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Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6

Browse Source 
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
  ALSA: emu10k1 - delay the PCM interrupts (add pcm_irq_delay parameter)
  ALSA: hda - Fix ALC680 base model capture
  ASoC: Remove DSP mode support for WM8776
  ALSA: hda - Add quirk for Dell Vostro 1220
  ALSA: riptide - Fix detection / load of firmware files
This commit is contained in:
Linus Torvalds 2010-08-18 09:30:08 -07:00
commit 7dfb2d4069
9 changed files with 188 additions and 50 deletions
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1
include/sound/emu10k1.h
View File

@ -1707,6 +1707,7 @@ struct snd_emu10k1 {
unsigned int card_type; /* EMU10K1_CARD_* */ unsigned int card_type; /* EMU10K1_CARD_* */
unsigned int ecard_ctrl; /* ecard control bits */ unsigned int ecard_ctrl; /* ecard control bits */
unsigned long dma_mask; /* PCI DMA mask */ unsigned long dma_mask; /* PCI DMA mask */
unsigned int delay_pcm_irq; /* in samples */
int max_cache_pages; /* max memory size / PAGE_SIZE */ int max_cache_pages; /* max memory size / PAGE_SIZE */
struct snd_dma_buffer silent_page; /* silent page */ struct snd_dma_buffer silent_page; /* silent page */
struct snd_dma_buffer ptb_pages; /* page table pages */ struct snd_dma_buffer ptb_pages; /* page table pages */

4
sound/core/pcm_native.c
View File

@ -978,6 +978,10 @@ static int snd_pcm_do_pause(struct snd_pcm_substream *substream, int push)
{ {
if (substream->runtime->trigger_master != substream) if (substream->runtime->trigger_master != substream)
return 0; return 0;
/* some drivers might use hw_ptr to recover from the pause -
update the hw_ptr now */
if (push)
snd_pcm_update_hw_ptr(substream);
/* The jiffies check in snd_pcm_update_hw_ptr*() is done by /* The jiffies check in snd_pcm_update_hw_ptr*() is done by
* a delta betwen the current jiffies, this gives a large enough * a delta betwen the current jiffies, this gives a large enough
* delta, effectively to skip the check once. * delta, effectively to skip the check once.

4
sound/pci/emu10k1/emu10k1.c
View File

@ -52,6 +52,7 @@ static int max_synth_voices[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 64};
static int max_buffer_size[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 128}; static int max_buffer_size[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 128};
static int enable_ir[SNDRV_CARDS]; static int enable_ir[SNDRV_CARDS];
static uint subsystem[SNDRV_CARDS]; /* Force card subsystem model */ static uint subsystem[SNDRV_CARDS]; /* Force card subsystem model */
static uint delay_pcm_irq[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 2};
module_param_array(index, int, NULL, 0444); module_param_array(index, int, NULL, 0444);
MODULE_PARM_DESC(index, "Index value for the EMU10K1 soundcard."); MODULE_PARM_DESC(index, "Index value for the EMU10K1 soundcard.");
@ -73,6 +74,8 @@ module_param_array(enable_ir, bool, NULL, 0444);
MODULE_PARM_DESC(enable_ir, "Enable IR."); MODULE_PARM_DESC(enable_ir, "Enable IR.");
module_param_array(subsystem, uint, NULL, 0444); module_param_array(subsystem, uint, NULL, 0444);
MODULE_PARM_DESC(subsystem, "Force card subsystem model."); MODULE_PARM_DESC(subsystem, "Force card subsystem model.");
module_param_array(delay_pcm_irq, uint, NULL, 0444);
MODULE_PARM_DESC(delay_pcm_irq, "Delay PCM interrupt by specified number of samples (default 0).");
/* /*
* Class 0401: 1102:0008 (rev 00) Subsystem: 1102:1001 -> Audigy2 Value Model:SB0400 * Class 0401: 1102:0008 (rev 00) Subsystem: 1102:1001 -> Audigy2 Value Model:SB0400
*/ */
@ -127,6 +130,7 @@ static int __devinit snd_card_emu10k1_probe(struct pci_dev *pci,
&emu)) < 0) &emu)) < 0)
goto error; goto error;
card->private_data = emu; card->private_data = emu;
emu->delay_pcm_irq = delay_pcm_irq[dev] & 0x1f;
if ((err = snd_emu10k1_pcm(emu, 0, NULL)) < 0) if ((err = snd_emu10k1_pcm(emu, 0, NULL)) < 0)
goto error; goto error;
if ((err = snd_emu10k1_pcm_mic(emu, 1, NULL)) < 0) if ((err = snd_emu10k1_pcm_mic(emu, 1, NULL)) < 0)

30
sound/pci/emu10k1/emupcm.c
View File

@ -332,7 +332,7 @@ static void snd_emu10k1_pcm_init_voice(struct snd_emu10k1 *emu,
evoice->epcm->ccca_start_addr = start_addr + ccis; evoice->epcm->ccca_start_addr = start_addr + ccis;
if (extra) { if (extra) {
start_addr += ccis; start_addr += ccis;
end_addr += ccis; end_addr += ccis + emu->delay_pcm_irq;
} }
if (stereo && !extra) { if (stereo && !extra) {
snd_emu10k1_ptr_write(emu, CPF, voice, CPF_STEREO_MASK); snd_emu10k1_ptr_write(emu, CPF, voice, CPF_STEREO_MASK);
@ -360,7 +360,9 @@ static void snd_emu10k1_pcm_init_voice(struct snd_emu10k1 *emu,
/* Assumption that PT is already 0 so no harm overwriting */ /* Assumption that PT is already 0 so no harm overwriting */
snd_emu10k1_ptr_write(emu, PTRX, voice, (send_amount[0] << 8) | send_amount[1]); snd_emu10k1_ptr_write(emu, PTRX, voice, (send_amount[0] << 8) | send_amount[1]);
snd_emu10k1_ptr_write(emu, DSL, voice, end_addr | (send_amount[3] << 24)); snd_emu10k1_ptr_write(emu, DSL, voice, end_addr | (send_amount[3] << 24));
snd_emu10k1_ptr_write(emu, PSST, voice, start_addr | (send_amount[2] << 24)); snd_emu10k1_ptr_write(emu, PSST, voice,
(start_addr + (extra ? emu->delay_pcm_irq : 0)) |
(send_amount[2] << 24));
if (emu->card_capabilities->emu_model) if (emu->card_capabilities->emu_model)
pitch_target = PITCH_48000; /* Disable interpolators on emu1010 card */ pitch_target = PITCH_48000; /* Disable interpolators on emu1010 card */
else else
@ -732,6 +734,23 @@ static void snd_emu10k1_playback_stop_voice(struct snd_emu10k1 *emu, struct snd_
snd_emu10k1_ptr_write(emu, IP, voice, 0); snd_emu10k1_ptr_write(emu, IP, voice, 0);
} }
static inline void snd_emu10k1_playback_mangle_extra(struct snd_emu10k1 *emu,
struct snd_emu10k1_pcm *epcm,
struct snd_pcm_substream *substream,
struct snd_pcm_runtime *runtime)
{
unsigned int ptr, period_pos;
/* try to sychronize the current position for the interrupt
source voice */
period_pos = runtime->status->hw_ptr - runtime->hw_ptr_interrupt;
period_pos %= runtime->period_size;
ptr = snd_emu10k1_ptr_read(emu, CCCA, epcm->extra->number);
ptr &= ~0x00ffffff;
ptr |= epcm->ccca_start_addr + period_pos;
snd_emu10k1_ptr_write(emu, CCCA, epcm->extra->number, ptr);
}
static int snd_emu10k1_playback_trigger(struct snd_pcm_substream *substream, static int snd_emu10k1_playback_trigger(struct snd_pcm_substream *substream,
int cmd) int cmd)
{ {
@ -753,6 +772,8 @@ static int snd_emu10k1_playback_trigger(struct snd_pcm_substream *substream,
/* follow thru */ /* follow thru */
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_RESUME:
if (cmd == SNDRV_PCM_TRIGGER_PAUSE_RELEASE)
snd_emu10k1_playback_mangle_extra(emu, epcm, substream, runtime);
mix = &emu->pcm_mixer[substream->number]; mix = &emu->pcm_mixer[substream->number];
snd_emu10k1_playback_prepare_voice(emu, epcm->voices[0], 1, 0, mix); snd_emu10k1_playback_prepare_voice(emu, epcm->voices[0], 1, 0, mix);
snd_emu10k1_playback_prepare_voice(emu, epcm->voices[1], 0, 0, mix); snd_emu10k1_playback_prepare_voice(emu, epcm->voices[1], 0, 0, mix);
@ -869,8 +890,9 @@ static snd_pcm_uframes_t snd_emu10k1_playback_pointer(struct snd_pcm_substream *
#endif #endif
/* /*
printk(KERN_DEBUG printk(KERN_DEBUG
"ptr = 0x%x, buffer_size = 0x%x, period_size = 0x%x\n", "ptr = 0x%lx, buffer_size = 0x%lx, period_size = 0x%lx\n",
ptr, runtime->buffer_size, runtime->period_size); (long)ptr, (long)runtime->buffer_size,
(long)runtime->period_size);
*/ */
return ptr; return ptr;
} }

4
sound/pci/emu10k1/memory.c
View File

@ -310,8 +310,10 @@ snd_emu10k1_alloc_pages(struct snd_emu10k1 *emu, struct snd_pcm_substream *subst
if (snd_BUG_ON(!hdr)) if (snd_BUG_ON(!hdr))
return NULL; return NULL;
idx = runtime->period_size >= runtime->buffer_size ?
(emu->delay_pcm_irq * 2) : 0;
mutex_lock(&hdr->block_mutex); mutex_lock(&hdr->block_mutex);
blk = search_empty(emu, runtime->dma_bytes); blk = search_empty(emu, runtime->dma_bytes + idx);
if (blk == NULL) { if (blk == NULL) {
mutex_unlock(&hdr->block_mutex); mutex_unlock(&hdr->block_mutex);
return NULL; return NULL;

1
sound/pci/hda/patch_conexant.c
View File

@ -3049,6 +3049,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = {
SND_PCI_QUIRK(0x1028, 0x02f5, "Dell", SND_PCI_QUIRK(0x1028, 0x02f5, "Dell",
CXT5066_DELL_LAPTOP), CXT5066_DELL_LAPTOP),
SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT5066_OLPC_XO_1_5), SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT5066_OLPC_XO_1_5),
SND_PCI_QUIRK(0x1028, 0x02d8, "Dell Vostro", CXT5066_DELL_VOSTO),
SND_PCI_QUIRK(0x1028, 0x0402, "Dell Vostro", CXT5066_DELL_VOSTO), SND_PCI_QUIRK(0x1028, 0x0402, "Dell Vostro", CXT5066_DELL_VOSTO),
SND_PCI_QUIRK(0x1028, 0x0408, "Dell Inspiron One 19T", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x1028, 0x0408, "Dell Inspiron One 19T", CXT5066_IDEAPAD),
SND_PCI_QUIRK(0x1179, 0xff50, "Toshiba Satellite P500-PSPGSC-01800T", CXT5066_OLPC_XO_1_5), SND_PCI_QUIRK(0x1179, 0xff50, "Toshiba Satellite P500-PSPGSC-01800T", CXT5066_OLPC_XO_1_5),

176
sound/pci/hda/patch_realtek.c
View File

@ -19030,6 +19030,7 @@ static int patch_alc888(struct hda_codec *codec)
/* /*
* ALC680 support * ALC680 support
*/ */
#define ALC680_DIGIN_NID ALC880_DIGIN_NID
#define ALC680_DIGOUT_NID ALC880_DIGOUT_NID #define ALC680_DIGOUT_NID ALC880_DIGOUT_NID
#define alc680_modes alc260_modes #define alc680_modes alc260_modes
@ -19044,23 +19045,93 @@ static hda_nid_t alc680_adc_nids[3] = {
0x07, 0x08, 0x09 0x07, 0x08, 0x09
}; };
/*
* Analog capture ADC cgange
*/
static int alc680_capture_pcm_prepare(struct hda_pcm_stream *hinfo,
struct hda_codec *codec,
unsigned int stream_tag,
unsigned int format,
struct snd_pcm_substream *substream)
{
struct alc_spec *spec = codec->spec;
struct auto_pin_cfg *cfg = &spec->autocfg;
unsigned int pre_mic, pre_line;
pre_mic = snd_hda_jack_detect(codec, cfg->input_pins[AUTO_PIN_MIC]);
pre_line = snd_hda_jack_detect(codec, cfg->input_pins[AUTO_PIN_LINE]);
spec->cur_adc_stream_tag = stream_tag;
spec->cur_adc_format = format;
if (pre_mic || pre_line) {
if (pre_mic)
snd_hda_codec_setup_stream(codec, 0x08, stream_tag, 0,
format);
else
snd_hda_codec_setup_stream(codec, 0x09, stream_tag, 0,
format);
} else
snd_hda_codec_setup_stream(codec, 0x07, stream_tag, 0, format);
return 0;
}
static int alc680_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
struct hda_codec *codec,
struct snd_pcm_substream *substream)
{
snd_hda_codec_cleanup_stream(codec, 0x07);
snd_hda_codec_cleanup_stream(codec, 0x08);
snd_hda_codec_cleanup_stream(codec, 0x09);
return 0;
}
static struct hda_pcm_stream alc680_pcm_analog_auto_capture = {
.substreams = 1, /* can be overridden */
.channels_min = 2,
.channels_max = 2,
/* NID is set in alc_build_pcms */
.ops = {
.prepare = alc680_capture_pcm_prepare,
.cleanup = alc680_capture_pcm_cleanup
},
};
static struct snd_kcontrol_new alc680_base_mixer[] = { static struct snd_kcontrol_new alc680_base_mixer[] = {
/* output mixer control */ /* output mixer control */
HDA_CODEC_VOLUME("Front Playback Volume", 0x2, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Front Playback Volume", 0x2, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Headphone Playback Volume", 0x4, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Headphone Playback Volume", 0x4, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x16, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x16, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Int Mic Boost", 0x12, 0, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
HDA_CODEC_VOLUME("Line In Boost", 0x19, 0, HDA_INPUT),
{ } { }
}; };
static struct snd_kcontrol_new alc680_capture_mixer[] = { static struct hda_bind_ctls alc680_bind_cap_vol = {
HDA_CODEC_VOLUME("Capture Volume", 0x07, 0x0, HDA_INPUT), .ops = &snd_hda_bind_vol,
HDA_CODEC_MUTE("Capture Switch", 0x07, 0x0, HDA_INPUT), .values = {
HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x08, 0x0, HDA_INPUT), HDA_COMPOSE_AMP_VAL(0x07, 3, 0, HDA_INPUT),
HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x08, 0x0, HDA_INPUT), HDA_COMPOSE_AMP_VAL(0x08, 3, 0, HDA_INPUT),
HDA_CODEC_VOLUME_IDX("Capture Volume", 2, 0x09, 0x0, HDA_INPUT), HDA_COMPOSE_AMP_VAL(0x09, 3, 0, HDA_INPUT),
HDA_CODEC_MUTE_IDX("Capture Switch", 2, 0x09, 0x0, HDA_INPUT), 0
},
};
static struct hda_bind_ctls alc680_bind_cap_switch = {
.ops = &snd_hda_bind_sw,
.values = {
HDA_COMPOSE_AMP_VAL(0x07, 3, 0, HDA_INPUT),
HDA_COMPOSE_AMP_VAL(0x08, 3, 0, HDA_INPUT),
HDA_COMPOSE_AMP_VAL(0x09, 3, 0, HDA_INPUT),
0
},
};
static struct snd_kcontrol_new alc680_master_capture_mixer[] = {
HDA_BIND_VOL("Capture Volume", &alc680_bind_cap_vol),
HDA_BIND_SW("Capture Switch", &alc680_bind_cap_switch),
{ } /* end */ { } /* end */
}; };
@ -19068,25 +19139,73 @@ static struct snd_kcontrol_new alc680_capture_mixer[] = {
* generic initialization of ADC, input mixers and output mixers * generic initialization of ADC, input mixers and output mixers
*/ */
static struct hda_verb alc680_init_verbs[] = { static struct hda_verb alc680_init_verbs[] = {
/* Unmute DAC0-1 and set vol = 0 */ {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0}, {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24}, {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20}, {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x16, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_MIC_EVENT | AC_USRSP_EN},
{ } { }
}; };
/* toggle speaker-output according to the hp-jack state */
static void alc680_base_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x16;
spec->autocfg.speaker_pins[0] = 0x14;
spec->autocfg.speaker_pins[1] = 0x15;
spec->autocfg.input_pins[AUTO_PIN_MIC] = 0x18;
spec->autocfg.input_pins[AUTO_PIN_LINE] = 0x19;
}
static void alc680_rec_autoswitch(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
struct auto_pin_cfg *cfg = &spec->autocfg;
unsigned int present;
hda_nid_t new_adc;
present = snd_hda_jack_detect(codec, cfg->input_pins[AUTO_PIN_MIC]);
new_adc = present ? 0x8 : 0x7;
__snd_hda_codec_cleanup_stream(codec, !present ? 0x8 : 0x7, 1);
snd_hda_codec_setup_stream(codec, new_adc,
spec->cur_adc_stream_tag, 0,
spec->cur_adc_format);
}
static void alc680_unsol_event(struct hda_codec *codec,
unsigned int res)
{
if ((res >> 26) == ALC880_HP_EVENT)
alc_automute_amp(codec);
if ((res >> 26) == ALC880_MIC_EVENT)
alc680_rec_autoswitch(codec);
}
static void alc680_inithook(struct hda_codec *codec)
{
alc_automute_amp(codec);
alc680_rec_autoswitch(codec);
}
/* create input playback/capture controls for the given pin */ /* create input playback/capture controls for the given pin */
static int alc680_new_analog_output(struct alc_spec *spec, hda_nid_t nid, static int alc680_new_analog_output(struct alc_spec *spec, hda_nid_t nid,
const char *ctlname, int idx) const char *ctlname, int idx)
@ -19197,13 +19316,7 @@ static void alc680_auto_init_hp_out(struct hda_codec *codec)
#define alc680_pcm_analog_capture alc880_pcm_analog_capture #define alc680_pcm_analog_capture alc880_pcm_analog_capture
#define alc680_pcm_analog_alt_capture alc880_pcm_analog_alt_capture #define alc680_pcm_analog_alt_capture alc880_pcm_analog_alt_capture
#define alc680_pcm_digital_playback alc880_pcm_digital_playback #define alc680_pcm_digital_playback alc880_pcm_digital_playback
#define alc680_pcm_digital_capture alc880_pcm_digital_capture
static struct hda_input_mux alc680_capture_source = {
.num_items = 1,
.items = {
{ "Mic", 0x0 },
},
};
/* /*
* BIOS auto configuration * BIOS auto configuration
@ -19218,6 +19331,7 @@ static int alc680_parse_auto_config(struct hda_codec *codec)
alc680_ignore); alc680_ignore);
if (err < 0) if (err < 0)
return err; return err;
if (!spec->autocfg.line_outs) { if (!spec->autocfg.line_outs) {
if (spec->autocfg.dig_outs || spec->autocfg.dig_in_pin) { if (spec->autocfg.dig_outs || spec->autocfg.dig_in_pin) {
spec->multiout.max_channels = 2; spec->multiout.max_channels = 2;
@ -19239,8 +19353,6 @@ static int alc680_parse_auto_config(struct hda_codec *codec)
add_mixer(spec, spec->kctls.list); add_mixer(spec, spec->kctls.list);
add_verb(spec, alc680_init_verbs); add_verb(spec, alc680_init_verbs);
spec->num_mux_defs = 1;
spec->input_mux = &alc680_capture_source;
err = alc_auto_add_mic_boost(codec); err = alc_auto_add_mic_boost(codec);
if (err < 0) if (err < 0)
@ -19279,17 +19391,17 @@ static struct snd_pci_quirk alc680_cfg_tbl[] = {
static struct alc_config_preset alc680_presets[] = { static struct alc_config_preset alc680_presets[] = {
[ALC680_BASE] = { [ALC680_BASE] = {
.mixers = { alc680_base_mixer }, .mixers = { alc680_base_mixer },
.cap_mixer = alc680_capture_mixer, .cap_mixer = alc680_master_capture_mixer,
.init_verbs = { alc680_init_verbs }, .init_verbs = { alc680_init_verbs },
.num_dacs = ARRAY_SIZE(alc680_dac_nids), .num_dacs = ARRAY_SIZE(alc680_dac_nids),
.dac_nids = alc680_dac_nids, .dac_nids = alc680_dac_nids,
.num_adc_nids = ARRAY_SIZE(alc680_adc_nids),
.adc_nids = alc680_adc_nids,
.hp_nid = 0x04,
.dig_out_nid = ALC680_DIGOUT_NID, .dig_out_nid = ALC680_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc680_modes), .num_channel_mode = ARRAY_SIZE(alc680_modes),
.channel_mode = alc680_modes, .channel_mode = alc680_modes,
.input_mux = &alc680_capture_source, .unsol_event = alc680_unsol_event,
.setup = alc680_base_setup,
.init_hook = alc680_inithook,
}, },
}; };
@ -19333,9 +19445,9 @@ static int patch_alc680(struct hda_codec *codec)
setup_preset(codec, &alc680_presets[board_config]); setup_preset(codec, &alc680_presets[board_config]);
spec->stream_analog_playback = &alc680_pcm_analog_playback; spec->stream_analog_playback = &alc680_pcm_analog_playback;
spec->stream_analog_capture = &alc680_pcm_analog_capture; spec->stream_analog_capture = &alc680_pcm_analog_auto_capture;
spec->stream_analog_alt_capture = &alc680_pcm_analog_alt_capture;
spec->stream_digital_playback = &alc680_pcm_digital_playback; spec->stream_digital_playback = &alc680_pcm_digital_playback;
spec->stream_digital_capture = &alc680_pcm_digital_capture;
if (!spec->adc_nids) { if (!spec->adc_nids) {
spec->adc_nids = alc680_adc_nids; spec->adc_nids = alc680_adc_nids;

11
sound/pci/riptide/riptide.c
View File

@ -1224,15 +1224,14 @@ static int try_to_load_firmware(struct cmdif *cif, struct snd_riptide *chip)
firmware.firmware.ASIC, firmware.firmware.CODEC, firmware.firmware.ASIC, firmware.firmware.CODEC,
firmware.firmware.AUXDSP, firmware.firmware.PROG); firmware.firmware.AUXDSP, firmware.firmware.PROG);
if (!chip)
return 1;
for (i = 0; i < FIRMWARE_VERSIONS; i++) { for (i = 0; i < FIRMWARE_VERSIONS; i++) {
if (!memcmp(&firmware_versions[i], &firmware, sizeof(firmware))) if (!memcmp(&firmware_versions[i], &firmware, sizeof(firmware)))
break; return 1; /* OK */
}
if (i >= FIRMWARE_VERSIONS)
return 0; /* no match */
if (!chip) }
return 1; /* OK */
snd_printdd("Writing Firmware\n"); snd_printdd("Writing Firmware\n");
if (!chip->fw_entry) { if (!chip->fw_entry) {

7
sound/soc/codecs/wm8776.c
View File

@ -178,13 +178,6 @@ static int wm8776_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
case SND_SOC_DAIFMT_LEFT_J: case SND_SOC_DAIFMT_LEFT_J:
iface |= 0x0001; iface |= 0x0001;
break; break;
/* FIXME: CHECK A/B */
case SND_SOC_DAIFMT_DSP_A:
iface |= 0x0003;
break;
case SND_SOC_DAIFMT_DSP_B:
iface |= 0x0007;
break;
default: default:
return -EINVAL; return -EINVAL;
} }