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

[alsa-devel] How to transfer more than 2 channels between i.MX25 & AD1938 using SSI?
by Sven Zeisberg 30 Aug '10

30 Aug '10

Hi, has anybody succeeded to transfer more than 2 audio channels between i.MX25 and a CODEC throught the i.MX25 SSI Port? We have succeeded to set the CODEC (AD1938) as Master to the required mode (TDM with 8 Timeslots and 32 Bit per Timeslot). However, as soon as we start to play a wav file with aplay, we can see on the logic analyzer that the data format on the SSI lines is reset to a stereo mode (2 channels with frame sync changing level between channels). In the imx-ssi driver, I can see that .channel_max = 2 - therefore my question: does this driver support more than two channels at all? BR Sven

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[alsa-devel] [PATCH v4 0/3] ASoC: Add iPAQ RX1950 support
by Vasily Khoruzhick 29 Aug '10

29 Aug '10

These three patches are necessary to add sound support for iPAQ RX1950 PDA, appropriate mach-rx1950.c patch is missing, as it depends on battery and LEDs patches which are still not merged nor reviewed. First patch improves situation with power consumption for uda1380 and also fixes suspend/resume. Second patch just adds glue driver. Third patch fixes s3c_device_iis name and exports s3c_device_pcm for all s3c24xx-based devices.

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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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