December 2008 - Alsa-devel - mailman.alsa-project.org

Re: [alsa-devel] [PATCH 01/01] alsa: sound/soc/pxa/pxa2xx_pcm.c, check and set invalid dma_ch to be <= to 0 in pxa2xx_pcm_hw_free()
by Mark Brown 20 Dec '08

20 Dec '08

On Fri, Dec 19, 2008 at 09:52:22PM -0800, sware wrote: > From: Stephen Ware <stephen.ware(a)eqware.net>, Dennis O'Brien <dennis.obrien(a)eqware.net> ... > Signed-off-by: Dennis O'Brien <dennis.obrien(a)eqware.net> The patch itself is good but could you please clarify what's going on with the author and signoff lines (apart from anything else the multiple authors appears to be confusing git, it's ignoring Dennis). The combination of git format-patch and send-email is handy for generating something that applies easily.

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[alsa-devel] [PATCH 1/4] ALSA: ASoC: tlv320aic3x add dsp_a
by Troy Kisky 20 Dec '08

20 Dec '08

Add SND_SOC_DAIFMT_DSP_A mode option. Signed-off-by: Troy Kisky <troy.kisky(a)boundarydevices.com> diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index b76bcc3..b9b67ed 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -847,6 +847,7 @@ static int aic3x_set_dai_fmt(struct snd_soc_dai *codec_dai, struct snd_soc_codec *codec = codec_dai->codec; struct aic3x_priv *aic3x = codec->private_data; u8 iface_areg, iface_breg; + int delay = 0; iface_areg = aic3x_read_reg_cache(codec, AIC3X_ASD_INTF_CTRLA) & 0x3f; iface_breg = aic3x_read_reg_cache(codec, AIC3X_ASD_INTF_CTRLB) & 0x3f; @@ -872,6 +873,8 @@ static int aic3x_set_dai_fmt(struct snd_soc_dai *codec_dai, SND_SOC_DAIFMT_INV_MASK)) { case (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF): break; + case (SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_NF): + delay = 1; case (SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF): iface_breg |= (0x01 << 6); break; @@ -888,6 +891,7 @@ static int aic3x_set_dai_fmt(struct snd_soc_dai *codec_dai, /* set iface */ aic3x_write(codec, AIC3X_ASD_INTF_CTRLA, iface_areg); aic3x_write(codec, AIC3X_ASD_INTF_CTRLB, iface_breg); + aic3x_write(codec, AIC3X_ASD_INTF_CTRLC, delay); return 0; } diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h index 00a195a..40a6c2a 100644 --- a/sound/soc/codecs/tlv320aic3x.h +++ b/sound/soc/codecs/tlv320aic3x.h @@ -35,6 +35,8 @@ #define AIC3X_ASD_INTF_CTRLA 8 /* Audio serial data interface control register B */ #define AIC3X_ASD_INTF_CTRLB 9 +/* Audio serial data interface control register C */ +#define AIC3X_ASD_INTF_CTRLC 10 /* Audio overflow status and PLL R value programming register */ #define AIC3X_OVRF_STATUS_AND_PLLR_REG 11 /* Audio codec digital filter control register */ -- 1.5.4.3

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[alsa-devel] "No codecs found" on Intel HDA + AD1988A
by Alexander Chemeris 19 Dec '08

19 Dec '08

Hi, I have Asus P5B-V motherboard with integrated Intel HDA audio and AD1988A codec (I see the chip on the board). About half a year ago or so it used to work, but now when snd-hda-intel module is loaded it claims it can't find a codec: $ dmesg | grep -i hda [ 13.648205] HDA Intel 0000:00:1b.0: PCI INT A -> GSI 22 (level, low) -> IRQ 22 [ 13.648236] HDA Intel 0000:00:1b.0: setting latency timer to 64 [ 13.680537] hda-intel: no codecs found! [ 13.680589] HDA Intel 0000:00:1b.0: PCI INT A disabled I tried different recent versions of ALSA to no avail. So far I've tried: * 1.0.17 with Ubuntu patches * 1.0.18 with Ubuntu patches * 1.0.18a downloaded half an hour ago from here: ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/alsa-driver-sna… I also tried to compile 1.0.16, but seems it is not compatible with the kernel: $ uname -a Linux nidus 2.6.27-10-generic #1 SMP Fri Nov 21 19:19:18 UTC 2008 x86_64 GNU/Linux When I tried 1.0.18a, I also manually loaded snd-hda-codec_analog before loading snd-hda-intel without any success. And yeah, when I just run `modprobe snd-hda-intel` only following modules are loaded: $ lsmod | grep snd snd_hda_intel 37864 0 snd_hda_codec 86784 1 snd_hda_intel snd_hwdep 17160 1 snd_hda_codec snd_pcm_oss 52256 0 snd_mixer_oss 25088 1 snd_pcm_oss snd_pcm 98824 3 snd_hda_intel,snd_hda_codec,snd_pcm_oss snd_seq_oss 43776 0 snd_seq_midi_event 16768 1 snd_seq_oss snd_seq 67744 4 snd_seq_oss,snd_seq_midi_event snd_timer 33424 2 snd_pcm,snd_seq snd_seq_device 16788 2 snd_seq_oss,snd_seq snd 83144 10 snd_hda_intel,snd_hda_codec,snd_hwdep,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_seq_oss,snd_seq,snd_timer,snd_seq_device soundcore 16800 1 snd snd_page_alloc 18576 2 snd_hda_intel,snd_pcm What can I do to help get my audio working again? I don't want to install different kernel or restart computer (well, as a last resort, may be...), but I easily can test different any patches, etc. PS I don't think it matters, but I use Ubuntu 8.10 x64. -- Regards, Alexander Chemeris. SIPez LLC. SIP VoIP, IM and Presence Consulting http://www.SIPez.com tel: +1 (617) 273-4000

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Re: [alsa-devel] No HDMI audio on Asus P5Q-EM
by Chris Pimlott 19 Dec '08

19 Dec '08

Forgot to join the list first, also here is the bugtrack issue: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4281 On Thu, Dec 11, 2008 at 3:41 PM, Chris Pimlott <pimlottc(a)gmail.com> wrote: > I have an Asus P5Q-EM motherboard, with Intel G45 chipset and Realtek ALC > 1200 integrated audio, with HDMI output. (Specifications: > http://www.asus.com/products.aspx?modelmenu=2&model=2413&l1=3&l2=11&l3=761&…)<http://www.asus.com/products.aspx?modelmenu=2&model=2413&l1=3&l2=11&l3=761&…>[ > ^<http://www.asus.com/products.aspx?modelmenu=2&model=2413&l1=3&l2=11&l3=761&…> > ] > > The analog output has always worked fine (from the default Mythbuntu > install), but I do not get any output when trying to use the HDMI. I am > testing with aplay: "aplay -D plughw:0,3 > /usr/share/sounds/alsa/Front_Center.wav". plughw:0,1 with the same command > works (on analog out). > > I see three devices, "IEC958", "IEC958 Default PCM" and "IEC958 1", which > are all unmuted. I have tried flipped the copyright setting with "iecset -n > 1 copy on" but that has made no difference. > > I have tried a number of versions, including the stock 2.6.27 kernel > package shipped in Mythbuntu, alsa-1.0.18 and alsa-1.0.18a, as well as > version of the sound-2.6 from 2008-11-23 and today. The most recent I'm > trying is sound-2.6 version is b13a9d80db0f1f4ace2863e8824a3d3299730d0a. > > I've notice there's been a fair amount of updates to this area of the > snd_hda_intel driver, and some changes to how the device has shown up in > recent releases. Should I even expect this to work, or is it still > "in-progress"? Thanks. > > Chris Pimlott >

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[alsa-devel] arecord ..xrun!!!!
by vasudha rao 19 Dec '08

19 Dec '08

Hi .. I am trying to record .wav files.. I did like, $arecord -D copy test.wav Recording WAVE 'test.wav' : Unsigned 8 bit, Rate 8000 Hz, Mono xrun!!! (at least 0.251 ms long) xrun!!! (at least 0.071 ms long) xrun!!! (at least 0.072 ms long) . . . When I run this test.wav, no sound!!!! Am I doing wrong anywhere?? Thanks:-)

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[alsa-devel] arecord working now..
by vasudha rao 19 Dec '08

19 Dec '08

Hi.. I am able to record now.. I used arecord like, $arecord -d 10 -f cd -t wav -D copy foobar.wav and I play like, $aplay -D hw:0,0 foobar.wav All works fine..:-) But the thing is I dont know why it was not working and why it is working now..?? Please tel me if u found it why??and if time permits for u.. Thank you ..

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Re: [alsa-devel] Patch to /sound/soc/pxa/pxa2xx_pcm.c, WHO IS INTERESTED?
by Mark Brown 19 Dec '08

19 Dec '08

On Thu, Dec 18, 2008 at 10:34:53AM -0800, sware wrote: > Patch is to 2.6.27.9 (yesterday), notice .10 is up today, > patch should still apply. Thanks for your patch. Unfortunately all this code has been moved about a lot in the latest code in order to share the code with the non-ASoC AC97 driver so your patch won't apply - see the topic/asoc branch of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git for the latest code. Could you rebase your work against that, please? Also, please include a Signed-off-by line for patch submissions - for details see Documentation/SubmittingPatches > diff -uprN -X linux-2.6.27.9.orig/Documentation/dontdiff linux-2.6.27.9.orig/sound/soc/pxa/pxa2xx-pcm.c linux-2.6.27.9.snd_modified/sound/soc/pxa/pxa2xx-pcm.c > --- linux-2.6.27.9.orig/sound/soc/pxa/pxa2xx-pcm.c 2008-12-13 15:56:16.000000000 -0800 > +++ linux-2.6.27.9.snd_modified/sound/soc/pxa/pxa2xx-pcm.c 2008-12-17 14:09:02.000000000 -0800 > @@ -137,13 +137,18 @@ static int pxa2xx_pcm_hw_free(struct snd > { > struct pxa2xx_runtime_data *prtd = substream->runtime->private_data; > > - if (prtd && prtd->params) > + if (!prtd) > + return 0; > + if (!prtd->params) > + return 0; > + prtd should always be there - the code should at least complain loudly if it's NULL. params might potentially be hit, though. > *prtd->params->drcmr = 0; > > - if (prtd->dma_ch) { > + if (prtd->dma_ch >= 0) { > snd_pcm_set_runtime_buffer(substream, NULL); > pxa_free_dma(prtd->dma_ch); > - prtd->dma_ch = 0; > + prtd->dma_ch = -1; /* invalid channel */ > + prtd->params = NULL; > } This bit is good. > @@ -266,6 +271,8 @@ static int pxa2xx_pcm_close(struct snd_p > dma_free_writecombine(substream->pcm->card->dev, PAGE_SIZE, > prtd->dma_desc_array, prtd->dma_desc_array_phys); > kfree(prtd); > + runtime->private_data = NULL; > + > return 0; > } As is this.

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[alsa-devel] Streams
by James Courtier-Dutton 19 Dec '08

19 Dec '08

Hi, There was some talk some time ago of maybe moving alsa to a streams approach instead of ring buffer and periods approach. Has anything become of this? Is there a plan to improve the abstraction between hardware and user space regarding constraints on period sizes when doing sample rate conversion? Kind Regards James

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[alsa-devel] 44.1khz capture on CA0106?
by Takashi Iwai 19 Dec '08

19 Dec '08

Hi, can anyone confirm that 44.1kHz capture on CA0106 causes the problem with 48kHz playback? Typically, recording via 44.kHz using hw device, then run 48kHz playback. % arecord -fcd -Dhw:0 foo.wav % aplay -Dhw:0 48khz-samples.wav I got a bug report about this, and disabling 44.1kHz capture fixes the problem. His case is with MSI K8N mobo, and I'd like to know whether this happens in general on CA0106. The bug report is found in https://bugzilla.novell.com/show_bug.cgi?id=447624 thanks, Takashi

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[alsa-devel] ca0106 SPDIF defaults
by Ben Stanley 19 Dec '08

19 Dec '08

Hi, I previously reported problems that are still plaguing me [1] to do with SPDIF settings that don't reset properly. I am now looking at doing a patch to the ca0106 driver to fix this problem. It appears I have to add a new control to be able to set the default settings. I presume this introduces complications with existing settings in asound.state? Is there some collected wisdom on how to add new controls while providing a migration/upgrade/interoperability path? Ben Stanley. [1] http://article.gmane.org/gmane.linux.alsa.devel/55406/match=independent+pcm…

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