B_SHIFT 0 /* EQ1_B1_B - [15:0] */ +#define WM5100_EQ1_B1_B_WIDTH 16 /* EQ1_B1_B - [15:0] */ + +/* + * R3604 (0xE14) - EQ1_5 + */ +#define WM5100_EQ1_B1_PG_MASK 0xFFFF /* EQ1_B1_PG - [15:0] */ +#define WM5100_EQ1_B1_PG_SHIFT 0 /* EQ1_B1_PG - [15:0] */ +#define WM5100_EQ1_B1_PG_WIDTH 16 /* EQ1_B1_PG - [15:0] */ + +/* + * R3605 (0xE15) - EQ1_6 + */ +#define WM5100_EQ1_B2_A_MASK 0xFFFF /* EQ1_B2_A - [15:0] */ +#define WM5100_EQ1_B2_A_SHIFT 0 /* EQ1_B2_A - [15:0] */ +#define WM5100_EQ1_B2_A_WIDTH 16 /* EQ1_B2_A - [15:0] */ + +/* + * R3606 (0xE16) - EQ1_7 + */ +#define WM5100_EQ1_B2_B_MASK 0xFFFF /* EQ1_B2_B - [15:0] */ +#define WM5100_EQ1_B2_B_SHIFT 0 /* EQ1_B2_B - [15:0] */ +#define WM5100_EQ1_B2_B_WIDTH 16 /* EQ1_B2_B - [15:0] */ + +/* + * R3607 (0xE17) - EQ1_8 + */ +#define WM5100_EQ1_B2_C_MASK 0xFFFF /* EQ1_B2_C - [15:0] */ +#define WM5100_EQ1_B2_C_SHIFT 0 /* EQ1_B2_C - [15:0] */ +#define WM5100_EQ1_B2_C_WIDTH 16 /* EQ1_B2_C - [15:0] */ + +/* + * R3608 (0xE18) - EQ1_9 + */ +#define WM5100_EQ1_B2_PG_MASK 0xFFFF /* EQ1_B2_PG - [15:0] */ +#define WM5100_EQ1_B2_PG_SHIFT 0 /* EQ1_B2_PG - [15:0] */ +#define WM5100_EQ1_B2_PG_WIDTH 16 /* EQ1_B2_PG - [15:0] */ + +/* + * R3609 (0xE19) - EQ1_10 + */ +#define WM5100_EQ1_B3_A_MASK 0xFFFF /* EQ1_B3_A - [15:0] */ +#define WM5100_EQ1_B3_A_SHIFT 0 /* EQ1_B3_A - [15:0] */ +#define WM5100_EQ1_B3_A_WIDTH 16 /* EQ1_B3_A - [15:0] */ + +/* + * R3610 (0xE1A) - EQ1_11 + */ +#define WM5100_EQ1_B3_B_MASK 0xFFFF /* EQ1_B3_B - [15:0] */ +#define WM5100_EQ1_B3_B_SHIFT 0 /* EQ1_B3_B - [15:0] */ +#define WM5100_EQ1_B3_B_WIDTH 16 /* EQ1_B3_B - [15:0] */ + +/* + * R3611 (0xE1B) - EQ1_12 + */ +#define WM5100_EQ1_B3_C_MASK 0xFFFF /* EQ1_B3_C - [15:0] */ +#define WM5100_EQ1_B3_C_SHIFT 0 /* EQ1_B3_C - [15:0] */ +#define WM5100_EQ1_B3_C_WIDTH 16 /* EQ1_B3_C - [15:0] */ + +/* + * R3612 (0xE1C) - EQ1_13 + */ +#define WM5100_EQ1_B3_PG_MASK 0xFFFF /* EQ1_B3_PG - [15:0] */ +#define WM5100_EQ1_B3_PG_SHIFT 0 /* EQ1_B3_PG - [15:0] */ +#define WM5100_EQ1_B3_PG_WIDTH 16 /* EQ1_B3_PG - [15:0] */ + +/* + * R3613 (0xE1D) - EQ1_14 + */ +#define WM5100_EQ1_B4_A_MASK 0xFFFF /* EQ1_B4_A - [15:0] */ +#define WM5100_EQ1_B4_A_SHIFT 0 /* EQ1_B4_A - [15:0] */ +#define WM5100_EQ1_B4_A_WIDTH 16 /* EQ1_B4_A - [15:0] */ + +/* + * R3614 (0xE1E) - EQ1_15 + */ +#define WM5100_EQ1_B4_B_MASK 0xFFFF /* EQ1_B4_B - [15:0] */ +#define WM5100_EQ1_B4_B_SHIFT 0 /* EQ1_B4_B - [15:0] */ +#define WM5100_EQ1_B4_B_WIDTH 16 /* EQ1_B4_B - [15:0] */ + +/* + * R3615 (0xE1F) - EQ1_16 + */ +#define WM5100_EQ1_B4_C_MASK 0xFFFF /* EQ1_B4_C - [15:0] */ +#define WM5100_EQ1_B4_C_SHIFT 0 /* EQ1_B4_C - [15:0] */ +#define WM5100_EQ1_B4_C_WIDTH 16 /* EQ1_B4_C - [15:0] */ + +/* + * R3616 (0xE20) - EQ1_17 + */ +#define WM5100_EQ1_B4_PG_MASK 0xFFFF /* EQ1_B4_PG - [15:0] */ +#define WM5100_EQ1_B4_PG_SHIFT 0 /* EQ1_B4_PG - [15:0] */ +#define WM5100_EQ1_B4_PG_WIDTH 16 /* EQ1_B4_PG - [15:0] */ + +/* + * R3617 (0xE21) - EQ1_18 + */ +#define WM5100_EQ1_B5_A_MASK 0xFFFF /* EQ1_B5_A - [15:0] */ +#define WM5100_EQ1_B5_A_SHIFT 0 /* EQ1_B5_A - [15:0] */ +#define WM5100_EQ1_B5_A_WIDTH 16 /* EQ1_B5_A - [15:0] */ + +/* + * R3618 (0xE22) - EQ1_19 + */ +#define WM5100_EQ1_B5_B_MASK 0xFFFF /* EQ1_B5_B - [15:0] */ +#define WM5100_EQ1_B5_B_SHIFT 0 /* EQ1_B5_B - [15:0] */ +#define WM5100_EQ1_B5_B_WIDTH 16 /* EQ1_B5_B - [15:0] */ + +/* + * R3619 (0xE23) - EQ1_20 + */ +#define WM5100_EQ1_B5_PG_MASK 0xFFFF /* EQ1_B5_PG - [15:0] */ +#define WM5100_EQ1_B5_PG_SHIFT 0 /* EQ1_B5_PG - [15:0] */ +#define WM5100_EQ1_B5_PG_WIDTH 16 /* EQ1_B5_PG - [15:0] */ + +/* + * R3622 (0xE26) - EQ2_1 + */ +#define WM5100_EQ2_B1_GAIN_MASK 0xF800 /* EQ2_B1_GAIN - [15:11] */ +#define WM5100_EQ2_B1_GAIN_SHIFT 11 /* EQ2_B1_GAIN - [15:11] */ +#define WM5100_EQ2_B1_GAIN_WIDTH 5 /* EQ2_B1_GAIN - [15:11] */ +#define WM5100_EQ2_B2_GAIN_MASK 0x07C0 /* EQ2_B2_GAIN - [10:6] */ +#define WM5100_EQ2_B2_GAIN_SHIFT 6 /* EQ2_B2_GAIN - [10:6] */ +#define WM5100_EQ2_B2_GAIN_WIDTH 5 /* EQ2_B2_GAIN - [10:6] */ +#define WM5100_EQ2_B3_GAIN_MASK 0x003E /* EQ2_B3_GAIN - [5:1] */ +#define WM5100_EQ2_B3_GAIN_SHIFT 1 /* EQ2_B3_GAIN - [5:1] */ +#define WM5100_EQ2_B3_GAIN_WIDTH 5 /* EQ2_B3_GAIN - [5:1] */ +#define WM5100_EQ2_ENA 0x0001 /* EQ2_ENA */ +#define WM5100_EQ2_ENA_MASK 0x0001 /* EQ2_ENA */ +#define WM5100_EQ2_ENA_SHIFT 0 /* EQ2_ENA */ +#define WM5100_EQ2_ENA_WIDTH 1 /* EQ2_ENA */ + +/* + * R3623 (0xE27) - EQ2_2 + */ +#define WM5100_EQ2_B4_GAIN_MASK 0xF800 /* EQ2_B4_GAIN - [15:11] */ +#define WM5100_EQ2_B4_GAIN_SHIFT 11 /* EQ2_B4_GAIN - [15:11] */ +#define WM5100_EQ2_B4_GAIN_WIDTH 5 /* EQ2_B4_GAIN - [15:11] */ +#define WM5100_EQ2_B5_GAIN_MASK 0x07C0 /* EQ2_B5_GAIN - [10:6] */ +#define WM5100_EQ2_B5_GAIN_SHIFT 6 /* EQ2_B5_GAIN - [10:6] */ +#define WM5100_EQ2_B5_GAIN_WIDTH 5 /* EQ2_B5_GAIN - [10:6] */ + +/* + * R3624 (0xE28) - EQ2_3 + */ +#define WM5100_EQ2_B1_A_MASK 0xFFFF /* EQ2_B1_A - [15:0] */ +#define WM5100_EQ2_B1_A_SHIFT 0 /* EQ2_B1_A - [15:0] */ +#define WM5100_EQ2_B1_A_WIDTH 16 /* EQ2_B1_A - [15:0] */ + +/* + * R3625 (0xE29) - EQ2_4 + */ +#define WM5100_EQ2_B1_B_MASK 0xFFFF /* EQ2_B1_B - [15:0] */ +#define WM5100_EQ2_B1_B_SHIFT 0 /* EQ2_B1_B - [15:0] */ +#define WM5100_EQ2_B1_B_WIDTH 16 /* EQ2_B1_B - [15:0] */ + +/* + * R3626 (0xE2A) - EQ2_5 + */ +#define WM5100_EQ2_B1_PG_MASK 0xFFFF /* EQ2_B1_PG - [15:0] */ +#define WM5100_EQ2_B1_PG_SHIFT 0 /* EQ2_B1_PG - [15:0] */ +#define WM5100_EQ2_B1_PG_WIDTH 16 /* EQ2_B1_PG - [15:0] */ + +/* + * R3627 (0xE2B) - EQ2_6 + */ +#define WM5100_EQ2_B2_A_MASK 0xFFFF /* EQ2_B2_A - [15:0] */ +#define WM5100_EQ2_B2_A_SHIFT 0 /* EQ2_B2_A - [15:0] */ +#define WM5100_EQ2_B2_A_WIDTH 16 /* EQ2_B2_A - [15:0] */ + +/* + * R3628 (0xE2C) - EQ2_7 + */ +#define WM5100_EQ2_B2_B_MASK 0xFFFF /* EQ2_B2_B - [15:0] */ +#define WM5100_EQ2_B2_B_SHIFT 0 /* EQ2_B2_B - [15:0] */ +#define WM5100_EQ2_B2_B_WIDTH 16 /* EQ2_B2_B - [15:0] */ + +/* + * R3629 (0xE2D) - EQ2_8 + */ +#define WM5100_EQ2_B2_C_MASK 0xFFFF /* EQ2_B2_C - [15:0] */ +#define WM5100_EQ2_B2_C_SHIFT 0 /* EQ2_B2_C - [15:0] */ +#define WM5100_EQ2_B2_C_WIDTH 16 /* EQ2_B2_C - [15:0] */ + +/* + * R3630 (0xE2E) - EQ2_9 + */ +#define WM5100_EQ2_B2_PG_MASK 0xFFFF /* EQ2_B2_PG - [15:0] */ +#define WM5100_EQ2_B2_PG_SHIFT 0 /* EQ2_B2_PG - [15:0] */ +#define WM5100_EQ2_B2_PG_WIDTH 16 /* EQ2_B2_PG - [15:0] */ + +/* + * R3631 (0xE2F) - EQ2_10 + */ +#define WM5100_EQ2_B3_A_MASK 0xFFFF /* EQ2_B3_A - [15:0] */ +#define WM5100_EQ2_B3_A_SHIFT 0 /* EQ2_B3_A - [15:0] */ +#define WM5100_EQ2_B3_A_WIDTH 16 /* EQ2_B3_A - [15:0] */ + +/* + * R3632 (0xE30) - EQ2_11 + */ +#define WM5100_EQ2_B3_B_MASK 0xFFFF /* EQ2_B3_B - [15:0] */ +#define WM5100_EQ2_B3_B_SHIFT 0 /* EQ2_B3_B - [15:0] */ +#define WM5100_EQ2_B3_B_WIDTH 16 /* EQ2_B3_B - [15:0] */ + +/* + * R3633 (0xE31) - EQ2_12 + */ +#define WM5100_EQ2_B3_C_MASK 0xFFFF /* EQ2_B3_C - [15:0] */ +#define WM5100_EQ2_B3_C_SHIFT 0 /* EQ2_B3_C - [15:0] */ +#define WM5100_EQ2_B3_C_WIDTH 16 /* EQ2_B3_C - [15:0] */ + +/* + * R3634 (0xE32) - EQ2_13 + */ +#define WM5100_EQ2_B3_PG_MASK 0xFFFF /* EQ2_B3_PG - [15:0] */ +#define WM5100_EQ2_B3_PG_SHIFT 0 /* EQ2_B3_PG - [15:0] */ +#define WM5100_EQ2_B3_PG_WIDTH 16 /* EQ2_B3_PG - [15:0] */ + +/* + * R3635 (0xE33) - EQ2_14 + */ +#define WM5100_EQ2_B4_A_MASK 0xFFFF /* EQ2_B4_A - [15:0] */ +#define WM5100_EQ2_B4_A_SHIFT 0 /* EQ2_B4_A - [15:0] */ +#define WM5100_EQ2_B4_A_WIDTH 16 /* EQ2_B4_A - [15:0] */ + +/* + * R3636 (0xE34) - EQ2_15 + */ +#define WM5100_EQ2_B4_B_MASK 0xFFFF /* EQ2_B4_B - [15:0] */ +#define WM5100_EQ2_B4_B_SHIFT 0 /* EQ2_B4_B - [15:0] */ +#define WM5100_EQ2_B4_B_WIDTH 16 /* EQ2_B4_B - [15:0] */ + +/* + * R3637 (0xE35) - EQ2_16 + */ +#define WM5100_EQ2_B4_C_MASK 0xFFFF /* EQ2_B4_C - [15:0] */ +#define WM5100_EQ2_B4_C_SHIFT 0 /* EQ2_B4_C - [15:0] */ +#define WM5100_EQ2_B4_C_WIDTH 16 /* EQ2_B4_C - [15:0] */ + +/* + * R3638 (0xE36) - EQ2_17 + */ +#define WM5100_EQ2_B4_PG_MASK 0xFFFF /* EQ2_B4_PG - [15:0] */ +#define WM5100_EQ2_B4_PG_SHIFT 0 /* EQ2_B4_PG - [15:0] */ +#define WM5100_EQ2_B4_PG_WIDTH 16 /* EQ2_B4_PG - [15:0] */ + +/* + * R3639 (0xE37) - EQ2_18 + */ +#define WM5100_EQ2_B5_A_MASK 0xFFFF /* EQ2_B5_A - [15:0] */ +#define WM5100_EQ2_B5_A_SHIFT 0 /* EQ2_B5_A - [15:0] */ +#define WM5100_EQ2_B5_A_WIDTH 16 /* EQ2_B5_A - [15:0] */ + +/* + * R3640 (0xE38) - EQ2_19 + */ +#define WM5100_EQ2_B5_B_MASK 0xFFFF /* EQ2_B5_B - [15:0] */ +#define WM5100_EQ2_B5_B_SHIFT 0 /* EQ2_B5_B - [15:0] */ +#define WM5100_EQ2_B5_B_WIDTH 16 /* EQ2_B5_B - [15:0] */ + +/* + * R3641 (0xE39) - EQ2_20 + */ +#define WM5100_EQ2_B5_PG_MASK 0xFFFF /* EQ2_B5_PG - [15:0] */ +#define WM5100_EQ2_B5_PG_SHIFT 0 /* EQ2_B5_PG - [15:0] */ +#define WM5100_EQ2_B5_PG_WIDTH 16 /* EQ2_B5_PG - [15:0] */ + +/* + * R3644 (0xE3C) - EQ3_1 + */ +#define WM5100_EQ3_B1_GAIN_MASK 0xF800 /* EQ3_B1_GAIN - [15:11] */ +#define WM5100_EQ3_B1_GAIN_SHIFT 11 /* EQ3_B1_GAIN - [15:11] */ +#define WM5100_EQ3_B1_GAIN_WIDTH 5 /* EQ3_B1_GAIN - [15:11] */ +#define WM5100_EQ3_B2_GAIN_MASK 0x07C0 /* EQ3_B2_GAIN - [10:6] */ +#define WM5100_EQ3_B2_GAIN_SHIFT 6 /* EQ3_B2_GAIN - [10:6] */ +#define WM5100_EQ3_B2_GAIN_WIDTH 5 /* EQ3_B2_GAIN - [10:6] */ +#define WM5100_EQ3_B3_GAIN_MASK 0x003E /* EQ3_B3_GAIN - [5:1] */ +#define WM5100_EQ3_B3_GAIN_SHIFT 1 /* EQ3_B3_GAIN - [5:1] */ +#define WM5100_EQ3_B3_GAIN_WIDTH 5 /* EQ3_B3_GAIN - [5:1] */ +#define WM5100_EQ3_ENA 0x0001 /* EQ3_ENA */ +#define WM5100_EQ3_ENA_MASK 0x0001 /* EQ3_ENA */ +#define WM5100_EQ3_ENA_SHIFT 0 /* EQ3_ENA */ +#define WM5100_EQ3_ENA_WIDTH 1 /* EQ3_ENA */ + +/* + * R3645 (0xE3D) - EQ3_2 + */ +#define WM5100_EQ3_B4_GAIN_MASK 0xF800 /* EQ3_B4_GAIN - [15:11] */ +#define WM5100_EQ3_B4_GAIN_SHIFT 11 /* EQ3_B4_GAIN - [15:11] */ +#define WM5100_EQ3_B4_GAIN_WIDTH 5 /* EQ3_B4_GAIN - [15:11] */ +#define WM5100_EQ3_B5_GAIN_MASK 0x07C0 /* EQ3_B5_GAIN - [10:6] */ +#define WM5100_EQ3_B5_GAIN_SHIFT 6 /* EQ3_B5_GAIN - [10:6] */ +#define WM5100_EQ3_B5_GAIN_WIDTH 5 /* EQ3_B5_GAIN - [10:6] */ + +/* + * R3646 (0xE3E) - EQ3_3 + */ +#define WM5100_EQ3_B1_A_MASK 0xFFFF /* EQ3_B1_A - [15:0] */ +#define WM5100_EQ3_B1_A_SHIFT 0 /* EQ3_B1_A - [15:0] */ +#define WM5100_EQ3_B1_A_WIDTH 16 /* EQ3_B1_A - [15:0] */ + +/* + * R3647 (0xE3F) - EQ3_4 + */ +#define WM5100_EQ3_B1_B_MASK 0xFFFF /* EQ3_B1_B - [15:0] */ +#define WM5100_EQ3_B1_B_SHIFT 0 /* EQ3_B1_B - [15:0] */ +#define WM5100_EQ3_B1_B_WIDTH 16 /* EQ3_B1_B - [15:0] */ + +/* + * R3648 (0xE40) - EQ3_5 + */ +#define WM5100_EQ3_B1_PG_MASK 0xFFFF /* EQ3_B1_PG - [15:0] */ +#define WM5100_EQ3_B1_PG_SHIFT 0 /* EQ3_B1_PG - [15:0] */ +#define WM5100_EQ3_B1_PG_WIDTH 16 /* EQ3_B1_PG - [15:0] */ + +/* + * R3649 (0xE41) - EQ3_6 + */ +#define WM5100_EQ3_B2_A_MASK 0xFFFF /* EQ3_B2_A - [15:0] */ +#define WM5100_EQ3_B2_A_SHIFT 0 /* EQ3_B2_A - [15:0] */ +#define WM5100_EQ3_B2_A_WIDTH 16 /* EQ3_B2_A - [15:0] */ + +/* + * R3650 (0xE42) - EQ3_7 + */ +#define WM5100_EQ3_B2_B_MASK 0xFFFF /* EQ3_B2_B - [15:0] */ +#define WM5100_EQ3_B2_B_SHIFT 0 /* EQ3_B2_B - [15:0] */ +#define WM5100_EQ3_B2_B_WIDTH 16 /* EQ3_B2_B - [15:0] */ + +/* + * R3651 (0xE43) - EQ3_8 + */ +#define WM5100_EQ3_B2_C_MASK 0xFFFF /* EQ3_B2_C - [15:0] */ +#define WM5100_EQ3_B2_C_SHIFT 0 /* EQ3_B2_C - [15:0] */ +#define WM5100_EQ3_B2_C_WIDTH 16 /* EQ3_B2_C - [15:0] */ + +/* + * R3652 (0xE44) - EQ3_9 + */ +#define WM5100_EQ3_B2_PG_MASK 0xFFFF /* EQ3_B2_PG - [15:0] */ +#define WM5100_EQ3_B2_PG_SHIFT 0 /* EQ3_B2_PG - [15:0] */ +#define WM5100_EQ3_B2_PG_WIDTH 16 /* EQ3_B2_PG - [15:0] */ + +/* + * R3653 (0xE45) - EQ3_10 + */ +#define WM5100_EQ3_B3_A_MASK 0xFFFF /* EQ3_B3_A - [15:0] */ +#define WM5100_EQ3_B3_A_SHIFT 0 /* EQ3_B3_A - [15:0] */ +#define WM5100_EQ3_B3_A_WIDTH 16 /* EQ3_B3_A - [15:0] */ + +/* + * R3654 (0xE46) - EQ3_11 + */ +#define WM5100_EQ3_B3_B_MASK 0xFFFF /* EQ3_B3_B - [15:0] */ +#define WM5100_EQ3_B3_B_SHIFT 0 /* EQ3_B3_B - [15:0] */ +#define WM5100_EQ3_B3_B_WIDTH 16 /* EQ3_B3_B - [15:0] */ + +/* + * R3655 (0xE47) - EQ3_12 + */ +#define WM5100_EQ3_B3_C_MASK 0xFFFF /* EQ3_B3_C - [15:0] */ +#define WM5100_EQ3_B3_C_SHIFT 0 /* EQ3_B3_C - [15:0] */ +#define WM5100_EQ3_B3_C_WIDTH 16 /* EQ3_B3_C - [15:0] */ + +/* + * R3656 (0xE48) - EQ3_13 + */ +#define WM5100_EQ3_B3_PG_MASK 0xFFFF /* EQ3_B3_PG - [15:0] */ +#define WM5100_EQ3_B3_PG_SHIFT 0 /* EQ3_B3_PG - [15:0] */ +#define WM5100_EQ3_B3_PG_WIDTH 16 /* EQ3_B3_PG - [15:0] */ + +/* + * R3657 (0xE49) - EQ3_14 + */ +#define WM5100_EQ3_B4_A_MASK 0xFFFF /* EQ3_B4_A - [15:0] */ +#define WM5100_EQ3_B4_A_SHIFT 0 /* EQ3_B4_A - [15:0] */ +#define WM5100_EQ3_B4_A_WIDTH 16 /* EQ3_B4_A - [15:0] */ + +/* + * R3658 (0xE4A) - EQ3_15 + */ +#define WM5100_EQ3_B4_B_MASK 0xFFFF /* EQ3_B4_B - [15:0] */ +#define WM5100_EQ3_B4_B_SHIFT 0 /* EQ3_B4_B - [15:0] */ +#define WM5100_EQ3_B4_B_WIDTH 16 /* EQ3_B4_B - [15:0] */ + +/* + * R3659 (0xE4B) - EQ3_16 + */ +#define WM5100_EQ3_B4_C_MASK 0xFFFF /* EQ3_B4_C - [15:0] */ +#define WM5100_EQ3_B4_C_SHIFT 0 /* EQ3_B4_C - [15:0] */ +#define WM5100_EQ3_B4_C_WIDTH 16 /* EQ3_B4_C - [15:0] */ + +/* + * R3660 (0xE4C) - EQ3_17 + */ +#define WM5100_EQ3_B4_PG_MASK 0xFFFF /* EQ3_B4_PG - [15:0] */ +#define WM5100_EQ3_B4_PG_SHIFT 0 /* EQ3_B4_PG - [15:0] */ +#define WM5100_EQ3_B4_PG_WIDTH 16 /* EQ3_B4_PG - [15:0] */ + +/* + * R3661 (0xE4D) - EQ3_18 + */ +#define WM5100_EQ3_B5_A_MASK 0xFFFF /* EQ3_B5_A - [15:0] */ +#define WM5100_EQ3_B5_A_SHIFT 0 /* EQ3_B5_A - [15:0] */ +#define WM5100_EQ3_B5_A_WIDTH 16 /* EQ3_B5_A - [15:0] */ + +/* + * R3662 (0xE4E) - EQ3_19 + */ +#define WM5100_EQ3_B5_B_MASK 0xFFFF /* EQ3_B5_B - [15:0] */ +#define WM5100_EQ3_B5_B_SHIFT 0 /* EQ3_B5_B - [15:0] */ +#define WM5100_EQ3_B5_B_WIDTH 16 /* EQ3_B5_B - [15:0] */ + +/* + * R3663 (0xE4F) - EQ3_20 + */ +#define WM5100_EQ3_B5_PG_MASK 0xFFFF /* EQ3_B5_PG - [15:0] */ +#define WM5100_EQ3_B5_PG_SHIFT 0 /* EQ3_B5_PG - [15:0] */ +#define WM5100_EQ3_B5_PG_WIDTH 16 /* EQ3_B5_PG - [15:0] */ + +/* + * R3666 (0xE52) - EQ4_1 + */ +#define WM5100_EQ4_B1_GAIN_MASK 0xF800 /* EQ4_B1_GAIN - [15:11] */ +#define WM5100_EQ4_B1_GAIN_SHIFT 11 /* EQ4_B1_GAIN - [15:11] */ +#define WM5100_EQ4_B1_GAIN_WIDTH 5 /* EQ4_B1_GAIN - [15:11] */ +#define WM5100_EQ4_B2_GAIN_MASK 0x07C0 /* EQ4_B2_GAIN - [10:6] */ +#define WM5100_EQ4_B2_GAIN_SHIFT 6 /* EQ4_B2_GAIN - [10:6] */ +#define WM5100_EQ4_B2_GAIN_WIDTH 5 /* EQ4_B2_GAIN - [10:6] */ +#define WM5100_EQ4_B3_GAIN_MASK 0x003E /* EQ4_B3_GAIN - [5:1] */ +#define WM5100_EQ4_B3_GAIN_SHIFT 1 /* EQ4_B3_GAIN - [5:1] */ +#define WM5100_EQ4_B3_GAIN_WIDTH 5 /* EQ4_B3_GAIN - [5:1] */ +#define WM5100_EQ4_ENA 0x0001 /* EQ4_ENA */ +#define WM5100_EQ4_ENA_MASK 0x0001 /* EQ4_ENA */ +#define WM5100_EQ4_ENA_SHIFT 0 /* EQ4_ENA */ +#define WM5100_EQ4_ENA_WIDTH 1 /* EQ4_ENA */ + +/* + * R3667 (0xE53) - EQ4_2 + */ +#define WM5100_EQ4_B4_GAIN_MASK 0xF800 /* EQ4_B4_GAIN - [15:11] */ +#define WM5100_EQ4_B4_GAIN_SHIFT 11 /* EQ4_B4_GAIN - [15:11] */ +#define WM5100_EQ4_B4_GAIN_WIDTH 5 /* EQ4_B4_GAIN - [15:11] */ +#define WM5100_EQ4_B5_GAIN_MASK 0x07C0 /* EQ4_B5_GAIN - [10:6] */ +#define WM5100_EQ4_B5_GAIN_SHIFT 6 /* EQ4_B5_GAIN - [10:6] */ +#define WM5100_EQ4_B5_GAIN_WIDTH 5 /* EQ4_B5_GAIN - [10:6] */ + +/* + * R3668 (0xE54) - EQ4_3 + */ +#define WM5100_EQ4_B1_A_MASK 0xFFFF /* EQ4_B1_A - [15:0] */ +#define WM5100_EQ4_B1_A_SHIFT 0 /* EQ4_B1_A - [15:0] */ +#define WM5100_EQ4_B1_A_WIDTH 16 /* EQ4_B1_A - [15:0] */ + +/* + * R3669 (0xE55) - EQ4_4 + */ +#define WM5100_EQ4_B1_B_MASK 0xFFFF /* EQ4_B1_B - [15:0] */ +#define WM5100_EQ4_B1_B_SHIFT 0 /* EQ4_B1_B - [15:0] */ +#define WM5100_EQ4_B1_B_WIDTH 16 /* EQ4_B1_B - [15:0] */ + +/* + * R3670 (0xE56) - EQ4_5 + */ +#define WM5100_EQ4_B1_PG_MASK 0xFFFF /* EQ4_B1_PG - [15:0] */ +#define WM5100_EQ4_B1_PG_SHIFT 0 /* EQ4_B1_PG - [15:0] */ +#define WM5100_EQ4_B1_PG_WIDTH 16 /* EQ4_B1_PG - [15:0] */ + +/* + * R3671 (0xE57) - EQ4_6 + */ +#define WM5100_EQ4_B2_A_MASK 0xFFFF /* EQ4_B2_A - [15:0] */ +#define WM5100_EQ4_B2_A_SHIFT 0 /* EQ4_B2_A - [15:0] */ +#define WM5100_EQ4_B2_A_WIDTH 16 /* EQ4_B2_A - [15:0] */ + +/* + * R3672 (0xE58) - EQ4_7 + */ +#define WM5100_EQ4_B2_B_MASK 0xFFFF /* EQ4_B2_B - [15:0] */ +#define WM5100_EQ4_B2_B_SHIFT 0 /* EQ4_B2_B - [15:0] */ +#define WM5100_EQ4_B2_B_WIDTH 16 /* EQ4_B2_B - [15:0] */ + +/* + * R3673 (0xE59) - EQ4_8 + */ +#define WM5100_EQ4_B2_C_MASK 0xFFFF /* EQ4_B2_C - [15:0] */ +#define WM5100_EQ4_B2_C_SHIFT 0 /* EQ4_B2_C - [15:0] */ +#define WM5100_EQ4_B2_C_WIDTH 16 /* EQ4_B2_C - [15:0] */ + +/* + * R3674 (0xE5A) - EQ4_9 + */ +#define WM5100_EQ4_B2_PG_MASK 0xFFFF /* EQ4_B2_PG - [15:0] */ +#define WM5100_EQ4_B2_PG_SHIFT 0 /* EQ4_B2_PG - [15:0] */ +#define WM5100_EQ4_B2_PG_WIDTH 16 /* EQ4_B2_PG - [15:0] */ + +/* + * R3675 (0xE5B) - EQ4_10 + */ +#define WM5100_EQ4_B3_A_MASK 0xFFFF /* EQ4_B3_A - [15:0] */ +#define WM5100_EQ4_B3_A_SHIFT 0 /* EQ4_B3_A - [15:0] */ +#define WM5100_EQ4_B3_A_WIDTH 16 /* EQ4_B3_A - [15:0] */ + +/* + * R3676 (0xE5C) - EQ4_11 + */ +#define WM5100_EQ4_B3_B_MASK 0xFFFF /* EQ4_B3_B - [15:0] */ +#define WM5100_EQ4_B3_B_SHIFT 0 /* EQ4_B3_B - [15:0] */ +#define WM5100_EQ4_B3_B_WIDTH 16 /* EQ4_B3_B - [15:0] */ + +/* + * R3677 (0xE5D) - EQ4_12 + */ +#define WM5100_EQ4_B3_C_MASK 0xFFFF /* EQ4_B3_C - [15:0] */ +#define WM5100_EQ4_B3_C_SHIFT 0 /* EQ4_B3_C - [15:0] */ +#define WM5100_EQ4_B3_C_WIDTH 16 /* EQ4_B3_C - [15:0] */ + +/* + * R3678 (0xE5E) - EQ4_13 + */ +#define WM5100_EQ4_B3_PG_MASK 0xFFFF /* EQ4_B3_PG - [15:0] */ +#define WM5100_EQ4_B3_PG_SHIFT 0 /* EQ4_B3_PG - [15:0] */ +#define WM5100_EQ4_B3_PG_WIDTH 16 /* EQ4_B3_PG - [15:0] */ + +/* + * R3679 (0xE5F) - EQ4_14 + */ +#define WM5100_EQ4_B4_A_MASK 0xFFFF /* EQ4_B4_A - [15:0] */ +#define WM5100_EQ4_B4_A_SHIFT 0 /* EQ4_B4_A - [15:0] */ +#define WM5100_EQ4_B4_A_WIDTH 16 /* EQ4_B4_A - [15:0] */ + +/* + * R3680 (0xE60) - EQ4_15 + */ +#define WM5100_EQ4_B4_B_MASK 0xFFFF /* EQ4_B4_B - [15:0] */ +#define WM5100_EQ4_B4_B_SHIFT 0 /* EQ4_B4_B - [15:0] */ +#define WM5100_EQ4_B4_B_WIDTH 16 /* EQ4_B4_B - [15:0] */ + +/* + * R3681 (0xE61) - EQ4_16 + */ +#define WM5100_EQ4_B4_C_MASK 0xFFFF /* EQ4_B4_C - [15:0] */ +#define WM5100_EQ4_B4_C_SHIFT 0 /* EQ4_B4_C - [15:0] */ +#define WM5100_EQ4_B4_C_WIDTH 16 /* EQ4_B4_C - [15:0] */ + +/* + * R3682 (0xE62) - EQ4_17 + */ +#define WM5100_EQ4_B4_PG_MASK 0xFFFF /* EQ4_B4_PG - [15:0] */ +#define WM5100_EQ4_B4_PG_SHIFT 0 /* EQ4_B4_PG - [15:0] */ +#define WM5100_EQ4_B4_PG_WIDTH 16 /* EQ4_B4_PG - [15:0] */ + +/* + * R3683 (0xE63) - EQ4_18 + */ +#define WM5100_EQ4_B5_A_MASK 0xFFFF /* EQ4_B5_A - [15:0] */ +#define WM5100_EQ4_B5_A_SHIFT 0 /* EQ4_B5_A - [15:0] */ +#define WM5100_EQ4_B5_A_WIDTH 16 /* EQ4_B5_A - [15:0] */ + +/* + * R3684 (0xE64) - EQ4_19 + */ +#define WM5100_EQ4_B5_B_MASK 0xFFFF /* EQ4_B5_B - [15:0] */ +#define WM5100_EQ4_B5_B_SHIFT 0 /* EQ4_B5_B - [15:0] */ +#define WM5100_EQ4_B5_B_WIDTH 16 /* EQ4_B5_B - [15:0] */ + +/* + * R3685 (0xE65) - EQ4_20 + */ +#define WM5100_EQ4_B5_PG_MASK 0xFFFF /* EQ4_B5_PG - [15:0] */ +#define WM5100_EQ4_B5_PG_SHIFT 0 /* EQ4_B5_PG - [15:0] */ +#define WM5100_EQ4_B5_PG_WIDTH 16 /* EQ4_B5_PG - [15:0] */ + +/* + * R3712 (0xE80) - DRC1 ctrl1 + */ +#define WM5100_DRC_SIG_DET_RMS_MASK 0xF800 /* DRC_SIG_DET_RMS - [15:11] */ +#define WM5100_DRC_SIG_DET_RMS_SHIFT 11 /* DRC_SIG_DET_RMS - [15:11] */ +#define WM5100_DRC_SIG_DET_RMS_WIDTH 5 /* DRC_SIG_DET_RMS - [15:11] */ +#define WM5100_DRC_SIG_DET_PK_MASK 0x0600 /* DRC_SIG_DET_PK - [10:9] */ +#define WM5100_DRC_SIG_DET_PK_SHIFT 9 /* DRC_SIG_DET_PK - [10:9] */ +#define WM5100_DRC_SIG_DET_PK_WIDTH 2 /* DRC_SIG_DET_PK - [10:9] */ +#define WM5100_DRC_NG_ENA 0x0100 /* DRC_NG_ENA */ +#define WM5100_DRC_NG_ENA_MASK 0x0100 /* DRC_NG_ENA */ +#define WM5100_DRC_NG_ENA_SHIFT 8 /* DRC_NG_ENA */ +#define WM5100_DRC_NG_ENA_WIDTH 1 /* DRC_NG_ENA */ +#define WM5100_DRC_SIG_DET_MODE 0x0080 /* DRC_SIG_DET_MODE */ +#define WM5100_DRC_SIG_DET_MODE_MASK 0x0080 /* DRC_SIG_DET_MODE */ +#define WM5100_DRC_SIG_DET_MODE_SHIFT 7 /* DRC_SIG_DET_MODE */ +#define WM5100_DRC_SIG_DET_MODE_WIDTH 1 /* DRC_SIG_DET_MODE */ +#define WM5100_DRC_SIG_DET 0x0040 /* DRC_SIG_DET */ +#define WM5100_DRC_SIG_DET_MASK 0x0040 /* DRC_SIG_DET */ +#define WM5100_DRC_SIG_DET_SHIFT 6 /* DRC_SIG_DET */ +#define WM5100_DRC_SIG_DET_WIDTH 1 /* DRC_SIG_DET */ +#define WM5100_DRC_KNEE2_OP_ENA 0x0020 /* DRC_KNEE2_OP_ENA */ +#define WM5100_DRC_KNEE2_OP_ENA_MASK 0x0020 /* DRC_KNEE2_OP_ENA */ +#define WM5100_DRC_KNEE2_OP_ENA_SHIFT 5 /* DRC_KNEE2_OP_ENA */ +#define WM5100_DRC_KNEE2_OP_ENA_WIDTH 1 /* DRC_KNEE2_OP_ENA */ +#define WM5100_DRC_QR 0x0010 /* DRC_QR */ +#define WM5100_DRC_QR_MASK 0x0010 /* DRC_QR */ +#define WM5100_DRC_QR_SHIFT 4 /* DRC_QR */ +#define WM5100_DRC_QR_WIDTH 1 /* DRC_QR */ +#define WM5100_DRC_ANTICLIP 0x0008 /* DRC_ANTICLIP */ +#define WM5100_DRC_ANTICLIP_MASK 0x0008 /* DRC_ANTICLIP */ +#define WM5100_DRC_ANTICLIP_SHIFT 3 /* DRC_ANTICLIP */ +#define WM5100_DRC_ANTICLIP_WIDTH 1 /* DRC_ANTICLIP */ +#define WM5100_DRCL_ENA 0x0002 /* DRCL_ENA */ +#define WM5100_DRCL_ENA_MASK 0x0002 /* DRCL_ENA */ +#define WM5100_DRCL_ENA_SHIFT 1 /* DRCL_ENA */ +#define WM5100_DRCL_ENA_WIDTH 1 /* DRCL_ENA */ +#define WM5100_DRCR_ENA 0x0001 /* DRCR_ENA */ +#define WM5100_DRCR_ENA_MASK 0x0001 /* DRCR_ENA */ +#define WM5100_DRCR_ENA_SHIFT 0 /* DRCR_ENA */ +#define WM5100_DRCR_ENA_WIDTH 1 /* DRCR_ENA */ + +/* + * R3713 (0xE81) - DRC1 ctrl2 + */ +#define WM5100_DRC_ATK_MASK 0x1E00 /* DRC_ATK - [12:9] */ +#define WM5100_DRC_ATK_SHIFT 9 /* DRC_ATK - [12:9] */ +#define WM5100_DRC_ATK_WIDTH 4 /* DRC_ATK - [12:9] */ +#define WM5100_DRC_DCY_MASK 0x01E0 /* DRC_DCY - [8:5] */ +#define WM5100_DRC_DCY_SHIFT 5 /* DRC_DCY - [8:5] */ +#define WM5100_DRC_DCY_WIDTH 4 /* DRC_DCY - [8:5] */ +#define WM5100_DRC_MINGAIN_MASK 0x001C /* DRC_MINGAIN - [4:2] */ +#define WM5100_DRC_MINGAIN_SHIFT 2 /* DRC_MINGAIN - [4:2] */ +#define WM5100_DRC_MINGAIN_WIDTH 3 /* DRC_MINGAIN - [4:2] */ +#define WM5100_DRC_MAXGAIN_MASK 0x0003 /* DRC_MAXGAIN - [1:0] */ +#define WM5100_DRC_MAXGAIN_SHIFT 0 /* DRC_MAXGAIN - [1:0] */ +#define WM5100_DRC_MAXGAIN_WIDTH 2 /* DRC_MAXGAIN - [1:0] */ + +/* + * R3714 (0xE82) - DRC1 ctrl3 + */ +#define WM5100_DRC_NG_MINGAIN_MASK 0xF000 /* DRC_NG_MINGAIN - [15:12] */ +#define WM5100_DRC_NG_MINGAIN_SHIFT 12 /* DRC_NG_MINGAIN - [15:12] */ +#define WM5100_DRC_NG_MINGAIN_WIDTH 4 /* DRC_NG_MINGAIN - [15:12] */ +#define WM5100_DRC_NG_EXP_MASK 0x0C00 /* DRC_NG_EXP - [11:10] */ +#define WM5100_DRC_NG_EXP_SHIFT 10 /* DRC_NG_EXP - [11:10] */ +#define WM5100_DRC_NG_EXP_WIDTH 2 /* DRC_NG_EXP - [11:10] */ +#define WM5100_DRC_QR_THR_MASK 0x0300 /* DRC_QR_THR - [9:8] */ +#define WM5100_DRC_QR_THR_SHIFT 8 /* DRC_QR_THR - [9:8] */ +#define WM5100_DRC_QR_THR_WIDTH 2 /* DRC_QR_THR - [9:8] */ +#define WM5100_DRC_QR_DCY_MASK 0x00C0 /* DRC_QR_DCY - [7:6] */ +#define WM5100_DRC_QR_DCY_SHIFT 6 /* DRC_QR_DCY - [7:6] */ +#define WM5100_DRC_QR_DCY_WIDTH 2 /* DRC_QR_DCY - [7:6] */ +#define WM5100_DRC_HI_COMP_MASK 0x0038 /* DRC_HI_COMP - [5:3] */ +#define WM5100_DRC_HI_COMP_SHIFT 3 /* DRC_HI_COMP - [5:3] */ +#define WM5100_DRC_HI_COMP_WIDTH 3 /* DRC_HI_COMP - [5:3] */ +#define WM5100_DRC_LO_COMP_MASK 0x0007 /* DRC_LO_COMP - [2:0] */ +#define WM5100_DRC_LO_COMP_SHIFT 0 /* DRC_LO_COMP - [2:0] */ +#define WM5100_DRC_LO_COMP_WIDTH 3 /* DRC_LO_COMP - [2:0] */ + +/* + * R3715 (0xE83) - DRC1 ctrl4 + */ +#define WM5100_DRC_KNEE_IP_MASK 0x07E0 /* DRC_KNEE_IP - [10:5] */ +#define WM5100_DRC_KNEE_IP_SHIFT 5 /* DRC_KNEE_IP - [10:5] */ +#define WM5100_DRC_KNEE_IP_WIDTH 6 /* DRC_KNEE_IP - [10:5] */ +#define WM5100_DRC_KNEE_OP_MASK 0x001F /* DRC_KNEE_OP - [4:0] */ +#define WM5100_DRC_KNEE_OP_SHIFT 0 /* DRC_KNEE_OP - [4:0] */ +#define WM5100_DRC_KNEE_OP_WIDTH 5 /* DRC_KNEE_OP - [4:0] */ + +/* + * R3716 (0xE84) - DRC1 ctrl5 + */ +#define WM5100_DRC_KNEE2_IP_MASK 0x03E0 /* DRC_KNEE2_IP - [9:5] */ +#define WM5100_DRC_KNEE2_IP_SHIFT 5 /* DRC_KNEE2_IP - [9:5] */ +#define WM5100_DRC_KNEE2_IP_WIDTH 5 /* DRC_KNEE2_IP - [9:5] */ +#define WM5100_DRC_KNEE2_OP_MASK 0x001F /* DRC_KNEE2_OP - [4:0] */ +#define WM5100_DRC_KNEE2_OP_SHIFT 0 /* DRC_KNEE2_OP - [4:0] */ +#define WM5100_DRC_KNEE2_OP_WIDTH 5 /* DRC_KNEE2_OP - [4:0] */ + +/* + * R3776 (0xEC0) - HPLPF1_1 + */ +#define WM5100_LHPF1_MODE 0x0002 /* LHPF1_MODE */ +#define WM5100_LHPF1_MODE_MASK 0x0002 /* LHPF1_MODE */ +#define WM5100_LHPF1_MODE_SHIFT 1 /* LHPF1_MODE */ +#define WM5100_LHPF1_MODE_WIDTH 1 /* LHPF1_MODE */ +#define WM5100_LHPF1_ENA 0x0001 /* LHPF1_ENA */ +#define WM5100_LHPF1_ENA_MASK 0x0001 /* LHPF1_ENA */ +#define WM5100_LHPF1_ENA_SHIFT 0 /* LHPF1_ENA */ +#define WM5100_LHPF1_ENA_WIDTH 1 /* LHPF1_ENA */ + +/* + * R3777 (0xEC1) - HPLPF1_2 + */ +#define WM5100_LHPF1_COEFF_MASK 0xFFFF /* LHPF1_COEFF - [15:0] */ +#define WM5100_LHPF1_COEFF_SHIFT 0 /* LHPF1_COEFF - [15:0] */ +#define WM5100_LHPF1_COEFF_WIDTH 16 /* LHPF1_COEFF - [15:0] */ + +/* + * R3780 (0xEC4) - HPLPF2_1 + */ +#define WM5100_LHPF2_MODE 0x0002 /* LHPF2_MODE */ +#define WM5100_LHPF2_MODE_MASK 0x0002 /* LHPF2_MODE */ +#define WM5100_LHPF2_MODE_SHIFT 1 /* LHPF2_MODE */ +#define WM5100_LHPF2_MODE_WIDTH 1 /* LHPF2_MODE */ +#define WM5100_LHPF2_ENA 0x0001 /* LHPF2_ENA */ +#define WM5100_LHPF2_ENA_MASK 0x0001 /* LHPF2_ENA */ +#define WM5100_LHPF2_ENA_SHIFT 0 /* LHPF2_ENA */ +#define WM5100_LHPF2_ENA_WIDTH 1 /* LHPF2_ENA */ + +/* + * R3781 (0xEC5) - HPLPF2_2 + */ +#define WM5100_LHPF2_COEFF_MASK 0xFFFF /* LHPF2_COEFF - [15:0] */ +#define WM5100_LHPF2_COEFF_SHIFT 0 /* LHPF2_COEFF - [15:0] */ +#define WM5100_LHPF2_COEFF_WIDTH 16 /* LHPF2_COEFF - [15:0] */ + +/* + * R3784 (0xEC8) - HPLPF3_1 + */ +#define WM5100_LHPF3_MODE 0x0002 /* LHPF3_MODE */ +#define WM5100_LHPF3_MODE_MASK 0x0002 /* LHPF3_MODE */ +#define WM5100_LHPF3_MODE_SHIFT 1 /* LHPF3_MODE */ +#define WM5100_LHPF3_MODE_WIDTH 1 /* LHPF3_MODE */ +#define WM5100_LHPF3_ENA 0x0001 /* LHPF3_ENA */ +#define WM5100_LHPF3_ENA_MASK 0x0001 /* LHPF3_ENA */ +#define WM5100_LHPF3_ENA_SHIFT 0 /* LHPF3_ENA */ +#define WM5100_LHPF3_ENA_WIDTH 1 /* LHPF3_ENA */ + +/* + * R3785 (0xEC9) - HPLPF3_2 + */ +#define WM5100_LHPF3_COEFF_MASK 0xFFFF /* LHPF3_COEFF - [15:0] */ +#define WM5100_LHPF3_COEFF_SHIFT 0 /* LHPF3_COEFF - [15:0] */ +#define WM5100_LHPF3_COEFF_WIDTH 16 /* LHPF3_COEFF - [15:0] */ + +/* + * R3788 (0xECC) - HPLPF4_1 + */ +#define WM5100_LHPF4_MODE 0x0002 /* LHPF4_MODE */ +#define WM5100_LHPF4_MODE_MASK 0x0002 /* LHPF4_MODE */ +#define WM5100_LHPF4_MODE_SHIFT 1 /* LHPF4_MODE */ +#define WM5100_LHPF4_MODE_WIDTH 1 /* LHPF4_MODE */ +#define WM5100_LHPF4_ENA 0x0001 /* LHPF4_ENA */ +#define WM5100_LHPF4_ENA_MASK 0x0001 /* LHPF4_ENA */ +#define WM5100_LHPF4_ENA_SHIFT 0 /* LHPF4_ENA */ +#define WM5100_LHPF4_ENA_WIDTH 1 /* LHPF4_ENA */ + +/* + * R3789 (0xECD) - HPLPF4_2 + */ +#define WM5100_LHPF4_COEFF_MASK 0xFFFF /* LHPF4_COEFF - [15:0] */ +#define WM5100_LHPF4_COEFF_SHIFT 0 /* LHPF4_COEFF - [15:0] */ +#define WM5100_LHPF4_COEFF_WIDTH 16 /* LHPF4_COEFF - [15:0] */ + +/* + * R16384 (0x4000) - DSP1 DM 0 + */ +#define WM5100_DSP1_DM_START_1_MASK 0x00FF /* DSP1_DM_START - [7:0] */ +#define WM5100_DSP1_DM_START_1_SHIFT 0 /* DSP1_DM_START - [7:0] */ +#define WM5100_DSP1_DM_START_1_WIDTH 8 /* DSP1_DM_START - [7:0] */ + +/* + * R16385 (0x4001) - DSP1 DM 1 + */ +#define WM5100_DSP1_DM_START_MASK 0xFFFF /* DSP1_DM_START - [15:0] */ +#define WM5100_DSP1_DM_START_SHIFT 0 /* DSP1_DM_START - [15:0] */ +#define WM5100_DSP1_DM_START_WIDTH 16 /* DSP1_DM_START - [15:0] */ + +/* + * R16386 (0x4002) - DSP1 DM 2 + */ +#define WM5100_DSP1_DM_1_1_MASK 0x00FF /* DSP1_DM_1 - [7:0] */ +#define WM5100_DSP1_DM_1_1_SHIFT 0 /* DSP1_DM_1 - [7:0] */ +#define WM5100_DSP1_DM_1_1_WIDTH 8 /* DSP1_DM_1 - [7:0] */ + +/* + * R16387 (0x4003) - DSP1 DM 3 + */ +#define WM5100_DSP1_DM_1_MASK 0xFFFF /* DSP1_DM_1 - [15:0] */ +#define WM5100_DSP1_DM_1_SHIFT 0 /* DSP1_DM_1 - [15:0] */ +#define WM5100_DSP1_DM_1_WIDTH 16 /* DSP1_DM_1 - [15:0] */ + +/* + * R16892 (0x41FC) - DSP1 DM 508 + */ +#define WM5100_DSP1_DM_254_1_MASK 0x00FF /* DSP1_DM_254 - [7:0] */ +#define WM5100_DSP1_DM_254_1_SHIFT 0 /* DSP1_DM_254 - [7:0] */ +#define WM5100_DSP1_DM_254_1_WIDTH 8 /* DSP1_DM_254 - [7:0] */ + +/* + * R16893 (0x41FD) - DSP1 DM 509 + */ +#define WM5100_DSP1_DM_254_MASK 0xFFFF /* DSP1_DM_254 - [15:0] */ +#define WM5100_DSP1_DM_254_SHIFT 0 /* DSP1_DM_254 - [15:0] */ +#define WM5100_DSP1_DM_254_WIDTH 16 /* DSP1_DM_254 - [15:0] */ + +/* + * R16894 (0x41FE) - DSP1 DM 510 + */ +#define WM5100_DSP1_DM_END_1_MASK 0x00FF /* DSP1_DM_END - [7:0] */ +#define WM5100_DSP1_DM_END_1_SHIFT 0 /* DSP1_DM_END - [7:0] */ +#define WM5100_DSP1_DM_END_1_WIDTH 8 /* DSP1_DM_END - [7:0] */ + +/* + * R16895 (0x41FF) - DSP1 DM 511 + */ +#define WM5100_DSP1_DM_END_MASK 0xFFFF /* DSP1_DM_END - [15:0] */ +#define WM5100_DSP1_DM_END_SHIFT 0 /* DSP1_DM_END - [15:0] */ +#define WM5100_DSP1_DM_END_WIDTH 16 /* DSP1_DM_END - [15:0] */ + +/* + * R18432 (0x4800) - DSP1 PM 0 + */ +#define WM5100_DSP1_PM_START_2_MASK 0x00FF /* DSP1_PM_START - [7:0] */ +#define WM5100_DSP1_PM_START_2_SHIFT 0 /* DSP1_PM_START - [7:0] */ +#define WM5100_DSP1_PM_START_2_WIDTH 8 /* DSP1_PM_START - [7:0] */ + +/* + * R18433 (0x4801) - DSP1 PM 1 + */ +#define WM5100_DSP1_PM_START_1_MASK 0xFFFF /* DSP1_PM_START - [15:0] */ +#define WM5100_DSP1_PM_START_1_SHIFT 0 /* DSP1_PM_START - [15:0] */ +#define WM5100_DSP1_PM_START_1_WIDTH 16 /* DSP1_PM_START - [15:0] */ + +/* + * R18434 (0x4802) - DSP1 PM 2 + */ +#define WM5100_DSP1_PM_START_MASK 0xFFFF /* DSP1_PM_START - [15:0] */ +#define WM5100_DSP1_PM_START_SHIFT 0 /* DSP1_PM_START - [15:0] */ +#define WM5100_DSP1_PM_START_WIDTH 16 /* DSP1_PM_START - [15:0] */ + +/* + * R18435 (0x4803) - DSP1 PM 3 + */ +#define WM5100_DSP1_PM_1_2_MASK 0x00FF /* DSP1_PM_1 - [7:0] */ +#define WM5100_DSP1_PM_1_2_SHIFT 0 /* DSP1_PM_1 - [7:0] */ +#define WM5100_DSP1_PM_1_2_WIDTH 8 /* DSP1_PM_1 - [7:0] */ + +/* + * R18436 (0x4804) - DSP1 PM 4 + */ +#define WM5100_DSP1_PM_1_1_MASK 0xFFFF /* DSP1_PM_1 - [15:0] */ +#define WM5100_DSP1_PM_1_1_SHIFT 0 /* DSP1_PM_1 - [15:0] */ +#define WM5100_DSP1_PM_1_1_WIDTH 16 /* DSP1_PM_1 - [15:0] */ + +/* + * R18437 (0x4805) - DSP1 PM 5 + */ +#define WM5100_DSP1_PM_1_MASK 0xFFFF /* DSP1_PM_1 - [15:0] */ +#define WM5100_DSP1_PM_1_SHIFT 0 /* DSP1_PM_1 - [15:0] */ +#define WM5100_DSP1_PM_1_WIDTH 16 /* DSP1_PM_1 - [15:0] */ + +/* + * R19962 (0x4DFA) - DSP1 PM 1530 + */ +#define WM5100_DSP1_PM_510_2_MASK 0x00FF /* DSP1_PM_510 - [7:0] */ +#define WM5100_DSP1_PM_510_2_SHIFT 0 /* DSP1_PM_510 - [7:0] */ +#define WM5100_DSP1_PM_510_2_WIDTH 8 /* DSP1_PM_510 - [7:0] */ + +/* + * R19963 (0x4DFB) - DSP1 PM 1531 + */ +#define WM5100_DSP1_PM_510_1_MASK 0xFFFF /* DSP1_PM_510 - [15:0] */ +#define WM5100_DSP1_PM_510_1_SHIFT 0 /* DSP1_PM_510 - [15:0] */ +#define WM5100_DSP1_PM_510_1_WIDTH 16 /* DSP1_PM_510 - [15:0] */ + +/* + * R19964 (0x4DFC) - DSP1 PM 1532 + */ +#define WM5100_DSP1_PM_510_MASK 0xFFFF /* DSP1_PM_510 - [15:0] */ +#define WM5100_DSP1_PM_510_SHIFT 0 /* DSP1_PM_510 - [15:0] */ +#define WM5100_DSP1_PM_510_WIDTH 16 /* DSP1_PM_510 - [15:0] */ + +/* + * R19965 (0x4DFD) - DSP1 PM 1533 + */ +#define WM5100_DSP1_PM_END_2_MASK 0x00FF /* DSP1_PM_END - [7:0] */ +#define WM5100_DSP1_PM_END_2_SHIFT 0 /* DSP1_PM_END - [7:0] */ +#define WM5100_DSP1_PM_END_2_WIDTH 8 /* DSP1_PM_END - [7:0] */ + +/* + * R19966 (0x4DFE) - DSP1 PM 1534 + */ +#define WM5100_DSP1_PM_END_1_MASK 0xFFFF /* DSP1_PM_END - [15:0] */ +#define WM5100_DSP1_PM_END_1_SHIFT 0 /* DSP1_PM_END - [15:0] */ +#define WM5100_DSP1_PM_END_1_WIDTH 16 /* DSP1_PM_END - [15:0] */ + +/* + * R19967 (0x4DFF) - DSP1 PM 1535 + */ +#define WM5100_DSP1_PM_END_MASK 0xFFFF /* DSP1_PM_END - [15:0] */ +#define WM5100_DSP1_PM_END_SHIFT 0 /* DSP1_PM_END - [15:0] */ +#define WM5100_DSP1_PM_END_WIDTH 16 /* DSP1_PM_END - [15:0] */ + +/* + * R20480 (0x5000) - DSP1 ZM 0 + */ +#define WM5100_DSP1_ZM_START_1_MASK 0x00FF /* DSP1_ZM_START - [7:0] */ +#define WM5100_DSP1_ZM_START_1_SHIFT 0 /* DSP1_ZM_START - [7:0] */ +#define WM5100_DSP1_ZM_START_1_WIDTH 8 /* DSP1_ZM_START - [7:0] */ + +/* + * R20481 (0x5001) - DSP1 ZM 1 + */ +#define WM5100_DSP1_ZM_START_MASK 0xFFFF /* DSP1_ZM_START - [15:0] */ +#define WM5100_DSP1_ZM_START_SHIFT 0 /* DSP1_ZM_START - [15:0] */ +#define WM5100_DSP1_ZM_START_WIDTH 16 /* DSP1_ZM_START - [15:0] */ + +/* + * R20482 (0x5002) - DSP1 ZM 2 + */ +#define WM5100_DSP1_ZM_1_1_MASK 0x00FF /* DSP1_ZM_1 - [7:0] */ +#define WM5100_DSP1_ZM_1_1_SHIFT 0 /* DSP1_ZM_1 - [7:0] */ +#define WM5100_DSP1_ZM_1_1_WIDTH 8 /* DSP1_ZM_1 - [7:0] */ + +/* + * R20483 (0x5003) - DSP1 ZM 3 + */ +#define WM5100_DSP1_ZM_1_MASK 0xFFFF /* DSP1_ZM_1 - [15:0] */ +#define WM5100_DSP1_ZM_1_SHIFT 0 /* DSP1_ZM_1 - [15:0] */ +#define WM5100_DSP1_ZM_1_WIDTH 16 /* DSP1_ZM_1 - [15:0] */ + +/* + * R22524 (0x57FC) - DSP1 ZM 2044 + */ +#define WM5100_DSP1_ZM_1022_1_MASK 0x00FF /* DSP1_ZM_1022 - [7:0] */ +#define WM5100_DSP1_ZM_1022_1_SHIFT 0 /* DSP1_ZM_1022 - [7:0] */ +#define WM5100_DSP1_ZM_1022_1_WIDTH 8 /* DSP1_ZM_1022 - [7:0] */ + +/* + * R22525 (0x57FD) - DSP1 ZM 2045 + */ +#define WM5100_DSP1_ZM_1022_MASK 0xFFFF /* DSP1_ZM_1022 - [15:0] */ +#define WM5100_DSP1_ZM_1022_SHIFT 0 /* DSP1_ZM_1022 - [15:0] */ +#define WM5100_DSP1_ZM_1022_WIDTH 16 /* DSP1_ZM_1022 - [15:0] */ + +/* + * R22526 (0x57FE) - DSP1 ZM 2046 + */ +#define WM5100_DSP1_ZM_END_1_MASK 0x00FF /* DSP1_ZM_END - [7:0] */ +#define WM5100_DSP1_ZM_END_1_SHIFT 0 /* DSP1_ZM_END - [7:0] */ +#define WM5100_DSP1_ZM_END_1_WIDTH 8 /* DSP1_ZM_END - [7:0] */ + +/* + * R22527 (0x57FF) - DSP1 ZM 2047 + */ +#define WM5100_DSP1_ZM_END_MASK 0xFFFF /* DSP1_ZM_END - [15:0] */ +#define WM5100_DSP1_ZM_END_SHIFT 0 /* DSP1_ZM_END - [15:0] */ +#define WM5100_DSP1_ZM_END_WIDTH 16 /* DSP1_ZM_END - [15:0] */ + +/* + * R24576 (0x6000) - DSP2 DM 0 + */ +#define WM5100_DSP2_DM_START_1_MASK 0x00FF /* DSP2_DM_START - [7:0] */ +#define WM5100_DSP2_DM_START_1_SHIFT 0 /* DSP2_DM_START - [7:0] */ +#define WM5100_DSP2_DM_START_1_WIDTH 8 /* DSP2_DM_START - [7:0] */ + +/* + * R24577 (0x6001) - DSP2 DM 1 + */ +#define WM5100_DSP2_DM_START_MASK 0xFFFF /* DSP2_DM_START - [15:0] */ +#define WM5100_DSP2_DM_START_SHIFT 0 /* DSP2_DM_START - [15:0] */ +#define WM5100_DSP2_DM_START_WIDTH 16 /* DSP2_DM_START - [15:0] */ + +/* + * R24578 (0x6002) - DSP2 DM 2 + */ +#define WM5100_DSP2_DM_1_1_MASK 0x00FF /* DSP2_DM_1 - [7:0] */ +#define WM5100_DSP2_DM_1_1_SHIFT 0 /* DSP2_DM_1 - [7:0] */ +#define WM5100_DSP2_DM_1_1_WIDTH 8 /* DSP2_DM_1 - [7:0] */ + +/* + * R24579 (0x6003) - DSP2 DM 3 + */ +#define WM5100_DSP2_DM_1_MASK 0xFFFF /* DSP2_DM_1 - [15:0] */ +#define WM5100_DSP2_DM_1_SHIFT 0 /* DSP2_DM_1 - [15:0] */ +#define WM5100_DSP2_DM_1_WIDTH 16 /* DSP2_DM_1 - [15:0] */ + +/* + * R25084 (0x61FC) - DSP2 DM 508 + */ +#define WM5100_DSP2_DM_254_1_MASK 0x00FF /* DSP2_DM_254 - [7:0] */ +#define WM5100_DSP2_DM_254_1_SHIFT 0 /* DSP2_DM_254 - [7:0] */ +#define WM5100_DSP2_DM_254_1_WIDTH 8 /* DSP2_DM_254 - [7:0] */ + +/* + * R25085 (0x61FD) - DSP2 DM 509 + */ +#define WM5100_DSP2_DM_254_MASK 0xFFFF /* DSP2_DM_254 - [15:0] */ +#define WM5100_DSP2_DM_254_SHIFT 0 /* DSP2_DM_254 - [15:0] */ +#define WM5100_DSP2_DM_254_WIDTH 16 /* DSP2_DM_254 - [15:0] */ + +/* + * R25086 (0x61FE) - DSP2 DM 510 + */ +#define WM5100_DSP2_DM_END_1_MASK 0x00FF /* DSP2_DM_END - [7:0] */ +#define WM5100_DSP2_DM_END_1_SHIFT 0 /* DSP2_DM_END - [7:0] */ +#define WM5100_DSP2_DM_END_1_WIDTH 8 /* DSP2_DM_END - [7:0] */ + +/* + * R25087 (0x61FF) - DSP2 DM 511 + */ +#define WM5100_DSP2_DM_END_MASK 0xFFFF /* DSP2_DM_END - [15:0] */ +#define WM5100_DSP2_DM_END_SHIFT 0 /* DSP2_DM_END - [15:0] */ +#define WM5100_DSP2_DM_END_WIDTH 16 /* DSP2_DM_END - [15:0] */ + +/* + * R26624 (0x6800) - DSP2 PM 0 + */ +#define WM5100_DSP2_PM_START_2_MASK 0x00FF /* DSP2_PM_START - [7:0] */ +#define WM5100_DSP2_PM_START_2_SHIFT 0 /* DSP2_PM_START - [7:0] */ +#define WM5100_DSP2_PM_START_2_WIDTH 8 /* DSP2_PM_START - [7:0] */ + +/* + * R26625 (0x6801) - DSP2 PM 1 + */ +#define WM5100_DSP2_PM_START_1_MASK 0xFFFF /* DSP2_PM_START - [15:0] */ +#define WM5100_DSP2_PM_START_1_SHIFT 0 /* DSP2_PM_START - [15:0] */ +#define WM5100_DSP2_PM_START_1_WIDTH 16 /* DSP2_PM_START - [15:0] */ + +/* + * R26626 (0x6802) - DSP2 PM 2 + */ +#define WM5100_DSP2_PM_START_MASK 0xFFFF /* DSP2_PM_START - [15:0] */ +#define WM5100_DSP2_PM_START_SHIFT 0 /* DSP2_PM_START - [15:0] */ +#define WM5100_DSP2_PM_START_WIDTH 16 /* DSP2_PM_START - [15:0] */ + +/* + * R26627 (0x6803) - DSP2 PM 3 + */ +#define WM5100_DSP2_PM_1_2_MASK 0x00FF /* DSP2_PM_1 - [7:0] */ +#define WM5100_DSP2_PM_1_2_SHIFT 0 /* DSP2_PM_1 - [7:0] */ +#define WM5100_DSP2_PM_1_2_WIDTH 8 /* DSP2_PM_1 - [7:0] */ + +/* + * R26628 (0x6804) - DSP2 PM 4 + */ +#define WM5100_DSP2_PM_1_1_MASK 0xFFFF /* DSP2_PM_1 - [15:0] */ +#define WM5100_DSP2_PM_1_1_SHIFT 0 /* DSP2_PM_1 - [15:0] */ +#define WM5100_DSP2_PM_1_1_WIDTH 16 /* DSP2_PM_1 - [15:0] */ + +/* + * R26629 (0x6805) - DSP2 PM 5 + */ +#define WM5100_DSP2_PM_1_MASK 0xFFFF /* DSP2_PM_1 - [15:0] */ +#define WM5100_DSP2_PM_1_SHIFT 0 /* DSP2_PM_1 - [15:0] */ +#define WM5100_DSP2_PM_1_WIDTH 16 /* DSP2_PM_1 - [15:0] */ + +/* + * R28154 (0x6DFA) - DSP2 PM 1530 + */ +#define WM5100_DSP2_PM_510_2_MASK 0x00FF /* DSP2_PM_510 - [7:0] */ +#define WM5100_DSP2_PM_510_2_SHIFT 0 /* DSP2_PM_510 - [7:0] */ +#define WM5100_DSP2_PM_510_2_WIDTH 8 /* DSP2_PM_510 - [7:0] */ + +/* + * R28155 (0x6DFB) - DSP2 PM 1531 + */ +#define WM5100_DSP2_PM_510_1_MASK 0xFFFF /* DSP2_PM_510 - [15:0] */ +#define WM5100_DSP2_PM_510_1_SHIFT 0 /* DSP2_PM_510 - [15:0] */ +#define WM5100_DSP2_PM_510_1_WIDTH 16 /* DSP2_PM_510 - [15:0] */ + +/* + * R28156 (0x6DFC) - DSP2 PM 1532 + */ +#define WM5100_DSP2_PM_510_MASK 0xFFFF /* DSP2_PM_510 - [15:0] */ +#define WM5100_DSP2_PM_510_SHIFT 0 /* DSP2_PM_510 - [15:0] */ +#define WM5100_DSP2_PM_510_WIDTH 16 /* DSP2_PM_510 - [15:0] */ + +/* + * R28157 (0x6DFD) - DSP2 PM 1533 + */ +#define WM5100_DSP2_PM_END_2_MASK 0x00FF /* DSP2_PM_END - [7:0] */ +#define WM5100_DSP2_PM_END_2_SHIFT 0 /* DSP2_PM_END - [7:0] */ +#define WM5100_DSP2_PM_END_2_WIDTH 8 /* DSP2_PM_END - [7:0] */ + +/* + * R28158 (0x6DFE) - DSP2 PM 1534 + */ +#define WM5100_DSP2_PM_END_1_MASK 0xFFFF /* DSP2_PM_END - [15:0] */ +#define WM5100_DSP2_PM_END_1_SHIFT 0 /* DSP2_PM_END - [15:0] */ +#define WM5100_DSP2_PM_END_1_WIDTH 16 /* DSP2_PM_END - [15:0] */ + +/* + * R28159 (0x6DFF) - DSP2 PM 1535 + */ +#define WM5100_DSP2_PM_END_MASK 0xFFFF /* DSP2_PM_END - [15:0] */ +#define WM5100_DSP2_PM_END_SHIFT 0 /* DSP2_PM_END - [15:0] */ +#define WM5100_DSP2_PM_END_WIDTH 16 /* DSP2_PM_END - [15:0] */ + +/* + * R28672 (0x7000) - DSP2 ZM 0 + */ +#define WM5100_DSP2_ZM_START_1_MASK 0x00FF /* DSP2_ZM_START - [7:0] */ +#define WM5100_DSP2_ZM_START_1_SHIFT 0 /* DSP2_ZM_START - [7:0] */ +#define WM5100_DSP2_ZM_START_1_WIDTH 8 /* DSP2_ZM_START - [7:0] */ + +/* + * R28673 (0x7001) - DSP2 ZM 1 + */ +#define WM5100_DSP2_ZM_START_MASK 0xFFFF /* DSP2_ZM_START - [15:0] */ +#define WM5100_DSP2_ZM_START_SHIFT 0 /* DSP2_ZM_START - [15:0] */ +#define WM5100_DSP2_ZM_START_WIDTH 16 /* DSP2_ZM_START - [15:0] */ + +/* + * R28674 (0x7002) - DSP2 ZM 2 + */ +#define WM5100_DSP2_ZM_1_1_MASK 0x00FF /* DSP2_ZM_1 - [7:0] */ +#define WM5100_DSP2_ZM_1_1_SHIFT 0 /* DSP2_ZM_1 - [7:0] */ +#define WM5100_DSP2_ZM_1_1_WIDTH 8 /* DSP2_ZM_1 - [7:0] */ + +/* + * R28675 (0x7003) - DSP2 ZM 3 + */ +#define WM5100_DSP2_ZM_1_MASK 0xFFFF /* DSP2_ZM_1 - [15:0] */ +#define WM5100_DSP2_ZM_1_SHIFT 0 /* DSP2_ZM_1 - [15:0] */ +#define WM5100_DSP2_ZM_1_WIDTH 16 /* DSP2_ZM_1 - [15:0] */ + +/* + * R30716 (0x77FC) - DSP2 ZM 2044 + */ +#define WM5100_DSP2_ZM_1022_1_MASK 0x00FF /* DSP2_ZM_1022 - [7:0] */ +#define WM5100_DSP2_ZM_1022_1_SHIFT 0 /* DSP2_ZM_1022 - [7:0] */ +#define WM5100_DSP2_ZM_1022_1_WIDTH 8 /* DSP2_ZM_1022 - [7:0] */ + +/* + * R30717 (0x77FD) - DSP2 ZM 2045 + */ +#define WM5100_DSP2_ZM_1022_MASK 0xFFFF /* DSP2_ZM_1022 - [15:0] */ +#define WM5100_DSP2_ZM_1022_SHIFT 0 /* DSP2_ZM_1022 - [15:0] */ +#define WM5100_DSP2_ZM_1022_WIDTH 16 /* DSP2_ZM_1022 - [15:0] */ + +/* + * R30718 (0x77FE) - DSP2 ZM 2046 + */ +#define WM5100_DSP2_ZM_END_1_MASK 0x00FF /* DSP2_ZM_END - [7:0] */ +#define WM5100_DSP2_ZM_END_1_SHIFT 0 /* DSP2_ZM_END - [7:0] */ +#define WM5100_DSP2_ZM_END_1_WIDTH 8 /* DSP2_ZM_END - [7:0] */ + +/* + * R30719 (0x77FF) - DSP2 ZM 2047 + */ +#define WM5100_DSP2_ZM_END_MASK 0xFFFF /* DSP2_ZM_END - [15:0] */ +#define WM5100_DSP2_ZM_END_SHIFT 0 /* DSP2_ZM_END - [15:0] */ +#define WM5100_DSP2_ZM_END_WIDTH 16 /* DSP2_ZM_END - [15:0] */ + +/* + * R32768 (0x8000) - DSP3 DM 0 + */ +#define WM5100_DSP3_DM_START_1_MASK 0x00FF /* DSP3_DM_START - [7:0] */ +#define WM5100_DSP3_DM_START_1_SHIFT 0 /* DSP3_DM_START - [7:0] */ +#define WM5100_DSP3_DM_START_1_WIDTH 8 /* DSP3_DM_START - [7:0] */ + +/* + * R32769 (0x8001) - DSP3 DM 1 + */ +#define WM5100_DSP3_DM_START_MASK 0xFFFF /* DSP3_DM_START - [15:0] */ +#define WM5100_DSP3_DM_START_SHIFT 0 /* DSP3_DM_START - [15:0] */ +#define WM5100_DSP3_DM_START_WIDTH 16 /* DSP3_DM_START - [15:0] */ + +/* + * R32770 (0x8002) - DSP3 DM 2 + */ +#define WM5100_DSP3_DM_1_1_MASK 0x00FF /* DSP3_DM_1 - [7:0] */ +#define WM5100_DSP3_DM_1_1_SHIFT 0 /* DSP3_DM_1 - [7:0] */ +#define WM5100_DSP3_DM_1_1_WIDTH 8 /* DSP3_DM_1 - [7:0] */ + +/* + * R32771 (0x8003) - DSP3 DM 3 + */ +#define WM5100_DSP3_DM_1_MASK 0xFFFF /* DSP3_DM_1 - [15:0] */ +#define WM5100_DSP3_DM_1_SHIFT 0 /* DSP3_DM_1 - [15:0] */ +#define WM5100_DSP3_DM_1_WIDTH 16 /* DSP3_DM_1 - [15:0] */ + +/* + * R33276 (0x81FC) - DSP3 DM 508 + */ +#define WM5100_DSP3_DM_254_1_MASK 0x00FF /* DSP3_DM_254 - [7:0] */ +#define WM5100_DSP3_DM_254_1_SHIFT 0 /* DSP3_DM_254 - [7:0] */ +#define WM5100_DSP3_DM_254_1_WIDTH 8 /* DSP3_DM_254 - [7:0] */ + +/* + * R33277 (0x81FD) - DSP3 DM 509 + */ +#define WM5100_DSP3_DM_254_MASK 0xFFFF /* DSP3_DM_254 - [15:0] */ +#define WM5100_DSP3_DM_254_SHIFT 0 /* DSP3_DM_254 - [15:0] */ +#define WM5100_DSP3_DM_254_WIDTH 16 /* DSP3_DM_254 - [15:0] */ + +/* + * R33278 (0x81FE) - DSP3 DM 510 + */ +#define WM5100_DSP3_DM_END_1_MASK 0x00FF /* DSP3_DM_END - [7:0] */ +#define WM5100_DSP3_DM_END_1_SHIFT 0 /* DSP3_DM_END - [7:0] */ +#define WM5100_DSP3_DM_END_1_WIDTH 8 /* DSP3_DM_END - [7:0] */ + +/* + * R33279 (0x81FF) - DSP3 DM 511 + */ +#define WM5100_DSP3_DM_END_MASK 0xFFFF /* DSP3_DM_END - [15:0] */ +#define WM5100_DSP3_DM_END_SHIFT 0 /* DSP3_DM_END - [15:0] */ +#define WM5100_DSP3_DM_END_WIDTH 16 /* DSP3_DM_END - [15:0] */ + +/* + * R34816 (0x8800) - DSP3 PM 0 + */ +#define WM5100_DSP3_PM_START_2_MASK 0x00FF /* DSP3_PM_START - [7:0] */ +#define WM5100_DSP3_PM_START_2_SHIFT 0 /* DSP3_PM_START - [7:0] */ +#define WM5100_DSP3_PM_START_2_WIDTH 8 /* DSP3_PM_START - [7:0] */ + +/* + * R34817 (0x8801) - DSP3 PM 1 + */ +#define WM5100_DSP3_PM_START_1_MASK 0xFFFF /* DSP3_PM_START - [15:0] */ +#define WM5100_DSP3_PM_START_1_SHIFT 0 /* DSP3_PM_START - [15:0] */ +#define WM5100_DSP3_PM_START_1_WIDTH 16 /* DSP3_PM_START - [15:0] */ + +/* + * R34818 (0x8802) - DSP3 PM 2 + */ +#define WM5100_DSP3_PM_START_MASK 0xFFFF /* DSP3_PM_START - [15:0] */ +#define WM5100_DSP3_PM_START_SHIFT 0 /* DSP3_PM_START - [15:0] */ +#define WM5100_DSP3_PM_START_WIDTH 16 /* DSP3_PM_START - [15:0] */ + +/* + * R34819 (0x8803) - DSP3 PM 3 + */ +#define WM5100_DSP3_PM_1_2_MASK 0x00FF /* DSP3_PM_1 - [7:0] */ +#define WM5100_DSP3_PM_1_2_SHIFT 0 /* DSP3_PM_1 - [7:0] */ +#define WM5100_DSP3_PM_1_2_WIDTH 8 /* DSP3_PM_1 - [7:0] */ + +/* + * R34820 (0x8804) - DSP3 PM 4 + */ +#define WM5100_DSP3_PM_1_1_MASK 0xFFFF /* DSP3_PM_1 - [15:0] */ +#define WM5100_DSP3_PM_1_1_SHIFT 0 /* DSP3_PM_1 - [15:0] */ +#define WM5100_DSP3_PM_1_1_WIDTH 16 /* DSP3_PM_1 - [15:0] */ + +/* + * R34821 (0x8805) - DSP3 PM 5 + */ +#define WM5100_DSP3_PM_1_MASK 0xFFFF /* DSP3_PM_1 - [15:0] */ +#define WM5100_DSP3_PM_1_SHIFT 0 /* DSP3_PM_1 - [15:0] */ +#define WM5100_DSP3_PM_1_WIDTH 16 /* DSP3_PM_1 - [15:0] */ + +/* + * R36346 (0x8DFA) - DSP3 PM 1530 + */ +#define WM5100_DSP3_PM_510_2_MASK 0x00FF /* DSP3_PM_510 - [7:0] */ +#define WM5100_DSP3_PM_510_2_SHIFT 0 /* DSP3_PM_510 - [7:0] */ +#define WM5100_DSP3_PM_510_2_WIDTH 8 /* DSP3_PM_510 - [7:0] */ + +/* + * R36347 (0x8DFB) - DSP3 PM 1531 + */ +#define WM5100_DSP3_PM_510_1_MASK 0xFFFF /* DSP3_PM_510 - [15:0] */ +#define WM5100_DSP3_PM_510_1_SHIFT 0 /* DSP3_PM_510 - [15:0] */ +#define WM5100_DSP3_PM_510_1_WIDTH 16 /* DSP3_PM_510 - [15:0] */ + +/* + * R36348 (0x8DFC) - DSP3 PM 1532 + */ +#define WM5100_DSP3_PM_510_MASK 0xFFFF /* DSP3_PM_510 - [15:0] */ +#define WM5100_DSP3_PM_510_SHIFT 0 /* DSP3_PM_510 - [15:0] */ +#define WM5100_DSP3_PM_510_WIDTH 16 /* DSP3_PM_510 - [15:0] */ + +/* + * R36349 (0x8DFD) - DSP3 PM 1533 + */ +#define WM5100_DSP3_PM_END_2_MASK 0x00FF /* DSP3_PM_END - [7:0] */ +#define WM5100_DSP3_PM_END_2_SHIFT 0 /* DSP3_PM_END - [7:0] */ +#define WM5100_DSP3_PM_END_2_WIDTH 8 /* DSP3_PM_END - [7:0] */ + +/* + * R36350 (0x8DFE) - DSP3 PM 1534 + */ +#define WM5100_DSP3_PM_END_1_MASK 0xFFFF /* DSP3_PM_END - [15:0] */ +#define WM5100_DSP3_PM_END_1_SHIFT 0 /* DSP3_PM_END - [15:0] */ +#define WM5100_DSP3_PM_END_1_WIDTH 16 /* DSP3_PM_END - [15:0] */ + +/* + * R36351 (0x8DFF) - DSP3 PM 1535 + */ +#define WM5100_DSP3_PM_END_MASK 0xFFFF /* DSP3_PM_END - [15:0] */ +#define WM5100_DSP3_PM_END_SHIFT 0 /* DSP3_PM_END - [15:0] */ +#define WM5100_DSP3_PM_END_WIDTH 16 /* DSP3_PM_END - [15:0] */ + +/* + * R36864 (0x9000) - DSP3 ZM 0 + */ +#define WM5100_DSP3_ZM_START_1_MASK 0x00FF /* DSP3_ZM_START - [7:0] */ +#define WM5100_DSP3_ZM_START_1_SHIFT 0 /* DSP3_ZM_START - [7:0] */ +#define WM5100_DSP3_ZM_START_1_WIDTH 8 /* DSP3_ZM_START - [7:0] */ + +/* + * R36865 (0x9001) - DSP3 ZM 1 + */ +#define WM5100_DSP3_ZM_START_MASK 0xFFFF /* DSP3_ZM_START - [15:0] */ +#define WM5100_DSP3_ZM_START_SHIFT 0 /* DSP3_ZM_START - [15:0] */ +#define WM5100_DSP3_ZM_START_WIDTH 16 /* DSP3_ZM_START - [15:0] */ + +/* + * R36866 (0x9002) - DSP3 ZM 2 + */ +#define WM5100_DSP3_ZM_1_1_MASK 0x00FF /* DSP3_ZM_1 - [7:0] */ +#define WM5100_DSP3_ZM_1_1_SHIFT 0 /* DSP3_ZM_1 - [7:0] */ +#define WM5100_DSP3_ZM_1_1_WIDTH 8 /* DSP3_ZM_1 - [7:0] */ + +/* + * R36867 (0x9003) - DSP3 ZM 3 + */ +#define WM5100_DSP3_ZM_1_MASK 0xFFFF /* DSP3_ZM_1 - [15:0] */ +#define WM5100_DSP3_ZM_1_SHIFT 0 /* DSP3_ZM_1 - [15:0] */ +#define WM5100_DSP3_ZM_1_WIDTH 16 /* DSP3_ZM_1 - [15:0] */ + +/* + * R38908 (0x97FC) - DSP3 ZM 2044 + */ +#define WM5100_DSP3_ZM_1022_1_MASK 0x00FF /* DSP3_ZM_1022 - [7:0] */ +#define WM5100_DSP3_ZM_1022_1_SHIFT 0 /* DSP3_ZM_1022 - [7:0] */ +#define WM5100_DSP3_ZM_1022_1_WIDTH 8 /* DSP3_ZM_1022 - [7:0] */ + +/* + * R38909 (0x97FD) - DSP3 ZM 2045 + */ +#define WM5100_DSP3_ZM_1022_MASK 0xFFFF /* DSP3_ZM_1022 - [15:0] */ +#define WM5100_DSP3_ZM_1022_SHIFT 0 /* DSP3_ZM_1022 - [15:0] */ +#define WM5100_DSP3_ZM_1022_WIDTH 16 /* DSP3_ZM_1022 - [15:0] */ + +/* + * R38910 (0x97FE) - DSP3 ZM 2046 + */ +#define WM5100_DSP3_ZM_END_1_MASK 0x00FF /* DSP3_ZM_END - [7:0] */ +#define WM5100_DSP3_ZM_END_1_SHIFT 0 /* DSP3_ZM_END - [7:0] */ +#define WM5100_DSP3_ZM_END_1_WIDTH 8 /* DSP3_ZM_END - [7:0] */ + +/* + * R38911 (0x97FF) - DSP3 ZM 2047 + */ +#define WM5100_DSP3_ZM_END_MASK 0xFFFF /* DSP3_ZM_END - [15:0] */ +#define WM5100_DSP3_ZM_END_SHIFT 0 /* DSP3_ZM_END - [15:0] */ +#define WM5100_DSP3_ZM_END_WIDTH 16 /* DSP3_ZM_END - [15:0] */ + +int wm5100_readable_register(struct snd_soc_codec *codec, unsigned int reg); +int wm5100_volatile_register(struct snd_soc_codec *codec, unsigned int reg); + +extern u16 wm5100_reg_defaults[WM5100_MAX_REGISTER + 1]; + +#endif diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 6d6dc9efe914..35f3ad83dfb6 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -355,7 +355,7 @@ static int wm8350_put_volsw_2r_vu(struct snd_kcontrol *kcontrol, return 1; } - ret = snd_soc_put_volsw_2r(kcontrol, ucontrol); + ret = snd_soc_put_volsw(kcontrol, ucontrol); if (ret < 0) return ret; @@ -392,23 +392,9 @@ static int wm8350_get_volsw_2r(struct snd_kcontrol *kcontrol, break; } - return snd_soc_get_volsw_2r(kcontrol, ucontrol); + return snd_soc_get_volsw(kcontrol, ucontrol); } -/* double control with volume update */ -#define SOC_WM8350_DOUBLE_R_TLV(xname, reg_left, reg_right, xshift, xmax, \ - xinvert, tlv_array) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ - .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ - SNDRV_CTL_ELEM_ACCESS_READWRITE | \ - SNDRV_CTL_ELEM_ACCESS_VOLATILE, \ - .tlv.p = (tlv_array), \ - .info = snd_soc_info_volsw_2r, \ - .get = wm8350_get_volsw_2r, .put = wm8350_put_volsw_2r_vu, \ - .private_value = (unsigned long)&(struct soc_mixer_control) \ - {.reg = reg_left, .rreg = reg_right, .shift = xshift, \ - .rshift = xshift, .max = xmax, .invert = xinvert}, } - static const char *wm8350_deemp[] = { "None", "32kHz", "44.1kHz", "48kHz" }; static const char *wm8350_pol[] = { "Normal", "Inv R", "Inv L", "Inv L & R" }; static const char *wm8350_dacmutem[] = { "Normal", "Soft" }; @@ -443,26 +429,29 @@ static const unsigned int capture_sd_tlv[] = { static const struct snd_kcontrol_new wm8350_snd_controls[] = { SOC_ENUM("Playback Deemphasis", wm8350_enum[0]), SOC_ENUM("Playback DAC Inversion", wm8350_enum[1]), - SOC_WM8350_DOUBLE_R_TLV("Playback PCM Volume", + SOC_DOUBLE_R_EXT_TLV("Playback PCM Volume", WM8350_DAC_DIGITAL_VOLUME_L, WM8350_DAC_DIGITAL_VOLUME_R, - 0, 255, 0, dac_pcm_tlv), + 0, 255, 0, wm8350_get_volsw_2r, + wm8350_put_volsw_2r_vu, dac_pcm_tlv), SOC_ENUM("Playback PCM Mute Function", wm8350_enum[2]), SOC_ENUM("Playback PCM Mute Speed", wm8350_enum[3]), SOC_ENUM("Capture PCM Filter", wm8350_enum[4]), SOC_ENUM("Capture PCM HP Filter", wm8350_enum[5]), SOC_ENUM("Capture ADC Inversion", wm8350_enum[6]), - SOC_WM8350_DOUBLE_R_TLV("Capture PCM Volume", + SOC_DOUBLE_R_EXT_TLV("Capture PCM Volume", WM8350_ADC_DIGITAL_VOLUME_L, WM8350_ADC_DIGITAL_VOLUME_R, - 0, 255, 0, adc_pcm_tlv), + 0, 255, 0, wm8350_get_volsw_2r, + wm8350_put_volsw_2r_vu, adc_pcm_tlv), SOC_DOUBLE_TLV("Capture Sidetone Volume", WM8350_ADC_DIVIDER, 8, 4, 15, 1, capture_sd_tlv), - SOC_WM8350_DOUBLE_R_TLV("Capture Volume", + SOC_DOUBLE_R_EXT_TLV("Capture Volume", WM8350_LEFT_INPUT_VOLUME, WM8350_RIGHT_INPUT_VOLUME, - 2, 63, 0, pre_amp_tlv), + 2, 63, 0, wm8350_get_volsw_2r, + wm8350_put_volsw_2r_vu, pre_amp_tlv), SOC_DOUBLE_R("Capture ZC Switch", WM8350_LEFT_INPUT_VOLUME, WM8350_RIGHT_INPUT_VOLUME, 13, 1, 0), @@ -490,17 +479,19 @@ static const struct snd_kcontrol_new wm8350_snd_controls[] = { SOC_SINGLE_TLV("Out4 Capture Volume", WM8350_INPUT_MIXER_VOLUME, 1, 7, 0, out_mix_tlv), - SOC_WM8350_DOUBLE_R_TLV("Out1 Playback Volume", + SOC_DOUBLE_R_EXT_TLV("Out1 Playback Volume", WM8350_LOUT1_VOLUME, WM8350_ROUT1_VOLUME, - 2, 63, 0, out_pga_tlv), + 2, 63, 0, wm8350_get_volsw_2r, + wm8350_put_volsw_2r_vu, out_pga_tlv), SOC_DOUBLE_R("Out1 Playback ZC Switch", WM8350_LOUT1_VOLUME, WM8350_ROUT1_VOLUME, 13, 1, 0), - SOC_WM8350_DOUBLE_R_TLV("Out2 Playback Volume", + SOC_DOUBLE_R_EXT_TLV("Out2 Playback Volume", WM8350_LOUT2_VOLUME, WM8350_ROUT2_VOLUME, - 2, 63, 0, out_pga_tlv), + 2, 63, 0, wm8350_get_volsw_2r, + wm8350_put_volsw_2r_vu, out_pga_tlv), SOC_DOUBLE_R("Out2 Playback ZC Switch", WM8350_LOUT2_VOLUME, WM8350_ROUT2_VOLUME, 13, 1, 0), SOC_SINGLE("Out2 Right Invert Switch", WM8350_ROUT2_VOLUME, 10, 1, 0), diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c index fbee556cbf35..dc13be2a09c5 100644 --- a/sound/soc/codecs/wm8400.c +++ b/sound/soc/codecs/wm8400.c @@ -383,7 +383,7 @@ static int inmixer_event (struct snd_soc_dapm_widget *w, (1 << WM8400_AINRMUX_PWR))) { reg |= WM8400_AINR_ENA; } else { - reg &= ~WM8400_AINL_ENA; + reg &= ~WM8400_AINR_ENA; } wm8400_write(w->codec, WM8400_POWER_MANAGEMENT_2, reg); diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index db0dced74843..07c9cc759e97 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -20,6 +20,7 @@ #include <linux/platform_device.h> #include <linux/spi/spi.h> #include <linux/slab.h> +#include <linux/of_device.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -479,6 +480,8 @@ static int wm8510_set_bias_level(struct snd_soc_codec *codec, power1 |= WM8510_POWER1_BIASEN | WM8510_POWER1_BUFIOEN; if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + snd_soc_cache_sync(codec); + /* Initial cap charge at VMID 5k */ snd_soc_write(codec, WM8510_POWER1, power1 | 0x3); mdelay(100); @@ -540,18 +543,7 @@ static int wm8510_suspend(struct snd_soc_codec *codec, pm_message_t state) static int wm8510_resume(struct snd_soc_codec *codec) { - int i; - u8 data[2]; - u16 *cache = codec->reg_cache; - - /* Sync reg_cache with the hardware */ - for (i = 0; i < ARRAY_SIZE(wm8510_reg); i++) { - data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001); - data[1] = cache[i] & 0x00ff; - codec->hw_write(codec->control_data, data, 2); - } wm8510_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - return 0; } @@ -598,6 +590,11 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8510 = { .reg_cache_default =wm8510_reg, }; +static const struct of_device_id wm8510_of_match[] = { + { .compatible = "wlf,wm8510" }, + { }, +}; + #if defined(CONFIG_SPI_MASTER) static int __devinit wm8510_spi_probe(struct spi_device *spi) { @@ -628,6 +625,7 @@ static struct spi_driver wm8510_spi_driver = { .driver = { .name = "wm8510", .owner = THIS_MODULE, + .of_match_table = wm8510_of_match, }, .probe = wm8510_spi_probe, .remove = __devexit_p(wm8510_spi_remove), @@ -671,6 +669,7 @@ static struct i2c_driver wm8510_i2c_driver = { .driver = { .name = "wm8510-codec", .owner = THIS_MODULE, + .of_match_table = wm8510_of_match, }, .probe = wm8510_i2c_probe, .remove = __devexit_p(wm8510_i2c_remove), diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c index 4fd4d8dca0fc..db7a6819499f 100644 --- a/sound/soc/codecs/wm8523.c +++ b/sound/soc/codecs/wm8523.c @@ -20,6 +20,7 @@ #include <linux/platform_device.h> #include <linux/regulator/consumer.h> #include <linux/slab.h> +#include <linux/of_device.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -84,7 +85,7 @@ static const char *wm8523_zd_count_text[] = { static const struct soc_enum wm8523_zc_count = SOC_ENUM_SINGLE(WM8523_ZERO_DETECT, 0, 2, wm8523_zd_count_text); -static const struct snd_kcontrol_new wm8523_snd_controls[] = { +static const struct snd_kcontrol_new wm8523_controls[] = { SOC_DOUBLE_R_TLV("Playback Volume", WM8523_DAC_GAINL, WM8523_DAC_GAINR, 0, 448, 0, dac_tlv), SOC_SINGLE("ZC Switch", WM8523_DAC_CTRL3, 4, 1, 0), @@ -101,22 +102,11 @@ SND_SOC_DAPM_OUTPUT("LINEVOUTL"), SND_SOC_DAPM_OUTPUT("LINEVOUTR"), }; -static const struct snd_soc_dapm_route intercon[] = { +static const struct snd_soc_dapm_route wm8523_dapm_routes[] = { { "LINEVOUTL", NULL, "DAC" }, { "LINEVOUTR", NULL, "DAC" }, }; -static int wm8523_add_widgets(struct snd_soc_codec *codec) -{ - struct snd_soc_dapm_context *dapm = &codec->dapm; - - snd_soc_dapm_new_controls(dapm, wm8523_dapm_widgets, - ARRAY_SIZE(wm8523_dapm_widgets)); - snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon)); - - return 0; -} - static struct { int value; int ratio; @@ -416,7 +406,6 @@ static int wm8523_probe(struct snd_soc_codec *codec) struct wm8523_priv *wm8523 = snd_soc_codec_get_drvdata(codec); int ret, i; - codec->hw_write = (hw_write_t)i2c_master_send; wm8523->rate_constraint.list = &wm8523->rate_constraint_list[0]; wm8523->rate_constraint.count = ARRAY_SIZE(wm8523->rate_constraint_list); @@ -479,10 +468,6 @@ static int wm8523_probe(struct snd_soc_codec *codec) /* Bias level configuration will have done an extra enable */ regulator_bulk_disable(ARRAY_SIZE(wm8523->supplies), wm8523->supplies); - snd_soc_add_controls(codec, wm8523_snd_controls, - ARRAY_SIZE(wm8523_snd_controls)); - wm8523_add_widgets(codec); - return 0; err_enable: @@ -512,6 +497,18 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8523 = { .reg_word_size = sizeof(u16), .reg_cache_default = wm8523_reg, .volatile_register = wm8523_volatile_register, + + .controls = wm8523_controls, + .num_controls = ARRAY_SIZE(wm8523_controls), + .dapm_widgets = wm8523_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm8523_dapm_widgets), + .dapm_routes = wm8523_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(wm8523_dapm_routes), +}; + +static const struct of_device_id wm8523_of_match[] = { + { .compatible = "wlf,wm8523" }, + { }, }; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) @@ -551,8 +548,9 @@ MODULE_DEVICE_TABLE(i2c, wm8523_i2c_id); static struct i2c_driver wm8523_i2c_driver = { .driver = { - .name = "wm8523-codec", + .name = "wm8523", .owner = THIS_MODULE, + .of_match_table = wm8523_of_match, }, .probe = wm8523_i2c_probe, .remove = __devexit_p(wm8523_i2c_remove), diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index 4bbc0a79f01e..8212b3c8bfdd 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -26,6 +26,7 @@ #include <linux/platform_device.h> #include <linux/regulator/consumer.h> #include <linux/slab.h> +#include <linux/of_device.h> #include <sound/core.h> #include <sound/pcm.h> @@ -212,7 +213,7 @@ static int wm8580_out_vu(struct snd_kcontrol *kcontrol, reg_cache[reg] = 0; reg_cache[reg2] = 0; - ret = snd_soc_put_volsw_2r(kcontrol, ucontrol); + ret = snd_soc_put_volsw(kcontrol, ucontrol); if (ret < 0) return ret; @@ -223,31 +224,19 @@ static int wm8580_out_vu(struct snd_kcontrol *kcontrol, return 0; } -#define SOC_WM8580_OUT_DOUBLE_R_TLV(xname, reg_left, reg_right, xshift, xmax, \ - xinvert, tlv_array) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ - .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ - SNDRV_CTL_ELEM_ACCESS_READWRITE, \ - .tlv.p = (tlv_array), \ - .info = snd_soc_info_volsw_2r, \ - .get = snd_soc_get_volsw_2r, .put = wm8580_out_vu, \ - .private_value = (unsigned long)&(struct soc_mixer_control) \ - {.reg = reg_left, .rreg = reg_right, .shift = xshift, \ - .max = xmax, .invert = xinvert} } - static const struct snd_kcontrol_new wm8580_snd_controls[] = { -SOC_WM8580_OUT_DOUBLE_R_TLV("DAC1 Playback Volume", - WM8580_DIGITAL_ATTENUATION_DACL1, - WM8580_DIGITAL_ATTENUATION_DACR1, - 0, 0xff, 0, dac_tlv), -SOC_WM8580_OUT_DOUBLE_R_TLV("DAC2 Playback Volume", - WM8580_DIGITAL_ATTENUATION_DACL2, - WM8580_DIGITAL_ATTENUATION_DACR2, - 0, 0xff, 0, dac_tlv), -SOC_WM8580_OUT_DOUBLE_R_TLV("DAC3 Playback Volume", - WM8580_DIGITAL_ATTENUATION_DACL3, - WM8580_DIGITAL_ATTENUATION_DACR3, - 0, 0xff, 0, dac_tlv), +SOC_DOUBLE_R_EXT_TLV("DAC1 Playback Volume", + WM8580_DIGITAL_ATTENUATION_DACL1, + WM8580_DIGITAL_ATTENUATION_DACR1, + 0, 0xff, 0, snd_soc_get_volsw, wm8580_out_vu, dac_tlv), +SOC_DOUBLE_R_EXT_TLV("DAC2 Playback Volume", + WM8580_DIGITAL_ATTENUATION_DACL2, + WM8580_DIGITAL_ATTENUATION_DACR2, + 0, 0xff, 0, snd_soc_get_volsw, wm8580_out_vu, dac_tlv), +SOC_DOUBLE_R_EXT_TLV("DAC3 Playback Volume", + WM8580_DIGITAL_ATTENUATION_DACL3, + WM8580_DIGITAL_ATTENUATION_DACR3, + 0, 0xff, 0, snd_soc_get_volsw, wm8580_out_vu, dac_tlv), SOC_SINGLE("DAC1 Deemphasis Switch", WM8580_DAC_CONTROL3, 0, 1, 0), SOC_SINGLE("DAC2 Deemphasis Switch", WM8580_DAC_CONTROL3, 1, 1, 0), @@ -441,8 +430,7 @@ static int wm8580_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, /* Always disable the PLL - it is not safe to leave it running * while reprogramming it. */ - reg = snd_soc_read(codec, WM8580_PWRDN2); - snd_soc_write(codec, WM8580_PWRDN2, reg | pwr_mask); + snd_soc_update_bits(codec, WM8580_PWRDN2, pwr_mask, pwr_mask); if (!freq_in || !freq_out) return 0; @@ -460,8 +448,7 @@ static int wm8580_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, snd_soc_write(codec, WM8580_PLLA4 + offset, reg); /* All done, turn it on */ - reg = snd_soc_read(codec, WM8580_PWRDN2); - snd_soc_write(codec, WM8580_PWRDN2, reg & ~pwr_mask); + snd_soc_update_bits(codec, WM8580_PWRDN2, pwr_mask, 0); return 0; } @@ -759,7 +746,6 @@ static int wm8580_digital_mute(struct snd_soc_dai *codec_dai, int mute) static int wm8580_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - u16 reg; switch (level) { case SND_SOC_BIAS_ON: case SND_SOC_BIAS_PREPARE: @@ -768,20 +754,19 @@ static int wm8580_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_STANDBY: if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { /* Power up and get individual control of the DACs */ - reg = snd_soc_read(codec, WM8580_PWRDN1); - reg &= ~(WM8580_PWRDN1_PWDN | WM8580_PWRDN1_ALLDACPD); - snd_soc_write(codec, WM8580_PWRDN1, reg); + snd_soc_update_bits(codec, WM8580_PWRDN1, + WM8580_PWRDN1_PWDN | + WM8580_PWRDN1_ALLDACPD, 0); /* Make VMID high impedance */ - reg = snd_soc_read(codec, WM8580_ADC_CONTROL1); - reg &= ~0x100; - snd_soc_write(codec, WM8580_ADC_CONTROL1, reg); + snd_soc_update_bits(codec, WM8580_ADC_CONTROL1, + 0x100, 0); } break; case SND_SOC_BIAS_OFF: - reg = snd_soc_read(codec, WM8580_PWRDN1); - snd_soc_write(codec, WM8580_PWRDN1, reg | WM8580_PWRDN1_PWDN); + snd_soc_update_bits(codec, WM8580_PWRDN1, + WM8580_PWRDN1_PWDN, WM8580_PWRDN1_PWDN); break; } codec->dapm.bias_level = level; @@ -907,6 +892,11 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8580 = { .reg_cache_default = wm8580_reg, }; +static const struct of_device_id wm8580_of_match[] = { + { .compatible = "wlf,wm8580" }, + { }, +}; + #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) static int wm8580_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) @@ -943,8 +933,9 @@ MODULE_DEVICE_TABLE(i2c, wm8580_i2c_id); static struct i2c_driver wm8580_i2c_driver = { .driver = { - .name = "wm8580-codec", + .name = "wm8580", .owner = THIS_MODULE, + .of_match_table = wm8580_of_match, }, .probe = wm8580_i2c_probe, .remove = wm8580_i2c_remove, diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c index a537e4af6ae7..8d0347cf0e9a 100644 --- a/sound/soc/codecs/wm8711.c +++ b/sound/soc/codecs/wm8711.c @@ -3,7 +3,7 @@ * * Copyright 2006 Wolfson Microelectronics * - * Author: Mike Arthur <linux@wolfsonmicro.com> + * Author: Mike Arthur <Mike.Arthur@wolfsonmicro.com> * * Based on wm8731.c by Richard Purdie * @@ -21,6 +21,7 @@ #include <linux/platform_device.h> #include <linux/spi/spi.h> #include <linux/slab.h> +#include <linux/of_device.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -286,7 +287,6 @@ static int wm8711_set_dai_fmt(struct snd_soc_dai *codec_dai, return 0; } - static int wm8711_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { @@ -299,6 +299,9 @@ static int wm8711_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_PREPARE: break; case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) + snd_soc_cache_sync(codec); + snd_soc_write(codec, WM8711_PWR, reg | 0x0040); break; case SND_SOC_BIAS_OFF: @@ -345,25 +348,14 @@ static int wm8711_suspend(struct snd_soc_codec *codec, pm_message_t state) static int wm8711_resume(struct snd_soc_codec *codec) { - int i; - u8 data[2]; - u16 *cache = codec->reg_cache; - - /* Sync reg_cache with the hardware */ - for (i = 0; i < ARRAY_SIZE(wm8711_reg); i++) { - data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001); - data[1] = cache[i] & 0x00ff; - codec->hw_write(codec->control_data, data, 2); - } wm8711_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - return 0; } static int wm8711_probe(struct snd_soc_codec *codec) { struct wm8711_priv *wm8711 = snd_soc_codec_get_drvdata(codec); - int ret, reg; + int ret; ret = snd_soc_codec_set_cache_io(codec, 7, 9, wm8711->bus_type); if (ret < 0) { @@ -380,10 +372,8 @@ static int wm8711_probe(struct snd_soc_codec *codec) wm8711_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* Latch the update bits */ - reg = snd_soc_read(codec, WM8711_LOUT1V); - snd_soc_write(codec, WM8711_LOUT1V, reg | 0x0100); - reg = snd_soc_read(codec, WM8711_ROUT1V); - snd_soc_write(codec, WM8711_ROUT1V, reg | 0x0100); + snd_soc_update_bits(codec, WM8711_LOUT1V, 0x0100, 0x0100); + snd_soc_update_bits(codec, WM8711_ROUT1V, 0x0100, 0x0100); snd_soc_add_controls(codec, wm8711_snd_controls, ARRAY_SIZE(wm8711_snd_controls)); @@ -414,6 +404,12 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8711 = { .num_dapm_routes = ARRAY_SIZE(wm8711_intercon), }; +static const struct of_device_id wm8711_of_match[] = { + { .compatible = "wlf,wm8711", }, + { } +}; +MODULE_DEVICE_TABLE(of, wm8711_of_match); + #if defined(CONFIG_SPI_MASTER) static int __devinit wm8711_spi_probe(struct spi_device *spi) { @@ -443,8 +439,9 @@ static int __devexit wm8711_spi_remove(struct spi_device *spi) static struct spi_driver wm8711_spi_driver = { .driver = { - .name = "wm8711-codec", + .name = "wm8711", .owner = THIS_MODULE, + .of_match_table = wm8711_of_match, }, .probe = wm8711_spi_probe, .remove = __devexit_p(wm8711_spi_remove), @@ -487,8 +484,9 @@ MODULE_DEVICE_TABLE(i2c, wm8711_i2c_id); static struct i2c_driver wm8711_i2c_driver = { .driver = { - .name = "wm8711-codec", + .name = "wm8711", .owner = THIS_MODULE, + .of_match_table = wm8711_of_match, }, .probe = wm8711_i2c_probe, .remove = __devexit_p(wm8711_i2c_remove), diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c index 86d4718d3a76..04b027efd5c0 100644 --- a/sound/soc/codecs/wm8728.c +++ b/sound/soc/codecs/wm8728.c @@ -19,6 +19,7 @@ #include <linux/platform_device.h> #include <linux/spi/spi.h> #include <linux/slab.h> +#include <linux/of_device.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -269,6 +270,12 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8728 = { .num_dapm_routes = ARRAY_SIZE(wm8728_intercon), }; +static const struct of_device_id wm8728_of_match[] = { + { .compatible = "wlf,wm8728", }, + { } +}; +MODULE_DEVICE_TABLE(of, wm8728_of_match); + #if defined(CONFIG_SPI_MASTER) static int __devinit wm8728_spi_probe(struct spi_device *spi) { @@ -298,8 +305,9 @@ static int __devexit wm8728_spi_remove(struct spi_device *spi) static struct spi_driver wm8728_spi_driver = { .driver = { - .name = "wm8728-codec", + .name = "wm8728", .owner = THIS_MODULE, + .of_match_table = wm8728_of_match, }, .probe = wm8728_spi_probe, .remove = __devexit_p(wm8728_spi_remove), @@ -342,8 +350,9 @@ MODULE_DEVICE_TABLE(i2c, wm8728_i2c_id); static struct i2c_driver wm8728_i2c_driver = { .driver = { - .name = "wm8728-codec", + .name = "wm8728", .owner = THIS_MODULE, + .of_match_table = wm8728_of_match, }, .probe = wm8728_i2c_probe, .remove = __devexit_p(wm8728_i2c_remove), diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 76b4361e9b80..7e5ec03f6f8d 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -22,6 +22,7 @@ #include <linux/platform_device.h> #include <linux/regulator/consumer.h> #include <linux/spi/spi.h> +#include <linux/of_device.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -426,9 +427,7 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { struct wm8731_priv *wm8731 = snd_soc_codec_get_drvdata(codec); - int i, ret; - u8 data[2]; - u16 *cache = codec->reg_cache; + int ret; u16 reg; switch (level) { @@ -443,16 +442,7 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec, if (ret != 0) return ret; - /* Sync reg_cache with the hardware */ - for (i = 0; i < ARRAY_SIZE(wm8731_reg); i++) { - if (cache[i] == wm8731_reg[i]) - continue; - - data[0] = (i << 1) | ((cache[i] >> 8) - & 0x0001); - data[1] = cache[i] & 0x00ff; - codec->hw_write(codec->control_data, data, 2); - } + snd_soc_cache_sync(codec); } /* Clear PWROFF, gate CLKOUT, everything else as-is */ @@ -607,6 +597,13 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8731 = { .num_dapm_routes = ARRAY_SIZE(wm8731_intercon), }; +static const struct of_device_id wm8731_of_match[] = { + { .compatible = "wlf,wm8731", }, + { } +}; + +MODULE_DEVICE_TABLE(of, wm8731_of_match); + #if defined(CONFIG_SPI_MASTER) static int __devinit wm8731_spi_probe(struct spi_device *spi) { @@ -638,6 +635,7 @@ static struct spi_driver wm8731_spi_driver = { .driver = { .name = "wm8731", .owner = THIS_MODULE, + .of_match_table = wm8731_of_match, }, .probe = wm8731_spi_probe, .remove = __devexit_p(wm8731_spi_remove), @@ -682,6 +680,7 @@ static struct i2c_driver wm8731_i2c_driver = { .driver = { .name = "wm8731", .owner = THIS_MODULE, + .of_match_table = wm8731_of_match, }, .probe = wm8731_i2c_probe, .remove = __devexit_p(wm8731_i2c_remove), diff --git a/sound/soc/codecs/wm8737.c b/sound/soc/codecs/wm8737.c index 30c67d06a904..f6aef58845c2 100644 --- a/sound/soc/codecs/wm8737.c +++ b/sound/soc/codecs/wm8737.c @@ -20,6 +20,7 @@ #include <linux/regulator/consumer.h> #include <linux/spi/spi.h> #include <linux/slab.h> +#include <linux/of_device.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -634,6 +635,13 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8737 = { .reg_cache_default = wm8737_reg, }; +static const struct of_device_id wm8737_of_match[] = { + { .compatible = "wlf,wm8737", }, + { } +}; + +MODULE_DEVICE_TABLE(of, wm8737_of_match); + #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) static __devinit int wm8737_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) @@ -673,6 +681,7 @@ static struct i2c_driver wm8737_i2c_driver = { .driver = { .name = "wm8737", .owner = THIS_MODULE, + .of_match_table = wm8737_of_match, }, .probe = wm8737_i2c_probe, .remove = __devexit_p(wm8737_i2c_remove), @@ -711,6 +720,7 @@ static struct spi_driver wm8737_spi_driver = { .driver = { .name = "wm8737", .owner = THIS_MODULE, + .of_match_table = wm8737_of_match, }, .probe = wm8737_spi_probe, .remove = __devexit_p(wm8737_spi_remove), diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index 25af901fe813..57ad22aacc51 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -17,9 +17,11 @@ #include <linux/delay.h> #include <linux/pm.h> #include <linux/i2c.h> +#include <linux/spi/spi.h> #include <linux/platform_device.h> #include <linux/regulator/consumer.h> #include <linux/slab.h> +#include <linux/of_device.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -337,10 +339,10 @@ static int wm8741_set_dai_fmt(struct snd_soc_dai *codec_dai, iface |= 0x0004; break; case SND_SOC_DAIFMT_DSP_A: - iface |= 0x0003; + iface |= 0x000C; break; case SND_SOC_DAIFMT_DSP_B: - iface |= 0x0013; + iface |= 0x001C; break; default: return -EINVAL; @@ -402,15 +404,7 @@ static struct snd_soc_dai_driver wm8741_dai = { #ifdef CONFIG_PM static int wm8741_resume(struct snd_soc_codec *codec) { - u16 *cache = codec->reg_cache; - int i; - - /* RESTORE REG Cache */ - for (i = 0; i < WM8741_REGISTER_COUNT; i++) { - if (cache[i] == wm8741_reg_defaults[i] || WM8741_RESET == i) - continue; - snd_soc_write(codec, i, cache[i]); - } + snd_soc_cache_sync(codec); return 0; } #else @@ -422,17 +416,35 @@ static int wm8741_probe(struct snd_soc_codec *codec) { struct wm8741_priv *wm8741 = snd_soc_codec_get_drvdata(codec); int ret = 0; + int i; + + for (i = 0; i < ARRAY_SIZE(wm8741->supplies); i++) + wm8741->supplies[i].supply = wm8741_supply_names[i]; + + ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(wm8741->supplies), + wm8741->supplies); + if (ret != 0) { + dev_err(codec->dev, "Failed to request supplies: %d\n", ret); + goto err; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(wm8741->supplies), + wm8741->supplies); + if (ret != 0) { + dev_err(codec->dev, "Failed to enable supplies: %d\n", ret); + goto err_get; + } ret = snd_soc_codec_set_cache_io(codec, 7, 9, wm8741->control_type); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; + goto err_enable; } ret = wm8741_reset(codec); if (ret < 0) { dev_err(codec->dev, "Failed to issue reset\n"); - return ret; + goto err_enable; } /* Change some default settings - latch VU */ @@ -442,7 +454,7 @@ static int wm8741_probe(struct snd_soc_codec *codec) WM8741_UPDATELM, WM8741_UPDATELM); snd_soc_update_bits(codec, WM8741_DACRLSB_ATTENUATION, WM8741_UPDATERL, WM8741_UPDATERL); - snd_soc_update_bits(codec, WM8741_DACRLSB_ATTENUATION, + snd_soc_update_bits(codec, WM8741_DACRMSB_ATTENUATION, WM8741_UPDATERM, WM8741_UPDATERM); snd_soc_add_controls(codec, wm8741_snd_controls, @@ -451,58 +463,61 @@ static int wm8741_probe(struct snd_soc_codec *codec) dev_dbg(codec->dev, "Successful registration\n"); return ret; + +err_enable: + regulator_bulk_disable(ARRAY_SIZE(wm8741->supplies), wm8741->supplies); +err_get: + regulator_bulk_free(ARRAY_SIZE(wm8741->supplies), wm8741->supplies); +err: + return ret; +} + +static int wm8741_remove(struct snd_soc_codec *codec) +{ + struct wm8741_priv *wm8741 = snd_soc_codec_get_drvdata(codec); + + regulator_bulk_disable(ARRAY_SIZE(wm8741->supplies), wm8741->supplies); + regulator_bulk_free(ARRAY_SIZE(wm8741->supplies), wm8741->supplies); + + return 0; } static struct snd_soc_codec_driver soc_codec_dev_wm8741 = { .probe = wm8741_probe, + .remove = wm8741_remove, .resume = wm8741_resume, .reg_cache_size = ARRAY_SIZE(wm8741_reg_defaults), .reg_word_size = sizeof(u16), .reg_cache_default = wm8741_reg_defaults, }; +static const struct of_device_id wm8741_of_match[] = { + { .compatible = "wlf,wm8741", }, + { } +}; +MODULE_DEVICE_TABLE(of, wm8741_of_match); + #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) static int wm8741_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct wm8741_priv *wm8741; - int ret, i; + int ret; wm8741 = kzalloc(sizeof(struct wm8741_priv), GFP_KERNEL); if (wm8741 == NULL) return -ENOMEM; - for (i = 0; i < ARRAY_SIZE(wm8741->supplies); i++) - wm8741->supplies[i].supply = wm8741_supply_names[i]; - - ret = regulator_bulk_get(&i2c->dev, ARRAY_SIZE(wm8741->supplies), - wm8741->supplies); - if (ret != 0) { - dev_err(&i2c->dev, "Failed to request supplies: %d\n", ret); - goto err; - } - - ret = regulator_bulk_enable(ARRAY_SIZE(wm8741->supplies), - wm8741->supplies); - if (ret != 0) { - dev_err(&i2c->dev, "Failed to enable supplies: %d\n", ret); - goto err_get; - } - i2c_set_clientdata(i2c, wm8741); wm8741->control_type = SND_SOC_I2C; - ret = snd_soc_register_codec(&i2c->dev, - &soc_codec_dev_wm8741, &wm8741_dai, 1); - if (ret < 0) - goto err_enable; - return ret; + ret = snd_soc_register_codec(&i2c->dev, + &soc_codec_dev_wm8741, &wm8741_dai, 1); + if (ret != 0) + goto err; -err_enable: - regulator_bulk_disable(ARRAY_SIZE(wm8741->supplies), wm8741->supplies); + return ret; -err_get: - regulator_bulk_free(ARRAY_SIZE(wm8741->supplies), wm8741->supplies); err: kfree(wm8741); return ret; @@ -510,10 +525,7 @@ err: static int wm8741_i2c_remove(struct i2c_client *client) { - struct wm8741_priv *wm8741 = i2c_get_clientdata(client); - snd_soc_unregister_codec(&client->dev); - regulator_bulk_free(ARRAY_SIZE(wm8741->supplies), wm8741->supplies); kfree(i2c_get_clientdata(client)); return 0; } @@ -526,8 +538,9 @@ MODULE_DEVICE_TABLE(i2c, wm8741_i2c_id); static struct i2c_driver wm8741_i2c_driver = { .driver = { - .name = "wm8741-codec", + .name = "wm8741", .owner = THIS_MODULE, + .of_match_table = wm8741_of_match, }, .probe = wm8741_i2c_probe, .remove = wm8741_i2c_remove, @@ -535,6 +548,44 @@ static struct i2c_driver wm8741_i2c_driver = { }; #endif +#if defined(CONFIG_SPI_MASTER) +static int __devinit wm8741_spi_probe(struct spi_device *spi) +{ + struct wm8741_priv *wm8741; + int ret; + + wm8741 = kzalloc(sizeof(struct wm8741_priv), GFP_KERNEL); + if (wm8741 == NULL) + return -ENOMEM; + + wm8741->control_type = SND_SOC_SPI; + spi_set_drvdata(spi, wm8741); + + ret = snd_soc_register_codec(&spi->dev, + &soc_codec_dev_wm8741, &wm8741_dai, 1); + if (ret < 0) + kfree(wm8741); + return ret; +} + +static int __devexit wm8741_spi_remove(struct spi_device *spi) +{ + snd_soc_unregister_codec(&spi->dev); + kfree(spi_get_drvdata(spi)); + return 0; +} + +static struct spi_driver wm8741_spi_driver = { + .driver = { + .name = "wm8741", + .owner = THIS_MODULE, + .of_match_table = wm8741_of_match, + }, + .probe = wm8741_spi_probe, + .remove = __devexit_p(wm8741_spi_remove), +}; +#endif /* CONFIG_SPI_MASTER */ + static int __init wm8741_modinit(void) { int ret = 0; @@ -544,6 +595,13 @@ static int __init wm8741_modinit(void) if (ret != 0) pr_err("Failed to register WM8741 I2C driver: %d\n", ret); #endif +#if defined(CONFIG_SPI_MASTER) + ret = spi_register_driver(&wm8741_spi_driver); + if (ret != 0) { + printk(KERN_ERR "Failed to register wm8741 SPI driver: %d\n", + ret); + } +#endif return ret; } @@ -551,6 +609,9 @@ module_init(wm8741_modinit); static void __exit wm8741_exit(void) { +#if defined(CONFIG_SPI_MASTER) + spi_unregister_driver(&wm8741_spi_driver); +#endif #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) i2c_del_driver(&wm8741_i2c_driver); #endif diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index d0003cc3bcd6..ca75a8180708 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -21,6 +21,7 @@ #include <linux/platform_device.h> #include <linux/spi/spi.h> #include <linux/slab.h> +#include <linux/of_device.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -615,6 +616,8 @@ static int wm8750_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_STANDBY: if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + snd_soc_cache_sync(codec); + /* Set VMID to 5k */ snd_soc_write(codec, WM8750_PWR1, pwr_reg | 0x01c1); @@ -672,28 +675,14 @@ static int wm8750_suspend(struct snd_soc_codec *codec, pm_message_t state) static int wm8750_resume(struct snd_soc_codec *codec) { - int i; - u8 data[2]; - u16 *cache = codec->reg_cache; - - /* Sync reg_cache with the hardware */ - for (i = 0; i < ARRAY_SIZE(wm8750_reg); i++) { - if (i == WM8750_RESET) - continue; - data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001); - data[1] = cache[i] & 0x00ff; - codec->hw_write(codec->control_data, data, 2); - } - wm8750_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - return 0; } static int wm8750_probe(struct snd_soc_codec *codec) { struct wm8750_priv *wm8750 = snd_soc_codec_get_drvdata(codec); - int reg, ret; + int ret; ret = snd_soc_codec_set_cache_io(codec, 7, 9, wm8750->control_type); if (ret < 0) { @@ -711,22 +700,14 @@ static int wm8750_probe(struct snd_soc_codec *codec) wm8750_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* set the update bits */ - reg = snd_soc_read(codec, WM8750_LDAC); - snd_soc_write(codec, WM8750_LDAC, reg | 0x0100); - reg = snd_soc_read(codec, WM8750_RDAC); - snd_soc_write(codec, WM8750_RDAC, reg | 0x0100); - reg = snd_soc_read(codec, WM8750_LOUT1V); - snd_soc_write(codec, WM8750_LOUT1V, reg | 0x0100); - reg = snd_soc_read(codec, WM8750_ROUT1V); - snd_soc_write(codec, WM8750_ROUT1V, reg | 0x0100); - reg = snd_soc_read(codec, WM8750_LOUT2V); - snd_soc_write(codec, WM8750_LOUT2V, reg | 0x0100); - reg = snd_soc_read(codec, WM8750_ROUT2V); - snd_soc_write(codec, WM8750_ROUT2V, reg | 0x0100); - reg = snd_soc_read(codec, WM8750_LINVOL); - snd_soc_write(codec, WM8750_LINVOL, reg | 0x0100); - reg = snd_soc_read(codec, WM8750_RINVOL); - snd_soc_write(codec, WM8750_RINVOL, reg | 0x0100); + snd_soc_update_bits(codec, WM8750_LDAC, 0x0100, 0x0100); + snd_soc_update_bits(codec, WM8750_RDAC, 0x0100, 0x0100); + snd_soc_update_bits(codec, WM8750_LOUT1V, 0x0100, 0x0100); + snd_soc_update_bits(codec, WM8750_ROUT1V, 0x0100, 0x0100); + snd_soc_update_bits(codec, WM8750_LOUT2V, 0x0100, 0x0100); + snd_soc_update_bits(codec, WM8750_ROUT2V, 0x0100, 0x0100); + snd_soc_update_bits(codec, WM8750_LINVOL, 0x0100, 0x0100); + snd_soc_update_bits(codec, WM8750_RINVOL, 0x0100, 0x0100); snd_soc_add_controls(codec, wm8750_snd_controls, ARRAY_SIZE(wm8750_snd_controls)); @@ -751,6 +732,13 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8750 = { .reg_cache_default = wm8750_reg, }; +static const struct of_device_id wm8750_of_match[] = { + { .compatible = "wlf,wm8750", }, + { .compatible = "wlf,wm8987", }, + { } +}; +MODULE_DEVICE_TABLE(of, wm8750_of_match); + #if defined(CONFIG_SPI_MASTER) static int __devinit wm8750_spi_probe(struct spi_device *spi) { @@ -787,8 +775,9 @@ MODULE_DEVICE_TABLE(spi, wm8750_spi_ids); static struct spi_driver wm8750_spi_driver = { .driver = { - .name = "wm8750-codec", + .name = "wm8750", .owner = THIS_MODULE, + .of_match_table = wm8750_of_match, }, .id_table = wm8750_spi_ids, .probe = wm8750_spi_probe, @@ -833,8 +822,9 @@ MODULE_DEVICE_TABLE(i2c, wm8750_i2c_id); static struct i2c_driver wm8750_i2c_driver = { .driver = { - .name = "wm8750-codec", + .name = "wm8750", .owner = THIS_MODULE, + .of_match_table = wm8750_of_match, }, .probe = wm8750_i2c_probe, .remove = __devexit_p(wm8750_i2c_remove), diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index aa091a0d8187..a9504710bb69 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -38,6 +38,7 @@ #include <linux/delay.h> #include <linux/pm.h> #include <linux/i2c.h> +#include <linux/of_device.h> #include <linux/platform_device.h> #include <linux/spi/spi.h> #include <linux/slab.h> @@ -1490,6 +1491,12 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8753 = { .reg_cache_default = wm8753_reg, }; +static const struct of_device_id wm8753_of_match[] = { + { .compatible = "wlf,wm8753", }, + { } +}; +MODULE_DEVICE_TABLE(of, wm8753_of_match); + #if defined(CONFIG_SPI_MASTER) static int __devinit wm8753_spi_probe(struct spi_device *spi) { @@ -1519,8 +1526,9 @@ static int __devexit wm8753_spi_remove(struct spi_device *spi) static struct spi_driver wm8753_spi_driver = { .driver = { - .name = "wm8753-codec", + .name = "wm8753", .owner = THIS_MODULE, + .of_match_table = wm8753_of_match, }, .probe = wm8753_spi_probe, .remove = __devexit_p(wm8753_spi_remove), @@ -1563,8 +1571,9 @@ MODULE_DEVICE_TABLE(i2c, wm8753_i2c_id); static struct i2c_driver wm8753_i2c_driver = { .driver = { - .name = "wm8753-codec", + .name = "wm8753", .owner = THIS_MODULE, + .of_match_table = wm8753_of_match, }, .probe = wm8753_i2c_probe, .remove = __devexit_p(wm8753_i2c_remove), diff --git a/sound/soc/codecs/wm8770.c b/sound/soc/codecs/wm8770.c index 19b92baa9e8c..aa05e6507f84 100644 --- a/sound/soc/codecs/wm8770.c +++ b/sound/soc/codecs/wm8770.c @@ -14,6 +14,7 @@ #include <linux/moduleparam.h> #include <linux/init.h> #include <linux/delay.h> +#include <linux/of_device.h> #include <linux/pm.h> #include <linux/platform_device.h> #include <linux/spi/spi.h> @@ -684,6 +685,12 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8770 = { .reg_cache_default = wm8770_reg_defs }; +static const struct of_device_id wm8770_of_match[] = { + { .compatible = "wlf,wm8770", }, + { } +}; +MODULE_DEVICE_TABLE(of, wm8770_of_match); + #if defined(CONFIG_SPI_MASTER) static int __devinit wm8770_spi_probe(struct spi_device *spi) { @@ -715,6 +722,7 @@ static struct spi_driver wm8770_spi_driver = { .driver = { .name = "wm8770", .owner = THIS_MODULE, + .of_match_table = wm8770_of_match, }, .probe = wm8770_spi_probe, .remove = __devexit_p(wm8770_spi_remove) diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c index 8e7953b1b790..bfdc52370ad0 100644 --- a/sound/soc/codecs/wm8776.c +++ b/sound/soc/codecs/wm8776.c @@ -18,6 +18,7 @@ #include <linux/delay.h> #include <linux/pm.h> #include <linux/i2c.h> +#include <linux/of_device.h> #include <linux/platform_device.h> #include <linux/spi/spi.h> #include <linux/slab.h> @@ -215,8 +216,6 @@ static int wm8776_hw_params(struct snd_pcm_substream *substream, int ratio_shift, master; int i; - iface = 0; - switch (dai->driver->id) { case WM8776_DAI_DAC: iface_reg = WM8776_DACIFCTRL; @@ -232,20 +231,23 @@ static int wm8776_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - /* Set word length */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: - break; - case SNDRV_PCM_FORMAT_S20_3LE: - iface |= 0x10; + switch (snd_pcm_format_width(params_format(params))) { + case 16: + iface = 0; + case 20: + iface = 0x10; break; - case SNDRV_PCM_FORMAT_S24_LE: - iface |= 0x20; + case 24: + iface = 0x20; break; - case SNDRV_PCM_FORMAT_S32_LE: - iface |= 0x30; + case 32: + iface = 0x30; break; + default: + dev_err(codec->dev, "Unsupported sample size: %i\n", + snd_pcm_format_width(params_format(params))); + return -EINVAL; } /* Only need to set MCLK/LRCLK ratio if we're master */ @@ -306,6 +308,8 @@ static int wm8776_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_STANDBY: if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + snd_soc_cache_sync(codec); + /* Disable the global powerdown; DAPM does the rest */ snd_soc_update_bits(codec, WM8776_PWRDOWN, 1, 0); } @@ -320,11 +324,6 @@ static int wm8776_set_bias_level(struct snd_soc_codec *codec, return 0; } -#define WM8776_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\ - SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 |\ - SNDRV_PCM_RATE_96000) - - #define WM8776_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) @@ -349,7 +348,9 @@ static struct snd_soc_dai_driver wm8776_dai[] = { .stream_name = "Playback", .channels_min = 2, .channels_max = 2, - .rates = WM8776_RATES, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 32000, + .rate_max = 192000, .formats = WM8776_FORMATS, }, .ops = &wm8776_dac_ops, @@ -361,7 +362,9 @@ static struct snd_soc_dai_driver wm8776_dai[] = { .stream_name = "Capture", .channels_min = 2, .channels_max = 2, - .rates = WM8776_RATES, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 32000, + .rate_max = 96000, .formats = WM8776_FORMATS, }, .ops = &wm8776_adc_ops, @@ -378,21 +381,7 @@ static int wm8776_suspend(struct snd_soc_codec *codec, pm_message_t state) static int wm8776_resume(struct snd_soc_codec *codec) { - int i; - u8 data[2]; - u16 *cache = codec->reg_cache; - - /* Sync reg_cache with the hardware */ - for (i = 0; i < ARRAY_SIZE(wm8776_reg); i++) { - if (cache[i] == wm8776_reg[i]) - continue; - data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001); - data[1] = cache[i] & 0x00ff; - codec->hw_write(codec->control_data, data, 2); - } - wm8776_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - return 0; } #else @@ -452,6 +441,12 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8776 = { .reg_cache_default = wm8776_reg, }; +static const struct of_device_id wm8776_of_match[] = { + { .compatible = "wlf,wm8776", }, + { } +}; +MODULE_DEVICE_TABLE(of, wm8776_of_match); + #if defined(CONFIG_SPI_MASTER) static int __devinit wm8776_spi_probe(struct spi_device *spi) { @@ -481,8 +476,9 @@ static int __devexit wm8776_spi_remove(struct spi_device *spi) static struct spi_driver wm8776_spi_driver = { .driver = { - .name = "wm8776-codec", + .name = "wm8776", .owner = THIS_MODULE, + .of_match_table = wm8776_of_match, }, .probe = wm8776_spi_probe, .remove = __devexit_p(wm8776_spi_remove), @@ -525,8 +521,9 @@ MODULE_DEVICE_TABLE(i2c, wm8776_i2c_id); static struct i2c_driver wm8776_i2c_driver = { .driver = { - .name = "wm8776-codec", + .name = "wm8776", .owner = THIS_MODULE, + .of_match_table = wm8776_of_match, }, .probe = wm8776_i2c_probe, .remove = __devexit_p(wm8776_i2c_remove), diff --git a/sound/soc/codecs/wm8782.c b/sound/soc/codecs/wm8782.c index a2a09f85ea99..f2ced71328b0 100644 --- a/sound/soc/codecs/wm8782.c +++ b/sound/soc/codecs/wm8782.c @@ -60,7 +60,7 @@ static struct platform_driver wm8782_codec_driver = { .owner = THIS_MODULE, }, .probe = wm8782_probe, - .remove = wm8782_remove, + .remove = __devexit_p(wm8782_remove), }; static int __init wm8782_init(void) diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index 9a5e67c5a6bd..9ee072b85975 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -16,6 +16,7 @@ #include <linux/delay.h> #include <linux/pm.h> #include <linux/i2c.h> +#include <linux/of_device.h> #include <linux/spi/spi.h> #include <linux/regulator/consumer.h> #include <linux/slab.h> @@ -717,6 +718,12 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8804 = { .volatile_register = wm8804_volatile }; +static const struct of_device_id wm8804_of_match[] = { + { .compatible = "wlf,wm8804", }, + { } +}; +MODULE_DEVICE_TABLE(of, wm8804_of_match); + #if defined(CONFIG_SPI_MASTER) static int __devinit wm8804_spi_probe(struct spi_device *spi) { @@ -748,6 +755,7 @@ static struct spi_driver wm8804_spi_driver = { .driver = { .name = "wm8804", .owner = THIS_MODULE, + .of_match_table = wm8804_of_match, }, .probe = wm8804_spi_probe, .remove = __devexit_p(wm8804_spi_remove) @@ -792,6 +800,7 @@ static struct i2c_driver wm8804_i2c_driver = { .driver = { .name = "wm8804", .owner = THIS_MODULE, + .of_match_table = wm8804_of_match, }, .probe = wm8804_i2c_probe, .remove = __devexit_p(wm8804_i2c_remove), diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index 082040eda8a2..3d0dc1591ecc 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -110,8 +110,8 @@ #define WM8900_REG_CLOCKING1_BCLK_DIR 0x1 #define WM8900_REG_CLOCKING1_MCLK_SRC 0x100 -#define WM8900_REG_CLOCKING1_BCLK_MASK (~0x01e) -#define WM8900_REG_CLOCKING1_OPCLK_MASK (~0x7000) +#define WM8900_REG_CLOCKING1_BCLK_MASK 0x01e +#define WM8900_REG_CLOCKING1_OPCLK_MASK 0x7000 #define WM8900_REG_CLOCKING2_ADC_CLKDIV 0xe0 #define WM8900_REG_CLOCKING2_DAC_CLKDIV 0x1c @@ -135,7 +135,7 @@ #define WM8900_REG_HPCTL1_HP_SHORT 0x08 #define WM8900_REG_HPCTL1_HP_SHORT2 0x04 -#define WM8900_LRC_MASK 0xfc00 +#define WM8900_LRC_MASK 0x03ff struct wm8900_priv { enum snd_soc_control_type control_type; @@ -742,26 +742,20 @@ static int wm8900_set_fll(struct snd_soc_codec *codec, { struct wm8900_priv *wm8900 = snd_soc_codec_get_drvdata(codec); struct _fll_div fll_div; - unsigned int reg; if (wm8900->fll_in == freq_in && wm8900->fll_out == freq_out) return 0; /* The digital side should be disabled during any change. */ - reg = snd_soc_read(codec, WM8900_REG_POWER1); - snd_soc_write(codec, WM8900_REG_POWER1, - reg & (~WM8900_REG_POWER1_FLL_ENA)); + snd_soc_update_bits(codec, WM8900_REG_POWER1, + WM8900_REG_POWER1_FLL_ENA, 0); /* Disable the FLL? */ if (!freq_in || !freq_out) { - reg = snd_soc_read(codec, WM8900_REG_CLOCKING1); - snd_soc_write(codec, WM8900_REG_CLOCKING1, - reg & (~WM8900_REG_CLOCKING1_MCLK_SRC)); - - reg = snd_soc_read(codec, WM8900_REG_FLLCTL1); - snd_soc_write(codec, WM8900_REG_FLLCTL1, - reg & (~WM8900_REG_FLLCTL1_OSC_ENA)); - + snd_soc_update_bits(codec, WM8900_REG_CLOCKING1, + WM8900_REG_CLOCKING1_MCLK_SRC, 0); + snd_soc_update_bits(codec, WM8900_REG_FLLCTL1, + WM8900_REG_FLLCTL1_OSC_ENA, 0); wm8900->fll_in = freq_in; wm8900->fll_out = freq_out; @@ -796,15 +790,14 @@ static int wm8900_set_fll(struct snd_soc_codec *codec, else snd_soc_write(codec, WM8900_REG_FLLCTL6, 0); - reg = snd_soc_read(codec, WM8900_REG_POWER1); - snd_soc_write(codec, WM8900_REG_POWER1, - reg | WM8900_REG_POWER1_FLL_ENA); + snd_soc_update_bits(codec, WM8900_REG_POWER1, + WM8900_REG_POWER1_FLL_ENA, + WM8900_REG_POWER1_FLL_ENA); reenable: - reg = snd_soc_read(codec, WM8900_REG_CLOCKING1); - snd_soc_write(codec, WM8900_REG_CLOCKING1, - reg | WM8900_REG_CLOCKING1_MCLK_SRC); - + snd_soc_update_bits(codec, WM8900_REG_CLOCKING1, + WM8900_REG_CLOCKING1_MCLK_SRC, + WM8900_REG_CLOCKING1_MCLK_SRC); return 0; } @@ -818,43 +811,35 @@ static int wm8900_set_dai_clkdiv(struct snd_soc_dai *codec_dai, int div_id, int div) { struct snd_soc_codec *codec = codec_dai->codec; - unsigned int reg; switch (div_id) { case WM8900_BCLK_DIV: - reg = snd_soc_read(codec, WM8900_REG_CLOCKING1); - snd_soc_write(codec, WM8900_REG_CLOCKING1, - div | (reg & WM8900_REG_CLOCKING1_BCLK_MASK)); + snd_soc_update_bits(codec, WM8900_REG_CLOCKING1, + WM8900_REG_CLOCKING1_BCLK_MASK, div); break; case WM8900_OPCLK_DIV: - reg = snd_soc_read(codec, WM8900_REG_CLOCKING1); - snd_soc_write(codec, WM8900_REG_CLOCKING1, - div | (reg & WM8900_REG_CLOCKING1_OPCLK_MASK)); + snd_soc_update_bits(codec, WM8900_REG_CLOCKING1, + WM8900_REG_CLOCKING1_OPCLK_MASK, div); break; case WM8900_DAC_LRCLK: - reg = snd_soc_read(codec, WM8900_REG_AUDIO4); - snd_soc_write(codec, WM8900_REG_AUDIO4, - div | (reg & WM8900_LRC_MASK)); + snd_soc_update_bits(codec, WM8900_REG_AUDIO4, + WM8900_LRC_MASK, div); break; case WM8900_ADC_LRCLK: - reg = snd_soc_read(codec, WM8900_REG_AUDIO3); - snd_soc_write(codec, WM8900_REG_AUDIO3, - div | (reg & WM8900_LRC_MASK)); + snd_soc_update_bits(codec, WM8900_REG_AUDIO3, + WM8900_LRC_MASK, div); break; case WM8900_DAC_CLKDIV: - reg = snd_soc_read(codec, WM8900_REG_CLOCKING2); - snd_soc_write(codec, WM8900_REG_CLOCKING2, - div | (reg & WM8900_REG_CLOCKING2_DAC_CLKDIV)); + snd_soc_update_bits(codec, WM8900_REG_CLOCKING2, + WM8900_REG_CLOCKING2_DAC_CLKDIV, div); break; case WM8900_ADC_CLKDIV: - reg = snd_soc_read(codec, WM8900_REG_CLOCKING2); - snd_soc_write(codec, WM8900_REG_CLOCKING2, - div | (reg & WM8900_REG_CLOCKING2_ADC_CLKDIV)); + snd_soc_update_bits(codec, WM8900_REG_CLOCKING2, + WM8900_REG_CLOCKING2_ADC_CLKDIV, div); break; case WM8900_LRCLK_MODE: - reg = snd_soc_read(codec, WM8900_REG_DACCTRL); - snd_soc_write(codec, WM8900_REG_DACCTRL, - div | (reg & WM8900_REG_DACCTRL_AIF_LRCLKRATE)); + snd_soc_update_bits(codec, WM8900_REG_DACCTRL, + WM8900_REG_DACCTRL_AIF_LRCLKRATE, div); break; default: return -EINVAL; @@ -1037,12 +1022,12 @@ static int wm8900_set_bias_level(struct snd_soc_codec *codec, switch (level) { case SND_SOC_BIAS_ON: /* Enable thermal shutdown */ - reg = snd_soc_read(codec, WM8900_REG_GPIO); - snd_soc_write(codec, WM8900_REG_GPIO, - reg | WM8900_REG_GPIO_TEMP_ENA); - reg = snd_soc_read(codec, WM8900_REG_ADDCTL); - snd_soc_write(codec, WM8900_REG_ADDCTL, - reg | WM8900_REG_ADDCTL_TEMP_SD); + snd_soc_update_bits(codec, WM8900_REG_GPIO, + WM8900_REG_GPIO_TEMP_ENA, + WM8900_REG_GPIO_TEMP_ENA); + snd_soc_update_bits(codec, WM8900_REG_ADDCTL, + WM8900_REG_ADDCTL_TEMP_SD, + WM8900_REG_ADDCTL_TEMP_SD); break; case SND_SOC_BIAS_PREPARE: @@ -1205,26 +1190,16 @@ static int wm8900_probe(struct snd_soc_codec *codec) wm8900_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* Latch the volume update bits */ - snd_soc_write(codec, WM8900_REG_LINVOL, - snd_soc_read(codec, WM8900_REG_LINVOL) | 0x100); - snd_soc_write(codec, WM8900_REG_RINVOL, - snd_soc_read(codec, WM8900_REG_RINVOL) | 0x100); - snd_soc_write(codec, WM8900_REG_LOUT1CTL, - snd_soc_read(codec, WM8900_REG_LOUT1CTL) | 0x100); - snd_soc_write(codec, WM8900_REG_ROUT1CTL, - snd_soc_read(codec, WM8900_REG_ROUT1CTL) | 0x100); - snd_soc_write(codec, WM8900_REG_LOUT2CTL, - snd_soc_read(codec, WM8900_REG_LOUT2CTL) | 0x100); - snd_soc_write(codec, WM8900_REG_ROUT2CTL, - snd_soc_read(codec, WM8900_REG_ROUT2CTL) | 0x100); - snd_soc_write(codec, WM8900_REG_LDAC_DV, - snd_soc_read(codec, WM8900_REG_LDAC_DV) | 0x100); - snd_soc_write(codec, WM8900_REG_RDAC_DV, - snd_soc_read(codec, WM8900_REG_RDAC_DV) | 0x100); - snd_soc_write(codec, WM8900_REG_LADC_DV, - snd_soc_read(codec, WM8900_REG_LADC_DV) | 0x100); - snd_soc_write(codec, WM8900_REG_RADC_DV, - snd_soc_read(codec, WM8900_REG_RADC_DV) | 0x100); + snd_soc_update_bits(codec, WM8900_REG_LINVOL, 0x100, 0x100); + snd_soc_update_bits(codec, WM8900_REG_RINVOL, 0x100, 0x100); + snd_soc_update_bits(codec, WM8900_REG_LOUT1CTL, 0x100, 0x100); + snd_soc_update_bits(codec, WM8900_REG_ROUT1CTL, 0x100, 0x100); + snd_soc_update_bits(codec, WM8900_REG_LOUT2CTL, 0x100, 0x100); + snd_soc_update_bits(codec, WM8900_REG_ROUT2CTL, 0x100, 0x100); + snd_soc_update_bits(codec, WM8900_REG_LDAC_DV, 0x100, 0x100); + snd_soc_update_bits(codec, WM8900_REG_RDAC_DV, 0x100, 0x100); + snd_soc_update_bits(codec, WM8900_REG_LADC_DV, 0x100, 0x100); + snd_soc_update_bits(codec, WM8900_REG_RADC_DV, 0x100, 0x100); /* Set the DAC and mixer output bias */ snd_soc_write(codec, WM8900_REG_OUTBIASCTL, 0x81); diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index b085575d4aa5..9fc8f4c0a9a9 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -50,7 +50,6 @@ static const char *wm8904_supply_names[WM8904_NUM_SUPPLIES] = { struct wm8904_priv { enum wm8904_type devtype; - void *control_data; struct regulator_bulk_data supplies[WM8904_NUM_SUPPLIES]; @@ -2540,7 +2539,6 @@ static __devinit int wm8904_i2c_probe(struct i2c_client *i2c, wm8904->devtype = id->driver_data; i2c_set_clientdata(i2c, wm8904); - wm8904->control_data = i2c; wm8904->pdata = i2c->dev.platform_data; ret = snd_soc_register_codec(&i2c->dev, diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index 056daa0010f9..dc5cb3150857 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -43,9 +43,19 @@ struct wm8940_priv { unsigned int sysclk; enum snd_soc_control_type control_type; - void *control_data; }; +static int wm8940_volatile_register(struct snd_soc_codec *codec, + unsigned int reg) +{ + switch (reg) { + case WM8940_SOFTRESET: + return 1; + default: + return 0; + } +} + static u16 wm8940_reg_defaults[] = { 0x8940, /* Soft Reset */ 0x0000, /* Power 1 */ @@ -460,6 +470,14 @@ static int wm8940_set_bias_level(struct snd_soc_codec *codec, ret = snd_soc_write(codec, WM8940_POWER1, pwr_reg | 0x1); break; case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + ret = snd_soc_cache_sync(codec); + if (ret < 0) { + dev_err(codec->dev, "Failed to sync cache: %d\n", ret); + return ret; + } + } + /* ensure bufioen and biasen */ pwr_reg |= (1 << 2) | (1 << 3); /* set vmid to 300k for standby */ @@ -470,6 +488,8 @@ static int wm8940_set_bias_level(struct snd_soc_codec *codec, break; } + codec->dapm.bias_level = level; + return ret; } @@ -660,30 +680,8 @@ static int wm8940_suspend(struct snd_soc_codec *codec, pm_message_t state) static int wm8940_resume(struct snd_soc_codec *codec) { - int i; - int ret; - u8 data[3]; - u16 *cache = codec->reg_cache; - - /* Sync reg_cache with the hardware - * Could use auto incremented writes to speed this up - */ - for (i = 0; i < ARRAY_SIZE(wm8940_reg_defaults); i++) { - data[0] = i; - data[1] = (cache[i] & 0xFF00) >> 8; - data[2] = cache[i] & 0x00FF; - ret = codec->hw_write(codec->control_data, data, 3); - if (ret < 0) - goto error_ret; - else if (ret != 3) { - ret = -EIO; - goto error_ret; - } - } - ret = wm8940_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - -error_ret: - return ret; + wm8940_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + return 0; } static int wm8940_probe(struct snd_soc_codec *codec) @@ -693,7 +691,6 @@ static int wm8940_probe(struct snd_soc_codec *codec) int ret; u16 reg; - codec->control_data = wm8940->control_data; ret = snd_soc_codec_set_cache_io(codec, 8, 16, wm8940->control_type); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); @@ -744,6 +741,7 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8940 = { .reg_cache_size = ARRAY_SIZE(wm8940_reg_defaults), .reg_word_size = sizeof(u16), .reg_cache_default = wm8940_reg_defaults, + .volatile_register = wm8940_volatile_register, }; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) @@ -758,7 +756,6 @@ static __devinit int wm8940_i2c_probe(struct i2c_client *i2c, return -ENOMEM; i2c_set_clientdata(i2c, wm8940); - wm8940->control_data = i2c; wm8940->control_type = SND_SOC_I2C; ret = snd_soc_register_codec(&i2c->dev, diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 4393394b7bc1..2df253c18568 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -72,7 +72,6 @@ static const u16 wm8960_reg[WM8960_CACHEREGNUM] = { struct wm8960_priv { enum snd_soc_control_type control_type; - void *control_data; int (*set_bias_level)(struct snd_soc_codec *, enum snd_soc_bias_level level); struct snd_soc_dapm_widget *lout1; @@ -575,6 +574,8 @@ static int wm8960_set_bias_level_out3(struct snd_soc_codec *codec, case SND_SOC_BIAS_STANDBY: if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + snd_soc_cache_sync(codec); + /* Enable anti-pop features */ snd_soc_write(codec, WM8960_APOP1, WM8960_POBCTRL | WM8960_SOFT_ST | @@ -677,6 +678,9 @@ static int wm8960_set_bias_level_capless(struct snd_soc_codec *codec, WM8960_VREF | WM8960_VMID_MASK, 0); break; + case SND_SOC_BIAS_OFF: + snd_soc_cache_sync(codec); + break; default: break; } @@ -902,16 +906,6 @@ static int wm8960_suspend(struct snd_soc_codec *codec, pm_message_t state) static int wm8960_resume(struct snd_soc_codec *codec) { struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); - int i; - u8 data[2]; - u16 *cache = codec->reg_cache; - - /* Sync reg_cache with the hardware */ - for (i = 0; i < ARRAY_SIZE(wm8960_reg); i++) { - data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001); - data[1] = cache[i] & 0x00ff; - codec->hw_write(codec->control_data, data, 2); - } wm8960->set_bias_level(codec, SND_SOC_BIAS_STANDBY); return 0; @@ -925,7 +919,6 @@ static int wm8960_probe(struct snd_soc_codec *codec) u16 reg; wm8960->set_bias_level = wm8960_set_bias_level_out3; - codec->control_data = wm8960->control_data; if (!pdata) { dev_warn(codec->dev, "No platform data supplied\n"); @@ -1015,7 +1008,6 @@ static __devinit int wm8960_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, wm8960); wm8960->control_type = SND_SOC_I2C; - wm8960->control_data = i2c; ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8960, &wm8960_dai, 1); diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c index cdee8103d09b..9568c8a49f96 100644 --- a/sound/soc/codecs/wm8961.c +++ b/sound/soc/codecs/wm8961.c @@ -974,7 +974,9 @@ static int wm8961_probe(struct snd_soc_codec *codec) } /* This isn't volatile - readback doesn't correspond to write */ - reg = codec->hw_read(codec, WM8961_RIGHT_INPUT_VOLUME); + codec->cache_bypass = 1; + reg = snd_soc_read(codec, WM8961_RIGHT_INPUT_VOLUME); + codec->cache_bypass = 0; dev_info(codec->dev, "WM8961 family %d revision %c\n", (reg & WM8961_DEVICE_ID_MASK) >> WM8961_DEVICE_ID_SHIFT, ((reg & WM8961_CHIP_REV_MASK) >> WM8961_CHIP_REV_SHIFT) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index d2c315fa1b9b..f60dfa16545e 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -63,6 +63,8 @@ struct wm8962_priv { int fll_fref; int fll_fout; + u16 dsp2_ena; + struct delayed_work mic_work; struct snd_soc_jack *jack; @@ -837,7 +839,7 @@ static const struct wm8962_reg_access { [40] = { 0x00FF, 0x01FF, 0x0000 }, /* R40 - SPKOUTL volume */ [41] = { 0x00FF, 0x01FF, 0x0000 }, /* R41 - SPKOUTR volume */ - [47] = { 0x000F, 0x0000, 0x0000 }, /* R47 - Thermal Shutdown Status */ + [47] = { 0x000F, 0x0000, 0xFFFF }, /* R47 - Thermal Shutdown Status */ [48] = { 0x7EC7, 0x7E07, 0xFFFF }, /* R48 - Additional Control (4) */ [49] = { 0x00D3, 0x00D7, 0xFFFF }, /* R49 - Class D Control 1 */ [51] = { 0x0047, 0x0047, 0x0000 }, /* R51 - Class D Control 2 */ @@ -965,7 +967,7 @@ static const struct wm8962_reg_access { [584] = { 0x002D, 0x002D, 0x0000 }, /* R584 - IRQ Debounce */ [586] = { 0xC000, 0xC000, 0x0000 }, /* R586 - MICINT Source Pol */ [768] = { 0x0001, 0x0001, 0x0000 }, /* R768 - DSP2 Power Management */ - [1037] = { 0x0000, 0x003F, 0x0000 }, /* R1037 - DSP2_ExecControl */ + [1037] = { 0x0000, 0x003F, 0xFFFF }, /* R1037 - DSP2_ExecControl */ [4096] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4096 - Write Sequencer 0 */ [4097] = { 0x00FF, 0x00FF, 0x0000 }, /* R4097 - Write Sequencer 1 */ [4098] = { 0x070F, 0x070F, 0x0000 }, /* R4098 - Write Sequencer 2 */ @@ -1986,6 +1988,122 @@ static const unsigned int classd_tlv[] = { }; static const DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); +static int wm8962_dsp2_write_config(struct snd_soc_codec *codec) +{ + return 0; +} + +static int wm8962_dsp2_set_enable(struct snd_soc_codec *codec, u16 val) +{ + u16 adcl = snd_soc_read(codec, WM8962_LEFT_ADC_VOLUME); + u16 adcr = snd_soc_read(codec, WM8962_RIGHT_ADC_VOLUME); + u16 dac = snd_soc_read(codec, WM8962_ADC_DAC_CONTROL_1); + + /* Mute the ADCs and DACs */ + snd_soc_write(codec, WM8962_LEFT_ADC_VOLUME, 0); + snd_soc_write(codec, WM8962_RIGHT_ADC_VOLUME, WM8962_ADC_VU); + snd_soc_update_bits(codec, WM8962_ADC_DAC_CONTROL_1, + WM8962_DAC_MUTE, WM8962_DAC_MUTE); + + snd_soc_write(codec, WM8962_SOUNDSTAGE_ENABLES_0, val); + + /* Restore the ADCs and DACs */ + snd_soc_write(codec, WM8962_LEFT_ADC_VOLUME, adcl); + snd_soc_write(codec, WM8962_RIGHT_ADC_VOLUME, adcr); + snd_soc_update_bits(codec, WM8962_ADC_DAC_CONTROL_1, + WM8962_DAC_MUTE, dac); + + return 0; +} + +static int wm8962_dsp2_start(struct snd_soc_codec *codec) +{ + struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); + + wm8962_dsp2_write_config(codec); + + snd_soc_write(codec, WM8962_DSP2_EXECCONTROL, WM8962_DSP2_RUNR); + + wm8962_dsp2_set_enable(codec, wm8962->dsp2_ena); + + return 0; +} + +static int wm8962_dsp2_stop(struct snd_soc_codec *codec) +{ + wm8962_dsp2_set_enable(codec, 0); + + snd_soc_write(codec, WM8962_DSP2_EXECCONTROL, WM8962_DSP2_STOP); + + return 0; +} + +#define WM8962_DSP2_ENABLE(xname, xshift) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = wm8962_dsp2_ena_info, \ + .get = wm8962_dsp2_ena_get, .put = wm8962_dsp2_ena_put, \ + .private_value = xshift } + +static int wm8962_dsp2_ena_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + + return 0; +} + +static int wm8962_dsp2_ena_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int shift = kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.integer.value[0] = !!(wm8962->dsp2_ena & 1 << shift); + + return 0; +} + +static int wm8962_dsp2_ena_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int shift = kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); + int old = wm8962->dsp2_ena; + int ret = 0; + int dsp2_running = snd_soc_read(codec, WM8962_DSP2_POWER_MANAGEMENT) & + WM8962_DSP2_ENA; + + mutex_lock(&codec->mutex); + + if (ucontrol->value.integer.value[0]) + wm8962->dsp2_ena |= 1 << shift; + else + wm8962->dsp2_ena &= ~(1 << shift); + + if (wm8962->dsp2_ena == old) + goto out; + + ret = 1; + + if (dsp2_running) { + if (wm8962->dsp2_ena) + wm8962_dsp2_set_enable(codec, wm8962->dsp2_ena); + else + wm8962_dsp2_stop(codec); + } + +out: + mutex_unlock(&codec->mutex); + + return ret; +} + /* The VU bits for the headphones are in a different register to the mute * bits and only take effect on the PGA if it is actually powered. */ @@ -2021,7 +2139,6 @@ static int wm8962_put_spk_sw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - u16 *reg_cache = codec->reg_cache; int ret; /* Apply the update (if any) */ @@ -2030,16 +2147,19 @@ static int wm8962_put_spk_sw(struct snd_kcontrol *kcontrol, return 0; /* If the left PGA is enabled hit that VU bit... */ - if (reg_cache[WM8962_PWR_MGMT_2] & WM8962_SPKOUTL_PGA_ENA) - return snd_soc_write(codec, WM8962_SPKOUTL_VOLUME, - reg_cache[WM8962_SPKOUTL_VOLUME]); + ret = snd_soc_read(codec, WM8962_PWR_MGMT_2); + if (ret & WM8962_SPKOUTL_PGA_ENA) { + snd_soc_write(codec, WM8962_SPKOUTL_VOLUME, + snd_soc_read(codec, WM8962_SPKOUTL_VOLUME)); + return 1; + } /* ...otherwise the right. The VU is stereo. */ - if (reg_cache[WM8962_PWR_MGMT_2] & WM8962_SPKOUTR_PGA_ENA) - return snd_soc_write(codec, WM8962_SPKOUTR_VOLUME, - reg_cache[WM8962_SPKOUTR_VOLUME]); + if (ret & WM8962_SPKOUTR_PGA_ENA) + snd_soc_write(codec, WM8962_SPKOUTR_VOLUME, + snd_soc_read(codec, WM8962_SPKOUTR_VOLUME)); - return 0; + return 1; } static const char *cap_hpf_mode_text[] = { @@ -2049,6 +2169,14 @@ static const char *cap_hpf_mode_text[] = { static const struct soc_enum cap_hpf_mode = SOC_ENUM_SINGLE(WM8962_ADC_DAC_CONTROL_2, 10, 2, cap_hpf_mode_text); + +static const char *cap_lhpf_mode_text[] = { + "LPF", "HPF" +}; + +static const struct soc_enum cap_lhpf_mode = + SOC_ENUM_SINGLE(WM8962_LHPF1, 1, 2, cap_lhpf_mode_text); + static const struct snd_kcontrol_new wm8962_snd_controls[] = { SOC_DOUBLE("Input Mixer Switch", WM8962_INPUT_MIXER_CONTROL_1, 3, 2, 1, 1), @@ -2077,6 +2205,8 @@ SOC_DOUBLE_R("Capture ZC Switch", WM8962_LEFT_INPUT_VOLUME, SOC_SINGLE("Capture HPF Switch", WM8962_ADC_DAC_CONTROL_1, 0, 1, 1), SOC_ENUM("Capture HPF Mode", cap_hpf_mode), SOC_SINGLE("Capture HPF Cutoff", WM8962_ADC_DAC_CONTROL_2, 7, 7, 0), +SOC_SINGLE("Capture LHPF Switch", WM8962_LHPF1, 0, 1, 0), +SOC_ENUM("Capture LHPF Mode", cap_lhpf_mode), SOC_DOUBLE_R_TLV("Sidetone Volume", WM8962_DAC_DSP_MIXING_1, WM8962_DAC_DSP_MIXING_2, 4, 12, 0, st_tlv), @@ -2134,6 +2264,11 @@ SOC_DOUBLE_R_TLV("EQ4 Volume", WM8962_EQ3, WM8962_EQ23, WM8962_EQL_B4_GAIN_SHIFT, 31, 0, eq_tlv), SOC_DOUBLE_R_TLV("EQ5 Volume", WM8962_EQ3, WM8962_EQ23, WM8962_EQL_B5_GAIN_SHIFT, 31, 0, eq_tlv), + +WM8962_DSP2_ENABLE("VSS Switch", WM8962_VSS_ENA_SHIFT), +WM8962_DSP2_ENABLE("HPF1 Switch", WM8962_HPF1_ENA_SHIFT), +WM8962_DSP2_ENABLE("HPF2 Switch", WM8962_HPF2_ENA_SHIFT), +WM8962_DSP2_ENABLE("HD Bass Switch", WM8962_HDBASS_ENA_SHIFT), }; static const struct snd_kcontrol_new wm8962_spk_mono_controls[] = { @@ -2365,7 +2500,6 @@ static int out_pga_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_codec *codec = w->codec; - u16 *reg_cache = codec->reg_cache; int reg; switch (w->shift) { @@ -2388,11 +2522,36 @@ static int out_pga_event(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_POST_PMU: - return snd_soc_write(codec, reg, reg_cache[reg]); + return snd_soc_write(codec, reg, snd_soc_read(codec, reg)); + default: + BUG(); + return -EINVAL; + } +} + +static int dsp2_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + if (wm8962->dsp2_ena) + wm8962_dsp2_start(codec); + break; + + case SND_SOC_DAPM_PRE_PMD: + if (wm8962->dsp2_ena) + wm8962_dsp2_stop(codec); + break; + default: BUG(); return -EINVAL; } + + return 0; } static const char *st_text[] = { "None", "Right", "Left" }; @@ -2509,7 +2668,7 @@ SND_SOC_DAPM_INPUT("IN4R"), SND_SOC_DAPM_INPUT("Beep"), SND_SOC_DAPM_INPUT("DMICDAT"), -SND_SOC_DAPM_MICBIAS("MICBIAS", WM8962_PWR_MGMT_1, 1, 0), +SND_SOC_DAPM_SUPPLY("MICBIAS", WM8962_PWR_MGMT_1, 1, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("Class G", WM8962_CHARGE_PUMP_B, 0, 1, NULL, 0), SND_SOC_DAPM_SUPPLY("SYSCLK", WM8962_CLOCKING2, 5, 0, sysclk_event, @@ -2517,6 +2676,9 @@ SND_SOC_DAPM_SUPPLY("SYSCLK", WM8962_CLOCKING2, 5, 0, sysclk_event, SND_SOC_DAPM_SUPPLY("Charge Pump", WM8962_CHARGE_PUMP_1, 0, 0, cp_event, SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_SUPPLY("TOCLK", WM8962_ADDITIONAL_CONTROL_1, 0, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY_S("DSP2", 1, WM8962_DSP2_POWER_MANAGEMENT, + WM8962_DSP2_ENA_SHIFT, 0, dsp2_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_MIXER("INPGAL", WM8962_LEFT_INPUT_PGA_CONTROL, 4, 0, inpgal, ARRAY_SIZE(inpgal)), @@ -2527,7 +2689,7 @@ SND_SOC_DAPM_MIXER("MIXINL", WM8962_PWR_MGMT_1, 5, 0, SND_SOC_DAPM_MIXER("MIXINR", WM8962_PWR_MGMT_1, 4, 0, mixinr, ARRAY_SIZE(mixinr)), -SND_SOC_DAPM_AIF_IN("DMIC", NULL, 0, WM8962_PWR_MGMT_1, 10, 0), +SND_SOC_DAPM_AIF_IN("DMIC_ENA", NULL, 0, WM8962_PWR_MGMT_1, 10, 0), SND_SOC_DAPM_ADC("ADCL", "Capture", WM8962_PWR_MGMT_1, 3, 0), SND_SOC_DAPM_ADC("ADCR", "Capture", WM8962_PWR_MGMT_1, 2, 0), @@ -2606,17 +2768,19 @@ static const struct snd_soc_dapm_route wm8962_intercon[] = { { "MICBIAS", NULL, "SYSCLK" }, - { "DMIC", NULL, "DMICDAT" }, + { "DMIC_ENA", NULL, "DMICDAT" }, { "ADCL", NULL, "SYSCLK" }, { "ADCL", NULL, "TOCLK" }, { "ADCL", NULL, "MIXINL" }, - { "ADCL", NULL, "DMIC" }, + { "ADCL", NULL, "DMIC_ENA" }, + { "ADCL", NULL, "DSP2" }, { "ADCR", NULL, "SYSCLK" }, { "ADCR", NULL, "TOCLK" }, { "ADCR", NULL, "MIXINR" }, - { "ADCR", NULL, "DMIC" }, + { "ADCR", NULL, "DMIC_ENA" }, + { "ADCR", NULL, "DSP2" }, { "STL", "Left", "ADCL" }, { "STL", "Right", "ADCR" }, @@ -2628,11 +2792,13 @@ static const struct snd_soc_dapm_route wm8962_intercon[] = { { "DACL", NULL, "TOCLK" }, { "DACL", NULL, "Beep" }, { "DACL", NULL, "STL" }, + { "DACL", NULL, "DSP2" }, { "DACR", NULL, "SYSCLK" }, { "DACR", NULL, "TOCLK" }, { "DACR", NULL, "Beep" }, { "DACR", NULL, "STR" }, + { "DACR", NULL, "DSP2" }, { "HPMIXL", "IN4L Switch", "IN4L" }, { "HPMIXL", "IN4R Switch", "IN4R" }, @@ -3058,9 +3224,9 @@ static int wm8962_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) int aif0 = 0; switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { - case SND_SOC_DAIFMT_DSP_A: - aif0 |= WM8962_LRCLK_INV; case SND_SOC_DAIFMT_DSP_B: + aif0 |= WM8962_LRCLK_INV | 3; + case SND_SOC_DAIFMT_DSP_A: aif0 |= 3; switch (fmt & SND_SOC_DAIFMT_INV_MASK) { @@ -3403,12 +3569,16 @@ static irqreturn_t wm8962_irq(int irq, void *data) struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); int mask; int active; + int reg; mask = snd_soc_read(codec, WM8962_INTERRUPT_STATUS_2_MASK); active = snd_soc_read(codec, WM8962_INTERRUPT_STATUS_2); active &= ~mask; + if (!active) + return IRQ_NONE; + /* Acknowledge the interrupts */ snd_soc_write(codec, WM8962_INTERRUPT_STATUS_2, active); @@ -3420,9 +3590,21 @@ static irqreturn_t wm8962_irq(int irq, void *data) if (active & WM8962_FIFOS_ERR_EINT) dev_err(codec->dev, "FIFO error\n"); - if (active & WM8962_TEMP_SHUT_EINT) + if (active & WM8962_TEMP_SHUT_EINT) { dev_crit(codec->dev, "Thermal shutdown\n"); + reg = snd_soc_read(codec, WM8962_THERMAL_SHUTDOWN_STATUS); + + if (reg & WM8962_TEMP_ERR_HP) + dev_crit(codec->dev, "Headphone thermal error\n"); + if (reg & WM8962_TEMP_WARN_HP) + dev_crit(codec->dev, "Headphone thermal warning\n"); + if (reg & WM8962_TEMP_ERR_SPK) + dev_crit(codec->dev, "Speaker thermal error\n"); + if (reg & WM8962_TEMP_WARN_SPK) + dev_crit(codec->dev, "Speaker thermal warning\n"); + } + if (active & (WM8962_MICSCD_EINT | WM8962_MICD_EINT)) { dev_dbg(codec->dev, "Microphone event detected\n"); diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index 572bb80627a4..b444b297d0b2 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -546,6 +546,9 @@ static int wm8971_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_PREPARE: break; case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) + snd_soc_cache_sync(codec); + /* mute dac and set vmid to 500k, enable VREF */ snd_soc_write(codec, WM8971_PWR1, pwr_reg | 0x0140); break; @@ -605,20 +608,8 @@ static int wm8971_suspend(struct snd_soc_codec *codec, pm_message_t state) static int wm8971_resume(struct snd_soc_codec *codec) { - int i; - u8 data[2]; - u16 *cache = codec->reg_cache; u16 reg; - /* Sync reg_cache with the hardware */ - for (i = 0; i < ARRAY_SIZE(wm8971_reg); i++) { - if (i + 1 == WM8971_RESET) - continue; - data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001); - data[1] = cache[i] & 0x00ff; - codec->hw_write(codec->control_data, data, 2); - } - wm8971_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* charge wm8971 caps */ @@ -660,25 +651,14 @@ static int wm8971_probe(struct snd_soc_codec *codec) msecs_to_jiffies(1000)); /* set the update bits */ - reg = snd_soc_read(codec, WM8971_LDAC); - snd_soc_write(codec, WM8971_LDAC, reg | 0x0100); - reg = snd_soc_read(codec, WM8971_RDAC); - snd_soc_write(codec, WM8971_RDAC, reg | 0x0100); - - reg = snd_soc_read(codec, WM8971_LOUT1V); - snd_soc_write(codec, WM8971_LOUT1V, reg | 0x0100); - reg = snd_soc_read(codec, WM8971_ROUT1V); - snd_soc_write(codec, WM8971_ROUT1V, reg | 0x0100); - - reg = snd_soc_read(codec, WM8971_LOUT2V); - snd_soc_write(codec, WM8971_LOUT2V, reg | 0x0100); - reg = snd_soc_read(codec, WM8971_ROUT2V); - snd_soc_write(codec, WM8971_ROUT2V, reg | 0x0100); - - reg = snd_soc_read(codec, WM8971_LINVOL); - snd_soc_write(codec, WM8971_LINVOL, reg | 0x0100); - reg = snd_soc_read(codec, WM8971_RINVOL); - snd_soc_write(codec, WM8971_RINVOL, reg | 0x0100); + snd_soc_update_bits(codec, WM8971_LDAC, 0x0100, 0x0100); + snd_soc_update_bits(codec, WM8971_RDAC, 0x0100, 0x0100); + snd_soc_update_bits(codec, WM8971_LOUT1V, 0x0100, 0x0100); + snd_soc_update_bits(codec, WM8971_ROUT1V, 0x0100, 0x0100); + snd_soc_update_bits(codec, WM8971_LOUT2V, 0x0100, 0x0100); + snd_soc_update_bits(codec, WM8971_ROUT2V, 0x0100, 0x0100); + snd_soc_update_bits(codec, WM8971_LINVOL, 0x0100, 0x0100); + snd_soc_update_bits(codec, WM8971_RINVOL, 0x0100, 0x0100); snd_soc_add_controls(codec, wm8971_snd_controls, ARRAY_SIZE(wm8971_snd_controls)); diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index ca646a822444..9352f1e088d2 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -3,7 +3,7 @@ * * Copyright 2006-2009 Wolfson Microelectronics PLC. * - * Author: Liam Girdwood <linux@wolfsonmicro.com> + * Author: Liam Girdwood <Liam.Girdwood@wolfsonmicro.com> * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License version 2 as @@ -530,6 +530,8 @@ static int wm8974_set_bias_level(struct snd_soc_codec *codec, power1 |= WM8974_POWER1_BIASEN | WM8974_POWER1_BUFIOEN; if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + snd_soc_cache_sync(codec); + /* Initial cap charge at VMID 5k */ snd_soc_write(codec, WM8974_POWER1, power1 | 0x3); mdelay(100); @@ -589,18 +591,7 @@ static int wm8974_suspend(struct snd_soc_codec *codec, pm_message_t state) static int wm8974_resume(struct snd_soc_codec *codec) { - int i; - u8 data[2]; - u16 *cache = codec->reg_cache; - - /* Sync reg_cache with the hardware */ - for (i = 0; i < ARRAY_SIZE(wm8974_reg); i++) { - data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001); - data[1] = cache[i] & 0x00ff; - codec->hw_write(codec->control_data, data, 2); - } wm8974_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - return 0; } diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index 85e3e630e763..41ca4d9ac20c 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -52,7 +52,6 @@ static const u16 wm8978_reg[WM8978_CACHEREGNUM] = { /* codec private data */ struct wm8978_priv { enum snd_soc_control_type control_type; - void *control_data; unsigned int f_pllout; unsigned int f_mclk; unsigned int f_256fs; @@ -955,7 +954,6 @@ static int wm8978_probe(struct snd_soc_codec *codec) * default hardware setting */ wm8978->sysclk = WM8978_PLL; - codec->control_data = wm8978->control_data; ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_I2C); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); @@ -1016,7 +1014,6 @@ static __devinit int wm8978_i2c_probe(struct i2c_client *i2c, return -ENOMEM; i2c_set_clientdata(i2c, wm8978); - wm8978->control_data = i2c; ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8978, &wm8978_dai, 1); diff --git a/sound/soc/codecs/wm8983.c b/sound/soc/codecs/wm8983.c index 17f04ec2b940..93ee28439be5 100644 --- a/sound/soc/codecs/wm8983.c +++ b/sound/soc/codecs/wm8983.c @@ -1007,7 +1007,7 @@ static int wm8983_probe(struct snd_soc_codec *codec) return ret; } - ret = snd_soc_write(codec, WM8983_SOFTWARE_RESET, 0x8983); + ret = snd_soc_write(codec, WM8983_SOFTWARE_RESET, 0); if (ret < 0) { dev_err(codec->dev, "Failed to issue reset: %d\n", ret); return ret; diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c index d7170f1381aa..2e9eba717d1a 100644 --- a/sound/soc/codecs/wm8988.c +++ b/sound/soc/codecs/wm8988.c @@ -55,7 +55,6 @@ struct wm8988_priv { struct snd_pcm_hw_constraint_list *sysclk_constraints; }; - #define wm8988_reset(c) snd_soc_write(c, WM8988_RESET, 0) /* @@ -676,6 +675,8 @@ static int wm8988_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_STANDBY: if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + snd_soc_cache_sync(codec); + /* VREF, VMID=2x5k */ snd_soc_write(codec, WM8988_PWR1, pwr_reg | 0x1c1); @@ -736,21 +737,7 @@ static int wm8988_suspend(struct snd_soc_codec *codec, pm_message_t state) static int wm8988_resume(struct snd_soc_codec *codec) { - int i; - u8 data[2]; - u16 *cache = codec->reg_cache; - - /* Sync reg_cache with the hardware */ - for (i = 0; i < WM8988_NUM_REG; i++) { - if (i == WM8988_RESET) - continue; - data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001); - data[1] = cache[i] & 0x00ff; - codec->hw_write(codec->control_data, data, 2); - } - wm8988_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - return 0; } @@ -759,7 +746,6 @@ static int wm8988_probe(struct snd_soc_codec *codec) struct wm8988_priv *wm8988 = snd_soc_codec_get_drvdata(codec); struct snd_soc_dapm_context *dapm = &codec->dapm; int ret = 0; - u16 reg; ret = snd_soc_codec_set_cache_io(codec, 7, 9, wm8988->control_type); if (ret < 0) { @@ -774,16 +760,11 @@ static int wm8988_probe(struct snd_soc_codec *codec) } /* set the update bits (we always update left then right) */ - reg = snd_soc_read(codec, WM8988_RADC); - snd_soc_write(codec, WM8988_RADC, reg | 0x100); - reg = snd_soc_read(codec, WM8988_RDAC); - snd_soc_write(codec, WM8988_RDAC, reg | 0x0100); - reg = snd_soc_read(codec, WM8988_ROUT1V); - snd_soc_write(codec, WM8988_ROUT1V, reg | 0x0100); - reg = snd_soc_read(codec, WM8988_ROUT2V); - snd_soc_write(codec, WM8988_ROUT2V, reg | 0x0100); - reg = snd_soc_read(codec, WM8988_RINVOL); - snd_soc_write(codec, WM8988_RINVOL, reg | 0x0100); + snd_soc_update_bits(codec, WM8988_RADC, 0x0100, 0x0100); + snd_soc_update_bits(codec, WM8988_RDAC, 0x0100, 0x0100); + snd_soc_update_bits(codec, WM8988_ROUT1V, 0x0100, 0x0100); + snd_soc_update_bits(codec, WM8988_ROUT2V, 0x0100, 0x0100); + snd_soc_update_bits(codec, WM8988_RINVOL, 0x0100, 0x0100); wm8988_set_bias_level(codec, SND_SOC_BIAS_STANDBY); diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 100aeee5ba96..d29a9622964c 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -36,10 +36,17 @@ struct wm8990_priv { unsigned int pcmclk; }; -/* - * wm8990 register cache. Note that register 0 is not included in the - * cache. - */ +static int wm8990_volatile_register(struct snd_soc_codec *codec, + unsigned int reg) +{ + switch (reg) { + case WM8990_RESET: + return 1; + default: + return 0; + } +} + static const u16 wm8990_reg[] = { 0x8990, /* R0 - Reset */ 0x0000, /* R1 - Power Management (1) */ @@ -394,7 +401,7 @@ static int inmixer_event(struct snd_soc_dapm_widget *w, (1 << WM8990_AINRMUX_PWR_BIT))) { reg |= WM8990_AINR_ENA; } else { - reg &= ~WM8990_AINL_ENA; + reg &= ~WM8990_AINR_ENA; } snd_soc_write(w->codec, WM8990_POWER_MANAGEMENT_2, reg); @@ -974,7 +981,6 @@ static void pll_factors(struct _pll_div *pll_div, unsigned int target, static int wm8990_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, int source, unsigned int freq_in, unsigned int freq_out) { - u16 reg; struct snd_soc_codec *codec = codec_dai->codec; struct _pll_div pll_div; @@ -982,13 +988,12 @@ static int wm8990_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, pll_factors(&pll_div, freq_out * 4, freq_in); /* Turn on PLL */ - reg = snd_soc_read(codec, WM8990_POWER_MANAGEMENT_2); - reg |= WM8990_PLL_ENA; - snd_soc_write(codec, WM8990_POWER_MANAGEMENT_2, reg); + snd_soc_update_bits(codec, WM8990_POWER_MANAGEMENT_2, + WM8990_PLL_ENA, WM8990_PLL_ENA); /* sysclk comes from PLL */ - reg = snd_soc_read(codec, WM8990_CLOCKING_2); - snd_soc_write(codec, WM8990_CLOCKING_2, reg | WM8990_SYSCLK_SRC); + snd_soc_update_bits(codec, WM8990_CLOCKING_2, + WM8990_SYSCLK_SRC, WM8990_SYSCLK_SRC); /* set up N , fractional mode and pre-divisor if necessary */ snd_soc_write(codec, WM8990_PLL1, pll_div.n | WM8990_SDM | @@ -996,10 +1001,9 @@ static int wm8990_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, snd_soc_write(codec, WM8990_PLL2, (u8)(pll_div.k>>8)); snd_soc_write(codec, WM8990_PLL3, (u8)(pll_div.k & 0xFF)); } else { - /* Turn on PLL */ - reg = snd_soc_read(codec, WM8990_POWER_MANAGEMENT_2); - reg &= ~WM8990_PLL_ENA; - snd_soc_write(codec, WM8990_POWER_MANAGEMENT_2, reg); + /* Turn off PLL */ + snd_soc_update_bits(codec, WM8990_POWER_MANAGEMENT_2, + WM8990_PLL_ENA, 0); } return 0; } @@ -1077,28 +1081,23 @@ static int wm8990_set_dai_clkdiv(struct snd_soc_dai *codec_dai, int div_id, int div) { struct snd_soc_codec *codec = codec_dai->codec; - u16 reg; switch (div_id) { case WM8990_MCLK_DIV: - reg = snd_soc_read(codec, WM8990_CLOCKING_2) & - ~WM8990_MCLK_DIV_MASK; - snd_soc_write(codec, WM8990_CLOCKING_2, reg | div); + snd_soc_update_bits(codec, WM8990_CLOCKING_2, + WM8990_MCLK_DIV_MASK, div); break; case WM8990_DACCLK_DIV: - reg = snd_soc_read(codec, WM8990_CLOCKING_2) & - ~WM8990_DAC_CLKDIV_MASK; - snd_soc_write(codec, WM8990_CLOCKING_2, reg | div); + snd_soc_update_bits(codec, WM8990_CLOCKING_2, + WM8990_DAC_CLKDIV_MASK, div); break; case WM8990_ADCCLK_DIV: - reg = snd_soc_read(codec, WM8990_CLOCKING_2) & - ~WM8990_ADC_CLKDIV_MASK; - snd_soc_write(codec, WM8990_CLOCKING_2, reg | div); + snd_soc_update_bits(codec, WM8990_CLOCKING_2, + WM8990_ADC_CLKDIV_MASK, div); break; case WM8990_BCLK_DIV: - reg = snd_soc_read(codec, WM8990_CLOCKING_1) & - ~WM8990_BCLK_DIV_MASK; - snd_soc_write(codec, WM8990_CLOCKING_1, reg | div); + snd_soc_update_bits(codec, WM8990_CLOCKING_1, + WM8990_BCLK_DIV_MASK, div); break; default: return -EINVAL; @@ -1156,7 +1155,7 @@ static int wm8990_mute(struct snd_soc_dai *dai, int mute) static int wm8990_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - u16 val; + int ret; switch (level) { case SND_SOC_BIAS_ON: @@ -1164,13 +1163,18 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_PREPARE: /* VMID=2*50k */ - val = snd_soc_read(codec, WM8990_POWER_MANAGEMENT_1) & - ~WM8990_VMID_MODE_MASK; - snd_soc_write(codec, WM8990_POWER_MANAGEMENT_1, val | 0x2); + snd_soc_update_bits(codec, WM8990_POWER_MANAGEMENT_1, + WM8990_VMID_MODE_MASK, 0x2); break; case SND_SOC_BIAS_STANDBY: if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + ret = snd_soc_cache_sync(codec); + if (ret < 0) { + dev_err(codec->dev, "Failed to sync cache: %d\n", ret); + return ret; + } + /* Enable all output discharge bits */ snd_soc_write(codec, WM8990_ANTIPOP1, WM8990_DIS_LLINE | WM8990_DIS_RLINE | WM8990_DIS_OUT3 | @@ -1225,9 +1229,8 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec, } /* VMID=2*250k */ - val = snd_soc_read(codec, WM8990_POWER_MANAGEMENT_1) & - ~WM8990_VMID_MODE_MASK; - snd_soc_write(codec, WM8990_POWER_MANAGEMENT_1, val | 0x4); + snd_soc_update_bits(codec, WM8990_POWER_MANAGEMENT_1, + WM8990_VMID_MODE_MASK, 0x4); break; case SND_SOC_BIAS_OFF: @@ -1241,8 +1244,8 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec, WM8990_BUFIOEN); /* mute DAC */ - val = snd_soc_read(codec, WM8990_DAC_CTRL); - snd_soc_write(codec, WM8990_DAC_CTRL, val | WM8990_DAC_MUTE); + snd_soc_update_bits(codec, WM8990_DAC_CTRL, + WM8990_DAC_MUTE, WM8990_DAC_MUTE); /* Enable any disabled outputs */ snd_soc_write(codec, WM8990_POWER_MANAGEMENT_1, 0x1f03); @@ -1319,19 +1322,6 @@ static int wm8990_suspend(struct snd_soc_codec *codec, pm_message_t state) static int wm8990_resume(struct snd_soc_codec *codec) { - int i; - u8 data[2]; - u16 *cache = codec->reg_cache; - - /* Sync reg_cache with the hardware */ - for (i = 0; i < ARRAY_SIZE(wm8990_reg); i++) { - if (i + 1 == WM8990_RESET) - continue; - data[0] = ((i + 1) << 1) | ((cache[i] >> 8) & 0x0001); - data[1] = cache[i] & 0x00ff; - codec->hw_write(codec->control_data, data, 2); - } - wm8990_set_bias_level(codec, SND_SOC_BIAS_STANDBY); return 0; } @@ -1343,7 +1333,6 @@ static int wm8990_resume(struct snd_soc_codec *codec) static int wm8990_probe(struct snd_soc_codec *codec) { int ret; - u16 reg; ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_I2C); if (ret < 0) { @@ -1356,15 +1345,14 @@ static int wm8990_probe(struct snd_soc_codec *codec) /* charge output caps */ wm8990_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - reg = snd_soc_read(codec, WM8990_AUDIO_INTERFACE_4); - snd_soc_write(codec, WM8990_AUDIO_INTERFACE_4, reg | WM8990_ALRCGPIO1); + snd_soc_update_bits(codec, WM8990_AUDIO_INTERFACE_4, + WM8990_ALRCGPIO1, WM8990_ALRCGPIO1); - reg = snd_soc_read(codec, WM8990_GPIO1_GPIO2) & - ~WM8990_GPIO1_SEL_MASK; - snd_soc_write(codec, WM8990_GPIO1_GPIO2, reg | 1); + snd_soc_update_bits(codec, WM8990_GPIO1_GPIO2, + WM8990_GPIO1_SEL_MASK, 1); - reg = snd_soc_read(codec, WM8990_POWER_MANAGEMENT_2); - snd_soc_write(codec, WM8990_POWER_MANAGEMENT_2, reg | WM8990_OPCLK_ENA); + snd_soc_update_bits(codec, WM8990_POWER_MANAGEMENT_2, + WM8990_OPCLK_ENA, WM8990_OPCLK_ENA); snd_soc_write(codec, WM8990_LEFT_OUTPUT_VOLUME, 0x50 | (1<<8)); snd_soc_write(codec, WM8990_RIGHT_OUTPUT_VOLUME, 0x50 | (1<<8)); @@ -1392,6 +1380,7 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8990 = { .reg_cache_size = ARRAY_SIZE(wm8990_reg), .reg_word_size = sizeof(u16), .reg_cache_default = wm8990_reg, + .volatile_register = wm8990_volatile_register, }; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c index 6af23d06870f..c9ab3ba9bced 100644 --- a/sound/soc/codecs/wm8991.c +++ b/sound/soc/codecs/wm8991.c @@ -3,7 +3,7 @@ * * Copyright 2007-2010 Wolfson Microelectronics PLC. * Author: Graeme Gregory - * linux@wolfsonmicro.com + * Graeme.Gregory@wolfsonmicro.com * * This program is free software; you can redistribute it and/or modify it * under the terms of the GNU General Public License as published by the @@ -393,7 +393,7 @@ static int inmixer_event(struct snd_soc_dapm_widget *w, (1 << WM8991_AINRMUX_PWR_BIT))) reg |= WM8991_AINR_ENA; else - reg &= ~WM8991_AINL_ENA; + reg &= ~WM8991_AINR_ENA; snd_soc_write(w->codec, WM8991_POWER_MANAGEMENT_2, reg); return 0; @@ -1264,7 +1264,6 @@ static int wm8991_probe(struct snd_soc_codec *codec) { struct wm8991_priv *wm8991; int ret; - unsigned int reg; wm8991 = snd_soc_codec_get_drvdata(codec); @@ -1282,19 +1281,18 @@ static int wm8991_probe(struct snd_soc_codec *codec) wm8991_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - reg = snd_soc_read(codec, WM8991_AUDIO_INTERFACE_4); - snd_soc_write(codec, WM8991_AUDIO_INTERFACE_4, reg | WM8991_ALRCGPIO1); + snd_soc_update_bits(codec, WM8991_AUDIO_INTERFACE_4, + WM8991_ALRCGPIO1, WM8991_ALRCGPIO1); - reg = snd_soc_read(codec, WM8991_GPIO1_GPIO2) & - ~WM8991_GPIO1_SEL_MASK; - snd_soc_write(codec, WM8991_GPIO1_GPIO2, reg | 1); + snd_soc_update_bits(codec, WM8991_GPIO1_GPIO2, + WM8991_GPIO1_SEL_MASK, 1); - reg = snd_soc_read(codec, WM8991_POWER_MANAGEMENT_1); - snd_soc_write(codec, WM8991_POWER_MANAGEMENT_1, reg | WM8991_VREF_ENA| - WM8991_VMID_MODE_MASK); + snd_soc_update_bits(codec, WM8991_POWER_MANAGEMENT_1, + WM8991_VREF_ENA | WM8991_VMID_MODE_MASK, + WM8991_VREF_ENA | WM8991_VMID_MODE_MASK); - reg = snd_soc_read(codec, WM8991_POWER_MANAGEMENT_2); - snd_soc_write(codec, WM8991_POWER_MANAGEMENT_2, reg | WM8991_OPCLK_ENA); + snd_soc_update_bits(codec, WM8991_POWER_MANAGEMENT_2, + WM8991_OPCLK_ENA, WM8991_OPCLK_ENA); snd_soc_write(codec, WM8991_DAC_CTRL, 0); snd_soc_write(codec, WM8991_LEFT_OUTPUT_VOLUME, 0x50 | (1<<8)); diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index 6e85b8869af7..eec8e1435116 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -847,6 +847,7 @@ SND_SOC_DAPM_SUPPLY("CLK_SYS", WM8993_BUS_CONTROL_1, 1, 0, clk_sys_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_SUPPLY("TOCLK", WM8993_CLOCKING_1, 14, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("CLK_DSP", WM8993_CLOCKING_3, 0, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("VMID", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_ADC("ADCL", NULL, WM8993_POWER_MANAGEMENT_2, 1, 0), SND_SOC_DAPM_ADC("ADCR", NULL, WM8993_POWER_MANAGEMENT_2, 0, 0), @@ -880,6 +881,9 @@ SND_SOC_DAPM_PGA("Direct Voice", SND_SOC_NOPM, 0, 0, NULL, 0), }; static const struct snd_soc_dapm_route routes[] = { + { "MICBIAS1", NULL, "VMID" }, + { "MICBIAS2", NULL, "VMID" }, + { "ADCL", NULL, "CLK_SYS" }, { "ADCL", NULL, "CLK_DSP" }, { "ADCR", NULL, "CLK_SYS" }, @@ -1433,7 +1437,8 @@ static int wm8993_probe(struct snd_soc_codec *codec) int ret, i, val; wm8993->hubs_data.hp_startup_mode = 1; - wm8993->hubs_data.dcs_codes = -2; + wm8993->hubs_data.dcs_codes_l = -2; + wm8993->hubs_data.dcs_codes_r = -2; wm8993->hubs_data.series_startup = 1; ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_I2C); diff --git a/sound/soc/codecs/wm8994-tables.c b/sound/soc/codecs/wm8994-tables.c index a87adbd05ee1..df5a8b9a250f 100644 --- a/sound/soc/codecs/wm8994-tables.c +++ b/sound/soc/codecs/wm8994-tables.c @@ -1073,8 +1073,8 @@ const struct wm8994_access_mask wm8994_access_masks[WM8994_CACHE_SIZE] = { { 0x0000, 0x0000 }, /* R1069 */ { 0x0000, 0x0000 }, /* R1070 */ { 0x0000, 0x0000 }, /* R1071 */ - { 0x0000, 0x0000 }, /* R1072 */ - { 0x0000, 0x0000 }, /* R1073 */ + { 0x006F, 0x006F }, /* R1072 - AIF1 DAC1 Noise Gate */ + { 0x006F, 0x006F }, /* R1073 - AIF1 DAC2 Noise Gate */ { 0x0000, 0x0000 }, /* R1074 */ { 0x0000, 0x0000 }, /* R1075 */ { 0x0000, 0x0000 }, /* R1076 */ @@ -1329,7 +1329,7 @@ const struct wm8994_access_mask wm8994_access_masks[WM8994_CACHE_SIZE] = { { 0x0000, 0x0000 }, /* R1325 */ { 0x0000, 0x0000 }, /* R1326 */ { 0x0000, 0x0000 }, /* R1327 */ - { 0x0000, 0x0000 }, /* R1328 */ + { 0x006F, 0x006F }, /* R1328 - AIF2 DAC Noise Gate */ { 0x0000, 0x0000 }, /* R1329 */ { 0x0000, 0x0000 }, /* R1330 */ { 0x0000, 0x0000 }, /* R1331 */ @@ -1635,8 +1635,8 @@ const u16 wm8994_reg_defaults[WM8994_CACHE_SIZE] = { 0x0000, /* R58 - MICBIAS */ 0x000D, /* R59 - LDO 1 */ 0x0003, /* R60 - LDO 2 */ - 0x0000, /* R61 */ - 0x0000, /* R62 */ + 0x0039, /* R61 - MICBIAS1 */ + 0x0039, /* R62 - MICBIAS2 */ 0x0000, /* R63 */ 0x0000, /* R64 */ 0x0000, /* R65 */ @@ -2646,8 +2646,8 @@ const u16 wm8994_reg_defaults[WM8994_CACHE_SIZE] = { 0x0000, /* R1069 */ 0x0000, /* R1070 */ 0x0000, /* R1071 */ - 0x0000, /* R1072 */ - 0x0000, /* R1073 */ + 0x0068, /* R1072 - AIF1 DAC1 Noise Gate */ + 0x0068, /* R1073 - AIF1 DAC2 Noise Gate */ 0x0000, /* R1074 */ 0x0000, /* R1075 */ 0x0000, /* R1076 */ @@ -2902,7 +2902,7 @@ const u16 wm8994_reg_defaults[WM8994_CACHE_SIZE] = { 0x0000, /* R1325 */ 0x0000, /* R1326 */ 0x0000, /* R1327 */ - 0x0000, /* R1328 */ + 0x0068, /* R1328 - AIF2 DAC Noise Gate */ 0x0000, /* R1329 */ 0x0000, /* R1330 */ 0x0000, /* R1331 */ diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index b393f9fac97a..6b73efd26991 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -107,6 +107,7 @@ static int wm8994_volatile(struct snd_soc_codec *codec, unsigned int reg) case WM8994_LDO_2: case WM8958_DSP2_EXECCONTROL: case WM8958_MIC_DETECT_3: + case WM8994_DC_SERVO_4E: return 1; default: return 0; @@ -207,7 +208,7 @@ static int configure_aif_clock(struct snd_soc_codec *codec, int aif) static int configure_clock(struct snd_soc_codec *codec) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); - int old, new; + int change, new; /* Bring up the AIF clocks first */ configure_aif_clock(codec, 0); @@ -228,14 +229,11 @@ static int configure_clock(struct snd_soc_codec *codec) else new = 0; - old = snd_soc_read(codec, WM8994_CLOCKING_1) & WM8994_SYSCLK_SRC; - - /* If there's no change then we're done. */ - if (old == new) + change = snd_soc_update_bits(codec, WM8994_CLOCKING_1, + WM8994_SYSCLK_SRC, new); + if (!change) return 0; - snd_soc_update_bits(codec, WM8994_CLOCKING_1, WM8994_SYSCLK_SRC, new); - snd_soc_dapm_sync(&codec->dapm); return 0; @@ -281,6 +279,8 @@ static const DECLARE_TLV_DB_SCALE(digital_tlv, -7200, 75, 1); static const DECLARE_TLV_DB_SCALE(st_tlv, -3600, 300, 0); static const DECLARE_TLV_DB_SCALE(wm8994_3d_tlv, -1600, 183, 0); static const DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); +static const DECLARE_TLV_DB_SCALE(ng_tlv, -10200, 600, 0); +static const DECLARE_TLV_DB_SCALE(mixin_boost_tlv, 0, 900, 0); #define WM8994_DRC_SWITCH(xname, reg, shift) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ @@ -660,8 +660,52 @@ SOC_SINGLE_TLV("AIF2 EQ5 Volume", WM8994_AIF2_EQ_GAINS_2, 6, 31, 0, eq_tlv), }; +static const char *wm8958_ng_text[] = { + "30ms", "125ms", "250ms", "500ms", +}; + +static const struct soc_enum wm8958_aif1dac1_ng_hold = + SOC_ENUM_SINGLE(WM8958_AIF1_DAC1_NOISE_GATE, + WM8958_AIF1DAC1_NG_THR_SHIFT, 4, wm8958_ng_text); + +static const struct soc_enum wm8958_aif1dac2_ng_hold = + SOC_ENUM_SINGLE(WM8958_AIF1_DAC2_NOISE_GATE, + WM8958_AIF1DAC2_NG_THR_SHIFT, 4, wm8958_ng_text); + +static const struct soc_enum wm8958_aif2dac_ng_hold = + SOC_ENUM_SINGLE(WM8958_AIF2_DAC_NOISE_GATE, + WM8958_AIF2DAC_NG_THR_SHIFT, 4, wm8958_ng_text); + static const struct snd_kcontrol_new wm8958_snd_controls[] = { SOC_SINGLE_TLV("AIF3 Boost Volume", WM8958_AIF3_CONTROL_2, 10, 3, 0, aif_tlv), + +SOC_SINGLE("AIF1DAC1 Noise Gate Switch", WM8958_AIF1_DAC1_NOISE_GATE, + WM8958_AIF1DAC1_NG_ENA_SHIFT, 1, 0), +SOC_ENUM("AIF1DAC1 Noise Gate Hold Time", wm8958_aif1dac1_ng_hold), +SOC_SINGLE_TLV("AIF1DAC1 Noise Gate Threshold Volume", + WM8958_AIF1_DAC1_NOISE_GATE, WM8958_AIF1DAC1_NG_THR_SHIFT, + 7, 1, ng_tlv), + +SOC_SINGLE("AIF1DAC2 Noise Gate Switch", WM8958_AIF1_DAC2_NOISE_GATE, + WM8958_AIF1DAC2_NG_ENA_SHIFT, 1, 0), +SOC_ENUM("AIF1DAC2 Noise Gate Hold Time", wm8958_aif1dac2_ng_hold), +SOC_SINGLE_TLV("AIF1DAC2 Noise Gate Threshold Volume", + WM8958_AIF1_DAC2_NOISE_GATE, WM8958_AIF1DAC2_NG_THR_SHIFT, + 7, 1, ng_tlv), + +SOC_SINGLE("AIF2DAC Noise Gate Switch", WM8958_AIF2_DAC_NOISE_GATE, + WM8958_AIF2DAC_NG_ENA_SHIFT, 1, 0), +SOC_ENUM("AIF2DAC Noise Gate Hold Time", wm8958_aif2dac_ng_hold), +SOC_SINGLE_TLV("AIF2DAC Noise Gate Threshold Volume", + WM8958_AIF2_DAC_NOISE_GATE, WM8958_AIF2DAC_NG_THR_SHIFT, + 7, 1, ng_tlv), +}; + +static const struct snd_kcontrol_new wm1811_snd_controls[] = { +SOC_SINGLE_TLV("MIXINL IN1LP Boost Volume", WM8994_INPUT_MIXER_1, 7, 1, 0, + mixin_boost_tlv), +SOC_SINGLE_TLV("MIXINL IN1RP Boost Volume", WM8994_INPUT_MIXER_1, 8, 1, 0, + mixin_boost_tlv), }; static int clk_sys_event(struct snd_soc_dapm_widget *w, @@ -681,6 +725,97 @@ static int clk_sys_event(struct snd_soc_dapm_widget *w, return 0; } +static void vmid_reference(struct snd_soc_codec *codec) +{ + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + + wm8994->vmid_refcount++; + + dev_dbg(codec->dev, "Referencing VMID, refcount is now %d\n", + wm8994->vmid_refcount); + + if (wm8994->vmid_refcount == 1) { + /* Startup bias, VMID ramp & buffer */ + snd_soc_update_bits(codec, WM8994_ANTIPOP_2, + WM8994_STARTUP_BIAS_ENA | + WM8994_VMID_BUF_ENA | + WM8994_VMID_RAMP_MASK, + WM8994_STARTUP_BIAS_ENA | + WM8994_VMID_BUF_ENA | + (0x11 << WM8994_VMID_RAMP_SHIFT)); + + /* Main bias enable, VMID=2x40k */ + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_1, + WM8994_BIAS_ENA | + WM8994_VMID_SEL_MASK, + WM8994_BIAS_ENA | 0x2); + + msleep(20); + } +} + +static void vmid_dereference(struct snd_soc_codec *codec) +{ + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + + wm8994->vmid_refcount--; + + dev_dbg(codec->dev, "Dereferencing VMID, refcount is now %d\n", + wm8994->vmid_refcount); + + if (wm8994->vmid_refcount == 0) { + /* Switch over to startup biases */ + snd_soc_update_bits(codec, WM8994_ANTIPOP_2, + WM8994_BIAS_SRC | + WM8994_STARTUP_BIAS_ENA | + WM8994_VMID_BUF_ENA | + WM8994_VMID_RAMP_MASK, + WM8994_BIAS_SRC | + WM8994_STARTUP_BIAS_ENA | + WM8994_VMID_BUF_ENA | + (1 << WM8994_VMID_RAMP_SHIFT)); + + /* Disable main biases */ + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_1, + WM8994_BIAS_ENA | + WM8994_VMID_SEL_MASK, 0); + + /* Discharge line */ + snd_soc_update_bits(codec, WM8994_ANTIPOP_1, + WM8994_LINEOUT1_DISCH | + WM8994_LINEOUT2_DISCH, + WM8994_LINEOUT1_DISCH | + WM8994_LINEOUT2_DISCH); + + msleep(5); + + /* Switch off startup biases */ + snd_soc_update_bits(codec, WM8994_ANTIPOP_2, + WM8994_BIAS_SRC | + WM8994_STARTUP_BIAS_ENA | + WM8994_VMID_BUF_ENA | + WM8994_VMID_RAMP_MASK, 0); + } +} + +static int vmid_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + vmid_reference(codec); + break; + + case SND_SOC_DAPM_POST_PMD: + vmid_dereference(codec); + break; + } + + return 0; +} + static void wm8994_update_class_w(struct snd_soc_codec *codec) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); @@ -1208,6 +1343,8 @@ SND_SOC_DAPM_INPUT("Clock"), SND_SOC_DAPM_SUPPLY_S("MICBIAS Supply", 1, SND_SOC_NOPM, 0, 0, micbias_ev, SND_SOC_DAPM_PRE_PMU), +SND_SOC_DAPM_SUPPLY("VMID", SND_SOC_NOPM, 0, 0, vmid_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_SUPPLY("CLK_SYS", SND_SOC_NOPM, 0, 0, clk_sys_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), @@ -1282,7 +1419,7 @@ SND_SOC_DAPM_MUX("AIF2DAC Mux", SND_SOC_NOPM, 0, 0, &aif2dac_mux), SND_SOC_DAPM_MUX("AIF2ADC Mux", SND_SOC_NOPM, 0, 0, &aif2adc_mux), SND_SOC_DAPM_AIF_IN("AIF3DACDAT", "AIF3 Playback", 0, SND_SOC_NOPM, 0, 0), -SND_SOC_DAPM_AIF_IN("AIF3ADCDAT", "AIF3 Capture", 0, SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_AIF_OUT("AIF3ADCDAT", "AIF3 Capture", 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_SUPPLY("TOCLK", WM8994_CLOCKING_1, 4, 0, NULL, 0), @@ -1525,6 +1662,8 @@ static const struct snd_soc_dapm_route wm8994_revd_intercon[] = { static const struct snd_soc_dapm_route wm8994_intercon[] = { { "AIF2DACL", NULL, "AIF2DAC Mux" }, { "AIF2DACR", NULL, "AIF2DAC Mux" }, + { "MICBIAS1", NULL, "VMID" }, + { "MICBIAS2", NULL, "VMID" }, }; static const struct snd_soc_dapm_route wm8958_intercon[] = { @@ -1629,10 +1768,12 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src, unsigned int freq_in, unsigned int freq_out) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + struct wm8994 *control = codec->control_data; int reg_offset, ret; struct fll_div fll; u16 reg, aif1, aif2; unsigned long timeout; + bool was_enabled; aif1 = snd_soc_read(codec, WM8994_AIF1_CLOCKING_1) & WM8994_AIF1CLK_ENA; @@ -1653,6 +1794,9 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src, return -EINVAL; } + reg = snd_soc_read(codec, WM8994_FLL1_CONTROL_1 + reg_offset); + was_enabled = reg & WM8994_FLL1_ENA; + switch (src) { case 0: /* Allow no source specification when stopping */ @@ -1719,6 +1863,21 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src, /* Enable (with fractional mode if required) */ if (freq_out) { + /* Enable VMID if we need it */ + if (!was_enabled) { + switch (control->type) { + case WM8994: + vmid_reference(codec); + break; + case WM8958: + if (wm8994->revision < 1) + vmid_reference(codec); + break; + default: + break; + } + } + if (fll.k) reg = WM8994_FLL1_ENA | WM8994_FLL1_FRAC; else @@ -1736,6 +1895,20 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src, } else { msleep(5); } + } else { + if (was_enabled) { + switch (control->type) { + case WM8994: + vmid_dereference(codec); + break; + case WM8958: + if (wm8994->revision < 1) + vmid_dereference(codec); + break; + default: + break; + } + } } wm8994->fll[id].in = freq_in; @@ -1852,9 +2025,6 @@ static int wm8994_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_PREPARE: - /* VMID=2x40k */ - snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_1, - WM8994_VMID_SEL_MASK, 0x2); break; case SND_SOC_BIAS_STANDBY: @@ -1888,6 +2058,15 @@ static int wm8994_set_bias_level(struct snd_soc_codec *codec, WM8958_CP_DISCH); } break; + + case WM1811: + if (wm8994->revision < 2) { + snd_soc_write(codec, 0x102, 0x3); + snd_soc_write(codec, 0x5d, 0x7e); + snd_soc_write(codec, 0x5e, 0x0); + snd_soc_write(codec, 0x102, 0x0); + } + break; } /* Discharge LINEOUT1 & 2 */ @@ -1896,65 +2075,13 @@ static int wm8994_set_bias_level(struct snd_soc_codec *codec, WM8994_LINEOUT2_DISCH, WM8994_LINEOUT1_DISCH | WM8994_LINEOUT2_DISCH); - - /* Startup bias, VMID ramp & buffer */ - snd_soc_update_bits(codec, WM8994_ANTIPOP_2, - WM8994_STARTUP_BIAS_ENA | - WM8994_VMID_BUF_ENA | - WM8994_VMID_RAMP_MASK, - WM8994_STARTUP_BIAS_ENA | - WM8994_VMID_BUF_ENA | - (0x11 << WM8994_VMID_RAMP_SHIFT)); - - /* Main bias enable, VMID=2x40k */ - snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_1, - WM8994_BIAS_ENA | - WM8994_VMID_SEL_MASK, - WM8994_BIAS_ENA | 0x2); - - msleep(20); } - /* VMID=2x500k */ - snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_1, - WM8994_VMID_SEL_MASK, 0x4); break; case SND_SOC_BIAS_OFF: if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY) { - /* Switch over to startup biases */ - snd_soc_update_bits(codec, WM8994_ANTIPOP_2, - WM8994_BIAS_SRC | - WM8994_STARTUP_BIAS_ENA | - WM8994_VMID_BUF_ENA | - WM8994_VMID_RAMP_MASK, - WM8994_BIAS_SRC | - WM8994_STARTUP_BIAS_ENA | - WM8994_VMID_BUF_ENA | - (1 << WM8994_VMID_RAMP_SHIFT)); - - /* Disable main biases */ - snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_1, - WM8994_BIAS_ENA | - WM8994_VMID_SEL_MASK, 0); - - /* Discharge line */ - snd_soc_update_bits(codec, WM8994_ANTIPOP_1, - WM8994_LINEOUT1_DISCH | - WM8994_LINEOUT2_DISCH, - WM8994_LINEOUT1_DISCH | - WM8994_LINEOUT2_DISCH); - - msleep(5); - - /* Switch off startup biases */ - snd_soc_update_bits(codec, WM8994_ANTIPOP_2, - WM8994_BIAS_SRC | - WM8994_STARTUP_BIAS_ENA | - WM8994_VMID_BUF_ENA | - WM8994_VMID_RAMP_MASK, 0); - wm8994->cur_fw = NULL; pm_runtime_put(codec->dev); @@ -2055,10 +2182,18 @@ static int wm8994_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) /* The AIF2 format configuration needs to be mirrored to AIF3 * on WM8958 if it's in use so just do it all the time. */ - if (control->type == WM8958 && dai->id == 2) - snd_soc_update_bits(codec, WM8958_AIF3_CONTROL_1, - WM8994_AIF1_LRCLK_INV | - WM8958_AIF3_FMT_MASK, aif1); + switch (control->type) { + case WM1811: + case WM8958: + if (dai->id == 2) + snd_soc_update_bits(codec, WM8958_AIF3_CONTROL_1, + WM8994_AIF1_LRCLK_INV | + WM8958_AIF3_FMT_MASK, aif1); + break; + + default: + break; + } snd_soc_update_bits(codec, aif1_reg, WM8994_AIF1_BCLK_INV | WM8994_AIF1_LRCLK_INV | @@ -2100,7 +2235,6 @@ static int wm8994_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_codec *codec = dai->codec; - struct wm8994 *control = codec->control_data; struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); int aif1_reg; int aif2_reg; @@ -2143,14 +2277,6 @@ static int wm8994_hw_params(struct snd_pcm_substream *substream, dev_dbg(codec->dev, "AIF2 using split LRCLK\n"); } break; - case 3: - switch (control->type) { - case WM8958: - aif1_reg = WM8958_AIF3_CONTROL_1; - break; - default: - return 0; - } default: return -EINVAL; } @@ -2271,6 +2397,7 @@ static int wm8994_aif3_hw_params(struct snd_pcm_substream *substream, switch (dai->id) { case 3: switch (control->type) { + case WM1811: case WM8958: aif1_reg = WM8958_AIF3_CONTROL_1; break; @@ -2311,7 +2438,7 @@ static void wm8994_aif_shutdown(struct snd_pcm_substream *substream, rate_reg = WM8994_AIF1_RATE; break; case 2: - rate_reg = WM8994_AIF1_RATE; + rate_reg = WM8994_AIF2_RATE; break; default: break; @@ -2384,6 +2511,21 @@ static int wm8994_set_tristate(struct snd_soc_dai *codec_dai, int tristate) return snd_soc_update_bits(codec, reg, mask, val); } +static int wm8994_aif2_probe(struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + + /* Disable the pulls on the AIF if we're using it to save power. */ + snd_soc_update_bits(codec, WM8994_GPIO_3, + WM8994_GPN_PU | WM8994_GPN_PD, 0); + snd_soc_update_bits(codec, WM8994_GPIO_4, + WM8994_GPN_PU | WM8994_GPN_PD, 0); + snd_soc_update_bits(codec, WM8994_GPIO_5, + WM8994_GPN_PU | WM8994_GPN_PD, 0); + + return 0; +} + #define WM8994_RATES SNDRV_PCM_RATE_8000_96000 #define WM8994_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ @@ -2451,6 +2593,7 @@ static struct snd_soc_dai_driver wm8994_dai[] = { .rates = WM8994_RATES, .formats = WM8994_FORMATS, }, + .probe = wm8994_aif2_probe, .ops = &wm8994_aif2_dai_ops, }, { @@ -2485,6 +2628,7 @@ static int wm8994_suspend(struct snd_soc_codec *codec, pm_message_t state) case WM8994: snd_soc_update_bits(codec, WM8994_MICBIAS, WM8994_MICD_ENA, 0); break; + case WM1811: case WM8958: snd_soc_update_bits(codec, WM8958_MIC_DETECT_1, WM8958_MICD_ENA, 0); @@ -2553,6 +2697,7 @@ static int wm8994_resume(struct snd_soc_codec *codec) snd_soc_update_bits(codec, WM8994_MICBIAS, WM8994_MICD_ENA, WM8994_MICD_ENA); break; + case WM1811: case WM8958: if (wm8994->jack_cb) snd_soc_update_bits(codec, WM8958_MIC_DETECT_1, @@ -2851,8 +2996,13 @@ int wm8958_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack, struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); struct wm8994 *control = codec->control_data; - if (control->type != WM8958) + switch (control->type) { + case WM1811: + case WM8958: + break; + default: return -EINVAL; + } if (jack) { if (!cb) { @@ -2916,6 +3066,24 @@ static irqreturn_t wm8994_fifo_error(int irq, void *data) return IRQ_HANDLED; } +static irqreturn_t wm8994_temp_warn(int irq, void *data) +{ + struct snd_soc_codec *codec = data; + + dev_err(codec->dev, "Thermal warning\n"); + + return IRQ_HANDLED; +} + +static irqreturn_t wm8994_temp_shut(int irq, void *data) +{ + struct snd_soc_codec *codec = data; + + dev_crit(codec->dev, "Thermal shutdown\n"); + + return IRQ_HANDLED; +} + static int wm8994_codec_probe(struct snd_soc_codec *codec) { struct wm8994 *control; @@ -2972,13 +3140,14 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) switch (wm8994->revision) { case 2: case 3: - wm8994->hubs.dcs_codes = -5; + wm8994->hubs.dcs_codes_l = -5; + wm8994->hubs.dcs_codes_r = -5; wm8994->hubs.hp_startup_mode = 1; wm8994->hubs.dcs_readback_mode = 1; wm8994->hubs.series_startup = 1; break; default: - wm8994->hubs.dcs_readback_mode = 1; + wm8994->hubs.dcs_readback_mode = 2; break; } break; @@ -2987,12 +3156,34 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) wm8994->hubs.dcs_readback_mode = 1; break; + case WM1811: + wm8994->hubs.dcs_readback_mode = 2; + wm8994->hubs.no_series_update = 1; + + switch (wm8994->revision) { + case 0: + case 1: + wm8994->hubs.dcs_codes_l = -9; + wm8994->hubs.dcs_codes_r = -5; + break; + default: + break; + } + + snd_soc_update_bits(codec, WM8994_ANALOGUE_HP_1, + WM1811_HPOUT1_ATTN, WM1811_HPOUT1_ATTN); + break; + default: break; } wm8994_request_irq(codec->control_data, WM8994_IRQ_FIFOS_ERR, wm8994_fifo_error, "FIFO error", codec); + wm8994_request_irq(wm8994->control_data, WM8994_IRQ_TEMP_WARN, + wm8994_temp_warn, "Thermal warning", codec); + wm8994_request_irq(wm8994->control_data, WM8994_IRQ_TEMP_SHUT, + wm8994_temp_shut, "Thermal shutdown", codec); ret = wm8994_request_irq(codec->control_data, WM8994_IRQ_DCS_DONE, wm_hubs_dcs_done, "DC servo done", @@ -3043,6 +3234,7 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) break; case WM8958: + case WM1811: if (wm8994->micdet_irq) { ret = request_threaded_irq(wm8994->micdet_irq, NULL, wm8958_mic_irq, @@ -3205,6 +3397,19 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) ARRAY_SIZE(wm8994_dac_widgets)); } break; + + case WM1811: + snd_soc_add_controls(codec, wm8958_snd_controls, + ARRAY_SIZE(wm8958_snd_controls)); + snd_soc_dapm_new_controls(dapm, wm8958_dapm_widgets, + ARRAY_SIZE(wm8958_dapm_widgets)); + snd_soc_dapm_new_controls(dapm, wm8994_lateclk_widgets, + ARRAY_SIZE(wm8994_lateclk_widgets)); + snd_soc_dapm_new_controls(dapm, wm8994_adc_widgets, + ARRAY_SIZE(wm8994_adc_widgets)); + snd_soc_dapm_new_controls(dapm, wm8994_dac_widgets, + ARRAY_SIZE(wm8994_dac_widgets)); + break; } @@ -3241,6 +3446,12 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) wm8958_dsp2_init(codec); break; + case WM1811: + snd_soc_dapm_add_routes(dapm, wm8994_lateclk_intercon, + ARRAY_SIZE(wm8994_lateclk_intercon)); + snd_soc_dapm_add_routes(dapm, wm8958_intercon, + ARRAY_SIZE(wm8958_intercon)); + break; } return 0; @@ -3257,6 +3468,8 @@ err_irq: wm8994_free_irq(codec->control_data, WM8994_IRQ_DCS_DONE, &wm8994->hubs); wm8994_free_irq(codec->control_data, WM8994_IRQ_FIFOS_ERR, codec); + wm8994_free_irq(codec->control_data, WM8994_IRQ_TEMP_SHUT, codec); + wm8994_free_irq(codec->control_data, WM8994_IRQ_TEMP_WARN, codec); err: kfree(wm8994); return ret; @@ -3279,6 +3492,8 @@ static int wm8994_codec_remove(struct snd_soc_codec *codec) wm8994_free_irq(codec->control_data, WM8994_IRQ_DCS_DONE, &wm8994->hubs); wm8994_free_irq(codec->control_data, WM8994_IRQ_FIFOS_ERR, codec); + wm8994_free_irq(codec->control_data, WM8994_IRQ_TEMP_SHUT, codec); + wm8994_free_irq(codec->control_data, WM8994_IRQ_TEMP_WARN, codec); switch (control->type) { case WM8994: @@ -3292,6 +3507,7 @@ static int wm8994_codec_remove(struct snd_soc_codec *codec) wm8994); break; + case WM1811: case WM8958: if (wm8994->micdet_irq) free_irq(wm8994->micdet_irq, wm8994); diff --git a/sound/soc/codecs/wm8994.h b/sound/soc/codecs/wm8994.h index 1ab2266039f7..f4f1355efc82 100644 --- a/sound/soc/codecs/wm8994.h +++ b/sound/soc/codecs/wm8994.h @@ -83,6 +83,8 @@ struct wm8994_priv { struct completion fll_locked[2]; bool fll_locked_irq; + int vmid_refcount; + int dac_rates[2]; int lrclk_shared[2]; diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c index 5ad873fda814..78eeb21e6696 100644 --- a/sound/soc/codecs/wm8995.c +++ b/sound/soc/codecs/wm8995.c @@ -485,7 +485,7 @@ static int configure_aif_clock(struct snd_soc_codec *codec, int aif) static int configure_clock(struct snd_soc_codec *codec) { struct wm8995_priv *wm8995; - int old, new; + int change, new; wm8995 = snd_soc_codec_get_drvdata(codec); @@ -509,15 +509,11 @@ static int configure_clock(struct snd_soc_codec *codec) else new = 0; - old = snd_soc_read(codec, WM8995_CLOCKING_1) & WM8995_SYSCLK_SRC; - - /* If there's no change then we're done. */ - if (old == new) + change = snd_soc_update_bits(codec, WM8995_CLOCKING_1, + WM8995_SYSCLK_SRC_MASK, new); + if (!change) return 0; - snd_soc_update_bits(codec, WM8995_CLOCKING_1, - WM8995_SYSCLK_SRC_MASK, new); - snd_soc_dapm_sync(&codec->dapm); return 0; @@ -1573,11 +1569,16 @@ static int wm8995_resume(struct snd_soc_codec *codec) static int wm8995_remove(struct snd_soc_codec *codec) { struct wm8995_priv *wm8995; - struct i2c_client *i2c; + int i; - i2c = container_of(codec->dev, struct i2c_client, dev); wm8995 = snd_soc_codec_get_drvdata(codec); wm8995_set_bias_level(codec, SND_SOC_BIAS_OFF); + + for (i = 0; i < ARRAY_SIZE(wm8995->supplies); ++i) + regulator_unregister_notifier(wm8995->supplies[i].consumer, + &wm8995->disable_nb[i]); + + regulator_bulk_free(ARRAY_SIZE(wm8995->supplies), wm8995->supplies); return 0; } @@ -1642,6 +1643,7 @@ static int wm8995_probe(struct snd_soc_codec *codec) if (ret != 0x8995) { dev_err(codec->dev, "Invalid device ID: %#x\n", ret); + ret = -EINVAL; goto err_reg_enable; } diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 0cdb9d105671..645c980d6b80 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -41,12 +41,11 @@ #define HPOUT2L 4 #define HPOUT2R 8 -#define WM8996_NUM_SUPPLIES 4 +#define WM8996_NUM_SUPPLIES 3 static const char *wm8996_supply_names[WM8996_NUM_SUPPLIES] = { "DBVDD", "AVDD1", "AVDD2", - "CPVDD", }; struct wm8996_priv { @@ -71,6 +70,8 @@ struct wm8996_priv { struct regulator_bulk_data supplies[WM8996_NUM_SUPPLIES]; struct notifier_block disable_nb[WM8996_NUM_SUPPLIES]; + struct regulator *cpvdd; + int bg_ena; struct wm8996_pdata pdata; @@ -112,7 +113,6 @@ static int wm8996_regulator_event_##n(struct notifier_block *nb, \ WM8996_REGULATOR_EVENT(0) WM8996_REGULATOR_EVENT(1) WM8996_REGULATOR_EVENT(2) -WM8996_REGULATOR_EVENT(3) static const u16 wm8996_reg[WM8996_MAX_REGISTER] = { [WM8996_SOFTWARE_RESET] = 0x8996, @@ -414,6 +414,7 @@ static const DECLARE_TLV_DB_SCALE(out_digital_tlv, -1200, 150, 0); static const DECLARE_TLV_DB_SCALE(out_tlv, -900, 75, 0); static const DECLARE_TLV_DB_SCALE(spk_tlv, -900, 150, 0); static const DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); +static const DECLARE_TLV_DB_SCALE(threedstereo_tlv, -1600, 183, 1); static const char *sidetone_hpf_text[] = { "2.9kHz", "1.5kHz", "735Hz", "403Hz", "196Hz", "98Hz", "49Hz" @@ -608,6 +609,14 @@ SOC_SINGLE("DAC High Performance Switch", WM8996_OVERSAMPLING, 0, 1, 0), SOC_SINGLE("DAC Soft Mute Switch", WM8996_DAC_SOFTMUTE, 1, 1, 0), SOC_SINGLE("DAC Slow Soft Mute Switch", WM8996_DAC_SOFTMUTE, 0, 1, 0), +SOC_SINGLE("DSP1 3D Stereo Switch", WM8996_DSP1_RX_FILTERS_2, 8, 1, 0), +SOC_SINGLE("DSP2 3D Stereo Switch", WM8996_DSP2_RX_FILTERS_2, 8, 1, 0), + +SOC_SINGLE_TLV("DSP1 3D Stereo Volume", WM8996_DSP1_RX_FILTERS_2, 10, 15, + 0, threedstereo_tlv), +SOC_SINGLE_TLV("DSP2 3D Stereo Volume", WM8996_DSP2_RX_FILTERS_2, 10, 15, + 0, threedstereo_tlv), + SOC_DOUBLE_TLV("Digital Output 1 Volume", WM8996_DAC1_HPOUT1_VOLUME, 0, 4, 8, 0, out_digital_tlv), SOC_DOUBLE_TLV("Digital Output 2 Volume", WM8996_DAC2_HPOUT2_VOLUME, 0, 4, @@ -632,6 +641,14 @@ SOC_DOUBLE_R("Speaker ZC Switch", WM8996_LEFT_PDM_SPEAKER, SOC_SINGLE("DSP1 EQ Switch", WM8996_DSP1_RX_EQ_GAINS_1, 0, 1, 0), SOC_SINGLE("DSP2 EQ Switch", WM8996_DSP2_RX_EQ_GAINS_1, 0, 1, 0), + +SOC_SINGLE("DSP1 DRC TXL Switch", WM8996_DSP1_DRC_1, 0, 1, 0), +SOC_SINGLE("DSP1 DRC TXR Switch", WM8996_DSP1_DRC_1, 1, 1, 0), +SOC_SINGLE("DSP1 DRC RX Switch", WM8996_DSP1_DRC_1, 2, 1, 0), + +SOC_SINGLE("DSP2 DRC TXL Switch", WM8996_DSP2_DRC_1, 0, 1, 0), +SOC_SINGLE("DSP2 DRC TXR Switch", WM8996_DSP2_DRC_1, 1, 1, 0), +SOC_SINGLE("DSP2 DRC RX Switch", WM8996_DSP2_DRC_1, 2, 1, 0), }; static const struct snd_kcontrol_new wm8996_eq_controls[] = { @@ -658,19 +675,75 @@ SOC_SINGLE_TLV("DSP2 EQ B5 Volume", WM8996_DSP2_RX_EQ_GAINS_2, 6, 31, 0, eq_tlv), }; +static void wm8996_bg_enable(struct snd_soc_codec *codec) +{ + struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec); + + wm8996->bg_ena++; + if (wm8996->bg_ena == 1) { + snd_soc_update_bits(codec, WM8996_POWER_MANAGEMENT_1, + WM8996_BG_ENA, WM8996_BG_ENA); + msleep(2); + } +} + +static void wm8996_bg_disable(struct snd_soc_codec *codec) +{ + struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec); + + wm8996->bg_ena--; + if (!wm8996->bg_ena) + snd_soc_update_bits(codec, WM8996_POWER_MANAGEMENT_1, + WM8996_BG_ENA, 0); +} + +static int bg_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + int ret = 0; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + wm8996_bg_enable(codec); + break; + case SND_SOC_DAPM_POST_PMD: + wm8996_bg_disable(codec); + break; + default: + BUG(); + ret = -EINVAL; + } + + return ret; +} + static int cp_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { + struct snd_soc_codec *codec = w->codec; + struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec); + int ret = 0; + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + ret = regulator_enable(wm8996->cpvdd); + if (ret != 0) + dev_err(codec->dev, "Failed to enable CPVDD: %d\n", + ret); + break; case SND_SOC_DAPM_POST_PMU: msleep(5); break; + case SND_SOC_DAPM_POST_PMD: + regulator_disable_deferred(wm8996->cpvdd, 20); + break; default: BUG(); - return -EINVAL; + ret = -EINVAL; } - return 0; + return ret; } static int rmv_short_event(struct snd_soc_dapm_widget *w, @@ -698,7 +771,7 @@ static void wait_for_dc_servo(struct snd_soc_codec *codec, u16 mask) { struct i2c_client *i2c = to_i2c_client(codec->dev); struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec); - int i, ret; + int ret; unsigned long timeout = 200; snd_soc_write(codec, WM8996_DC_SERVO_2, mask); @@ -713,15 +786,12 @@ static void wait_for_dc_servo(struct snd_soc_codec *codec, u16 mask) } else { msleep(1); - if (--i) { - timeout = 0; - break; - } + timeout--; } ret = snd_soc_read(codec, WM8996_DC_SERVO_2); dev_dbg(codec->dev, "DC servo state: %x\n", ret); - } while (ret & mask); + } while (timeout && ret & mask); if (timeout == 0) dev_err(codec->dev, "DC servo timed out for %x\n", mask); @@ -979,9 +1049,12 @@ SND_SOC_DAPM_SUPPLY_S("SYSCLK", 1, WM8996_AIF_CLOCKING_1, 0, 0, NULL, 0), SND_SOC_DAPM_SUPPLY_S("SYSDSPCLK", 2, WM8996_CLOCKING_1, 1, 0, NULL, 0), SND_SOC_DAPM_SUPPLY_S("AIFCLK", 2, WM8996_CLOCKING_1, 2, 0, NULL, 0), SND_SOC_DAPM_SUPPLY_S("Charge Pump", 2, WM8996_CHARGE_PUMP_1, 15, 0, cp_event, - SND_SOC_DAPM_POST_PMU), - + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +SND_SOC_DAPM_SUPPLY("Bandgap", SND_SOC_NOPM, 0, 0, bg_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_SUPPLY("LDO2", WM8996_POWER_MANAGEMENT_2, 1, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("MICB1 Audio", WM8996_MICBIAS_1, 4, 1, NULL, 0), +SND_SOC_DAPM_SUPPLY("MICB2 Audio", WM8996_MICBIAS_2, 4, 1, NULL, 0), SND_SOC_DAPM_MICBIAS("MICB2", WM8996_POWER_MANAGEMENT_1, 9, 0), SND_SOC_DAPM_MICBIAS("MICB1", WM8996_POWER_MANAGEMENT_1, 8, 0), @@ -1035,14 +1108,14 @@ SND_SOC_DAPM_DAC("DAC2R", NULL, WM8996_POWER_MANAGEMENT_5, 2, 0), SND_SOC_DAPM_DAC("DAC1L", NULL, WM8996_POWER_MANAGEMENT_5, 1, 0), SND_SOC_DAPM_DAC("DAC1R", NULL, WM8996_POWER_MANAGEMENT_5, 0, 0), -SND_SOC_DAPM_AIF_IN("AIF2RX1", "AIF2 Playback", 1, +SND_SOC_DAPM_AIF_IN("AIF2RX1", "AIF2 Playback", 0, WM8996_POWER_MANAGEMENT_4, 9, 0), -SND_SOC_DAPM_AIF_IN("AIF2RX0", "AIF2 Playback", 2, +SND_SOC_DAPM_AIF_IN("AIF2RX0", "AIF2 Playback", 1, WM8996_POWER_MANAGEMENT_4, 8, 0), -SND_SOC_DAPM_AIF_IN("AIF2TX1", "AIF2 Capture", 1, +SND_SOC_DAPM_AIF_OUT("AIF2TX1", "AIF2 Capture", 0, WM8996_POWER_MANAGEMENT_6, 9, 0), -SND_SOC_DAPM_AIF_IN("AIF2TX0", "AIF2 Capture", 2, +SND_SOC_DAPM_AIF_OUT("AIF2TX0", "AIF2 Capture", 1, WM8996_POWER_MANAGEMENT_6, 8, 0), SND_SOC_DAPM_AIF_IN("AIF1RX5", "AIF1 Playback", 5, @@ -1137,17 +1210,23 @@ static const struct snd_soc_dapm_route wm8996_dapm_routes[] = { { "Charge Pump", NULL, "SYSCLK" }, { "MICB1", NULL, "LDO2" }, + { "MICB1", NULL, "MICB1 Audio" }, + { "MICB1", NULL, "Bandgap" }, { "MICB2", NULL, "LDO2" }, + { "MICB2", NULL, "MICB2 Audio" }, + { "MICB2", NULL, "Bandgap" }, { "IN1L PGA", NULL, "IN2LN" }, { "IN1L PGA", NULL, "IN2LP" }, { "IN1L PGA", NULL, "IN1LN" }, { "IN1L PGA", NULL, "IN1LP" }, + { "IN1L PGA", NULL, "Bandgap" }, { "IN1R PGA", NULL, "IN2RN" }, { "IN1R PGA", NULL, "IN2RP" }, { "IN1R PGA", NULL, "IN1RN" }, { "IN1R PGA", NULL, "IN1RP" }, + { "IN1R PGA", NULL, "Bandgap" }, { "ADCL", NULL, "IN1L PGA" }, @@ -1281,6 +1360,7 @@ static const struct snd_soc_dapm_route wm8996_dapm_routes[] = { { "DAC2R", NULL, "DAC2R Mixer" }, { "HPOUT2L PGA", NULL, "Charge Pump" }, + { "HPOUT2L PGA", NULL, "Bandgap" }, { "HPOUT2L PGA", NULL, "DAC2L" }, { "HPOUT2L_DLY", NULL, "HPOUT2L PGA" }, { "HPOUT2L_DCS", NULL, "HPOUT2L_DLY" }, @@ -1288,6 +1368,7 @@ static const struct snd_soc_dapm_route wm8996_dapm_routes[] = { { "HPOUT2L_RMV_SHORT", NULL, "HPOUT2L_OUTP" }, { "HPOUT2R PGA", NULL, "Charge Pump" }, + { "HPOUT2R PGA", NULL, "Bandgap" }, { "HPOUT2R PGA", NULL, "DAC2R" }, { "HPOUT2R_DLY", NULL, "HPOUT2R PGA" }, { "HPOUT2R_DCS", NULL, "HPOUT2R_DLY" }, @@ -1295,6 +1376,7 @@ static const struct snd_soc_dapm_route wm8996_dapm_routes[] = { { "HPOUT2R_RMV_SHORT", NULL, "HPOUT2R_OUTP" }, { "HPOUT1L PGA", NULL, "Charge Pump" }, + { "HPOUT1L PGA", NULL, "Bandgap" }, { "HPOUT1L PGA", NULL, "DAC1L" }, { "HPOUT1L_DLY", NULL, "HPOUT1L PGA" }, { "HPOUT1L_DCS", NULL, "HPOUT1L_DLY" }, @@ -1302,6 +1384,7 @@ static const struct snd_soc_dapm_route wm8996_dapm_routes[] = { { "HPOUT1L_RMV_SHORT", NULL, "HPOUT1L_OUTP" }, { "HPOUT1R PGA", NULL, "Charge Pump" }, + { "HPOUT1R PGA", NULL, "Bandgap" }, { "HPOUT1R PGA", NULL, "DAC1R" }, { "HPOUT1R_DLY", NULL, "HPOUT1R PGA" }, { "HPOUT1R_DCS", NULL, "HPOUT1R_DLY" }, @@ -1620,14 +1703,7 @@ static int wm8996_set_bias_level(struct snd_soc_codec *codec, switch (level) { case SND_SOC_BIAS_ON: - break; - case SND_SOC_BIAS_PREPARE: - if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY) { - snd_soc_update_bits(codec, WM8996_POWER_MANAGEMENT_1, - WM8996_BG_ENA, WM8996_BG_ENA); - msleep(2); - } break; case SND_SOC_BIAS_STANDBY: @@ -1650,9 +1726,6 @@ static int wm8996_set_bias_level(struct snd_soc_codec *codec, codec->cache_only = false; snd_soc_cache_sync(codec); } - - snd_soc_update_bits(codec, WM8996_POWER_MANAGEMENT_1, - WM8996_BG_ENA, 0); break; case SND_SOC_BIAS_OFF: @@ -1847,7 +1920,7 @@ static int wm8996_hw_params(struct snd_pcm_substream *substream, snd_soc_update_bits(codec, lrclk_reg, WM8996_AIF1RX_RATE_MASK, lrclk); snd_soc_update_bits(codec, WM8996_AIF_CLOCKING_2, - WM8996_DSP1_DIV_SHIFT << dsp_shift, dsp); + WM8996_DSP1_DIV_MASK << dsp_shift, dsp); return 0; } @@ -2041,7 +2114,7 @@ static int wm8996_set_fll(struct snd_soc_codec *codec, int fll_id, int source, struct i2c_client *i2c = to_i2c_client(codec->dev); struct _fll_div fll_div; unsigned long timeout; - int ret, reg; + int ret, reg, retry; /* Any change? */ if (source == wm8996->fll_src && Fref == wm8996->fll_fref && @@ -2057,6 +2130,8 @@ static int wm8996_set_fll(struct snd_soc_codec *codec, int fll_id, int source, snd_soc_update_bits(codec, WM8996_FLL_CONTROL_1, WM8996_FLL_ENA, 0); + wm8996_bg_disable(codec); + return 0; } @@ -2111,6 +2186,11 @@ static int wm8996_set_fll(struct snd_soc_codec *codec, int fll_id, int source, snd_soc_write(codec, WM8996_FLL_EFS_1, fll_div.lambda); + /* Enable the bandgap if it's not already enabled */ + ret = snd_soc_read(codec, WM8996_FLL_CONTROL_1); + if (!(ret & WM8996_FLL_ENA)) + wm8996_bg_enable(codec); + /* Clear any pending completions (eg, from failed startups) */ try_wait_for_completion(&wm8996->fll_lock); @@ -2128,17 +2208,29 @@ static int wm8996_set_fll(struct snd_soc_codec *codec, int fll_id, int source, else timeout = msecs_to_jiffies(2); - /* Allow substantially longer if we've actually got the IRQ */ + /* Allow substantially longer if we've actually got the IRQ, poll + * at a slightly higher rate if we don't. + */ if (i2c->irq) - timeout *= 1000; + timeout *= 10; + else + timeout /= 2; - ret = wait_for_completion_timeout(&wm8996->fll_lock, timeout); + for (retry = 0; retry < 10; retry++) { + ret = wait_for_completion_timeout(&wm8996->fll_lock, + timeout); + if (ret != 0) { + WARN_ON(!i2c->irq); + break; + } - if (ret == 0 && i2c->irq) { + ret = snd_soc_read(codec, WM8996_INTERRUPT_RAW_STATUS_2); + if (ret & WM8996_FLL_LOCK_STS) + break; + } + if (retry == 10) { dev_err(codec->dev, "Timed out waiting for FLL\n"); ret = -ETIMEDOUT; - } else { - ret = 0; } dev_dbg(codec->dev, "FLL configured for %dHz->%dHz\n", Fref, Fout); @@ -2297,12 +2389,94 @@ int wm8996_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack, /* Enable interrupts and we're off */ snd_soc_update_bits(codec, WM8996_INTERRUPT_STATUS_2_MASK, - WM8996_IM_MICD_EINT, 0); + WM8996_IM_MICD_EINT | WM8996_HP_DONE_EINT, 0); return 0; } EXPORT_SYMBOL_GPL(wm8996_detect); +static void wm8996_hpdet_irq(struct snd_soc_codec *codec) +{ + struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec); + int val, reg, report; + + /* Assume headphone in error conditions; we need to report + * something or we stall our state machine. + */ + report = SND_JACK_HEADPHONE; + + reg = snd_soc_read(codec, WM8996_HEADPHONE_DETECT_2); + if (reg < 0) { + dev_err(codec->dev, "Failed to read HPDET status\n"); + goto out; + } + + if (!(reg & WM8996_HP_DONE)) { + dev_err(codec->dev, "Got HPDET IRQ but HPDET is busy\n"); + goto out; + } + + val = reg & WM8996_HP_LVL_MASK; + + dev_dbg(codec->dev, "HPDET measured %d ohms\n", val); + + /* If we've got high enough impedence then report as line, + * otherwise assume headphone. + */ + if (val >= 126) + report = SND_JACK_LINEOUT; + else + report = SND_JACK_HEADPHONE; + +out: + if (wm8996->jack_mic) + report |= SND_JACK_MICROPHONE; + + snd_soc_jack_report(wm8996->jack, report, + SND_JACK_LINEOUT | SND_JACK_HEADSET); + + wm8996->detecting = false; + + /* If the output isn't running re-clamp it */ + if (!(snd_soc_read(codec, WM8996_POWER_MANAGEMENT_1) & + (WM8996_HPOUT1L_ENA | WM8996_HPOUT1R_RMV_SHORT))) + snd_soc_update_bits(codec, WM8996_ANALOGUE_HP_1, + WM8996_HPOUT1L_RMV_SHORT | + WM8996_HPOUT1R_RMV_SHORT, 0); + + /* Go back to looking at the microphone */ + snd_soc_update_bits(codec, WM8996_ACCESSORY_DETECT_MODE_1, + WM8996_JD_MODE_MASK, 0); + snd_soc_update_bits(codec, WM8996_MIC_DETECT_1, WM8996_MICD_ENA, + WM8996_MICD_ENA); + + snd_soc_dapm_disable_pin(&codec->dapm, "Bandgap"); + snd_soc_dapm_sync(&codec->dapm); +} + +static void wm8996_hpdet_start(struct snd_soc_codec *codec) +{ + /* Unclamp the output, we can't measure while we're shorting it */ + snd_soc_update_bits(codec, WM8996_ANALOGUE_HP_1, + WM8996_HPOUT1L_RMV_SHORT | + WM8996_HPOUT1R_RMV_SHORT, + WM8996_HPOUT1L_RMV_SHORT | + WM8996_HPOUT1R_RMV_SHORT); + + /* We need bandgap for HPDET */ + snd_soc_dapm_force_enable_pin(&codec->dapm, "Bandgap"); + snd_soc_dapm_sync(&codec->dapm); + + /* Go into headphone detect left mode */ + snd_soc_update_bits(codec, WM8996_MIC_DETECT_1, WM8996_MICD_ENA, 0); + snd_soc_update_bits(codec, WM8996_ACCESSORY_DETECT_MODE_1, + WM8996_JD_MODE_MASK, 1); + + /* Trigger a measurement */ + snd_soc_update_bits(codec, WM8996_HEADPHONE_DETECT_1, + WM8996_HP_POLL, WM8996_HP_POLL); +} + static void wm8996_micd(struct snd_soc_codec *codec) { struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec); @@ -2323,28 +2497,36 @@ static void wm8996_micd(struct snd_soc_codec *codec) wm8996->jack_mic = false; wm8996->detecting = true; snd_soc_jack_report(wm8996->jack, 0, - SND_JACK_HEADSET | SND_JACK_BTN_0); + SND_JACK_LINEOUT | SND_JACK_HEADSET | + SND_JACK_BTN_0); + snd_soc_update_bits(codec, WM8996_MIC_DETECT_1, WM8996_MICD_RATE_MASK, WM8996_MICD_RATE_MASK); return; } - /* If the measurement is very high we've got a microphone but - * do a little debounce to account for mechanical issues. + /* If the measurement is very high we've got a microphone, + * either we just detected one or if we already reported then + * we've got a button release event. */ if (val & 0x400) { - dev_dbg(codec->dev, "Microphone detected\n"); - snd_soc_jack_report(wm8996->jack, SND_JACK_HEADSET, - SND_JACK_HEADSET | SND_JACK_BTN_0); - wm8996->jack_mic = true; - wm8996->detecting = false; - - /* Increase poll rate to give better responsiveness - * for buttons */ - snd_soc_update_bits(codec, WM8996_MIC_DETECT_1, - WM8996_MICD_RATE_MASK, - 5 << WM8996_MICD_RATE_SHIFT); + if (wm8996->detecting) { + dev_dbg(codec->dev, "Microphone detected\n"); + wm8996->jack_mic = true; + wm8996_hpdet_start(codec); + + /* Increase poll rate to give better responsiveness + * for buttons */ + snd_soc_update_bits(codec, WM8996_MIC_DETECT_1, + WM8996_MICD_RATE_MASK, + 5 << WM8996_MICD_RATE_SHIFT); + } else { + dev_dbg(codec->dev, "Mic button up\n"); + snd_soc_jack_report(wm8996->jack, 0, SND_JACK_BTN_0); + } + + return; } /* If we detected a lower impedence during initial startup @@ -2376,15 +2558,11 @@ static void wm8996_micd(struct snd_soc_codec *codec) if (val & 0x3fc) { if (wm8996->jack_mic) { dev_dbg(codec->dev, "Mic button detected\n"); - snd_soc_jack_report(wm8996->jack, - SND_JACK_HEADSET | SND_JACK_BTN_0, - SND_JACK_HEADSET | SND_JACK_BTN_0); - } else { - dev_dbg(codec->dev, "Headphone detected\n"); - snd_soc_jack_report(wm8996->jack, - SND_JACK_HEADPHONE, - SND_JACK_HEADSET | + snd_soc_jack_report(wm8996->jack, SND_JACK_BTN_0, SND_JACK_BTN_0); + } else if (wm8996->detecting) { + dev_dbg(codec->dev, "Headphone detected\n"); + wm8996_hpdet_start(codec); /* Increase the detection rate a bit for * responsiveness. @@ -2392,8 +2570,6 @@ static void wm8996_micd(struct snd_soc_codec *codec) snd_soc_update_bits(codec, WM8996_MIC_DETECT_1, WM8996_MICD_RATE_MASK, 7 << WM8996_MICD_RATE_SHIFT); - - wm8996->detecting = false; } } } @@ -2412,6 +2588,9 @@ static irqreturn_t wm8996_irq(int irq, void *data) } irq_val &= ~snd_soc_read(codec, WM8996_INTERRUPT_STATUS_2_MASK); + if (!irq_val) + return IRQ_NONE; + snd_soc_write(codec, WM8996_INTERRUPT_STATUS_2, irq_val); if (irq_val & (WM8996_DCS_DONE_01_EINT | WM8996_DCS_DONE_23_EINT)) { @@ -2430,10 +2609,10 @@ static irqreturn_t wm8996_irq(int irq, void *data) if (irq_val & WM8996_MICD_EINT) wm8996_micd(codec); - if (irq_val) - return IRQ_HANDLED; - else - return IRQ_NONE; + if (irq_val & WM8996_HP_DONE_EINT) + wm8996_hpdet_irq(codec); + + return IRQ_HANDLED; } static irqreturn_t wm8996_edge_irq(int irq, void *data) @@ -2527,7 +2706,6 @@ static int wm8996_probe(struct snd_soc_codec *codec) init_completion(&wm8996->fll_lock); dapm->idle_bias_off = true; - dapm->bias_level = SND_SOC_BIAS_OFF; ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_I2C); if (ret != 0) { @@ -2548,7 +2726,13 @@ static int wm8996_probe(struct snd_soc_codec *codec) wm8996->disable_nb[0].notifier_call = wm8996_regulator_event_0; wm8996->disable_nb[1].notifier_call = wm8996_regulator_event_1; wm8996->disable_nb[2].notifier_call = wm8996_regulator_event_2; - wm8996->disable_nb[3].notifier_call = wm8996_regulator_event_3; + + wm8996->cpvdd = regulator_get(&i2c->dev, "CPVDD"); + if (IS_ERR(wm8996->cpvdd)) { + ret = PTR_ERR(wm8996->cpvdd); + dev_err(&i2c->dev, "Failed to get CPVDD: %d\n", ret); + goto err_get; + } /* This should really be moved into the regulator core */ for (i = 0; i < ARRAY_SIZE(wm8996->supplies); i++) { @@ -2565,7 +2749,7 @@ static int wm8996_probe(struct snd_soc_codec *codec) wm8996->supplies); if (ret != 0) { dev_err(codec->dev, "Failed to enable supplies: %d\n", ret); - goto err_get; + goto err_cpvdd; } if (wm8996->pdata.ldo_ena >= 0) { @@ -2808,6 +2992,8 @@ err_enable: gpio_set_value_cansleep(wm8996->pdata.ldo_ena, 0); regulator_bulk_disable(ARRAY_SIZE(wm8996->supplies), wm8996->supplies); +err_cpvdd: + regulator_put(wm8996->cpvdd); err_get: regulator_bulk_free(ARRAY_SIZE(wm8996->supplies), wm8996->supplies); err: @@ -2831,6 +3017,7 @@ static int wm8996_remove(struct snd_soc_codec *codec) for (i = 0; i < ARRAY_SIZE(wm8996->supplies); i++) regulator_unregister_notifier(wm8996->supplies[i].consumer, &wm8996->disable_nb[i]); + regulator_put(wm8996->cpvdd); regulator_bulk_free(ARRAY_SIZE(wm8996->supplies), wm8996->supplies); return 0; diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index a4691321f9b3..3cd35a02c28c 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -157,7 +157,6 @@ static struct { struct wm9081_priv { enum snd_soc_control_type control_type; - void *control_data; int sysclk_source; int mclk_rate; int sysclk_rate; @@ -174,6 +173,7 @@ static int wm9081_volatile_register(struct snd_soc_codec *codec, unsigned int re { switch (reg) { case WM9081_SOFTWARE_RESET: + case WM9081_INTERRUPT_STATUS: return 1; default: return 0; @@ -820,7 +820,7 @@ static int wm9081_set_bias_level(struct snd_soc_codec *codec, /* VMID 2*240k */ reg = snd_soc_read(codec, WM9081_BIAS_CONTROL_1); reg &= ~WM9081_VMID_SEL_MASK; - reg |= 0x40; + reg |= 0x04; snd_soc_write(codec, WM9081_VMID_CONTROL, reg); /* Standby bias current on */ @@ -1120,8 +1120,8 @@ static int wm9081_digital_mute(struct snd_soc_dai *codec_dai, int mute) return 0; } -static int wm9081_set_sysclk(struct snd_soc_codec *codec, - int clk_id, unsigned int freq, int dir) +static int wm9081_set_sysclk(struct snd_soc_codec *codec, int clk_id, + int source, unsigned int freq, int dir) { struct wm9081_priv *wm9081 = snd_soc_codec_get_drvdata(codec); @@ -1213,7 +1213,6 @@ static int wm9081_probe(struct snd_soc_codec *codec) int ret; u16 reg; - codec->control_data = wm9081->control_data; ret = snd_soc_codec_set_cache_io(codec, 8, 16, wm9081->control_type); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); @@ -1250,8 +1249,6 @@ static int wm9081_probe(struct snd_soc_codec *codec) snd_soc_write(codec, WM9081_ANALOGUE_SPEAKER_PGA, reg | WM9081_SPKPGAZC); - snd_soc_add_controls(codec, wm9081_snd_controls, - ARRAY_SIZE(wm9081_snd_controls)); if (!wm9081->pdata.num_retune_configs) { dev_dbg(codec->dev, "No ReTune Mobile data, using normal EQ\n"); @@ -1311,6 +1308,8 @@ static struct snd_soc_codec_driver soc_codec_dev_wm9081 = { .reg_cache_default = wm9081_reg_defaults, .volatile_register = wm9081_volatile_register, + .controls = wm9081_snd_controls, + .num_controls = ARRAY_SIZE(wm9081_snd_controls), .dapm_widgets = wm9081_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(wm9081_dapm_widgets), .dapm_routes = wm9081_audio_paths, @@ -1330,7 +1329,6 @@ static __devinit int wm9081_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, wm9081); wm9081->control_type = SND_SOC_I2C; - wm9081->control_data = i2c; if (dev_get_platdata(&i2c->dev)) memcpy(&wm9081->pdata, dev_get_platdata(&i2c->dev), diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c index 4de12203e611..2b5252c9e377 100644 --- a/sound/soc/codecs/wm9090.c +++ b/sound/soc/codecs/wm9090.c @@ -139,9 +139,7 @@ static const u16 wm9090_reg_defaults[] = { /* This struct is used to save the context */ struct wm9090_priv { - struct mutex mutex; struct wm9090_platform_data pdata; - void *control_data; }; static int wm9090_volatile(struct snd_soc_codec *codec, unsigned int reg) @@ -550,10 +548,8 @@ static int wm9090_set_bias_level(struct snd_soc_codec *codec, static int wm9090_probe(struct snd_soc_codec *codec) { - struct wm9090_priv *wm9090 = snd_soc_codec_get_drvdata(codec); int ret; - codec->control_data = wm9090->control_data; ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_I2C); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); @@ -662,8 +658,6 @@ static int wm9090_i2c_probe(struct i2c_client *i2c, sizeof(wm9090->pdata)); i2c_set_clientdata(i2c, wm9090); - wm9090->control_data = i2c; - mutex_init(&wm9090->mutex); ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm9090, NULL, 0); @@ -684,6 +678,7 @@ static int __devexit wm9090_i2c_remove(struct i2c_client *i2c) static const struct i2c_device_id wm9090_id[] = { { "wm9090", 0 }, + { "wm9093", 0 }, { } }; MODULE_DEVICE_TABLE(i2c, wm9090_id); diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index e763c54c55dc..84f33d4ea2cd 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -18,6 +18,7 @@ #include <linux/pm.h> #include <linux/i2c.h> #include <linux/platform_device.h> +#include <linux/mfd/wm8994/registers.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -116,14 +117,23 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec) { struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); s8 offset; - u16 reg, reg_l, reg_r, dcs_cfg; + u16 reg, reg_l, reg_r, dcs_cfg, dcs_reg; + + switch (hubs->dcs_readback_mode) { + case 2: + dcs_reg = WM8994_DC_SERVO_4E; + break; + default: + dcs_reg = WM8993_DC_SERVO_3; + break; + } /* If we're using a digital only path and have a previously * callibrated DC servo offset stored then use that. */ if (hubs->class_w && hubs->class_w_dcs) { dev_dbg(codec->dev, "Using cached DC servo offset %x\n", hubs->class_w_dcs); - snd_soc_write(codec, WM8993_DC_SERVO_3, hubs->class_w_dcs); + snd_soc_write(codec, dcs_reg, hubs->class_w_dcs); wait_for_dc_servo(codec, WM8993_DCS_TRIG_DAC_WR_0 | WM8993_DCS_TRIG_DAC_WR_1); @@ -154,8 +164,9 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec) reg_r = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_2) & WM8993_DCS_INTEG_CHAN_1_MASK; break; + case 2: case 1: - reg = snd_soc_read(codec, WM8993_DC_SERVO_3); + reg = snd_soc_read(codec, dcs_reg); reg_r = (reg & WM8993_DCS_DAC_WR_VAL_1_MASK) >> WM8993_DCS_DAC_WR_VAL_1_SHIFT; reg_l = reg & WM8993_DCS_DAC_WR_VAL_0_MASK; @@ -168,24 +179,25 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec) dev_dbg(codec->dev, "DCS input: %x %x\n", reg_l, reg_r); /* Apply correction to DC servo result */ - if (hubs->dcs_codes) { - dev_dbg(codec->dev, "Applying %d code DC servo correction\n", - hubs->dcs_codes); + if (hubs->dcs_codes_l || hubs->dcs_codes_r) { + dev_dbg(codec->dev, + "Applying %d/%d code DC servo correction\n", + hubs->dcs_codes_l, hubs->dcs_codes_r); /* HPOUT1R */ offset = reg_r; - offset += hubs->dcs_codes; + offset += hubs->dcs_codes_r; dcs_cfg = (u8)offset << WM8993_DCS_DAC_WR_VAL_1_SHIFT; /* HPOUT1L */ offset = reg_l; - offset += hubs->dcs_codes; + offset += hubs->dcs_codes_l; dcs_cfg |= (u8)offset; dev_dbg(codec->dev, "DCS result: %x\n", dcs_cfg); /* Do it */ - snd_soc_write(codec, WM8993_DC_SERVO_3, dcs_cfg); + snd_soc_write(codec, dcs_reg, dcs_cfg); wait_for_dc_servo(codec, WM8993_DCS_TRIG_DAC_WR_0 | WM8993_DCS_TRIG_DAC_WR_1); @@ -210,14 +222,14 @@ static int wm8993_put_dc_servo(struct snd_kcontrol *kcontrol, struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); int ret; - ret = snd_soc_put_volsw_2r(kcontrol, ucontrol); + ret = snd_soc_put_volsw(kcontrol, ucontrol); /* Updating the analogue gains invalidates the DC servo cache */ hubs->class_w_dcs = 0; /* If we're applying an offset correction then updating the * callibration would be likely to introduce further offsets. */ - if (hubs->dcs_codes || hubs->no_series_update) + if (hubs->dcs_codes_l || hubs->dcs_codes_r || hubs->no_series_update) return ret; /* Only need to do this if the outputs are active */ @@ -350,19 +362,11 @@ SOC_DOUBLE_TLV("Speaker Boost Volume", WM8993_SPKOUT_BOOST, 3, 0, 7, 0, SOC_ENUM("Speaker Reference", speaker_ref), SOC_ENUM("Speaker Mode", speaker_mode), -{ - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Headphone Volume", - .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | - SNDRV_CTL_ELEM_ACCESS_READWRITE, - .tlv.p = outpga_tlv, - .info = snd_soc_info_volsw_2r, - .get = snd_soc_get_volsw_2r, .put = wm8993_put_dc_servo, - .private_value = (unsigned long)&(struct soc_mixer_control) { - .reg = WM8993_LEFT_OUTPUT_VOLUME, - .rreg = WM8993_RIGHT_OUTPUT_VOLUME, - .shift = 0, .max = 63 - }, -}, +SOC_DOUBLE_R_EXT_TLV("Headphone Volume", + WM8993_LEFT_OUTPUT_VOLUME, WM8993_RIGHT_OUTPUT_VOLUME, + 0, 63, 0, snd_soc_get_volsw, wm8993_put_dc_servo, + outpga_tlv), + SOC_DOUBLE_R("Headphone Switch", WM8993_LEFT_OUTPUT_VOLUME, WM8993_RIGHT_OUTPUT_VOLUME, 6, 1, 0), SOC_DOUBLE_R("Headphone ZC Switch", WM8993_LEFT_OUTPUT_VOLUME, @@ -699,6 +703,11 @@ static const struct snd_soc_dapm_route analogue_routes[] = { { "IN1L PGA", "IN1LP Switch", "IN1LP" }, { "IN1L PGA", "IN1LN Switch", "IN1LN" }, + { "IN1L PGA", NULL, "VMID" }, + { "IN1R PGA", NULL, "VMID" }, + { "IN2L PGA", NULL, "VMID" }, + { "IN2R PGA", NULL, "VMID" }, + { "IN1R PGA", "IN1RP Switch", "IN1RP" }, { "IN1R PGA", "IN1RN Switch", "IN1RN" }, @@ -716,12 +725,14 @@ static const struct snd_soc_dapm_route analogue_routes[] = { { "MIXINL", NULL, "Direct Voice" }, { "MIXINL", NULL, "IN1LP" }, { "MIXINL", NULL, "Left Output Mixer" }, + { "MIXINL", NULL, "VMID" }, { "MIXINR", "IN1R Switch", "IN1R PGA" }, { "MIXINR", "IN2R Switch", "IN2R PGA" }, { "MIXINR", NULL, "Direct Voice" }, { "MIXINR", NULL, "IN1RP" }, { "MIXINR", NULL, "Right Output Mixer" }, + { "MIXINR", NULL, "VMID" }, { "ADCL", NULL, "MIXINL" }, { "ADCR", NULL, "MIXINR" }, @@ -752,6 +763,7 @@ static const struct snd_soc_dapm_route analogue_routes[] = { { "Earpiece Mixer", "Left Output Switch", "Left Output PGA" }, { "Earpiece Mixer", "Right Output Switch", "Right Output PGA" }, + { "Earpiece Driver", NULL, "VMID" }, { "Earpiece Driver", NULL, "Earpiece Mixer" }, { "HPOUT2N", NULL, "Earpiece Driver" }, { "HPOUT2P", NULL, "Earpiece Driver" }, @@ -774,9 +786,11 @@ static const struct snd_soc_dapm_route analogue_routes[] = { { "SPKR Boost", "SPKR Switch", "SPKR" }, { "SPKR Boost", "SPKL Switch", "SPKL" }, + { "SPKL Driver", NULL, "VMID" }, { "SPKL Driver", NULL, "SPKL Boost" }, { "SPKL Driver", NULL, "CLK_SYS" }, + { "SPKR Driver", NULL, "VMID" }, { "SPKR Driver", NULL, "SPKR Boost" }, { "SPKR Driver", NULL, "CLK_SYS" }, @@ -790,12 +804,18 @@ static const struct snd_soc_dapm_route analogue_routes[] = { { "Headphone PGA", NULL, "Left Headphone Mux" }, { "Headphone PGA", NULL, "Right Headphone Mux" }, + { "Headphone PGA", NULL, "VMID" }, { "Headphone PGA", NULL, "CLK_SYS" }, { "Headphone PGA", NULL, "Headphone Supply" }, { "HPOUT1L", NULL, "Headphone PGA" }, { "HPOUT1R", NULL, "Headphone PGA" }, + { "LINEOUT1N Driver", NULL, "VMID" }, + { "LINEOUT1P Driver", NULL, "VMID" }, + { "LINEOUT2N Driver", NULL, "VMID" }, + { "LINEOUT2P Driver", NULL, "VMID" }, + { "LINEOUT1N", NULL, "LINEOUT1N Driver" }, { "LINEOUT1P", NULL, "LINEOUT1P Driver" }, { "LINEOUT2N", NULL, "LINEOUT2N Driver" }, diff --git a/sound/soc/codecs/wm_hubs.h b/sound/soc/codecs/wm_hubs.h index 676b1252ab91..c674c7a502a6 100644 --- a/sound/soc/codecs/wm_hubs.h +++ b/sound/soc/codecs/wm_hubs.h @@ -23,7 +23,8 @@ extern const unsigned int wm_hubs_spkmix_tlv[]; /* This *must* be the first element of the codec->private_data struct */ struct wm_hubs_data { - int dcs_codes; + int dcs_codes_l; + int dcs_codes_r; int dcs_readback_mode; int hp_startup_mode; int series_startup; diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index fe7984221eb9..f78c3f0f280c 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -150,8 +150,6 @@ static int evm_aic3x_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_enable_pin(dapm, "Mic Jack"); snd_soc_dapm_enable_pin(dapm, "Line In"); - snd_soc_dapm_sync(dapm); - return 0; } diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index d0d60b8a54d4..300e12118c00 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -265,6 +265,7 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, struct davinci_mcbsp_dev *dev = snd_soc_dai_get_drvdata(cpu_dai); unsigned int pcr; unsigned int srgr; + bool inv_fs = false; /* Attention srgr is updated by hw_params! */ srgr = DAVINCI_MCBSP_SRGR_FSGM | DAVINCI_MCBSP_SRGR_FPER(DEFAULT_BITPERSAMPLE * 2 - 1) | @@ -330,7 +331,7 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, * more empty bit clock slots between channels as the sample * rate is lowered. */ - fmt ^= SND_SOC_DAIFMT_NB_IF; + inv_fs = true; case SND_SOC_DAIFMT_DSP_A: dev->mode = MOD_DSP_A; break; @@ -394,6 +395,8 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, default: return -EINVAL; } + if (inv_fs == true) + pcr ^= (DAVINCI_MCBSP_PCR_FSXP | DAVINCI_MCBSP_PCR_FSRP); davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SRGR_REG, srgr); dev->pcr = pcr; davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_PCR_REG, pcr); diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 8566238db2a5..7173df254a91 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -732,16 +732,19 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, davinci_hw_param(dev, substream->stream); switch (params_format(params)) { + case SNDRV_PCM_FORMAT_U8: case SNDRV_PCM_FORMAT_S8: dma_params->data_type = 1; word_length = DAVINCI_AUDIO_WORD_8; break; + case SNDRV_PCM_FORMAT_U16_LE: case SNDRV_PCM_FORMAT_S16_LE: dma_params->data_type = 2; word_length = DAVINCI_AUDIO_WORD_16; break; + case SNDRV_PCM_FORMAT_U32_LE: case SNDRV_PCM_FORMAT_S32_LE: dma_params->data_type = 4; word_length = DAVINCI_AUDIO_WORD_32; @@ -818,6 +821,13 @@ static struct snd_soc_dai_ops davinci_mcasp_dai_ops = { }; +#define DAVINCI_MCASP_PCM_FMTS (SNDRV_PCM_FMTBIT_S8 | \ + SNDRV_PCM_FMTBIT_U8 | \ + SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_U16_LE | \ + SNDRV_PCM_FMTBIT_S32_LE | \ + SNDRV_PCM_FMTBIT_U32_LE) + static struct snd_soc_dai_driver davinci_mcasp_dai[] = { { .name = "davinci-mcasp.0", @@ -825,17 +835,13 @@ static struct snd_soc_dai_driver davinci_mcasp_dai[] = { .channels_min = 2, .channels_max = 2, .rates = DAVINCI_MCASP_RATES, - .formats = SNDRV_PCM_FMTBIT_S8 | - SNDRV_PCM_FMTBIT_S16_LE | - SNDRV_PCM_FMTBIT_S32_LE, + .formats = DAVINCI_MCASP_PCM_FMTS, }, .capture = { .channels_min = 2, .channels_max = 2, .rates = DAVINCI_MCASP_RATES, - .formats = SNDRV_PCM_FMTBIT_S8 | - SNDRV_PCM_FMTBIT_S16_LE | - SNDRV_PCM_FMTBIT_S32_LE, + .formats = DAVINCI_MCASP_PCM_FMTS, }, .ops = &davinci_mcasp_dai_ops, @@ -846,7 +852,7 @@ static struct snd_soc_dai_driver davinci_mcasp_dai[] = { .channels_min = 1, .channels_max = 384, .rates = DAVINCI_MCASP_RATES, - .formats = SNDRV_PCM_FMTBIT_S16_LE, + .formats = DAVINCI_MCASP_PCM_FMTS, }, .ops = &davinci_mcasp_dai_ops, }, diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index a49e667373bc..d5fe08cc5db7 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -180,7 +180,6 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) { struct davinci_runtime_data *prtd = substream->runtime->private_data; struct snd_pcm_runtime *runtime = substream->runtime; - int link = prtd->asp_link[0]; unsigned int period_size; unsigned int dma_offset; dma_addr_t dma_pos; @@ -198,7 +197,8 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) fifo_level = prtd->params->fifo_level; pr_debug("davinci_pcm: audio_set_dma_params_play channel = %d " - "dma_ptr = %x period_size=%x\n", link, dma_pos, period_size); + "dma_ptr = %x period_size=%x\n", prtd->asp_link[0], dma_pos, + period_size); data_type = prtd->params->data_type; count = period_size / data_type; @@ -222,17 +222,19 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) } acnt = prtd->params->acnt; - edma_set_src(link, src, INCR, W8BIT); - edma_set_dest(link, dst, INCR, W8BIT); + edma_set_src(prtd->asp_link[0], src, INCR, W8BIT); + edma_set_dest(prtd->asp_link[0], dst, INCR, W8BIT); - edma_set_src_index(link, src_bidx, src_cidx); - edma_set_dest_index(link, dst_bidx, dst_cidx); + edma_set_src_index(prtd->asp_link[0], src_bidx, src_cidx); + edma_set_dest_index(prtd->asp_link[0], dst_bidx, dst_cidx); if (!fifo_level) - edma_set_transfer_params(link, acnt, count, 1, 0, ASYNC); + edma_set_transfer_params(prtd->asp_link[0], acnt, count, 1, 0, + ASYNC); else - edma_set_transfer_params(link, acnt, fifo_level, count, - fifo_level, ABSYNC); + edma_set_transfer_params(prtd->asp_link[0], acnt, fifo_level, + count, fifo_level, + ABSYNC); } static void davinci_pcm_dma_irq(unsigned link, u16 ch_status, void *data) @@ -305,7 +307,6 @@ static int ping_pong_dma_setup(struct snd_pcm_substream *substream) unsigned int acnt = params->acnt; /* divide by 2 for ping/pong */ unsigned int ping_size = snd_pcm_lib_period_bytes(substream) >> 1; - int link = prtd->asp_link[1]; unsigned int fifo_level = prtd->params->fifo_level; unsigned int count; if ((data_type == 0) || (data_type > 4)) { @@ -316,28 +317,26 @@ static int ping_pong_dma_setup(struct snd_pcm_substream *substream) dma_addr_t asp_src_pong = iram_dma->addr + ping_size; ram_src_cidx = ping_size; ram_dst_cidx = -ping_size; - edma_set_src(link, asp_src_pong, INCR, W8BIT); + edma_set_src(prtd->asp_link[1], asp_src_pong, INCR, W8BIT); - link = prtd->asp_link[0]; - edma_set_src_index(link, data_type, data_type * fifo_level); - link = prtd->asp_link[1]; - edma_set_src_index(link, data_type, data_type * fifo_level); + edma_set_src_index(prtd->asp_link[0], data_type, + data_type * fifo_level); + edma_set_src_index(prtd->asp_link[1], data_type, + data_type * fifo_level); - link = prtd->ram_link; - edma_set_src(link, runtime->dma_addr, INCR, W32BIT); + edma_set_src(prtd->ram_link, runtime->dma_addr, INCR, W32BIT); } else { dma_addr_t asp_dst_pong = iram_dma->addr + ping_size; ram_src_cidx = -ping_size; ram_dst_cidx = ping_size; - edma_set_dest(link, asp_dst_pong, INCR, W8BIT); + edma_set_dest(prtd->asp_link[1], asp_dst_pong, INCR, W8BIT); - link = prtd->asp_link[0]; - edma_set_dest_index(link, data_type, data_type * fifo_level); - link = prtd->asp_link[1]; - edma_set_dest_index(link, data_type, data_type * fifo_level); + edma_set_dest_index(prtd->asp_link[0], data_type, + data_type * fifo_level); + edma_set_dest_index(prtd->asp_link[1], data_type, + data_type * fifo_level); - link = prtd->ram_link; - edma_set_dest(link, runtime->dma_addr, INCR, W32BIT); + edma_set_dest(prtd->ram_link, runtime->dma_addr, INCR, W32BIT); } if (!fifo_level) { @@ -354,10 +353,9 @@ static int ping_pong_dma_setup(struct snd_pcm_substream *substream) count, fifo_level, ABSYNC); } - link = prtd->ram_link; - edma_set_src_index(link, ping_size, ram_src_cidx); - edma_set_dest_index(link, ping_size, ram_dst_cidx); - edma_set_transfer_params(link, ping_size, 2, + edma_set_src_index(prtd->ram_link, ping_size, ram_src_cidx); + edma_set_dest_index(prtd->ram_link, ping_size, ram_dst_cidx); + edma_set_transfer_params(prtd->ram_link, ping_size, 2, runtime->periods, 2, ASYNC); /* init master params */ @@ -406,32 +404,32 @@ static int request_ping_pong(struct snd_pcm_substream *substream, { dma_addr_t asp_src_ping; dma_addr_t asp_dst_ping; - int link; + int ret; struct davinci_pcm_dma_params *params = prtd->params; /* Request ram master channel */ - link = prtd->ram_channel = edma_alloc_channel(EDMA_CHANNEL_ANY, + ret = prtd->ram_channel = edma_alloc_channel(EDMA_CHANNEL_ANY, davinci_pcm_dma_irq, substream, prtd->params->ram_chan_q); - if (link < 0) + if (ret < 0) goto exit1; /* Request ram link channel */ - link = prtd->ram_link = edma_alloc_slot( + ret = prtd->ram_link = edma_alloc_slot( EDMA_CTLR(prtd->ram_channel), EDMA_SLOT_ANY); - if (link < 0) + if (ret < 0) goto exit2; - link = prtd->asp_link[1] = edma_alloc_slot( + ret = prtd->asp_link[1] = edma_alloc_slot( EDMA_CTLR(prtd->asp_channel), EDMA_SLOT_ANY); - if (link < 0) + if (ret < 0) goto exit3; prtd->ram_link2 = -1; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - link = prtd->ram_link2 = edma_alloc_slot( + ret = prtd->ram_link2 = edma_alloc_slot( EDMA_CTLR(prtd->ram_channel), EDMA_SLOT_ANY); - if (link < 0) + if (ret < 0) goto exit4; } /* circle ping-pong buffers */ @@ -448,36 +446,33 @@ static int request_ping_pong(struct snd_pcm_substream *substream, asp_dst_ping = iram_dma->addr; } /* ping */ - link = prtd->asp_link[0]; - edma_set_src(link, asp_src_ping, INCR, W16BIT); - edma_set_dest(link, asp_dst_ping, INCR, W16BIT); - edma_set_src_index(link, 0, 0); - edma_set_dest_index(link, 0, 0); + edma_set_src(prtd->asp_link[0], asp_src_ping, INCR, W16BIT); + edma_set_dest(prtd->asp_link[0], asp_dst_ping, INCR, W16BIT); + edma_set_src_index(prtd->asp_link[0], 0, 0); + edma_set_dest_index(prtd->asp_link[0], 0, 0); - edma_read_slot(link, &prtd->asp_params); + edma_read_slot(prtd->asp_link[0], &prtd->asp_params); prtd->asp_params.opt &= ~(TCCMODE | EDMA_TCC(0x3f) | TCINTEN); prtd->asp_params.opt |= TCCHEN | EDMA_TCC(prtd->ram_channel & 0x3f); - edma_write_slot(link, &prtd->asp_params); + edma_write_slot(prtd->asp_link[0], &prtd->asp_params); /* pong */ - link = prtd->asp_link[1]; - edma_set_src(link, asp_src_ping, INCR, W16BIT); - edma_set_dest(link, asp_dst_ping, INCR, W16BIT); - edma_set_src_index(link, 0, 0); - edma_set_dest_index(link, 0, 0); + edma_set_src(prtd->asp_link[1], asp_src_ping, INCR, W16BIT); + edma_set_dest(prtd->asp_link[1], asp_dst_ping, INCR, W16BIT); + edma_set_src_index(prtd->asp_link[1], 0, 0); + edma_set_dest_index(prtd->asp_link[1], 0, 0); - edma_read_slot(link, &prtd->asp_params); + edma_read_slot(prtd->asp_link[1], &prtd->asp_params); prtd->asp_params.opt &= ~(TCCMODE | EDMA_TCC(0x3f)); /* interrupt after every pong completion */ prtd->asp_params.opt |= TCINTEN | TCCHEN | EDMA_TCC(prtd->ram_channel & 0x3f); - edma_write_slot(link, &prtd->asp_params); + edma_write_slot(prtd->asp_link[1], &prtd->asp_params); /* ram */ - link = prtd->ram_link; - edma_set_src(link, iram_dma->addr, INCR, W32BIT); - edma_set_dest(link, iram_dma->addr, INCR, W32BIT); + edma_set_src(prtd->ram_link, iram_dma->addr, INCR, W32BIT); + edma_set_dest(prtd->ram_link, iram_dma->addr, INCR, W32BIT); pr_debug("%s: audio dma channels/slots in use for ram:%u %u %u," "for asp:%u %u %u\n", __func__, prtd->ram_channel, prtd->ram_link, prtd->ram_link2, @@ -494,7 +489,7 @@ exit2: edma_free_channel(prtd->ram_channel); prtd->ram_channel = -1; exit1: - return link; + return ret; } static int davinci_pcm_dma_request(struct snd_pcm_substream *substream) @@ -502,22 +497,22 @@ static int davinci_pcm_dma_request(struct snd_pcm_substream *substream) struct snd_dma_buffer *iram_dma; struct davinci_runtime_data *prtd = substream->runtime->private_data; struct davinci_pcm_dma_params *params = prtd->params; - int link; + int ret; if (!params) return -ENODEV; /* Request asp master DMA channel */ - link = prtd->asp_channel = edma_alloc_channel(params->channel, + ret = prtd->asp_channel = edma_alloc_channel(params->channel, davinci_pcm_dma_irq, substream, prtd->params->asp_chan_q); - if (link < 0) + if (ret < 0) goto exit1; /* Request asp link channels */ - link = prtd->asp_link[0] = edma_alloc_slot( + ret = prtd->asp_link[0] = edma_alloc_slot( EDMA_CTLR(prtd->asp_channel), EDMA_SLOT_ANY); - if (link < 0) + if (ret < 0) goto exit2; iram_dma = (struct snd_dma_buffer *)substream->dma_buffer.private_data; @@ -537,17 +532,17 @@ static int davinci_pcm_dma_request(struct snd_pcm_substream *substream) * the buffer and its length (ccnt) ... use it as a template * so davinci_pcm_enqueue_dma() takes less time in IRQ. */ - edma_read_slot(link, &prtd->asp_params); + edma_read_slot(prtd->asp_link[0], &prtd->asp_params); prtd->asp_params.opt |= TCINTEN | EDMA_TCC(EDMA_CHAN_SLOT(prtd->asp_channel)); - prtd->asp_params.link_bcntrld = EDMA_CHAN_SLOT(link) << 5; - edma_write_slot(link, &prtd->asp_params); + prtd->asp_params.link_bcntrld = EDMA_CHAN_SLOT(prtd->asp_link[0]) << 5; + edma_write_slot(prtd->asp_link[0], &prtd->asp_params); return 0; exit2: edma_free_channel(prtd->asp_channel); prtd->asp_channel = -1; exit1: - return link; + return ret; } static int davinci_pcm_trigger(struct snd_pcm_substream *substream, int cmd) diff --git a/sound/soc/ep93xx/edb93xx.c b/sound/soc/ep93xx/edb93xx.c index d3aa15119d26..0134d4e9131c 100644 --- a/sound/soc/ep93xx/edb93xx.c +++ b/sound/soc/ep93xx/edb93xx.c @@ -28,12 +28,6 @@ #include <mach/hardware.h> #include "ep93xx-pcm.h" -#define edb93xx_has_audio() (machine_is_edb9301() || \ - machine_is_edb9302() || \ - machine_is_edb9302a() || \ - machine_is_edb9307a() || \ - machine_is_edb9315a()) - static int edb93xx_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -94,49 +88,61 @@ static struct snd_soc_card snd_soc_edb93xx = { .num_links = 1, }; -static struct platform_device *edb93xx_snd_device; - -static int __init edb93xx_init(void) +static int __devinit edb93xx_probe(struct platform_device *pdev) { + struct snd_soc_card *card = &snd_soc_edb93xx; int ret; - if (!edb93xx_has_audio()) - return -ENODEV; - ret = ep93xx_i2s_acquire(EP93XX_SYSCON_DEVCFG_I2SONAC97, EP93XX_SYSCON_I2SCLKDIV_ORIDE | EP93XX_SYSCON_I2SCLKDIV_SPOL); if (ret) return ret; - edb93xx_snd_device = platform_device_alloc("soc-audio", -1); - if (!edb93xx_snd_device) { - ret = -ENOMEM; - goto free_i2s; + card->dev = &pdev->dev; + + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", + ret); + ep93xx_i2s_release(); } - platform_set_drvdata(edb93xx_snd_device, &snd_soc_edb93xx); - ret = platform_device_add(edb93xx_snd_device); - if (ret) - goto device_put; + return ret; +} - return 0; +static int __devexit edb93xx_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); -device_put: - platform_device_put(edb93xx_snd_device); -free_i2s: + snd_soc_unregister_card(card); ep93xx_i2s_release(); - return ret; + + return 0; +} + +static struct platform_driver edb93xx_driver = { + .driver = { + .name = "edb93xx-audio", + .owner = THIS_MODULE, + }, + .probe = edb93xx_probe, + .remove = __devexit_p(edb93xx_remove), +}; + +static int __init edb93xx_init(void) +{ + return platform_driver_register(&edb93xx_driver); } module_init(edb93xx_init); static void __exit edb93xx_exit(void) { - platform_device_unregister(edb93xx_snd_device); - ep93xx_i2s_release(); + platform_driver_unregister(&edb93xx_driver); } module_exit(edb93xx_exit); MODULE_AUTHOR("Alexander Sverdlin <subaparts@yandex.ru>"); MODULE_DESCRIPTION("ALSA SoC EDB93xx"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:edb93xx-audio"); diff --git a/sound/soc/ep93xx/ep93xx-ac97.c b/sound/soc/ep93xx/ep93xx-ac97.c index c7417c76552b..3cd6158d83e1 100644 --- a/sound/soc/ep93xx/ep93xx-ac97.c +++ b/sound/soc/ep93xx/ep93xx-ac97.c @@ -335,7 +335,7 @@ static struct snd_soc_dai_ops ep93xx_ac97_dai_ops = { .trigger = ep93xx_ac97_trigger, }; -struct snd_soc_dai_driver ep93xx_ac97_dai = { +static struct snd_soc_dai_driver ep93xx_ac97_dai = { .name = "ep93xx-ac97", .id = 0, .ac97_control = 1, diff --git a/sound/soc/ep93xx/ep93xx-pcm.c b/sound/soc/ep93xx/ep93xx-pcm.c index 8dfd3ad84b19..d00230a591b1 100644 --- a/sound/soc/ep93xx/ep93xx-pcm.c +++ b/sound/soc/ep93xx/ep93xx-pcm.c @@ -355,3 +355,4 @@ module_exit(ep93xx_soc_platform_exit); MODULE_AUTHOR("Ryan Mallon"); MODULE_DESCRIPTION("EP93xx ALSA PCM interface"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:ep93xx-pcm-audio"); diff --git a/sound/soc/ep93xx/simone.c b/sound/soc/ep93xx/simone.c index 286817946c56..968cb316d511 100644 --- a/sound/soc/ep93xx/simone.c +++ b/sound/soc/ep93xx/simone.c @@ -39,53 +39,61 @@ static struct snd_soc_card snd_soc_simone = { }; static struct platform_device *simone_snd_ac97_device; -static struct platform_device *simone_snd_device; -static int __init simone_init(void) +static int __devinit simone_probe(struct platform_device *pdev) { + struct snd_soc_card *card = &snd_soc_simone; int ret; - if (!machine_is_sim_one()) - return -ENODEV; - - simone_snd_ac97_device = platform_device_alloc("ac97-codec", -1); - if (!simone_snd_ac97_device) - return -ENOMEM; + simone_snd_ac97_device = platform_device_register_simple("ac97-codec", + -1, NULL, 0); + if (IS_ERR(simone_snd_ac97_device)) + return PTR_ERR(simone_snd_ac97_device); - ret = platform_device_add(simone_snd_ac97_device); - if (ret) - goto fail1; + card->dev = &pdev->dev; - simone_snd_device = platform_device_alloc("soc-audio", -1); - if (!simone_snd_device) { - ret = -ENOMEM; - goto fail2; + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", + ret); + platform_device_unregister(simone_snd_ac97_device); } - platform_set_drvdata(simone_snd_device, &snd_soc_simone); - ret = platform_device_add(simone_snd_device); - if (ret) - goto fail3; + return ret; +} + +static int __devexit simone_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + + snd_soc_unregister_card(card); + platform_device_unregister(simone_snd_ac97_device); return 0; +} -fail3: - platform_device_put(simone_snd_device); -fail2: - platform_device_del(simone_snd_ac97_device); -fail1: - platform_device_put(simone_snd_ac97_device); - return ret; +static struct platform_driver simone_driver = { + .driver = { + .name = "simone-audio", + .owner = THIS_MODULE, + }, + .probe = simone_probe, + .remove = __devexit_p(simone_remove), +}; + +static int __init simone_init(void) +{ + return platform_driver_register(&simone_driver); } module_init(simone_init); static void __exit simone_exit(void) { - platform_device_unregister(simone_snd_device); - platform_device_unregister(simone_snd_ac97_device); + platform_driver_unregister(&simone_driver); } module_exit(simone_exit); MODULE_DESCRIPTION("ALSA SoC Simplemachines Sim.One"); MODULE_AUTHOR("Mika Westerberg <mika.westerberg@iki.fi>"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:simone-audio"); diff --git a/sound/soc/ep93xx/snappercl15.c b/sound/soc/ep93xx/snappercl15.c index c8aa8a5003ca..f74ac54c285a 100644 --- a/sound/soc/ep93xx/snappercl15.c +++ b/sound/soc/ep93xx/snappercl15.c @@ -104,37 +104,56 @@ static struct snd_soc_card snd_soc_snappercl15 = { .num_links = 1, }; -static struct platform_device *snappercl15_snd_device; - -static int __init snappercl15_init(void) +static int __devinit snappercl15_probe(struct platform_device *pdev) { + struct snd_soc_card *card = &snd_soc_snappercl15; int ret; - if (!machine_is_snapper_cl15()) - return -ENODEV; - ret = ep93xx_i2s_acquire(EP93XX_SYSCON_DEVCFG_I2SONAC97, EP93XX_SYSCON_I2SCLKDIV_ORIDE | EP93XX_SYSCON_I2SCLKDIV_SPOL); if (ret) return ret; - snappercl15_snd_device = platform_device_alloc("soc-audio", -1); - if (!snappercl15_snd_device) - return -ENOMEM; - - platform_set_drvdata(snappercl15_snd_device, &snd_soc_snappercl15); - ret = platform_device_add(snappercl15_snd_device); - if (ret) - platform_device_put(snappercl15_snd_device); + card->dev = &pdev->dev; + + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", + ret); + ep93xx_i2s_release(); + } return ret; } -static void __exit snappercl15_exit(void) +static int __devexit snappercl15_remove(struct platform_device *pdev) { - platform_device_unregister(snappercl15_snd_device); + struct snd_soc_card *card = platform_get_drvdata(pdev); + + snd_soc_unregister_card(card); ep93xx_i2s_release(); + + return 0; +} + +static struct platform_driver snappercl15_driver = { + .driver = { + .name = "snappercl15-audio", + .owner = THIS_MODULE, + }, + .probe = snappercl15_probe, + .remove = __devexit_p(snappercl15_remove), +}; + +static int __init snappercl15_init(void) +{ + return platform_driver_register(&snappercl15_driver); +} + +static void __exit snappercl15_exit(void) +{ + platform_driver_unregister(&snappercl15_driver); } module_init(snappercl15_init); @@ -143,4 +162,4 @@ module_exit(snappercl15_exit); MODULE_AUTHOR("Ryan Mallon"); MODULE_DESCRIPTION("ALSA SoC Snapper CL15"); MODULE_LICENSE("GPL"); - +MODULE_ALIAS("platform:snappercl15-audio"); diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index cb50598338e9..ef15402a3bc4 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -297,7 +297,6 @@ static irqreturn_t fsl_dma_isr(int irq, void *dev_id) static int fsl_dma_new(struct snd_soc_pcm_runtime *rtd) { struct snd_card *card = rtd->card->snd_card; - struct snd_soc_dai *dai = rtd->cpu_dai; struct snd_pcm *pcm = rtd->pcm; static u64 fsl_dma_dmamask = DMA_BIT_MASK(36); int ret; diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index d48afea5d93d..0268cf989736 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -78,7 +78,6 @@ * @second_stream: pointer to second stream * @playback: the number of playback streams opened * @capture: the number of capture streams opened - * @asynchronous: 0=synchronous mode, 1=asynchronous mode * @cpu_dai: the CPU DAI for this device * @dev_attr: the sysfs device attribute structure * @stats: SSI statistics @@ -90,9 +89,6 @@ struct fsl_ssi_private { unsigned int irq; struct snd_pcm_substream *first_stream; struct snd_pcm_substream *second_stream; - unsigned int playback; - unsigned int capture; - int asynchronous; unsigned int fifo_depth; struct snd_soc_dai_driver cpu_dai_drv; struct device_attribute dev_attr; @@ -281,24 +277,18 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(rtd->cpu_dai); + struct fsl_ssi_private *ssi_private = + snd_soc_dai_get_drvdata(rtd->cpu_dai); + int synchronous = ssi_private->cpu_dai_drv.symmetric_rates; /* * If this is the first stream opened, then request the IRQ * and initialize the SSI registers. */ - if (!ssi_private->playback && !ssi_private->capture) { + if (!ssi_private->first_stream) { struct ccsr_ssi __iomem *ssi = ssi_private->ssi; - int ret; - - /* The 'name' should not have any slashes in it. */ - ret = request_irq(ssi_private->irq, fsl_ssi_isr, 0, - ssi_private->name, ssi_private); - if (ret < 0) { - dev_err(substream->pcm->card->dev, - "could not claim irq %u\n", ssi_private->irq); - return ret; - } + + ssi_private->first_stream = substream; /* * Section 16.5 of the MPC8610 reference manual says that the @@ -316,7 +306,7 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, clrsetbits_be32(&ssi->scr, CCSR_SSI_SCR_I2S_MODE_MASK | CCSR_SSI_SCR_SYN, CCSR_SSI_SCR_TFR_CLK_DIS | CCSR_SSI_SCR_I2S_MODE_SLAVE - | (ssi_private->asynchronous ? 0 : CCSR_SSI_SCR_SYN)); + | (synchronous ? CCSR_SSI_SCR_SYN : 0)); out_be32(&ssi->stcr, CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TFEN0 | @@ -333,7 +323,7 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, * master. */ - /* 4. Enable the interrupts and DMA requests */ + /* Enable the interrupts and DMA requests */ out_be32(&ssi->sier, SIER_FLAGS); /* @@ -362,58 +352,47 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, * this is bad is because at this point, the PCM driver has not * finished initializing the DMA controller. */ - } + } else { + if (synchronous) { + struct snd_pcm_runtime *first_runtime = + ssi_private->first_stream->runtime; + /* + * This is the second stream open, and we're in + * synchronous mode, so we need to impose sample + * sample size constraints. This is because STCCR is + * used for playback and capture in synchronous mode, + * so there's no way to specify different word + * lengths. + * + * Note that this can cause a race condition if the + * second stream is opened before the first stream is + * fully initialized. We provide some protection by + * checking to make sure the first stream is + * initialized, but it's not perfect. ALSA sometimes + * re-initializes the driver with a different sample + * rate or size. If the second stream is opened + * before the first stream has received its final + * parameters, then the second stream may be + * constrained to the wrong sample rate or size. + */ + if (!first_runtime->sample_bits) { + dev_err(substream->pcm->card->dev, + "set sample size in %s stream first\n", + substream->stream == + SNDRV_PCM_STREAM_PLAYBACK + ? "capture" : "playback"); + return -EAGAIN; + } - if (!ssi_private->first_stream) - ssi_private->first_stream = substream; - else { - /* This is the second stream open, so we need to impose sample - * rate and maybe sample size constraints. Note that this can - * cause a race condition if the second stream is opened before - * the first stream is fully initialized. - * - * We provide some protection by checking to make sure the first - * stream is initialized, but it's not perfect. ALSA sometimes - * re-initializes the driver with a different sample rate or - * size. If the second stream is opened before the first stream - * has received its final parameters, then the second stream may - * be constrained to the wrong sample rate or size. - * - * FIXME: This code does not handle opening and closing streams - * repeatedly. If you open two streams and then close the first - * one, you may not be able to open another stream until you - * close the second one as well. - */ - struct snd_pcm_runtime *first_runtime = - ssi_private->first_stream->runtime; - - if (!first_runtime->sample_bits) { - dev_err(substream->pcm->card->dev, - "set sample size in %s stream first\n", - substream->stream == SNDRV_PCM_STREAM_PLAYBACK - ? "capture" : "playback"); - return -EAGAIN; - } - - /* If we're in synchronous mode, then we need to constrain - * the sample size as well. We don't support independent sample - * rates in asynchronous mode. - */ - if (!ssi_private->asynchronous) snd_pcm_hw_constraint_minmax(substream->runtime, SNDRV_PCM_HW_PARAM_SAMPLE_BITS, first_runtime->sample_bits, first_runtime->sample_bits); + } ssi_private->second_stream = substream; } - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - ssi_private->playback++; - - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) - ssi_private->capture++; - return 0; } @@ -434,24 +413,35 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hw_params, struct snd_soc_dai *cpu_dai) { struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(cpu_dai); + struct ccsr_ssi __iomem *ssi = ssi_private->ssi; + unsigned int sample_size = + snd_pcm_format_width(params_format(hw_params)); + u32 wl = CCSR_SSI_SxCCR_WL(sample_size); + int enabled = in_be32(&ssi->scr) & CCSR_SSI_SCR_SSIEN; - if (substream == ssi_private->first_stream) { - struct ccsr_ssi __iomem *ssi = ssi_private->ssi; - unsigned int sample_size = - snd_pcm_format_width(params_format(hw_params)); - u32 wl = CCSR_SSI_SxCCR_WL(sample_size); + /* + * If we're in synchronous mode, and the SSI is already enabled, + * then STCCR is already set properly. + */ + if (enabled && ssi_private->cpu_dai_drv.symmetric_rates) + return 0; - /* The SSI should always be disabled at this points (SSIEN=0) */ + /* + * FIXME: The documentation says that SxCCR[WL] should not be + * modified while the SSI is enabled. The only time this can + * happen is if we're trying to do simultaneous playback and + * capture in asynchronous mode. Unfortunately, I have been enable + * to get that to work at all on the P1022DS. Therefore, we don't + * bother to disable/enable the SSI when setting SxCCR[WL], because + * the SSI will stop anyway. Maybe one day, this will get fixed. + */ - /* In synchronous mode, the SSI uses STCCR for capture */ - if ((substream->stream == SNDRV_PCM_STREAM_PLAYBACK) || - !ssi_private->asynchronous) - clrsetbits_be32(&ssi->stccr, - CCSR_SSI_SxCCR_WL_MASK, wl); - else - clrsetbits_be32(&ssi->srccr, - CCSR_SSI_SxCCR_WL_MASK, wl); - } + /* In synchronous mode, the SSI uses STCCR for capture */ + if ((substream->stream == SNDRV_PCM_STREAM_PLAYBACK) || + ssi_private->cpu_dai_drv.symmetric_rates) + clrsetbits_be32(&ssi->stccr, CCSR_SSI_SxCCR_WL_MASK, wl); + else + clrsetbits_be32(&ssi->srccr, CCSR_SSI_SxCCR_WL_MASK, wl); return 0; } @@ -474,7 +464,6 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd, switch (cmd) { case SNDRV_PCM_TRIGGER_START: - clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) setbits32(&ssi->scr, @@ -510,27 +499,18 @@ static void fsl_ssi_shutdown(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(rtd->cpu_dai); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - ssi_private->playback--; - - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) - ssi_private->capture--; - if (ssi_private->first_stream == substream) ssi_private->first_stream = ssi_private->second_stream; ssi_private->second_stream = NULL; /* - * If this is the last active substream, disable the SSI and release - * the IRQ. + * If this is the last active substream, disable the SSI. */ - if (!ssi_private->playback && !ssi_private->capture) { + if (!ssi_private->first_stream) { struct ccsr_ssi __iomem *ssi = ssi_private->ssi; clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); - - free_irq(ssi_private->irq, ssi_private); } } @@ -675,22 +655,33 @@ static int __devinit fsl_ssi_probe(struct platform_device *pdev) ret = of_address_to_resource(np, 0, &res); if (ret) { dev_err(&pdev->dev, "could not determine device resources\n"); - kfree(ssi_private); - return ret; + goto error_kmalloc; } ssi_private->ssi = of_iomap(np, 0); if (!ssi_private->ssi) { dev_err(&pdev->dev, "could not map device resources\n"); - kfree(ssi_private); - return -ENOMEM; + ret = -ENOMEM; + goto error_kmalloc; } ssi_private->ssi_phys = res.start; + ssi_private->irq = irq_of_parse_and_map(np, 0); + if (ssi_private->irq == NO_IRQ) { + dev_err(&pdev->dev, "no irq for node %s\n", np->full_name); + ret = -ENXIO; + goto error_iomap; + } + + /* The 'name' should not have any slashes in it. */ + ret = request_irq(ssi_private->irq, fsl_ssi_isr, 0, ssi_private->name, + ssi_private); + if (ret < 0) { + dev_err(&pdev->dev, "could not claim irq %u\n", ssi_private->irq); + goto error_irqmap; + } /* Are the RX and the TX clocks locked? */ - if (of_find_property(np, "fsl,ssi-asynchronous", NULL)) - ssi_private->asynchronous = 1; - else + if (!of_find_property(np, "fsl,ssi-asynchronous", NULL)) ssi_private->cpu_dai_drv.symmetric_rates = 1; /* Determine the FIFO depth. */ @@ -711,7 +702,7 @@ static int __devinit fsl_ssi_probe(struct platform_device *pdev) if (ret) { dev_err(&pdev->dev, "could not create sysfs %s file\n", ssi_private->dev_attr.attr.name); - goto error; + goto error_irq; } /* Register with ASoC */ @@ -720,7 +711,7 @@ static int __devinit fsl_ssi_probe(struct platform_device *pdev) ret = snd_soc_register_dai(&pdev->dev, &ssi_private->cpu_dai_drv); if (ret) { dev_err(&pdev->dev, "failed to register DAI: %d\n", ret); - goto error; + goto error_dev; } /* Trigger the machine driver's probe function. The platform driver @@ -741,18 +732,28 @@ static int __devinit fsl_ssi_probe(struct platform_device *pdev) if (IS_ERR(ssi_private->pdev)) { ret = PTR_ERR(ssi_private->pdev); dev_err(&pdev->dev, "failed to register platform: %d\n", ret); - goto error; + goto error_dai; } return 0; -error: +error_dai: snd_soc_unregister_dai(&pdev->dev); + +error_dev: dev_set_drvdata(&pdev->dev, NULL); - if (dev_attr) - device_remove_file(&pdev->dev, dev_attr); + device_remove_file(&pdev->dev, dev_attr); + +error_irq: + free_irq(ssi_private->irq, ssi_private); + +error_irqmap: irq_dispose_mapping(ssi_private->irq); + +error_iomap: iounmap(ssi_private->ssi); + +error_kmalloc: kfree(ssi_private); return ret; @@ -766,6 +767,9 @@ static int fsl_ssi_remove(struct platform_device *pdev) snd_soc_unregister_dai(&pdev->dev); device_remove_file(&pdev->dev, &ssi_private->dev_attr); + free_irq(ssi_private->irq, ssi_private); + irq_dispose_mapping(ssi_private->irq); + kfree(ssi_private); dev_set_drvdata(&pdev->dev, NULL); diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index 358f0baaf71b..31af405bda84 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -505,7 +505,7 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev) return 0; error_sound: - platform_device_unregister(sound_device); + platform_device_put(sound_device); error: kfree(machine_data); error_alloc: diff --git a/sound/soc/fsl/p1022_ds.c b/sound/soc/fsl/p1022_ds.c index fcb862eb0c73..2c064a9824ad 100644 --- a/sound/soc/fsl/p1022_ds.c +++ b/sound/soc/fsl/p1022_ds.c @@ -267,7 +267,7 @@ static int codec_node_dev_name(struct device_node *np, char *buf, size_t len) if (bus < 0) return bus; - snprintf(buf, len, "%s-codec.%u-%04x", temp, bus, addr); + snprintf(buf, len, "%s.%u-%04x", temp, bus, addr); return 0; } @@ -506,7 +506,7 @@ static int p1022_ds_probe(struct platform_device *pdev) error: if (sound_device) - platform_device_unregister(sound_device); + platform_device_put(sound_device); kfree(mdata); error_put: diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig index bb699bb55a50..b133bfcc5848 100644 --- a/sound/soc/imx/Kconfig +++ b/sound/soc/imx/Kconfig @@ -29,7 +29,7 @@ config SND_MXC_SOC_WM1133_EV1 config SND_SOC_MX27VIS_AIC32X4 tristate "SoC audio support for Visstrim M10 boards" depends on MACH_IMX27_VISSTRIM_M10 - select SND_SOC_TVL320AIC32X4 + select SND_SOC_TLV320AIC32X4 select SND_MXC_SOC_MX2 help Say Y if you want to add support for SoC audio on Visstrim SM10 @@ -50,6 +50,7 @@ config SND_SOC_EUKREA_TLV320 || MACH_EUKREA_MBIMXSD25_BASEBOARD \ || MACH_EUKREA_MBIMXSD35_BASEBOARD \ || MACH_EUKREA_MBIMXSD51_BASEBOARD + depends on I2C select SND_SOC_TLV320AIC23 select SND_MXC_SOC_FIQ help diff --git a/sound/soc/imx/imx-pcm-fiq.c b/sound/soc/imx/imx-pcm-fiq.c index 7945625e0e08..8df0fae21943 100644 --- a/sound/soc/imx/imx-pcm-fiq.c +++ b/sound/soc/imx/imx-pcm-fiq.c @@ -240,25 +240,23 @@ static int ssi_irq = 0; static int imx_pcm_fiq_new(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_dai *dai = rtd->cpu_dai; struct snd_pcm *pcm = rtd->pcm; + struct snd_pcm_substream *substream; int ret; ret = imx_pcm_new(rtd); if (ret) return ret; - if (dai->driver->playback.channels_min) { - struct snd_pcm_substream *substream = - pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream; + substream = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream; + if (substream) { struct snd_dma_buffer *buf = &substream->dma_buffer; imx_ssi_fiq_tx_buffer = (unsigned long)buf->area; } - if (dai->driver->capture.channels_min) { - struct snd_pcm_substream *substream = - pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream; + substream = pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream; + if (substream) { struct snd_dma_buffer *buf = &substream->dma_buffer; imx_ssi_fiq_rx_buffer = (unsigned long)buf->area; diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c index 10a8e2783751..4c05e2b8f4d2 100644 --- a/sound/soc/imx/imx-ssi.c +++ b/sound/soc/imx/imx-ssi.c @@ -357,8 +357,8 @@ int snd_imx_pcm_mmap(struct snd_pcm_substream *substream, struct snd_pcm_runtime *runtime = substream->runtime; int ret; - ret = dma_mmap_coherent(NULL, vma, runtime->dma_area, - runtime->dma_addr, runtime->dma_bytes); + ret = dma_mmap_writecombine(substream->pcm->card->dev, vma, + runtime->dma_area, runtime->dma_addr, runtime->dma_bytes); pr_debug("%s: ret: %d %p 0x%08x 0x%08x\n", __func__, ret, runtime->dma_area, @@ -391,7 +391,6 @@ static u64 imx_pcm_dmamask = DMA_BIT_MASK(32); int imx_pcm_new(struct snd_soc_pcm_runtime *rtd) { struct snd_card *card = rtd->card->snd_card; - struct snd_soc_dai *dai = rtd->cpu_dai; struct snd_pcm *pcm = rtd->pcm; int ret = 0; @@ -399,14 +398,14 @@ int imx_pcm_new(struct snd_soc_pcm_runtime *rtd) card->dev->dma_mask = &imx_pcm_dmamask; if (!card->dev->coherent_dma_mask) card->dev->coherent_dma_mask = DMA_BIT_MASK(32); - if (dai->driver->playback.channels_min) { + if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { ret = imx_pcm_preallocate_dma_buffer(pcm, SNDRV_PCM_STREAM_PLAYBACK); if (ret) goto out; } - if (dai->driver->capture.channels_min) { + if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { ret = imx_pcm_preallocate_dma_buffer(pcm, SNDRV_PCM_STREAM_CAPTURE); if (ret) diff --git a/sound/soc/imx/imx-ssi.h b/sound/soc/imx/imx-ssi.h index 0a84cec3599e..1072dfb53e47 100644 --- a/sound/soc/imx/imx-ssi.h +++ b/sound/soc/imx/imx-ssi.h @@ -218,12 +218,6 @@ struct imx_ssi { struct platform_device *soc_platform_pdev_fiq; }; -struct snd_soc_platform *imx_ssi_fiq_init(struct platform_device *pdev, - struct imx_ssi *ssi); -void imx_ssi_fiq_exit(struct platform_device *pdev, struct imx_ssi *ssi); -struct snd_soc_platform *imx_ssi_dma_mx2_init(struct platform_device *pdev, - struct imx_ssi *ssi); - int snd_imx_pcm_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *vma); int imx_pcm_new(struct snd_soc_pcm_runtime *rtd); void imx_pcm_free(struct snd_pcm *pcm); diff --git a/sound/soc/jz4740/jz4740-pcm.c b/sound/soc/jz4740/jz4740-pcm.c index a7c9578be983..d1989cde9f14 100644 --- a/sound/soc/jz4740/jz4740-pcm.c +++ b/sound/soc/jz4740/jz4740-pcm.c @@ -299,7 +299,7 @@ static void jz4740_pcm_free(struct snd_pcm *pcm) static u64 jz4740_pcm_dmamask = DMA_BIT_MASK(32); -int jz4740_pcm_new(struct snd_soc_pcm_runtime *rtd) +static int jz4740_pcm_new(struct snd_soc_pcm_runtime *rtd) { struct snd_card *card = rtd->card->snd_card; struct snd_soc_dai *dai = rtd->cpu_dai; diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index d0bcf3fcea01..715e841c0507 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -476,7 +476,7 @@ static __devexit int kirkwood_i2s_dev_remove(struct platform_device *pdev) static struct platform_driver kirkwood_i2s_driver = { .probe = kirkwood_i2s_dev_probe, - .remove = kirkwood_i2s_dev_remove, + .remove = __devexit_p(kirkwood_i2s_dev_remove), .driver = { .name = DRV_NAME, .owner = THIS_MODULE, diff --git a/sound/soc/kirkwood/kirkwood-t5325.c b/sound/soc/kirkwood/kirkwood-t5325.c index c8d21956ab52..c772b3cf4039 100644 --- a/sound/soc/kirkwood/kirkwood-t5325.c +++ b/sound/soc/kirkwood/kirkwood-t5325.c @@ -79,8 +79,6 @@ static int t5325_dai_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_enable_pin(dapm, "Headphone Jack"); snd_soc_dapm_enable_pin(dapm, "Speaker"); - snd_soc_dapm_sync(dapm); - return 0; } diff --git a/sound/soc/mid-x86/mfld_machine.c b/sound/soc/mid-x86/mfld_machine.c index 429aa1be2cff..598f48c0d8f5 100644 --- a/sound/soc/mid-x86/mfld_machine.c +++ b/sound/soc/mid-x86/mfld_machine.c @@ -54,9 +54,7 @@ static unsigned int hs_switch; static unsigned int lo_dac; struct mfld_mc_private { - struct platform_device *socdev; void __iomem *int_base; - struct snd_soc_codec *codec; u8 interrupt_status; }; @@ -235,7 +233,6 @@ static int mfld_init(struct snd_soc_pcm_runtime *runtime) /* always connected */ snd_soc_dapm_enable_pin(dapm, "Headphones"); snd_soc_dapm_enable_pin(dapm, "Mic"); - snd_soc_dapm_sync(dapm); ret_val = snd_soc_add_controls(codec, mfld_snd_controls, ARRAY_SIZE(mfld_snd_controls)); @@ -253,7 +250,6 @@ static int mfld_init(struct snd_soc_pcm_runtime *runtime) /* we dont use linein in this so set to NC */ snd_soc_dapm_disable_pin(dapm, "LINEINL"); snd_soc_dapm_disable_pin(dapm, "LINEINR"); - snd_soc_dapm_sync(dapm); /* Headset and button jack detection */ ret_val = snd_soc_jack_new(codec, "Intel(R) MID Audio Jack", diff --git a/sound/soc/mid-x86/sst_platform.c b/sound/soc/mid-x86/sst_platform.c index 3e7826058efe..7df8c58ba50a 100644 --- a/sound/soc/mid-x86/sst_platform.c +++ b/sound/soc/mid-x86/sst_platform.c @@ -63,7 +63,7 @@ static struct snd_pcm_hardware sst_platform_pcm_hw = { }; /* MFLD - MSIC */ -struct snd_soc_dai_driver sst_platform_dai[] = { +static struct snd_soc_dai_driver sst_platform_dai[] = { { .name = "Headset-cpu-dai", .id = 0, @@ -226,13 +226,18 @@ static int sst_platform_init_stream(struct snd_pcm_substream *substream) static int sst_platform_open(struct snd_pcm_substream *substream) { - struct snd_pcm_runtime *runtime; + struct snd_pcm_runtime *runtime = substream->runtime; struct sst_runtime_stream *stream; int ret_val = 0; pr_debug("sst_platform_open called\n"); - runtime = substream->runtime; - runtime->hw = sst_platform_pcm_hw; + + snd_soc_set_runtime_hwparams(substream, &sst_platform_pcm_hw); + ret_val = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret_val < 0) + return ret_val; + stream = kzalloc(sizeof(*stream), GFP_KERNEL); if (!stream) return -ENOMEM; @@ -259,8 +264,8 @@ static int sst_platform_open(struct snd_pcm_substream *substream) return ret_val; } runtime->private_data = stream; - return snd_pcm_hw_constraint_integer(runtime, - SNDRV_PCM_HW_PARAM_PERIODS); + + return 0; } static int sst_platform_close(struct snd_pcm_substream *substream) @@ -469,7 +474,7 @@ static struct platform_driver sst_platform_driver = { static int __init sst_soc_platform_init(void) { pr_debug("sst_soc_platform_init called\n"); - return platform_driver_register(&sst_platform_driver); + return platform_driver_register(&sst_platform_driver); } module_init(sst_soc_platform_init); diff --git a/sound/soc/mxs/Kconfig b/sound/soc/mxs/Kconfig new file mode 100644 index 000000000000..e4ba8d5f25fa --- /dev/null +++ b/sound/soc/mxs/Kconfig @@ -0,0 +1,20 @@ +menuconfig SND_MXS_SOC + tristate "SoC Audio for Freescale MXS CPUs" + depends on ARCH_MXS + select SND_PCM + help + Say Y or M if you want to add support for codecs attached to + the MXS SAIF interface. + + +if SND_MXS_SOC + +config SND_SOC_MXS_SGTL5000 + tristate "SoC Audio support for i.MX boards with sgtl5000" + depends on I2C + select SND_SOC_SGTL5000 + help + Say Y if you want to add support for SoC audio on an MXS board with + a sgtl5000 codec. + +endif # SND_MXS_SOC diff --git a/sound/soc/mxs/Makefile b/sound/soc/mxs/Makefile new file mode 100644 index 000000000000..565b5b51e8b7 --- /dev/null +++ b/sound/soc/mxs/Makefile @@ -0,0 +1,10 @@ +# MXS Platform Support +snd-soc-mxs-objs := mxs-saif.o +snd-soc-mxs-pcm-objs := mxs-pcm.o + +obj-$(CONFIG_SND_MXS_SOC) += snd-soc-mxs.o snd-soc-mxs-pcm.o + +# i.MX Machine Support +snd-soc-mxs-sgtl5000-objs := mxs-sgtl5000.o + +obj-$(CONFIG_SND_SOC_MXS_SGTL5000) += snd-soc-mxs-sgtl5000.o diff --git a/sound/soc/mxs/mxs-pcm.c b/sound/soc/mxs/mxs-pcm.c new file mode 100644 index 000000000000..dea5aa4aa647 --- /dev/null +++ b/sound/soc/mxs/mxs-pcm.c @@ -0,0 +1,359 @@ +/* + * Copyright (C) 2011 Freescale Semiconductor, Inc. All Rights Reserved. + * + * Based on sound/soc/imx/imx-pcm-dma-mx2.c + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#include <linux/clk.h> +#include <linux/delay.h> +#include <linux/device.h> +#include <linux/dma-mapping.h> +#include <linux/init.h> +#include <linux/interrupt.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/dmaengine.h> + +#include <sound/core.h> +#include <sound/initval.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include <mach/dma.h> +#include "mxs-pcm.h" + +static struct snd_pcm_hardware snd_mxs_hardware = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_RESUME | + SNDRV_PCM_INFO_INTERLEAVED, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S20_3LE | + SNDRV_PCM_FMTBIT_S24_LE, + .channels_min = 2, + .channels_max = 2, + .period_bytes_min = 32, + .period_bytes_max = 8192, + .periods_min = 1, + .periods_max = 52, + .buffer_bytes_max = 64 * 1024, + .fifo_size = 32, + +}; + +static void audio_dma_irq(void *data) +{ + struct snd_pcm_substream *substream = (struct snd_pcm_substream *)data; + struct snd_pcm_runtime *runtime = substream->runtime; + struct mxs_pcm_runtime_data *iprtd = runtime->private_data; + + iprtd->offset += iprtd->period_bytes; + iprtd->offset %= iprtd->period_bytes * iprtd->periods; + snd_pcm_period_elapsed(substream); +} + +static bool filter(struct dma_chan *chan, void *param) +{ + struct mxs_pcm_runtime_data *iprtd = param; + struct mxs_pcm_dma_params *dma_params = iprtd->dma_params; + + if (!mxs_dma_is_apbx(chan)) + return false; + + if (chan->chan_id != dma_params->chan_num) + return false; + + chan->private = &iprtd->dma_data; + + return true; +} + +static int mxs_dma_alloc(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_pcm_runtime *runtime = substream->runtime; + struct mxs_pcm_runtime_data *iprtd = runtime->private_data; + dma_cap_mask_t mask; + + iprtd->dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + + dma_cap_zero(mask); + dma_cap_set(DMA_SLAVE, mask); + iprtd->dma_data.chan_irq = iprtd->dma_params->chan_irq; + iprtd->dma_chan = dma_request_channel(mask, filter, iprtd); + if (!iprtd->dma_chan) + return -EINVAL; + + return 0; +} + +static int snd_mxs_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct mxs_pcm_runtime_data *iprtd = runtime->private_data; + unsigned long dma_addr; + struct dma_chan *chan; + int ret; + + ret = mxs_dma_alloc(substream, params); + if (ret) + return ret; + chan = iprtd->dma_chan; + + iprtd->size = params_buffer_bytes(params); + iprtd->periods = params_periods(params); + iprtd->period_bytes = params_period_bytes(params); + iprtd->offset = 0; + iprtd->period_time = HZ / (params_rate(params) / + params_period_size(params)); + + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + + dma_addr = runtime->dma_addr; + + iprtd->buf = substream->dma_buffer.area; + + iprtd->desc = chan->device->device_prep_dma_cyclic(chan, dma_addr, + iprtd->period_bytes * iprtd->periods, + iprtd->period_bytes, + substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? + DMA_TO_DEVICE : DMA_FROM_DEVICE); + if (!iprtd->desc) { + dev_err(&chan->dev->device, "cannot prepare slave dma\n"); + return -EINVAL; + } + + iprtd->desc->callback = audio_dma_irq; + iprtd->desc->callback_param = substream; + + return 0; +} + +static int snd_mxs_pcm_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct mxs_pcm_runtime_data *iprtd = runtime->private_data; + + if (iprtd->dma_chan) { + dma_release_channel(iprtd->dma_chan); + iprtd->dma_chan = NULL; + } + + return 0; +} + +static int snd_mxs_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct mxs_pcm_runtime_data *iprtd = runtime->private_data; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + dmaengine_submit(iprtd->desc); + + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + dmaengine_terminate_all(iprtd->dma_chan); + + break; + default: + return -EINVAL; + } + + return 0; +} + +static snd_pcm_uframes_t snd_mxs_pcm_pointer( + struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct mxs_pcm_runtime_data *iprtd = runtime->private_data; + + return bytes_to_frames(substream->runtime, iprtd->offset); +} + +static int snd_mxs_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct mxs_pcm_runtime_data *iprtd; + int ret; + + iprtd = kzalloc(sizeof(*iprtd), GFP_KERNEL); + if (iprtd == NULL) + return -ENOMEM; + runtime->private_data = iprtd; + + ret = snd_pcm_hw_constraint_integer(substream->runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) { + kfree(iprtd); + return ret; + } + + snd_soc_set_runtime_hwparams(substream, &snd_mxs_hardware); + + return 0; +} + +static int snd_mxs_close(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct mxs_pcm_runtime_data *iprtd = runtime->private_data; + + kfree(iprtd); + + return 0; +} + +static int snd_mxs_pcm_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + return dma_mmap_writecombine(substream->pcm->card->dev, vma, + runtime->dma_area, + runtime->dma_addr, + runtime->dma_bytes); +} + +static struct snd_pcm_ops mxs_pcm_ops = { + .open = snd_mxs_open, + .close = snd_mxs_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_mxs_pcm_hw_params, + .hw_free = snd_mxs_pcm_hw_free, + .trigger = snd_mxs_pcm_trigger, + .pointer = snd_mxs_pcm_pointer, + .mmap = snd_mxs_pcm_mmap, +}; + +static int mxs_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) +{ + struct snd_pcm_substream *substream = pcm->streams[stream].substream; + struct snd_dma_buffer *buf = &substream->dma_buffer; + size_t size = snd_mxs_hardware.buffer_bytes_max; + + buf->dev.type = SNDRV_DMA_TYPE_DEV; + buf->dev.dev = pcm->card->dev; + buf->private_data = NULL; + buf->area = dma_alloc_writecombine(pcm->card->dev, size, + &buf->addr, GFP_KERNEL); + if (!buf->area) + return -ENOMEM; + buf->bytes = size; + + return 0; +} + +static u64 mxs_pcm_dmamask = DMA_BIT_MASK(32); +static int mxs_pcm_new(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_card *card = rtd->card->snd_card; + struct snd_pcm *pcm = rtd->pcm; + int ret = 0; + + if (!card->dev->dma_mask) + card->dev->dma_mask = &mxs_pcm_dmamask; + if (!card->dev->coherent_dma_mask) + card->dev->coherent_dma_mask = DMA_BIT_MASK(32); + + if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { + ret = mxs_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_PLAYBACK); + if (ret) + goto out; + } + + if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { + ret = mxs_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_CAPTURE); + if (ret) + goto out; + } + +out: + return ret; +} + +static void mxs_pcm_free(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream; + struct snd_dma_buffer *buf; + int stream; + + for (stream = 0; stream < 2; stream++) { + substream = pcm->streams[stream].substream; + if (!substream) + continue; + + buf = &substream->dma_buffer; + if (!buf->area) + continue; + + dma_free_writecombine(pcm->card->dev, buf->bytes, + buf->area, buf->addr); + buf->area = NULL; + } +} + +static struct snd_soc_platform_driver mxs_soc_platform = { + .ops = &mxs_pcm_ops, + .pcm_new = mxs_pcm_new, + .pcm_free = mxs_pcm_free, +}; + +static int __devinit mxs_soc_platform_probe(struct platform_device *pdev) +{ + return snd_soc_register_platform(&pdev->dev, &mxs_soc_platform); +} + +static int __devexit mxs_soc_platform_remove(struct platform_device *pdev) +{ + snd_soc_unregister_platform(&pdev->dev); + + return 0; +} + +static struct platform_driver mxs_pcm_driver = { + .driver = { + .name = "mxs-pcm-audio", + .owner = THIS_MODULE, + }, + .probe = mxs_soc_platform_probe, + .remove = __devexit_p(mxs_soc_platform_remove), +}; + +static int __init snd_mxs_pcm_init(void) +{ + return platform_driver_register(&mxs_pcm_driver); +} +module_init(snd_mxs_pcm_init); + +static void __exit snd_mxs_pcm_exit(void) +{ + platform_driver_unregister(&mxs_pcm_driver); +} +module_exit(snd_mxs_pcm_exit); diff --git a/sound/soc/mxs/mxs-pcm.h b/sound/soc/mxs/mxs-pcm.h new file mode 100644 index 000000000000..f55ac4f7a76a --- /dev/null +++ b/sound/soc/mxs/mxs-pcm.h @@ -0,0 +1,43 @@ +/* + * Copyright (C) 2011 Freescale Semiconductor, Inc. All Rights Reserved. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#ifndef _MXS_PCM_H +#define _MXS_PCM_H + +#include <mach/dma.h> + +struct mxs_pcm_dma_params { + int chan_irq; + int chan_num; +}; + +struct mxs_pcm_runtime_data { + int period_bytes; + int periods; + int dma; + unsigned long offset; + unsigned long size; + void *buf; + int period_time; + struct dma_async_tx_descriptor *desc; + struct dma_chan *dma_chan; + struct mxs_dma_data dma_data; + struct mxs_pcm_dma_params *dma_params; +}; + +#endif diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c new file mode 100644 index 000000000000..76dc74d24fc2 --- /dev/null +++ b/sound/soc/mxs/mxs-saif.c @@ -0,0 +1,798 @@ +/* + * Copyright 2011 Freescale Semiconductor, Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/dma-mapping.h> +#include <linux/clk.h> +#include <linux/delay.h> +#include <linux/time.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/saif.h> +#include <mach/dma.h> +#include <asm/mach-types.h> +#include <mach/hardware.h> +#include <mach/mxs.h> + +#include "mxs-saif.h" + +static struct mxs_saif *mxs_saif[2]; + +/* + * SAIF is a little different with other normal SOC DAIs on clock using. + * + * For MXS, two SAIF modules are instantiated on-chip. + * Each SAIF has a set of clock pins and can be operating in master + * mode simultaneously if they are connected to different off-chip codecs. + * Also, one of the two SAIFs can master or drive the clock pins while the + * other SAIF, in slave mode, receives clocking from the master SAIF. + * This also means that both SAIFs must operate at the same sample rate. + * + * We abstract this as each saif has a master, the master could be + * himself or other saifs. In the generic saif driver, saif does not need + * to know the different clkmux. Saif only needs to know who is his master + * and operating his master to generate the proper clock rate for him. + * The master id is provided in mach-specific layer according to different + * clkmux setting. + */ + +static int mxs_saif_set_dai_sysclk(struct snd_soc_dai *cpu_dai, + int clk_id, unsigned int freq, int dir) +{ + struct mxs_saif *saif = snd_soc_dai_get_drvdata(cpu_dai); + + switch (clk_id) { + case MXS_SAIF_MCLK: + saif->mclk = freq; + break; + default: + return -EINVAL; + } + return 0; +} + +/* + * Since SAIF may work on EXTMASTER mode, IOW, it's working BITCLK&LRCLK + * is provided by other SAIF, we provide a interface here to get its master + * from its master_id. + * Note that the master could be himself. + */ +static inline struct mxs_saif *mxs_saif_get_master(struct mxs_saif * saif) +{ + return mxs_saif[saif->master_id]; +} + +/* + * Set SAIF clock and MCLK + */ +static int mxs_saif_set_clk(struct mxs_saif *saif, + unsigned int mclk, + unsigned int rate) +{ + u32 scr; + int ret; + struct mxs_saif *master_saif; + + dev_dbg(saif->dev, "mclk %d rate %d\n", mclk, rate); + + /* Set master saif to generate proper clock */ + master_saif = mxs_saif_get_master(saif); + if (!master_saif) + return -EINVAL; + + dev_dbg(saif->dev, "master saif%d\n", master_saif->id); + + /* Checking if can playback and capture simutaneously */ + if (master_saif->ongoing && rate != master_saif->cur_rate) { + dev_err(saif->dev, + "can not change clock, master saif%d(rate %d) is ongoing\n", + master_saif->id, master_saif->cur_rate); + return -EINVAL; + } + + scr = __raw_readl(master_saif->base + SAIF_CTRL); + scr &= ~BM_SAIF_CTRL_BITCLK_MULT_RATE; + scr &= ~BM_SAIF_CTRL_BITCLK_BASE_RATE; + + /* + * Set SAIF clock + * + * The SAIF clock should be either 384*fs or 512*fs. + * If MCLK is used, the SAIF clk ratio need to match mclk ratio. + * For 32x mclk, set saif clk as 512*fs. + * For 48x mclk, set saif clk as 384*fs. + * + * If MCLK is not used, we just set saif clk to 512*fs. + */ + if (master_saif->mclk_in_use) { + if (mclk % 32 == 0) { + scr &= ~BM_SAIF_CTRL_BITCLK_BASE_RATE; + ret = clk_set_rate(master_saif->clk, 512 * rate); + } else if (mclk % 48 == 0) { + scr |= BM_SAIF_CTRL_BITCLK_BASE_RATE; + ret = clk_set_rate(master_saif->clk, 384 * rate); + } else { + /* SAIF MCLK should be either 32x or 48x */ + return -EINVAL; + } + } else { + ret = clk_set_rate(master_saif->clk, 512 * rate); + scr &= ~BM_SAIF_CTRL_BITCLK_BASE_RATE; + } + + if (ret) + return ret; + + master_saif->cur_rate = rate; + + if (!master_saif->mclk_in_use) { + __raw_writel(scr, master_saif->base + SAIF_CTRL); + return 0; + } + + /* + * Program the over-sample rate for MCLK output + * + * The available MCLK range is 32x, 48x... 512x. The rate + * could be from 8kHz to 192kH. + */ + switch (mclk / rate) { + case 32: + scr |= BF_SAIF_CTRL_BITCLK_MULT_RATE(4); + break; + case 64: + scr |= BF_SAIF_CTRL_BITCLK_MULT_RATE(3); + break; + case 128: + scr |= BF_SAIF_CTRL_BITCLK_MULT_RATE(2); + break; + case 256: + scr |= BF_SAIF_CTRL_BITCLK_MULT_RATE(1); + break; + case 512: + scr |= BF_SAIF_CTRL_BITCLK_MULT_RATE(0); + break; + case 48: + scr |= BF_SAIF_CTRL_BITCLK_MULT_RATE(3); + break; + case 96: + scr |= BF_SAIF_CTRL_BITCLK_MULT_RATE(2); + break; + case 192: + scr |= BF_SAIF_CTRL_BITCLK_MULT_RATE(1); + break; + case 384: + scr |= BF_SAIF_CTRL_BITCLK_MULT_RATE(0); + break; + default: + return -EINVAL; + } + + __raw_writel(scr, master_saif->base + SAIF_CTRL); + + return 0; +} + +/* + * Put and disable MCLK. + */ +int mxs_saif_put_mclk(unsigned int saif_id) +{ + struct mxs_saif *saif = mxs_saif[saif_id]; + u32 stat; + + if (!saif) + return -EINVAL; + + stat = __raw_readl(saif->base + SAIF_STAT); + if (stat & BM_SAIF_STAT_BUSY) { + dev_err(saif->dev, "error: busy\n"); + return -EBUSY; + } + + clk_disable(saif->clk); + + /* disable MCLK output */ + __raw_writel(BM_SAIF_CTRL_CLKGATE, + saif->base + SAIF_CTRL + MXS_SET_ADDR); + __raw_writel(BM_SAIF_CTRL_RUN, + saif->base + SAIF_CTRL + MXS_CLR_ADDR); + + saif->mclk_in_use = 0; + return 0; +} + +/* + * Get MCLK and set clock rate, then enable it + * + * This interface is used for codecs who are using MCLK provided + * by saif. + */ +int mxs_saif_get_mclk(unsigned int saif_id, unsigned int mclk, + unsigned int rate) +{ + struct mxs_saif *saif = mxs_saif[saif_id]; + u32 stat; + int ret; + struct mxs_saif *master_saif; + + if (!saif) + return -EINVAL; + + /* Clear Reset */ + __raw_writel(BM_SAIF_CTRL_SFTRST, + saif->base + SAIF_CTRL + MXS_CLR_ADDR); + + /* FIXME: need clear clk gate for register r/w */ + __raw_writel(BM_SAIF_CTRL_CLKGATE, + saif->base + SAIF_CTRL + MXS_CLR_ADDR); + + master_saif = mxs_saif_get_master(saif); + if (saif != master_saif) { + dev_err(saif->dev, "can not get mclk from a non-master saif\n"); + return -EINVAL; + } + + stat = __raw_readl(saif->base + SAIF_STAT); + if (stat & BM_SAIF_STAT_BUSY) { + dev_err(saif->dev, "error: busy\n"); + return -EBUSY; + } + + saif->mclk_in_use = 1; + ret = mxs_saif_set_clk(saif, mclk, rate); + if (ret) + return ret; + + ret = clk_enable(saif->clk); + if (ret) + return ret; + + /* enable MCLK output */ + __raw_writel(BM_SAIF_CTRL_RUN, + saif->base + SAIF_CTRL + MXS_SET_ADDR); + + return 0; +} + +/* + * SAIF DAI format configuration. + * Should only be called when port is inactive. + */ +static int mxs_saif_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) +{ + u32 scr, stat; + u32 scr0; + struct mxs_saif *saif = snd_soc_dai_get_drvdata(cpu_dai); + + stat = __raw_readl(saif->base + SAIF_STAT); + if (stat & BM_SAIF_STAT_BUSY) { + dev_err(cpu_dai->dev, "error: busy\n"); + return -EBUSY; + } + + scr0 = __raw_readl(saif->base + SAIF_CTRL); + scr0 = scr0 & ~BM_SAIF_CTRL_BITCLK_EDGE & ~BM_SAIF_CTRL_LRCLK_POLARITY \ + & ~BM_SAIF_CTRL_JUSTIFY & ~BM_SAIF_CTRL_DELAY; + scr = 0; + + /* DAI mode */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + /* data frame low 1clk before data */ + scr |= BM_SAIF_CTRL_DELAY; + scr &= ~BM_SAIF_CTRL_LRCLK_POLARITY; + break; + case SND_SOC_DAIFMT_LEFT_J: + /* data frame high with data */ + scr &= ~BM_SAIF_CTRL_DELAY; + scr &= ~BM_SAIF_CTRL_LRCLK_POLARITY; + scr &= ~BM_SAIF_CTRL_JUSTIFY; + break; + default: + return -EINVAL; + } + + /* DAI clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_IB_IF: + scr |= BM_SAIF_CTRL_BITCLK_EDGE; + scr |= BM_SAIF_CTRL_LRCLK_POLARITY; + break; + case SND_SOC_DAIFMT_IB_NF: + scr |= BM_SAIF_CTRL_BITCLK_EDGE; + scr &= ~BM_SAIF_CTRL_LRCLK_POLARITY; + break; + case SND_SOC_DAIFMT_NB_IF: + scr &= ~BM_SAIF_CTRL_BITCLK_EDGE; + scr |= BM_SAIF_CTRL_LRCLK_POLARITY; + break; + case SND_SOC_DAIFMT_NB_NF: + scr &= ~BM_SAIF_CTRL_BITCLK_EDGE; + scr &= ~BM_SAIF_CTRL_LRCLK_POLARITY; + break; + } + + /* + * Note: We simply just support master mode since SAIF TX can only + * work as master. + * Here the master is relative to codec side. + * Saif internally could be slave when working on EXTMASTER mode. + * We just hide this to machine driver. + */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + if (saif->id == saif->master_id) + scr &= ~BM_SAIF_CTRL_SLAVE_MODE; + else + scr |= BM_SAIF_CTRL_SLAVE_MODE; + + __raw_writel(scr | scr0, saif->base + SAIF_CTRL); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int mxs_saif_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct mxs_saif *saif = snd_soc_dai_get_drvdata(cpu_dai); + snd_soc_dai_set_dma_data(cpu_dai, substream, &saif->dma_param); + + /* clear error status to 0 for each re-open */ + saif->fifo_underrun = 0; + saif->fifo_overrun = 0; + + /* Clear Reset for normal operations */ + __raw_writel(BM_SAIF_CTRL_SFTRST, + saif->base + SAIF_CTRL + MXS_CLR_ADDR); + + /* clear clock gate */ + __raw_writel(BM_SAIF_CTRL_CLKGATE, + saif->base + SAIF_CTRL + MXS_CLR_ADDR); + + return 0; +} + +/* + * Should only be called when port is inactive. + * although can be called multiple times by upper layers. + */ +static int mxs_saif_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *cpu_dai) +{ + struct mxs_saif *saif = snd_soc_dai_get_drvdata(cpu_dai); + u32 scr, stat; + int ret; + + /* mclk should already be set */ + if (!saif->mclk && saif->mclk_in_use) { + dev_err(cpu_dai->dev, "set mclk first\n"); + return -EINVAL; + } + + stat = __raw_readl(saif->base + SAIF_STAT); + if (stat & BM_SAIF_STAT_BUSY) { + dev_err(cpu_dai->dev, "error: busy\n"); + return -EBUSY; + } + + /* + * Set saif clk based on sample rate. + * If mclk is used, we also set mclk, if not, saif->mclk is + * default 0, means not used. + */ + ret = mxs_saif_set_clk(saif, saif->mclk, params_rate(params)); + if (ret) { + dev_err(cpu_dai->dev, "unable to get proper clk\n"); + return ret; + } + + scr = __raw_readl(saif->base + SAIF_CTRL); + + scr &= ~BM_SAIF_CTRL_WORD_LENGTH; + scr &= ~BM_SAIF_CTRL_BITCLK_48XFS_ENABLE; + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + scr |= BF_SAIF_CTRL_WORD_LENGTH(0); + break; + case SNDRV_PCM_FORMAT_S20_3LE: + scr |= BF_SAIF_CTRL_WORD_LENGTH(4); + scr |= BM_SAIF_CTRL_BITCLK_48XFS_ENABLE; + break; + case SNDRV_PCM_FORMAT_S24_LE: + scr |= BF_SAIF_CTRL_WORD_LENGTH(8); + scr |= BM_SAIF_CTRL_BITCLK_48XFS_ENABLE; + break; + default: + return -EINVAL; + } + + /* Tx/Rx config */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + /* enable TX mode */ + scr &= ~BM_SAIF_CTRL_READ_MODE; + } else { + /* enable RX mode */ + scr |= BM_SAIF_CTRL_READ_MODE; + } + + __raw_writel(scr, saif->base + SAIF_CTRL); + return 0; +} + +static int mxs_saif_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct mxs_saif *saif = snd_soc_dai_get_drvdata(cpu_dai); + + /* enable FIFO error irqs */ + __raw_writel(BM_SAIF_CTRL_FIFO_ERROR_IRQ_EN, + saif->base + SAIF_CTRL + MXS_SET_ADDR); + + return 0; +} + +static int mxs_saif_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *cpu_dai) +{ + struct mxs_saif *saif = snd_soc_dai_get_drvdata(cpu_dai); + struct mxs_saif *master_saif; + u32 delay; + + master_saif = mxs_saif_get_master(saif); + if (!master_saif) + return -EINVAL; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + dev_dbg(cpu_dai->dev, "start\n"); + + clk_enable(master_saif->clk); + if (!master_saif->mclk_in_use) + __raw_writel(BM_SAIF_CTRL_RUN, + master_saif->base + SAIF_CTRL + MXS_SET_ADDR); + + /* + * If the saif's master is not himself, we also need to enable + * itself clk for its internal basic logic to work. + */ + if (saif != master_saif) { + clk_enable(saif->clk); + __raw_writel(BM_SAIF_CTRL_RUN, + saif->base + SAIF_CTRL + MXS_SET_ADDR); + } + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + /* + * write a data to saif data register to trigger + * the transfer + */ + __raw_writel(0, saif->base + SAIF_DATA); + } else { + /* + * read a data from saif data register to trigger + * the receive + */ + __raw_readl(saif->base + SAIF_DATA); + } + + master_saif->ongoing = 1; + + dev_dbg(saif->dev, "CTRL 0x%x STAT 0x%x\n", + __raw_readl(saif->base + SAIF_CTRL), + __raw_readl(saif->base + SAIF_STAT)); + + dev_dbg(master_saif->dev, "CTRL 0x%x STAT 0x%x\n", + __raw_readl(master_saif->base + SAIF_CTRL), + __raw_readl(master_saif->base + SAIF_STAT)); + break; + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + dev_dbg(cpu_dai->dev, "stop\n"); + + /* wait a while for the current sample to complete */ + delay = USEC_PER_SEC / master_saif->cur_rate; + + if (!master_saif->mclk_in_use) { + __raw_writel(BM_SAIF_CTRL_RUN, + master_saif->base + SAIF_CTRL + MXS_CLR_ADDR); + udelay(delay); + } + clk_disable(master_saif->clk); + + if (saif != master_saif) { + __raw_writel(BM_SAIF_CTRL_RUN, + saif->base + SAIF_CTRL + MXS_CLR_ADDR); + udelay(delay); + clk_disable(saif->clk); + } + + master_saif->ongoing = 0; + + break; + default: + return -EINVAL; + } + + return 0; +} + +#define MXS_SAIF_RATES SNDRV_PCM_RATE_8000_192000 +#define MXS_SAIF_FORMATS \ + (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_ops mxs_saif_dai_ops = { + .startup = mxs_saif_startup, + .trigger = mxs_saif_trigger, + .prepare = mxs_saif_prepare, + .hw_params = mxs_saif_hw_params, + .set_sysclk = mxs_saif_set_dai_sysclk, + .set_fmt = mxs_saif_set_dai_fmt, +}; + +static int mxs_saif_dai_probe(struct snd_soc_dai *dai) +{ + struct mxs_saif *saif = dev_get_drvdata(dai->dev); + + snd_soc_dai_set_drvdata(dai, saif); + + return 0; +} + +static struct snd_soc_dai_driver mxs_saif_dai = { + .name = "mxs-saif", + .probe = mxs_saif_dai_probe, + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = MXS_SAIF_RATES, + .formats = MXS_SAIF_FORMATS, + }, + .capture = { + .channels_min = 2, + .channels_max = 2, + .rates = MXS_SAIF_RATES, + .formats = MXS_SAIF_FORMATS, + }, + .ops = &mxs_saif_dai_ops, +}; + +static irqreturn_t mxs_saif_irq(int irq, void *dev_id) +{ + struct mxs_saif *saif = dev_id; + unsigned int stat; + + stat = __raw_readl(saif->base + SAIF_STAT); + if (!(stat & (BM_SAIF_STAT_FIFO_UNDERFLOW_IRQ | + BM_SAIF_STAT_FIFO_OVERFLOW_IRQ))) + return IRQ_NONE; + + if (stat & BM_SAIF_STAT_FIFO_UNDERFLOW_IRQ) { + dev_dbg(saif->dev, "underrun!!! %d\n", ++saif->fifo_underrun); + __raw_writel(BM_SAIF_STAT_FIFO_UNDERFLOW_IRQ, + saif->base + SAIF_STAT + MXS_CLR_ADDR); + } + + if (stat & BM_SAIF_STAT_FIFO_OVERFLOW_IRQ) { + dev_dbg(saif->dev, "overrun!!! %d\n", ++saif->fifo_overrun); + __raw_writel(BM_SAIF_STAT_FIFO_OVERFLOW_IRQ, + saif->base + SAIF_STAT + MXS_CLR_ADDR); + } + + dev_dbg(saif->dev, "SAIF_CTRL %x SAIF_STAT %x\n", + __raw_readl(saif->base + SAIF_CTRL), + __raw_readl(saif->base + SAIF_STAT)); + + return IRQ_HANDLED; +} + +static int mxs_saif_probe(struct platform_device *pdev) +{ + struct resource *iores, *dmares; + struct mxs_saif *saif; + struct mxs_saif_platform_data *pdata; + int ret = 0; + + if (pdev->id >= ARRAY_SIZE(mxs_saif)) + return -EINVAL; + + pdata = pdev->dev.platform_data; + if (pdata && pdata->init) { + ret = pdata->init(); + if (ret) + return ret; + } + + saif = kzalloc(sizeof(*saif), GFP_KERNEL); + if (!saif) + return -ENOMEM; + + mxs_saif[pdev->id] = saif; + saif->id = pdev->id; + + saif->master_id = saif->id; + if (pdata && pdata->get_master_id) { + saif->master_id = pdata->get_master_id(saif->id); + if (saif->master_id < 0 || + saif->master_id >= ARRAY_SIZE(mxs_saif)) + return -EINVAL; + } + + saif->clk = clk_get(&pdev->dev, NULL); + if (IS_ERR(saif->clk)) { + ret = PTR_ERR(saif->clk); + dev_err(&pdev->dev, "Cannot get the clock: %d\n", + ret); + goto failed_clk; + } + + iores = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!iores) { + ret = -ENODEV; + dev_err(&pdev->dev, "failed to get io resource: %d\n", + ret); + goto failed_get_resource; + } + + if (!request_mem_region(iores->start, resource_size(iores), + "mxs-saif")) { + dev_err(&pdev->dev, "request_mem_region failed\n"); + ret = -EBUSY; + goto failed_get_resource; + } + + saif->base = ioremap(iores->start, resource_size(iores)); + if (!saif->base) { + dev_err(&pdev->dev, "ioremap failed\n"); + ret = -ENODEV; + goto failed_ioremap; + } + + dmares = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!dmares) { + ret = -ENODEV; + dev_err(&pdev->dev, "failed to get dma resource: %d\n", + ret); + goto failed_ioremap; + } + saif->dma_param.chan_num = dmares->start; + + saif->irq = platform_get_irq(pdev, 0); + if (saif->irq < 0) { + ret = saif->irq; + dev_err(&pdev->dev, "failed to get irq resource: %d\n", + ret); + goto failed_get_irq1; + } + + saif->dev = &pdev->dev; + ret = request_irq(saif->irq, mxs_saif_irq, 0, "mxs-saif", saif); + if (ret) { + dev_err(&pdev->dev, "failed to request irq\n"); + goto failed_get_irq1; + } + + saif->dma_param.chan_irq = platform_get_irq(pdev, 1); + if (saif->dma_param.chan_irq < 0) { + ret = saif->dma_param.chan_irq; + dev_err(&pdev->dev, "failed to get dma irq resource: %d\n", + ret); + goto failed_get_irq2; + } + + platform_set_drvdata(pdev, saif); + + ret = snd_soc_register_dai(&pdev->dev, &mxs_saif_dai); + if (ret) { + dev_err(&pdev->dev, "register DAI failed\n"); + goto failed_register; + } + + saif->soc_platform_pdev = platform_device_alloc( + "mxs-pcm-audio", pdev->id); + if (!saif->soc_platform_pdev) { + ret = -ENOMEM; + goto failed_pdev_alloc; + } + + platform_set_drvdata(saif->soc_platform_pdev, saif); + ret = platform_device_add(saif->soc_platform_pdev); + if (ret) { + dev_err(&pdev->dev, "failed to add soc platform device\n"); + goto failed_pdev_add; + } + + return 0; + +failed_pdev_add: + platform_device_put(saif->soc_platform_pdev); +failed_pdev_alloc: + snd_soc_unregister_dai(&pdev->dev); +failed_register: +failed_get_irq2: + free_irq(saif->irq, saif); +failed_get_irq1: + iounmap(saif->base); +failed_ioremap: + release_mem_region(iores->start, resource_size(iores)); +failed_get_resource: + clk_put(saif->clk); +failed_clk: + kfree(saif); + + return ret; +} + +static int __devexit mxs_saif_remove(struct platform_device *pdev) +{ + struct resource *res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + struct mxs_saif *saif = platform_get_drvdata(pdev); + + platform_device_unregister(saif->soc_platform_pdev); + + snd_soc_unregister_dai(&pdev->dev); + + iounmap(saif->base); + release_mem_region(res->start, resource_size(res)); + free_irq(saif->irq, saif); + + clk_put(saif->clk); + kfree(saif); + + return 0; +} + +static struct platform_driver mxs_saif_driver = { + .probe = mxs_saif_probe, + .remove = __devexit_p(mxs_saif_remove), + + .driver = { + .name = "mxs-saif", + .owner = THIS_MODULE, + }, +}; + +static int __init mxs_saif_init(void) +{ + return platform_driver_register(&mxs_saif_driver); +} + +static void __exit mxs_saif_exit(void) +{ + platform_driver_unregister(&mxs_saif_driver); +} + +module_init(mxs_saif_init); +module_exit(mxs_saif_exit); +MODULE_AUTHOR("Freescale Semiconductor, Inc."); +MODULE_DESCRIPTION("MXS ASoC SAIF driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/mxs/mxs-saif.h b/sound/soc/mxs/mxs-saif.h new file mode 100644 index 000000000000..12c91e4eb941 --- /dev/null +++ b/sound/soc/mxs/mxs-saif.h @@ -0,0 +1,134 @@ +/* + * Copyright (C) 2011 Freescale Semiconductor, Inc. All Rights Reserved. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + + +#ifndef _MXS_SAIF_H +#define _MXS_SAIF_H + +#define SAIF_CTRL 0x0 +#define SAIF_STAT 0x10 +#define SAIF_DATA 0x20 +#define SAIF_VERSION 0X30 + +/* SAIF_CTRL */ +#define BM_SAIF_CTRL_SFTRST 0x80000000 +#define BM_SAIF_CTRL_CLKGATE 0x40000000 +#define BP_SAIF_CTRL_BITCLK_MULT_RATE 27 +#define BM_SAIF_CTRL_BITCLK_MULT_RATE 0x38000000 +#define BF_SAIF_CTRL_BITCLK_MULT_RATE(v) \ + (((v) << 27) & BM_SAIF_CTRL_BITCLK_MULT_RATE) +#define BM_SAIF_CTRL_BITCLK_BASE_RATE 0x04000000 +#define BM_SAIF_CTRL_FIFO_ERROR_IRQ_EN 0x02000000 +#define BM_SAIF_CTRL_FIFO_SERVICE_IRQ_EN 0x01000000 +#define BP_SAIF_CTRL_RSRVD2 21 +#define BM_SAIF_CTRL_RSRVD2 0x00E00000 + +#define BP_SAIF_CTRL_DMAWAIT_COUNT 16 +#define BM_SAIF_CTRL_DMAWAIT_COUNT 0x001F0000 +#define BF_SAIF_CTRL_DMAWAIT_COUNT(v) \ + (((v) << 16) & BM_SAIF_CTRL_DMAWAIT_COUNT) +#define BP_SAIF_CTRL_CHANNEL_NUM_SELECT 14 +#define BM_SAIF_CTRL_CHANNEL_NUM_SELECT 0x0000C000 +#define BF_SAIF_CTRL_CHANNEL_NUM_SELECT(v) \ + (((v) << 14) & BM_SAIF_CTRL_CHANNEL_NUM_SELECT) +#define BM_SAIF_CTRL_LRCLK_PULSE 0x00002000 +#define BM_SAIF_CTRL_BIT_ORDER 0x00001000 +#define BM_SAIF_CTRL_DELAY 0x00000800 +#define BM_SAIF_CTRL_JUSTIFY 0x00000400 +#define BM_SAIF_CTRL_LRCLK_POLARITY 0x00000200 +#define BM_SAIF_CTRL_BITCLK_EDGE 0x00000100 +#define BP_SAIF_CTRL_WORD_LENGTH 4 +#define BM_SAIF_CTRL_WORD_LENGTH 0x000000F0 +#define BF_SAIF_CTRL_WORD_LENGTH(v) \ + (((v) << 4) & BM_SAIF_CTRL_WORD_LENGTH) +#define BM_SAIF_CTRL_BITCLK_48XFS_ENABLE 0x00000008 +#define BM_SAIF_CTRL_SLAVE_MODE 0x00000004 +#define BM_SAIF_CTRL_READ_MODE 0x00000002 +#define BM_SAIF_CTRL_RUN 0x00000001 + +/* SAIF_STAT */ +#define BM_SAIF_STAT_PRESENT 0x80000000 +#define BP_SAIF_STAT_RSRVD2 17 +#define BM_SAIF_STAT_RSRVD2 0x7FFE0000 +#define BF_SAIF_STAT_RSRVD2(v) \ + (((v) << 17) & BM_SAIF_STAT_RSRVD2) +#define BM_SAIF_STAT_DMA_PREQ 0x00010000 +#define BP_SAIF_STAT_RSRVD1 7 +#define BM_SAIF_STAT_RSRVD1 0x0000FF80 +#define BF_SAIF_STAT_RSRVD1(v) \ + (((v) << 7) & BM_SAIF_STAT_RSRVD1) + +#define BM_SAIF_STAT_FIFO_UNDERFLOW_IRQ 0x00000040 +#define BM_SAIF_STAT_FIFO_OVERFLOW_IRQ 0x00000020 +#define BM_SAIF_STAT_FIFO_SERVICE_IRQ 0x00000010 +#define BP_SAIF_STAT_RSRVD0 1 +#define BM_SAIF_STAT_RSRVD0 0x0000000E +#define BF_SAIF_STAT_RSRVD0(v) \ + (((v) << 1) & BM_SAIF_STAT_RSRVD0) +#define BM_SAIF_STAT_BUSY 0x00000001 + +/* SAFI_DATA */ +#define BP_SAIF_DATA_PCM_RIGHT 16 +#define BM_SAIF_DATA_PCM_RIGHT 0xFFFF0000 +#define BF_SAIF_DATA_PCM_RIGHT(v) \ + (((v) << 16) & BM_SAIF_DATA_PCM_RIGHT) +#define BP_SAIF_DATA_PCM_LEFT 0 +#define BM_SAIF_DATA_PCM_LEFT 0x0000FFFF +#define BF_SAIF_DATA_PCM_LEFT(v) \ + (((v) << 0) & BM_SAIF_DATA_PCM_LEFT) + +/* SAIF_VERSION */ +#define BP_SAIF_VERSION_MAJOR 24 +#define BM_SAIF_VERSION_MAJOR 0xFF000000 +#define BF_SAIF_VERSION_MAJOR(v) \ + (((v) << 24) & BM_SAIF_VERSION_MAJOR) +#define BP_SAIF_VERSION_MINOR 16 +#define BM_SAIF_VERSION_MINOR 0x00FF0000 +#define BF_SAIF_VERSION_MINOR(v) \ + (((v) << 16) & BM_SAIF_VERSION_MINOR) +#define BP_SAIF_VERSION_STEP 0 +#define BM_SAIF_VERSION_STEP 0x0000FFFF +#define BF_SAIF_VERSION_STEP(v) \ + (((v) << 0) & BM_SAIF_VERSION_STEP) + +#define MXS_SAIF_MCLK 0 + +#include "mxs-pcm.h" + +struct mxs_saif { + struct device *dev; + struct clk *clk; + unsigned int mclk; + unsigned int mclk_in_use; + void __iomem *base; + int irq; + struct mxs_pcm_dma_params dma_param; + unsigned int id; + unsigned int master_id; + unsigned int cur_rate; + unsigned int ongoing; + + struct platform_device *soc_platform_pdev; + u32 fifo_underrun; + u32 fifo_overrun; +}; + +extern int mxs_saif_put_mclk(unsigned int saif_id); +extern int mxs_saif_get_mclk(unsigned int saif_id, unsigned int mclk, + unsigned int rate); +#endif diff --git a/sound/soc/mxs/mxs-sgtl5000.c b/sound/soc/mxs/mxs-sgtl5000.c new file mode 100644 index 000000000000..7fbeaec06eb4 --- /dev/null +++ b/sound/soc/mxs/mxs-sgtl5000.c @@ -0,0 +1,173 @@ +/* + * Copyright 2011 Freescale Semiconductor, Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#include <linux/module.h> +#include <linux/device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/jack.h> +#include <sound/soc-dapm.h> +#include <asm/mach-types.h> + +#include "../codecs/sgtl5000.h" +#include "mxs-saif.h" + +static int mxs_sgtl5000_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + unsigned int rate = params_rate(params); + u32 dai_format, mclk; + int ret; + + /* sgtl5000 does not support 512*rate when in 96000 fs */ + switch (rate) { + case 96000: + mclk = 256 * rate; + break; + default: + mclk = 512 * rate; + break; + } + + /* Sgtl5000 sysclk should be >= 8MHz and <= 27M */ + if (mclk < 8000000 || mclk > 27000000) + return -EINVAL; + + /* Set SGTL5000's SYSCLK (provided by SAIF MCLK) */ + ret = snd_soc_dai_set_sysclk(codec_dai, SGTL5000_SYSCLK, mclk, 0); + if (ret) + return ret; + + /* The SAIF MCLK should be the same as SGTL5000_SYSCLK */ + ret = snd_soc_dai_set_sysclk(cpu_dai, MXS_SAIF_MCLK, mclk, 0); + if (ret) + return ret; + + /* set codec to slave mode */ + dai_format = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS; + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, dai_format); + if (ret) + return ret; + + /* set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, dai_format); + if (ret) + return ret; + + return 0; +} + +static struct snd_soc_ops mxs_sgtl5000_hifi_ops = { + .hw_params = mxs_sgtl5000_hw_params, +}; + +static struct snd_soc_dai_link mxs_sgtl5000_dai[] = { + { + .name = "HiFi Tx", + .stream_name = "HiFi Playback", + .codec_dai_name = "sgtl5000", + .codec_name = "sgtl5000.0-000a", + .cpu_dai_name = "mxs-saif.0", + .platform_name = "mxs-pcm-audio.0", + .ops = &mxs_sgtl5000_hifi_ops, + }, { + .name = "HiFi Rx", + .stream_name = "HiFi Capture", + .codec_dai_name = "sgtl5000", + .codec_name = "sgtl5000.0-000a", + .cpu_dai_name = "mxs-saif.1", + .platform_name = "mxs-pcm-audio.1", + .ops = &mxs_sgtl5000_hifi_ops, + }, +}; + +static struct snd_soc_card mxs_sgtl5000 = { + .name = "mxs_sgtl5000", + .dai_link = mxs_sgtl5000_dai, + .num_links = ARRAY_SIZE(mxs_sgtl5000_dai), +}; + +static int __devinit mxs_sgtl5000_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &mxs_sgtl5000; + int ret; + + /* + * Set an init clock(11.28Mhz) for sgtl5000 initialization(i2c r/w). + * The Sgtl5000 sysclk is derived from saif0 mclk and it's range + * should be >= 8MHz and <= 27M. + */ + ret = mxs_saif_get_mclk(0, 44100 * 256, 44100); + if (ret) + return ret; + + card->dev = &pdev->dev; + platform_set_drvdata(pdev, card); + + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", + ret); + return ret; + } + + return 0; +} + +static int __devexit mxs_sgtl5000_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + + mxs_saif_put_mclk(0); + + snd_soc_unregister_card(card); + + return 0; +} + +static struct platform_driver mxs_sgtl5000_audio_driver = { + .driver = { + .name = "mxs-sgtl5000", + .owner = THIS_MODULE, + }, + .probe = mxs_sgtl5000_probe, + .remove = __devexit_p(mxs_sgtl5000_remove), +}; + +static int __init mxs_sgtl5000_init(void) +{ + return platform_driver_register(&mxs_sgtl5000_audio_driver); +} +module_init(mxs_sgtl5000_init); + +static void __exit mxs_sgtl5000_exit(void) +{ + platform_driver_unregister(&mxs_sgtl5000_audio_driver); +} +module_exit(mxs_sgtl5000_exit); + +MODULE_AUTHOR("Freescale Semiconductor, Inc."); +MODULE_DESCRIPTION("MXS ALSA SoC Machine driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/nuc900/nuc900-pcm.c b/sound/soc/nuc900/nuc900-pcm.c index d589ef14e917..ae8d6806966b 100644 --- a/sound/soc/nuc900/nuc900-pcm.c +++ b/sound/soc/nuc900/nuc900-pcm.c @@ -227,7 +227,7 @@ static int nuc900_dma_trigger(struct snd_pcm_substream *substream, int cmd) return ret; } -int nuc900_dma_getposition(struct snd_pcm_substream *substream, +static int nuc900_dma_getposition(struct snd_pcm_substream *substream, dma_addr_t *src, dma_addr_t *dst) { struct snd_pcm_runtime *runtime = substream->runtime; @@ -268,7 +268,7 @@ static int nuc900_dma_open(struct snd_pcm_substream *substream) nuc900_audio = nuc900_ac97_data; if (request_irq(nuc900_audio->irq_num, nuc900_dma_interrupt, - IRQF_DISABLED, "nuc900-dma", substream)) + 0, "nuc900-dma", substream)) return -EBUSY; runtime->private_data = nuc900_audio; @@ -318,7 +318,6 @@ static u64 nuc900_pcm_dmamask = DMA_BIT_MASK(32); static int nuc900_dma_new(struct snd_soc_pcm_runtime *rtd) { struct snd_card *card = rtd->card->snd_card; - struct snd_soc_dai *dai = rtd->cpu_dai; struct snd_pcm *pcm = rtd->pcm; if (!card->dev->dma_mask) diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index 59e2c8d1e38d..052fd758722e 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -1,7 +1,7 @@ # OMAP Platform Support snd-soc-omap-objs := omap-pcm.o snd-soc-omap-mcbsp-objs := omap-mcbsp.o -snd-soc-omap-mcpdm-objs := omap-mcpdm.o mcpdm.o +snd-soc-omap-mcpdm-objs := omap-mcpdm.o snd-soc-omap-hdmi-objs := omap-hdmi.o obj-$(CONFIG_SND_OMAP_SOC) += snd-soc-omap.o diff --git a/sound/soc/omap/am3517evm.c b/sound/soc/omap/am3517evm.c index 73dde4a1adc3..8da55e916451 100644 --- a/sound/soc/omap/am3517evm.c +++ b/sound/soc/omap/am3517evm.c @@ -43,26 +43,6 @@ static int am3517evm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai = rtd->cpu_dai; int ret; - /* Set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, - SND_SOC_DAIFMT_DSP_B | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) { - printk(KERN_ERR "can't set codec DAI configuration\n"); - return ret; - } - - /* Set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, - SND_SOC_DAIFMT_DSP_B | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) { - printk(KERN_ERR "can't set cpu DAI configuration\n"); - return ret; - } - /* Set the codec system clock for DAC and ADC */ ret = snd_soc_dai_set_sysclk(codec_dai, 0, CODEC_CLOCK, SND_SOC_CLOCK_IN); @@ -110,28 +90,6 @@ static const struct snd_soc_dapm_route audio_map[] = { {"MICIN", NULL, "Mic In"}, }; -static int am3517evm_aic23_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_codec *codec = rtd->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; - - /* Add am3517-evm specific widgets */ - snd_soc_dapm_new_controls(dapm, tlv320aic23_dapm_widgets, - ARRAY_SIZE(tlv320aic23_dapm_widgets)); - - /* Set up davinci-evm specific audio path audio_map */ - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - - /* always connected */ - snd_soc_dapm_enable_pin(dapm, "Line Out"); - snd_soc_dapm_enable_pin(dapm, "Line In"); - snd_soc_dapm_enable_pin(dapm, "Mic In"); - - snd_soc_dapm_sync(dapm); - - return 0; -} - /* Digital audio interface glue - connects codec <--> CPU */ static struct snd_soc_dai_link am3517evm_dai = { .name = "TLV320AIC23", @@ -140,7 +98,8 @@ static struct snd_soc_dai_link am3517evm_dai = { .codec_dai_name = "tlv320aic23-hifi", .platform_name = "omap-pcm-audio", .codec_name = "tlv320aic23-codec.2-001a", - .init = am3517evm_aic23_init, + .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, .ops = &am3517evm_ops, }; @@ -149,6 +108,11 @@ static struct snd_soc_card snd_soc_am3517evm = { .name = "am3517evm", .dai_link = &am3517evm_dai, .num_links = 1, + + .dapm_widgets = tlv320aic23_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tlv320aic23_dapm_widgets), + .dapm_routes = audio_map, + .num_dapm_routes = ARRAY_SIZE(audio_map), }; static struct platform_device *am3517evm_snd_device; diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c index 0aa475f92efa..dcb7b689a4ea 100644 --- a/sound/soc/omap/ams-delta.c +++ b/sound/soc/omap/ams-delta.c @@ -569,7 +569,6 @@ static int ams_delta_cx20442_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_disable_pin(dapm, "Speaker"); snd_soc_dapm_disable_pin(dapm, "AGCIN"); snd_soc_dapm_disable_pin(dapm, "AGCOUT"); - snd_soc_dapm_sync(dapm); /* Add virtual switch */ ret = snd_soc_add_controls(codec, ams_delta_audio_controls, diff --git a/sound/soc/omap/igep0020.c b/sound/soc/omap/igep0020.c index 0ae34702995b..84615a7de6ad 100644 --- a/sound/soc/omap/igep0020.c +++ b/sound/soc/omap/igep0020.c @@ -38,29 +38,8 @@ static int igep2_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; int ret; - /* Set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, - SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) { - printk(KERN_ERR "can't set codec DAI configuration\n"); - return ret; - } - - /* Set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, - SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) { - printk(KERN_ERR "can't set cpu DAI configuration\n"); - return ret; - } - /* Set the codec system clock for DAC and ADC */ ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000, SND_SOC_CLOCK_IN); @@ -84,6 +63,8 @@ static struct snd_soc_dai_link igep2_dai = { .codec_dai_name = "twl4030-hifi", .platform_name = "omap-pcm-audio", .codec_name = "twl4030-codec", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, .ops = &igep2_ops, }; diff --git a/sound/soc/omap/mcpdm.c b/sound/soc/omap/mcpdm.c deleted file mode 100644 index 50e59194ad81..000000000000 --- a/sound/soc/omap/mcpdm.c +++ /dev/null @@ -1,470 +0,0 @@ -/* - * mcpdm.c -- McPDM interface driver - * - * Author: Jorge Eduardo Candelaria <x0107209@ti.com> - * Copyright (C) 2009 - Texas Instruments, Inc. - * - * This program is free software; you can redistribute it and/or - * modify it under the terms of the GNU General Public License - * version 2 as published by the Free Software Foundation. - * - * This program is distributed in the hope that it will be useful, but - * WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA - * 02110-1301 USA - * - */ - -#include <linux/module.h> -#include <linux/init.h> -#include <linux/device.h> -#include <linux/platform_device.h> -#include <linux/wait.h> -#include <linux/slab.h> -#include <linux/interrupt.h> -#include <linux/err.h> -#include <linux/clk.h> -#include <linux/delay.h> -#include <linux/io.h> -#include <linux/irq.h> - -#include "mcpdm.h" - -static struct omap_mcpdm *mcpdm; - -static inline void omap_mcpdm_write(u16 reg, u32 val) -{ - __raw_writel(val, mcpdm->io_base + reg); -} - -static inline int omap_mcpdm_read(u16 reg) -{ - return __raw_readl(mcpdm->io_base + reg); -} - -static void omap_mcpdm_reg_dump(void) -{ - dev_dbg(mcpdm->dev, "***********************\n"); - dev_dbg(mcpdm->dev, "IRQSTATUS_RAW: 0x%04x\n", - omap_mcpdm_read(MCPDM_IRQSTATUS_RAW)); - dev_dbg(mcpdm->dev, "IRQSTATUS: 0x%04x\n", - omap_mcpdm_read(MCPDM_IRQSTATUS)); - dev_dbg(mcpdm->dev, "IRQENABLE_SET: 0x%04x\n", - omap_mcpdm_read(MCPDM_IRQENABLE_SET)); - dev_dbg(mcpdm->dev, "IRQENABLE_CLR: 0x%04x\n", - omap_mcpdm_read(MCPDM_IRQENABLE_CLR)); - dev_dbg(mcpdm->dev, "IRQWAKE_EN: 0x%04x\n", - omap_mcpdm_read(MCPDM_IRQWAKE_EN)); - dev_dbg(mcpdm->dev, "DMAENABLE_SET: 0x%04x\n", - omap_mcpdm_read(MCPDM_DMAENABLE_SET)); - dev_dbg(mcpdm->dev, "DMAENABLE_CLR: 0x%04x\n", - omap_mcpdm_read(MCPDM_DMAENABLE_CLR)); - dev_dbg(mcpdm->dev, "DMAWAKEEN: 0x%04x\n", - omap_mcpdm_read(MCPDM_DMAWAKEEN)); - dev_dbg(mcpdm->dev, "CTRL: 0x%04x\n", - omap_mcpdm_read(MCPDM_CTRL)); - dev_dbg(mcpdm->dev, "DN_DATA: 0x%04x\n", - omap_mcpdm_read(MCPDM_DN_DATA)); - dev_dbg(mcpdm->dev, "UP_DATA: 0x%04x\n", - omap_mcpdm_read(MCPDM_UP_DATA)); - dev_dbg(mcpdm->dev, "FIFO_CTRL_DN: 0x%04x\n", - omap_mcpdm_read(MCPDM_FIFO_CTRL_DN)); - dev_dbg(mcpdm->dev, "FIFO_CTRL_UP: 0x%04x\n", - omap_mcpdm_read(MCPDM_FIFO_CTRL_UP)); - dev_dbg(mcpdm->dev, "DN_OFFSET: 0x%04x\n", - omap_mcpdm_read(MCPDM_DN_OFFSET)); - dev_dbg(mcpdm->dev, "***********************\n"); -} - -/* - * Takes the McPDM module in and out of reset state. - * Uplink and downlink can be reset individually. - */ -static void omap_mcpdm_reset_capture(int reset) -{ - int ctrl = omap_mcpdm_read(MCPDM_CTRL); - - if (reset) - ctrl |= SW_UP_RST; - else - ctrl &= ~SW_UP_RST; - - omap_mcpdm_write(MCPDM_CTRL, ctrl); -} - -static void omap_mcpdm_reset_playback(int reset) -{ - int ctrl = omap_mcpdm_read(MCPDM_CTRL); - - if (reset) - ctrl |= SW_DN_RST; - else - ctrl &= ~SW_DN_RST; - - omap_mcpdm_write(MCPDM_CTRL, ctrl); -} - -/* - * Enables the transfer through the PDM interface to/from the Phoenix - * codec by enabling the corresponding UP or DN channels. - */ -void omap_mcpdm_start(int stream) -{ - int ctrl = omap_mcpdm_read(MCPDM_CTRL); - - if (stream) - ctrl |= mcpdm->up_channels; - else - ctrl |= mcpdm->dn_channels; - - omap_mcpdm_write(MCPDM_CTRL, ctrl); -} - -/* - * Disables the transfer through the PDM interface to/from the Phoenix - * codec by disabling the corresponding UP or DN channels. - */ -void omap_mcpdm_stop(int stream) -{ - int ctrl = omap_mcpdm_read(MCPDM_CTRL); - - if (stream) - ctrl &= ~mcpdm->up_channels; - else - ctrl &= ~mcpdm->dn_channels; - - omap_mcpdm_write(MCPDM_CTRL, ctrl); -} - -/* - * Configures McPDM uplink for audio recording. - * This function should be called before omap_mcpdm_start. - */ -int omap_mcpdm_capture_open(struct omap_mcpdm_link *uplink) -{ - int irq_mask = 0; - int ctrl; - - if (!uplink) - return -EINVAL; - - mcpdm->uplink = uplink; - - /* Enable irq request generation */ - irq_mask |= uplink->irq_mask & MCPDM_UPLINK_IRQ_MASK; - omap_mcpdm_write(MCPDM_IRQENABLE_SET, irq_mask); - - /* Configure uplink threshold */ - if (uplink->threshold > UP_THRES_MAX) - uplink->threshold = UP_THRES_MAX; - - omap_mcpdm_write(MCPDM_FIFO_CTRL_UP, uplink->threshold); - - /* Configure DMA controller */ - omap_mcpdm_write(MCPDM_DMAENABLE_SET, DMA_UP_ENABLE); - - /* Set pdm out format */ - ctrl = omap_mcpdm_read(MCPDM_CTRL); - ctrl &= ~PDMOUTFORMAT; - ctrl |= uplink->format & PDMOUTFORMAT; - - /* Uplink channels */ - mcpdm->up_channels = uplink->channels & (PDM_UP_MASK | PDM_STATUS_MASK); - - omap_mcpdm_write(MCPDM_CTRL, ctrl); - - return 0; -} - -/* - * Configures McPDM downlink for audio playback. - * This function should be called before omap_mcpdm_start. - */ -int omap_mcpdm_playback_open(struct omap_mcpdm_link *downlink) -{ - int irq_mask = 0; - int ctrl; - - if (!downlink) - return -EINVAL; - - mcpdm->downlink = downlink; - - /* Enable irq request generation */ - irq_mask |= downlink->irq_mask & MCPDM_DOWNLINK_IRQ_MASK; - omap_mcpdm_write(MCPDM_IRQENABLE_SET, irq_mask); - - /* Configure uplink threshold */ - if (downlink->threshold > DN_THRES_MAX) - downlink->threshold = DN_THRES_MAX; - - omap_mcpdm_write(MCPDM_FIFO_CTRL_DN, downlink->threshold); - - /* Enable DMA request generation */ - omap_mcpdm_write(MCPDM_DMAENABLE_SET, DMA_DN_ENABLE); - - /* Set pdm out format */ - ctrl = omap_mcpdm_read(MCPDM_CTRL); - ctrl &= ~PDMOUTFORMAT; - ctrl |= downlink->format & PDMOUTFORMAT; - - /* Downlink channels */ - mcpdm->dn_channels = downlink->channels & (PDM_DN_MASK | PDM_CMD_MASK); - - omap_mcpdm_write(MCPDM_CTRL, ctrl); - - return 0; -} - -/* - * Cleans McPDM uplink configuration. - * This function should be called when the stream is closed. - */ -int omap_mcpdm_capture_close(struct omap_mcpdm_link *uplink) -{ - int irq_mask = 0; - - if (!uplink) - return -EINVAL; - - /* Disable irq request generation */ - irq_mask |= uplink->irq_mask & MCPDM_UPLINK_IRQ_MASK; - omap_mcpdm_write(MCPDM_IRQENABLE_CLR, irq_mask); - - /* Disable DMA request generation */ - omap_mcpdm_write(MCPDM_DMAENABLE_CLR, DMA_UP_ENABLE); - - /* Clear Downlink channels */ - mcpdm->up_channels = 0; - - mcpdm->uplink = NULL; - - return 0; -} - -/* - * Cleans McPDM downlink configuration. - * This function should be called when the stream is closed. - */ -int omap_mcpdm_playback_close(struct omap_mcpdm_link *downlink) -{ - int irq_mask = 0; - - if (!downlink) - return -EINVAL; - - /* Disable irq request generation */ - irq_mask |= downlink->irq_mask & MCPDM_DOWNLINK_IRQ_MASK; - omap_mcpdm_write(MCPDM_IRQENABLE_CLR, irq_mask); - - /* Disable DMA request generation */ - omap_mcpdm_write(MCPDM_DMAENABLE_CLR, DMA_DN_ENABLE); - - /* clear Downlink channels */ - mcpdm->dn_channels = 0; - - mcpdm->downlink = NULL; - - return 0; -} - -static irqreturn_t omap_mcpdm_irq_handler(int irq, void *dev_id) -{ - struct omap_mcpdm *mcpdm_irq = dev_id; - int irq_status; - - irq_status = omap_mcpdm_read(MCPDM_IRQSTATUS); - - /* Acknowledge irq event */ - omap_mcpdm_write(MCPDM_IRQSTATUS, irq_status); - - if (irq & MCPDM_DN_IRQ_FULL) { - dev_err(mcpdm_irq->dev, "DN FIFO error %x\n", irq_status); - omap_mcpdm_reset_playback(1); - omap_mcpdm_playback_open(mcpdm_irq->downlink); - omap_mcpdm_reset_playback(0); - } - - if (irq & MCPDM_DN_IRQ_EMPTY) { - dev_err(mcpdm_irq->dev, "DN FIFO error %x\n", irq_status); - omap_mcpdm_reset_playback(1); - omap_mcpdm_playback_open(mcpdm_irq->downlink); - omap_mcpdm_reset_playback(0); - } - - if (irq & MCPDM_DN_IRQ) { - dev_dbg(mcpdm_irq->dev, "DN write request\n"); - } - - if (irq & MCPDM_UP_IRQ_FULL) { - dev_err(mcpdm_irq->dev, "UP FIFO error %x\n", irq_status); - omap_mcpdm_reset_capture(1); - omap_mcpdm_capture_open(mcpdm_irq->uplink); - omap_mcpdm_reset_capture(0); - } - - if (irq & MCPDM_UP_IRQ_EMPTY) { - dev_err(mcpdm_irq->dev, "UP FIFO error %x\n", irq_status); - omap_mcpdm_reset_capture(1); - omap_mcpdm_capture_open(mcpdm_irq->uplink); - omap_mcpdm_reset_capture(0); - } - - if (irq & MCPDM_UP_IRQ) { - dev_dbg(mcpdm_irq->dev, "UP write request\n"); - } - - return IRQ_HANDLED; -} - -int omap_mcpdm_request(void) -{ - int ret; - - clk_enable(mcpdm->clk); - - spin_lock(&mcpdm->lock); - - if (!mcpdm->free) { - dev_err(mcpdm->dev, "McPDM interface is in use\n"); - spin_unlock(&mcpdm->lock); - ret = -EBUSY; - goto err; - } - mcpdm->free = 0; - - spin_unlock(&mcpdm->lock); - - /* Disable lines while request is ongoing */ - omap_mcpdm_write(MCPDM_CTRL, 0x00); - - ret = request_irq(mcpdm->irq, omap_mcpdm_irq_handler, - 0, "McPDM", (void *)mcpdm); - if (ret) { - dev_err(mcpdm->dev, "Request for McPDM IRQ failed\n"); - goto err; - } - - return 0; - -err: - clk_disable(mcpdm->clk); - return ret; -} - -void omap_mcpdm_free(void) -{ - spin_lock(&mcpdm->lock); - if (mcpdm->free) { - dev_err(mcpdm->dev, "McPDM interface is already free\n"); - spin_unlock(&mcpdm->lock); - return; - } - mcpdm->free = 1; - spin_unlock(&mcpdm->lock); - - clk_disable(mcpdm->clk); - - free_irq(mcpdm->irq, (void *)mcpdm); -} - -/* Enable/disable DC offset cancelation for the analog - * headset path (PDM channels 1 and 2). - */ -int omap_mcpdm_set_offset(int offset1, int offset2) -{ - int offset; - - if ((offset1 > DN_OFST_MAX) || (offset2 > DN_OFST_MAX)) - return -EINVAL; - - offset = (offset1 << DN_OFST_RX1) | (offset2 << DN_OFST_RX2); - - /* offset cancellation for channel 1 */ - if (offset1) - offset |= DN_OFST_RX1_EN; - else - offset &= ~DN_OFST_RX1_EN; - - /* offset cancellation for channel 2 */ - if (offset2) - offset |= DN_OFST_RX2_EN; - else - offset &= ~DN_OFST_RX2_EN; - - omap_mcpdm_write(MCPDM_DN_OFFSET, offset); - - return 0; -} - -int __devinit omap_mcpdm_probe(struct platform_device *pdev) -{ - struct resource *res; - int ret = 0; - - mcpdm = kzalloc(sizeof(struct omap_mcpdm), GFP_KERNEL); - if (!mcpdm) { - ret = -ENOMEM; - goto exit; - } - - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (res == NULL) { - dev_err(&pdev->dev, "no resource\n"); - goto err_resource; - } - - spin_lock_init(&mcpdm->lock); - mcpdm->free = 1; - mcpdm->io_base = ioremap(res->start, resource_size(res)); - if (!mcpdm->io_base) { - ret = -ENOMEM; - goto err_resource; - } - - mcpdm->irq = platform_get_irq(pdev, 0); - - mcpdm->clk = clk_get(&pdev->dev, "pdm_ck"); - if (IS_ERR(mcpdm->clk)) { - ret = PTR_ERR(mcpdm->clk); - dev_err(&pdev->dev, "unable to get pdm_ck: %d\n", ret); - goto err_clk; - } - - mcpdm->dev = &pdev->dev; - platform_set_drvdata(pdev, mcpdm); - - return 0; - -err_clk: - iounmap(mcpdm->io_base); -err_resource: - kfree(mcpdm); -exit: - return ret; -} - -int omap_mcpdm_remove(struct platform_device *pdev) -{ - struct omap_mcpdm *mcpdm_ptr = platform_get_drvdata(pdev); - - platform_set_drvdata(pdev, NULL); - - clk_put(mcpdm_ptr->clk); - - iounmap(mcpdm_ptr->io_base); - - mcpdm_ptr->clk = NULL; - mcpdm_ptr->free = 0; - mcpdm_ptr->dev = NULL; - - kfree(mcpdm_ptr); - - return 0; -} - diff --git a/sound/soc/omap/mcpdm.h b/sound/soc/omap/mcpdm.h deleted file mode 100644 index 20c20a8649fe..000000000000 --- a/sound/soc/omap/mcpdm.h +++ /dev/null @@ -1,153 +0,0 @@ -/* - * mcpdm.h -- Defines for McPDM driver - * - * Author: Jorge Eduardo Candelaria <x0107209@ti.com> - * - * This program is free software; you can redistribute it and/or - * modify it under the terms of the GNU General Public License - * version 2 as published by the Free Software Foundation. - * - * This program is distributed in the hope that it will be useful, but - * WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA - * 02110-1301 USA - * - */ - -/* McPDM registers */ - -#define MCPDM_REVISION 0x00 -#define MCPDM_SYSCONFIG 0x10 -#define MCPDM_IRQSTATUS_RAW 0x24 -#define MCPDM_IRQSTATUS 0x28 -#define MCPDM_IRQENABLE_SET 0x2C -#define MCPDM_IRQENABLE_CLR 0x30 -#define MCPDM_IRQWAKE_EN 0x34 -#define MCPDM_DMAENABLE_SET 0x38 -#define MCPDM_DMAENABLE_CLR 0x3C -#define MCPDM_DMAWAKEEN 0x40 -#define MCPDM_CTRL 0x44 -#define MCPDM_DN_DATA 0x48 -#define MCPDM_UP_DATA 0x4C -#define MCPDM_FIFO_CTRL_DN 0x50 -#define MCPDM_FIFO_CTRL_UP 0x54 -#define MCPDM_DN_OFFSET 0x58 - -/* - * MCPDM_IRQ bit fields - * IRQSTATUS_RAW, IRQSTATUS, IRQENABLE_SET, IRQENABLE_CLR - */ - -#define MCPDM_DN_IRQ (1 << 0) -#define MCPDM_DN_IRQ_EMPTY (1 << 1) -#define MCPDM_DN_IRQ_ALMST_EMPTY (1 << 2) -#define MCPDM_DN_IRQ_FULL (1 << 3) - -#define MCPDM_UP_IRQ (1 << 8) -#define MCPDM_UP_IRQ_EMPTY (1 << 9) -#define MCPDM_UP_IRQ_ALMST_FULL (1 << 10) -#define MCPDM_UP_IRQ_FULL (1 << 11) - -#define MCPDM_DOWNLINK_IRQ_MASK 0x00F -#define MCPDM_UPLINK_IRQ_MASK 0xF00 - -/* - * MCPDM_DMAENABLE bit fields - */ - -#define DMA_DN_ENABLE 0x1 -#define DMA_UP_ENABLE 0x2 - -/* - * MCPDM_CTRL bit fields - */ - -#define PDM_UP1_EN 0x0001 -#define PDM_UP2_EN 0x0002 -#define PDM_UP3_EN 0x0004 -#define PDM_DN1_EN 0x0008 -#define PDM_DN2_EN 0x0010 -#define PDM_DN3_EN 0x0020 -#define PDM_DN4_EN 0x0040 -#define PDM_DN5_EN 0x0080 -#define PDMOUTFORMAT 0x0100 -#define CMD_INT 0x0200 -#define STATUS_INT 0x0400 -#define SW_UP_RST 0x0800 -#define SW_DN_RST 0x1000 -#define PDM_UP_MASK 0x007 -#define PDM_DN_MASK 0x0F8 -#define PDM_CMD_MASK 0x200 -#define PDM_STATUS_MASK 0x400 - - -#define PDMOUTFORMAT_LJUST (0 << 8) -#define PDMOUTFORMAT_RJUST (1 << 8) - -/* - * MCPDM_FIFO_CTRL bit fields - */ - -#define UP_THRES_MAX 0xF -#define DN_THRES_MAX 0xF - -/* - * MCPDM_DN_OFFSET bit fields - */ - -#define DN_OFST_RX1_EN 0x0001 -#define DN_OFST_RX2_EN 0x0100 - -#define DN_OFST_RX1 1 -#define DN_OFST_RX2 9 -#define DN_OFST_MAX 0x1F - -#define MCPDM_UPLINK 1 -#define MCPDM_DOWNLINK 2 - -struct omap_mcpdm_link { - int irq_mask; - int threshold; - int format; - int channels; -}; - -struct omap_mcpdm_platform_data { - unsigned long phys_base; - u16 irq; -}; - -struct omap_mcpdm { - struct device *dev; - unsigned long phys_base; - void __iomem *io_base; - u8 free; - int irq; - - spinlock_t lock; - struct omap_mcpdm_platform_data *pdata; - struct clk *clk; - struct omap_mcpdm_link *downlink; - struct omap_mcpdm_link *uplink; - struct completion irq_completion; - - int dn_channels; - int up_channels; -}; - -extern void omap_mcpdm_start(int stream); -extern void omap_mcpdm_stop(int stream); -extern int omap_mcpdm_capture_open(struct omap_mcpdm_link *uplink); -extern int omap_mcpdm_playback_open(struct omap_mcpdm_link *downlink); -extern int omap_mcpdm_capture_close(struct omap_mcpdm_link *uplink); -extern int omap_mcpdm_playback_close(struct omap_mcpdm_link *downlink); -extern int omap_mcpdm_request(void); -extern void omap_mcpdm_free(void); -extern int omap_mcpdm_set_offset(int offset1, int offset2); -int __devinit omap_mcpdm_probe(struct platform_device *pdev); -int omap_mcpdm_remove(struct platform_device *pdev); diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index 62e292f49313..7e3c20c965c6 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -115,25 +115,8 @@ static int n810_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; int err; - /* Set codec DAI configuration */ - err = snd_soc_dai_set_fmt(codec_dai, - SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM); - if (err < 0) - return err; - - /* Set cpu DAI configuration */ - err = snd_soc_dai_set_fmt(cpu_dai, - SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM); - if (err < 0) - return err; - /* Set the codec system clock for DAC and ADC */ err = snd_soc_dai_set_sysclk(codec_dai, 0, 12000000, SND_SOC_CLOCK_IN); @@ -274,7 +257,6 @@ static int n810_aic33_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; struct snd_soc_dapm_context *dapm = &codec->dapm; - int err; /* Not connected */ snd_soc_dapm_nc_pin(dapm, "MONO_LOUT"); @@ -286,21 +268,6 @@ static int n810_aic33_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_nc_pin(dapm, "LINE2L"); snd_soc_dapm_nc_pin(dapm, "LINE2R"); - /* Add N810 specific controls */ - err = snd_soc_add_controls(codec, aic33_n810_controls, - ARRAY_SIZE(aic33_n810_controls)); - if (err < 0) - return err; - - /* Add N810 specific widgets */ - snd_soc_dapm_new_controls(dapm, aic33_dapm_widgets, - ARRAY_SIZE(aic33_dapm_widgets)); - - /* Set up N810 specific audio path audio_map */ - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - - snd_soc_dapm_sync(dapm); - return 0; } @@ -312,6 +279,8 @@ static struct snd_soc_dai_link n810_dai = { .platform_name = "omap-pcm-audio", .codec_name = "tlv320aic3x-codec.2-0018", .codec_dai_name = "tlv320aic3x-hifi", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, .init = n810_aic33_init, .ops = &n810_ops, }; @@ -321,6 +290,13 @@ static struct snd_soc_card snd_soc_n810 = { .name = "N810", .dai_link = &n810_dai, .num_links = 1, + + .controls = aic33_n810_controls, + .num_controls = ARRAY_SIZE(aic33_n810_controls), + .dapm_widgets = aic33_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(aic33_dapm_widgets), + .dapm_routes = audio_map, + .num_dapm_routes = ARRAY_SIZE(audio_map), }; static struct platform_device *n810_snd_device; diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 478d60778453..4314647e735e 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -317,6 +317,10 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, return 0; } + regs->rcr2 &= ~(RPHASE | RFRLEN2(0x7f) | RWDLEN2(7)); + regs->xcr2 &= ~(RPHASE | XFRLEN2(0x7f) | XWDLEN2(7)); + regs->rcr1 &= ~(RFRLEN1(0x7f) | RWDLEN1(7)); + regs->xcr1 &= ~(XFRLEN1(0x7f) | XWDLEN1(7)); format = mcbsp_data->fmt & SND_SOC_DAIFMT_FORMAT_MASK; wpf = channels = params_channels(params); if (channels == 2 && (format == SND_SOC_DAIFMT_I2S || @@ -369,6 +373,8 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, framesize = wlen * channels; /* Set FS period and length in terms of bit clock periods */ + regs->srgr2 &= ~FPER(0xfff); + regs->srgr1 &= ~FWID(0xff); switch (format) { case SND_SOC_DAIFMT_I2S: case SND_SOC_DAIFMT_LEFT_J: @@ -398,7 +404,7 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, { struct omap_mcbsp_data *mcbsp_data = snd_soc_dai_get_drvdata(cpu_dai); struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs; - unsigned int temp_fmt = fmt; + bool inv_fs = false; if (mcbsp_data->configured) return 0; @@ -430,21 +436,21 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, regs->xcr2 |= XDATDLY(0); regs->spcr1 |= RJUST(2); /* Invert FS polarity configuration */ - temp_fmt ^= SND_SOC_DAIFMT_NB_IF; + inv_fs = true; break; case SND_SOC_DAIFMT_DSP_A: /* 1-bit data delay */ regs->rcr2 |= RDATDLY(1); regs->xcr2 |= XDATDLY(1); /* Invert FS polarity configuration */ - temp_fmt ^= SND_SOC_DAIFMT_NB_IF; + inv_fs = true; break; case SND_SOC_DAIFMT_DSP_B: /* 0-bit data delay */ regs->rcr2 |= RDATDLY(0); regs->xcr2 |= XDATDLY(0); /* Invert FS polarity configuration */ - temp_fmt ^= SND_SOC_DAIFMT_NB_IF; + inv_fs = true; break; default: /* Unsupported data format */ @@ -468,7 +474,7 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, } /* Set bit clock (CLKX/CLKR) and FS polarities */ - switch (temp_fmt & SND_SOC_DAIFMT_INV_MASK) { + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { case SND_SOC_DAIFMT_NB_NF: /* * Normal BCLK + FS. @@ -489,6 +495,8 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, default: return -EINVAL; } + if (inv_fs == true) + regs->pcr0 ^= FSXP | FSRP; return 0; } @@ -503,6 +511,7 @@ static int omap_mcbsp_dai_set_clkdiv(struct snd_soc_dai *cpu_dai, return -ENODEV; mcbsp_data->clk_div = div; + regs->srgr1 &= ~CLKGDV(0xff); regs->srgr1 |= CLKGDV(div - 1); return 0; @@ -516,11 +525,12 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs; int err = 0; - if (mcbsp_data->active) + if (mcbsp_data->active) { if (freq == mcbsp_data->in_freq) return 0; else return -EBUSY; + } /* The McBSP signal muxing functions are only available on McBSP1 */ if (clk_id == OMAP_MCBSP_CLKR_SRC_CLKR || @@ -531,6 +541,8 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, return -EINVAL; mcbsp_data->in_freq = freq; + regs->srgr2 &= ~CLKSM; + regs->pcr0 &= ~SCLKME; switch (clk_id) { case OMAP_MCBSP_SYSCLK_CLK: @@ -605,8 +617,7 @@ static int mcbsp_dai_probe(struct snd_soc_dai *dai) return 0; } -static struct snd_soc_dai_driver omap_mcbsp_dai = -{ +static struct snd_soc_dai_driver omap_mcbsp_dai = { .probe = mcbsp_dai_probe, .playback = { .channels_min = 1, diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c index bed09c27e44c..41d17067cc73 100644 --- a/sound/soc/omap/omap-mcpdm.c +++ b/sound/soc/omap/omap-mcpdm.c @@ -1,11 +1,12 @@ /* * omap-mcpdm.c -- OMAP ALSA SoC DAI driver using McPDM port * - * Copyright (C) 2009 Texas Instruments + * Copyright (C) 2009 - 2011 Texas Instruments * - * Author: Misael Lopez Cruz <x0052729@ti.com> + * Author: Misael Lopez Cruz <misael.lopez@ti.com> * Contact: Jorge Eduardo Candelaria <x0107209@ti.com> * Margarita Olaya <magi.olaya@ti.com> + * Peter Ujfalusi <peter.ujfalusi@ti.com> * * This program is free software; you can redistribute it and/or * modify it under the terms of the GNU General Public License @@ -25,41 +26,42 @@ #include <linux/init.h> #include <linux/module.h> -#include <linux/device.h> +#include <linux/platform_device.h> +#include <linux/interrupt.h> +#include <linux/err.h> +#include <linux/io.h> +#include <linux/irq.h> +#include <linux/slab.h> +#include <linux/pm_runtime.h> + #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> -#include <sound/initval.h> #include <sound/soc.h> #include <plat/dma.h> -#include <plat/mcbsp.h> -#include "mcpdm.h" +#include <plat/omap_hwmod.h> +#include "omap-mcpdm.h" #include "omap-pcm.h" -struct omap_mcpdm_data { - struct omap_mcpdm_link *links; - int active; -}; +struct omap_mcpdm { + struct device *dev; + unsigned long phys_base; + void __iomem *io_base; + int irq; -static struct omap_mcpdm_link omap_mcpdm_links[] = { - /* downlink */ - { - .irq_mask = MCPDM_DN_IRQ_EMPTY | MCPDM_DN_IRQ_FULL, - .threshold = 1, - .format = PDMOUTFORMAT_LJUST, - }, - /* uplink */ - { - .irq_mask = MCPDM_UP_IRQ_EMPTY | MCPDM_UP_IRQ_FULL, - .threshold = 1, - .format = PDMOUTFORMAT_LJUST, - }, -}; + struct mutex mutex; + + /* channel data */ + u32 dn_channels; + u32 up_channels; + + /* McPDM FIFO thresholds */ + u32 dn_threshold; + u32 up_threshold; -static struct omap_mcpdm_data mcpdm_data = { - .links = omap_mcpdm_links, - .active = 0, + /* McPDM dn offsets for rx1, and 2 channels */ + u32 dn_rx_offset; }; /* @@ -71,88 +73,259 @@ static struct omap_pcm_dma_data omap_mcpdm_dai_dma_params[] = { .dma_req = OMAP44XX_DMA_MCPDM_DL, .data_type = OMAP_DMA_DATA_TYPE_S32, .sync_mode = OMAP_DMA_SYNC_PACKET, - .packet_size = 16, - .port_addr = OMAP44XX_MCPDM_L3_BASE + MCPDM_DN_DATA, + .port_addr = OMAP44XX_MCPDM_L3_BASE + MCPDM_REG_DN_DATA, }, { .name = "Audio capture", .dma_req = OMAP44XX_DMA_MCPDM_UP, .data_type = OMAP_DMA_DATA_TYPE_S32, .sync_mode = OMAP_DMA_SYNC_PACKET, - .packet_size = 16, - .port_addr = OMAP44XX_MCPDM_L3_BASE + MCPDM_UP_DATA, + .port_addr = OMAP44XX_MCPDM_L3_BASE + MCPDM_REG_UP_DATA, }, }; -static int omap_mcpdm_dai_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) +static inline void omap_mcpdm_write(struct omap_mcpdm *mcpdm, u16 reg, u32 val) { - int err = 0; + __raw_writel(val, mcpdm->io_base + reg); +} - if (!dai->active) - err = omap_mcpdm_request(); +static inline int omap_mcpdm_read(struct omap_mcpdm *mcpdm, u16 reg) +{ + return __raw_readl(mcpdm->io_base + reg); +} - return err; +#ifdef DEBUG +static void omap_mcpdm_reg_dump(struct omap_mcpdm *mcpdm) +{ + dev_dbg(mcpdm->dev, "***********************\n"); + dev_dbg(mcpdm->dev, "IRQSTATUS_RAW: 0x%04x\n", + omap_mcpdm_read(mcpdm, MCPDM_REG_IRQSTATUS_RAW)); + dev_dbg(mcpdm->dev, "IRQSTATUS: 0x%04x\n", + omap_mcpdm_read(mcpdm, MCPDM_REG_IRQSTATUS)); + dev_dbg(mcpdm->dev, "IRQENABLE_SET: 0x%04x\n", + omap_mcpdm_read(mcpdm, MCPDM_REG_IRQENABLE_SET)); + dev_dbg(mcpdm->dev, "IRQENABLE_CLR: 0x%04x\n", + omap_mcpdm_read(mcpdm, MCPDM_REG_IRQENABLE_CLR)); + dev_dbg(mcpdm->dev, "IRQWAKE_EN: 0x%04x\n", + omap_mcpdm_read(mcpdm, MCPDM_REG_IRQWAKE_EN)); + dev_dbg(mcpdm->dev, "DMAENABLE_SET: 0x%04x\n", + omap_mcpdm_read(mcpdm, MCPDM_REG_DMAENABLE_SET)); + dev_dbg(mcpdm->dev, "DMAENABLE_CLR: 0x%04x\n", + omap_mcpdm_read(mcpdm, MCPDM_REG_DMAENABLE_CLR)); + dev_dbg(mcpdm->dev, "DMAWAKEEN: 0x%04x\n", + omap_mcpdm_read(mcpdm, MCPDM_REG_DMAWAKEEN)); + dev_dbg(mcpdm->dev, "CTRL: 0x%04x\n", + omap_mcpdm_read(mcpdm, MCPDM_REG_CTRL)); + dev_dbg(mcpdm->dev, "DN_DATA: 0x%04x\n", + omap_mcpdm_read(mcpdm, MCPDM_REG_DN_DATA)); + dev_dbg(mcpdm->dev, "UP_DATA: 0x%04x\n", + omap_mcpdm_read(mcpdm, MCPDM_REG_UP_DATA)); + dev_dbg(mcpdm->dev, "FIFO_CTRL_DN: 0x%04x\n", + omap_mcpdm_read(mcpdm, MCPDM_REG_FIFO_CTRL_DN)); + dev_dbg(mcpdm->dev, "FIFO_CTRL_UP: 0x%04x\n", + omap_mcpdm_read(mcpdm, MCPDM_REG_FIFO_CTRL_UP)); + dev_dbg(mcpdm->dev, "***********************\n"); } +#else +static void omap_mcpdm_reg_dump(struct omap_mcpdm *mcpdm) {} +#endif -static void omap_mcpdm_dai_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) +/* + * Enables the transfer through the PDM interface to/from the Phoenix + * codec by enabling the corresponding UP or DN channels. + */ +static void omap_mcpdm_start(struct omap_mcpdm *mcpdm) +{ + u32 ctrl = omap_mcpdm_read(mcpdm, MCPDM_REG_CTRL); + + ctrl |= (MCPDM_SW_DN_RST | MCPDM_SW_UP_RST); + omap_mcpdm_write(mcpdm, MCPDM_REG_CTRL, ctrl); + + ctrl |= mcpdm->dn_channels | mcpdm->up_channels; + omap_mcpdm_write(mcpdm, MCPDM_REG_CTRL, ctrl); + + ctrl &= ~(MCPDM_SW_DN_RST | MCPDM_SW_UP_RST); + omap_mcpdm_write(mcpdm, MCPDM_REG_CTRL, ctrl); +} + +/* + * Disables the transfer through the PDM interface to/from the Phoenix + * codec by disabling the corresponding UP or DN channels. + */ +static void omap_mcpdm_stop(struct omap_mcpdm *mcpdm) +{ + u32 ctrl = omap_mcpdm_read(mcpdm, MCPDM_REG_CTRL); + + ctrl |= (MCPDM_SW_DN_RST | MCPDM_SW_UP_RST); + omap_mcpdm_write(mcpdm, MCPDM_REG_CTRL, ctrl); + + ctrl &= ~(mcpdm->dn_channels | mcpdm->up_channels); + omap_mcpdm_write(mcpdm, MCPDM_REG_CTRL, ctrl); + + ctrl &= ~(MCPDM_SW_DN_RST | MCPDM_SW_UP_RST); + omap_mcpdm_write(mcpdm, MCPDM_REG_CTRL, ctrl); + +} + +/* + * Is the physical McPDM interface active. + */ +static inline int omap_mcpdm_active(struct omap_mcpdm *mcpdm) +{ + return omap_mcpdm_read(mcpdm, MCPDM_REG_CTRL) & + (MCPDM_PDM_DN_MASK | MCPDM_PDM_UP_MASK); +} + +/* + * Configures McPDM uplink, and downlink for audio. + * This function should be called before omap_mcpdm_start. + */ +static void omap_mcpdm_open_streams(struct omap_mcpdm *mcpdm) +{ + omap_mcpdm_write(mcpdm, MCPDM_REG_IRQENABLE_SET, + MCPDM_DN_IRQ_EMPTY | MCPDM_DN_IRQ_FULL | + MCPDM_UP_IRQ_EMPTY | MCPDM_UP_IRQ_FULL); + + /* Enable DN RX1/2 offset cancellation feature, if configured */ + if (mcpdm->dn_rx_offset) { + u32 dn_offset = mcpdm->dn_rx_offset; + + omap_mcpdm_write(mcpdm, MCPDM_REG_DN_OFFSET, dn_offset); + dn_offset |= (MCPDM_DN_OFST_RX1_EN | MCPDM_DN_OFST_RX2_EN); + omap_mcpdm_write(mcpdm, MCPDM_REG_DN_OFFSET, dn_offset); + } + + omap_mcpdm_write(mcpdm, MCPDM_REG_FIFO_CTRL_DN, mcpdm->dn_threshold); + omap_mcpdm_write(mcpdm, MCPDM_REG_FIFO_CTRL_UP, mcpdm->up_threshold); + + omap_mcpdm_write(mcpdm, MCPDM_REG_DMAENABLE_SET, + MCPDM_DMA_DN_ENABLE | MCPDM_DMA_UP_ENABLE); +} + +/* + * Cleans McPDM uplink, and downlink configuration. + * This function should be called when the stream is closed. + */ +static void omap_mcpdm_close_streams(struct omap_mcpdm *mcpdm) +{ + /* Disable irq request generation for downlink */ + omap_mcpdm_write(mcpdm, MCPDM_REG_IRQENABLE_CLR, + MCPDM_DN_IRQ_EMPTY | MCPDM_DN_IRQ_FULL); + + /* Disable DMA request generation for downlink */ + omap_mcpdm_write(mcpdm, MCPDM_REG_DMAENABLE_CLR, MCPDM_DMA_DN_ENABLE); + + /* Disable irq request generation for uplink */ + omap_mcpdm_write(mcpdm, MCPDM_REG_IRQENABLE_CLR, + MCPDM_UP_IRQ_EMPTY | MCPDM_UP_IRQ_FULL); + + /* Disable DMA request generation for uplink */ + omap_mcpdm_write(mcpdm, MCPDM_REG_DMAENABLE_CLR, MCPDM_DMA_UP_ENABLE); + + /* Disable RX1/2 offset cancellation */ + if (mcpdm->dn_rx_offset) + omap_mcpdm_write(mcpdm, MCPDM_REG_DN_OFFSET, 0); +} + +static irqreturn_t omap_mcpdm_irq_handler(int irq, void *dev_id) +{ + struct omap_mcpdm *mcpdm = dev_id; + int irq_status; + + irq_status = omap_mcpdm_read(mcpdm, MCPDM_REG_IRQSTATUS); + + /* Acknowledge irq event */ + omap_mcpdm_write(mcpdm, MCPDM_REG_IRQSTATUS, irq_status); + + if (irq_status & MCPDM_DN_IRQ_FULL) + dev_dbg(mcpdm->dev, "DN (playback) FIFO Full\n"); + + if (irq_status & MCPDM_DN_IRQ_EMPTY) + dev_dbg(mcpdm->dev, "DN (playback) FIFO Empty\n"); + + if (irq_status & MCPDM_DN_IRQ) + dev_dbg(mcpdm->dev, "DN (playback) write request\n"); + + if (irq_status & MCPDM_UP_IRQ_FULL) + dev_dbg(mcpdm->dev, "UP (capture) FIFO Full\n"); + + if (irq_status & MCPDM_UP_IRQ_EMPTY) + dev_dbg(mcpdm->dev, "UP (capture) FIFO Empty\n"); + + if (irq_status & MCPDM_UP_IRQ) + dev_dbg(mcpdm->dev, "UP (capture) write request\n"); + + return IRQ_HANDLED; +} + +static int omap_mcpdm_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { - if (!dai->active) - omap_mcpdm_free(); + struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai); + + mutex_lock(&mcpdm->mutex); + + if (!dai->active) { + pm_runtime_get_sync(mcpdm->dev); + + /* Enable watch dog for ES above ES 1.0 to avoid saturation */ + if (omap_rev() != OMAP4430_REV_ES1_0) { + u32 ctrl = omap_mcpdm_read(mcpdm, MCPDM_REG_CTRL); + + omap_mcpdm_write(mcpdm, MCPDM_REG_CTRL, + ctrl | MCPDM_WD_EN); + } + omap_mcpdm_open_streams(mcpdm); + } + + mutex_unlock(&mcpdm->mutex); + + return 0; } -static int omap_mcpdm_dai_trigger(struct snd_pcm_substream *substream, int cmd, +static void omap_mcpdm_dai_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct omap_mcpdm_data *mcpdm_priv = snd_soc_dai_get_drvdata(dai); - int stream = substream->stream; - int err = 0; - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - if (!mcpdm_priv->active++) - omap_mcpdm_start(stream); - break; + struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai); - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - if (!--mcpdm_priv->active) - omap_mcpdm_stop(stream); - break; - default: - err = -EINVAL; + mutex_lock(&mcpdm->mutex); + + if (!dai->active) { + if (omap_mcpdm_active(mcpdm)) { + omap_mcpdm_stop(mcpdm); + omap_mcpdm_close_streams(mcpdm); + } + + if (!omap_mcpdm_active(mcpdm)) + pm_runtime_put_sync(mcpdm->dev); } - return err; + mutex_unlock(&mcpdm->mutex); } static int omap_mcpdm_dai_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct omap_mcpdm_data *mcpdm_priv = snd_soc_dai_get_drvdata(dai); - struct omap_mcpdm_link *mcpdm_links = mcpdm_priv->links; + struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai); int stream = substream->stream; - int channels, err, link_mask = 0; - - snd_soc_dai_set_dma_data(dai, substream, - &omap_mcpdm_dai_dma_params[stream]); + struct omap_pcm_dma_data *dma_data; + int channels; + int link_mask = 0; channels = params_channels(params); switch (channels) { + case 5: + if (stream == SNDRV_PCM_STREAM_CAPTURE) + /* up to 3 channels for capture */ + return -EINVAL; + link_mask |= 1 << 4; case 4: if (stream == SNDRV_PCM_STREAM_CAPTURE) - /* up to 2 channels for capture */ + /* up to 3 channels for capture */ return -EINVAL; link_mask |= 1 << 3; case 3: - if (stream == SNDRV_PCM_STREAM_CAPTURE) - /* up to 2 channels for capture */ - return -EINVAL; link_mask |= 1 << 2; case 2: link_mask |= 1 << 1; @@ -164,95 +337,187 @@ static int omap_mcpdm_dai_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } + dma_data = &omap_mcpdm_dai_dma_params[stream]; + + /* Configure McPDM channels, and DMA packet size */ if (stream == SNDRV_PCM_STREAM_PLAYBACK) { - mcpdm_links[stream].channels = link_mask << 3; - err = omap_mcpdm_playback_open(&mcpdm_links[stream]); + mcpdm->dn_channels = link_mask << 3; + dma_data->packet_size = + (MCPDM_DN_THRES_MAX - mcpdm->dn_threshold) * channels; } else { - mcpdm_links[stream].channels = link_mask << 0; - err = omap_mcpdm_capture_open(&mcpdm_links[stream]); + mcpdm->up_channels = link_mask << 0; + dma_data->packet_size = mcpdm->up_threshold * channels; } - return err; + snd_soc_dai_set_dma_data(dai, substream, dma_data); + + return 0; } -static int omap_mcpdm_dai_hw_free(struct snd_pcm_substream *substream, +static int omap_mcpdm_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct omap_mcpdm_data *mcpdm_priv = snd_soc_dai_get_drvdata(dai); - struct omap_mcpdm_link *mcpdm_links = mcpdm_priv->links; - int stream = substream->stream; - int err; + struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - err = omap_mcpdm_playback_close(&mcpdm_links[stream]); - else - err = omap_mcpdm_capture_close(&mcpdm_links[stream]); + if (!omap_mcpdm_active(mcpdm)) { + omap_mcpdm_start(mcpdm); + omap_mcpdm_reg_dump(mcpdm); + } - return err; + return 0; } static struct snd_soc_dai_ops omap_mcpdm_dai_ops = { .startup = omap_mcpdm_dai_startup, .shutdown = omap_mcpdm_dai_shutdown, - .trigger = omap_mcpdm_dai_trigger, .hw_params = omap_mcpdm_dai_hw_params, - .hw_free = omap_mcpdm_dai_hw_free, + .prepare = omap_mcpdm_prepare, }; -#define OMAP_MCPDM_RATES (SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) -#define OMAP_MCPDM_FORMATS (SNDRV_PCM_FMTBIT_S32_LE) +static int omap_mcpdm_probe(struct snd_soc_dai *dai) +{ + struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai); + int ret; -static int omap_mcpdm_dai_probe(struct snd_soc_dai *dai) + pm_runtime_enable(mcpdm->dev); + + /* Disable lines while request is ongoing */ + pm_runtime_get_sync(mcpdm->dev); + omap_mcpdm_write(mcpdm, MCPDM_REG_CTRL, 0x00); + + ret = request_irq(mcpdm->irq, omap_mcpdm_irq_handler, + 0, "McPDM", (void *)mcpdm); + + pm_runtime_put_sync(mcpdm->dev); + + if (ret) { + dev_err(mcpdm->dev, "Request for IRQ failed\n"); + pm_runtime_disable(mcpdm->dev); + } + + /* Configure McPDM threshold values */ + mcpdm->dn_threshold = 2; + mcpdm->up_threshold = MCPDM_UP_THRES_MAX - 3; + return ret; +} + +static int omap_mcpdm_remove(struct snd_soc_dai *dai) { - snd_soc_dai_set_drvdata(dai, &mcpdm_data); + struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai); + + free_irq(mcpdm->irq, (void *)mcpdm); + pm_runtime_disable(mcpdm->dev); + return 0; } +#define OMAP_MCPDM_RATES (SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) +#define OMAP_MCPDM_FORMATS SNDRV_PCM_FMTBIT_S32_LE + static struct snd_soc_dai_driver omap_mcpdm_dai = { - .probe = omap_mcpdm_dai_probe, + .probe = omap_mcpdm_probe, + .remove = omap_mcpdm_remove, + .probe_order = SND_SOC_COMP_ORDER_LATE, + .remove_order = SND_SOC_COMP_ORDER_EARLY, .playback = { .channels_min = 1, - .channels_max = 4, + .channels_max = 5, .rates = OMAP_MCPDM_RATES, .formats = OMAP_MCPDM_FORMATS, }, .capture = { .channels_min = 1, - .channels_max = 2, + .channels_max = 3, .rates = OMAP_MCPDM_RATES, .formats = OMAP_MCPDM_FORMATS, }, .ops = &omap_mcpdm_dai_ops, }; +void omap_mcpdm_configure_dn_offsets(struct snd_soc_pcm_runtime *rtd, + u8 rx1, u8 rx2) +{ + struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(rtd->cpu_dai); + + mcpdm->dn_rx_offset = MCPDM_DNOFST_RX1(rx1) | MCPDM_DNOFST_RX2(rx2); +} +EXPORT_SYMBOL_GPL(omap_mcpdm_configure_dn_offsets); + static __devinit int asoc_mcpdm_probe(struct platform_device *pdev) { - int ret; + struct omap_mcpdm *mcpdm; + struct resource *res; + int ret = 0; + + mcpdm = kzalloc(sizeof(struct omap_mcpdm), GFP_KERNEL); + if (!mcpdm) + return -ENOMEM; + + platform_set_drvdata(pdev, mcpdm); + + mutex_init(&mcpdm->mutex); + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (res == NULL) { + dev_err(&pdev->dev, "no resource\n"); + goto err_res; + } + + if (!request_mem_region(res->start, resource_size(res), "McPDM")) { + ret = -EBUSY; + goto err_res; + } + + mcpdm->io_base = ioremap(res->start, resource_size(res)); + if (!mcpdm->io_base) { + ret = -ENOMEM; + goto err_iomap; + } + + mcpdm->irq = platform_get_irq(pdev, 0); + if (mcpdm->irq < 0) { + ret = mcpdm->irq; + goto err_irq; + } + + mcpdm->dev = &pdev->dev; - ret = omap_mcpdm_probe(pdev); - if (ret < 0) - return ret; ret = snd_soc_register_dai(&pdev->dev, &omap_mcpdm_dai); - if (ret < 0) - omap_mcpdm_remove(pdev); + if (!ret) + return 0; + +err_irq: + iounmap(mcpdm->io_base); +err_iomap: + release_mem_region(res->start, resource_size(res)); +err_res: + kfree(mcpdm); return ret; } static int __devexit asoc_mcpdm_remove(struct platform_device *pdev) { + struct omap_mcpdm *mcpdm = platform_get_drvdata(pdev); + struct resource *res; + snd_soc_unregister_dai(&pdev->dev); - omap_mcpdm_remove(pdev); + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + iounmap(mcpdm->io_base); + release_mem_region(res->start, resource_size(res)); + + kfree(mcpdm); return 0; } static struct platform_driver asoc_mcpdm_driver = { .driver = { - .name = "omap-mcpdm-dai", - .owner = THIS_MODULE, + .name = "omap-mcpdm", + .owner = THIS_MODULE, }, - .probe = asoc_mcpdm_probe, - .remove = __devexit_p(asoc_mcpdm_remove), + .probe = asoc_mcpdm_probe, + .remove = __devexit_p(asoc_mcpdm_remove), }; static int __init snd_omap_mcpdm_init(void) @@ -267,6 +532,6 @@ static void __exit snd_omap_mcpdm_exit(void) } module_exit(snd_omap_mcpdm_exit); -MODULE_AUTHOR("Misael Lopez Cruz <x0052729@ti.com>"); +MODULE_AUTHOR("Misael Lopez Cruz <misael.lopez@ti.com>"); MODULE_DESCRIPTION("OMAP PDM SoC Interface"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/omap/omap-mcpdm.h b/sound/soc/omap/omap-mcpdm.h new file mode 100644 index 000000000000..de8cf26595b1 --- /dev/null +++ b/sound/soc/omap/omap-mcpdm.h @@ -0,0 +1,107 @@ +/* + * omap-mcpdm.h + * + * Copyright (C) 2009 - 2011 Texas Instruments + * + * Contact: Misael Lopez Cruz <misael.lopez@ti.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#ifndef __OMAP_MCPDM_H__ +#define __OMAP_MCPDM_H__ + +#define MCPDM_REG_REVISION 0x00 +#define MCPDM_REG_SYSCONFIG 0x10 +#define MCPDM_REG_IRQSTATUS_RAW 0x24 +#define MCPDM_REG_IRQSTATUS 0x28 +#define MCPDM_REG_IRQENABLE_SET 0x2C +#define MCPDM_REG_IRQENABLE_CLR 0x30 +#define MCPDM_REG_IRQWAKE_EN 0x34 +#define MCPDM_REG_DMAENABLE_SET 0x38 +#define MCPDM_REG_DMAENABLE_CLR 0x3C +#define MCPDM_REG_DMAWAKEEN 0x40 +#define MCPDM_REG_CTRL 0x44 +#define MCPDM_REG_DN_DATA 0x48 +#define MCPDM_REG_UP_DATA 0x4C +#define MCPDM_REG_FIFO_CTRL_DN 0x50 +#define MCPDM_REG_FIFO_CTRL_UP 0x54 +#define MCPDM_REG_DN_OFFSET 0x58 + +/* + * MCPDM_IRQ bit fields + * IRQSTATUS_RAW, IRQSTATUS, IRQENABLE_SET, IRQENABLE_CLR + */ + +#define MCPDM_DN_IRQ (1 << 0) +#define MCPDM_DN_IRQ_EMPTY (1 << 1) +#define MCPDM_DN_IRQ_ALMST_EMPTY (1 << 2) +#define MCPDM_DN_IRQ_FULL (1 << 3) + +#define MCPDM_UP_IRQ (1 << 8) +#define MCPDM_UP_IRQ_EMPTY (1 << 9) +#define MCPDM_UP_IRQ_ALMST_FULL (1 << 10) +#define MCPDM_UP_IRQ_FULL (1 << 11) + +#define MCPDM_DOWNLINK_IRQ_MASK 0x00F +#define MCPDM_UPLINK_IRQ_MASK 0xF00 + +/* + * MCPDM_DMAENABLE bit fields + */ + +#define MCPDM_DMA_DN_ENABLE (1 << 0) +#define MCPDM_DMA_UP_ENABLE (1 << 1) + +/* + * MCPDM_CTRL bit fields + */ + +#define MCPDM_PDM_UPLINK_EN(x) (1 << (x - 1)) /* ch1 is at bit 0 */ +#define MCPDM_PDM_DOWNLINK_EN(x) (1 << (x + 2)) /* ch1 is at bit 3 */ +#define MCPDM_PDMOUTFORMAT (1 << 8) +#define MCPDM_CMD_INT (1 << 9) +#define MCPDM_STATUS_INT (1 << 10) +#define MCPDM_SW_UP_RST (1 << 11) +#define MCPDM_SW_DN_RST (1 << 12) +#define MCPDM_WD_EN (1 << 14) +#define MCPDM_PDM_UP_MASK 0x7 +#define MCPDM_PDM_DN_MASK (0x1f << 3) + + +#define MCPDM_PDMOUTFORMAT_LJUST (0 << 8) +#define MCPDM_PDMOUTFORMAT_RJUST (1 << 8) + +/* + * MCPDM_FIFO_CTRL bit fields + */ + +#define MCPDM_UP_THRES_MAX 0xF +#define MCPDM_DN_THRES_MAX 0xF + +/* + * MCPDM_DN_OFFSET bit fields + */ + +#define MCPDM_DN_OFST_RX1_EN (1 << 0) +#define MCPDM_DNOFST_RX1(x) ((x & 0x1f) << 1) +#define MCPDM_DN_OFST_RX2_EN (1 << 8) +#define MCPDM_DNOFST_RX2(x) ((x & 0x1f) << 9) + +void omap_mcpdm_configure_dn_offsets(struct snd_soc_pcm_runtime *rtd, + u8 rx1, u8 rx2); + +#endif /* End of __OMAP_MCPDM_H__ */ diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index 9b5c88ac35b9..5e37ec915de2 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -198,6 +198,14 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream) OMAP_DMA_LAST_IRQ | OMAP_DMA_BLOCK_IRQ); else if (!substream->runtime->no_period_wakeup) omap_enable_dma_irq(prtd->dma_ch, OMAP_DMA_FRAME_IRQ); + else { + /* + * No period wakeup: + * we need to disable BLOCK_IRQ, which is enabled by the omap + * dma core at request dma time. + */ + omap_disable_dma_irq(prtd->dma_ch, OMAP_DMA_BLOCK_IRQ); + } if (!(cpu_class_is_omap1())) { omap_set_dma_src_burst_mode(prtd->dma_ch, diff --git a/sound/soc/omap/omap3evm.c b/sound/soc/omap/omap3evm.c index 0daa04469836..bf9ae2a6f901 100644 --- a/sound/soc/omap/omap3evm.c +++ b/sound/soc/omap/omap3evm.c @@ -36,29 +36,8 @@ static int omap3evm_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; int ret; - /* Set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, - SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) { - printk(KERN_ERR "Can't set codec DAI configuration\n"); - return ret; - } - - /* Set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, - SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) { - printk(KERN_ERR "Can't set cpu DAI configuration\n"); - return ret; - } - /* Set the codec system clock for DAC and ADC */ ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000, SND_SOC_CLOCK_IN); @@ -82,6 +61,8 @@ static struct snd_soc_dai_link omap3evm_dai = { .codec_dai_name = "twl4030-hifi", .platform_name = "omap-pcm-audio", .codec_name = "twl4030-codec", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, .ops = &omap3evm_ops, }; diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c index 8047c521e318..30a75b406aea 100644 --- a/sound/soc/omap/omap3pandora.c +++ b/sound/soc/omap/omap3pandora.c @@ -48,24 +48,8 @@ static int omap3pandora_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - int fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS; int ret; - /* Set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, fmt); - if (ret < 0) { - pr_err(PREFIX "can't set codec DAI configuration\n"); - return ret; - } - - /* Set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, fmt); - if (ret < 0) { - pr_err(PREFIX "can't set cpu DAI configuration\n"); - return ret; - } - /* Set the codec system clock for DAC and ADC */ ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000, SND_SOC_CLOCK_IN); @@ -189,10 +173,8 @@ static int omap3pandora_out_init(struct snd_soc_pcm_runtime *rtd) if (ret < 0) return ret; - snd_soc_dapm_add_routes(dapm, omap3pandora_out_map, + return snd_soc_dapm_add_routes(dapm, omap3pandora_out_map, ARRAY_SIZE(omap3pandora_out_map)); - - return snd_soc_dapm_sync(dapm); } static int omap3pandora_in_init(struct snd_soc_pcm_runtime *rtd) @@ -212,10 +194,8 @@ static int omap3pandora_in_init(struct snd_soc_pcm_runtime *rtd) if (ret < 0) return ret; - snd_soc_dapm_add_routes(dapm, omap3pandora_in_map, + return snd_soc_dapm_add_routes(dapm, omap3pandora_in_map, ARRAY_SIZE(omap3pandora_in_map)); - - return snd_soc_dapm_sync(dapm); } static struct snd_soc_ops omap3pandora_ops = { @@ -231,6 +211,8 @@ static struct snd_soc_dai_link omap3pandora_dai[] = { .codec_dai_name = "twl4030-hifi", .platform_name = "omap-pcm-audio", .codec_name = "twl4030-codec", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, .ops = &omap3pandora_ops, .init = omap3pandora_out_init, }, { @@ -240,6 +222,8 @@ static struct snd_soc_dai_link omap3pandora_dai[] = { .codec_dai_name = "twl4030-hifi", .platform_name = "omap-pcm-audio", .codec_name = "twl4030-codec", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, .ops = &omap3pandora_ops, .init = omap3pandora_in_init, } diff --git a/sound/soc/omap/osk5912.c b/sound/soc/omap/osk5912.c index 7e75e775fb4a..db91ccaf6c97 100644 --- a/sound/soc/omap/osk5912.c +++ b/sound/soc/omap/osk5912.c @@ -55,29 +55,8 @@ static int osk_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; int err; - /* Set codec DAI configuration */ - err = snd_soc_dai_set_fmt(codec_dai, - SND_SOC_DAIFMT_DSP_B | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM); - if (err < 0) { - printk(KERN_ERR "can't set codec DAI configuration\n"); - return err; - } - - /* Set cpu DAI configuration */ - err = snd_soc_dai_set_fmt(cpu_dai, - SND_SOC_DAIFMT_DSP_B | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM); - if (err < 0) { - printk(KERN_ERR "can't set cpu DAI configuration\n"); - return err; - } - /* Set the codec system clock for DAC and ADC */ err = snd_soc_dai_set_sysclk(codec_dai, 0, CODEC_CLOCK, SND_SOC_CLOCK_IN); @@ -112,27 +91,6 @@ static const struct snd_soc_dapm_route audio_map[] = { {"MICIN", NULL, "Mic Jack"}, }; -static int osk_tlv320aic23_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_codec *codec = rtd->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; - - /* Add osk5912 specific widgets */ - snd_soc_dapm_new_controls(dapm, tlv320aic23_dapm_widgets, - ARRAY_SIZE(tlv320aic23_dapm_widgets)); - - /* Set up osk5912 specific audio path audio_map */ - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - - snd_soc_dapm_enable_pin(dapm, "Headphone Jack"); - snd_soc_dapm_enable_pin(dapm, "Line In"); - snd_soc_dapm_enable_pin(dapm, "Mic Jack"); - - snd_soc_dapm_sync(dapm); - - return 0; -} - /* Digital audio interface glue - connects codec <--> CPU */ static struct snd_soc_dai_link osk_dai = { .name = "TLV320AIC23", @@ -141,7 +99,8 @@ static struct snd_soc_dai_link osk_dai = { .codec_dai_name = "tlv320aic23-hifi", .platform_name = "omap-pcm-audio", .codec_name = "tlv320aic23-codec", - .init = osk_tlv320aic23_init, + .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, .ops = &osk_ops, }; @@ -150,6 +109,11 @@ static struct snd_soc_card snd_soc_card_osk = { .name = "OSK5912", .dai_link = &osk_dai, .num_links = 1, + + .dapm_widgets = tlv320aic23_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tlv320aic23_dapm_widgets), + .dapm_routes = audio_map, + .num_dapm_routes = ARRAY_SIZE(audio_map), }; static struct platform_device *osk_snd_device; diff --git a/sound/soc/omap/overo.c b/sound/soc/omap/overo.c index bbcf380bfb56..739efe9e327a 100644 --- a/sound/soc/omap/overo.c +++ b/sound/soc/omap/overo.c @@ -38,29 +38,8 @@ static int overo_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; int ret; - /* Set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, - SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) { - printk(KERN_ERR "can't set codec DAI configuration\n"); - return ret; - } - - /* Set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, - SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) { - printk(KERN_ERR "can't set cpu DAI configuration\n"); - return ret; - } - /* Set the codec system clock for DAC and ADC */ ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000, SND_SOC_CLOCK_IN); @@ -84,6 +63,8 @@ static struct snd_soc_dai_link overo_dai = { .codec_dai_name = "twl4030-hifi", .platform_name = "omap-pcm-audio", .codec_name = "twl4030-codec", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, .ops = &overo_ops, }; diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c index 893300a53bab..a56842380c72 100644 --- a/sound/soc/omap/rx51.c +++ b/sound/soc/omap/rx51.c @@ -115,24 +115,6 @@ static int rx51_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - int err; - - /* Set codec DAI configuration */ - err = snd_soc_dai_set_fmt(codec_dai, - SND_SOC_DAIFMT_DSP_A | - SND_SOC_DAIFMT_IB_NF | - SND_SOC_DAIFMT_CBM_CFM); - if (err < 0) - return err; - - /* Set cpu DAI configuration */ - err = snd_soc_dai_set_fmt(cpu_dai, - SND_SOC_DAIFMT_DSP_A | - SND_SOC_DAIFMT_IB_NF | - SND_SOC_DAIFMT_CBM_CFM); - if (err < 0) - return err; /* Set the codec system clock for DAC and ADC */ return snd_soc_dai_set_sysclk(codec_dai, 0, 19200000, @@ -335,8 +317,6 @@ static int rx51_aic34_init(struct snd_soc_pcm_runtime *rtd) if (err < 0) return err; - snd_soc_dapm_sync(dapm); - /* AV jack detection */ err = snd_soc_jack_new(codec, "AV Jack", SND_JACK_HEADSET | SND_JACK_VIDEOOUT, @@ -377,6 +357,8 @@ static struct snd_soc_dai_link rx51_dai[] = { .codec_dai_name = "tlv320aic3x-hifi", .platform_name = "omap-pcm-audio", .codec_name = "tlv320aic3x-codec.2-0018", + .dai_fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_NF | + SND_SOC_DAIFMT_CBM_CFM, .init = rx51_aic34_init, .ops = &rx51_ops, }, diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c index 9f6a758029d1..4f1969de91a7 100644 --- a/sound/soc/omap/sdp3430.c +++ b/sound/soc/omap/sdp3430.c @@ -53,29 +53,8 @@ static int sdp3430_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; int ret; - /* Set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, - SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) { - printk(KERN_ERR "can't set codec DAI configuration\n"); - return ret; - } - - /* Set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, - SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) { - printk(KERN_ERR "can't set cpu DAI configuration\n"); - return ret; - } - /* Set the codec system clock for DAC and ADC */ ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000, SND_SOC_CLOCK_IN); @@ -91,49 +70,6 @@ static struct snd_soc_ops sdp3430_ops = { .hw_params = sdp3430_hw_params, }; -static int sdp3430_hw_voice_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - int ret; - - /* Set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, - SND_SOC_DAIFMT_DSP_A | - SND_SOC_DAIFMT_IB_NF | - SND_SOC_DAIFMT_CBM_CFM); - if (ret) { - printk(KERN_ERR "can't set codec DAI configuration\n"); - return ret; - } - - /* Set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, - SND_SOC_DAIFMT_DSP_A | - SND_SOC_DAIFMT_IB_NF | - SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) { - printk(KERN_ERR "can't set cpu DAI configuration\n"); - return ret; - } - - /* Set the codec system clock for DAC and ADC */ - ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000, - SND_SOC_CLOCK_IN); - if (ret < 0) { - printk(KERN_ERR "can't set codec system clock\n"); - return ret; - } - - return 0; -} - -static struct snd_soc_ops sdp3430_voice_ops = { - .hw_params = sdp3430_hw_voice_params, -}; - /* Headset jack */ static struct snd_soc_jack hs_jack; @@ -193,15 +129,6 @@ static int sdp3430_twl4030_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_dapm_context *dapm = &codec->dapm; int ret; - /* Add SDP3430 specific widgets */ - ret = snd_soc_dapm_new_controls(dapm, sdp3430_twl4030_dapm_widgets, - ARRAY_SIZE(sdp3430_twl4030_dapm_widgets)); - if (ret) - return ret; - - /* Set up SDP3430 specific audio path audio_map */ - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - /* SDP3430 connected pins */ snd_soc_dapm_enable_pin(dapm, "Ext Mic"); snd_soc_dapm_enable_pin(dapm, "Ext Spk"); @@ -223,10 +150,6 @@ static int sdp3430_twl4030_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_nc_pin(dapm, "CARKITL"); snd_soc_dapm_nc_pin(dapm, "CARKITR"); - ret = snd_soc_dapm_sync(dapm); - if (ret) - return ret; - /* Headset jack detection */ ret = snd_soc_jack_new(codec, "Headset Jack", SND_JACK_HEADSET, &hs_jack); @@ -267,6 +190,8 @@ static struct snd_soc_dai_link sdp3430_dai[] = { .codec_dai_name = "twl4030-hifi", .platform_name = "omap-pcm-audio", .codec_name = "twl4030-codec", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, .init = sdp3430_twl4030_init, .ops = &sdp3430_ops, }, @@ -277,8 +202,10 @@ static struct snd_soc_dai_link sdp3430_dai[] = { .codec_dai_name = "twl4030-voice", .platform_name = "omap-pcm-audio", .codec_name = "twl4030-codec", + .dai_fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_NF | + SND_SOC_DAIFMT_CBM_CFM, .init = sdp3430_twl4030_voice_init, - .ops = &sdp3430_voice_ops, + .ops = &sdp3430_ops, }, }; @@ -287,6 +214,11 @@ static struct snd_soc_card snd_soc_sdp3430 = { .name = "SDP3430", .dai_link = sdp3430_dai, .num_links = ARRAY_SIZE(sdp3430_dai), + + .dapm_widgets = sdp3430_twl4030_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(sdp3430_twl4030_dapm_widgets), + .dapm_routes = audio_map, + .num_dapm_routes = ARRAY_SIZE(audio_map), }; static struct platform_device *sdp3430_snd_device; diff --git a/sound/soc/omap/sdp4430.c b/sound/soc/omap/sdp4430.c index b80efb02bfca..cc3d792af5ea 100644 --- a/sound/soc/omap/sdp4430.c +++ b/sound/soc/omap/sdp4430.c @@ -32,7 +32,7 @@ #include <plat/hardware.h> #include <plat/mux.h> -#include "mcpdm.h" +#include "omap-mcpdm.h" #include "omap-pcm.h" #include "../codecs/twl6040.h" @@ -88,7 +88,7 @@ static const struct snd_soc_dapm_widget sdp4430_twl6040_dapm_widgets[] = { SND_SOC_DAPM_MIC("Headset Mic", NULL), SND_SOC_DAPM_HP("Headset Stereophone", NULL), SND_SOC_DAPM_SPK("Earphone Spk", NULL), - SND_SOC_DAPM_INPUT("Aux/FM Stereo In"), + SND_SOC_DAPM_INPUT("FM Stereo In"), }; static const struct snd_soc_dapm_route audio_map[] = { @@ -113,36 +113,22 @@ static const struct snd_soc_dapm_route audio_map[] = { {"Earphone Spk", NULL, "EP"}, /* Aux/FM Stereo In: AFML, AFMR */ - {"AFML", NULL, "Aux/FM Stereo In"}, - {"AFMR", NULL, "Aux/FM Stereo In"}, + {"AFML", NULL, "FM Stereo In"}, + {"AFMR", NULL, "FM Stereo In"}, }; static int sdp4430_twl6040_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; - int ret; - - /* Add SDP4430 specific widgets */ - ret = snd_soc_dapm_new_controls(dapm, sdp4430_twl6040_dapm_widgets, - ARRAY_SIZE(sdp4430_twl6040_dapm_widgets)); - if (ret) - return ret; - - /* Set up SDP4430 specific audio path audio_map */ - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); + int ret, hs_trim; - /* SDP4430 connected pins */ - snd_soc_dapm_enable_pin(dapm, "Ext Mic"); - snd_soc_dapm_enable_pin(dapm, "Ext Spk"); - snd_soc_dapm_enable_pin(dapm, "AFML"); - snd_soc_dapm_enable_pin(dapm, "AFMR"); - snd_soc_dapm_enable_pin(dapm, "Headset Mic"); - snd_soc_dapm_enable_pin(dapm, "Headset Stereophone"); - - ret = snd_soc_dapm_sync(dapm); - if (ret) - return ret; + /* + * Configure McPDM offset cancellation based on the HSOTRIM value from + * twl6040. + */ + hs_trim = twl6040_get_trim_value(codec, TWL6040_TRIM_HSOTRIM); + omap_mcpdm_configure_dn_offsets(rtd, TWL6040_HSF_TRIM_LEFT(hs_trim), + TWL6040_HSF_TRIM_RIGHT(hs_trim)); /* Headset jack detection */ ret = snd_soc_jack_new(codec, "Headset Jack", @@ -165,8 +151,8 @@ static int sdp4430_twl6040_init(struct snd_soc_pcm_runtime *rtd) static struct snd_soc_dai_link sdp4430_dai = { .name = "TWL6040", .stream_name = "TWL6040", - .cpu_dai_name ="omap-mcpdm-dai", - .codec_dai_name = "twl6040-hifi", + .cpu_dai_name = "omap-mcpdm", + .codec_dai_name = "twl6040-legacy", .platform_name = "omap-pcm-audio", .codec_name = "twl6040-codec", .init = sdp4430_twl6040_init, @@ -178,6 +164,11 @@ static struct snd_soc_card snd_soc_sdp4430 = { .name = "SDP4430", .dai_link = &sdp4430_dai, .num_links = 1, + + .dapm_widgets = sdp4430_twl6040_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(sdp4430_twl6040_dapm_widgets), + .dapm_routes = audio_map, + .num_dapm_routes = ARRAY_SIZE(audio_map), }; static struct platform_device *sdp4430_snd_device; diff --git a/sound/soc/omap/zoom2.c b/sound/soc/omap/zoom2.c index 9a2666ffc16c..7cf35c82368a 100644 --- a/sound/soc/omap/zoom2.c +++ b/sound/soc/omap/zoom2.c @@ -44,29 +44,8 @@ static int zoom2_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; int ret; - /* Set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, - SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) { - printk(KERN_ERR "can't set codec DAI configuration\n"); - return ret; - } - - /* Set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, - SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) { - printk(KERN_ERR "can't set cpu DAI configuration\n"); - return ret; - } - /* Set the codec system clock for DAC and ADC */ ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000, SND_SOC_CLOCK_IN); @@ -82,49 +61,6 @@ static struct snd_soc_ops zoom2_ops = { .hw_params = zoom2_hw_params, }; -static int zoom2_hw_voice_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - int ret; - - /* Set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, - SND_SOC_DAIFMT_DSP_A | - SND_SOC_DAIFMT_IB_NF | - SND_SOC_DAIFMT_CBM_CFM); - if (ret) { - printk(KERN_ERR "can't set codec DAI configuration\n"); - return ret; - } - - /* Set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, - SND_SOC_DAIFMT_DSP_A | - SND_SOC_DAIFMT_IB_NF | - SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) { - printk(KERN_ERR "can't set cpu DAI configuration\n"); - return ret; - } - - /* Set the codec system clock for DAC and ADC */ - ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000, - SND_SOC_CLOCK_IN); - if (ret < 0) { - printk(KERN_ERR "can't set codec system clock\n"); - return ret; - } - - return 0; -} - -static struct snd_soc_ops zoom2_voice_ops = { - .hw_params = zoom2_hw_voice_params, -}; - /* Zoom2 machine DAPM */ static const struct snd_soc_dapm_widget zoom2_twl4030_dapm_widgets[] = { SND_SOC_DAPM_MIC("Ext Mic", NULL), @@ -162,23 +98,6 @@ static int zoom2_twl4030_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; struct snd_soc_dapm_context *dapm = &codec->dapm; - int ret; - - /* Add Zoom2 specific widgets */ - ret = snd_soc_dapm_new_controls(dapm, zoom2_twl4030_dapm_widgets, - ARRAY_SIZE(zoom2_twl4030_dapm_widgets)); - if (ret) - return ret; - - /* Set up Zoom2 specific audio path audio_map */ - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - - /* Zoom2 connected pins */ - snd_soc_dapm_enable_pin(dapm, "Ext Mic"); - snd_soc_dapm_enable_pin(dapm, "Ext Spk"); - snd_soc_dapm_enable_pin(dapm, "Headset Mic"); - snd_soc_dapm_enable_pin(dapm, "Headset Stereophone"); - snd_soc_dapm_enable_pin(dapm, "Aux In"); /* TWL4030 not connected pins */ snd_soc_dapm_nc_pin(dapm, "CARKITMIC"); @@ -190,9 +109,7 @@ static int zoom2_twl4030_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_nc_pin(dapm, "CARKITL"); snd_soc_dapm_nc_pin(dapm, "CARKITR"); - ret = snd_soc_dapm_sync(dapm); - - return ret; + return 0; } static int zoom2_twl4030_voice_init(struct snd_soc_pcm_runtime *rtd) @@ -217,6 +134,8 @@ static struct snd_soc_dai_link zoom2_dai[] = { .codec_dai_name = "twl4030-hifi", .platform_name = "omap-pcm-audio", .codec_name = "twl4030-codec", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, .init = zoom2_twl4030_init, .ops = &zoom2_ops, }, @@ -227,8 +146,10 @@ static struct snd_soc_dai_link zoom2_dai[] = { .codec_dai_name = "twl4030-voice", .platform_name = "omap-pcm-audio", .codec_name = "twl4030-codec", + .dai_fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_NF | + SND_SOC_DAIFMT_CBM_CFM, .init = zoom2_twl4030_voice_init, - .ops = &zoom2_voice_ops, + .ops = &zoom2_ops, }, }; @@ -237,6 +158,11 @@ static struct snd_soc_card snd_soc_zoom2 = { .name = "Zoom2", .dai_link = zoom2_dai, .num_links = ARRAY_SIZE(zoom2_dai), + + .dapm_widgets = zoom2_twl4030_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(zoom2_twl4030_dapm_widgets), + .dapm_routes = audio_map, + .num_dapm_routes = ARRAY_SIZE(audio_map), }; static struct platform_device *zoom2_snd_device; diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 33ebc46b45b5..ffd2242e305f 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -121,6 +121,7 @@ config SND_PXA2XX_SOC_PALM27X config SND_SOC_SAARB tristate "SoC Audio support for Marvell Saarb" depends on SND_PXA2XX_SOC && MACH_SAARB + select MFD_88PM860X select SND_PXA_SOC_SSP select SND_SOC_88PM860X help @@ -130,6 +131,7 @@ config SND_SOC_SAARB config SND_SOC_TAVOREVB3 tristate "SoC Audio support for Marvell Tavor EVB3" depends on SND_PXA2XX_SOC && MACH_TAVOREVB3 + select MFD_88PM860X select SND_PXA_SOC_SSP select SND_SOC_88PM860X help diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index 28757fb9df31..b0e2fb720910 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -299,7 +299,6 @@ static int corgi_wm8731_init(struct snd_soc_pcm_runtime *rtd) /* Set up corgi specific audio path audio_map */ snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_sync(dapm); return 0; } diff --git a/sound/soc/pxa/e740_wm9705.c b/sound/soc/pxa/e740_wm9705.c index dc65650a6fa1..35ed7eb8cff2 100644 --- a/sound/soc/pxa/e740_wm9705.c +++ b/sound/soc/pxa/e740_wm9705.c @@ -108,8 +108,6 @@ static int e740_ac97_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_sync(dapm); - return 0; } diff --git a/sound/soc/pxa/e750_wm9705.c b/sound/soc/pxa/e750_wm9705.c index 51897fcd911b..ce5f056009a7 100644 --- a/sound/soc/pxa/e750_wm9705.c +++ b/sound/soc/pxa/e750_wm9705.c @@ -90,8 +90,6 @@ static int e750_ac97_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_sync(dapm); - return 0; } diff --git a/sound/soc/pxa/e800_wm9712.c b/sound/soc/pxa/e800_wm9712.c index 053ed208e59f..6a8f38b6c379 100644 --- a/sound/soc/pxa/e800_wm9712.c +++ b/sound/soc/pxa/e800_wm9712.c @@ -80,7 +80,6 @@ static int e800_ac97_init(struct snd_soc_pcm_runtime *rtd) ARRAY_SIZE(e800_dapm_widgets)); snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_sync(dapm); return 0; } diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c index 67dcc36cd621..e79f516c400e 100644 --- a/sound/soc/pxa/magician.c +++ b/sound/soc/pxa/magician.c @@ -92,11 +92,10 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - unsigned int acps, acds, width, rate; + unsigned int acps, acds, width; unsigned int div4 = PXA_SSP_CLK_SCDB_4; int ret = 0; - rate = params_rate(params); width = snd_pcm_format_physical_width(params_format(params)); /* @@ -424,7 +423,6 @@ static int magician_uda1380_init(struct snd_soc_pcm_runtime *rtd) /* Set up magician specific audio path interconnects */ snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_sync(dapm); return 0; } diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c index 38ca6759907e..0b8d1ee738a4 100644 --- a/sound/soc/pxa/mioa701_wm9713.c +++ b/sound/soc/pxa/mioa701_wm9713.c @@ -151,7 +151,6 @@ static int mioa701_wm9713_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_enable_pin(dapm, "Front Mic"); snd_soc_dapm_enable_pin(dapm, "GSM Line In"); snd_soc_dapm_enable_pin(dapm, "GSM Line Out"); - snd_soc_dapm_sync(dapm); return 0; } diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c index 504e4004f004..7edc1fb71fae 100644 --- a/sound/soc/pxa/palm27x.c +++ b/sound/soc/pxa/palm27x.c @@ -107,10 +107,6 @@ static int palm27x_ac97_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_nc_pin(dapm, "PHONE"); snd_soc_dapm_nc_pin(dapm, "MIC2"); - err = snd_soc_dapm_sync(dapm); - if (err) - return err; - /* Jack detection API stuff */ err = snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE, &hs_jack); diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c index da3ae4316cf2..4c29bc1f9cfe 100644 --- a/sound/soc/pxa/poodle.c +++ b/sound/soc/pxa/poodle.c @@ -265,7 +265,6 @@ static int poodle_wm8731_init(struct snd_soc_pcm_runtime *rtd) /* Set up poodle specific audio path audio_map */ snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_sync(dapm); return 0; } diff --git a/sound/soc/pxa/raumfeld.c b/sound/soc/pxa/raumfeld.c index 1a591f1ebfbd..b899a3bc8f42 100644 --- a/sound/soc/pxa/raumfeld.c +++ b/sound/soc/pxa/raumfeld.c @@ -306,8 +306,10 @@ static int __init raumfeld_audio_init(void) &snd_soc_raumfeld_connector); ret = platform_device_add(raumfeld_audio_device); - if (ret < 0) + if (ret < 0) { + platform_device_put(raumfeld_audio_device); return ret; + } raumfeld_enable_audio(true); return 0; diff --git a/sound/soc/pxa/saarb.c b/sound/soc/pxa/saarb.c index 9595189fc681..d9467a2c6de0 100644 --- a/sound/soc/pxa/saarb.c +++ b/sound/soc/pxa/saarb.c @@ -146,10 +146,6 @@ static int saarb_pm860x_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_disable_pin(dapm, "Headset Mic 2"); snd_soc_dapm_disable_pin(dapm, "Headset Stereophone"); - ret = snd_soc_dapm_sync(dapm); - if (ret) - return ret; - /* Headset jack detection */ snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE | SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2, diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index b253d864868a..c2d6ff9b1588 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -301,7 +301,6 @@ static int spitz_wm8750_init(struct snd_soc_pcm_runtime *rtd) /* Set up spitz specific audio paths */ snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_sync(dapm); return 0; } @@ -312,7 +311,7 @@ static struct snd_soc_dai_link spitz_dai = { .cpu_dai_name = "pxa2xx-i2s", .codec_dai_name = "wm8750-hifi", .platform_name = "pxa-pcm-audio", - .codec_name = "wm8750-codec.0-001b", + .codec_name = "wm8750.0-001b", .init = spitz_wm8750_init, .ops = &spitz_ops, }; diff --git a/sound/soc/pxa/tavorevb3.c b/sound/soc/pxa/tavorevb3.c index f881f65ec172..eeec892e0e04 100644 --- a/sound/soc/pxa/tavorevb3.c +++ b/sound/soc/pxa/tavorevb3.c @@ -146,10 +146,6 @@ static int evb3_pm860x_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_disable_pin(dapm, "Headset Mic 2"); snd_soc_dapm_disable_pin(dapm, "Headset Stereophone"); - ret = snd_soc_dapm_sync(dapm); - if (ret) - return ret; - /* Headset jack detection */ snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE | SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2, diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c index 9a2351366957..620fc69ae632 100644 --- a/sound/soc/pxa/tosa.c +++ b/sound/soc/pxa/tosa.c @@ -211,7 +211,6 @@ static int tosa_ac97_init(struct snd_soc_pcm_runtime *rtd) /* set up tosa specific audio path audio_map */ snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_sync(dapm); return 0; } diff --git a/sound/soc/pxa/z2.c b/sound/soc/pxa/z2.c index d69d9fc32233..b311ffe04b71 100644 --- a/sound/soc/pxa/z2.c +++ b/sound/soc/pxa/z2.c @@ -161,10 +161,6 @@ static int z2_wm8750_init(struct snd_soc_pcm_runtime *rtd) /* Set up z2 specific audio paths */ snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - ret = snd_soc_dapm_sync(dapm); - if (ret) - goto err; - /* Jack detection API stuff */ ret = snd_soc_jack_new(codec, "Headset Jack", SND_JACK_HEADSET, &hs_jack); @@ -198,7 +194,7 @@ static struct snd_soc_dai_link z2_dai = { .cpu_dai_name = "pxa2xx-i2s", .codec_dai_name = "wm8750-hifi", .platform_name = "pxa-pcm-audio", - .codec_name = "wm8750-codec.0-001b", + .codec_name = "wm8750.0-001b", .init = z2_wm8750_init, .ops = &z2_ops, }; diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c index 2b8350b52232..580aae38e502 100644 --- a/sound/soc/pxa/zylonite.c +++ b/sound/soc/pxa/zylonite.c @@ -87,7 +87,6 @@ static int zylonite_wm9713_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_enable_pin(dapm, "Headphone"); snd_soc_dapm_enable_pin(dapm, "Headset Earpiece"); - snd_soc_dapm_sync(dapm); return 0; } diff --git a/sound/soc/s6000/s6000-pcm.c b/sound/soc/s6000/s6000-pcm.c index 80c85fd64e1a..55efc2bdf0bd 100644 --- a/sound/soc/s6000/s6000-pcm.c +++ b/sound/soc/s6000/s6000-pcm.c @@ -446,7 +446,6 @@ static u64 s6000_pcm_dmamask = DMA_BIT_MASK(32); static int s6000_pcm_new(struct snd_soc_pcm_runtime *runtime) { struct snd_card *card = runtime->card->snd_card; - struct snd_soc_dai *dai = runtime->cpu_dai; struct snd_pcm *pcm = runtime->pcm; struct s6000_pcm_dma_params *params; int res; diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 65f980ef2870..53aaa69eda03 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -63,7 +63,9 @@ config SND_SOC_SAMSUNG_SMDK_WM8580 config SND_SOC_SAMSUNG_SMDK_WM8994 tristate "SoC I2S Audio support for WM8994 on SMDK" - depends on SND_SOC_SAMSUNG && (MACH_SMDKV310 || MACH_SMDKC210) + depends on SND_SOC_SAMSUNG && (MACH_SMDKV310 || MACH_SMDKC210 || MACH_SMDK4212) + depends on I2C=y && GENERIC_HARDIRQS + select MFD_WM8994 select SND_SOC_WM8994 select SND_SAMSUNG_I2S help @@ -150,7 +152,9 @@ config SND_SOC_SMARTQ config SND_SOC_GONI_AQUILA_WM8994 tristate "SoC I2S Audio support for AQUILA/GONI - WM8994" depends on SND_SOC_SAMSUNG && (MACH_GONI || MACH_AQUILA) + depends on I2C=y && GENERIC_HARDIRQS select SND_SAMSUNG_I2S + select MFD_WM8994 select SND_SOC_WM8994 help Say Y if you want to add support for SoC audio on goni or aquila @@ -158,7 +162,7 @@ config SND_SOC_GONI_AQUILA_WM8994 config SND_SOC_SAMSUNG_SMDK_SPDIF tristate "SoC S/PDIF Audio support for SMDK" - depends on SND_SOC_SAMSUNG && (MACH_SMDKC100 || MACH_SMDKC110 || MACH_SMDKV210 || MACH_SMDKV310) + depends on SND_SOC_SAMSUNG && (MACH_SMDKC100 || MACH_SMDKC110 || MACH_SMDKV210 || MACH_SMDKV310 || MACH_SMDK4212) select SND_SAMSUNG_SPDIF help Say Y if you want to add support for SoC S/PDIF audio on the SMDK. @@ -173,7 +177,9 @@ config SND_SOC_SMDK_WM8580_PCM config SND_SOC_SMDK_WM8994_PCM tristate "SoC PCM Audio support for WM8994 on SMDK" - depends on SND_SOC_SAMSUNG && (MACH_SMDKC210 || MACH_SMDKV310) + depends on SND_SOC_SAMSUNG && (MACH_SMDKC210 || MACH_SMDKV310 || MACH_SMDK4212) + depends on I2C=y && GENERIC_HARDIRQS + select MFD_WM8994 select SND_SOC_WM8994 select SND_SAMSUNG_PCM help diff --git a/sound/soc/samsung/ac97.c b/sound/soc/samsung/ac97.c index f97110e72e85..b5e922f469d5 100644 --- a/sound/soc/samsung/ac97.c +++ b/sound/soc/samsung/ac97.c @@ -444,7 +444,7 @@ static __devinit int s3c_ac97_probe(struct platform_device *pdev) } ret = request_irq(irq_res->start, s3c_ac97_irq, - IRQF_DISABLED, "AC97", NULL); + 0, "AC97", NULL); if (ret < 0) { dev_err(&pdev->dev, "ac97: interrupt request failed.\n"); goto err4; @@ -495,7 +495,7 @@ static __devexit int s3c_ac97_remove(struct platform_device *pdev) static struct platform_driver s3c_ac97_driver = { .probe = s3c_ac97_probe, - .remove = s3c_ac97_remove, + .remove = __devexit_p(s3c_ac97_remove), .driver = { .name = "samsung-ac97", .owner = THIS_MODULE, diff --git a/sound/soc/samsung/goni_wm8994.c b/sound/soc/samsung/goni_wm8994.c index eb6d72ed55a7..4a34f608e131 100644 --- a/sound/soc/samsung/goni_wm8994.c +++ b/sound/soc/samsung/goni_wm8994.c @@ -99,14 +99,6 @@ static int goni_wm8994_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_dapm_context *dapm = &codec->dapm; int ret; - /* add goni specific widgets */ - snd_soc_dapm_new_controls(dapm, goni_dapm_widgets, - ARRAY_SIZE(goni_dapm_widgets)); - - /* set up goni specific audio routes */ - snd_soc_dapm_add_routes(dapm, goni_dapm_routes, - ARRAY_SIZE(goni_dapm_routes)); - /* set endpoints to not connected */ snd_soc_dapm_nc_pin(dapm, "IN2LP:VXRN"); snd_soc_dapm_nc_pin(dapm, "IN2RP:VXRP"); @@ -120,8 +112,6 @@ static int goni_wm8994_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_nc_pin(dapm, "SPKOUTRP"); } - snd_soc_dapm_sync(dapm); - /* Headset jack detection */ ret = snd_soc_jack_new(codec, "Headset Jack", SND_JACK_HEADSET | SND_JACK_MECHANICAL | SND_JACK_AVOUT, @@ -255,6 +245,11 @@ static struct snd_soc_card goni = { .name = "goni", .dai_link = goni_dai, .num_links = ARRAY_SIZE(goni_dai), + + .dapm_widgets = goni_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(goni_dapm_widgets), + .dapm_routes = goni_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(goni_dapm_routes), }; static int __init goni_init(void) diff --git a/sound/soc/samsung/h1940_uda1380.c b/sound/soc/samsung/h1940_uda1380.c index c6c65892294e..f75a4b60cf38 100644 --- a/sound/soc/samsung/h1940_uda1380.c +++ b/sound/soc/samsung/h1940_uda1380.c @@ -182,24 +182,10 @@ static int h1940_uda1380_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_dapm_context *dapm = &codec->dapm; int err; - /* Add h1940 specific widgets */ - err = snd_soc_dapm_new_controls(dapm, uda1380_dapm_widgets, - ARRAY_SIZE(uda1380_dapm_widgets)); - if (err) - return err; - - /* Set up h1940 specific audio path audio_mapnects */ - err = snd_soc_dapm_add_routes(dapm, audio_map, - ARRAY_SIZE(audio_map)); - if (err) - return err; - snd_soc_dapm_enable_pin(dapm, "Headphone Jack"); snd_soc_dapm_enable_pin(dapm, "Speaker"); snd_soc_dapm_enable_pin(dapm, "Mic Jack"); - snd_soc_dapm_sync(dapm); - snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE, &hp_jack); @@ -230,6 +216,11 @@ static struct snd_soc_card h1940_asoc = { .name = "h1940", .dai_link = h1940_uda1380_dai, .num_links = ARRAY_SIZE(h1940_uda1380_dai), + + .dapm_widgets = uda1380_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(uda1380_dapm_widgets), + .dapm_routes = audio_map, + .num_dapm_routes = ARRAY_SIZE(audio_map), }; static int __init h1940_init(void) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index c086b78539ee..0c9ac20d2223 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1136,7 +1136,7 @@ static __devexit int samsung_i2s_remove(struct platform_device *pdev) static struct platform_driver samsung_i2s_driver = { .probe = samsung_i2s_probe, - .remove = samsung_i2s_remove, + .remove = __devexit_p(samsung_i2s_remove), .driver = { .name = "samsung-i2s", .owner = THIS_MODULE, diff --git a/sound/soc/samsung/jive_wm8750.c b/sound/soc/samsung/jive_wm8750.c index 14eb6ea69e7c..f5f7c6f822d5 100644 --- a/sound/soc/samsung/jive_wm8750.c +++ b/sound/soc/samsung/jive_wm8750.c @@ -110,18 +110,6 @@ static int jive_wm8750_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_nc_pin(dapm, "OUT3"); snd_soc_dapm_nc_pin(dapm, "MONO"); - /* Add jive specific widgets */ - err = snd_soc_dapm_new_controls(dapm, wm8750_dapm_widgets, - ARRAY_SIZE(wm8750_dapm_widgets)); - if (err) { - printk(KERN_ERR "%s: failed to add widgets (%d)\n", - __func__, err); - return err; - } - - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_sync(dapm); - return 0; } @@ -131,7 +119,7 @@ static struct snd_soc_dai_link jive_dai = { .cpu_dai_name = "s3c2412-i2s", .codec_dai_name = "wm8750-hifi", .platform_name = "samsung-audio", - .codec_name = "wm8750-codec.0-001a", + .codec_name = "wm8750.0-001a", .init = jive_wm8750_init, .ops = &jive_ops, }; @@ -141,6 +129,11 @@ static struct snd_soc_card snd_soc_machine_jive = { .name = "Jive", .dai_link = &jive_dai, .num_links = 1, + + .dapm_widgtets = wm8750_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm8750_dapm_widgets), + .dapm_routes = audio_map, + .num_dapm_routes = ARRAY_SIZE(audio_map), }; static struct platform_device *jive_snd_device; diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c index 16152ed08648..7207189cd211 100644 --- a/sound/soc/samsung/neo1973_wm8753.c +++ b/sound/soc/samsung/neo1973_wm8753.c @@ -367,8 +367,6 @@ static int neo1973_wm8753_init(struct snd_soc_pcm_runtime *rtd) return ret; } - snd_soc_dapm_sync(dapm); - return 0; } @@ -409,8 +407,6 @@ static int neo1973_lm4857_init(struct snd_soc_dapm_context *dapm) snd_soc_dapm_ignore_suspend(dapm, "Handset Spk"); snd_soc_dapm_ignore_suspend(dapm, "Headphone"); - snd_soc_dapm_sync(dapm); - return 0; } diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c index 9c7e8b48aed6..e55d7a5c4bdc 100644 --- a/sound/soc/samsung/pcm.c +++ b/sound/soc/samsung/pcm.c @@ -624,7 +624,7 @@ static __devexit int s3c_pcm_dev_remove(struct platform_device *pdev) static struct platform_driver s3c_pcm_driver = { .probe = s3c_pcm_dev_probe, - .remove = s3c_pcm_dev_remove, + .remove = __devexit_p(s3c_pcm_dev_remove), .driver = { .name = "samsung-pcm", .owner = THIS_MODULE, diff --git a/sound/soc/samsung/rx1950_uda1380.c b/sound/soc/samsung/rx1950_uda1380.c index bc8c1676459f..aea7f1b24e6b 100644 --- a/sound/soc/samsung/rx1950_uda1380.c +++ b/sound/soc/samsung/rx1950_uda1380.c @@ -90,12 +90,6 @@ static struct snd_soc_dai_link rx1950_uda1380_dai[] = { }, }; -static struct snd_soc_card rx1950_asoc = { - .name = "rx1950", - .dai_link = rx1950_uda1380_dai, - .num_links = ARRAY_SIZE(rx1950_uda1380_dai), -}; - /* rx1950 machine dapm widgets */ static const struct snd_soc_dapm_widget uda1380_dapm_widgets[] = { SND_SOC_DAPM_HP("Headphone Jack", NULL), @@ -117,6 +111,17 @@ static const struct snd_soc_dapm_route audio_map[] = { {"VINM", NULL, "Mic Jack"}, }; +static struct snd_soc_card rx1950_asoc = { + .name = "rx1950", + .dai_link = rx1950_uda1380_dai, + .num_links = ARRAY_SIZE(rx1950_uda1380_dai), + + .dapm_widgets = uda1380_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(uda1380_dapm_widgets), + .dapm_routes = audio_map, + .num_dapm_routes = ARRAY_SIZE(audio_map), +}; + static struct platform_device *s3c24xx_snd_device; static int rx1950_startup(struct snd_pcm_substream *substream) @@ -220,26 +225,10 @@ static int rx1950_uda1380_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_dapm_context *dapm = &codec->dapm; int err; - /* Add rx1950 specific widgets */ - err = snd_soc_dapm_new_controls(dapm, uda1380_dapm_widgets, - ARRAY_SIZE(uda1380_dapm_widgets)); - - if (err) - return err; - - /* Set up rx1950 specific audio path audio_mapnects */ - err = snd_soc_dapm_add_routes(dapm, audio_map, - ARRAY_SIZE(audio_map)); - - if (err) - return err; - snd_soc_dapm_enable_pin(dapm, "Headphone Jack"); snd_soc_dapm_enable_pin(dapm, "Speaker"); snd_soc_dapm_enable_pin(dapm, "Mic Jack"); - snd_soc_dapm_sync(dapm); - snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE, &hp_jack); diff --git a/sound/soc/samsung/s3c-i2s-v2.c b/sound/soc/samsung/s3c-i2s-v2.c index 52074a2b0696..7a73380b3560 100644 --- a/sound/soc/samsung/s3c-i2s-v2.c +++ b/sound/soc/samsung/s3c-i2s-v2.c @@ -16,6 +16,7 @@ * option) any later version. */ +#include <linux/module.h> #include <linux/delay.h> #include <linux/clk.h> #include <linux/io.h> diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c index 841ab14c1100..f26a8bfb2357 100644 --- a/sound/soc/samsung/s3c2412-i2s.c +++ b/sound/soc/samsung/s3c2412-i2s.c @@ -69,10 +69,10 @@ static int s3c2412_i2s_probe(struct snd_soc_dai *dai) s3c2412_i2s.dma_playback = &s3c2412_i2s_pcm_stereo_out; s3c2412_i2s.iis_cclk = clk_get(dai->dev, "i2sclk"); - if (s3c2412_i2s.iis_cclk == NULL) { + if (IS_ERR(s3c2412_i2s.iis_cclk)) { pr_err("failed to get i2sclk clock\n"); iounmap(s3c2412_i2s.regs); - return -ENODEV; + return PTR_ERR(s3c2412_i2s.iis_cclk); } /* Set MPLL as the source for IIS CLK */ @@ -176,7 +176,7 @@ static __devexit int s3c2412_iis_dev_remove(struct platform_device *pdev) static struct platform_driver s3c2412_iis_driver = { .probe = s3c2412_iis_dev_probe, - .remove = s3c2412_iis_dev_remove, + .remove = __devexit_p(s3c2412_iis_dev_remove), .driver = { .name = "s3c2412-iis", .owner = THIS_MODULE, diff --git a/sound/soc/samsung/s3c24xx-i2s.c b/sound/soc/samsung/s3c24xx-i2s.c index 63d8849d80bd..c08117e658db 100644 --- a/sound/soc/samsung/s3c24xx-i2s.c +++ b/sound/soc/samsung/s3c24xx-i2s.c @@ -383,10 +383,10 @@ static int s3c24xx_i2s_probe(struct snd_soc_dai *dai) return -ENXIO; s3c24xx_i2s.iis_clk = clk_get(dai->dev, "iis"); - if (s3c24xx_i2s.iis_clk == NULL) { + if (IS_ERR(s3c24xx_i2s.iis_clk)) { pr_err("failed to get iis_clock\n"); iounmap(s3c24xx_i2s.regs); - return -ENODEV; + return PTR_ERR(s3c24xx_i2s.iis_clk); } clk_enable(s3c24xx_i2s.iis_clk); @@ -481,7 +481,7 @@ static __devexit int s3c24xx_iis_dev_remove(struct platform_device *pdev) static struct platform_driver s3c24xx_iis_driver = { .probe = s3c24xx_iis_dev_probe, - .remove = s3c24xx_iis_dev_remove, + .remove = __devexit_p(s3c24xx_iis_dev_remove), .driver = { .name = "s3c24xx-iis", .owner = THIS_MODULE, diff --git a/sound/soc/samsung/s3c24xx_simtec.c b/sound/soc/samsung/s3c24xx_simtec.c index 349566f0686b..c8d525bf6122 100644 --- a/sound/soc/samsung/s3c24xx_simtec.c +++ b/sound/soc/samsung/s3c24xx_simtec.c @@ -300,7 +300,7 @@ static void detach_gpio_amp(struct s3c24xx_audio_simtec_pdata *pd) } #ifdef CONFIG_PM -int simtec_audio_resume(struct device *dev) +static int simtec_audio_resume(struct device *dev) { simtec_call_startup(pdata); return 0; diff --git a/sound/soc/samsung/s3c24xx_simtec_hermes.c b/sound/soc/samsung/s3c24xx_simtec_hermes.c index ce6aef604179..6bc5a36af1d9 100644 --- a/sound/soc/samsung/s3c24xx_simtec_hermes.c +++ b/sound/soc/samsung/s3c24xx_simtec_hermes.c @@ -65,18 +65,12 @@ static int simtec_hermes_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_codec *codec = rtd->codec; struct snd_soc_dapm_context *dapm = &codec->dapm; - snd_soc_dapm_new_controls(dapm, dapm_widgets, - ARRAY_SIZE(dapm_widgets)); - - snd_soc_dapm_add_routes(dapm, base_map, ARRAY_SIZE(base_map)); - snd_soc_dapm_enable_pin(dapm, "Headphone Jack"); snd_soc_dapm_enable_pin(dapm, "Line In"); snd_soc_dapm_enable_pin(dapm, "Line Out"); snd_soc_dapm_enable_pin(dapm, "Mic Jack"); simtec_audio_init(rtd); - snd_soc_dapm_sync(dapm); return 0; } @@ -96,6 +90,11 @@ static struct snd_soc_card snd_soc_machine_simtec_aic33 = { .name = "Simtec-Hermes", .dai_link = &simtec_dai_aic33, .num_links = 1, + + .dapm_widgets = dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(dapm_widgets), + .dapm_routes = base_map, + .num_dapm_routes = ARRAY_SIZE(base_map), }; static int __devinit simtec_audio_hermes_probe(struct platform_device *pd) diff --git a/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c b/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c index a7ef7db54687..7bdda7674008 100644 --- a/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c +++ b/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c @@ -54,18 +54,12 @@ static int simtec_tlv320aic23_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_codec *codec = rtd->codec; struct snd_soc_dapm_context *dapm = &codec->dapm; - snd_soc_dapm_new_controls(dapm, dapm_widgets, - ARRAY_SIZE(dapm_widgets)); - - snd_soc_dapm_add_routes(dapm, base_map, ARRAY_SIZE(base_map)); - snd_soc_dapm_enable_pin(dapm, "Headphone Jack"); snd_soc_dapm_enable_pin(dapm, "Line In"); snd_soc_dapm_enable_pin(dapm, "Line Out"); snd_soc_dapm_enable_pin(dapm, "Mic Jack"); simtec_audio_init(rtd); - snd_soc_dapm_sync(dapm); return 0; } @@ -85,6 +79,11 @@ static struct snd_soc_card snd_soc_machine_simtec_aic23 = { .name = "Simtec", .dai_link = &simtec_dai_aic23, .num_links = 1, + + .dapm_widgets = dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(dapm_widgets), + .dapm_routes = base_map, + .num_dapm_routes = ARRAY_SIZE(base_map), }; static int __devinit simtec_audio_tlv320aic23_probe(struct platform_device *pd) diff --git a/sound/soc/samsung/s3c24xx_uda134x.c b/sound/soc/samsung/s3c24xx_uda134x.c index dc9d551f6788..65c1cfd47d8a 100644 --- a/sound/soc/samsung/s3c24xx_uda134x.c +++ b/sound/soc/samsung/s3c24xx_uda134x.c @@ -66,17 +66,17 @@ static int s3c24xx_uda134x_startup(struct snd_pcm_substream *substream) pr_debug("%s %d\n", __func__, clk_users); if (clk_users == 0) { xtal = clk_get(&s3c24xx_uda134x_snd_device->dev, "xtal"); - if (!xtal) { + if (IS_ERR(xtal)) { printk(KERN_ERR "%s cannot get xtal\n", __func__); - ret = -EBUSY; + ret = PTR_ERR(xtal); } else { pclk = clk_get(&s3c24xx_uda134x_snd_device->dev, "pclk"); - if (!pclk) { + if (IS_ERR(pclk)) { printk(KERN_ERR "%s cannot get pclk\n", __func__); clk_put(xtal); - ret = -EBUSY; + ret = PTR_ERR(pclk); } } if (!ret) { diff --git a/sound/soc/samsung/smartq_wm8987.c b/sound/soc/samsung/smartq_wm8987.c index 0a2c4f223038..6ac6bc2bcc4e 100644 --- a/sound/soc/samsung/smartq_wm8987.c +++ b/sound/soc/samsung/smartq_wm8987.c @@ -153,20 +153,6 @@ static int smartq_wm8987_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_dapm_context *dapm = &codec->dapm; int err = 0; - /* Add SmartQ specific widgets */ - snd_soc_dapm_new_controls(dapm, wm8987_dapm_widgets, - ARRAY_SIZE(wm8987_dapm_widgets)); - - /* add SmartQ specific controls */ - err = snd_soc_add_controls(codec, wm8987_smartq_controls, - ARRAY_SIZE(wm8987_smartq_controls)); - - if (err < 0) - return err; - - /* setup SmartQ specific audio path */ - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - /* set endpoints to not connected */ snd_soc_dapm_nc_pin(dapm, "LINPUT1"); snd_soc_dapm_nc_pin(dapm, "RINPUT1"); @@ -178,10 +164,6 @@ static int smartq_wm8987_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_enable_pin(dapm, "Internal Mic"); snd_soc_dapm_disable_pin(dapm, "Headphone Jack"); - err = snd_soc_dapm_sync(dapm); - if (err) - return err; - /* Headphone jack detection */ err = snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE, &smartq_jack); @@ -207,7 +189,7 @@ static struct snd_soc_dai_link smartq_dai[] = { .cpu_dai_name = "samsung-i2s.0", .codec_dai_name = "wm8750-hifi", .platform_name = "samsung-audio", - .codec_name = "wm8750-codec.0-0x1a", + .codec_name = "wm8750.0-0x1a", .init = smartq_wm8987_init, .ops = &smartq_hifi_ops, }, @@ -217,6 +199,13 @@ static struct snd_soc_card snd_soc_smartq = { .name = "SmartQ", .dai_link = smartq_dai, .num_links = ARRAY_SIZE(smartq_dai), + + .dapm_widgets = wm8987_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm8987_dapm_widgets), + .dapm_routes = audio_map, + .num_dapm_routes = ARRAY_SIZE(audio_map), + .controls = wm8987_smartq_controls, + .num_controls = ARRAY_SIZE(wm8987_smartq_controls), }; static struct platform_device *smartq_snd_device; diff --git a/sound/soc/samsung/smdk_wm8580.c b/sound/soc/samsung/smdk_wm8580.c index 3d26f6607aa4..8f92ffceb5ca 100644 --- a/sound/soc/samsung/smdk_wm8580.c +++ b/sound/soc/samsung/smdk_wm8580.c @@ -119,30 +119,24 @@ static struct snd_soc_ops smdk_ops = { }; /* SMDK Playback widgets */ -static const struct snd_soc_dapm_widget wm8580_dapm_widgets_pbk[] = { +static const struct snd_soc_dapm_widget smdk_wm8580_dapm_widgets[] = { SND_SOC_DAPM_HP("Front", NULL), SND_SOC_DAPM_HP("Center+Sub", NULL), SND_SOC_DAPM_HP("Rear", NULL), -}; -/* SMDK Capture widgets */ -static const struct snd_soc_dapm_widget wm8580_dapm_widgets_cpt[] = { SND_SOC_DAPM_MIC("MicIn", NULL), SND_SOC_DAPM_LINE("LineIn", NULL), }; /* SMDK-PAIFTX connections */ -static const struct snd_soc_dapm_route audio_map_tx[] = { +static const struct snd_soc_dapm_route smdk_wm8580_audio_map[] = { /* MicIn feeds AINL */ {"AINL", NULL, "MicIn"}, /* LineIn feeds AINL/R */ {"AINL", NULL, "LineIn"}, {"AINR", NULL, "LineIn"}, -}; -/* SMDK-PAIFRX connections */ -static const struct snd_soc_dapm_route audio_map_rx[] = { /* Front Left/Right are fed VOUT1L/R */ {"Front", NULL, "VOUT1L"}, {"Front", NULL, "VOUT1R"}, @@ -161,39 +155,11 @@ static int smdk_wm8580_init_paiftx(struct snd_soc_pcm_runtime *rtd) struct snd_soc_codec *codec = rtd->codec; struct snd_soc_dapm_context *dapm = &codec->dapm; - /* Add smdk specific Capture widgets */ - snd_soc_dapm_new_controls(dapm, wm8580_dapm_widgets_cpt, - ARRAY_SIZE(wm8580_dapm_widgets_cpt)); - - /* Set up PAIFTX audio path */ - snd_soc_dapm_add_routes(dapm, audio_map_tx, ARRAY_SIZE(audio_map_tx)); - /* Enabling the microphone requires the fitting of a 0R * resistor to connect the line from the microphone jack. */ snd_soc_dapm_disable_pin(dapm, "MicIn"); - /* signal a DAPM event */ - snd_soc_dapm_sync(dapm); - - return 0; -} - -static int smdk_wm8580_init_paifrx(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_codec *codec = rtd->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; - - /* Add smdk specific Playback widgets */ - snd_soc_dapm_new_controls(dapm, wm8580_dapm_widgets_pbk, - ARRAY_SIZE(wm8580_dapm_widgets_pbk)); - - /* Set up PAIFRX audio path */ - snd_soc_dapm_add_routes(dapm, audio_map_rx, ARRAY_SIZE(audio_map_rx)); - - /* signal a DAPM event */ - snd_soc_dapm_sync(dapm); - return 0; } @@ -210,8 +176,7 @@ static struct snd_soc_dai_link smdk_dai[] = { .cpu_dai_name = "samsung-i2s.0", .codec_dai_name = "wm8580-hifi-playback", .platform_name = "samsung-audio", - .codec_name = "wm8580-codec.0-001b", - .init = smdk_wm8580_init_paifrx, + .codec_name = "wm8580.0-001b", .ops = &smdk_ops, }, [PRI_CAPTURE] = { /* Primary Capture i/f */ @@ -220,7 +185,7 @@ static struct snd_soc_dai_link smdk_dai[] = { .cpu_dai_name = "samsung-i2s.0", .codec_dai_name = "wm8580-hifi-capture", .platform_name = "samsung-audio", - .codec_name = "wm8580-codec.0-001b", + .codec_name = "wm8580.0-001b", .init = smdk_wm8580_init_paiftx, .ops = &smdk_ops, }, @@ -230,8 +195,7 @@ static struct snd_soc_dai_link smdk_dai[] = { .cpu_dai_name = "samsung-i2s.x", .codec_dai_name = "wm8580-hifi-playback", .platform_name = "samsung-audio", - .codec_name = "wm8580-codec.0-001b", - .init = smdk_wm8580_init_paifrx, + .codec_name = "wm8580.0-001b", .ops = &smdk_ops, }, }; @@ -240,6 +204,11 @@ static struct snd_soc_card smdk = { .name = "SMDK-I2S", .dai_link = smdk_dai, .num_links = 2, + + .dapm_widgets = smdk_wm8580_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(smdk_wm8580_dapm_widgets), + .dapm_routes = smdk_wm8580_audio_map, + .num_dapm_routes = ARRAY_SIZE(smdk_wm8580_audio_map), }; static struct platform_device *smdk_snd_device; diff --git a/sound/soc/samsung/smdk_wm8580pcm.c b/sound/soc/samsung/smdk_wm8580pcm.c index 0d12092df164..4b9c73477ce0 100644 --- a/sound/soc/samsung/smdk_wm8580pcm.c +++ b/sound/soc/samsung/smdk_wm8580pcm.c @@ -127,7 +127,7 @@ static struct snd_soc_dai_link smdk_dai[] = { .cpu_dai_name = "samsung-pcm.0", .codec_dai_name = "wm8580-hifi-playback", .platform_name = "samsung-audio", - .codec_name = "wm8580-codec.0-001b", + .codec_name = "wm8580.0-001b", .ops = &smdk_wm8580_pcm_ops, }, { .name = "WM8580 PAIF PCM TX", @@ -135,7 +135,7 @@ static struct snd_soc_dai_link smdk_dai[] = { .cpu_dai_name = "samsung-pcm.0", .codec_dai_name = "wm8580-hifi-capture", .platform_name = "samsung-audio", - .codec_name = "wm8580-codec.0-001b", + .codec_name = "wm8580.0-001b", .ops = &smdk_wm8580_pcm_ops, }, }; diff --git a/sound/soc/samsung/smdk_wm8994.c b/sound/soc/samsung/smdk_wm8994.c index 45fbe2b3727f..f75e43997d5b 100644 --- a/sound/soc/samsung/smdk_wm8994.c +++ b/sound/soc/samsung/smdk_wm8994.c @@ -117,8 +117,6 @@ static int smdk_wm8994_init_paiftx(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_nc_pin(dapm, "IN1RP"); snd_soc_dapm_nc_pin(dapm, "IN2RP:VXRP"); - snd_soc_dapm_sync(dapm); - return 0; } diff --git a/sound/soc/samsung/spdif.c b/sound/soc/samsung/spdif.c index 28c491dacf7a..3122f3154bfa 100644 --- a/sound/soc/samsung/spdif.c +++ b/sound/soc/samsung/spdif.c @@ -340,7 +340,7 @@ static struct snd_soc_dai_ops spdif_dai_ops = { .shutdown = spdif_shutdown, }; -struct snd_soc_dai_driver samsung_spdif_dai = { +static struct snd_soc_dai_driver samsung_spdif_dai = { .name = "samsung-spdif", .playback = { .stream_name = "S/PDIF Playback", @@ -475,7 +475,7 @@ static __devexit int spdif_remove(struct platform_device *pdev) static struct platform_driver samsung_spdif_driver = { .probe = spdif_probe, - .remove = spdif_remove, + .remove = __devexit_p(spdif_remove), .driver = { .name = "samsung-spdif", .owner = THIS_MODULE, diff --git a/sound/soc/samsung/speyside.c b/sound/soc/samsung/speyside.c index 590e9274b062..b9e213f6cc06 100644 --- a/sound/soc/samsung/speyside.c +++ b/sound/soc/samsung/speyside.c @@ -125,10 +125,6 @@ static struct snd_soc_jack_pin speyside_headset_pins[] = { .pin = "Headset Mic", .mask = SND_JACK_MICROPHONE, }, - { - .pin = "Headphone", - .mask = SND_JACK_HEADPHONE, - }, }; /* Default the headphone selection to active high */ @@ -171,7 +167,8 @@ static int speyside_wm8996_init(struct snd_soc_pcm_runtime *rtd) gpio_direction_output(WM8996_HPSEL_GPIO, speyside_jack_polarity); ret = snd_soc_jack_new(codec, "Headset", - SND_JACK_HEADSET | SND_JACK_BTN_0, + SND_JACK_LINEOUT | SND_JACK_HEADSET | + SND_JACK_BTN_0, &speyside_headset); if (ret) return ret; @@ -227,7 +224,7 @@ static int speyside_wm9081_init(struct snd_soc_dapm_context *dapm) snd_soc_dapm_nc_pin(dapm, "LINEOUT"); /* At any time the WM9081 is active it will have this clock */ - return snd_soc_codec_set_sysclk(dapm->codec, WM9081_SYSCLK_MCLK, + return snd_soc_codec_set_sysclk(dapm->codec, WM9081_SYSCLK_MCLK, 0, 48000 * 256, 0); } @@ -252,6 +249,7 @@ static const struct snd_kcontrol_new controls[] = { SOC_DAPM_PIN_SWITCH("Main AMIC"), SOC_DAPM_PIN_SWITCH("WM1250 Input"), SOC_DAPM_PIN_SWITCH("WM1250 Output"), + SOC_DAPM_PIN_SWITCH("Headphone"), }; static struct snd_soc_dapm_widget widgets[] = { diff --git a/sound/soc/samsung/speyside_wm8962.c b/sound/soc/samsung/speyside_wm8962.c index 72535f2daaf2..8a082044436e 100644 --- a/sound/soc/samsung/speyside_wm8962.c +++ b/sound/soc/samsung/speyside_wm8962.c @@ -16,6 +16,8 @@ #include "../codecs/wm8962.h" +static int sample_rate = 44100; + static int speyside_wm8962_set_bias_level(struct snd_soc_card *card, struct snd_soc_dapm_context *dapm, enum snd_soc_bias_level level) @@ -31,13 +33,13 @@ static int speyside_wm8962_set_bias_level(struct snd_soc_card *card, if (dapm->bias_level == SND_SOC_BIAS_STANDBY) { ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, WM8962_FLL_MCLK, 32768, - 44100 * 256); + sample_rate * 512); if (ret < 0) pr_err("Failed to start FLL: %d\n", ret); ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_FLL, - 44100 * 256, + sample_rate * 512, SND_SOC_CLOCK_IN); if (ret < 0) { pr_err("Failed to set SYSCLK: %d\n", ret); @@ -92,22 +94,7 @@ static int speyside_wm8962_set_bias_level_post(struct snd_soc_card *card, static int speyside_wm8962_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - int ret; - - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S - | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S - | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; + sample_rate = params_rate(params); return 0; } @@ -124,12 +111,15 @@ static struct snd_soc_dai_link speyside_wm8962_dai[] = { .codec_dai_name = "wm8962", .platform_name = "samsung-audio", .codec_name = "wm8962.1-001a", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, .ops = &speyside_wm8962_ops, }, }; static const struct snd_kcontrol_new controls[] = { SOC_DAPM_PIN_SWITCH("Main Speaker"), + SOC_DAPM_PIN_SWITCH("DMIC"), }; static struct snd_soc_dapm_widget widgets[] = { @@ -137,6 +127,7 @@ static struct snd_soc_dapm_widget widgets[] = { SND_SOC_DAPM_MIC("Headset Mic", NULL), SND_SOC_DAPM_MIC("DMIC", NULL), + SND_SOC_DAPM_MIC("AMIC", NULL), SND_SOC_DAPM_SPK("Main Speaker", NULL), }; @@ -148,12 +139,16 @@ static struct snd_soc_dapm_route audio_paths[] = { { "Main Speaker", NULL, "SPKOUTL" }, { "Main Speaker", NULL, "SPKOUTR" }, - { "MICBIAS", NULL, "Headset Mic" }, - { "IN4L", NULL, "MICBIAS" }, - { "IN4R", NULL, "MICBIAS" }, + { "Headset Mic", NULL, "MICBIAS" }, + { "IN4L", NULL, "Headset Mic" }, + { "IN4R", NULL, "Headset Mic" }, + + { "AMIC", NULL, "MICBIAS" }, + { "IN1L", NULL, "AMIC" }, + { "IN1R", NULL, "AMIC" }, - { "MICBIAS", NULL, "DMIC" }, - { "DMICDAT", NULL, "MICBIAS" }, + { "DMIC", NULL, "MICBIAS" }, + { "DMICDAT", NULL, "DMIC" }, }; static struct snd_soc_jack speyside_wm8962_headset; diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 8e112ccffb13..a32fd16ad668 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -210,7 +210,7 @@ struct fsi_master { * basic read write function */ -static void __fsi_reg_write(u32 reg, u32 data) +static void __fsi_reg_write(u32 __iomem *reg, u32 data) { /* valid data area is 24bit */ data &= 0x00ffffff; @@ -218,12 +218,12 @@ static void __fsi_reg_write(u32 reg, u32 data) __raw_writel(data, reg); } -static u32 __fsi_reg_read(u32 reg) +static u32 __fsi_reg_read(u32 __iomem *reg) { return __raw_readl(reg); } -static void __fsi_reg_mask_set(u32 reg, u32 mask, u32 data) +static void __fsi_reg_mask_set(u32 __iomem *reg, u32 mask, u32 data) { u32 val = __fsi_reg_read(reg); @@ -250,7 +250,7 @@ static u32 _fsi_master_read(struct fsi_master *master, u32 reg) unsigned long flags; spin_lock_irqsave(&master->lock, flags); - ret = __fsi_reg_read((u32)(master->base + reg)); + ret = __fsi_reg_read(master->base + reg); spin_unlock_irqrestore(&master->lock, flags); return ret; @@ -264,7 +264,7 @@ static void _fsi_master_mask_set(struct fsi_master *master, unsigned long flags; spin_lock_irqsave(&master->lock, flags); - __fsi_reg_mask_set((u32)(master->base + reg), mask, data); + __fsi_reg_mask_set(master->base + reg, mask, data); spin_unlock_irqrestore(&master->lock, flags); } @@ -1285,7 +1285,7 @@ static int fsi_probe(struct platform_device *pdev) pm_runtime_enable(&pdev->dev); dev_set_drvdata(&pdev->dev, master); - ret = request_irq(irq, &fsi_interrupt, IRQF_DISABLED, + ret = request_irq(irq, &fsi_interrupt, 0, id_entry->name, master); if (ret) { dev_err(&pdev->dev, "irq request err\n"); diff --git a/sound/soc/sh/sh7760-ac97.c b/sound/soc/sh/sh7760-ac97.c index 917d3ceadc9d..c62ae689c4a1 100644 --- a/sound/soc/sh/sh7760-ac97.c +++ b/sound/soc/sh/sh7760-ac97.c @@ -20,12 +20,6 @@ extern struct snd_soc_dai_driver sh4_hac_dai[2]; extern struct snd_soc_platform_driver sh7760_soc_platform; -static int machine_init(struct snd_soc_pcm_runtime *rtd) -{ - snd_soc_dapm_sync(&rtd->codec->dapm); - return 0; -} - static struct snd_soc_dai_link sh7760_ac97_dai = { .name = "AC97", .stream_name = "AC97 HiFi", @@ -33,7 +27,6 @@ static struct snd_soc_dai_link sh7760_ac97_dai = { .codec_dai_name = "ac97-hifi", .platform_name = "sh7760-pcm-audio", .codec_name = "ac97-codec", - .init = machine_init, .ops = NULL, }; diff --git a/sound/soc/sh/ssi.c b/sound/soc/sh/ssi.c index 05192d97b377..e0c621c0553b 100644 --- a/sound/soc/sh/ssi.c +++ b/sound/soc/sh/ssi.c @@ -342,7 +342,7 @@ static struct snd_soc_dai_ops ssi_dai_ops = { .set_fmt = ssi_set_fmt, }; -struct snd_soc_dai_driver sh4_ssi_dai[] = { +static struct snd_soc_dai_driver sh4_ssi_dai[] = { { .name = "ssi-dai.0", .playback = { diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 20b7f3b003a3..143c705ac27b 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -548,9 +548,6 @@ static inline int snd_soc_lzo_get_blkpos(struct snd_soc_codec *codec, static inline int snd_soc_lzo_get_blksize(struct snd_soc_codec *codec) { - const struct snd_soc_codec_driver *codec_drv; - - codec_drv = codec->driver; return DIV_ROUND_UP(codec->reg_size, snd_soc_lzo_block_count()); } @@ -868,10 +865,6 @@ static int snd_soc_flat_cache_exit(struct snd_soc_codec *codec) static int snd_soc_flat_cache_init(struct snd_soc_codec *codec) { - const struct snd_soc_codec_driver *codec_drv; - - codec_drv = codec->driver; - if (codec->reg_def_copy) codec->reg_cache = kmemdup(codec->reg_def_copy, codec->reg_size, GFP_KERNEL); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index ef69f5a02709..a5d3685a5d38 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -106,7 +106,7 @@ static int format_register_str(struct snd_soc_codec *codec, if (wordsize + regsize + 2 + 1 != len) return -EINVAL; - ret = snd_soc_read(codec , reg); + ret = snd_soc_read(codec, reg); if (ret < 0) { memset(regbuf, 'X', regsize); regbuf[regsize] = '\0'; @@ -144,7 +144,7 @@ static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf, step = codec->driver->reg_cache_step; for (i = 0; i < codec->driver->reg_cache_size; i += step) { - if (codec->readable_register && !codec->readable_register(codec, i)) + if (!snd_soc_codec_readable_register(codec, i)) continue; if (codec->driver->display_register) { count += codec->driver->display_register(codec, buf + count, @@ -245,7 +245,6 @@ static ssize_t codec_reg_write_file(struct file *file, size_t buf_size; char *start = buf; unsigned long reg, value; - int step = 1; struct snd_soc_codec *codec = file->private_data; buf_size = min(count, (sizeof(buf)-1)); @@ -253,9 +252,6 @@ static ssize_t codec_reg_write_file(struct file *file, return -EFAULT; buf[buf_size] = 0; - if (codec->driver->reg_cache_step) - step = codec->driver->reg_cache_step; - while (*start == ' ') start++; reg = simple_strtoul(start, &start, 16); @@ -957,6 +953,8 @@ static int soc_probe_codec(struct snd_soc_card *card, snd_soc_dapm_new_controls(&codec->dapm, driver->dapm_widgets, driver->num_dapm_widgets); + codec->dapm.idle_bias_off = driver->idle_bias_off; + if (driver->probe) { ret = driver->probe(codec); if (ret < 0) { @@ -1057,6 +1055,9 @@ static int soc_post_component_init(struct snd_soc_card *card, } rtd->card = card; + /* Make sure all DAPM widgets are instantiated */ + snd_soc_dapm_new_widgets(&codec->dapm); + /* machine controls, routes and widgets are not prefixed */ temp = codec->name_prefix; codec->name_prefix = NULL; @@ -1072,9 +1073,6 @@ static int soc_post_component_init(struct snd_soc_card *card, } codec->name_prefix = temp; - /* Make sure all DAPM widgets are instantiated */ - snd_soc_dapm_new_widgets(&codec->dapm); - /* register the rtd device */ rtd->codec = codec; rtd->dev.parent = card->dev; @@ -1319,6 +1317,7 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) struct snd_soc_codec *codec; struct snd_soc_codec_conf *codec_conf; enum snd_soc_compress_type compress_type; + struct snd_soc_dai_link *dai_link; int ret, i, order; mutex_lock(&card->mutex); @@ -1431,6 +1430,28 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) snd_soc_dapm_add_routes(&card->dapm, card->dapm_routes, card->num_dapm_routes); + snd_soc_dapm_new_widgets(&card->dapm); + + for (i = 0; i < card->num_links; i++) { + dai_link = &card->dai_link[i]; + + if (dai_link->dai_fmt) { + ret = snd_soc_dai_set_fmt(card->rtd[i].codec_dai, + dai_link->dai_fmt); + if (ret != 0) + dev_warn(card->rtd[i].codec_dai->dev, + "Failed to set DAI format: %d\n", + ret); + + ret = snd_soc_dai_set_fmt(card->rtd[i].cpu_dai, + dai_link->dai_fmt); + if (ret != 0) + dev_warn(card->rtd[i].cpu_dai->dev, + "Failed to set DAI format: %d\n", + ret); + } + } + snprintf(card->snd_card->shortname, sizeof(card->snd_card->shortname), "%s", card->name); snprintf(card->snd_card->longname, sizeof(card->snd_card->longname), @@ -1459,6 +1480,8 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) } } + snd_soc_dapm_new_widgets(&card->dapm); + ret = snd_card_register(card->snd_card); if (ret < 0) { printk(KERN_ERR "asoc: failed to register soundcard for %s\n", card->name); @@ -1479,6 +1502,7 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) #endif card->instantiated = 1; + snd_soc_dapm_sync(&card->dapm); mutex_unlock(&card->mutex); return; @@ -2229,7 +2253,8 @@ EXPORT_SYMBOL_GPL(snd_soc_info_volsw_ext); * @kcontrol: mixer control * @uinfo: control element information * - * Callback to provide information about a single mixer control. + * Callback to provide information about a single mixer control, or a double + * mixer control that spans 2 registers. * * Returns 0 for success. */ @@ -2239,8 +2264,6 @@ int snd_soc_info_volsw(struct snd_kcontrol *kcontrol, struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; int platform_max; - unsigned int shift = mc->shift; - unsigned int rshift = mc->rshift; if (!mc->platform_max) mc->platform_max = mc->max; @@ -2251,7 +2274,7 @@ int snd_soc_info_volsw(struct snd_kcontrol *kcontrol, else uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = shift == rshift ? 1 : 2; + uinfo->count = snd_soc_volsw_is_stereo(mc) ? 2 : 1; uinfo->value.integer.min = 0; uinfo->value.integer.max = platform_max; return 0; @@ -2263,7 +2286,8 @@ EXPORT_SYMBOL_GPL(snd_soc_info_volsw); * @kcontrol: mixer control * @ucontrol: control element information * - * Callback to get the value of a single mixer control. + * Callback to get the value of a single mixer control, or a double mixer + * control that spans 2 registers. * * Returns 0 for success. */ @@ -2274,6 +2298,7 @@ int snd_soc_get_volsw(struct snd_kcontrol *kcontrol, (struct soc_mixer_control *)kcontrol->private_value; struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); unsigned int reg = mc->reg; + unsigned int reg2 = mc->rreg; unsigned int shift = mc->shift; unsigned int rshift = mc->rshift; int max = mc->max; @@ -2282,13 +2307,18 @@ int snd_soc_get_volsw(struct snd_kcontrol *kcontrol, ucontrol->value.integer.value[0] = (snd_soc_read(codec, reg) >> shift) & mask; - if (shift != rshift) - ucontrol->value.integer.value[1] = - (snd_soc_read(codec, reg) >> rshift) & mask; - if (invert) { + if (invert) ucontrol->value.integer.value[0] = max - ucontrol->value.integer.value[0]; - if (shift != rshift) + + if (snd_soc_volsw_is_stereo(mc)) { + if (reg == reg2) + ucontrol->value.integer.value[1] = + (snd_soc_read(codec, reg) >> rshift) & mask; + else + ucontrol->value.integer.value[1] = + (snd_soc_read(codec, reg2) >> shift) & mask; + if (invert) ucontrol->value.integer.value[1] = max - ucontrol->value.integer.value[1]; } @@ -2302,7 +2332,8 @@ EXPORT_SYMBOL_GPL(snd_soc_get_volsw); * @kcontrol: mixer control * @ucontrol: control element information * - * Callback to set the value of a single mixer control. + * Callback to set the value of a single mixer control, or a double mixer + * control that spans 2 registers. * * Returns 0 for success. */ @@ -2313,143 +2344,44 @@ int snd_soc_put_volsw(struct snd_kcontrol *kcontrol, (struct soc_mixer_control *)kcontrol->private_value; struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); unsigned int reg = mc->reg; + unsigned int reg2 = mc->rreg; unsigned int shift = mc->shift; unsigned int rshift = mc->rshift; int max = mc->max; unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; - unsigned int val, val2, val_mask; + int err; + bool type_2r = 0; + unsigned int val2 = 0; + unsigned int val, val_mask; val = (ucontrol->value.integer.value[0] & mask); if (invert) val = max - val; val_mask = mask << shift; val = val << shift; - if (shift != rshift) { + if (snd_soc_volsw_is_stereo(mc)) { val2 = (ucontrol->value.integer.value[1] & mask); if (invert) val2 = max - val2; - val_mask |= mask << rshift; - val |= val2 << rshift; - } - return snd_soc_update_bits_locked(codec, reg, val_mask, val); -} -EXPORT_SYMBOL_GPL(snd_soc_put_volsw); - -/** - * snd_soc_info_volsw_2r - double mixer info callback - * @kcontrol: mixer control - * @uinfo: control element information - * - * Callback to provide information about a double mixer control that - * spans 2 codec registers. - * - * Returns 0 for success. - */ -int snd_soc_info_volsw_2r(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - int platform_max; - - if (!mc->platform_max) - mc->platform_max = mc->max; - platform_max = mc->platform_max; - - if (platform_max == 1 && !strstr(kcontrol->id.name, " Volume")) - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - else - uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - - uinfo->count = 2; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = platform_max; - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_info_volsw_2r); - -/** - * snd_soc_get_volsw_2r - double mixer get callback - * @kcontrol: mixer control - * @ucontrol: control element information - * - * Callback to get the value of a double mixer control that spans 2 registers. - * - * Returns 0 for success. - */ -int snd_soc_get_volsw_2r(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - unsigned int reg = mc->reg; - unsigned int reg2 = mc->rreg; - unsigned int shift = mc->shift; - int max = mc->max; - unsigned int mask = (1 << fls(max)) - 1; - unsigned int invert = mc->invert; - - ucontrol->value.integer.value[0] = - (snd_soc_read(codec, reg) >> shift) & mask; - ucontrol->value.integer.value[1] = - (snd_soc_read(codec, reg2) >> shift) & mask; - if (invert) { - ucontrol->value.integer.value[0] = - max - ucontrol->value.integer.value[0]; - ucontrol->value.integer.value[1] = - max - ucontrol->value.integer.value[1]; - } - - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_get_volsw_2r); - -/** - * snd_soc_put_volsw_2r - double mixer set callback - * @kcontrol: mixer control - * @ucontrol: control element information - * - * Callback to set the value of a double mixer control that spans 2 registers. - * - * Returns 0 for success. - */ -int snd_soc_put_volsw_2r(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - unsigned int reg = mc->reg; - unsigned int reg2 = mc->rreg; - unsigned int shift = mc->shift; - int max = mc->max; - unsigned int mask = (1 << fls(max)) - 1; - unsigned int invert = mc->invert; - int err; - unsigned int val, val2, val_mask; - - val_mask = mask << shift; - val = (ucontrol->value.integer.value[0] & mask); - val2 = (ucontrol->value.integer.value[1] & mask); - - if (invert) { - val = max - val; - val2 = max - val2; + if (reg == reg2) { + val_mask |= mask << rshift; + val |= val2 << rshift; + } else { + val2 = val2 << shift; + type_2r = 1; + } } - - val = val << shift; - val2 = val2 << shift; - err = snd_soc_update_bits_locked(codec, reg, val_mask, val); if (err < 0) return err; - err = snd_soc_update_bits_locked(codec, reg2, val_mask, val2); + if (type_2r) + err = snd_soc_update_bits_locked(codec, reg2, val_mask, val2); + return err; } -EXPORT_SYMBOL_GPL(snd_soc_put_volsw_2r); +EXPORT_SYMBOL_GPL(snd_soc_put_volsw); /** * snd_soc_info_volsw_s8 - signed mixer info callback @@ -2680,7 +2612,7 @@ int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id, if (dai->driver && dai->driver->ops->set_sysclk) return dai->driver->ops->set_sysclk(dai, clk_id, freq, dir); else if (dai->codec && dai->codec->driver->set_sysclk) - return dai->codec->driver->set_sysclk(dai->codec, clk_id, + return dai->codec->driver->set_sysclk(dai->codec, clk_id, 0, freq, dir); else return -EINVAL; @@ -2691,16 +2623,18 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_sysclk); * snd_soc_codec_set_sysclk - configure CODEC system or master clock. * @codec: CODEC * @clk_id: DAI specific clock ID + * @source: Source for the clock * @freq: new clock frequency in Hz * @dir: new clock direction - input/output. * * Configures the CODEC master (MCLK) or system (SYSCLK) clocking. */ int snd_soc_codec_set_sysclk(struct snd_soc_codec *codec, int clk_id, - unsigned int freq, int dir) + int source, unsigned int freq, int dir) { if (codec->driver->set_sysclk) - return codec->driver->set_sysclk(codec, clk_id, freq, dir); + return codec->driver->set_sysclk(codec, clk_id, source, + freq, dir); else return -EINVAL; } @@ -2895,6 +2829,7 @@ int snd_soc_register_card(struct snd_soc_card *card) card->rtd[i].dai_link = &card->dai_link[i]; INIT_LIST_HEAD(&card->list); + INIT_LIST_HEAD(&card->dapm_dirty); card->instantiated = 0; mutex_init(&card->mutex); @@ -3153,6 +3088,7 @@ int snd_soc_register_platform(struct device *dev, platform->driver = platform_drv; platform->dapm.dev = dev; platform->dapm.platform = platform; + platform->dapm.stream_event = platform_drv->stream_event; mutex_lock(&client_mutex); list_add(&platform->list, &platform_list); @@ -3265,6 +3201,7 @@ int snd_soc_register_codec(struct device *dev, codec->dapm.dev = dev; codec->dapm.codec = codec; codec->dapm.seq_notifier = codec_drv->seq_notifier; + codec->dapm.stream_event = codec_drv->stream_event; codec->dev = dev; codec->driver = codec_drv; codec->num_dai = num_dai; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index d67c637557a7..f42e8b9fb17d 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -48,6 +48,8 @@ #include <trace/events/asoc.h> +#define DAPM_UPDATE_STAT(widget, val) widget->dapm->card->dapm_stats.val++; + /* dapm power sequences - make this per codec in the future */ static int dapm_up_seq[] = { [snd_soc_dapm_pre] = 0, @@ -117,6 +119,21 @@ static void pop_dbg(struct device *dev, u32 pop_time, const char *fmt, ...) kfree(buf); } +static bool dapm_dirty_widget(struct snd_soc_dapm_widget *w) +{ + return !list_empty(&w->dirty); +} + +void dapm_mark_dirty(struct snd_soc_dapm_widget *w, const char *reason) +{ + if (!dapm_dirty_widget(w)) { + dev_vdbg(w->dapm->dev, "Marking %s dirty due to %s\n", + w->name, reason); + list_add_tail(&w->dirty, &w->dapm->card->dapm_dirty); + } +} +EXPORT_SYMBOL_GPL(dapm_mark_dirty); + /* create a new dapm widget */ static inline struct snd_soc_dapm_widget *dapm_cnew_widget( const struct snd_soc_dapm_widget *_widget) @@ -316,7 +333,7 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, } } break; - /* does not effect routing - always connected */ + /* does not affect routing - always connected */ case snd_soc_dapm_pga: case snd_soc_dapm_out_drv: case snd_soc_dapm_output: @@ -328,13 +345,13 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, case snd_soc_dapm_supply: case snd_soc_dapm_aif_in: case snd_soc_dapm_aif_out: - p->connect = 1; - break; - /* does effect routing - dynamically connected */ case snd_soc_dapm_hp: case snd_soc_dapm_mic: case snd_soc_dapm_spk: case snd_soc_dapm_line: + p->connect = 1; + break; + /* does affect routing - dynamically connected */ case snd_soc_dapm_pre: case snd_soc_dapm_post: p->connect = 0; @@ -443,6 +460,11 @@ static int dapm_new_mixer(struct snd_soc_dapm_widget *w) if (path->name != (char *)w->kcontrol_news[i].name) continue; + if (w->kcontrols[i]) { + path->kcontrol = w->kcontrols[i]; + continue; + } + wlistsize = sizeof(struct snd_soc_dapm_widget_list) + sizeof(struct snd_soc_dapm_widget *), wlist = kzalloc(wlistsize, GFP_KERNEL); @@ -579,8 +601,8 @@ static int dapm_new_mux(struct snd_soc_dapm_widget *w) name + prefix_len, prefix); ret = snd_ctl_add(card, kcontrol); if (ret < 0) { - dev_err(dapm->dev, - "asoc: failed to add kcontrol %s\n", w->name); + dev_err(dapm->dev, "failed to add kcontrol %s: %d\n", + w->name, ret); kfree(wlist); return ret; } @@ -644,30 +666,45 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget) struct snd_soc_dapm_path *path; int con = 0; + if (widget->outputs >= 0) + return widget->outputs; + + DAPM_UPDATE_STAT(widget, path_checks); + if (widget->id == snd_soc_dapm_supply) return 0; switch (widget->id) { case snd_soc_dapm_adc: case snd_soc_dapm_aif_out: - if (widget->active) - return snd_soc_dapm_suspend_check(widget); + if (widget->active) { + widget->outputs = snd_soc_dapm_suspend_check(widget); + return widget->outputs; + } default: break; } if (widget->connected) { /* connected pin ? */ - if (widget->id == snd_soc_dapm_output && !widget->ext) - return snd_soc_dapm_suspend_check(widget); + if (widget->id == snd_soc_dapm_output && !widget->ext) { + widget->outputs = snd_soc_dapm_suspend_check(widget); + return widget->outputs; + } /* connected jack or spk ? */ - if (widget->id == snd_soc_dapm_hp || widget->id == snd_soc_dapm_spk || - (widget->id == snd_soc_dapm_line && !list_empty(&widget->sources))) - return snd_soc_dapm_suspend_check(widget); + if (widget->id == snd_soc_dapm_hp || + widget->id == snd_soc_dapm_spk || + (widget->id == snd_soc_dapm_line && + !list_empty(&widget->sources))) { + widget->outputs = snd_soc_dapm_suspend_check(widget); + return widget->outputs; + } } list_for_each_entry(path, &widget->sinks, list_source) { + DAPM_UPDATE_STAT(widget, neighbour_checks); + if (path->weak) continue; @@ -680,6 +717,8 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget) } } + widget->outputs = con; + return con; } @@ -692,6 +731,11 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget) struct snd_soc_dapm_path *path; int con = 0; + if (widget->inputs >= 0) + return widget->inputs; + + DAPM_UPDATE_STAT(widget, path_checks); + if (widget->id == snd_soc_dapm_supply) return 0; @@ -699,28 +743,40 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget) switch (widget->id) { case snd_soc_dapm_dac: case snd_soc_dapm_aif_in: - if (widget->active) - return snd_soc_dapm_suspend_check(widget); + if (widget->active) { + widget->inputs = snd_soc_dapm_suspend_check(widget); + return widget->inputs; + } default: break; } if (widget->connected) { /* connected pin ? */ - if (widget->id == snd_soc_dapm_input && !widget->ext) - return snd_soc_dapm_suspend_check(widget); + if (widget->id == snd_soc_dapm_input && !widget->ext) { + widget->inputs = snd_soc_dapm_suspend_check(widget); + return widget->inputs; + } /* connected VMID/Bias for lower pops */ - if (widget->id == snd_soc_dapm_vmid) - return snd_soc_dapm_suspend_check(widget); + if (widget->id == snd_soc_dapm_vmid) { + widget->inputs = snd_soc_dapm_suspend_check(widget); + return widget->inputs; + } /* connected jack ? */ if (widget->id == snd_soc_dapm_mic || - (widget->id == snd_soc_dapm_line && !list_empty(&widget->sinks))) - return snd_soc_dapm_suspend_check(widget); + (widget->id == snd_soc_dapm_line && + !list_empty(&widget->sinks))) { + widget->inputs = snd_soc_dapm_suspend_check(widget); + return widget->inputs; + } + } list_for_each_entry(path, &widget->sources, list_sink) { + DAPM_UPDATE_STAT(widget, neighbour_checks); + if (path->weak) continue; @@ -733,6 +789,8 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget) } } + widget->inputs = con; + return con; } @@ -756,12 +814,29 @@ int dapm_reg_event(struct snd_soc_dapm_widget *w, } EXPORT_SYMBOL_GPL(dapm_reg_event); +static int dapm_widget_power_check(struct snd_soc_dapm_widget *w) +{ + if (w->power_checked) + return w->new_power; + + if (w->force) + w->new_power = 1; + else + w->new_power = w->power_check(w); + + w->power_checked = true; + + return w->new_power; +} + /* Generic check to see if a widget should be powered. */ static int dapm_generic_check_power(struct snd_soc_dapm_widget *w) { int in, out; + DAPM_UPDATE_STAT(w, power_checks); + in = is_connected_input_ep(w); dapm_clear_walk(w->dapm); out = is_connected_output_ep(w); @@ -774,6 +849,8 @@ static int dapm_adc_check_power(struct snd_soc_dapm_widget *w) { int in; + DAPM_UPDATE_STAT(w, power_checks); + if (w->active) { in = is_connected_input_ep(w); dapm_clear_walk(w->dapm); @@ -788,6 +865,8 @@ static int dapm_dac_check_power(struct snd_soc_dapm_widget *w) { int out; + DAPM_UPDATE_STAT(w, power_checks); + if (w->active) { out = is_connected_output_ep(w); dapm_clear_walk(w->dapm); @@ -801,10 +880,13 @@ static int dapm_dac_check_power(struct snd_soc_dapm_widget *w) static int dapm_supply_check_power(struct snd_soc_dapm_widget *w) { struct snd_soc_dapm_path *path; - int power = 0; + + DAPM_UPDATE_STAT(w, power_checks); /* Check if one of our outputs is connected */ list_for_each_entry(path, &w->sinks, list_source) { + DAPM_UPDATE_STAT(w, neighbour_checks); + if (path->weak) continue; @@ -815,21 +897,18 @@ static int dapm_supply_check_power(struct snd_soc_dapm_widget *w) if (!path->sink) continue; - if (path->sink->force) { - power = 1; - break; - } - - if (path->sink->power_check && - path->sink->power_check(path->sink)) { - power = 1; - break; - } + if (dapm_widget_power_check(path->sink)) + return 1; } dapm_clear_walk(w->dapm); - return power; + return 0; +} + +static int dapm_always_on_check_power(struct snd_soc_dapm_widget *w) +{ + return 1; } static int dapm_seq_compare(struct snd_soc_dapm_widget *a, @@ -1172,6 +1251,85 @@ static void dapm_post_sequence_async(void *data, async_cookie_t cookie) } } +static void dapm_widget_set_peer_power(struct snd_soc_dapm_widget *peer, + bool power, bool connect) +{ + /* If a connection is being made or broken then that update + * will have marked the peer dirty, otherwise the widgets are + * not connected and this update has no impact. */ + if (!connect) + return; + + /* If the peer is already in the state we're moving to then we + * won't have an impact on it. */ + if (power != peer->power) + dapm_mark_dirty(peer, "peer state change"); +} + +static void dapm_widget_set_power(struct snd_soc_dapm_widget *w, bool power, + struct list_head *up_list, + struct list_head *down_list) +{ + struct snd_soc_dapm_path *path; + + if (w->power == power) + return; + + trace_snd_soc_dapm_widget_power(w, power); + + /* If we changed our power state perhaps our neigbours changed + * also. + */ + list_for_each_entry(path, &w->sources, list_sink) { + if (path->source) { + dapm_widget_set_peer_power(path->source, power, + path->connect); + } + } + switch (w->id) { + case snd_soc_dapm_supply: + /* Supplies can't affect their outputs, only their inputs */ + break; + default: + list_for_each_entry(path, &w->sinks, list_source) { + if (path->sink) { + dapm_widget_set_peer_power(path->sink, power, + path->connect); + } + } + break; + } + + if (power) + dapm_seq_insert(w, up_list, true); + else + dapm_seq_insert(w, down_list, false); + + w->power = power; +} + +static void dapm_power_one_widget(struct snd_soc_dapm_widget *w, + struct list_head *up_list, + struct list_head *down_list) +{ + int power; + + switch (w->id) { + case snd_soc_dapm_pre: + dapm_seq_insert(w, down_list, false); + break; + case snd_soc_dapm_post: + dapm_seq_insert(w, up_list, true); + break; + + default: + power = dapm_widget_power_check(w); + + dapm_widget_set_power(w, power, up_list, down_list); + break; + } +} + /* * Scan each dapm widget for complete audio path. * A complete path is a route that has valid endpoints i.e.:- @@ -1190,7 +1348,6 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) LIST_HEAD(down_list); LIST_HEAD(async_domain); enum snd_soc_bias_level bias; - int power; trace_snd_soc_dapm_start(card); @@ -1203,61 +1360,47 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) } } - /* Check which widgets we need to power and store them in - * lists indicating if they should be powered up or down. - */ + memset(&card->dapm_stats, 0, sizeof(card->dapm_stats)); + list_for_each_entry(w, &card->widgets, list) { - switch (w->id) { - case snd_soc_dapm_pre: - dapm_seq_insert(w, &down_list, false); - break; - case snd_soc_dapm_post: - dapm_seq_insert(w, &up_list, true); - break; + w->power_checked = false; + w->inputs = -1; + w->outputs = -1; + } - default: - if (!w->power_check) - continue; + /* Check which widgets we need to power and store them in + * lists indicating if they should be powered up or down. We + * only check widgets that have been flagged as dirty but note + * that new widgets may be added to the dirty list while we + * iterate. + */ + list_for_each_entry(w, &card->dapm_dirty, dirty) { + dapm_power_one_widget(w, &up_list, &down_list); + } - if (!w->force) - power = w->power_check(w); - else - power = 1; + list_for_each_entry(w, &card->widgets, list) { + list_del_init(&w->dirty); - if (power) { - d = w->dapm; + if (w->power) { + d = w->dapm; - /* Supplies and micbiases only bring - * the context up to STANDBY as unless - * something else is active and - * passing audio they generally don't - * require full power. - */ - switch (w->id) { - case snd_soc_dapm_supply: - case snd_soc_dapm_micbias: - if (d->target_bias_level < SND_SOC_BIAS_STANDBY) - d->target_bias_level = SND_SOC_BIAS_STANDBY; - break; - default: - d->target_bias_level = SND_SOC_BIAS_ON; - break; - } + /* Supplies and micbiases only bring the + * context up to STANDBY as unless something + * else is active and passing audio they + * generally don't require full power. + */ + switch (w->id) { + case snd_soc_dapm_supply: + case snd_soc_dapm_micbias: + if (d->target_bias_level < SND_SOC_BIAS_STANDBY) + d->target_bias_level = SND_SOC_BIAS_STANDBY; + break; + default: + d->target_bias_level = SND_SOC_BIAS_ON; + break; } - - if (w->power == power) - continue; - - trace_snd_soc_dapm_widget_power(w, power); - - if (power) - dapm_seq_insert(w, &up_list, true); - else - dapm_seq_insert(w, &down_list, false); - - w->power = power; - break; } + } /* If there are no DAPM widgets then try to figure out power from the @@ -1286,14 +1429,18 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) } } - /* Force all contexts in the card to the same bias state */ + /* Force all contexts in the card to the same bias state if + * they're not ground referenced. + */ bias = SND_SOC_BIAS_OFF; list_for_each_entry(d, &card->dapm_list, list) if (d->target_bias_level > bias) bias = d->target_bias_level; list_for_each_entry(d, &card->dapm_list, list) - d->target_bias_level = bias; + if (!d->idle_bias_off) + d->target_bias_level = bias; + trace_snd_soc_dapm_walk_done(card); /* Run all the bias changes in parallel */ list_for_each_entry(d, &dapm->card->dapm_list, list) @@ -1524,14 +1671,21 @@ static int dapm_mux_update_power(struct snd_soc_dapm_widget *widget, found = 1; /* we now need to match the string in the enum to the path */ - if (!(strcmp(path->name, e->texts[mux]))) + if (!(strcmp(path->name, e->texts[mux]))) { path->connect = 1; /* new connection */ - else + dapm_mark_dirty(path->source, "mux connection"); + } else { + if (path->connect) + dapm_mark_dirty(path->source, + "mux disconnection"); path->connect = 0; /* old connection must be powered down */ + } } - if (found) + if (found) { + dapm_mark_dirty(widget, "mux change"); dapm_power_widgets(widget->dapm, SND_SOC_DAPM_STREAM_NOP); + } return 0; } @@ -1556,11 +1710,13 @@ static int dapm_mixer_update_power(struct snd_soc_dapm_widget *widget, /* found, now check type */ found = 1; path->connect = connect; - break; + dapm_mark_dirty(path->source, "mixer connection"); } - if (found) + if (found) { + dapm_mark_dirty(widget, "mixer update"); dapm_power_widgets(widget->dapm, SND_SOC_DAPM_STREAM_NOP); + } return 0; } @@ -1704,6 +1860,7 @@ static int snd_soc_dapm_set_pin(struct snd_soc_dapm_context *dapm, w->connected = status; if (status == 0) w->force = 0; + dapm_mark_dirty(w, "pin configuration"); return 0; } @@ -1719,6 +1876,13 @@ static int snd_soc_dapm_set_pin(struct snd_soc_dapm_context *dapm, */ int snd_soc_dapm_sync(struct snd_soc_dapm_context *dapm) { + /* + * Suppress early reports (eg, jacks syncing their state) to avoid + * silly DAPM runs during card startup. + */ + if (!dapm->card || !dapm->card->instantiated) + return 0; + return dapm_power_widgets(dapm, SND_SOC_DAPM_STREAM_NOP); } EXPORT_SYMBOL_GPL(snd_soc_dapm_sync); @@ -2004,42 +2168,18 @@ int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm) case snd_soc_dapm_switch: case snd_soc_dapm_mixer: case snd_soc_dapm_mixer_named_ctl: - w->power_check = dapm_generic_check_power; dapm_new_mixer(w); break; case snd_soc_dapm_mux: case snd_soc_dapm_virt_mux: case snd_soc_dapm_value_mux: - w->power_check = dapm_generic_check_power; dapm_new_mux(w); break; - case snd_soc_dapm_adc: - case snd_soc_dapm_aif_out: - w->power_check = dapm_adc_check_power; - break; - case snd_soc_dapm_dac: - case snd_soc_dapm_aif_in: - w->power_check = dapm_dac_check_power; - break; case snd_soc_dapm_pga: case snd_soc_dapm_out_drv: - w->power_check = dapm_generic_check_power; dapm_new_pga(w); break; - case snd_soc_dapm_input: - case snd_soc_dapm_output: - case snd_soc_dapm_micbias: - case snd_soc_dapm_spk: - case snd_soc_dapm_hp: - case snd_soc_dapm_mic: - case snd_soc_dapm_line: - w->power_check = dapm_generic_check_power; - break; - case snd_soc_dapm_supply: - w->power_check = dapm_supply_check_power; - case snd_soc_dapm_vmid: - case snd_soc_dapm_pre: - case snd_soc_dapm_post: + default: break; } @@ -2056,6 +2196,7 @@ int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm) w->new = 1; + dapm_mark_dirty(w, "new widget"); dapm_debugfs_add_widget(w); } @@ -2530,6 +2671,44 @@ int snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, else snprintf(w->name, name_len, "%s", widget->name); + switch (w->id) { + case snd_soc_dapm_switch: + case snd_soc_dapm_mixer: + case snd_soc_dapm_mixer_named_ctl: + w->power_check = dapm_generic_check_power; + break; + case snd_soc_dapm_mux: + case snd_soc_dapm_virt_mux: + case snd_soc_dapm_value_mux: + w->power_check = dapm_generic_check_power; + break; + case snd_soc_dapm_adc: + case snd_soc_dapm_aif_out: + w->power_check = dapm_adc_check_power; + break; + case snd_soc_dapm_dac: + case snd_soc_dapm_aif_in: + w->power_check = dapm_dac_check_power; + break; + case snd_soc_dapm_pga: + case snd_soc_dapm_out_drv: + case snd_soc_dapm_input: + case snd_soc_dapm_output: + case snd_soc_dapm_micbias: + case snd_soc_dapm_spk: + case snd_soc_dapm_hp: + case snd_soc_dapm_mic: + case snd_soc_dapm_line: + w->power_check = dapm_generic_check_power; + break; + case snd_soc_dapm_supply: + w->power_check = dapm_supply_check_power; + break; + default: + w->power_check = dapm_always_on_check_power; + break; + } + dapm->n_widgets++; w->dapm = dapm; w->codec = dapm->codec; @@ -2537,6 +2716,7 @@ int snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, INIT_LIST_HEAD(&w->sources); INIT_LIST_HEAD(&w->sinks); INIT_LIST_HEAD(&w->list); + INIT_LIST_HEAD(&w->dirty); list_add(&w->list, &dapm->card->widgets); /* machine layer set ups unconnected pins and insertions */ @@ -2584,9 +2764,10 @@ static void soc_dapm_stream_event(struct snd_soc_dapm_context *dapm, { if (!w->sname || w->dapm != dapm) continue; - dev_dbg(w->dapm->dev, "widget %s\n %s stream %s event %d\n", + dev_vdbg(w->dapm->dev, "widget %s\n %s stream %s event %d\n", w->name, w->sname, stream, event); if (strstr(w->sname, stream)) { + dapm_mark_dirty(w, "stream event"); switch(event) { case SND_SOC_DAPM_STREAM_START: w->active = 1; @@ -2604,6 +2785,10 @@ static void soc_dapm_stream_event(struct snd_soc_dapm_context *dapm, } dapm_power_widgets(dapm, event); + + /* do we need to notify any clients that DAPM stream is complete */ + if (dapm->stream_event) + dapm->stream_event(dapm, event); } /** @@ -2672,6 +2857,7 @@ int snd_soc_dapm_force_enable_pin(struct snd_soc_dapm_context *dapm, dev_dbg(w->dapm->dev, "dapm: force enable pin %s\n", pin); w->connected = 1; w->force = 1; + dapm_mark_dirty(w, "force enable"); return 0; } diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c index a62f7dd4ba96..dd89933e2c72 100644 --- a/sound/soc/soc-io.c +++ b/sound/soc/soc-io.c @@ -13,26 +13,14 @@ #include <linux/i2c.h> #include <linux/spi/spi.h> +#include <linux/regmap.h> #include <sound/soc.h> #include <trace/events/asoc.h> -#ifdef CONFIG_SPI_MASTER -static int do_spi_write(void *control, const char *data, int len) -{ - struct spi_device *spi = control; - int ret; - - ret = spi_write(spi, data, len); - if (ret < 0) - return ret; - - return len; -} -#endif - -static int do_hw_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value, const void *data, int len) +#ifdef CONFIG_REGMAP +static int hw_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) { int ret; @@ -49,13 +37,7 @@ static int do_hw_write(struct snd_soc_codec *codec, unsigned int reg, return 0; } - ret = codec->hw_write(codec->control_data, data, len); - if (ret == len) - return 0; - if (ret < 0) - return ret; - else - return -EIO; + return regmap_write(codec->control_data, reg, value); } static unsigned int hw_read(struct snd_soc_codec *codec, unsigned int reg) @@ -69,8 +51,11 @@ static unsigned int hw_read(struct snd_soc_codec *codec, unsigned int reg) if (codec->cache_only) return -1; - BUG_ON(!codec->hw_read); - return codec->hw_read(codec, reg); + ret = regmap_read(codec->control_data, reg, &val); + if (ret == 0) + return val; + else + return -1; } ret = snd_soc_cache_read(codec, reg, &val); @@ -79,202 +64,18 @@ static unsigned int hw_read(struct snd_soc_codec *codec, unsigned int reg) return val; } -static int snd_soc_4_12_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - u16 data; - - data = cpu_to_be16((reg << 12) | (value & 0xffffff)); - - return do_hw_write(codec, reg, value, &data, 2); -} - -static int snd_soc_7_9_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - u16 data; - - data = cpu_to_be16((reg << 9) | (value & 0x1ff)); - - return do_hw_write(codec, reg, value, &data, 2); -} - -static int snd_soc_8_8_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - u8 data[2]; - - reg &= 0xff; - data[0] = reg; - data[1] = value & 0xff; - - return do_hw_write(codec, reg, value, data, 2); -} - -static int snd_soc_8_16_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - u8 data[3]; - u16 val = cpu_to_be16(value); - - data[0] = reg; - memcpy(&data[1], &val, sizeof(val)); - - return do_hw_write(codec, reg, value, data, 3); -} - -#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) -static unsigned int do_i2c_read(struct snd_soc_codec *codec, - void *reg, int reglen, - void *data, int datalen) -{ - struct i2c_msg xfer[2]; - int ret; - struct i2c_client *client = codec->control_data; - - /* Write register */ - xfer[0].addr = client->addr; - xfer[0].flags = 0; - xfer[0].len = reglen; - xfer[0].buf = reg; - - /* Read data */ - xfer[1].addr = client->addr; - xfer[1].flags = I2C_M_RD; - xfer[1].len = datalen; - xfer[1].buf = data; - - ret = i2c_transfer(client->adapter, xfer, 2); - if (ret == 2) - return 0; - else if (ret < 0) - return ret; - else - return -EIO; -} -#endif - -#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) -static unsigned int snd_soc_8_8_read_i2c(struct snd_soc_codec *codec, - unsigned int r) -{ - u8 reg = r; - u8 data; - int ret; - - ret = do_i2c_read(codec, ®, 1, &data, 1); - if (ret < 0) - return 0; - return data; -} -#else -#define snd_soc_8_8_read_i2c NULL -#endif - -#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) -static unsigned int snd_soc_8_16_read_i2c(struct snd_soc_codec *codec, - unsigned int r) -{ - u8 reg = r; - u16 data; - int ret; - - ret = do_i2c_read(codec, ®, 1, &data, 2); - if (ret < 0) - return 0; - return (data >> 8) | ((data & 0xff) << 8); -} -#else -#define snd_soc_8_16_read_i2c NULL -#endif - -#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) -static unsigned int snd_soc_16_8_read_i2c(struct snd_soc_codec *codec, - unsigned int r) -{ - u16 reg = r; - u8 data; - int ret; - - ret = do_i2c_read(codec, ®, 2, &data, 1); - if (ret < 0) - return 0; - return data; -} -#else -#define snd_soc_16_8_read_i2c NULL -#endif - -#if defined(CONFIG_SPI_MASTER) -static unsigned int snd_soc_16_8_read_spi(struct snd_soc_codec *codec, - unsigned int r) -{ - struct spi_device *spi = codec->control_data; - - const u16 reg = cpu_to_be16(r | 0x100); - u8 data; - int ret; - - ret = spi_write_then_read(spi, ®, 2, &data, 1); - if (ret < 0) - return 0; - return data; -} -#else -#define snd_soc_16_8_read_spi NULL -#endif - -static int snd_soc_16_8_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - u8 data[3]; - u16 rval = cpu_to_be16(reg); - - memcpy(data, &rval, sizeof(rval)); - data[2] = value; - - return do_hw_write(codec, reg, value, data, 3); -} - -#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) -static unsigned int snd_soc_16_16_read_i2c(struct snd_soc_codec *codec, - unsigned int r) -{ - u16 reg = cpu_to_be16(r); - u16 data; - int ret; - - ret = do_i2c_read(codec, ®, 2, &data, 2); - if (ret < 0) - return 0; - return be16_to_cpu(data); -} -#else -#define snd_soc_16_16_read_i2c NULL -#endif - -static int snd_soc_16_16_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - u16 data[2]; - - data[0] = cpu_to_be16(reg); - data[1] = cpu_to_be16(value); - - return do_hw_write(codec, reg, value, data, sizeof(data)); -} - /* Primitive bulk write support for soc-cache. The data pointed to by - * `data' needs to already be in the form the hardware expects - * including any leading register specific data. Any data written - * through this function will not go through the cache as it only - * handles writing to volatile or out of bounds registers. + * `data' needs to already be in the form the hardware expects. Any + * data written through this function will not go through the cache as + * it only handles writing to volatile or out of bounds registers. + * + * This is currently only supported for devices using the regmap API + * wrappers. */ -static int snd_soc_hw_bulk_write_raw(struct snd_soc_codec *codec, unsigned int reg, +static int snd_soc_hw_bulk_write_raw(struct snd_soc_codec *codec, + unsigned int reg, const void *data, size_t len) { - int ret; - /* To ensure that we don't get out of sync with the cache, check * whether the base register is volatile or if we've directly asked * to bypass the cache. Out of bounds registers are considered @@ -285,68 +86,9 @@ static int snd_soc_hw_bulk_write_raw(struct snd_soc_codec *codec, unsigned int r && reg < codec->driver->reg_cache_size) return -EINVAL; - switch (codec->control_type) { -#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) - case SND_SOC_I2C: - ret = i2c_master_send(to_i2c_client(codec->dev), data, len); - break; -#endif -#if defined(CONFIG_SPI_MASTER) - case SND_SOC_SPI: - ret = spi_write(to_spi_device(codec->dev), data, len); - break; -#endif - default: - BUG(); - } - - if (ret == len) - return 0; - if (ret < 0) - return ret; - else - return -EIO; + return regmap_raw_write(codec->control_data, reg, data, len); } -static struct { - int addr_bits; - int data_bits; - int (*write)(struct snd_soc_codec *codec, unsigned int, unsigned int); - unsigned int (*read)(struct snd_soc_codec *, unsigned int); - unsigned int (*i2c_read)(struct snd_soc_codec *, unsigned int); - unsigned int (*spi_read)(struct snd_soc_codec *, unsigned int); -} io_types[] = { - { - .addr_bits = 4, .data_bits = 12, - .write = snd_soc_4_12_write, - }, - { - .addr_bits = 7, .data_bits = 9, - .write = snd_soc_7_9_write, - }, - { - .addr_bits = 8, .data_bits = 8, - .write = snd_soc_8_8_write, - .i2c_read = snd_soc_8_8_read_i2c, - }, - { - .addr_bits = 8, .data_bits = 16, - .write = snd_soc_8_16_write, - .i2c_read = snd_soc_8_16_read_i2c, - }, - { - .addr_bits = 16, .data_bits = 8, - .write = snd_soc_16_8_write, - .i2c_read = snd_soc_16_8_read_i2c, - .spi_read = snd_soc_16_8_read_spi, - }, - { - .addr_bits = 16, .data_bits = 16, - .write = snd_soc_16_16_write, - .i2c_read = snd_soc_16_16_read_i2c, - }, -}; - /** * snd_soc_codec_set_cache_io: Set up standard I/O functions. * @@ -370,50 +112,51 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, int addr_bits, int data_bits, enum snd_soc_control_type control) { - int i; - - for (i = 0; i < ARRAY_SIZE(io_types); i++) - if (io_types[i].addr_bits == addr_bits && - io_types[i].data_bits == data_bits) - break; - if (i == ARRAY_SIZE(io_types)) { - printk(KERN_ERR - "No I/O functions for %d bit address %d bit data\n", - addr_bits, data_bits); - return -EINVAL; - } + struct regmap_config config; - codec->write = io_types[i].write; + memset(&config, 0, sizeof(config)); + codec->write = hw_write; codec->read = hw_read; codec->bulk_write_raw = snd_soc_hw_bulk_write_raw; + config.reg_bits = addr_bits; + config.val_bits = data_bits; + switch (control) { +#if defined(CONFIG_REGMAP_I2C) || defined(CONFIG_REGMAP_I2C_MODULE) case SND_SOC_I2C: -#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) - codec->hw_write = (hw_write_t)i2c_master_send; -#endif - if (io_types[i].i2c_read) - codec->hw_read = io_types[i].i2c_read; - - codec->control_data = container_of(codec->dev, - struct i2c_client, - dev); + codec->control_data = regmap_init_i2c(to_i2c_client(codec->dev), + &config); break; +#endif +#if defined(CONFIG_REGMAP_SPI) || defined(CONFIG_REGMAP_SPI_MODULE) case SND_SOC_SPI: -#ifdef CONFIG_SPI_MASTER - codec->hw_write = do_spi_write; + codec->control_data = regmap_init_spi(to_spi_device(codec->dev), + &config); + break; #endif - if (io_types[i].spi_read) - codec->hw_read = io_types[i].spi_read; - codec->control_data = container_of(codec->dev, - struct spi_device, - dev); + case SND_SOC_REGMAP: + /* Device has made its own regmap arrangements */ break; + + default: + return -EINVAL; } + if (IS_ERR(codec->control_data)) + return PTR_ERR(codec->control_data); + return 0; } EXPORT_SYMBOL_GPL(snd_soc_codec_set_cache_io); - +#else +int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, + int addr_bits, int data_bits, + enum snd_soc_control_type control) +{ + return -ENOTSUPP; +} +EXPORT_SYMBOL_GPL(snd_soc_codec_set_cache_io); +#endif diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index fa31d9c2abd8..52db96636290 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -188,6 +188,8 @@ int snd_soc_jack_add_pins(struct snd_soc_jack *jack, int count, list_add(&(pins[i].list), &jack->pins); } + snd_soc_dapm_new_widgets(&jack->codec->card->dapm); + /* Update to reflect the last reported status; canned jack * implementations are likely to set their state before the * card has an opportunity to associate pins. diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 2879c883eebc..ee15337353fa 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -27,17 +27,13 @@ #include <sound/soc.h> #include <sound/initval.h> -static DEFINE_MUTEX(pcm_mutex); - -static int soc_pcm_apply_symmetry(struct snd_pcm_substream *substream) +static int soc_pcm_apply_symmetry(struct snd_pcm_substream *substream, + struct snd_soc_dai *soc_dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; int ret; - if (!codec_dai->driver->symmetric_rates && - !cpu_dai->driver->symmetric_rates && + if (!soc_dai->driver->symmetric_rates && !rtd->dai_link->symmetric_rates) return 0; @@ -45,19 +41,19 @@ static int soc_pcm_apply_symmetry(struct snd_pcm_substream *substream) * the second can need to get its constraints before the first has * picked a rate. Complain and allow the application to carry on. */ - if (!rtd->rate) { - dev_warn(&rtd->dev, + if (!soc_dai->rate) { + dev_warn(soc_dai->dev, "Not enforcing symmetric_rates due to race\n"); return 0; } - dev_dbg(&rtd->dev, "Symmetry forces %dHz rate\n", rtd->rate); + dev_dbg(soc_dai->dev, "Symmetry forces %dHz rate\n", soc_dai->rate); ret = snd_pcm_hw_constraint_minmax(substream->runtime, SNDRV_PCM_HW_PARAM_RATE, - rtd->rate, rtd->rate); + soc_dai->rate, soc_dai->rate); if (ret < 0) { - dev_err(&rtd->dev, + dev_err(soc_dai->dev, "Unable to apply rate symmetry constraint: %d\n", ret); return ret; } @@ -187,8 +183,14 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) } /* Symmetry only applies if we've already got an active stream. */ - if (cpu_dai->active || codec_dai->active) { - ret = soc_pcm_apply_symmetry(substream); + if (cpu_dai->active) { + ret = soc_pcm_apply_symmetry(substream, cpu_dai); + if (ret != 0) + goto config_err; + } + + if (codec_dai->active) { + ret = soc_pcm_apply_symmetry(substream, codec_dai); if (ret != 0) goto config_err; } @@ -290,8 +292,12 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) codec_dai->active--; codec->active--; - if (!cpu_dai->active && !codec_dai->active) - rtd->rate = 0; + /* clear the corresponding DAIs rate when inactive */ + if (!cpu_dai->active) + cpu_dai->rate = 0; + + if (!codec_dai->active) + codec_dai->rate = 0; /* Muting the DAC suppresses artifacts caused during digital * shutdown, for example from stopping clocks. @@ -313,10 +319,17 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) cpu_dai->runtime = NULL; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - /* start delayed pop wq here for playback streams */ - codec_dai->pop_wait = 1; - schedule_delayed_work(&rtd->delayed_work, - msecs_to_jiffies(rtd->pmdown_time)); + if (unlikely(codec->ignore_pmdown_time)) { + /* powered down playback stream now */ + snd_soc_dapm_stream_event(rtd, + codec_dai->driver->playback.stream_name, + SND_SOC_DAPM_STREAM_STOP); + } else { + /* start delayed pop wq here for playback streams */ + codec_dai->pop_wait = 1; + schedule_delayed_work(&rtd->delayed_work, + msecs_to_jiffies(rtd->pmdown_time)); + } } else { /* capture streams can be powered down now */ snd_soc_dapm_stream_event(rtd, @@ -449,7 +462,9 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, } } - rtd->rate = params_rate(params); + /* store the rate for each DAIs */ + cpu_dai->rate = params_rate(params); + codec_dai->rate = params_rate(params); out: mutex_unlock(&rtd->pcm_mutex); diff --git a/sound/soc/tegra/tegra_das.c b/sound/soc/tegra/tegra_das.c index 9f24ef73f2cb..3b55a44146af 100644 --- a/sound/soc/tegra/tegra_das.c +++ b/sound/soc/tegra/tegra_das.c @@ -212,7 +212,7 @@ err_release: release_mem_region(res->start, resource_size(res)); err_free: kfree(das); - das = 0; + das = NULL; exit: return ret; } @@ -234,7 +234,7 @@ static int __devexit tegra_das_remove(struct platform_device *pdev) release_mem_region(res->start, resource_size(res)); kfree(das); - das = 0; + das = NULL; return 0; } diff --git a/sound/soc/tegra/tegra_i2s.c b/sound/soc/tegra/tegra_i2s.c index f36b9969cfec..6728fab8c411 100644 --- a/sound/soc/tegra/tegra_i2s.c +++ b/sound/soc/tegra/tegra_i2s.c @@ -312,7 +312,7 @@ static struct snd_soc_dai_ops tegra_i2s_dai_ops = { .trigger = tegra_i2s_trigger, }; -struct snd_soc_dai_driver tegra_i2s_dai[] = { +static struct snd_soc_dai_driver tegra_i2s_dai[] = { { .name = DRV_NAME ".0", .probe = tegra_i2s_probe, diff --git a/sound/soc/tegra/tegra_pcm.c b/sound/soc/tegra/tegra_pcm.c index c7cfd96e991e..436def1dfa39 100644 --- a/sound/soc/tegra/tegra_pcm.c +++ b/sound/soc/tegra/tegra_pcm.c @@ -367,7 +367,7 @@ static void tegra_pcm_free(struct snd_pcm *pcm) tegra_pcm_deallocate_dma_buffer(pcm, SNDRV_PCM_STREAM_PLAYBACK); } -struct snd_soc_platform_driver tegra_pcm_platform = { +static struct snd_soc_platform_driver tegra_pcm_platform = { .ops = &tegra_pcm_ops, .pcm_new = tegra_pcm_new, .pcm_free = tegra_pcm_free, diff --git a/sound/soc/tegra/tegra_spdif.c b/sound/soc/tegra/tegra_spdif.c index abe606b0a29e..dd11d0c63474 100644 --- a/sound/soc/tegra/tegra_spdif.c +++ b/sound/soc/tegra/tegra_spdif.c @@ -127,7 +127,7 @@ static int tegra_spdif_hw_params(struct snd_pcm_substream *substream, { struct device *dev = substream->pcm->card->dev; struct tegra_spdif *spdif = snd_soc_dai_get_drvdata(dai); - int ret, srate, spdifclock; + int ret, spdifclock; spdif->reg_ctrl &= ~TEGRA_SPDIF_CTRL_PACK; spdif->reg_ctrl &= ~TEGRA_SPDIF_CTRL_BIT_MODE_MASK; @@ -140,7 +140,6 @@ static int tegra_spdif_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - srate = params_rate(params); switch (params_rate(params)) { case 32000: spdifclock = 4096000; @@ -232,7 +231,7 @@ static struct snd_soc_dai_ops tegra_spdif_dai_ops = { .trigger = tegra_spdif_trigger, }; -struct snd_soc_dai_driver tegra_spdif_dai = { +static struct snd_soc_dai_driver tegra_spdif_dai = { .name = DRV_NAME, .probe = tegra_spdif_probe, .playback = { diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index be27f1d229af..a81cf39257bf 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -339,8 +339,6 @@ static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_nc_pin(dapm, "LINEOUTL"); } - snd_soc_dapm_sync(dapm); - return 0; } diff --git a/sound/soc/tegra/trimslice.c b/sound/soc/tegra/trimslice.c index 8fc07e9adf2e..b3a7efa6d960 100644 --- a/sound/soc/tegra/trimslice.c +++ b/sound/soc/tegra/trimslice.c @@ -124,8 +124,6 @@ static int trimslice_asoc_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_nc_pin(dapm, "RHPOUT"); snd_soc_dapm_nc_pin(dapm, "MICIN"); - snd_soc_dapm_sync(dapm); - return 0; } diff --git a/sound/soc/txx9/txx9aclc-ac97.c b/sound/soc/txx9/txx9aclc-ac97.c index 743d07b82c06..a4e3f5501847 100644 --- a/sound/soc/txx9/txx9aclc-ac97.c +++ b/sound/soc/txx9/txx9aclc-ac97.c @@ -201,7 +201,7 @@ static int __devinit txx9aclc_ac97_dev_probe(struct platform_device *pdev) if (!drvdata->base) return -EBUSY; err = devm_request_irq(&pdev->dev, irq, txx9aclc_ac97_irq, - IRQF_DISABLED, dev_name(&pdev->dev), drvdata); + 0, dev_name(&pdev->dev), drvdata); if (err < 0) return err; diff --git a/sound/soc/txx9/txx9aclc-generic.c b/sound/soc/txx9/txx9aclc-generic.c index 6770e7166be4..9b5e283af51c 100644 --- a/sound/soc/txx9/txx9aclc-generic.c +++ b/sound/soc/txx9/txx9aclc-generic.c @@ -62,7 +62,7 @@ static int __exit txx9aclc_generic_remove(struct platform_device *pdev) } static struct platform_driver txx9aclc_generic_driver = { - .remove = txx9aclc_generic_remove, + .remove = __exit_p(txx9aclc_generic_remove), .driver = { .name = "txx9aclc-generic", .owner = THIS_MODULE, diff --git a/sound/sparc/amd7930.c b/sound/sparc/amd7930.c index ad7d4d7d9237..f036776380b5 100644 --- a/sound/sparc/amd7930.c +++ b/sound/sparc/amd7930.c @@ -962,7 +962,7 @@ static int __devinit snd_amd7930_create(struct snd_card *card, amd7930_idle(amd); if (request_irq(irq, snd_amd7930_interrupt, - IRQF_DISABLED | IRQF_SHARED, "amd7930", amd)) { + IRQF_SHARED, "amd7930", amd)) { snd_printk(KERN_ERR "amd7930-%d: Unable to grab IRQ %d\n", dev, irq); snd_amd7930_free(amd); diff --git a/sound/usb/6fire/firmware.c b/sound/usb/6fire/firmware.c index 1e3ae3327dd3..07bcfe4d18a7 100644 --- a/sound/usb/6fire/firmware.c +++ b/sound/usb/6fire/firmware.c @@ -16,6 +16,7 @@ #include <linux/firmware.h> #include <linux/bitrev.h> +#include <linux/kernel.h> #include "firmware.h" #include "chip.h" @@ -59,21 +60,19 @@ struct ihex_record { unsigned int txt_offset; /* current position in txt_data */ }; -static u8 usb6fire_fw_ihex_nibble(const u8 n) -{ - if (n >= '0' && n <= '9') - return n - '0'; - else if (n >= 'A' && n <= 'F') - return n - ('A' - 10); - else if (n >= 'a' && n <= 'f') - return n - ('a' - 10); - return 0; -} - static u8 usb6fire_fw_ihex_hex(const u8 *data, u8 *crc) { - u8 val = (usb6fire_fw_ihex_nibble(data[0]) << 4) | - usb6fire_fw_ihex_nibble(data[1]); + u8 val = 0; + int hval; + + hval = hex_to_bin(data[0]); + if (hval >= 0) + val |= (hval << 4); + + hval = hex_to_bin(data[1]); + if (hval >= 0) + val |= hval; + *crc += val; return val; } diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig index 8beb77563da2..3efc21c3d67c 100644 --- a/sound/usb/Kconfig +++ b/sound/usb/Kconfig @@ -67,6 +67,7 @@ config SND_USB_CAIAQ * Native Instruments Guitar Rig mobile * Native Instruments Traktor Kontrol X1 * Native Instruments Traktor Kontrol S4 + * Native Instruments Maschine Controller To compile this driver as a module, choose M here: the module will be called snd-usb-caiaq. @@ -85,6 +86,7 @@ config SND_USB_CAIAQ_INPUT * Native Instruments Kore Controller 2 * Native Instruments Audio Kontrol 1 * Native Instruments Traktor Kontrol S4 + * Native Instruments Maschine Controller config SND_USB_US122L tristate "Tascam US-122L USB driver" diff --git a/sound/usb/Makefile b/sound/usb/Makefile index cf9ed66445fa..ac256dc4c6be 100644 --- a/sound/usb/Makefile +++ b/sound/usb/Makefile @@ -3,16 +3,16 @@ # snd-usb-audio-objs := card.o \ + clock.o \ + endpoint.o \ + format.o \ + helper.o \ mixer.o \ mixer_quirks.o \ + pcm.o \ proc.o \ quirks.o \ - format.o \ - endpoint.o \ - urb.o \ - pcm.o \ - helper.o \ - clock.o + stream.o snd-usbmidi-lib-objs := midi.o diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c index 45bc4a2dc6f0..3eb605bd9503 100644 --- a/sound/usb/caiaq/device.c +++ b/sound/usb/caiaq/device.c @@ -50,7 +50,8 @@ MODULE_SUPPORTED_DEVICE("{{Native Instruments, RigKontrol2}," "{Native Instruments, Session I/O}," "{Native Instruments, GuitarRig mobile}" "{Native Instruments, Traktor Kontrol X1}" - "{Native Instruments, Traktor Kontrol S4}"); + "{Native Instruments, Traktor Kontrol S4}" + "{Native Instruments, Maschine Controller}"); static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-max */ static char* id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* Id for this card */ @@ -146,6 +147,11 @@ static struct usb_device_id snd_usb_id_table[] = { .idVendor = USB_VID_NATIVEINSTRUMENTS, .idProduct = USB_PID_TRAKTORAUDIO2 }, + { + .match_flags = USB_DEVICE_ID_MATCH_DEVICE, + .idVendor = USB_VID_NATIVEINSTRUMENTS, + .idProduct = USB_PID_MASCHINECONTROLLER + }, { /* terminator */ } }; diff --git a/sound/usb/caiaq/device.h b/sound/usb/caiaq/device.h index 3f9c6339ae90..562b0bff9c41 100644 --- a/sound/usb/caiaq/device.h +++ b/sound/usb/caiaq/device.h @@ -18,6 +18,7 @@ #define USB_PID_TRAKTORKONTROLX1 0x2305 #define USB_PID_TRAKTORKONTROLS4 0xbaff #define USB_PID_TRAKTORAUDIO2 0x041d +#define USB_PID_MASCHINECONTROLLER 0x0808 #define EP1_BUFSIZE 64 #define EP4_BUFSIZE 512 diff --git a/sound/usb/caiaq/input.c b/sound/usb/caiaq/input.c index a213813487bd..26a121b42c3c 100644 --- a/sound/usb/caiaq/input.c +++ b/sound/usb/caiaq/input.c @@ -67,6 +67,61 @@ static unsigned short keycode_kore[] = { KEY_BRL_DOT5 }; +#define MASCHINE_BUTTONS (42) +#define MASCHINE_BUTTON(X) ((X) + BTN_MISC) +#define MASCHINE_PADS (16) +#define MASCHINE_PAD(X) ((X) + ABS_PRESSURE) + +static unsigned short keycode_maschine[] = { + MASCHINE_BUTTON(40), /* mute */ + MASCHINE_BUTTON(39), /* solo */ + MASCHINE_BUTTON(38), /* select */ + MASCHINE_BUTTON(37), /* duplicate */ + MASCHINE_BUTTON(36), /* navigate */ + MASCHINE_BUTTON(35), /* pad mode */ + MASCHINE_BUTTON(34), /* pattern */ + MASCHINE_BUTTON(33), /* scene */ + KEY_RESERVED, /* spacer */ + + MASCHINE_BUTTON(30), /* rec */ + MASCHINE_BUTTON(31), /* erase */ + MASCHINE_BUTTON(32), /* shift */ + MASCHINE_BUTTON(28), /* grid */ + MASCHINE_BUTTON(27), /* > */ + MASCHINE_BUTTON(26), /* < */ + MASCHINE_BUTTON(25), /* restart */ + + MASCHINE_BUTTON(21), /* E */ + MASCHINE_BUTTON(22), /* F */ + MASCHINE_BUTTON(23), /* G */ + MASCHINE_BUTTON(24), /* H */ + MASCHINE_BUTTON(20), /* D */ + MASCHINE_BUTTON(19), /* C */ + MASCHINE_BUTTON(18), /* B */ + MASCHINE_BUTTON(17), /* A */ + + MASCHINE_BUTTON(0), /* control */ + MASCHINE_BUTTON(2), /* browse */ + MASCHINE_BUTTON(4), /* < */ + MASCHINE_BUTTON(6), /* snap */ + MASCHINE_BUTTON(7), /* autowrite */ + MASCHINE_BUTTON(5), /* > */ + MASCHINE_BUTTON(3), /* sampling */ + MASCHINE_BUTTON(1), /* step */ + + MASCHINE_BUTTON(15), /* 8 softkeys */ + MASCHINE_BUTTON(14), + MASCHINE_BUTTON(13), + MASCHINE_BUTTON(12), + MASCHINE_BUTTON(11), + MASCHINE_BUTTON(10), + MASCHINE_BUTTON(9), + MASCHINE_BUTTON(8), + + MASCHINE_BUTTON(16), /* note repeat */ + MASCHINE_BUTTON(29) /* play */ +}; + #define KONTROLX1_INPUTS (40) #define KONTROLS4_BUTTONS (12 * 8) #define KONTROLS4_AXIS (46) @@ -218,6 +273,29 @@ static void snd_caiaq_input_read_erp(struct snd_usb_caiaqdev *dev, input_report_abs(input_dev, ABS_HAT3Y, i); input_sync(input_dev); break; + + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_MASCHINECONTROLLER): + /* 4 under the left screen */ + input_report_abs(input_dev, ABS_HAT0X, decode_erp(buf[21], buf[20])); + input_report_abs(input_dev, ABS_HAT0Y, decode_erp(buf[15], buf[14])); + input_report_abs(input_dev, ABS_HAT1X, decode_erp(buf[9], buf[8])); + input_report_abs(input_dev, ABS_HAT1Y, decode_erp(buf[3], buf[2])); + + /* 4 under the right screen */ + input_report_abs(input_dev, ABS_HAT2X, decode_erp(buf[19], buf[18])); + input_report_abs(input_dev, ABS_HAT2Y, decode_erp(buf[13], buf[12])); + input_report_abs(input_dev, ABS_HAT3X, decode_erp(buf[7], buf[6])); + input_report_abs(input_dev, ABS_HAT3Y, decode_erp(buf[1], buf[0])); + + /* volume */ + input_report_abs(input_dev, ABS_RX, decode_erp(buf[17], buf[16])); + /* tempo */ + input_report_abs(input_dev, ABS_RY, decode_erp(buf[11], buf[10])); + /* swing */ + input_report_abs(input_dev, ABS_RZ, decode_erp(buf[5], buf[4])); + + input_sync(input_dev); + break; } } @@ -400,6 +478,25 @@ static void snd_usb_caiaq_tks4_dispatch(struct snd_usb_caiaqdev *dev, input_sync(dev->input_dev); } +#define MASCHINE_MSGBLOCK_SIZE 2 + +static void snd_usb_caiaq_maschine_dispatch(struct snd_usb_caiaqdev *dev, + const unsigned char *buf, + unsigned int len) +{ + unsigned int i, pad_id; + uint16_t pressure; + + for (i = 0; i < MASCHINE_PADS; i++) { + pressure = be16_to_cpu(buf[i * 2] << 8 | buf[(i * 2) + 1]); + pad_id = pressure >> 12; + + input_report_abs(dev->input_dev, MASCHINE_PAD(pad_id), pressure & 0xfff); + } + + input_sync(dev->input_dev); +} + static void snd_usb_caiaq_ep4_reply_dispatch(struct urb *urb) { struct snd_usb_caiaqdev *dev = urb->context; @@ -425,6 +522,13 @@ static void snd_usb_caiaq_ep4_reply_dispatch(struct urb *urb) case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLS4): snd_usb_caiaq_tks4_dispatch(dev, buf, urb->actual_length); break; + + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_MASCHINECONTROLLER): + if (urb->actual_length < (MASCHINE_PADS * MASCHINE_MSGBLOCK_SIZE)) + goto requeue; + + snd_usb_caiaq_maschine_dispatch(dev, buf, urb->actual_length); + break; } requeue: @@ -444,6 +548,7 @@ static int snd_usb_caiaq_input_open(struct input_dev *idev) switch (dev->chip.usb_id) { case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLX1): case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLS4): + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_MASCHINECONTROLLER): if (usb_submit_urb(dev->ep4_in_urb, GFP_KERNEL) != 0) return -EIO; break; @@ -462,6 +567,7 @@ static void snd_usb_caiaq_input_close(struct input_dev *idev) switch (dev->chip.usb_id) { case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLX1): case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLS4): + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_MASCHINECONTROLLER): usb_kill_urb(dev->ep4_in_urb); break; } @@ -652,6 +758,50 @@ int snd_usb_caiaq_input_init(struct snd_usb_caiaqdev *dev) break; + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_MASCHINECONTROLLER): + input->evbit[0] = BIT_MASK(EV_KEY) | BIT_MASK(EV_ABS); + input->absbit[0] = BIT_MASK(ABS_HAT0X) | BIT_MASK(ABS_HAT0Y) | + BIT_MASK(ABS_HAT1X) | BIT_MASK(ABS_HAT1Y) | + BIT_MASK(ABS_HAT2X) | BIT_MASK(ABS_HAT2Y) | + BIT_MASK(ABS_HAT3X) | BIT_MASK(ABS_HAT3Y) | + BIT_MASK(ABS_RX) | BIT_MASK(ABS_RY) | + BIT_MASK(ABS_RZ); + + BUILD_BUG_ON(sizeof(dev->keycode) < sizeof(keycode_maschine)); + memcpy(dev->keycode, keycode_maschine, sizeof(keycode_maschine)); + input->keycodemax = ARRAY_SIZE(keycode_maschine); + + for (i = 0; i < MASCHINE_PADS; i++) { + input->absbit[0] |= MASCHINE_PAD(i); + input_set_abs_params(input, MASCHINE_PAD(i), 0, 0xfff, 5, 10); + } + + input_set_abs_params(input, ABS_HAT0X, 0, 999, 0, 10); + input_set_abs_params(input, ABS_HAT0Y, 0, 999, 0, 10); + input_set_abs_params(input, ABS_HAT1X, 0, 999, 0, 10); + input_set_abs_params(input, ABS_HAT1Y, 0, 999, 0, 10); + input_set_abs_params(input, ABS_HAT2X, 0, 999, 0, 10); + input_set_abs_params(input, ABS_HAT2Y, 0, 999, 0, 10); + input_set_abs_params(input, ABS_HAT3X, 0, 999, 0, 10); + input_set_abs_params(input, ABS_HAT3Y, 0, 999, 0, 10); + input_set_abs_params(input, ABS_RX, 0, 999, 0, 10); + input_set_abs_params(input, ABS_RY, 0, 999, 0, 10); + input_set_abs_params(input, ABS_RZ, 0, 999, 0, 10); + + dev->ep4_in_urb = usb_alloc_urb(0, GFP_KERNEL); + if (!dev->ep4_in_urb) { + ret = -ENOMEM; + goto exit_free_idev; + } + + usb_fill_bulk_urb(dev->ep4_in_urb, usb_dev, + usb_rcvbulkpipe(usb_dev, 0x4), + dev->ep4_in_buf, EP4_BUFSIZE, + snd_usb_caiaq_ep4_reply_dispatch, dev); + + snd_usb_caiaq_set_auto_msg(dev, 1, 10, 5); + break; + default: /* no input methods supported on this device */ goto exit_free_idev; @@ -664,15 +814,17 @@ int snd_usb_caiaq_input_init(struct snd_usb_caiaqdev *dev) for (i = 0; i < input->keycodemax; i++) __set_bit(dev->keycode[i], input->keybit); + dev->input_dev = input; + ret = input_register_device(input); if (ret < 0) goto exit_free_idev; - dev->input_dev = input; return 0; exit_free_idev: input_free_device(input); + dev->input_dev = NULL; return ret; } @@ -688,4 +840,3 @@ void snd_usb_caiaq_input_free(struct snd_usb_caiaqdev *dev) input_unregister_device(dev->input_dev); dev->input_dev = NULL; } - diff --git a/sound/usb/card.c b/sound/usb/card.c index 3068f043099a..05c1aae0b010 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -65,9 +65,9 @@ #include "helper.h" #include "debug.h" #include "pcm.h" -#include "urb.h" #include "format.h" #include "power.h" +#include "stream.h" MODULE_AUTHOR("Takashi Iwai <tiwai@suse.de>"); MODULE_DESCRIPTION("USB Audio"); @@ -185,7 +185,7 @@ static int snd_usb_create_stream(struct snd_usb_audio *chip, int ctrlif, int int return -EINVAL; } - if (! snd_usb_parse_audio_endpoints(chip, interface)) { + if (! snd_usb_parse_audio_interface(chip, interface)) { usb_set_interface(dev, interface, 0); /* reset the current interface */ usb_driver_claim_interface(&usb_audio_driver, iface, (void *)-1L); return -EINVAL; diff --git a/sound/usb/card.h b/sound/usb/card.h index ae4251d5abf7..a39edcc32a93 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -94,6 +94,8 @@ struct snd_usb_substream { spinlock_t lock; struct snd_urb_ops ops; /* callbacks (must be filled at init) */ + int last_frame_number; /* stored frame number */ + int last_delay; /* stored delay */ }; struct snd_usb_stream { diff --git a/sound/usb/clock.c b/sound/usb/clock.c index 075195e8661a..379baad3d5ad 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -91,7 +91,7 @@ static int uac_clock_selector_get_val(struct snd_usb_audio *chip, int selector_i USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, UAC2_CX_CLOCK_SELECTOR << 8, snd_usb_ctrl_intf(chip) | (selector_id << 8), - &buf, sizeof(buf), 1000); + &buf, sizeof(buf)); if (ret < 0) return ret; @@ -118,7 +118,7 @@ static bool uac_clock_source_is_valid(struct snd_usb_audio *chip, int source_id) USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN, UAC2_CS_CONTROL_CLOCK_VALID << 8, snd_usb_ctrl_intf(chip) | (source_id << 8), - &data, sizeof(data), 1000); + &data, sizeof(data)); if (err < 0) { snd_printk(KERN_WARNING "%s(): cannot get clock validity for id %d\n", @@ -222,7 +222,7 @@ static int set_sample_rate_v1(struct snd_usb_audio *chip, int iface, if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC_SET_CUR, USB_TYPE_CLASS | USB_RECIP_ENDPOINT | USB_DIR_OUT, UAC_EP_CS_ATTR_SAMPLE_RATE << 8, ep, - data, sizeof(data), 1000)) < 0) { + data, sizeof(data))) < 0) { snd_printk(KERN_ERR "%d:%d:%d: cannot set freq %d to ep %#x\n", dev->devnum, iface, fmt->altsetting, rate, ep); return err; @@ -231,7 +231,7 @@ static int set_sample_rate_v1(struct snd_usb_audio *chip, int iface, if ((err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC_GET_CUR, USB_TYPE_CLASS | USB_RECIP_ENDPOINT | USB_DIR_IN, UAC_EP_CS_ATTR_SAMPLE_RATE << 8, ep, - data, sizeof(data), 1000)) < 0) { + data, sizeof(data))) < 0) { snd_printk(KERN_WARNING "%d:%d:%d: cannot get freq at ep %#x\n", dev->devnum, iface, fmt->altsetting, ep); return 0; /* some devices don't support reading */ @@ -273,7 +273,7 @@ static int set_sample_rate_v2(struct snd_usb_audio *chip, int iface, USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_OUT, UAC2_CS_CONTROL_SAM_FREQ << 8, snd_usb_ctrl_intf(chip) | (clock << 8), - data, sizeof(data), 1000)) < 0) { + data, sizeof(data))) < 0) { snd_printk(KERN_ERR "%d:%d:%d: cannot set freq %d (v2)\n", dev->devnum, iface, fmt->altsetting, rate); return err; @@ -283,7 +283,7 @@ static int set_sample_rate_v2(struct snd_usb_audio *chip, int iface, USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN, UAC2_CS_CONTROL_SAM_FREQ << 8, snd_usb_ctrl_intf(chip) | (clock << 8), - data, sizeof(data), 1000)) < 0) { + data, sizeof(data))) < 0) { snd_printk(KERN_WARNING "%d:%d:%d: cannot get freq (v2)\n", dev->devnum, iface, fmt->altsetting); return err; diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 7d46e482375d..81c6edecd862 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -15,436 +15,951 @@ * */ +#include <linux/gfp.h> #include <linux/init.h> -#include <linux/slab.h> #include <linux/usb.h> #include <linux/usb/audio.h> -#include <linux/usb/audio-v2.h> #include <sound/core.h> #include <sound/pcm.h> #include "usbaudio.h" +#include "helper.h" #include "card.h" -#include "proc.h" -#include "quirks.h" #include "endpoint.h" -#include "urb.h" #include "pcm.h" -#include "helper.h" -#include "format.h" -#include "clock.h" /* - * free a substream + * convert a sampling rate into our full speed format (fs/1000 in Q16.16) + * this will overflow at approx 524 kHz */ -static void free_substream(struct snd_usb_substream *subs) +static inline unsigned get_usb_full_speed_rate(unsigned int rate) { - struct list_head *p, *n; - - if (!subs->num_formats) - return; /* not initialized */ - list_for_each_safe(p, n, &subs->fmt_list) { - struct audioformat *fp = list_entry(p, struct audioformat, list); - kfree(fp->rate_table); - kfree(fp); - } - kfree(subs->rate_list.list); + return ((rate << 13) + 62) / 125; } +/* + * convert a sampling rate into USB high speed format (fs/8000 in Q16.16) + * this will overflow at approx 4 MHz + */ +static inline unsigned get_usb_high_speed_rate(unsigned int rate) +{ + return ((rate << 10) + 62) / 125; +} /* - * free a usb stream instance + * unlink active urbs. */ -static void snd_usb_audio_stream_free(struct snd_usb_stream *stream) +static int deactivate_urbs(struct snd_usb_substream *subs, int force, int can_sleep) { - free_substream(&stream->substream[0]); - free_substream(&stream->substream[1]); - list_del(&stream->list); - kfree(stream); + struct snd_usb_audio *chip = subs->stream->chip; + unsigned int i; + int async; + + subs->running = 0; + + if (!force && subs->stream->chip->shutdown) /* to be sure... */ + return -EBADFD; + + async = !can_sleep && chip->async_unlink; + + if (!async && in_interrupt()) + return 0; + + for (i = 0; i < subs->nurbs; i++) { + if (test_bit(i, &subs->active_mask)) { + if (!test_and_set_bit(i, &subs->unlink_mask)) { + struct urb *u = subs->dataurb[i].urb; + if (async) + usb_unlink_urb(u); + else + usb_kill_urb(u); + } + } + } + if (subs->syncpipe) { + for (i = 0; i < SYNC_URBS; i++) { + if (test_bit(i+16, &subs->active_mask)) { + if (!test_and_set_bit(i+16, &subs->unlink_mask)) { + struct urb *u = subs->syncurb[i].urb; + if (async) + usb_unlink_urb(u); + else + usb_kill_urb(u); + } + } + } + } + return 0; } -static void snd_usb_audio_pcm_free(struct snd_pcm *pcm) + +/* + * release a urb data + */ +static void release_urb_ctx(struct snd_urb_ctx *u) { - struct snd_usb_stream *stream = pcm->private_data; - if (stream) { - stream->pcm = NULL; - snd_usb_audio_stream_free(stream); + if (u->urb) { + if (u->buffer_size) + usb_free_coherent(u->subs->dev, u->buffer_size, + u->urb->transfer_buffer, + u->urb->transfer_dma); + usb_free_urb(u->urb); + u->urb = NULL; } } +/* + * wait until all urbs are processed. + */ +static int wait_clear_urbs(struct snd_usb_substream *subs) +{ + unsigned long end_time = jiffies + msecs_to_jiffies(1000); + unsigned int i; + int alive; + + do { + alive = 0; + for (i = 0; i < subs->nurbs; i++) { + if (test_bit(i, &subs->active_mask)) + alive++; + } + if (subs->syncpipe) { + for (i = 0; i < SYNC_URBS; i++) { + if (test_bit(i + 16, &subs->active_mask)) + alive++; + } + } + if (! alive) + break; + schedule_timeout_uninterruptible(1); + } while (time_before(jiffies, end_time)); + if (alive) + snd_printk(KERN_ERR "timeout: still %d active urbs..\n", alive); + return 0; +} /* - * add this endpoint to the chip instance. - * if a stream with the same endpoint already exists, append to it. - * if not, create a new pcm stream. + * release a substream */ -int snd_usb_add_audio_endpoint(struct snd_usb_audio *chip, int stream, struct audioformat *fp) +void snd_usb_release_substream_urbs(struct snd_usb_substream *subs, int force) { - struct list_head *p; - struct snd_usb_stream *as; - struct snd_usb_substream *subs; - struct snd_pcm *pcm; - int err; + int i; + + /* stop urbs (to be sure) */ + deactivate_urbs(subs, force, 1); + wait_clear_urbs(subs); + + for (i = 0; i < MAX_URBS; i++) + release_urb_ctx(&subs->dataurb[i]); + for (i = 0; i < SYNC_URBS; i++) + release_urb_ctx(&subs->syncurb[i]); + usb_free_coherent(subs->dev, SYNC_URBS * 4, + subs->syncbuf, subs->sync_dma); + subs->syncbuf = NULL; + subs->nurbs = 0; +} - list_for_each(p, &chip->pcm_list) { - as = list_entry(p, struct snd_usb_stream, list); - if (as->fmt_type != fp->fmt_type) - continue; - subs = &as->substream[stream]; - if (!subs->endpoint) - continue; - if (subs->endpoint == fp->endpoint) { - list_add_tail(&fp->list, &subs->fmt_list); - subs->num_formats++; - subs->formats |= fp->formats; - return 0; +/* + * complete callback from data urb + */ +static void snd_complete_urb(struct urb *urb) +{ + struct snd_urb_ctx *ctx = urb->context; + struct snd_usb_substream *subs = ctx->subs; + struct snd_pcm_substream *substream = ctx->subs->pcm_substream; + int err = 0; + + if ((subs->running && subs->ops.retire(subs, substream->runtime, urb)) || + !subs->running || /* can be stopped during retire callback */ + (err = subs->ops.prepare(subs, substream->runtime, urb)) < 0 || + (err = usb_submit_urb(urb, GFP_ATOMIC)) < 0) { + clear_bit(ctx->index, &subs->active_mask); + if (err < 0) { + snd_printd(KERN_ERR "cannot submit urb (err = %d)\n", err); + snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); } } - /* look for an empty stream */ - list_for_each(p, &chip->pcm_list) { - as = list_entry(p, struct snd_usb_stream, list); - if (as->fmt_type != fp->fmt_type) - continue; - subs = &as->substream[stream]; - if (subs->endpoint) - continue; - err = snd_pcm_new_stream(as->pcm, stream, 1); - if (err < 0) - return err; - snd_usb_init_substream(as, stream, fp); - return 0; +} + + +/* + * complete callback from sync urb + */ +static void snd_complete_sync_urb(struct urb *urb) +{ + struct snd_urb_ctx *ctx = urb->context; + struct snd_usb_substream *subs = ctx->subs; + struct snd_pcm_substream *substream = ctx->subs->pcm_substream; + int err = 0; + + if ((subs->running && subs->ops.retire_sync(subs, substream->runtime, urb)) || + !subs->running || /* can be stopped during retire callback */ + (err = subs->ops.prepare_sync(subs, substream->runtime, urb)) < 0 || + (err = usb_submit_urb(urb, GFP_ATOMIC)) < 0) { + clear_bit(ctx->index + 16, &subs->active_mask); + if (err < 0) { + snd_printd(KERN_ERR "cannot submit sync urb (err = %d)\n", err); + snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); + } } +} + - /* create a new pcm */ - as = kzalloc(sizeof(*as), GFP_KERNEL); - if (!as) - return -ENOMEM; - as->pcm_index = chip->pcm_devs; - as->chip = chip; - as->fmt_type = fp->fmt_type; - err = snd_pcm_new(chip->card, "USB Audio", chip->pcm_devs, - stream == SNDRV_PCM_STREAM_PLAYBACK ? 1 : 0, - stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1, - &pcm); - if (err < 0) { - kfree(as); - return err; +/* + * initialize a substream for plaback/capture + */ +int snd_usb_init_substream_urbs(struct snd_usb_substream *subs, + unsigned int period_bytes, + unsigned int rate, + unsigned int frame_bits) +{ + unsigned int maxsize, i; + int is_playback = subs->direction == SNDRV_PCM_STREAM_PLAYBACK; + unsigned int urb_packs, total_packs, packs_per_ms; + struct snd_usb_audio *chip = subs->stream->chip; + + /* calculate the frequency in 16.16 format */ + if (snd_usb_get_speed(subs->dev) == USB_SPEED_FULL) + subs->freqn = get_usb_full_speed_rate(rate); + else + subs->freqn = get_usb_high_speed_rate(rate); + subs->freqm = subs->freqn; + subs->freqshift = INT_MIN; + /* calculate max. frequency */ + if (subs->maxpacksize) { + /* whatever fits into a max. size packet */ + maxsize = subs->maxpacksize; + subs->freqmax = (maxsize / (frame_bits >> 3)) + << (16 - subs->datainterval); + } else { + /* no max. packet size: just take 25% higher than nominal */ + subs->freqmax = subs->freqn + (subs->freqn >> 2); + maxsize = ((subs->freqmax + 0xffff) * (frame_bits >> 3)) + >> (16 - subs->datainterval); } - as->pcm = pcm; - pcm->private_data = as; - pcm->private_free = snd_usb_audio_pcm_free; - pcm->info_flags = 0; - if (chip->pcm_devs > 0) - sprintf(pcm->name, "USB Audio #%d", chip->pcm_devs); + subs->phase = 0; + + if (subs->fill_max) + subs->curpacksize = subs->maxpacksize; else - strcpy(pcm->name, "USB Audio"); + subs->curpacksize = maxsize; - snd_usb_init_substream(as, stream, fp); + if (snd_usb_get_speed(subs->dev) != USB_SPEED_FULL) + packs_per_ms = 8 >> subs->datainterval; + else + packs_per_ms = 1; + + if (is_playback) { + urb_packs = max(chip->nrpacks, 1); + urb_packs = min(urb_packs, (unsigned int)MAX_PACKS); + } else + urb_packs = 1; + urb_packs *= packs_per_ms; + if (subs->syncpipe) + urb_packs = min(urb_packs, 1U << subs->syncinterval); + + /* decide how many packets to be used */ + if (is_playback) { + unsigned int minsize, maxpacks; + /* determine how small a packet can be */ + minsize = (subs->freqn >> (16 - subs->datainterval)) + * (frame_bits >> 3); + /* with sync from device, assume it can be 12% lower */ + if (subs->syncpipe) + minsize -= minsize >> 3; + minsize = max(minsize, 1u); + total_packs = (period_bytes + minsize - 1) / minsize; + /* we need at least two URBs for queueing */ + if (total_packs < 2) { + total_packs = 2; + } else { + /* and we don't want too long a queue either */ + maxpacks = max(MAX_QUEUE * packs_per_ms, urb_packs * 2); + total_packs = min(total_packs, maxpacks); + } + } else { + while (urb_packs > 1 && urb_packs * maxsize >= period_bytes) + urb_packs >>= 1; + total_packs = MAX_URBS * urb_packs; + } + subs->nurbs = (total_packs + urb_packs - 1) / urb_packs; + if (subs->nurbs > MAX_URBS) { + /* too much... */ + subs->nurbs = MAX_URBS; + total_packs = MAX_URBS * urb_packs; + } else if (subs->nurbs < 2) { + /* too little - we need at least two packets + * to ensure contiguous playback/capture + */ + subs->nurbs = 2; + } - list_add(&as->list, &chip->pcm_list); - chip->pcm_devs++; + /* allocate and initialize data urbs */ + for (i = 0; i < subs->nurbs; i++) { + struct snd_urb_ctx *u = &subs->dataurb[i]; + u->index = i; + u->subs = subs; + u->packets = (i + 1) * total_packs / subs->nurbs + - i * total_packs / subs->nurbs; + u->buffer_size = maxsize * u->packets; + if (subs->fmt_type == UAC_FORMAT_TYPE_II) + u->packets++; /* for transfer delimiter */ + u->urb = usb_alloc_urb(u->packets, GFP_KERNEL); + if (!u->urb) + goto out_of_memory; + u->urb->transfer_buffer = + usb_alloc_coherent(subs->dev, u->buffer_size, + GFP_KERNEL, &u->urb->transfer_dma); + if (!u->urb->transfer_buffer) + goto out_of_memory; + u->urb->pipe = subs->datapipe; + u->urb->transfer_flags = URB_ISO_ASAP | URB_NO_TRANSFER_DMA_MAP; + u->urb->interval = 1 << subs->datainterval; + u->urb->context = u; + u->urb->complete = snd_complete_urb; + } + + if (subs->syncpipe) { + /* allocate and initialize sync urbs */ + subs->syncbuf = usb_alloc_coherent(subs->dev, SYNC_URBS * 4, + GFP_KERNEL, &subs->sync_dma); + if (!subs->syncbuf) + goto out_of_memory; + for (i = 0; i < SYNC_URBS; i++) { + struct snd_urb_ctx *u = &subs->syncurb[i]; + u->index = i; + u->subs = subs; + u->packets = 1; + u->urb = usb_alloc_urb(1, GFP_KERNEL); + if (!u->urb) + goto out_of_memory; + u->urb->transfer_buffer = subs->syncbuf + i * 4; + u->urb->transfer_dma = subs->sync_dma + i * 4; + u->urb->transfer_buffer_length = 4; + u->urb->pipe = subs->syncpipe; + u->urb->transfer_flags = URB_ISO_ASAP | + URB_NO_TRANSFER_DMA_MAP; + u->urb->number_of_packets = 1; + u->urb->interval = 1 << subs->syncinterval; + u->urb->context = u; + u->urb->complete = snd_complete_sync_urb; + } + } + return 0; - snd_usb_proc_pcm_format_add(as); +out_of_memory: + snd_usb_release_substream_urbs(subs, 0); + return -ENOMEM; +} +/* + * prepare urb for full speed capture sync pipe + * + * fill the length and offset of each urb descriptor. + * the fixed 10.14 frequency is passed through the pipe. + */ +static int prepare_capture_sync_urb(struct snd_usb_substream *subs, + struct snd_pcm_runtime *runtime, + struct urb *urb) +{ + unsigned char *cp = urb->transfer_buffer; + struct snd_urb_ctx *ctx = urb->context; + + urb->dev = ctx->subs->dev; /* we need to set this at each time */ + urb->iso_frame_desc[0].length = 3; + urb->iso_frame_desc[0].offset = 0; + cp[0] = subs->freqn >> 2; + cp[1] = subs->freqn >> 10; + cp[2] = subs->freqn >> 18; return 0; } -static int parse_uac_endpoint_attributes(struct snd_usb_audio *chip, - struct usb_host_interface *alts, - int protocol, int iface_no) +/* + * prepare urb for high speed capture sync pipe + * + * fill the length and offset of each urb descriptor. + * the fixed 12.13 frequency is passed as 16.16 through the pipe. + */ +static int prepare_capture_sync_urb_hs(struct snd_usb_substream *subs, + struct snd_pcm_runtime *runtime, + struct urb *urb) { - /* parsed with a v1 header here. that's ok as we only look at the - * header first which is the same for both versions */ - struct uac_iso_endpoint_descriptor *csep; - struct usb_interface_descriptor *altsd = get_iface_desc(alts); - int attributes = 0; - - csep = snd_usb_find_desc(alts->endpoint[0].extra, alts->endpoint[0].extralen, NULL, USB_DT_CS_ENDPOINT); - - /* Creamware Noah has this descriptor after the 2nd endpoint */ - if (!csep && altsd->bNumEndpoints >= 2) - csep = snd_usb_find_desc(alts->endpoint[1].extra, alts->endpoint[1].extralen, NULL, USB_DT_CS_ENDPOINT); - - if (!csep || csep->bLength < 7 || - csep->bDescriptorSubtype != UAC_EP_GENERAL) { - snd_printk(KERN_WARNING "%d:%u:%d : no or invalid" - " class specific endpoint descriptor\n", - chip->dev->devnum, iface_no, - altsd->bAlternateSetting); - return 0; - } + unsigned char *cp = urb->transfer_buffer; + struct snd_urb_ctx *ctx = urb->context; + + urb->dev = ctx->subs->dev; /* we need to set this at each time */ + urb->iso_frame_desc[0].length = 4; + urb->iso_frame_desc[0].offset = 0; + cp[0] = subs->freqn; + cp[1] = subs->freqn >> 8; + cp[2] = subs->freqn >> 16; + cp[3] = subs->freqn >> 24; + return 0; +} - if (protocol == UAC_VERSION_1) { - attributes = csep->bmAttributes; - } else { - struct uac2_iso_endpoint_descriptor *csep2 = - (struct uac2_iso_endpoint_descriptor *) csep; +/* + * process after capture sync complete + * - nothing to do + */ +static int retire_capture_sync_urb(struct snd_usb_substream *subs, + struct snd_pcm_runtime *runtime, + struct urb *urb) +{ + return 0; +} - attributes = csep->bmAttributes & UAC_EP_CS_ATTR_FILL_MAX; +/* + * prepare urb for capture data pipe + * + * fill the offset and length of each descriptor. + * + * we use a temporary buffer to write the captured data. + * since the length of written data is determined by host, we cannot + * write onto the pcm buffer directly... the data is thus copied + * later at complete callback to the global buffer. + */ +static int prepare_capture_urb(struct snd_usb_substream *subs, + struct snd_pcm_runtime *runtime, + struct urb *urb) +{ + int i, offs; + struct snd_urb_ctx *ctx = urb->context; + + offs = 0; + urb->dev = ctx->subs->dev; /* we need to set this at each time */ + for (i = 0; i < ctx->packets; i++) { + urb->iso_frame_desc[i].offset = offs; + urb->iso_frame_desc[i].length = subs->curpacksize; + offs += subs->curpacksize; + } + urb->transfer_buffer_length = offs; + urb->number_of_packets = ctx->packets; + return 0; +} - /* emulate the endpoint attributes of a v1 device */ - if (csep2->bmControls & UAC2_CONTROL_PITCH) - attributes |= UAC_EP_CS_ATTR_PITCH_CONTROL; +/* + * process after capture complete + * + * copy the data from each desctiptor to the pcm buffer, and + * update the current position. + */ +static int retire_capture_urb(struct snd_usb_substream *subs, + struct snd_pcm_runtime *runtime, + struct urb *urb) +{ + unsigned long flags; + unsigned char *cp; + int i; + unsigned int stride, frames, bytes, oldptr; + int period_elapsed = 0; + + stride = runtime->frame_bits >> 3; + + for (i = 0; i < urb->number_of_packets; i++) { + cp = (unsigned char *)urb->transfer_buffer + urb->iso_frame_desc[i].offset; + if (urb->iso_frame_desc[i].status) { + snd_printd(KERN_ERR "frame %d active: %d\n", i, urb->iso_frame_desc[i].status); + // continue; + } + bytes = urb->iso_frame_desc[i].actual_length; + frames = bytes / stride; + if (!subs->txfr_quirk) + bytes = frames * stride; + if (bytes % (runtime->sample_bits >> 3) != 0) { +#ifdef CONFIG_SND_DEBUG_VERBOSE + int oldbytes = bytes; +#endif + bytes = frames * stride; + snd_printdd(KERN_ERR "Corrected urb data len. %d->%d\n", + oldbytes, bytes); + } + /* update the current pointer */ + spin_lock_irqsave(&subs->lock, flags); + oldptr = subs->hwptr_done; + subs->hwptr_done += bytes; + if (subs->hwptr_done >= runtime->buffer_size * stride) + subs->hwptr_done -= runtime->buffer_size * stride; + frames = (bytes + (oldptr % stride)) / stride; + subs->transfer_done += frames; + if (subs->transfer_done >= runtime->period_size) { + subs->transfer_done -= runtime->period_size; + period_elapsed = 1; + } + spin_unlock_irqrestore(&subs->lock, flags); + /* copy a data chunk */ + if (oldptr + bytes > runtime->buffer_size * stride) { + unsigned int bytes1 = + runtime->buffer_size * stride - oldptr; + memcpy(runtime->dma_area + oldptr, cp, bytes1); + memcpy(runtime->dma_area, cp + bytes1, bytes - bytes1); + } else { + memcpy(runtime->dma_area + oldptr, cp, bytes); + } } + if (period_elapsed) + snd_pcm_period_elapsed(subs->pcm_substream); + return 0; +} - return attributes; +/* + * Process after capture complete when paused. Nothing to do. + */ +static int retire_paused_capture_urb(struct snd_usb_substream *subs, + struct snd_pcm_runtime *runtime, + struct urb *urb) +{ + return 0; } -static struct uac2_input_terminal_descriptor * - snd_usb_find_input_terminal_descriptor(struct usb_host_interface *ctrl_iface, - int terminal_id) + +/* + * prepare urb for playback sync pipe + * + * set up the offset and length to receive the current frequency. + */ +static int prepare_playback_sync_urb(struct snd_usb_substream *subs, + struct snd_pcm_runtime *runtime, + struct urb *urb) { - struct uac2_input_terminal_descriptor *term = NULL; + struct snd_urb_ctx *ctx = urb->context; + + urb->dev = ctx->subs->dev; /* we need to set this at each time */ + urb->iso_frame_desc[0].length = min(4u, ctx->subs->syncmaxsize); + urb->iso_frame_desc[0].offset = 0; + return 0; +} - while ((term = snd_usb_find_csint_desc(ctrl_iface->extra, - ctrl_iface->extralen, - term, UAC_INPUT_TERMINAL))) { - if (term->bTerminalID == terminal_id) - return term; +/* + * process after playback sync complete + * + * Full speed devices report feedback values in 10.14 format as samples per + * frame, high speed devices in 16.16 format as samples per microframe. + * Because the Audio Class 1 spec was written before USB 2.0, many high speed + * devices use a wrong interpretation, some others use an entirely different + * format. Therefore, we cannot predict what format any particular device uses + * and must detect it automatically. + */ +static int retire_playback_sync_urb(struct snd_usb_substream *subs, + struct snd_pcm_runtime *runtime, + struct urb *urb) +{ + unsigned int f; + int shift; + unsigned long flags; + + if (urb->iso_frame_desc[0].status != 0 || + urb->iso_frame_desc[0].actual_length < 3) + return 0; + + f = le32_to_cpup(urb->transfer_buffer); + if (urb->iso_frame_desc[0].actual_length == 3) + f &= 0x00ffffff; + else + f &= 0x0fffffff; + if (f == 0) + return 0; + + if (unlikely(subs->freqshift == INT_MIN)) { + /* + * The first time we see a feedback value, determine its format + * by shifting it left or right until it matches the nominal + * frequency value. This assumes that the feedback does not + * differ from the nominal value more than +50% or -25%. + */ + shift = 0; + while (f < subs->freqn - subs->freqn / 4) { + f <<= 1; + shift++; + } + while (f > subs->freqn + subs->freqn / 2) { + f >>= 1; + shift--; + } + subs->freqshift = shift; + } + else if (subs->freqshift >= 0) + f <<= subs->freqshift; + else + f >>= -subs->freqshift; + + if (likely(f >= subs->freqn - subs->freqn / 8 && f <= subs->freqmax)) { + /* + * If the frequency looks valid, set it. + * This value is referred to in prepare_playback_urb(). + */ + spin_lock_irqsave(&subs->lock, flags); + subs->freqm = f; + spin_unlock_irqrestore(&subs->lock, flags); + } else { + /* + * Out of range; maybe the shift value is wrong. + * Reset it so that we autodetect again the next time. + */ + subs->freqshift = INT_MIN; } - return NULL; + return 0; } -static struct uac2_output_terminal_descriptor * - snd_usb_find_output_terminal_descriptor(struct usb_host_interface *ctrl_iface, - int terminal_id) +/* determine the number of frames in the next packet */ +static int snd_usb_audio_next_packet_size(struct snd_usb_substream *subs) { - struct uac2_output_terminal_descriptor *term = NULL; - - while ((term = snd_usb_find_csint_desc(ctrl_iface->extra, - ctrl_iface->extralen, - term, UAC_OUTPUT_TERMINAL))) { - if (term->bTerminalID == terminal_id) - return term; + if (subs->fill_max) + return subs->maxframesize; + else { + subs->phase = (subs->phase & 0xffff) + + (subs->freqm << subs->datainterval); + return min(subs->phase >> 16, subs->maxframesize); } +} - return NULL; +/* + * Prepare urb for streaming before playback starts or when paused. + * + * We don't have any data, so we send silence. + */ +static int prepare_nodata_playback_urb(struct snd_usb_substream *subs, + struct snd_pcm_runtime *runtime, + struct urb *urb) +{ + unsigned int i, offs, counts; + struct snd_urb_ctx *ctx = urb->context; + int stride = runtime->frame_bits >> 3; + + offs = 0; + urb->dev = ctx->subs->dev; + for (i = 0; i < ctx->packets; ++i) { + counts = snd_usb_audio_next_packet_size(subs); + urb->iso_frame_desc[i].offset = offs * stride; + urb->iso_frame_desc[i].length = counts * stride; + offs += counts; + } + urb->number_of_packets = ctx->packets; + urb->transfer_buffer_length = offs * stride; + memset(urb->transfer_buffer, + runtime->format == SNDRV_PCM_FORMAT_U8 ? 0x80 : 0, + offs * stride); + return 0; } -int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) +/* + * prepare urb for playback data pipe + * + * Since a URB can handle only a single linear buffer, we must use double + * buffering when the data to be transferred overflows the buffer boundary. + * To avoid inconsistencies when updating hwptr_done, we use double buffering + * for all URBs. + */ +static int prepare_playback_urb(struct snd_usb_substream *subs, + struct snd_pcm_runtime *runtime, + struct urb *urb) { - struct usb_device *dev; - struct usb_interface *iface; - struct usb_host_interface *alts; - struct usb_interface_descriptor *altsd; - int i, altno, err, stream; - int format = 0, num_channels = 0; - struct audioformat *fp = NULL; - int num, protocol, clock = 0; - struct uac_format_type_i_continuous_descriptor *fmt; + int i, stride; + unsigned int counts, frames, bytes; + unsigned long flags; + int period_elapsed = 0; + struct snd_urb_ctx *ctx = urb->context; + + stride = runtime->frame_bits >> 3; + + frames = 0; + urb->dev = ctx->subs->dev; /* we need to set this at each time */ + urb->number_of_packets = 0; + spin_lock_irqsave(&subs->lock, flags); + for (i = 0; i < ctx->packets; i++) { + counts = snd_usb_audio_next_packet_size(subs); + /* set up descriptor */ + urb->iso_frame_desc[i].offset = frames * stride; + urb->iso_frame_desc[i].length = counts * stride; + frames += counts; + urb->number_of_packets++; + subs->transfer_done += counts; + if (subs->transfer_done >= runtime->period_size) { + subs->transfer_done -= runtime->period_size; + period_elapsed = 1; + if (subs->fmt_type == UAC_FORMAT_TYPE_II) { + if (subs->transfer_done > 0) { + /* FIXME: fill-max mode is not + * supported yet */ + frames -= subs->transfer_done; + counts -= subs->transfer_done; + urb->iso_frame_desc[i].length = + counts * stride; + subs->transfer_done = 0; + } + i++; + if (i < ctx->packets) { + /* add a transfer delimiter */ + urb->iso_frame_desc[i].offset = + frames * stride; + urb->iso_frame_desc[i].length = 0; + urb->number_of_packets++; + } + break; + } + } + if (period_elapsed) /* finish at the period boundary */ + break; + } + bytes = frames * stride; + if (subs->hwptr_done + bytes > runtime->buffer_size * stride) { + /* err, the transferred area goes over buffer boundary. */ + unsigned int bytes1 = + runtime->buffer_size * stride - subs->hwptr_done; + memcpy(urb->transfer_buffer, + runtime->dma_area + subs->hwptr_done, bytes1); + memcpy(urb->transfer_buffer + bytes1, + runtime->dma_area, bytes - bytes1); + } else { + memcpy(urb->transfer_buffer, + runtime->dma_area + subs->hwptr_done, bytes); + } + subs->hwptr_done += bytes; + if (subs->hwptr_done >= runtime->buffer_size * stride) + subs->hwptr_done -= runtime->buffer_size * stride; + + /* update delay with exact number of samples queued */ + runtime->delay = subs->last_delay; + runtime->delay += frames; + subs->last_delay = runtime->delay; + + /* realign last_frame_number */ + subs->last_frame_number = usb_get_current_frame_number(subs->dev); + subs->last_frame_number &= 0xFF; /* keep 8 LSBs */ + + spin_unlock_irqrestore(&subs->lock, flags); + urb->transfer_buffer_length = bytes; + if (period_elapsed) + snd_pcm_period_elapsed(subs->pcm_substream); + return 0; +} - dev = chip->dev; +/* + * process after playback data complete + * - decrease the delay count again + */ +static int retire_playback_urb(struct snd_usb_substream *subs, + struct snd_pcm_runtime *runtime, + struct urb *urb) +{ + unsigned long flags; + int stride = runtime->frame_bits >> 3; + int processed = urb->transfer_buffer_length / stride; + int est_delay; - /* parse the interface's altsettings */ - iface = usb_ifnum_to_if(dev, iface_no); + spin_lock_irqsave(&subs->lock, flags); - num = iface->num_altsetting; + est_delay = snd_usb_pcm_delay(subs, runtime->rate); + /* update delay with exact number of samples played */ + if (processed > subs->last_delay) + subs->last_delay = 0; + else + subs->last_delay -= processed; + runtime->delay = subs->last_delay; /* - * Dallas DS4201 workaround: It presents 5 altsettings, but the last - * one misses syncpipe, and does not produce any sound. + * Report when delay estimate is off by more than 2ms. + * The error should be lower than 2ms since the estimate relies + * on two reads of a counter updated every ms. */ - if (chip->usb_id == USB_ID(0x04fa, 0x4201)) - num = 4; - - for (i = 0; i < num; i++) { - alts = &iface->altsetting[i]; - altsd = get_iface_desc(alts); - protocol = altsd->bInterfaceProtocol; - /* skip invalid one */ - if ((altsd->bInterfaceClass != USB_CLASS_AUDIO && - altsd->bInterfaceClass != USB_CLASS_VENDOR_SPEC) || - (altsd->bInterfaceSubClass != USB_SUBCLASS_AUDIOSTREAMING && - altsd->bInterfaceSubClass != USB_SUBCLASS_VENDOR_SPEC) || - altsd->bNumEndpoints < 1 || - le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize) == 0) - continue; - /* must be isochronous */ - if ((get_endpoint(alts, 0)->bmAttributes & USB_ENDPOINT_XFERTYPE_MASK) != - USB_ENDPOINT_XFER_ISOC) - continue; - /* check direction */ - stream = (get_endpoint(alts, 0)->bEndpointAddress & USB_DIR_IN) ? - SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK; - altno = altsd->bAlternateSetting; - - if (snd_usb_apply_interface_quirk(chip, iface_no, altno)) - continue; - - /* get audio formats */ - switch (protocol) { - default: - snd_printdd(KERN_WARNING "%d:%u:%d: unknown interface protocol %#02x, assuming v1\n", - dev->devnum, iface_no, altno, protocol); - protocol = UAC_VERSION_1; - /* fall through */ - - case UAC_VERSION_1: { - struct uac1_as_header_descriptor *as = - snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, UAC_AS_GENERAL); - - if (!as) { - snd_printk(KERN_ERR "%d:%u:%d : UAC_AS_GENERAL descriptor not found\n", - dev->devnum, iface_no, altno); - continue; - } + if (abs(est_delay - subs->last_delay) * 1000 > runtime->rate * 2) + snd_printk(KERN_DEBUG "delay: estimated %d, actual %d\n", + est_delay, subs->last_delay); - if (as->bLength < sizeof(*as)) { - snd_printk(KERN_ERR "%d:%u:%d : invalid UAC_AS_GENERAL desc\n", - dev->devnum, iface_no, altno); - continue; - } + spin_unlock_irqrestore(&subs->lock, flags); + return 0; +} - format = le16_to_cpu(as->wFormatTag); /* remember the format value */ - break; - } +static const char *usb_error_string(int err) +{ + switch (err) { + case -ENODEV: + return "no device"; + case -ENOENT: + return "endpoint not enabled"; + case -EPIPE: + return "endpoint stalled"; + case -ENOSPC: + return "not enough bandwidth"; + case -ESHUTDOWN: + return "device disabled"; + case -EHOSTUNREACH: + return "device suspended"; + case -EINVAL: + case -EAGAIN: + case -EFBIG: + case -EMSGSIZE: + return "internal error"; + default: + return "unknown error"; + } +} - case UAC_VERSION_2: { - struct uac2_input_terminal_descriptor *input_term; - struct uac2_output_terminal_descriptor *output_term; - struct uac2_as_header_descriptor *as = - snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, UAC_AS_GENERAL); +/* + * set up and start data/sync urbs + */ +static int start_urbs(struct snd_usb_substream *subs, struct snd_pcm_runtime *runtime) +{ + unsigned int i; + int err; - if (!as) { - snd_printk(KERN_ERR "%d:%u:%d : UAC_AS_GENERAL descriptor not found\n", - dev->devnum, iface_no, altno); - continue; + if (subs->stream->chip->shutdown) + return -EBADFD; + + for (i = 0; i < subs->nurbs; i++) { + if (snd_BUG_ON(!subs->dataurb[i].urb)) + return -EINVAL; + if (subs->ops.prepare(subs, runtime, subs->dataurb[i].urb) < 0) { + snd_printk(KERN_ERR "cannot prepare datapipe for urb %d\n", i); + goto __error; + } + } + if (subs->syncpipe) { + for (i = 0; i < SYNC_URBS; i++) { + if (snd_BUG_ON(!subs->syncurb[i].urb)) + return -EINVAL; + if (subs->ops.prepare_sync(subs, runtime, subs->syncurb[i].urb) < 0) { + snd_printk(KERN_ERR "cannot prepare syncpipe for urb %d\n", i); + goto __error; } + } + } - if (as->bLength < sizeof(*as)) { - snd_printk(KERN_ERR "%d:%u:%d : invalid UAC_AS_GENERAL desc\n", - dev->devnum, iface_no, altno); - continue; + subs->active_mask = 0; + subs->unlink_mask = 0; + subs->running = 1; + for (i = 0; i < subs->nurbs; i++) { + err = usb_submit_urb(subs->dataurb[i].urb, GFP_ATOMIC); + if (err < 0) { + snd_printk(KERN_ERR "cannot submit datapipe " + "for urb %d, error %d: %s\n", + i, err, usb_error_string(err)); + goto __error; + } + set_bit(i, &subs->active_mask); + } + if (subs->syncpipe) { + for (i = 0; i < SYNC_URBS; i++) { + err = usb_submit_urb(subs->syncurb[i].urb, GFP_ATOMIC); + if (err < 0) { + snd_printk(KERN_ERR "cannot submit syncpipe " + "for urb %d, error %d: %s\n", + i, err, usb_error_string(err)); + goto __error; } + set_bit(i + 16, &subs->active_mask); + } + } + return 0; - num_channels = as->bNrChannels; - format = le32_to_cpu(as->bmFormats); + __error: + // snd_pcm_stop(subs->pcm_substream, SNDRV_PCM_STATE_XRUN); + deactivate_urbs(subs, 0, 0); + return -EPIPE; +} - /* lookup the terminal associated to this interface - * to extract the clock */ - input_term = snd_usb_find_input_terminal_descriptor(chip->ctrl_intf, - as->bTerminalLink); - if (input_term) { - clock = input_term->bCSourceID; - break; - } - output_term = snd_usb_find_output_terminal_descriptor(chip->ctrl_intf, - as->bTerminalLink); - if (output_term) { - clock = output_term->bCSourceID; - break; - } +/* + */ +static struct snd_urb_ops audio_urb_ops[2] = { + { + .prepare = prepare_nodata_playback_urb, + .retire = retire_playback_urb, + .prepare_sync = prepare_playback_sync_urb, + .retire_sync = retire_playback_sync_urb, + }, + { + .prepare = prepare_capture_urb, + .retire = retire_capture_urb, + .prepare_sync = prepare_capture_sync_urb, + .retire_sync = retire_capture_sync_urb, + }, +}; - snd_printk(KERN_ERR "%d:%u:%d : bogus bTerminalLink %d\n", - dev->devnum, iface_no, altno, as->bTerminalLink); - continue; - } - } +/* + * initialize the substream instance. + */ - /* get format type */ - fmt = snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, UAC_FORMAT_TYPE); - if (!fmt) { - snd_printk(KERN_ERR "%d:%u:%d : no UAC_FORMAT_TYPE desc\n", - dev->devnum, iface_no, altno); - continue; - } - if (((protocol == UAC_VERSION_1) && (fmt->bLength < 8)) || - ((protocol == UAC_VERSION_2) && (fmt->bLength < 6))) { - snd_printk(KERN_ERR "%d:%u:%d : invalid UAC_FORMAT_TYPE desc\n", - dev->devnum, iface_no, altno); - continue; - } +void snd_usb_init_substream(struct snd_usb_stream *as, + int stream, struct audioformat *fp) +{ + struct snd_usb_substream *subs = &as->substream[stream]; + + INIT_LIST_HEAD(&subs->fmt_list); + spin_lock_init(&subs->lock); + + subs->stream = as; + subs->direction = stream; + subs->dev = as->chip->dev; + subs->txfr_quirk = as->chip->txfr_quirk; + subs->ops = audio_urb_ops[stream]; + if (snd_usb_get_speed(subs->dev) >= USB_SPEED_HIGH) + subs->ops.prepare_sync = prepare_capture_sync_urb_hs; + + snd_usb_set_pcm_ops(as->pcm, stream); + + list_add_tail(&fp->list, &subs->fmt_list); + subs->formats |= fp->formats; + subs->endpoint = fp->endpoint; + subs->num_formats++; + subs->fmt_type = fp->fmt_type; +} - /* - * Blue Microphones workaround: The last altsetting is identical - * with the previous one, except for a larger packet size, but - * is actually a mislabeled two-channel setting; ignore it. - */ - if (fmt->bNrChannels == 1 && - fmt->bSubframeSize == 2 && - altno == 2 && num == 3 && - fp && fp->altsetting == 1 && fp->channels == 1 && - fp->formats == SNDRV_PCM_FMTBIT_S16_LE && - protocol == UAC_VERSION_1 && - le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize) == - fp->maxpacksize * 2) - continue; - - fp = kzalloc(sizeof(*fp), GFP_KERNEL); - if (! fp) { - snd_printk(KERN_ERR "cannot malloc\n"); - return -ENOMEM; - } +int snd_usb_substream_playback_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_usb_substream *subs = substream->runtime->private_data; - fp->iface = iface_no; - fp->altsetting = altno; - fp->altset_idx = i; - fp->endpoint = get_endpoint(alts, 0)->bEndpointAddress; - fp->ep_attr = get_endpoint(alts, 0)->bmAttributes; - fp->datainterval = snd_usb_parse_datainterval(chip, alts); - fp->maxpacksize = le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize); - /* num_channels is only set for v2 interfaces */ - fp->channels = num_channels; - if (snd_usb_get_speed(dev) == USB_SPEED_HIGH) - fp->maxpacksize = (((fp->maxpacksize >> 11) & 3) + 1) - * (fp->maxpacksize & 0x7ff); - fp->attributes = parse_uac_endpoint_attributes(chip, alts, protocol, iface_no); - fp->clock = clock; - - /* some quirks for attributes here */ - - switch (chip->usb_id) { - case USB_ID(0x0a92, 0x0053): /* AudioTrak Optoplay */ - /* Optoplay sets the sample rate attribute although - * it seems not supporting it in fact. - */ - fp->attributes &= ~UAC_EP_CS_ATTR_SAMPLE_RATE; - break; - case USB_ID(0x041e, 0x3020): /* Creative SB Audigy 2 NX */ - case USB_ID(0x0763, 0x2003): /* M-Audio Audiophile USB */ - /* doesn't set the sample rate attribute, but supports it */ - fp->attributes |= UAC_EP_CS_ATTR_SAMPLE_RATE; - break; - case USB_ID(0x0763, 0x2001): /* M-Audio Quattro USB */ - case USB_ID(0x0763, 0x2012): /* M-Audio Fast Track Pro USB */ - case USB_ID(0x047f, 0x0ca1): /* plantronics headset */ - case USB_ID(0x077d, 0x07af): /* Griffin iMic (note that there is - an older model 77d:223) */ - /* - * plantronics headset and Griffin iMic have set adaptive-in - * although it's really not... - */ - fp->ep_attr &= ~USB_ENDPOINT_SYNCTYPE; - if (stream == SNDRV_PCM_STREAM_PLAYBACK) - fp->ep_attr |= USB_ENDPOINT_SYNC_ADAPTIVE; - else - fp->ep_attr |= USB_ENDPOINT_SYNC_SYNC; - break; - } + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + subs->ops.prepare = prepare_playback_urb; + return 0; + case SNDRV_PCM_TRIGGER_STOP: + return deactivate_urbs(subs, 0, 0); + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + subs->ops.prepare = prepare_nodata_playback_urb; + return 0; + } - /* ok, let's parse further... */ - if (snd_usb_parse_audio_format(chip, fp, format, fmt, stream, alts) < 0) { - kfree(fp->rate_table); - kfree(fp); - fp = NULL; - continue; - } + return -EINVAL; +} - snd_printdd(KERN_INFO "%d:%u:%d: add audio endpoint %#x\n", dev->devnum, iface_no, altno, fp->endpoint); - err = snd_usb_add_audio_endpoint(chip, stream, fp); - if (err < 0) { - kfree(fp->rate_table); - kfree(fp); - return err; - } - /* try to set the interface... */ - usb_set_interface(chip->dev, iface_no, altno); - snd_usb_init_pitch(chip, iface_no, alts, fp); - snd_usb_init_sample_rate(chip, iface_no, alts, fp, fp->rate_max); +int snd_usb_substream_capture_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_usb_substream *subs = substream->runtime->private_data; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + subs->ops.retire = retire_capture_urb; + return start_urbs(subs, substream->runtime); + case SNDRV_PCM_TRIGGER_STOP: + return deactivate_urbs(subs, 0, 0); + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + subs->ops.retire = retire_paused_capture_urb; + return 0; + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + subs->ops.retire = retire_capture_urb; + return 0; } + + return -EINVAL; +} + +int snd_usb_substream_prepare(struct snd_usb_substream *subs, + struct snd_pcm_runtime *runtime) +{ + /* clear urbs (to be sure) */ + deactivate_urbs(subs, 0, 1); + wait_clear_urbs(subs); + + /* for playback, submit the URBs now; otherwise, the first hwptr_done + * updates for all URBs would happen at the same time when starting */ + if (subs->direction == SNDRV_PCM_STREAM_PLAYBACK) { + subs->ops.prepare = prepare_nodata_playback_urb; + return start_urbs(subs, runtime); + } + return 0; } diff --git a/sound/usb/endpoint.h b/sound/usb/endpoint.h index 64dd0db023b2..88eb63a636eb 100644 --- a/sound/usb/endpoint.h +++ b/sound/usb/endpoint.h @@ -1,11 +1,21 @@ #ifndef __USBAUDIO_ENDPOINT_H #define __USBAUDIO_ENDPOINT_H -int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, - int iface_no); +void snd_usb_init_substream(struct snd_usb_stream *as, + int stream, + struct audioformat *fp); -int snd_usb_add_audio_endpoint(struct snd_usb_audio *chip, - int stream, - struct audioformat *fp); +int snd_usb_init_substream_urbs(struct snd_usb_substream *subs, + unsigned int period_bytes, + unsigned int rate, + unsigned int frame_bits); + +void snd_usb_release_substream_urbs(struct snd_usb_substream *subs, int force); + +int snd_usb_substream_prepare(struct snd_usb_substream *subs, + struct snd_pcm_runtime *runtime); + +int snd_usb_substream_playback_trigger(struct snd_pcm_substream *substream, int cmd); +int snd_usb_substream_capture_trigger(struct snd_pcm_substream *substream, int cmd); #endif /* __USBAUDIO_ENDPOINT_H */ diff --git a/sound/usb/format.c b/sound/usb/format.c index 8d042dce0d16..89421d176570 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -286,7 +286,7 @@ static int parse_audio_format_rates_v2(struct snd_usb_audio *chip, USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN, UAC2_CS_CONTROL_SAM_FREQ << 8, snd_usb_ctrl_intf(chip) | (clock << 8), - tmp, sizeof(tmp), 1000); + tmp, sizeof(tmp)); if (ret < 0) { snd_printk(KERN_ERR "%s(): unable to retrieve number of sample rates (clock %d)\n", @@ -307,7 +307,7 @@ static int parse_audio_format_rates_v2(struct snd_usb_audio *chip, USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN, UAC2_CS_CONTROL_SAM_FREQ << 8, snd_usb_ctrl_intf(chip) | (clock << 8), - data, data_size, 1000); + data, data_size); if (ret < 0) { snd_printk(KERN_ERR "%s(): unable to retrieve sample rate range (clock %d)\n", diff --git a/sound/usb/helper.c b/sound/usb/helper.c index f280c1903c25..9eed8f40b179 100644 --- a/sound/usb/helper.c +++ b/sound/usb/helper.c @@ -81,7 +81,7 @@ void *snd_usb_find_csint_desc(void *buffer, int buflen, void *after, u8 dsubtype */ int snd_usb_ctl_msg(struct usb_device *dev, unsigned int pipe, __u8 request, __u8 requesttype, __u16 value, __u16 index, void *data, - __u16 size, int timeout) + __u16 size) { int err; void *buf = NULL; @@ -92,7 +92,7 @@ int snd_usb_ctl_msg(struct usb_device *dev, unsigned int pipe, __u8 request, return -ENOMEM; } err = usb_control_msg(dev, pipe, request, requesttype, - value, index, buf, size, timeout); + value, index, buf, size, 1000); if (size > 0) { memcpy(data, buf, size); kfree(buf); diff --git a/sound/usb/helper.h b/sound/usb/helper.h index 09bd943c43bf..805c300dd004 100644 --- a/sound/usb/helper.h +++ b/sound/usb/helper.h @@ -8,7 +8,7 @@ void *snd_usb_find_csint_desc(void *descstart, int desclen, void *after, u8 dsub int snd_usb_ctl_msg(struct usb_device *dev, unsigned int pipe, __u8 request, __u8 requesttype, __u16 value, __u16 index, - void *data, __u16 size, int timeout); + void *data, __u16 size); unsigned char snd_usb_parse_datainterval(struct snd_usb_audio *chip, struct usb_host_interface *alts); diff --git a/sound/usb/midi.c b/sound/usb/midi.c index f9289102886a..e21f026d9577 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -816,6 +816,22 @@ static struct usb_protocol_ops snd_usbmidi_raw_ops = { .output = snd_usbmidi_raw_output, }; +/* + * FTDI protocol: raw MIDI bytes, but input packets have two modem status bytes. + */ + +static void snd_usbmidi_ftdi_input(struct snd_usb_midi_in_endpoint* ep, + uint8_t* buffer, int buffer_length) +{ + if (buffer_length > 2) + snd_usbmidi_input_data(ep, 0, buffer + 2, buffer_length - 2); +} + +static struct usb_protocol_ops snd_usbmidi_ftdi_ops = { + .input = snd_usbmidi_ftdi_input, + .output = snd_usbmidi_raw_output, +}; + static void snd_usbmidi_us122l_input(struct snd_usb_midi_in_endpoint *ep, uint8_t *buffer, int buffer_length) { @@ -2163,6 +2179,17 @@ int snd_usbmidi_create(struct snd_card *card, /* endpoint 1 is input-only */ endpoints[1].out_cables = 0; break; + case QUIRK_MIDI_FTDI: + umidi->usb_protocol_ops = &snd_usbmidi_ftdi_ops; + + /* set baud rate to 31250 (48 MHz / 16 / 96) */ + err = usb_control_msg(umidi->dev, usb_sndctrlpipe(umidi->dev, 0), + 3, 0x40, 0x60, 0, NULL, 0, 1000); + if (err < 0) + break; + + err = snd_usbmidi_detect_per_port_endpoints(umidi, endpoints); + break; default: snd_printd(KERN_ERR "invalid quirk type %d\n", quirk->type); err = -ENXIO; diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index cdd19d7fe500..60f65ace7474 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -296,7 +296,7 @@ static int get_ctl_value_v1(struct usb_mixer_elem_info *cval, int request, int v if (snd_usb_ctl_msg(chip->dev, usb_rcvctrlpipe(chip->dev, 0), request, USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, validx, snd_usb_ctrl_intf(chip) | (cval->id << 8), - buf, val_len, 100) >= val_len) { + buf, val_len) >= val_len) { *value_ret = convert_signed_value(cval, snd_usb_combine_bytes(buf, val_len)); snd_usb_autosuspend(cval->mixer->chip); return 0; @@ -333,7 +333,7 @@ static int get_ctl_value_v2(struct usb_mixer_elem_info *cval, int request, int v ret = snd_usb_ctl_msg(chip->dev, usb_rcvctrlpipe(chip->dev, 0), bRequest, USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, validx, snd_usb_ctrl_intf(chip) | (cval->id << 8), - buf, size, 1000); + buf, size); snd_usb_autosuspend(chip); if (ret < 0) { @@ -445,7 +445,7 @@ int snd_usb_mixer_set_ctl_value(struct usb_mixer_elem_info *cval, usb_sndctrlpipe(chip->dev, 0), request, USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_OUT, validx, snd_usb_ctrl_intf(chip) | (cval->id << 8), - buf, val_len, 100) >= 0) { + buf, val_len) >= 0) { snd_usb_autosuspend(chip); return 0; } @@ -881,8 +881,17 @@ static int mixer_ctl_feature_info(struct snd_kcontrol *kcontrol, struct snd_ctl_ uinfo->value.integer.min = 0; uinfo->value.integer.max = 1; } else { - if (! cval->initialized) - get_min_max(cval, 0); + if (!cval->initialized) { + get_min_max(cval, 0); + if (cval->initialized && cval->dBmin >= cval->dBmax) { + kcontrol->vd[0].access &= + ~(SNDRV_CTL_ELEM_ACCESS_TLV_READ | + SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK); + snd_ctl_notify(cval->mixer->chip->card, + SNDRV_CTL_EVENT_MASK_INFO, + &kcontrol->id); + } + } uinfo->value.integer.min = 0; uinfo->value.integer.max = (cval->max - cval->min + cval->res - 1) / cval->res; @@ -1250,7 +1259,7 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, void build_feature_ctl(state, _ftr, 0, i, &iterm, unitid, 0); } } else { /* UAC_VERSION_2 */ - for (i = 0; i < 30/2; i++) { + for (i = 0; i < ARRAY_SIZE(audio_feature_info); i++) { unsigned int ch_bits = 0; unsigned int ch_read_only = 0; diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 3d0f4873112b..ab125ee0b0f0 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -190,18 +190,18 @@ static int snd_audigy2nx_led_put(struct snd_kcontrol *kcontrol, struct snd_ctl_e err = snd_usb_ctl_msg(mixer->chip->dev, usb_sndctrlpipe(mixer->chip->dev, 0), 0x24, USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER, - !value, 0, NULL, 0, 100); + !value, 0, NULL, 0); /* USB X-Fi S51 Pro */ if (mixer->chip->usb_id == USB_ID(0x041e, 0x30df)) err = snd_usb_ctl_msg(mixer->chip->dev, usb_sndctrlpipe(mixer->chip->dev, 0), 0x24, USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER, - !value, 0, NULL, 0, 100); + !value, 0, NULL, 0); else err = snd_usb_ctl_msg(mixer->chip->dev, usb_sndctrlpipe(mixer->chip->dev, 0), 0x24, USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER, - value, index + 2, NULL, 0, 100); + value, index + 2, NULL, 0); if (err < 0) return err; mixer->audigy2nx_leds[index] = value; @@ -299,7 +299,7 @@ static void snd_audigy2nx_proc_read(struct snd_info_entry *entry, usb_rcvctrlpipe(mixer->chip->dev, 0), UAC_GET_MEM, USB_DIR_IN | USB_TYPE_CLASS | USB_RECIP_INTERFACE, 0, - jacks[i].unitid << 8, buf, 3, 100); + jacks[i].unitid << 8, buf, 3); if (err == 3 && (buf[0] == 3 || buf[0] == 6)) snd_iprintf(buffer, "%02x %02x\n", buf[1], buf[2]); else @@ -332,7 +332,7 @@ static int snd_xonar_u1_switch_put(struct snd_kcontrol *kcontrol, err = snd_usb_ctl_msg(mixer->chip->dev, usb_sndctrlpipe(mixer->chip->dev, 0), 0x08, USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER, - 50, 0, &new_status, 1, 100); + 50, 0, &new_status, 1); if (err < 0) return err; mixer->xonar_u1_status = new_status; diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index b8dcbf407bbb..0220b0f335b9 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -28,12 +28,36 @@ #include "card.h" #include "quirks.h" #include "debug.h" -#include "urb.h" +#include "endpoint.h" #include "helper.h" #include "pcm.h" #include "clock.h" #include "power.h" +/* return the estimated delay based on USB frame counters */ +snd_pcm_uframes_t snd_usb_pcm_delay(struct snd_usb_substream *subs, + unsigned int rate) +{ + int current_frame_number; + int frame_diff; + int est_delay; + + current_frame_number = usb_get_current_frame_number(subs->dev); + /* + * HCD implementations use different widths, use lower 8 bits. + * The delay will be managed up to 256ms, which is more than + * enough + */ + frame_diff = (current_frame_number - subs->last_frame_number) & 0xff; + + /* Approximation based on number of samples per USB frame (ms), + some truncation for 44.1 but the estimate is good enough */ + est_delay = subs->last_delay - (frame_diff * rate / 1000); + if (est_delay < 0) + est_delay = 0; + return est_delay; +} + /* * return the current pcm pointer. just based on the hwptr_done value. */ @@ -45,6 +69,8 @@ static snd_pcm_uframes_t snd_usb_pcm_pointer(struct snd_pcm_substream *substream subs = (struct snd_usb_substream *)substream->runtime->private_data; spin_lock(&subs->lock); hwptr_done = subs->hwptr_done; + substream->runtime->delay = snd_usb_pcm_delay(subs, + substream->runtime->rate); spin_unlock(&subs->lock); return hwptr_done / (substream->runtime->frame_bits >> 3); } @@ -126,7 +152,7 @@ static int init_pitch_v1(struct snd_usb_audio *chip, int iface, if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC_SET_CUR, USB_TYPE_CLASS|USB_RECIP_ENDPOINT|USB_DIR_OUT, UAC_EP_CS_ATTR_PITCH_CONTROL << 8, ep, - data, sizeof(data), 1000)) < 0) { + data, sizeof(data))) < 0) { snd_printk(KERN_ERR "%d:%d:%d: cannot set enable PITCH\n", dev->devnum, iface, ep); return err; @@ -150,7 +176,7 @@ static int init_pitch_v2(struct snd_usb_audio *chip, int iface, if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC2_CS_CUR, USB_TYPE_CLASS | USB_RECIP_ENDPOINT | USB_DIR_OUT, UAC2_EP_CS_PITCH << 8, 0, - data, sizeof(data), 1000)) < 0) { + data, sizeof(data))) < 0) { snd_printk(KERN_ERR "%d:%d:%d: cannot set enable PITCH (v2)\n", dev->devnum, iface, fmt->altsetting); return err; @@ -417,6 +443,8 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream) subs->hwptr_done = 0; subs->transfer_done = 0; subs->phase = 0; + subs->last_delay = 0; + subs->last_frame_number = 0; runtime->delay = 0; return snd_usb_substream_prepare(subs, runtime); diff --git a/sound/usb/pcm.h b/sound/usb/pcm.h index ed3e283f618d..df7a003682ad 100644 --- a/sound/usb/pcm.h +++ b/sound/usb/pcm.h @@ -1,6 +1,9 @@ #ifndef __USBAUDIO_PCM_H #define __USBAUDIO_PCM_H +snd_pcm_uframes_t snd_usb_pcm_delay(struct snd_usb_substream *subs, + unsigned int rate); + void snd_usb_set_pcm_ops(struct snd_pcm *pcm, int stream); int snd_usb_init_pitch(struct snd_usb_audio *chip, int iface, diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index a42e3ef3832d..b61945f3af9e 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -39,6 +39,17 @@ .idProduct = prod, \ .bInterfaceClass = USB_CLASS_VENDOR_SPEC +/* FTDI devices */ +{ + USB_DEVICE(0x0403, 0xb8d8), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + /* .vendor_name = "STARR LABS", */ + /* .product_name = "Starr Labs MIDI USB device", */ + .ifnum = 0, + .type = QUIRK_MIDI_FTDI + } +}, + /* Creative/Toshiba Multimedia Center SB-0500 */ { USB_DEVICE(0x041e, 0x3048), @@ -1678,6 +1689,20 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { + /* Added support for Roland UM-ONE which differs from UM-1 */ + USB_DEVICE(0x0582, 0x012a), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + /* .vendor_name = "ROLAND", */ + /* .product_name = "UM-ONE", */ + .ifnum = 0, + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = & (const struct snd_usb_midi_endpoint_info) { + .out_cables = 0x0001, + .in_cables = 0x0003 + } + } +}, +{ USB_DEVICE(0x0582, 0x011e), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { /* .vendor_name = "BOSS", */ diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 81e07d842581..2e5bc7344026 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -34,6 +34,7 @@ #include "endpoint.h" #include "pcm.h" #include "clock.h" +#include "stream.h" /* * handle the quirks for the contained interfaces @@ -106,7 +107,7 @@ static int create_standard_audio_quirk(struct snd_usb_audio *chip, alts = &iface->altsetting[0]; altsd = get_iface_desc(alts); - err = snd_usb_parse_audio_endpoints(chip, altsd->bInterfaceNumber); + err = snd_usb_parse_audio_interface(chip, altsd->bInterfaceNumber); if (err < 0) { snd_printk(KERN_ERR "cannot setup if %d: error %d\n", altsd->bInterfaceNumber, err); @@ -147,7 +148,7 @@ static int create_fixed_stream_quirk(struct snd_usb_audio *chip, stream = (fp->endpoint & USB_DIR_IN) ? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK; - err = snd_usb_add_audio_endpoint(chip, stream, fp); + err = snd_usb_add_audio_stream(chip, stream, fp); if (err < 0) { kfree(fp); kfree(rate_table); @@ -254,7 +255,7 @@ static int create_uaxx_quirk(struct snd_usb_audio *chip, stream = (fp->endpoint & USB_DIR_IN) ? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK; - err = snd_usb_add_audio_endpoint(chip, stream, fp); + err = snd_usb_add_audio_stream(chip, stream, fp); if (err < 0) { kfree(fp); return err; @@ -306,6 +307,7 @@ int snd_usb_create_quirk(struct snd_usb_audio *chip, [QUIRK_MIDI_EMAGIC] = create_any_midi_quirk, [QUIRK_MIDI_CME] = create_any_midi_quirk, [QUIRK_MIDI_AKAI] = create_any_midi_quirk, + [QUIRK_MIDI_FTDI] = create_any_midi_quirk, [QUIRK_AUDIO_STANDARD_INTERFACE] = create_standard_audio_quirk, [QUIRK_AUDIO_FIXED_ENDPOINT] = create_fixed_stream_quirk, [QUIRK_AUDIO_EDIROL_UAXX] = create_uaxx_quirk, @@ -338,7 +340,7 @@ static int snd_usb_extigy_boot_quirk(struct usb_device *dev, struct usb_interfac snd_printdd("sending Extigy boot sequence...\n"); /* Send message to force it to reconnect with full interface. */ err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev,0), - 0x10, 0x43, 0x0001, 0x000a, NULL, 0, 1000); + 0x10, 0x43, 0x0001, 0x000a, NULL, 0); if (err < 0) snd_printdd("error sending boot message: %d\n", err); err = usb_get_descriptor(dev, USB_DT_DEVICE, 0, &dev->descriptor, sizeof(dev->descriptor)); @@ -359,11 +361,11 @@ static int snd_usb_audigy2nx_boot_quirk(struct usb_device *dev) snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), 0x2a, USB_DIR_IN | USB_TYPE_VENDOR | USB_RECIP_OTHER, - 0, 0, &buf, 1, 1000); + 0, 0, &buf, 1); if (buf == 0) { snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), 0x29, USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER, - 1, 2000, NULL, 0, 1000); + 1, 2000, NULL, 0); return -ENODEV; } return 0; @@ -406,7 +408,7 @@ static int snd_usb_cm106_write_int_reg(struct usb_device *dev, int reg, u16 valu buf[3] = reg; return snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), USB_REQ_SET_CONFIGURATION, USB_DIR_OUT | USB_TYPE_CLASS | USB_RECIP_ENDPOINT, - 0, 0, &buf, 4, 1000); + 0, 0, &buf, 4); } static int snd_usb_cm106_boot_quirk(struct usb_device *dev) diff --git a/sound/usb/stream.c b/sound/usb/stream.c new file mode 100644 index 000000000000..5ff8010b2d6f --- /dev/null +++ b/sound/usb/stream.c @@ -0,0 +1,452 @@ +/* + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + + +#include <linux/init.h> +#include <linux/slab.h> +#include <linux/usb.h> +#include <linux/usb/audio.h> +#include <linux/usb/audio-v2.h> + +#include <sound/core.h> +#include <sound/pcm.h> + +#include "usbaudio.h" +#include "card.h" +#include "proc.h" +#include "quirks.h" +#include "endpoint.h" +#include "pcm.h" +#include "helper.h" +#include "format.h" +#include "clock.h" +#include "stream.h" + +/* + * free a substream + */ +static void free_substream(struct snd_usb_substream *subs) +{ + struct list_head *p, *n; + + if (!subs->num_formats) + return; /* not initialized */ + list_for_each_safe(p, n, &subs->fmt_list) { + struct audioformat *fp = list_entry(p, struct audioformat, list); + kfree(fp->rate_table); + kfree(fp); + } + kfree(subs->rate_list.list); +} + + +/* + * free a usb stream instance + */ +static void snd_usb_audio_stream_free(struct snd_usb_stream *stream) +{ + free_substream(&stream->substream[0]); + free_substream(&stream->substream[1]); + list_del(&stream->list); + kfree(stream); +} + +static void snd_usb_audio_pcm_free(struct snd_pcm *pcm) +{ + struct snd_usb_stream *stream = pcm->private_data; + if (stream) { + stream->pcm = NULL; + snd_usb_audio_stream_free(stream); + } +} + + +/* + * add this endpoint to the chip instance. + * if a stream with the same endpoint already exists, append to it. + * if not, create a new pcm stream. + */ +int snd_usb_add_audio_stream(struct snd_usb_audio *chip, + int stream, + struct audioformat *fp) +{ + struct list_head *p; + struct snd_usb_stream *as; + struct snd_usb_substream *subs; + struct snd_pcm *pcm; + int err; + + list_for_each(p, &chip->pcm_list) { + as = list_entry(p, struct snd_usb_stream, list); + if (as->fmt_type != fp->fmt_type) + continue; + subs = &as->substream[stream]; + if (!subs->endpoint) + continue; + if (subs->endpoint == fp->endpoint) { + list_add_tail(&fp->list, &subs->fmt_list); + subs->num_formats++; + subs->formats |= fp->formats; + return 0; + } + } + /* look for an empty stream */ + list_for_each(p, &chip->pcm_list) { + as = list_entry(p, struct snd_usb_stream, list); + if (as->fmt_type != fp->fmt_type) + continue; + subs = &as->substream[stream]; + if (subs->endpoint) + continue; + err = snd_pcm_new_stream(as->pcm, stream, 1); + if (err < 0) + return err; + snd_usb_init_substream(as, stream, fp); + return 0; + } + + /* create a new pcm */ + as = kzalloc(sizeof(*as), GFP_KERNEL); + if (!as) + return -ENOMEM; + as->pcm_index = chip->pcm_devs; + as->chip = chip; + as->fmt_type = fp->fmt_type; + err = snd_pcm_new(chip->card, "USB Audio", chip->pcm_devs, + stream == SNDRV_PCM_STREAM_PLAYBACK ? 1 : 0, + stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1, + &pcm); + if (err < 0) { + kfree(as); + return err; + } + as->pcm = pcm; + pcm->private_data = as; + pcm->private_free = snd_usb_audio_pcm_free; + pcm->info_flags = 0; + if (chip->pcm_devs > 0) + sprintf(pcm->name, "USB Audio #%d", chip->pcm_devs); + else + strcpy(pcm->name, "USB Audio"); + + snd_usb_init_substream(as, stream, fp); + + list_add(&as->list, &chip->pcm_list); + chip->pcm_devs++; + + snd_usb_proc_pcm_format_add(as); + + return 0; +} + +static int parse_uac_endpoint_attributes(struct snd_usb_audio *chip, + struct usb_host_interface *alts, + int protocol, int iface_no) +{ + /* parsed with a v1 header here. that's ok as we only look at the + * header first which is the same for both versions */ + struct uac_iso_endpoint_descriptor *csep; + struct usb_interface_descriptor *altsd = get_iface_desc(alts); + int attributes = 0; + + csep = snd_usb_find_desc(alts->endpoint[0].extra, alts->endpoint[0].extralen, NULL, USB_DT_CS_ENDPOINT); + + /* Creamware Noah has this descriptor after the 2nd endpoint */ + if (!csep && altsd->bNumEndpoints >= 2) + csep = snd_usb_find_desc(alts->endpoint[1].extra, alts->endpoint[1].extralen, NULL, USB_DT_CS_ENDPOINT); + + if (!csep || csep->bLength < 7 || + csep->bDescriptorSubtype != UAC_EP_GENERAL) { + snd_printk(KERN_WARNING "%d:%u:%d : no or invalid" + " class specific endpoint descriptor\n", + chip->dev->devnum, iface_no, + altsd->bAlternateSetting); + return 0; + } + + if (protocol == UAC_VERSION_1) { + attributes = csep->bmAttributes; + } else { + struct uac2_iso_endpoint_descriptor *csep2 = + (struct uac2_iso_endpoint_descriptor *) csep; + + attributes = csep->bmAttributes & UAC_EP_CS_ATTR_FILL_MAX; + + /* emulate the endpoint attributes of a v1 device */ + if (csep2->bmControls & UAC2_CONTROL_PITCH) + attributes |= UAC_EP_CS_ATTR_PITCH_CONTROL; + } + + return attributes; +} + +static struct uac2_input_terminal_descriptor * + snd_usb_find_input_terminal_descriptor(struct usb_host_interface *ctrl_iface, + int terminal_id) +{ + struct uac2_input_terminal_descriptor *term = NULL; + + while ((term = snd_usb_find_csint_desc(ctrl_iface->extra, + ctrl_iface->extralen, + term, UAC_INPUT_TERMINAL))) { + if (term->bTerminalID == terminal_id) + return term; + } + + return NULL; +} + +static struct uac2_output_terminal_descriptor * + snd_usb_find_output_terminal_descriptor(struct usb_host_interface *ctrl_iface, + int terminal_id) +{ + struct uac2_output_terminal_descriptor *term = NULL; + + while ((term = snd_usb_find_csint_desc(ctrl_iface->extra, + ctrl_iface->extralen, + term, UAC_OUTPUT_TERMINAL))) { + if (term->bTerminalID == terminal_id) + return term; + } + + return NULL; +} + +int snd_usb_parse_audio_interface(struct snd_usb_audio *chip, int iface_no) +{ + struct usb_device *dev; + struct usb_interface *iface; + struct usb_host_interface *alts; + struct usb_interface_descriptor *altsd; + int i, altno, err, stream; + int format = 0, num_channels = 0; + struct audioformat *fp = NULL; + int num, protocol, clock = 0; + struct uac_format_type_i_continuous_descriptor *fmt; + + dev = chip->dev; + + /* parse the interface's altsettings */ + iface = usb_ifnum_to_if(dev, iface_no); + + num = iface->num_altsetting; + + /* + * Dallas DS4201 workaround: It presents 5 altsettings, but the last + * one misses syncpipe, and does not produce any sound. + */ + if (chip->usb_id == USB_ID(0x04fa, 0x4201)) + num = 4; + + for (i = 0; i < num; i++) { + alts = &iface->altsetting[i]; + altsd = get_iface_desc(alts); + protocol = altsd->bInterfaceProtocol; + /* skip invalid one */ + if ((altsd->bInterfaceClass != USB_CLASS_AUDIO && + altsd->bInterfaceClass != USB_CLASS_VENDOR_SPEC) || + (altsd->bInterfaceSubClass != USB_SUBCLASS_AUDIOSTREAMING && + altsd->bInterfaceSubClass != USB_SUBCLASS_VENDOR_SPEC) || + altsd->bNumEndpoints < 1 || + le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize) == 0) + continue; + /* must be isochronous */ + if ((get_endpoint(alts, 0)->bmAttributes & USB_ENDPOINT_XFERTYPE_MASK) != + USB_ENDPOINT_XFER_ISOC) + continue; + /* check direction */ + stream = (get_endpoint(alts, 0)->bEndpointAddress & USB_DIR_IN) ? + SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK; + altno = altsd->bAlternateSetting; + + if (snd_usb_apply_interface_quirk(chip, iface_no, altno)) + continue; + + /* get audio formats */ + switch (protocol) { + default: + snd_printdd(KERN_WARNING "%d:%u:%d: unknown interface protocol %#02x, assuming v1\n", + dev->devnum, iface_no, altno, protocol); + protocol = UAC_VERSION_1; + /* fall through */ + + case UAC_VERSION_1: { + struct uac1_as_header_descriptor *as = + snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, UAC_AS_GENERAL); + + if (!as) { + snd_printk(KERN_ERR "%d:%u:%d : UAC_AS_GENERAL descriptor not found\n", + dev->devnum, iface_no, altno); + continue; + } + + if (as->bLength < sizeof(*as)) { + snd_printk(KERN_ERR "%d:%u:%d : invalid UAC_AS_GENERAL desc\n", + dev->devnum, iface_no, altno); + continue; + } + + format = le16_to_cpu(as->wFormatTag); /* remember the format value */ + break; + } + + case UAC_VERSION_2: { + struct uac2_input_terminal_descriptor *input_term; + struct uac2_output_terminal_descriptor *output_term; + struct uac2_as_header_descriptor *as = + snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, UAC_AS_GENERAL); + + if (!as) { + snd_printk(KERN_ERR "%d:%u:%d : UAC_AS_GENERAL descriptor not found\n", + dev->devnum, iface_no, altno); + continue; + } + + if (as->bLength < sizeof(*as)) { + snd_printk(KERN_ERR "%d:%u:%d : invalid UAC_AS_GENERAL desc\n", + dev->devnum, iface_no, altno); + continue; + } + + num_channels = as->bNrChannels; + format = le32_to_cpu(as->bmFormats); + + /* lookup the terminal associated to this interface + * to extract the clock */ + input_term = snd_usb_find_input_terminal_descriptor(chip->ctrl_intf, + as->bTerminalLink); + if (input_term) { + clock = input_term->bCSourceID; + break; + } + + output_term = snd_usb_find_output_terminal_descriptor(chip->ctrl_intf, + as->bTerminalLink); + if (output_term) { + clock = output_term->bCSourceID; + break; + } + + snd_printk(KERN_ERR "%d:%u:%d : bogus bTerminalLink %d\n", + dev->devnum, iface_no, altno, as->bTerminalLink); + continue; + } + } + + /* get format type */ + fmt = snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, UAC_FORMAT_TYPE); + if (!fmt) { + snd_printk(KERN_ERR "%d:%u:%d : no UAC_FORMAT_TYPE desc\n", + dev->devnum, iface_no, altno); + continue; + } + if (((protocol == UAC_VERSION_1) && (fmt->bLength < 8)) || + ((protocol == UAC_VERSION_2) && (fmt->bLength < 6))) { + snd_printk(KERN_ERR "%d:%u:%d : invalid UAC_FORMAT_TYPE desc\n", + dev->devnum, iface_no, altno); + continue; + } + + /* + * Blue Microphones workaround: The last altsetting is identical + * with the previous one, except for a larger packet size, but + * is actually a mislabeled two-channel setting; ignore it. + */ + if (fmt->bNrChannels == 1 && + fmt->bSubframeSize == 2 && + altno == 2 && num == 3 && + fp && fp->altsetting == 1 && fp->channels == 1 && + fp->formats == SNDRV_PCM_FMTBIT_S16_LE && + protocol == UAC_VERSION_1 && + le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize) == + fp->maxpacksize * 2) + continue; + + fp = kzalloc(sizeof(*fp), GFP_KERNEL); + if (! fp) { + snd_printk(KERN_ERR "cannot malloc\n"); + return -ENOMEM; + } + + fp->iface = iface_no; + fp->altsetting = altno; + fp->altset_idx = i; + fp->endpoint = get_endpoint(alts, 0)->bEndpointAddress; + fp->ep_attr = get_endpoint(alts, 0)->bmAttributes; + fp->datainterval = snd_usb_parse_datainterval(chip, alts); + fp->maxpacksize = le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize); + /* num_channels is only set for v2 interfaces */ + fp->channels = num_channels; + if (snd_usb_get_speed(dev) == USB_SPEED_HIGH) + fp->maxpacksize = (((fp->maxpacksize >> 11) & 3) + 1) + * (fp->maxpacksize & 0x7ff); + fp->attributes = parse_uac_endpoint_attributes(chip, alts, protocol, iface_no); + fp->clock = clock; + + /* some quirks for attributes here */ + + switch (chip->usb_id) { + case USB_ID(0x0a92, 0x0053): /* AudioTrak Optoplay */ + /* Optoplay sets the sample rate attribute although + * it seems not supporting it in fact. + */ + fp->attributes &= ~UAC_EP_CS_ATTR_SAMPLE_RATE; + break; + case USB_ID(0x041e, 0x3020): /* Creative SB Audigy 2 NX */ + case USB_ID(0x0763, 0x2003): /* M-Audio Audiophile USB */ + /* doesn't set the sample rate attribute, but supports it */ + fp->attributes |= UAC_EP_CS_ATTR_SAMPLE_RATE; + break; + case USB_ID(0x0763, 0x2001): /* M-Audio Quattro USB */ + case USB_ID(0x0763, 0x2012): /* M-Audio Fast Track Pro USB */ + case USB_ID(0x047f, 0x0ca1): /* plantronics headset */ + case USB_ID(0x077d, 0x07af): /* Griffin iMic (note that there is + an older model 77d:223) */ + /* + * plantronics headset and Griffin iMic have set adaptive-in + * although it's really not... + */ + fp->ep_attr &= ~USB_ENDPOINT_SYNCTYPE; + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + fp->ep_attr |= USB_ENDPOINT_SYNC_ADAPTIVE; + else + fp->ep_attr |= USB_ENDPOINT_SYNC_SYNC; + break; + } + + /* ok, let's parse further... */ + if (snd_usb_parse_audio_format(chip, fp, format, fmt, stream, alts) < 0) { + kfree(fp->rate_table); + kfree(fp); + fp = NULL; + continue; + } + + snd_printdd(KERN_INFO "%d:%u:%d: add audio endpoint %#x\n", dev->devnum, iface_no, altno, fp->endpoint); + err = snd_usb_add_audio_stream(chip, stream, fp); + if (err < 0) { + kfree(fp->rate_table); + kfree(fp); + return err; + } + /* try to set the interface... */ + usb_set_interface(chip->dev, iface_no, altno); + snd_usb_init_pitch(chip, iface_no, alts, fp); + snd_usb_init_sample_rate(chip, iface_no, alts, fp, fp->rate_max); + } + return 0; +} + diff --git a/sound/usb/stream.h b/sound/usb/stream.h new file mode 100644 index 000000000000..c97f679fc84f --- /dev/null +++ b/sound/usb/stream.h @@ -0,0 +1,12 @@ +#ifndef __USBAUDIO_STREAM_H +#define __USBAUDIO_STREAM_H + +int snd_usb_parse_audio_interface(struct snd_usb_audio *chip, + int iface_no); + +int snd_usb_add_audio_stream(struct snd_usb_audio *chip, + int stream, + struct audioformat *fp); + +#endif /* __USBAUDIO_STREAM_H */ + diff --git a/sound/usb/urb.c b/sound/usb/urb.c deleted file mode 100644 index e184349aee83..000000000000 --- a/sound/usb/urb.c +++ /dev/null @@ -1,941 +0,0 @@ -/* - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA - * - */ - -#include <linux/gfp.h> -#include <linux/init.h> -#include <linux/usb.h> -#include <linux/usb/audio.h> - -#include <sound/core.h> -#include <sound/pcm.h> - -#include "usbaudio.h" -#include "helper.h" -#include "card.h" -#include "urb.h" -#include "pcm.h" - -/* - * convert a sampling rate into our full speed format (fs/1000 in Q16.16) - * this will overflow at approx 524 kHz - */ -static inline unsigned get_usb_full_speed_rate(unsigned int rate) -{ - return ((rate << 13) + 62) / 125; -} - -/* - * convert a sampling rate into USB high speed format (fs/8000 in Q16.16) - * this will overflow at approx 4 MHz - */ -static inline unsigned get_usb_high_speed_rate(unsigned int rate) -{ - return ((rate << 10) + 62) / 125; -} - -/* - * unlink active urbs. - */ -static int deactivate_urbs(struct snd_usb_substream *subs, int force, int can_sleep) -{ - struct snd_usb_audio *chip = subs->stream->chip; - unsigned int i; - int async; - - subs->running = 0; - - if (!force && subs->stream->chip->shutdown) /* to be sure... */ - return -EBADFD; - - async = !can_sleep && chip->async_unlink; - - if (!async && in_interrupt()) - return 0; - - for (i = 0; i < subs->nurbs; i++) { - if (test_bit(i, &subs->active_mask)) { - if (!test_and_set_bit(i, &subs->unlink_mask)) { - struct urb *u = subs->dataurb[i].urb; - if (async) - usb_unlink_urb(u); - else - usb_kill_urb(u); - } - } - } - if (subs->syncpipe) { - for (i = 0; i < SYNC_URBS; i++) { - if (test_bit(i+16, &subs->active_mask)) { - if (!test_and_set_bit(i+16, &subs->unlink_mask)) { - struct urb *u = subs->syncurb[i].urb; - if (async) - usb_unlink_urb(u); - else - usb_kill_urb(u); - } - } - } - } - return 0; -} - - -/* - * release a urb data - */ -static void release_urb_ctx(struct snd_urb_ctx *u) -{ - if (u->urb) { - if (u->buffer_size) - usb_free_coherent(u->subs->dev, u->buffer_size, - u->urb->transfer_buffer, - u->urb->transfer_dma); - usb_free_urb(u->urb); - u->urb = NULL; - } -} - -/* - * wait until all urbs are processed. - */ -static int wait_clear_urbs(struct snd_usb_substream *subs) -{ - unsigned long end_time = jiffies + msecs_to_jiffies(1000); - unsigned int i; - int alive; - - do { - alive = 0; - for (i = 0; i < subs->nurbs; i++) { - if (test_bit(i, &subs->active_mask)) - alive++; - } - if (subs->syncpipe) { - for (i = 0; i < SYNC_URBS; i++) { - if (test_bit(i + 16, &subs->active_mask)) - alive++; - } - } - if (! alive) - break; - schedule_timeout_uninterruptible(1); - } while (time_before(jiffies, end_time)); - if (alive) - snd_printk(KERN_ERR "timeout: still %d active urbs..\n", alive); - return 0; -} - -/* - * release a substream - */ -void snd_usb_release_substream_urbs(struct snd_usb_substream *subs, int force) -{ - int i; - - /* stop urbs (to be sure) */ - deactivate_urbs(subs, force, 1); - wait_clear_urbs(subs); - - for (i = 0; i < MAX_URBS; i++) - release_urb_ctx(&subs->dataurb[i]); - for (i = 0; i < SYNC_URBS; i++) - release_urb_ctx(&subs->syncurb[i]); - usb_free_coherent(subs->dev, SYNC_URBS * 4, - subs->syncbuf, subs->sync_dma); - subs->syncbuf = NULL; - subs->nurbs = 0; -} - -/* - * complete callback from data urb - */ -static void snd_complete_urb(struct urb *urb) -{ - struct snd_urb_ctx *ctx = urb->context; - struct snd_usb_substream *subs = ctx->subs; - struct snd_pcm_substream *substream = ctx->subs->pcm_substream; - int err = 0; - - if ((subs->running && subs->ops.retire(subs, substream->runtime, urb)) || - !subs->running || /* can be stopped during retire callback */ - (err = subs->ops.prepare(subs, substream->runtime, urb)) < 0 || - (err = usb_submit_urb(urb, GFP_ATOMIC)) < 0) { - clear_bit(ctx->index, &subs->active_mask); - if (err < 0) { - snd_printd(KERN_ERR "cannot submit urb (err = %d)\n", err); - snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); - } - } -} - - -/* - * complete callback from sync urb - */ -static void snd_complete_sync_urb(struct urb *urb) -{ - struct snd_urb_ctx *ctx = urb->context; - struct snd_usb_substream *subs = ctx->subs; - struct snd_pcm_substream *substream = ctx->subs->pcm_substream; - int err = 0; - - if ((subs->running && subs->ops.retire_sync(subs, substream->runtime, urb)) || - !subs->running || /* can be stopped during retire callback */ - (err = subs->ops.prepare_sync(subs, substream->runtime, urb)) < 0 || - (err = usb_submit_urb(urb, GFP_ATOMIC)) < 0) { - clear_bit(ctx->index + 16, &subs->active_mask); - if (err < 0) { - snd_printd(KERN_ERR "cannot submit sync urb (err = %d)\n", err); - snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); - } - } -} - - -/* - * initialize a substream for plaback/capture - */ -int snd_usb_init_substream_urbs(struct snd_usb_substream *subs, - unsigned int period_bytes, - unsigned int rate, - unsigned int frame_bits) -{ - unsigned int maxsize, i; - int is_playback = subs->direction == SNDRV_PCM_STREAM_PLAYBACK; - unsigned int urb_packs, total_packs, packs_per_ms; - struct snd_usb_audio *chip = subs->stream->chip; - - /* calculate the frequency in 16.16 format */ - if (snd_usb_get_speed(subs->dev) == USB_SPEED_FULL) - subs->freqn = get_usb_full_speed_rate(rate); - else - subs->freqn = get_usb_high_speed_rate(rate); - subs->freqm = subs->freqn; - subs->freqshift = INT_MIN; - /* calculate max. frequency */ - if (subs->maxpacksize) { - /* whatever fits into a max. size packet */ - maxsize = subs->maxpacksize; - subs->freqmax = (maxsize / (frame_bits >> 3)) - << (16 - subs->datainterval); - } else { - /* no max. packet size: just take 25% higher than nominal */ - subs->freqmax = subs->freqn + (subs->freqn >> 2); - maxsize = ((subs->freqmax + 0xffff) * (frame_bits >> 3)) - >> (16 - subs->datainterval); - } - subs->phase = 0; - - if (subs->fill_max) - subs->curpacksize = subs->maxpacksize; - else - subs->curpacksize = maxsize; - - if (snd_usb_get_speed(subs->dev) != USB_SPEED_FULL) - packs_per_ms = 8 >> subs->datainterval; - else - packs_per_ms = 1; - - if (is_playback) { - urb_packs = max(chip->nrpacks, 1); - urb_packs = min(urb_packs, (unsigned int)MAX_PACKS); - } else - urb_packs = 1; - urb_packs *= packs_per_ms; - if (subs->syncpipe) - urb_packs = min(urb_packs, 1U << subs->syncinterval); - - /* decide how many packets to be used */ - if (is_playback) { - unsigned int minsize, maxpacks; - /* determine how small a packet can be */ - minsize = (subs->freqn >> (16 - subs->datainterval)) - * (frame_bits >> 3); - /* with sync from device, assume it can be 12% lower */ - if (subs->syncpipe) - minsize -= minsize >> 3; - minsize = max(minsize, 1u); - total_packs = (period_bytes + minsize - 1) / minsize; - /* we need at least two URBs for queueing */ - if (total_packs < 2) { - total_packs = 2; - } else { - /* and we don't want too long a queue either */ - maxpacks = max(MAX_QUEUE * packs_per_ms, urb_packs * 2); - total_packs = min(total_packs, maxpacks); - } - } else { - while (urb_packs > 1 && urb_packs * maxsize >= period_bytes) - urb_packs >>= 1; - total_packs = MAX_URBS * urb_packs; - } - subs->nurbs = (total_packs + urb_packs - 1) / urb_packs; - if (subs->nurbs > MAX_URBS) { - /* too much... */ - subs->nurbs = MAX_URBS; - total_packs = MAX_URBS * urb_packs; - } else if (subs->nurbs < 2) { - /* too little - we need at least two packets - * to ensure contiguous playback/capture - */ - subs->nurbs = 2; - } - - /* allocate and initialize data urbs */ - for (i = 0; i < subs->nurbs; i++) { - struct snd_urb_ctx *u = &subs->dataurb[i]; - u->index = i; - u->subs = subs; - u->packets = (i + 1) * total_packs / subs->nurbs - - i * total_packs / subs->nurbs; - u->buffer_size = maxsize * u->packets; - if (subs->fmt_type == UAC_FORMAT_TYPE_II) - u->packets++; /* for transfer delimiter */ - u->urb = usb_alloc_urb(u->packets, GFP_KERNEL); - if (!u->urb) - goto out_of_memory; - u->urb->transfer_buffer = - usb_alloc_coherent(subs->dev, u->buffer_size, - GFP_KERNEL, &u->urb->transfer_dma); - if (!u->urb->transfer_buffer) - goto out_of_memory; - u->urb->pipe = subs->datapipe; - u->urb->transfer_flags = URB_ISO_ASAP | URB_NO_TRANSFER_DMA_MAP; - u->urb->interval = 1 << subs->datainterval; - u->urb->context = u; - u->urb->complete = snd_complete_urb; - } - - if (subs->syncpipe) { - /* allocate and initialize sync urbs */ - subs->syncbuf = usb_alloc_coherent(subs->dev, SYNC_URBS * 4, - GFP_KERNEL, &subs->sync_dma); - if (!subs->syncbuf) - goto out_of_memory; - for (i = 0; i < SYNC_URBS; i++) { - struct snd_urb_ctx *u = &subs->syncurb[i]; - u->index = i; - u->subs = subs; - u->packets = 1; - u->urb = usb_alloc_urb(1, GFP_KERNEL); - if (!u->urb) - goto out_of_memory; - u->urb->transfer_buffer = subs->syncbuf + i * 4; - u->urb->transfer_dma = subs->sync_dma + i * 4; - u->urb->transfer_buffer_length = 4; - u->urb->pipe = subs->syncpipe; - u->urb->transfer_flags = URB_ISO_ASAP | - URB_NO_TRANSFER_DMA_MAP; - u->urb->number_of_packets = 1; - u->urb->interval = 1 << subs->syncinterval; - u->urb->context = u; - u->urb->complete = snd_complete_sync_urb; - } - } - return 0; - -out_of_memory: - snd_usb_release_substream_urbs(subs, 0); - return -ENOMEM; -} - -/* - * prepare urb for full speed capture sync pipe - * - * fill the length and offset of each urb descriptor. - * the fixed 10.14 frequency is passed through the pipe. - */ -static int prepare_capture_sync_urb(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime, - struct urb *urb) -{ - unsigned char *cp = urb->transfer_buffer; - struct snd_urb_ctx *ctx = urb->context; - - urb->dev = ctx->subs->dev; /* we need to set this at each time */ - urb->iso_frame_desc[0].length = 3; - urb->iso_frame_desc[0].offset = 0; - cp[0] = subs->freqn >> 2; - cp[1] = subs->freqn >> 10; - cp[2] = subs->freqn >> 18; - return 0; -} - -/* - * prepare urb for high speed capture sync pipe - * - * fill the length and offset of each urb descriptor. - * the fixed 12.13 frequency is passed as 16.16 through the pipe. - */ -static int prepare_capture_sync_urb_hs(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime, - struct urb *urb) -{ - unsigned char *cp = urb->transfer_buffer; - struct snd_urb_ctx *ctx = urb->context; - - urb->dev = ctx->subs->dev; /* we need to set this at each time */ - urb->iso_frame_desc[0].length = 4; - urb->iso_frame_desc[0].offset = 0; - cp[0] = subs->freqn; - cp[1] = subs->freqn >> 8; - cp[2] = subs->freqn >> 16; - cp[3] = subs->freqn >> 24; - return 0; -} - -/* - * process after capture sync complete - * - nothing to do - */ -static int retire_capture_sync_urb(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime, - struct urb *urb) -{ - return 0; -} - -/* - * prepare urb for capture data pipe - * - * fill the offset and length of each descriptor. - * - * we use a temporary buffer to write the captured data. - * since the length of written data is determined by host, we cannot - * write onto the pcm buffer directly... the data is thus copied - * later at complete callback to the global buffer. - */ -static int prepare_capture_urb(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime, - struct urb *urb) -{ - int i, offs; - struct snd_urb_ctx *ctx = urb->context; - - offs = 0; - urb->dev = ctx->subs->dev; /* we need to set this at each time */ - for (i = 0; i < ctx->packets; i++) { - urb->iso_frame_desc[i].offset = offs; - urb->iso_frame_desc[i].length = subs->curpacksize; - offs += subs->curpacksize; - } - urb->transfer_buffer_length = offs; - urb->number_of_packets = ctx->packets; - return 0; -} - -/* - * process after capture complete - * - * copy the data from each desctiptor to the pcm buffer, and - * update the current position. - */ -static int retire_capture_urb(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime, - struct urb *urb) -{ - unsigned long flags; - unsigned char *cp; - int i; - unsigned int stride, frames, bytes, oldptr; - int period_elapsed = 0; - - stride = runtime->frame_bits >> 3; - - for (i = 0; i < urb->number_of_packets; i++) { - cp = (unsigned char *)urb->transfer_buffer + urb->iso_frame_desc[i].offset; - if (urb->iso_frame_desc[i].status) { - snd_printd(KERN_ERR "frame %d active: %d\n", i, urb->iso_frame_desc[i].status); - // continue; - } - bytes = urb->iso_frame_desc[i].actual_length; - frames = bytes / stride; - if (!subs->txfr_quirk) - bytes = frames * stride; - if (bytes % (runtime->sample_bits >> 3) != 0) { -#ifdef CONFIG_SND_DEBUG_VERBOSE - int oldbytes = bytes; -#endif - bytes = frames * stride; - snd_printdd(KERN_ERR "Corrected urb data len. %d->%d\n", - oldbytes, bytes); - } - /* update the current pointer */ - spin_lock_irqsave(&subs->lock, flags); - oldptr = subs->hwptr_done; - subs->hwptr_done += bytes; - if (subs->hwptr_done >= runtime->buffer_size * stride) - subs->hwptr_done -= runtime->buffer_size * stride; - frames = (bytes + (oldptr % stride)) / stride; - subs->transfer_done += frames; - if (subs->transfer_done >= runtime->period_size) { - subs->transfer_done -= runtime->period_size; - period_elapsed = 1; - } - spin_unlock_irqrestore(&subs->lock, flags); - /* copy a data chunk */ - if (oldptr + bytes > runtime->buffer_size * stride) { - unsigned int bytes1 = - runtime->buffer_size * stride - oldptr; - memcpy(runtime->dma_area + oldptr, cp, bytes1); - memcpy(runtime->dma_area, cp + bytes1, bytes - bytes1); - } else { - memcpy(runtime->dma_area + oldptr, cp, bytes); - } - } - if (period_elapsed) - snd_pcm_period_elapsed(subs->pcm_substream); - return 0; -} - -/* - * Process after capture complete when paused. Nothing to do. - */ -static int retire_paused_capture_urb(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime, - struct urb *urb) -{ - return 0; -} - - -/* - * prepare urb for playback sync pipe - * - * set up the offset and length to receive the current frequency. - */ -static int prepare_playback_sync_urb(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime, - struct urb *urb) -{ - struct snd_urb_ctx *ctx = urb->context; - - urb->dev = ctx->subs->dev; /* we need to set this at each time */ - urb->iso_frame_desc[0].length = min(4u, ctx->subs->syncmaxsize); - urb->iso_frame_desc[0].offset = 0; - return 0; -} - -/* - * process after playback sync complete - * - * Full speed devices report feedback values in 10.14 format as samples per - * frame, high speed devices in 16.16 format as samples per microframe. - * Because the Audio Class 1 spec was written before USB 2.0, many high speed - * devices use a wrong interpretation, some others use an entirely different - * format. Therefore, we cannot predict what format any particular device uses - * and must detect it automatically. - */ -static int retire_playback_sync_urb(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime, - struct urb *urb) -{ - unsigned int f; - int shift; - unsigned long flags; - - if (urb->iso_frame_desc[0].status != 0 || - urb->iso_frame_desc[0].actual_length < 3) - return 0; - - f = le32_to_cpup(urb->transfer_buffer); - if (urb->iso_frame_desc[0].actual_length == 3) - f &= 0x00ffffff; - else - f &= 0x0fffffff; - if (f == 0) - return 0; - - if (unlikely(subs->freqshift == INT_MIN)) { - /* - * The first time we see a feedback value, determine its format - * by shifting it left or right until it matches the nominal - * frequency value. This assumes that the feedback does not - * differ from the nominal value more than +50% or -25%. - */ - shift = 0; - while (f < subs->freqn - subs->freqn / 4) { - f <<= 1; - shift++; - } - while (f > subs->freqn + subs->freqn / 2) { - f >>= 1; - shift--; - } - subs->freqshift = shift; - } - else if (subs->freqshift >= 0) - f <<= subs->freqshift; - else - f >>= -subs->freqshift; - - if (likely(f >= subs->freqn - subs->freqn / 8 && f <= subs->freqmax)) { - /* - * If the frequency looks valid, set it. - * This value is referred to in prepare_playback_urb(). - */ - spin_lock_irqsave(&subs->lock, flags); - subs->freqm = f; - spin_unlock_irqrestore(&subs->lock, flags); - } else { - /* - * Out of range; maybe the shift value is wrong. - * Reset it so that we autodetect again the next time. - */ - subs->freqshift = INT_MIN; - } - - return 0; -} - -/* determine the number of frames in the next packet */ -static int snd_usb_audio_next_packet_size(struct snd_usb_substream *subs) -{ - if (subs->fill_max) - return subs->maxframesize; - else { - subs->phase = (subs->phase & 0xffff) - + (subs->freqm << subs->datainterval); - return min(subs->phase >> 16, subs->maxframesize); - } -} - -/* - * Prepare urb for streaming before playback starts or when paused. - * - * We don't have any data, so we send silence. - */ -static int prepare_nodata_playback_urb(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime, - struct urb *urb) -{ - unsigned int i, offs, counts; - struct snd_urb_ctx *ctx = urb->context; - int stride = runtime->frame_bits >> 3; - - offs = 0; - urb->dev = ctx->subs->dev; - for (i = 0; i < ctx->packets; ++i) { - counts = snd_usb_audio_next_packet_size(subs); - urb->iso_frame_desc[i].offset = offs * stride; - urb->iso_frame_desc[i].length = counts * stride; - offs += counts; - } - urb->number_of_packets = ctx->packets; - urb->transfer_buffer_length = offs * stride; - memset(urb->transfer_buffer, - runtime->format == SNDRV_PCM_FORMAT_U8 ? 0x80 : 0, - offs * stride); - return 0; -} - -/* - * prepare urb for playback data pipe - * - * Since a URB can handle only a single linear buffer, we must use double - * buffering when the data to be transferred overflows the buffer boundary. - * To avoid inconsistencies when updating hwptr_done, we use double buffering - * for all URBs. - */ -static int prepare_playback_urb(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime, - struct urb *urb) -{ - int i, stride; - unsigned int counts, frames, bytes; - unsigned long flags; - int period_elapsed = 0; - struct snd_urb_ctx *ctx = urb->context; - - stride = runtime->frame_bits >> 3; - - frames = 0; - urb->dev = ctx->subs->dev; /* we need to set this at each time */ - urb->number_of_packets = 0; - spin_lock_irqsave(&subs->lock, flags); - for (i = 0; i < ctx->packets; i++) { - counts = snd_usb_audio_next_packet_size(subs); - /* set up descriptor */ - urb->iso_frame_desc[i].offset = frames * stride; - urb->iso_frame_desc[i].length = counts * stride; - frames += counts; - urb->number_of_packets++; - subs->transfer_done += counts; - if (subs->transfer_done >= runtime->period_size) { - subs->transfer_done -= runtime->period_size; - period_elapsed = 1; - if (subs->fmt_type == UAC_FORMAT_TYPE_II) { - if (subs->transfer_done > 0) { - /* FIXME: fill-max mode is not - * supported yet */ - frames -= subs->transfer_done; - counts -= subs->transfer_done; - urb->iso_frame_desc[i].length = - counts * stride; - subs->transfer_done = 0; - } - i++; - if (i < ctx->packets) { - /* add a transfer delimiter */ - urb->iso_frame_desc[i].offset = - frames * stride; - urb->iso_frame_desc[i].length = 0; - urb->number_of_packets++; - } - break; - } - } - if (period_elapsed) /* finish at the period boundary */ - break; - } - bytes = frames * stride; - if (subs->hwptr_done + bytes > runtime->buffer_size * stride) { - /* err, the transferred area goes over buffer boundary. */ - unsigned int bytes1 = - runtime->buffer_size * stride - subs->hwptr_done; - memcpy(urb->transfer_buffer, - runtime->dma_area + subs->hwptr_done, bytes1); - memcpy(urb->transfer_buffer + bytes1, - runtime->dma_area, bytes - bytes1); - } else { - memcpy(urb->transfer_buffer, - runtime->dma_area + subs->hwptr_done, bytes); - } - subs->hwptr_done += bytes; - if (subs->hwptr_done >= runtime->buffer_size * stride) - subs->hwptr_done -= runtime->buffer_size * stride; - runtime->delay += frames; - spin_unlock_irqrestore(&subs->lock, flags); - urb->transfer_buffer_length = bytes; - if (period_elapsed) - snd_pcm_period_elapsed(subs->pcm_substream); - return 0; -} - -/* - * process after playback data complete - * - decrease the delay count again - */ -static int retire_playback_urb(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime, - struct urb *urb) -{ - unsigned long flags; - int stride = runtime->frame_bits >> 3; - int processed = urb->transfer_buffer_length / stride; - - spin_lock_irqsave(&subs->lock, flags); - if (processed > runtime->delay) - runtime->delay = 0; - else - runtime->delay -= processed; - spin_unlock_irqrestore(&subs->lock, flags); - return 0; -} - -static const char *usb_error_string(int err) -{ - switch (err) { - case -ENODEV: - return "no device"; - case -ENOENT: - return "endpoint not enabled"; - case -EPIPE: - return "endpoint stalled"; - case -ENOSPC: - return "not enough bandwidth"; - case -ESHUTDOWN: - return "device disabled"; - case -EHOSTUNREACH: - return "device suspended"; - case -EINVAL: - case -EAGAIN: - case -EFBIG: - case -EMSGSIZE: - return "internal error"; - default: - return "unknown error"; - } -} - -/* - * set up and start data/sync urbs - */ -static int start_urbs(struct snd_usb_substream *subs, struct snd_pcm_runtime *runtime) -{ - unsigned int i; - int err; - - if (subs->stream->chip->shutdown) - return -EBADFD; - - for (i = 0; i < subs->nurbs; i++) { - if (snd_BUG_ON(!subs->dataurb[i].urb)) - return -EINVAL; - if (subs->ops.prepare(subs, runtime, subs->dataurb[i].urb) < 0) { - snd_printk(KERN_ERR "cannot prepare datapipe for urb %d\n", i); - goto __error; - } - } - if (subs->syncpipe) { - for (i = 0; i < SYNC_URBS; i++) { - if (snd_BUG_ON(!subs->syncurb[i].urb)) - return -EINVAL; - if (subs->ops.prepare_sync(subs, runtime, subs->syncurb[i].urb) < 0) { - snd_printk(KERN_ERR "cannot prepare syncpipe for urb %d\n", i); - goto __error; - } - } - } - - subs->active_mask = 0; - subs->unlink_mask = 0; - subs->running = 1; - for (i = 0; i < subs->nurbs; i++) { - err = usb_submit_urb(subs->dataurb[i].urb, GFP_ATOMIC); - if (err < 0) { - snd_printk(KERN_ERR "cannot submit datapipe " - "for urb %d, error %d: %s\n", - i, err, usb_error_string(err)); - goto __error; - } - set_bit(i, &subs->active_mask); - } - if (subs->syncpipe) { - for (i = 0; i < SYNC_URBS; i++) { - err = usb_submit_urb(subs->syncurb[i].urb, GFP_ATOMIC); - if (err < 0) { - snd_printk(KERN_ERR "cannot submit syncpipe " - "for urb %d, error %d: %s\n", - i, err, usb_error_string(err)); - goto __error; - } - set_bit(i + 16, &subs->active_mask); - } - } - return 0; - - __error: - // snd_pcm_stop(subs->pcm_substream, SNDRV_PCM_STATE_XRUN); - deactivate_urbs(subs, 0, 0); - return -EPIPE; -} - - -/* - */ -static struct snd_urb_ops audio_urb_ops[2] = { - { - .prepare = prepare_nodata_playback_urb, - .retire = retire_playback_urb, - .prepare_sync = prepare_playback_sync_urb, - .retire_sync = retire_playback_sync_urb, - }, - { - .prepare = prepare_capture_urb, - .retire = retire_capture_urb, - .prepare_sync = prepare_capture_sync_urb, - .retire_sync = retire_capture_sync_urb, - }, -}; - -/* - * initialize the substream instance. - */ - -void snd_usb_init_substream(struct snd_usb_stream *as, - int stream, struct audioformat *fp) -{ - struct snd_usb_substream *subs = &as->substream[stream]; - - INIT_LIST_HEAD(&subs->fmt_list); - spin_lock_init(&subs->lock); - - subs->stream = as; - subs->direction = stream; - subs->dev = as->chip->dev; - subs->txfr_quirk = as->chip->txfr_quirk; - subs->ops = audio_urb_ops[stream]; - if (snd_usb_get_speed(subs->dev) >= USB_SPEED_HIGH) - subs->ops.prepare_sync = prepare_capture_sync_urb_hs; - - snd_usb_set_pcm_ops(as->pcm, stream); - - list_add_tail(&fp->list, &subs->fmt_list); - subs->formats |= fp->formats; - subs->endpoint = fp->endpoint; - subs->num_formats++; - subs->fmt_type = fp->fmt_type; -} - -int snd_usb_substream_playback_trigger(struct snd_pcm_substream *substream, int cmd) -{ - struct snd_usb_substream *subs = substream->runtime->private_data; - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - subs->ops.prepare = prepare_playback_urb; - return 0; - case SNDRV_PCM_TRIGGER_STOP: - return deactivate_urbs(subs, 0, 0); - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - subs->ops.prepare = prepare_nodata_playback_urb; - return 0; - } - - return -EINVAL; -} - -int snd_usb_substream_capture_trigger(struct snd_pcm_substream *substream, int cmd) -{ - struct snd_usb_substream *subs = substream->runtime->private_data; - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - subs->ops.retire = retire_capture_urb; - return start_urbs(subs, substream->runtime); - case SNDRV_PCM_TRIGGER_STOP: - return deactivate_urbs(subs, 0, 0); - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - subs->ops.retire = retire_paused_capture_urb; - return 0; - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - subs->ops.retire = retire_capture_urb; - return 0; - } - - return -EINVAL; -} - -int snd_usb_substream_prepare(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime) -{ - /* clear urbs (to be sure) */ - deactivate_urbs(subs, 0, 1); - wait_clear_urbs(subs); - - /* for playback, submit the URBs now; otherwise, the first hwptr_done - * updates for all URBs would happen at the same time when starting */ - if (subs->direction == SNDRV_PCM_STREAM_PLAYBACK) { - subs->ops.prepare = prepare_nodata_playback_urb; - return start_urbs(subs, runtime); - } - - return 0; -} - diff --git a/sound/usb/urb.h b/sound/usb/urb.h deleted file mode 100644 index 888da38079cf..000000000000 --- a/sound/usb/urb.h +++ /dev/null @@ -1,21 +0,0 @@ -#ifndef __USBAUDIO_URB_H -#define __USBAUDIO_URB_H - -void snd_usb_init_substream(struct snd_usb_stream *as, - int stream, - struct audioformat *fp); - -int snd_usb_init_substream_urbs(struct snd_usb_substream *subs, - unsigned int period_bytes, - unsigned int rate, - unsigned int frame_bits); - -void snd_usb_release_substream_urbs(struct snd_usb_substream *subs, int force); - -int snd_usb_substream_prepare(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime); - -int snd_usb_substream_playback_trigger(struct snd_pcm_substream *substream, int cmd); -int snd_usb_substream_capture_trigger(struct snd_pcm_substream *substream, int cmd); - -#endif /* __USBAUDIO_URB_H */ diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 1e79986b5777..3e2b03577936 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -80,6 +80,7 @@ enum quirk_type { QUIRK_MIDI_CME, QUIRK_MIDI_AKAI, QUIRK_MIDI_US122L, + QUIRK_MIDI_FTDI, QUIRK_AUDIO_STANDARD_INTERFACE, QUIRK_AUDIO_FIXED_ENDPOINT, QUIRK_AUDIO_EDIROL_UAXX, |