[Alsa-devel] Re: [alsa-dev]Question regarding period/buffer and error handling
Cournapeau David [EMAIL PROTECTED] writes: Hi there, For my research, I need to use audio with matlab under linux, and sound support of matlab is kind of... well, crappy (basically, it is opening the /dev/audio file and write to it; on my computer, it doesn't seem to work). That's why I am thinking about using a wrapper to alsa. If you are playing fairly short pieces of sound you might be able to use this function I wrote some time ago: function playsnd(y,fs,bits) wf = tempname; ws = warning; warning off wavwrite(y, fs, bits, wf); warning ws wf = [wf '.wav']; [s,o] = unix(sprintf('aplay %s', wf)); delete(wf); -- Måns Rullgård [EMAIL PROTECTED] --- 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] Re: Question regarding period/buffer and error handling
Cournapeau David [EMAIL PROTECTED] writes: Måns Rullgård wrote: If you are playing fairly short pieces of sound you might be able to use this function I wrote some time ago: function playsnd(y,fs,bits) wf = tempname; ws = warning; warning off wavwrite(y, fs, bits, wf); warning ws wf = [wf '.wav']; [s,o] = unix(sprintf('aplay %s', wf)); delete(wf); Well, I precisely want to avoid writing to a temporary file :) If the files are small they will probably never hit the physical disk. Of course, your sounds might be longer than that. -- Måns Rullgård [EMAIL PROTECTED] --- 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] alsa-lib spamming terminal
Is there a way to make alsa-lib stop spamming the terminal with error messages, for instance when a non-blocking open fails? -- Måns Rullgård [EMAIL PROTECTED] --- 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
[Alsa-devel] Re: alsa-lib spamming terminal
Clemens Ladisch [EMAIL PROTECTED] writes: Måns Rullgård wrote: Is there a way to make alsa-lib stop spamming the terminal with error messages, for instance when a non-blocking open fails? snd_lib_error_set_handler() Thanks. I wonder why I didn't find that when reading the docs. -- Måns Rullgård [EMAIL PROTECTED] --- 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] Re: [Alsa-dev]detecing if a device is already used by an other app
Cournapeau David [EMAIL PROTECTED] writes: Hi there, I am currently using two computers which have a crappy intl8x0 audio chipset, and these chipset cannot be used by two different apps in the same time. For example, if an audio app is using the soundcard, trying to launch xmms ( with alsa output plugin) gives an error message which says something like check that no other app is blocking the device. But if I launch alsaplayer (alsa output plugin) instead of xmms, the app doesn't complain: it 'just' blocks on the snd_pcm_open call. Basically, after having looked at the source, is seems like this difference is coming from the flag in snd_pcm_open: if SND_PCM_NONBLOCK is used, opening an already opened device failed, if SND_PCM_ASYNC is used, the app just blocks, until the other app releases the device. This is correct. Is there a (simple) way to use the blocking call and detecting if the device is already opened ? You can open it in non-blocking mode and set it to blocking later, if you want that behavior when writing data. -- Måns Rullgård [EMAIL PROTECTED] --- 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] Re: Audio and Video sync. Getting Timestamps
James Courtier-Dutton [EMAIL PROTECTED] writes: Juan Carlos Granda wrote: That's my app does: 1.- Open the device for capture 2.- Set the access mode SND_PCM_ACCESS_RW_INTERLEAVED 3.- Set format 16 bits (SND_PCM_FORMAT_S16_LE) 4.- Set channels 2 (stereo) 5.- Set buffer time near 1 second 6.- Set period time near 0.1 seconds 7.- Copy the hardware params to alsa 8.- Set start threshold (0x7f - Explicit start) 9.- Set transfer align 1 10.- Copy the software params to alsa read loop: - Read 'period size' frames (snd_pcm_readi) How can i get the timestamp of the first frame read? If i use snd_pcm_status_get_trigger_tstamp i obtain the same timestamp all the times. If i use snd_pcm_status_get_tstamp i obtain the now time. Is there any way to get the timestamp of every period. Thanks I help develope xinehq.de. We have needed to do audio, video sync, but we don't need any audio timestamps using the get_trigger_tstamp etc. We just use the snd_pcm_delay() call. Take an example. Sample rate 1 Samples a second (1 just to make things easy to explain.) We call snd_pcm_delay() and it returns 4000. 4000 samples at a sample rate of 1 = 400ms. If we now write some samples to the sound device, we know that they will arrive at the speakers in 400ms time. The OP was recording. -- Måns Rullgård [EMAIL PROTECTED] --- This SF.Net email is sponsored by Sleepycat Software Learn developer strategies Cisco, Motorola, Ericsson Lucent use to deliver higher performing products faster, at low TCO. http://www.sleepycat.com/telcomwpreg.php?From=osdnemail3 ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
[Alsa-devel] assert failures in snd_pcm_mmap_commit
When playing music using alsa-lib sometimes, very rarely, the player crashes and dumps core. GDB gives me this stack trace: #0 0xe410 in ?? () #1 0x4be05684 in ?? () #2 0x0006 in ?? () #3 0x7126 in ?? () #4 0x4007e859 in *__GI_raise (sig=6) at ../nptl/sysdeps/unix/sysv/linux/raise.c:66 #5 0x40080032 in *__GI_abort () at ../sysdeps/generic/abort.c:88 #6 0x4007805f in *__GI___assert_fail (assertion=0x0, file=0x0, line=0, function=0x48c9c136 snd_pcm_mmap_commit) at assert.c:83 #7 0x48c3ef71 in snd_pcm_mmap_commit (pcm=0x81232f8, offset=3840, frames=480) at pcm_local.h:289 #8 0x48c5307e in snd_pcm_rate_commit_next_period (pcm=0x80b0e68, appl_offset=6248) at pcm_rate.c:1005 #9 0x48c53397 in snd_pcm_rate_mmap_commit (pcm=0x80b0e68, offset=6028, size=220) at pcm_rate.c:1194 #10 0x48c3ef18 in snd_pcm_mmap_commit (pcm=0x80b0e68, offset=6028, frames=440) at pcm.c:5920 #11 0x48c5716a in snd_pcm_mmap_write_areas (pcm=0x80b0e68, areas=0x4be059b0, offset=0, size=440) at pcm_mmap.c:120 #12 0x48c3f373 in snd_pcm_write_areas (pcm=0x80b0e68, areas=0x4be059b0, offset=0, size=6380, func=0x48c57090 snd_pcm_mmap_write_areas) at pcm.c:6094 #13 0x48c57361 in snd_pcm_mmap_writei (pcm=0x80b0e68, buffer=0x0, size=0) at pcm_mmap.c:180 #14 0x48c381d2 in snd_pcm_writei (pcm=0x7126, buffer=0x80fb420, size=6500) at pcm_local.h:370 #15 0x48bfbb81 in alsa_write (ad=0x0, data=0x0, samples=0) at ../../tcvp/src/output/drivers/alsa/alsa.c:132 #16 0x4ae01689 in audio_play (p=0x80f56f0) at ../../tcvp/src/output/audio/audio.c:218 #17 0x4002b853 in start_thread (arg=0x0) at pthread_create.c:264 #18 0x4012e3aa in clone () from /lib/libc.so.6 I am using Linux 2.6.4 with ALSA driver 1.0.3 and alsa-lib 1.0.3a (which I see now is a little out-dated). The sound card is a SiS7012 in an Asus laptop. Could this be caused by a bug in the player? -- Måns Rullgård [EMAIL PROTECTED] --- 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
[Alsa-devel] Re: Creative MP3+ or Hercules Muse Pocket
Frank Barknecht [EMAIL PROTECTED] writes: Hallo, Robin Cook hat gesagt: // Robin Cook wrote: Trying to find out if either or both of the Creative Soundblaster MP3+ or the Hercules Muse Pocket usb sound cards are supported by the usb driver in ALSA? If so are is there anything special that needs to be done to get them setup under alsa? Normally these cards just all use the snd-usb-audio driver. If you run kernel 2.6 you can just modprobe snd-usb-audio to get sound. I never hear of the Hercules card, but it looks very cool with that knob, which might even work as an input device on Linux. This would then maybe require the hid/event interface of the USB system in the kernel, which is not part of ALSA. Isn't the knob just a volume control? It wouldn't need to send anything to the computer. -- Måns Rullgård [EMAIL PROTECTED] --- 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] Re: An alsa lib, that works over OSS drivers ?
Jaroslav Kysela [EMAIL PROTECTED] writes: On Mon, 19 Apr 2004, David Balazic wrote: This new library would have the same name and API as the existing alsa library. How does this sound ? Feasible ? It's enough to write OSS plugin to alsa-lib. Has it been done? -- Måns Rullgård [EMAIL PROTECTED] --- This SF.Net email is sponsored by: IBM Linux Tutorials Free Linux tutorial presented by Daniel Robbins, President and CEO of GenToo technologies. Learn everything from fundamentals to system administration.http://ads.osdn.com/?ad_id=1470alloc_id=3638op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
[Alsa-devel] Re: An alsa lib, that works over OSS drivers ?
Me Eby [EMAIL PROTECTED] writes: Can't be done that way. That is why there is an OSS wrapper for alsa. The OSS drivers do not support all of the capabilities that ALSA drivers do. So now there's one person saying it can't be done, and another saying it has been done. There seems to be slight contradiction here. -- Måns Rullgård [EMAIL PROTECTED] --- This SF.Net email is sponsored by: IBM Linux Tutorials Free Linux tutorial presented by Daniel Robbins, President and CEO of GenToo technologies. Learn everything from fundamentals to system administration.http://ads.osdn.com/?ad_id=1470alloc_id=3638op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
[Alsa-devel] Re: An alsa lib, that works over OSS drivers ?
Jaroslav Kysela [EMAIL PROTECTED] writes: On Mon, 19 Apr 2004, M?ns Rullg?rd wrote: Me Eby [EMAIL PROTECTED] writes: Can't be done that way. That is why there is an OSS wrapper for alsa. The OSS drivers do not support all of the capabilities that ALSA drivers do. So now there's one person saying it can't be done, and another saying it has been done. There seems to be slight contradiction here. Not really. This comment is for the OSS API-ALSA driver conversion, but the former question was for the ALSA API-OSS driver conversion. They were both replying to the same message, so I assumed they were referring to the same idea. I must have missed something. -- Måns Rullgård [EMAIL PROTECTED] --- This SF.Net email is sponsored by: IBM Linux Tutorials Free Linux tutorial presented by Daniel Robbins, President and CEO of GenToo technologies. Learn everything from fundamentals to system administration.http://ads.osdn.com/?ad_id=1470alloc_id=3638op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
[Alsa-devel] Re: Fw: [Bug 76413] arts does not follow ALSA API
James Courtier-Dutton [EMAIL PROTECTED] writes: Joerg Mayer wrote: On Fri, Mar 19, 2004 at 08:30:13PM +, James Courtier-Dutton wrote: The guys there just don't understand what you told them. If they took the time to actually read what you said, they could easily fix their problem. Summary: - No bug in alsa. Summary 2: There seem to be quite a lot of projects that have problems to get native alsa support right: wine, winex, arts, xmms... wine is only just starting to support alsa properly, due mainly to lack of real need to alsa support for a long time. They are starting to support it well now. xmms don't support it well for obvious reasons. They charge people for OSS drivers, so the xmms alsa driver has to look worse! The funny thing is that they can't even get that right, unless, of course, you consider random skips a sign of correctness. -- Måns Rullgård [EMAIL PROTECTED] --- This SF.Net email is sponsored by: IBM Linux Tutorials Free Linux tutorial presented by Daniel Robbins, President and CEO of GenToo technologies. Learn everything from fundamentals to system administration.http://ads.osdn.com/?ad_id=1470alloc_id=3638op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] snd_pcm_wait strangeness when resampling
Jaroslav Kysela [EMAIL PROTECTED] writes: On Thu, 11 Mar 2004, M?ns Rullg?rd wrote: I've noticed some strange things with snd_pcm_wait. My application opens the sound device in non-blocking mode and uses snd_pcm_wait whenever snd_pcm_writei return EAGAIN. This works fine as long as ALSA isn't doing any resampling. However, when resampling (e.g. from 44.1 kHz to 48 kHz) snd_pcm_wait returns immediately, insisting that there is room in the buffer, after which snd_pcm_writei obviously returns EAGAIN once more. The same thing happens if I use OSS emulation and select() or poll(). Replacing the snd_pcm_wait call with usleep(1) makes the CPU load look normal again, while the application still works. I am using Linux 2.6.4, ALSA driver 1.0.3 and library 1.0.3a with a SIS7012 chip. It behaves the same way with kernel 2.6.3 and driver 1.0.2c. I'm not quite sure when it started. This application used to work properly. Has something changed so I need to modify my application? Could you give me your code to analyze the problem? The full code is in http://tcvp.sf.net/tcvp-snapshot.tar.gz, if you feel like digging through 20k lines of code. The ALSA part is in here: http://tcvp.bkbits.net:8080/tcvp/anno/src/output/drivers/alsa/alsa.c%401.9 It is called from here: http://tcvp.bkbits.net:8080/tcvp/anno/src/output/audio/audio.c%401.13 in the audio_play function -- Måns Rullgård [EMAIL PROTECTED] --- This SF.Net email is sponsored by: IBM Linux Tutorials Free Linux tutorial presented by Daniel Robbins, President and CEO of GenToo technologies. Learn everything from fundamentals to system administration.http://ads.osdn.com/?ad_id70alloc_id638op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] snd_pcm_wait strangeness when resampling
Jaroslav Kysela [EMAIL PROTECTED] writes: On Thu, 11 Mar 2004, M?ns Rullg?rd wrote: I've noticed some strange things with snd_pcm_wait. My application opens the sound device in non-blocking mode and uses snd_pcm_wait whenever snd_pcm_writei return EAGAIN. This works fine as long as ALSA isn't doing any resampling. However, when resampling (e.g. from 44.1 kHz to 48 kHz) snd_pcm_wait returns immediately, insisting that there is room in the buffer, after which snd_pcm_writei obviously returns EAGAIN once more. The same thing happens if I use OSS emulation and select() or poll(). Replacing the snd_pcm_wait call with usleep(1) makes the CPU load look normal again, while the application still works. I am using Linux 2.6.4, ALSA driver 1.0.3 and library 1.0.3a with a SIS7012 chip. It behaves the same way with kernel 2.6.3 and driver 1.0.2c. I'm not quite sure when it started. This application used to work properly. Has something changed so I need to modify my application? Could you give me your code to analyze the problem? I've noticed that it works properly using alsa-lib 1.0.0rc2. That's what I happened to have compiled, it might work with later versions too. Did you manage to build the thing yet? -- Måns Rullgård [EMAIL PROTECTED] --- This SF.Net email is sponsored by: IBM Linux Tutorials Free Linux tutorial presented by Daniel Robbins, President and CEO of GenToo technologies. Learn everything from fundamentals to system administration.http://ads.osdn.com/?ad_id70alloc_id638op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
[Alsa-devel] Re: [2.6.3] snd_intel8x0: playing to fast
Luca [EMAIL PROTECTED] writes: Il Wed, Mar 10, 2004 at 11:08:22PM +0100, M?ns Rullg?rd ha scritto: *This message was transferred with a trial version of CommuniGate(tm) Pro* Luca [EMAIL PROTECTED] writes: Hi, I'm using linux 2.6.3 with alsalib 1.0.2c. When using alsa output plugin of XMMS or mplayer the song is played too fast. Using OSS emulation works fine. This is from bootlog: intel8x0_measure_ac97_clock: measured 49482 usecs intel8x0: clocking to 48000 Make sure the programs are using device default or plughw:0,0 (or whatever device numbers you like). If the programs can't be persuaded to do that they are buggy. mplayer is using hw:0,0. Bug. alsa-init: requested format: 44100 Hz, 2 channels, Signed 16-bit (Little-Endian) alsa-init: compiled for ALSA-1.0.3 alsa-init: soundcard set to hw:0 alsa-init: pcm opend in block-mode alsa-init: chunksize set to 1024 alsa-init: current val=16, fragcount=16 alsa-init: got buffersize=65536 alsa9: 44100 Hz/2 channels/4 bpf/65536 bytes buffer/Signed 16 bit Little Endian It doens't like plugwh:0,0: alsa-init: requested format: 44100 Hz, 2 channels, Signed 16-bit (Little-Endian) alsa-init: compiled for ALSA-1.0.3 alsa-init: soundcard set to plughw alsa-init: pcm opend in block-mode alsa-init: unable to set periodsize: Invalid argument Bug. Btw, mplayer from CVS (checked out today) works nicely with -ao alsa1x: alsa-init: requested format: 44100 Hz, 2 channels, Signed 16-bit (Little-Endian) alsa-init: compiled for ALSA-1.0.3 alsa-init: got device=0, subdevice=0 alsa-init: 1 soundcard found, using: hw:0,0 alsa-init: pcm opened in block-mode alsa-init: chunksize set to 1024 alsa-init: fragcount=16 alsa-init: got buffersize=65536 alsa1x: 48000 Hz/2 channels/4 bpf/65536 bytes buffer/Signed 16 bit Little Endian ... [resample] Changing sample rate to 48000Hz [resample] Changing sample rate to 48000Hz [resample] Using linear interpolation. So my sound card accept sound only at 48000Hz but old mplayer didn't notice that alsa-lib doesn't accept its request for 44100Hz. Correct? So it seems. -- Måns Rullgård [EMAIL PROTECTED] --- This SF.Net email is sponsored by: IBM Linux Tutorials Free Linux tutorial presented by Daniel Robbins, President and CEO of GenToo technologies. Learn everything from fundamentals to system administration.http://ads.osdn.com/?ad_id=1470alloc_id=3638op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
[Alsa-devel] snd_pcm_wait strangeness when resampling
I've noticed some strange things with snd_pcm_wait. My application opens the sound device in non-blocking mode and uses snd_pcm_wait whenever snd_pcm_writei return EAGAIN. This works fine as long as ALSA isn't doing any resampling. However, when resampling (e.g. from 44.1 kHz to 48 kHz) snd_pcm_wait returns immediately, insisting that there is room in the buffer, after which snd_pcm_writei obviously returns EAGAIN once more. The same thing happens if I use OSS emulation and select() or poll(). Replacing the snd_pcm_wait call with usleep(1) makes the CPU load look normal again, while the application still works. I am using Linux 2.6.4, ALSA driver 1.0.3 and library 1.0.3a with a SIS7012 chip. It behaves the same way with kernel 2.6.3 and driver 1.0.2c. I'm not quite sure when it started. This application used to work properly. Has something changed so I need to modify my application? -- Måns Rullgård [EMAIL PROTECTED] --- This SF.Net email is sponsored by: IBM Linux Tutorials Free Linux tutorial presented by Daniel Robbins, President and CEO of GenToo technologies. Learn everything from fundamentals to system administration.http://ads.osdn.com/?ad_id70alloc_id638op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
[Alsa-devel] Re: [2.6.3] snd_intel8x0: playing to fast
Luca [EMAIL PROTECTED] writes: Hi, I'm using linux 2.6.3 with alsalib 1.0.2c. When using alsa output plugin of XMMS or mplayer the song is played too fast. Using OSS emulation works fine. This is from bootlog: intel8x0_measure_ac97_clock: measured 49482 usecs intel8x0: clocking to 48000 Make sure the programs are using device default or plughw:0,0 (or whatever device numbers you like). If the programs can't be persuaded to do that they are buggy. Well, those two are buggy anyway. -- Måns Rullgård [EMAIL PROTECTED] --- This SF.Net email is sponsored by: IBM Linux Tutorials Free Linux tutorial presented by Daniel Robbins, President and CEO of GenToo technologies. Learn everything from fundamentals to system administration.http://ads.osdn.com/?ad_id=1470alloc_id=3638op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
[Alsa-devel] Re: When I use xmms with OSS plugin sound is normal with the ALSA plugin sound skips sometimes
Kristof vansant [EMAIL PROTECTED] writes: When I use xmms with OSS plugin sound is normal but with the ALSA plugin sound skips sometimes. Is this a xmms problem? Or makes OSS emulation the sound stable? It's an xmms problem. Read the code and you'll see why. It can skip with OSS output too. -- Måns Rullgård [EMAIL PROTECTED] --- SF.Net is sponsored by: Speed Start Your Linux Apps Now. Build and deploy apps Web services for Linux with a free DVD software kit from IBM. Click Now! http://ads.osdn.com/?ad_id=1356alloc_id=3438op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
[Alsa-devel] Re: Some proposed ALSA sysfs changes to stock kernel from GregKH
Jaroslav Kysela [EMAIL PROTECTED] writes: On Tue, 20 Jan 2004, M?ns Rullg?rd wrote: R Chan [EMAIL PROTECTED] writes: Greg KH is proposing some sysfs changes to the stock kernel - any impact on CVS ALSA? Is it ok to rely solely on CVS ALSA - I always build 2.6 from the stock tree and then immediately overwrite the sound directory with ALSA CVS. Those patches do not apply cleanly to the latest ALSA version. There are some minor conflicts, all easily resolved. I'm not sure whether anything else depends on the ALSA changes, but I wouldn't think so. I've been running ALSA 1.0.1 merged with Greg's patches for some time, and it seems to be working well. We will merge these patches when they're merged to mainstream. I suppose someone will also update the drivers as needed. Only the intel8x0 driver has been updated. The rest should still work as before these patches. -- Måns Rullgård [EMAIL PROTECTED] --- The SF.Net email is sponsored by EclipseCon 2004 Premiere Conference on Open Tools Development and Integration See the breadth of Eclipse activity. February 3-5 in Anaheim, CA. http://www.eclipsecon.org/osdn ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
[Alsa-devel] Using timestamp features when capturing
I need to find out at what system time a particular sample was captured. The various snd_pcm_*_tstamp functions seem to do something like that, but I can't find any proper documentation for them. I've already searched the mail archives without finding anything useful. Could someone give me a hint? -- Måns Rullgård [EMAIL PROTECTED] --- The SF.Net email is sponsored by EclipseCon 2004 Premiere Conference on Open Tools Development and Integration See the breadth of Eclipse activity. February 3-5 in Anaheim, CA. http://www.eclipsecon.org/osdn ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
[Alsa-devel] Re: Moving from OSS to ALSA
Lorn Potter [EMAIL PROTECTED] writes: On Monday 12 January 2004 4:13 am, James Wright wrote: I am currently looking into rewriting our current OSS sound routines to native ALSA, as it seems OSS will invariably be phased out now that the ALSA driver is distrubuted with the Linux kernel, plus ALSA seems to have a great number of benefits for us. Personally, I hope OSS compat will never be phased out. Why? OSS is simple, and concise. If I am writing a simple audio recording/playing app, I can get the job done using OSS code in _much_ less lines of code. I have to disagree. For my music/video player, I've written both ALSA and OSS output modules. The ALSA module is 261 lines, the OSS module 258 lines, including comments and whitespace. The ALSA module does things that are more or less impossible with OSS, such as sample accurate timing. -- Måns Rullgård [EMAIL PROTECTED] --- This SF.net email is sponsored by: Perforce Software. Perforce is the Fast Software Configuration Management System offering advanced branching capabilities and atomic changes on 50+ platforms. Free Eval! http://www.perforce.com/perforce/loadprog.html ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
[Alsa-devel] Re: weird thing: sound will skip sometimes with OSS in xmms but not if I use esound
Kristof vansant [EMAIL PROTECTED] writes: Does esound do something special to keep sound stable? It happens with snd-via82xx + OSS emulation. I'd rather think xmms does something that causes skips. I've seen it with a cmi8738. It's one of the reasons I stopped using xmms. It's actually quite simple to play sound without skips. -- Måns Rullgård [EMAIL PROTECTED] --- This SF.net email is sponsored by: Perforce Software. Perforce is the Fast Software Configuration Management System offering advanced branching capabilities and atomic changes on 50+ platforms. Free Eval! http://www.perforce.com/perforce/loadprog.html ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
[Alsa-devel] Re: ALSA release 1.0.1
Jaroslav Kysela [EMAIL PROTECTED] writes: On Thu, 8 Jan 2004, Kristof vansant wrote: Does this mean alsa 1.0.1 will be added to kernel 2.6.1 ? No, sorry. This is beyond of my hands. I will send update patches to Andrew Morton and Linus ASAP, but the inclusion is in their hands. Is there a BK tree anywhere for ALSA? It would simplify manual upgrades a bit. -- Måns Rullgård [EMAIL PROTECTED] --- This SF.net email is sponsored by: Perforce Software. Perforce is the Fast Software Configuration Management System offering advanced branching capabilities and atomic changes on 50+ platforms. Free Eval! http://www.perforce.com/perforce/loadprog.html ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
[Alsa-devel] Re: ALSA release 1.0.1
Jaroslav Kysela [EMAIL PROTECTED] writes: On Thu, 8 Jan 2004, M?ns Rullg?rd wrote: Is there a BK tree anywhere for ALSA? It would simplify manual upgrades a bit. Yes: http://linux-sound.bkbits.net/linux-sound The 1.0.1 code is in. Perfect. It's working with Linux 2.6.1-rc3 and a sis7012. -- Måns Rullgård [EMAIL PROTECTED] --- This SF.net email is sponsored by: Perforce Software. Perforce is the Fast Software Configuration Management System offering advanced branching capabilities and atomic changes on 50+ platforms. Free Eval! http://www.perforce.com/perforce/loadprog.html ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
[Alsa-devel] Re: alsa timer slippage
Jan Depner [EMAIL PROTECTED] writes: On Sun, 2003-12-14 at 08:07, Chris Cannam wrote: While trying to track down the source of some poor timing in sequencing, I've noticed that my ALSA sequencer queue timer has a tendency to fall suddenly behind. I have a little test program (available on request) that just starts a queue and every second or so compares the queue timer against real time as returned by gettimeofday(). It doesn't mind if the two don't quite match, but it does complain if the difference between the two timers changes dramatically between two sample points. When I run it, it never lasts for more than about a minute before the ALSA queue timer suddenly slips by anything from 10 to 60 milliseconds. If I'm not mistaken the timing for your audio is coming from your sound card not your system clock. The gettimeofday is from the system clock. They probably won't match. That's true. However, you typically see a gradual drift, not a sudden jump. -- Måns Rullgård [EMAIL PROTECTED] --- 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
[Alsa-devel] Re: plz fix this before alsa 1.0.0 get's released
Kristof vansant [EMAIL PROTECTED] writes: alsa won't compile on kernel 2.6 removing sndversions.h in vxp440.c and vxpocket.c does the trick. /home/lupus/alsa-driver-1.0.0rc2/pcmcia/vx/vxp440.c:6:31: linux/modversions.h: Onbekend bestand of map /home/lupus/alsa-driver-1.0.0rc2/pcmcia/vx/vxp440.c:7:25: sndversions.h: Onbekend bestand of map That's the Dutch message for No such file or directory, right? It's nice to set the language to English when posting error messages. -- Måns Rullgård [EMAIL PROTECTED] --- 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
[Alsa-devel] Re: Tools to help install alsa.
James Courtier-Dutton [EMAIL PROTECTED] writes: 2) For alsa to work in kernel 2.6, the first item in the OSS menu has to be set to M, otherwise module soundcore does not seem to exist. (This is unconfirmed, so might just be user error) That is not the case. It works fine here, without anything under OSS selected. -- Måns Rullgård [EMAIL PROTECTED] --- 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
[Alsa-devel] Re: Can't record properly with cmi8738 card
Tobias Deiminger [EMAIL PROTECTED] writes: I've got this card (Anubis Typhoon): [CMI8738MC6 ]: CMI8738-MC6 - C-Media PCI CMI8738-MC6 C-Media PCI CMI8738-MC6 (model 55) at 0xd400, irq 10 and i'm using alsa 0.9.6. I want to record something via mic-in using DAP for example. When I tried it the first time, i could only record a very low signal (although the signal was monitored clean and loud to the speakers). Then I turned off the mic-to-center switch as described at alsa-project.org. Now the recorded signal was higher, but still to low to work properly with it. What's wrong? Mic-capture is on, mic and mic-caputre are set to max and mic-boost is on. Is this a general problem with this card/driver? Mic input works fine with my Hercules Gamesurround DVD 5.1 (or whatever the name was) based on the same chip, so it can work. What type of microphone do you use? -- Måns Rullgård [EMAIL PROTECTED] --- 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
[Alsa-devel] Re: Can't record properly with cmi8738 card
Tobias Deiminger [EMAIL PROTECTED] writes: Maxtone Moving-Coil Mic Imp. 600 Ohm but i've got the same problem if I use my guitar instead of a mic. I tried also to use my v-amp as preamp for the guitar and this works a little better, but if I turn the preamp master too loud then the recorded signal gets distorted. Most sound cards expect a cheap electret microphone. These have much higher output levels than coil-based ones. I suggest you use a pre-amp with output level suitable for the line-in input. Whatever the usual hifi stuff uses for interconnection is appropriate. -- Måns Rullgård [EMAIL PROTECTED] --- 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
[Alsa-devel] Memory leaks in alsa-lib
(../../../../tcvp/src/tcvp/tcvp.c:328) ==3078==by 0x424D76B9: t_open (../../../../tcvp/src/tcvp/tcvp.c:453) ==3078==by 0x424D7AC9: t_event (../../../../tcvp/src/tcvp/tcvp.c:581) ==3078==by 0x402737B3: thread_wrapper (../../../valgrind/coregrind/vg_libpthread.c:667) ==3078==by 0x40175153: do__quit (../../../valgrind/coregrind/vg_scheduler.c:2146) ==3078== ==3078== ==3078== 5504 bytes in 172 blocks are still reachable in loss record 17 of 20 ==3078==at 0x4002AC23: calloc (../../../valgrind/coregrind/vg_replace_malloc.c:273) ==3078==by 0x4376C73A: _snd_config_make (conf.c:836) ==3078==by 0x4376DD52: snd_config_top (conf.c:1525) ==3078==by 0x43770927: snd_config_update_r (conf.c:3045) ==3078==by 0x43770D67: snd_config_update (conf.c:3095) ==3078==by 0x43789CC6: snd_pcm_open (pcm.c:1929) ==3078==by 0x4373C262: alsa_open (../../../../../tcvp/src/output/alsa/alsa.c:355) ==3078==by 0x424D6E13: new_pipe (../../../../tcvp/src/tcvp/tcvp.c:328) ==3078==by 0x424D76B9: t_open (../../../../tcvp/src/tcvp/tcvp.c:453) ==3078==by 0x424D7AC9: t_event (../../../../tcvp/src/tcvp/tcvp.c:581) ==3078==by 0x402737B3: thread_wrapper (../../../valgrind/coregrind/vg_libpthread.c:667) ==3078==by 0x40175153: do__quit (../../../valgrind/coregrind/vg_scheduler.c:2146) ==3078== ==3078== ==3078== 9024 bytes in 282 blocks are possibly lost in loss record 19 of 20 ==3078==at 0x4002AC23: calloc (../../../valgrind/coregrind/vg_replace_malloc.c:273) ==3078==by 0x4376C73A: _snd_config_make (conf.c:836) ==3078==by 0x4376C80B: _snd_config_make_add (conf.c:862) ==3078==by 0x4376CA4D: parse_value (conf.c:956) ==3078==by 0x4376D0F3: parse_def (conf.c:1184) ==3078==by 0x4376D34D: parse_defs (conf.c:1213) ==3078==by 0x4376CD90: parse_array_def (conf.c:1008) ==3078==by 0x4376CE7B: parse_array_defs (conf.c:1051) ==3078==by 0x4376D16F: parse_def (conf.c:1170) ==3078==by 0x4376D34D: parse_defs (conf.c:1213) ==3078==by 0x4376DE43: snd_config_load1 (conf.c:1544) ==3078==by 0x4376DFDF: snd_config_load (conf.c:1593) ==3078==by 0x437709C5: snd_config_update_r (conf.c:3054) ==3078==by 0x43770D67: snd_config_update (conf.c:3095) ==3078==by 0x43789CC6: snd_pcm_open (pcm.c:1929) ==3078==by 0x4373C262: alsa_open (../../../../../tcvp/src/output/alsa/alsa.c:355) ==3078==by 0x424D6E13: new_pipe (../../../../tcvp/src/tcvp/tcvp.c:328) ==3078==by 0x424D76B9: t_open (../../../../tcvp/src/tcvp/tcvp.c:453) -- Måns Rullgård [EMAIL PROTECTED] --- This SF.net email is sponsored by: VM Ware With VMware you can run multiple operating systems on a single machine. WITHOUT REBOOTING! Mix Linux / Windows / Novell virtual machines at the same time. Free trial click here:http://www.vmware.com/wl/offer/358/0 ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] Memory leaks in alsa-lib
Jaroslav Kysela [EMAIL PROTECTED] writes: PID USER PR NI VIRT RES SHR S %CPU %MEMTIME+ COMMAND 7148 mru 11 0 25160 27m 2740 S 0.0 12.4 0:00.08 tcvp Now the man page for top tells me that VIRT = SWAP + RES, so unless a negative amount has been swapped (what would that mean?), something strange is going on. The odd thing is that this happens only when I use the ALSA output module. Could it be that alsa-lib triggers some bug somewhere in top, or in the kernel? This seems a little unlikely, though. I will try to investigate this problem. I seems to be a kernel problem. See my recent post to linux-kernel on the subject for more details. -- Måns Rullgård [EMAIL PROTECTED] --- This SF.net email is sponsored by: VM Ware With VMware you can run multiple operating systems on a single machine. WITHOUT REBOOTING! Mix Linux / Windows / Novell virtual machines at the same time. Free trial click here:http://www.vmware.com/wl/offer/358/0 ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
[Alsa-devel] Re: plughw versus hw (fwd)
[EMAIL PROTECTED] writes: i read that plughw does some magic conversion for samplerate and stuff. how does this magic work. whats the main differenc in the concept of hw and plughw. hw interfaces directly to the physical card. Thus you can only use modes supported by the hardware. plughw will detect whether the hardware can do what you are requesting, and automatically convert the data to something that can be played, at the expense of CPU time. -- Måns Rullgård [EMAIL PROTECTED] --- This SF.Net email sponsored by: Free pre-built ASP.NET sites including Data Reports, E-commerce, Portals, and Forums are available now. Download today and enter to win an XBOX or Visual Studio .NET. http://aspnet.click-url.com/go/psa0013ave/direct;at.aspnet_072303_01/01 ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
[Alsa-devel] Re: aplay not playing MPEG
Eliot Blennerhassett [EMAIL PROTECTED] writes: Can aplay/alsa send raw MPEG data to a card? aplay -Dhw:0 -v -t raw --channels=2 --format=MPEG --rate=44100 ~/audio/44khz Playing raw data '/home/eliot/audio/44khz' : MPEG, Rate 44100 Hz, Stereo aplay: set_params:805: Sample format non available This command works fine if I specify S16_LE, S23, FLOAT, S24, but not MPEG. Any suggestions as to whe - do I have a hope of making it work? (My card supports all the above including MPEG in exactly the same way as far as ALSA driver is concerned) What model is your card? Are you sure that it really supports mpeg, and that it isn't just a windows driver faking it for marketing reasons? -- Måns Rullgård [EMAIL PROTECTED] --- This SF.net email is sponsored by: VM Ware With VMware you can run multiple operating systems on a single machine. WITHOUT REBOOTING! Mix Linux / Windows / Novell virtual machines at the same time. Free trial click here: http://www.vmware.com/wl/offer/345/0 ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
[Alsa-devel] Re: aplay not playing MPEG
Eliot Blennerhassett [EMAIL PROTECTED] writes: What model is your card? Are you sure that it really supports mpeg, and that it isn't just a windows driver faking it for marketing reasons? Audioscience ASI6244, definitely has MPEG decode on the card (I am one of the cards designers) Well, I guess you should know. I've some Soundblaster cards marketed as MP3, even though they can't do anything of the kind in hardware. I am writing the driver for this card, but can't figure out why the MPEG doesn't even make it to the driver. My lack of understanding of alsa-lib I expect. It's nice that someone writes drivers for their own cards, but I'm afraid I can't help you. -- Måns Rullgård [EMAIL PROTECTED] --- This SF.net email is sponsored by: VM Ware With VMware you can run multiple operating systems on a single machine. WITHOUT REBOOTING! Mix Linux / Windows / Novell virtual machines at the same time. Free trial click here: http://www.vmware.com/wl/offer/345/0 ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
[Alsa-devel] Re: ring buffer pointer accuracy.
Giuliano Pochini [EMAIL PROTECTED] writes: I would like some idea of how accurate the snd_pcm_delay() result is likely to be. If I am trying to syncronise video to audio, and I notice a difference between audio and video, I need to know how accurate I can make it. I suppose you need a precision of 42ms (the duration of a cinema frame). You can read the position at intervals = 42ms. If the value you read is different then the previous one, you know that value is up to date. You know exactly when you started to play the sound. If you do your own timing and you take into account the period time, you only need to read position 3-4 times every 10s or so to check if you are losing sync. You could use the ALSA timer interface. It gives a resolution equal to the period time. If that's too long time, you can use the system clock to interpolate. Even if you have a really bad soundcard, it's unlikely the sample clock skews more than 1%. 1% would be about a minute at the end of normal movie. That's not what I like to call acceptable. If some cards only update delay once per period, but others update it all the time, I would like the application to know about it, so it can decide how accurately it can actually get audio and video in sync. You can use shorter periods if your cards allows it. Most cards I've come across support period lengths of down to 10 ms or less. That's plenty enough for video playback. -- Måns Rullgård [EMAIL PROTECTED] --- This SF.Net email sponsored by: Parasoft Error proof Web apps, automate testing more. Download eval WebKing and get a free book. www.parasoft.com/bulletproofapps1 ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
[Alsa-devel] Re: Fwd: RE: To ESS Tech Support
Warren Turkal [EMAIL PROTECTED] writes: Another message from the ESS people. If some other people could take the time to request this info from ESS also, maybe they would be more forthcoming with the info. Tell them you want the complete specs for the chip, and are willing to sign an NDA. That might help. -- Måns Rullgård [EMAIL PROTECTED] --- This SF.NET email is sponsored by: eBay Great deals on office technology -- on eBay now! Click here: http://adfarm.mediaplex.com/ad/ck/711-11697-6916-5 ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
[Alsa-devel] Re: Fwd: RE: To ESS Tech Support
P. Christeas [EMAIL PROTECTED] writes: Warren Turkal [EMAIL PROTECTED] writes: Another message from the ESS people. If some other people could take the time to request this info from ESS also, maybe they would be more forthcoming with the info. Tell them you want the complete specs for the chip, and are willing to sign an NDA. That might help. That may not be accurate. If you're writing a driver for linux, you most probably want to publish it under GPL, which definitely conflicts with the NDA. The NDA should be about the chip only, not the driver. You could get permission to release source code based on the docs. I got that from 3Dlabs. The real case you should stick at (the one for any such case) is that nobody could actually use the linux driver without owning the product, for which we've paid. Tell them that by supporting (it's so easy for them) linux, they get themselves out of the non-desirable hardware list. We (linux users) have several lists with hw and companies we should avoid. ESS is one of them. This is of course preferred, but it doesn't always help. -- Måns Rullgård [EMAIL PROTECTED] --- This SF.NET email is sponsored by: eBay Great deals on office technology -- on eBay now! Click here: http://adfarm.mediaplex.com/ad/ck/711-11697-6916-5 ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
[Alsa-devel] Re: Fwd: RE: To ESS Tech Support
Allan Klinbail [EMAIL PROTECTED] writes: Here is a message from ESS regarding getting some information for the ESS1988/Allegro-1 series cards in order to get better support for them in linux. I hope that some others will help me badger the ESS people in order to (A) release info needed to make the Linux driver or (B) release the Windows driver source so that it can be used for ideas for the Linux driver. Looks like they didn't read your email... Essentially you are offering them a favour and they see the word linux and run away scared.. It may very well have been an automatic reply triggered by the work linux in the question. I've seen such things before. -- Måns Rullgård [EMAIL PROTECTED] --- This SF.NET email is sponsored by: eBay Great deals on office technology -- on eBay now! Click here: http://adfarm.mediaplex.com/ad/ck/711-11697-6916-5 ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
[Alsa-devel] [PATCH] Fix pause function
This patch fixes the kernel side of snd_pcm_pause() to actually pause instead of always returning -EBADFD. The patch is against alsa-driver-0.9.4, same as CVS HEAD for this file. --- acore/pcm_native.c~ Tue Jun 3 13:48:09 2003 +++ acore/pcm_native.c Tue Jun 3 19:27:06 2003 @@ -863,7 +863,7 @@ static int snd_pcm_pause(snd_pcm_substream_t *substream, int push) { - return snd_pcm_action(snd_pcm_action_pause, substream, 0); + return snd_pcm_action(snd_pcm_action_pause, substream, push); } #ifdef CONFIG_PM -- Måns Rullgård [EMAIL PROTECTED] --- This SF.net email is sponsored by: eBay Get office equipment for less on eBay! http://adfarm.mediaplex.com/ad/ck/711-11697-6916-5 ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
[Alsa-devel] 4/6-channel output on CMI8738
I'm trying to get 4/6-channel playback to work with a CMI8738 chip. It's behaving a little oddly, however. First I have to switch off Four Channel Mode, which I guess is normal. With that switch off, I can play 4 channels on hw:0,1 or surround40. The problem is that the front output is replicated on the rear jack. This is cured by turning on Exchange DAC. The PCM volume control also works in this mode, but the Master control only affects front output. Anyhow, 4-channel playback is pretty much OK. Next I try 6-channel playback. With the same settings it works, except that the volume control doesn't affect channels 5 and 6. With Exchange DAC off, the front output is silent, but channels 1 and 2 are sent to the rear instead. All this would be acceptable if it wasn't for some quirks. Sometimes, the right rear output goes to maximum volume, with no apparent reason, and a few times the output has suddenly stopped completely. This has happened while playing with the mixer settings. Stopping and restarting the playback program helps. Finally, the surround51 device behaves like a duplicate of hw:0,0. Am I doing something wrong, or is the driver just not perfect? Here's my /proc/asound/cards: 0 [CMI8738MC6 ]: CMI8738-MC6 - C-Media PCI CMI8738-MC6 C-Media PCI CMI8738-MC6 (model 55) at 0x8800, irq 24 and /proc/asound/version: Advanced Linux Sound Architecture Driver Version 0.9.2. Compiled on Mar 31 2003 for kernel 2.4.21-pre5. Maybe I should upgrade, but I haven't noticed any changes to the cmipci driver that seem related to the problems. -- Måns Rullgård [EMAIL PROTECTED] --- This SF.net email is sponsored by: eBay Get office equipment for less on eBay! http://adfarm.mediaplex.com/ad/ck/711-11697-6916-5 ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel