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[alsa-devel] [PATCH 1/4] SOUND: atiixp, use uninitialized_var(chip)
by Jiri Slaby 26 Aug '09

26 Aug '09

To avoid wrong compiler warnings, use unitialized_var(chip) in snd_atiixp_probe's from atiixp and atiixp_modem drivers. 'chip' is unused when unset due to retval being nonzero. Signed-off-by: Jiri Slaby <jirislaby(a)gmail.com> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Jaroslav Kysela <perex(a)perex.cz> --- sound/pci/atiixp.c | 2 +- sound/pci/atiixp_modem.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c index d6752df..436ffea 100644 --- a/sound/pci/atiixp.c +++ b/sound/pci/atiixp.c @@ -1648,7 +1648,7 @@ static int __devinit snd_atiixp_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) { struct snd_card *card; - struct atiixp *chip; + struct atiixp *uninitialized_var(chip); int err; err = snd_card_create(index, id, THIS_MODULE, 0, &card); diff --git a/sound/pci/atiixp_modem.c b/sound/pci/atiixp_modem.c index e7e147b..b6191f2 100644 --- a/sound/pci/atiixp_modem.c +++ b/sound/pci/atiixp_modem.c @@ -1285,7 +1285,7 @@ static int __devinit snd_atiixp_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) { struct snd_card *card; - struct atiixp_modem *chip; + struct atiixp_modem *uninitialized_var(chip); int err; err = snd_card_create(index, id, THIS_MODULE, 0, &card); -- 1.6.3.3

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[alsa-devel] alsa random crash
by some guy 26 Aug '09

26 Aug '09

Problem: Alsa randomly crashes after ~10 hours of use; this varies from 2 to 20 hours. Symptoms: Sound stops while media is playing (in web browser, or mplayer), or convulses sporatically. The issue is not always reproducible (i.e I can reproduce the result when I want), but it is perisistant. Info: http://www.alsa-project.org/db/?f=fe434884ea7a16b1ae7c06cfd282832971f41fd6 lspci erhttp://pastebin.ca/1542518 kernel version is gentoo-2.6.28-r5 I really appreciate it. -anonymous alsa'er. _________________________________________________________________ More storage. Better anti-spam and antivirus protection. Hotmail makes it simple. http://go.microsoft.com/?linkid=9671357

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[alsa-devel] [PATCH] ASoC: SDP3430: Fix TWL GPIO6 pin mux request
by Candelaria Villareal, Jorge 25 Aug '09

25 Aug '09

Fix the write to PMBR1 register through I2C. Also, the constant which holds the value to write is now called TWL4030_GPIO6_PWM0_MUTE. This name is based on TRM to avoid confusion. Signed-off-by: Jorge Eduardo Candelaria <x0107209(a)ti.com> --- sound/soc/omap/sdp3430.c | 15 +++++++++++---- 1 files changed, 11 insertions(+), 4 deletions(-) diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c index f7e5b74..4a3f62d 100644 --- a/sound/soc/omap/sdp3430.c +++ b/sound/soc/omap/sdp3430.c @@ -40,8 +40,10 @@ #include "omap-pcm.h" #include "../codecs/twl4030.h" -#define TWL4030_INTBR_PMBR1 0x0D -#define EXTMUTE(value) (value << 2) +/* TWL4030 PMBR1 Register */ +#define TWL4030_INTBR_PMBR1 0x0D +/* TWL4030 PMBR1 Register GPIO6 mux bit */ +#define TWL4030_GPIO6_PWM0_MUTE(value) (value << 2) static struct snd_soc_card snd_soc_sdp3430; @@ -299,6 +301,7 @@ static struct platform_device *sdp3430_snd_device; static int __init sdp3430_soc_init(void) { int ret; + u8 pin_mux; if (!machine_is_omap_3430sdp()) { pr_debug("Not SDP3430!\n"); @@ -318,8 +321,12 @@ static int __init sdp3430_soc_init(void) *(unsigned int *)sdp3430_dai[1].cpu_dai->private_data = 2; /* McBSP3 */ /* Set TWL4030 GPIO6 as EXTMUTE signal */ - twl4030_i2c_write_u8(TWL4030_MODULE_INTBR, EXTMUTE(0x02), - TWL4030_MODULE_INTBR); + twl4030_i2c_read_u8(TWL4030_MODULE_INTBR, &pin_mux, + TWL4030_INTBR_PMBR1); + pin_mux &= ~TWL4030_GPIO6_PWM0_MUTE(0x03); + pin_mux |= TWL4030_GPIO6_PWM0_MUTE(0x02); + twl4030_i2c_write_u8(TWL4030_MODULE_INTBR, pin_mux, + TWL4030_INTBR_PMBR1); ret = platform_device_add(sdp3430_snd_device); if (ret) -- 1.6.0.4

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Re: [alsa-devel] Acer Aspire 7730G support
by Denis Kup 25 Aug '09

25 Aug '09

2009/8/25 Takashi Iwai <tiwai(a)suse.de> > BTW, did you check, as I suggested, whether acer-aspire-6530g with > changing the channel mode from 2ch to 6ch works for 7730g? > I guess we can merge acer-aspire-* variants to one or two models. > Can you describe in details what I need to do, please?

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Re: [alsa-devel] Acer Aspire 7730G support
by Takashi Iwai 25 Aug '09

25 Aug '09

At Mon, 24 Aug 2009 19:57:58 +0400, Denis Kup wrote: > > 2009/8/24 Takashi Iwai <tiwai(a)suse.de> > > > At Mon, 24 Aug 2009 17:10:10 +0400, > > Denis Kup wrote: > > > > > > Sorry, I was very busy last two months. We don't finish with this theme. > > > What about my patch? > > > > Please check with the latest sound GIT tree (or alsa-driver snapshot) > > whether it still work and repost the patch. There have been lots of > > changes in HD-audio driver code, so an old patch might not work any > > more. > > > > > > thanks, > > > > Takashi > > > I tested patch with alsa-driver-1.0.20.40.g13311.579.gcf341. All works as it > should. Try rather with the snapshot tarball in ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/alsa-driver-sna… The version you tried isn't the latest one. > It's diff between patched version( patch_realtek_new.c ) and unpatched: Make a patch with diff -u option (and better with -p, too). thanks, Takashi

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[alsa-devel] [PATCH] ASoC: SDP3430: Fix TWL GPIO6 pin mux request
by Candelaria Villareal, Jorge 25 Aug '09

25 Aug '09

Fix the write to PMBR1 register through I2C. Also, the constant which holds the value to write is now called TWL4030_GPIO6_PWM0_MUTE. This name is based on TRM to avoid confusion. Signed-off-by: Jorge Eduardo Candelaria <x0107209(a)ti.com> --- sound/soc/omap/sdp3430.c | 15 +++++++++++---- 1 files changed, 11 insertions(+), 4 deletions(-) diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c index f7e5b74..7ad99d2 100644 --- a/sound/soc/omap/sdp3430.c +++ b/sound/soc/omap/sdp3430.c @@ -40,8 +40,10 @@ #include "omap-pcm.h" #include "../codecs/twl4030.h" -#define TWL4030_INTBR_PMBR1 0x0D -#define EXTMUTE(value) (value << 2) +/* TWL4030 PMBR1 Register */ +#define TWL4030_INTBR_PMBR1 0x0D +/* TWL4030 PMBR1 Register GPIO6 mux bit */ +#define TWL4030_GPIO6_PWM0_MUTE(value) (value << 2) static struct snd_soc_card snd_soc_sdp3430; @@ -299,6 +301,7 @@ static struct platform_device *sdp3430_snd_device; static int __init sdp3430_soc_init(void) { int ret; + u8 pin_mux; if (!machine_is_omap_3430sdp()) { pr_debug("Not SDP3430!\n"); @@ -318,8 +321,12 @@ static int __init sdp3430_soc_init(void) *(unsigned int *)sdp3430_dai[1].cpu_dai->private_data = 2; /* McBSP3 */ /* Set TWL4030 GPIO6 as EXTMUTE signal */ - twl4030_i2c_write_u8(TWL4030_MODULE_INTBR, EXTMUTE(0x02), - TWL4030_MODULE_INTBR); + twl4030_i2c_read_u8(TWL4030_MODULE_INTBR, &pin_mux, + TWL4030_INTBR_PMBR1); + pin_mux &= ~TWL4030_GPIO6_PWM0_MUTE(0x3); + pin_mux |= TWL4030_GPIO6_PWM0_MUTE(0x2); + twl4030_i2c_write_u8(TWL4030_MODULE_INTBR, pin_mux, + TWL4030_INTBR_PMBR1); ret = platform_device_add(sdp3430_snd_device); if (ret) -- 1.6.0.4

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[alsa-devel] [PATCH] ASoC: s3c24xx platform: Fix s3c2410_dma_started called at wrong time
by Shine Liu 25 Aug '09

25 Aug '09

s3c24xx dma has the auto reload feature, when the the trnasfer is done, CURR_TC(DSTAT[19:0], current value of transfer count) reaches 0, and DMA ACK becomes 1, and then, TC(DCON[19:0]) will be loaded into CURR_TC. So the transmission is repeated. IRQ is issued while auto reload occurs. We change the DISRC and DCON[19:0] in the ISR, but at this time, the auto reload has been performed already. The first block is being re-transmitted by the DMA. So we need rewrite the DISRC and DCON[19:0] for the next block immediatly after the this block has been started to be transported. The function s3c2410_dma_started() is for this perpose, which is called in the form of "s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_STARTED);" in s3c24xx_pcm_trigger(). But it is not correct. DMA transmission won't start until DMA REQ signal arrived, it is the time s3c24xx_snd_txctrl(1) or s3c24xx_snd_rxctrl(1) is called in s3c24xx_i2s_trigger(). In the current framework, s3c24xx_pcm_trigger() is always called before s3c24xx_pcm_trigger(). So the s3c2410_dma_started() should be called in s3c24xx_pcm_trigger() after s3c24xx_snd_txctrl(1) or s3c24xx_snd_rxctrl(1) is called in this function. However, s3c2410_dma_started() is dma related, to call this function we should provide the channel number, which is given by substream->runtime->private_data->params->channel. The private_data points to a struct s3c24xx_runtime_data object, which is define in s3c24xx_pcm.c, so s3c2410_dma_started() can't be called in s3c24xx_i2s.c To move the struct s3c24xx_runtime_data defination to s3c24xx-pcm.h is one solution. If not moved, the patch shold be like: Signed-off-by: Shine Liu <shinel(a)foxmail.com> ---------------------------------------------------------------- --- a/sound/soc/s3c24xx/s3c24xx-pcm.h 2009-08-14 06:43:34.000000000 +0800 +++ b/sound/soc/s3c24xx/s3c24xx-pcm.h 2009-08-20 16:06:14.000000000 +0800 @@ -28,4 +28,6 @@ extern struct snd_soc_platform s3c24xx_soc_platform; extern struct snd_ac97_bus_ops s3c24xx_ac97_ops; +void s3c24xx_pcm_dma_started(struct snd_pcm_substream *substream); + #endif --- a/sound/soc/s3c24xx/s3c24xx-pcm.c 2009-08-14 06:43:34.000000000 +0800 +++ b/sound/soc/s3c24xx/s3c24xx-pcm.c 2009-08-20 15:56:46.000000000 +0800 @@ -66,6 +66,28 @@ struct s3c24xx_pcm_dma_params *params; }; +/* s3c24xx_pcm_dma_started + * + * To load the second dma buffer immediately after the + * first dma buffer has been started transferring, so that + * in the dma autoreload situation, the first data block + * won't be re-transferred after the first irq issued. + * + * This function is called in the s3c24xx-i2s-trigger() of + * s3c24xx-i2s.c, just after the dma transfer is really started. + * + * It's ugly here. Need a elegant way to fix this bug. + * FIXME + * + */ + +void s3c24xx_pcm_dma_started(struct snd_pcm_substream *substream) +{ + struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; + s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_STARTED); + +} + /* s3c24xx_pcm_enqueue * * place a dma buffer onto the queue for the dma system @@ -255,7 +277,6 @@ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: prtd->state |= ST_RUNNING; s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_START); - s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_STARTED); break; case SNDRV_PCM_TRIGGER_STOP: --- a/sound/soc/s3c24xx/s3c24xx-i2s.c 2009-08-14 06:43:34.000000000 +0800 +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c 2009-08-20 16:08:06.000000000 +0800 @@ -296,6 +296,9 @@ s3c24xx_snd_rxctrl(1); else s3c24xx_snd_txctrl(1); + + /* DMA transfer is really started, equeue the next buffer */ + s3c24xx_pcm_dma_started(substream); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND:

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[alsa-devel] ASoC updates for 2.6.32
by Mark Brown 25 Aug '09

25 Aug '09

The following changes since commit c6ea2af76ab79b82a0bfed313e4c2b68b10c2bc1: Takashi Iwai (1): ASoC: Remove unneeded inclusion of linux/regulator/consumer.h are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.32 Janusz Krzysztofik (1): ARM: OMAP: DMA: Add support for DMA channel self linking on OMAP1510 Jarkko Nikula (3): OMAP: McBSP: Use textual values in DMA operating mode sysfs files ASoC: OMAP: Fix setup of XCCR and RCCR registers in McBSP DAI ARM: OMAP: McBSP: Merge two functions into omap_mcbsp_start/_stop Mark Brown (4): ASoC: Add input and output AIF widgets ASoC: WM8993 digital mixing support ASoC: Select core DMA when building for S3C64xx Merge branch 'topic/digital-mixing' into for-2.6.32 arch/arm/plat-omap/dma.c | 10 +++ arch/arm/plat-omap/include/mach/mcbsp.h | 2 - arch/arm/plat-omap/mcbsp.c | 132 ++++++++++++------------------ include/sound/soc-dapm.h | 8 ++ sound/soc/codecs/wm8993.c | 88 ++++++++++++++++++++- sound/soc/omap/omap-mcbsp.c | 13 +--- sound/soc/s3c24xx/Kconfig | 1 + sound/soc/soc-dapm.c | 58 ++++++++++---- 8 files changed, 199 insertions(+), 113 deletions(-)

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[alsa-devel] [PATCH 0/3] Few patches to recent OMAP McBSP and ASoC changes
by Jarkko Nikula 25 Aug '09

25 Aug '09

Hi I would like to add these tree patches to recent set: [PATCHv5 00/20] OMAP ASoC changes in DMA utilization http://mailman.alsa-project.org/pipermail/alsa-devel/2009-August/020619.html These are actually independent patches but probably easier to handle as with the set above since some of the patches are touching both ASoC and OMAP subsystems. First patch is the most important since it changes the user space API for McBSP DMA operating mode selection. It would be more difficult to change it later if software starts to use current numeric value based operating mode selection in sysfs files. -- Jarkko

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[alsa-devel] multiple devices or one device w/ multiple subdevices?
by pl bossart 25 Aug '09

25 Aug '09

Hi, I have a piece of custom hardware that provides 7 independent PCM outputs. This is not intended for multichannel playback, the content on each output can be completely different; playback will not be synchronized and will start separately. The only restriction is that all outputs rely on the same baserate and share the same bitclock, i.e. all the outputs work at 48kHz. _From the mailling list archives, subdevices seem to be relevant when doing hw mixing or when lots of streams are handled. None of this applies to my case. Also it looks like PulseAudio knows nothing about subdevices in its detection loop, meaning the user would need to manually add some load-module commands in the PulseAudio configuration files. And last but not least, I didn't see very many lines of code (and no documentation) that showed how subdevices are registered/handled, etc. My conclusion is that multiple devices would seem like a much better option. But I wanted to check with the community that I am on the right track. Thanks for your feedback! - P

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