diff options
| author | Mauro Carvalho Chehab <mchehab@redhat.com> | 2012-03-19 13:41:24 -0300 |
|---|---|---|
| committer | Mauro Carvalho Chehab <mchehab@redhat.com> | 2012-03-19 13:41:24 -0300 |
| commit | 9ce28d827f74d0acdd058bded8bab5309b0f5c8f (patch) | |
| tree | 634f22e8df9c7fd3966b3639e3e997436751ca50 /sound | |
| parent | f074ff92b5b26f3a559fab1203c36e140ea8d067 (diff) | |
| parent | c16fa4f2ad19908a47c63d8fa436a1178438c7e7 (diff) | |
| download | linux-9ce28d827f74d0acdd058bded8bab5309b0f5c8f.tar.gz linux-9ce28d827f74d0acdd058bded8bab5309b0f5c8f.tar.bz2 linux-9ce28d827f74d0acdd058bded8bab5309b0f5c8f.zip | |
Merge tag 'v3.3' into staging/for_v3.4
* tag 'v3.3': (1646 commits)
Linux 3.3
Don't limit non-nested epoll paths
netfilter: ctnetlink: fix race between delete and timeout expiration
ipv6: Don't dev_hold(dev) in ip6_mc_find_dev_rcu.
nilfs2: fix NULL pointer dereference in nilfs_load_super_block()
nilfs2: clamp ns_r_segments_percentage to [1, 99]
afs: Remote abort can cause BUG in rxrpc code
afs: Read of file returns EBADMSG
C6X: remove dead code from entry.S
wimax/i2400m: fix erroneous NETDEV_TX_BUSY use
net/hyperv: fix erroneous NETDEV_TX_BUSY use
net/usbnet: reserve headroom on rx skbs
bnx2x: fix memory leak in bnx2x_init_firmware()
bnx2x: fix a crash on corrupt firmware file
sch_sfq: revert dont put new flow at the end of flows
ipv6: fix icmp6_dst_alloc()
MAINTAINERS: Add Serge as maintainer of capabilities
drivers/video/backlight/s6e63m0.c: fix corruption storing gamma mode
MAINTAINERS: add entry for exynos mipi display drivers
MAINTAINERS: fix link to Gustavo Padovans tree
...
Diffstat (limited to 'sound')
43 files changed, 638 insertions, 482 deletions
diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index dac3633507c9..a68aed7fce02 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -441,19 +441,22 @@ snd_compr_set_params(struct snd_compr_stream *stream, unsigned long arg) params = kmalloc(sizeof(*params), GFP_KERNEL); if (!params) return -ENOMEM; - if (copy_from_user(params, (void __user *)arg, sizeof(*params))) - return -EFAULT; + if (copy_from_user(params, (void __user *)arg, sizeof(*params))) { + retval = -EFAULT; + goto out; + } retval = snd_compr_allocate_buffer(stream, params); if (retval) { - kfree(params); - return -ENOMEM; + retval = -ENOMEM; + goto out; } retval = stream->ops->set_params(stream, params); if (retval) goto out; stream->runtime->state = SNDRV_PCM_STATE_SETUP; - } else + } else { return -EPERM; + } out: kfree(params); return retval; diff --git a/sound/isa/sb/emu8000_patch.c b/sound/isa/sb/emu8000_patch.c index e09f144177f5..c99c6078be33 100644 --- a/sound/isa/sb/emu8000_patch.c +++ b/sound/isa/sb/emu8000_patch.c @@ -22,7 +22,6 @@ #include "emu8000_local.h" #include <asm/uaccess.h> #include <linux/moduleparam.h> -#include <linux/moduleparam.h> static int emu8000_reset_addr; module_param(emu8000_reset_addr, int, 0444); diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c index 95ffa6a9db6e..496f14c1a731 100644 --- a/sound/pci/azt3328.c +++ b/sound/pci/azt3328.c @@ -2684,10 +2684,9 @@ snd_azf3328_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) err = snd_opl3_hwdep_new(opl3, 0, 1, NULL); if (err < 0) goto out_err; + opl3->private_data = chip; } - opl3->private_data = chip; - sprintf(card->longname, "%s at 0x%lx, irq %i", card->shortname, chip->ctrl_io, chip->irq); diff --git a/sound/pci/hda/alc880_quirks.c b/sound/pci/hda/alc880_quirks.c index 5b68435d195b..501501ef36a9 100644 --- a/sound/pci/hda/alc880_quirks.c +++ b/sound/pci/hda/alc880_quirks.c @@ -762,16 +762,22 @@ static void alc880_uniwill_unsol_event(struct hda_codec *codec, /* Looks like the unsol event is incompatible with the standard * definition. 4bit tag is placed at 28 bit! */ - switch (res >> 28) { + res >>= 28; + switch (res) { case ALC_MIC_EVENT: alc88x_simple_mic_automute(codec); break; default: - alc_sku_unsol_event(codec, res); + alc_exec_unsol_event(codec, res); break; } } +static void alc880_unsol_event(struct hda_codec *codec, unsigned int res) +{ + alc_exec_unsol_event(codec, res >> 28); +} + static void alc880_uniwill_p53_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -800,10 +806,11 @@ static void alc880_uniwill_p53_unsol_event(struct hda_codec *codec, /* Looks like the unsol event is incompatible with the standard * definition. 4bit tag is placed at 28 bit! */ - if ((res >> 28) == ALC_DCVOL_EVENT) + res >>= 28; + if (res == ALC_DCVOL_EVENT) alc880_uniwill_p53_dcvol_automute(codec); else - alc_sku_unsol_event(codec, res); + alc_exec_unsol_event(codec, res); } /* @@ -1677,7 +1684,7 @@ static const struct alc_config_preset alc880_presets[] = { .channel_mode = alc880_lg_ch_modes, .need_dac_fix = 1, .input_mux = &alc880_lg_capture_source, - .unsol_event = alc_sku_unsol_event, + .unsol_event = alc880_unsol_event, .setup = alc880_lg_setup, .init_hook = alc_hp_automute, #ifdef CONFIG_SND_HDA_POWER_SAVE diff --git a/sound/pci/hda/alc882_quirks.c b/sound/pci/hda/alc882_quirks.c index bdf0ed4ab3e2..bb364a53f546 100644 --- a/sound/pci/hda/alc882_quirks.c +++ b/sound/pci/hda/alc882_quirks.c @@ -730,6 +730,11 @@ static void alc889A_mb31_unsol_event(struct hda_codec *codec, unsigned int res) alc889A_mb31_automute(codec); } +static void alc882_unsol_event(struct hda_codec *codec, unsigned int res) +{ + alc_exec_unsol_event(codec, res >> 26); +} + /* * configuration and preset */ @@ -775,7 +780,7 @@ static const struct alc_config_preset alc882_presets[] = { .channel_mode = alc885_mba21_ch_modes, .num_channel_mode = ARRAY_SIZE(alc885_mba21_ch_modes), .input_mux = &alc882_capture_source, - .unsol_event = alc_sku_unsol_event, + .unsol_event = alc882_unsol_event, .setup = alc885_mba21_setup, .init_hook = alc_hp_automute, }, @@ -791,7 +796,7 @@ static const struct alc_config_preset alc882_presets[] = { .input_mux = &alc882_capture_source, .dig_out_nid = ALC882_DIGOUT_NID, .dig_in_nid = ALC882_DIGIN_NID, - .unsol_event = alc_sku_unsol_event, + .unsol_event = alc882_unsol_event, .setup = alc885_mbp3_setup, .init_hook = alc_hp_automute, }, @@ -806,7 +811,7 @@ static const struct alc_config_preset alc882_presets[] = { .input_mux = &mb5_capture_source, .dig_out_nid = ALC882_DIGOUT_NID, .dig_in_nid = ALC882_DIGIN_NID, - .unsol_event = alc_sku_unsol_event, + .unsol_event = alc882_unsol_event, .setup = alc885_mb5_setup, .init_hook = alc_hp_automute, }, @@ -821,7 +826,7 @@ static const struct alc_config_preset alc882_presets[] = { .input_mux = &macmini3_capture_source, .dig_out_nid = ALC882_DIGOUT_NID, .dig_in_nid = ALC882_DIGIN_NID, - .unsol_event = alc_sku_unsol_event, + .unsol_event = alc882_unsol_event, .setup = alc885_macmini3_setup, .init_hook = alc_hp_automute, }, @@ -836,7 +841,7 @@ static const struct alc_config_preset alc882_presets[] = { .input_mux = &alc889A_imac91_capture_source, .dig_out_nid = ALC882_DIGOUT_NID, .dig_in_nid = ALC882_DIGIN_NID, - .unsol_event = alc_sku_unsol_event, + .unsol_event = alc882_unsol_event, .setup = alc885_imac91_setup, .init_hook = alc_hp_automute, }, diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 4df72c0e8c37..684307372d73 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1447,7 +1447,7 @@ void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid, for (i = 0; i < c->cvt_setups.used; i++) { p = snd_array_elem(&c->cvt_setups, i); if (!p->active && p->stream_tag == stream_tag && - get_wcaps_type(get_wcaps(codec, p->nid)) == type) + get_wcaps_type(get_wcaps(c, p->nid)) == type) p->dirty = 1; } } @@ -1759,7 +1759,11 @@ static void put_vol_mute(struct hda_codec *codec, struct hda_amp_info *info, parm = ch ? AC_AMP_SET_RIGHT : AC_AMP_SET_LEFT; parm |= direction == HDA_OUTPUT ? AC_AMP_SET_OUTPUT : AC_AMP_SET_INPUT; parm |= index << AC_AMP_SET_INDEX_SHIFT; - parm |= val; + if ((val & HDA_AMP_MUTE) && !(info->amp_caps & AC_AMPCAP_MUTE) && + (info->amp_caps & AC_AMPCAP_MIN_MUTE)) + ; /* set the zero value as a fake mute */ + else + parm |= val; snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, parm); info->vol[ch] = val; } @@ -2026,7 +2030,7 @@ int snd_hda_mixer_amp_tlv(struct snd_kcontrol *kcontrol, int op_flag, val1 = -((caps & AC_AMPCAP_OFFSET) >> AC_AMPCAP_OFFSET_SHIFT); val1 += ofs; val1 = ((int)val1) * ((int)val2); - if (min_mute) + if (min_mute || (caps & AC_AMPCAP_MIN_MUTE)) val2 |= TLV_DB_SCALE_MUTE; if (put_user(SNDRV_CTL_TLVT_DB_SCALE, _tlv)) return -EFAULT; @@ -5114,7 +5118,7 @@ static int fill_audio_out_name(struct hda_codec *codec, hda_nid_t nid, const char *pfx = "", *sfx = ""; /* handle as a speaker if it's a fixed line-out */ - if (!strcmp(name, "Line-Out") && attr == INPUT_PIN_ATTR_INT) + if (!strcmp(name, "Line Out") && attr == INPUT_PIN_ATTR_INT) name = "Speaker"; /* check the location */ switch (attr) { @@ -5173,7 +5177,7 @@ int snd_hda_get_pin_label(struct hda_codec *codec, hda_nid_t nid, switch (get_defcfg_device(def_conf)) { case AC_JACK_LINE_OUT: - return fill_audio_out_name(codec, nid, cfg, "Line-Out", + return fill_audio_out_name(codec, nid, cfg, "Line Out", label, maxlen, indexp); case AC_JACK_SPEAKER: return fill_audio_out_name(codec, nid, cfg, "Speaker", diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index e9f71dc0d464..f0f1943a4b2c 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -298,6 +298,9 @@ enum { #define AC_AMPCAP_MUTE (1<<31) /* mute capable */ #define AC_AMPCAP_MUTE_SHIFT 31 +/* driver-specific amp-caps: using bits 24-30 */ +#define AC_AMPCAP_MIN_MUTE (1 << 30) /* min-volume = mute */ + /* Connection list */ #define AC_CLIST_LENGTH (0x7f<<0) #define AC_CLIST_LONG (1<<7) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index fb35474c1203..95dfb6874941 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -469,6 +469,7 @@ struct azx { unsigned int irq_pending_warned :1; unsigned int probing :1; /* codec probing phase */ unsigned int snoop:1; + unsigned int align_buffer_size:1; /* for debugging */ unsigned int last_cmd[AZX_MAX_CODECS]; @@ -1690,7 +1691,7 @@ 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) + if (chip->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 @@ -2773,8 +2774,9 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, } /* disable buffer size rounding to 128-byte multiples if supported */ + chip->align_buffer_size = align_buffer_size; if (chip->driver_caps & AZX_DCAPS_BUFSIZE) - align_buffer_size = 0; + chip->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))) diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index d8a35da0803f..9d819c4b4923 100644 --- a/sound/pci/hda/hda_jack.c +++ b/sound/pci/hda/hda_jack.c @@ -282,7 +282,8 @@ int snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid, EXPORT_SYMBOL_HDA(snd_hda_jack_add_kctl); static int add_jack_kctl(struct hda_codec *codec, hda_nid_t nid, - const struct auto_pin_cfg *cfg) + const struct auto_pin_cfg *cfg, + char *lastname, int *lastidx) { unsigned int def_conf, conn; char name[44]; @@ -298,6 +299,10 @@ static int add_jack_kctl(struct hda_codec *codec, hda_nid_t nid, return 0; snd_hda_get_pin_label(codec, nid, cfg, name, sizeof(name), &idx); + if (!strcmp(name, lastname) && idx == *lastidx) + idx++; + strncpy(lastname, name, 44); + *lastidx = idx; err = snd_hda_jack_add_kctl(codec, nid, name, idx); if (err < 0) return err; @@ -311,41 +316,42 @@ int snd_hda_jack_add_kctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg) { const hda_nid_t *p; - int i, err; + int i, err, lastidx = 0; + char lastname[44] = ""; for (i = 0, p = cfg->line_out_pins; i < cfg->line_outs; i++, p++) { - err = add_jack_kctl(codec, *p, cfg); + err = add_jack_kctl(codec, *p, cfg, lastname, &lastidx); if (err < 0) return err; } for (i = 0, p = cfg->hp_pins; i < cfg->hp_outs; i++, p++) { if (*p == *cfg->line_out_pins) /* might be duplicated */ break; - err = add_jack_kctl(codec, *p, cfg); + err = add_jack_kctl(codec, *p, cfg, lastname, &lastidx); if (err < 0) return err; } for (i = 0, p = cfg->speaker_pins; i < cfg->speaker_outs; i++, p++) { if (*p == *cfg->line_out_pins) /* might be duplicated */ break; - err = add_jack_kctl(codec, *p, cfg); + err = add_jack_kctl(codec, *p, cfg, lastname, &lastidx); if (err < 0) return err; } for (i = 0; i < cfg->num_inputs; i++) { - err = add_jack_kctl(codec, cfg->inputs[i].pin, cfg); + err = add_jack_kctl(codec, cfg->inputs[i].pin, cfg, lastname, &lastidx); if (err < 0) return err; } for (i = 0, p = cfg->dig_out_pins; i < cfg->dig_outs; i++, p++) { - err = add_jack_kctl(codec, *p, cfg); + err = add_jack_kctl(codec, *p, cfg, lastname, &lastidx); if (err < 0) return err; } - err = add_jack_kctl(codec, cfg->dig_in_pin, cfg); + err = add_jack_kctl(codec, cfg->dig_in_pin, cfg, lastname, &lastidx); if (err < 0) return err; - err = add_jack_kctl(codec, cfg->mono_out_pin, cfg); + err = add_jack_kctl(codec, cfg->mono_out_pin, cfg, lastname, &lastidx); if (err < 0) return err; return 0; diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 35abe3c62908..21d91d580da8 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -728,18 +728,19 @@ static int ca0132_hp_switch_put(struct snd_kcontrol *kcontrol, err = chipio_read(codec, REG_CODEC_MUTE, &data); if (err < 0) - return err; + goto exit; /* *valp 0 is mute, 1 is unmute */ data = (data & 0x7f) | (*valp ? 0 : 0x80); - chipio_write(codec, REG_CODEC_MUTE, data); + err = chipio_write(codec, REG_CODEC_MUTE, data); if (err < 0) - return err; + goto exit; spec->curr_hp_switch = *valp; + exit: snd_hda_power_down(codec); - return 1; + return err < 0 ? err : 1; } static int ca0132_speaker_switch_get(struct snd_kcontrol *kcontrol, @@ -770,18 +771,19 @@ static int ca0132_speaker_switch_put(struct snd_kcontrol *kcontrol, err = chipio_read(codec, REG_CODEC_MUTE, &data); if (err < 0) - return err; + goto exit; /* *valp 0 is mute, 1 is unmute */ data = (data & 0xef) | (*valp ? 0 : 0x10); - chipio_write(codec, REG_CODEC_MUTE, data); + err = chipio_write(codec, REG_CODEC_MUTE, data); if (err < 0) - return err; + goto exit; spec->curr_speaker_switch = *valp; + exit: snd_hda_power_down(codec); - return 1; + return err < 0 ? err : 1; } static int ca0132_hp_volume_get(struct snd_kcontrol *kcontrol, @@ -819,25 +821,26 @@ static int ca0132_hp_volume_put(struct snd_kcontrol *kcontrol, err = chipio_read(codec, REG_CODEC_HP_VOL_L, &data); if (err < 0) - return err; + goto exit; val = 31 - left_vol; data = (data & 0xe0) | val; - chipio_write(codec, REG_CODEC_HP_VOL_L, data); + err = chipio_write(codec, REG_CODEC_HP_VOL_L, data); if (err < 0) - return err; + goto exit; val = 31 - right_vol; data = (data & 0xe0) | val; - chipio_write(codec, REG_CODEC_HP_VOL_R, data); + err = chipio_write(codec, REG_CODEC_HP_VOL_R, data); if (err < 0) - return err; + goto exit; spec->curr_hp_volume[0] = left_vol; spec->curr_hp_volume[1] = right_vol; + exit: snd_hda_power_down(codec); - return 1; + return err < 0 ? err : 1; } static int add_hp_switch(struct hda_codec *codec, hda_nid_t nid) @@ -936,6 +939,8 @@ static int ca0132_build_controls(struct hda_codec *codec) if (err < 0) return err; err = add_in_volume(codec, spec->dig_in, "IEC958"); + if (err < 0) + return err; } return 0; } diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 0e99357e822c..c83ccdba1e5a 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -609,7 +609,7 @@ static int add_output(struct hda_codec *codec, hda_nid_t dac, int idx, "Front Speaker", "Surround Speaker", "Bass Speaker" }; static const char * const line_outs[] = { - "Front Line-Out", "Surround Line-Out", "Bass Line-Out" + "Front Line Out", "Surround Line Out", "Bass Line Out" }; fix_volume_caps(codec, dac); @@ -635,7 +635,7 @@ static int add_output(struct hda_codec *codec, hda_nid_t dac, int idx, if (num_ctls > 1) name = line_outs[idx]; else - name = "Line-Out"; + name = "Line Out"; break; } @@ -988,8 +988,10 @@ static void cs_automic(struct hda_codec *codec) change_cur_input(codec, !spec->automic_idx, 0); } else { if (present) { - spec->last_input = spec->cur_input; - spec->cur_input = spec->automic_idx; + if (spec->cur_input != spec->automic_idx) { + spec->last_input = spec->cur_input; + spec->cur_input = spec->automic_idx; + } } else { spec->cur_input = spec->last_input; } diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 8a32a69c83c3..d29d6d377904 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3027,7 +3027,7 @@ static const struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400s", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x21c5, "Thinkpad Edge 13", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x21c6, "Thinkpad Edge 13", CXT5066_ASUS), - SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo Thinkpad", CXT5066_THINKPAD), + SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo T510", CXT5066_AUTO), SND_PCI_QUIRK(0x17aa, 0x21cf, "Lenovo T520 & W520", CXT5066_AUTO), SND_PCI_QUIRK(0x17aa, 0x21da, "Lenovo X220", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x21db, "Lenovo X220-tablet", CXT5066_THINKPAD), @@ -3482,7 +3482,7 @@ static int cx_automute_mode_info(struct snd_kcontrol *kcontrol, "Disabled", "Enabled" }; static const char * const texts3[] = { - "Disabled", "Speaker Only", "Line-Out+Speaker" + "Disabled", "Speaker Only", "Line Out+Speaker" }; const char * const *texts; @@ -4079,7 +4079,8 @@ static int cx_auto_add_volume_idx(struct hda_codec *codec, const char *basename, err = snd_hda_ctl_add(codec, nid, kctl); if (err < 0) return err; - if (!(query_amp_caps(codec, nid, hda_dir) & AC_AMPCAP_MUTE)) + if (!(query_amp_caps(codec, nid, hda_dir) & + (AC_AMPCAP_MUTE | AC_AMPCAP_MIN_MUTE))) break; } return 0; @@ -4379,6 +4380,22 @@ static const struct snd_pci_quirk cxt_fixups[] = { {} }; +/* add "fake" mute amp-caps to DACs on cx5051 so that mixer mute switches + * can be created (bko#42825) + */ +static void add_cx5051_fake_mutes(struct hda_codec *codec) +{ + static hda_nid_t out_nids[] = { + 0x10, 0x11, 0 + }; + hda_nid_t *p; + + for (p = out_nids; *p; p++) + snd_hda_override_amp_caps(codec, *p, HDA_OUTPUT, + AC_AMPCAP_MIN_MUTE | + query_amp_caps(codec, *p, HDA_OUTPUT)); +} + static int patch_conexant_auto(struct hda_codec *codec) { struct conexant_spec *spec; @@ -4397,6 +4414,9 @@ static int patch_conexant_auto(struct hda_codec *codec) case 0x14f15045: spec->single_adc_amp = 1; break; + case 0x14f15051: + add_cx5051_fake_mutes(codec); + break; } apply_pin_fixup(codec, cxt_fixups, cxt_pincfg_tbl); diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 5e82acf77c5a..22c73b78ac6f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -80,6 +80,8 @@ enum { ALC_AUTOMUTE_MIXER, /* mute/unmute mixer widget AMP */ }; +#define MAX_VOL_NIDS 0x40 + struct alc_spec { /* codec parameterization */ const struct snd_kcontrol_new *mixers[5]; /* mixer arrays */ @@ -118,8 +120,8 @@ 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); + DECLARE_BITMAP(vol_ctls, MAX_VOL_NIDS << 1); + DECLARE_BITMAP(sw_ctls, MAX_VOL_NIDS << 1); /* capture setup for dynamic dual-adc switch */ hda_nid_t cur_adc; @@ -177,6 +179,7 @@ struct alc_spec { 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 */ + unsigned int keep_vref_in_automute:1; /* Don't clear VREF in automute */ /* other flags */ unsigned int no_analog :1; /* digital I/O only */ @@ -185,7 +188,6 @@ struct alc_spec { unsigned int vol_in_capsrc:1; /* use capsrc volume (ADC has no vol) */ unsigned int parse_flags; /* passed to snd_hda_parse_pin_defcfg() */ unsigned int shared_mic_hp:1; /* HP/Mic-in sharing */ - unsigned int use_jack_tbl:1; /* 1 for model=auto */ /* auto-mute control */ int automute_mode; @@ -496,13 +498,24 @@ static void do_automute(struct hda_codec *codec, int num_pins, hda_nid_t *pins, for (i = 0; i < num_pins; i++) { hda_nid_t nid = pins[i]; + unsigned int val; if (!nid) break; switch (spec->automute_mode) { case ALC_AUTOMUTE_PIN: + /* don't reset VREF value in case it's controlling + * the amp (see alc861_fixup_asus_amp_vref_0f()) + */ + if (spec->keep_vref_in_automute) { + val = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + val &= ~PIN_HP; + } else + val = 0; + val |= pin_bits; snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, - pin_bits); + val); break; case ALC_AUTOMUTE_AMP: snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0, @@ -621,17 +634,10 @@ static void alc_mic_automute(struct hda_codec *codec) alc_mux_select(codec, 0, spec->int_mic_idx, false); } -/* unsolicited event for HP jack sensing */ -static void alc_sku_unsol_event(struct hda_codec *codec, unsigned int res) +/* handle the specified unsol action (ALC_XXX_EVENT) */ +static void alc_exec_unsol_event(struct hda_codec *codec, int action) { - struct alc_spec *spec = codec->spec; - if (codec->vendor_id == 0x10ec0880) - res >>= 28; - else - res >>= 26; - if (spec->use_jack_tbl) - res = snd_hda_jack_get_action(codec, res); - switch (res) { + switch (action) { case ALC_HP_EVENT: alc_hp_automute(codec); break; @@ -645,6 +651,17 @@ static void alc_sku_unsol_event(struct hda_codec *codec, unsigned int res) snd_hda_jack_report_sync(codec); } +/* unsolicited event for HP jack sensing */ +static void alc_sku_unsol_event(struct hda_codec *codec, unsigned int res) +{ + if (codec->vendor_id == 0x10ec0880) + res >>= 28; + else + res >>= 26; + res = snd_hda_jack_get_action(codec, res); + alc_exec_unsol_event(codec, res); +} + /* call init functions of standard auto-mute helpers */ static void alc_inithook(struct hda_codec *codec) { @@ -785,7 +802,7 @@ static int alc_automute_mode_info(struct snd_kcontrol *kcontrol, "Disabled", "Enabled" }; static const char * const texts3[] = { - "Disabled", "Speaker Only", "Line-Out+Speaker" + "Disabled", "Speaker Only", "Line Out+Speaker" }; const char * const *texts; @@ -1839,7 +1856,9 @@ static const char * const alc_slave_vols[] = { "Headphone Playback Volume", "Speaker Playback Volume", "Mono Playback Volume", - "Line-Out Playback Volume", + "Line Out Playback Volume", + "CLFE Playback Volume", + "Bass Speaker Playback Volume", "PCM Playback Volume", NULL, }; @@ -1854,7 +1873,9 @@ static const char * const alc_slave_sws[] = { "Speaker Playback Switch", "Mono Playback Switch", "IEC958 Playback Switch", - "Line-Out Playback Switch", + "Line Out Playback Switch", + "CLFE Playback Switch", + "Bass Speaker Playback Switch", "PCM Playback Switch", NULL, }; @@ -1883,7 +1904,7 @@ static const struct snd_kcontrol_new alc_beep_mixer[] = { }; #endif -static int alc_build_controls(struct hda_codec *codec) +static int __alc_build_controls(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; struct snd_kcontrol *kctl = NULL; @@ -2029,11 +2050,16 @@ static int alc_build_controls(struct hda_codec *codec) alc_free_kctls(codec); /* no longer needed */ - err = snd_hda_jack_add_kctls(codec, &spec->autocfg); + return 0; +} + +static int alc_build_controls(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + int err = __alc_build_controls(codec); if (err < 0) return err; - - return 0; + return snd_hda_jack_add_kctls(codec, &spec->autocfg); } @@ -2042,12 +2068,16 @@ static int alc_build_controls(struct hda_codec *codec) */ static void alc_init_special_input_src(struct hda_codec *codec); +static int alc269_fill_coef(struct hda_codec *codec); static int alc_init(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; unsigned int i; + if (codec->vendor_id == 0x10ec0269) + alc269_fill_coef(codec); + alc_fix_pll(codec); alc_auto_init_amp(codec, spec->init_amp); @@ -2298,7 +2328,7 @@ static int alc_build_pcms(struct hda_codec *codec) "%s Analog", codec->chip_name); info->name = spec->stream_name_analog; - if (spec->multiout.dac_nids > 0) { + if (spec->multiout.num_dacs > 0) { p = spec->stream_analog_playback; if (!p) p = &alc_pcm_analog_playback; @@ -3125,7 +3155,10 @@ static int alc_auto_fill_dac_nids(struct hda_codec *codec) 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); + unsigned int dir; + if (snd_BUG_ON(nid >= MAX_VOL_NIDS)) + return 0; + dir = get_amp_direction_(data); return (nid << 1) | dir; } @@ -3233,7 +3266,7 @@ static int alc_auto_create_multi_out_ctls(struct hda_codec *codec, int i, err, noutputs; noutputs = cfg->line_outs; - if (spec->multi_ios > 0) + if (spec->multi_ios > 0 && cfg->line_outs < 3) noutputs += spec->multi_ios; for (i = 0; i < noutputs; i++) { @@ -3768,7 +3801,7 @@ static void alc_auto_init_input_src(struct hda_codec *codec) else nums = spec->num_adc_nids; for (c = 0; c < nums; c++) - alc_mux_select(codec, 0, spec->cur_mux[c], true); + alc_mux_select(codec, c, spec->cur_mux[c], true); } /* add mic boosts if needed */ @@ -3904,7 +3937,6 @@ static void set_capture_mixer(struct hda_codec *codec) static void alc_auto_init_std(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - spec->use_jack_tbl = 1; alc_auto_init_multi_out(codec); alc_auto_init_extra_out(codec); alc_auto_init_analog_input(codec); @@ -4168,6 +4200,8 @@ static int patch_alc880(struct hda_codec *codec) codec->patch_ops = alc_patch_ops; if (board_config == ALC_MODEL_AUTO) spec->init_hook = alc_auto_init_std; + else + codec->patch_ops.build_controls = __alc_build_controls; #ifdef CONFIG_SND_HDA_POWER_SAVE if (!spec->loopback.amplist) spec->loopback.amplist = alc880_loopbacks; @@ -4297,6 +4331,8 @@ static int patch_alc260(struct hda_codec *codec) codec->patch_ops = alc_patch_ops; if (board_config == ALC_MODEL_AUTO) spec->init_hook = alc_auto_init_std; + else + codec->patch_ops.build_controls = __alc_build_controls; spec->shutup = alc_eapd_shutup; #ifdef CONFIG_SND_HDA_POWER_SAVE if (!spec->loopback.amplist) @@ -4335,6 +4371,7 @@ enum { ALC882_FIXUP_PB_M5210, ALC882_FIXUP_ACER_ASPIRE_7736, ALC882_FIXUP_ASUS_W90V, + ALC889_FIXUP_CD, ALC889_FIXUP_VAIO_TT, ALC888_FIXUP_EEE1601, ALC882_FIXUP_EAPD, @@ -4347,6 +4384,7 @@ enum { ALC882_FIXUP_ACER_ASPIRE_8930G, ALC882_FIXUP_ASPIRE_8930G_VERBS, ALC885_FIXUP_MACPRO_GPIO, + ALC889_FIXUP_DAC_ROUTE, }; static void alc889_fixup_coef(struct hda_codec *codec, @@ -4400,6 +4438,31 @@ static void alc885_fixup_macpro_gpio(struct hda_codec *codec, alc882_gpio_mute(codec, 1, 0); } +/* Fix the connection of some pins for ALC889: + * At least, Acer Aspire 5935 shows the connections to DAC3/4 don't + * work correctly (bko#42740) + */ +static void alc889_fixup_dac_route(struct hda_codec *codec, + const struct alc_fixup *fix, int action) +{ + if (action == ALC_FIXUP_ACT_PRE_PROBE) { + /* fake the connections during parsing the tree */ + hda_nid_t conn1[2] = { 0x0c, 0x0d }; + hda_nid_t conn2[2] = { 0x0e, 0x0f }; + snd_hda_override_conn_list(codec, 0x14, 2, conn1); + snd_hda_override_conn_list(codec, 0x15, 2, conn1); + snd_hda_override_conn_list(codec, 0x18, 2, conn2); + snd_hda_override_conn_list(codec, 0x1a, 2, conn2); + } else if (action == ALC_FIXUP_ACT_PROBE) { + /* restore the connections */ + hda_nid_t conn[5] = { 0x0c, 0x0d, 0x0e, 0x0f, 0x26 }; + snd_hda_override_conn_list(codec, 0x14, 5, conn); + snd_hda_override_conn_list(codec, 0x15, 5, conn); + snd_hda_override_conn_list(codec, 0x18, 5, conn); + snd_hda_override_conn_list(codec, 0x1a, 5, conn); + } +} + static const struct alc_fixup alc882_fixups[] = { [ALC882_FIXUP_ABIT_AW9D_MAX] = { .type = ALC_FIXUP_PINS, @@ -4436,6 +4499,13 @@ static const struct alc_fixup alc882_fixups[] = { { } } }, + [ALC889_FIXUP_CD] = { + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { + { 0x1c, 0x993301f0 }, /* CD */ + { } + } + }, [ALC889_FIXUP_VAIO_TT] = { .type = ALC_FIXUP_PINS, .v.pins = (const struct alc_pincfg[]) { @@ -4547,6 +4617,10 @@ static const struct alc_fixup alc882_fixups[] = { .type = ALC_FIXUP_FUNC, .v.func = alc885_fixup_macpro_gpio, }, + [ALC889_FIXUP_DAC_ROUTE] = { + .type = ALC_FIXUP_FUNC, + .v.func = alc889_fixup_dac_route, + }, }; static const struct snd_pci_quirk alc882_fixup_tbl[] = { @@ -4571,6 +4645,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x0142, "Acer Aspire 7730G", ALC882_FIXUP_ACER_ASPIRE_4930G), SND_PCI_QUIRK(0x1025, 0x0155, "Packard-Bell M5120", ALC882_FIXUP_PB_M5210), + SND_PCI_QUIRK(0x1025, 0x0259, "Acer Aspire 5935", ALC889_FIXUP_DAC_ROUTE), SND_PCI_QUIRK(0x1025, 0x0296, "Acer Aspire 7736z", ALC882_FIXUP_ACER_ASPIRE_7736), SND_PCI_QUIRK(0x1043, 0x13c2, "Asus A7M", ALC882_FIXUP_EAPD), SND_PCI_QUIRK(0x1043, 0x1873, "ASUS W90V", ALC882_FIXUP_ASUS_W90V), @@ -4587,6 +4662,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC882_FIXUP_EAPD), SND_PCI_QUIRK_VENDOR(0x1462, "MSI", ALC882_FIXUP_GPIO3), + SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte EP45-DS3", ALC889_FIXUP_CD), SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", ALC882_FIXUP_ABIT_AW9D_MAX), SND_PCI_QUIRK_VENDOR(0x1558, "Clevo laptop", ALC882_FIXUP_EAPD), SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_FIXUP_EAPD), @@ -4691,6 +4767,8 @@ static int patch_alc882(struct hda_codec *codec) codec->patch_ops = alc_patch_ops; if (board_config == ALC_MODEL_AUTO) spec->init_hook = alc_auto_init_std; + else + codec->patch_ops.build_controls = __alc_build_controls; #ifdef CONFIG_SND_HDA_POWER_SAVE if (!spec->loopback.amplist) @@ -4722,7 +4800,6 @@ enum { ALC262_FIXUP_FSC_H270, ALC262_FIXUP_HP_Z200, ALC262_FIXUP_TYAN, - ALC262_FIXUP_TOSHIBA_RX1, ALC262_FIXUP_LENOVO_3000, ALC262_FIXUP_BENQ, ALC262_FIXUP_BENQ_T31, @@ -4752,16 +4829,6 @@ static const struct alc_fixup alc262_fixups[] = { { } } }, - [ALC262_FIXUP_TOSHIBA_RX1] = { - .type = ALC_FIXUP_PINS, - .v.pins = (const struct alc_pincfg[]) { - { 0x14, 0x90170110 }, /* speaker */ - { 0x15, 0x0421101f }, /* HP */ - { 0x1a, 0x40f000f0 }, /* N/A */ - { 0x1b, 0x40f000f0 }, /* N/A */ - { 0x1e, 0x40f000f0 }, /* N/A */ - } - }, [ALC262_FIXUP_LENOVO_3000] = { .type = ALC_FIXUP_VERBS, .v.verbs = (const struct hda_verb[]) { @@ -4794,8 +4861,6 @@ static const struct snd_pci_quirk alc262_fixup_tbl[] = { SND_PCI_QUIRK(0x10cf, 0x1397, "Fujitsu", ALC262_FIXUP_BENQ), SND_PCI_QUIRK(0x10cf, 0x142d, "Fujitsu Lifebook E8410", ALC262_FIXUP_BENQ), SND_PCI_QUIRK(0x10f1, 0x2915, "Tyan Thunder n6650W", ALC262_FIXUP_TYAN), - SND_PCI_QUIRK(0x1179, 0x0001, "Toshiba dynabook SS RX1", - ALC262_FIXUP_TOSHIBA_RX1), SND_PCI_QUIRK(0x1734, 0x1147, "FSC Celsius H270", ALC262_FIXUP_FSC_H270), SND_PCI_QUIRK(0x17aa, 0x384e, "Lenovo 3000", ALC262_FIXUP_LENOVO_3000), SND_PCI_QUIRK(0x17ff, 0x0560, "Benq ED8", ALC262_FIXUP_BENQ), @@ -5364,7 +5429,6 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { 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), @@ -5416,8 +5480,12 @@ static const struct alc_model_fixup alc269_fixup_models[] = { static int alc269_fill_coef(struct hda_codec *codec) { + struct alc_spec *spec = codec->spec; int val; + if (spec->codec_variant != ALC269_TYPE_ALC269VB) + return 0; + if ((alc_get_coef0(codec) & 0x00ff) < 0x015) { alc_write_coef_idx(codec, 0xf, 0x960b); alc_write_coef_idx(codec, 0xe, 0x8817); @@ -5573,8 +5641,28 @@ static const struct hda_amp_list alc861_loopbacks[] = { /* Pin config fixes */ enum { PINFIX_FSC_AMILO_PI1505, + PINFIX_ASUS_A6RP, }; +/* On some laptops, VREF of pin 0x0f is abused for controlling the main amp */ +static void alc861_fixup_asus_amp_vref_0f(struct hda_codec *codec, + const struct alc_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + unsigned int val; + + if (action != ALC_FIXUP_ACT_INIT) + return; + val = snd_hda_codec_read(codec, 0x0f, 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + if (!(val & (AC_PINCTL_IN_EN | AC_PINCTL_OUT_EN))) + val |= AC_PINCTL_IN_EN; + val |= AC_PINCTL_VREF_50; + snd_hda_codec_write(codec, 0x0f, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, val); + spec->keep_vref_in_automute = 1; +} + static const struct alc_fixup alc861_fixups[] = { [PINFIX_FSC_AMILO_PI1505] = { .type = ALC_FIXUP_PINS, @@ -5584,9 +5672,16 @@ static const struct alc_fixup alc861_fixups[] = { { } } }, + [PINFIX_ASUS_A6RP] = { + .type = ALC_FIXUP_FUNC, + .v.func = alc861_fixup_asus_amp_vref_0f, + }, }; static const struct snd_pci_quirk alc861_fixup_tbl[] = { + SND_PCI_QUIRK_VENDOR(0x1043, "ASUS laptop", PINFIX_ASUS_A6RP), + SND_PCI_QUIRK(0x1584, 0x0000, "Uniwill ECS M31EI", PINFIX_ASUS_A6RP), + SND_PCI_QUIRK(0x1584, 0x2b01, "Haier W18", PINFIX_ASUS_A6RP), SND_PCI_QUIRK(0x1734, 0x10c7, "FSC Amilo Pi1505", PINFIX_FSC_AMILO_PI1505), {} }; diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 3556408d6ece..9dbb5735d778 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1608,7 +1608,7 @@ static const struct snd_pci_quirk stac92hd73xx_codec_id_cfg_tbl[] = { SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x043a, "Alienware M17x", STAC_ALIENWARE_M17X), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0490, - "Alienware M17x", STAC_ALIENWARE_M17X), + "Alienware M17x R3", STAC_DELL_EQ), {} /* terminator */ }; @@ -4163,13 +4163,15 @@ static int enable_pin_detect(struct hda_codec *codec, hda_nid_t nid, return 1; } -static int is_nid_hp_pin(struct auto_pin_cfg *cfg, hda_nid_t nid) +static int is_nid_out_jack_pin(struct auto_pin_cfg *cfg, hda_nid_t nid) { int i; for (i = 0; i < cfg->hp_outs; i++) if (cfg->hp_pins[i] == nid) return 1; /* nid is a HP-Out */ - + for (i = 0; i < cfg->line_outs; i++) + if (cfg->line_out_pins[i] == nid) + return 1; /* nid is a line-Out */ return 0; /* nid is not a HP-Out */ }; @@ -4375,7 +4377,7 @@ static int stac92xx_init(struct hda_codec *codec) continue; } - if (is_nid_hp_pin(cfg, nid)) + if (is_nid_out_jack_pin(cfg, nid)) continue; /* already has an unsol event */ pinctl = snd_hda_codec_read(codec, nid, 0, @@ -4627,7 +4629,7 @@ static void stac92xx_hp_detect(struct hda_codec *codec) unsigned int val = AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN; if (no_hp_sensing(spec, i)) continue; - if (presence) + if (1 /*presence*/) stac92xx_set_pinctl(codec, cfg->hp_pins[i], val); #if 0 /* FIXME */ /* Resetting the pinctl like below may lead to (a sort of) regressions @@ -4868,7 +4870,14 @@ static int find_mute_led_cfg(struct hda_codec *codec, int default_polarity) /* BIOS bug: unfilled OEM string */ if (strstr(dev->name, "HP_Mute_LED_P_G")) { set_hp_led_gpio(codec); - spec->gpio_led_polarity = 1; + switch (codec->subsystem_id) { + case 0x103c148a: + spec->gpio_led_polarity = 0; + break; + default: + spec->gpio_led_polarity = 1; + break; + } return 1; } } @@ -5069,9 +5078,9 @@ static int stac92xx_update_led_status(struct hda_codec *codec) spec->gpio_dir, spec->gpio_data); } else { notmtd_lvl = spec->gpio_led_polarity ? - AC_PINCTL_VREF_HIZ : AC_PINCTL_VREF_GRD; + AC_PINCTL_VREF_50 : AC_PINCTL_VREF_GRD; muted_lvl = spec->gpio_led_polarity ? - AC_PINCTL_VREF_GRD : AC_PINCTL_VREF_HIZ; + AC_PINCTL_VREF_GRD : AC_PINCTL_VREF_50; spec->vref_led = muted ? muted_lvl : notmtd_lvl; stac_vrefout_set(codec, spec->vref_mute_led_nid, spec->vref_led); diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 03e63fed9caf..dff9a00ee8fb 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -199,6 +199,9 @@ struct via_spec { unsigned int no_pin_power_ctl; enum VIA_HDA_CODEC codec_type; + /* analog low-power control */ + bool alc_mode; + /* smart51 setup */ unsigned int smart51_nums; hda_nid_t smart51_pins[2]; @@ -663,6 +666,9 @@ static void via_auto_init_analog_input(struct hda_codec *codec) /* init input-src */ for (i = 0; i < spec->num_adc_nids; i++) { int adc_idx = spec->inputs[spec->cur_mux[i]].adc_idx; + /* secondary ADCs must have the unique MUX */ + if (i > 0 && !spec->mux_nids[i]) + break; if (spec->mux_nids[adc_idx]) { int mux_idx = spec->inputs[spec->cur_mux[i]].mux_idx; snd_hda_codec_write(codec, spec->mux_nids[adc_idx], 0, @@ -687,6 +693,15 @@ static void via_auto_init_analog_input(struct hda_codec *codec) } } +static void update_power_state(struct hda_codec *codec, hda_nid_t nid, + unsigned int parm) +{ + if (snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_POWER_STATE, 0) == parm) + return; + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_POWER_STATE, parm); +} + static void set_pin_power_state(struct hda_codec *codec, hda_nid_t nid, unsigned int *affected_parm) { @@ -709,7 +724,7 @@ static void set_pin_power_state(struct hda_codec *codec, hda_nid_t nid, } else parm = AC_PWRST_D3; - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, nid, parm); } static int via_pin_power_ctl_info(struct snd_kcontrol *kcontrol, @@ -749,6 +764,7 @@ static int via_pin_power_ctl_put(struct snd_kcontrol *kcontrol, return 0; spec->no_pin_power_ctl = val; set_widgets_power_state(codec); + analog_low_current_mode(codec); return 1; } @@ -1036,13 +1052,19 @@ static bool is_aa_path_mute(struct hda_codec *codec) } /* enter/exit analog low-current mode */ -static void analog_low_current_mode(struct hda_codec *codec) +static void __analog_low_current_mode(struct hda_codec *codec, bool force) { struct via_spec *spec = codec->spec; bool enable; unsigned int verb, parm; - enable = is_aa_path_mute(codec) && (spec->opened_streams != 0); + if (spec->no_pin_power_ctl) + enable = false; + else + enable = is_aa_path_mute(codec) && !spec->opened_streams; + if (enable == spec->alc_mode && !force) + return; + spec->alc_mode = enable; /* decide low current mode's verb & parameter */ switch (spec->codec_type) { @@ -1074,6 +1096,11 @@ static void analog_low_current_mode(struct hda_codec *codec) snd_hda_codec_write(codec, codec->afg, 0, verb, parm); } +static void analog_low_current_mode(struct hda_codec *codec) +{ + return __analog_low_current_mode(codec, false); +} + /* * generic initialization of ADC, input mixers and output mixers */ @@ -1446,6 +1473,7 @@ static int via_build_controls(struct hda_codec *codec) struct snd_kcontrol *kctl; int err, i; + spec->no_pin_power_ctl = 1; if (spec->set_widgets_power_state) if (!via_clone_control(spec, &via_pin_power_ctl_enum)) return -ENOMEM; @@ -1499,10 +1527,6 @@ static int via_build_controls(struct hda_codec *codec) return err; } - /* init power states */ - set_widgets_power_state(codec); - analog_low_current_mode(codec); - via_free_kctls(codec); /* no longer needed */ err = snd_hda_jack_add_kctls(codec, &spec->autocfg); @@ -2295,10 +2319,7 @@ static int via_mux_enum_put(struct snd_kcontrol *kcontrol, if (mux) { /* switch to D0 beofre change index */ - if (snd_hda_codec_read(codec, mux, 0, - AC_VERB_GET_POWER_STATE, 0x00) != AC_PWRST_D0) - snd_hda_codec_write(codec, mux, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + update_power_state(codec, mux, AC_PWRST_D0); snd_hda_codec_write(codec, mux, 0, AC_VERB_SET_CONNECT_SEL, spec->inputs[cur].mux_idx); @@ -2776,6 +2797,10 @@ static int via_init(struct hda_codec *codec) for (i = 0; i < spec->num_iverbs; i++) snd_hda_sequence_write(codec, spec->init_verbs[i]); + /* init power states */ + set_widgets_power_state(codec); + __analog_low_current_mode(codec, true); + via_auto_init_multi_out(codec); via_auto_init_hp_out(codec); via_auto_init_speaker_out(codec); @@ -2922,9 +2947,9 @@ static void set_widgets_power_state_vt1708B(struct hda_codec *codec) if (imux_is_smixer) parm = AC_PWRST_D0; /* SW0 (17h), AIW 0/1 (13h/14h) */ - snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x13, 0, AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x14, 0, AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x17, parm); + update_power_state(codec, 0x13, parm); + update_power_state(codec, 0x14, parm); /* outputs */ /* PW0 (19h), SW1 (18h), AOW1 (11h) */ @@ -2932,8 +2957,8 @@ static void set_widgets_power_state_vt1708B(struct hda_codec *codec) set_pin_power_state(codec, 0x19, &parm); if (spec->smart51_enabled) set_pin_power_state(codec, 0x1b, &parm); - snd_hda_codec_write(codec, 0x18, 0, AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x18, parm); + update_power_state(codec, 0x11, parm); /* PW6 (22h), SW2 (26h), AOW2 (24h) */ if (is_8ch) { @@ -2941,20 +2966,16 @@ static void set_widgets_power_state_vt1708B(struct hda_codec *codec) set_pin_power_state(codec, 0x22, &parm); if (spec->smart51_enabled) set_pin_power_state(codec, 0x1a, &parm); - snd_hda_codec_write(codec, 0x26, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x24, 0, - AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x26, parm); + update_power_state(codec, 0x24, parm); } else if (codec->vendor_id == 0x11064397) { /* PW7(23h), SW2(27h), AOW2(25h) */ parm = AC_PWRST_D3; set_pin_power_state(codec, 0x23, &parm); if (spec->smart51_enabled) set_pin_power_state(codec, 0x1a, &parm); - snd_hda_codec_write(codec, 0x27, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x25, 0, - AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x27, parm); + update_power_state(codec, 0x25, parm); } /* PW 3/4/7 (1ch/1dh/23h) */ @@ -2966,17 +2987,13 @@ static void set_widgets_power_state_vt1708B(struct hda_codec *codec) set_pin_power_state(codec, 0x23, &parm); /* MW0 (16h), Sw3 (27h), AOW 0/3 (10h/25h) */ - snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_POWER_STATE, - imux_is_smixer ? AC_PWRST_D0 : parm); - snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x16, imux_is_smixer ? AC_PWRST_D0 : parm); + update_power_state(codec, 0x10, parm); if (is_8ch) { - snd_hda_codec_write(codec, 0x25, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x27, 0, - AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x25, parm); + update_power_state(codec, 0x27, parm); } else if (codec->vendor_id == 0x11064397 && spec->hp_independent_mode) - snd_hda_codec_write(codec, 0x25, 0, - AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x25, parm); } static int patch_vt1708S(struct hda_codec *codec); @@ -3149,10 +3166,10 @@ static void set_widgets_power_state_vt1702(struct hda_codec *codec) if (imux_is_smixer) parm = AC_PWRST_D0; /* SW0 (13h) = stereo mixer (idx 3) */ /* SW0 (13h), AIW 0/1/2 (12h/1fh/20h) */ - snd_hda_codec_write(codec, 0x13, 0, AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x12, 0, AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x13, parm); + update_power_state(codec, 0x12, parm); + update_power_state(codec, 0x1f, parm); + update_power_state(codec, 0x20, parm); /* outputs */ /* PW 3/4 (16h/17h) */ @@ -3160,10 +3177,9 @@ static void set_widgets_power_state_vt1702(struct hda_codec *codec) set_pin_power_state(codec, 0x17, &parm); set_pin_power_state(codec, 0x16, &parm); /* MW0 (1ah), AOW 0/1 (10h/1dh) */ - snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_POWER_STATE, - imux_is_smixer ? AC_PWRST_D0 : parm); - snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x1d, 0, AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x1a, imux_is_smixer ? AC_PWRST_D0 : parm); + update_power_state(codec, 0x10, parm); + update_power_state(codec, 0x1d, parm); } static int patch_vt1702(struct hda_codec *codec) @@ -3228,52 +3244,48 @@ static void set_widgets_power_state_vt1718S(struct hda_codec *codec) if (imux_is_smixer) parm = AC_PWRST_D0; /* MUX6/7 (1eh/1fh), AIW 0/1 (10h/11h) */ - snd_hda_codec_write(codec, 0x1e, 0, AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x1e, parm); + update_power_state(codec, 0x1f, parm); + update_power_state(codec, 0x10, parm); + update_power_state(codec, 0x11, parm); /* outputs */ /* PW3 (27h), MW2 (1ah), AOW3 (bh) */ parm = AC_PWRST_D3; set_pin_power_state(codec, 0x27, &parm); - snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0xb, 0, AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x1a, parm); + update_power_state(codec, 0xb, parm); /* PW2 (26h), AOW2 (ah) */ parm = AC_PWRST_D3; set_pin_power_state(codec, 0x26, &parm); if (spec->smart51_enabled) set_pin_power_state(codec, 0x2b, &parm); - snd_hda_codec_write(codec, 0xa, 0, AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0xa, parm); /* PW0 (24h), AOW0 (8h) */ parm = AC_PWRST_D3; set_pin_power_state(codec, 0x24, &parm); if (!spec->hp_independent_mode) /* check for redirected HP */ set_pin_power_state(codec, 0x28, &parm); - snd_hda_codec_write(codec, 0x8, 0, AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x8, parm); /* MW9 (21h), Mw2 (1ah), AOW0 (8h) */ - snd_hda_codec_write(codec, 0x21, 0, AC_VERB_SET_POWER_STATE, - imux_is_smixer ? AC_PWRST_D0 : parm); + update_power_state(codec, 0x21, imux_is_smixer ? AC_PWRST_D0 : parm); /* PW1 (25h), AOW1 (9h) */ parm = AC_PWRST_D3; set_pin_power_state(codec, 0x25, &parm); if (spec->smart51_enabled) set_pin_power_state(codec, 0x2a, &parm); - snd_hda_codec_write(codec, 0x9, 0, AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x9, parm); if (spec->hp_independent_mode) { /* PW4 (28h), MW3 (1bh), MUX1(34h), AOW4 (ch) */ parm = AC_PWRST_D3; set_pin_power_state(codec, 0x28, &parm); - snd_hda_codec_write(codec, 0x1b, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x34, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0xc, 0, - AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x1b, parm); + update_power_state(codec, 0x34, parm); + update_power_state(codec, 0xc, parm); } } @@ -3433,8 +3445,8 @@ static void set_widgets_power_state_vt1716S(struct hda_codec *codec) if (imux_is_smixer) parm = AC_PWRST_D0; /* SW0 (17h), AIW0(13h) */ - snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x13, 0, AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x17, parm); + update_power_state(codec, 0x13, parm); parm = AC_PWRST_D3; set_pin_power_state(codec, 0x1e, &parm); @@ -3442,12 +3454,11 @@ static void set_widgets_power_state_vt1716S(struct hda_codec *codec) if (spec->dmic_enabled) set_pin_power_state(codec, 0x22, &parm); else - snd_hda_codec_write(codec, 0x22, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + update_power_state(codec, 0x22, AC_PWRST_D3); /* SW2(26h), AIW1(14h) */ - snd_hda_codec_write(codec, 0x26, 0, AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x14, 0, AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x26, parm); + update_power_state(codec, 0x14, parm); /* outputs */ /* PW0 (19h), SW1 (18h), AOW1 (11h) */ @@ -3456,8 +3467,8 @@ static void set_widgets_power_state_vt1716S(struct hda_codec *codec) /* Smart 5.1 PW2(1bh) */ if (spec->smart51_enabled) set_pin_power_state(codec, 0x1b, &parm); - snd_hda_codec_write(codec, 0x18, 0, AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x18, parm); + update_power_state(codec, 0x11, parm); /* PW7 (23h), SW3 (27h), AOW3 (25h) */ parm = AC_PWRST_D3; @@ -3465,12 +3476,12 @@ static void set_widgets_power_state_vt1716S(struct hda_codec *codec) /* Smart 5.1 PW1(1ah) */ if (spec->smart51_enabled) set_pin_power_state(codec, 0x1a, &parm); - snd_hda_codec_write(codec, 0x27, 0, AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x27, parm); /* Smart 5.1 PW5(1eh) */ if (spec->smart51_enabled) set_pin_power_state(codec, 0x1e, &parm); - snd_hda_codec_write(codec, 0x25, 0, AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x25, parm); /* Mono out */ /* SW4(28h)->MW1(29h)-> PW12 (2ah)*/ @@ -3486,9 +3497,9 @@ static void set_widgets_power_state_vt1716S(struct hda_codec *codec) mono_out = 1; } parm = mono_out ? AC_PWRST_D0 : AC_PWRST_D3; - snd_hda_codec_write(codec, 0x28, 0, AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x29, 0, AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x2a, 0, AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x28, parm); + update_power_state(codec, 0x29, parm); + update_power_state(codec, 0x2a, parm); /* PW 3/4 (1ch/1dh) */ parm = AC_PWRST_D3; @@ -3496,15 +3507,12 @@ static void set_widgets_power_state_vt1716S(struct hda_codec *codec) set_pin_power_state(codec, 0x1d, &parm); /* HP Independent Mode, power on AOW3 */ if (spec->hp_independent_mode) - snd_hda_codec_write(codec, 0x25, 0, - AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x25, parm); /* force to D0 for internal Speaker */ /* MW0 (16h), AOW0 (10h) */ - snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_POWER_STATE, - imux_is_smixer ? AC_PWRST_D0 : parm); - snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, - mono_out ? AC_PWRST_D0 : parm); + update_power_state(codec, 0x16, imux_is_smixer ? AC_PWRST_D0 : parm); + update_power_state(codec, 0x10, mono_out ? AC_PWRST_D0 : parm); } static int patch_vt1716S(struct hda_codec *codec) @@ -3580,54 +3588,45 @@ static void set_widgets_power_state_vt2002P(struct hda_codec *codec) set_pin_power_state(codec, 0x2b, &parm); parm = AC_PWRST_D0; /* MUX9/10 (1eh/1fh), AIW 0/1 (10h/11h) */ - snd_hda_codec_write(codec, 0x1e, 0, AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x1e, parm); + update_power_state(codec, 0x1f, parm); + update_power_state(codec, 0x10, parm); + update_power_state(codec, 0x11, parm); /* outputs */ /* AOW0 (8h)*/ - snd_hda_codec_write(codec, 0x8, 0, AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x8, parm); if (spec->codec_type == VT1802) { /* PW4 (28h), MW4 (18h), MUX4(38h) */ parm = AC_PWRST_D3; set_pin_power_state(codec, 0x28, &parm); - snd_hda_codec_write(codec, 0x18, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x38, 0, - AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x18, parm); + update_power_state(codec, 0x38, parm); } else { /* PW4 (26h), MW4 (1ch), MUX4(37h) */ parm = AC_PWRST_D3; set_pin_power_state(codec, 0x26, &parm); - snd_hda_codec_write(codec, 0x1c, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x37, 0, - AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x1c, parm); + update_power_state(codec, 0x37, parm); } if (spec->codec_type == VT1802) { /* PW1 (25h), MW1 (15h), MUX1(35h), AOW1 (9h) */ parm = AC_PWRST_D3; set_pin_power_state(codec, 0x25, &parm); - snd_hda_codec_write(codec, 0x15, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x35, 0, - AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x15, parm); + update_power_state(codec, 0x35, parm); } else { /* PW1 (25h), MW1 (19h), MUX1(35h), AOW1 (9h) */ parm = AC_PWRST_D3; set_pin_power_state(codec, 0x25, &parm); - snd_hda_codec_write(codec, 0x19, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x35, 0, - AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x19, parm); + update_power_state(codec, 0x35, parm); } if (spec->hp_independent_mode) - snd_hda_codec_write(codec, 0x9, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + update_power_state(codec, 0x9, AC_PWRST_D0); /* Class-D */ /* PW0 (24h), MW0(18h/14h), MUX0(34h) */ @@ -3637,12 +3636,10 @@ static void set_widgets_power_state_vt2002P(struct hda_codec *codec) set_pin_power_state(codec, 0x24, &parm); parm = present ? AC_PWRST_D3 : AC_PWRST_D0; if (spec->codec_type == VT1802) - snd_hda_codec_write(codec, 0x14, 0, - AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x14, parm); else - snd_hda_codec_write(codec, 0x18, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x34, 0, AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x18, parm); + update_power_state(codec, 0x34, parm); /* Mono Out */ present = snd_hda_jack_detect(codec, 0x26); @@ -3650,28 +3647,20 @@ static void set_widgets_power_state_vt2002P(struct hda_codec *codec) parm = present ? AC_PWRST_D3 : AC_PWRST_D0; if (spec->codec_type == VT1802) { /* PW15 (33h), MW8(1ch), MUX8(3ch) */ - snd_hda_codec_write(codec, 0x33, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x1c, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x3c, 0, - AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x33, parm); + update_power_state(codec, 0x1c, parm); + update_power_state(codec, 0x3c, parm); } else { /* PW15 (31h), MW8(17h), MUX8(3bh) */ - snd_hda_codec_write(codec, 0x31, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x17, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x3b, 0, - AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x31, parm); + update_power_state(codec, 0x17, parm); + update_power_state(codec, 0x3b, parm); } /* MW9 (21h) */ if (imux_is_smixer || !is_aa_path_mute(codec)) - snd_hda_codec_write(codec, 0x21, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + update_power_state(codec, 0x21, AC_PWRST_D0); else - snd_hda_codec_write(codec, 0x21, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + update_power_state(codec, 0x21, AC_PWRST_D3); } /* patch for vt2002P */ @@ -3731,30 +3720,28 @@ static void set_widgets_power_state_vt1812(struct hda_codec *codec) set_pin_power_state(codec, 0x2b, &parm); parm = AC_PWRST_D0; /* MUX10/11 (1eh/1fh), AIW 0/1 (10h/11h) */ - snd_hda_codec_write(codec, 0x1e, 0, AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x1e, parm); + update_power_state(codec, 0x1f, parm); + update_power_state(codec, 0x10, parm); + update_power_state(codec, 0x11, parm); /* outputs */ /* AOW0 (8h)*/ - snd_hda_codec_write(codec, 0x8, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + update_power_state(codec, 0x8, AC_PWRST_D0); /* PW4 (28h), MW4 (18h), MUX4(38h) */ parm = AC_PWRST_D3; set_pin_power_state(codec, 0x28, &parm); - snd_hda_codec_write(codec, 0x18, 0, AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x38, 0, AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x18, parm); + update_power_state(codec, 0x38, parm); /* PW1 (25h), MW1 (15h), MUX1(35h), AOW1 (9h) */ parm = AC_PWRST_D3; set_pin_power_state(codec, 0x25, &parm); - snd_hda_codec_write(codec, 0x15, 0, AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x35, 0, AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x15, parm); + update_power_state(codec, 0x35, parm); if (spec->hp_independent_mode) - snd_hda_codec_write(codec, 0x9, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + update_power_state(codec, 0x9, AC_PWRST_D0); /* Internal Speaker */ /* PW0 (24h), MW0(14h), MUX0(34h) */ @@ -3763,15 +3750,11 @@ static void set_widgets_power_state_vt1812(struct hda_codec *codec) parm = AC_PWRST_D3; set_pin_power_state(codec, 0x24, &parm); if (present) { - snd_hda_codec_write(codec, 0x14, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D3); - snd_hda_codec_write(codec, 0x34, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + update_power_state(codec, 0x14, AC_PWRST_D3); + update_power_state(codec, 0x34, AC_PWRST_D3); } else { - snd_hda_codec_write(codec, 0x14, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D0); - snd_hda_codec_write(codec, 0x34, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + update_power_state(codec, 0x14, AC_PWRST_D0); + update_power_state(codec, 0x34, AC_PWRST_D0); } @@ -3782,26 +3765,20 @@ static void set_widgets_power_state_vt1812(struct hda_codec *codec) parm = AC_PWRST_D3; set_pin_power_state(codec, 0x31, &parm); if (present) { - snd_hda_codec_write(codec, 0x1c, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D3); - snd_hda_codec_write(codec, 0x3c, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D3); - snd_hda_codec_write(codec, 0x3e, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + update_power_state(codec, 0x1c, AC_PWRST_D3); + update_power_state(codec, 0x3c, AC_PWRST_D3); + update_power_state(codec, 0x3e, AC_PWRST_D3); } else { - snd_hda_codec_write(codec, 0x1c, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D0); - snd_hda_codec_write(codec, 0x3c, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D0); - snd_hda_codec_write(codec, 0x3e, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + update_power_state(codec, 0x1c, AC_PWRST_D0); + update_power_state(codec, 0x3c, AC_PWRST_D0); + update_power_state(codec, 0x3e, AC_PWRST_D0); } /* PW15 (33h), MW15 (1dh), MUX15(3dh) */ parm = AC_PWRST_D3; set_pin_power_state(codec, 0x33, &parm); - snd_hda_codec_write(codec, 0x1d, 0, AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x3d, 0, AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x1d, parm); + update_power_state(codec, 0x3d, parm); } diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index 9f3b01bb72c8..e0a4263baa20 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -2102,6 +2102,12 @@ static struct ac97_quirk ac97_quirks[] __devinitdata = { }, { .subvendor = 0x161f, + .subdevice = 0x202f, + .name = "Gateway M520", + .type = AC97_TUNE_INV_EAPD + }, + { + .subvendor = 0x161f, .subdevice = 0x203a, .name = "Gateway 4525GZ", /* AD1981B */ .type = AC97_TUNE_INV_EAPD diff --git a/sound/pci/oxygen/oxygen_mixer.c b/sound/pci/oxygen/oxygen_mixer.c index 26c7e8bcb229..c0dbb52d45be 100644 --- a/sound/pci/oxygen/oxygen_mixer.c +++ b/sound/pci/oxygen/oxygen_mixer.c @@ -618,9 +618,12 @@ static int ac97_volume_get(struct snd_kcontrol *ctl, mutex_lock(&chip->mutex); reg = oxygen_read_ac97(chip, codec, index); mutex_unlock(&chip->mutex); - value->value.integer.value[0] = 31 - (reg & 0x1f); - if (stereo) - value->value.integer.value[1] = 31 - ((reg >> 8) & 0x1f); + if (!stereo) { + value->value.integer.value[0] = 31 - (reg & 0x1f); + } else { + value->value.integer.value[0] = 31 - ((reg >> 8) & 0x1f); + value->value.integer.value[1] = 31 - (reg & 0x1f); + } return 0; } @@ -636,14 +639,14 @@ static int ac97_volume_put(struct snd_kcontrol *ctl, mutex_lock(&chip->mutex); oldreg = oxygen_read_ac97(chip, codec, index); - newreg = oldreg; - newreg = (newreg & ~0x1f) | - (31 - (value->value.integer.value[0] & 0x1f)); - if (stereo) - newreg = (newreg & ~0x1f00) | - ((31 - (value->value.integer.value[1] & 0x1f)) << 8); - else - newreg = (newreg & ~0x1f00) | ((newreg & 0x1f) << 8); + if (!stereo) { + newreg = oldreg & ~0x1f; + newreg |= 31 - (value->value.integer.value[0] & 0x1f); + } else { + newreg = oldreg & ~0x1f1f; + newreg |= (31 - (value->value.integer.value[0] & 0x1f)) << 8; + newreg |= 31 - (value->value.integer.value[1] & 0x1f); + } change = newreg != oldreg; if (change) oxygen_write_ac97(chip, codec, index, newreg); diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index cc9f6c83d661..bc030a2088da 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -6333,6 +6333,7 @@ static int __devinit snd_hdspm_create_hwdep(struct snd_card *card, hw->ops.open = snd_hdspm_hwdep_dummy_op; hw->ops.ioctl = snd_hdspm_hwdep_ioctl; + hw->ops.ioctl_compat = snd_hdspm_hwdep_ioctl; hw->ops.release = snd_hdspm_hwdep_dummy_op; return 0; diff --git a/sound/pci/ymfpci/ymfpci.c b/sound/pci/ymfpci/ymfpci.c index e57b89e8aa89..94ab728f5ca8 100644 --- a/sound/pci/ymfpci/ymfpci.c +++ b/sound/pci/ymfpci/ymfpci.c @@ -286,17 +286,22 @@ static int __devinit snd_card_ymfpci_probe(struct pci_dev *pci, snd_card_free(card); return err; } - if ((err = snd_ymfpci_pcm_4ch(chip, 2, NULL)) < 0) { + err = snd_ymfpci_mixer(chip, rear_switch[dev]); + if (err < 0) { snd_card_free(card); return err; } - if ((err = snd_ymfpci_pcm2(chip, 3, NULL)) < 0) { - snd_card_free(card); - return err; - } - if ((err = snd_ymfpci_mixer(chip, rear_switch[dev])) < 0) { - snd_card_free(card); - return err; + if (chip->ac97->ext_id & AC97_EI_SDAC) { + err = snd_ymfpci_pcm_4ch(chip, 2, NULL); + if (err < 0) { + snd_card_free(card); + return err; + } + err = snd_ymfpci_pcm2(chip, 3, NULL); + if (err < 0) { + snd_card_free(card); + return err; + } } if ((err = snd_ymfpci_timer(chip, 0)) < 0) { snd_card_free(card); diff --git a/sound/pci/ymfpci/ymfpci_main.c b/sound/pci/ymfpci/ymfpci_main.c index 03ee4e365311..12a9a2b03387 100644 --- a/sound/pci/ymfpci/ymfpci_main.c +++ b/sound/pci/ymfpci/ymfpci_main.c @@ -1614,6 +1614,14 @@ static int snd_ymfpci_put_dup4ch(struct snd_kcontrol *kcontrol, struct snd_ctl_e return change; } +static struct snd_kcontrol_new snd_ymfpci_dup4ch __devinitdata = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "4ch Duplication", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_ymfpci_info_dup4ch, + .get = snd_ymfpci_get_dup4ch, + .put = snd_ymfpci_put_dup4ch, +}; static struct snd_kcontrol_new snd_ymfpci_controls[] __devinitdata = { { @@ -1642,13 +1650,6 @@ YMFPCI_DOUBLE(SNDRV_CTL_NAME_IEC958("",CAPTURE,VOLUME), 1, YDSXGR_SPDIFLOOPVOL), YMFPCI_SINGLE(SNDRV_CTL_NAME_IEC958("",PLAYBACK,SWITCH), 0, YDSXGR_SPDIFOUTCTRL, 0), YMFPCI_SINGLE(SNDRV_CTL_NAME_IEC958("",CAPTURE,SWITCH), 0, YDSXGR_SPDIFINCTRL, 0), YMFPCI_SINGLE(SNDRV_CTL_NAME_IEC958("Loop",NONE,NONE), 0, YDSXGR_SPDIFINCTRL, 4), -{ - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "4ch Duplication", - .info = snd_ymfpci_info_dup4ch, - .get = snd_ymfpci_get_dup4ch, - .put = snd_ymfpci_put_dup4ch, -}, }; @@ -1838,6 +1839,12 @@ int __devinit snd_ymfpci_mixer(struct snd_ymfpci *chip, int rear_switch) if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_ymfpci_controls[idx], chip))) < 0) return err; } + if (chip->ac97->ext_id & AC97_EI_SDAC) { + kctl = snd_ctl_new1(&snd_ymfpci_dup4ch, chip); + err = snd_ctl_add(chip->card, kctl); + if (err < 0) + return err; + } /* add S/PDIF control */ if (snd_BUG_ON(!chip->pcm_spdif)) diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 5ef70b5d27e4..278c0a0575f5 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -146,13 +146,10 @@ static const struct snd_kcontrol_new ak4642_snd_controls[] = { SOC_DOUBLE_R_TLV("Digital Playback Volume", L_DVC, R_DVC, 0, 0xFF, 1, out_tlv), - - SOC_SINGLE("Headphone Switch", PW_MGMT2, 6, 1, 0), }; -static const struct snd_kcontrol_new ak4642_hpout_mixer_controls[] = { - SOC_DAPM_SINGLE("DACH", MD_CTL4, 0, 1, 0), -}; +static const struct snd_kcontrol_new ak4642_headphone_control = + SOC_DAPM_SINGLE("Switch", PW_MGMT2, 6, 1, 0); static const struct snd_kcontrol_new ak4642_lout_mixer_controls[] = { SOC_DAPM_SINGLE("DACL", SG_SL1, 4, 1, 0), @@ -165,13 +162,12 @@ static const struct snd_soc_dapm_widget ak4642_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("HPOUTR"), SND_SOC_DAPM_OUTPUT("LINEOUT"), - SND_SOC_DAPM_MIXER("HPOUTL Mixer", PW_MGMT2, 5, 0, - &ak4642_hpout_mixer_controls[0], - ARRAY_SIZE(ak4642_hpout_mixer_controls)), + SND_SOC_DAPM_PGA("HPL Out", PW_MGMT2, 5, 0, NULL, 0), + SND_SOC_DAPM_PGA("HPR Out", PW_MGMT2, 4, 0, NULL, 0), + SND_SOC_DAPM_SWITCH("Headphone Enable", SND_SOC_NOPM, 0, 0, + &ak4642_headphone_control), - SND_SOC_DAPM_MIXER("HPOUTR Mixer", PW_MGMT2, 4, 0, - &ak4642_hpout_mixer_controls[0], - ARRAY_SIZE(ak4642_hpout_mixer_controls)), + SND_SOC_DAPM_PGA("DACH", MD_CTL4, 0, 0, NULL, 0), SND_SOC_DAPM_MIXER("LINEOUT Mixer", PW_MGMT1, 3, 0, &ak4642_lout_mixer_controls[0], @@ -184,12 +180,17 @@ static const struct snd_soc_dapm_widget ak4642_dapm_widgets[] = { static const struct snd_soc_dapm_route ak4642_intercon[] = { /* Outputs */ - {"HPOUTL", NULL, "HPOUTL Mixer"}, - {"HPOUTR", NULL, "HPOUTR Mixer"}, + {"HPOUTL", NULL, "HPL Out"}, + {"HPOUTR", NULL, "HPR Out"}, {"LINEOUT", NULL, "LINEOUT Mixer"}, - {"HPOUTL Mixer", "DACH", "DAC"}, - {"HPOUTR Mixer", "DACH", "DAC"}, + {"HPL Out", NULL, "Headphone Enable"}, + {"HPR Out", NULL, "Headphone Enable"}, + + {"Headphone Enable", "Switch", "DACH"}, + + {"DACH", NULL, "DAC"}, + {"LINEOUT Mixer", "DACL", "DAC"}, }; diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index 9d38db8f1919..78979b3e0e95 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -1113,7 +1113,7 @@ static int cs42l73_pcm_hw_params(struct snd_pcm_substream *substream, priv->config[id].mmcc &= 0xC0; priv->config[id].mmcc |= cs42l73_mclk_coeffs[mclk_coeff].mmcc; priv->config[id].spc &= 0xFC; - priv->config[id].spc &= MCK_SCLK_64FS; + priv->config[id].spc |= MCK_SCLK_MCLK; } else { /* CS42L73 Slave */ priv->config[id].spc &= 0xFC; diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index f8863ebb4304..7f4ba819a9f6 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -987,12 +987,12 @@ static int sgtl5000_restore_regs(struct snd_soc_codec *codec) /* restore regular registers */ for (reg = 0; reg <= SGTL5000_CHIP_SHORT_CTRL; reg += 2) { - /* this regs depends on the others */ + /* These regs should restore in particular order */ if (reg == SGTL5000_CHIP_ANA_POWER || reg == SGTL5000_CHIP_CLK_CTRL || reg == SGTL5000_CHIP_LINREG_CTRL || reg == SGTL5000_CHIP_LINE_OUT_CTRL || - reg == SGTL5000_CHIP_CLK_CTRL) + reg == SGTL5000_CHIP_REF_CTRL) continue; snd_soc_write(codec, reg, cache[reg]); @@ -1003,8 +1003,17 @@ static int sgtl5000_restore_regs(struct snd_soc_codec *codec) snd_soc_write(codec, reg, cache[reg]); /* - * restore power and other regs according - * to set_power() and set_clock() + * restore these regs according to the power setting sequence in + * sgtl5000_set_power_regs() and clock setting sequence in + * sgtl5000_set_clock(). + * + * The order of restore is: + * 1. SGTL5000_CHIP_CLK_CTRL MCLK_FREQ bits (1:0) should be restore after + * SGTL5000_CHIP_ANA_POWER PLL bits set + * 2. SGTL5000_CHIP_LINREG_CTRL should be set before + * SGTL5000_CHIP_ANA_POWER LINREG_D restored + * 3. SGTL5000_CHIP_REF_CTRL controls Analog Ground Voltage, + * prefer to resotre it after SGTL5000_CHIP_ANA_POWER restored */ snd_soc_write(codec, SGTL5000_CHIP_LINREG_CTRL, cache[SGTL5000_CHIP_LINREG_CTRL]); diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index eb401ef021fb..372b0b83bd9f 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -60,7 +60,6 @@ struct aic32x4_rate_divs { struct aic32x4_priv { u32 sysclk; - s32 master; u8 page_no; void *control_data; u32 power_cfg; @@ -369,7 +368,6 @@ static int aic32x4_set_dai_sysclk(struct snd_soc_dai *codec_dai, static int aic32x4_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; - struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec); u8 iface_reg_1; u8 iface_reg_2; u8 iface_reg_3; @@ -384,11 +382,9 @@ static int aic32x4_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) /* set master/slave audio interface */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: - aic32x4->master = 1; iface_reg_1 |= AIC32X4_BCLKMASTER | AIC32X4_WCLKMASTER; break; case SND_SOC_DAIFMT_CBS_CFS: - aic32x4->master = 0; break; default: printk(KERN_ERR "aic32x4: invalid DAI master/slave interface\n"); @@ -526,64 +522,58 @@ static int aic32x4_mute(struct snd_soc_dai *dai, int mute) 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); - switch (level) { case SND_SOC_BIAS_ON: - if (aic32x4->master) { - /* Switch on PLL */ - snd_soc_update_bits(codec, AIC32X4_PLLPR, - AIC32X4_PLLEN, AIC32X4_PLLEN); - - /* Switch on NDAC Divider */ - snd_soc_update_bits(codec, AIC32X4_NDAC, - AIC32X4_NDACEN, AIC32X4_NDACEN); - - /* Switch on MDAC Divider */ - snd_soc_update_bits(codec, AIC32X4_MDAC, - AIC32X4_MDACEN, AIC32X4_MDACEN); - - /* Switch on NADC Divider */ - snd_soc_update_bits(codec, AIC32X4_NADC, - AIC32X4_NADCEN, AIC32X4_NADCEN); - - /* Switch on MADC Divider */ - snd_soc_update_bits(codec, AIC32X4_MADC, - AIC32X4_MADCEN, AIC32X4_MADCEN); - - /* Switch on BCLK_N Divider */ - snd_soc_update_bits(codec, AIC32X4_BCLKN, - AIC32X4_BCLKEN, AIC32X4_BCLKEN); - } + /* Switch on PLL */ + snd_soc_update_bits(codec, AIC32X4_PLLPR, + AIC32X4_PLLEN, AIC32X4_PLLEN); + + /* Switch on NDAC Divider */ + snd_soc_update_bits(codec, AIC32X4_NDAC, + AIC32X4_NDACEN, AIC32X4_NDACEN); + + /* Switch on MDAC Divider */ + snd_soc_update_bits(codec, AIC32X4_MDAC, + AIC32X4_MDACEN, AIC32X4_MDACEN); + + /* Switch on NADC Divider */ + snd_soc_update_bits(codec, AIC32X4_NADC, + AIC32X4_NADCEN, AIC32X4_NADCEN); + + /* Switch on MADC Divider */ + snd_soc_update_bits(codec, AIC32X4_MADC, + AIC32X4_MADCEN, AIC32X4_MADCEN); + + /* Switch on BCLK_N Divider */ + snd_soc_update_bits(codec, AIC32X4_BCLKN, + AIC32X4_BCLKEN, AIC32X4_BCLKEN); break; case SND_SOC_BIAS_PREPARE: break; case SND_SOC_BIAS_STANDBY: - if (aic32x4->master) { - /* Switch off PLL */ - snd_soc_update_bits(codec, AIC32X4_PLLPR, - AIC32X4_PLLEN, 0); - - /* Switch off NDAC Divider */ - snd_soc_update_bits(codec, AIC32X4_NDAC, - AIC32X4_NDACEN, 0); - - /* Switch off MDAC Divider */ - snd_soc_update_bits(codec, AIC32X4_MDAC, - AIC32X4_MDACEN, 0); - - /* Switch off NADC Divider */ - snd_soc_update_bits(codec, AIC32X4_NADC, - AIC32X4_NADCEN, 0); - - /* Switch off MADC Divider */ - snd_soc_update_bits(codec, AIC32X4_MADC, - AIC32X4_MADCEN, 0); - - /* Switch off BCLK_N Divider */ - snd_soc_update_bits(codec, AIC32X4_BCLKN, - AIC32X4_BCLKEN, 0); - } + /* Switch off PLL */ + snd_soc_update_bits(codec, AIC32X4_PLLPR, + AIC32X4_PLLEN, 0); + + /* Switch off NDAC Divider */ + snd_soc_update_bits(codec, AIC32X4_NDAC, + AIC32X4_NDACEN, 0); + + /* Switch off MDAC Divider */ + snd_soc_update_bits(codec, AIC32X4_MDAC, + AIC32X4_MDACEN, 0); + + /* Switch off NADC Divider */ + snd_soc_update_bits(codec, AIC32X4_NADC, + AIC32X4_NADCEN, 0); + + /* Switch off MADC Divider */ + snd_soc_update_bits(codec, AIC32X4_MADC, + AIC32X4_MADCEN, 0); + + /* Switch off BCLK_N Divider */ + snd_soc_update_bits(codec, AIC32X4_BCLKN, + AIC32X4_BCLKEN, 0); break; case SND_SOC_BIAS_OFF: break; @@ -651,9 +641,11 @@ static int aic32x4_probe(struct snd_soc_codec *codec) if (aic32x4->power_cfg & AIC32X4_PWR_AVDD_DVDD_WEAK_DISABLE) { snd_soc_write(codec, AIC32X4_PWRCFG, AIC32X4_AVDDWEAKDISABLE); } - if (aic32x4->power_cfg & AIC32X4_PWR_AIC32X4_LDO_ENABLE) { - snd_soc_write(codec, AIC32X4_LDOCTL, AIC32X4_LDOCTLEN); - } + + tmp_reg = (aic32x4->power_cfg & AIC32X4_PWR_AIC32X4_LDO_ENABLE) ? + AIC32X4_LDOCTLEN : 0; + snd_soc_write(codec, AIC32X4_LDOCTL, tmp_reg); + tmp_reg = snd_soc_read(codec, AIC32X4_CMMODE); if (aic32x4->power_cfg & AIC32X4_PWR_CMMODE_LDOIN_RANGE_18_36) { tmp_reg |= AIC32X4_LDOIN_18_36; diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index c2880907fced..a75c3766aede 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -733,8 +733,9 @@ static int __devinit wm2000_i2c_probe(struct i2c_client *i2c, struct wm2000_priv *wm2000; struct wm2000_platform_data *pdata; const char *filename; - const struct firmware *fw; - int reg, ret; + const struct firmware *fw = NULL; + int ret; + int reg; u16 id; wm2000 = devm_kzalloc(&i2c->dev, sizeof(struct wm2000_priv), @@ -751,7 +752,7 @@ static int __devinit wm2000_i2c_probe(struct i2c_client *i2c, ret = PTR_ERR(wm2000->regmap); dev_err(&i2c->dev, "Failed to allocate register map: %d\n", ret); - goto err; + goto out; } /* Verify that this is a WM2000 */ @@ -763,7 +764,7 @@ static int __devinit wm2000_i2c_probe(struct i2c_client *i2c, if (id != 0x2000) { dev_err(&i2c->dev, "Device is not a WM2000 - ID %x\n", id); ret = -ENODEV; - goto err_regmap; + goto out_regmap_exit; } reg = wm2000_read(i2c, WM2000_REG_REVISON); @@ -782,7 +783,7 @@ static int __devinit wm2000_i2c_probe(struct i2c_client *i2c, ret = request_firmware(&fw, filename, &i2c->dev); if (ret != 0) { dev_err(&i2c->dev, "Failed to acquire ANC data: %d\n", ret); - goto err_regmap; + goto out_regmap_exit; } /* Pre-cook the concatenation of the register address onto the image */ @@ -793,15 +794,13 @@ static int __devinit wm2000_i2c_probe(struct i2c_client *i2c, if (wm2000->anc_download == NULL) { dev_err(&i2c->dev, "Out of memory\n"); ret = -ENOMEM; - goto err_fw; + goto out_regmap_exit; } wm2000->anc_download[0] = 0x80; wm2000->anc_download[1] = 0x00; memcpy(wm2000->anc_download + 2, fw->data, fw->size); - release_firmware(fw); - wm2000->anc_eng_ena = 1; wm2000->anc_active = 1; wm2000->spk_ena = 1; @@ -809,18 +808,14 @@ static int __devinit wm2000_i2c_probe(struct i2c_client *i2c, wm2000_reset(wm2000); - ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm2000, - NULL, 0); - if (ret != 0) - goto err_fw; + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm2000, NULL, 0); + if (!ret) + goto out; - return 0; - -err_fw: - release_firmware(fw); -err_regmap: +out_regmap_exit: regmap_exit(wm2000->regmap); -err: +out: + release_firmware(fw); return ret; } diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 8b24323d6b2c..89f2af77b1c3 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -1377,6 +1377,7 @@ static int wm5100_set_bias_level(struct snd_soc_codec *codec, switch (wm5100->rev) { case 0: + regcache_cache_bypass(wm5100->regmap, true); snd_soc_write(codec, 0x11, 0x3); snd_soc_write(codec, 0x203, 0xc); snd_soc_write(codec, 0x206, 0); @@ -1392,6 +1393,7 @@ static int wm5100_set_bias_level(struct snd_soc_codec *codec, snd_soc_write(codec, wm5100_reva_patches[i].reg, wm5100_reva_patches[i].val); + regcache_cache_bypass(wm5100->regmap, false); break; default: break; @@ -1402,6 +1404,8 @@ static int wm5100_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_OFF: + regcache_cache_only(wm5100->regmap, true); + regcache_mark_dirty(wm5100->regmap); if (wm5100->pdata.ldo_ena) gpio_set_value_cansleep(wm5100->pdata.ldo_ena, 0); regulator_bulk_disable(ARRAY_SIZE(wm5100->core_supplies), @@ -2180,6 +2184,7 @@ static void wm5100_micd_irq(struct snd_soc_codec *codec) if (wm5100->jack_detecting) { dev_dbg(codec->dev, "Microphone detected\n"); wm5100->jack_mic = true; + wm5100->jack_detecting = false; snd_soc_jack_report(wm5100->jack, SND_JACK_HEADSET, SND_JACK_HEADSET | SND_JACK_BTN_0); @@ -2218,6 +2223,7 @@ static void wm5100_micd_irq(struct snd_soc_codec *codec) SND_JACK_BTN_0); } else if (wm5100->jack_detecting) { dev_dbg(codec->dev, "Headphone detected\n"); + wm5100->jack_detecting = false; snd_soc_jack_report(wm5100->jack, SND_JACK_HEADPHONE, SND_JACK_HEADPHONE); @@ -2607,6 +2613,13 @@ static const struct regmap_config wm5100_regmap = { .cache_type = REGCACHE_RBTREE, }; +static const unsigned int wm5100_mic_ctrl_reg[] = { + WM5100_IN1L_CONTROL, + WM5100_IN2L_CONTROL, + WM5100_IN3L_CONTROL, + WM5100_IN4L_CONTROL, +}; + static __devinit int wm5100_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -2739,7 +2752,7 @@ static __devinit int wm5100_i2c_probe(struct i2c_client *i2c, } for (i = 0; i < ARRAY_SIZE(wm5100->pdata.in_mode); i++) { - regmap_update_bits(wm5100->regmap, WM5100_IN1L_CONTROL, + regmap_update_bits(wm5100->regmap, wm5100_mic_ctrl_reg[i], WM5100_IN1_MODE_MASK | WM5100_IN1_DMIC_SUP_MASK, (wm5100->pdata.in_mode[i] << diff --git a/sound/soc/codecs/wm8958-dsp2.c b/sound/soc/codecs/wm8958-dsp2.c index 8d4ea43d40a3..40ac888faf3d 100644 --- a/sound/soc/codecs/wm8958-dsp2.c +++ b/sound/soc/codecs/wm8958-dsp2.c @@ -55,7 +55,7 @@ static int wm8958_dsp2_fw(struct snd_soc_codec *codec, const char *name, return 0; if (fw->size < 32) { - dev_err(codec->dev, "%s: firmware too short (%d bytes)\n", + dev_err(codec->dev, "%s: firmware too short (%zd bytes)\n", name, fw->size); goto err; } diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 296de4e30d26..0ac228b7dc04 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -96,7 +96,7 @@ static int wm8962_regulator_event_##n(struct notifier_block *nb, \ struct wm8962_priv *wm8962 = container_of(nb, struct wm8962_priv, \ disable_nb[n]); \ if (event & REGULATOR_EVENT_DISABLE) { \ - regcache_cache_only(wm8962->regmap, true); \ + regcache_mark_dirty(wm8962->regmap); \ } \ return 0; \ } @@ -2564,7 +2564,7 @@ static int dsp2_event(struct snd_soc_dapm_widget *w, return 0; } -static const char *st_text[] = { "None", "Right", "Left" }; +static const char *st_text[] = { "None", "Left", "Right" }; static const struct soc_enum str_enum = SOC_ENUM_SINGLE(WM8962_DAC_DSP_MIXING_1, 2, 3, st_text); @@ -3159,13 +3159,13 @@ static int wm8962_hw_params(struct snd_pcm_substream *substream, case SNDRV_PCM_FORMAT_S16_LE: break; case SNDRV_PCM_FORMAT_S20_3LE: - aif0 |= 0x40; + aif0 |= 0x4; break; case SNDRV_PCM_FORMAT_S24_LE: - aif0 |= 0x80; + aif0 |= 0x8; break; case SNDRV_PCM_FORMAT_S32_LE: - aif0 |= 0xc0; + aif0 |= 0xc; break; default: return -EINVAL; diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 93d27b660257..ec69a6c152fe 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -770,6 +770,8 @@ static void vmid_reference(struct snd_soc_codec *codec) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + pm_runtime_get_sync(codec->dev); + wm8994->vmid_refcount++; dev_dbg(codec->dev, "Referencing VMID, refcount is now %d\n", @@ -783,7 +785,12 @@ static void vmid_reference(struct snd_soc_codec *codec) WM8994_VMID_RAMP_MASK, WM8994_STARTUP_BIAS_ENA | WM8994_VMID_BUF_ENA | - (0x11 << WM8994_VMID_RAMP_SHIFT)); + (0x3 << WM8994_VMID_RAMP_SHIFT)); + + /* Remove discharge for line out */ + snd_soc_update_bits(codec, WM8994_ANTIPOP_1, + WM8994_LINEOUT1_DISCH | + WM8994_LINEOUT2_DISCH, 0); /* Main bias enable, VMID=2x40k */ snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_1, @@ -837,6 +844,8 @@ static void vmid_dereference(struct snd_soc_codec *codec) WM8994_VMID_BUF_ENA | WM8994_VMID_RAMP_MASK, 0); } + + pm_runtime_put(codec->dev); } static int vmid_event(struct snd_soc_dapm_widget *w, @@ -2753,11 +2762,6 @@ static int wm8994_resume(struct snd_soc_codec *codec) codec->cache_only = 0; } - /* Restore the registers */ - ret = snd_soc_cache_sync(codec); - if (ret != 0) - dev_err(codec->dev, "Failed to sync cache: %d\n", ret); - wm8994_set_bias_level(codec, SND_SOC_BIAS_STANDBY); for (i = 0; i < ARRAY_SIZE(wm8994->fll); i++) { diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index d8da10fe5b52..61f7daa4d0e6 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -108,7 +108,7 @@ static int wm8996_regulator_event_##n(struct notifier_block *nb, \ struct wm8996_priv *wm8996 = container_of(nb, struct wm8996_priv, \ disable_nb[n]); \ if (event & REGULATOR_EVENT_DISABLE) { \ - regcache_cache_only(wm8996->regmap, true); \ + regcache_mark_dirty(wm8996->regmap); \ } \ return 0; \ } @@ -1120,7 +1120,8 @@ 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_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_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), @@ -2007,6 +2008,7 @@ static int wm8996_set_sysclk(struct snd_soc_dai *dai, struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec); int lfclk = 0; int ratediv = 0; + int sync = WM8996_REG_SYNC; int src; int old; @@ -2051,6 +2053,7 @@ static int wm8996_set_sysclk(struct snd_soc_dai *dai, case 32000: case 32768: lfclk = WM8996_LFCLK_ENA; + sync = 0; break; default: dev_warn(codec->dev, "Unsupported clock rate %dHz\n", @@ -2064,6 +2067,8 @@ static int wm8996_set_sysclk(struct snd_soc_dai *dai, WM8996_SYSCLK_SRC_MASK | WM8996_SYSCLK_DIV_MASK, src << WM8996_SYSCLK_SRC_SHIFT | ratediv); snd_soc_update_bits(codec, WM8996_CLOCKING_1, WM8996_LFCLK_ENA, lfclk); + snd_soc_update_bits(codec, WM8996_CONTROL_INTERFACE_1, + WM8996_REG_SYNC, sync); snd_soc_update_bits(codec, WM8996_AIF_CLOCKING_1, WM8996_SYSCLK_ENA, old); diff --git a/sound/soc/codecs/wm8996.h b/sound/soc/codecs/wm8996.h index 0fde643194ce..de9ac3e44aec 100644 --- a/sound/soc/codecs/wm8996.h +++ b/sound/soc/codecs/wm8996.h @@ -1567,6 +1567,10 @@ int wm8996_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack, /* * R257 (0x101) - Control Interface (1) */ +#define WM8996_REG_SYNC 0x8000 /* REG_SYNC */ +#define WM8996_REG_SYNC_MASK 0x8000 /* REG_SYNC */ +#define WM8996_REG_SYNC_SHIFT 15 /* REG_SYNC */ +#define WM8996_REG_SYNC_WIDTH 1 /* REG_SYNC */ #define WM8996_AUTO_INC 0x0004 /* AUTO_INC */ #define WM8996_AUTO_INC_MASK 0x0004 /* AUTO_INC */ #define WM8996_AUTO_INC_SHIFT 2 /* AUTO_INC */ diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 2a61094075f8..8a68cea4a3ee 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -586,14 +586,14 @@ SOC_DAPM_SINGLE("Left Output Switch", WM8993_LINE_MIXER1, 0, 1, 0), }; static const struct snd_kcontrol_new line2_mix[] = { -SOC_DAPM_SINGLE("IN2R Switch", WM8993_LINE_MIXER2, 2, 1, 0), -SOC_DAPM_SINGLE("IN2L Switch", WM8993_LINE_MIXER2, 1, 1, 0), +SOC_DAPM_SINGLE("IN1L Switch", WM8993_LINE_MIXER2, 2, 1, 0), +SOC_DAPM_SINGLE("IN1R Switch", WM8993_LINE_MIXER2, 1, 1, 0), SOC_DAPM_SINGLE("Output Switch", WM8993_LINE_MIXER2, 0, 1, 0), }; static const struct snd_kcontrol_new line2n_mix[] = { -SOC_DAPM_SINGLE("Left Output Switch", WM8993_LINE_MIXER2, 6, 1, 0), -SOC_DAPM_SINGLE("Right Output Switch", WM8993_LINE_MIXER2, 5, 1, 0), +SOC_DAPM_SINGLE("Left Output Switch", WM8993_LINE_MIXER2, 5, 1, 0), +SOC_DAPM_SINGLE("Right Output Switch", WM8993_LINE_MIXER2, 6, 1, 0), }; static const struct snd_kcontrol_new line2p_mix[] = { @@ -613,6 +613,8 @@ SND_SOC_DAPM_INPUT("IN2RP:VXRP"), SND_SOC_DAPM_SUPPLY("MICBIAS2", WM8993_POWER_MANAGEMENT_1, 5, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("MICBIAS1", WM8993_POWER_MANAGEMENT_1, 4, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("LINEOUT_VMID_BUF", WM8993_ANTIPOP1, 7, 0, NULL, 0), + SND_SOC_DAPM_MIXER("IN1L PGA", WM8993_POWER_MANAGEMENT_2, 6, 0, in1l_pga, ARRAY_SIZE(in1l_pga)), SND_SOC_DAPM_MIXER("IN1R PGA", WM8993_POWER_MANAGEMENT_2, 4, 0, @@ -834,9 +836,11 @@ static const struct snd_soc_dapm_route lineout1_diff_routes[] = { }; static const struct snd_soc_dapm_route lineout1_se_routes[] = { + { "LINEOUT1N Mixer", NULL, "LINEOUT_VMID_BUF" }, { "LINEOUT1N Mixer", "Left Output Switch", "Left Output PGA" }, { "LINEOUT1N Mixer", "Right Output Switch", "Right Output PGA" }, + { "LINEOUT1P Mixer", NULL, "LINEOUT_VMID_BUF" }, { "LINEOUT1P Mixer", "Left Output Switch", "Left Output PGA" }, { "LINEOUT1N Driver", NULL, "LINEOUT1N Mixer" }, @@ -844,8 +848,8 @@ static const struct snd_soc_dapm_route lineout1_se_routes[] = { }; static const struct snd_soc_dapm_route lineout2_diff_routes[] = { - { "LINEOUT2 Mixer", "IN2L Switch", "IN2L PGA" }, - { "LINEOUT2 Mixer", "IN2R Switch", "IN2R PGA" }, + { "LINEOUT2 Mixer", "IN1L Switch", "IN1L PGA" }, + { "LINEOUT2 Mixer", "IN1R Switch", "IN1R PGA" }, { "LINEOUT2 Mixer", "Output Switch", "Right Output PGA" }, { "LINEOUT2N Driver", NULL, "LINEOUT2 Mixer" }, @@ -853,9 +857,11 @@ static const struct snd_soc_dapm_route lineout2_diff_routes[] = { }; static const struct snd_soc_dapm_route lineout2_se_routes[] = { + { "LINEOUT2N Mixer", NULL, "LINEOUT_VMID_BUF" }, { "LINEOUT2N Mixer", "Left Output Switch", "Left Output PGA" }, { "LINEOUT2N Mixer", "Right Output Switch", "Right Output PGA" }, + { "LINEOUT2P Mixer", NULL, "LINEOUT_VMID_BUF" }, { "LINEOUT2P Mixer", "Right Output Switch", "Right Output PGA" }, { "LINEOUT2N Driver", NULL, "LINEOUT2N Mixer" }, diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c index 01d1f749cf02..b6adbed6e506 100644 --- a/sound/soc/imx/imx-ssi.c +++ b/sound/soc/imx/imx-ssi.c @@ -112,7 +112,7 @@ static int imx_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) break; case SND_SOC_DAIFMT_DSP_A: /* data on rising edge of bclk, frame high 1clk before data */ - strcr |= SSI_STCR_TFSL | SSI_STCR_TEFS; + strcr |= SSI_STCR_TFSL | SSI_STCR_TXBIT0 | SSI_STCR_TEFS; break; } diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index dccfb37a9626..f204dbac11d4 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -124,6 +124,8 @@ static int mxs_saif_set_clk(struct mxs_saif *saif, * * If MCLK is not used, we just set saif clk to 512*fs. */ + clk_prepare_enable(master_saif->clk); + if (master_saif->mclk_in_use) { if (mclk % 32 == 0) { scr &= ~BM_SAIF_CTRL_BITCLK_BASE_RATE; @@ -133,6 +135,7 @@ static int mxs_saif_set_clk(struct mxs_saif *saif, ret = clk_set_rate(master_saif->clk, 384 * rate); } else { /* SAIF MCLK should be either 32x or 48x */ + clk_disable_unprepare(master_saif->clk); return -EINVAL; } } else { @@ -140,6 +143,8 @@ static int mxs_saif_set_clk(struct mxs_saif *saif, scr &= ~BM_SAIF_CTRL_BITCLK_BASE_RATE; } + clk_disable_unprepare(master_saif->clk); + if (ret) return ret; diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c index 7ac0ba2025c3..d23b19a59d83 100644 --- a/sound/soc/samsung/neo1973_wm8753.c +++ b/sound/soc/samsung/neo1973_wm8753.c @@ -230,8 +230,6 @@ static const struct snd_kcontrol_new neo1973_wm8753_controls[] = { /* GTA02 specific routes and controls */ -#ifdef CONFIG_MACH_NEO1973_GTA02 - static int gta02_speaker_enabled; static int lm4853_set_spk(struct snd_kcontrol *kcontrol, @@ -311,10 +309,6 @@ static int neo1973_gta02_wm8753_init(struct snd_soc_codec *codec) return 0; } -#else -static int neo1973_gta02_wm8753_init(struct snd_soc_code *codec) { return 0; } -#endif - static int neo1973_wm8753_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; @@ -322,10 +316,6 @@ static int neo1973_wm8753_init(struct snd_soc_pcm_runtime *rtd) int ret; /* set up NC codec pins */ - if (machine_is_neo1973_gta01()) { - snd_soc_dapm_nc_pin(dapm, "LOUT2"); - snd_soc_dapm_nc_pin(dapm, "ROUT2"); - } snd_soc_dapm_nc_pin(dapm, "OUT3"); snd_soc_dapm_nc_pin(dapm, "OUT4"); snd_soc_dapm_nc_pin(dapm, "LINE1"); @@ -370,50 +360,6 @@ static int neo1973_wm8753_init(struct snd_soc_pcm_runtime *rtd) return 0; } -/* GTA01 specific controls */ - -#ifdef CONFIG_MACH_NEO1973_GTA01 - -static const struct snd_soc_dapm_route neo1973_lm4857_routes[] = { - {"Amp IN", NULL, "ROUT1"}, - {"Amp IN", NULL, "LOUT1"}, - - {"Handset Spk", NULL, "Amp EP"}, - {"Stereo Out", NULL, "Amp LS"}, - {"Headphone", NULL, "Amp HP"}, -}; - -static const struct snd_soc_dapm_widget neo1973_lm4857_dapm_widgets[] = { - SND_SOC_DAPM_SPK("Handset Spk", NULL), - SND_SOC_DAPM_SPK("Stereo Out", NULL), - SND_SOC_DAPM_HP("Headphone", NULL), -}; - -static int neo1973_lm4857_init(struct snd_soc_dapm_context *dapm) -{ - int ret; - - ret = snd_soc_dapm_new_controls(dapm, neo1973_lm4857_dapm_widgets, - ARRAY_SIZE(neo1973_lm4857_dapm_widgets)); - if (ret) - return ret; - - ret = snd_soc_dapm_add_routes(dapm, neo1973_lm4857_routes, - ARRAY_SIZE(neo1973_lm4857_routes)); - if (ret) - return ret; - - snd_soc_dapm_ignore_suspend(dapm, "Stereo Out"); - snd_soc_dapm_ignore_suspend(dapm, "Handset Spk"); - snd_soc_dapm_ignore_suspend(dapm, "Headphone"); - - return 0; -} - -#else -static int neo1973_lm4857_init(struct snd_soc_dapm_context *dapm) { return 0; }; -#endif - static struct snd_soc_dai_link neo1973_dai[] = { { /* Hifi Playback - for similatious use with voice below */ .name = "WM8753", @@ -421,7 +367,7 @@ static struct snd_soc_dai_link neo1973_dai[] = { .platform_name = "samsung-audio", .cpu_dai_name = "s3c24xx-iis", .codec_dai_name = "wm8753-hifi", - .codec_name = "wm8753-codec.0-001a", + .codec_name = "wm8753.0-001a", .init = neo1973_wm8753_init, .ops = &neo1973_hifi_ops, }, @@ -430,7 +376,7 @@ static struct snd_soc_dai_link neo1973_dai[] = { .stream_name = "Voice", .cpu_dai_name = "dfbmcs320-pcm", .codec_dai_name = "wm8753-voice", - .codec_name = "wm8753-codec.0-001a", + .codec_name = "wm8753.0-001a", .ops = &neo1973_voice_ops, }, }; @@ -440,11 +386,6 @@ static struct snd_soc_aux_dev neo1973_aux_devs[] = { .name = "dfbmcs320", .codec_name = "dfbmcs320.0", }, - { - .name = "lm4857", - .codec_name = "lm4857.0-007c", - .init = neo1973_lm4857_init, - }, }; static struct snd_soc_codec_conf neo1973_codec_conf[] = { @@ -454,14 +395,10 @@ static struct snd_soc_codec_conf neo1973_codec_conf[] = { }, }; -#ifdef CONFIG_MACH_NEO1973_GTA02 static const struct gpio neo1973_gta02_gpios[] = { { GTA02_GPIO_HP_IN, GPIOF_OUT_INIT_HIGH, "GTA02_HP_IN" }, { GTA02_GPIO_AMP_SHUT, GPIOF_OUT_INIT_HIGH, "GTA02_AMP_SHUT" }, }; -#else -static const struct gpio neo1973_gta02_gpios[] = {}; -#endif static struct snd_soc_card neo1973 = { .name = "neo1973", @@ -480,7 +417,7 @@ static int __init neo1973_init(void) { int ret; - if (!machine_is_neo1973_gta01() && !machine_is_neo1973_gta02()) + if (!machine_is_neo1973_gta02()) return -ENODEV; if (machine_is_neo1973_gta02()) { diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index db6c89a28bda..ea4a82d01160 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -1152,12 +1152,8 @@ static snd_pcm_uframes_t fsi_pointer(struct snd_pcm_substream *substream) { struct fsi_priv *fsi = fsi_get_priv(substream); struct fsi_stream *io = fsi_get_stream(fsi, fsi_is_play(substream)); - int samples_pos = io->buff_sample_pos - 1; - if (samples_pos < 0) - samples_pos = 0; - - return fsi_sample2frame(fsi, samples_pos); + return fsi_sample2frame(fsi, io->buff_sample_pos); } static struct snd_pcm_ops fsi_pcm_ops = { diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b5ecf6d23214..92cee24ed2dc 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -567,6 +567,17 @@ int snd_soc_suspend(struct device *dev) if (!codec->suspended && codec->driver->suspend) { switch (codec->dapm.bias_level) { case SND_SOC_BIAS_STANDBY: + /* + * If the CODEC is capable of idle + * bias off then being in STANDBY + * means it's doing something, + * otherwise fall through. + */ + if (codec->dapm.idle_bias_off) { + dev_dbg(codec->dev, + "idle_bias_off CODEC on over suspend\n"); + break; + } case SND_SOC_BIAS_OFF: codec->driver->suspend(codec); codec->suspended = 1; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 1f55ded4047f..1315663c1c09 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3068,9 +3068,13 @@ static void soc_dapm_shutdown_codec(struct snd_soc_dapm_context *dapm) * standby. */ if (powerdown) { - snd_soc_dapm_set_bias_level(dapm, SND_SOC_BIAS_PREPARE); + if (dapm->bias_level == SND_SOC_BIAS_ON) + snd_soc_dapm_set_bias_level(dapm, + SND_SOC_BIAS_PREPARE); dapm_seq_run(dapm, &down_list, 0, false); - snd_soc_dapm_set_bias_level(dapm, SND_SOC_BIAS_STANDBY); + if (dapm->bias_level == SND_SOC_BIAS_PREPARE) + snd_soc_dapm_set_bias_level(dapm, + SND_SOC_BIAS_STANDBY); } } @@ -3083,7 +3087,9 @@ void snd_soc_dapm_shutdown(struct snd_soc_card *card) list_for_each_entry(codec, &card->codec_dev_list, list) { soc_dapm_shutdown_codec(&codec->dapm); - snd_soc_dapm_set_bias_level(&codec->dapm, SND_SOC_BIAS_OFF); + if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY) + snd_soc_dapm_set_bias_level(&codec->dapm, + SND_SOC_BIAS_OFF); } } diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c index 2cf87f5afed4..fde9a7a29cb6 100644 --- a/sound/usb/caiaq/audio.c +++ b/sound/usb/caiaq/audio.c @@ -311,8 +311,10 @@ snd_usb_caiaq_pcm_pointer(struct snd_pcm_substream *sub) spin_lock(&dev->spinlock); - if (dev->input_panic || dev->output_panic) + if (dev->input_panic || dev->output_panic) { ptr = SNDRV_PCM_POS_XRUN; + goto unlock; + } if (sub->stream == SNDRV_PCM_STREAM_PLAYBACK) ptr = bytes_to_frames(sub->runtime, @@ -321,6 +323,7 @@ snd_usb_caiaq_pcm_pointer(struct snd_pcm_substream *sub) ptr = bytes_to_frames(sub->runtime, dev->audio_in_buf_pos[index]); +unlock: spin_unlock(&dev->spinlock); return ptr; } diff --git a/sound/usb/card.h b/sound/usb/card.h index a39edcc32a93..da5fa1ac4eda 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -1,6 +1,7 @@ #ifndef __USBAUDIO_CARD_H #define __USBAUDIO_CARD_H +#define MAX_NR_RATES 1024 #define MAX_PACKS 20 #define MAX_PACKS_HS (MAX_PACKS * 8) /* in high speed mode */ #define MAX_URBS 8 diff --git a/sound/usb/format.c b/sound/usb/format.c index e09aba19375c..ddfef57c4c9f 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -209,8 +209,6 @@ static int parse_audio_format_rates_v1(struct snd_usb_audio *chip, struct audiof return 0; } -#define MAX_UAC2_NR_RATES 1024 - /* * Helper function to walk the array of sample rate triplets reported by * the device. The problem is that we need to parse whole array first to @@ -255,7 +253,7 @@ static int parse_uac2_sample_rate_range(struct audioformat *fp, int nr_triplets, fp->rates |= snd_pcm_rate_to_rate_bit(rate); nr_rates++; - if (nr_rates >= MAX_UAC2_NR_RATES) { + if (nr_rates >= MAX_NR_RATES) { snd_printk(KERN_ERR "invalid uac2 rates\n"); break; } diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 8edc5035fc8f..d89ab4c7d44b 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -1618,6 +1618,14 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { + /* Edirol UM-3G */ + USB_DEVICE_VENDOR_SPEC(0x0582, 0x0108), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .ifnum = 0, + .type = QUIRK_MIDI_STANDARD_INTERFACE + } +}, +{ /* Boss JS-8 Jam Station */ USB_DEVICE(0x0582, 0x0109), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index a3ddac0deffd..27817266867a 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -132,10 +132,14 @@ static int create_fixed_stream_quirk(struct snd_usb_audio *chip, unsigned *rate_table = NULL; fp = kmemdup(quirk->data, sizeof(*fp), GFP_KERNEL); - if (! fp) { + if (!fp) { snd_printk(KERN_ERR "cannot memdup\n"); return -ENOMEM; } + if (fp->nr_rates > MAX_NR_RATES) { + kfree(fp); + return -EINVAL; + } if (fp->nr_rates > 0) { rate_table = kmemdup(fp->rate_table, sizeof(int) * fp->nr_rates, GFP_KERNEL); |