November 2010 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 17/17][trivial] Sound, OSS: Remove unnecessary casts of void ptr returning alloc function return values
by Jesper Juhl 09 Nov '10

09 Nov '10

Hi, The [vk][cmz]alloc(_node) family of functions return void pointers which it's completely unnecessary/pointless to cast to other pointer types since that happens implicitly. This patch removes such casts from sound/oss/ Signed-off-by: Jesper Juhl <jj(a)chaosbits.net> --- midibuf.c | 4 ++-- pss.c | 6 +++--- sequencer.c | 4 ++-- 3 files changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/oss/midibuf.c b/sound/oss/midibuf.c index 782b3b8..ceedb1e 100644 --- a/sound/oss/midibuf.c +++ b/sound/oss/midibuf.c @@ -178,7 +178,7 @@ int MIDIbuf_open(int dev, struct file *file) return err; parms[dev].prech_timeout = MAX_SCHEDULE_TIMEOUT; - midi_in_buf[dev] = (struct midi_buf *) vmalloc(sizeof(struct midi_buf)); + midi_in_buf[dev] = vmalloc(sizeof(struct midi_buf)); if (midi_in_buf[dev] == NULL) { @@ -188,7 +188,7 @@ int MIDIbuf_open(int dev, struct file *file) } midi_in_buf[dev]->len = midi_in_buf[dev]->head = midi_in_buf[dev]->tail = 0; - midi_out_buf[dev] = (struct midi_buf *) vmalloc(sizeof(struct midi_buf)); + midi_out_buf[dev] = vmalloc(sizeof(struct midi_buf)); if (midi_out_buf[dev] == NULL) { diff --git a/sound/oss/pss.c b/sound/oss/pss.c index e19dd5d..9b800ce 100644 --- a/sound/oss/pss.c +++ b/sound/oss/pss.c @@ -859,7 +859,7 @@ static int pss_coproc_ioctl(void *dev_info, unsigned int cmd, void __user *arg, return 0; case SNDCTL_COPR_LOAD: - buf = (copr_buffer *) vmalloc(sizeof(copr_buffer)); + buf = vmalloc(sizeof(copr_buffer)); if (buf == NULL) return -ENOSPC; if (copy_from_user(buf, arg, sizeof(copr_buffer))) { @@ -871,7 +871,7 @@ static int pss_coproc_ioctl(void *dev_info, unsigned int cmd, void __user *arg, return err; case SNDCTL_COPR_SENDMSG: - mbuf = (copr_msg *)vmalloc(sizeof(copr_msg)); + mbuf = vmalloc(sizeof(copr_msg)); if (mbuf == NULL) return -ENOSPC; if (copy_from_user(mbuf, arg, sizeof(copr_msg))) { @@ -895,7 +895,7 @@ static int pss_coproc_ioctl(void *dev_info, unsigned int cmd, void __user *arg, case SNDCTL_COPR_RCVMSG: err = 0; - mbuf = (copr_msg *)vmalloc(sizeof(copr_msg)); + mbuf = vmalloc(sizeof(copr_msg)); if (mbuf == NULL) return -ENOSPC; data = (unsigned short *)mbuf->data; diff --git a/sound/oss/sequencer.c b/sound/oss/sequencer.c index e85789e..5ea1098 100644 --- a/sound/oss/sequencer.c +++ b/sound/oss/sequencer.c @@ -1646,13 +1646,13 @@ void sequencer_init(void) { if (sequencer_ok) return; - queue = (unsigned char *)vmalloc(SEQ_MAX_QUEUE * EV_SZ); + queue = vmalloc(SEQ_MAX_QUEUE * EV_SZ); if (queue == NULL) { printk(KERN_ERR "sequencer: Can't allocate memory for sequencer output queue\n"); return; } - iqueue = (unsigned char *)vmalloc(SEQ_MAX_QUEUE * IEV_SZ); + iqueue = vmalloc(SEQ_MAX_QUEUE * IEV_SZ); if (iqueue == NULL) { printk(KERN_ERR "sequencer: Can't allocate memory for sequencer input queue\n"); -- Jesper Juhl <jj(a)chaosbits.net> http://www.chaosbits.net/ Don't top-post http://www.catb.org/~esr/jargon/html/T/top-post.html Plain text mails only, please.

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[alsa-devel] [PATCH] ALSA: add support for disabling period irq
by Pierre-Louis Bossart 08 Nov '10

08 Nov '10

Merged and cleaned patch based on earlier patches posted on alsa-devel by Clemens Ladisch <clemens(a)ladisch.de> and Pierre-Louis Bossart <pierre-louis.bossart(a)intel.com> This patch disables period interrupts which are not needed when the application relies on a system timer to wake-up and refill the ring buffer. The behavior of the driver is left unchanged, and interrupts are only disabled if the application requests this configuration. The behavior in case of underruns is slightly different, instead of being detected during the period interrupts the underruns are detected when the application calls snd_pcm_update_avail, which in turns forces a refresh of the hw pointer and shows the buffer is empty. More specifically this patch makes a lot of sense when PulseAudio relies on timer-based scheduling to access audio devices such as HDAudio or Intel SST. Disabling interrupts removes two unwanted wake-ups due to period elapsed events in low-power playback modes. It also simplifies PulseAudio voice modules used for speech calls. To quote Lennart "This patch looks very interesting and desirable. This is something have long been waiting for." Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)intel.com> --- core/pcm_lib.c | 8 ++++++-- core/pcm_native.c | 6 ++++++ include/asound.h | 4 ++++ include/pcm.h | 1 + pci/hda/hda_intel.c | 9 ++++++--- pci/oxygen/oxygen_pcm.c | 14 ++++++++++---- 6 files changed, 33 insertions(+), 9 deletions(-) diff --git a/core/pcm_lib.c b/core/pcm_lib.c index a1707cc..4963d8f 100644 --- a/core/pcm_lib.c +++ b/core/pcm_lib.c @@ -374,7 +374,8 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, (unsigned long)runtime->hw_ptr_base); } /* something must be really wrong */ - if (delta >= runtime->buffer_size + runtime->period_size) { + if (delta >= runtime->buffer_size + runtime->period_size && + !runtime->no_period_irq) { hw_ptr_error(substream, "Unexpected hw_pointer value %s" "(stream=%i, pos=%ld, new_hw_ptr=%ld, " @@ -395,6 +396,8 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, */ if (runtime->hw.info & SNDRV_PCM_INFO_BATCH) goto no_jiffies_check; + if (runtime->no_period_irq) + goto no_jiffies_check; hdelta = delta; if (hdelta < runtime->delay) goto no_jiffies_check; @@ -431,7 +434,8 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, hw_base = new_hw_ptr - (new_hw_ptr % runtime->buffer_size); } no_jiffies_check: - if (delta > runtime->period_size + runtime->period_size / 2) { + if (delta > runtime->period_size + runtime->period_size / 2 && + !runtime->no_period_irq) { hw_ptr_error(substream, "Lost interrupts? %s" "(stream=%i, delta=%ld, new_hw_ptr=%ld, " diff --git a/core/pcm_native.c b/core/pcm_native.c index 8bc7cb3..92c8c59 100644 --- a/core/pcm_native.c +++ b/core/pcm_native.c @@ -412,6 +412,12 @@ static int snd_pcm_hw_params(struct snd_pcm_substream *substream, goto _error; } + if (params->info & SNDRV_PCM_INFO_NO_PERIOD_IRQ) + runtime->no_period_irq = !!(params->flags & + SNDRV_PCM_HW_PARAMS_NO_PERIOD_IRQ); + else + runtime->no_period_irq = 0; + runtime->access = params_access(params); runtime->format = params_format(params); runtime->subformat = params_subformat(params); diff --git a/include/asound.h b/include/asound.h index a1803ec..d4b5f2d 100644 --- a/include/asound.h +++ b/include/asound.h @@ -259,6 +259,7 @@ typedef int __bitwise snd_pcm_subformat_t; #define SNDRV_PCM_INFO_HALF_DUPLEX 0x00100000 /* only half duplex */ #define SNDRV_PCM_INFO_JOINT_DUPLEX 0x00200000 /* playback and capture stream are somewhat correlated */ #define SNDRV_PCM_INFO_SYNC_START 0x00400000 /* pcm support some kind of sync go */ +#define SNDRV_PCM_INFO_NO_PERIOD_IRQ 0x00800000 /* period interrupt can be disabled */ #define SNDRV_PCM_INFO_FIFO_IN_FRAMES 0x80000000 /* internal kernel flag - FIFO size is in frames */ typedef int __bitwise snd_pcm_state_t; @@ -334,6 +335,9 @@ typedef int snd_pcm_hw_param_t; #define SNDRV_PCM_HW_PARAM_LAST_INTERVAL SNDRV_PCM_HW_PARAM_TICK_TIME #define SNDRV_PCM_HW_PARAMS_NORESAMPLE (1<<0) /* avoid rate resampling */ +#define SNDRV_PCM_HW_PARAMS_EXPORT_BUFFER (1<<1) /* export buffer */ +#define SNDRV_PCM_HW_PARAMS_NO_PERIOD_IRQ (1<<2) /* disable period + interrupts */ struct snd_interval { unsigned int min, max; diff --git a/include/pcm.h b/include/pcm.h index dfd9b76..9abb4aa 100644 --- a/include/pcm.h +++ b/include/pcm.h @@ -297,6 +297,7 @@ struct snd_pcm_runtime { unsigned int info; unsigned int rate_num; unsigned int rate_den; + bool no_period_irq; /* -- SW params -- */ int tstamp_mode; /* mmap timestamp is updated */ diff --git a/pci/hda/hda_intel.c b/pci/hda/hda_intel.c index 38b063e..9f456a8 100644 --- a/pci/hda/hda_intel.c +++ b/pci/hda/hda_intel.c @@ -1227,7 +1227,8 @@ static int azx_setup_periods(struct azx *chip, pos_adj = 0; } else { ofs = setup_bdle(substream, azx_dev, - &bdl, ofs, pos_adj, 1); + &bdl, ofs, pos_adj, + !substream->runtime->no_period_irq); if (ofs < 0) goto error; } @@ -1239,7 +1240,8 @@ static int azx_setup_periods(struct azx *chip, period_bytes - pos_adj, 0); else ofs = setup_bdle(substream, azx_dev, &bdl, ofs, - period_bytes, 1); + period_bytes, + !substream->runtime->no_period_irq); if (ofs < 0) goto error; } @@ -1507,7 +1509,8 @@ static struct snd_pcm_hardware azx_pcm_hw = { /* No full-resume yet implemented */ /* SNDRV_PCM_INFO_RESUME |*/ SNDRV_PCM_INFO_PAUSE | - SNDRV_PCM_INFO_SYNC_START), + SNDRV_PCM_INFO_SYNC_START | + SNDRV_PCM_INFO_NO_PERIOD_IRQ), .formats = SNDRV_PCM_FMTBIT_S16_LE, .rates = SNDRV_PCM_RATE_48000, .rate_min = 48000, diff --git a/pci/oxygen/oxygen_pcm.c b/pci/oxygen/oxygen_pcm.c index 8146674..a32ee97 100644 --- a/pci/oxygen/oxygen_pcm.c +++ b/pci/oxygen/oxygen_pcm.c @@ -39,7 +39,8 @@ static const struct snd_pcm_hardware oxygen_stereo_hardware = { SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_PAUSE | - SNDRV_PCM_INFO_SYNC_START, + SNDRV_PCM_INFO_SYNC_START | + SNDRV_PCM_INFO_NO_PERIOD_IRQ, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE, .rates = SNDRV_PCM_RATE_32000 | @@ -65,7 +66,8 @@ static const struct snd_pcm_hardware oxygen_multichannel_hardware = { SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_PAUSE | - SNDRV_PCM_INFO_SYNC_START, + SNDRV_PCM_INFO_SYNC_START | + SNDRV_PCM_INFO_NO_PERIOD_IRQ, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE, .rates = SNDRV_PCM_RATE_32000 | @@ -91,7 +93,8 @@ static const struct snd_pcm_hardware oxygen_ac97_hardware = { SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_PAUSE | - SNDRV_PCM_INFO_SYNC_START, + SNDRV_PCM_INFO_SYNC_START | + SNDRV_PCM_INFO_NO_PERIOD_IRQ, .formats = SNDRV_PCM_FMTBIT_S16_LE, .rates = SNDRV_PCM_RATE_48000, .rate_min = 48000, @@ -530,7 +533,10 @@ static int oxygen_prepare(struct snd_pcm_substream *substream) oxygen_set_bits8(chip, OXYGEN_DMA_FLUSH, channel_mask); oxygen_clear_bits8(chip, OXYGEN_DMA_FLUSH, channel_mask); - chip->interrupt_mask |= channel_mask; + if (substream->runtime->no_period_irq) + chip->interrupt_mask &= ~channel_mask; + else + chip->interrupt_mask |= channel_mask; oxygen_write16(chip, OXYGEN_INTERRUPT_MASK, chip->interrupt_mask); spin_unlock_irq(&chip->reg_lock); return 0; -- 1.7.2.3

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[alsa-devel] [PATCH 1/2] pcm: pass hw_params flags to slave
by Pierre-Louis Bossart 08 Nov '10

08 Nov '10

fix required before interrupt disabling routines patch can be applied. Without this fix, the interrupts are only disabled when accessing directly hw devices. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)intel.com> --- src/pcm/pcm_params.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/src/pcm/pcm_params.c b/src/pcm/pcm_params.c index 3a90bcd..416b081 100644 --- a/src/pcm/pcm_params.c +++ b/src/pcm/pcm_params.c @@ -2126,6 +2126,7 @@ int _snd_pcm_hw_params_refine(snd_pcm_hw_params_t *params, err = changed; } params->info &= src->info; + params->flags = src->flags; /* propagate flags to slave */ return err; } -- 1.7.2.3

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[alsa-devel] [PATCH] ASoC: soc-cache: Use BUG_ON() for unsupported hw_read() calls
by Dimitris Papastamos 08 Nov '10

08 Nov '10

Instead of dereferencing a NULL function pointer and falling apart use BUG_ON() for any unimplemented hw_read() calls. Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- sound/soc/soc-cache.c | 6 ++++++ 1 files changed, 6 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 8785a0c..e747998 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -25,6 +25,7 @@ static unsigned int snd_soc_4_12_read(struct snd_soc_codec *codec, if (codec->cache_only) return -1; + BUG_ON(!codec->hw_read); return codec->hw_read(codec, reg); } @@ -99,6 +100,7 @@ static unsigned int snd_soc_7_9_read(struct snd_soc_codec *codec, if (codec->cache_only) return -1; + BUG_ON(!codec->hw_read); return codec->hw_read(codec, reg); } @@ -199,6 +201,7 @@ static unsigned int snd_soc_8_8_read(struct snd_soc_codec *codec, if (codec->cache_only) return -1; + BUG_ON(!codec->hw_read); return codec->hw_read(codec, reg); } @@ -270,6 +273,7 @@ static unsigned int snd_soc_8_16_read(struct snd_soc_codec *codec, if (codec->cache_only) return -1; + BUG_ON(!codec->hw_read); return codec->hw_read(codec, reg); } else { return cache[reg]; @@ -420,6 +424,7 @@ static unsigned int snd_soc_16_8_read(struct snd_soc_codec *codec, if (codec->cache_only) return -1; + BUG_ON(!codec->hw_read); return codec->hw_read(codec, reg); } @@ -531,6 +536,7 @@ static unsigned int snd_soc_16_16_read(struct snd_soc_codec *codec, if (codec->cache_only) return -1; + BUG_ON(!codec->hw_read); return codec->hw_read(codec, reg); } -- 1.7.3.2

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[alsa-devel] [PATCH] ASoC: s3c24xx: Fix compilation problem.
by Marek Belisko 08 Nov '10

08 Nov '10

When make mini2440_defconfig compilation end with undefined references to DMA functions. There was missing selection for S3C2410_DMA when compile ASoC audio for S3C24xx CPU. Tested on mini2440 board. Signed-off-by: Marek Belisko <marek.belisko(a)gmail.com> --- sound/soc/s3c24xx/Kconfig | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index 8a6b53c..d85bf8a 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -2,6 +2,7 @@ config SND_S3C24XX_SOC tristate "SoC Audio for the Samsung S3CXXXX chips" depends on ARCH_S3C2410 || ARCH_S3C64XX || ARCH_S5PC100 || ARCH_S5PV210 select S3C64XX_DMA if ARCH_S3C64XX + select S3C2410_DMA if ARCH_S3C2410 help Say Y or M if you want to add support for codecs attached to the S3C24XX AC97 or I2S interfaces. You will also need to -- 1.7.1

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[alsa-devel] [PATCH] ASoC: Remove unneeded use of address-of operator
by Dimitris Papastamos 08 Nov '10

08 Nov '10

There is no need to use '&' in this case. Either way, if a is an array of some type, then a == &a == &a[0]. Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8580.c | 2 +- sound/soc/codecs/wm8741.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index a2e0ed5..eda0e90 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -905,7 +905,7 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8580 = { .set_bias_level = wm8580_set_bias_level, .reg_cache_size = ARRAY_SIZE(wm8580_reg), .reg_word_size = sizeof(u16), - .reg_cache_default = &wm8580_reg, + .reg_cache_default = wm8580_reg, }; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index 90e31e9..9304075 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -455,7 +455,7 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8741 = { .resume = wm8741_resume, .reg_cache_size = ARRAY_SIZE(wm8741_reg_defaults), .reg_word_size = sizeof(u16), - .reg_cache_default = &wm8741_reg_defaults, + .reg_cache_default = wm8741_reg_defaults, }; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) -- 1.7.3.2

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[alsa-devel] SoundBlaster 16 PnP CT2950 Technical Specification
by Florian Scholz 08 Nov '10

08 Nov '10

Good evening. I need the technical specification for the sound card sb16 ct2950 for academic purpose. Thank you Florian Scholz

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[alsa-devel] ALSA processor usage is too high
by Adam Rosenberg 08 Nov '10

08 Nov '10

I am trying to decode 8 MP3 files simultaneously. Each file is 3 minutes 11 seconds long. When I just decode the files and copy the PCM data to memory it takes 2 minutes 36 seconds. When I do the same test after opening 8 ALSA PCM streams and while writing data to them it takes 4 minutes 37 seconds. This is much too slow for what should be a simple copy operation. I need to have 8 MP3 files decoding and playing on our custom board. I have taken the driver as far as I can on my own. I am seeking expert help to get this working as soon as possible. I am sorry if this was not the appropriate place to post such a request. Please contact me directly if you are interested in working on this project. The custom board is setup as follows: Analog Devices BF537 processor running uClinux from svn://blackfin.uclinux.org/uclinux-dist/trunk uclinux-dist Using alsa-lib-1.0.23 from http://www.alsa-project.org Two Cirrus Logic CS42448 CODECs connected to BF537 SPORT0 Primary and Secondary data lines in Multichannel mode CS42448 configured for TDM (32 bits per channel * 8 channels = 256 bits per frame) ICS661 Audio Clock at (256*48000Hz) connected to BF537 SPORT0 and Both CS42448 CODECs for bit clocking You can find the driver here: http://www.alcorn.com/ftp/swap/sound_cs42448.zip Thank you, Adam Adam Rosenberg Software Engineer Alcorn McBride Inc. 3300 South Hiawassee Building 105 Orlando, FL 32835 (407) 296 - 5800 ext. 5490

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[alsa-devel] Notebook ASUS X5DE-EX068V statt 699,00 bei uns nur 349,00
by Notebook2000 07 Nov '10

07 Nov '10

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[alsa-devel] [PATCH] ASoC: Lock the CODEC in PXA external jack controls
by Mark Brown 06 Nov '10

06 Nov '10

When doing anything with the system, especially DAPM, we need to hold the CODEC mutex. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/pxa/corgi.c | 5 +++++ sound/soc/pxa/magician.c | 4 ++++ sound/soc/pxa/poodle.c | 5 +++++ sound/soc/pxa/spitz.c | 5 +++++ sound/soc/pxa/tosa.c | 5 +++++ 5 files changed, 24 insertions(+), 0 deletions(-) diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index 97e9423..f451acd 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -100,8 +100,13 @@ static int corgi_startup(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_codec *codec = rtd->codec; + mutex_lock(&codec->mutex); + /* check the jack status at stream startup */ corgi_ext_control(codec); + + mutex_unlock(&codec->mutex); + return 0; } diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c index b8207ce..5ef0526 100644 --- a/sound/soc/pxa/magician.c +++ b/sound/soc/pxa/magician.c @@ -72,9 +72,13 @@ static int magician_startup(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_codec *codec = rtd->codec; + mutex_lock(&codec->mutex); + /* check the jack status at stream startup */ magician_ext_control(codec); + mutex_unlock(&codec->mutex); + return 0; } diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c index af84ee9..84edd03 100644 --- a/sound/soc/pxa/poodle.c +++ b/sound/soc/pxa/poodle.c @@ -77,8 +77,13 @@ static int poodle_startup(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_codec *codec = rtd->codec; + mutex_lock(&codec->mutex); + /* check the jack status at stream startup */ poodle_ext_control(codec); + + mutex_unlock(&codec->mutex); + return 0; } diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index f470f36..0b30d7d 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -108,8 +108,13 @@ static int spitz_startup(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_codec *codec = rtd->codec; + mutex_lock(&codec->mutex); + /* check the jack status at stream startup */ spitz_ext_control(codec); + + mutex_unlock(&codec->mutex); + return 0; } diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c index 73d0edd..7b983f9 100644 --- a/sound/soc/pxa/tosa.c +++ b/sound/soc/pxa/tosa.c @@ -81,8 +81,13 @@ static int tosa_startup(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_codec *codec = rtd->codec; + mutex_lock(&codec->mutex); + /* check the jack status at stream startup */ tosa_ext_control(codec); + + mutex_unlock(&codec->mutex); + return 0; } -- 1.7.1

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