diff options
author | Takashi Iwai <tiwai@suse.de> | 2010-09-03 22:38:52 +0200 |
---|---|---|
committer | Takashi Iwai <tiwai@suse.de> | 2010-09-03 22:38:52 +0200 |
commit | 68885a3ff38ed51fa02f241feb405c9922a90ee0 (patch) | |
tree | 2fc626df39d5e0e1f6b065238141f7d49187c737 /sound | |
parent | 7b28079b3284ccb15ad4f003fb7073890600d0c1 (diff) | |
parent | a2acad8298a42b7be684a32fafaf83332bba9c2b (diff) |
Merge branch 'fix/misc' into topic/misc
Diffstat (limited to 'sound')
-rw-r--r-- | sound/core/pcm.c | 6 | ||||
-rw-r--r-- | sound/oss/sound_timer.c | 2 | ||||
-rw-r--r-- | sound/pci/asihpi/hpi6205.c | 7 | ||||
-rw-r--r-- | sound/pci/hda/hda_codec.c | 33 | ||||
-rw-r--r-- | sound/pci/hda/hda_codec.h | 2 | ||||
-rw-r--r-- | sound/pci/hda/hda_eld.c | 4 | ||||
-rw-r--r-- | sound/pci/hda/patch_conexant.c | 2 | ||||
-rw-r--r-- | sound/pci/hda/patch_hdmi.c | 21 | ||||
-rw-r--r-- | sound/pci/hda/patch_intelhdmi.c | 8 | ||||
-rw-r--r-- | sound/pci/hda/patch_nvhdmi.c | 8 | ||||
-rw-r--r-- | sound/pci/hda/patch_realtek.c | 177 | ||||
-rw-r--r-- | sound/pci/hda/patch_sigmatel.c | 15 | ||||
-rw-r--r-- | sound/pci/intel8x0.c | 6 | ||||
-rw-r--r-- | sound/soc/codecs/wm8776.c | 7 | ||||
-rw-r--r-- | sound/soc/imx/imx-ssi.c | 3 | ||||
-rw-r--r-- | sound/soc/soc-core.c | 2 | ||||
-rw-r--r-- | sound/usb/card.c | 17 | ||||
-rw-r--r-- | sound/usb/clock.c | 3 | ||||
-rw-r--r-- | sound/usb/endpoint.c | 11 | ||||
-rw-r--r-- | sound/usb/format.c | 22 | ||||
-rw-r--r-- | sound/usb/mixer.c | 10 | ||||
-rw-r--r-- | sound/usb/pcm.c | 3 |
22 files changed, 247 insertions, 122 deletions
diff --git a/sound/core/pcm.c b/sound/core/pcm.c index cbe815dfbdc8..204af48c5cc1 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -203,10 +203,16 @@ static char *snd_pcm_format_names[] = { FORMAT(S18_3BE), FORMAT(U18_3LE), FORMAT(U18_3BE), + FORMAT(G723_24), + FORMAT(G723_24_1B), + FORMAT(G723_40), + FORMAT(G723_40_1B), }; const char *snd_pcm_format_name(snd_pcm_format_t format) { + if (format >= ARRAY_SIZE(snd_pcm_format_names)) + return "Unknown"; return snd_pcm_format_names[format]; } EXPORT_SYMBOL_GPL(snd_pcm_format_name); diff --git a/sound/oss/sound_timer.c b/sound/oss/sound_timer.c index f0f0c19fbff7..48cda6c4c257 100644 --- a/sound/oss/sound_timer.c +++ b/sound/oss/sound_timer.c @@ -26,7 +26,7 @@ static unsigned long prev_event_time; static volatile unsigned long usecs_per_tmr; /* Length of the current interval */ static struct sound_lowlev_timer *tmr; -static spinlock_t lock; +static DEFINE_SPINLOCK(lock); static unsigned long tmr2ticks(int tmr_value) { diff --git a/sound/pci/asihpi/hpi6205.c b/sound/pci/asihpi/hpi6205.c index 3b4413448226..22c5fc625533 100644 --- a/sound/pci/asihpi/hpi6205.c +++ b/sound/pci/asihpi/hpi6205.c @@ -941,8 +941,7 @@ static void outstream_host_buffer_free(struct hpi_adapter_obj *pao, } -static u32 outstream_get_space_available(struct hpi_hostbuffer_status - *status) +static u32 outstream_get_space_available(struct hpi_hostbuffer_status *status) { return status->size_in_bytes - (status->host_index - status->dSP_index); @@ -987,6 +986,10 @@ static void outstream_write(struct hpi_adapter_obj *pao, /* write it */ phm->function = HPI_OSTREAM_WRITE; hw_message(pao, phm, phr); + + if (phr->error) + return; + /* update status information that the DSP would typically * update (and will update next time the DSP * buffer update task reads data from the host BBM buffer) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index dd8fb86c842b..3827092cc1d2 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -589,6 +589,7 @@ int /*__devinit*/ snd_hda_bus_new(struct snd_card *card, bus->ops = temp->ops; mutex_init(&bus->cmd_mutex); + mutex_init(&bus->prepare_mutex); INIT_LIST_HEAD(&bus->codec_list); snprintf(bus->workq_name, sizeof(bus->workq_name), @@ -1068,7 +1069,6 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, codec->addr = codec_addr; mutex_init(&codec->spdif_mutex); mutex_init(&codec->control_mutex); - mutex_init(&codec->prepare_mutex); init_hda_cache(&codec->amp_cache, sizeof(struct hda_amp_info)); init_hda_cache(&codec->cmd_cache, sizeof(struct hda_cache_head)); snd_array_init(&codec->mixers, sizeof(struct hda_nid_item), 32); @@ -1213,6 +1213,7 @@ void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid, u32 stream_tag, int channel_id, int format) { + struct hda_codec *c; struct hda_cvt_setup *p; unsigned int oldval, newval; int i; @@ -1253,10 +1254,12 @@ void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid, p->dirty = 0; /* make other inactive cvts with the same stream-tag dirty */ - for (i = 0; i < codec->cvt_setups.used; i++) { - p = snd_array_elem(&codec->cvt_setups, i); - if (!p->active && p->stream_tag == stream_tag) - p->dirty = 1; + list_for_each_entry(c, &codec->bus->codec_list, list) { + 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) + p->dirty = 1; + } } } EXPORT_SYMBOL_HDA(snd_hda_codec_setup_stream); @@ -1306,12 +1309,16 @@ static void really_cleanup_stream(struct hda_codec *codec, /* clean up the all conflicting obsolete streams */ static void purify_inactive_streams(struct hda_codec *codec) { + struct hda_codec *c; int i; - for (i = 0; i < codec->cvt_setups.used; i++) { - struct hda_cvt_setup *p = snd_array_elem(&codec->cvt_setups, i); - if (p->dirty) - really_cleanup_stream(codec, p); + list_for_each_entry(c, &codec->bus->codec_list, list) { + for (i = 0; i < c->cvt_setups.used; i++) { + struct hda_cvt_setup *p; + p = snd_array_elem(&c->cvt_setups, i); + if (p->dirty) + really_cleanup_stream(c, p); + } } } @@ -3502,11 +3509,11 @@ int snd_hda_codec_prepare(struct hda_codec *codec, struct snd_pcm_substream *substream) { int ret; - mutex_lock(&codec->prepare_mutex); + mutex_lock(&codec->bus->prepare_mutex); ret = hinfo->ops.prepare(hinfo, codec, stream, format, substream); if (ret >= 0) purify_inactive_streams(codec); - mutex_unlock(&codec->prepare_mutex); + mutex_unlock(&codec->bus->prepare_mutex); return ret; } EXPORT_SYMBOL_HDA(snd_hda_codec_prepare); @@ -3515,9 +3522,9 @@ void snd_hda_codec_cleanup(struct hda_codec *codec, struct hda_pcm_stream *hinfo, struct snd_pcm_substream *substream) { - mutex_lock(&codec->prepare_mutex); + mutex_lock(&codec->bus->prepare_mutex); hinfo->ops.cleanup(hinfo, codec, substream); - mutex_unlock(&codec->prepare_mutex); + mutex_unlock(&codec->bus->prepare_mutex); } EXPORT_SYMBOL_HDA(snd_hda_codec_cleanup); diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 4303353feda9..62c702240108 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -648,6 +648,7 @@ struct hda_bus { struct hda_codec *caddr_tbl[HDA_MAX_CODEC_ADDRESS + 1]; struct mutex cmd_mutex; + struct mutex prepare_mutex; /* unsolicited event queue */ struct hda_bus_unsolicited *unsol; @@ -826,7 +827,6 @@ struct hda_codec { struct mutex spdif_mutex; struct mutex control_mutex; - struct mutex prepare_mutex; unsigned int spdif_status; /* IEC958 status bits */ unsigned short spdif_ctls; /* SPDIF control bits */ unsigned int spdif_in_enable; /* SPDIF input enable? */ diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c index 803b298f7411..26c3ade73583 100644 --- a/sound/pci/hda/hda_eld.c +++ b/sound/pci/hda/hda_eld.c @@ -596,6 +596,8 @@ void snd_hda_eld_proc_free(struct hda_codec *codec, struct hdmi_eld *eld) } EXPORT_SYMBOL_HDA(snd_hda_eld_proc_free); +#endif /* CONFIG_PROC_FS */ + /* update PCM info based on ELD */ void hdmi_eld_update_pcm_info(struct hdmi_eld *eld, struct hda_pcm_stream *pcm, struct hda_pcm_stream *codec_pars) @@ -644,5 +646,3 @@ void hdmi_eld_update_pcm_info(struct hdmi_eld *eld, struct hda_pcm_stream *pcm, pcm->maxbps = min(pcm->maxbps, codec_pars->maxbps); } EXPORT_SYMBOL_HDA(hdmi_eld_update_pcm_info); - -#endif /* CONFIG_PROC_FS */ diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 31b5d9eeba68..5cdb80edbd7f 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3049,6 +3049,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x1028, 0x02f5, "Dell", CXT5066_DELL_LAPTOP), 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, 0x0408, "Dell Inspiron One 19T", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x1179, 0xff50, "Toshiba Satellite P500-PSPGSC-01800T", CXT5066_OLPC_XO_1_5), @@ -3058,6 +3059,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x21b4, "Thinkpad Edge", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo Thinkpad", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G series", CXT5066_IDEAPAD), + SND_PCI_QUIRK(0x17aa, 0x390a, "Lenovo S10-3t", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x3938, "Lenovo G series (AMD)", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x3a0d, "ideapad", CXT5066_IDEAPAD), {} diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 2bc0f07cf33f..afd6022a96a7 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -707,8 +707,6 @@ static int hdmi_setup_stream(struct hda_codec *codec, hda_nid_t nid, u32 stream_tag, int format) { struct hdmi_spec *spec = codec->spec; - int tag; - int fmt; int pinctl; int new_pinctl = 0; int i; @@ -745,24 +743,7 @@ static int hdmi_setup_stream(struct hda_codec *codec, hda_nid_t nid, return -EINVAL; } - tag = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONV, 0) >> 4; - fmt = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_STREAM_FORMAT, 0); - - snd_printdd("hdmi_setup_stream: " - "NID=0x%x, %sstream=0x%x, %sformat=0x%x\n", - nid, - tag == stream_tag ? "" : "new-", - stream_tag, - fmt == format ? "" : "new-", - format); - - if (tag != stream_tag) - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_CHANNEL_STREAMID, - stream_tag << 4); - if (fmt != format) - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_STREAM_FORMAT, format); + snd_hda_codec_setup_stream(codec, nid, stream_tag, 0, format); return 0; } diff --git a/sound/pci/hda/patch_intelhdmi.c b/sound/pci/hda/patch_intelhdmi.c index d382d3c81c0f..36a9b83a6174 100644 --- a/sound/pci/hda/patch_intelhdmi.c +++ b/sound/pci/hda/patch_intelhdmi.c @@ -69,20 +69,12 @@ static int intel_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, return hdmi_setup_stream(codec, hinfo->nid, stream_tag, format); } -static int intel_hdmi_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - return 0; -} - static struct hda_pcm_stream intel_hdmi_pcm_playback = { .substreams = 1, .channels_min = 2, .ops = { .open = hdmi_pcm_open, .prepare = intel_hdmi_playback_pcm_prepare, - .cleanup = intel_hdmi_playback_pcm_cleanup, }, }; diff --git a/sound/pci/hda/patch_nvhdmi.c b/sound/pci/hda/patch_nvhdmi.c index f636870dc718..69b950d527c3 100644 --- a/sound/pci/hda/patch_nvhdmi.c +++ b/sound/pci/hda/patch_nvhdmi.c @@ -326,13 +326,6 @@ static int nvhdmi_dig_playback_pcm_prepare_8ch(struct hda_pcm_stream *hinfo, return 0; } -static int nvhdmi_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - return 0; -} - static int nvhdmi_dig_playback_pcm_prepare_2ch(struct hda_pcm_stream *hinfo, struct hda_codec *codec, unsigned int stream_tag, @@ -350,7 +343,6 @@ static struct hda_pcm_stream nvhdmi_pcm_digital_playback_8ch_89 = { .ops = { .open = hdmi_pcm_open, .prepare = nvhdmi_dig_playback_pcm_prepare_8ch_89, - .cleanup = nvhdmi_playback_pcm_cleanup, }, }; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 2cd1ae809e46..627bf9963368 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -14467,6 +14467,7 @@ static const struct alc_fixup alc269_fixups[] = { static struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x104d, 0x9071, "Sony VAIO", ALC269_FIXUP_SONY_VAIO), + SND_PCI_QUIRK(0x104d, 0x9077, "Sony VAIO", ALC269_FIXUP_SONY_VAIO), {} }; @@ -19030,6 +19031,7 @@ static int patch_alc888(struct hda_codec *codec) /* * ALC680 support */ +#define ALC680_DIGIN_NID ALC880_DIGIN_NID #define ALC680_DIGOUT_NID ALC880_DIGOUT_NID #define alc680_modes alc260_modes @@ -19044,23 +19046,93 @@ static hda_nid_t alc680_adc_nids[3] = { 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[] = { /* output mixer control */ HDA_CODEC_VOLUME("Front Playback Volume", 0x2, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Headphone Playback Volume", 0x4, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x16, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Int Mic Boost", 0x12, 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[] = { - HDA_CODEC_VOLUME("Capture Volume", 0x07, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x07, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x08, 0x0, HDA_INPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x08, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME_IDX("Capture Volume", 2, 0x09, 0x0, HDA_INPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 2, 0x09, 0x0, HDA_INPUT), +static struct hda_bind_ctls alc680_bind_cap_vol = { + .ops = &snd_hda_bind_vol, + .values = { + HDA_COMPOSE_AMP_VAL(0x07, 3, 0, HDA_INPUT), + HDA_COMPOSE_AMP_VAL(0x08, 3, 0, HDA_INPUT), + HDA_COMPOSE_AMP_VAL(0x09, 3, 0, HDA_INPUT), + 0 + }, +}; + +static 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 */ }; @@ -19068,25 +19140,73 @@ static struct snd_kcontrol_new alc680_capture_mixer[] = { * generic initialization of ADC, input mixers and output mixers */ static struct hda_verb alc680_init_verbs[] = { - /* Unmute DAC0-1 and set vol = 0 */ - {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, - {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0}, - {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24}, - {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20}, + {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + + {0x16, AC_VERB_SET_UNSOLICITED_ENABLE, 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 */ static int alc680_new_analog_output(struct alc_spec *spec, hda_nid_t nid, const char *ctlname, int idx) @@ -19197,13 +19317,7 @@ static void alc680_auto_init_hp_out(struct hda_codec *codec) #define alc680_pcm_analog_capture alc880_pcm_analog_capture #define alc680_pcm_analog_alt_capture alc880_pcm_analog_alt_capture #define alc680_pcm_digital_playback alc880_pcm_digital_playback - -static struct hda_input_mux alc680_capture_source = { - .num_items = 1, - .items = { - { "Mic", 0x0 }, - }, -}; +#define alc680_pcm_digital_capture alc880_pcm_digital_capture /* * BIOS auto configuration @@ -19218,6 +19332,7 @@ static int alc680_parse_auto_config(struct hda_codec *codec) alc680_ignore); if (err < 0) return err; + if (!spec->autocfg.line_outs) { if (spec->autocfg.dig_outs || spec->autocfg.dig_in_pin) { spec->multiout.max_channels = 2; @@ -19239,8 +19354,6 @@ static int alc680_parse_auto_config(struct hda_codec *codec) add_mixer(spec, spec->kctls.list); add_verb(spec, alc680_init_verbs); - spec->num_mux_defs = 1; - spec->input_mux = &alc680_capture_source; err = alc_auto_add_mic_boost(codec); if (err < 0) @@ -19279,17 +19392,17 @@ static struct snd_pci_quirk alc680_cfg_tbl[] = { static struct alc_config_preset alc680_presets[] = { [ALC680_BASE] = { .mixers = { alc680_base_mixer }, - .cap_mixer = alc680_capture_mixer, + .cap_mixer = alc680_master_capture_mixer, .init_verbs = { alc680_init_verbs }, .num_dacs = ARRAY_SIZE(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, .num_channel_mode = ARRAY_SIZE(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 +19446,9 @@ static int patch_alc680(struct hda_codec *codec) setup_preset(codec, &alc680_presets[board_config]); spec->stream_analog_playback = &alc680_pcm_analog_playback; - spec->stream_analog_capture = &alc680_pcm_analog_capture; - spec->stream_analog_alt_capture = &alc680_pcm_analog_alt_capture; + spec->stream_analog_capture = &alc680_pcm_analog_auto_capture; spec->stream_digital_playback = &alc680_pcm_digital_playback; + spec->stream_digital_capture = &alc680_pcm_digital_capture; if (!spec->adc_nids) { spec->adc_nids = alc680_adc_nids; diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index f3f861bd1bf8..95148e58026c 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -6303,6 +6303,21 @@ static struct hda_codec_preset snd_hda_preset_sigmatel[] = { { .id = 0x111d76b5, .name = "92HD71B6X", .patch = patch_stac92hd71bxx }, { .id = 0x111d76b6, .name = "92HD71B5X", .patch = patch_stac92hd71bxx }, { .id = 0x111d76b7, .name = "92HD71B5X", .patch = patch_stac92hd71bxx }, + { .id = 0x111d76c0, .name = "92HD89C3", .patch = patch_stac92hd73xx }, + { .id = 0x111d76c1, .name = "92HD89C2", .patch = patch_stac92hd73xx }, + { .id = 0x111d76c2, .name = "92HD89C1", .patch = patch_stac92hd73xx }, + { .id = 0x111d76c3, .name = "92HD89B3", .patch = patch_stac92hd73xx }, + { .id = 0x111d76c4, .name = "92HD89B2", .patch = patch_stac92hd73xx }, + { .id = 0x111d76c5, .name = "92HD89B1", .patch = patch_stac92hd73xx }, + { .id = 0x111d76c6, .name = "92HD89E3", .patch = patch_stac92hd73xx }, + { .id = 0x111d76c7, .name = "92HD89E2", .patch = patch_stac92hd73xx }, + { .id = 0x111d76c8, .name = "92HD89E1", .patch = patch_stac92hd73xx }, + { .id = 0x111d76c9, .name = "92HD89D3", .patch = patch_stac92hd73xx }, + { .id = 0x111d76ca, .name = "92HD89D2", .patch = patch_stac92hd73xx }, + { .id = 0x111d76cb, .name = "92HD89D1", .patch = patch_stac92hd73xx }, + { .id = 0x111d76cc, .name = "92HD89F3", .patch = patch_stac92hd73xx }, + { .id = 0x111d76cd, .name = "92HD89F2", .patch = patch_stac92hd73xx }, + { .id = 0x111d76ce, .name = "92HD89F1", .patch = patch_stac92hd73xx }, {} /* terminator */ }; diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index 6433e65c9507..467749249576 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -1776,6 +1776,12 @@ static struct ac97_quirk ac97_quirks[] __devinitdata = { }, { .subvendor = 0x1014, + .subdevice = 0x0534, + .name = "ThinkPad X31", + .type = AC97_TUNE_INV_EAPD + }, + { + .subvendor = 0x1014, .subdevice = 0x1f00, .name = "MS-9128", .type = AC97_TUNE_ALC_JACK diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c index 4e212ed62ea6..f8154e661524 100644 --- a/sound/soc/codecs/wm8776.c +++ b/sound/soc/codecs/wm8776.c @@ -178,13 +178,6 @@ static int wm8776_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) case SND_SOC_DAIFMT_LEFT_J: iface |= 0x0001; break; - /* FIXME: CHECK A/B */ - case SND_SOC_DAIFMT_DSP_A: - iface |= 0x0003; - break; - case SND_SOC_DAIFMT_DSP_B: - iface |= 0x0007; - break; default: return -EINVAL; } diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c index a11daa1e905b..c81da05a4f11 100644 --- a/sound/soc/imx/imx-ssi.c +++ b/sound/soc/imx/imx-ssi.c @@ -254,6 +254,9 @@ static int imx_ssi_hw_params(struct snd_pcm_substream *substream, dma_data = &ssi->dma_params_rx; } + if (ssi->flags & IMX_SSI_SYN) + reg = SSI_STCCR; + snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data); sccr = readl(ssi->base + reg) & ~SSI_STCCR_WL_MASK; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 844ae8221a3a..acc91daa1c55 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -251,7 +251,7 @@ static void soc_init_codec_debugfs(struct snd_soc_codec *codec) printk(KERN_WARNING "ASoC: Failed to create codec register debugfs file\n"); - codec->debugfs_pop_time = debugfs_create_u32("dapm_pop_time", 0744, + codec->debugfs_pop_time = debugfs_create_u32("dapm_pop_time", 0644, codec->debugfs_codec_root, &codec->pop_time); if (!codec->debugfs_pop_time) diff --git a/sound/usb/card.c b/sound/usb/card.c index 498a2d8fa4bb..4aa4678e0a01 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -216,6 +216,11 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif) } switch (protocol) { + default: + snd_printdd(KERN_WARNING "unknown interface protocol %#02x, assuming v1\n", + protocol); + /* fall through */ + case UAC_VERSION_1: { struct uac1_ac_header_descriptor *h1 = control_header; @@ -253,10 +258,6 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif) break; } - - default: - snd_printk(KERN_ERR "unknown protocol version 0x%02x\n", protocol); - return -EINVAL; } return 0; @@ -480,7 +481,13 @@ static void *snd_usb_audio_probe(struct usb_device *dev, goto __error; } - chip->ctrl_intf = alts; + /* + * For devices with more than one control interface, we assume the + * first contains the audio controls. We might need a more specific + * check here in the future. + */ + if (!chip->ctrl_intf) + chip->ctrl_intf = alts; if (err > 0) { /* create normal USB audio interfaces */ diff --git a/sound/usb/clock.c b/sound/usb/clock.c index b853f8df794f..7754a1034545 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -295,12 +295,11 @@ int snd_usb_init_sample_rate(struct snd_usb_audio *chip, int iface, switch (altsd->bInterfaceProtocol) { case UAC_VERSION_1: + default: return set_sample_rate_v1(chip, iface, alts, fmt, rate); case UAC_VERSION_2: return set_sample_rate_v2(chip, iface, alts, fmt, rate); } - - return -EINVAL; } diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index bb9f938558fd..b0ef9f501896 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -275,6 +275,12 @@ int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) /* get audio formats */ switch (protocol) { + default: + snd_printdd(KERN_WARNING "%d:%u:%d: unknown interface protocol %#02x, assuming v1\n", + dev->devnum, iface_no, altno, protocol); + protocol = UAC_VERSION_1; + /* fall through */ + case UAC_VERSION_1: { struct uac1_as_header_descriptor *as = snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, UAC_AS_GENERAL); @@ -336,11 +342,6 @@ int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) dev->devnum, iface_no, altno, as->bTerminalLink); continue; } - - default: - snd_printk(KERN_ERR "%d:%u:%d : unknown interface protocol %04x\n", - dev->devnum, iface_no, altno, protocol); - continue; } /* get format type */ diff --git a/sound/usb/format.c b/sound/usb/format.c index 3a1375459c06..69148212aa70 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -49,7 +49,8 @@ static u64 parse_audio_format_i_type(struct snd_usb_audio *chip, u64 pcm_formats; switch (protocol) { - case UAC_VERSION_1: { + case UAC_VERSION_1: + default: { struct uac_format_type_i_discrete_descriptor *fmt = _fmt; sample_width = fmt->bBitResolution; sample_bytes = fmt->bSubframeSize; @@ -64,9 +65,6 @@ static u64 parse_audio_format_i_type(struct snd_usb_audio *chip, format <<= 1; break; } - - default: - return -EINVAL; } pcm_formats = 0; @@ -384,6 +382,10 @@ static int parse_audio_format_i(struct snd_usb_audio *chip, * audio class v2 uses class specific EP0 range requests for that. */ switch (protocol) { + default: + snd_printdd(KERN_WARNING "%d:%u:%d : invalid protocol version %d, assuming v1\n", + chip->dev->devnum, fp->iface, fp->altsetting, protocol); + /* fall through */ case UAC_VERSION_1: fp->channels = fmt->bNrChannels; ret = parse_audio_format_rates_v1(chip, fp, (unsigned char *) fmt, 7); @@ -392,10 +394,6 @@ static int parse_audio_format_i(struct snd_usb_audio *chip, /* fp->channels is already set in this case */ ret = parse_audio_format_rates_v2(chip, fp); break; - default: - snd_printk(KERN_ERR "%d:%u:%d : invalid protocol version %d\n", - chip->dev->devnum, fp->iface, fp->altsetting, protocol); - return -EINVAL; } if (fp->channels < 1) { @@ -438,6 +436,10 @@ static int parse_audio_format_ii(struct snd_usb_audio *chip, fp->channels = 1; switch (protocol) { + default: + snd_printdd(KERN_WARNING "%d:%u:%d : invalid protocol version %d, assuming v1\n", + chip->dev->devnum, fp->iface, fp->altsetting, protocol); + /* fall through */ case UAC_VERSION_1: { struct uac_format_type_ii_discrete_descriptor *fmt = _fmt; brate = le16_to_cpu(fmt->wMaxBitRate); @@ -456,10 +458,6 @@ static int parse_audio_format_ii(struct snd_usb_audio *chip, ret = parse_audio_format_rates_v2(chip, fp); break; } - default: - snd_printk(KERN_ERR "%d:%u:%d : invalid protocol version %d\n", - chip->dev->devnum, fp->iface, fp->altsetting, protocol); - return -EINVAL; } return ret; diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index c166db0057d3..3ed3901369ce 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -2175,7 +2175,15 @@ int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif, } host_iface = &usb_ifnum_to_if(chip->dev, ctrlif)->altsetting[0]; - mixer->protocol = get_iface_desc(host_iface)->bInterfaceProtocol; + switch (get_iface_desc(host_iface)->bInterfaceProtocol) { + case UAC_VERSION_1: + default: + mixer->protocol = UAC_VERSION_1; + break; + case UAC_VERSION_2: + mixer->protocol = UAC_VERSION_2; + break; + } if ((err = snd_usb_mixer_controls(mixer)) < 0 || (err = snd_usb_mixer_status_create(mixer)) < 0) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 58ed6820a8cf..f49756c1b837 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -173,13 +173,12 @@ int snd_usb_init_pitch(struct snd_usb_audio *chip, int iface, switch (altsd->bInterfaceProtocol) { case UAC_VERSION_1: + default: return init_pitch_v1(chip, iface, alts, fmt); case UAC_VERSION_2: return init_pitch_v2(chip, iface, alts, fmt); } - - return -EINVAL; } /* |