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[alsa-devel] [PATCH 1/6] ASoC: ad1980: Stay in 20bit mode for architectures other than blackfin
by Uwe Kleine-König 29 Aug '10

29 Aug '10

From: Sascha Hauer <s.hauer(a)pengutronix.de> Signed-off-by: Sascha Hauer <s.hauer(a)pengutronix.de> Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> --- sound/soc/codecs/ad1980.c | 10 +++++++++- 1 files changed, 9 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c index 70cfaec..35c0580 100644 --- a/sound/soc/codecs/ad1980.c +++ b/sound/soc/codecs/ad1980.c @@ -168,13 +168,21 @@ retry: } soc_ac97_ops.reset(codec->ac97); + +#ifdef SND_BF5XX_SOC_AD1980 /* Set bit 16slot in register 74h, then every slot will has only 16 * bits. This command is sent out in 20bit mode, in which case the - * first nibble of data is eaten by the addr. (Tag is always 16 bit)*/ + * first nibble of data is eaten by the addr. (Tag is always 16 bit). + * + * This seems to be blackfin specific. On other systems we want 20bit + * mode. + */ ac97_write(codec, AC97_AD_SERIAL_CFG, 0x9900); +#endif if (ac97_read(codec, AC97_RESET) != 0x0090) goto err; + return 0; err: -- 1.7.1

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[alsa-devel] [patch] ALSA: pcm: add more format names
by Dan Carpenter 28 Aug '10

28 Aug '10

There were some new formats added in commit 15c0cee6c809 "ALSA: pcm: Define G723 3-bit and 5-bit formats". That commit increased SNDRV_PCM_FORMAT_LAST as well. My concern is that there are a couple places which do: for (i = 0; i < SNDRV_PCM_FORMAT_LAST; i++) { if (dummy->pcm_hw.formats & (1ULL << i)) snd_iprintf(buffer, " %s", snd_pcm_format_name(i)); } I haven't tested these but it looks like if "i" were equal to SNDRV_PCM_FORMAT_G723_24 or higher then we might read past the end of the array. Signed-off-by: Dan Carpenter <error27(a)gmail.com> --- Resending because I screwed up the CC lis on my first mail. diff --git a/sound/core/pcm.c b/sound/core/pcm.c index cbe815d..204af48 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -203,10 +203,16 @@ static char *snd_pcm_format_names[] = { FORMAT(S18_3BE), FORMAT(U18_3LE), FORMAT(U18_3BE), + FORMAT(G723_24), + FORMAT(G723_24_1B), + FORMAT(G723_40), + FORMAT(G723_40_1B), }; const char *snd_pcm_format_name(snd_pcm_format_t format) { + if (format >= ARRAY_SIZE(snd_pcm_format_names)) + return "Unknown"; return snd_pcm_format_names[format]; } EXPORT_SYMBOL_GPL(snd_pcm_format_name);

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[alsa-devel] [PATCH] sound: oss: fix uninitialized spinlock
by Akinobu Mita 28 Aug '10

28 Aug '10

The spinlock lock in sound_timer.c is used without initialization. Signed-off-by: Akinobu Mita <akinobu.mita(a)gmail.com> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Jaroslav Kysela <perex(a)perex.cz> --- sound/oss/sound_timer.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/oss/sound_timer.c b/sound/oss/sound_timer.c index f0f0c19..48cda6c 100644 --- a/sound/oss/sound_timer.c +++ b/sound/oss/sound_timer.c @@ -26,7 +26,7 @@ static unsigned long prev_event_time; static volatile unsigned long usecs_per_tmr; /* Length of the current interval */ static struct sound_lowlev_timer *tmr; -static spinlock_t lock; +static DEFINE_SPINLOCK(lock); static unsigned long tmr2ticks(int tmr_value) { -- 1.6.0.6

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[alsa-devel] [PATCH] Return hw error directly from oustream_write.
by linux＠audioscience.com 28 Aug '10

28 Aug '10

From: Eliot Blennerhassett <eblennerhassett(a)audioscience.com> If hw error is ignored, status is updated with invalid info. Signed-off-by: Eliot Blennerhassett <eblennerhassett(a)audioscience.com> --- sound/pci/asihpi/hpi6205.c | 7 +++++-- 1 files changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/pci/asihpi/hpi6205.c b/sound/pci/asihpi/hpi6205.c index 3b44134..22c5fc6 100644 --- a/sound/pci/asihpi/hpi6205.c +++ b/sound/pci/asihpi/hpi6205.c @@ -941,8 +941,7 @@ static void outstream_host_buffer_free(struct hpi_adapter_obj *pao, } -static u32 outstream_get_space_available(struct hpi_hostbuffer_status - *status) +static u32 outstream_get_space_available(struct hpi_hostbuffer_status *status) { return status->size_in_bytes - (status->host_index - status->dSP_index); @@ -987,6 +986,10 @@ static void outstream_write(struct hpi_adapter_obj *pao, /* write it */ phm->function = HPI_OSTREAM_WRITE; hw_message(pao, phm, phr); + + if (phr->error) + return; + /* update status information that the DSP would typically * update (and will update next time the DSP * buffer update task reads data from the host BBM buffer) -- 1.7.0.4

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[alsa-devel] ASoC update for 2.6.36
by Mark Brown 28 Aug '10

28 Aug '10

The following changes since commit 70bf043b137aa9ff2711b16532774465e07a8f47: ASoC: i.MX ssi: use SSI_STCCR in synchronous mode (2010-08-23 13:33:05 +0100) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.36 Axel Lin (1): ASoC: soc-core: fix debugfs_pop_time file permissions You can't get much smaller and less invasive than a one character fix! sound/soc/soc-core.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-)

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[alsa-devel] [PATCH 0/4] ASoC: tlv320aic3x: Output stage cleanup
by Jarkko Nikula 28 Aug '10

28 Aug '10

Hi I have had always difficulties to remember which aic3x mixer control controls what route but haven't bothered to look any deeper until very recently after wondering why the swapped routes aren't working. It turned out that there were two reasons: 1. Output mixers were misdefined to mix output pins to internal signals where DAPM logic and hw as well does opposite 2. Swapped routes are not working due this misdefinition Then there wasn't uniform way to represent and name volume controls for those output mixer stages. This patchset does first some minor cleanups, then fixes the mixer stage also adding a few missing routes and then puts hopefully some logic to output controls. Unfortunately this means control name changes but hopefully driver is now more understandable and complete. -- Jarkko

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Re: [alsa-devel] [PATCH] ASoC: Complete supported clock ratios and rate constraints for wm8741.
by Mark Brown 28 Aug '10

28 Aug '10

On Fri, Aug 27, 2010 at 03:26:27PM +0100, Ian Lartey wrote: > Signed-off-by: Ian Lartey <ian(a)opensource.wolfsonmicro.com> Applied, thanks - but you probably want to remove alsa-devel-request from your address book.

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Re: [alsa-devel] explicit commandline control of speakers vs headphone output
by Eric Holstege 28 Aug '10

28 Aug '10

Thanks, Clemens; How were you able to determine that from the "amixer scontents" output. Although I can't disable *automatic* speaker mute on headphone insertion, can I explicitly unmute them again (e.g. with amixer sset or cset)? If not.... "cat /proc/asound/card*/codec*" says it is the Realtek ALC888. Does this mean I have to somehow patch the kernel sound module file .../linux-source-2.6.*/sound/pci/hda/patch_realtek.c to fix this somehow? Regards -Eric Clemens Ladisch wrote: > Eric Holstege wrote: > >> Simple mixer control ... >> > > It looks as if your codec cannot be configured to disable the automatic > headphone switching. > > > Regards, > Clemens > > >

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[alsa-devel] Problem with underrun on new CS42448 x2 driver
by Adam Rosenberg 28 Aug '10

28 Aug '10

I am trying to write a driver for the Blackfin BF537 processor on a custom board that has two CS42448 CODECs connected to SPORT0. I am using TDM and there are 8 channels per frame on each data line (primary and secondary). My DMA buffer must interleave the data for the two data lines (for a total of 16 channels of data per frame). I have completed all of this and the driver will sometimes work properly but I am having underrun problems. I would like to use snd_async_add_pcm_handler as it is used in the /test/pcm.c example but it seems as though an underrun occurs at regular intervals causing the PCM prepare function to be called again (this results in a tone that sounds good but only for a short time before it stops and then starts again). There is plenty of time for the data to be transferred. I feel as though the driver is not calling snd_pcm_period_elapsed fast enough. At the moment, I call snd_pcm_period_elapsed from within a callback function that occurs every time a DMA period elapse occurs. I have been working on this for quite a while now. Is there someone who can look at the driver code for me or provide detailed information on when ALSA expects to have snd_pcm_period_elapsed get called to avoid an underrun? The other thing that is strange is that I have to explicitly set the period size for aplay to work. For example: aplay --period-size=1024 test.wav which gives me a period time of 21333 but when I use this period time with the /test/pcm.c sample that is provided the tone still does not sound right (although it does seem to improve) I am about ready to bypass ALSA entirely but I hate the idea of breaking from standards (not to mention having to rewrite an entire audio interface). Any help would be greatly appreciated! You can check out the driver here: http://www.alcorn.com/ftp/swap/ami_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] xine AC3 dropout problem - Please help!
by VDR User 27 Aug '10

27 Aug '10

Hi. Xine seems to have a problem with ac3 audio dropping out frequently, which ruins the ability to watch hdtv for example. I have made a sample which exhibits this problem, which plays just fine in mplayer so I assume that means root causes such as a buggy system scheduler can be ruled out. The sample also plays fine on my main Windows desktop as well. Adjusting things such as engine.buffers.audio_num_buffers in xine's config has no affect. It has been suggested that the problem may be xine not properly handling a stream error. If anyone can and is willing to take a look at this and hopefully resolve the problem once and for all, that would be great. The sample is 20MB so it can't be attached to this post. Until I can find a better place to host the sample, I've temporarily uploaded it to: http://rapidshare.com/files/415479559/xine_ac3_dropout_bug.tar.bz2 System details are: debian testing stable kernel 2.6.35.3 (quiet notsc clocksource=hpet) gcc 4.4.4-8 alsa-base 1.0.23+dfsg-1 alsa-utils 1.0.23-2 xine-lib-1.2 hg r11577 xine-ui hg r3054 VDR-1.7.15 mplayer-nogui 2:1.0~rc3++svn20100804-0.0 I recently tried the notsc and clocksource=hpet kernel options, which also had no affect. I want to reiterate that the dropout problem only occurs in xine. The error I see in the xine log when the dropout occurs is: pcm_hw.c: snd_pcm_hw_delay() SNDRV_PCM_IOCTL_DELAY failed (-32). Best regards, Derek

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