diff options
Diffstat (limited to 'sound')
47 files changed, 518 insertions, 197 deletions
diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c index dc78272fc39f..1f0f8213e2d5 100644 --- a/sound/arm/aaci.c +++ b/sound/arm/aaci.c @@ -937,6 +937,7 @@ static int __devinit aaci_probe_ac97(struct aaci *aaci) struct snd_ac97 *ac97; int ret; + writel(0, aaci->base + AC97_POWERDOWN); /* * Assert AACIRESET for 2us */ diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 0c1440121c22..c69c60b2a48a 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -953,11 +953,12 @@ static int snd_pcm_dev_register(struct snd_device *device) struct snd_pcm_substream *substream; struct snd_pcm_notify *notify; char str[16]; - struct snd_pcm *pcm = device->device_data; + struct snd_pcm *pcm; struct device *dev; - if (snd_BUG_ON(!pcm || !device)) + if (snd_BUG_ON(!device || !device->device_data)) return -ENXIO; + pcm = device->device_data; mutex_lock(®ister_mutex); err = snd_pcm_add(pcm); if (err) { diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index c0adc14c91f0..70d6f25ba526 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -248,7 +248,8 @@ static int assign_substream(struct snd_rawmidi *rmidi, int subdevice, list_for_each_entry(substream, &s->substreams, list) { if (substream->opened) { if (stream == SNDRV_RAWMIDI_STREAM_INPUT || - !(mode & SNDRV_RAWMIDI_LFLG_APPEND)) + !(mode & SNDRV_RAWMIDI_LFLG_APPEND) || + !substream->append) continue; } if (subdevice < 0 || subdevice == substream->number) { @@ -266,17 +267,21 @@ static int open_substream(struct snd_rawmidi *rmidi, { int err; - err = snd_rawmidi_runtime_create(substream); - if (err < 0) - return err; - err = substream->ops->open(substream); - if (err < 0) - return err; - substream->opened = 1; - if (substream->use_count++ == 0) + if (substream->use_count == 0) { + err = snd_rawmidi_runtime_create(substream); + if (err < 0) + return err; + err = substream->ops->open(substream); + if (err < 0) { + snd_rawmidi_runtime_free(substream); + return err; + } + substream->opened = 1; substream->active_sensing = 0; - if (mode & SNDRV_RAWMIDI_LFLG_APPEND) - substream->append = 1; + if (mode & SNDRV_RAWMIDI_LFLG_APPEND) + substream->append = 1; + } + substream->use_count++; rmidi->streams[substream->stream].substream_opened++; return 0; } @@ -297,27 +302,27 @@ static int rawmidi_open_priv(struct snd_rawmidi *rmidi, int subdevice, int mode, SNDRV_RAWMIDI_STREAM_INPUT, mode, &sinput); if (err < 0) - goto __error; + return err; } if (mode & SNDRV_RAWMIDI_LFLG_OUTPUT) { err = assign_substream(rmidi, subdevice, SNDRV_RAWMIDI_STREAM_OUTPUT, mode, &soutput); if (err < 0) - goto __error; + return err; } if (sinput) { err = open_substream(rmidi, sinput, mode); if (err < 0) - goto __error; + return err; } if (soutput) { err = open_substream(rmidi, soutput, mode); if (err < 0) { if (sinput) close_substream(rmidi, sinput, 0); - goto __error; + return err; } } @@ -325,13 +330,6 @@ static int rawmidi_open_priv(struct snd_rawmidi *rmidi, int subdevice, int mode, rfile->input = sinput; rfile->output = soutput; return 0; - - __error: - if (sinput && sinput->runtime) - snd_rawmidi_runtime_free(sinput); - if (soutput && soutput->runtime) - snd_rawmidi_runtime_free(soutput); - return err; } /* called from sound/core/seq/seq_midi.c */ diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c index 6ba066c41d2e..252e04ce602f 100644 --- a/sound/drivers/dummy.c +++ b/sound/drivers/dummy.c @@ -165,7 +165,7 @@ MODULE_PARM_DESC(enable, "Enable this dummy soundcard."); module_param_array(pcm_devs, int, NULL, 0444); MODULE_PARM_DESC(pcm_devs, "PCM devices # (0-4) for dummy driver."); module_param_array(pcm_substreams, int, NULL, 0444); -MODULE_PARM_DESC(pcm_substreams, "PCM substreams # (1-16) for dummy driver."); +MODULE_PARM_DESC(pcm_substreams, "PCM substreams # (1-128) for dummy driver."); //module_param_array(midi_devs, int, NULL, 0444); //MODULE_PARM_DESC(midi_devs, "MIDI devices # (0-2) for dummy driver."); module_param(fake_buffer, bool, 0444); @@ -808,8 +808,6 @@ static int __devinit snd_card_dummy_new_mixer(struct snd_dummy *dummy) unsigned int idx; int err; - if (snd_BUG_ON(!dummy)) - return -EINVAL; spin_lock_init(&dummy->mixer_lock); strcpy(card->mixername, "Dummy Mixer"); diff --git a/sound/drivers/opl3/opl3_midi.c b/sound/drivers/opl3/opl3_midi.c index 6e7d09ae0e82..7d722a025d0d 100644 --- a/sound/drivers/opl3/opl3_midi.c +++ b/sound/drivers/opl3/opl3_midi.c @@ -29,6 +29,8 @@ extern char snd_opl3_regmap[MAX_OPL2_VOICES][4]; extern int use_internal_drums; +static void snd_opl3_note_off_unsafe(void *p, int note, int vel, + struct snd_midi_channel *chan); /* * The next table looks magical, but it certainly is not. Its values have * been calculated as table[i]=8*log(i/64)/log(2) with an obvious exception @@ -242,16 +244,20 @@ void snd_opl3_timer_func(unsigned long data) int again = 0; int i; - spin_lock_irqsave(&opl3->sys_timer_lock, flags); + spin_lock_irqsave(&opl3->voice_lock, flags); for (i = 0; i < opl3->max_voices; i++) { struct snd_opl3_voice *vp = &opl3->voices[i]; if (vp->state > 0 && vp->note_off_check) { if (vp->note_off == jiffies) - snd_opl3_note_off(opl3, vp->note, 0, vp->chan); + snd_opl3_note_off_unsafe(opl3, vp->note, 0, + vp->chan); else again++; } } + spin_unlock_irqrestore(&opl3->voice_lock, flags); + + spin_lock_irqsave(&opl3->sys_timer_lock, flags); if (again) { opl3->tlist.expires = jiffies + 1; /* invoke again */ add_timer(&opl3->tlist); @@ -658,15 +664,14 @@ static void snd_opl3_kill_voice(struct snd_opl3 *opl3, int voice) /* * Release a note in response to a midi note off. */ -void snd_opl3_note_off(void *p, int note, int vel, struct snd_midi_channel *chan) +static void snd_opl3_note_off_unsafe(void *p, int note, int vel, + struct snd_midi_channel *chan) { struct snd_opl3 *opl3; int voice; struct snd_opl3_voice *vp; - unsigned long flags; - opl3 = p; #ifdef DEBUG_MIDI @@ -674,12 +679,9 @@ void snd_opl3_note_off(void *p, int note, int vel, struct snd_midi_channel *chan chan->number, chan->midi_program, note); #endif - spin_lock_irqsave(&opl3->voice_lock, flags); - if (opl3->synth_mode == SNDRV_OPL3_MODE_SEQ) { if (chan->drum_channel && use_internal_drums) { snd_opl3_drum_switch(opl3, note, vel, 0, chan); - spin_unlock_irqrestore(&opl3->voice_lock, flags); return; } /* this loop will hopefully kill all extra voices, because @@ -697,6 +699,16 @@ void snd_opl3_note_off(void *p, int note, int vel, struct snd_midi_channel *chan snd_opl3_kill_voice(opl3, voice); } } +} + +void snd_opl3_note_off(void *p, int note, int vel, + struct snd_midi_channel *chan) +{ + struct snd_opl3 *opl3 = p; + unsigned long flags; + + spin_lock_irqsave(&opl3->voice_lock, flags); + snd_opl3_note_off_unsafe(p, note, vel, chan); spin_unlock_irqrestore(&opl3->voice_lock, flags); } diff --git a/sound/drivers/pcsp/pcsp_lib.c b/sound/drivers/pcsp/pcsp_lib.c index 84cc2658c05b..e1145ac6e908 100644 --- a/sound/drivers/pcsp/pcsp_lib.c +++ b/sound/drivers/pcsp/pcsp_lib.c @@ -39,25 +39,20 @@ static DECLARE_TASKLET(pcsp_pcm_tasklet, pcsp_call_pcm_elapsed, 0); /* write the port and returns the next expire time in ns; * called at the trigger-start and in hrtimer callback */ -static unsigned long pcsp_timer_update(struct hrtimer *handle) +static u64 pcsp_timer_update(struct snd_pcsp *chip) { unsigned char timer_cnt, val; u64 ns; struct snd_pcm_substream *substream; struct snd_pcm_runtime *runtime; - struct snd_pcsp *chip = container_of(handle, struct snd_pcsp, timer); unsigned long flags; if (chip->thalf) { outb(chip->val61, 0x61); chip->thalf = 0; - if (!atomic_read(&chip->timer_active)) - return 0; return chip->ns_rem; } - if (!atomic_read(&chip->timer_active)) - return 0; substream = chip->playback_substream; if (!substream) return 0; @@ -88,24 +83,17 @@ static unsigned long pcsp_timer_update(struct hrtimer *handle) return ns; } -enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle) +static void pcsp_pointer_update(struct snd_pcsp *chip) { - struct snd_pcsp *chip = container_of(handle, struct snd_pcsp, timer); struct snd_pcm_substream *substream; - int periods_elapsed, pointer_update; size_t period_bytes, buffer_bytes; - unsigned long ns; + int periods_elapsed; unsigned long flags; - pointer_update = !chip->thalf; - ns = pcsp_timer_update(handle); - if (!ns) - return HRTIMER_NORESTART; - /* update the playback position */ substream = chip->playback_substream; if (!substream) - return HRTIMER_NORESTART; + return; period_bytes = snd_pcm_lib_period_bytes(substream); buffer_bytes = snd_pcm_lib_buffer_bytes(substream); @@ -134,6 +122,26 @@ enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle) if (periods_elapsed) tasklet_schedule(&pcsp_pcm_tasklet); +} + +enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle) +{ + struct snd_pcsp *chip = container_of(handle, struct snd_pcsp, timer); + int pointer_update; + u64 ns; + + if (!atomic_read(&chip->timer_active) || !chip->playback_substream) + return HRTIMER_NORESTART; + + pointer_update = !chip->thalf; + ns = pcsp_timer_update(chip); + if (!ns) { + printk(KERN_WARNING "PCSP: unexpected stop\n"); + return HRTIMER_NORESTART; + } + + if (pointer_update) + pcsp_pointer_update(chip); hrtimer_forward(handle, hrtimer_get_expires(handle), ns_to_ktime(ns)); @@ -142,8 +150,6 @@ enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle) static int pcsp_start_playing(struct snd_pcsp *chip) { - unsigned long ns; - #if PCSP_DEBUG printk(KERN_INFO "PCSP: start_playing called\n"); #endif @@ -159,11 +165,7 @@ static int pcsp_start_playing(struct snd_pcsp *chip) atomic_set(&chip->timer_active, 1); chip->thalf = 0; - ns = pcsp_timer_update(&pcsp_chip.timer); - if (!ns) - return -EIO; - - hrtimer_start(&pcsp_chip.timer, ktime_set(0, ns), HRTIMER_MODE_REL); + hrtimer_start(&pcsp_chip.timer, ktime_set(0, 0), HRTIMER_MODE_REL); return 0; } @@ -232,21 +234,22 @@ static int snd_pcsp_playback_hw_free(struct snd_pcm_substream *substream) static int snd_pcsp_playback_prepare(struct snd_pcm_substream *substream) { struct snd_pcsp *chip = snd_pcm_substream_chip(substream); + pcsp_sync_stop(chip); + chip->playback_ptr = 0; + chip->period_ptr = 0; + chip->fmt_size = + snd_pcm_format_physical_width(substream->runtime->format) >> 3; + chip->is_signed = snd_pcm_format_signed(substream->runtime->format); #if PCSP_DEBUG printk(KERN_INFO "PCSP: prepare called, " - "size=%zi psize=%zi f=%zi f1=%i\n", + "size=%zi psize=%zi f=%zi f1=%i fsize=%i\n", snd_pcm_lib_buffer_bytes(substream), snd_pcm_lib_period_bytes(substream), snd_pcm_lib_buffer_bytes(substream) / snd_pcm_lib_period_bytes(substream), - substream->runtime->periods); + substream->runtime->periods, + chip->fmt_size); #endif - pcsp_sync_stop(chip); - chip->playback_ptr = 0; - chip->period_ptr = 0; - chip->fmt_size = - snd_pcm_format_physical_width(substream->runtime->format) >> 3; - chip->is_signed = snd_pcm_format_signed(substream->runtime->format); return 0; } diff --git a/sound/drivers/pcsp/pcsp_mixer.c b/sound/drivers/pcsp/pcsp_mixer.c index 199b03377142..903bc846763f 100644 --- a/sound/drivers/pcsp/pcsp_mixer.c +++ b/sound/drivers/pcsp/pcsp_mixer.c @@ -72,7 +72,7 @@ static int pcsp_treble_put(struct snd_kcontrol *kcontrol, if (treble != chip->treble) { chip->treble = treble; #if PCSP_DEBUG - printk(KERN_INFO "PCSP: rate set to %i\n", PCSP_RATE()); + printk(KERN_INFO "PCSP: rate set to %li\n", PCSP_RATE()); #endif changed = 1; } diff --git a/sound/oss/dmasound/dmasound_core.c b/sound/oss/dmasound/dmasound_core.c index 793b7f478433..3f3c3f71db4b 100644 --- a/sound/oss/dmasound/dmasound_core.c +++ b/sound/oss/dmasound/dmasound_core.c @@ -219,7 +219,9 @@ static int shared_resources_initialised; * Mid level stuff */ -struct sound_settings dmasound = { .lock = SPIN_LOCK_UNLOCKED }; +struct sound_settings dmasound = { + .lock = __SPIN_LOCK_UNLOCKED(dmasound.lock) +}; static inline void sound_silence(void) { diff --git a/sound/oss/hex2hex.c b/sound/oss/hex2hex.c index 5460faae98c9..041ef5c52bc2 100644 --- a/sound/oss/hex2hex.c +++ b/sound/oss/hex2hex.c @@ -12,7 +12,7 @@ #define MAX_SIZE (256*1024) unsigned char buf[MAX_SIZE]; -int loadhex(FILE *inf, unsigned char *buf) +static int loadhex(FILE *inf, unsigned char *buf) { int l=0, c, i; diff --git a/sound/oss/sb_common.c b/sound/oss/sb_common.c index 77d0e5efda76..ce4db49291f7 100644 --- a/sound/oss/sb_common.c +++ b/sound/oss/sb_common.c @@ -157,7 +157,7 @@ static void sb_intr (sb_devc *devc) break; default: - /* printk(KERN_WARN "Sound Blaster: Unexpected interrupt\n"); */ + /* printk(KERN_WARNING "Sound Blaster: Unexpected interrupt\n"); */ ; } } @@ -177,7 +177,7 @@ static void sb_intr (sb_devc *devc) break; default: - /* printk(KERN_WARN "Sound Blaster: Unexpected interrupt\n"); */ + /* printk(KERN_WARNING "Sound Blaster: Unexpected interrupt\n"); */ ; } } diff --git a/sound/oss/sb_ess.c b/sound/oss/sb_ess.c index 180e95c87e3e..51a3d381a59e 100644 --- a/sound/oss/sb_ess.c +++ b/sound/oss/sb_ess.c @@ -782,7 +782,7 @@ printk(KERN_INFO "FKS: ess_handle_channel %s irq_mode=%d\n", channel, irq_mode); break; default:; - /* printk(KERN_WARN "ESS: Unexpected interrupt\n"); */ + /* printk(KERN_WARNING "ESS: Unexpected interrupt\n"); */ } } diff --git a/sound/parisc/harmony.c b/sound/parisc/harmony.c index e924492df21d..f47f9e226b08 100644 --- a/sound/parisc/harmony.c +++ b/sound/parisc/harmony.c @@ -624,6 +624,9 @@ snd_harmony_pcm_init(struct snd_harmony *h) struct snd_pcm *pcm; int err; + if (snd_BUG_ON(!h)) + return -EINVAL; + harmony_disable_interrupts(h); err = snd_pcm_new(h->card, "harmony", 0, 1, 1, &pcm); @@ -865,11 +868,12 @@ snd_harmony_mixer_reset(struct snd_harmony *h) static int __devinit snd_harmony_mixer_init(struct snd_harmony *h) { - struct snd_card *card = h->card; + struct snd_card *card; int idx, err; if (snd_BUG_ON(!h)) return -EINVAL; + card = h->card; strcpy(card->mixername, "Harmony Gain control interface"); for (idx = 0; idx < HARMONY_CONTROLS; idx++) { diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index fb5ee3cc3968..75c602b5b132 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -259,7 +259,6 @@ config SND_CS5530 config SND_CS5535AUDIO tristate "CS5535/CS5536 Audio" - depends on X86 && !X86_64 select SND_PCM select SND_AC97_CODEC help diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c index b458d208720b..aaf4da68969c 100644 --- a/sound/pci/ali5451/ali5451.c +++ b/sound/pci/ali5451/ali5451.c @@ -973,7 +973,7 @@ static void snd_ali_free_voice(struct snd_ali * codec, void *private_data; snd_ali_printk("free_voice: channel=%d\n",pvoice->number); - if (pvoice == NULL || !pvoice->use) + if (!pvoice->use) return; snd_ali_clear_voices(codec, pvoice->number, pvoice->number); spin_lock_irq(&codec->voice_alloc); diff --git a/sound/pci/bt87x.c b/sound/pci/bt87x.c index 24585c6c6d01..4e2b925a94cc 100644 --- a/sound/pci/bt87x.c +++ b/sound/pci/bt87x.c @@ -808,6 +808,8 @@ static struct pci_device_id snd_bt87x_ids[] = { BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x1002, 0x0001, GENERIC), /* Leadtek Winfast tv 2000xp delux */ BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x107d, 0x6606, GENERIC), + /* Pinnacle PCTV */ + BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x11bd, 0x0012, GENERIC), /* Voodoo TV 200 */ BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x121a, 0x3000, GENERIC), /* Askey Computer Corp. MagicTView'99 */ diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index c9ad182e1b4b..6517f589d01d 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -722,9 +722,10 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus, chip->last_cmd[addr]); chip->single_cmd = 1; bus->response_reset = 0; - /* re-initialize CORB/RIRB */ + /* release CORB/RIRB */ azx_free_cmd_io(chip); - azx_init_cmd_io(chip); + /* disable unsolicited responses */ + azx_writel(chip, GCTL, azx_readl(chip, GCTL) & ~ICH6_GCTL_UNSOL); return -1; } @@ -865,7 +866,9 @@ static int azx_reset(struct azx *chip) } /* Accept unsolicited responses */ - azx_writel(chip, GCTL, azx_readl(chip, GCTL) | ICH6_GCTL_UNSOL); + if (!chip->single_cmd) + azx_writel(chip, GCTL, azx_readl(chip, GCTL) | + ICH6_GCTL_UNSOL); /* detect codecs */ if (!chip->codec_mask) { @@ -980,7 +983,8 @@ static void azx_init_chip(struct azx *chip) azx_int_enable(chip); /* initialize the codec command I/O */ - azx_init_cmd_io(chip); + if (!chip->single_cmd) + azx_init_cmd_io(chip); /* program the position buffer */ azx_writel(chip, DPLBASE, (u32)chip->posbuf.addr); @@ -2674,6 +2678,7 @@ static struct pci_device_id azx_ids[] = { { PCI_DEVICE(0x10de, 0x044b), .driver_data = AZX_DRIVER_NVIDIA }, { PCI_DEVICE(0x10de, 0x055c), .driver_data = AZX_DRIVER_NVIDIA }, { PCI_DEVICE(0x10de, 0x055d), .driver_data = AZX_DRIVER_NVIDIA }, + { PCI_DEVICE(0x10de, 0x0590), .driver_data = AZX_DRIVER_NVIDIA }, { PCI_DEVICE(0x10de, 0x0774), .driver_data = AZX_DRIVER_NVIDIA }, { PCI_DEVICE(0x10de, 0x0775), .driver_data = AZX_DRIVER_NVIDIA }, { PCI_DEVICE(0x10de, 0x0776), .driver_data = AZX_DRIVER_NVIDIA }, diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 3fbbc8c01e70..905859d4f4df 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -110,6 +110,7 @@ struct conexant_spec { unsigned int dell_automute; unsigned int port_d_mode; + unsigned char ext_mic_bias; }; static int conexant_playback_pcm_open(struct hda_pcm_stream *hinfo, @@ -1927,6 +1928,11 @@ static hda_nid_t cxt5066_adc_nids[3] = { 0x14, 0x15, 0x16 }; static hda_nid_t cxt5066_capsrc_nids[1] = { 0x17 }; #define CXT5066_SPDIF_OUT 0x21 +/* OLPC's microphone port is DC coupled for use with external sensors, + * therefore we use a 50% mic bias in order to center the input signal with + * the DC input range of the codec. */ +#define CXT5066_OLPC_EXT_MIC_BIAS PIN_VREF50 + static struct hda_channel_mode cxt5066_modes[1] = { { 2, NULL }, }; @@ -1980,9 +1986,10 @@ static int cxt5066_hp_master_sw_put(struct snd_kcontrol *kcontrol, /* toggle input of built-in and mic jack appropriately */ static void cxt5066_automic(struct hda_codec *codec) { - static struct hda_verb ext_mic_present[] = { + struct conexant_spec *spec = codec->spec; + struct hda_verb ext_mic_present[] = { /* enable external mic, port B */ - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, spec->ext_mic_bias}, /* switch to external mic input */ {0x17, AC_VERB_SET_CONNECT_SEL, 0}, @@ -2235,7 +2242,7 @@ static struct hda_verb cxt5066_init_verbs_olpc[] = { {0x19, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */ /* Port B: external microphone */ - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, CXT5066_OLPC_EXT_MIC_BIAS}, /* Port C: internal microphone */ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, @@ -2325,6 +2332,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = { CXT5066_LAPTOP), SND_PCI_QUIRK(0x1028, 0x02f5, "Dell", CXT5066_DELL_LAPTOP), + SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT5066_OLPC_XO_1_5), {} }; @@ -2352,6 +2360,7 @@ static int patch_cxt5066(struct hda_codec *codec) spec->input_mux = &cxt5066_capture_source; spec->port_d_mode = PIN_HP; + spec->ext_mic_bias = PIN_VREF80; spec->num_init_verbs = 1; spec->init_verbs[0] = cxt5066_init_verbs; @@ -2383,6 +2392,7 @@ static int patch_cxt5066(struct hda_codec *codec) spec->mixers[spec->num_mixers++] = cxt5066_mixer_master_olpc; spec->mixers[spec->num_mixers++] = cxt5066_mixers; spec->port_d_mode = 0; + spec->ext_mic_bias = CXT5066_OLPC_EXT_MIC_BIAS; /* no S/PDIF out */ spec->multiout.dig_out_nid = 0; diff --git a/sound/pci/hda/patch_nvhdmi.c b/sound/pci/hda/patch_nvhdmi.c index c8435c9a97f9..6afdab09bab7 100644 --- a/sound/pci/hda/patch_nvhdmi.c +++ b/sound/pci/hda/patch_nvhdmi.c @@ -29,6 +29,9 @@ #include "hda_codec.h" #include "hda_local.h" +/* define below to restrict the supported rates and formats */ +/* #define LIMITED_RATE_FMT_SUPPORT */ + struct nvhdmi_spec { struct hda_multi_out multiout; @@ -60,6 +63,22 @@ static struct hda_verb nvhdmi_basic_init[] = { {} /* terminator */ }; +#ifdef LIMITED_RATE_FMT_SUPPORT +/* support only the safe format and rate */ +#define SUPPORTED_RATES SNDRV_PCM_RATE_48000 +#define SUPPORTED_MAXBPS 16 +#define SUPPORTED_FORMATS SNDRV_PCM_FMTBIT_S16_LE +#else +/* support all rates and formats */ +#define SUPPORTED_RATES \ + (SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\ + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_176400 |\ + SNDRV_PCM_RATE_192000) +#define SUPPORTED_MAXBPS 24 +#define SUPPORTED_FORMATS \ + (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE) +#endif + /* * Controls */ @@ -258,9 +277,9 @@ static struct hda_pcm_stream nvhdmi_pcm_digital_playback_8ch = { .channels_min = 2, .channels_max = 8, .nid = Nv_Master_Convert_nid, - .rates = SNDRV_PCM_RATE_48000, - .maxbps = 16, - .formats = SNDRV_PCM_FMTBIT_S16_LE, + .rates = SUPPORTED_RATES, + .maxbps = SUPPORTED_MAXBPS, + .formats = SUPPORTED_FORMATS, .ops = { .open = nvhdmi_dig_playback_pcm_open, .close = nvhdmi_dig_playback_pcm_close_8ch, @@ -273,9 +292,9 @@ static struct hda_pcm_stream nvhdmi_pcm_digital_playback_2ch = { .channels_min = 2, .channels_max = 2, .nid = Nv_Master_Convert_nid, - .rates = SNDRV_PCM_RATE_48000, - .maxbps = 16, - .formats = SNDRV_PCM_FMTBIT_S16_LE, + .rates = SUPPORTED_RATES, + .maxbps = SUPPORTED_MAXBPS, + .formats = SUPPORTED_FORMATS, .ops = { .open = nvhdmi_dig_playback_pcm_open, .close = nvhdmi_dig_playback_pcm_close_2ch, @@ -378,6 +397,7 @@ static int patch_nvhdmi_2ch(struct hda_codec *codec) static struct hda_codec_preset snd_hda_preset_nvhdmi[] = { { .id = 0x10de0002, .name = "MCP78 HDMI", .patch = patch_nvhdmi_8ch }, { .id = 0x10de0003, .name = "MCP78 HDMI", .patch = patch_nvhdmi_8ch }, + { .id = 0x10de0005, .name = "MCP78 HDMI", .patch = patch_nvhdmi_8ch }, { .id = 0x10de0006, .name = "MCP78 HDMI", .patch = patch_nvhdmi_8ch }, { .id = 0x10de0007, .name = "MCP7A HDMI", .patch = patch_nvhdmi_8ch }, { .id = 0x10de0067, .name = "MCP67 HDMI", .patch = patch_nvhdmi_2ch }, @@ -387,6 +407,7 @@ static struct hda_codec_preset snd_hda_preset_nvhdmi[] = { MODULE_ALIAS("snd-hda-codec-id:10de0002"); MODULE_ALIAS("snd-hda-codec-id:10de0003"); +MODULE_ALIAS("snd-hda-codec-id:10de0005"); MODULE_ALIAS("snd-hda-codec-id:10de0006"); MODULE_ALIAS("snd-hda-codec-id:10de0007"); MODULE_ALIAS("snd-hda-codec-id:10de0067"); diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 7810d3dcad83..70583719282b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -275,7 +275,7 @@ struct alc_spec { struct snd_kcontrol_new *cap_mixer; /* capture mixer */ unsigned int beep_amp; /* beep amp value, set via set_beep_amp() */ - const struct hda_verb *init_verbs[5]; /* initialization verbs + const struct hda_verb *init_verbs[10]; /* initialization verbs * don't forget NULL * termination! */ @@ -965,6 +965,8 @@ static void alc_automute_pin(struct hda_codec *codec) unsigned int nid = spec->autocfg.hp_pins[0]; int i; + if (!nid) + return; pincap = snd_hda_query_pin_caps(codec, nid); if (pincap & AC_PINCAP_TRIG_REQ) /* need trigger? */ snd_hda_codec_read(codec, nid, 0, AC_VERB_SET_PIN_SENSE, 0); @@ -1332,15 +1334,20 @@ do_sku: * when the external headphone out jack is plugged" */ if (!spec->autocfg.hp_pins[0]) { + hda_nid_t nid; tmp = (ass >> 11) & 0x3; /* HP to chassis */ if (tmp == 0) - spec->autocfg.hp_pins[0] = porta; + nid = porta; else if (tmp == 1) - spec->autocfg.hp_pins[0] = porte; + nid = porte; else if (tmp == 2) - spec->autocfg.hp_pins[0] = portd; + nid = portd; else return 1; + for (i = 0; i < spec->autocfg.line_outs; i++) + if (spec->autocfg.line_out_pins[i] == nid) + return 1; + spec->autocfg.hp_pins[0] = nid; } alc_init_auto_hp(codec); @@ -1362,7 +1369,7 @@ static void alc_ssid_check(struct hda_codec *codec, } /* - * Fix-up pin default configurations + * Fix-up pin default configurations and add default verbs */ struct alc_pincfg { @@ -1370,9 +1377,14 @@ struct alc_pincfg { u32 val; }; -static void alc_fix_pincfg(struct hda_codec *codec, +struct alc_fixup { + const struct alc_pincfg *pins; + const struct hda_verb *verbs; +}; + +static void alc_pick_fixup(struct hda_codec *codec, const struct snd_pci_quirk *quirk, - const struct alc_pincfg **pinfix) + const struct alc_fixup *fix) { const struct alc_pincfg *cfg; @@ -1380,9 +1392,14 @@ static void alc_fix_pincfg(struct hda_codec *codec, if (!quirk) return; - cfg = pinfix[quirk->value]; - for (; cfg->nid; cfg++) - snd_hda_codec_set_pincfg(codec, cfg->nid, cfg->val); + fix += quirk->value; + cfg = fix->pins; + if (cfg) { + for (; cfg->nid; cfg++) + snd_hda_codec_set_pincfg(codec, cfg->nid, cfg->val); + } + if (fix->verbs) + add_verb(codec->spec, fix->verbs); } /* @@ -4667,9 +4684,9 @@ static int alc880_parse_auto_config(struct hda_codec *codec) spec->multiout.dig_out_nid = dig_nid; else { spec->multiout.slave_dig_outs = spec->slave_dig_outs; - spec->slave_dig_outs[i - 1] = dig_nid; - if (i == ARRAY_SIZE(spec->slave_dig_outs) - 1) + if (i >= ARRAY_SIZE(spec->slave_dig_outs) - 1) break; + spec->slave_dig_outs[i - 1] = dig_nid; } } if (spec->autocfg.dig_in_pin) @@ -6232,7 +6249,7 @@ static struct snd_pci_quirk alc260_cfg_tbl[] = { SND_PCI_QUIRK(0x1025, 0x008f, "Acer", ALC260_ACER), SND_PCI_QUIRK(0x1509, 0x4540, "Favorit 100XS", ALC260_FAVORIT100), SND_PCI_QUIRK(0x103c, 0x2808, "HP d5700", ALC260_HP_3013), - SND_PCI_QUIRK(0x103c, 0x280a, "HP d5750", ALC260_HP_3013), + SND_PCI_QUIRK(0x103c, 0x280a, "HP d5750", ALC260_AUTO), /* no quirk */ SND_PCI_QUIRK(0x103c, 0x3010, "HP", ALC260_HP_3013), SND_PCI_QUIRK(0x103c, 0x3011, "HP", ALC260_HP_3013), SND_PCI_QUIRK(0x103c, 0x3012, "HP", ALC260_HP_DC7600), @@ -8894,10 +8911,11 @@ static struct snd_pci_quirk alc882_ssid_cfg_tbl[] = { SND_PCI_QUIRK(0x106b, 0x3800, "MacbookPro 4,1", ALC885_MBP3), SND_PCI_QUIRK(0x106b, 0x3e00, "iMac 24 Aluminum", ALC885_IMAC24), SND_PCI_QUIRK(0x106b, 0x3f00, "Macbook 5,1", ALC885_MB5), - /* FIXME: HP jack sense seems not working for MBP 5,1, so apparently - * no perfect solution yet + /* FIXME: HP jack sense seems not working for MBP 5,1 or 5,2, + * so apparently no perfect solution yet */ SND_PCI_QUIRK(0x106b, 0x4000, "MacbookPro 5,1", ALC885_MB5), + SND_PCI_QUIRK(0x106b, 0x4600, "MacbookPro 5,2", ALC885_MB5), {} /* terminator */ }; @@ -9593,11 +9611,13 @@ static struct alc_pincfg alc882_abit_aw9d_pinfix[] = { { } }; -static const struct alc_pincfg *alc882_pin_fixes[] = { - [PINFIX_ABIT_AW9D_MAX] = alc882_abit_aw9d_pinfix, +static const struct alc_fixup alc882_fixups[] = { + [PINFIX_ABIT_AW9D_MAX] = { + .pins = alc882_abit_aw9d_pinfix + }, }; -static struct snd_pci_quirk alc882_pinfix_tbl[] = { +static struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", PINFIX_ABIT_AW9D_MAX), {} }; @@ -9794,9 +9814,9 @@ static int alc882_parse_auto_config(struct hda_codec *codec) spec->multiout.dig_out_nid = dig_nid; else { spec->multiout.slave_dig_outs = spec->slave_dig_outs; - spec->slave_dig_outs[i - 1] = dig_nid; - if (i == ARRAY_SIZE(spec->slave_dig_outs) - 1) + if (i >= ARRAY_SIZE(spec->slave_dig_outs) - 1) break; + spec->slave_dig_outs[i - 1] = dig_nid; } } if (spec->autocfg.dig_in_pin) @@ -9869,7 +9889,7 @@ static int patch_alc882(struct hda_codec *codec) board_config = ALC882_AUTO; } - alc_fix_pincfg(codec, alc882_pinfix_tbl, alc882_pin_fixes); + alc_pick_fixup(codec, alc882_fixup_tbl, alc882_fixups); if (board_config == ALC882_AUTO) { /* automatic parse from the BIOS config */ @@ -11441,6 +11461,8 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = { SND_PCI_QUIRK(0x104d, 0x820f, "Sony ASSAMD", ALC262_SONY_ASSAMD), SND_PCI_QUIRK(0x104d, 0x9016, "Sony VAIO", ALC262_AUTO), /* dig-only */ SND_PCI_QUIRK(0x104d, 0x9025, "Sony VAIO Z21MN", ALC262_TOSHIBA_S06), + SND_PCI_QUIRK(0x104d, 0x9035, "Sony VAIO VGN-FW170J", ALC262_AUTO), + SND_PCI_QUIRK(0x104d, 0x9047, "Sony VAIO Type G", ALC262_AUTO), SND_PCI_QUIRK_MASK(0x104d, 0xff00, 0x9000, "Sony VAIO", ALC262_SONY_ASSAMD), SND_PCI_QUIRK(0x1179, 0x0001, "Toshiba dynabook SS RX1", @@ -12585,7 +12607,8 @@ static struct snd_pci_quirk alc268_cfg_tbl[] = { SND_PCI_QUIRK(0x1025, 0x015b, "Acer Aspire One", ALC268_ACER_ASPIRE_ONE), SND_PCI_QUIRK(0x1028, 0x0253, "Dell OEM", ALC268_DELL), - SND_PCI_QUIRK(0x1028, 0x02b0, "Dell Inspiron Mini9", ALC268_DELL), + SND_PCI_QUIRK_MASK(0x1028, 0xfff0, 0x02b0, + "Dell Inspiron Mini9/Vostro A90", ALC268_DELL), /* almost compatible with toshiba but with optional digital outs; * auto-probing seems working fine */ @@ -12842,12 +12865,15 @@ static int patch_alc268(struct hda_codec *codec) unsigned int wcap = get_wcaps(codec, 0x07); int i; + spec->capsrc_nids = alc268_capsrc_nids; /* get type */ wcap = get_wcaps_type(wcap); if (spec->auto_mic || wcap != AC_WID_AUD_IN || spec->input_mux->num_items == 1) { spec->adc_nids = alc268_adc_nids_alt; spec->num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt); + if (spec->auto_mic) + fixup_automic_adc(codec); if (spec->auto_mic || spec->input_mux->num_items == 1) add_mixer(spec, alc268_capture_nosrc_mixer); else @@ -12857,7 +12883,6 @@ static int patch_alc268(struct hda_codec *codec) spec->num_adc_nids = ARRAY_SIZE(alc268_adc_nids); add_mixer(spec, alc268_capture_mixer); } - spec->capsrc_nids = alc268_capsrc_nids; /* set default input source */ for (i = 0; i < spec->num_adc_nids; i++) snd_hda_codec_write_cache(codec, alc268_capsrc_nids[i], @@ -14357,15 +14382,16 @@ static void alc861_auto_init_multi_out(struct hda_codec *codec) static void alc861_auto_init_hp_out(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - hda_nid_t pin; - pin = spec->autocfg.hp_pins[0]; - if (pin) - alc861_auto_set_output_and_unmute(codec, pin, PIN_HP, + if (spec->autocfg.hp_outs) + alc861_auto_set_output_and_unmute(codec, + spec->autocfg.hp_pins[0], + PIN_HP, spec->multiout.hp_nid); - pin = spec->autocfg.speaker_pins[0]; - if (pin) - alc861_auto_set_output_and_unmute(codec, pin, PIN_OUT, + if (spec->autocfg.speaker_outs) + alc861_auto_set_output_and_unmute(codec, + spec->autocfg.speaker_pins[0], + PIN_OUT, spec->multiout.dac_nids[0]); } @@ -15158,7 +15184,7 @@ static struct snd_pci_quirk alc861vd_cfg_tbl[] = { SND_PCI_QUIRK(0x1019, 0xa88d, "Realtek ALC660 demo", ALC660VD_3ST), SND_PCI_QUIRK(0x103c, 0x30bf, "HP TX1000", ALC861VD_HP), SND_PCI_QUIRK(0x1043, 0x12e2, "Asus z35m", ALC660VD_3ST), - SND_PCI_QUIRK(0x1043, 0x1339, "Asus G1", ALC660VD_3ST), + /*SND_PCI_QUIRK(0x1043, 0x1339, "Asus G1", ALC660VD_3ST),*/ /* auto */ SND_PCI_QUIRK(0x1043, 0x1633, "Asus V1Sn", ALC660VD_ASUS_V1S), SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS", ALC660VD_3ST_DIG), SND_PCI_QUIRK(0x10de, 0x03f0, "Realtek ALC660 demo", ALC660VD_3ST), @@ -15551,6 +15577,29 @@ static void alc861vd_auto_init(struct hda_codec *codec) alc_inithook(codec); } +enum { + ALC660VD_FIX_ASUS_GPIO1 +}; + +/* reset GPIO1 */ +static const struct hda_verb alc660vd_fix_asus_gpio1_verbs[] = { + {0x01, AC_VERB_SET_GPIO_MASK, 0x03}, + {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01}, + {0x01, AC_VERB_SET_GPIO_DATA, 0x01}, + { } +}; + +static const struct alc_fixup alc861vd_fixups[] = { + [ALC660VD_FIX_ASUS_GPIO1] = { + .verbs = alc660vd_fix_asus_gpio1_verbs, + }, +}; + +static struct snd_pci_quirk alc861vd_fixup_tbl[] = { + SND_PCI_QUIRK(0x1043, 0x1339, "ASUS A7-K", ALC660VD_FIX_ASUS_GPIO1), + {} +}; + static int patch_alc861vd(struct hda_codec *codec) { struct alc_spec *spec; @@ -15572,6 +15621,8 @@ static int patch_alc861vd(struct hda_codec *codec) board_config = ALC861VD_AUTO; } + alc_pick_fixup(codec, alc861vd_fixup_tbl, alc861vd_fixups); + if (board_config == ALC861VD_AUTO) { /* automatic parse from the BIOS config */ err = alc861vd_parse_auto_config(codec); @@ -17329,7 +17380,7 @@ static int alc662_auto_create_extra_out(struct hda_codec *codec, hda_nid_t pin, /* create playback/capture controls for input pins */ #define alc662_auto_create_input_ctls \ - alc880_auto_create_input_ctls + alc882_auto_create_input_ctls static void alc662_auto_set_output_and_unmute(struct hda_codec *codec, hda_nid_t nid, int pin_type, diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index a9b26828a651..86de305fc9f2 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -28,6 +28,7 @@ #include <linux/delay.h> #include <linux/slab.h> #include <linux/pci.h> +#include <linux/dmi.h> #include <sound/core.h> #include <sound/asoundef.h> #include <sound/jack.h> @@ -158,6 +159,7 @@ enum { STAC_D965_5ST_NO_FP, STAC_DELL_3ST, STAC_DELL_BIOS, + STAC_927X_VOLKNOB, STAC_927X_MODELS }; @@ -907,6 +909,16 @@ static struct hda_verb d965_core_init[] = { {} }; +static struct hda_verb dell_3st_core_init[] = { + /* don't set delta bit */ + {0x24, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0x7f}, + /* unmute node 0x1b */ + {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, + /* select node 0x03 as DAC */ + {0x0b, AC_VERB_SET_CONNECT_SEL, 0x01}, + {} +}; + static struct hda_verb stac927x_core_init[] = { /* set master volume and direct control */ { 0x24, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, @@ -915,6 +927,14 @@ static struct hda_verb stac927x_core_init[] = { {} }; +static struct hda_verb stac927x_volknob_core_init[] = { + /* don't set delta bit */ + {0x24, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0x7f}, + /* enable analog pc beep path */ + {0x01, AC_VERB_SET_DIGI_CONVERT_2, 1 << 5}, + {} +}; + static struct hda_verb stac9205_core_init[] = { /* set master volume and direct control */ { 0x24, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, @@ -1570,6 +1590,8 @@ static struct snd_pci_quirk stac92hd73xx_cfg_tbl[] = { "Dell Studio 17", STAC_DELL_M6_DMIC), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02be, "Dell Studio 1555", STAC_DELL_M6_DMIC), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02bd, + "Dell Studio 1557", STAC_DELL_M6_DMIC), {} /* terminator */ }; @@ -1674,6 +1696,8 @@ static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = { "DFI LanParty", STAC_92HD71BXX_REF), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30fb, "HP dv4-1222nr", STAC_HP_DV4_1222NR), + SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x1720, + "HP", STAC_HP_DV5), SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x3080, "HP", STAC_HP_DV5), SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x30f0, @@ -1999,6 +2023,7 @@ static unsigned int *stac927x_brd_tbl[STAC_927X_MODELS] = { [STAC_D965_5ST_NO_FP] = d965_5st_no_fp_pin_configs, [STAC_DELL_3ST] = dell_3st_pin_configs, [STAC_DELL_BIOS] = NULL, + [STAC_927X_VOLKNOB] = NULL, }; static const char *stac927x_models[STAC_927X_MODELS] = { @@ -2010,6 +2035,7 @@ static const char *stac927x_models[STAC_927X_MODELS] = { [STAC_D965_5ST_NO_FP] = "5stack-no-fp", [STAC_DELL_3ST] = "dell-3stack", [STAC_DELL_BIOS] = "dell-bios", + [STAC_927X_VOLKNOB] = "volknob", }; static struct snd_pci_quirk stac927x_cfg_tbl[] = { @@ -2045,6 +2071,8 @@ static struct snd_pci_quirk stac927x_cfg_tbl[] = { "Intel D965", STAC_D965_5ST), SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_INTEL, 0xff00, 0x2500, "Intel D965", STAC_D965_5ST), + /* volume-knob fixes */ + SND_PCI_QUIRK_VENDOR(0x10cf, "FSC", STAC_927X_VOLKNOB), {} /* terminator */ }; @@ -4642,6 +4670,26 @@ static void stac92xx_unsol_event(struct hda_codec *codec, unsigned int res) } } +static int hp_bseries_system(u32 subsystem_id) +{ + switch (subsystem_id) { + case 0x103c307e: + case 0x103c307f: + case 0x103c3080: + case 0x103c3081: + case 0x103c1722: + case 0x103c1723: + case 0x103c1724: + case 0x103c1725: + case 0x103c1726: + case 0x103c1727: + case 0x103c1728: + case 0x103c1729: + return 1; + } + return 0; +} + #ifdef CONFIG_PROC_FS static void stac92hd_proc_hook(struct snd_info_buffer *buffer, struct hda_codec *codec, hda_nid_t nid) @@ -4731,6 +4779,11 @@ static int stac92xx_hp_check_power_status(struct hda_codec *codec, else spec->gpio_data |= spec->gpio_led; /* white */ + if (hp_bseries_system(codec->subsystem_id)) { + /* LED state is inverted on these systems */ + spec->gpio_data ^= spec->gpio_led; + } + stac_gpio_set(codec, spec->gpio_mask, spec->gpio_dir, spec->gpio_data); @@ -5220,6 +5273,7 @@ static int patch_stac92hd71bxx(struct hda_codec *codec) { struct sigmatel_spec *spec; struct hda_verb *unmute_init = stac92hd71bxx_unmute_core_init; + unsigned int pin_cfg; int err = 0; spec = kzalloc(sizeof(*spec), GFP_KERNEL); @@ -5403,6 +5457,45 @@ again: break; } + if (hp_bseries_system(codec->subsystem_id)) { + pin_cfg = snd_hda_codec_get_pincfg(codec, 0x0f); + if (get_defcfg_device(pin_cfg) == AC_JACK_LINE_OUT || + get_defcfg_device(pin_cfg) == AC_JACK_SPEAKER || + get_defcfg_device(pin_cfg) == AC_JACK_HP_OUT) { + /* It was changed in the BIOS to just satisfy MS DTM. + * Lets turn it back into slaved HP + */ + pin_cfg = (pin_cfg & (~AC_DEFCFG_DEVICE)) + | (AC_JACK_HP_OUT << + AC_DEFCFG_DEVICE_SHIFT); + pin_cfg = (pin_cfg & (~(AC_DEFCFG_DEF_ASSOC + | AC_DEFCFG_SEQUENCE))) + | 0x1f; + snd_hda_codec_set_pincfg(codec, 0x0f, pin_cfg); + } + } + + if ((codec->subsystem_id >> 16) == PCI_VENDOR_ID_HP) { + const struct dmi_device *dev = NULL; + while ((dev = dmi_find_device(DMI_DEV_TYPE_OEM_STRING, + NULL, dev))) { + if (strcmp(dev->name, "HP_Mute_LED_1")) { + switch (codec->vendor_id) { + case 0x111d7608: + spec->gpio_led = 0x01; + break; + case 0x111d7600: + case 0x111d7601: + case 0x111d7602: + case 0x111d7603: + spec->gpio_led = 0x08; + break; + } + break; + } + } + } + #ifdef CONFIG_SND_HDA_POWER_SAVE if (spec->gpio_led) { spec->gpio_mask |= spec->gpio_led; @@ -5612,10 +5705,14 @@ static int patch_stac927x(struct hda_codec *codec) spec->dmic_nids = stac927x_dmic_nids; spec->num_dmics = STAC927X_NUM_DMICS; - spec->init = d965_core_init; + spec->init = dell_3st_core_init; spec->dmux_nids = stac927x_dmux_nids; spec->num_dmuxes = ARRAY_SIZE(stac927x_dmux_nids); break; + case STAC_927X_VOLKNOB: + spec->num_dmics = 0; + spec->init = stac927x_volknob_core_init; + break; default: spec->num_dmics = 0; spec->init = stac927x_core_init; diff --git a/sound/pci/ice1712/amp.c b/sound/pci/ice1712/amp.c index 37564300b50d..6da21a2bcade 100644 --- a/sound/pci/ice1712/amp.c +++ b/sound/pci/ice1712/amp.c @@ -52,11 +52,13 @@ static int __devinit snd_vt1724_amp_init(struct snd_ice1712 *ice) /* only use basic functionality for now */ - ice->num_total_dacs = 2; /* only PSDOUT0 is connected */ + /* VT1616 6ch codec connected to PSDOUT0 using packed mode */ + ice->num_total_dacs = 6; ice->num_total_adcs = 2; - /* Chaintech AV-710 has another codecs, which need initialization */ - /* initialize WM8728 codec */ + /* Chaintech AV-710 has another WM8728 codec connected to PSDOUT4 + (shared with the SPDIF output). Mixer control for this codec + is not yet supported. */ if (ice->eeprom.subvendor == VT1724_SUBDEVICE_AV710) { for (i = 0; i < ARRAY_SIZE(wm_inits); i += 2) wm_put(ice, wm_inits[i], wm_inits[i+1]); diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index cecf1ffeeaaa..d74033a2cfbe 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -2259,7 +2259,7 @@ static int snd_ice1712_pro_peak_get(struct snd_kcontrol *kcontrol, } static struct snd_kcontrol_new snd_ice1712_mixer_pro_peak __devinitdata = { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .iface = SNDRV_CTL_ELEM_IFACE_PCM, .name = "Multi Track Peak", .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, .info = snd_ice1712_pro_peak_info, diff --git a/sound/pci/ice1712/ice1712.h b/sound/pci/ice1712/ice1712.h index 9da2dae64c5b..d063149e7047 100644 --- a/sound/pci/ice1712/ice1712.h +++ b/sound/pci/ice1712/ice1712.h @@ -382,8 +382,8 @@ struct snd_ice1712 { #ifdef CONFIG_PM int (*pm_suspend)(struct snd_ice1712 *); int (*pm_resume)(struct snd_ice1712 *); - int pm_suspend_enabled:1; - int pm_saved_is_spdif_master:1; + unsigned int pm_suspend_enabled:1; + unsigned int pm_saved_is_spdif_master:1; unsigned int pm_saved_spdif_ctrl; unsigned char pm_saved_spdif_cfg; unsigned int pm_saved_route; diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index af6e00148621..10fc92c05574 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -648,7 +648,7 @@ static int snd_vt1724_set_pro_rate(struct snd_ice1712 *ice, unsigned int rate, (inb(ICEMT1724(ice, DMA_PAUSE)) & DMA_PAUSES)) { /* running? we cannot change the rate now... */ spin_unlock_irqrestore(&ice->reg_lock, flags); - return -EBUSY; + return ((rate == ice->cur_rate) && !force) ? 0 : -EBUSY; } if (!force && is_pro_rate_locked(ice)) { spin_unlock_irqrestore(&ice->reg_lock, flags); @@ -1294,7 +1294,7 @@ static int __devinit snd_vt1724_pcm_spdif(struct snd_ice1712 *ice, int device) snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(ice->pci), - 64*1024, 64*1024); + 256*1024, 256*1024); ice->pcm = pcm; @@ -1408,7 +1408,7 @@ static int __devinit snd_vt1724_pcm_indep(struct snd_ice1712 *ice, int device) snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(ice->pci), - 64*1024, 64*1024); + 256*1024, 256*1024); ice->pcm_ds = pcm; @@ -2110,7 +2110,7 @@ static int snd_vt1724_pro_peak_get(struct snd_kcontrol *kcontrol, } static struct snd_kcontrol_new snd_vt1724_mixer_pro_peak __devinitdata = { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .iface = SNDRV_CTL_ELEM_IFACE_PCM, .name = "Multi Track Peak", .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, .info = snd_vt1724_pro_peak_info, diff --git a/sound/pci/ice1712/prodigy_hifi.c b/sound/pci/ice1712/prodigy_hifi.c index c75515f5be6f..6a9fee3ee78f 100644 --- a/sound/pci/ice1712/prodigy_hifi.c +++ b/sound/pci/ice1712/prodigy_hifi.c @@ -1100,7 +1100,7 @@ static void ak4396_init(struct snd_ice1712 *ice) } #ifdef CONFIG_PM -static int __devinit prodigy_hd2_resume(struct snd_ice1712 *ice) +static int prodigy_hd2_resume(struct snd_ice1712 *ice) { /* initialize ak4396 codec and restore previous mixer volumes */ struct prodigy_hifi_spec *spec = ice->spec; diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index 754867ed4785..aac20fb4aad2 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -1950,6 +1950,12 @@ static struct ac97_quirk ac97_quirks[] __devinitdata = { }, { .subvendor = 0x104d, + .subdevice = 0x8144, + .name = "Sony", + .type = AC97_TUNE_INV_EAPD + }, + { + .subvendor = 0x104d, .subdevice = 0x8197, .name = "Sony S1XP", .type = AC97_TUNE_INV_EAPD diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index acfa4760da49..8a332d2f615c 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -386,6 +386,7 @@ struct via82xx { struct snd_pcm *pcms[2]; struct snd_rawmidi *rmidi; + struct snd_kcontrol *dxs_controls[4]; struct snd_ac97_bus *ac97_bus; struct snd_ac97 *ac97; @@ -1216,9 +1217,9 @@ static int snd_via82xx_pcm_open(struct via82xx *chip, struct viadev *viadev, /* - * open callback for playback on via686 and via823x DSX + * open callback for playback on via686 */ -static int snd_via82xx_playback_open(struct snd_pcm_substream *substream) +static int snd_via686_playback_open(struct snd_pcm_substream *substream) { struct via82xx *chip = snd_pcm_substream_chip(substream); struct viadev *viadev = &chip->devs[chip->playback_devno + substream->number]; @@ -1230,6 +1231,32 @@ static int snd_via82xx_playback_open(struct snd_pcm_substream *substream) } /* + * open callback for playback on via823x DXS + */ +static int snd_via8233_playback_open(struct snd_pcm_substream *substream) +{ + struct via82xx *chip = snd_pcm_substream_chip(substream); + struct viadev *viadev; + unsigned int stream; + int err; + + viadev = &chip->devs[chip->playback_devno + substream->number]; + if ((err = snd_via82xx_pcm_open(chip, viadev, substream)) < 0) + return err; + stream = viadev->reg_offset / 0x10; + if (chip->dxs_controls[stream]) { + chip->playback_volume[stream][0] = 0; + chip->playback_volume[stream][1] = 0; + chip->dxs_controls[stream]->vd[0].access &= + ~SNDRV_CTL_ELEM_ACCESS_INACTIVE; + snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE | + SNDRV_CTL_EVENT_MASK_INFO, + &chip->dxs_controls[stream]->id); + } + return 0; +} + +/* * open callback for playback on via823x multi-channel */ static int snd_via8233_multi_open(struct snd_pcm_substream *substream) @@ -1302,10 +1329,26 @@ static int snd_via82xx_pcm_close(struct snd_pcm_substream *substream) return 0; } +static int snd_via8233_playback_close(struct snd_pcm_substream *substream) +{ + struct via82xx *chip = snd_pcm_substream_chip(substream); + struct viadev *viadev = substream->runtime->private_data; + unsigned int stream; + + stream = viadev->reg_offset / 0x10; + if (chip->dxs_controls[stream]) { + chip->dxs_controls[stream]->vd[0].access |= + SNDRV_CTL_ELEM_ACCESS_INACTIVE; + snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_INFO, + &chip->dxs_controls[stream]->id); + } + return snd_via82xx_pcm_close(substream); +} + /* via686 playback callbacks */ static struct snd_pcm_ops snd_via686_playback_ops = { - .open = snd_via82xx_playback_open, + .open = snd_via686_playback_open, .close = snd_via82xx_pcm_close, .ioctl = snd_pcm_lib_ioctl, .hw_params = snd_via82xx_hw_params, @@ -1331,8 +1374,8 @@ static struct snd_pcm_ops snd_via686_capture_ops = { /* via823x DSX playback callbacks */ static struct snd_pcm_ops snd_via8233_playback_ops = { - .open = snd_via82xx_playback_open, - .close = snd_via82xx_pcm_close, + .open = snd_via8233_playback_open, + .close = snd_via8233_playback_close, .ioctl = snd_pcm_lib_ioctl, .hw_params = snd_via82xx_hw_params, .hw_free = snd_via82xx_hw_free, @@ -1626,7 +1669,7 @@ static int snd_via8233_dxs_volume_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct via82xx *chip = snd_kcontrol_chip(kcontrol); - unsigned int idx = snd_ctl_get_ioff(kcontrol, &ucontrol->id); + unsigned int idx = kcontrol->id.subdevice; ucontrol->value.integer.value[0] = VIA_DXS_MAX_VOLUME - chip->playback_volume[idx][0]; ucontrol->value.integer.value[1] = VIA_DXS_MAX_VOLUME - chip->playback_volume[idx][1]; @@ -1646,7 +1689,7 @@ static int snd_via8233_dxs_volume_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct via82xx *chip = snd_kcontrol_chip(kcontrol); - unsigned int idx = snd_ctl_get_ioff(kcontrol, &ucontrol->id); + unsigned int idx = kcontrol->id.subdevice; unsigned long port = chip->port + 0x10 * idx; unsigned char val; int i, change = 0; @@ -1705,11 +1748,13 @@ static struct snd_kcontrol_new snd_via8233_pcmdxs_volume_control __devinitdata = }; static struct snd_kcontrol_new snd_via8233_dxs_volume_control __devinitdata = { - .name = "VIA DXS Playback Volume", - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | - SNDRV_CTL_ELEM_ACCESS_TLV_READ), - .count = 4, + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .device = 0, + /* .subdevice set later */ + .name = "PCM Playback Volume", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ | + SNDRV_CTL_ELEM_ACCESS_INACTIVE, .info = snd_via8233_dxs_volume_info, .get = snd_via8233_dxs_volume_get, .put = snd_via8233_dxs_volume_put, @@ -1936,10 +1981,19 @@ static int __devinit snd_via8233_init_misc(struct via82xx *chip) } else /* Using DXS when PCM emulation is enabled is really weird */ { - /* Standalone DXS controls */ - err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_via8233_dxs_volume_control, chip)); - if (err < 0) - return err; + for (i = 0; i < 4; ++i) { + struct snd_kcontrol *kctl; + + kctl = snd_ctl_new1( + &snd_via8233_dxs_volume_control, chip); + if (!kctl) + return -ENOMEM; + kctl->id.subdevice = i; + err = snd_ctl_add(chip->card, kctl); + if (err < 0) + return err; + chip->dxs_controls[i] = kctl; + } } } /* select spdif data slot 10/11 */ diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf.c b/sound/pcmcia/pdaudiocf/pdaudiocf.c index 7dea74b71cf1..64b859925c0b 100644 --- a/sound/pcmcia/pdaudiocf/pdaudiocf.c +++ b/sound/pcmcia/pdaudiocf/pdaudiocf.c @@ -217,20 +217,25 @@ static void snd_pdacf_detach(struct pcmcia_device *link) * configuration callback */ -#define CS_CHECK(fn, ret) \ -do { last_fn = (fn); if ((last_ret = (ret)) != 0) goto cs_failed; } while (0) - static int pdacf_config(struct pcmcia_device *link) { struct snd_pdacf *pdacf = link->priv; - int last_fn, last_ret; + int ret; snd_printdd(KERN_DEBUG "pdacf_config called\n"); link->conf.ConfigIndex = 0x5; - CS_CHECK(RequestIO, pcmcia_request_io(link, &link->io)); - CS_CHECK(RequestIRQ, pcmcia_request_irq(link, &link->irq)); - CS_CHECK(RequestConfiguration, pcmcia_request_configuration(link, &link->conf)); + ret = pcmcia_request_io(link, &link->io); + if (ret) + goto failed; + + ret = pcmcia_request_irq(link, &link->irq); + if (ret) + goto failed; + + ret = pcmcia_request_configuration(link, &link->conf); + if (ret) + goto failed; if (snd_pdacf_assign_resources(pdacf, link->io.BasePort1, link->irq.AssignedIRQ) < 0) goto failed; @@ -238,8 +243,6 @@ static int pdacf_config(struct pcmcia_device *link) link->dev_node = &pdacf->node; return 0; -cs_failed: - cs_error(link, last_fn, last_ret); failed: pcmcia_disable_device(link); return -ENODEV; diff --git a/sound/pcmcia/vx/vxpocket.c b/sound/pcmcia/vx/vxpocket.c index 7445cc8a47d3..1492744ad67f 100644 --- a/sound/pcmcia/vx/vxpocket.c +++ b/sound/pcmcia/vx/vxpocket.c @@ -213,14 +213,11 @@ static int snd_vxpocket_assign_resources(struct vx_core *chip, int port, int irq * configuration callback */ -#define CS_CHECK(fn, ret) \ -do { last_fn = (fn); if ((last_ret = (ret)) != 0) goto cs_failed; } while (0) - static int vxpocket_config(struct pcmcia_device *link) { struct vx_core *chip = link->priv; struct snd_vxpocket *vxp = (struct snd_vxpocket *)chip; - int last_fn, last_ret; + int ret; snd_printdd(KERN_DEBUG "vxpocket_config called\n"); @@ -235,9 +232,17 @@ static int vxpocket_config(struct pcmcia_device *link) strcpy(chip->card->driver, vxp440_hw.name); } - CS_CHECK(RequestIO, pcmcia_request_io(link, &link->io)); - CS_CHECK(RequestIRQ, pcmcia_request_irq(link, &link->irq)); - CS_CHECK(RequestConfiguration, pcmcia_request_configuration(link, &link->conf)); + ret = pcmcia_request_io(link, &link->io); + if (ret) + goto failed; + + ret = pcmcia_request_irq(link, &link->irq); + if (ret) + goto failed; + + ret = pcmcia_request_configuration(link, &link->conf); + if (ret) + goto failed; chip->dev = &handle_to_dev(link); snd_card_set_dev(chip->card, chip->dev); @@ -248,8 +253,6 @@ static int vxpocket_config(struct pcmcia_device *link) link->dev_node = &vxp->node; return 0; -cs_failed: - cs_error(link, last_fn, last_ret); failed: pcmcia_disable_device(link); return -ENODEV; diff --git a/sound/ppc/Kconfig b/sound/ppc/Kconfig index bd2338ab2ced..0519c60f5be1 100644 --- a/sound/ppc/Kconfig +++ b/sound/ppc/Kconfig @@ -2,7 +2,7 @@ menuconfig SND_PPC bool "PowerPC sound devices" - depends on PPC64 || PPC32 + depends on PPC default y help Support for sound devices specific to PowerPC architectures. diff --git a/sound/sh/aica.c b/sound/sh/aica.c index 583a3693df75..a0df401ebb9f 100644 --- a/sound/sh/aica.c +++ b/sound/sh/aica.c @@ -49,6 +49,7 @@ MODULE_AUTHOR("Adrian McMenamin <adrian@mcmen.demon.co.uk>"); MODULE_DESCRIPTION("Dreamcast AICA sound (pcm) driver"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Yamaha/SEGA, AICA}}"); +MODULE_FIRMWARE("aica_firmware.bin"); /* module parameters */ #define CARD_NAME "AICA" diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index 0b8dcb5cd729..6b24d8bb02bb 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -265,8 +265,8 @@ static const int bosr_usb_divisor_table[] = { #define UPPER_GROUP ((1<<8) | (1<<9) | (1<<10) | (1<<11) | (1<<15)) static const unsigned short sr_valid_mask[] = { LOWER_GROUP|UPPER_GROUP, /* Normal, bosr - 0*/ - LOWER_GROUP|UPPER_GROUP, /* Normal, bosr - 1*/ LOWER_GROUP, /* Usb, bosr - 0*/ + LOWER_GROUP|UPPER_GROUP, /* Normal, bosr - 1*/ UPPER_GROUP, /* Usb, bosr - 1*/ }; /* diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 3ff0373dff89..593d5b9c9f03 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -579,7 +579,7 @@ static const struct snd_kcontrol_new wm8350_left_capt_mixer_controls[] = { SOC_DAPM_SINGLE_TLV("L3 Capture Volume", WM8350_INPUT_MIXER_VOLUME_L, 9, 7, 0, out_mix_tlv), SOC_DAPM_SINGLE("PGA Capture Switch", - WM8350_LEFT_INPUT_VOLUME, 14, 1, 0), + WM8350_LEFT_INPUT_VOLUME, 14, 1, 1), }; /* Right Input Mixer */ @@ -589,7 +589,7 @@ static const struct snd_kcontrol_new wm8350_right_capt_mixer_controls[] = { SOC_DAPM_SINGLE_TLV("L3 Capture Volume", WM8350_INPUT_MIXER_VOLUME_R, 13, 7, 0, out_mix_tlv), SOC_DAPM_SINGLE("PGA Capture Switch", - WM8350_RIGHT_INPUT_VOLUME, 14, 1, 0), + WM8350_RIGHT_INPUT_VOLUME, 14, 1, 1), }; /* Left Mic Mixer */ diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index da97aae475a2..1ef2454c5205 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -790,7 +790,7 @@ static int wm8940_register(struct wm8940_priv *wm8940, codec->reg_cache = &wm8940->reg_cache; ret = snd_soc_codec_set_cache_io(codec, 8, 16, control); - if (ret == 0) { + if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; } diff --git a/sound/soc/imx/mxc-ssi.c b/sound/soc/imx/mxc-ssi.c index 3806ff2c0cd4..ccdefe60e752 100644 --- a/sound/soc/imx/mxc-ssi.c +++ b/sound/soc/imx/mxc-ssi.c @@ -397,14 +397,6 @@ static int imx_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, break; } - /* sync */ - if (!(fmt & SND_SOC_DAIFMT_ASYNC)) - scr |= SSI_SCR_SYN; - - /* tdm - only for stereo atm */ - if (fmt & SND_SOC_DAIFMT_TDM) - scr |= SSI_SCR_NET; - if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) { SSI1_STCR = stcr; SSI1_SRCR = srcr; diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index 2dee9839be86..653a362425df 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -21,7 +21,18 @@ config SND_OMAP_SOC_AMS_DELTA select SND_OMAP_SOC_MCBSP select SND_SOC_CX20442 help - Say Y if you want to add support for SoC audio on Amstrad Delta. + Say Y if you want to add support for SoC audio device connected to + a handset and a speakerphone found on Amstrad E3 (Delta) videophone. + + Note that in order to get those devices fully supported, you have to + build the kernel with standard serial port driver included and + configured for at least 4 ports. Then, from userspace, you must load + a line discipline #19 on the modem (ttyS3) serial line. The simplest + way to achieve this is to install util-linux-ng and use the included + ldattach utility. This can be started automatically from udev, + a simple rule like this one should do the trick (it does for me): + ACTION=="add", KERNEL=="controlC0", \ + RUN+="/usr/sbin/ldattach 19 /dev/ttyS3" config SND_OMAP_SOC_OSK5912 tristate "SoC Audio support for omap osk5912" diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index 5735945788bf..6a829eef2a4f 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -195,8 +195,12 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream) else omap_enable_dma_irq(prtd->dma_ch, OMAP_DMA_FRAME_IRQ); - omap_set_dma_src_burst_mode(prtd->dma_ch, OMAP_DMA_DATA_BURST_16); - omap_set_dma_dest_burst_mode(prtd->dma_ch, OMAP_DMA_DATA_BURST_16); + if (!(cpu_class_is_omap1())) { + omap_set_dma_src_burst_mode(prtd->dma_ch, + OMAP_DMA_DATA_BURST_16); + omap_set_dma_dest_burst_mode(prtd->dma_ch, + OMAP_DMA_DATA_BURST_16); + } return 0; } diff --git a/sound/soc/omap/omap3evm.c b/sound/soc/omap/omap3evm.c index 9114c263077b..13aa380de162 100644 --- a/sound/soc/omap/omap3evm.c +++ b/sound/soc/omap/omap3evm.c @@ -144,4 +144,4 @@ module_exit(omap3evm_soc_exit); MODULE_AUTHOR("Anuj Aggarwal <anuj.aggarwal@ti.com>"); MODULE_DESCRIPTION("ALSA SoC OMAP3 EVM"); -MODULE_LICENSE("GPLv2"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c index ad219aaf7cb8..0cd06f5dd356 100644 --- a/sound/soc/omap/omap3pandora.c +++ b/sound/soc/omap/omap3pandora.c @@ -134,7 +134,7 @@ static int omap3pandora_hp_event(struct snd_soc_dapm_widget *w, * |P| <--- TWL4030 <--------- Line In and MICs */ static const struct snd_soc_dapm_widget omap3pandora_out_dapm_widgets[] = { - SND_SOC_DAPM_DAC("PCM DAC", "Playback", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("PCM DAC", "HiFi Playback", SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_PGA_E("Headphone Amplifier", SND_SOC_NOPM, 0, 0, NULL, 0, omap3pandora_hp_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), @@ -181,6 +181,7 @@ static int omap3pandora_out_init(struct snd_soc_codec *codec) snd_soc_dapm_nc_pin(codec, "CARKITR"); snd_soc_dapm_nc_pin(codec, "HFL"); snd_soc_dapm_nc_pin(codec, "HFR"); + snd_soc_dapm_nc_pin(codec, "VIBRA"); ret = snd_soc_dapm_new_controls(codec, omap3pandora_out_dapm_widgets, ARRAY_SIZE(omap3pandora_out_dapm_widgets)); diff --git a/sound/soc/s3c24xx/s3c24xx-pcm.c b/sound/soc/s3c24xx/s3c24xx-pcm.c index 5cbbdc80fde3..1f35c6fcf5fd 100644 --- a/sound/soc/s3c24xx/s3c24xx-pcm.c +++ b/sound/soc/s3c24xx/s3c24xx-pcm.c @@ -75,11 +75,19 @@ static void s3c24xx_pcm_enqueue(struct snd_pcm_substream *substream) { struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; dma_addr_t pos = prtd->dma_pos; + unsigned int limit; int ret; pr_debug("Entered %s\n", __func__); - while (prtd->dma_loaded < prtd->dma_limit) { + if (s3c_dma_has_circular()) { + limit = (prtd->dma_end - prtd->dma_start) / prtd->dma_period; + } else + limit = prtd->dma_limit; + + pr_debug("%s: loaded %d, limit %d\n", __func__, prtd->dma_loaded, limit); + + while (prtd->dma_loaded < limit) { unsigned long len = prtd->dma_period; pr_debug("dma_loaded: %d\n", prtd->dma_loaded); @@ -123,7 +131,7 @@ static void s3c24xx_audio_buffdone(struct s3c2410_dma_chan *channel, snd_pcm_period_elapsed(substream); spin_lock(&prtd->lock); - if (prtd->state & ST_RUNNING) { + if (prtd->state & ST_RUNNING && !s3c_dma_has_circular()) { prtd->dma_loaded--; s3c24xx_pcm_enqueue(substream); } @@ -164,6 +172,11 @@ static int s3c24xx_pcm_hw_params(struct snd_pcm_substream *substream, printk(KERN_ERR "failed to get dma channel\n"); return ret; } + + /* use the circular buffering if we have it available. */ + if (s3c_dma_has_circular()) + s3c2410_dma_setflags(prtd->params->channel, + S3C2410_DMAF_CIRCULAR); } s3c2410_dma_set_buffdone_fn(prtd->params->channel, diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.c b/sound/soc/s3c24xx/s3c64xx-i2s.c index 3c06c401d0fb..105a77eeded0 100644 --- a/sound/soc/s3c24xx/s3c64xx-i2s.c +++ b/sound/soc/s3c24xx/s3c64xx-i2s.c @@ -220,6 +220,8 @@ static __devinit int s3c64xx_iis_dev_probe(struct platform_device *pdev) goto err; } + clk_enable(i2s->iis_cclk); + ret = s3c_i2sv2_probe(pdev, dai, i2s, 0); if (ret) goto err_clk; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 7ff04ad2a97e..0a1b2f64bbee 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -834,6 +834,9 @@ EXPORT_SYMBOL_GPL(snd_soc_resume_device); #define soc_resume NULL #endif +static struct snd_soc_dai_ops null_dai_ops = { +}; + static void snd_soc_instantiate_card(struct snd_soc_card *card) { struct platform_device *pdev = container_of(card->dev, @@ -877,6 +880,11 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) ac97 = 1; } + for (i = 0; i < card->num_links; i++) { + if (!card->dai_link[i].codec_dai->ops) + card->dai_link[i].codec_dai->ops = &null_dai_ops; + } + /* If we have AC97 in the system then don't wait for the * codec. This will need revisiting if we have to handle * systems with mixed AC97 and non-AC97 parts. Only check for @@ -2329,9 +2337,6 @@ static int snd_soc_unregister_card(struct snd_soc_card *card) return 0; } -static struct snd_soc_dai_ops null_dai_ops = { -}; - /** * snd_soc_register_dai - Register a DAI with the ASoC core * diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index f79711b9fa5b..d89f6dc00908 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -524,7 +524,7 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget) /* connected jack or spk ? */ if (widget->id == snd_soc_dapm_hp || widget->id == snd_soc_dapm_spk || - widget->id == snd_soc_dapm_line) + (widget->id == snd_soc_dapm_line && !list_empty(&widget->sources))) return 1; } @@ -573,7 +573,8 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget) return 1; /* connected jack ? */ - if (widget->id == snd_soc_dapm_mic || widget->id == snd_soc_dapm_line) + if (widget->id == snd_soc_dapm_mic || + (widget->id == snd_soc_dapm_line && !list_empty(&widget->sinks))) return 1; } @@ -2071,9 +2072,9 @@ int snd_soc_dapm_stream_event(struct snd_soc_codec *codec, } } } - mutex_unlock(&codec->mutex); dapm_power_widgets(codec, event); + mutex_unlock(&codec->mutex); dump_dapm(codec, __func__); return 0; } diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c index 121af0644fd9..86b2c3b92df5 100644 --- a/sound/usb/caiaq/audio.c +++ b/sound/usb/caiaq/audio.c @@ -62,10 +62,14 @@ static void activate_substream(struct snd_usb_caiaqdev *dev, struct snd_pcm_substream *sub) { + spin_lock(&dev->spinlock); + if (sub->stream == SNDRV_PCM_STREAM_PLAYBACK) dev->sub_playback[sub->number] = sub; else dev->sub_capture[sub->number] = sub; + + spin_unlock(&dev->spinlock); } static void @@ -269,16 +273,22 @@ snd_usb_caiaq_pcm_pointer(struct snd_pcm_substream *sub) { int index = sub->number; struct snd_usb_caiaqdev *dev = snd_pcm_substream_chip(sub); + snd_pcm_uframes_t ptr; + + spin_lock(&dev->spinlock); if (dev->input_panic || dev->output_panic) - return SNDRV_PCM_POS_XRUN; + ptr = SNDRV_PCM_POS_XRUN; if (sub->stream == SNDRV_PCM_STREAM_PLAYBACK) - return bytes_to_frames(sub->runtime, + ptr = bytes_to_frames(sub->runtime, dev->audio_out_buf_pos[index]); else - return bytes_to_frames(sub->runtime, + ptr = bytes_to_frames(sub->runtime, dev->audio_in_buf_pos[index]); + + spin_unlock(&dev->spinlock); + return ptr; } /* operators for both playback and capture */ diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c index 83e6c1312d47..a3f02dd97440 100644 --- a/sound/usb/caiaq/device.c +++ b/sound/usb/caiaq/device.c @@ -35,7 +35,7 @@ #include "input.h" MODULE_AUTHOR("Daniel Mack <daniel@caiaq.de>"); -MODULE_DESCRIPTION("caiaq USB audio, version 1.3.19"); +MODULE_DESCRIPTION("caiaq USB audio, version 1.3.20"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Native Instruments, RigKontrol2}," "{Native Instruments, RigKontrol3}," diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 8e7f78941ba6..e9a3a9dca15c 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -210,7 +210,7 @@ struct snd_usb_midi_endpoint_info { /* */ -#define combine_word(s) ((*s) | ((unsigned int)(s)[1] << 8)) +#define combine_word(s) ((*(s)) | ((unsigned int)(s)[1] << 8)) #define combine_triple(s) (combine_word(s) | ((unsigned int)(s)[2] << 16)) #define combine_quad(s) (combine_triple(s) | ((unsigned int)(s)[3] << 24)) diff --git a/sound/usb/usbmixer.c b/sound/usb/usbmixer.c index 9efcfd08d747..c998220b99c6 100644 --- a/sound/usb/usbmixer.c +++ b/sound/usb/usbmixer.c @@ -1071,6 +1071,15 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, unsig channels = (ftr[0] - 7) / csize - 1; master_bits = snd_usb_combine_bytes(ftr + 6, csize); + /* master configuration quirks */ + switch (state->chip->usb_id) { + case USB_ID(0x08bb, 0x2702): + snd_printk(KERN_INFO + "usbmixer: master volume quirk for PCM2702 chip\n"); + /* disable non-functional volume control */ + master_bits &= ~(1 << (USB_FEATURE_VOLUME - 1)); + break; + } if (channels > 0) first_ch_bits = snd_usb_combine_bytes(ftr + 6 + csize, csize); else |