Re: [Alsa-devel] configuration space never flattened? Bug?
On Fri, 30 Apr 2004, Clemens Ladisch wrote: Erik Inge Bolsø wrote: I'm having trouble doing snd_pcm_hw_params_can_pause() on a certain pcm, even though it has already (hopefully?) been fixed to one config by snd_pcm_hw_params()... snd_pcm_hw_params has been called, at least. How can I check? fileja { type file file /tmp/alsa slave.pcm null } xine: pcm.c:2604: snd_pcm_hw_params_can_pause: Assertion `params params-info != ~0U' failed. Apparently, the null pcm doesn't initialize params-info. The patch below should help. Yep, works fine. Thanks! -- Erik I. Bolsø | email: knan at mo.himolde.no The UNIX philosophy basically involves giving you enough rope to hang yourself. And then a couple of feet more, just to be sure. --- This SF.Net email is sponsored by: Oracle 10g Get certified on the hottest thing ever to hit the market... Oracle 10g. Take an Oracle 10g class now, and we'll give you the exam FREE. http://ads.osdn.com/?ad_id=3149alloc_id=8166op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
[Alsa-devel] Software volume via LADSPA and .asoundrc
On Mon, 3 May 2004, Clemens Ladisch wrote: Nico Schottelius wrote: anyone knows if there is the possibility to adjust sound volume or not. CMI hardware doesn't have this capability. It would be possible to write a plugin that scales sound data in software before sending it to the device, but nobody has done this yet. Install LADSPA, then try this in your .asoundrc ... adjust the 0.5 to taste. Having it show up as an adjustable mixer control for runtime tuning would be even neater, of course. pcm._volume { type ladspa slave.pcm plughw:0,0; path /usr/local/lib/ladspa; playback_plugins [ { label amp_mono input { controls [ 0.5 ] } } ] } pcm.!default { type plug slave.pcm _volume; } -- Erik I. Bolsø | email: knan at mo.himolde.no The UNIX philosophy basically involves giving you enough rope to hang yourself. And then a couple of feet more, just to be sure. --- This SF.Net email is sponsored by: Oracle 10g Get certified on the hottest thing ever to hit the market... Oracle 10g. Take an Oracle 10g class now, and we'll give you the exam FREE. http://ads.osdn.com/?ad_id149alloc_id66op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] pcm_ladspa and multichannel filters
On Fri, 23 Apr 2004, Erik Inge Bolsø wrote: See subject. Has anyone got this working? I've followed the examples and got a few different filters working (chorus, delay), but am having trouble with more complex ones - and suspect a bug in alsa-lib. Current ALSA CVS version, as of a few hours ago. The plugin in question is from swh-plugins-0.4.3.tar.gz. Had the same problem with a plugin from ladspa's cmt_src_1.15.tgz The .asoundrc below pcm._novoice { type ladspa slave.pcm plughw:0,0; path /usr/local/lib/ladspa; playback_plugins [ { label karaoke policy none input { controls { Vocal volume (dB) -30 } bindings { 0 Left in 1 Right in } } output { bindings { 0 Left out 1 Right out } } } ] } pcm.novoice { type plug slave.pcm _novoice; } gives me [EMAIL PROTECTED]:~$ aplay -Dnovoice PCM1.wav Playing WAVE 'PCM1.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo ALSA lib pcm_ladspa.c:439:(snd_pcm_ladspa_allocate_instances) Unable to connect input port of plugin 'Karaoke' channel 1 depth 0 aplay: set_params:880: Unable to install hw params: ACCESS: RW_INTERLEAVED FORMAT: S16_LE SUBFORMAT: STD SAMPLE_BITS: 16 FRAME_BITS: 32 CHANNELS: 2 RATE: 44100 PERIOD_TIME: (74285 74286) PERIOD_SIZE: 3276 PERIOD_BYTES: 13104 PERIODS: (2 3) BUFFER_TIME: (148594 148595) BUFFER_SIZE: 6553 BUFFER_BYTES: 26212 TICK_TIME: 1 I've tried tracing it in gdb, but gdb keeps playing tricks on me - so no luck yet. At one point, in snd_pcm_ladspa_connect_plugin, I saw the parameters plugin and io resolve to the same address, but haven't been able to reproduce yet. plugin-input looked correct, though. Red herring. printf's show that this was gdb playing tricks. I get this, though. snd_pcm_ladspa_connect_plugin: io-port_bindings_size: 2 snd_pcm_ladspa_connect: instance-channel = 0, channel = 0 snd_pcm_ladspa_connect_plugin: io-port_bindings_size: 2 snd_pcm_ladspa_connect: instance-channel = 0, channel = 1 ALSA lib pcm_ladspa.c:455:(snd_pcm_ladspa_allocate_instances) Unable to connect input port of plugin 'Karaoke' channel 1 depth 0 aplay: set_params:880: Unable to install hw params: Hmmm. On second reading, it seems alsa-lib only handles mono-input and mono-output ladspa plugins, even though the config parser is smart enough to understand the concept of multichannel plugins. Damn. -- Erik I. Bolsø | email: knan at mo.himolde.no The UNIX philosophy basically involves giving you enough rope to hang yourself. And then a couple of feet more, just to be sure. --- This SF.Net email is sponsored by: Oracle 10g Get certified on the hottest thing ever to hit the market... Oracle 10g. Take an Oracle 10g class now, and we'll give you the exam FREE. http://ads.osdn.com/?ad_id=3149alloc_id=8166op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
[Alsa-devel] [PATCH] Fix a few debug messages in alsa-lib pcm_ladspa
Index: src/pcm/pcm_ladspa.c === RCS file: /cvsroot/alsa/alsa-lib/src/pcm/pcm_ladspa.c,v retrieving revision 1.16 diff -u -r1.16 pcm_ladspa.c --- src/pcm/pcm_ladspa.c25 Feb 2004 11:24:30 - 1.16 +++ src/pcm/pcm_ladspa.c28 Apr 2004 21:40:48 - @@ -492,7 +492,7 @@ list_for_each(pos1, plugin-instances) { instance = list_entry(pos1, snd_pcm_ladspa_instance_t, list); if (instance-channel == NO_ASSIGN) { - SNDERR(channel %u is not assigned for plugin '%s' depth %u, plugin-desc-Name, instance-channel, instance-depth); + SNDERR(channel %u is not assigned for plugin '%s' depth %u, instance-channel, plugin-desc-Name, instance-depth); return -EINVAL; } if (instance-channel != channel) { @@ -902,7 +902,7 @@ if (err = 0) { err = snd_pcm_ladspa_find_port(array[channel], lplug, io-pdesc | LADSPA_PORT_AUDIO, port); if (err 0) { - SNDERR(Unable to find an audio port (%li) for channel %s, port); + SNDERR(Unable to find an audio port (%li) for channel %s, port, id); return err; } continue; -- Erik I. Bolsø | email: knan at mo.himolde.no The UNIX philosophy basically involves giving you enough rope to hang yourself. And then a couple of feet more, just to be sure. --- This SF.Net email is sponsored by: Oracle 10g Get certified on the hottest thing ever to hit the market... Oracle 10g. Take an Oracle 10g class now, and we'll give you the exam FREE. http://ads.osdn.com/?ad_id=3149alloc_id=8166op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] aoss failures under Alsa-1.0.0rc1
On Thu, 4 Dec 2003, Mark Knecht wrote: pcm.playback_5_6 { type dshare slave hdsp ipc_key 314159265# some unique number ipc_key_add_uid yes # no to let multiple users share it bindings { 0 5 1 6 } } Uhmm... for _playback_, shouldn't that be type dmix? Or am I confused again? -- Erik I. Bolsø | email: knan at mo.himolde.no The UNIX philosophy basically involves giving you enough rope to hang yourself. And then a couple of feet more, just to be sure. --- This SF.net email is sponsored by: SF.net Giveback Program. Does SourceForge.net help you be more productive? Does it help you create better code? SHARE THE LOVE, and help us help YOU! Click Here: http://sourceforge.net/donate/ ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] crash with rc8
On Wed, 5 Mar 2003, alfp wrote: Also, dmesg gives me Unable to handle kernel NULL pointer dereference at virtual address 0008 printing eip: f8b410b7 *pde = Oops: 0002 CPU:0 EIP:0010:[f8b410b7]Not tainted EFLAGS: 00010296 eax: f6384bd8 ebx: ecx: edx: f6384bd8 esi: 00015888 edi: 00015888 ebp: f638c000 esp: f6277c5c ds: 0018 es: 0018 ss: 0018 Process aplay (pid: 2288, stackpage=f6277000) Stack: f63170f4 f6384ba0 f754b800 f8b561a3 f6384bd8 00015888 f63170e0 f6384ba0 f6277ce4 f8b66f21 f6384ba0 00015888 f754b800 f6277ce4 f6384ba0 f8b4b4e0 Call Trace:[f8b561a3] [f8b66f21] [f8b4b4e0] [f8b4b594] [f8b4ed4a] [c01b87b0] [c0129666] [c01563ff] [c0109627] Code: c7 43 08 00 00 00 00 8b 02 83 f8 02 0f 84 97 00 00 00 83 f8 (one for each crashed aplay) Install ksymoops, run dmesg | ksymoops to decode the oopsen, send them here so we can find out what's wrong. Could be alsa at fault, could be the mandrake vendor kernel at fault. The first oops is usually the most reliable. Download and compile ksymoops from http://www.kernel.org/pub/linux/utils/kernel/ksymoops/v2.4/ if you can't find it in a mandrake rpm. -- Erik I. Bolsø | email: knan at mo.himolde.no The UNIX philosophy basically involves giving you enough rope to hang yourself. And then a couple of feet more, just to be sure. --- This sf.net email is sponsored by:ThinkGeek Welcome to geek heaven. http://thinkgeek.com/sf ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] Native Sample Format
On Sun, 14 Apr 2002, Frank Uepping wrote: This is not a ALSA specific question (so I apologize if this is inappropriate) and directed to driver programmers. Are the sample formats ALAW and MULAW supported by all soundcards in Hardware? What is a native sample format supported by all soundcards? There is none, I suspect. Some pro sundcards support only one sample format (24bit) in hardware, some old cards do only 8bit in hardware... and A/MULAW is not supported in hardware by all cards, no. -- Erik I. Bolsø | email: knan at mo.himolde.no The UNIX philosophy basically involves giving you enough rope to hang yourself. And then a couple of feet more, just to be sure. ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] ALSA AMD OOPS
On Thu, 21 Mar 2002, Mark Constable wrote: -- Unable to handle kernel NULL pointer dereference at virtual address printing eip: c01112a3 *pde = Oops: CPU:0 EIP:0010:[c01112a3]Not tainted EFLAGS: 00010097 eax: f693693c ebx: f6936938 ecx: edx: 0003 esi: f571cc08 edi: 0001 ebp: f5909c84 esp: f5909c6c ds: 0018 es: 0018 ss: 0018 Process alsamixer (pid: 333, stackpage=f5909000) Stack: f6fec8a0 f571cc08 f6fec8e8 f693693c 0082 0003 f6936920 f8883d61 f6936920 0282 b5b0 f571cc08 f5909cfc f7718a00 f88847e1 f7718a00 0001 f571cbc8 f7718a00 0028 0002 Call Trace: [f8883d61] [f88847e1] [c01395f7] [c0106d93] Code: 8b 01 85 45 fc 74 4d 31 c0 9c 5e fa c7 01 00 00 00 00 83 79 Segmentation fault Does not tell us anything at all. Please, run the oops through ksymoops URL: ftp://ftp.us.kernel.org/pub/linux/utils/kernel/ksymoops/v2.4/ and resend. -- Erik I. Bolsø | email: knan at mo.himolde.no The UNIX philosophy basically involves giving you enough rope to hang yourself. And then a couple of feet more, just to be sure. ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] alsa5 API and alsa9 API
On Fri, 15 Mar 2002, Mathieu Dube wrote: The wave.c example and the API's doc doesnt work with alsa 0.9 right? is there any doc anywhere about the alsa 0.9 api? http://www.alsa-project.org/ Click on Documentation on the left. A little down the page, there's a heading ALSA 0.9.x Developer documentation. Click on on-line documentation. You're probably interested in the normal digital audio interface? Heading API Links = PCM (digital audio) interface There's a quite detailed page. At the bottom is links to full examples with cross-links to the API. That's a start, right? Yes, the API link on the front page is outdated. Could someone please update it to point to the same place as ALSA 0.9.x Developer documentation = on-line documentation ? -- Erik I. Bolsø | email: knan at mo.himolde.no The UNIX philosophy basically involves giving you enough rope to hang yourself. And then a couple of feet more, just to be sure. ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
RE: [Alsa-devel] ymfpci : four speakers with OSS-emulation ?
On Wed, 13 Mar 2002, Adam K Kirchhoff wrote: Well, the ymfpci driver (with native Alsa support) only uses the front speakers by default, unless the application (ie. aplay, alsaplayer) is told the use the rear speakers. In either case, you can still only get output on either the front or the rear :-) So rewriting the application wouldn't necessarily solve the problem. Having fun with the a type = multi pcm on top of hw:0,0 and hw:0,2, then a type = plug pcm with route_policy = copy parameter on top of that and the aoss wrapper on top of that again should also do the trick, I imagine. -- Erik I. Bolsø | email: knan at mo.himolde.no The UNIX philosophy basically involves giving you enough rope to hang yourself. And then a couple of feet more, just to be sure. ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
[Alsa-devel] CMIPCI @ Asus A7M266-D bug...
Greets. On my new PC, there's a cmipci chip built-in to the mainboard. It's noisy, but that's as expected. But there's a weird issue with it. I can play 16bit stereo 44.1khz sound just fine. 22 khz, though, is inaudible. In fact, everything but 44khz stereo 16bit gives me either white noise or inaudibility. Facts about the board: The sound part has three jacks... which can be configured as 6-channel output, or output/mic/line_in. No SPDIF, no joystick/midi port, no FM part. Board: Asus A7M266-D ALSA version: 0.9.0beta11 Hope this gives someone an idea what to look for? More info available on request, just ask... root@monster:/home/knan# lspci -n -s 02:04 02:04.0 Class 0401: 13f6:0111 (rev 10) root@monster:/home/knan# lspci -vvvx -s 02:04.0 02:04.0 Multimedia audio controller: C-Media Electronics Inc CM8738 (rev 10) Subsystem: Asustek Computer, Inc.: Unknown device 8077 Control: I/O+ Mem- BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping+ SERR- FastB2B- Status: Cap+ 66Mhz- UDF- FastB2B- ParErr- DEVSEL=medium TAbort- TAbort- MAbort- SERR- PERR- Latency: 32 (500ns min, 6000ns max) Interrupt: pin A routed to IRQ 10 Region 0: I/O ports at c800 [size=256] Capabilities: [c0] Power Management version 2 Flags: PMEClk- DSI- D1+ D2+ AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot-,D3cold-) Status: D0 PME-Enable- DSel=0 DScale=0 PME- 00: f6 13 11 01 85 00 10 02 10 00 01 04 00 20 00 00 10: 01 c8 00 00 00 00 00 00 00 00 00 00 00 00 00 00 20: 00 00 00 00 00 00 00 00 00 00 00 00 43 10 77 80 30: 00 00 00 00 c0 00 00 00 00 00 00 00 0a 01 02 18 knan@monster:~$ cat /proc/asound/cmi/cmipci C-Media PCI CMI8738-MC6 (model 55) at 0xc800, irq 10 00: 02 00 00 00 04: c4 28 00 00 08: 0f 00 09 00 0c: 00 00 00 0a 10: c0 00 00 00 14: 00 60 c0 00 18: 00 00 c9 04 1c: 00 00 fe ff 20: 10 40 00 00 24: a3 30 00 06 28: ff ff ff ff 2c: ff ff ff ff 30: 00 00 00 00 34: 00 00 00 00 38: 00 00 00 00 3c: 00 00 00 00 When playing silently at 22khz: knan@monster:~$ cat /proc/asound/cmi/pcm0p/sub0/hw_params access: RW_INTERLEAVED format: S16_LE subformat: STD channels: 2 rate: 22050 (22050/1) period_size: 2048 buffer_size: 16384 tick_time: 1 OSS format: S16_LE OSS channels: 2 OSS rate: 22050 OSS period bytes: 8192 OSS periods: 8 knan@monster:~$ cat /proc/asound/cmi/pcm0p/sub0/sw_params tstamp_mode: NONE period_step: 1 sleep_min: 0 avail_min: 2048 xfer_align: 1 start_threshold: 1 stop_threshold: 16384 silence_threshold: 0 silence_size: 0 boundary: 1073741824 knan@monster:~$ cat /proc/asound/cmi/cmipci C-Media PCI CMI8738-MC6 (model 55) at 0xc800, irq 10 00: 02 00 01 00 04: c4 28 00 00 08: 0f 00 09 00 0c: 00 00 01 0a 10: c0 00 00 00 14: 00 60 c0 00 18: 00 00 c9 04 1c: 00 00 fe ff 20: 10 40 00 00 24: a3 30 00 06 28: ff ff ff ff 2c: ff ff ff ff 30: 00 00 00 00 34: 00 00 00 00 38: 00 00 00 00 3c: 00 00 00 00 When playing just fine at 44kHz: knan@monster:~$ cat /proc/asound/cmi/cmipci C-Media PCI CMI8738-MC6 (model 55) at 0xc800, irq 10 00: 02 00 01 00 04: c4 2d 00 00 08: 0f 00 09 00 0c: 00 00 01 0a 10: c0 00 00 00 14: 00 60 c0 00 18: 00 00 c9 04 1c: 00 00 fe ff 20: 10 40 00 00 24: a3 30 00 06 28: ff ff ff ff 2c: ff ff ff ff 30: 00 00 00 00 34: 00 00 00 00 38: 00 00 00 00 3c: 00 00 00 00 -- Erik I. Bolsø | email: knan at mo.himolde.no The UNIX philosophy basically involves giving you enough rope to hang yourself. And then a couple of feet more, just to be sure. ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
[Alsa-devel] [Solved] Re: CMIPCI @ Asus A7M266-D bug...
On Tue, 26 Feb 2002, Takashi Iwai wrote: At Tue, 26 Feb 2002 17:55:36 +0100 (CET), Erik Inge Bolsø wrote: On my new PC, there's a cmipci chip built-in to the mainboard. It's noisy, but that's as expected. But there's a weird issue with it. I can play 16bit stereo 44.1khz sound just fine. 22 khz, though, is inaudible. In fact, everything but 44khz stereo 16bit gives me either white noise or inaudibility. comparison between registers on two states gives only the difference of regisister 0x05.. that is quite normal. sorry, i have no idea... well, could you run once arecord with 22kHz and then try aplay with 22kHz again? also, doesn't 48khz playback work, too? Aha! By accident, the Exchange DAC mixer switch was on. Turning that off, all works as expected... Or rather... by experimentation, I've determined this: Exchange DAC switch on, IEC958 In Monitor switch on: 22050 Hz silent, 44100 Hz fine Exchange DAC switch on, IEC958 In Monitor switch off: 22050 Hz silent, 44100 Hz silent Exchange DAC switch off, IEC958 In Monitor switch on: 22050 Hz fine, 44100 Hz fine Exchange DAC switch off, IEC958 In Monitor switch off: 22050 Hz fine, 44100 Hz silent Just extremely weird signal routing on this one card, or is this common? Please document this somewhere for others :) ( Perhaps match by subsystem id and modify mixer element names accordingly to something sane? And/or mark all switches that have something to do with signal routing as such, and let a specialized mixer app take care of such things? ) -- Erik I. Bolsø | email: knan at mo.himolde.no The UNIX philosophy basically involves giving you enough rope to hang yourself. And then a couple of feet more, just to be sure. ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] asoundrc
On Wed, 19 Dec 2001, Alexander Ehlert wrote: Hi, is there any documentation yet for the format of an asoundrc file? alsa-lib-0.9.0beta10a.tar.bz2: alsa-lib-0.9.0beta10a/doc/asoundrc.txt Hopefully even reasonably up-to-date... -- Erik I. Bolsø | email: knan at mo.himolde.no The UNIX philosophy basically involves giving you enough rope to hang yourself. And then a couple of feet more, just to be sure. ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
RE: [Alsa-devel] Philips Acoustic Edge
On Fri, 12 Oct 2001, James Courtier-Dutton wrote: All chip only manufactures publish the details. I don't have any Philips hardware, but there is a principal here. Well, the DSP chip on the Acoustic Edge series_does_ have a 68 page datasheet available. It may not be enough to write a driver, but it's something, right? ( Philips Semiconductors SAA7785 Thunderbird Avenger DSP ) http://www-us6.semiconductors.com/acrobat/datasheets/SAA7785_0.pdf -- Erik I. Bolsø | email: knan at mo.himolde.no The UNIX philosophy basically involves giving you enough rope to hang yourself. And then a couple of feet more, just to be sure. ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
RE: [Alsa-devel] Update: locked modules with no users...
On Wed, 10 Oct 2001, James Courtier-Dutton wrote: I think you will find that X likes to play the odd wave file as you click on the wrong thing etc. It could be X which is using the audio device. Well, X does not use the audio device. The Dreaded Desktop Environment (KDE or GNOME or whatever) might very well start something that eats the audio device, though. A little killall kaudioserver artsd esd should take care of that, if you want to free it. -- Erik I. Bolsø | email: knan at mo.himolde.no The UNIX philosophy basically involves giving you enough rope to hang yourself. And then a couple of feet more, just to be sure. ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] hang up during latency test
On Fri, 5 Oct 2001, Takashi Iwai wrote: At Fri, 05 Oct 2001 10:24:26 -0400, Paul Davis wrote: A good news: after several tries and hacks, I got 1msec latency. from what card? or is this using the h/w pointer location directly, and not relying on interrupts ? i don't know any cards that can provide 1msec output latency by using interrupts ... SB Live with 2 x 256 bytes period. The test program uses simple loop of snd_pcm_writei(). The pcm interface is hw, of course. I tried two patched kernels, LL patch and preemption patch. Both results are shown at http://www.alsa-project.org/~iwai/latency-results/rf-ll2-alsa/2x256.html and http://www.alsa-project.org/~iwai/latency-results/rf-pe2-alsa/2x256.html respectively. nitpick Now what is our definition of latency, exactly? 256 byte periods = 1.45ms between interrupts, according to simple calculations - so we're not at the sub-1ms level yet, at least :)= /nitpick -- Erik I. Bolsø | email: knan at mo.himolde.no The UNIX philosophy basically involves giving you enough rope to hang yourself. And then a couple of feet more, just to be sure. ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel