April 2011 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] HDA record fails with FIFO error
by Xu, Andiry 11 Apr '11

11 Apr '11

Hi Takashi, Currently I encountered an HD audio issue reported by customer on a Lenovo x100e system. Sometimes, but not always, when starting a recording stream through the HDA controller, the controller generates a large amount of interrupts (~40 000 interrupts per second). After this has happened, jack sense (i e unsolicited events from the codec) stops working until the next system reboot. SD0STS returns a FIFO error (0x28) in interrupt handler. The interrupt service routine acknowledges this error but does not do anything to counteract the root cause to the problem, so it appears again and again. Restarting the stream does not seem to help. Enable MSI or not does not help either. The issue occurs on 2.6.35 and 2.6.38-rc8+, have not tried latest kernel yet but I think it's also there. FIFO error indicates FIFO overrun occurring while the RUN bit is set, but the driver simply acknowledge and clear the error. I wonder what the root cause and the right treatment are in this case. Any suggestions? Thanks & Best regards, Andiry

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[alsa-devel] Question about snd_mixer_selem_get_playback_dB_range()
by Tanu Kaskinen 11 Apr '11

11 Apr '11

Hi, I have added code to Pulseaudio that checks that the kernel driver returns the same values if I call snd_mixer_selem_get_playback_dB_range() or snd_mixer_selem_ask_playback_vol_dB() with the lowest and highest volume steps. The reason I do this is that I've seen a driver that didn't return correct values from _dB_range() (it's fixed nowadays), so it's possible that the same kind of bugs will be seen in the future too. Doing this check at Pulseaudio startup helps to detect such bugs early. My question is about the case where the lowest volume step is mute. Is it a bug in the driver if _dB_range() returns something else than SND_CTL_TLV_DB_GAIN_MUTE as the min gain in such case? -- Tanu

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[alsa-devel] emu10k1 44100Hz and pulseaudio
by Raymond Yau 11 Apr '11

11 Apr '11

https://bugs.launchpad.net/ubuntu/+source/alsa-plugins/+bug/57089 It seem that emu10k1 has problem with 44100Hz aplay -Dhw:0 udial.wav The sound is distorted but there is no distortion when use dmix at 48000Hz defaults.pcm.dmix.rate 48000 aplay -Dplug:dmix:0 udial.wav Does emu10k1 really support 44100Hz playback ? or udial.wav is a special case

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[alsa-devel] [PATCH] Avoid useless loop
by keroid1＠gmail.com 10 Apr '11

10 Apr '11

From: xingchao <keroid1(a)gmail.com> Only drop to the for loop to bind DAI when the work is not finished. Signed-off-by: xingchao <keroid1(a)gmail.com> --- sound/soc/soc-core.c | 6 ++++-- 1 files changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4dda589..0380a26 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1810,8 +1810,10 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) } /* bind DAIs */ - for (i = 0; i < card->num_links; i++) - soc_bind_dai_link(card, i); + if (card->num_rtds != card->num_links) { + for (i = 0; i < card->num_links; i++) + soc_bind_dai_link(card, i); + } /* bind completed ? */ if (card->num_rtd != card->num_links) { -- 1.7.1

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[alsa-devel] Conexant CX20585 recording does not work
by Gabriel Schulhof 09 Apr '11

09 Apr '11

Hi! My mom has a laptop with the following sound card: lspci: 00:14.2 Audio device: ATI Technologies Inc SBx00 Azalia (Intel HDA) (rev 40) driver: snd_hda_intel alsamixer: Card: HDA ATI SB Chip: Conexant CX20585 I can't record from this card no matter what I try. It seems that alsamixer has a lot of options: Mic B <Capture switch> Mic C <Capture switch> Mic E <Capture switch> Mic F <Capture switch> Capture <0% -- 100% slider> Analog Mic Boost <0 db -- 40 db slider with 10 db increment> I have tried to record from each mic in turn, I have tried the different boost levels (0 db - 40 db), but I have had no luck picking up anything. I have attached an external mic to the line-in, but no difference. When I say "I have tried to record" I mean that I logged into her computer, chose one of the mics from the above list, and then I told my mom to say something while looking at the input sound level indicator in the GNOME sound settings panel. On other computers, I have found this a sufficient indication of whether the mic works. Is this inability to record a known limitation of this sound card? If not, is there some special setting I have to modify to make it work? Please find attached the output from alsa-info. TIA for your help, Gabriel

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[alsa-devel] emu10k1 and pulseaudio I: (alsa-lib)setup.c: Cannot lock ctl
by Raymond Yau 09 Apr '11

09 Apr '11

In debugging emu10k1 with pulseaudio by printing elem->id->name SNDERR("Cannot lock ctl elem %s", elem->id->name); I discovered that sometime "EMU10K1 PCM Send Volume" is locked Is this a bug in alsa-lib or pulseaudio ? D: alsa-util.c: Checking for superset analog-stereo (front:%f) D: alsa-util.c: Trying front:0 with SND_PCM_NO_AUTO_FORMAT ... D: alsa-util.c: Managed to open front:0 I: alsa-util.c: cannot disable ALSA period wakeups D: alsa-util.c: Maximum hw buffer size is 371 ms I: (alsa-lib)setup.c: Cannot lock ctl elem EMU10K1 PCM Send Volume I: (alsa-lib)setup.c: Cannot lock ctl elem EMU10K1 PCM Send Volume I: (alsa-lib)setup.c: Cannot lock ctl elem EMU10K1 PCM Send Volume I: (alsa-lib)setup.c: Cannot lock ctl elem EMU10K1 PCM Send Volume D: alsa-util.c: Set neither period nor buffer size. I: (alsa-lib)setup.c: Cannot lock ctl elem EMU10K1 PCM Send Volume I: alsa-util.c: snd_pcm_hw_params failed: Device or resource busy

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Re: [alsa-devel] [PATCH] MAINTAINERS: Update Samsung ASoC maintainer's id
by Mark Brown 09 Apr '11

09 Apr '11

On Fri, Apr 08, 2011 at 11:29:42AM +0530, Jassi Brar wrote: > Signed-off-by: Jassi Brar <jassisinghbrar(a)gmail.com> > --- > MAINTAINERS | 2 +- Applied, thanks.

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[alsa-devel] Vertex distortion bug in xonotic
by Sebastian H. 08 Apr '11

08 Apr '11

Hello list In the Mesa versions 7.9.2 - 7.10.2 I get strange triangle distortion bugs in the game Xonotic. They don't appear in 7.8.2. http://xwmw.org/misc/triangle_bug_01.jpg http://xwmw.org/misc/triangle_bug_02.jpg glxinfo | grep string server glx vendor string: SGI server glx version string: 1.2 client glx vendor string: Mesa Project and SGI client glx version string: 1.4 OpenGL vendor string: X.Org R300 Project OpenGL renderer string: Gallium 0.4 on ATI RV570 OpenGL version string: 2.1 Mesa 7.10.2 OpenGL shading language version string: 1.20 Regards, Sebastian H.

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[alsa-devel] Shipping ALSA External Plugin
by Kip Warner 08 Apr '11

08 Apr '11

Hey list, Is it possible for an application to ship an external ALSA plugin with itself, and have it available at runtime to an existing ALSA installation on the user's system? -- Kip Warner -- Software Engineer OpenPGP encrypted/signed mail preferred http://www.thevertigo.com

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[alsa-devel] usb-audio erro during playback
by Helian 08 Apr '11

08 Apr '11

Hi, I'm trying to use a KingRex UC-192 usb/spdif with my linux setup but I can't play anything. I'm using slackware 13.1 with alsa 1.0.24. Alsa recognizes the device, the output of "lsusb -v" is in the gzip file. The "/proc/asound/cardX/stream0" shows KingRex technology co.,ltd KingRex UC-192 USB TO Spdif/I2S at usb-0000:00:0b.1- : USB Audio Playback: Status: Stop Interface 1 Altset 1 Format: S16_LE Channels: 2 Endpoint: 4 OUT (ADAPTIVE) Rates: 44100, 88200, 176400, 48000, 96000, 192000 Data packet interval: 1000 us Interface 1 Altset 2 Format: S24_3LE Channels: 2 Endpoint: 4 OUT (ADAPTIVE) Rates: 44100, 88200, 176400, 48000, 96000, 192000 Data packet interval: 500 us Interface 1 Altset 3 Format: S32_LE Channels: 2 Endpoint: 4 OUT (ADAPTIVE) Rates: 44100, 88200, 176400, 48000, 96000, 192000 Data packet interval: 500 us but when I try to play anything I only get a distorted sound and the syslog message "ALSA clock.c:116: uac_clock_source_is_valid(): cannot get clock validity for 128" Any suggestions on how to fix this problem ? Thanks

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