June 2014 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 1/2] ALSA: soundcore: coding style fixes
by Daniel Mack 05 Jun '14

05 Jun '14

sound/sounc_core.c has a larger number of style issues. Fix them to make the code more readable. No functional change. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- sound/sound_core.c | 192 ++++++++++++++++++++++++----------------------------- 1 file changed, 85 insertions(+), 107 deletions(-) diff --git a/sound/sound_core.c b/sound/sound_core.c index 11e953a..bf9e41e 100644 --- a/sound/sound_core.c +++ b/sound/sound_core.c @@ -68,21 +68,16 @@ module_exit(cleanup_soundcore); #ifdef CONFIG_SOUND_OSS_CORE /* * OSS sound core handling. Breaks out sound functions to submodules - * - * Author: Alan Cox <alan(a)lxorguk.ukuu.org.uk> - * - * Fixes: * + * Author: Alan Cox <alan(a)lxorguk.ukuu.org.uk> * * This program is free software; you can redistribute it and/or * modify it under the terms of the GNU General Public License * as published by the Free Software Foundation; either version * 2 of the License, or (at your option) any later version. * - * -------------------- - * * Top level handler for the sound subsystem. Various devices can - * plug into this. The fact they don't all go via OSS doesn't mean + * plug into this. The fact they don't all go via OSS doesn't mean * they don't have to implement the OSS API. There is a lot of logic * to keeping much of the OSS weight out of the code in a compatibility * module, but it's up to the driver to rember to load it... @@ -111,8 +106,7 @@ module_exit(cleanup_soundcore); #define SOUND_STEP 16 -struct sound_unit -{ +struct sound_unit { int unit_minor; const struct file_operations *unit_fops; struct sound_unit *next; @@ -158,8 +152,7 @@ module_param(preclaim_oss, int, 0444); static int soundcore_open(struct inode *, struct file *); -static const struct file_operations soundcore_fops = -{ +static const struct file_operations soundcore_fops = { /* We must have an owner or the module locking fails */ .owner = THIS_MODULE, .open = soundcore_open, @@ -171,71 +164,66 @@ static const struct file_operations soundcore_fops = * join into it. Called with the lock asserted */ -static int __sound_insert_unit(struct sound_unit * s, struct sound_unit **list, const struct file_operations *fops, int index, int low, int top) +static int __sound_insert_unit(struct sound_unit *s, struct sound_unit **list, + const struct file_operations *fops, + int index, int low, int top) { - int n=low; + int n = low; if (index < 0) { /* first free */ + while (*list && (*list)->unit_minor < n) + list = &((*list)->next); - while (*list && (*list)->unit_minor<n) - list=&((*list)->next); - - while(n<top) - { + while (n < top) { /* Found a hole ? */ - if(*list==NULL || (*list)->unit_minor>n) + if (*list == NULL || (*list)->unit_minor > n) break; - list=&((*list)->next); - n+=SOUND_STEP; + + list = &((*list)->next); + n += SOUND_STEP; } - if(n>=top) + if (n >= top) return -ENOENT; } else { - n = low+(index*16); + n = low + (index * 16); while (*list) { - if ((*list)->unit_minor==n) + if ((*list)->unit_minor == n) return -EBUSY; - if ((*list)->unit_minor>n) + if ((*list)->unit_minor > n) break; - list=&((*list)->next); + list = &((*list)->next); } - } - - /* - * Fill it in - */ - - s->unit_minor=n; - s->unit_fops=fops; - - /* - * Link it - */ - - s->next=*list; - *list=s; - - + } + + /* Fill it in */ + s->unit_minor = n; + s->unit_fops = fops; + + /* Link it */ + s->next = *list; + *list = s; + return n; } /* * Remove a node from the chain. Called with the lock asserted */ - -static struct sound_unit *__sound_remove_unit(struct sound_unit **list, int unit) +static struct sound_unit *__sound_remove_unit(struct sound_unit **list, + int unit) { - while(*list) - { - struct sound_unit *p=*list; - if(p->unit_minor==unit) - { - *list=p->next; + while (*list) { + struct sound_unit *p = *list; + + if (p->unit_minor == unit) { + *list = p->next; return p; } - list=&(p->next); + + list = &p->next; } + printk(KERN_ERR "Sound device %d went missing!\n", unit); return NULL; } @@ -250,12 +238,15 @@ static DEFINE_SPINLOCK(sound_loader_lock); * Allocate the controlling structure and add it to the sound driver * list. Acquires locks as needed */ - -static int sound_insert_unit(struct sound_unit **list, const struct file_operations *fops, int index, int low, int top, const char *name, umode_t mode, struct device *dev) +static int sound_insert_unit(struct sound_unit **list, + const struct file_operations *fops, + int index, int low, int top, const char *name, + umode_t mode, struct device *dev) { - struct sound_unit *s = kmalloc(sizeof(*s), GFP_KERNEL); + struct sound_unit *s; int r; + s = kmalloc(sizeof(*s), GFP_KERNEL); if (!s) return -ENOMEM; @@ -263,7 +254,7 @@ static int sound_insert_unit(struct sound_unit **list, const struct file_operati retry: r = __sound_insert_unit(s, list, fops, index, low, top); spin_unlock(&sound_loader_lock); - + if (r < 0) goto fail; else if (r < SOUND_STEP) @@ -292,7 +283,8 @@ retry: } device_create(sound_class, dev, MKDEV(SOUND_MAJOR, s->unit_minor), - NULL, "%s", s->name+6); + NULL, "%s", s->name + 6); + return s->unit_minor; fail: @@ -305,7 +297,6 @@ fail: * completed the removal before their file operations become * invalid. */ - static void sound_remove_unit(struct sound_unit **list, int unit) { struct sound_unit *p; @@ -357,7 +348,6 @@ static struct sound_unit *chains[SOUND_STEP]; * Return: The allocated number is returned on success. On failure, * a negative error code is returned. */ - int register_sound_special_device(const struct file_operations *fops, int unit, struct device *dev) { @@ -367,69 +357,66 @@ int register_sound_special_device(const struct file_operations *fops, int unit, char _name[16]; switch (chain) { - case 0: + case 0: name = "mixer"; break; - case 1: + case 1: name = "sequencer"; if (unit >= SOUND_STEP) goto __unknown; max_unit = unit + 1; break; - case 2: + case 2: name = "midi"; break; - case 3: + case 3: name = "dsp"; break; - case 4: + case 4: name = "audio"; break; - case 5: + case 5: name = "dspW"; break; - case 8: + case 8: name = "sequencer2"; if (unit >= SOUND_STEP) goto __unknown; max_unit = unit + 1; break; - case 9: + case 9: name = "dmmidi"; break; - case 10: + case 10: name = "dmfm"; break; - case 12: + case 12: name = "adsp"; break; - case 13: + case 13: name = "amidi"; break; - case 14: + case 14: name = "admmidi"; break; - default: - { - __unknown: - sprintf(_name, "unknown%d", chain); - if (unit >= SOUND_STEP) - strcat(_name, "-"); - name = _name; - } + default: +__unknown: + sprintf(_name, "unknown%d", chain); + if (unit >= SOUND_STEP) + strcat(_name, "-"); + name = _name; break; } + return sound_insert_unit(&chains[chain], fops, -1, unit, max_unit, name, S_IRUSR | S_IWUSR, dev); } - EXPORT_SYMBOL(register_sound_special_device); int register_sound_special(const struct file_operations *fops, int unit) { return register_sound_special_device(fops, unit, NULL); } - EXPORT_SYMBOL(register_sound_special); /** @@ -443,13 +430,11 @@ EXPORT_SYMBOL(register_sound_special); * Return: On success, the allocated number is returned. On failure, * a negative error code is returned. */ - int register_sound_mixer(const struct file_operations *fops, int dev) { return sound_insert_unit(&chains[0], fops, dev, 0, 128, "mixer", S_IRUSR | S_IWUSR, NULL); } - EXPORT_SYMBOL(register_sound_mixer); /** @@ -469,14 +454,13 @@ int register_sound_midi(const struct file_operations *fops, int dev) return sound_insert_unit(&chains[2], fops, dev, 2, 130, "midi", S_IRUSR | S_IWUSR, NULL); } - EXPORT_SYMBOL(register_sound_midi); /* * DSP's are registered as a triple. Register only one and cheat * in open - see below. */ - + /** * register_sound_dsp - register a DSP device * @fops: File operations for the driver @@ -497,7 +481,6 @@ int register_sound_dsp(const struct file_operations *fops, int dev) return sound_insert_unit(&chains[3], fops, dev, 3, 131, "dsp", S_IWUSR | S_IRUSR, NULL); } - EXPORT_SYMBOL(register_sound_dsp); /** @@ -514,7 +497,6 @@ void unregister_sound_special(int unit) { sound_remove_unit(&chains[unit % SOUND_STEP], unit); } - EXPORT_SYMBOL(unregister_sound_special); /** @@ -529,7 +511,6 @@ void unregister_sound_mixer(int unit) { sound_remove_unit(&chains[0], unit); } - EXPORT_SYMBOL(unregister_sound_mixer); /** @@ -544,7 +525,6 @@ void unregister_sound_midi(int unit) { sound_remove_unit(&chains[2], unit); } - EXPORT_SYMBOL(unregister_sound_midi); /** @@ -561,21 +541,19 @@ void unregister_sound_dsp(int unit) { sound_remove_unit(&chains[3], unit); } - - EXPORT_SYMBOL(unregister_sound_dsp); static struct sound_unit *__look_for_unit(int chain, int unit) { struct sound_unit *s; - - s=chains[chain]; - while(s && s->unit_minor <= unit) - { - if(s->unit_minor==unit) + + s = chains[chain]; + while (s && s->unit_minor <= unit) { + if (s->unit_minor == unit) return s; - s=s->next; + s = s->next; } + return NULL; } @@ -586,14 +564,13 @@ static int soundcore_open(struct inode *inode, struct file *file) struct sound_unit *s; const struct file_operations *new_fops = NULL; - chain=unit&0x0F; - if(chain==4 || chain==5) /* dsp/audio/dsp16 */ - { - unit&=0xF0; - unit|=3; - chain=3; + chain = unit & 0x0F; + if (chain == 4 || chain == 5) { /* dsp/audio/dsp16 */ + unit &= 0xF0; + unit |= 3; + chain = 3; } - + spin_lock(&sound_loader_lock); s = __look_for_unit(chain, unit); if (s) @@ -608,8 +585,8 @@ static int soundcore_open(struct inode *inode, struct file *file) * ALSA toplevel modules for soundcards, thus we need * load them at first. [Jaroslav Kysela <perex(a)jcu.cz>] */ - request_module("sound-slot-%i", unit>>4); - request_module("sound-service-%i-%i", unit>>4, chain); + request_module("sound-slot-%i", unit >> 4); + request_module("sound-service-%i-%i", unit >> 4, chain); /* * sound-slot/service-* module aliases are scheduled @@ -636,10 +613,11 @@ static int soundcore_open(struct inode *inode, struct file *file) replace_fops(file, new_fops); if (file->f_op->open) - err = file->f_op->open(inode,file); + err = file->f_op->open(inode, file); return err; } + return -ENODEV; } -- 1.9.3

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[alsa-devel] New codec support for ALC891
by Kailang 05 Jun '14

05 Jun '14

Hi Takashi, We have a new codec ALC891. The support patch as attach. Many Thanks. BR, Kailang

2 1

[alsa-devel] [PATCH - ucm 1/1] ucm: set active_verb when set_verb was successful
by nrubinstein＠aldebaran-robotics.com 05 Jun '14

05 Jun '14

From: Noé Rubinstein <nrubinstein(a)aldebaran-robotics.com> This seems to be a bug: execute_sequence returns zero on success, therefore the current code changes the active_verb iff execute_sequence failed. Signed-off-by: Noé Rubinstein <nrubinstein(a)aldebaran-robotics.com> diff --git a/src/ucm/main.c b/src/ucm/main.c index 0494728..3c0d7e4 100644 --- a/src/ucm/main.c +++ b/src/ucm/main.c @@ -623,7 +623,7 @@ static int set_verb(snd_use_case_mgr_t *uc_mgr, &verb->value_list, &uc_mgr->value_list, NULL); - if (enable && err >= 0) + if (enable && err == 0) uc_mgr->active_verb = verb; return err; } -- 1.9.2

2 1

[alsa-devel] Rate or plug plugin issue with snd-pcsp
by Alexander E. Patrakov 05 Jun '14

05 Jun '14

Hello. While testing application support for weird-rate cards (such as snd-pcsp), I found a bug in alsa-lib. Alsa-plugins are installed, so the "libspeex (builtin)" rate converter is used. This bug already exists in alsa-plugins 1.0.27.2, so not a regression and thus should not block 1.0.28. $ file test.wav test.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, stereo 32000 Hz $ aplay -vvv -D plughw:pcsp test.wav Playing WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 32000 Hz, Stereo aplay: set_params:1297: Unable to install hw params: ACCESS: RW_INTERLEAVED FORMAT: S16_LE SUBFORMAT: STD SAMPLE_BITS: 16 FRAME_BITS: 32 CHANNELS: 2 RATE: NONE PERIOD_TIME: (125006 125007) PERIOD_SIZE: NONE PERIOD_BYTES: (16000 16004) PERIODS: (3 4) BUFFER_TIME: 500000 BUFFER_SIZE: 16000 BUFFER_BYTES: 64000 TICK_TIME: 0 The pcsp device supports the sample rate of 37286 Hz. Files with 44100 Hz sample rate work fine. $ aplay -vv -D plughw:pcsp -f cd /dev/zero Playing raw data '/dev/zero' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo Plug PCM: Route conversion PCM (sformat=S16_LE) Transformation table: 0 <- 0*0,5 + 1*0,5 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 44100 exact rate : 44100 (44100/1) msbits : 16 buffer_size : 22050 period_size : 5512 period_time : 125006 tstamp_mode : NONE period_step : 1 avail_min : 5512 period_event : 0 start_threshold : 22050 stop_threshold : 22050 silence_threshold: 0 silence_size : 0 boundary : 6206523236469964800 Slave: Rate conversion PCM (37286, sformat=S16_LE) Converter: libspeex (builtin) Protocol version: 10002 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 1 rate : 44100 exact rate : 44100 (44100/1) msbits : 16 buffer_size : 22050 period_size : 5512 period_time : 125006 tstamp_mode : NONE period_step : 1 avail_min : 5512 period_event : 0 start_threshold : 22050 stop_threshold : 22050 silence_threshold: 0 silence_size : 0 boundary : 6206523236469964800 Slave: Hardware PCM card 2 'pcsp' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 1 rate : 37286 exact rate : 37286 (37286/1) msbits : 16 buffer_size : 18644 period_size : 4661 period_time : 125006 tstamp_mode : NONE period_step : 1 avail_min : 4661 period_event : 0 start_threshold : 18644 stop_threshold : 18644 silence_threshold: 0 silence_size : 0 boundary : 5247819465793470464 appl_ptr : 0 hw_ptr : 0 #+ | 00%^C -- Alexander E. Patrakov

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[alsa-devel] [PATCH] ASoC: pxa: add I2C dependencies as needed
by Arnd Bergmann 04 Jun '14

04 Jun '14

We have in the past added 'depends on I2C' for some of the PXA boards after hitting randconfig build bugs. I have seens a couple of new bugs in this area during the linux-next cycle for 3.16, after it became possible to build some more PXA machines with I2C disabled. To shut this up for good, this adds the dependency to every board that uses I2C as the interface to the codec. I have gone through all board files and verified that they all either use AC97 or I2C, and this annotates the latter. Some of these already enable I2C from mach-pxa/Kconfig, but since that can change it's better to be explicit here. The link error that can result otherwise happens when CONFIG_I2C is set to 'm' and the codec driver is built-in as a result of being selected by the platform specific glue. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index dcf0c08..e195b038 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -40,7 +40,7 @@ config SND_MMP_SOC_SSPA config SND_PXA2XX_SOC_CORGI tristate "SoC Audio support for Sharp Zaurus SL-C7x0" - depends on SND_PXA2XX_SOC && PXA_SHARP_C7xx + depends on SND_PXA2XX_SOC && PXA_SHARP_C7xx && I2C select SND_PXA2XX_SOC_I2S select SND_SOC_WM8731 help @@ -49,7 +49,7 @@ config SND_PXA2XX_SOC_CORGI config SND_PXA2XX_SOC_SPITZ tristate "SoC Audio support for Sharp Zaurus SL-Cxx00" - depends on SND_PXA2XX_SOC && PXA_SHARP_Cxx00 + depends on SND_PXA2XX_SOC && PXA_SHARP_Cxx00 && I2C select SND_PXA2XX_SOC_I2S select SND_SOC_WM8750 help @@ -58,7 +58,7 @@ config SND_PXA2XX_SOC_SPITZ config SND_PXA2XX_SOC_Z2 tristate "SoC Audio support for Zipit Z2" - depends on SND_PXA2XX_SOC && MACH_ZIPIT2 + depends on SND_PXA2XX_SOC && MACH_ZIPIT2 && I2C select SND_PXA2XX_SOC_I2S select SND_SOC_WM8750 help @@ -66,7 +66,7 @@ config SND_PXA2XX_SOC_Z2 config SND_PXA2XX_SOC_POODLE tristate "SoC Audio support for Poodle" - depends on SND_PXA2XX_SOC && MACH_POODLE + depends on SND_PXA2XX_SOC && MACH_POODLE && I2C select SND_PXA2XX_SOC_I2S select SND_SOC_WM8731 help @@ -181,7 +181,7 @@ config SND_PXA2XX_SOC_HX4700 config SND_PXA2XX_SOC_MAGICIAN tristate "SoC Audio support for HTC Magician" - depends on SND_PXA2XX_SOC && MACH_MAGICIAN + depends on SND_PXA2XX_SOC && MACH_MAGICIAN && I2C select SND_PXA2XX_SOC_I2S select SND_PXA_SOC_SSP select SND_SOC_UDA1380

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[alsa-devel] [PATCH] sound: tlv320aic32x4: Fixed Coding style issues
by ShaRab Shaik 04 Jun '14

04 Jun '14

>From 617d7a25f3c0c08e0b95b24de2d1ea0f3f4b98c7 Mon Sep 17 00:00:00 2001 From: Shahina Shaik <sharab.shaik(a)gmail.com> Date: Wed, 4 Jun 2014 16:27:19 +0530 Subject: [PATCH] sound: tlv320aic32x4: Fixed Coding style issues. Fixed brace coding style, lines morethan 80 characters issues. Replaced printk with pr_err as pr_err is prefered than printk. Signed-off-by: Shahina Shaik <sharab.shaik(a)gmail.com> --- sound/soc/codecs/tlv320aic32x4.c | 34 +++++++++++++++++++--------------- 1 files changed, 19 insertions(+), 15 deletions(-) diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 1d9b117..9e45490 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -306,7 +306,7 @@ static inline int aic32x4_get_divs(int mclk, int rate) return i; } } - printk(KERN_ERR "aic32x4: master clock and sample rate is not supported\n"); + pr_err(KERN_ERR "aic32x4: master clock and sample rate is not supported\n"); return -EINVAL; } @@ -323,7 +323,7 @@ static int aic32x4_set_dai_sysclk(struct snd_soc_dai *codec_dai, aic32x4->sysclk = freq; return 0; } - printk(KERN_ERR "aic32x4: invalid frequency to set DAI system clock\n"); + pr_err(KERN_ERR "aic32x4: invalid frequency to set DAI system clock\n"); return -EINVAL; } @@ -349,7 +349,7 @@ static int aic32x4_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) case SND_SOC_DAIFMT_CBS_CFS: break; default: - printk(KERN_ERR "aic32x4: invalid DAI master/slave interface\n"); + pr_err(KERN_ERR "aic32x4: invalid DAI master/slave interface\n"); return -EINVAL; } @@ -374,7 +374,7 @@ static int aic32x4_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) (AIC32X4_LEFT_JUSTIFIED_MODE << AIC32X4_PLLJ_SHIFT); break; default: - printk(KERN_ERR "aic32x4: invalid DAI interface format\n"); + pr_err(KERN_ERR "aic32x4: invalid DAI interface format\n"); return -EINVAL; } @@ -395,7 +395,7 @@ static int aic32x4_hw_params(struct snd_pcm_substream *substream, i = aic32x4_get_divs(aic32x4->sysclk, params_rate(params)); if (i < 0) { - printk(KERN_ERR "aic32x4: sampling rate not supported\n"); + pr_err(KERN_ERR "aic32x4: sampling rate not supported\n"); return i; } @@ -626,32 +626,36 @@ static int aic32x4_probe(struct snd_soc_codec *codec) snd_soc_write(codec, AIC32X4_MICBIAS, AIC32X4_MICBIAS_LDOIN | AIC32X4_MICBIAS_2075V); } - if (aic32x4->power_cfg & AIC32X4_PWR_AVDD_DVDD_WEAK_DISABLE) { + if (aic32x4->power_cfg & AIC32X4_PWR_AVDD_DVDD_WEAK_DISABLE) snd_soc_write(codec, AIC32X4_PWRCFG, AIC32X4_AVDDWEAKDISABLE); - } + tmp_reg = (aic32x4->power_cfg & AIC32X4_PWR_AIC32X4_LDO_ENABLE) ? AIC32X4_LDOCTLEN : 0; snd_soc_write(codec, AIC32X4_LDOCTL, tmp_reg); tmp_reg = snd_soc_read(codec, AIC32X4_CMMODE); - if (aic32x4->power_cfg & AIC32X4_PWR_CMMODE_LDOIN_RANGE_18_36) { + if (aic32x4->power_cfg & AIC32X4_PWR_CMMODE_LDOIN_RANGE_18_36) tmp_reg |= AIC32X4_LDOIN_18_36; - } - if (aic32x4->power_cfg & AIC32X4_PWR_CMMODE_HP_LDOIN_POWERED) { + + if (aic32x4->power_cfg & AIC32X4_PWR_CMMODE_HP_LDOIN_POWERED) tmp_reg |= AIC32X4_LDOIN2HP; - } + snd_soc_write(codec, AIC32X4_CMMODE, tmp_reg); /* Mic PGA routing */ if (aic32x4->micpga_routing & AIC32X4_MICPGA_ROUTE_LMIC_IN2R_10K) - snd_soc_write(codec, AIC32X4_LMICPGANIN, AIC32X4_LMICPGANIN_IN2R_10K); + snd_soc_write(codec, AIC32X4_LMICPGANIN, + AIC32X4_LMICPGANIN_IN2R_10K); else - snd_soc_write(codec, AIC32X4_LMICPGANIN, AIC32X4_LMICPGANIN_CM1L_10K); + snd_soc_write(codec, AIC32X4_LMICPGANIN, + AIC32X4_LMICPGANIN_CM1L_10K); if (aic32x4->micpga_routing & AIC32X4_MICPGA_ROUTE_RMIC_IN1L_10K) - snd_soc_write(codec, AIC32X4_RMICPGANIN, AIC32X4_RMICPGANIN_IN1L_10K); + snd_soc_write(codec, AIC32X4_RMICPGANIN, + AIC32X4_RMICPGANIN_IN1L_10K); else - snd_soc_write(codec, AIC32X4_RMICPGANIN, AIC32X4_RMICPGANIN_CM1R_10K); + snd_soc_write(codec, AIC32X4_RMICPGANIN, + AIC32X4_RMICPGANIN_CM1R_10K); aic32x4_set_bias_level(codec, SND_SOC_BIAS_STANDBY);

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[alsa-devel] [PATCH] ASoC: samsung: Add I2C dependency for snow
by Arnd Bergmann 04 Jun '14

04 Jun '14

Both codecs used by snow, max98090 and max98095 require the use of I2C, so we can only select this driver if I2C is there, otherwise we get a build error like: codecs/max98090.c:2494:1: warning: data definition has no type or storage class [enabled by default] module_i2c_driver(max98090_i2c_driver); ^ codecs/max98095.c:2443:1: warning: data definition has no type or storage class [enabled by default] module_i2c_driver(max98095_i2c_driver); ^ This adds one more I2C dependency to the hundreds we already have. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Cc: Tushar Behera <tushar.behera(a)linaro.org> --- diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index a7cb8f6..cb7be4f 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -232,7 +232,7 @@ config SND_SOC_LITTLEMILL config SND_SOC_SNOW tristate "Audio support for Google Snow boards" - depends on SND_SOC_SAMSUNG + depends on SND_SOC_SAMSUNG && I2C select SND_SOC_MAX98090 select SND_SOC_MAX98095 select SND_SAMSUNG_I2S

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[alsa-devel] [PATCH v2] ASoC: arizona: Implement TDM support for Arizona devices
by Charles Keepax 04 Jun '14

04 Jun '14

Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- Changes since v1: - Removed check of TDM width to requested stream word length include/linux/mfd/arizona/core.h | 3 + sound/soc/codecs/arizona.c | 78 +++++++++++++++++++++++++++++++++++--- 2 files changed, 75 insertions(+), 6 deletions(-) diff --git a/include/linux/mfd/arizona/core.h b/include/linux/mfd/arizona/core.h index 5cf8b91..11783b5 100644 --- a/include/linux/mfd/arizona/core.h +++ b/include/linux/mfd/arizona/core.h @@ -110,6 +110,9 @@ struct arizona { int clk32k_ref; struct snd_soc_dapm_context *dapm; + + int tdm_width[ARIZONA_MAX_AIF]; + int tdm_slots[ARIZONA_MAX_AIF]; }; int arizona_clk32k_enable(struct arizona *arizona); diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 29e198f..e77f61c 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1185,7 +1185,10 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, int base = dai->driver->base; const int *rates; int i, ret, val; + int channels = params_channels(params); int chan_limit = arizona->pdata.max_channels_clocked[dai->id - 1]; + int tdm_width = arizona->tdm_width[dai->id - 1]; + int tdm_slots = arizona->tdm_slots[dai->id - 1]; int bclk, lrclk, wl, frame, bclk_target; if (params_rate(params) % 8000) @@ -1193,18 +1196,27 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, else rates = &arizona_48k_bclk_rates[0]; - bclk_target = snd_soc_params_to_bclk(params); - if (chan_limit && chan_limit < params_channels(params)) { + if (tdm_slots) { + arizona_aif_dbg(dai, "Configuring for %d %d bit TDM slots\n", + tdm_slots, tdm_width); + bclk_target = tdm_slots * tdm_width * params_rate(params); + channels = tdm_slots; + } else { + bclk_target = snd_soc_params_to_bclk(params); + } + + if (chan_limit && chan_limit < channels) { arizona_aif_dbg(dai, "Limiting to %d channels\n", chan_limit); - bclk_target /= params_channels(params); + bclk_target /= channels; bclk_target *= chan_limit; } - /* Force stereo for I2S mode */ + /* Force multiple of 2 channels for I2S mode */ val = snd_soc_read(codec, base + ARIZONA_AIF_FORMAT); - if (params_channels(params) == 1 && (val & ARIZONA_AIF1_FMT_MASK)) { + if ((channels & 1) && (val & ARIZONA_AIF1_FMT_MASK)) { arizona_aif_dbg(dai, "Forcing stereo mode\n"); - bclk_target *= 2; + bclk_target /= channels; + bclk_target *= channels + 1; } for (i = 0; i < ARRAY_SIZE(arizona_44k1_bclk_rates); i++) { @@ -1324,9 +1336,63 @@ static int arizona_set_tristate(struct snd_soc_dai *dai, int tristate) ARIZONA_AIF1_TRI, reg); } +static void arizona_set_channels_to_mask(struct snd_soc_dai *dai, + unsigned int base, + int channels, unsigned int mask) +{ + struct snd_soc_codec *codec = dai->codec; + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + struct arizona *arizona = priv->arizona; + int slot, i; + + for (i = 0; i < channels; ++i) { + slot = ffs(mask) - 1; + if (slot < 0) + return; + + regmap_write(arizona->regmap, base + i, slot); + + mask &= ~(1 << slot); + } + + if (mask) + arizona_aif_warn(dai, "Too many channels in TDM mask\n"); +} + +static int arizona_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, + unsigned int rx_mask, int slots, int slot_width) +{ + struct snd_soc_codec *codec = dai->codec; + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + struct arizona *arizona = priv->arizona; + int base = dai->driver->base; + int rx_max_chan = dai->driver->playback.channels_max; + int tx_max_chan = dai->driver->capture.channels_max; + + /* Only support TDM for the physical AIFs */ + if (dai->id > ARIZONA_MAX_AIF) + return -ENOTSUPP; + + if (slots == 0) { + tx_mask = (1 << tx_max_chan) - 1; + rx_mask = (1 << rx_max_chan) - 1; + } + + arizona_set_channels_to_mask(dai, base + ARIZONA_AIF_FRAME_CTRL_3, + tx_max_chan, tx_mask); + arizona_set_channels_to_mask(dai, base + ARIZONA_AIF_FRAME_CTRL_11, + rx_max_chan, rx_mask); + + arizona->tdm_width[dai->id - 1] = slot_width; + arizona->tdm_slots[dai->id - 1] = slots; + + return 0; +} + const struct snd_soc_dai_ops arizona_dai_ops = { .startup = arizona_startup, .set_fmt = arizona_set_fmt, + .set_tdm_slot = arizona_set_tdm_slot, .hw_params = arizona_hw_params, .set_sysclk = arizona_dai_set_sysclk, .set_tristate = arizona_set_tristate, -- 1.7.2.5

2 1

[alsa-devel] [PATCH v2 1/1] ALSA: seq: Continue broadcasting events to ports if one of them fails
by Adam Goode 04 Jun '14

04 Jun '14

Sometimes PORT_EXIT messages are lost when a process is exiting. This happens if you subscribe to the announce port with client A, then subscribe to the announce port with client B, then kill client A. Client B will not see the PORT_EXIT message because client A's port is closing and is earlier in the announce port subscription list. The for each loop will try to send the announcement to client A and fail, then will stop trying to broadcast to other ports. Killing B works fine since the announcement will already have gone to A. The CLIENT_EXIT message does not get lost. How to reproduce problem: *** termA $ aseqdump -p 0:1 0:1 Port subscribed 0:1 -> 128:0 *** termB $ aseqdump -p 0:1 *** termA 0:1 Client start client 129 0:1 Port start 129:0 0:1 Port subscribed 0:1 -> 129:0 *** termB 0:1 Port subscribed 0:1 -> 129:0 *** termA ^C *** termB 0:1 Client exit client 128 <--- expected Port exit as well (before client exit) Signed-off-by: Adam Goode <agoode(a)google.com> diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c index 9ca5e64..225c7315 100644 --- a/sound/core/seq/seq_clientmgr.c +++ b/sound/core/seq/seq_clientmgr.c @@ -660,7 +660,7 @@ static int deliver_to_subscribers(struct snd_seq_client *client, int atomic, int hop) { struct snd_seq_subscribers *subs; - int err = 0, num_ev = 0; + int err, result = 0, num_ev = 0; struct snd_seq_event event_saved; struct snd_seq_client_port *src_port; struct snd_seq_port_subs_info *grp; @@ -685,8 +685,12 @@ static int deliver_to_subscribers(struct snd_seq_client *client, subs->info.flags & SNDRV_SEQ_PORT_SUBS_TIME_REAL); err = snd_seq_deliver_single_event(client, event, 0, atomic, hop); - if (err < 0) - break; + if (err < 0) { + /* save first error that occurs and continue */ + if (!result) + result = err; + continue; + } num_ev++; /* restore original event record */ *event = event_saved; @@ -697,7 +701,7 @@ static int deliver_to_subscribers(struct snd_seq_client *client, up_read(&grp->list_mutex); *event = event_saved; /* restore */ snd_seq_port_unlock(src_port); - return (err < 0) ? err : num_ev; + return (result < 0) ? result : num_ev; } @@ -709,7 +713,7 @@ static int port_broadcast_event(struct snd_seq_client *client, struct snd_seq_event *event, int atomic, int hop) { - int num_ev = 0, err = 0; + int num_ev = 0, err, result = 0; struct snd_seq_client *dest_client; struct snd_seq_client_port *port; @@ -724,14 +728,18 @@ static int port_broadcast_event(struct snd_seq_client *client, err = snd_seq_deliver_single_event(NULL, event, SNDRV_SEQ_FILTER_BROADCAST, atomic, hop); - if (err < 0) - break; + if (err < 0) { + /* save first error that occurs and continue */ + if (!result) + result = err; + continue; + } num_ev++; } read_unlock(&dest_client->ports_lock); snd_seq_client_unlock(dest_client); event->dest.port = SNDRV_SEQ_ADDRESS_BROADCAST; /* restore */ - return (err < 0) ? err : num_ev; + return (result < 0) ? result : num_ev; } /* @@ -741,7 +749,7 @@ static int port_broadcast_event(struct snd_seq_client *client, static int broadcast_event(struct snd_seq_client *client, struct snd_seq_event *event, int atomic, int hop) { - int err = 0, num_ev = 0; + int err, result = 0, num_ev = 0; int dest; struct snd_seq_addr addr; @@ -760,12 +768,16 @@ static int broadcast_event(struct snd_seq_client *client, err = snd_seq_deliver_single_event(NULL, event, SNDRV_SEQ_FILTER_BROADCAST, atomic, hop); - if (err < 0) - break; + if (err < 0) { + /* save first error that occurs and continue */ + if (!result) + result = err; + continue; + } num_ev += err; } event->dest = addr; /* restore */ - return (err < 0) ? err : num_ev; + return (result < 0) ? result : num_ev; } -- 2.0.0.526.g5318336

2 1

[alsa-devel] Correction for comment about firewire stack for tag 'sound-3.16-rc1'
by Takashi Sakamoto 04 Jun '14

04 Jun '14

Iwai-san, I find a typo in the comment. If it's impossible to correct it, please ignore this message. > FireWire-audio: > - FireWire audio stack enhancement (AMDTP, MIDI) > - BeBbB-based devices support > - Fireworks-based device support s/BeBbB/BeBoB/ ~ ~ You can see "what's BeBoB" in sound/firewire/bebob/bebob.c:10. I use BeBoB as a name of module, names of directory/files and Kconfig so it's better to correct this typo. And I strongly hope to describe a support for 'incoming isochronous stream' and 'duplex streams with timestamp synchronization'. They're main mechanisms which BeBoB/Fireworks need but previous stack didn't have. If it's too detailed for comment of tag, please keep it what it is. Regards Takashi Sakamoto o-takashi(a)sakamocchi.jp

2 1

[alsa-devel] [PATCH 0/6] Fix some minor issues for Fireworks/BeBoB drivers
by Takashi Sakamoto 04 Jun '14

04 Jun '14

Remove some unused or meaningless functions and improve some codes and comments. Takashi Sakamoto (6): fireworks/bebob: Shorten critical section for stream_stop_duplex() fireworks: Use safer way to arrange ring buffer pointer fireworks: Improve comments about Fireworks transaction fireworks: Remove a constant over width to which it's applied fireworks: Remove meaningless mutex_unlock() bebob: Remove unused function prototype sound/firewire/bebob/bebob.h | 2 -- sound/firewire/bebob/bebob_stream.c | 4 ++-- sound/firewire/fireworks/fireworks.c | 1 - sound/firewire/fireworks/fireworks.h | 1 - sound/firewire/fireworks/fireworks_hwdep.c | 2 +- sound/firewire/fireworks/fireworks_stream.c | 4 ++-- sound/firewire/fireworks/fireworks_transaction.c | 18 +++++++++--------- 7 files changed, 14 insertions(+), 18 deletions(-) -- 1.8.3.2

2 13

[alsa-devel] [PATCH v6] ASoC: add RT286 CODEC driver
by bardliao＠realtek.com 04 Jun '14

04 Jun '14

From: Bard Liao <bardliao(a)realtek.com> ALC286 is a dual mode codec, which can run as HD-A or I2S mode. It is controlled by HD-A verb commands via I2C protocol. The following is the I/O difference between ALC286 and general I2S codecs. 1. A HD-A verb command contains three parts, NID, VID, and PID. And an I2S command contains only two parts: address and data. 2. Not only the register address is written, but the read command also includes the entire write command. 3. rt286 uses different registers for read and write the same bits. As a result, regmap is difficult to be used on ALC286. We don't request a standard I/O by snd_soc_codec_set_cache_io anymore. Now we have specific write and read functions for ALC286's I/O. A verb command contains three part: NID, VID, and PID. We combine NID and VID as reg address, and PID as reg value. And do corresponding change in rt286_hw_read to get the data properly. Signed-off-by: Bard Liao <bardliao(a)realtek.com> Signed-off-by: Gustaw Lewandowski <gustaw.lewandowski(a)intel.com> --- The jack detection function is done by Lewandowski Gustaw. So I add Gustaw's Signed-off-by line in this patch The difference between this version and previous version is listed below. * Implement specific read/write functions rather than use regmap. * Use dapm widget to control power bit instead of setting it in _event * Add device ID check Verb command use two bit to set power state(D0, D1, D2, D3). But dapm widget use only one bit to set power up/down. So we use dapm widget to switch each node between D0 and D1 power state. And will set audio group to D3 when bias level = STANDBY. --- include/sound/rt286.h | 19 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rt286.c | 1054 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt286.h | 186 ++++++++ 5 files changed, 1265 insertions(+) create mode 100644 include/sound/rt286.h create mode 100644 sound/soc/codecs/rt286.c create mode 100644 sound/soc/codecs/rt286.h diff --git a/include/sound/rt286.h b/include/sound/rt286.h new file mode 100644 index 0000000..eb773d1 --- /dev/null +++ b/include/sound/rt286.h @@ -0,0 +1,19 @@ +/* + * linux/sound/rt286.h -- Platform data for RT286 + * + * Copyright 2013 Realtek Microelectronics + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __LINUX_SND_RT286_H +#define __LINUX_SND_RT286_H + +struct rt286_platform_data { + bool cbj_en; /*combo jack enable*/ + bool gpio2_en; /*GPIO2 enable*/ +}; + +#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index f0e8401..5818391 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -69,6 +69,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_PCM3008 select SND_SOC_PCM512x_I2C if I2C select SND_SOC_PCM512x_SPI if SPI_MASTER + select SND_SOC_RT286 if I2C select SND_SOC_RT5631 if I2C select SND_SOC_RT5640 if I2C select SND_SOC_SGTL5000 if I2C @@ -390,6 +391,9 @@ config SND_SOC_PCM512x_SPI select SND_SOC_PCM512x select REGMAP_SPI +config SND_SOC_RT286 + tristate + config SND_SOC_RT5631 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 3c4d275..5ef8adc 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -58,6 +58,7 @@ snd-soc-pcm3008-objs := pcm3008.o snd-soc-pcm512x-objs := pcm512x.o snd-soc-pcm512x-i2c-objs := pcm512x-i2c.o snd-soc-pcm512x-spi-objs := pcm512x-spi.o +snd-soc-rt286-objs := rt286.o snd-soc-rt5631-objs := rt5631.o snd-soc-rt5640-objs := rt5640.o snd-soc-sgtl5000-objs := sgtl5000.o @@ -209,6 +210,7 @@ obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_PCM512x) += snd-soc-pcm512x.o obj-$(CONFIG_SND_SOC_PCM512x_I2C) += snd-soc-pcm512x-i2c.o obj-$(CONFIG_SND_SOC_PCM512x_SPI) += snd-soc-pcm512x-spi.o +obj-$(CONFIG_SND_SOC_RT286) += snd-soc-rt286.o obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o obj-$(CONFIG_SND_SOC_RT5640) += snd-soc-rt5640.o obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c new file mode 100644 index 0000000..a88b3500 --- /dev/null +++ b/sound/soc/codecs/rt286.c @@ -0,0 +1,1054 @@ +/* + * rt286.c -- RT286 ALSA SoC audio codec driver + * + * Copyright 2013 Realtek Semiconductor Corp. + * Author: Bard Liao <bardliao(a)realtek.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/spi/spi.h> +#include <linux/acpi.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <sound/jack.h> +#include <linux/workqueue.h> +#include <sound/rt286.h> +#include <sound/hda_verbs.h> + +#include "rt286.h" + +#define RT286_VENDOR_ID 0x10ec0286 + +#define RT286_INDEX_REG_BASE 0x100 +#define RT286_INDEX_REG_RANGE 0xff +#define RT286_INDEX_REG_MAX (RT286_INDEX_REG_BASE + RT286_INDEX_REG_RANGE) + +struct rt286_priv { + struct snd_soc_codec *codec; + struct rt286_platform_data pdata; + struct i2c_client *i2c; + struct snd_soc_jack *jack; + struct delayed_work jack_detect_work; + int sys_clk; +}; + +static unsigned int rt286_hw_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + struct i2c_client *client = rt286->i2c; + struct i2c_msg xfer[2]; + int ret; + __be32 be_reg; + unsigned int index, vid, buf = 0x0; + + /*index registers*/ + if (reg >= RT286_INDEX_REG_BASE && reg < RT286_INDEX_REG_MAX) { + ret = snd_soc_write(codec, + VERB_CMD(AC_VERB_SET_COEF_INDEX, + RT286_VENDOR_REGISTERS, 0), + reg - RT286_INDEX_REG_BASE); + + if (ret < 0) { + dev_err(codec->dev, + "Failed to set private addr: %d\n", ret); + return ret; + } + + reg = VERB_CMD(AC_VERB_GET_PROC_COEF, + RT286_VENDOR_REGISTERS, 0); + } + reg = reg | 0x80000; + vid = (reg >> 8) & 0xfff; + + if (AC_VERB_GET_AMP_GAIN_MUTE == (vid & 0xf00)) { + index = (reg >> 8) & 0xf; + reg = (reg & ~0xf0f) | index; + } + be_reg = cpu_to_be32(reg); + + /* Write register */ + xfer[0].addr = client->addr; + xfer[0].flags = 0; + xfer[0].len = 4; + xfer[0].buf = (u8 *)&be_reg; + + /* Read data */ + xfer[1].addr = client->addr; + xfer[1].flags = I2C_M_RD; + xfer[1].len = 4; + xfer[1].buf = (u8 *)&buf; + + ret = i2c_transfer(client->adapter, xfer, 2); + if (ret < 0) + return ret; + else if (ret != 2) + return -EIO; + + return be32_to_cpu(buf); +} + +static int rt286_hw_write(struct snd_soc_codec *codec, + unsigned int reg, unsigned int value) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + struct i2c_client *client = rt286->i2c; + u8 data[4]; + int ret; + + /*index registers*/ + if (reg >= RT286_INDEX_REG_BASE && reg < RT286_INDEX_REG_MAX) { + ret = snd_soc_write(codec, + VERB_CMD(AC_VERB_SET_COEF_INDEX, + RT286_VENDOR_REGISTERS, 0), + reg - RT286_INDEX_REG_BASE); + if (ret < 0) { + dev_err(codec->dev, + "Failed to set private addr: %d\n", ret); + return -EIO; + } + reg = VERB_CMD(AC_VERB_SET_PROC_COEF, + RT286_VENDOR_REGISTERS, 0); + } + + data[0] = (reg >> 24) & 0xff; + data[1] = (reg >> 16) & 0xff; + /* + * 4 bit VID: reg should be 0 + * 12 bit VID: value should be 0 + * So we use an OR operator to handle it rather than use if condition. + */ + data[2] = ((reg >> 8) & 0xff) | ((value >> 8) & 0xff); + data[3] = value & 0xff; + + ret = i2c_master_send(client, data, 4); + if (ret == 4) + return 0; + if (ret < 0) + return ret; + else + return -EIO; +} + +static int rt286_support_power_controls[] = { + RT286_DAC_OUT1, + RT286_DAC_OUT2, + RT286_ADC_IN1, + RT286_ADC_IN2, + RT286_MIC1, + RT286_DMIC1, + RT286_DMIC2, + RT286_SPK_OUT, + RT286_HP_OUT, +}; +#define RT286_POWER_REG_LEN ARRAY_SIZE(rt286_support_power_controls) + +static int rt286_jack_detect(struct snd_soc_codec *codec, bool *hp, bool *mic) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + unsigned int val, buf; + int i; + + *hp = false; + *mic = false; + + if (rt286->pdata.cbj_en) { + buf = snd_soc_read(codec, RT286_GET_HP_SENSE); + *hp = buf & 0x80000000; + if (*hp) { + /* power on HV,VERF */ + snd_soc_update_bits(codec, RT286_INDEX_REG_BASE + + RT286_POWER_CTRL1, 0x1001, 0x0); + /* power LDO1 */ + snd_soc_update_bits(codec, RT286_INDEX_REG_BASE + + RT286_POWER_CTRL2, 0x4, 0x4); + snd_soc_write(codec, RT286_SET_MIC1, 0x24); + val = snd_soc_read(codec, RT286_INDEX_REG_BASE + + RT286_CBJ_CTRL2); + + msleep(200); + i = 40; + while (((val & 0x0800) == 0) && (i > 0)) { + val = snd_soc_read(codec, + RT286_INDEX_REG_BASE + + RT286_CBJ_CTRL2); + i--; + msleep(20); + } + + if (0x0400 == (val & 0x0700)) { + *mic = false; + + snd_soc_write(codec, + RT286_SET_MIC1, 0x20); + /* power off HV,VERF */ + snd_soc_update_bits(codec, + RT286_INDEX_REG_BASE + + RT286_POWER_CTRL1, 0x1001, 0x1001); + snd_soc_update_bits(codec, + RT286_INDEX_REG_BASE + + RT286_A_BIAS_CTRL3, 0xc000, 0x0000); + snd_soc_update_bits(codec, + RT286_INDEX_REG_BASE + + RT286_CBJ_CTRL1, 0x0030, 0x0000); + snd_soc_update_bits(codec, + RT286_INDEX_REG_BASE + + RT286_A_BIAS_CTRL2, 0xc000, 0x0000); + } else if ((0x0200 == (val & 0x0700)) || + (0x0100 == (val & 0x0700))) { + *mic = true; + snd_soc_update_bits(codec, + RT286_INDEX_REG_BASE + + RT286_A_BIAS_CTRL3, 0xc000, 0x8000); + snd_soc_update_bits(codec, + RT286_INDEX_REG_BASE + + RT286_CBJ_CTRL1, 0x0030, 0x0020); + snd_soc_update_bits(codec, + RT286_INDEX_REG_BASE + + RT286_A_BIAS_CTRL2, 0xc000, 0x8000); + } else { + *mic = false; + } + + snd_soc_update_bits(codec, RT286_INDEX_REG_BASE + + RT286_MISC_CTRL1, + 0x0060, 0x0000); + } else { + snd_soc_update_bits(codec, RT286_INDEX_REG_BASE + + RT286_MISC_CTRL1, + 0x0060, 0x0020); + snd_soc_update_bits(codec, RT286_INDEX_REG_BASE + + RT286_A_BIAS_CTRL3, + 0xc000, 0x8000); + snd_soc_update_bits(codec, RT286_INDEX_REG_BASE + + RT286_CBJ_CTRL1, + 0x0030, 0x0020); + snd_soc_update_bits(codec, RT286_INDEX_REG_BASE + + RT286_A_BIAS_CTRL2, + 0xc000, 0x8000); + + *mic = false; + } + } else { + buf = snd_soc_read(codec, RT286_GET_HP_SENSE); + *hp = buf & 0x80000000; + buf = snd_soc_read(codec, RT286_GET_MIC1_SENSE); + *mic = buf & 0x80000000; + } + + return 0; +} + +static void rt286_jack_detect_work(struct work_struct *work) +{ + struct rt286_priv *rt286 = + container_of(work, struct rt286_priv, jack_detect_work.work); + int status = 0; + + bool hp = false; + bool mic = false; + + rt286_jack_detect(rt286->codec, &hp, &mic); + + if (hp == true) + status |= SND_JACK_HEADPHONE; + + if (mic == true) + status |= SND_JACK_MICROPHONE; + + snd_soc_jack_report(rt286->jack, status, + SND_JACK_MICROPHONE | SND_JACK_HEADPHONE); +} + +int rt286_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + rt286->jack = jack; + + /* Send an initial empty report */ + snd_soc_jack_report(rt286->jack, 0, + SND_JACK_MICROPHONE | SND_JACK_HEADPHONE); + + return 0; +} +EXPORT_SYMBOL_GPL(rt286_mic_detect); + +static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -6350, 50, 0); +static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, 0, 1000, 0); + + + +static const struct snd_kcontrol_new rt286_snd_controls[] = { + SOC_DOUBLE_R_TLV("DAC0 Playback Volume", RT286_DACL_GAIN, + RT286_DACR_GAIN, 0, 0x7f, 0, out_vol_tlv), + SOC_DOUBLE_R_TLV("ADC0 Capture Volume", RT286_ADCL_GAIN, + RT286_ADCR_GAIN, 0, 0x7f, 0, out_vol_tlv), + SOC_SINGLE_TLV("AMIC Volume", RT286_MIC_GAIN, + 0, 0x3, 0, mic_vol_tlv), + SOC_DOUBLE_R("Speaker Playback Switch", RT286_SPOL_GAIN, + RT286_SPOR_GAIN, RT286_MUTE_SFT, 1, 1), +}; + +/* Digital Mixer */ +static const struct snd_kcontrol_new rt286_front_mix[] = { + SOC_DAPM_SINGLE("DAC Switch", RT286_F_DAC_SWITCH, + RT286_MUTE_SFT, 1, 1), + SOC_DAPM_SINGLE("RECMIX Switch", RT286_F_RECMIX_SWITCH, + RT286_MUTE_SFT, 1, 1), +}; + +/* Analog Input Mixer */ +static const struct snd_kcontrol_new rt286_rec_mix[] = { + SOC_DAPM_SINGLE("Mic1 Switch", RT286_REC_MIC_SWITCH, + RT286_MUTE_SFT, 1, 1), + SOC_DAPM_SINGLE("I2S Switch", RT286_REC_I2S_SWITCH, + RT286_MUTE_SFT, 1, 1), + SOC_DAPM_SINGLE("Line1 Switch", RT286_REC_LINE_SWITCH, + RT286_MUTE_SFT, 1, 1), + SOC_DAPM_SINGLE("Beep Switch", RT286_REC_BEEP_SWITCH, + RT286_MUTE_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new spo_enable_control = + SOC_DAPM_SINGLE("Switch", RT286_SET_PIN_SPK, + RT286_SET_PIN_SFT, 1, 0); + +static const struct snd_kcontrol_new hpol_enable_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT286_HPOL_GAIN, + RT286_MUTE_SFT, 1, 1); + +static const struct snd_kcontrol_new hpor_enable_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT286_HPOR_GAIN, + RT286_MUTE_SFT, 1, 1); + +/* ADC0 source */ +static const char * const rt286_adc_src[] = { + "Mic", "RECMIX", "Dmic" +}; + +static const int rt286_adc_values[] = { + 0, 4, 5, +}; + +static SOC_VALUE_ENUM_SINGLE_DECL( + rt286_adc0_enum, RT286_ADC0_MUX, RT286_ADC_SEL_SFT, + RT286_ADC_SEL_MASK, rt286_adc_src, rt286_adc_values); + +static const struct snd_kcontrol_new rt286_adc0_mux = + SOC_DAPM_VALUE_ENUM("ADC 0 source", rt286_adc0_enum); + +static SOC_VALUE_ENUM_SINGLE_DECL( + rt286_adc1_enum, RT286_ADC1_MUX, RT286_ADC_SEL_SFT, + RT286_ADC_SEL_MASK, rt286_adc_src, rt286_adc_values); + +static const struct snd_kcontrol_new rt286_adc1_mux = + SOC_DAPM_VALUE_ENUM("ADC 1 source", rt286_adc1_enum); + +static const char * const rt286_dac_src[] = { + "Front", "Surround" +}; +/* HP-OUT source */ +static SOC_ENUM_SINGLE_DECL(rt286_hpo_enum, RT286_HPO_MUX, + 0, rt286_dac_src); + +static const struct snd_kcontrol_new rt286_hpo_mux = +SOC_DAPM_ENUM("HPO source", rt286_hpo_enum); + +/* SPK-OUT source */ +static SOC_ENUM_SINGLE_DECL(rt286_spo_enum, RT286_SPK_MUX, + 0, rt286_dac_src); + +static const struct snd_kcontrol_new rt286_spo_mux = +SOC_DAPM_ENUM("SPO source", rt286_spo_enum); + +static int rt286_spk_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_write(codec, + RT286_SPK_EAPD, RT286_SET_EAPD_HIGH); + break; + case SND_SOC_DAPM_PRE_PMD: + snd_soc_write(codec, + RT286_SPK_EAPD, RT286_SET_EAPD_LOW); + break; + + default: + return 0; + } + + return 0; +} + +static int rt286_set_dmic1_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_write(codec, RT286_SET_PIN_DMIC1, 0x20); + break; + case SND_SOC_DAPM_PRE_PMD: + snd_soc_write(codec, RT286_SET_PIN_DMIC1, 0); + break; + default: + return 0; + } + + return 0; +} + +static int rt286_adc_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + unsigned int nid; + + nid = (w->reg >> 20) & 0xff; + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_update_bits(codec, + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, nid, 0), + 0x7080, 0x7000); + break; + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, nid, 0), + 0x7080, 0x7080); + break; + default: + return 0; + } + + return 0; +} + +static const struct snd_soc_dapm_widget rt286_dapm_widgets[] = { + /* Input Lines */ + SND_SOC_DAPM_INPUT("DMIC1 Pin"), + SND_SOC_DAPM_INPUT("DMIC2 Pin"), + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_INPUT("LINE1"), + SND_SOC_DAPM_INPUT("Beep"), + + /* DMIC */ + SND_SOC_DAPM_PGA_E("DMIC1", RT286_SET_POWER(RT286_DMIC1), 0, 1, + NULL, 0, rt286_set_dmic1_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA("DMIC2", RT286_SET_POWER(RT286_DMIC2), 0, 1, + NULL, 0), + SND_SOC_DAPM_SUPPLY("DMIC Receiver", SND_SOC_NOPM, + 0, 0, NULL, 0), + + /* REC Mixer */ + SND_SOC_DAPM_MIXER("RECMIX", SND_SOC_NOPM, 0, 0, + rt286_rec_mix, ARRAY_SIZE(rt286_rec_mix)), + + /* ADCs */ + SND_SOC_DAPM_ADC("ADC 0", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("ADC 1", NULL, SND_SOC_NOPM, 0, 0), + + /* ADC Mux */ + SND_SOC_DAPM_MUX_E("ADC 0 Mux", RT286_SET_POWER(RT286_ADC_IN1), 0, 1, + &rt286_adc0_mux, rt286_adc_event, SND_SOC_DAPM_PRE_PMD | + SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_MUX_E("ADC 1 Mux", RT286_SET_POWER(RT286_ADC_IN2), 0, 1, + &rt286_adc1_mux, rt286_adc_event, SND_SOC_DAPM_PRE_PMD | + SND_SOC_DAPM_POST_PMU), + + /* Audio Interface */ + SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIF2RX", "AIF2 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF2TX", "AIF2 Capture", 0, SND_SOC_NOPM, 0, 0), + + /* Output Side */ + /* DACs */ + SND_SOC_DAPM_DAC("DAC 0", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("DAC 1", NULL, SND_SOC_NOPM, 0, 0), + + /* Output Mux */ + SND_SOC_DAPM_MUX("SPK Mux", SND_SOC_NOPM, 0, 0, &rt286_spo_mux), + SND_SOC_DAPM_MUX("HPO Mux", SND_SOC_NOPM, 0, 0, &rt286_hpo_mux), + + SND_SOC_DAPM_SUPPLY("HP Power", RT286_SET_PIN_HPO, + RT286_SET_PIN_SFT, 0, NULL, 0), + + /* Output Mixer */ + SND_SOC_DAPM_MIXER("Front", RT286_SET_POWER(RT286_DAC_OUT1), 0, 1, + rt286_front_mix, ARRAY_SIZE(rt286_front_mix)), + SND_SOC_DAPM_PGA("Surround", RT286_SET_POWER(RT286_DAC_OUT2), 0, 1, + NULL, 0), + + /* Output Pga */ + SND_SOC_DAPM_SWITCH_E("SPO", SND_SOC_NOPM, 0, 0, + &spo_enable_control, rt286_spk_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_SWITCH("HPO L", SND_SOC_NOPM, 0, 0, + &hpol_enable_control), + SND_SOC_DAPM_SWITCH("HPO R", SND_SOC_NOPM, 0, 0, + &hpor_enable_control), + + /* Output Lines */ + SND_SOC_DAPM_OUTPUT("SPOL"), + SND_SOC_DAPM_OUTPUT("SPOR"), + SND_SOC_DAPM_OUTPUT("HPO Pin"), + SND_SOC_DAPM_OUTPUT("SPDIF"), +}; + +static const struct snd_soc_dapm_route rt286_dapm_routes[] = { + {"DMIC1", NULL, "DMIC1 Pin"}, + {"DMIC2", NULL, "DMIC2 Pin"}, + {"DMIC1", NULL, "DMIC Receiver"}, + {"DMIC2", NULL, "DMIC Receiver"}, + + {"RECMIX", "Beep Switch", "Beep"}, + {"RECMIX", "Line1 Switch", "LINE1"}, + {"RECMIX", "Mic1 Switch", "MIC1"}, + + {"ADC 0 Mux", "Dmic", "DMIC1"}, + {"ADC 0 Mux", "RECMIX", "RECMIX"}, + {"ADC 0 Mux", "Mic", "MIC1"}, + {"ADC 1 Mux", "Dmic", "DMIC2"}, + {"ADC 1 Mux", "RECMIX", "RECMIX"}, + {"ADC 1 Mux", "Mic", "MIC1"}, + + {"ADC 0", NULL, "ADC 0 Mux"}, + {"ADC 1", NULL, "ADC 1 Mux"}, + + {"AIF1TX", NULL, "ADC 0"}, + {"AIF2TX", NULL, "ADC 1"}, + + {"DAC 0", NULL, "AIF1RX"}, + {"DAC 1", NULL, "AIF2RX"}, + + {"Front", "DAC Switch", "DAC 0"}, + {"Front", "RECMIX Switch", "RECMIX"}, + + {"Surround", NULL, "DAC 1"}, + + {"SPK Mux", "Front", "Front"}, + {"SPK Mux", "Surround", "Surround"}, + + {"HPO Mux", "Front", "Front"}, + {"HPO Mux", "Surround", "Surround"}, + + {"SPO", "Switch", "SPK Mux"}, + {"HPO L", "Switch", "HPO Mux"}, + {"HPO R", "Switch", "HPO Mux"}, + {"HPO L", NULL, "HP Power"}, + {"HPO R", NULL, "HP Power"}, + + {"SPOL", NULL, "SPO"}, + {"SPOR", NULL, "SPO"}, + {"HPO Pin", NULL, "HPO L"}, + {"HPO Pin", NULL, "HPO R"}, +}; + +static int rt286_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + unsigned int val = 0; + int d_len_code; + + switch (params_rate(params)) { + /* bit 14 0:48K 1:44.1K */ + case 44100: + val |= 0x4000; + break; + case 48000: + break; + default: + dev_err(codec->dev, "Unsupported sample rate %d\n", + params_rate(params)); + return -EINVAL; + } + switch (rt286->sys_clk) { + case 12288000: + case 24576000: + if (params_rate(params) != 48000) { + dev_err(codec->dev, "Sys_clk is not matched (%d %d)\n", + params_rate(params), rt286->sys_clk); + return -EINVAL; + } + break; + case 11289600: + case 22579200: + if (params_rate(params) != 44100) { + dev_err(codec->dev, "Sys_clk is not matched (%d %d)\n", + params_rate(params), rt286->sys_clk); + return -EINVAL; + } + break; + } + + if (params_channels(params) <= 16) { + /* bit 3:0 Number of Channel */ + val |= (params_channels(params) - 1); + } else { + dev_err(codec->dev, "Unsupported channels %d\n", + params_channels(params)); + return -EINVAL; + } + + d_len_code = 0; + switch (params_format(params)) { + /* bit 6:4 Bits per Sample */ + case SNDRV_PCM_FORMAT_S16_LE: + d_len_code = 0; + val |= (0x1 << 4); + break; + case SNDRV_PCM_FORMAT_S32_LE: + d_len_code = 2; + val |= (0x4 << 4); + break; + case SNDRV_PCM_FORMAT_S20_3LE: + d_len_code = 1; + val |= (0x2 << 4); + break; + case SNDRV_PCM_FORMAT_S24_LE: + d_len_code = 2; + val |= (0x3 << 4); + break; + case SNDRV_PCM_FORMAT_S8: + break; + default: + return -EINVAL; + } + snd_soc_update_bits(codec, RT286_INDEX_REG_BASE + + RT286_I2S_CTRL1, 0x0018, d_len_code << 3); + dev_dbg(codec->dev, "format val = 0x%x\n", val); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + snd_soc_update_bits(codec, RT286_DAC_FORMAT, 0x407f, val); + else + snd_soc_update_bits(codec, RT286_ADC_FORMAT, 0x407f, val); + + return 0; +} + +static int rt286_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + snd_soc_update_bits(codec, RT286_INDEX_REG_BASE + + RT286_I2S_CTRL1, 0x800, 0x800); + break; + case SND_SOC_DAIFMT_CBS_CFS: + snd_soc_update_bits(codec, RT286_INDEX_REG_BASE + + RT286_I2S_CTRL1, 0x800, 0x0); + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + snd_soc_update_bits(codec, RT286_INDEX_REG_BASE + + RT286_I2S_CTRL1, 0x300, 0x0); + break; + case SND_SOC_DAIFMT_LEFT_J: + snd_soc_update_bits(codec, RT286_INDEX_REG_BASE + + RT286_I2S_CTRL1, 0x300, 0x1 << 8); + break; + case SND_SOC_DAIFMT_DSP_A: + snd_soc_update_bits(codec, RT286_INDEX_REG_BASE + + RT286_I2S_CTRL1, 0x300, 0x2 << 8); + break; + case SND_SOC_DAIFMT_DSP_B: + snd_soc_update_bits(codec, RT286_INDEX_REG_BASE + + RT286_I2S_CTRL1, 0x300, 0x3 << 8); + break; + default: + return -EINVAL; + } + /* bit 15 Stream Type 0:PCM 1:Non-PCM */ + snd_soc_update_bits(codec, RT286_DAC_FORMAT, 0x8000, 0); + snd_soc_update_bits(codec, RT286_ADC_FORMAT, 0x8000, 0); + + return 0; +} + +static int rt286_set_dai_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + + dev_dbg(codec->dev, "%s freq=%d\n", __func__, freq); + + if (RT286_SCLK_S_MCLK == clk_id) { + snd_soc_update_bits(codec, RT286_INDEX_REG_BASE + + RT286_I2S_CTRL2, 0x0100, 0x0); + snd_soc_update_bits(codec, RT286_INDEX_REG_BASE + + RT286_PLL_CTRL1, 0x20, 0x20); + } else { + snd_soc_update_bits(codec, RT286_INDEX_REG_BASE + + RT286_I2S_CTRL2, 0x0100, 0x0100); + snd_soc_update_bits(codec, RT286_INDEX_REG_BASE + + RT286_PLL_CTRL, 0x4, 0x4); + snd_soc_update_bits(codec, RT286_INDEX_REG_BASE + + RT286_PLL_CTRL1, 0x20, 0x0); + } + + switch (freq) { + case 19200000: + if (RT286_SCLK_S_MCLK == clk_id) { + dev_err(codec->dev, "Should not use MCLK\n"); + return -EINVAL; + } + snd_soc_update_bits(codec, RT286_INDEX_REG_BASE + + RT286_I2S_CTRL2, 0x40, 0x40); + break; + case 24000000: + if (RT286_SCLK_S_MCLK == clk_id) { + dev_err(codec->dev, "Should not use MCLK\n"); + return -EINVAL; + } + snd_soc_update_bits(codec, RT286_INDEX_REG_BASE + + RT286_I2S_CTRL2, 0x40, 0x0); + break; + case 12288000: + case 11289600: + snd_soc_update_bits(codec, RT286_INDEX_REG_BASE + + RT286_I2S_CTRL2, 0x8, 0x0); + snd_soc_update_bits(codec, RT286_INDEX_REG_BASE + + RT286_CLK_DIV, 0xfc1e, 0x0004); + break; + case 24576000: + case 22579200: + snd_soc_update_bits(codec, RT286_INDEX_REG_BASE + + RT286_I2S_CTRL2, 0x8, 0x8); + snd_soc_update_bits(codec, RT286_INDEX_REG_BASE + + RT286_CLK_DIV, 0xfc1e, 0x5406); + break; + default: + dev_err(codec->dev, "Unsupported system clock\n"); + return -EINVAL; + } + + rt286->sys_clk = freq; + + return 0; +} + +static int rt286_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio) +{ + struct snd_soc_codec *codec = dai->codec; + + dev_dbg(codec->dev, "%s ratio=%d\n", __func__, ratio); + if (50 == ratio) + snd_soc_update_bits(codec, RT286_INDEX_REG_BASE + + RT286_I2S_CTRL1, 0x1000, 0x1000); + else + snd_soc_update_bits(codec, RT286_INDEX_REG_BASE + + RT286_I2S_CTRL1, 0x1000, 0x0); + + + return 0; +} + +static int rt286_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_PREPARE: + if (SND_SOC_BIAS_STANDBY == codec->dapm.bias_level) + snd_soc_write(codec, + RT286_SET_AUDIO_POWER, AC_PWRST_D0); + break; + + case SND_SOC_BIAS_STANDBY: + snd_soc_write(codec, + RT286_SET_AUDIO_POWER, AC_PWRST_D3); + break; + + default: + break; + } + codec->dapm.bias_level = level; + + return 0; +} + +static irqreturn_t rt286_irq(int irq, void *data) +{ + struct rt286_priv *rt286 = data; + bool hp = false; + bool mic = false; + int status = 0; + + rt286_jack_detect(rt286->codec, &hp, &mic); + + /* Clear IRQ */ + snd_soc_update_bits(rt286->codec, RT286_INDEX_REG_BASE + + RT286_IRQ_CTRL, 0x1, 0x1); + + if (hp == true) + status |= SND_JACK_HEADPHONE; + + if (mic == true) + status |= SND_JACK_MICROPHONE; + + snd_soc_jack_report(rt286->jack, status, + SND_JACK_MICROPHONE | SND_JACK_HEADPHONE); + + pm_wakeup_event(&rt286->i2c->dev, 300); + + return IRQ_HANDLED; +} + +static int rt286_probe(struct snd_soc_codec *codec) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + int i, ret; + + ret = snd_soc_read(codec, + RT286_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID)); + if (ret != RT286_VENDOR_ID) { + dev_err(codec->dev, + "Device with ID register %x is not rt286\n", ret); + return -ENODEV; + } + + snd_soc_write(codec, RT286_SET_AUDIO_POWER, AC_PWRST_D3); + + for (i = 0; i < RT286_POWER_REG_LEN; i++) + snd_soc_write(codec, + RT286_SET_POWER(rt286_support_power_controls[i]), + AC_PWRST_D1); + + if (!rt286->pdata.cbj_en) { + snd_soc_write(codec, RT286_INDEX_REG_BASE + + RT286_CBJ_CTRL2, 0x0000); + snd_soc_write(codec, RT286_INDEX_REG_BASE + + RT286_MIC1_DET_CTRL, 0x0816); + snd_soc_write(codec, RT286_INDEX_REG_BASE + + RT286_MISC_CTRL1, 0x0000); + snd_soc_update_bits(codec, RT286_INDEX_REG_BASE + + RT286_CBJ_CTRL1, 0xf000, 0xb000); + } else { + snd_soc_update_bits(codec, RT286_INDEX_REG_BASE + + RT286_CBJ_CTRL1, 0xf000, 0x5000); + } + + mdelay(10); + + if (!rt286->pdata.gpio2_en) + snd_soc_write(codec, RT286_SET_DMIC2_DEFAULT, 0x4000); + else + snd_soc_write(codec, RT286_SET_DMIC2_DEFAULT, 0); + + mdelay(10); + + /*Power down LDO2*/ + snd_soc_update_bits(codec, RT286_INDEX_REG_BASE + + RT286_POWER_CTRL2, 0x8, 0x0); + + codec->dapm.bias_level = SND_SOC_BIAS_OFF; + rt286->codec = codec; + + if (rt286->i2c->irq) { + snd_soc_update_bits(codec, RT286_INDEX_REG_BASE + + RT286_IRQ_CTRL, 0x2, 0x2); + + INIT_DELAYED_WORK(&rt286->jack_detect_work, + rt286_jack_detect_work); + schedule_delayed_work(&rt286->jack_detect_work, + msecs_to_jiffies(1250)); + + ret = request_threaded_irq(rt286->i2c->irq, NULL, rt286_irq, + IRQF_TRIGGER_HIGH | IRQF_ONESHOT, "rt286", rt286); + if (ret != 0) { + dev_err(codec->dev, + "Failed to reguest IRQ: %d\n", ret); + return ret; + } + } + + return 0; +} + +static int rt286_remove(struct snd_soc_codec *codec) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + cancel_delayed_work_sync(&rt286->jack_detect_work); + + return 0; +} + +#define RT286_STEREO_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) +#define RT286_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) + +static const struct snd_soc_dai_ops rt286_aif_dai_ops = { + .hw_params = rt286_hw_params, + .set_fmt = rt286_set_dai_fmt, + .set_sysclk = rt286_set_dai_sysclk, + .set_bclk_ratio = rt286_set_bclk_ratio, +}; + +static struct snd_soc_dai_driver rt286_dai[] = { + { + .name = "rt286-aif1", + .id = RT286_AIF1, + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT286_STEREO_RATES, + .formats = RT286_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT286_STEREO_RATES, + .formats = RT286_FORMATS, + }, + .ops = &rt286_aif_dai_ops, + .symmetric_rates = 1, + }, + { + .name = "rt286-aif2", + .id = RT286_AIF2, + .playback = { + .stream_name = "AIF2 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT286_STEREO_RATES, + .formats = RT286_FORMATS, + }, + .capture = { + .stream_name = "AIF2 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT286_STEREO_RATES, + .formats = RT286_FORMATS, + }, + .ops = &rt286_aif_dai_ops, + .symmetric_rates = 1, + }, + +}; + +static struct snd_soc_codec_driver soc_codec_dev_rt286 = { + .probe = rt286_probe, + .remove = rt286_remove, + .set_bias_level = rt286_set_bias_level, + .idle_bias_off = true, + .write = rt286_hw_write, + .read = rt286_hw_read, + .controls = rt286_snd_controls, + .num_controls = ARRAY_SIZE(rt286_snd_controls), + .dapm_widgets = rt286_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rt286_dapm_widgets), + .dapm_routes = rt286_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rt286_dapm_routes), +}; + +static const struct i2c_device_id rt286_i2c_id[] = { + {"rt286", 0}, + {} +}; +MODULE_DEVICE_TABLE(i2c, rt286_i2c_id); + +static const struct acpi_device_id rt286_acpi_match[] = { + { "INT343A", 0 }, + {}, +}; +MODULE_DEVICE_TABLE(acpi, rt286_acpi_match); + +static int rt286_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct rt286_platform_data *pdata = dev_get_platdata(&i2c->dev); + struct rt286_priv *rt286; + int ret; + + rt286 = devm_kzalloc(&i2c->dev, sizeof(*rt286), + GFP_KERNEL); + if (NULL == rt286) + return -ENOMEM; + + rt286->i2c = i2c; + i2c_set_clientdata(i2c, rt286); + + if (pdata) + rt286->pdata = *pdata; + + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt286, + rt286_dai, ARRAY_SIZE(rt286_dai)); + + return ret; +} + +static int rt286_i2c_remove(struct i2c_client *i2c) +{ + struct rt286_priv *rt286 = i2c_get_clientdata(i2c); + + if (i2c->irq) + free_irq(i2c->irq, rt286); + snd_soc_unregister_codec(&i2c->dev); + + return 0; +} + + +struct i2c_driver rt286_i2c_driver = { + .driver = { + .name = "rt286", + .owner = THIS_MODULE, + .acpi_match_table = ACPI_PTR(rt286_acpi_match), + }, + .probe = rt286_i2c_probe, + .remove = rt286_i2c_remove, + .id_table = rt286_i2c_id, +}; + +module_i2c_driver(rt286_i2c_driver); + +MODULE_DESCRIPTION("ASoC RT286 driver"); +MODULE_AUTHOR("Bard Liao <bardliao(a)realtek.com>"); +MODULE_LICENSE("GPL"); + +/* + * ALC286 is a dual mode codec, which can run as HD-A or I2S mode. + * It is controlled by HD-A verb commands via I2C protocol. + * The following is the I/O difference between ALC286 and general I2S codecs. + * 1. A HD-A verb command contains three parts, NID, VID, and PID. + * And an I2S command contains only two parts: address and data. + * 2. Not only the register address is written, but the read command also + includes the entire write command. + * 3. rt286 uses different registers for read and write the same bits. + * As a result, regmap is difficult to be used on ALC286. + * We don't request a standard I/O by snd_soc_codec_set_cache_io anymore. + * Now we have specific write and read functions for ALC286's I/O. + * A verb command contains three part: NID, VID, and PID. + * We combine NID and VID as reg address, and PID as reg value. + * And do corresponding change in rt286_hw_read to get the data properly. + */ diff --git a/sound/soc/codecs/rt286.h b/sound/soc/codecs/rt286.h new file mode 100644 index 0000000..eaa093f --- /dev/null +++ b/sound/soc/codecs/rt286.h @@ -0,0 +1,186 @@ +/* + * rt286.h -- RT286 ALSA SoC audio driver + * + * Copyright 2011 Realtek Microelectronics + * Author: Johnny Hsu <johnnyhsu(a)realtek.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __RT286_H__ +#define __RT286_H__ + +#define VERB_CMD(V, N, D) ((N << 20) | (V << 8) | D) + +#define RT286_AUDIO_FUNCTION_GROUP 0x01 +#define RT286_DAC_OUT1 0x02 +#define RT286_DAC_OUT2 0x03 +#define RT286_ADC_IN1 0x09 +#define RT286_ADC_IN2 0x08 +#define RT286_MIXER_IN 0x0b +#define RT286_MIXER_OUT1 0x0c +#define RT286_MIXER_OUT2 0x0d +#define RT286_DMIC1 0x12 +#define RT286_DMIC2 0x13 +#define RT286_SPK_OUT 0x14 +#define RT286_MIC1 0x18 +#define RT286_LINE1 0x1a +#define RT286_BEEP 0x1d +#define RT286_SPDIF 0x1e +#define RT286_VENDOR_REGISTERS 0x20 +#define RT286_HP_OUT 0x21 +#define RT286_MIXER_IN1 0x22 +#define RT286_MIXER_IN2 0x23 + +#define RT286_SET_PIN_SFT 6 +#define RT286_SET_PIN_ENABLE 0x40 +#define RT286_SET_PIN_DISABLE 0 +#define RT286_SET_EAPD_HIGH 0x2 +#define RT286_SET_EAPD_LOW 0 + +#define RT286_MUTE_SFT 7 + +/* Verb commands */ +#define RT286_GET_PARAM(NID, PARAM) VERB_CMD(AC_VERB_PARAMETERS, NID, PARAM) +#define RT286_SET_POWER(NID) VERB_CMD(AC_VERB_SET_POWER_STATE, NID, 0) +#define RT286_SET_AUDIO_POWER RT286_SET_POWER(RT286_AUDIO_FUNCTION_GROUP) +#define RT286_SET_HPO_POWER RT286_SET_POWER(RT286_HP_OUT) +#define RT286_SET_SPK_POWER RT286_SET_POWER(RT286_SPK_OUT) +#define RT286_SET_DMIC1_POWER RT286_SET_POWER(RT286_DMIC1) +#define RT286_SPK_MUX\ + VERB_CMD(AC_VERB_SET_CONNECT_SEL, RT286_SPK_OUT, 0) +#define RT286_HPO_MUX\ + VERB_CMD(AC_VERB_SET_CONNECT_SEL, RT286_HP_OUT, 0) +#define RT286_ADC0_MUX\ + VERB_CMD(AC_VERB_SET_CONNECT_SEL, RT286_MIXER_IN1, 0) +#define RT286_ADC1_MUX\ + VERB_CMD(AC_VERB_SET_CONNECT_SEL, RT286_MIXER_IN2, 0) +#define RT286_SET_MIC1\ + VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT286_MIC1, 0) +#define RT286_SET_PIN_HPO\ + VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT286_HP_OUT, 0) +#define RT286_SET_PIN_SPK\ + VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT286_SPK_OUT, 0) +#define RT286_SET_PIN_DMIC1\ + VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT286_DMIC1, 0) +#define RT286_SPK_EAPD\ + VERB_CMD(AC_VERB_SET_EAPD_BTLENABLE, RT286_SPK_OUT, 0) +#define RT286_SET_AMP_GAIN_HPO\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_HP_OUT, 0) +#define RT286_GET_HP_SENSE\ + VERB_CMD(AC_VERB_GET_PIN_SENSE, RT286_HP_OUT, 0) +#define RT286_GET_MIC1_SENSE\ + VERB_CMD(AC_VERB_GET_PIN_SENSE, RT286_MIC1, 0) +#define RT286_SET_DMIC2_DEFAULT\ + VERB_CMD(AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, RT286_DMIC2, 0) +#define RT286_DACL_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_DAC_OUT1, 0xa000) +#define RT286_DACR_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_DAC_OUT1, 0x9000) +#define RT286_ADCL_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_ADC_IN1, 0x6000) +#define RT286_ADCR_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_ADC_IN1, 0x5000) +#define RT286_MIC_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIC1, 0x7000) +#define RT286_SPOL_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_SPK_OUT, 0xa000) +#define RT286_SPOR_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_SPK_OUT, 0x9000) +#define RT286_HPOL_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_HP_OUT, 0xa000) +#define RT286_HPOR_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_HP_OUT, 0x9000) +#define RT286_F_DAC_SWITCH\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIXER_OUT1, 0x7000) +#define RT286_F_RECMIX_SWITCH\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIXER_OUT1, 0x7100) +#define RT286_REC_MIC_SWITCH\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIXER_IN, 0x7000) +#define RT286_REC_I2S_SWITCH\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIXER_IN, 0x7100) +#define RT286_REC_LINE_SWITCH\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIXER_IN, 0x7200) +#define RT286_REC_BEEP_SWITCH\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIXER_IN, 0x7300) +#define RT286_DAC_FORMAT\ + VERB_CMD(AC_VERB_SET_STREAM_FORMAT, RT286_DAC_OUT1, 0) +#define RT286_ADC_FORMAT\ + VERB_CMD(AC_VERB_SET_STREAM_FORMAT, RT286_ADC_IN1, 0) + + +/* Index registers */ +#define RT286_A_BIAS_CTRL1 0x01 +#define RT286_A_BIAS_CTRL2 0x02 +#define RT286_POWER_CTRL1 0x03 +#define RT286_A_BIAS_CTRL3 0x04 +#define RT286_POWER_CTRL2 0x08 +#define RT286_I2S_CTRL1 0x09 +#define RT286_I2S_CTRL2 0x0a +#define RT286_CLK_DIV 0x0b +#define RT286_POWER_CTRL3 0x0f +#define RT286_MIC1_DET_CTRL 0x19 +#define RT286_MISC_CTRL1 0x20 +#define RT286_IRQ_CTRL 0x33 +#define RT286_PLL_CTRL1 0x49 +#define RT286_CBJ_CTRL1 0x4f +#define RT286_CBJ_CTRL2 0x50 +#define RT286_PLL_CTRL 0x63 + +/* SPDIF (0x06) */ +#define RT286_SPDIF_SEL_SFT 0 +#define RT286_SPDIF_SEL_PCM0 0 +#define RT286_SPDIF_SEL_PCM1 1 +#define RT286_SPDIF_SEL_SPOUT 2 +#define RT286_SPDIF_SEL_PP 3 + +/* RECMIX (0x0b) */ +#define RT286_M_REC_BEEP_SFT 0 +#define RT286_M_REC_LINE1_SFT 1 +#define RT286_M_REC_MIC1_SFT 2 +#define RT286_M_REC_I2S_SFT 3 + +/* Front (0x0c) */ +#define RT286_M_FRONT_DAC_SFT 0 +#define RT286_M_FRONT_REC_SFT 1 + +/* SPK-OUT (0x14) */ +#define RT286_M_SPK_MUX_SFT 14 +#define RT286_SPK_SEL_MASK 0x1 +#define RT286_SPK_SEL_SFT 0 +#define RT286_SPK_SEL_F 0 +#define RT286_SPK_SEL_S 1 + +/* HP-OUT (0x21) */ +#define RT286_M_HP_MUX_SFT 14 +#define RT286_HP_SEL_MASK 0x1 +#define RT286_HP_SEL_SFT 0 +#define RT286_HP_SEL_F 0 +#define RT286_HP_SEL_S 1 + +/* ADC (0x22) (0x23) */ +#define RT286_ADC_SEL_MASK 0x7 +#define RT286_ADC_SEL_SFT 0 +#define RT286_ADC_SEL_SURR 0 +#define RT286_ADC_SEL_FRONT 1 +#define RT286_ADC_SEL_DMIC 2 +#define RT286_ADC_SEL_BEEP 4 +#define RT286_ADC_SEL_LINE1 5 +#define RT286_ADC_SEL_I2S 6 +#define RT286_ADC_SEL_MIC1 7 + +#define RT286_SCLK_S_MCLK 0 +#define RT286_SCLK_S_PLL 1 + +enum { + RT286_AIF1, + RT286_AIF2, + RT286_AIFS, +}; + +int rt286_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack); + +#endif /* __RT286_H__ */ + -- 1.8.1.1.439.g50a6b54

5 19

[alsa-devel] [PATCH] ASoC: arizona: Implement TDM support for Arizona devices
by Charles Keepax 04 Jun '14

04 Jun '14

Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- include/linux/mfd/arizona/core.h | 3 + sound/soc/codecs/arizona.c | 81 +++++++++++++++++++++++++++++++++++--- 2 files changed, 78 insertions(+), 6 deletions(-) diff --git a/include/linux/mfd/arizona/core.h b/include/linux/mfd/arizona/core.h index 5cf8b91..11783b5 100644 --- a/include/linux/mfd/arizona/core.h +++ b/include/linux/mfd/arizona/core.h @@ -110,6 +110,9 @@ struct arizona { int clk32k_ref; struct snd_soc_dapm_context *dapm; + + int tdm_width[ARIZONA_MAX_AIF]; + int tdm_slots[ARIZONA_MAX_AIF]; }; int arizona_clk32k_enable(struct arizona *arizona); diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 29e198f..d98bd77 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1185,7 +1185,10 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, int base = dai->driver->base; const int *rates; int i, ret, val; + int channels = params_channels(params); int chan_limit = arizona->pdata.max_channels_clocked[dai->id - 1]; + int tdm_width = arizona->tdm_width[dai->id - 1]; + int tdm_slots = arizona->tdm_slots[dai->id - 1]; int bclk, lrclk, wl, frame, bclk_target; if (params_rate(params) % 8000) @@ -1193,18 +1196,27 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, else rates = &arizona_48k_bclk_rates[0]; - bclk_target = snd_soc_params_to_bclk(params); - if (chan_limit && chan_limit < params_channels(params)) { + if (tdm_slots) { + arizona_aif_dbg(dai, "Configuring for %d %d bit TDM slots\n", + tdm_slots, tdm_width); + bclk_target = tdm_slots * tdm_width * params_rate(params); + channels = tdm_slots; + } else { + bclk_target = snd_soc_params_to_bclk(params); + } + + if (chan_limit && chan_limit < channels) { arizona_aif_dbg(dai, "Limiting to %d channels\n", chan_limit); - bclk_target /= params_channels(params); + bclk_target /= channels; bclk_target *= chan_limit; } - /* Force stereo for I2S mode */ + /* Force multiple of 2 channels for I2S mode */ val = snd_soc_read(codec, base + ARIZONA_AIF_FORMAT); - if (params_channels(params) == 1 && (val & ARIZONA_AIF1_FMT_MASK)) { + if ((channels & 1) && (val & ARIZONA_AIF1_FMT_MASK)) { arizona_aif_dbg(dai, "Forcing stereo mode\n"); - bclk_target *= 2; + bclk_target /= channels; + bclk_target *= channels + 1; } for (i = 0; i < ARRAY_SIZE(arizona_44k1_bclk_rates); i++) { @@ -1228,6 +1240,9 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, wl = snd_pcm_format_width(params_format(params)); frame = wl << ARIZONA_AIF1TX_WL_SHIFT | wl; + if (tdm_width && wl != tdm_width) + arizona_aif_warn(dai, "Word length not equal to TDM width\n"); + ret = arizona_hw_params_rate(substream, params, dai); if (ret != 0) return ret; @@ -1324,9 +1339,63 @@ static int arizona_set_tristate(struct snd_soc_dai *dai, int tristate) ARIZONA_AIF1_TRI, reg); } +static void arizona_set_channels_to_mask(struct snd_soc_dai *dai, + unsigned int base, + int channels, unsigned int mask) +{ + struct snd_soc_codec *codec = dai->codec; + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + struct arizona *arizona = priv->arizona; + int slot, i; + + for (i = 0; i < channels; ++i) { + slot = ffs(mask) - 1; + if (slot < 0) + return; + + regmap_write(arizona->regmap, base + i, slot); + + mask &= ~(1 << slot); + } + + if (mask) + arizona_aif_warn(dai, "Too many channels in TDM mask\n"); +} + +static int arizona_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, + unsigned int rx_mask, int slots, int slot_width) +{ + struct snd_soc_codec *codec = dai->codec; + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + struct arizona *arizona = priv->arizona; + int base = dai->driver->base; + int rx_max_chan = dai->driver->playback.channels_max; + int tx_max_chan = dai->driver->capture.channels_max; + + /* Only support TDM for the physical AIFs */ + if (dai->id > ARIZONA_MAX_AIF) + return -ENOTSUPP; + + if (slots == 0) { + tx_mask = (1 << tx_max_chan) - 1; + rx_mask = (1 << rx_max_chan) - 1; + } + + arizona_set_channels_to_mask(dai, base + ARIZONA_AIF_FRAME_CTRL_3, + tx_max_chan, tx_mask); + arizona_set_channels_to_mask(dai, base + ARIZONA_AIF_FRAME_CTRL_11, + rx_max_chan, rx_mask); + + arizona->tdm_width[dai->id - 1] = slot_width; + arizona->tdm_slots[dai->id - 1] = slots; + + return 0; +} + const struct snd_soc_dai_ops arizona_dai_ops = { .startup = arizona_startup, .set_fmt = arizona_set_fmt, + .set_tdm_slot = arizona_set_tdm_slot, .hw_params = arizona_hw_params, .set_sysclk = arizona_dai_set_sysclk, .set_tristate = arizona_set_tristate, -- 1.7.2.5

2 2

[alsa-devel] PROBLEM: [Lenovo ThinkPad X61s] Speaker volume mutes itself in random intervals after upgrading from 3.11 to 3.15.
by Alex Fedora 04 Jun '14

04 Jun '14

Hi, Please see problem description below. I've captured alsa info output for the latest kernel. Kernel just before the regression. And info from the kernel with the offending commit. But I had to calculate diff from first one to fit to 100K limit. If you want, I can send all 3 files as an attachments. Also the latest version of the kernel was took from http://kernel.ubuntu.com/~kernel-ppa/mainline/daily/current/linux-image-3.1… . Thank you, Alex [1.] One line summary of the problem: [Lenovo ThinkPad X61s] Speaker volume mutes itself in random intervals after upgrade to latest kernel [2.] Full description of the problem/report: Speaker mutes itself in random intervals. Headphones are working fine at that moment. I can see what volume of the Speaker drops to 0% and then back to 100% using alsamixer. It seems like Internal Mic Boost also jumps from 100% to 0% and back. It is regression from 5ccc618fee67f0f0b2122dd4b32a02fd2b6a1569 (ALSA: hda - Remove static quirks for AD1884/1984 & variants). I've tested previous commit aa95d61b43e0fcb0b2ce68e5efa37174fd9e5cd3 (ALSA: hda - Remove static quirks for AD1882) and audio works as expected. New kernel shows two controls "Headphone Playback Volume" and "Speaker Playback Volume". But old one shows single control "PCM Playback Volume". "Speaker Playback Volume" is the one which gets muted randomly. [3.] Keywords (i.e., modules, networking, kernel): [4.] Kernel version (from /proc/version): Linux version 3.15.0-999-generic (apw@gomeisa) (gcc version 4.6.3 (Ubuntu/Linaro 4.6.3-1ubuntu5) ) #201405310205 SMP Sat May 31 06:06:08 UTC 2014 [5.] Output of Oops.. message (if applicable) with symbolic information resolved (see Documentation/oops-tracing.txt) [6.] A small shell script or example program which triggers the problem (if possible) Tested manually. [7.] Environment Description: Ubuntu 14.04 LTS Release: 14.04 [7.1.] Software (add the output of the ver_linux script here) If some fields are empty or look unusual you may have an old version. Compare to the current minimal requirements in Documentation/Changes. Linux x61s 3.15.0-999-generic #201405310205 SMP Sat May 31 06:06:08 UTC 2014 x86_64 x86_64 x86_64 GNU/Linux Gnu C 4.8 Gnu make 3.81 binutils 2.24 util-linux 2.20.1 mount support module-init-tools 15 e2fsprogs 1.42.9 pcmciautils 018 Linux C Library 2.19 Dynamic linker (ldd) 2.19 Procps 3.3.9 Net-tools 1.60 Kbd 1.15.5 Sh-utils 8.21 wireless-tools 30 Modules Loaded ctr ccm nls_iso8859_1 ipt_MASQUERADE iptable_nat nf_nat_ipv4 nf_nat nf_conntrack_ipv4 nf_defrag_ipv4 xt_conntrack nf_conntrack ipt_REJECT xt_CHECKSUM iptable_mangle xt_tcpudp bridge stp llc ip6table_filter ip6_tables iptable_filter ip_tables ebtable_nat ebtables x_tables rfcomm bnep bluetooth 6lowpan_iphc binfmt_misc snd_hda_codec_analog snd_hda_codec_generic snd_hda_intel snd_hda_controller snd_hda_codec snd_hwdep thinkpad_acpi dm_multipath coretemp snd_pcm nvram kvm_intel scsi_dh snd_seq_midi arc4 kvm snd_seq_midi_event iwl4965 iwlegacy mac80211 snd_rawmidi snd_seq cfg80211 snd_seq_device pcmcia snd_timer microcode snd serio_raw yenta_socket soundcore mac_hid pcmcia_rsrc ppdev lpc_ich pcmcia_core parport_pc lp parport xts gf128mul dm_crypt mmc_block ahci sdhci_pci psmouse libahci sdhci i915 video i2c_algo_bit e1000e drm_kms_helper drm ptp pps_core [7.2.] Processor information (from /proc/cpuinfo): processor : 0 vendor_id : GenuineIntel cpu family : 6 model : 15 model name : Intel(R) Core(TM)2 Duo CPU L7500 @ 1.60GHz stepping : 11 microcode : 0xba cpu MHz : 800.000 cache size : 4096 KB physical id : 0 siblings : 2 core id : 0 cpu cores : 2 apicid : 0 initial apicid : 0 fpu : yes fpu_exception : yes cpuid level : 10 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe syscall nx lm constant_tsc arch_perfmon pebs bts rep_good nopl aperfmperf pni dtes64 monitor ds_cpl vmx est tm2 ssse3 cx16 xtpr pdcm lahf_lm ida dtherm tpr_shadow vnmi flexpriority bogomips : 3191.89 clflush size : 64 cache_alignment : 64 address sizes : 36 bits physical, 48 bits virtual power management: processor : 1 vendor_id : GenuineIntel cpu family : 6 model : 15 model name : Intel(R) Core(TM)2 Duo CPU L7500 @ 1.60GHz stepping : 11 microcode : 0xba cpu MHz : 800.000 cache size : 4096 KB physical id : 0 siblings : 2 core id : 1 cpu cores : 2 apicid : 1 initial apicid : 1 fpu : yes fpu_exception : yes cpuid level : 10 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe syscall nx lm constant_tsc arch_perfmon pebs bts rep_good nopl aperfmperf pni dtes64 monitor ds_cpl vmx est tm2 ssse3 cx16 xtpr pdcm lahf_lm ida dtherm tpr_shadow vnmi flexpriority bogomips : 3191.89 clflush size : 64 cache_alignment : 64 address sizes : 36 bits physical, 48 bits virtual power management: [7.3.] Module information (from /proc/modules): ctr 13193 2 - Live 0x0000000000000000 ccm 17856 2 - Live 0x0000000000000000 nls_iso8859_1 12713 1 - Live 0x0000000000000000 ipt_MASQUERADE 12880 3 - Live 0x0000000000000000 iptable_nat 13151 1 - Live 0x0000000000000000 nf_nat_ipv4 13316 1 iptable_nat, Live 0x0000000000000000 nf_nat 26147 3 ipt_MASQUERADE,iptable_nat,nf_nat_ipv4, Live 0x0000000000000000 nf_conntrack_ipv4 14857 2 - Live 0x0000000000000000 nf_defrag_ipv4 12758 1 nf_conntrack_ipv4, Live 0x0000000000000000 xt_conntrack 12760 1 - Live 0x0000000000000000 nf_conntrack 105687 6 ipt_MASQUERADE,iptable_nat,nf_nat_ipv4,nf_nat,nf_conntrack_ipv4,xt_conntrack, Live 0x0000000000000000 ipt_REJECT 12579 2 - Live 0x0000000000000000 xt_CHECKSUM 12549 1 - Live 0x0000000000000000 iptable_mangle 12734 1 - Live 0x0000000000000000 xt_tcpudp 12924 6 - Live 0x0000000000000000 bridge 116335 0 - Live 0x0000000000000000 stp 12976 1 bridge, Live 0x0000000000000000 llc 14441 2 bridge,stp, Live 0x0000000000000000 ip6table_filter 12815 0 - Live 0x0000000000000000 ip6_tables 27504 1 ip6table_filter, Live 0x0000000000000000 iptable_filter 12810 1 - Live 0x0000000000000000 ip_tables 27718 3 iptable_nat,iptable_mangle,iptable_filter, Live 0x0000000000000000 ebtable_nat 12807 0 - Live 0x0000000000000000 ebtables 35305 1 ebtable_nat, Live 0x0000000000000000 x_tables 34194 11 ipt_MASQUERADE,xt_conntrack,ipt_REJECT,xt_CHECKSUM,iptable_mangle,xt_tcpudp,ip6table_filter,ip6_tables,iptable_filter,ip_tables,ebtables, Live 0x0000000000000000 rfcomm 75078 0 - Live 0x0000000000000000 bnep 19884 2 - Live 0x0000000000000000 bluetooth 461775 10 rfcomm,bnep, Live 0x0000000000000000 6lowpan_iphc 18968 1 bluetooth, Live 0x0000000000000000 binfmt_misc 17508 1 - Live 0x0000000000000000 snd_hda_codec_analog 15049 1 - Live 0x0000000000000000 snd_hda_codec_generic 70087 1 snd_hda_codec_analog, Live 0x0000000000000000 snd_hda_intel 30608 6 - Live 0x0000000000000000 snd_hda_controller 35518 1 snd_hda_intel, Live 0x0000000000000000 snd_hda_codec 144671 4 snd_hda_codec_analog,snd_hda_codec_generic,snd_hda_intel,snd_hda_controller, Live 0x0000000000000000 snd_hwdep 13613 1 snd_hda_codec, Live 0x0000000000000000 thinkpad_acpi 82198 0 - Live 0x0000000000000000 dm_multipath 23188 0 - Live 0x0000000000000000 coretemp 13638 0 - Live 0x0000000000000000 snd_pcm 113863 3 snd_hda_intel,snd_hda_controller,snd_hda_codec, Live 0x0000000000000000 nvram 14462 1 thinkpad_acpi, Live 0x0000000000000000 kvm_intel 148919 0 - Live 0x0000000000000000 scsi_dh 14873 1 dm_multipath, Live 0x0000000000000000 snd_seq_midi 13564 0 - Live 0x0000000000000000 arc4 12573 2 - Live 0x0000000000000000 kvm 463855 1 kvm_intel, Live 0x0000000000000000 snd_seq_midi_event 14899 1 snd_seq_midi, Live 0x0000000000000000 iwl4965 118803 0 - Live 0x0000000000000000 iwlegacy 105101 1 iwl4965, Live 0x0000000000000000 mac80211 663883 2 iwl4965,iwlegacy, Live 0x0000000000000000 snd_rawmidi 30865 1 snd_seq_midi, Live 0x0000000000000000 snd_seq 67636 2 snd_seq_midi,snd_seq_midi_event, Live 0x0000000000000000 cfg80211 515506 3 iwl4965,iwlegacy,mac80211, Live 0x0000000000000000 snd_seq_device 14497 3 snd_seq_midi,snd_rawmidi,snd_seq, Live 0x0000000000000000 pcmcia 62760 0 - Live 0x0000000000000000 snd_timer 30118 2 snd_pcm,snd_seq, Live 0x0000000000000000 microcode 24391 0 - Live 0x0000000000000000 snd 74195 23 snd_hda_codec_analog,snd_hda_codec_generic,snd_hda_intel,snd_hda_codec,snd_hwdep,thinkpad_acpi,snd_pcm,snd_rawmidi,snd_seq,snd_seq_device,snd_timer, Live 0x0000000000000000 serio_raw 13483 0 - Live 0x0000000000000000 yenta_socket 45447 0 - Live 0x0000000000000000 soundcore 12680 2 snd_hda_codec,snd, Live 0x0000000000000000 mac_hid 13275 0 - Live 0x0000000000000000 pcmcia_rsrc 18591 1 yenta_socket, Live 0x0000000000000000 ppdev 17711 0 - Live 0x0000000000000000 lpc_ich 21176 0 - Live 0x0000000000000000 pcmcia_core 23715 3 pcmcia,yenta_socket,pcmcia_rsrc, Live 0x0000000000000000 parport_pc 32906 0 - Live 0x0000000000000000 lp 17799 0 - Live 0x0000000000000000 parport 42481 3 ppdev,parport_pc,lp, Live 0x0000000000000000 xts 12951 1 - Live 0x0000000000000000 gf128mul 14951 1 xts, Live 0x0000000000000000 dm_crypt 23456 1 - Live 0x0000000000000000 mmc_block 36596 2 - Live 0x0000000000000000 ahci 30167 3 - Live 0x0000000000000000 sdhci_pci 23347 0 - Live 0x0000000000000000 psmouse 113095 0 - Live 0x0000000000000000 libahci 32191 1 ahci, Live 0x0000000000000000 sdhci 43409 1 sdhci_pci, Live 0x0000000000000000 i915 873516 3 - Live 0x0000000000000000 video 19932 1 i915, Live 0x0000000000000000 i2c_algo_bit 13564 1 i915, Live 0x0000000000000000 e1000e 262482 0 - Live 0x0000000000000000 drm_kms_helper 59729 1 i915, Live 0x0000000000000000 drm 310655 5 i915,drm_kms_helper, Live 0x0000000000000000 ptp 19534 1 e1000e, Live 0x0000000000000000 pps_core 19381 1 ptp, Live 0x0000000000000000 [7.4.] Loaded driver and hardware information (/proc/ioports, /proc/iomem) ioports: 0000-0cf7 : PCI Bus 0000:00 0000-001f : dma1 0020-0021 : pic1 0040-0043 : timer0 0050-0053 : timer1 0060-0060 : keyboard 0062-0062 : EC data 0064-0064 : keyboard 0066-0066 : EC cmd 0070-0071 : rtc0 0080-008f : dma page reg 00a0-00a1 : pic2 00c0-00df : dma2 00f0-00ff : fpu 03c0-03df : vesafb 0800-080f : pnp 00:01 0cf8-0cff : PCI conf1 0d00-ffff : PCI Bus 0000:00 1000-1003 : ACPI PM1a_EVT_BLK 1004-1005 : ACPI PM1a_CNT_BLK 1008-100b : ACPI PM_TMR 1010-1015 : ACPI CPU throttle 1020-1020 : ACPI PM2_CNT_BLK 1028-102f : ACPI GPE0_BLK 1030-1033 : iTCO_wdt 1060-107f : iTCO_wdt 1180-11bf : 0000:00:1f.0 1180-11bf : pnp 00:01 15e0-15ef : pnp 00:01 164e-164f : pnp 00:01 1800-1807 : 0000:00:02.0 1838-183b : 0000:00:1f.2 1838-183b : ahci 183c-183f : 0000:00:1f.2 183c-183f : ahci 1840-185f : 0000:00:19.0 1860-187f : 0000:00:1a.0 1860-187f : uhci_hcd 1880-189f : 0000:00:1a.1 1880-189f : uhci_hcd 18a0-18bf : 0000:00:1d.0 18a0-18bf : uhci_hcd 18c0-18df : 0000:00:1d.1 18c0-18df : uhci_hcd 18e0-18ff : 0000:00:1f.2 18e0-18ff : ahci 1c00-1c07 : 0000:00:1f.2 1c00-1c07 : ahci 1c08-1c0f : 0000:00:1f.2 1c08-1c0f : ahci 1c20-1c3f : 0000:00:1f.3 2000-2fff : PCI Bus 0000:02 3000-3fff : PCI Bus 0000:03 4000-7fff : PCI Bus 0000:05 4000-40ff : PCI CardBus 0000:06 4400-44ff : PCI CardBus 0000:06 iomem: 00000000-00000fff : reserved 00001000-0009d7ff : System RAM 0009d800-0009ffff : reserved 000a0000-000bffff : PCI Bus 0000:00 000c0000-000c7fff : Video ROM 000c8000-000cbfff : pnp 00:00 000cc000-000cffff : pnp 00:00 000d0000-000d3fff : PCI Bus 0000:00 000d4000-000d7fff : PCI Bus 0000:00 000d8000-000dbfff : PCI Bus 0000:00 000dc000-000dffff : PCI Bus 0000:00 000e0000-000fffff : reserved 000f0000-000fffff : System ROM 00100000-bf6affff : System RAM 01000000-0177cf58 : Kernel code 0177cf59-01d20f7f : Kernel data 01e7d000-01fe2fff : Kernel bss 2c000000-33ffffff : Crash kernel bf6b0000-bf6cbfff : ACPI Tables bf6cc000-bf6fffff : ACPI Non-volatile Storage bf700000-bfffffff : reserved bf800000-bfffffff : Graphics Stolen Memory c0000000-febfffff : PCI Bus 0000:00 c0000000-c3ffffff : PCI CardBus 0000:06 c4000000-c4000fff : Intel Flush Page d4000000-d7efffff : PCI Bus 0000:05 d7efec00-d7efecff : 0000:05:00.2 d7efec00-d7efecff : mmc0 d7eff000-d7efffff : 0000:05:00.0 d7eff000-d7efffff : yenta_socket d8000000-dbffffff : PCI Bus 0000:05 d8000000-dbffffff : PCI CardBus 0000:06 e0000000-efffffff : 0000:00:02.0 f0000000-f3ffffff : PCI MMCONFIG 0000 [bus 00-3f] f0000000-f3ffffff : reserved f0000000-f3ffffff : pnp 00:01 f4000000-f5ffffff : PCI Bus 0000:02 f6000000-f7ffffff : PCI Bus 0000:03 f7f00000-f7f01fff : 0000:03:00.0 f7f00000-f7f01fff : iwl4965 f8000000-f80fffff : 0000:00:02.0 f8100000-f81fffff : 0000:00:02.1 f8200000-f821ffff : 0000:00:19.0 f8200000-f821ffff : e1000e f8220000-f8223fff : 0000:00:1b.0 f8220000-f8223fff : ICH HD audio f8225000-f8225fff : 0000:00:19.0 f8225000-f8225fff : e1000e f8426000-f84267ff : 0000:00:1f.2 f8426000-f84267ff : ahci f8426c00-f8426fff : 0000:00:1a.7 f8426c00-f8426fff : ehci_hcd f8427000-f84273ff : 0000:00:1d.7 f8427000-f84273ff : ehci_hcd f8427400-f84274ff : 0000:00:1f.3 f8500000-f85fffff : PCI Bus 0000:02 f8600000-f86fffff : PCI Bus 0000:03 fec00000-fec0ffff : reserved fec00000-fec003ff : IOAPIC 0 fed00000-fed003ff : HPET 0 fed00000-fed003ff : reserved fed14000-fed19fff : reserved fed14000-fed17fff : pnp 00:01 fed18000-fed18fff : pnp 00:01 fed19000-fed19fff : pnp 00:01 fed1c000-fed8ffff : reserved fed1c000-fed1ffff : pnp 00:01 fed1f410-fed1f414 : iTCO_wdt fed40000-fed4bfff : PCI Bus 0000:00 fed45000-fed4bfff : pnp 00:01 fee00000-fee00fff : Local APIC fee00000-fee00fff : reserved ff000000-ffffffff : reserved 100000000-13bffffff : System RAM [7.5.] PCI information ('lspci -vvv' as root) 00:00.0 Host bridge: Intel Corporation Mobile PM965/GM965/GL960 Memory Controller Hub (rev 0c) Subsystem: Lenovo ThinkPad T61/R61 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B+ ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort+ >SERR- <PERR- INTx- Latency: 0 Capabilities: [e0] Vendor Specific Information: Len=0a <?> Kernel driver in use: agpgart-intel 00:02.0 VGA compatible controller: Intel Corporation Mobile GM965/GL960 Integrated Graphics Controller (primary) (rev 0c) (prog-if 00 [VGA controller]) Subsystem: Lenovo ThinkPad T61/R61 Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx+ Status: Cap+ 66MHz- UDF- FastB2B+ ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0 Interrupt: pin A routed to IRQ 43 Region 0: Memory at f8000000 (64-bit, non-prefetchable) [size=1M] Region 2: Memory at e0000000 (64-bit, prefetchable) [size=256M] Region 4: I/O ports at 1800 [size=8] Expansion ROM at <unassigned> [disabled] Capabilities: [90] MSI: Enable+ Count=1/1 Maskable- 64bit- Address: fee0300c Data: 41e1 Capabilities: [d0] Power Management version 3 Flags: PMEClk- DSI+ D1- D2- AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot-,D3cold-) Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- Bridge: PM- B3+ Kernel driver in use: i915 00:02.1 Display controller: Intel Corporation Mobile GM965/GL960 Integrated Graphics Controller (secondary) (rev 0c) Subsystem: Lenovo ThinkPad T61/R61 Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B+ ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0 Region 0: Memory at f8100000 (64-bit, non-prefetchable) [size=1M] Capabilities: [d0] Power Management version 3 Flags: PMEClk- DSI+ D1- D2- AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot-,D3cold-) Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- Bridge: PM- B3+ 00:19.0 Ethernet controller: Intel Corporation 82566MM Gigabit Network Connection (rev 03) Subsystem: Lenovo Device 20de Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- DisINTx+ Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0 Interrupt: pin A routed to IRQ 42 Region 0: Memory at f8200000 (32-bit, non-prefetchable) [size=128K] Region 1: Memory at f8225000 (32-bit, non-prefetchable) [size=4K] Region 2: I/O ports at 1840 [size=32] Capabilities: [c8] Power Management version 2 Flags: PMEClk- DSI+ D1- D2- AuxCurrent=0mA PME(D0+,D1-,D2-,D3hot+,D3cold+) Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=1 PME- Capabilities: [d0] MSI: Enable+ Count=1/1 Maskable- 64bit+ Address: 00000000fee0300c Data: 4172 Kernel driver in use: e1000e 00:1a.0 USB controller: Intel Corporation 82801H (ICH8 Family) USB UHCI Controller #4 (rev 03) (prog-if 00 [UHCI]) Subsystem: Lenovo ThinkPad T61/R61 Control: I/O+ Mem- BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap- 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0 Interrupt: pin A routed to IRQ 20 Region 4: I/O ports at 1860 [size=32] Kernel driver in use: uhci_hcd 00:1a.1 USB controller: Intel Corporation 82801H (ICH8 Family) USB UHCI Controller #5 (rev 03) (prog-if 00 [UHCI]) Subsystem: Lenovo Thinkpad T61/R61 Control: I/O+ Mem- BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap- 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0 Interrupt: pin B routed to IRQ 21 Region 4: I/O ports at 1880 [size=32] Kernel driver in use: uhci_hcd 00:1a.7 USB controller: Intel Corporation 82801H (ICH8 Family) USB2 EHCI Controller #2 (rev 03) (prog-if 20 [EHCI]) Subsystem: Lenovo ThinkPad T61/R61 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0 Interrupt: pin C routed to IRQ 22 Region 0: Memory at f8426c00 (32-bit, non-prefetchable) [size=1K] Capabilities: [50] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=375mA PME(D0+,D1-,D2-,D3hot+,D3cold+) Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- Kernel driver in use: ehci-pci 00:1b.0 Audio device: Intel Corporation 82801H (ICH8 Family) HD Audio Controller (rev 03) Subsystem: Lenovo ThinkPad T61/R61 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- DisINTx+ Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0, Cache Line Size: 64 bytes Interrupt: pin B routed to IRQ 46 Region 0: Memory at f8220000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=55mA PME(D0+,D1-,D2-,D3hot+,D3cold+) Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- Capabilities: [60] MSI: Enable+ Count=1/1 Maskable- 64bit+ Address: 00000000fee0300c Data: 4152 Capabilities: [70] Express (v1) Root Complex Integrated Endpoint, MSI 00 DevCap: MaxPayload 128 bytes, PhantFunc 0 ExtTag- RBE- DevCtl: Report errors: Correctable- Non-Fatal- Fatal- Unsupported- RlxdOrd- ExtTag- PhantFunc- AuxPwr- NoSnoop+ MaxPayload 128 bytes, MaxReadReq 128 bytes DevSta: CorrErr- UncorrErr- FatalErr- UnsuppReq- AuxPwr+ TransPend- Capabilities: [100 v1] Virtual Channel Caps: LPEVC=0 RefClk=100ns PATEntryBits=1 Arb: Fixed- WRR32- WRR64- WRR128- Ctrl: ArbSelect=Fixed Status: InProgress- VC0: Caps: PATOffset=00 MaxTimeSlots=1 RejSnoopTrans- Arb: Fixed- WRR32- WRR64- WRR128- TWRR128- WRR256- Ctrl: Enable+ ID=0 ArbSelect=Fixed TC/VC=01 Status: NegoPending- InProgress- VC1: Caps: PATOffset=00 MaxTimeSlots=1 RejSnoopTrans- Arb: Fixed- WRR32- WRR64- WRR128- TWRR128- WRR256- Ctrl: Enable+ ID=1 ArbSelect=Fixed TC/VC=80 Status: NegoPending- InProgress- Capabilities: [130 v1] Root Complex Link Desc: PortNumber=0f ComponentID=02 EltType=Config Link0: Desc: TargetPort=00 TargetComponent=02 AssocRCRB- LinkType=MemMapped LinkValid+ Addr: 00000000fed1c000 Kernel driver in use: snd_hda_intel 00:1c.0 PCI bridge: Intel Corporation 82801H (ICH8 Family) PCI Express Port 1 (rev 03) (prog-if 00 [Normal decode]) Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- DisINTx+ Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0, Cache Line Size: 64 bytes Bus: primary=00, secondary=02, subordinate=02, sec-latency=0 I/O behind bridge: 00002000-00002fff Memory behind bridge: f4000000-f5ffffff Prefetchable memory behind bridge: 00000000f8500000-00000000f85fffff Secondary status: 66MHz- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort+ <SERR- <PERR- BridgeCtl: Parity- SERR- NoISA+ VGA- MAbort- >Reset- FastB2B- PriDiscTmr- SecDiscTmr- DiscTmrStat- DiscTmrSERREn- Capabilities: [40] Express (v1) Root Port (Slot+), MSI 00 DevCap: MaxPayload 128 bytes, PhantFunc 0 ExtTag- RBE+ DevCtl: Report errors: Correctable- Non-Fatal- Fatal- Unsupported- RlxdOrd- ExtTag- PhantFunc- AuxPwr- NoSnoop- MaxPayload 128 bytes, MaxReadReq 128 bytes DevSta: CorrErr- UncorrErr- FatalErr- UnsuppReq- AuxPwr+ TransPend- LnkCap: Port #1, Speed 2.5GT/s, Width x1, ASPM L0s L1, Exit Latency L0s <1us, L1 <4us ClockPM- Surprise- LLActRep+ BwNot- LnkCtl: ASPM Disabled; RCB 64 bytes Disabled- CommClk- ExtSynch- ClockPM- AutWidDis- BWInt- AutBWInt- LnkSta: Speed 2.5GT/s, Width x0, TrErr- Train- SlotClk+ DLActive- BWMgmt- ABWMgmt- SltCap: AttnBtn- PwrCtrl- MRL- AttnInd- PwrInd- HotPlug+ Surprise+ Slot #2, PowerLimit 6.500W; Interlock- NoCompl- SltCtl: Enable: AttnBtn- PwrFlt- MRL- PresDet+ CmdCplt+ HPIrq+ LinkChg+ Control: AttnInd Unknown, PwrInd Unknown, Power- Interlock- SltSta: Status: AttnBtn- PowerFlt- MRL- CmdCplt- PresDet- Interlock- Changed: MRL- PresDet- LinkState- RootCtl: ErrCorrectable- ErrNon-Fatal- ErrFatal- PMEIntEna+ CRSVisible- RootCap: CRSVisible- RootSta: PME ReqID 0000, PMEStatus- PMEPending- Capabilities: [80] MSI: Enable+ Count=1/1 Maskable- 64bit- Address: fee0300c Data: 41a1 Capabilities: [90] Subsystem: Lenovo ThinkPad T61/R61 Capabilities: [a0] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=0mA PME(D0+,D1-,D2-,D3hot+,D3cold+) Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- Capabilities: [100 v1] Virtual Channel Caps: LPEVC=0 RefClk=100ns PATEntryBits=1 Arb: Fixed+ WRR32- WRR64- WRR128- Ctrl: ArbSelect=Fixed Status: InProgress- VC0: Caps: PATOffset=00 MaxTimeSlots=1 RejSnoopTrans- Arb: Fixed+ WRR32- WRR64- WRR128- TWRR128- WRR256- Ctrl: Enable+ ID=0 ArbSelect=Fixed TC/VC=ff Status: NegoPending- InProgress- Capabilities: [180 v1] Root Complex Link Desc: PortNumber=01 ComponentID=02 EltType=Config Link0: Desc: TargetPort=00 TargetComponent=02 AssocRCRB- LinkType=MemMapped LinkValid+ Addr: 00000000fed1c001 Kernel driver in use: pcieport 00:1c.1 PCI bridge: Intel Corporation 82801H (ICH8 Family) PCI Express Port 2 (rev 03) (prog-if 00 [Normal decode]) Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- DisINTx+ Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0, Cache Line Size: 64 bytes Bus: primary=00, secondary=03, subordinate=03, sec-latency=0 I/O behind bridge: 00003000-00003fff Memory behind bridge: f6000000-f7ffffff Prefetchable memory behind bridge: 00000000f8600000-00000000f86fffff Secondary status: 66MHz- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort+ <SERR+ <PERR- BridgeCtl: Parity- SERR- NoISA+ VGA- MAbort- >Reset- FastB2B- PriDiscTmr- SecDiscTmr- DiscTmrStat- DiscTmrSERREn- Capabilities: [40] Express (v1) Root Port (Slot+), MSI 00 DevCap: MaxPayload 128 bytes, PhantFunc 0 ExtTag- RBE+ DevCtl: Report errors: Correctable- Non-Fatal- Fatal- Unsupported- RlxdOrd- ExtTag- PhantFunc- AuxPwr- NoSnoop- MaxPayload 128 bytes, MaxReadReq 128 bytes DevSta: CorrErr- UncorrErr- FatalErr- UnsuppReq- AuxPwr+ TransPend- LnkCap: Port #2, Speed 2.5GT/s, Width x1, ASPM L0s L1, Exit Latency L0s <256ns, L1 <4us ClockPM- Surprise- LLActRep+ BwNot- LnkCtl: ASPM Disabled; RCB 64 bytes Disabled- CommClk+ ExtSynch- ClockPM- AutWidDis- BWInt- AutBWInt- LnkSta: Speed 2.5GT/s, Width x1, TrErr- Train- SlotClk+ DLActive+ BWMgmt- ABWMgmt- SltCap: AttnBtn- PwrCtrl- MRL- AttnInd- PwrInd- HotPlug+ Surprise+ Slot #3, PowerLimit 6.500W; Interlock- NoCompl- SltCtl: Enable: AttnBtn- PwrFlt- MRL- PresDet+ CmdCplt+ HPIrq+ LinkChg+ Control: AttnInd Unknown, PwrInd Unknown, Power- Interlock- SltSta: Status: AttnBtn- PowerFlt- MRL- CmdCplt- PresDet+ Interlock- Changed: MRL- PresDet- LinkState+ RootCtl: ErrCorrectable- ErrNon-Fatal- ErrFatal- PMEIntEna+ CRSVisible- RootCap: CRSVisible- RootSta: PME ReqID 0000, PMEStatus- PMEPending- Capabilities: [80] MSI: Enable+ Count=1/1 Maskable- 64bit- Address: fee0300c Data: 41b1 Capabilities: [90] Subsystem: Lenovo ThinkPad T61/R61 Capabilities: [a0] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=0mA PME(D0+,D1-,D2-,D3hot+,D3cold+) Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- Capabilities: [100 v1] Virtual Channel Caps: LPEVC=0 RefClk=100ns PATEntryBits=1 Arb: Fixed+ WRR32- WRR64- WRR128- Ctrl: ArbSelect=Fixed Status: InProgress- VC0: Caps: PATOffset=00 MaxTimeSlots=1 RejSnoopTrans- Arb: Fixed+ WRR32- WRR64- WRR128- TWRR128- WRR256- Ctrl: Enable+ ID=0 ArbSelect=Fixed TC/VC=01 Status: NegoPending- InProgress- Capabilities: [180 v1] Root Complex Link Desc: PortNumber=02 ComponentID=02 EltType=Config Link0: Desc: TargetPort=00 TargetComponent=02 AssocRCRB- LinkType=MemMapped LinkValid+ Addr: 00000000fed1c001 Kernel driver in use: pcieport 00:1d.0 USB controller: Intel Corporation 82801H (ICH8 Family) USB UHCI Controller #1 (rev 03) (prog-if 00 [UHCI]) Subsystem: Lenovo ThinkPad T61/R61 Control: I/O+ Mem- BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap- 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0 Interrupt: pin A routed to IRQ 16 Region 4: I/O ports at 18a0 [size=32] Kernel driver in use: uhci_hcd 00:1d.1 USB controller: Intel Corporation 82801H (ICH8 Family) USB UHCI Controller #2 (rev 03) (prog-if 00 [UHCI]) Subsystem: Lenovo ThinkPad T61/R61 Control: I/O+ Mem- BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap- 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0 Interrupt: pin B routed to IRQ 17 Region 4: I/O ports at 18c0 [size=32] Kernel driver in use: uhci_hcd 00:1d.7 USB controller: Intel Corporation 82801H (ICH8 Family) USB2 EHCI Controller #1 (rev 03) (prog-if 20 [EHCI]) Subsystem: Lenovo ThinkPad T61/R61 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0 Interrupt: pin D routed to IRQ 19 Region 0: Memory at f8427000 (32-bit, non-prefetchable) [size=1K] Capabilities: [50] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=375mA PME(D0+,D1-,D2-,D3hot+,D3cold+) Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- Capabilities: [58] Debug port: BAR=1 offset=00a0 Kernel driver in use: ehci-pci 00:1e.0 PCI bridge: Intel Corporation 82801 Mobile PCI Bridge (rev f3) (prog-if 01 [Subtractive decode]) Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0 Bus: primary=00, secondary=05, subordinate=08, sec-latency=32 I/O behind bridge: 00004000-00007fff Memory behind bridge: d4000000-d7efffff Prefetchable memory behind bridge: 00000000d8000000-00000000dbffffff Secondary status: 66MHz- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort+ <SERR- <PERR- BridgeCtl: Parity- SERR- NoISA- VGA- MAbort- >Reset- FastB2B- PriDiscTmr- SecDiscTmr- DiscTmrStat- DiscTmrSERREn- Capabilities: [50] Subsystem: Lenovo ThinkPad T61/R61 00:1f.0 ISA bridge: Intel Corporation 82801HEM (ICH8M-E) LPC Interface Controller (rev 03) Subsystem: Lenovo ThinkPad T61/R61 Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0 Capabilities: [e0] Vendor Specific Information: Len=0c <?> Kernel driver in use: lpc_ich 00:1f.2 SATA controller: Intel Corporation 82801HM/HEM (ICH8M/ICH8M-E) SATA Controller [AHCI mode] (rev 03) (prog-if 01 [AHCI 1.0]) Subsystem: Lenovo ThinkPad T61/R61 Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx+ Status: Cap+ 66MHz+ UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0 Interrupt: pin B routed to IRQ 44 Region 0: I/O ports at 1c08 [size=8] Region 1: I/O ports at 183c [size=4] Region 2: I/O ports at 1c00 [size=8] Region 3: I/O ports at 1838 [size=4] Region 4: I/O ports at 18e0 [size=32] Region 5: Memory at f8426000 (32-bit, non-prefetchable) [size=2K] Capabilities: [80] MSI: Enable+ Count=1/4 Maskable- 64bit- Address: fee0300c Data: 4122 Capabilities: [70] Power Management version 3 Flags: PMEClk- DSI- D1- D2- AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot+,D3cold-) Status: D0 NoSoftRst+ PME-Enable- DSel=0 DScale=0 PME- Capabilities: [a8] SATA HBA v1.0 BAR4 Offset=00000004 Kernel driver in use: ahci 00:1f.3 SMBus: Intel Corporation 82801H (ICH8 Family) SMBus Controller (rev 03) Subsystem: Lenovo ThinkPad T61/R61 Control: I/O+ Mem+ BusMaster- SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- DisINTx- Status: Cap- 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Interrupt: pin A routed to IRQ 11 Region 0: Memory at f8427400 (32-bit, non-prefetchable) [size=256] Region 4: I/O ports at 1c20 [size=32] 03:00.0 Network controller: Intel Corporation PRO/Wireless 4965 AG or AGN [Kedron] Network Connection (rev 61) Subsystem: Intel Corporation Lenovo ThinkPad T51 Physical Slot: 3 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR+ <PERR- INTx- Latency: 0, Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 45 Region 0: Memory at f7f00000 (64-bit, non-prefetchable) [size=8K] Capabilities: [c8] Power Management version 3 Flags: PMEClk- DSI+ D1- D2- AuxCurrent=0mA PME(D0+,D1-,D2-,D3hot+,D3cold+) Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- Capabilities: [d0] MSI: Enable+ Count=1/1 Maskable- 64bit+ Address: 00000000fee0300c Data: 4142 Capabilities: [e0] Express (v1) Endpoint, MSI 00 DevCap: MaxPayload 128 bytes, PhantFunc 0, Latency L0s <512ns, L1 unlimited ExtTag- AttnBtn- AttnInd- PwrInd- RBE+ FLReset- DevCtl: Report errors: Correctable- Non-Fatal- Fatal- Unsupported- RlxdOrd+ ExtTag- PhantFunc- AuxPwr- NoSnoop+ MaxPayload 128 bytes, MaxReadReq 128 bytes DevSta: CorrErr+ UncorrErr+ FatalErr- UnsuppReq+ AuxPwr+ TransPend- LnkCap: Port #0, Speed 2.5GT/s, Width x1, ASPM L0s L1, Exit Latency L0s <128ns, L1 <64us ClockPM+ Surprise- LLActRep- BwNot- LnkCtl: ASPM Disabled; RCB 64 bytes Disabled- CommClk+ ExtSynch- ClockPM- AutWidDis- BWInt- AutBWInt- LnkSta: Speed 2.5GT/s, Width x1, TrErr- Train- SlotClk+ DLActive- BWMgmt- ABWMgmt- Capabilities: [100 v1] Advanced Error Reporting UESta: DLP- SDES- TLP- FCP- CmpltTO- CmpltAbrt- UnxCmplt- RxOF- MalfTLP- ECRC- UnsupReq+ ACSViol- UEMsk: DLP- SDES- TLP- FCP- CmpltTO- CmpltAbrt- UnxCmplt- RxOF- MalfTLP- ECRC- UnsupReq- ACSViol- UESvrt: DLP+ SDES- TLP- FCP+ CmpltTO- CmpltAbrt- UnxCmplt- RxOF+ MalfTLP+ ECRC- UnsupReq- ACSViol- CESta: RxErr+ BadTLP- BadDLLP- Rollover- Timeout- NonFatalErr+ CEMsk: RxErr- BadTLP- BadDLLP- Rollover- Timeout- NonFatalErr+ AERCap: First Error Pointer: 14, GenCap- CGenEn- ChkCap- ChkEn- Capabilities: [140 v1] Device Serial Number 00-1d-e0-ff-ff-6b-9c-3f Kernel driver in use: iwl4965 05:00.0 CardBus bridge: Ricoh Co Ltd RL5c476 II (rev ba) Subsystem: Lenovo ThinkPad R61 Physical Slot: 1 Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 168, Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 16 Region 0: Memory at d7eff000 (32-bit, non-prefetchable) [size=4K] Bus: primary=05, secondary=06, subordinate=07, sec-latency=176 Memory window 0: d8000000-dbffffff (prefetchable) Memory window 1: c0000000-c3ffffff I/O window 0: 00004000-000040ff I/O window 1: 00004400-000044ff BridgeCtl: Parity- SERR- ISA- VGA- MAbort- >Reset- 16bInt+ PostWrite+ 16-bit legacy interface ports at 0001 Capabilities: [dc] Power Management version 2 Flags: PMEClk+ DSI- D1+ D2+ AuxCurrent=0mA PME(D0+,D1+,D2+,D3hot+,D3cold+) Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=2 PME- Bridge: PM- B3+ Kernel driver in use: yenta_cardbus 05:00.2 SD Host controller: Ricoh Co Ltd R5C822 SD/SDIO/MMC/MS/MSPro Host Adapter (rev 21) Subsystem: Lenovo ThinkPad W500 Physical Slot: 1 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 64 Interrupt: pin C routed to IRQ 18 Region 0: Memory at d7efec00 (32-bit, non-prefetchable) [size=256] Capabilities: [80] Power Management version 2 Flags: PMEClk- DSI- D1+ D2+ AuxCurrent=0mA PME(D0+,D1+,D2+,D3hot+,D3cold+) Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=2 PME- Kernel driver in use: sdhci-pci [7.6.] SCSI information (from /proc/scsi/scsi) Attached devices: Host: scsi0 Channel: 00 Id: 00 Lun: 00 Vendor: ATA Model: ST9200420AS Rev: 3.AA Type: Direct-Access ANSI SCSI revision: 05 [7.7.] Other information that might be relevant to the problem (please look in /proc and include all information that you think to be relevant): [X.] Other notes, patches, fixes, workarounds: https://bugs.launchpad.net/ubuntu/+source/linux/+bug/1324384 ---- ALSA info for 5ccc618fee67f0f0b2122dd4b32a02fd2b6a1569 BEGIN ---- upload=true&script=true&cardinfo= !!################################ !!ALSA Information Script v 0.4.60 !!################################ !!Script ran on: Sun Jun 1 02:56:49 UTC 2014 !!Linux Distribution Ubuntu 14.04 LTS \n \l DISTRIB_ID=Ubuntu DISTRIB_DESCRIPTION="Ubuntu 14.04 LTS" NAME="Ubuntu" ID=ubuntu ID_LIKE=debian PRETTY_NAME="Ubuntu 14.04 LTS" HOME_URL="http://www.ubuntu.com/" SUPPORT_URL="http://help.ubuntu.com/" BUG_REPORT_URL="http://bugs.launchpad.net/ubuntu/" !!DMI Information Manufacturer: LENOVO Product Name: 76693JG Product Version: ThinkPad X61s !!Kernel Information Kernel release: 3.10-rc7-359-g5ccc618-generic Operating System: GNU/Linux Architecture: x86_64 Processor: x86_64 SMP Enabled: Yes !!ALSA Version Driver version: k3.10-rc7-359-g5ccc618-generic Library version: 1.0.27.2 Utilities version: 1.0.27.2 !!Loaded ALSA modules snd_hda_intel thinkpad_acpi !!Sound Servers on this system Pulseaudio: Installed - Yes (/usr/bin/pulseaudio) Running - Yes !!Soundcards recognised by ALSA 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0xf8220000 irq 46 4 [ThinkPadEC ]: ThinkPad EC - ThinkPad Console Audio Control ThinkPad Console Audio Control at EC reg 0x30, fw 7MHT25WW-1.03 !!PCI Soundcards installed in the system 00:1b.0 Audio device: Intel Corporation 82801H (ICH8 Family) HD Audio Controller (rev 03) !!Advanced information - PCI Vendor/Device/Subsystem ID's 00:1b.0 0403: 8086:284b (rev 03) Subsystem: 17aa:20ac !!Modprobe options (Sound related) snd_atiixp_modem: index=-2 snd_intel8x0m: index=-2 snd_via82xx_modem: index=-2 snd_usb_audio: index=-2 snd_usb_caiaq: index=-2 snd_usb_ua101: index=-2 snd_usb_us122l: index=-2 snd_usb_usx2y: index=-2 snd_cmipci: mpu_port=0x330 fm_port=0x388 snd_pcsp: index=-2 snd_usb_audio: index=-2 !!Loaded sound module options !!Module: snd_hda_intel align_buffer_size : -1 bdl_pos_adj : 1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 beep_mode : N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N enable : Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y enable_msi : -1 id : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) index : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 jackpoll_ms : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 model : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) patch : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) position_fix : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 power_save : 1 power_save_controller : Y probe_mask : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 probe_only : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 single_cmd : N snoop : Y !!Module: thinkpad_acpi brightness_enable : 2 brightness_mode : 4 dbg_bluetoothemul : 0 dbg_uwbemul : 0 dbg_wlswemul : 0 dbg_wwanemul : 0 enable : Y experimental : 0 fan_control : N force_load : N hotkey_report_mode : 0 id : ThinkPadEC index : -536870912 volume_capabilities : 0 volume_control : N volume_mode : 3 !!HDA-Intel Codec information --startcollapse-- Codec: Analog Devices AD1984 Address: 0 AFG Function Id: 0x1 (unsol 0) Vendor Id: 0x11d41984 Subsystem Id: 0x17aa20d6 Revision Id: 0x100400 No Modem Function Group found Default PCM: rates [0x7ff]: 8000 11025 16000 22050 32000 44100 48000 88200 96000 176400 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Default Amp-Out caps: ofs=0x27, nsteps=0x27, stepsize=0x05, mute=0 State of AFG node 0x01: Power states: D0 D3 Power: setting=D0, actual=D0 GPIO: io=3, o=0, i=0, unsolicited=1, wake=0 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[1]: enable=1, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[2]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 Node 0x02 [Audio Output] wcaps 0x30311: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x7e0]: 44100 48000 88200 96000 176400 192000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Delay: 3 samples Connection: 3 0x01* 0x08 0x09 Node 0x03 [Audio Output] wcaps 0x405: Stereo Amp-Out Control: name="Headphone Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Device: name="AD1984 Analog", type="Audio", device=0 Device: name="AD1984 Alt Analog", type="Audio", device=2 Amp-Out caps: ofs=0x27, nsteps=0x27, stepsize=0x05, mute=0 Amp-Out vals: [0x27 0x27] Converter: stream=0, channel=0 Power states: D0 D3 Power: setting=D0, actual=D0 Node 0x04 [Audio Output] wcaps 0x405: Stereo Amp-Out Control: name="Speaker Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x27, nsteps=0x27, stepsize=0x05, mute=0 Amp-Out vals: [0x00 0x00] Converter: stream=0, channel=0 Power states: D0 D3 Power: setting=D0, actual=D0 Node 0x05 [Audio Input] wcaps 0x10050b: Stereo Amp-In Amp-In caps: ofs=0x27, nsteps=0x36, stepsize=0x05, mute=1 Amp-In vals: [0xa7 0xa7] Converter: stream=0, channel=0 SDI-Select: 0 Power states: D0 D3 Power: setting=D0, actual=D0 Connection: 1 0x17 Node 0x06 [Audio Input] wcaps 0x10050b: Stereo Amp-In Amp-In caps: ofs=0x27, nsteps=0x36, stepsize=0x05, mute=1 Amp-In vals: [0xa7 0xa7] Converter: stream=0, channel=0 SDI-Select: 0 Power states: D0 D3 Power: setting=D0, actual=D0 Connection: 1 0x18 Node 0x07 [Audio Mixer] wcaps 0x200103: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] Connection: 2 0x22 0x21 Node 0x08 [Audio Input] wcaps 0x100501: Stereo Device: name="AD1984 Analog", type="Audio", device=0 Converter: stream=0, channel=0 SDI-Select: 0 Power states: D0 D3 Power: setting=D0, actual=D0 Connection: 1 0x0c Node 0x09 [Audio Input] wcaps 0x100501: Stereo Converter: stream=0, channel=0 SDI-Select: 0 Power states: D0 D3 Power: setting=D0, actual=D0 Connection: 1 0x0d Node 0x0a [Audio Mixer] wcaps 0x200103: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] Connection: 2 0x04 0x21 Node 0x0b [Audio Mixer] wcaps 0x200103: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] Connection: 2 0x0f 0x21 Node 0x0c [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Control: name="Capture Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Capture Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x27, nsteps=0x36, stepsize=0x05, mute=1 Amp-Out vals: [0x28 0x28] Connection: 5 0x14 0x15* 0x16 0x20 0x25 Node 0x0d [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x27, nsteps=0x36, stepsize=0x05, mute=1 Amp-Out vals: [0xa7 0xa7] Connection: 5 0x14* 0x15 0x16 0x20 0x25 Node 0x0e [Audio Selector] wcaps 0x300101: Stereo Connection: 2 0x03* 0x04 Node 0x0f [Audio Selector] wcaps 0x300101: Stereo Connection: 2 0x03* 0x04 Node 0x10 [Beep Generator Widget] wcaps 0x70000c: Mono Amp-Out Control: name="Beep Playback Volume", index=0, device=0 ControlAmp: chs=1, dir=Out, idx=0, ofs=0 Control: name="Beep Playback Switch", index=0, device=0 ControlAmp: chs=1, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x0f, nsteps=0x0f, stepsize=0x0b, mute=1 Amp-Out vals: [0x0f] Node 0x11 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Control: name="Headphone Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Headphone Jack", index=0, device=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000001f: OUT HP Detect Trigger ImpSense Pin Default 0x0421401f: [Jack] HP Out at Ext Right Conn = 1/8, Color = Green DefAssociation = 0x1, Sequence = 0xf Pin-ctls: 0xc0: OUT HP Unsolicited: tag=01, enabled=1 Connection: 1 0x07 Node 0x12 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Control: name="Speaker Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Speaker Phantom Jack", index=0, device=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0001001f: OUT HP EAPD Detect Trigger ImpSense EAPD 0x0: Pin Default 0x90170110: [Fixed] Speaker at Int N/A Conn = Analog, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power states: D0 D3 Power: setting=D0, actual=D0 Connection: 1 0x0a Node 0x13 [Pin Complex] wcaps 0x40050c: Mono Amp-Out Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x9f] Pincap 0x00010010: OUT EAPD EAPD 0x0: Pin Default 0x511301f0: [N/A] Speaker at Int Rear Conn = ATAPI, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Power states: D0 D3 Power: setting=D0, actual=D0 Connection: 1 0x1f Node 0x14 [Pin Complex] wcaps 0x40008b: Stereo Amp-In Control: name="Mic Boost Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Mic Jack", index=0, device=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Pincap 0x00003727: IN Detect Trigger ImpSense Vref caps: HIZ 50 GRD 80 100 Pin Default 0x04a15021: [Jack] Mic at Ext Right Conn = 1/8, Color = Red DefAssociation = 0x2, Sequence = 0x1 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=03, enabled=1 Node 0x15 [Pin Complex] wcaps 0x40008b: Stereo Amp-In Control: name="Internal Mic Boost Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Internal Mic Phantom Jack", index=0, device=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Pincap 0x00003727: IN Detect Trigger ImpSense Vref caps: HIZ 50 GRD 80 100 Pin Default 0x90a7012e: [Fixed] Mic at Int N/A Conn = Analog, Color = Unknown DefAssociation = 0x2, Sequence = 0xe Misc = NO_PRESENCE Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=00, enabled=0 Node 0x16 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x00000037: IN OUT Detect Trigger ImpSense Pin Default 0x593301f0: [N/A] CD at Int ATAPI Conn = ATAPI, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Connection: 1 0x0b Node 0x17 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x00000020: IN Pin Default 0x55a601f0: [N/A] Mic at Int Top Conn = Digital, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x24: IN Node 0x18 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x00000020: IN Pin Default 0x55a601f0: [N/A] Mic at Int Top Conn = Digital, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Node 0x19 [Power Widget] wcaps 0x500500: Mono Power states: D0 D3 Power: setting=D0, actual=D0 Connection: 2 0x20 0x21 Node 0x1a [Pin Complex] wcaps 0x400000: Mono Pincap 0x00000020: IN Pin Default 0x91f311f0: [Fixed] Other at Int Rear Conn = ATAPI, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Node 0x1b [Pin Complex] wcaps 0x40030d: Stereo Digital Amp-Out Amp-Out caps: ofs=0x27, nsteps=0x27, stepsize=0x05, mute=1 Amp-Out vals: [0xa7 0xa7] Pincap 0x00000010: OUT Pin Default 0x614411f0: [N/A] SPDIF Out at Sep Rear Conn = RCA, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Connection: 1 0x02 Node 0x1c [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Control: name="Dock Mic Jack", index=0, device=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x00003737: IN OUT Detect Trigger ImpSense Vref caps: HIZ 50 GRD 80 100 Pin Default 0x21a15020: [Jack] Mic at Sep Rear Conn = 1/8, Color = Red DefAssociation = 0x2, Sequence = 0x0 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=02, enabled=1 Connection: 1 0x24 Node 0x1d [Vendor Defined Widget] wcaps 0xf00100: Mono Connection: 25 0x07* 0x19 0x0a 0x0b 0x0c 0x0d 0x0e 0x0f 0x1a 0x1c 0x11 0x12 0x13 0x14 0x15 0x16 0x1e 0x1f 0x20 0x21 0x22 0x23 0x24 0x25 0x26 Node 0x1e [Audio Mixer] wcaps 0x200103: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] Connection: 2 0x0e 0x21 Node 0x1f [Audio Mixer] wcaps 0x200100: Mono Connection: 1 0x1e Node 0x20 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Control: name="Internal Mic Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=1, ofs=0 Control: name="Internal Mic Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=1, ofs=0 Control: name="Dock Mic Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=4, ofs=0 Control: name="Dock Mic Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=4, ofs=0 Control: name="Mic Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Mic Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x80 0x80] [0x9f 0x9f] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 5 0x14 0x15 0x16 0x1a 0x25 Node 0x21 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x9f 0x9f] Connection: 1 0x20 Node 0x22 [Audio Selector] wcaps 0x300101: Stereo Connection: 2 0x03* 0x04 Node 0x23 [Audio Selector] wcaps 0x300101: Stereo Connection: 2 0x03* 0x04 Node 0x24 [Audio Mixer] wcaps 0x200103: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] Connection: 2 0x23 0x21 Node 0x25 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Control: name="Dock Mic Boost Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-Out vals: [0x00 0x00] Connection: 1 0x1c Node 0x26 [Vendor Defined Widget] wcaps 0xf00100: Mono Connection: 3 0x14* 0x15 0x1c --endcollapse-- !!ALSA Device nodes crw-rw----+ 1 root audio 116, 7 May 31 18:02 /dev/snd/controlC0 crw-rw----+ 1 root audio 116, 2 May 31 18:02 /dev/snd/controlC4 crw-rw----+ 1 root audio 116, 6 May 31 18:02 /dev/snd/hwC0D0 crw-rw----+ 1 root audio 116, 5 May 31 18:07 /dev/snd/pcmC0D0c crw-rw----+ 1 root audio 116, 4 May 31 19:13 /dev/snd/pcmC0D0p crw-rw----+ 1 root audio 116, 3 May 31 18:02 /dev/snd/pcmC0D2p crw-rw----+ 1 root audio 116, 1 May 31 18:02 /dev/snd/seq crw-rw----+ 1 root audio 116, 33 May 31 18:02 /dev/snd/timer /dev/snd/by-path: total 0 drwxr-xr-x 2 root root 80 May 31 18:02 . drwxr-xr-x 3 root root 220 May 31 18:02 .. lrwxrwxrwx 1 root root 12 May 31 18:02 pci-0000:00:1b.0 -> ../controlC0 lrwxrwxrwx 1 root root 12 May 31 18:02 platform-thinkpad_acpi -> ../controlC4 !!Aplay/Arecord output APLAY **** List of PLAYBACK Hardware Devices **** card 0: Intel [HDA Intel], device 0: AD1984 Analog [AD1984 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Intel [HDA Intel], device 2: AD1984 Alt Analog [AD1984 Alt Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 ARECORD **** List of CAPTURE Hardware Devices **** card 0: Intel [HDA Intel], device 0: AD1984 Analog [AD1984 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 !!Amixer output !!-------Mixer controls for card 0 [Intel] Card hw:0 'Intel'/'HDA Intel at 0xf8220000 irq 46' Mixer name : 'Analog Devices AD1984' Components : 'HDA:11d41984,17aa20d6,00100400' Controls : 31 Simple ctrls : 15 Simple mixer control 'Master',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 39 Mono: Playback 39 [100%] [0.00dB] [on] Simple mixer control 'Headphone',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 39 Mono: Front Left: Playback 39 [100%] [0.00dB] [on] Front Right: Playback 39 [100%] [0.00dB] [on] Simple mixer control 'Speaker',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 39 Mono: Front Left: Playback 0 [0%] [-58.50dB] [off] Front Right: Playback 0 [0%] [-58.50dB] [off] Simple mixer control 'PCM',0 Capabilities: pvolume Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 255 [100%] [0.00dB] Front Right: Playback 255 [100%] [0.00dB] Simple mixer control 'Mic',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 0 [0%] [-34.50dB] [off] Front Right: Playback 0 [0%] [-34.50dB] [off] Simple mixer control 'Mic Boost',0 Capabilities: volume Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 3 Front Left: 0 [0%] [0.00dB] Front Right: 0 [0%] [0.00dB] Simple mixer control 'Beep',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 15 Mono: Playback 15 [100%] [0.00dB] [on] Simple mixer control 'Capture',0 Capabilities: cvolume cswitch Capture channels: Front Left - Front Right Limits: Capture 0 - 54 Front Left: Capture 40 [74%] [1.50dB] [on] Front Right: Capture 40 [74%] [1.50dB] [on] Simple mixer control 'Auto-Mute Mode',0 Capabilities: enum Items: 'Disabled' 'Enabled' Item0: 'Enabled' Simple mixer control 'Digital',0 Capabilities: cvolume Capture channels: Front Left - Front Right Limits: Capture 0 - 120 Front Left: Capture 60 [50%] [0.00dB] Front Right: Capture 60 [50%] [0.00dB] Simple mixer control 'Dock Mic',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 0 [0%] [-34.50dB] [off] Front Right: Playback 0 [0%] [-34.50dB] [off] Simple mixer control 'Dock Mic Boost',0 Capabilities: volume Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 3 Front Left: 0 [0%] [0.00dB] Front Right: 0 [0%] [0.00dB] Simple mixer control 'Independent HP',0 Capabilities: enum Items: 'Disabled' 'Enabled' Item0: 'Disabled' Simple mixer control 'Internal Mic',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 31 [100%] [12.00dB] [off] Front Right: Playback 31 [100%] [12.00dB] [off] Simple mixer control 'Internal Mic Boost',0 Capabilities: volume Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 3 Front Left: 0 [0%] [0.00dB] Front Right: 0 [0%] [0.00dB] !!-------Mixer controls for card 4 [ThinkPadEC] Card hw:4 'ThinkPadEC'/'ThinkPad Console Audio Control at EC reg 0x30, fw 7MHT25WW-1.03' Mixer name : 'ThinkPad EC 7MHT25WW-1.03' Components : '' Controls : 1 Simple ctrls : 1 Simple mixer control 'Console',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] !!Alsactl output --startcollapse-- state.Intel { control.1 { iface MIXER name 'Headphone Playback Volume' value.0 39 value.1 39 comment { access 'read write' type INTEGER count 2 range '0 - 39' dbmin -5850 dbmax 0 dbvalue.0 0 dbvalue.1 0 } } control.2 { iface MIXER name 'Headphone Playback Switch' value.0 true value.1 true comment { access 'read write' type BOOLEAN count 2 } } control.3 { iface MIXER name 'Speaker Playback Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 39' dbmin -5850 dbmax 0 dbvalue.0 -5850 dbvalue.1 -5850 } } control.4 { iface MIXER name 'Speaker Playback Switch' value.0 false value.1 false comment { access 'read write' type BOOLEAN count 2 } } control.5 { iface MIXER name 'Independent HP' value Disabled comment { access 'read write' type ENUMERATED count 1 item.0 Disabled item.1 Enabled } } control.6 { iface MIXER name 'Internal Mic Playback Volume' value.0 31 value.1 31 comment { access 'read write' type INTEGER count 2 range '0 - 31' dbmin -3450 dbmax 1200 dbvalue.0 1200 dbvalue.1 1200 } } control.7 { iface MIXER name 'Internal Mic Playback Switch' value.0 false value.1 false comment { access 'read write' type BOOLEAN count 2 } } control.8 { iface MIXER name 'Dock Mic Playback Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 31' dbmin -3450 dbmax 1200 dbvalue.0 -3450 dbvalue.1 -3450 } } control.9 { iface MIXER name 'Dock Mic Playback Switch' value.0 false value.1 false comment { access 'read write' type BOOLEAN count 2 } } control.10 { iface MIXER name 'Mic Playback Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 31' dbmin -3450 dbmax 1200 dbvalue.0 -3450 dbvalue.1 -3450 } } control.11 { iface MIXER name 'Mic Playback Switch' value.0 false value.1 false comment { access 'read write' type BOOLEAN count 2 } } control.12 { iface MIXER name 'Auto-Mute Mode' value Enabled comment { access 'read write' type ENUMERATED count 1 item.0 Disabled item.1 Enabled } } control.13 { iface MIXER name 'Capture Volume' value.0 40 value.1 40 comment { access 'read write' type INTEGER count 2 range '0 - 54' dbmin -5850 dbmax 2250 dbvalue.0 150 dbvalue.1 150 } } control.14 { iface MIXER name 'Capture Switch' value.0 true value.1 true comment { access 'read write' type BOOLEAN count 2 } } control.15 { iface MIXER name 'Internal Mic Boost Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 3' dbmin 0 dbmax 3000 dbvalue.0 0 dbvalue.1 0 } } control.16 { iface MIXER name 'Dock Mic Boost Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 3' dbmin 0 dbmax 3000 dbvalue.0 0 dbvalue.1 0 } } control.17 { iface MIXER name 'Mic Boost Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 3' dbmin 0 dbmax 3000 dbvalue.0 0 dbvalue.1 0 } } control.18 { iface MIXER name 'Master Playback Volume' value 39 comment { access 'read write' type INTEGER count 1 range '0 - 39' dbmin -5850 dbmax 0 dbvalue.0 0 } } control.19 { iface MIXER name 'Master Playback Switch' value true comment { access 'read write' type BOOLEAN count 1 } } control.20 { iface CARD name 'Internal Mic Phantom Jack' value true comment { access read type BOOLEAN count 1 } } control.21 { iface CARD name 'Dock Mic Jack' value false comment { access read type BOOLEAN count 1 } } control.22 { iface CARD name 'Mic Jack' value false comment { access read type BOOLEAN count 1 } } control.23 { iface CARD name 'Headphone Jack' value true comment { access read type BOOLEAN count 1 } } control.24 { iface CARD name 'Speaker Phantom Jack' value true comment { access read type BOOLEAN count 1 } } control.25 { iface MIXER name 'Beep Playback Volume' value 15 comment { access 'read write' type INTEGER count 1 range '0 - 15' dbmin -4500 dbmax 0 dbvalue.0 0 } } control.26 { iface MIXER name 'Beep Playback Switch' value true comment { access 'read write' type BOOLEAN count 1 } } control.27 { iface PCM name 'Playback Channel Map' value.0 0 value.1 0 comment { access read type INTEGER count 2 range '0 - 36' } } control.28 { iface PCM name 'Capture Channel Map' value.0 0 value.1 0 comment { access read type INTEGER count 2 range '0 - 36' } } control.29 { iface PCM device 2 name 'Playback Channel Map' value.0 0 value.1 0 comment { access read type INTEGER count 2 range '0 - 36' } } control.30 { iface MIXER name 'Digital Capture Volume' value.0 60 value.1 60 comment { access 'read write user' type INTEGER count 2 range '0 - 120' tlv '0000000100000008fffff44800000032' dbmin -3000 dbmax 3000 dbvalue.0 0 dbvalue.1 0 } } control.31 { iface MIXER name 'PCM Playback Volume' value.0 255 value.1 255 comment { access 'read write user' type INTEGER count 2 range '0 - 255' tlv '0000000100000008ffffec1400000014' dbmin -5100 dbmax 0 dbvalue.0 0 dbvalue.1 0 } } } state.ThinkPadEC { control.1 { iface MIXER name 'Console Playback Switch' value true comment { access read type BOOLEAN count 1 } } } --endcollapse-- !!All Loaded Modules Module btrfs raid6_pq zlib_deflate xor ufs qnx4 hfsplus hfs minix ntfs msdos jfs xfs libcrc32c nls_iso8859_1 ipt_MASQUERADE iptable_nat nf_nat_ipv4 nf_nat nf_conntrack_ipv4 nf_defrag_ipv4 xt_conntrack nf_conntrack ipt_REJECT xt_CHECKSUM iptable_mangle xt_tcpudp bridge stp llc ip6table_filter ip6_tables iptable_filter ip_tables ebtable_nat ebtables x_tables bnep rfcomm bluetooth binfmt_misc snd_hda_codec_analog dm_multipath snd_hda_intel arc4 snd_hda_codec scsi_dh snd_hwdep snd_pcm iwl4965 thinkpad_acpi iwlegacy mac80211 nvram snd_page_alloc snd_seq_midi snd_seq_midi_event cfg80211 coretemp snd_rawmidi kvm_intel snd_seq snd_seq_device pcmcia kvm snd_timer serio_raw snd parport_pc soundcore ppdev yenta_socket pcmcia_rsrc pcmcia_core lpc_ich lp mac_hid parport xts gf128mul dm_crypt mmc_block psmouse sdhci_pci sdhci ahci libahci i915 i2c_algo_bit drm_kms_helper drm e1000e video ptp pps_core !!Sysfs Files /sys/class/sound/hwC0D0/init_pin_configs: 0x11 0x0421401f 0x12 0x90170110 0x13 0x511301f0 0x14 0x04a15021 0x15 0x90a7012e 0x16 0x593301f0 0x17 0x55a601f0 0x18 0x55a601f0 0x1a 0x91f311f0 0x1b 0x614411f0 0x1c 0x21a15020 /sys/class/sound/hwC0D0/driver_pin_configs: /sys/class/sound/hwC0D0/user_pin_configs: /sys/class/sound/hwC0D0/init_verbs: !!ALSA/HDA dmesg [ 35.045988] ieee80211 phy0: Selected rate control algorithm 'iwl-4965-rs' [ 35.076727] hda_intel: probe_mask set to 0x1 for device 17aa:20ac [ 35.076782] snd_hda_intel 0000:00:1b.0: irq 46 for MSI/MSI-X [ 35.084674] device-mapper: multipath: version 1.5.1 loaded -- [ 35.496579] Mic=0x14 [ 35.504968] input: HDA Intel Headphone as /devices/pci0000:00/0000:00:1b.0/sound/card0/input7 [ 35.505383] input: HDA Intel Mic as /devices/pci0000:00/0000:00:1b.0/sound/card0/input8 [ 35.506228] input: HDA Intel Dock Mic as /devices/pci0000:00/0000:00:1b.0/sound/card0/input9 [ 35.569962] pcmcia_socket pcmcia_socket0: cs: memory probe 0x0c0000-0x0fffff: ---- ALSA info for 5ccc618fee67f0f0b2122dd4b32a02fd2b6a1569 END ---- I had do do diff between 5ccc618f and aa95d61b4 to fit 100k email limit. ---- ALSA info for aa95d61b43e0fcb0b2ce68e5efa37174fd9e5cd3 BEGIN ---- 6c6 < !!Script ran on: Sun Jun 1 02:56:49 UTC 2014 --- > !!Script ran on: Sun Jun 1 05:51:12 UTC 2014 26c26 < Kernel release: 3.10-rc7-359-g5ccc618-generic --- > Kernel release: 3.10-rc7-358-gaa95d61-generic 36c36 < Driver version: k3.10-rc7-359-g5ccc618-generic --- > Driver version: k3.10-rc7-358-gaa95d61-generic 60c60 < HDA Intel at 0xf8220000 irq 46 --- > HDA Intel at 0xf8220000 irq 45 156c156 < IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 --- > IO[0]: enable=0, dir=0, wake=0, sticky=0, data=1, unsol=0 160c160,167 < Converter: stream=0, channel=0 --- > Control: name="IEC958 Playback Con Mask", index=0, device=0 > Control: name="IEC958 Playback Pro Mask", index=0, device=0 > Control: name="IEC958 Playback Default", index=0, device=0 > Control: name="IEC958 Playback Switch", index=0, device=0 > Control: name="IEC958 Default PCM Playback Switch", index=0, device=0 > Control: name="IEC958 Playback Source", index=0, device=0 > Device: name="AD198x Digital", type="SPDIF", device=1 > Converter: stream=8, channel=0 172,175d178 < Control: name="Headphone Playback Volume", index=0, device=0 < ControlAmp: chs=3, dir=Out, idx=0, ofs=0 < Device: name="AD1984 Analog", type="Audio", device=0 < Device: name="AD1984 Alt Analog", type="Audio", device=2 177c180 < Amp-Out vals: [0x27 0x27] --- > Amp-Out vals: [0x00 0x00] 182c185 < Control: name="Speaker Playback Volume", index=0, device=0 --- > Control: name="PCM Playback Volume", index=0, device=0 183a187 > Device: name="AD198x Analog", type="Audio", device=0 185,186c189,190 < Amp-Out vals: [0x00 0x00] < Converter: stream=0, channel=0 --- > Amp-Out vals: [0x27 0x27] > Converter: stream=8, channel=0 209c213 < Amp-In vals: [0x00 0x00] [0x80 0x80] --- > Amp-In vals: [0x00 0x00] [0x00 0x00] 213c217 < Device: name="AD1984 Analog", type="Audio", device=0 --- > Device: name="AD198x Analog", type="Audio", device=0 229c233 < Amp-In vals: [0x00 0x00] [0x80 0x80] --- > Amp-In vals: [0x00 0x00] [0x00 0x00] 241a246 > Control: name="Input Source", index=0, device=0 246a252,256 > Control: name="Capture Volume", index=1, device=0 > ControlAmp: chs=3, dir=Out, idx=0, ofs=0 > Control: name="Capture Switch", index=1, device=0 > ControlAmp: chs=3, dir=Out, idx=0, ofs=0 > Control: name="Input Source", index=1, device=0 253c263 < 0x03* 0x04 --- > 0x03 0x04* 258c268 < Control: name="Beep Playback Volume", index=0, device=0 --- > Control: name="Digital Beep Playback Volume", index=0, device=0 260c270 < Control: name="Beep Playback Switch", index=0, device=0 --- > Control: name="Digital Beep Playback Switch", index=0, device=0 263c273 < Amp-Out vals: [0x0f] --- > Amp-Out vals: [0x8f] 267d276 < Control: name="Headphone Jack", index=0, device=0 275c284 < Unsolicited: tag=01, enabled=1 --- > Unsolicited: tag=00, enabled=0 281d289 < Control: name="Speaker Phantom Jack", index=0, device=0 283c291 < Amp-Out vals: [0x80 0x80] --- > Amp-Out vals: [0x00 0x00] 285c293 < EAPD 0x0: --- > EAPD 0x2: EAPD 290c298 < Pin-ctls: 0x40: OUT --- > Pin-ctls: 0xc0: OUT HP 298c306 < Amp-Out vals: [0x9f] --- > Amp-Out vals: [0x80] 313d320 < Control: name="Mic Jack", index=0, device=0 322c329 < Unsolicited: tag=03, enabled=1 --- > Unsolicited: tag=00, enabled=0 326d332 < Control: name="Internal Mic Phantom Jack", index=0, device=0 375a382,383 > Control: name="IEC958 Playback Volume", index=0, device=0 > ControlAmp: chs=3, dir=Out, idx=0, ofs=0 377c385 < Amp-Out vals: [0xa7 0xa7] --- > Amp-Out vals: [0x27 0x27] 387d394 < Control: name="Dock Mic Jack", index=0, device=0 396c403 < Unsolicited: tag=02, enabled=1 --- > Unsolicited: tag=00, enabled=0 404c411 < Amp-In vals: [0x80 0x80] [0x80 0x80] --- > Amp-In vals: [0x00 0x00] [0x00 0x00] 410a418,421 > Control: name="Mic Playback Volume", index=0, device=0 > ControlAmp: chs=3, dir=In, idx=0, ofs=0 > Control: name="Mic Playback Switch", index=0, device=0 > ControlAmp: chs=3, dir=In, idx=0, ofs=0 414a426,429 > Control: name="Beep Playback Volume", index=0, device=0 > ControlAmp: chs=3, dir=In, idx=3, ofs=0 > Control: name="Beep Playback Switch", index=0, device=0 > ControlAmp: chs=3, dir=In, idx=3, ofs=0 419,422d433 < Control: name="Mic Playback Volume", index=0, device=0 < ControlAmp: chs=3, dir=In, idx=0, ofs=0 < Control: name="Mic Playback Switch", index=0, device=0 < ControlAmp: chs=3, dir=In, idx=0, ofs=0 424c435 < Amp-In vals: [0x80 0x80] [0x9f 0x9f] [0x80 0x80] [0x80 0x80] [0x80 0x80] --- > Amp-In vals: [0x80 0x80] [0x9f 0x9f] [0x80 0x80] [0x17 0x17] [0x80 0x80] 429c440 < Amp-Out vals: [0x9f 0x9f] --- > Amp-Out vals: [0x1f 0x1f] 434c445 < 0x03* 0x04 --- > 0x03 0x04* 459,466c470,477 < crw-rw----+ 1 root audio 116, 7 May 31 18:02 /dev/snd/controlC0 < crw-rw----+ 1 root audio 116, 2 May 31 18:02 /dev/snd/controlC4 < crw-rw----+ 1 root audio 116, 6 May 31 18:02 /dev/snd/hwC0D0 < crw-rw----+ 1 root audio 116, 5 May 31 18:07 /dev/snd/pcmC0D0c < crw-rw----+ 1 root audio 116, 4 May 31 19:13 /dev/snd/pcmC0D0p < crw-rw----+ 1 root audio 116, 3 May 31 18:02 /dev/snd/pcmC0D2p < crw-rw----+ 1 root audio 116, 1 May 31 18:02 /dev/snd/seq < crw-rw----+ 1 root audio 116, 33 May 31 18:02 /dev/snd/timer --- > crw-rw----+ 1 root audio 116, 7 May 31 22:39 /dev/snd/controlC0 > crw-rw----+ 1 root audio 116, 2 May 31 22:39 /dev/snd/controlC4 > crw-rw----+ 1 root audio 116, 6 May 31 22:39 /dev/snd/hwC0D0 > crw-rw----+ 1 root audio 116, 5 May 31 22:44 /dev/snd/pcmC0D0c > crw-rw----+ 1 root audio 116, 4 May 31 22:44 /dev/snd/pcmC0D0p > crw-rw----+ 1 root audio 116, 3 May 31 22:44 /dev/snd/pcmC0D1p > crw-rw----+ 1 root audio 116, 1 May 31 22:39 /dev/snd/seq > crw-rw----+ 1 root audio 116, 33 May 31 22:39 /dev/snd/timer 470,473c481,484 < drwxr-xr-x 2 root root 80 May 31 18:02 . < drwxr-xr-x 3 root root 220 May 31 18:02 .. < lrwxrwxrwx 1 root root 12 May 31 18:02 pci-0000:00:1b.0 -> ../controlC0 < lrwxrwxrwx 1 root root 12 May 31 18:02 platform-thinkpad_acpi -> ../controlC4 --- > drwxr-xr-x 2 root root 80 May 31 22:39 . > drwxr-xr-x 3 root root 220 May 31 22:39 .. > lrwxrwxrwx 1 root root 12 May 31 22:39 pci-0000:00:1b.0 -> ../controlC0 > lrwxrwxrwx 1 root root 12 May 31 22:39 platform-thinkpad_acpi -> ../controlC4 482,483c493,494 < card 0: Intel [HDA Intel], device 0: AD1984 Analog [AD1984 Analog] < Subdevices: 1/1 --- > card 0: Intel [HDA Intel], device 0: AD198x Analog [AD198x Analog] > Subdevices: 0/1 485c496 < card 0: Intel [HDA Intel], device 2: AD1984 Alt Analog [AD1984 Alt Analog] --- > card 0: Intel [HDA Intel], device 1: AD198x Digital [AD198x Digital] 492,493c503,504 < card 0: Intel [HDA Intel], device 0: AD1984 Analog [AD1984 Analog] < Subdevices: 1/1 --- > card 0: Intel [HDA Intel], device 0: AD198x Analog [AD198x Analog] > Subdevices: 2/2 494a506 > Subdevice #1: subdevice #1 501c513 < Card hw:0 'Intel'/'HDA Intel at 0xf8220000 irq 46' --- > Card hw:0 'Intel'/'HDA Intel at 0xf8220000 irq 45' 504,505c516,517 < Controls : 31 < Simple ctrls : 15 --- > Controls : 36 > Simple ctrls : 20 512c524 < Capabilities: pvolume pswitch --- > Capabilities: pswitch 514d525 < Limits: Playback 0 - 39 516,517c527,528 < Front Left: Playback 39 [100%] [0.00dB] [on] < Front Right: Playback 39 [100%] [0.00dB] [on] --- > Front Left: Playback [on] > Front Right: Playback [on] 519c530 < Capabilities: pvolume pswitch --- > Capabilities: pswitch 521d531 < Limits: Playback 0 - 39 523,524c533,534 < Front Left: Playback 0 [0%] [-58.50dB] [off] < Front Right: Playback 0 [0%] [-58.50dB] [off] --- > Front Left: Playback [on] > Front Right: Playback [on] 528c538 < Limits: Playback 0 - 255 --- > Limits: Playback 0 - 39 530,531c540,541 < Front Left: Playback 255 [100%] [0.00dB] < Front Right: Playback 255 [100%] [0.00dB] --- > Front Left: Playback 39 [100%] [0.00dB] > Front Right: Playback 39 [100%] [0.00dB] 546,547c556,564 < Simple mixer control 'Beep',0 < Capabilities: pvolume pvolume-joined pswitch pswitch-joined --- > Simple mixer control 'IEC958',0 > Capabilities: pvolume pswitch pswitch-joined > Playback channels: Front Left - Front Right > Limits: Playback 0 - 39 > Mono: > Front Left: Playback 39 [100%] [0.00dB] [off] > Front Right: Playback 39 [100%] [0.00dB] [off] > Simple mixer control 'IEC958 Default PCM',0 > Capabilities: pswitch pswitch-joined 549,550c566,577 < Limits: Playback 0 - 15 < Mono: Playback 15 [100%] [0.00dB] [on] --- > Mono: Playback [on] > Simple mixer control 'IEC958 Playback Source',0 > Capabilities: enum > Items: 'PCM' 'ADC' > Item0: 'PCM' > Simple mixer control 'Beep',0 > Capabilities: pvolume pswitch > Playback channels: Front Left - Front Right > Limits: Playback 0 - 31 > Mono: > Front Left: Playback 23 [74%] [0.00dB] [on] > Front Right: Playback 23 [74%] [0.00dB] [on] 557,560c584,589 < Simple mixer control 'Auto-Mute Mode',0 < Capabilities: enum < Items: 'Disabled' 'Enabled' < Item0: 'Enabled' --- > Simple mixer control 'Capture',1 > Capabilities: cvolume cswitch > Capture channels: Front Left - Front Right > Limits: Capture 0 - 54 > Front Left: Capture 39 [72%] [0.00dB] [off] > Front Right: Capture 39 [72%] [0.00dB] [off] 566a596,600 > Simple mixer control 'Digital Beep',0 > Capabilities: pvolume pvolume-joined pswitch pswitch-joined > Playback channels: Mono > Limits: Playback 0 - 15 > Mono: Playback 15 [100%] [0.00dB] [off] 581,584c615,622 < Simple mixer control 'Independent HP',0 < Capabilities: enum < Items: 'Disabled' 'Enabled' < Item0: 'Disabled' --- > Simple mixer control 'Input Source',0 > Capabilities: cenum > Items: 'Mic' 'Internal Mic' 'Mix' 'Dock Mic' > Item0: 'Internal Mic' > Simple mixer control 'Input Source',1 > Capabilities: cenum > Items: 'Mic' 'Internal Mic' 'Mix' 'Dock Mic' > Item0: 'Mic' 620c658 < name 'Headphone Playback Volume' --- > name 'PCM Playback Volume' 647,649c685,687 < name 'Speaker Playback Volume' < value.0 0 < value.1 0 --- > name 'Speaker Playback Switch' > value.0 true > value.1 true 652c690 < type INTEGER --- > type BOOLEAN 654,658d691 < range '0 - 39' < dbmin -5850 < dbmax 0 < dbvalue.0 -5850 < dbvalue.1 -5850 663,665c696,698 < name 'Speaker Playback Switch' < value.0 false < value.1 false --- > name 'Mic Playback Volume' > value.0 0 > value.1 0 668c701 < type BOOLEAN --- > type INTEGER 669a703,707 > range '0 - 31' > dbmin -3450 > dbmax 1200 > dbvalue.0 -3450 > dbvalue.1 -3450 674,675c712,714 < name 'Independent HP' < value Disabled --- > name 'Mic Playback Switch' > value.0 false > value.1 false 678,681c717,718 < type ENUMERATED < count 1 < item.0 Disabled < item.1 Enabled --- > type BOOLEAN > count 2 713,715c750,752 < name 'Dock Mic Playback Volume' < value.0 0 < value.1 0 --- > name 'Beep Playback Volume' > value.0 23 > value.1 23 723,724c760,761 < dbvalue.0 -3450 < dbvalue.1 -3450 --- > dbvalue.0 0 > dbvalue.1 0 729,731c766,768 < name 'Dock Mic Playback Switch' < value.0 false < value.1 false --- > name 'Beep Playback Switch' > value.0 true > value.1 true 740c777 < name 'Mic Playback Volume' --- > name 'Dock Mic Playback Volume' 756c793 < name 'Mic Playback Switch' --- > name 'Dock Mic Playback Switch' 767,768c804,806 < name 'Auto-Mute Mode' < value Enabled --- > name 'Mic Boost Volume' > value.0 0 > value.1 0 771,774c809,815 < type ENUMERATED < count 1 < item.0 Disabled < item.1 Enabled --- > type INTEGER > count 2 > range '0 - 3' > dbmin 0 > dbmax 3000 > dbvalue.0 0 > dbvalue.1 0 778a820,851 > name 'Internal Mic Boost Volume' > value.0 0 > value.1 0 > comment { > access 'read write' > type INTEGER > count 2 > range '0 - 3' > dbmin 0 > dbmax 3000 > dbvalue.0 0 > dbvalue.1 0 > } > } > control.14 { > iface MIXER > name 'Dock Mic Boost Volume' > value.0 0 > value.1 0 > comment { > access 'read write' > type INTEGER > count 2 > range '0 - 3' > dbmin 0 > dbmax 3000 > dbvalue.0 0 > dbvalue.1 0 > } > } > control.15 { > iface MIXER 793c866 < control.14 { --- > control.16 { 804c877 < control.15 { --- > control.17 { 806,808c879,882 < name 'Internal Mic Boost Volume' < value.0 0 < value.1 0 --- > name 'Capture Volume' > index 1 > value.0 39 > value.1 39 813,815c887,889 < range '0 - 3' < dbmin 0 < dbmax 3000 --- > range '0 - 54' > dbmin -5850 > dbmax 2250 820c894 < control.16 { --- > control.18 { 822,824c896,899 < name 'Dock Mic Boost Volume' < value.0 0 < value.1 0 --- > name 'Capture Switch' > index 1 > value.0 false > value.1 false 827c902 < type INTEGER --- > type BOOLEAN 829,833d903 < range '0 - 3' < dbmin 0 < dbmax 3000 < dbvalue.0 0 < dbvalue.1 0 836c906 < control.17 { --- > control.19 { 838,840c908,909 < name 'Mic Boost Volume' < value.0 0 < value.1 0 --- > name 'Input Source' > value 'Internal Mic' 843,849c912,917 < type INTEGER < count 2 < range '0 - 3' < dbmin 0 < dbmax 3000 < dbvalue.0 0 < dbvalue.1 0 --- > type ENUMERATED > count 1 > item.0 Mic > item.1 'Internal Mic' > item.2 Mix > item.3 'Dock Mic' 852c920 < control.18 { --- > control.20 { 854,855c922,924 < name 'Master Playback Volume' < value 39 --- > name 'Input Source' > index 1 > value Mic 858c927 < type INTEGER --- > type ENUMERATED 859a929,943 > item.0 Mic > item.1 'Internal Mic' > item.2 Mix > item.3 'Dock Mic' > } > } > control.21 { > iface MIXER > name 'IEC958 Playback Volume' > value.0 39 > value.1 39 > comment { > access 'read write' > type INTEGER > count 2 863a948 > dbvalue.1 0 866c951 < control.19 { --- > control.22 { 868,869c953,954 < name 'Master Playback Switch' < value true --- > name 'IEC958 Playback Source' > value PCM 872c957 < type BOOLEAN --- > type ENUMERATED 873a959,960 > item.0 PCM > item.1 ADC 876,879c963,966 < control.20 { < iface CARD < name 'Internal Mic Phantom Jack' < value true --- > control.23 { > iface MIXER > name 'IEC958 Playback Con Mask' > value '0fff000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' 882c969 < type BOOLEAN --- > type IEC958 886,889c973,976 < control.21 { < iface CARD < name 'Dock Mic Jack' < value false --- > control.24 { > iface MIXER > name 'IEC958 Playback Pro Mask' > value '0f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' 892c979 < type BOOLEAN --- > type IEC958 896,899c983,986 < control.22 { < iface CARD < name 'Mic Jack' < value false --- > control.25 { > iface MIXER > name 'IEC958 Playback Default' > value '0400000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' 901,902c988,989 < access read < type BOOLEAN --- > access 'read write' > type IEC958 906,909c993,996 < control.23 { < iface CARD < name 'Headphone Jack' < value true --- > control.26 { > iface MIXER > name 'IEC958 Playback Switch' > value false 911c998 < access read --- > access 'read write' 916,918c1003,1005 < control.24 { < iface CARD < name 'Speaker Phantom Jack' --- > control.27 { > iface MIXER > name 'IEC958 Default PCM Playback Switch' 921c1008 < access read --- > access 'read write' 926c1013 < control.25 { --- > control.28 { 928c1015 < name 'Beep Playback Volume' --- > name 'Digital Beep Playback Volume' 940c1027 < control.26 { --- > control.29 { 942c1029,1053 < name 'Beep Playback Switch' --- > name 'Digital Beep Playback Switch' > value false > comment { > access 'read write' > type BOOLEAN > count 1 > } > } > control.30 { > iface MIXER > name 'Master Playback Volume' > value 39 > comment { > access 'read write' > type INTEGER > count 1 > range '0 - 39' > dbmin -5850 > dbmax 0 > dbvalue.0 0 > } > } > control.31 { > iface MIXER > name 'Master Playback Switch' 950c1061 < control.27 { --- > control.32 { 952a1064,1075 > value.0 3 > value.1 4 > comment { > access read > type INTEGER > count 2 > range '0 - 36' > } > } > control.33 { > iface PCM > name 'Capture Channel Map' 962c1085 < control.28 { --- > control.34 { 964a1088 > index 1 974c1098 < control.29 { --- > control.35 { 976c1100 < device 2 --- > device 1 987c1111 < control.30 { --- > control.36 { 1004,1020d1127 < control.31 { < iface MIXER < name 'PCM Playback Volume' < value.0 255 < value.1 255 < comment { < access 'read write user' < type INTEGER < count 2 < range '0 - 255' < tlv '0000000100000008ffffec1400000014' < dbmin -5100 < dbmax 0 < dbvalue.0 0 < dbvalue.1 0 < } < } 1041,1054d1147 < btrfs < raid6_pq < zlib_deflate < xor < ufs < qnx4 < hfsplus < hfs < minix < ntfs < msdos < jfs < xfs < libcrc32c 1078d1170 < bnep 1079a1172 > bnep 1082c1175 < snd_hda_codec_analog --- > arc4 1083a1177,1182 > snd_hda_codec_analog > iwl4965 > scsi_dh > iwlegacy > mac80211 > coretemp 1085c1184 < arc4 --- > cfg80211 1087c1186,1187 < scsi_dh --- > kvm_intel > kvm 1089,1090d1188 < snd_pcm < iwl4965 1092,1094c1190 < iwlegacy < mac80211 < nvram --- > snd_pcm 1095a1192,1193 > nvram > pcmcia 1098,1099d1195 < cfg80211 < coretemp 1101c1197 < kvm_intel --- > serio_raw 1102a1199 > yenta_socket 1104,1105c1201 < pcmcia < kvm --- > pcmcia_rsrc 1107c1203,1204 < serio_raw --- > lpc_ich > pcmcia_core 1109d1205 < parport_pc 1110a1207,1208 > parport_pc > mac_hid 1112,1115d1209 < yenta_socket < pcmcia_rsrc < pcmcia_core < lpc_ich 1117d1210 < mac_hid 1123,1127d1215 < psmouse < sdhci_pci < sdhci < ahci < libahci 1128a1217,1219 > e1000e > ahci > psmouse 1130a1222 > libahci 1132,1133c1224,1225 < e1000e < video --- > sdhci_pci > sdhci 1135a1228 > video 1164,1173c1257,1260 < [ 35.045988] ieee80211 phy0: Selected rate control algorithm 'iwl-4965-rs' < [ 35.076727] hda_intel: probe_mask set to 0x1 for device 17aa:20ac < [ 35.076782] snd_hda_intel 0000:00:1b.0: irq 46 for MSI/MSI-X < [ 35.084674] device-mapper: multipath: version 1.5.1 loaded < -- < [ 35.496579] Mic=0x14 < [ 35.504968] input: HDA Intel Headphone as /devices/pci0000:00/0000:00:1b.0/sound/card0/input7 < [ 35.505383] input: HDA Intel Mic as /devices/pci0000:00/0000:00:1b.0/sound/card0/input8 < [ 35.506228] input: HDA Intel Dock Mic as /devices/pci0000:00/0000:00:1b.0/sound/card0/input9 < [ 35.569962] pcmcia_socket pcmcia_socket0: cs: memory probe 0x0c0000-0x0fffff: --- > [ 38.133419] cfg80211: Calling CRDA to update world regulatory domain > [ 38.179214] hda_intel: probe_mask set to 0x1 for device 17aa:20ac > [ 38.179301] snd_hda_intel 0000:00:1b.0: irq 45 for MSI/MSI-X > [ 38.531005] EXT4-fs (sda3): mounted filesystem with ordered data mode. Opts: (null) ---- ALSA info for aa95d61b43e0fcb0b2ce68e5efa37174fd9e5cd3 END ---- ---- ALSA info for latest daily upstream BEGIN ---- 6c6 < !!Script ran on: Sun Jun 1 02:56:49 UTC 2014 --- > !!Script ran on: Sun Jun 1 06:36:37 UTC 2014 26c26 < Kernel release: 3.10-rc7-359-g5ccc618-generic --- > Kernel release: 3.15.0-999-generic 36c36 < Driver version: k3.10-rc7-359-g5ccc618-generic --- > Driver version: k3.15.0-999-generic 61c61 < 4 [ThinkPadEC ]: ThinkPad EC - ThinkPad Console Audio Control --- > 29 [ThinkPadEC ]: ThinkPad EC - ThinkPad Console Audio Control 109c109 < power_save : 1 --- > power_save : 0 127d126 < hotkey_report_mode : 0 156c155 < IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 --- > IO[0]: enable=0, dir=0, wake=0, sticky=0, data=1, unsol=0 185c184 < Amp-Out vals: [0x00 0x00] --- > Amp-Out vals: [0x27 0x27] 209c208 < Amp-In vals: [0x00 0x00] [0x80 0x80] --- > Amp-In vals: [0x00 0x00] [0x00 0x00] 229c228 < Amp-In vals: [0x00 0x00] [0x80 0x80] --- > Amp-In vals: [0x00 0x00] [0x00 0x00] 245c244 < 0x14 0x15* 0x16 0x20 0x25 --- > 0x14* 0x15 0x16 0x20 0x25 258,261d256 < Control: name="Beep Playback Volume", index=0, device=0 < ControlAmp: chs=1, dir=Out, idx=0, ofs=0 < Control: name="Beep Playback Switch", index=0, device=0 < ControlAmp: chs=1, dir=Out, idx=0, ofs=0 263c258 < Amp-Out vals: [0x0f] --- > Amp-Out vals: [0x8f] 283c278 < Amp-Out vals: [0x80 0x80] --- > Amp-Out vals: [0x00 0x00] 285c280 < EAPD 0x0: --- > EAPD 0x2: EAPD 422a418,421 > Control: name="Beep Playback Volume", index=0, device=0 > ControlAmp: chs=3, dir=In, idx=3, ofs=0 > Control: name="Beep Playback Switch", index=0, device=0 > ControlAmp: chs=3, dir=In, idx=3, ofs=0 424c423 < Amp-In vals: [0x80 0x80] [0x9f 0x9f] [0x80 0x80] [0x80 0x80] [0x80 0x80] --- > Amp-In vals: [0x80 0x80] [0x9f 0x9f] [0x80 0x80] [0x17 0x17] [0x80 0x80] 429c428 < Amp-Out vals: [0x9f 0x9f] --- > Amp-Out vals: [0x1f 0x1f] 459,466c458,465 < crw-rw----+ 1 root audio 116, 7 May 31 18:02 /dev/snd/controlC0 < crw-rw----+ 1 root audio 116, 2 May 31 18:02 /dev/snd/controlC4 < crw-rw----+ 1 root audio 116, 6 May 31 18:02 /dev/snd/hwC0D0 < crw-rw----+ 1 root audio 116, 5 May 31 18:07 /dev/snd/pcmC0D0c < crw-rw----+ 1 root audio 116, 4 May 31 19:13 /dev/snd/pcmC0D0p < crw-rw----+ 1 root audio 116, 3 May 31 18:02 /dev/snd/pcmC0D2p < crw-rw----+ 1 root audio 116, 1 May 31 18:02 /dev/snd/seq < crw-rw----+ 1 root audio 116, 33 May 31 18:02 /dev/snd/timer --- > crw-rw----+ 1 root audio 116, 3 May 31 23:24 /dev/snd/controlC0 > crw-rw----+ 1 root audio 116, 2 May 31 23:24 /dev/snd/controlC29 > crw-rw----+ 1 root audio 116, 7 May 31 23:24 /dev/snd/hwC0D0 > crw-rw----+ 1 root audio 116, 5 May 31 23:32 /dev/snd/pcmC0D0c > crw-rw----+ 1 root audio 116, 4 May 31 23:31 /dev/snd/pcmC0D0p > crw-rw----+ 1 root audio 116, 6 May 31 23:24 /dev/snd/pcmC0D2p > crw-rw----+ 1 root audio 116, 1 May 31 23:24 /dev/snd/seq > crw-rw----+ 1 root audio 116, 33 May 31 23:24 /dev/snd/timer 470,473c469,472 < drwxr-xr-x 2 root root 80 May 31 18:02 . < drwxr-xr-x 3 root root 220 May 31 18:02 .. < lrwxrwxrwx 1 root root 12 May 31 18:02 pci-0000:00:1b.0 -> ../controlC0 < lrwxrwxrwx 1 root root 12 May 31 18:02 platform-thinkpad_acpi -> ../controlC4 --- > drwxr-xr-x 2 root root 80 May 31 23:24 . > drwxr-xr-x 3 root root 220 May 31 23:24 .. > lrwxrwxrwx 1 root root 12 May 31 23:24 pci-0000:00:1b.0 -> ../controlC0 > lrwxrwxrwx 1 root root 13 May 31 23:24 platform-thinkpad_acpi -> ../controlC29 523,524c522,523 < Front Left: Playback 0 [0%] [-58.50dB] [off] < Front Right: Playback 0 [0%] [-58.50dB] [off] --- > Front Left: Playback 39 [100%] [0.00dB] [on] > Front Right: Playback 39 [100%] [0.00dB] [on] 547,550c546,551 < Capabilities: pvolume pvolume-joined pswitch pswitch-joined < Playback channels: Mono < Limits: Playback 0 - 15 < Mono: Playback 15 [100%] [0.00dB] [on] --- > Capabilities: pvolume pswitch > Playback channels: Front Left - Front Right > Limits: Playback 0 - 31 > Mono: > Front Left: Playback 23 [74%] [0.00dB] [on] > Front Right: Playback 23 [74%] [0.00dB] [on] 600c601 < !!-------Mixer controls for card 4 [ThinkPadEC] --- > !!-------Mixer controls for card 29 [ThinkPadEC] 602c603 < Card hw:4 'ThinkPadEC'/'ThinkPad Console Audio Control at EC reg 0x30, fw 7MHT25WW-1.03' --- > Card hw:29 'ThinkPadEC'/'ThinkPad Console Audio Control at EC reg 0x30, fw 7MHT25WW-1.03' 648,649c649,650 < value.0 0 < value.1 0 --- > value.0 39 > value.1 39 657,658c658,659 < dbvalue.0 -5850 < dbvalue.1 -5850 --- > dbvalue.0 0 > dbvalue.1 0 664,665c665,666 < value.0 false < value.1 false --- > value.0 true > value.1 true 899c900 < value false --- > value true 909c910 < value true --- > value false 929c930,931 < value 15 --- > value.0 23 > value.1 23 933,936c935,938 < count 1 < range '0 - 15' < dbmin -4500 < dbmax 0 --- > count 2 > range '0 - 31' > dbmin -3450 > dbmax 1200 937a940 > dbvalue.1 0 943c946,947 < value true --- > value.0 true > value.1 true 947c951 < count 1 --- > count 2 1041,1054c1045,1046 < btrfs < raid6_pq < zlib_deflate < xor < ufs < qnx4 < hfsplus < hfs < minix < ntfs < msdos < jfs < xfs < libcrc32c --- > ctr > ccm 1078d1069 < bnep 1079a1071 > bnep 1080a1073 > 6lowpan_iphc 1083c1076 < dm_multipath --- > snd_hda_codec_generic 1085c1078 < arc4 --- > snd_hda_controller 1087d1079 < scsi_dh 1089,1090d1080 < snd_pcm < iwl4965 1092,1093c1082,1084 < iwlegacy < mac80211 --- > dm_multipath > coretemp > snd_pcm 1095c1086,1087 < snd_page_alloc --- > kvm_intel > scsi_dh 1096a1089,1090 > arc4 > kvm 1098,1099c1092,1094 < cfg80211 < coretemp --- > iwl4965 > iwlegacy > mac80211 1101d1095 < kvm_intel 1102a1097 > cfg80211 1105d1099 < kvm 1107c1101 < serio_raw --- > microcode 1109,1111c1103 < parport_pc < soundcore < ppdev --- > serio_raw 1112a1105,1106 > soundcore > mac_hid 1114c1108 < pcmcia_core --- > ppdev 1115a1110,1111 > pcmcia_core > parport_pc 1117d1112 < mac_hid 1123,1125d1117 < psmouse < sdhci_pci < sdhci 1126a1119,1120 > sdhci_pci > psmouse 1127a1122 > sdhci 1128a1124 > video 1129a1126 > e1000e 1132,1133d1128 < e1000e < video 1164,1167c1159,1162 < [ 35.045988] ieee80211 phy0: Selected rate control algorithm 'iwl-4965-rs' < [ 35.076727] hda_intel: probe_mask set to 0x1 for device 17aa:20ac < [ 35.076782] snd_hda_intel 0000:00:1b.0: irq 46 for MSI/MSI-X < [ 35.084674] device-mapper: multipath: version 1.5.1 loaded --- > [ 43.432228] input: ThinkPad Extra Buttons as /devices/platform/thinkpad_acpi/input/input7 > [ 43.624371] snd_hda_intel 0000:00:1b.0: probe_mask set to 0x1 for device 17aa:20ac > [ 43.624479] snd_hda_intel 0000:00:1b.0: irq 46 for MSI/MSI-X > [ 43.800655] systemd-udevd[578]: renamed network interface wlan0 to wlan2 1169,1173c1164,1176 < [ 35.496579] Mic=0x14 < [ 35.504968] input: HDA Intel Headphone as /devices/pci0000:00/0000:00:1b.0/sound/card0/input7 < [ 35.505383] input: HDA Intel Mic as /devices/pci0000:00/0000:00:1b.0/sound/card0/input8 < [ 35.506228] input: HDA Intel Dock Mic as /devices/pci0000:00/0000:00:1b.0/sound/card0/input9 < [ 35.569962] pcmcia_socket pcmcia_socket0: cs: memory probe 0x0c0000-0x0fffff: --- > [ 44.022500] cfg80211: (5735000 KHz - 5835000 KHz @ 40000 KHz), (300 mBi, 2000 mBm), (N/A) > [ 44.101116] sound hdaudioC0D0: autoconfig: line_outs=1 (0x12/0x0/0x0/0x0/0x0) type:speaker > [ 44.101123] sound hdaudioC0D0: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) > [ 44.101127] sound hdaudioC0D0: hp_outs=1 (0x11/0x0/0x0/0x0/0x0) > [ 44.101130] sound hdaudioC0D0: mono: mono_out=0x0 > [ 44.101133] sound hdaudioC0D0: inputs: > [ 44.101137] sound hdaudioC0D0: Internal Mic=0x15 > [ 44.101140] sound hdaudioC0D0: Dock Mic=0x1c > [ 44.101144] sound hdaudioC0D0: Mic=0x14 > [ 44.112515] input: HDA Intel Dock Mic as /devices/pci0000:00/0000:00:1b.0/sound/card0/input8 > [ 44.112657] input: HDA Intel Mic as /devices/pci0000:00/0000:00:1b.0/sound/card0/input9 > [ 44.112794] input: HDA Intel Headphone as /devices/pci0000:00/0000:00:1b.0/sound/card0/input10 > [ 45.815258] init: failsafe main process (883) killed by TERM signal ---- ALSA info for latest daily upstream END ----

3 10

Re: [alsa-devel] Set eapd to follow stream on or off
by Kailang 04 Jun '14

04 Jun '14

Hi Takashi, I abandon this update. Playing a slient stream will be had pop noise. It is hardware issue. It is not to avoid. But it need to read and write our hidden register to control eapd. Resume back need to fill eapd to keep low state. I think I just send the patch to ubuntu team. Thanks. > > >> At Thu, 22 May 2014 14:26:04 +0800, >> Kailang wrote: >>> >>> >>> >>> > At Wed, 21 May 2014 17:24:16 +0800, >>> > Kailang wrote: >>> >> >>> >> > At Wed, 21 May 2014 14:28:12 +0800, >>> >> > Kailang wrote: >>> >> >> >>> >> >> Hi Takashi, >>> >> >> >>> >> >> Attach is the update patch. >>> >> > >>> >> > Keep the function in lower letters. >>> >> > >>> >> > Also, update the patch description as well (and better to put more >>> >> > words). >>> >> >>> >> Update as attach. >>> >> Thanks. >>> > >>> > Well, do you really want to turn off EAPD at closing the stream? >>> >>> Yes. >>> >>> > As I repeatedly explained, this results in more frequent EAPD on/off, >>> > i.e. more pop noises. >>> > >>> >>> I have sent the patch for HP. >>> I will wait for feedback from customer. >> >> Did you send the test script, too? > No. > >> >>> If it is solve the issue, I will advise you. >>> >>> Our AE says, windows use this method. HP is agree this usage. >> >> Windows does this likely because of possible power saving, not because >> of pop noises. >> >> I'm fine to take the patch, but only when I'm convinced why it really >> fixes the pop noise problem in the scenario below: >> - playing a silent stream >> - frequent open / close a stream >> > > You are right. It is the key points. But I think first stream will show > pop noise. It was hard to remove. > Keep EAPD to high after first stream start maybe a good idea. > Thanks. > I think to request the machine is better. > I will let our AE to request the machine. > >> >> Takashi >> >> >>> >>> > >>> > Takashi >>> > >>> >> >>> >> > >>> >> > >>> >> > thanks, >>> >> > >>> >> > Takashi >>> >> > >>> >> >> Many thanks. >>> >> >> >>> >> >> BR, >>> >> >> Kailang >>> >> >> >>> >> >> >>> >> >> > At Wed, 21 May 2014 13:58:14 +0800, >>> >> >> > Kailang wrote: >>> >> >> >> >>> >> >> >> >>> >> >> >> > At Tue, 20 May 2014 18:05:42 +0800, >>> >> >> >> > Kailang wrote: >>> >> >> >> >> >>> >> >> >> >> > At Tue, 20 May 2014 17:20:44 +0800, >>> >> >> >> >> > Kailang wrote: >>> >> >> >> >> >> >>> >> >> >> >> >> >>> >> >> >> >> >> > At Tue, 20 May 2014 16:52:43 +0800, >>> >> >> >> >> >> > Kailang wrote: >>> >> >> >> >> >> >> >>> >> >> >> >> >> >> >>> >> >> >> >> >> >> > At Tue, 20 May 2014 15:15:49 +0800, >>> >> >> >> >> >> >> > Kailang wrote: >>> >> >> >> >> >> >> >> >>> >> >> >> >> >> >> >> Hi Takashi, >>> >> >> >> >> >> >> >> >>> >> >> >> >> >> >> >> HP machine had pop noise via speaker. >>> >> >> >> >> >> >> >> If EAPD on or off, speaker will have pop noise. >>> >> >> >> >> >> >> >> So, they want to set eapd to follow with stream on >>> >> >> >> >> >> >> >> or >>> >> >> >> >> >> >> >> off. >>> >> >> >> >> >> >> > >>> >> >> >> >> >> >> > I can apply this as is, but still I wonder whether >>> >> >> >> >> >> >> > this >>> >> >> >> >> >> >> > results >>> >> >> >> >> >> >> > in >>> >> >> >> >> >> >> > other problems. Doing EAPD on/off there means that >>> >> >> >> >> >> >> > it >>> >> >> >> >> >> >> > happens >>> >> >> >> >> >> >> > far >>> >> >> >> >> >> >> > more often. For example, what happens if you do >>> >> >> >> >> >> >> > like >>> >> >> >> >> >> >> > below? >>> >> >> >> >> >> >> > >>> >> >> >> >> >> >> > while true; do >>> >> >> >> >> >> >> > echo -n > /dev/snd/pcmC0D0p >>> >> >> >> >> >> >> > done >>> >> >> >> >> >> >> > >>> >> >> >> >> >> >> > And, what if the device is resumed from power-saving >>> >> >> >> >> >> >> > by >>> >> >> >> >> >> >> > an >>> >> >> >> >> >> >> > audio >>> >> >> >> >> >> >> > playback? Immediately after the power up, it'll >>> >> >> >> >> >> >> > start >>> >> >> >> >> >> >> > the >>> >> >> >> >> >> >> > streaming, >>> >> >> >> >> >> >> > so it'll turn on EAPD. >>> >> >> >> >> >> >> > >>> >> >> >> >> >> >> > That said, isn't it just a need for some delay >>> >> >> >> >> >> >> > before >>> >> >> >> >> >> >> > EAPD >>> >> >> >> >> >> >> > on/off? >>> >> >> >> >> >> >> >>> >> >> >> >> >> >> No, eapd on or off will cause pop noise. >>> >> >> >> >> >> > >>> >> >> >> >> >> > Does it do always? >>> >> >> >> >> >> No, pop noise just one pop sound. It can not remove by >>> >> >> >> >> >> software. >>> >> >> >> >> >> But this machine need to do this always. >>> >> >> >> >> > >>> >> >> >> >> > So, doing EAPD on/off often is rather worse. It's what >>> >> >> >> >> > your >>> >> >> >> >> > patch >>> >> >> >> >> > does. >>> >> >> >> >> > >>> >> >> >> >> > Once again: did you try the script above? If the pop >>> >> >> >> >> > noise >>> >> >> >> >> > always >>> >> >> >> >> >>> >> >> >> >> I had try it on my develop machine. I have no HP issue >>> >> >> >> >> machine >>> >> >> >> >> on >>> >> >> >> >> my >>> >> >> >> >> site. >>> >> >> >> >> >>> >> >> >> >> > happens, the operation like above is the worst scenario. >>> >> >> >> >> > >>> >> >> >> >> > Then, let's think again how to *reduce* it. We need one >>> >> >> >> >> > EAPD >>> >> >> >> >> > on >>> >> >> >> >> > at >>> >> >> >> >> > least after each power up. But it's needed only when you >>> >> >> >> >> > actually >>> >> >> >> >> > plays a stream. That is, we need only enable EAPD at >>> >> >> >> >> > playback >>> >> >> >> >> > stream >>> >> >> >> >> > open, but leave EAPD as is at close. >>> >> >> >> >> > >>> >> >> >> >> >>> >> >> >> >> [we need only enable EAPD at playback stream open, but leave >>> >> >> >> >> EAPD >>> >> >> >> >> as >>> >> >> >> >> is >>> >> >> >> >> at >>> >> >> >> >> close.] >>> >> >> >> >> Yes. >>> >> >> >> >> So, I need to create eapd low in alc269_shutup. Right? >>> >> >> >> > >>> >> >> >> > Better to create own shutup callback (you can call >>> >> >> >> > alc269_shutup() >>> >> >> >> > from there, too) and set it to spec->shutup in the fixup. >>> >> >> >> > The >>> >> >> >> > pincfg >>> >> >> >> > cleanup isn't often needed or may have bad effect. You >>> >> >> >> > should >>> >> >> >> > test >>> >> >> >> > with real machines what are needed and what are not. >>> >> >> >> >>> >> >> >> static void alc290_fixup_pop_noise(struct hda_codec *codec, >>> >> >> >> const struct hda_fixup *fix, int action) >>> >> >> >> { >>> >> >> >> struct alc_spec *spec = codec->spec; >>> >> >> >> >>> >> >> >> if (action == HDA_FIXUP_ACT_PRE_PROBE) { >>> >> >> >> spec->gen.pcm_playback_hook = ALC290_playback_pcm_hook; >>> >> >> >> } >>> >> >> >> if (action == HDA_FIXUP_ACT_INIT) { >>> >> >> >> set_eapd(codec, 0x14, false); >>> >> >> >> } >>> >> >> >> } >>> >> >> >> >>> >> >> >> Could I modify as upper code? >>> >> >> >> To set eapd low in HDA_FIXUP_ACT_INIT state. >>> >> >> >> If system resume, it will initial eapd to low. >>> >> >> >> To write eapd to same value, it will have no pop noise. >>> >> >> >> It will keep always low for eapd, only stream open and close >>> >> >> >> have >>> >> >> >> change >>> >> >> >> state for eapd. >>> >> >> > >>> >> >> > Yes, it looks OK. >>> >> >> > >>> >> >> > >>> >> >> > Takashi >>> >> >> > >>> >> >> > >>> >> >> >> >>> >> >> >> > >>> >> >> >> > >>> >> >> >> > Takashi >>> >> >> >> > >>> >> >> >> >> >>> >> >> >> >> > >>> >> >> >> >> > Takashi >>> >> >> >> >> > >>> >> >> >> >> >> >> Our customer just want to do eapd on when stream is >>> >> >> >> >> >> >> open. >>> >> >> >> >> >> >> The pop noise will not clear. Because music gain was >>> >> >> >> >> >> >> biger >>> >> >> >> >> >> >> than >>> >> >> >> >> >> >> pop >>> >> >> >> >> >> >> noise. >>> >> >> >> >> >> > >>> >> >> >> >> >> > Opening a stream doesn't mean that it follows the loud >>> >> >> >> >> >> > sound >>> >> >> >> >> >> > immediately. The script above wouldn't produce any >>> >> >> >> >> >> > sound >>> >> >> >> >> >> > but >>> >> >> >> >> >> > just >>> >> >> >> >> >> > does open/close the stream. Did you try that? >>> >> >> >> >> >> > >>> >> >> >> >> >> > >>> >> >> >> >> >> > Takashi >>> >> >> >> >> >> > >>> >> >> >> >> >> > ------Please consider the environment before printing >>> >> >> >> >> >> > this >>> >> >> >> >> >> > e-mail. >>> >> >> >> >> >> >>> >> >> >> >> >>> >> >> >> >>> >> >> [2 0001-set-eapd-follow-stream-on-off.patch >>> >> >> <application/octet-stream >>> >> >> (quoted-printable)>] >>> >> >> >>> >> > >>> >> > ------Please consider the environment before printing this e-mail. >>> >> [2 0002-set-eapd-follow-stream-on-off.patch <application/octet-stream >>> >> (quoted-printable)>] >>> >> >>> >

1 0

[alsa-devel] [PATCH 1/1] Alsa: seq: correctly detect input buffer overflow
by Adam Goode 04 Jun '14

04 Jun '14

snd_seq_event_dup returns -ENOMEM in some buffer-full conditions, but usually returns -EAGAIN. Make -EAGAIN trigger the overflow condition in snd_seq_fifo_event_in so that the fifo is cleared and -ENOSPC is returned to userspace as stated in the alsa-lib docs. Signed-off-by: Adam Goode <agoode(a)google.com> diff --git a/sound/core/seq/seq_fifo.c b/sound/core/seq/seq_fifo.c index 5599899..53a403e 100644 --- a/sound/core/seq/seq_fifo.c +++ b/sound/core/seq/seq_fifo.c @@ -124,7 +124,7 @@ int snd_seq_fifo_event_in(struct snd_seq_fifo *f, snd_use_lock_use(&f->use_lock); err = snd_seq_event_dup(f->pool, event, &cell, 1, NULL); /* always non-blocking */ if (err < 0) { - if (err == -ENOMEM) + if ((err == -ENOMEM) || (err == -EAGAIN)) atomic_inc(&f->overflow); snd_use_lock_free(&f->use_lock); return err; -- 1.9.1.423.g4596e3a

2 1

[alsa-devel] [PATCH - seq 1/1] ALSA: seq: Continue broadcasting events to ports if one of them fails
by Adam Goode 04 Jun '14

04 Jun '14

Sometimes PORT_EXIT messages are lost when a process is exiting. This happens if you subscribe to the announce port with client A, then subscribe to the announce port with client B, then kill client A. Client B will not see the PORT_EXIT message because client A's port is closing and is earlier in the announce port subscription list. The for each loop will try to send the announcement to client A and fail, then will stop trying to broadcast to other ports. Killing B works fine since the announcement will already have gone to A. The CLIENT_EXIT message does not get lost. How to reproduce problem: *** termA $ aseqdump -p 0:1 0:1 Port subscribed 0:1 -> 128:0 *** termB $ aseqdump -p 0:1 *** termA 0:1 Client start client 129 0:1 Port start 129:0 0:1 Port subscribed 0:1 -> 129:0 *** termB 0:1 Port subscribed 0:1 -> 129:0 *** termA ^C *** termB 0:1 Client exit client 128 <--- expected Port exit as well (before client exit) Signed-off-by: Adam Goode <agoode(a)google.com> diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c index 9ca5e64..f9b5315 100644 --- a/sound/core/seq/seq_clientmgr.c +++ b/sound/core/seq/seq_clientmgr.c @@ -660,7 +660,7 @@ static int deliver_to_subscribers(struct snd_seq_client *client, int atomic, int hop) { struct snd_seq_subscribers *subs; - int err = 0, num_ev = 0; + int err = 0, sticky_err = 0, num_ev = 0; struct snd_seq_event event_saved; struct snd_seq_client_port *src_port; struct snd_seq_port_subs_info *grp; @@ -685,8 +685,12 @@ static int deliver_to_subscribers(struct snd_seq_client *client, subs->info.flags & SNDRV_SEQ_PORT_SUBS_TIME_REAL); err = snd_seq_deliver_single_event(client, event, 0, atomic, hop); - if (err < 0) - break; + if (err < 0) { + /* save first error that occurs and continue */ + if (!sticky_err) + sticky_err = err; + continue; + } num_ev++; /* restore original event record */ *event = event_saved; @@ -697,7 +701,7 @@ static int deliver_to_subscribers(struct snd_seq_client *client, up_read(&grp->list_mutex); *event = event_saved; /* restore */ snd_seq_port_unlock(src_port); - return (err < 0) ? err : num_ev; + return (sticky_err < 0) ? sticky_err : num_ev; } @@ -709,7 +713,7 @@ static int port_broadcast_event(struct snd_seq_client *client, struct snd_seq_event *event, int atomic, int hop) { - int num_ev = 0, err = 0; + int num_ev = 0, err = 0, sticky_err = 0; struct snd_seq_client *dest_client; struct snd_seq_client_port *port; @@ -724,14 +728,18 @@ static int port_broadcast_event(struct snd_seq_client *client, err = snd_seq_deliver_single_event(NULL, event, SNDRV_SEQ_FILTER_BROADCAST, atomic, hop); - if (err < 0) - break; + if (err < 0) { + /* save first error that occurs and continue */ + if (!sticky_err) + sticky_err = err; + continue; + } num_ev++; } read_unlock(&dest_client->ports_lock); snd_seq_client_unlock(dest_client); event->dest.port = SNDRV_SEQ_ADDRESS_BROADCAST; /* restore */ - return (err < 0) ? err : num_ev; + return (sticky_err < 0) ? sticky_err : num_ev; } /* @@ -741,7 +749,7 @@ static int port_broadcast_event(struct snd_seq_client *client, static int broadcast_event(struct snd_seq_client *client, struct snd_seq_event *event, int atomic, int hop) { - int err = 0, num_ev = 0; + int err = 0, sticky_err = 0, num_ev = 0; int dest; struct snd_seq_addr addr; @@ -760,12 +768,16 @@ static int broadcast_event(struct snd_seq_client *client, err = snd_seq_deliver_single_event(NULL, event, SNDRV_SEQ_FILTER_BROADCAST, atomic, hop); - if (err < 0) - break; + if (err < 0) { + /* save first error that occurs and continue */ + if (!sticky_err) + sticky_err = err; + continue; + } num_ev += err; } event->dest = addr; /* restore */ - return (err < 0) ? err : num_ev; + return (sticky_err < 0) ? sticky_err : num_ev; } -- 1.9.1.423.g4596e3a

2 2

[alsa-devel] [GIT PULL] ASoC updates for v3.16
by Mark Brown 03 Jun '14

03 Jun '14

The following changes since commit cee429e5c53f7ffc1dd500001d9dff567833c23d: Merge remote-tracking branches 'asoc/topic/ux500', 'asoc/topic/wm8731', 'asoc/topic/wm8804', 'asoc/topic/wm8955' and 'asoc/topic/wm8985' into asoc-next (2014-05-22 00:24:04 +0100) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git asoc-v3.16-2 for you to fetch changes up to e1d4d3c854f25cff6c6c139588570e124d5e8fa4: ASoC: free jack GPIOs before the sound card is freed (2014-06-03 10:41:16 +0100) ---------------------------------------------------------------- ASoC: Final updates for v3.16 A few more updates from the last week of development, nothing too exciting. Highlights include: - GPIO descriptor support for jacks - More updates and fixes to the Freescale SSI, Intel and rsnd drivers. - New drivers for Analog Devices ADAU1361, ADAU1381, ADAU1761 and ADAU1781, and Realtek RT5677. ---------------------------------------------------------------- Alexander Shiyan (1): ASoC: fsl: Remove dependencies of boards for SND_SOC_EUKREA_TLV320 Andrew Lunn (1): ASoC: simple-card: Support setting mclk via a fixed factor Andy Shevchenko (1): ASoC: Intel: remove duplicate headers Axel Lin (2): ASoC: sirf-audio-codec: Simplify the new bitmask value in regmap_update_bits ASoC: cs42l56: Fix new value argument in snd_soc_update_bits calls Charles Keepax (1): ASoC: wm_adsp: Use adsp_err/warn instead of dev_err/warn Chen Zhen (1): ASoC: max98090: Add NI/MI values for user pclk 19.2 MHz Daniel Matuschek (1): ASoC: wm8804: Allow control of master clock divider in PLL generation Fabio Estevam (2): ASoC: fsl_ssi: Add suspend/resume support ASoC: sgtl5000: Fix the cache handling Imre Deak (1): ASoC: Intel: byt/hsw: Add missing kthread_stop to error/cleanup path Jarkko Nikula (8): ASoC: jack: Basic GPIO descriptor conversion ASoC: jack: Add support for GPIO descriptor defined jack pins ASoC: jack: Clarify GPIO descriptor lookup in struct snd_soc_jack_gpio doc ASoC: jack: Fix multiple definition of `snd_soc_jack_add_gpiods' ASoC: Intel: Add Baytrail byt-max98090 machine driver ASoC: Intel: Clear stored Baytrail DSP DMA pointer before stream start ASoC: Intel: Use devm_snd_soc_register_card ASoC: Intel: byt-rt5640: Use card PM ops from core Jyri Sarha (2): ASoC: simple-card: is_top_level_node parameter to simple_card_dai_link_of() ASoC: omap-pcm: Move omap-pcm under include/sound Kees Cook (1): ASoC: Intel: avoid format string leak to thread name Kuninori Morimoto (6): ASoC: rsnd: DT node clean up by using the of_node_put() ASoC: rsnd: save platform_device instead of device ASoC: rsnd: remove rsnd_src_non_ops ASoC: rsnd: module name is unified ASoC: rsnd: care DMA slave channel name for DT ASoC: rsnd: add rsnd_gen_dma_addr() for DMAC addr Lars-Peter Clausen (7): ASoC: davinci-evm: Replace instances of rtd->codec->card with rtd->card ASoC: Add ADAU1X61 and ADAU1X81 CODECs common code ASoC: Add ADAU1361/ADAU1761 audio CODEC support ASoC: Add ADAU1381/ADAU1781 audio CODEC support ASoC: Blackfin: ADAU1X61 eval board support ASoC: Blackfin: ADAU1X81 eval board support ASoC: Add helper functions to cast from DAPM context to CODEC/platform Mark Brown (19): Merge branch 'topic/davinci' of git://git.kernel.org/.../broonie/sound into asoc-omap Merge branch 'topic/fsl' of git://git.kernel.org/.../broonie/sound into asoc-fsl-ssi Merge branches 'topic/rt5640', 'topic/rt5645' and 'topic/rt5651' of git://git.kernel.org/.../broonie/sound into asoc-rl6231 ASoC: intel: The Baytrail/MAX98090 driver depends on I2C ASoC: cache: Fix error code when not using ASoC level cache Merge remote-tracking branches 'asoc/fix/pxa' and 'asoc/fix/tlv320aic3x' into asoc-linus Merge remote-tracking branch 'asoc/topic/core' into asoc-next Merge remote-tracking branches 'asoc/topic/adau' and 'asoc/topic/adsp' into asoc-next Merge remote-tracking branch 'asoc/topic/alc5623' into asoc-next Merge remote-tracking branch 'asoc/topic/cs42l56' into asoc-next Merge remote-tracking branch 'asoc/topic/davinci' into asoc-next Merge remote-tracking branch 'asoc/topic/fsl-ssi' into asoc-next Merge remote-tracking branches 'asoc/topic/gpio' and 'asoc/topic/intel' into asoc-next Merge remote-tracking branches 'asoc/topic/max98090' and 'asoc/topic/max98095' into asoc-next Merge remote-tracking branches 'asoc/topic/omap' and 'asoc/topic/rcar' into asoc-next Merge remote-tracking branches 'asoc/topic/rl6231' and 'asoc/topic/rt5677' into asoc-next Merge remote-tracking branches 'asoc/topic/samsung', 'asoc/topic/sgtl5000', 'asoc/topic/simple' and 'asoc/topic/sirf' into asoc-next Merge remote-tracking branch 'asoc/topic/tegra' into asoc-next Merge remote-tracking branches 'asoc/topic/wm8804' and 'asoc/topic/wm9713' into asoc-next Markus Pargmann (4): ASoC: fsl-ssi: make fsl,mode property optional ASoC: fsl-ssi: Fix baudclock handling ASoC: fsl-ssi: reorder and document fsl_ssi_private ASoC: fsl-ssi: Use regmap Matt Reimer (1): ASoC: wm9713: correct mono out PGA sources Oder Chiou (4): ASoC: rt5640: Add RL6231 class device shared support for RT5640, RT5645 and RT5651 ASoC: rt5640: Add the function of the PLL clock calculation to RL6231 shared support ASoC: rt5640: Add the function "get_clk_info" to RL6231 shared support ASoC: add RT5677 CODEC driver Peter Ujfalusi (1): ASoC: tlv320aci3x: Fix custom snd_soc_dapm_put_volsw_aic3x() function Sascha Hauer (4): ASoC: fsl-ssi: introduce SoC specific data ASoC: fsl-ssi: set bitclock in master mode from hw_params ASoC: fsl-ssi: remove unnecessary spinlock ASoC: fsl-ssi: Set framerate divider correctly for i2s master mode Stephen Boyd (1): ASoC: pxa: pxa-ssp: Terminate of match table Stephen Warren (2): ASoC: tegra: free jack GPIOs before the sound card is freed ASoC: free jack GPIOs before the sound card is freed Sylwester Nawrocki (1): ASoC: samsung: Handle errors when getting the op_clk clock Takashi Iwai (1): ASoC: alc5623: Fix Kconfig dependency Tushar Behera (3): ASoC: max98090: Add master clock handling ASoC: max98095: Add master clock handling ASoC: samsung: Use params_width() Xiubo Li (1): ASoC: cache: Fix possible ZERO_SIZE_PTR pointer dereferencing error. .../devicetree/bindings/sound/max98090.txt | 6 + .../devicetree/bindings/sound/max98095.txt | 6 + .../devicetree/bindings/sound/renesas,rsnd.txt | 1 + .../devicetree/bindings/sound/simple-card.txt | 3 + include/linux/platform_data/adau17x1.h | 109 + {sound/soc/omap => include/sound}/omap-pcm.h | 0 include/sound/rt5677.h | 21 + include/sound/soc.h | 49 +- sound/soc/blackfin/Kconfig | 26 + sound/soc/blackfin/Makefile | 4 + sound/soc/blackfin/bfin-eval-adau1x61.c | 142 + sound/soc/blackfin/bfin-eval-adau1x81.c | 130 + sound/soc/codecs/Kconfig | 52 +- sound/soc/codecs/Makefile | 22 +- sound/soc/codecs/adau1761-i2c.c | 60 + sound/soc/codecs/adau1761-spi.c | 77 + sound/soc/codecs/adau1761.c | 803 +++++ sound/soc/codecs/adau1761.h | 23 + sound/soc/codecs/adau1781-i2c.c | 58 + sound/soc/codecs/adau1781-spi.c | 75 + sound/soc/codecs/adau1781.c | 511 +++ sound/soc/codecs/adau1781.h | 23 + sound/soc/codecs/adau17x1.c | 866 +++++ sound/soc/codecs/adau17x1.h | 124 + sound/soc/codecs/cs42l56.c | 70 +- sound/soc/codecs/cs42l56.h | 10 +- sound/soc/codecs/max98090.c | 31 +- sound/soc/codecs/max98090.h | 1 + sound/soc/codecs/max98095.c | 24 + sound/soc/codecs/rl6231.c | 152 + sound/soc/codecs/rl6231.h | 34 + sound/soc/codecs/rt5640.c | 108 +- sound/soc/codecs/rt5640.h | 8 - sound/soc/codecs/rt5645.c | 111 +- sound/soc/codecs/rt5645.h | 7 - sound/soc/codecs/rt5651.c | 107 +- sound/soc/codecs/rt5651.h | 1 - sound/soc/codecs/rt5677.c | 3498 ++++++++++++++++++++ sound/soc/codecs/rt5677.h | 1451 ++++++++ sound/soc/codecs/sgtl5000.c | 75 +- sound/soc/codecs/sirf-audio-codec.c | 8 +- sound/soc/codecs/tlv320aic3x.c | 2 +- sound/soc/codecs/wm8804.c | 17 +- sound/soc/codecs/wm8804.h | 4 + sound/soc/codecs/wm9713.c | 3 +- sound/soc/codecs/wm_adsp.c | 37 +- sound/soc/davinci/davinci-evm.c | 9 +- sound/soc/davinci/davinci-mcasp.c | 2 +- sound/soc/fsl/Kconfig | 8 +- sound/soc/fsl/fsl_ssi.c | 576 ++-- sound/soc/fsl/fsl_ssi.h | 50 +- sound/soc/generic/simple-card.c | 39 +- sound/soc/intel/Kconfig | 9 + sound/soc/intel/Makefile | 2 + sound/soc/intel/byt-max98090.c | 203 ++ sound/soc/intel/byt-rt5640.c | 27 +- sound/soc/intel/haswell.c | 9 +- sound/soc/intel/sst-acpi.c | 1 + sound/soc/intel/sst-baytrail-ipc.c | 8 +- sound/soc/intel/sst-baytrail-pcm.c | 1 + sound/soc/intel/sst-haswell-ipc.c | 15 +- sound/soc/intel/sst-haswell-pcm.c | 1 - sound/soc/omap/ams-delta.c | 14 +- sound/soc/omap/omap-dmic.c | 2 +- sound/soc/omap/omap-hdmi.c | 2 +- sound/soc/omap/omap-mcbsp.c | 2 +- sound/soc/omap/omap-mcpdm.c | 2 +- sound/soc/omap/omap-twl4030.c | 28 +- sound/soc/omap/rx51.c | 18 +- sound/soc/pxa/hx4700.c | 9 +- sound/soc/pxa/pxa-ssp.c | 1 + sound/soc/samsung/h1940_uda1380.c | 11 +- sound/soc/samsung/i2s.c | 14 +- sound/soc/samsung/pcm.c | 4 +- sound/soc/samsung/rx1950_uda1380.c | 12 +- sound/soc/samsung/s3c-i2s-v2.c | 8 +- sound/soc/samsung/s3c2412-i2s.c | 6 +- sound/soc/samsung/s3c24xx-i2s.c | 6 +- sound/soc/samsung/smartq_wm8987.c | 11 +- sound/soc/samsung/smdk_wm8580.c | 8 +- sound/soc/samsung/smdk_wm8994.c | 2 +- sound/soc/samsung/spdif.c | 4 +- sound/soc/sh/rcar/core.c | 87 +- sound/soc/sh/rcar/dvc.c | 20 +- sound/soc/sh/rcar/gen.c | 95 + sound/soc/sh/rcar/rsnd.h | 10 +- sound/soc/sh/rcar/src.c | 50 +- sound/soc/sh/rcar/ssi.c | 28 +- sound/soc/soc-cache.c | 3 + sound/soc/soc-jack.c | 88 +- sound/soc/tegra/tegra_alc5632.c | 16 +- sound/soc/tegra/tegra_max98090.c | 16 +- sound/soc/tegra/tegra_rt5640.c | 16 +- sound/soc/tegra/tegra_wm8903.c | 11 +- 94 files changed, 9610 insertions(+), 904 deletions(-) create mode 100644 include/linux/platform_data/adau17x1.h rename {sound/soc/omap => include/sound}/omap-pcm.h (100%) create mode 100644 include/sound/rt5677.h create mode 100644 sound/soc/blackfin/bfin-eval-adau1x61.c create mode 100644 sound/soc/blackfin/bfin-eval-adau1x81.c create mode 100644 sound/soc/codecs/adau1761-i2c.c create mode 100644 sound/soc/codecs/adau1761-spi.c create mode 100644 sound/soc/codecs/adau1761.c create mode 100644 sound/soc/codecs/adau1761.h create mode 100644 sound/soc/codecs/adau1781-i2c.c create mode 100644 sound/soc/codecs/adau1781-spi.c create mode 100644 sound/soc/codecs/adau1781.c create mode 100644 sound/soc/codecs/adau1781.h create mode 100644 sound/soc/codecs/adau17x1.c create mode 100644 sound/soc/codecs/adau17x1.h create mode 100644 sound/soc/codecs/rl6231.c create mode 100644 sound/soc/codecs/rl6231.h create mode 100644 sound/soc/codecs/rt5677.c create mode 100644 sound/soc/codecs/rt5677.h create mode 100644 sound/soc/intel/byt-max98090.c

3 7

[alsa-devel] gitcompile fails
by Mark Brand 03 Jun '14

03 Jun '14

Fresh git clones of alsa-driver and alsa-kmirror produce a ./gitcompile failure for me: ln -s ../../alsa-kernel/include/pcm-indirect.h cp ../../alsa-kernel/include/pcm.h . patch -p0 -i pcm.patch pcm.h patching file pcm.h Hunk #1 FAILED at 280. Hunk #2 succeeded at 291 (offset -3 lines). [snip] Hunk #11 succeeded at 566 (offset -4 lines). 1 out of 11 hunks FAILED -- saving rejects to file pcm.h.rej make[3]: *** [pcm.h] Error 1 make[3]: Leaving directory `/home/mark/workspace/alsa/alsa-driver/alsa/include/sound' Since I've seen this before (http://mailman.alsa-project.org/pipermail/alsa-devel/2012-July/053419.html), would I be forgiven for assuming that this is once again as a result of a mismatch between alsa-driver and alsa-kmirror? Any suggestion will be much appreciated, thank you. Mark Brand.

1 0

[alsa-devel] [PATCH] ASoC: free jack GPIOs before the sound card is freed
by Stephen Warren 03 Jun '14

03 Jun '14

From: Stephen Warren <swarren(a)nvidia.com> This is the same change as commit fb6b8e71448a "ASoC: tegra: free jack GPIOs before the sound card is freed", but applied to all other ASoC machine drivers where code inspection indicates the same problem exists. That commit's description is: ========== snd_soc_jack_add_gpios() schedules a work queue item to poll the GPIO to generate an initial jack status report. If sound card initialization fails, that work item needs to be cancelled, so it doesn't run after the card has been freed. Specifically, freeing the card calls snd_jack_dev_free() which calls snd_jack_dev_disconnect() which sets jack->input_dev = NULL, and input_dev is used by snd_jack_report(), which is called from the work queue item. snd_soc_jack_free_gpios() cancels the work item. The Tegra ASoC machine drivers do call this function in the platform driver remove() callback. However, this happens after the sound card is freed, at least when the card is freed due to errors late during snd_soc_instantiate_card(). This leaves a window where the work item can execute after the card is freed. In next-20140522, sound card initialization does fail for unrelated reasons, and hits the problem described above. To solve this, fix the Tegra ASoC machine drivers to clean up the Jack GPIOs during the snd_soc_card's .remove() callback, which is executed before the overall card object is freed. also, guard the cleanup call based on whether we actually setup up the GPIOs in the first place. Ideally, we'd do the cleanup in a struct snd_soc_dai_link .fini/remove function to match where the GPIOs get set up. However, there is no such callback. ========== Note that I have not even compile-tested this in most cases, since most of the drivers rely on specific mach-* support I don't have enabled, and don't support COMPILE_TEST. Testing by the relevant board maintainers would be useful. Cc: Peter Ujfalusi <peter.ujfalusi(a)ti.com> (maintainer:OMAP AUDIO SUPPORT) Cc: Jarkko Nikula <jarkko.nikula(a)bitmer.com> (maintainer:OMAP AUDIO SUPPORT) Cc: Philipp Zabel <philipp.zabel(a)gmail.com> (maintainer:ARM/H4700 (HP IPA...) Cc: Paul Parsons <lost.distance(a)yahoo.com> (maintainer:ARM/H4700 (HP IPA...) Cc: Eric Miao <eric.y.miao(a)gmail.com> (maintainer:PXA2xx/PXA3xx SUP...) Cc: Russell King <linux(a)arm.linux.org.uk> (maintainer:PXA2xx/PXA3xx SUP...) Cc: Haojian Zhuang <haojian.zhuang(a)gmail.com> (maintainer:PXA2xx/PXA3xx SUP...) Cc: Ben Dooks <ben-linux(a)fluff.org> (maintainer:ARM/SAMSUNG ARM A...) Cc: Kukjin Kim <kgene.kim(a)samsung.com> (maintainer:ARM/SAMSUNG ARM A...) Cc: Sangbeom Kim <sbkim73(a)samsung.com> (supporter:SAMSUNG AUDIO (AS...) Cc: linux-omap(a)vger.kernel.org (open list:OMAP AUDIO SUPPORT) Cc: linux-samsung-soc(a)vger.kernel.org (moderated list:ARM/SAMSUNG ARM A...) Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- Mark, do you need this patch split up by architecture for topic branches? I assume not if you've already sent your pull requests. This applies fine on top of tag asoc-v3.16. --- sound/soc/omap/ams-delta.c | 14 ++++++++++---- sound/soc/omap/omap-twl4030.c | 28 ++++++++++++++-------------- sound/soc/omap/rx51.c | 18 +++++++++--------- sound/soc/pxa/hx4700.c | 9 ++++++++- sound/soc/samsung/h1940_uda1380.c | 11 +++++++++-- sound/soc/samsung/rx1950_uda1380.c | 12 ++++++++++-- sound/soc/samsung/smartq_wm8987.c | 11 +++++++++-- 7 files changed, 69 insertions(+), 34 deletions(-) diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c index bb243c663e6b..1f41951d8b7f 100644 --- a/sound/soc/omap/ams-delta.c +++ b/sound/soc/omap/ams-delta.c @@ -527,6 +527,15 @@ static int ams_delta_cx20442_init(struct snd_soc_pcm_runtime *rtd) return 0; } +static int ams_delta_card_remove(struct snd_soc_pcm_runtime *rtd) +{ + snd_soc_jack_free_gpios(&ams_delta_hook_switch, + ARRAY_SIZE(ams_delta_hook_switch_gpios), + ams_delta_hook_switch_gpios); + + return 0; +} + /* DAI glue - connects codec <--> CPU */ static struct snd_soc_dai_link ams_delta_dai_link = { .name = "CX20442", @@ -543,6 +552,7 @@ static struct snd_soc_dai_link ams_delta_dai_link = { static struct snd_soc_card ams_delta_audio_card = { .name = "AMS_DELTA", .owner = THIS_MODULE, + .remove = ams_delta_card_remove, .dai_link = &ams_delta_dai_link, .num_links = 1, @@ -579,10 +589,6 @@ static int ams_delta_remove(struct platform_device *pdev) dev_warn(&pdev->dev, "failed to unregister V253 line discipline\n"); - snd_soc_jack_free_gpios(&ams_delta_hook_switch, - ARRAY_SIZE(ams_delta_hook_switch_gpios), - ams_delta_hook_switch_gpios); - snd_soc_unregister_card(card); card->dev = NULL; return 0; diff --git a/sound/soc/omap/omap-twl4030.c b/sound/soc/omap/omap-twl4030.c index 64141db311b2..b4e282871658 100644 --- a/sound/soc/omap/omap-twl4030.c +++ b/sound/soc/omap/omap-twl4030.c @@ -231,6 +231,19 @@ static int omap_twl4030_init(struct snd_soc_pcm_runtime *rtd) return ret; } +static int omap_twl4030_card_remove(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_card *card = rtd->card; + struct omap_twl4030 *priv = snd_soc_card_get_drvdata(card); + + if (priv->jack_detect > 0) + snd_soc_jack_free_gpios(&priv->hs_jack, + ARRAY_SIZE(hs_jack_gpios), + hs_jack_gpios); + + return 0; +} + /* Digital audio interface glue - connects codec <--> CPU */ static struct snd_soc_dai_link omap_twl4030_dai_links[] = { { @@ -258,6 +271,7 @@ static struct snd_soc_dai_link omap_twl4030_dai_links[] = { /* Audio machine driver */ static struct snd_soc_card omap_twl4030_card = { .owner = THIS_MODULE, + .remove = omap_twl4030_card_remove, .dai_link = omap_twl4030_dai_links, .num_links = ARRAY_SIZE(omap_twl4030_dai_links), @@ -353,19 +367,6 @@ static int omap_twl4030_probe(struct platform_device *pdev) return 0; } -static int omap_twl4030_remove(struct platform_device *pdev) -{ - struct snd_soc_card *card = platform_get_drvdata(pdev); - struct omap_twl4030 *priv = snd_soc_card_get_drvdata(card); - - if (priv->jack_detect > 0) - snd_soc_jack_free_gpios(&priv->hs_jack, - ARRAY_SIZE(hs_jack_gpios), - hs_jack_gpios); - - return 0; -} - static const struct of_device_id omap_twl4030_of_match[] = { {.compatible = "ti,omap-twl4030", }, { }, @@ -380,7 +381,6 @@ static struct platform_driver omap_twl4030_driver = { .of_match_table = omap_twl4030_of_match, }, .probe = omap_twl4030_probe, - .remove = omap_twl4030_remove, }; module_platform_driver(omap_twl4030_driver); diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c index 6951dc812055..47a10290535b 100644 --- a/sound/soc/omap/rx51.c +++ b/sound/soc/omap/rx51.c @@ -334,6 +334,14 @@ static int rx51_aic34_init(struct snd_soc_pcm_runtime *rtd) return err; } +static int rx51_card_remove(struct snd_soc_pcm_runtime *rtd) +{ + snd_soc_jack_free_gpios(&rx51_av_jack, ARRAY_SIZE(rx51_av_jack_gpios), + rx51_av_jack_gpios); + + return 0; +} + /* Digital audio interface glue - connects codec <--> CPU */ static struct snd_soc_dai_link rx51_dai[] = { { @@ -368,6 +376,7 @@ static struct snd_soc_codec_conf rx51_codec_conf[] = { static struct snd_soc_card rx51_sound_card = { .name = "RX-51", .owner = THIS_MODULE, + .remove = rx51_card_remove, .dai_link = rx51_dai, .num_links = ARRAY_SIZE(rx51_dai), .aux_dev = rx51_aux_dev, @@ -499,14 +508,6 @@ static int rx51_soc_probe(struct platform_device *pdev) return 0; } -static int rx51_soc_remove(struct platform_device *pdev) -{ - snd_soc_jack_free_gpios(&rx51_av_jack, ARRAY_SIZE(rx51_av_jack_gpios), - rx51_av_jack_gpios); - - return 0; -} - #if defined(CONFIG_OF) static const struct of_device_id rx51_audio_of_match[] = { { .compatible = "nokia,n900-audio", }, @@ -522,7 +523,6 @@ static struct platform_driver rx51_soc_driver = { .of_match_table = of_match_ptr(rx51_audio_of_match), }, .probe = rx51_soc_probe, - .remove = rx51_soc_remove, }; module_platform_driver(rx51_soc_driver); diff --git a/sound/soc/pxa/hx4700.c b/sound/soc/pxa/hx4700.c index dcc9b04bd92c..6b81acaffddd 100644 --- a/sound/soc/pxa/hx4700.c +++ b/sound/soc/pxa/hx4700.c @@ -152,6 +152,13 @@ static int hx4700_ak4641_init(struct snd_soc_pcm_runtime *rtd) return err; } +static int hx4700_card_remove(struct snd_soc_pcm_runtime *rtd) +{ + snd_soc_jack_free_gpios(&hs_jack, 1, &hs_jack_gpio); + + return 0; +} + /* hx4700 digital audio interface glue - connects codec <--> CPU */ static struct snd_soc_dai_link hx4700_dai = { .name = "ak4641", @@ -170,6 +177,7 @@ static struct snd_soc_dai_link hx4700_dai = { static struct snd_soc_card snd_soc_card_hx4700 = { .name = "iPAQ hx4700", .owner = THIS_MODULE, + .remove = hx4700_card_remove, .dai_link = &hx4700_dai, .num_links = 1, .dapm_widgets = hx4700_dapm_widgets, @@ -206,7 +214,6 @@ static int hx4700_audio_probe(struct platform_device *pdev) static int hx4700_audio_remove(struct platform_device *pdev) { - snd_soc_jack_free_gpios(&hs_jack, 1, &hs_jack_gpio); snd_soc_unregister_card(&snd_soc_card_hx4700); gpio_set_value(GPIO92_HX4700_HP_DRIVER, 0); diff --git a/sound/soc/samsung/h1940_uda1380.c b/sound/soc/samsung/h1940_uda1380.c index 9f2fb69dbaae..720357f11a7f 100644 --- a/sound/soc/samsung/h1940_uda1380.c +++ b/sound/soc/samsung/h1940_uda1380.c @@ -189,6 +189,14 @@ static int h1940_uda1380_init(struct snd_soc_pcm_runtime *rtd) return 0; } +static int h1940_uda1380_card_remove(struct snd_soc_pcm_runtime *rtd) +{ + snd_soc_jack_free_gpios(&hp_jack, ARRAY_SIZE(hp_jack_gpios), + hp_jack_gpios); + + return 0; +} + /* s3c24xx digital audio interface glue - connects codec <--> CPU */ static struct snd_soc_dai_link h1940_uda1380_dai[] = { { @@ -206,6 +214,7 @@ static struct snd_soc_dai_link h1940_uda1380_dai[] = { static struct snd_soc_card h1940_asoc = { .name = "h1940", .owner = THIS_MODULE, + .remove = h1940_uda1380_card_remove, .dai_link = h1940_uda1380_dai, .num_links = ARRAY_SIZE(h1940_uda1380_dai), @@ -257,8 +266,6 @@ err_out: static void __exit h1940_exit(void) { platform_device_unregister(s3c24xx_snd_device); - snd_soc_jack_free_gpios(&hp_jack, ARRAY_SIZE(hp_jack_gpios), - hp_jack_gpios); gpio_free(S3C_GPIO_END + 9); } diff --git a/sound/soc/samsung/rx1950_uda1380.c b/sound/soc/samsung/rx1950_uda1380.c index 5b3e504d3a32..192aa9fc102f 100644 --- a/sound/soc/samsung/rx1950_uda1380.c +++ b/sound/soc/samsung/rx1950_uda1380.c @@ -31,6 +31,7 @@ #include "s3c24xx-i2s.h" static int rx1950_uda1380_init(struct snd_soc_pcm_runtime *rtd); +static int rx1950_uda1380_card_remove(struct snd_soc_pcm_runtime *rtd); static int rx1950_startup(struct snd_pcm_substream *substream); static int rx1950_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params); @@ -116,6 +117,7 @@ static const struct snd_soc_dapm_route audio_map[] = { static struct snd_soc_card rx1950_asoc = { .name = "rx1950", .owner = THIS_MODULE, + .remove = rx1950_uda1380_card_remove, .dai_link = rx1950_uda1380_dai, .num_links = ARRAY_SIZE(rx1950_uda1380_dai), @@ -234,6 +236,14 @@ static int rx1950_uda1380_init(struct snd_soc_pcm_runtime *rtd) return 0; } +static int rx1950_uda1380_card_remove(struct snd_soc_pcm_runtime *rtd) +{ + snd_soc_jack_free_gpios(&hp_jack, ARRAY_SIZE(hp_jack_gpios), + hp_jack_gpios); + + return 0; +} + static int __init rx1950_init(void) { int ret; @@ -278,8 +288,6 @@ err_gpio: static void __exit rx1950_exit(void) { platform_device_unregister(s3c24xx_snd_device); - snd_soc_jack_free_gpios(&hp_jack, ARRAY_SIZE(hp_jack_gpios), - hp_jack_gpios); gpio_free(S3C2410_GPA(1)); } diff --git a/sound/soc/samsung/smartq_wm8987.c b/sound/soc/samsung/smartq_wm8987.c index df55db5d3554..271a904277a1 100644 --- a/sound/soc/samsung/smartq_wm8987.c +++ b/sound/soc/samsung/smartq_wm8987.c @@ -182,6 +182,14 @@ static int smartq_wm8987_init(struct snd_soc_pcm_runtime *rtd) return err; } +static int smartq_wm8987_card_remove(struct snd_soc_pcm_runtime *rtd) +{ + snd_soc_jack_free_gpios(&smartq_jack, ARRAY_SIZE(smartq_jack_gpios), + smartq_jack_gpios); + + return 0; +} + static struct snd_soc_dai_link smartq_dai[] = { { .name = "wm8987", @@ -198,6 +206,7 @@ static struct snd_soc_dai_link smartq_dai[] = { static struct snd_soc_card snd_soc_smartq = { .name = "SmartQ", .owner = THIS_MODULE, + .remove = smartq_wm8987_card_remove, .dai_link = smartq_dai, .num_links = ARRAY_SIZE(smartq_dai), @@ -259,8 +268,6 @@ err_unregister_device: static void __exit smartq_exit(void) { gpio_free(S3C64XX_GPK(12)); - snd_soc_jack_free_gpios(&smartq_jack, ARRAY_SIZE(smartq_jack_gpios), - smartq_jack_gpios); platform_device_unregister(smartq_snd_device); } -- 1.8.1.5

2 1

[alsa-devel] [RFC] set up an sync channel between audio and display driver (i.e. ALSA and DRM)
by Lin, Mengdong 03 Jun '14

03 Jun '14

This RFC is based on previous discussion to set up a generic communication channel between display and audio driver and an internal design of Intel MCG/VPG HDMI audio driver. It's still an initial draft and your advice would be appreciated to improve the design. The basic idea is to create a new avsink module and let both drm and alsa depend on it. This new module provides a framework and APIs for synchronization between the display and audio driver. 1. Display/Audio Client The avsink core provides APIs to create, register and lookup a display/audio client. A specific display driver (eg. i915) or audio driver (eg. HD-Audio driver) can create a client, add some resources objects (shared power wells, display outputs, and audio inputs, register ops) to the client, and then register this client to avisink core. The peer driver can look up a registered client by a name or type, or both. If a client gives a valid peer client name on registration, avsink core will bind the two clients as peer for each other. And we expect a display client and an audio client to be peers for each other in a system. int avsink_new_client ( const char *name, int type, /* client type, display or audio */ struct module *module, void *context, const char *peer_name, struct avsink_client **client_ret); int avsink_free_client (struct avsink_client *client); int avsink_register_client(struct avsink_client *client); int avisink_unregister_client(int client_handle); struct avsink_client *avsink_lookup_client(const char *name, int type); struct avsink_client { const char *name; /* client name */ int type; /* client type*/ void *context; struct module *module; /* top-level module for locking */ struct avsink_client *peer; /* peer client */ /* shared power wells */ struct avsink_power_well *power_well; int num_power_wells; /* endpoints, display outputs or audio inputs */ struct avsink_endpoint * endpoint; int num_endpints; struct avsink_registers_ops *reg_ops; /* ops to access registers of a client */ void *private_data; ... }; On system boots, the avsink module is loaded before the display and audio driver module. And the display and audio driver may be loaded on parallel. * If a specific display driver (eg. i915) supports avsink, it can create a display client, add power wells and display outputs to the client, and then register the display client to the avsink core. Then it may look up if there is any audio client registered, by name or type, and may find an audio client registered by some audio driver. * If an audio driver supports avsink, it usually should look up a registered display client by name or type at first, because it may need the shared power well in GPU and check the display outputs' name to bind the audio inputs. If the display client is not registered yet, the audio driver can choose to wait (maybe in a work queue) or return -EAGAIN for a deferred probe. After the display client is found, the audio driver can register an audio client with the display client's name as the peer name, the avsink core will bind the display and audio clients to each other. Open question: If the display or audio driver is disabled by the black list, shall we introduce a time out to avoid waiting for the other client registered endlessly? 2. Shared power wells (optional) The audio and display devices, maybe only part of them, may share a common power well (e.g. for Intel Haswell and Broadwell). If so, the driver that controls the power well should define a power well object, implement the get/put ops, and add it to its avsink client before registering the client to avsink core. Then the peer client can look up this power well by its name, and get/put this power well as a user. A client can have multiple power well objects. struct avsink_power_well { const char *name; /* name of the power well */ void *context; /* parameter of get/put ops, maybe device pointer for this power well */ struct avsink_power_well_ops *ops }; struct avsink_power_well_ops { int (*get)(void *context); int (*put)(void *context); }; API: int avsink_new_power(struct avsink_client *client, const char *power_name, void * power_context, struct avsink_power_well_ops *ops, struct avsink_power_well **power_ret); struct avsink_power_well *avisnk_lookup_power(const char *name); int avsink_get_power(struct avsink_power_well *power); /* Reqesut the power */ int avsink_put_power(struct avsink_power_well *power); /* Release the power */ For example, the i915 display driver can create a device for the shared power well in Haswell GPU, implement its PM functions, and use the device pointer as the context when creating the power well object, like below struct avsink_power_well_ops i915_power_well_ops = { .get = pm_runtime_get_sync; .put = pm_runtime_put_sync; }; ... avsink_new_power ( display_client, "i915_display_power_well", pdev, /* pointer of the power well device */ &i915_power_well_ops, ...) Power domain is not used here since a single device seems enough to represent a power well. 3. Display output and audio input endpoints A display client should register the display output endpoints and its audio peer client should register the audio input endpoints. A client can have multiple endpoints. The avsink core will bind an audio input and a display output as peer to each other. This is to allow the audio and display driver to synchronize with each other for each display pipeline. All endpoints should be added to a client before the client is registered to avsink core. Dynamic endpoints are not supported now. A display out here represents a physical HDMI/DP output port. And as long as it's usable in the system (i.e. physically connected to the HDMP/DP port on the machine board), the display output should be registered not matter the port is connected to an external display device or not. And if HW and display driver can support DP1.2 daisy chain (multiple DP display devices can be connected to a single port), multiple static displays outputs should be defined for the DP port according to the HW capability. The port & display device number can be indicated by the name (e.g. "i915_DDI_B", "i915_DDI_B_DEV0", "i915_DDI_B_DEV1", or "i915_DDI_B_DEV2"), defined by the display driver. The audio driver can check the endpoints of its peer display client and use an display endpoint's name, or a presumed display endpoint name, as peer name when registering an audio endpoint, thus the avsink core will bind the two display and audio endpoints as peers. struct avsink_endpoint { const char *name; /*name of the endpoint */ int type; /* DISPLAY_OUTPUT or AUDIO_INPUT */ void *context; /* private data, used as parameter of the ops */ struct avsink_endpoint_ops *ops; struct avsink_endpoint *peer; /* peer endpoint */ }; struct avsink_endpoint_ops { int (*get_caps) (enum had_caps_list query_element, void *capabilities, void *context); int (*set_caps) (enum had_caps_list set_element, void *capabilities, void *context); int (*event_handler) (enum avsink_event_type event_type, void *context); }; API: int avsink_new_endpoint (struct avsink_client *client, const char *name, int type, /* DISPLAY_OUTPUT or AUDIO_INPUT*/ void *context, const char *peer_name, /* can be NULL if no clue */ avsink_endpoint_ops *ops, struct avsink_endpoint **endpoint_ret); int avsink_endpoint_get_caps(struct avsink_endpoint *endpoint, enum avsink_caps_list t get_element, void *capabilities); int avsink_endpoint_set_caps(struct avsink_endpoint *endpoint, enum had_caps_list set_element, void *capabilities); int avsink_endpoint_post_event(struct avsink_endpoint *endpoint, enum avsink_event_type event_type); 4. Get/Set caps on an endpoint The display or audio driver can get or set capabilities on an endpoint. Depending on the capability ID, the avsink core will call get_caps/set_caps ops of this endpoint, or call get_caps/set_caps ops of its peer endpoint and return the result to the caller. enum avsink_caps_list { /* capabilities for display output endpoints */ AVSINK_GET_DISPLAY_ELD = 1, AVSINK_GET_DISPLAY_TYPE, /* HDMI or DisplayPort */ AVSINK_GET_DISPLAY_NAME, /* Hope to use display device name under /sys/class/drm, like "card0-DP-1", for user * space to figure out which HDMI/DP output on the drm side corresponds to which audio * stream device on the alsa side */ AVSINK_GET_DISPLAY_SAMPLING_FREQ, /* HDMI TMDS clock or DP link symbol clock, for audio driver to * program N value */ AVSINK_GET_DISPLAY_HDCP_STATUS, AVSINK_GET_DISPLAY_AUDIO_STATUS, /* Whether audio is enabled */ AVSINK_SET_DISPLAY_ENABLE_AUDIO, /* Enable audio */ AVSINK_SET_DISPLAY_DISABLE_AUDIO, /* Disable audio */ AVSINK_SET_DISPLAY_ENABLE_AUDIO_INT, /* Enable audio interrupt */ AVSINK_SET_DISPLAY_DISABLE_AUDIO_INT, /* Disable audio interrupt */ /* capabilities for audio input endpoints */ AVSINK_GET_AUDIO_IS_BUSY, /* Whether there is an active audio streaming */ OTHERS_TBD, }; For example, the audio driver can query ELD info on an audio input endpoint by using caps AVSINK_GET_DISPLAY_ELD, and avsink core will call get_caps() on the peer display output endpoint and return the ELD info to the audio driver. Some audio driver may only use part of these caps. E.g. HD-Audio driver can use bus commands instead of the ops to control the audio on gfx side, so it doesn't use caps like ENABLE/DISABLE_AUDIO or ENABLE/DISABLE_AUDIO. When the display driver want to disable a display pipeline for hot-plug, mode change or power saving, it can use caps AVSINK_GET_AUDIO_IS_BUSY to check if the audio input is busy (active streaming) on this display pipeline. And if audio is busy, the display driver can choose to wait or go ahead to disable display pipeline anyway. For the latter case, the audio input endpoint will be notified by an event and should abort audio streaming. 5. Event handling of endpoints A driver can post events on an endpoint. Depending on the event type, the avsink core will call the endpoint's event handler or pass the event to its peer endpoint and trigger the peer's event handler function if defined. int avsink_endpoint_post_event(struct avsink_endpoint *endpoint, enum avsink_event_type event_type); Now we only defined event types which should be handled by the audio input endpoints. The event types can be extended in the future. enum avsink_event_type { AVSINK_EVENT_DISPLAY_DISABLE = 1, /* The display pipeline is disabled for hot-plug, mode change or * suspend. Audio driver should stop any active streaming. */ AVSINK_EVENT_DISPLAY_ENABLE, /* The display pipeline is enabled after hot-plug, mode change or * resume. Audio driver can restore previously interrupted streaming */ AVSINK_EVENT_DISPLAY_MODE_CHANGE, /* Display mode change event. At this time, the new display mode is * configured but the display pipeline is not enabled yet. Audio driver * can do some configurations such as programing N value */ AVSINK_EVENT_DISPLAY_AUDIO_BUFFER_DONE, /* Audio Buffer done interrupts. Only for audio drivers if DMA and * interrupt are handled by GPU */ AVSINK_EVENT_DISPLAY_AUDIO_BUFFER_UNDERRUN, /* Audio Buffer under run interrupts. Only for audio drivers if * DMA and interrupt are handled by GPU */ }; So for a display driver, it can post an event on a display output endpoint and get processed by the peer audio input endpoint. Or it can also directly post an event on a peer audio input endpoint, by using the 'peer' pointer on a display output endpoint. 6. Display register operation (optional) Some audio driver needs to access GPU audio registers. The register ops are provided by the peer display client. struct avsink_registers_ops { int (*read_register) (uint32_t reg_addr, uint32_t *data, void *context); int (*write_register) (uint32_t reg_addr, uint32_t data, void *context); int (*read_modify_register) (uint32_t reg_addr, uint32_t data, uint32_t mask, void *context); int avsink_define_reg_ops (struct avsink_client *client, struct avsink_registers_ops *ops); And avsink core provides API for the audio driver to access the display registers: int avsink_read_display_register(struct avsink_client *client , uint32_t offset, uint32_t *data); int avsink_write_display_register(struct avsink_client *client , uint32_t offset, uint32_t data); int avsink_read_modify_display_register(struct avsink_client *client, uint32_t offset, uint32_t data, uint32_t mask); If the client is an audio client, the avsink core will find it peer display client and call its register ops; and if the client is a display client, the avsink core will just call its own register ops. Thanks Mengdong

10 23

Re: [alsa-devel] [PATCH] ARM: i.MX: Compile FIQ code for AVIC & TZIC only
by Lothar Waßmann 03 Jun '14

03 Jun '14

Hi, Alexander Shiyan wrote: > There are no reason to compile FIQ specific code for GIC IRQ > controller (i.MX6). This patch restrict compilation for FIQ > specific code for AVIC and TZIC only. > > Signed-off-by: Alexander Shiyan <shc_work(a)mail.ru> > --- > arch/arm/mach-imx/Makefile | 6 +++--- > 1 file changed, 3 insertions(+), 3 deletions(-) > > diff --git a/arch/arm/mach-imx/Makefile b/arch/arm/mach-imx/Makefile > index bbe93bb..620a760 100644 > --- a/arch/arm/mach-imx/Makefile > +++ b/arch/arm/mach-imx/Makefile > @@ -1,4 +1,4 @@ > -obj-y := time.o cpu.o system.o irq-common.o > +obj-y := time.o cpu.o system.o > > obj-$(CONFIG_SOC_IMX1) += clk-imx1.o mm-imx1.o > obj-$(CONFIG_SOC_IMX21) += clk-imx21.o mm-imx21.o > @@ -21,8 +21,8 @@ obj-$(CONFIG_COMMON_CLK) += clk-pllv1.o clk-pllv2.o clk-pllv3.o clk-gate2.o \ > obj-$(CONFIG_IMX_HAVE_IOMUX_V1) += iomux-v1.o > obj-$(CONFIG_ARCH_MXC_IOMUX_V3) += iomux-v3.o > > -obj-$(CONFIG_MXC_TZIC) += tzic.o > -obj-$(CONFIG_MXC_AVIC) += avic.o > +obj-$(CONFIG_MXC_TZIC) += tzic.o irq-common.o > +obj-$(CONFIG_MXC_AVIC) += avic.o irq-common.o > > obj-$(CONFIG_MXC_USE_EPIT) += epit.o > obj-$(CONFIG_MXC_DEBUG_BOARD) += 3ds_debugboard.o > This patch introduces a build error in sound/soc/fsl/imx-pcm-fiq.c: sound/built-in.o: In function `imx_pcm_fiq_free': last.c:(.text+0x2aca8): undefined reference to `mxc_set_irq_fiq' sound/built-in.o: In function `imx_pcm_fiq_init': last.c:(.text+0x2b1a4): undefined reference to `mxc_set_irq_fiq' last.c:(.text+0x2b200): undefined reference to `mxc_set_irq_fiq' make[1]: *** [vmlinux] Error 1 make: *** [sub-make] Error 2 This could be fixed with the following patch: --- mxc-fiq.orig/sound/soc/fsl/Kconfig 2014-06-02 11:25:38.315960451 +0200 +++ mxc-fiq/sound/soc/fsl/Kconfig 2014-06-02 11:25:38.411960595 +0200 @@ -15,7 +15,7 @@ config SND_SOC_FSL_SSI tristate "Synchronous Serial Interface module support" select SND_SOC_IMX_PCM_DMA if SND_IMX_SOC != n - select SND_SOC_IMX_PCM_FIQ if SND_IMX_SOC != n && ARCH_MXC + select SND_SOC_IMX_PCM_FIQ if SND_IMX_SOC != n && (MXC_AVIC || MXC_TZIC) help Say Y if you want to add Synchronous Serial Interface (SSI) support for the Freescale CPUs. Lothar Waßmann -- ___________________________________________________________ Ka-Ro electronics GmbH | Pascalstraße 22 | D - 52076 Aachen Phone: +49 2408 1402-0 | Fax: +49 2408 1402-10 Geschäftsführer: Matthias Kaussen Handelsregistereintrag: Amtsgericht Aachen, HRB 4996 www.karo-electronics.de | info(a)karo-electronics.de ___________________________________________________________

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[alsa-devel] [PATCH] firewire-lib: Remove a comment about restriction of asynchronous operation
by Takashi Sakamoto 03 Jun '14

03 Jun '14

The comment for fcp_avc_transaction() describes it doesn't support this type of operation. But it was already supported by this commit. 00a7bb81c20f3e81711e28e0f6c08cee8fd18514 ALSA: firewire-lib: Add support for deferred transaction Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/fcp.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/firewire/fcp.c b/sound/firewire/fcp.c index d48e326..0619597 100644 --- a/sound/firewire/fcp.c +++ b/sound/firewire/fcp.c @@ -218,8 +218,6 @@ struct fcp_transaction { * * @command and @response can point to the same buffer. * - * Asynchronous operation (INTERIM, NOTIFY) is not supported at the moment. - * * Returns the actual size of the response frame, or a negative error code. */ int fcp_avc_transaction(struct fw_unit *unit, -- 1.8.3.2

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[alsa-devel] [PATCH 0/6] ASoC: rsnd: tidyup for DMA support for DT
by Kuninori Morimoto 03 Jun '14

03 Jun '14

Hi Mark These patches are tidyup code and fixup for DMA support of DT #1 - #4 are cleanup code patches #5 - #6 are fixup for DMA for DT Kuninori Morimoto (6): ASoC: rsnd: DT node clean up by using the of_node_put() ASoC: rsnd: save platform_device instead of device ASoC: rsnd: remove rsnd_src_non_ops ASoC: rsnd: module name is unified ASoC: rsnd: care DMA slave channel name for DT ASoC: rsnd: add rsnd_gen_dma_addr() for DMAC addr .../devicetree/bindings/sound/renesas,rsnd.txt | 1 + sound/soc/sh/rcar/core.c | 87 ++++++++++++++++-- sound/soc/sh/rcar/dvc.c | 20 ++++- sound/soc/sh/rcar/gen.c | 95 ++++++++++++++++++++ sound/soc/sh/rcar/rsnd.h | 10 ++- sound/soc/sh/rcar/src.c | 50 +++++++---- sound/soc/sh/rcar/ssi.c | 28 ++++-- 7 files changed, 260 insertions(+), 31 deletions(-)

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[alsa-devel] [PATCH] ASoC: rsnd: fixup SSI interrupts style of DeviceTree
by Kuninori Morimoto 03 Jun '14

03 Jun '14

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Current SSI interrupts has been set to each port. But it is too ugly style. SSI interrupts will be listed under rcar_sound,ssi node by this patch Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- >> Mark There is still no DT user for this driver at this point. No conflict happens on platform. .../devicetree/bindings/sound/renesas,rsnd.txt | 50 ++++++++------------ sound/soc/sh/rcar/ssi.c | 2 +- 2 files changed, 21 insertions(+), 31 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt index 8346cab..1fba56b 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -53,36 +53,26 @@ rcar_sound: rcar_sound@0xffd90000 { }; rcar_sound,ssi { - ssi0: ssi@0 { - interrupts = <0 370 IRQ_TYPE_LEVEL_HIGH>; - }; - ssi1: ssi@1 { - interrupts = <0 371 IRQ_TYPE_LEVEL_HIGH>; - }; - ssi2: ssi@2 { - interrupts = <0 372 IRQ_TYPE_LEVEL_HIGH>; - }; - ssi3: ssi@3 { - interrupts = <0 373 IRQ_TYPE_LEVEL_HIGH>; - }; - ssi4: ssi@4 { - interrupts = <0 374 IRQ_TYPE_LEVEL_HIGH>; - }; - ssi5: ssi@5 { - interrupts = <0 375 IRQ_TYPE_LEVEL_HIGH>; - }; - ssi6: ssi@6 { - interrupts = <0 376 IRQ_TYPE_LEVEL_HIGH>; - }; - ssi7: ssi@7 { - interrupts = <0 377 IRQ_TYPE_LEVEL_HIGH>; - }; - ssi8: ssi@8 { - interrupts = <0 378 IRQ_TYPE_LEVEL_HIGH>; - }; - ssi9: ssi@9 { - interrupts = <0 379 IRQ_TYPE_LEVEL_HIGH>; - }; + interrupts = <0 370 IRQ_TYPE_LEVEL_HIGH>, + <0 371 IRQ_TYPE_LEVEL_HIGH>, + <0 372 IRQ_TYPE_LEVEL_HIGH>, + <0 373 IRQ_TYPE_LEVEL_HIGH>, + <0 374 IRQ_TYPE_LEVEL_HIGH>, + <0 375 IRQ_TYPE_LEVEL_HIGH>, + <0 376 IRQ_TYPE_LEVEL_HIGH>, + <0 377 IRQ_TYPE_LEVEL_HIGH>, + <0 378 IRQ_TYPE_LEVEL_HIGH>, + <0 379 IRQ_TYPE_LEVEL_HIGH>; + ssi0: ssi@0 { }; + ssi1: ssi@1 { }; + ssi2: ssi@2 { }; + ssi3: ssi@3 { }; + ssi4: ssi@4 { }; + ssi5: ssi@5 { }; + ssi6: ssi@6 { }; + ssi7: ssi@7 { }; + ssi8: ssi@8 { }; + ssi9: ssi@9 { }; }; rcar_sound,dai { diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 2df723d..b311bc6 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -588,7 +588,7 @@ static void rsnd_of_parse_ssi(struct platform_device *pdev, /* * irq */ - ssi_info->pio_irq = irq_of_parse_and_map(np, 0); + ssi_info->pio_irq = irq_of_parse_and_map(node, i); /* * DMA -- 1.7.9.5

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