From 18f98ab54735f66ea84bf679b70fcec5e8b3df66 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 14 Dec 2009 13:22:04 +0900 Subject: ASoC: fsi-ak4642: Remove ak4642_add_i2c_device I2C devices should be registered when platform board setting in latest ASoC. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/fsi-ak4642.c | 30 ------------------------------ 1 file changed, 30 deletions(-) (limited to 'sound') diff --git a/sound/soc/sh/fsi-ak4642.c b/sound/soc/sh/fsi-ak4642.c index c7af09729c6e..5263ab18f827 100644 --- a/sound/soc/sh/fsi-ak4642.c +++ b/sound/soc/sh/fsi-ak4642.c @@ -42,42 +42,12 @@ static struct snd_soc_device fsi_snd_devdata = { .codec_dev = &soc_codec_dev_ak4642, }; -#define AK4642_BUS 0 -#define AK4642_ADR 0x12 -static int ak4642_add_i2c_device(void) -{ - struct i2c_board_info info; - struct i2c_adapter *adapter; - struct i2c_client *client; - - memset(&info, 0, sizeof(struct i2c_board_info)); - info.addr = AK4642_ADR; - strlcpy(info.type, "ak4642", I2C_NAME_SIZE); - - adapter = i2c_get_adapter(AK4642_BUS); - if (!adapter) { - printk(KERN_DEBUG "can't get i2c adapter\n"); - return -ENODEV; - } - - client = i2c_new_device(adapter, &info); - i2c_put_adapter(adapter); - if (!client) { - printk(KERN_DEBUG "can't add i2c device\n"); - return -ENODEV; - } - - return 0; -} - static struct platform_device *fsi_snd_device; static int __init fsi_ak4642_init(void) { int ret = -ENOMEM; - ak4642_add_i2c_device(); - fsi_snd_device = platform_device_alloc("soc-audio", -1); if (!fsi_snd_device) goto out; -- cgit v1.2.2 From b3172f222ab5afdc91ea058bd11c42cf169728f3 Mon Sep 17 00:00:00 2001 From: Guennadi Liakhovetski Date: Thu, 24 Dec 2009 01:13:51 +0100 Subject: ASoC: fix params_rate() macro use in several codecs Sevelar ASoC codec drivers wrongly assume, that the params_rate() macro returns one of SNDRV_PCM_RATE_* defines instead of the actual numerical sampling rate. Fix them. Signed-off-by: Guennadi Liakhovetski Acked-by: Liam Girdwood Signed-off-by: Mark Brown Cc: stable@kernel.org --- sound/soc/codecs/wm8510.c | 14 +++++++------- sound/soc/codecs/wm8940.c | 14 +++++++------- sound/soc/codecs/wm8974.c | 14 +++++++------- 3 files changed, 21 insertions(+), 21 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index 265e68c75df8..af8cb6995a1f 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -424,23 +424,23 @@ static int wm8510_pcm_hw_params(struct snd_pcm_substream *substream, /* filter coefficient */ switch (params_rate(params)) { - case SNDRV_PCM_RATE_8000: + case 8000: adn |= 0x5 << 1; break; - case SNDRV_PCM_RATE_11025: + case 11025: adn |= 0x4 << 1; break; - case SNDRV_PCM_RATE_16000: + case 16000: adn |= 0x3 << 1; break; - case SNDRV_PCM_RATE_22050: + case 22050: adn |= 0x2 << 1; break; - case SNDRV_PCM_RATE_32000: + case 32000: adn |= 0x1 << 1; break; - case SNDRV_PCM_RATE_44100: - case SNDRV_PCM_RATE_48000: + case 44100: + case 48000: break; } diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index 3d850b97037a..31e39ffd1d8e 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -378,23 +378,23 @@ static int wm8940_i2s_hw_params(struct snd_pcm_substream *substream, iface |= (1 << 9); switch (params_rate(params)) { - case SNDRV_PCM_RATE_8000: + case 8000: addcntrl |= (0x5 << 1); break; - case SNDRV_PCM_RATE_11025: + case 11025: addcntrl |= (0x4 << 1); break; - case SNDRV_PCM_RATE_16000: + case 16000: addcntrl |= (0x3 << 1); break; - case SNDRV_PCM_RATE_22050: + case 22050: addcntrl |= (0x2 << 1); break; - case SNDRV_PCM_RATE_32000: + case 32000: addcntrl |= (0x1 << 1); break; - case SNDRV_PCM_RATE_44100: - case SNDRV_PCM_RATE_48000: + case 44100: + case 48000: break; } ret = snd_soc_write(codec, WM8940_ADDCNTRL, addcntrl); diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index a808675388fc..8812751da8c9 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -482,23 +482,23 @@ static int wm8974_pcm_hw_params(struct snd_pcm_substream *substream, /* filter coefficient */ switch (params_rate(params)) { - case SNDRV_PCM_RATE_8000: + case 8000: adn |= 0x5 << 1; break; - case SNDRV_PCM_RATE_11025: + case 11025: adn |= 0x4 << 1; break; - case SNDRV_PCM_RATE_16000: + case 16000: adn |= 0x3 << 1; break; - case SNDRV_PCM_RATE_22050: + case 22050: adn |= 0x2 << 1; break; - case SNDRV_PCM_RATE_32000: + case 32000: adn |= 0x1 << 1; break; - case SNDRV_PCM_RATE_44100: - case SNDRV_PCM_RATE_48000: + case 44100: + case 48000: break; } -- cgit v1.2.2 From 8b90ca08821fee79e181bfcbc3bbd41ef5637136 Mon Sep 17 00:00:00 2001 From: Guennadi Liakhovetski Date: Thu, 24 Dec 2009 01:17:46 +0100 Subject: ALSA: Fix indentation in pcm_native.c Signed-off-by: Guennadi Liakhovetski Signed-off-by: Takashi Iwai --- sound/core/pcm_native.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 29ab46a12e11..25b0641e6b8c 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1918,13 +1918,13 @@ int snd_pcm_hw_constraints_complete(struct snd_pcm_substream *substream) err = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_RATE, hw->rate_min, hw->rate_max); - if (err < 0) - return err; + if (err < 0) + return err; err = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, hw->period_bytes_min, hw->period_bytes_max); - if (err < 0) - return err; + if (err < 0) + return err; err = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIODS, hw->periods_min, hw->periods_max); -- cgit v1.2.2 From ef18beded8ddbaafdf4914bab209f77e60ae3a18 Mon Sep 17 00:00:00 2001 From: Wu Fengguang Date: Fri, 25 Dec 2009 13:14:27 +0800 Subject: ALSA: hda - HDMI sticky stream tag support When we run the following commands in turn (with CONFIG_SND_HDA_POWER_SAVE_DEFAULT=0), speaker-test -Dhw:0,3 -c2 -twav # HDMI speaker-test -Dhw:0,0 -c2 -twav # Analog The second command will produce sound in the analog lineout _as well as_ HDMI sink. The root cause is, device 0 "reuses" the same stream tag that was used by device 3, and the "intelhdmi - sticky stream id" patch leaves the HDMI codec in a functional state. So the HDMI codec happily accepts the audio samples which reuse its stream tag. The proposed solution is to remember the last device each azx_dev was assigned to, and prefer to 1) reuse the azx_dev (and hence the stream tag) the HDMI codec last used 2) or assign a never-used azx_dev for HDMI With this patch and the above two speaker-test commands, HDMI codec will use stream tag 8 and Analog codec will use 5. The stream tag used by HDMI codec won't be reused by others, as long as we don't run out of the 4 playback azx_dev's. The legacy Analog codec will continue to use stream tag 5 because its device id is 0 (this is a bit tricky). Signed-off-by: Wu Fengguang Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 21 +++++++++++++++------ 1 file changed, 15 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index ff8ad46cc50e..ec9c348336cc 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -356,6 +356,7 @@ struct azx_dev { */ unsigned char stream_tag; /* assigned stream */ unsigned char index; /* stream index */ + int device; /* last device number assigned to */ unsigned int opened :1; unsigned int running :1; @@ -1441,10 +1442,13 @@ static int __devinit azx_codec_configure(struct azx *chip) */ /* assign a stream for the PCM */ -static inline struct azx_dev *azx_assign_device(struct azx *chip, int stream) +static inline struct azx_dev * +azx_assign_device(struct azx *chip, struct snd_pcm_substream *substream) { int dev, i, nums; - if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + struct azx_dev *res = NULL; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { dev = chip->playback_index_offset; nums = chip->playback_streams; } else { @@ -1453,10 +1457,15 @@ static inline struct azx_dev *azx_assign_device(struct azx *chip, int stream) } for (i = 0; i < nums; i++, dev++) if (!chip->azx_dev[dev].opened) { - chip->azx_dev[dev].opened = 1; - return &chip->azx_dev[dev]; + res = &chip->azx_dev[dev]; + if (res->device == substream->pcm->device) + break; } - return NULL; + if (res) { + res->opened = 1; + res->device = substream->pcm->device; + } + return res; } /* release the assigned stream */ @@ -1505,7 +1514,7 @@ static int azx_pcm_open(struct snd_pcm_substream *substream) int err; mutex_lock(&chip->open_mutex); - azx_dev = azx_assign_device(chip, substream->stream); + azx_dev = azx_assign_device(chip, substream); if (azx_dev == NULL) { mutex_unlock(&chip->open_mutex); return -EBUSY; -- cgit v1.2.2 From 729d55ba972348234759f8e40abf8de020f0d505 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 25 Dec 2009 22:49:01 +0100 Subject: ALSA: hda - Disable tigger at pin-sensing on AD codecs Analog Device codecs seem to have problems with the triggering of pin-sensing although their pincaps give the trigger requirements. Some reported that constant CPU load on HP laptops with AD codecs. For avoiding this regression, add a flag to codec struct to notify explicitly that the codec doesn't suppot the trigger at pin-sensing. Tested-by: Maciej Rutecki Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 10 ++++++---- sound/pci/hda/hda_codec.h | 1 + sound/pci/hda/patch_analog.c | 16 ++++++++++++++++ 3 files changed, 23 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 950ee5cfcacf..f98b47cd6cfb 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1327,11 +1327,13 @@ EXPORT_SYMBOL_HDA(snd_hda_query_pin_caps); */ u32 snd_hda_pin_sense(struct hda_codec *codec, hda_nid_t nid) { - u32 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); + u32 pincap; + if (!codec->no_trigger_sense) { + 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); + } return snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_PIN_SENSE, 0); } diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 1d541b7f5547..0a770a28e71f 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -817,6 +817,7 @@ struct hda_codec { unsigned int pin_amp_workaround:1; /* pin out-amp takes index * (e.g. Conexant codecs) */ + unsigned int no_trigger_sense:1; /* don't trigger at pin-sensing */ #ifdef CONFIG_SND_HDA_POWER_SAVE unsigned int power_on :1; /* current (global) power-state */ unsigned int power_transition :1; /* power-state in transition */ diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 1a36137e13ec..69a941c7b158 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -1186,6 +1186,8 @@ static int patch_ad1986a(struct hda_codec *codec) */ spec->multiout.no_share_stream = 1; + codec->no_trigger_sense = 1; + return 0; } @@ -1371,6 +1373,8 @@ static int patch_ad1983(struct hda_codec *codec) codec->patch_ops = ad198x_patch_ops; + codec->no_trigger_sense = 1; + return 0; } @@ -1813,6 +1817,9 @@ static int patch_ad1981(struct hda_codec *codec) codec->patch_ops.unsol_event = ad1981_hp_unsol_event; break; } + + codec->no_trigger_sense = 1; + return 0; } @@ -3118,6 +3125,8 @@ static int patch_ad1988(struct hda_codec *codec) #endif spec->vmaster_nid = 0x04; + codec->no_trigger_sense = 1; + return 0; } @@ -3330,6 +3339,8 @@ static int patch_ad1884(struct hda_codec *codec) codec->patch_ops = ad198x_patch_ops; + codec->no_trigger_sense = 1; + return 0; } @@ -4287,6 +4298,8 @@ static int patch_ad1884a(struct hda_codec *codec) break; } + codec->no_trigger_sense = 1; + return 0; } @@ -4623,6 +4636,9 @@ static int patch_ad1882(struct hda_codec *codec) spec->mixers[2] = ad1882_6stack_mixers; break; } + + codec->no_trigger_sense = 1; + return 0; } -- cgit v1.2.2 From a252c81a69c4f9a5a8782f33b91bd837e9dcd406 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 25 Dec 2009 22:56:20 +0100 Subject: ALSA: hda - use snd_hda_jack_detect() again in patch_sigmatel.c Use snd_hda_jack_detect() again for jack-sensing. The triggering problem can be worked around with codec->no_trigger_sense flag now. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 18 ++++++++++-------- 1 file changed, 10 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index eeda7beeb57a..2291a8396817 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4453,14 +4453,7 @@ static inline int get_pin_presence(struct hda_codec *codec, hda_nid_t nid) { if (!nid) return 0; - /* NOTE: we can't use snd_hda_jack_detect() here because STAC/IDT - * codecs behave wrongly when SET_PIN_SENSE is triggered, although - * the pincap gives TRIG_REQ bit. - */ - if (snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_PIN_SENSE, 0) & - AC_PINSENSE_PRESENCE) - return 1; - return 0; + return snd_hda_jack_detect(codec, nid); } static void stac92xx_line_out_detect(struct hda_codec *codec, @@ -4962,6 +4955,7 @@ static int patch_stac9200(struct hda_codec *codec) if (spec == NULL) return -ENOMEM; + codec->no_trigger_sense = 1; codec->spec = spec; spec->num_pins = ARRAY_SIZE(stac9200_pin_nids); spec->pin_nids = stac9200_pin_nids; @@ -5024,6 +5018,7 @@ static int patch_stac925x(struct hda_codec *codec) if (spec == NULL) return -ENOMEM; + codec->no_trigger_sense = 1; codec->spec = spec; spec->num_pins = ARRAY_SIZE(stac925x_pin_nids); spec->pin_nids = stac925x_pin_nids; @@ -5108,6 +5103,7 @@ static int patch_stac92hd73xx(struct hda_codec *codec) if (spec == NULL) return -ENOMEM; + codec->no_trigger_sense = 1; codec->spec = spec; codec->slave_dig_outs = stac92hd73xx_slave_dig_outs; spec->num_pins = ARRAY_SIZE(stac92hd73xx_pin_nids); @@ -5255,6 +5251,7 @@ static int patch_stac92hd83xxx(struct hda_codec *codec) if (spec == NULL) return -ENOMEM; + codec->no_trigger_sense = 1; codec->spec = spec; codec->slave_dig_outs = stac92hd83xxx_slave_dig_outs; spec->digbeep_nid = 0x21; @@ -5418,6 +5415,7 @@ static int patch_stac92hd71bxx(struct hda_codec *codec) if (spec == NULL) return -ENOMEM; + codec->no_trigger_sense = 1; codec->spec = spec; codec->patch_ops = stac92xx_patch_ops; spec->num_pins = STAC92HD71BXX_NUM_PINS; @@ -5661,6 +5659,7 @@ static int patch_stac922x(struct hda_codec *codec) if (spec == NULL) return -ENOMEM; + codec->no_trigger_sense = 1; codec->spec = spec; spec->num_pins = ARRAY_SIZE(stac922x_pin_nids); spec->pin_nids = stac922x_pin_nids; @@ -5764,6 +5763,7 @@ static int patch_stac927x(struct hda_codec *codec) if (spec == NULL) return -ENOMEM; + codec->no_trigger_sense = 1; codec->spec = spec; codec->slave_dig_outs = stac927x_slave_dig_outs; spec->num_pins = ARRAY_SIZE(stac927x_pin_nids); @@ -5898,6 +5898,7 @@ static int patch_stac9205(struct hda_codec *codec) if (spec == NULL) return -ENOMEM; + codec->no_trigger_sense = 1; codec->spec = spec; spec->num_pins = ARRAY_SIZE(stac9205_pin_nids); spec->pin_nids = stac9205_pin_nids; @@ -6053,6 +6054,7 @@ static int patch_stac9872(struct hda_codec *codec) spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) return -ENOMEM; + codec->no_trigger_sense = 1; codec->spec = spec; spec->num_pins = ARRAY_SIZE(stac9872_pin_nids); spec->pin_nids = stac9872_pin_nids; -- cgit v1.2.2 From 903b0eb39e3284cbf4d5a3773858e9586d16cbda Mon Sep 17 00:00:00 2001 From: Peter Huewe Date: Sat, 26 Dec 2009 03:27:45 +0100 Subject: ALSA: sound/arm: Fix build failure caused by missing struct aaci definition This patch fixes a build failure introduced by the patch ALSA: AACI: factor common hw_params logic into aaci_pcm_hw_params [1] by adding/moving the aaci struct to the right position. The patch mentioned above merged common source parts into one function, but unfortunately left out the aaci struct and consequently caused a build failure e.g. for arm versatile_config [2] References: [1] http://git.kernel.org/?p=linux/kernel/git/torvalds/linux-2.6.git;a=commitdiff;h=d3aee7996c30f928bbbbfd0994148e35d2e83084 [2] http://kisskb.ellerman.id.au/kisskb/buildresult/1893605/ Patch against Linus' tree. Signed-off-by: Peter Huewe Signed-off-by: Takashi Iwai --- sound/arm/aaci.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c index c5699863643b..656e474dca47 100644 --- a/sound/arm/aaci.c +++ b/sound/arm/aaci.c @@ -441,6 +441,7 @@ static int aaci_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { int err; + struct aaci *aaci = substream->private_data; aaci_pcm_hw_free(substream); if (aacirun->pcm_open) { @@ -560,7 +561,6 @@ static int aaci_pcm_open(struct snd_pcm_substream *substream) static int aaci_pcm_playback_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { - struct aaci *aaci = substream->private_data; struct aaci_runtime *aacirun = substream->runtime->private_data; unsigned int channels = params_channels(params); int ret; @@ -659,7 +659,6 @@ static struct snd_pcm_ops aaci_playback_ops = { static int aaci_pcm_capture_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { - struct aaci *aaci = substream->private_data; struct aaci_runtime *aacirun = substream->runtime->private_data; int ret; -- cgit v1.2.2 From 411fe85c7653f51403c2a6fd9026b0db2ab19478 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 27 Dec 2009 10:25:58 +0100 Subject: ALSA: hda - Don't cache beep controls The beep control verbs don't need to be cached for resume. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_beep.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index 5fe34a8d8c81..ca3c57a5f888 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -42,7 +42,7 @@ static void snd_hda_generate_beep(struct work_struct *work) return; /* generate tone */ - snd_hda_codec_write_cache(codec, beep->nid, 0, + snd_hda_codec_write(codec, beep->nid, 0, AC_VERB_SET_BEEP_CONTROL, beep->tone); } @@ -119,7 +119,7 @@ static void snd_hda_do_detach(struct hda_beep *beep) beep->dev = NULL; cancel_work_sync(&beep->beep_work); /* turn off beep for sure */ - snd_hda_codec_write_cache(beep->codec, beep->nid, 0, + snd_hda_codec_write(beep->codec, beep->nid, 0, AC_VERB_SET_BEEP_CONTROL, 0); } @@ -192,7 +192,7 @@ int snd_hda_enable_beep_device(struct hda_codec *codec, int enable) beep->enabled = enable; if (!enable) { /* turn off beep */ - snd_hda_codec_write_cache(beep->codec, beep->nid, 0, + snd_hda_codec_write(beep->codec, beep->nid, 0, AC_VERB_SET_BEEP_CONTROL, 0); } if (beep->mode == HDA_BEEP_MODE_SWREG) { -- cgit v1.2.2 From 54f7190b23080c7ac32078ed6a346bdc591ebef1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 27 Dec 2009 13:27:39 +0100 Subject: ALSA: hda - Fix Oops at reloading beep devices The recent change for supporting dynamic beep device allocation caused a problem resulting in Oops at reloading the driver. Also, it ignores the error from input device registration. This patch fixes the wrong check in snd_hda_detach_beep_device(), and returns an error when the input device registration fails properly. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_beep.c | 10 +++++++--- 1 file changed, 7 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index ca3c57a5f888..e4581a42ace5 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -239,8 +239,12 @@ int snd_hda_attach_beep_device(struct hda_codec *codec, int nid) mutex_init(&beep->mutex); if (beep->mode == HDA_BEEP_MODE_ON) { - beep->enabled = 1; - snd_hda_do_register(&beep->register_work); + int err = snd_hda_do_attach(beep); + if (err < 0) { + kfree(beep); + codec->beep = NULL; + return err; + } } return 0; @@ -253,7 +257,7 @@ void snd_hda_detach_beep_device(struct hda_codec *codec) if (beep) { cancel_work_sync(&beep->register_work); cancel_delayed_work(&beep->unregister_work); - if (beep->enabled) + if (beep->dev) snd_hda_do_detach(beep); codec->beep = NULL; kfree(beep); -- cgit v1.2.2 From dfb12eeb0f04b37e5eb3858864d074af4ecd2ac7 Mon Sep 17 00:00:00 2001 From: Daniel T Chen Date: Sun, 27 Dec 2009 15:48:40 -0500 Subject: ALSA: atiixp: Specify codec for Foxconn RC4107MA-RS2 BugLink: https://bugs.launchpad.net/ubuntu/+bug/498863 This mainboard needs ac97_codec=0. Cc: stable@kernel.org Tested-by: Apoorv Parle Signed-off-by: Daniel T Chen Signed-off-by: Takashi Iwai --- sound/pci/atiixp.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c index d6752dff2a44..42b4fbbd8e2b 100644 --- a/sound/pci/atiixp.c +++ b/sound/pci/atiixp.c @@ -297,6 +297,7 @@ static struct pci_device_id snd_atiixp_ids[] = { MODULE_DEVICE_TABLE(pci, snd_atiixp_ids); static struct snd_pci_quirk atiixp_quirks[] __devinitdata = { + SND_PCI_QUIRK(0x105b, 0x0c81, "Foxconn RC4107MA-RS2", 0), SND_PCI_QUIRK(0x15bd, 0x3100, "DFI RS482", 0), { } /* terminator */ }; -- cgit v1.2.2 From 9980c6209ebc2a02eb3ca51a4fae1e17f645c868 Mon Sep 17 00:00:00 2001 From: Roel Kluin Date: Sun, 27 Dec 2009 22:26:47 +0100 Subject: ALSA: test off by one in setsamplerate() With `while (i++ < MAX_WRITE_RETRY)' i reaches MAX_WRITE_RETRY + 1 after the loop Signed-off-by: Roel Kluin Signed-off-by: Takashi Iwai --- sound/pci/riptide/riptide.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c index b5ca02e2038c..e66ef2b69b5d 100644 --- a/sound/pci/riptide/riptide.c +++ b/sound/pci/riptide/riptide.c @@ -1058,7 +1058,7 @@ setsamplerate(struct cmdif *cif, unsigned char *intdec, unsigned int rate) rptr.retwords[2] != M && rptr.retwords[3] != N && i++ < MAX_WRITE_RETRY); - if (i == MAX_WRITE_RETRY) { + if (i > MAX_WRITE_RETRY) { snd_printdd("sent samplerate %d: %d failed\n", *intdec, rate); return -EIO; -- cgit v1.2.2 From 78b8d5d2ee280c463908fd75f3bdf246bcb6ac8d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 28 Dec 2009 12:24:22 +0100 Subject: ALSA: usb-audio - Avoid Oops after disconnect As the release of substreams may be done asynchronously from the disconnection, close callback needs to check the shutdown flag before actually accessing the usb interface. Reference: Novell bnc#505027 http://bugzilla.novell.com/show_bug.cgi?id=565027 Signed-off-by: Takashi Iwai --- sound/usb/usbaudio.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index 4963defee18a..9edef4684978 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -1936,7 +1936,7 @@ static int snd_usb_pcm_close(struct snd_pcm_substream *substream, int direction) struct snd_usb_stream *as = snd_pcm_substream_chip(substream); struct snd_usb_substream *subs = &as->substream[direction]; - if (subs->interface >= 0) { + if (!as->chip->shutdown && subs->interface >= 0) { usb_set_interface(subs->dev, subs->interface, 0); subs->interface = -1; } -- cgit v1.2.2 From ecbec242961ec66e900b5649ded1e40f5d5edc41 Mon Sep 17 00:00:00 2001 From: Manuel Lauss Date: Mon, 4 Jan 2010 16:29:49 +0100 Subject: ASoC: fixup oops in generic AC97 codec glue Initialize the glue by calling snd_soc_new_ac97_codec() as is done in other ASoC AC97 codecs. Fixes an oops caused by dereferencing uninitialized members in snd_soc_new_pcms(). Run-tested on Au1250. Signed-off-by: Manuel Lauss Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/ac97.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index 69bd0acc81c8..a1bbe16b7f96 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -102,6 +102,12 @@ static int ac97_soc_probe(struct platform_device *pdev) INIT_LIST_HEAD(&codec->dapm_widgets); INIT_LIST_HEAD(&codec->dapm_paths); + ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); + if (ret < 0) { + printk(KERN_ERR "ASoC: failed to init gen ac97 glue\n"); + goto err; + } + /* register pcms */ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); if (ret < 0) -- cgit v1.2.2 From 444c1953d496d272208902ff7010dc70d1f887f0 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Sun, 3 Jan 2010 12:39:27 +0100 Subject: sound: oss: off by one bug The problem is that in the original code sound_nblocks could go up to 1024 which would be an array overflow. This was found with a static checker and has been compile tested only. Signed-off-by: Dan Carpenter Signed-off-by: Jaroslav Kysela --- sound/oss/dev_table.c | 16 +++++++++------- sound/oss/sound_config.h | 2 ++ sound/oss/soundcard.c | 4 ++-- 3 files changed, 13 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/oss/dev_table.c b/sound/oss/dev_table.c index 08274c995d06..727bdb9ba2dc 100644 --- a/sound/oss/dev_table.c +++ b/sound/oss/dev_table.c @@ -67,14 +67,15 @@ int sound_install_audiodrv(int vers, char *name, struct audio_driver *driver, return -(EBUSY); } d = (struct audio_driver *) (sound_mem_blocks[sound_nblocks] = vmalloc(sizeof(struct audio_driver))); - - if (sound_nblocks < 1024) - sound_nblocks++; + sound_nblocks++; + if (sound_nblocks >= MAX_MEM_BLOCKS) + sound_nblocks = MAX_MEM_BLOCKS - 1; op = (struct audio_operations *) (sound_mem_blocks[sound_nblocks] = vmalloc(sizeof(struct audio_operations))); + sound_nblocks++; + if (sound_nblocks >= MAX_MEM_BLOCKS) + sound_nblocks = MAX_MEM_BLOCKS - 1; - if (sound_nblocks < 1024) - sound_nblocks++; if (d == NULL || op == NULL) { printk(KERN_ERR "Sound: Can't allocate driver for (%s)\n", name); sound_unload_audiodev(num); @@ -128,9 +129,10 @@ int sound_install_mixer(int vers, char *name, struct mixer_operations *driver, until you unload sound! */ op = (struct mixer_operations *) (sound_mem_blocks[sound_nblocks] = vmalloc(sizeof(struct mixer_operations))); + sound_nblocks++; + if (sound_nblocks >= MAX_MEM_BLOCKS) + sound_nblocks = MAX_MEM_BLOCKS - 1; - if (sound_nblocks < 1024) - sound_nblocks++; if (op == NULL) { printk(KERN_ERR "Sound: Can't allocate mixer driver for (%s)\n", name); return -ENOMEM; diff --git a/sound/oss/sound_config.h b/sound/oss/sound_config.h index 55271fbe7f49..9d35c4c65b9b 100644 --- a/sound/oss/sound_config.h +++ b/sound/oss/sound_config.h @@ -142,4 +142,6 @@ static inline int translate_mode(struct file *file) #define TIMER_ARMED 121234 #define TIMER_NOT_ARMED 1 +#define MAX_MEM_BLOCKS 1024 + #endif diff --git a/sound/oss/soundcard.c b/sound/oss/soundcard.c index 61aaedae6b7e..c62530943888 100644 --- a/sound/oss/soundcard.c +++ b/sound/oss/soundcard.c @@ -56,7 +56,7 @@ /* * Table for permanently allocated memory (used when unloading the module) */ -void * sound_mem_blocks[1024]; +void * sound_mem_blocks[MAX_MEM_BLOCKS]; int sound_nblocks = 0; /* Persistent DMA buffers */ @@ -574,7 +574,7 @@ static int __init oss_init(void) NULL, "%s%d", dev_list[i].name, j); } - if (sound_nblocks >= 1024) + if (sound_nblocks >= MAX_MEM_BLOCKS - 1) printk(KERN_ERR "Sound warning: Deallocation table was too small.\n"); return 0; -- cgit v1.2.2 From edf12b4af6e1d2b7c42c75ff00e55a9c52c06d70 Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Mon, 4 Jan 2010 22:23:34 +0100 Subject: sbawe: fix memory detection part 2 The patch "sbawe: fix memory detection" fixed detection for memoryless SB32 cards but broke detection of memory above 512KB. This patch fixes the regression. The patch has been tested on the SB32 card (CT3670) with 0MB, 2MB and 8MB memory installed. Signed-off-by: Krzysztof Helt Signed-off-by: Jaroslav Kysela --- sound/isa/sb/emu8000.c | 11 +++++++---- 1 file changed, 7 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/isa/sb/emu8000.c b/sound/isa/sb/emu8000.c index 751762f1c59a..0c40951b6523 100644 --- a/sound/isa/sb/emu8000.c +++ b/sound/isa/sb/emu8000.c @@ -377,12 +377,13 @@ init_arrays(struct snd_emu8000 *emu) static void __devinit size_dram(struct snd_emu8000 *emu) { - int i, size; + int i, size, detected_size; if (emu->dram_checked) return; size = 0; + detected_size = 0; /* write out a magic number */ snd_emu8000_dma_chan(emu, 0, EMU8000_RAM_WRITE); @@ -393,6 +394,8 @@ size_dram(struct snd_emu8000 *emu) while (size < EMU8000_MAX_DRAM) { + size += 512 * 1024; /* increment 512kbytes */ + /* Write a unique data on the test address. * if the address is out of range, the data is written on * 0x200000(=EMU8000_DRAM_OFFSET). Then the id word is @@ -414,7 +417,7 @@ size_dram(struct snd_emu8000 *emu) if (EMU8000_SMLD_READ(emu) != UNIQUE_ID2) break; /* no memory at this address */ - size += 512 * 1024; /* increment 512kbytes */ + detected_size = size; snd_emu8000_read_wait(emu); @@ -442,9 +445,9 @@ size_dram(struct snd_emu8000 *emu) snd_emu8000_dma_chan(emu, 1, EMU8000_RAM_CLOSE); snd_printdd("EMU8000 [0x%lx]: %d Kb on-board memory detected\n", - emu->port1, size/1024); + emu->port1, detected_size/1024); - emu->mem_size = size; + emu->mem_size = detected_size; emu->dram_checked = 1; } -- cgit v1.2.2 From 5ee518ecbcb5934e284ea51a19a939c891f5f7ea Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 7 Jan 2010 16:29:20 +0000 Subject: ASoC: Fix WM8350 DSP mode B configuration We need to set the LRCLK inversion bit to select DSP mode. Signed-off-by: Mark Brown Acked-by: Liam Girdwood Cc: stable@kernel.org --- sound/soc/codecs/wm8350.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index ebbf11b653a4..718ef912e758 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -925,7 +925,7 @@ static int wm8350_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) iface |= 0x3 << 8; break; case SND_SOC_DAIFMT_DSP_B: - iface |= 0x3 << 8; /* lg not sure which mode */ + iface |= 0x3 << 8 | WM8350_AIF_LRCLK_INV; break; default: return -EINVAL; -- cgit v1.2.2 From af9a75dd1a1f8a9aa406466cc8bb16208120488a Mon Sep 17 00:00:00 2001 From: Daniel T Chen Date: Sat, 9 Jan 2010 01:22:29 -0500 Subject: ALSA: ac97: Add Dell Dimension 2400 to Headphone/Line Jack Sense blacklist This model needs both 'Headphone Jack Sense' and 'Line Jack Sense' muted for audible playback, so just add it to the ad1981 jack sense blacklist. Cc: stable@kernel.org Tested-by: Pete Signed-off-by: Daniel T Chen Signed-off-by: Jaroslav Kysela --- sound/pci/ac97/ac97_patch.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c index 139cf3b2b9d7..d9266bae2849 100644 --- a/sound/pci/ac97/ac97_patch.c +++ b/sound/pci/ac97/ac97_patch.c @@ -1870,6 +1870,7 @@ static unsigned int ad1981_jacks_blacklist[] = { 0x10140554, /* Thinkpad T42p/R50p */ 0x10140567, /* Thinkpad T43p 2668-G7U */ 0x10140581, /* Thinkpad X41-2527 */ + 0x10280160, /* Dell Dimension 2400 */ 0x104380b0, /* Asus A7V8X-MX */ 0x11790241, /* Toshiba Satellite A-15 S127 */ 0x144dc01a, /* Samsung NP-X20C004/SEG */ -- cgit v1.2.2 From c68db7175f4dcb3d5789bb50bea6376fb81f87fe Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Sun, 10 Jan 2010 17:21:14 +0100 Subject: ALSA: ac97: add AC97 STMicroelectronics' codecs Add the STMicroelectronics ST7597 codec and an unknown codec from the same manufacturer found on the Creative SB 128 card (CT4810). Signed-off-by: Krzysztof Helt Signed-off-by: Jaroslav Kysela --- sound/pci/ac97/ac97_codec.c | 10 ++++++++++ sound/pci/ac97/ac97_id.h | 2 ++ 2 files changed, 12 insertions(+) (limited to 'sound') diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index c11920623009..a7630e9edf8a 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -83,6 +83,7 @@ static const struct ac97_codec_id snd_ac97_codec_id_vendors[] = { { 0x4e534300, 0xffffff00, "National Semiconductor", NULL, NULL }, { 0x50534300, 0xffffff00, "Philips", NULL, NULL }, { 0x53494c00, 0xffffff00, "Silicon Laboratory", NULL, NULL }, +{ 0x53544d00, 0xffffff00, "STMicroelectronics", NULL, NULL }, { 0x54524100, 0xffffff00, "TriTech", NULL, NULL }, { 0x54584e00, 0xffffff00, "Texas Instruments", NULL, NULL }, { 0x56494100, 0xffffff00, "VIA Technologies", NULL, NULL }, @@ -161,6 +162,7 @@ static const struct ac97_codec_id snd_ac97_codec_ids[] = { { 0x4e534350, 0xffffffff, "LM4550", patch_lm4550, NULL }, // volume wrap fix { 0x50534304, 0xffffffff, "UCB1400", patch_ucb1400, NULL }, { 0x53494c20, 0xffffffe0, "Si3036,8", mpatch_si3036, mpatch_si3036, AC97_MODEM_PATCH }, +{ 0x53544d02, 0xffffffff, "ST7597", NULL, NULL }, { 0x54524102, 0xffffffff, "TR28022", NULL, NULL }, { 0x54524103, 0xffffffff, "TR28023", NULL, NULL }, { 0x54524106, 0xffffffff, "TR28026", NULL, NULL }, @@ -213,6 +215,14 @@ static int snd_ac97_valid_reg(struct snd_ac97 *ac97, unsigned short reg) { /* filter some registers for buggy codecs */ switch (ac97->id) { + case AC97_ID_ST_AC97_ID4: + if (reg == 0x08) + return 0; + /* fall through */ + case AC97_ID_ST7597: + if (reg == 0x22 || reg == 0x7a) + return 1; + /* fall through */ case AC97_ID_AK4540: case AC97_ID_AK4542: if (reg <= 0x1c || reg == 0x20 || reg == 0x26 || reg >= 0x7c) diff --git a/sound/pci/ac97/ac97_id.h b/sound/pci/ac97/ac97_id.h index c129492c82b3..d603147c4a96 100644 --- a/sound/pci/ac97/ac97_id.h +++ b/sound/pci/ac97/ac97_id.h @@ -62,3 +62,5 @@ #define AC97_ID_CM9761_78 0x434d4978 #define AC97_ID_CM9761_82 0x434d4982 #define AC97_ID_CM9761_83 0x434d4983 +#define AC97_ID_ST7597 0x53544d02 +#define AC97_ID_ST_AC97_ID4 0x53544d04 -- cgit v1.2.2 From 9c0afc861a7228f718cb6a79fa7f9d46bf9ff300 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 12 Jan 2010 14:00:11 +0100 Subject: ALSA: hda - Fix ALC861-VD capture source mixer The capture source or input source mixer element wasn't created properly for ALC861-VD codec due to the wrong NID passed to alc_auto_create_input_ctls(). References: Novell bnc#568305 http://bugzilla.novell.com/show_bug.cgi?id=568305 Signed-off-by: Takashi Iwai Cc: --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c7465053d6bb..e3caa78ccd54 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -15493,7 +15493,7 @@ static struct alc_config_preset alc861vd_presets[] = { static int alc861vd_auto_create_input_ctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg) { - return alc_auto_create_input_ctls(codec, cfg, 0x15, 0x09, 0); + return alc_auto_create_input_ctls(codec, cfg, 0x15, 0x22, 0); } -- cgit v1.2.2 From 4dee8baa18d611b6dc854e1cc193550ff6f687be Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 13 Jan 2010 17:20:08 +0100 Subject: ALSA: hda - Fix Toshiba NB20x quirk entry The alc664-mode4 model doesn't seem to fit with Toshiba NB205 correctly. NB205 uses the pin 0x17 connected with the mixer 0x0f for the speaker output, which isn't controlled by mode4 model at all. Rather model=auto works fine as is on the latest driver, so let it back again. Tested-by: Nickolas Lloyd Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e3caa78ccd54..bff60cea7777 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -17251,7 +17251,7 @@ static struct snd_pci_quirk alc662_cfg_tbl[] = { SND_PCI_QUIRK(0x105b, 0x0cd6, "Foxconn", ALC662_ECS), SND_PCI_QUIRK(0x105b, 0x0d47, "Foxconn 45CMX/45GMX/45CMX-K", ALC662_3ST_6ch_DIG), - SND_PCI_QUIRK(0x1179, 0xff6e, "Toshiba NB200", ALC663_ASUS_MODE4), + SND_PCI_QUIRK(0x1179, 0xff6e, "Toshiba NB20x", ALC662_AUTO), SND_PCI_QUIRK(0x144d, 0xca00, "Samsung NC10", ALC272_SAMSUNG_NC10), SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte 945GCM-S2L", ALC662_3ST_6ch_DIG), -- cgit v1.2.2 From a76221d47ef2b73ff16c0fef00a784026308ea02 Mon Sep 17 00:00:00 2001 From: Alex Murray Date: Wed, 13 Jan 2010 23:15:03 +1030 Subject: ALSA: hda - Improved MacBook (Pro) 5,1 / 5,2 support This patch adds support for automatically muting the speakers when headphones are inserted, as well as relabelling the headphone widgets from the non-standard "HP" to the standard "Headphone" for the mb5 model. Signed-off-by: Alex Murray Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 28 ++++++++++++++++++++++++++-- 1 file changed, 26 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index bff60cea7777..11b989bacd3c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7094,8 +7094,8 @@ static struct snd_kcontrol_new alc885_mb5_mixer[] = { HDA_BIND_MUTE ("Surround Playback Switch", 0x0d, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("LFE Playback Volume", 0x0e, 0x00, HDA_OUTPUT), HDA_BIND_MUTE ("LFE Playback Switch", 0x0e, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("HP Playback Volume", 0x0f, 0x00, HDA_OUTPUT), - HDA_BIND_MUTE ("HP Playback Switch", 0x0f, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0f, 0x00, HDA_OUTPUT), + HDA_BIND_MUTE ("Headphone Playback Switch", 0x0f, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_MUTE ("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), @@ -7496,6 +7496,7 @@ static struct hda_verb alc885_mb5_init_verbs[] = { {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x14, AC_VERB_SET_CONNECT_SEL, 0x03}, + {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, /* Front Mic pin: input vref at 80% */ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, @@ -7680,6 +7681,27 @@ static void alc885_mbp3_setup(struct hda_codec *codec) spec->autocfg.speaker_pins[0] = 0x14; } +static void alc885_mb5_automute(struct hda_codec *codec) +{ + unsigned int present; + + present = snd_hda_codec_read(codec, 0x14, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + snd_hda_codec_amp_stereo(codec, 0x18, HDA_OUTPUT, 0, + HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); + snd_hda_codec_amp_stereo(codec, 0x1a, HDA_OUTPUT, 0, + HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); + +} + +static void alc885_mb5_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + /* Headphone insertion or removal. */ + if ((res >> 26) == ALC880_HP_EVENT) + alc885_mb5_automute(codec); +} + static void alc885_imac91_automute(struct hda_codec *codec) { unsigned int present; @@ -9126,6 +9148,8 @@ static struct alc_config_preset alc882_presets[] = { .input_mux = &mb5_capture_source, .dig_out_nid = ALC882_DIGOUT_NID, .dig_in_nid = ALC882_DIGIN_NID, + .unsol_event = alc885_mb5_unsol_event, + .init_hook = alc885_mb5_automute, }, [ALC885_MACPRO] = { .mixers = { alc882_macpro_mixer }, -- cgit v1.2.2 From c7a8eb103248a110cdbe0530d8c5ce987f099eee Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 14 Jan 2010 12:39:02 +0100 Subject: ALSA: hda - Fix missing capture mixer for ALC861/660 codecs The capture-related mixer elements are missing with ALC861/ALC660 codecs when quirks are present, due to missing call of set_capture_mixer(). Reference: Novell bnc#567340 http://bugzilla.novell.com/show_bug.cgi?id=567340 Signed-off-by: Takashi Iwai Cc: --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 11b989bacd3c..abae1007cea2 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -14879,6 +14879,8 @@ static int patch_alc861(struct hda_codec *codec) spec->stream_digital_playback = &alc861_pcm_digital_playback; spec->stream_digital_capture = &alc861_pcm_digital_capture; + if (!spec->cap_mixer) + set_capture_mixer(codec); set_beep_amp(spec, 0x23, 0, HDA_OUTPUT); spec->vmaster_nid = 0x03; -- cgit v1.2.2 From c181a13a41ef32c9037393f4b42b780e1a36eb91 Mon Sep 17 00:00:00 2001 From: Thadeu Lima de Souza Cascardo Date: Tue, 12 Jan 2010 20:20:39 -0200 Subject: ALSA: use subsys_initcall for sound core instead of module_init This is needed for built-in drivers which are built before the sound directory, like thinkpad_acpi. Otherwise, registering a card fails. Signed-off-by: Thadeu Lima de Souza Cascardo Signed-off-by: Takashi Iwai --- sound/core/sound.c | 4 ++-- sound/sound_core.c | 2 +- 2 files changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/core/sound.c b/sound/core/sound.c index 7872a02f6ca9..563d1967a0ad 100644 --- a/sound/core/sound.c +++ b/sound/core/sound.c @@ -468,5 +468,5 @@ static void __exit alsa_sound_exit(void) unregister_chrdev(major, "alsa"); } -module_init(alsa_sound_init) -module_exit(alsa_sound_exit) +subsys_initcall(alsa_sound_init); +module_exit(alsa_sound_exit); diff --git a/sound/sound_core.c b/sound/sound_core.c index dbca7c909a31..7c2d677a2df5 100644 --- a/sound/sound_core.c +++ b/sound/sound_core.c @@ -61,7 +61,7 @@ static void __exit cleanup_soundcore(void) class_destroy(sound_class); } -module_init(init_soundcore); +subsys_initcall(init_soundcore); module_exit(cleanup_soundcore); -- cgit v1.2.2 From d38cce7046cfd0011f69d5dcf6a22525438154f6 Mon Sep 17 00:00:00 2001 From: Kunal Gangakhedkar Date: Fri, 15 Jan 2010 21:01:47 +0530 Subject: ALSA: hda - Fix mute led GPIO on HP dv-series notebooks On my laptop (HP dv6-1110ax), there are no OEM strings in SMBIOS of type "HP_Mute_LED*". Hence, the GPIO for the mute button LED doesn't get set properly. I didn't find the strings in my cousin's laptop (HP dv9500t CTO) either. As per the documentation of find_mute_led_gpio(), these strings occur in HP B-series systems - so, before scanning the SMBIOS strings, we need to check if we're dealing with a B-series system. Need to get confirmation from HP if this logic takes care of all the systems. I'm trying to poke a friend there. Signed-off-by: Kunal Gangakhedkar Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 61 +++++++++++++++++++++++++++++++----------- 1 file changed, 45 insertions(+), 16 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 2291a8396817..799ba2570902 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4730,6 +4730,26 @@ static void stac92xx_unsol_event(struct hda_codec *codec, unsigned int res) } } +static int hp_blike_system(u32 subsystem_id); + +static void set_hp_led_gpio(struct hda_codec *codec) +{ + struct sigmatel_spec *spec = codec->spec; + switch (codec->vendor_id) { + case 0x111d7608: + /* GPIO 0 */ + spec->gpio_led = 0x01; + break; + case 0x111d7600: + case 0x111d7601: + case 0x111d7602: + case 0x111d7603: + /* GPIO 3 */ + spec->gpio_led = 0x08; + break; + } +} + /* * This method searches for the mute LED GPIO configuration * provided as OEM string in SMBIOS. The format of that string @@ -4741,6 +4761,14 @@ static void stac92xx_unsol_event(struct hda_codec *codec, unsigned int res) * * So, HP B-series like systems may have HP_Mute_LED_0 (current models) * or HP_Mute_LED_0_3 (future models) OEM SMBIOS strings + * + * + * The dv-series laptops don't seem to have the HP_Mute_LED* strings in + * SMBIOS - at least the ones I have seen do not have them - which include + * my own system (HP Pavilion dv6-1110ax) and my cousin's + * HP Pavilion dv9500t CTO. + * Need more information on whether it is true across the entire series. + * -- kunal */ static int find_mute_led_gpio(struct hda_codec *codec) { @@ -4751,28 +4779,27 @@ static int find_mute_led_gpio(struct hda_codec *codec) while ((dev = dmi_find_device(DMI_DEV_TYPE_OEM_STRING, NULL, dev))) { if (sscanf(dev->name, "HP_Mute_LED_%d_%d", - &spec->gpio_led_polarity, - &spec->gpio_led) == 2) { + &spec->gpio_led_polarity, + &spec->gpio_led) == 2) { spec->gpio_led = 1 << spec->gpio_led; return 1; } if (sscanf(dev->name, "HP_Mute_LED_%d", - &spec->gpio_led_polarity) == 1) { - switch (codec->vendor_id) { - case 0x111d7608: - /* GPIO 0 */ - spec->gpio_led = 0x01; - return 1; - case 0x111d7600: - case 0x111d7601: - case 0x111d7602: - case 0x111d7603: - /* GPIO 3 */ - spec->gpio_led = 0x08; - return 1; - } + &spec->gpio_led_polarity) == 1) { + set_hp_led_gpio(codec); + return 1; } } + + /* + * Fallback case - if we don't find the DMI strings, + * we statically set the GPIO - if not a B-series system. + */ + if (!hp_blike_system(codec->subsystem_id)) { + set_hp_led_gpio(codec); + spec->gpio_led_polarity = 1; + return 1; + } } return 0; } @@ -5548,6 +5575,8 @@ again: spec->num_dmuxes = ARRAY_SIZE(stac92hd71bxx_dmux_nids); spec->num_smuxes = stac92hd71bxx_connected_smuxes(codec, 0x1e); + snd_printdd("Found board config: %d\n", spec->board_config); + switch (spec->board_config) { case STAC_HP_M4: /* enable internal microphone */ -- cgit v1.2.2 From eaa9b3a748539651f50e3a234c8854e1b42a839a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 17 Jan 2010 13:09:33 +0100 Subject: ALSA: hda - Fix capture on Sony VAIO with single input Sony VAIO VGN-P11G with ALC262 codec has only one input pin, and the recording doesn't work with model=auto because ALC262 parser sets the wrong cap NIDs to choose the route and the default route for the sole input pin wasn't initialized properly. This patch solves these issues. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 62 +++++++++++++++++++++++++++++++++++++------ 1 file changed, 54 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index abae1007cea2..3f92def752fd 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1230,6 +1230,8 @@ static void alc_init_auto_mic(struct hda_codec *codec) return; /* invalid entry */ } } + if (!ext || !fixed) + return; if (!(get_wcaps(codec, ext) & AC_WCAP_UNSOL_CAP)) return; /* no unsol support */ snd_printdd("realtek: Enable auto-mic switch on NID 0x%x/0x%x\n", @@ -4812,6 +4814,49 @@ static void fixup_automic_adc(struct hda_codec *codec) spec->auto_mic = 0; /* disable auto-mic to be sure */ } +/* choose the ADC/MUX containing the input pin and initialize the setup */ +static void fixup_single_adc(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + hda_nid_t pin; + int i; + + /* search for the input pin; there must be only one */ + for (i = 0; i < AUTO_PIN_LAST; i++) { + if (spec->autocfg.input_pins[i]) { + pin = spec->autocfg.input_pins[i]; + break; + } + } + if (!pin) + return; + + /* set the default connection to that pin */ + for (i = 0; i < spec->num_adc_nids; i++) { + hda_nid_t cap = spec->capsrc_nids ? + spec->capsrc_nids[i] : spec->adc_nids[i]; + int idx; + + idx = get_connection_index(codec, cap, pin); + if (idx < 0) + continue; + /* use only this ADC */ + if (spec->capsrc_nids) + spec->capsrc_nids += i; + spec->adc_nids += i; + spec->num_adc_nids = 1; + /* select or unmute this route */ + if (get_wcaps_type(get_wcaps(codec, cap)) == AC_WID_AUD_MIX) { + snd_hda_codec_amp_stereo(codec, cap, HDA_INPUT, idx, + HDA_AMP_MUTE, 0); + } else { + snd_hda_codec_write_cache(codec, cap, 0, + AC_VERB_SET_CONNECT_SEL, idx); + } + return; + } +} + static void set_capture_mixer(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -4824,14 +4869,15 @@ static void set_capture_mixer(struct hda_codec *codec) alc_capture_mixer3 }, }; if (spec->num_adc_nids > 0 && spec->num_adc_nids <= 3) { - int mux; - if (spec->auto_mic) { - mux = 0; + int mux = 0; + if (spec->auto_mic) fixup_automic_adc(codec); - } else if (spec->input_mux && spec->input_mux->num_items > 1) - mux = 1; - else - mux = 0; + else if (spec->input_mux) { + if (spec->input_mux->num_items > 1) + mux = 1; + else if (spec->input_mux->num_items == 1) + fixup_single_adc(codec); + } spec->cap_mixer = caps[mux][spec->num_adc_nids - 1]; } } @@ -11203,7 +11249,7 @@ static int alc262_auto_create_multi_out_ctls(struct alc_spec *spec, } #define alc262_auto_create_input_ctls \ - alc880_auto_create_input_ctls + alc882_auto_create_input_ctls /* * generic initialization of ADC, input mixers and output mixers -- cgit v1.2.2 From 808c569f3609b37642d1e08373e3de829b99d0f8 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 18 Jan 2010 14:16:24 +0100 Subject: ALSA: Remove warning message for invalid OSS minor ranges When a card instance with a higher card number is registered, warning messages are spewed eventually with stack traces due to the invalid minor number for OSS device registration. For example, thinkpad-acpi registers the card number 29 as default, and you'll see always these messages. This is rather confusing (and worries users), thus better to return simply the error code. Signed-off-by: Takashi Iwai --- sound/core/sound_oss.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/core/sound_oss.c b/sound/core/sound_oss.c index 7fe12264ff80..0c164e5e4322 100644 --- a/sound/core/sound_oss.c +++ b/sound/core/sound_oss.c @@ -93,7 +93,7 @@ static int snd_oss_kernel_minor(int type, struct snd_card *card, int dev) default: return -EINVAL; } - if (snd_BUG_ON(minor < 0 || minor >= SNDRV_OSS_MINORS)) + if (minor < 0 || minor >= SNDRV_OSS_MINORS) return -EINVAL; return minor; } -- cgit v1.2.2 From 4feabefe53eb3742f0b2773a43200d1686f3a288 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 19 Jan 2010 15:38:44 +0100 Subject: ALSA: hda - Fix parsing pin node 0x21 on ALC259 ALC259 has a widget NID 0x21 for the output pin, but it wasn't handled properly in alc268_new_analog_output(). Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3f92def752fd..79cdae324c5e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -12541,6 +12541,7 @@ static int alc268_new_analog_output(struct alc_spec *spec, hda_nid_t nid, dac = 0x02; break; case 0x15: + case 0x21: dac = 0x03; break; default: -- cgit v1.2.2 From 3fb4a508b8e7957aa899f32cd6d9d462e102c7ca Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 19 Jan 2010 15:46:37 +0100 Subject: ALSA: hda - Turn on EAPD only if available for Realtek codecs Some codecs disable widgets used for output pins and reserve as vendor- spec widgets. Thus we need to check the widget type and pin cap before actually sending SET_EAPD verbs in the auto-configuration mode. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 29 +++++++++++++++++------------ 1 file changed, 17 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 79cdae324c5e..6ae610c0111e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1093,6 +1093,16 @@ static void alc889_coef_init(struct hda_codec *codec) snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_PROC_COEF, tmp|0x2010); } +/* turn on/off EAPD control (only if available) */ +static void set_eapd(struct hda_codec *codec, hda_nid_t nid, int on) +{ + if (get_wcaps_type(get_wcaps(codec, nid)) != AC_WID_PIN) + return; + if (snd_hda_query_pin_caps(codec, nid) & AC_PINCAP_EAPD) + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_EAPD_BTLENABLE, + on ? 2 : 0); +} + static void alc_auto_init_amp(struct hda_codec *codec, int type) { unsigned int tmp; @@ -1110,25 +1120,22 @@ static void alc_auto_init_amp(struct hda_codec *codec, int type) case ALC_INIT_DEFAULT: switch (codec->vendor_id) { case 0x10ec0260: - snd_hda_codec_write(codec, 0x0f, 0, - AC_VERB_SET_EAPD_BTLENABLE, 2); - snd_hda_codec_write(codec, 0x10, 0, - AC_VERB_SET_EAPD_BTLENABLE, 2); + set_eapd(codec, 0x0f, 1); + set_eapd(codec, 0x10, 1); break; case 0x10ec0262: case 0x10ec0267: case 0x10ec0268: case 0x10ec0269: + case 0x10ec0270: case 0x10ec0272: case 0x10ec0660: case 0x10ec0662: case 0x10ec0663: case 0x10ec0862: case 0x10ec0889: - snd_hda_codec_write(codec, 0x14, 0, - AC_VERB_SET_EAPD_BTLENABLE, 2); - snd_hda_codec_write(codec, 0x15, 0, - AC_VERB_SET_EAPD_BTLENABLE, 2); + set_eapd(codec, 0x14, 1); + set_eapd(codec, 0x15, 1); break; } switch (codec->vendor_id) { @@ -1836,10 +1843,8 @@ static void alc889_acer_aspire_8930g_setup(struct hda_codec *codec) #ifdef CONFIG_SND_HDA_POWER_SAVE static void alc889_power_eapd(struct hda_codec *codec, int power) { - snd_hda_codec_write(codec, 0x14, 0, - AC_VERB_SET_EAPD_BTLENABLE, power ? 2 : 0); - snd_hda_codec_write(codec, 0x15, 0, - AC_VERB_SET_EAPD_BTLENABLE, power ? 2 : 0); + set_eapd(codec, 0x14, power); + set_eapd(codec, 0x15, power); } #endif -- cgit v1.2.2 From dc99be47667c56046555e89e62f1ac17fa06329a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 20 Jan 2010 08:35:06 +0100 Subject: ALSA: hda - Fix HP T5735 automute This patch fixes the aut-mute setup on HP T5735 with ALC262 codec. Instead of wrong amp, use pin control toggling for muting the speaker now. Tested-by: Lee Trager Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6ae610c0111e..d00e6d1da085 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10382,7 +10382,7 @@ static void alc262_hp_t5735_setup(struct hda_codec *codec) struct alc_spec *spec = codec->spec; spec->autocfg.hp_pins[0] = 0x15; - spec->autocfg.speaker_pins[0] = 0x0c; /* HACK: not actually a pin */ + spec->autocfg.speaker_pins[0] = 0x14; } static struct snd_kcontrol_new alc262_hp_t5735_mixer[] = { @@ -11793,9 +11793,9 @@ static struct alc_config_preset alc262_presets[] = { .num_channel_mode = ARRAY_SIZE(alc262_modes), .channel_mode = alc262_modes, .input_mux = &alc262_capture_source, - .unsol_event = alc_automute_amp_unsol_event, + .unsol_event = alc_sku_unsol_event, .setup = alc262_hp_t5735_setup, - .init_hook = alc_automute_amp, + .init_hook = alc_inithook, }, [ALC262_HP_RP5700] = { .mixers = { alc262_hp_rp5700_mixer }, -- cgit v1.2.2 From 973b8cb0ead3e0b1dd3ee7b2df52e4dff1ffc707 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?=C5=81ukasz=20Wojni=C5=82owicz?= Date: Sun, 24 Jan 2010 14:12:37 +0100 Subject: ALSA: hda - add possibility to choose speakers configuration for 4930g MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Now one can choose speaker configuration in e.g. PulseAudio mixer Signed-off-by: Łukasz Wojniłowicz Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d00e6d1da085..da34095c707f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9478,6 +9478,7 @@ static struct alc_config_preset alc882_presets[] = { .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes), .channel_mode = alc883_3ST_6ch_modes, .need_dac_fix = 1, + .const_channel_count = 6, .num_mux_defs = ARRAY_SIZE(alc888_2_capture_sources), .input_mux = alc888_2_capture_sources, -- cgit v1.2.2 From 40aa7030e5213a43e9e0554fd7f95534ea310bf3 Mon Sep 17 00:00:00 2001 From: Guennadi Liakhovetski Date: Fri, 22 Jan 2010 18:00:03 +0100 Subject: ASoC: fix a memory-leak in wm8903 Remember to free the temporary register-cache. Signed-off-by: Guennadi Liakhovetski Acked-by: Liam Girdwood Signed-off-by: Mark Brown Cc: stable@kernel.org --- sound/soc/codecs/wm8903.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index ce5515e3f2b0..3595bd57c4eb 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1504,7 +1504,7 @@ static int wm8903_resume(struct platform_device *pdev) struct i2c_client *i2c = codec->control_data; int i; u16 *reg_cache = codec->reg_cache; - u16 *tmp_cache = kmemdup(codec->reg_cache, sizeof(wm8903_reg_defaults), + u16 *tmp_cache = kmemdup(reg_cache, sizeof(wm8903_reg_defaults), GFP_KERNEL); /* Bring the codec back up to standby first to minimise pop/clicks */ @@ -1516,6 +1516,7 @@ static int wm8903_resume(struct platform_device *pdev) for (i = 2; i < ARRAY_SIZE(wm8903_reg_defaults); i++) if (tmp_cache[i] != reg_cache[i]) snd_soc_write(codec, i, tmp_cache[i]); + kfree(tmp_cache); } else { dev_err(&i2c->dev, "Failed to allocate temporary cache\n"); } -- cgit v1.2.2 From 8ce28d6abff34886d3797b25324c940471b99164 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 27 Jan 2010 20:26:08 +0100 Subject: ALSA: hda - Add an ASUS mobo to MSI blacklist Sid Boyce reported that his machine locks up without enable_msi=0 option. This looks like another ASUS mobo with Nvidia combo. Reported-by: Sid Boyce Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index ec9c348336cc..565de38a3fc7 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2332,6 +2332,7 @@ static void __devinit check_probe_mask(struct azx *chip, int dev) */ static struct snd_pci_quirk msi_black_list[] __devinitdata = { SND_PCI_QUIRK(0x1043, 0x81f2, "ASUS", 0), /* Athlon64 X2 + nvidia */ + SND_PCI_QUIRK(0x1043, 0x829c, "ASUS", 0), /* nvidia */ {} }; -- cgit v1.2.2 From 3e59aaa7ae9de49af8810102f12857860d5bd0ed Mon Sep 17 00:00:00 2001 From: Anuj Aggarwal Date: Fri, 29 Jan 2010 13:58:55 +0530 Subject: ASoC: AIC23: Fixing writes to non-existing registers in resume function Commit e9ff5eb2 (Fixing infinite loop in resume path) uses wrong AIC23 register in resume function because of which register writes happen on some non-existing registers. Signed-off-by: Anuj Aggarwal Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic23.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index a9dc5fb54774..da589d8664d0 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -627,7 +627,7 @@ static int tlv320aic23_resume(struct platform_device *pdev) u16 reg; /* Sync reg_cache with the hardware */ - for (reg = 0; reg < TLV320AIC23_RESET; reg++) { + for (reg = 0; reg <= TLV320AIC23_ACTIVE; reg++) { u16 val = tlv320aic23_read_reg_cache(codec, reg); tlv320aic23_write(codec, reg, val); } -- cgit v1.2.2 From 5bbd4953a4fb5d8d597b4a53b8da97eee320b634 Mon Sep 17 00:00:00 2001 From: Anuj Aggarwal Date: Fri, 29 Jan 2010 15:49:22 +0530 Subject: ASoC: AM3517: ASoC driver not getting compiled Commit 761c9d45 (ASoC: Fix build of OMAP sound drivers) changes CONFIG_MACH_OMAP3517EVM -> CONFIG_SND_OMAP_SOC_OMAP3517EVM in the Makefile. Whereas the config option defined in Kconfig is SND_OMAP_SOC_AM3517EVM. Because of this, ASoC driver for AM3517 was not getting compiled. Signed-off-by: Anuj Aggarwal Signed-off-by: Mark Brown --- sound/soc/omap/Makefile | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index 3db8a6c523f4..19283e5edfbf 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -25,7 +25,7 @@ obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o obj-$(CONFIG_SND_OMAP_SOC_OVERO) += snd-soc-overo.o obj-$(CONFIG_SND_OMAP_SOC_OMAP2EVM) += snd-soc-omap2evm.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3EVM) += snd-soc-omap3evm.o -obj-$(CONFIG_SND_OMAP_SOC_OMAP3517EVM) += snd-soc-am3517evm.o +obj-$(CONFIG_SND_OMAP_SOC_AM3517EVM) += snd-soc-am3517evm.o obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3_BEAGLE) += snd-soc-omap3beagle.o -- cgit v1.2.2 From 21956b61f594f7924d98240da74bc81c28601fa9 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Tue, 2 Feb 2010 19:58:25 +0100 Subject: ALSA: ctxfi - fix PTP address initialization After hours of debugging, I finally found the reason why some source and runtime combination does not work. The PTP (page table pages) address must be aligned. I am not sure how much, but alignment to PAGE_SIZE is sufficient. Also, use ALSA's page allocation routines to ensure proper virtual -> physical address translation. Cc: Signed-off-by: Jaroslav Kysela --- sound/pci/ctxfi/ctatc.c | 15 ++------------- sound/pci/ctxfi/ctvmem.c | 38 ++++++++++++++++++-------------------- sound/pci/ctxfi/ctvmem.h | 8 +++++--- 3 files changed, 25 insertions(+), 36 deletions(-) (limited to 'sound') diff --git a/sound/pci/ctxfi/ctatc.c b/sound/pci/ctxfi/ctatc.c index cb65bd0dd35b..459c1f62783b 100644 --- a/sound/pci/ctxfi/ctatc.c +++ b/sound/pci/ctxfi/ctatc.c @@ -166,18 +166,7 @@ static void ct_unmap_audio_buffer(struct ct_atc *atc, struct ct_atc_pcm *apcm) static unsigned long atc_get_ptp_phys(struct ct_atc *atc, int index) { - struct ct_vm *vm; - void *kvirt_addr; - unsigned long phys_addr; - - vm = atc->vm; - kvirt_addr = vm->get_ptp_virt(vm, index); - if (kvirt_addr == NULL) - phys_addr = (~0UL); - else - phys_addr = virt_to_phys(kvirt_addr); - - return phys_addr; + return atc->vm->get_ptp_phys(atc->vm, index); } static unsigned int convert_format(snd_pcm_format_t snd_format) @@ -1669,7 +1658,7 @@ int __devinit ct_atc_create(struct snd_card *card, struct pci_dev *pci, } /* Set up device virtual memory management object */ - err = ct_vm_create(&atc->vm); + err = ct_vm_create(&atc->vm, pci); if (err < 0) goto error1; diff --git a/sound/pci/ctxfi/ctvmem.c b/sound/pci/ctxfi/ctvmem.c index 6b78752e9503..65da6e466f80 100644 --- a/sound/pci/ctxfi/ctvmem.c +++ b/sound/pci/ctxfi/ctvmem.c @@ -138,7 +138,7 @@ ct_vm_map(struct ct_vm *vm, struct snd_pcm_substream *substream, int size) return NULL; } - ptp = vm->ptp[0]; + ptp = (unsigned long *)vm->ptp[0].area; pte_start = (block->addr >> CT_PAGE_SHIFT); pages = block->size >> CT_PAGE_SHIFT; for (i = 0; i < pages; i++) { @@ -158,25 +158,25 @@ static void ct_vm_unmap(struct ct_vm *vm, struct ct_vm_block *block) } /* * - * return the host (kmalloced) addr of the @index-th device - * page talbe page on success, or NULL on failure. - * The first returned NULL indicates the termination. + * return the host physical addr of the @index-th device + * page table page on success, or ~0UL on failure. + * The first returned ~0UL indicates the termination. * */ -static void * -ct_get_ptp_virt(struct ct_vm *vm, int index) +static dma_addr_t +ct_get_ptp_phys(struct ct_vm *vm, int index) { - void *addr; + dma_addr_t addr; - addr = (index >= CT_PTP_NUM) ? NULL : vm->ptp[index]; + addr = (index >= CT_PTP_NUM) ? ~0UL : vm->ptp[index].addr; return addr; } -int ct_vm_create(struct ct_vm **rvm) +int ct_vm_create(struct ct_vm **rvm, struct pci_dev *pci) { struct ct_vm *vm; struct ct_vm_block *block; - int i; + int i, err = 0; *rvm = NULL; @@ -188,23 +188,21 @@ int ct_vm_create(struct ct_vm **rvm) /* Allocate page table pages */ for (i = 0; i < CT_PTP_NUM; i++) { - vm->ptp[i] = kmalloc(PAGE_SIZE, GFP_KERNEL); - if (!vm->ptp[i]) + err = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, + snd_dma_pci_data(pci), + PAGE_SIZE, &vm->ptp[i]); + if (err < 0) break; } - if (!i) { + if (err < 0) { /* no page table pages are allocated */ - kfree(vm); + ct_vm_destroy(vm); return -ENOMEM; } vm->size = CT_ADDRS_PER_PAGE * i; - /* Initialise remaining ptps */ - for (; i < CT_PTP_NUM; i++) - vm->ptp[i] = NULL; - vm->map = ct_vm_map; vm->unmap = ct_vm_unmap; - vm->get_ptp_virt = ct_get_ptp_virt; + vm->get_ptp_phys = ct_get_ptp_phys; INIT_LIST_HEAD(&vm->unused); INIT_LIST_HEAD(&vm->used); block = kzalloc(sizeof(*block), GFP_KERNEL); @@ -242,7 +240,7 @@ void ct_vm_destroy(struct ct_vm *vm) /* free allocated page table pages */ for (i = 0; i < CT_PTP_NUM; i++) - kfree(vm->ptp[i]); + snd_dma_free_pages(&vm->ptp[i]); vm->size = 0; diff --git a/sound/pci/ctxfi/ctvmem.h b/sound/pci/ctxfi/ctvmem.h index 01e4fd0386a3..b23adfca4de6 100644 --- a/sound/pci/ctxfi/ctvmem.h +++ b/sound/pci/ctxfi/ctvmem.h @@ -22,6 +22,8 @@ #include #include +#include +#include /* The chip can handle the page table of 4k pages * (emu20k1 can handle even 8k pages, but we don't use it right now) @@ -41,7 +43,7 @@ struct snd_pcm_substream; /* Virtual memory management object for card device */ struct ct_vm { - void *ptp[CT_PTP_NUM]; /* Device page table pages */ + struct snd_dma_buffer ptp[CT_PTP_NUM]; /* Device page table pages */ unsigned int size; /* Available addr space in bytes */ struct list_head unused; /* List of unused blocks */ struct list_head used; /* List of used blocks */ @@ -52,10 +54,10 @@ struct ct_vm { int size); /* Unmap device logical addr area. */ void (*unmap)(struct ct_vm *, struct ct_vm_block *block); - void *(*get_ptp_virt)(struct ct_vm *vm, int index); + dma_addr_t (*get_ptp_phys)(struct ct_vm *vm, int index); }; -int ct_vm_create(struct ct_vm **rvm); +int ct_vm_create(struct ct_vm **rvm, struct pci_dev *pci); void ct_vm_destroy(struct ct_vm *vm); #endif /* CTVMEM_H */ -- cgit v1.2.2 From 1eb6dc7dabcb4aa762d96f4f6978f3ef86321d68 Mon Sep 17 00:00:00 2001 From: Maxim Levitsky Date: Thu, 4 Feb 2010 22:21:47 +0200 Subject: ALSA: hda - Delay switching to polling mode if an interrupt was missing My sound codec seems sometimes (very rarely) to omit interrupts (ALC268) However, interrupt mode still works. Thus if we get timeout, poll the codec once. If we get 3 such polls in a row, then switch to polling mode. This patch is maybe an bandaid, but this might be a workaround for hardware bug. Signed-off-by: Maxim Levitsky Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 19 +++++++++++++++++-- 1 file changed, 17 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 565de38a3fc7..d853e2c33bb7 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -426,6 +426,7 @@ struct azx { /* flags */ int position_fix; + int poll_count; unsigned int running :1; unsigned int initialized :1; unsigned int single_cmd :1; @@ -506,7 +507,7 @@ static char *driver_short_names[] __devinitdata = { #define get_azx_dev(substream) (substream->runtime->private_data) static int azx_acquire_irq(struct azx *chip, int do_disconnect); - +static int azx_send_cmd(struct hda_bus *bus, unsigned int val); /* * Interface for HD codec */ @@ -664,11 +665,12 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus, { struct azx *chip = bus->private_data; unsigned long timeout; + int do_poll = 0; again: timeout = jiffies + msecs_to_jiffies(1000); for (;;) { - if (chip->polling_mode) { + if (chip->polling_mode || do_poll) { spin_lock_irq(&chip->reg_lock); azx_update_rirb(chip); spin_unlock_irq(&chip->reg_lock); @@ -676,6 +678,9 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus, if (!chip->rirb.cmds[addr]) { smp_rmb(); bus->rirb_error = 0; + + if (!do_poll) + chip->poll_count = 0; return chip->rirb.res[addr]; /* the last value */ } if (time_after(jiffies, timeout)) @@ -688,6 +693,16 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus, } } + if (!chip->polling_mode && chip->poll_count < 2) { + snd_printdd(SFX "azx_get_response timeout, " + "polling the codec once: last cmd=0x%08x\n", + chip->last_cmd[addr]); + do_poll = 1; + chip->poll_count++; + goto again; + } + + if (!chip->polling_mode) { snd_printk(KERN_WARNING SFX "azx_get_response timeout, " "switching to polling mode: last cmd=0x%08x\n", -- cgit v1.2.2 From 9492837a6f54b069e13e40e3c89898bb8837a386 Mon Sep 17 00:00:00 2001 From: Maxim Levitsky Date: Thu, 4 Feb 2010 22:26:37 +0200 Subject: ALSA: cosmetic: make hda intel interrupt name consistent with others This renames the interrupt name in /proc/interrupt. HDA Intel -> hda_intel This also eliminates space from the name, probably helping some parsers. Don't think anybody depends on this name in userspace Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index d853e2c33bb7..b8faa6dc5abe 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2058,7 +2058,7 @@ static int azx_acquire_irq(struct azx *chip, int do_disconnect) { if (request_irq(chip->pci->irq, azx_interrupt, chip->msi ? 0 : IRQF_SHARED, - "HDA Intel", chip)) { + "hda_intel", chip)) { printk(KERN_ERR "hda-intel: unable to grab IRQ %d, " "disabling device\n", chip->pci->irq); if (do_disconnect) -- cgit v1.2.2 From 9d4c7464458770d309169f7a7ce1ea6f8a4a7de5 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Fri, 5 Feb 2010 10:19:41 +0100 Subject: ALSA: ice1724 - aureon - fix wm8770 volume offset The volume register is from 0..0x7f and 0..0x1a range is mute. Also, fix mute combining in wm_vol_put(). The wrong behaviour was noticed by Peter Christensen. Signed-off-by: Jaroslav Kysela --- sound/pci/ice1712/aureon.c | 12 +++++++----- 1 file changed, 7 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/pci/ice1712/aureon.c b/sound/pci/ice1712/aureon.c index 765d7bd4c3d4..9e66f6d306f8 100644 --- a/sound/pci/ice1712/aureon.c +++ b/sound/pci/ice1712/aureon.c @@ -703,11 +703,13 @@ static void wm_set_vol(struct snd_ice1712 *ice, unsigned int index, unsigned sho { unsigned char nvol; - if ((master & WM_VOL_MUTE) || (vol & WM_VOL_MUTE)) + if ((master & WM_VOL_MUTE) || (vol & WM_VOL_MUTE)) { nvol = 0; - else + } else { nvol = ((vol % WM_VOL_CNT) * (master % WM_VOL_CNT)) / WM_VOL_MAX; + nvol += 0x1b; + } wm_put(ice, index, nvol); wm_put_nocache(ice, index, 0x180 | nvol); @@ -778,7 +780,7 @@ static int wm_master_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_ for (ch = 0; ch < 2; ch++) { unsigned int vol = ucontrol->value.integer.value[ch]; if (vol > WM_VOL_MAX) - continue; + vol = WM_VOL_MAX; vol |= spec->master[ch] & WM_VOL_MUTE; if (vol != spec->master[ch]) { int dac; @@ -834,8 +836,8 @@ static int wm_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value * for (i = 0; i < voices; i++) { unsigned int vol = ucontrol->value.integer.value[i]; if (vol > WM_VOL_MAX) - continue; - vol |= spec->vol[ofs+i]; + vol = WM_VOL_MAX; + vol |= spec->vol[ofs+i] & WM_VOL_MUTE; if (vol != spec->vol[ofs+i]) { spec->vol[ofs+i] = vol; idx = WM_DAC_ATTEN + ofs + i; -- cgit v1.2.2 From 3b9447fb7fa1829731290e64ef928d4f6461310a Mon Sep 17 00:00:00 2001 From: Grazvydas Ignotas Date: Fri, 5 Feb 2010 00:55:33 +0200 Subject: ASoC: pandora: Add APLL supply to fix audio output Pandora's external DAC is using 256*Fs output from the TWL4030 codec, and TWL4030 needs to have APLL enabled for it's 256*Fs output to function. Signed-off-by: Grazvydas Ignotas Acked-by: Peter Ujfalusi Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/omap/omap3pandora.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c index 71b2c161158d..68980c19a3bc 100644 --- a/sound/soc/omap/omap3pandora.c +++ b/sound/soc/omap/omap3pandora.c @@ -145,6 +145,7 @@ static const struct snd_soc_dapm_widget omap3pandora_in_dapm_widgets[] = { }; static const struct snd_soc_dapm_route omap3pandora_out_map[] = { + {"PCM DAC", NULL, "APLL Enable"}, {"Headphone Amplifier", NULL, "PCM DAC"}, {"Line Out", NULL, "PCM DAC"}, {"Headphone Jack", NULL, "Headphone Amplifier"}, -- cgit v1.2.2