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

[alsa-devel] [RFC] virtual master control for HD-audio
by Takashi Iwai 06 May '08

06 May '08

Hi, another thing I'd like to push into the next kernel is the virtual master volume control. As I posted in earlier posts, it adds virtual controls that have several slave controls with the same types, e.g. Front, Surround, Center, LFE, etc. Then these are adjusted simultaneously via the master control. It'd be appreciated if some one can test the patches with HD-audio h/w that has no master control yet (e.g. some STAC codecs). Note that this won't add the master volumes if there are no real volume controls. Some codecs have really no volume control, and this won't help for such devices. Two (and one for driver) patches will follow after this. Takashi

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[alsa-devel] Where can I find ALSA driver for smdk2410-UDA1341?
by yushilin 06 May '08

06 May '08

Any one knows? Thanks！

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[alsa-devel] [PATCH 1/3] soc - S3C2410 pcm - Update for changed s3c2410 DMA API change
by Mark Brown 05 May '08

05 May '08

From: ing. Davide Rizzo <davide(a)elpa.it> S3C2410 pcm doesn't work. s3c2410_dma_request() now returns the channel number and not 0 if OK. Davide This patch to update to a s3c2410_dma_request return value change Signed-off-by: Davide Rizzo <davide(a)elpa.it> Acked-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/s3c24xx/s3c24xx-pcm.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/s3c24xx/s3c24xx-pcm.c b/sound/soc/s3c24xx/s3c24xx-pcm.c index 6c70a81..7806ae6 100644 --- a/sound/soc/s3c24xx/s3c24xx-pcm.c +++ b/sound/soc/s3c24xx/s3c24xx-pcm.c @@ -171,7 +171,7 @@ static int s3c24xx_pcm_hw_params(struct snd_pcm_substream *substream, ret = s3c2410_dma_request(prtd->params->channel, prtd->params->client, NULL); - if (ret) { + if (ret < 0) { DBG(KERN_ERR "failed to get dma channel\n"); return ret; } -- 1.5.5.1

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[alsa-devel] Aloop master volume and oss
by Gustavo da Silva Serra 05 May '08

05 May '08

Hi! I am using aloop and everything works fine, but when I try to increase or decrease master volume in alsamixer, oss applications continue to play at maximum volume. When I am using a sound card, master volume affects oss applications. Is this an aloop configuration issue? Thanks in advance.

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Re: [alsa-devel] [pulseaudio-discuss] alsa pulse bugs
by tom＠dbservice.com 05 May '08

05 May '08

Quoting Colin Guthrie <gmane(a)colin.guthr.ie>: > Actually just having a very quick glance at the Alsa API docs, it > doesn't mention that this value should be 0 if there are no samples to play: > > http://www.alsa-project.org/alsa-doc/alsa-lib/group___p_c_m.html#ga0d9e14a4… > > Closest it says is: "It's positive and less than buffer size in normal > situation". > > So perhaps this is an invalid assumption at the wine side? However this part: Delay is distance between current application frame position and sound frame position. suggests that it indeed can be zero. Or why would a soundcard have a gap between the read and write positions if it has played everything? If Wine uses a wrong assumption, then please clarify the function description. > Is there perhaps a more appropriate API call they can use to do whatever > test they are doing? What Wine needs to know, is whether a particular sample has already been played or not. It does 'bytes_written - delay_in_bytes' to find out how much of the written data the soundcard has already played. If there is a better test for it, please explain. tom

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Re: [alsa-devel] [pulseaudio-discuss] alsa pulse bugs
by tom＠dbservice.com 05 May '08

05 May '08

Quoting Colin Guthrie <gmane(a)colin.guthr.ie>: Attached patches to #3943 and #3944. Please read the comments there. > tom(a)dbservice.com wrote: >>> Re the snd_pcm_delay() including network latency (#3945), this clearly >>> makes sense for network streams. Does you proposed fix include this >>> delay (albeit with the improvement that it also will drop to 0 if there >>> are no samples queued)? >> >> snd_pcm_delay() should not include any network latency. The API is >> defined as 'read pointer - write pointer', and applications expect >> that. Or at least they expect that when all samples are played that >> the delay drops to zero. > > With the caveat of very limited technical knowledge, I can agree on the > latter point (drop to 0 when all samples are played), but if it was > implemented sans net-delay in pulse would this not cause e.g. a-v sync > issues when playing via alsa to a networked PA server? If so then this > fix would introduce another bug. Your arguments seem reasonable. Let's see what others have to say to it. It would be interesting to see how much delay such change introduces. I can't test PA over network at home, but if someone wants, the patch is attached to the bug entry in the alsa bugtracker. >>> Also re #3942, I believe (but not sure) that the max buffer in the >>> glitch free branch has been increased significantly (as it now needs to >>> keep some degree of past sound as well as future buffer to allow the >>> rewinds to work properly (this is no doubt an inaccurate description of >>> why it's needed.... :p)) I'm not sure how this would affect this >>> solution, but Lennart will be able to advise better. >> >> I'd suggest to export MAX_MEMBLOCKQ_LENGTH in a public API so that >> applications can make use of it. If the max buffer length has been >> increased, then the alsa pulse plugin should be able to propagate that >> to applications using the alsa API. > > Well I'm not sure of the internals of the glitch-free stuff, but I doubt > it's a buffer that can be used in quite the same way as that. Lennart > will be able to advise if I've got the wrong end of the stick in my > comments :) tom

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Re: [alsa-devel] [pulseaudio-discuss] alsa pulse bugs
by tom＠dbservice.com 05 May '08

05 May '08

Quoting Colin Guthrie <gmane(a)colin.guthr.ie>: > tom(a)dbservice.com wrote: >> I reported three bugs to the alsa bugtracker: >> >> https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3942 >> https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3943 >> https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3944 >> >> It's three bugs after all. Once these are fixed I expect Wine to work. >> *fingers crossed* > > Ace Tom :) > > I've cross posted this to the alsa devel list. Hopefully Takashi (who's > committed some pulse-related stuff recently) will have some insight here. > > Re the snd_pcm_delay() including network latency (#3945), this clearly > makes sense for network streams. Does you proposed fix include this > delay (albeit with the improvement that it also will drop to 0 if there > are no samples queued)? snd_pcm_delay() should not include any network latency. The API is defined as 'read pointer - write pointer', and applications expect that. Or at least they expect that when all samples are played that the delay drops to zero. And no, my current patch does not include network latency at all. It simply does what alsa does, 'read pointer - write pointer'. > Also re #3942, I believe (but not sure) that the max buffer in the > glitch free branch has been increased significantly (as it now needs to > keep some degree of past sound as well as future buffer to allow the > rewinds to work properly (this is no doubt an inaccurate description of > why it's needed.... :p)) I'm not sure how this would affect this > solution, but Lennart will be able to advise better. I'd suggest to export MAX_MEMBLOCKQ_LENGTH in a public API so that applications can make use of it. If the max buffer length has been increased, then the alsa pulse plugin should be able to propagate that to applications using the alsa API. tom

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Re: [alsa-devel] [PATCH 2.6.25.1] S3C2410 i2s programming error
by Liam Girdwood 04 May '08

04 May '08

On Sun, 2008-05-04 at 15:00 +0200, ing. Davide Rizzo wrote: > S3C2410 i2s driver currently manages only i2s protocol (and not left > justified one) and slave mode. > With this small patch, other modes are possible. > Davide > > > plain text document attachment (s3c24xx-i2s.patch) > This patch to correctly manage codecs in slave mode and left-justified format > > Signed-off-by: Davide Rizzo <davide(a)elpa.it> > ================================================================================ > --- linux-2.6.25.1/sound/soc/s3c24xx/s3c24xx-i2s.c 2008-04-17 04:49:44.000000000 +0200 > +++ linux-2.6.25.1.elpa/sound/soc/s3c24xx/s3c24xx-i2s.c 2008-05-04 09:23:34.000000000 +0200 > @@ -224,6 +224,7 @@ static int s3c24xx_i2s_set_fmt(struct sn > iismod |= S3C2410_IISMOD_SLAVE; > break; > case SND_SOC_DAIFMT_CBS_CFS: > + iismod &= ~S3C2410_IISMOD_SLAVE; > break; > default: > return -EINVAL; > @@ -234,6 +235,7 @@ static int s3c24xx_i2s_set_fmt(struct sn > iismod |= S3C2410_IISMOD_MSB; > break; > case SND_SOC_DAIFMT_I2S: > + iismod &= ~S3C2410_IISMOD_MSB; > break; > default: > return -EINVAL; > ------------------------------------------------------------------- Acked-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com>

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Re: [alsa-devel] [PATCH 2.6.25.1] S3C2410 pcm audio broken
by Liam Girdwood 04 May '08

04 May '08

On Sun, 2008-05-04 at 14:57 +0200, ing. Davide Rizzo wrote: > S3C2410 pcm doesn't work. > s3c2410_dma_request() now returns the channel number and not 0 if OK. > Davide > plain text document attachment (s3c24xx-pcm.patch) > This patch to update to a s3c2410_dma_request return value change > > Signed-off-by: Davide Rizzo <davide(a)elpa.it> > ================================================================================ > --- linux-2.6.25.1/sound/soc/s3c24xx/s3c24xx-pcm.c 2008-04-17 04:49:44.000000000 +0200 > +++ linux-2.6.25.1.elpa/sound/soc/s3c24xx/s3c24xx-pcm.c 2008-05-04 09:27:46.000000000 +0200 > @@ -171,7 +171,7 @@ static int s3c24xx_pcm_hw_params(struct > ret = s3c2410_dma_request(prtd->params->channel, > prtd->params->client, NULL); > > - if (ret) { > + if (ret < 0) { > DBG(KERN_ERR "failed to get dma channel\n"); > return ret; > } > ------------------------------------------------------------------- Acked-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com>

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[alsa-devel] Spdif clock switching?
by Bart de Boer 04 May '08

04 May '08

Hi all, I have some weird behavior with my Realtek ALC650F chip and it was suggested at Alsa-user (by Dominique Dumont) to post my case here. It may concern a bug in the via82xx driver. I've used the same subject at Alsa-user in case someone wants to check the discussion over there. I'm trying to get all audio (44.1 kHz music and 48 kHz video) through my SPDIF link. I have succeeded at this. I am able to get bitperfect 44.1 kHz digital audio with MythMusic and I'm able to get bitperfect 48 kHz digital audio with mplayer. However, after playing 48 kHz material the system doesn't switch back to 44.1 kHz mode! It seems to be stuck at 48 kHz. The only way to trigger the system back into 44.1 kHz is to play some 44.1 kHz sample through the iec958 device (which will play the 44.1 kHz sample too fast at 48 kHz). After this I'm able to get 44.1 kHz digital audio through the analog device again. :| These are the settings I'm using: alsamixer ========= Unmute "IEC958" Set "IEC958 Playback AC97-SPSA" to 0 (0%) MythTV (General) ================ Audio output device: ALSA:default SPDIF passthrough device: ALSA:iec958:{AES0 0x02} [x] enable AC3 to SPDIF passthrough [x] enable DTS to SPDIF passthrough MythTV (Music) ============== Audio output device: default Ubuntu 8.04 - Sound Preferences =============================== Everything set to: Intel ICH5 - IEC958 .asoundrc ========= pcm.!default { type plug slave.pcm "analog-hw" # slave.pcm "digital-hw" # slave.pcm "dmix-digital" } ctl.!default { type hw card 0 } pcm.analog { type plug slave.pcm "analog-hw" } ctl.analog { type hw card 0 } pcm.analog-hw { type hw card 0 ## The default value for device is 0, so no need to specify ## - Uncomment one of the below or create a new "device N" line as appropriate ## for your sound card or # device 1 # device 4 } ctl.analog-hw { type hw card 0 } pcm.digital-hw { type hw card 0 ## - Comment out "device 1" above and uncomment one of the below or create a ## new "device N" line as appropriate for your sound card or # device 1 # device 2 device 4 } ctl.digital-hw { type hw card 0 } pcm.dmix-digital { type dmix ipc_key 1235 slave { pcm "digital-hw" period_time 0 period_size 1024 buffer_size 4096 rate 48000 } } ctl.dmix-digital { type hw card 0 } mplayer config ============== monitoraspect=16:9 fs=yes vo=xv # ao=alsa ao=alsa:device=iec958 # ao=alsa:device=hw=0.4 # ao=alsa:device=spdif afm=hwac3 $ cat /proc/asound/card0/codec97#0/ac97#0-0 =========================================== 0-0/0: Realtek ALC650F PCI Subsys Vendor: 0x1509 PCI Subsys Device: 0x9202 Capabilities : DAC resolution : 20-bit ADC resolution : 18-bit 3D enhancement : Realtek 3D Stereo Enhancement Current setup Mic gain : +0dB [+0dB] POP path : pre 3D Sim. stereo : off 3D enhancement : off Loudness : off Mono output : MIX Mic select : Mic1 ADC/DAC loopback : off Double rate slots: 10/11 Extended ID : codec=0 rev=1 LDAC SDAC CDAC DSA=0 SPDIF DRA VRA Extended status : SPCV LDAC SDAC CDAC SPDIF=3/4 SPDIF VRA PCM front DAC : 44100Hz PCM Surr DAC : 44100Hz PCM LFE DAC : 44100Hz PCM ADC : 44100Hz SPDIF Control : Consumer PCM Copyright Category=0x2 Generation=1 Rate=44.1kHz SPDIF In Status : Not Locked I hope someone can shed some light. :) Bart

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