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

[alsa-devel] n810 diablo and alsa ?
by John Simpson 13 Aug '10

13 Aug '10

Is there a how-to for installing the alsa-lib correctly on the n810? (hopefully from a fresh install of Maemo/Diablo?) I've been hitting brick walls in every direction. :-( The alsa-lib in the diablo repository seems to be outdated: [1|user@tablet|~/src]cat alsa-lib_1.0.14sa-2maemo2.dsc Format: 1.0 Source: alsa-lib Version: 1.0.14sa-2maemo2 Binary: lib32asound2-dev, lib64asound2, lib32asound2, libasound2-dev, libasound2, libasound2-doc, libasound2-dbg, lib64asound2-dev Maintainer: Debian ALSA Maintainers <pkg-alsa-devel(a)lists.alioth.debian.org> Architecture: any Standards-Version: 3.7.2 Build-Depends: debhelper (>= 4.2.32), quilt (>= 0.40), autotools-dev, doxygen, libc6-dev-powerpc [ppc64], libc6-dev-i386 [amd64], libc6-dev-ppc64 [powerpc], libc6-dev-s390x [s390], libc6-dev-sparc64 [sparc], lib32gcc1 [amd64 ppc64], libc6-dev (>= 2.5) [armel] Vcs-Browser: http://svn.debian.org/wsvn/pkg-alsa/trunk/alsa-lib/ Uploaders: Jordi Mallach <jordi(a)debian.org>, Mikael Magnusson <mikma(a)users.sourceforge.net>, Elimar Riesebieter <riesebie(a)lxtec.de>, Marc-Andre Lureau <marc-andre.lureau(a)nokia.com> Vcs-Svn: svn://svn.debian.org/pkg-alsa/trunk/alsa-lib

2 2

Re: [alsa-devel] intel 8x0 ALSA driver in virtualbox - choppy sound
by Lex F 13 Aug '10

13 Aug '10

Hi Thanks for your reply. I tried to answer your questions below. On 13 August 2010 17:06, Takashi Iwai <tiwai(a)suse.de> wrote: > We have had many problems with VMware, too :) > Which sound backend are you using? Direct ALSA / dmix or PA? I believe it's direct ALSA, I have choppy output even with aplay /usr/share/sounds/alsa/*.wav > Actually, there is no real interrupt, but just the virtual one, > thus it can be never accurate. So, rather the question is how the > driver is tolerant for sloppy IRQ updates. > > Usually, when the period size is large and the buffer have many > periods (not two), the delayed update works more stably. I am afraid I don't understand this :) I am compiling a kernel with the following options enabled: CONFIG_SND_PCM_XRUN_DEBUG, CONFIG_SND_VERBOSE_PROCFS, CONFIG_SND_DEBUG Would any of these help to determine the cause and suggest a fix? Lex

2 2

[alsa-devel] [PATCH 3/5] ASoC: update setting for pxa ssp slave mode
by Haojian Zhuang 13 Aug '10

13 Aug '10

2 1

[alsa-devel] ASoC updates for 2.6.36
by Mark Brown 13 Aug '10

13 Aug '10

The following changes since commit 2e2211a387e70c62e1249822a94f4a33933ecd62: ASoC: add AD1980 obsolete information (2010-08-10 15:43:45 +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 A couple of focused patches for the WM8994, plus the MAINTAINERS additions. They're not 100% ASoC but mostly so by volume of code covered so hopefully not a problem for you to take via your tree - let me know if it's an issue and I'll drop them. Mark Brown (4): ASoC: Optimise DSP performance for WM8994 ASoC: Only tweak WM8994 chip configuration on devices up to rev D MAINTAINERS: Make Wolfson entry also cover CODEC drivers MAINTAINERS: Add Ian Lartey as comaintaner for Wolfson devices MAINTAINERS | 7 ++++--- sound/soc/codecs/wm8994.c | 23 +++++++++++++++-------- 2 files changed, 19 insertions(+), 11 deletions(-)

2 1

[alsa-devel] [PATCH 5/5] ASoc: add machine driver for 88pm860x
by Haojian Zhuang 13 Aug '10

13 Aug '10

2 3

[alsa-devel] [PATCH 4/5] ASoC: add 88pm860x codec driver
by Haojian Zhuang 13 Aug '10

13 Aug '10

2 3

[alsa-devel] alsa-lib: Rate conversion broken for non-S16?
by Mark Hills 13 Aug '10

13 Aug '10

I have a device (snd-usb-caiaq) whose native sample format is S24_3BE. When sample rate conversion is used, the audio becomes near silent, distorted, and only in the left channel. It is reproducable with aplay (see example below). Recording is not affected. Playback is correct if only format conversion is used. It looks something is causing the incorrect put function to be used by the rate conversion. I traced it to linear_expand() and others, alsa-lib-src/pcm_rate_linear.c. Forcing the function to put S16 format into the destination buffer allows, suprisingly, the sound to be heard correctly and at full volume from both aplay and xwax: if (rate->put_idx == 10) rate->put_idx = 6 I suspected that linear_init() is looking in the wrong place to choose the attributes for the rate conversion. After some time spent I cannot see it. But is there any reason that the output buffer for the rate conversion would not be in S24_3BE? And for the rate conversion to be misinformed? $ uname -s -r -v -p -i -m Linux 2.6.34.1-mh #83 SMP PREEMPT Wed Jul 21 23:25:33 BST 2010 i686 pentium4 i386 $ alsacap Card 0, ID `Audio8DJ', name `Audio 8 DJ' Device 0, ID `Audio 8 DJ', name `Audio 8 DJ', 4 subdevices (4 available) 2 channels, sampling rate 44100..96000 Hz Sample formats: S24_3BE Subdevice 0, name `subdevice #0' Subdevice 1, name `subdevice #1' Subdevice 2, name `subdevice #2' Subdevice 3, name `subdevice #3' $ aplay -v -D plughw:Audio8DJ,0,0 halcyon-24000.wav Playing WAVE 'halcyon-24000.wav' : Signed 16 bit Little Endian, Rate 24000 Hz, Stereo Plug PCM: Rate conversion PCM (48000, sformat=S24_3BE) Converter: linear-interpolation Protocol version: 10002 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 24000 exact rate : 24000 (24000/1) msbits : 16 buffer_size : 10922 period_size : 2730 period_time : 113770 tstamp_mode : NONE period_step : 1 avail_min : 2730 period_event : 0 start_threshold : 10922 stop_threshold : 10922 silence_threshold: 0 silence_size : 0 boundary : 715784192 Slave: Hardware PCM card 0 'Audio 8 DJ' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S24_3BE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 24 buffer_size : 21845 period_size : 5461 period_time : 113770 tstamp_mode : NONE period_step : 1 avail_min : 5461 period_event : 0 start_threshold : 21844 stop_threshold : 21845 silence_threshold: 0 silence_size : 0 boundary : 1431633920 appl_ptr : 0 hw_ptr : 0 -- Mark

2 6

[alsa-devel] [PATCH 0/5] ASoc: support 88pm860x codec driver
by Haojian Zhuang 13 Aug '10

13 Aug '10

Add 88pm860x codec driver. This series patch is based on ASoC for-2.6.37 branch. Thanks Haojian

2 1

[alsa-devel] [PATCH] ISA: New Aztech Sound Galaxy driver
by René Herman 13 Aug '10

13 Aug '10

Hi Takashi. I've dusted of the snd-galaxy Aztech Sound Galaxy driver that I posted back in 2007. Have not been around for a long time but found some time again currently. Earlier this year I was contacted about the driver so there's still at least some demand for it. It is the same version as posted way back then: http://www.spinics.net/lists/alsa-devel/msg09791.html only brought up to date to 2.6.35, and with lib.c folded back in to fix the remaining problem from the earlier post. It's also been tested again on all previously tested cards and two new ones. Everything still found working. More information in the changelog accompanying the patch below. Cleaning up and moving AZT2320 to the galaxy directory as well would be a next step, as would removing the old sgalaxy driver. The patch has been generated against 2.6.35 and I've just now checked that it applies cleanly to current git. Hope that'll do since I'm not really setup to work against alsa-driver or anything. When updating the driver I noticed fairly little seems to have happened inside sound/isa anyway over the last few years, so I assumed it would be okay. Hope it can be applied as is, but I have some time to deal with problems currently. @ Ram: Do you even still have the card? ;-) Regards, Rene

2 4

[alsa-devel] [PATCH 1/4] ASoC: Optimise DSP performance for WM8994
by Mark Brown 13 Aug '10

13 Aug '10

Change the chip defaults to optimise performance of some of the DSP functionality. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 5 +++-- 1 files changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index a87046a..2dc9daa 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3082,10 +3082,11 @@ static int wm8994_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_STANDBY: if (codec->bias_level == SND_SOC_BIAS_OFF) { - /* Tweak DC servo configuration for improved - * performance. */ + /* Tweak DC servo and DSP configuration for + * improved performance. */ snd_soc_write(codec, 0x102, 0x3); snd_soc_write(codec, 0x56, 0x3); + snd_soc_write(codec, 0x817, 0); snd_soc_write(codec, 0x102, 0); /* Discharge LINEOUT1 & 2 */ -- 1.7.1

2 4