Re: [Alsa-user] Is ALSA the right tool for my app?
John Sigler wrote: To keep the level constant, it might need to add or remove a few frames once in a while. This won't work with Dolby data (except by dropping or adding a complete Dolby frame). Using a compressed format requires that the decoder is completely synchronized to the source. Regards, Clemens - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] C-media on board audio
Bob van der Poel wrote: But, I can't get a alsamixer to work: bob$ alsamixer No mixer elems found alsamixer works; it just tells you that this device apparently has no mixer controls. Please show the output of lsusb -v for this device. bob$ aplay mix.wav aplay: main:550: audio open error: No such file or directory Do you have /etc/asound.conf or ~/.asoundrc files? Do the device nodes in /dev/snd/ exist? Regards, Clemens - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] C-media on board audio
Sergei Steshenko wrote: On the one hand, the original poster mentions onboard audio, on the other it looks like a USB audio device is present. Is this - two different soundcards - onboard + USB the case and should it be the case ? Or 6 channel onboard audio is USB ? It's 8 channels, and the CM6501 is indeed an onboard USB chip. Regards, Clemens - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Redirecting Mic Input in software (no arecord / aplay involved)
Thierry Bouchard wrote: Your comment bring me another question though, why would enabling the input monitoring on my sound card have fixed my problem and would have been a better solution? What exactly is input monitoring? Input monitoring means that the recorded data is played back immediately. You are implementing this in software, but doing it in hardware would be much easier (for you) and have less latency. Regards, Clemens - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] 2 loopback device
klondike wrote: 2008/1/8, Clemens Ladisch [EMAIL PROTECTED]: Carlos Hernandez wrote: I got a loopback device by doing modproble snd-aloop. But now, I need a second loopback device. How can I up that second device? do I need to modprobe with some parameter? pcm_devs=2 Hi, and sorry for giving this message such an up, but i have tried to modprobe snd-aloop with this parameter but it doesn't work :( Oops, this parameter wasn't used and has been removed in the latest driver version. Use enable=1,1. HTH Clemens - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] asound.conf
Thierry Bouchard wrote:
I wrote a simple plugin for my microphone which is supposed to convert the
data into a 32 bps format. Here is how it looks like :
pcm.jcb-in-1 {
type hw
card 0
device 2
}
pcm.MicPlug
{
type plug
slave
{
pcm jcb-in-1
format S32_LE
}
}
So now if Im opening the PCM device named MicPlug and start reading on it, Im
expecting (which may totally be wrong) that the data I read will be in a
32bps format, which is never happening. In fact, the data I read is exactly
in the format that I set using snd_pcm_hw_params_set_format. Is the
conversion supposed to happen or am I expecting something that is totally
wrong?
This works the other way round: The slave device (the hardware device)
is forced to use 32-bit samples, but the MicPlug device converts the
samples to whatever format is requested (that's what plug does).
If your application wants to use 32-bit samples, it should just request
this format.
Also is there any good documentation about how the asound.conf file works?
Not really. What do you want to do?
Regards,
Clemens
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Re: [Alsa-user] Redirecting Mic Input in software (no arecord / aplay involved)
Thierry Bouchard wrote:
Im trying to capture the Microphone input and redirect it right away into
speakers. The problem is that whenever I use snd_pcm_readi
and snd_pcm_writei one after each other in my capture loop, the pipe
becomes broken on the output side, i.e. every call to snd_pcm_writei fails
after it succeeded for like 2 passes.
In the time that snd_pcm_readi needs to wait for a block of data to
become available, the data written by the last snd_pcm_writei call is
played by the sound card. Just when snd_pcm_readi returns, the playback
buffer has become completely empty, and even the slightest delay in
writing the next data to it will result in an underrun.
To ensure that the playback buffer contains enough samples to adjust for
timing variations, do not start playing when the first block of data is
written to it but when the buffer is (almost) completely filled.
The buffer fullness at which a playback device is started is called the
start threshold. The default value is 1, i.e., after at least one
sample has been written to the buffer; you can change it with the
snd_pcm_sw_params_start_threshold function.
The following is from the implementation of snd_pcm_set_params:
snd_pcm_sw_params_t *swparams;
snd_pcm_sw_params_alloca(swparams);
...
err = snd_pcm_sw_params_current(pcm, swparams);
if (err 0) {
SNDERR(Unable to determine current swparams for %s: %s, s,
snd_strerror(err));
return err;
}
/* start the transfer when the buffer is almost full: */
/* (buffer_size / avail_min) * avail_min */
err = snd_pcm_sw_params_set_start_threshold(pcm, swparams, (buffer_size
/ period_size) * period_size);
if (err 0) {
SNDERR(Unable to set start threshold mode for %s: %s, s,
snd_strerror(err));
return err;
}
/* allow the transfer when at least period_size samples can be
processed */
err = snd_pcm_sw_params_set_avail_min(pcm, swparams, period_size);
if (err 0) {
SNDERR(Unable to set avail min for %s: %s, s,
snd_strerror(err));
return err;
}
/* write the parameters to the playback device */
err = snd_pcm_sw_params(pcm, swparams);
if (err 0) {
SNDERR(Unable to set sw params for %s: %s, s,
snd_strerror(err));
return err;
}
However, an even better solution for your problem would be to enable the
input monitoring function of your sound card. This should be supported
by the CMI8788 chip, but it isn't supported by the driver because so far
I have not been able to find out how it works. I'll look into it.
Regards,
Clemens
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Re: [Alsa-user] Soundcard hda-intel unsupported on linux kernel 2.4
David wrote: I am running on a Linux kernel 2.4 (CentOS 3) and I was just trying to compile the latest alsa drivers (1.0.16) for a Realtek ALC260 soundcard, but when running ./configure --with-cards=hda-intel I get the following error: checking for which soundcards to compile driver for... configure: error: Unsupported soundcard hda-intel By looking inside the ./configure script I can see that this driver is only compiled if the following test holds true: ( test $kversion.$kpatchlevel = 2.6 ) Which is obviously false on my machine running a 2.4 kernel. Is there any workaround this issue? Is the support of the 2.4 kernel on this driver expected any time soon? Or should I simply give up having sound on this machine (I cannot move up to the 2.6 kernel due to other constraints)? The snd-hda-intel driver does not compile on 2.2 kernels, so it has an entry in the kconfig-vers file to disable it on those kernels. However, the kconfig-vers entry says 2.6 because the utils/mod-deps utility that generates that part of the configure script does not support any version checks with numbers less than 2.6. If you don't want to fix mod-deps, just remove the entry from kconfig-vers and run ./hgcompile --with-cards=hda-intel. HTH Clemens - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Monitoring arecord.
James Shatto wrote: Is there any way to monitor arecord's progress? In terms of time recorded, space used, and mixer levels? Specify the -v option two or three times to see recorded levels. Time and space aren't monitored. HTH Clemens - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] driver for theatron agrippa DTS 7.1 and Xi-Fi
Amaury De Ganseman wrote: I want to purchase a club3D theatron agrippa DTS 7.1. On the website they say that the chip is CMI8770 but I don't find this on the cmedia website and no info if it works on alsa or not... In fact I'm confused because on the alsa website thay say it's CMI8788 and on their website CMI8770. The ALSA website says that the Theatron DTS uses a CMI8788 chip. The Theatron Agrippa DTS is a different card. The CMI8770 is just the same as the CMI8768+, which is just the same as the CMI8768 except that the Windows driver comes with software Dolby/ DTS encoders. I've updated the Club3D page. See also http://www.cmedia.com.tw/forums/viewtopic.php?t=1583 I'm also looking for the xi-fi xtrem audio but don't see a lot of info on alsa website...I see that there are 2 chips on for the xtrem audio and another for other xi-fi (fatality for ex). Some (all?) X-Fi Extreme Audio models don't use an X-Fi chip but the same chip that used on the Audigy LS. The Audigy LS chip is supported, but not the other X-Fi chips. HTH Clemens - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] driver found my card, but utils (lib) doesn't
Janos Makadi wrote: I have a system which has to run, with two soundcard. The old ones are died, so I bought two Genius Soundmaker 5.1. My old kernel is 2.6.4, and the alsa version, which cames with this kernel is 1.0.2. The two card, was shown by the kernel, but just the first was adjustable with alsamixer. When I started alsamixer with the second one, it hangs or didn't do anything, it was only respond to the ESC button. I compiled the last (2.6.24.2) kernel, and the 1.0.15 lib, and the utils then. The driver shows, the card, but every util tells: no soundcards found This is the last state withe the 2.6.24.2 kernel, and the 1.0.15 lib, and utils: devices: 2:: timer 3:: sequencer 4: [ 0- 2]: digital audio playback 5: [ 0- 2]: digital audio capture 6: [ 0- 1]: digital audio playback 7: [ 0- 0]: digital audio playback 8: [ 0- 0]: digital audio capture 9: [ 0] : control cards: 0 [CMI8738]: CMI8738-MC6 - C-Media CMI8738 C-Media CMI8738 (model 55) at 0xa000, irq 10 Shouldn't there be two cards? arecord -l: arecord: device_list:205: no soundcards found... Please show the output of ls -l /dev/snd/. Regards, Clemens - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] problem with edirol um-1 ex
Stefan Thomas wrote: I have a problem with the Edirol UM-1EX. It's a USB midi interface. I connected it to my USB-port, but I could neither hear any sound from my soundcard ... I use Aureon 5.1. Fun PCI as soundcard. Your sound card does not have a MIDI synthesizer. You'll have to run a software synthesizer like Timidity or fluidsynth. nor could I input notes with rosegarden. Do you want to input notes by hand, or are you using a MIDI keyboard? Regards, Clemens - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] A52 encoding on Fedora 8
Justin wrote: I'm trying to get a52 encoding to work on fedora 8. ... What are the prereqs required to building this module? http://www.alsa-project.org/main/index.php/A52_plugin HTH Clemens - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] multiple instances of snd-virmidi
Martin Kemp wrote: Does anyone know how to set up multiple instances of snd-virmidi or to gain more than the standard 4 ports. modprobe snd-virmidi enable=1,1,1,1 or put the options into /etc/modprobe.conf (or whatever module configuration file your distribution uses): options snd-virmidi enable=1,1,1,1 To force the virtual cards to a specific position, use multiple values for the index option: options snd-virmidi enable=1,1,1,1 index=4,5,6,7 HTH Clemens - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] random system crashes when outputting midi using snd-mpu401
Phil Carter wrote: I'm trying to get MIDI output (through the game port) to work, but my system locks up randomly when playing MIDI files using aplaymidi. Everything works fine for a few seconds to a few minutes, and then my system freezes and even the magic sysrq keys don't do anything, so I have to hit the reset button on my computer. Any ideas on what I can do to fix this, or to figure out what's causing it? I have ALSA 1.0.14 and ALSA lib 1.0.14a, with Slackware 12, kernel 2.6.21.5. This may be cause by a bug that was fixed in 1.0.16rc1: http://hg.alsa-project.org/alsa-kernel/rev/1fe79ca6c78e Please try updating to newer ALSA drivers (or a newer kernel). Regards, Clemens - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] powermac / tumbler sound stops after few seconds
not disclosed wrote: I have linux install on a powerbook G4 titanium and the sound does not work. At best I get a short burst of sound 1-5seconds long and then nothing and the application hangs (in this case aplay). This bug has been fixed in ALSA 1.0.16rc2. HTH Clemens - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Still get sound with Master or PCM set to zero :(
* wrote: When either my Master or PCM channel is set to zero, there is still sound. Is there any way to calibrate the gain levels that alsa is using for Master=0 and PCM=0? No. These controls are implemented in hardware. Regards, Clemens - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] error : file descriptor is in bad state
pramod gurav wrote: But when i try to writei() to interface . i get the error error : in writei : file descriptor is in bad state I'd guess you didn't set the sample format with snd_pcm_hw_params(). HTH Clemens - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] X-Fi drivers?
The Source wrote: I'm Creative X-Fi Platinum Fatal1ty Champ1on owner and it works fine in Vista, but alsa does not support this card so I don't have sound in linux. I have found drivers on creative site but they are incompatible with newer kernels (with SLUB for example). And it looks like Creative is not going to write new ones. Can alsa include these drivers to it's package No, those drivers contain binary code for a single architecture (x86-64). It might be possible to fix the incompatibilities with newer kernels for someone who has time. or alsa developers should make there own drivers? Creative doesn't give out documentation; this would require major reverse engineering efforts. Regards, Clemens - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] AC3 pass-through possible with SB Audigy 2NX (USB) ?
Nils Rennebarth wrote: I have a creative sound blaster Audigy 2 NX USB sound card because the internal sound card of my laptop isn't that great. The card is connected to my Denon ADV 500SC amplifier by optical S/PDIF. This works very well for stereo output. The amplifier however does support Dolby surround sound, and I would like to hear DVDs with 5.1 Dolby digital tracks by hardware AC3 pass-through. Does the card support this? No. If not, is that a driver limitation or a hardware limitation? I think it is possible, but I don't know how to program this device for non-audio data. Regards, Clemens - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Sound card recognized differently sometimes after reboot
Thomas Börkel wrote: My Soundblaster 5.1 is sometimes recognized differently after boot. Sometimes it has 223 controls, sometimes 224 controls. ... I have saved the 2 different configurations that are being recognized with alsactl: http://www.boerkel.de/asound.state.1 http://www.boerkel.de/asound.state.2 The CD Playback Switch control is missing. So, when alsa wants to restore the mixer settings during boot, it fails and PCM control is muted. The startup script of your distribution shouldn't fail in this case. I'd guess it lacks the -F option when running alsactl restore. Regards, Clemens - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Edirol UA-20
W. Ryan Nestor wrote: I have an Edirol UA-20 which works fine as an audio device in it's 'normal' mode, without MIDI functionality. In it's 'advanced' mode, however, it doesn't seem to work at all. kernel 2.6.23.11-rt14 Alsa1.0.14 proc/asound/cards: (advanced) 1 [UA20 ]: USB-Audio - UA-20 EDIROL UA-20 at usb-:00:07.2-1, full speed Well, it's detected. Does aplay -D default:1 something.wav work? Does it appear in the output of aplaymidi -l? Regards, Clemens - Check out the new SourceForge.net Marketplace. It's the best place to buy or sell services for just about anything Open Source. http://ad.doubleclick.net/clk;164216239;13503038;w?http://sf.net/marketplace ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Edirol UA-20
immanuel litzroth wrote: Quoting Clemens Ladisch [EMAIL PROTECTED]: W. Ryan Nestor wrote: I have an Edirol UA-20 which works fine as an audio device in it's 'normal' mode, without MIDI functionality. In it's 'advanced' mode, however, it doesn't seem to work at all. kernel 2.6.23.11-rt14 Alsa1.0.14 proc/asound/cards: (advanced) 1 [UA20 ]: USB-Audio - UA-20 EDIROL UA-20 at usb-:00:07.2-1, full speed Well, it's detected. Does aplay -D default:1 something.wav work? Does it appear in the output of aplaymidi -l? The device supports different sampling modes in advanced and normal mode. That might explain why it works in one mode and not in the other. It should work in both modes. I think I still have the .alsarc I used with this device and I will send it this evening, but without a plugin doing on the fly resampling you'll have problems. Neither a custom .asoundrc file nor a resampler plugin are needed, this is handled by ALSA in recent versions. Regards, Clemens - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Configure sound card for hundreds devices at once
Quasi Steady State wrote: Everytime the system is booting I run the command 'alsactl -F restore'. Everything goes perfect. But if I *change the harddisk (flash-card) from the device to another (same hardware) device I get an error during booting:* Error if '*alsactl -F restore* ' is run I get the following errors: 1. alsactl: set_control:970: failed to obtain info for control #3 (No such file or directory) 2. alsactl: set_control:1085: bad control.16.value type This means that the sound card on the other device has different mixer controls. Please compare the output of amixer controls on both devices. HTH Clemens - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] ALSA support for toslink
Michelle Dupuis wrote: My MB supports both toslink and SPDIF - from what I've read toslink has a much higher bandwidth. Will the ALSA driver support moving more data through toslink? It's very likely that in your case, Toslink means optical as opposed to the usual electrical coaxial SPDIF connector. The data at both connectors will be exactly the same. Regards, Clemens - Check out the new SourceForge.net Marketplace. It's the best place to buy or sell services for just about anything Open Source. http://ad.doubleclick.net/clk;164216239;13503038;w?http://sf.net/marketplace ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Alsa compilation - lasound
Felipe Uderman wrote: I have successfully ported the alsa library to the arm platform. It shouldn't need porting. What did you have to change? But I don't know what are the binaries of alsa, and where they are located. make install installs the binaries and the configuration files to the default location (/usr/lib and /usr/share/alsa). If you have cross- compiled alsa-lib, you usually use make DESTDIR=/somewhere to put the files below the root of the target machine's file system. HTH Clemens - Check out the new SourceForge.net Marketplace. It's the best place to buy or sell services for just about anything Open Source. http://ad.doubleclick.net/clk;164216239;13503038;w?http://sf.net/marketplace ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Getting a driver to support alsa natively
Adrian McMenamin wrote: On 10/01/2008, Clemens Ladisch [EMAIL PROTECTED] wrote: Adrian McMenamin wrote: $ aplay -D hw:0,0 /aine-email.wav Playing WAVE '/aine-email.wav' : Unsigned 8 bit, Rate 8000 Hz, Mono aplay: set_params:895: Access type not available This means that the device does not support SND_PCM_ACCESS_RW_INTERLEAVED. Could explain this a bit further? The driver doesn't support interleaved because the hardware doesn't. But I assume alsa-lib can fix that? Not when using hw, but the plug plugin should be able to do this (when the access types don't match, it inserts the copy plugin). I don't know why this doesn't work when you're using plughw. Regards, Clemens - Check out the new SourceForge.net Marketplace. It's the best place to buy or sell services for just about anything Open Source. http://ad.doubleclick.net/clk;164216239;13503038;w?http://sf.net/marketplace ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Getting a driver to support alsa natively
Adrian McMenamin wrote: Lee Revell wrote: You also need to find some way to confirm or deny that alsa-lib is trying to open the right files. Given this is nfs-root, the only realistic way is to hack the sources and find out what the path is. Done: http://hg.alsa-project.org/alsa-lib/rev/5593d06d30ee. Regards, Clemens - Check out the new SourceForge.net Marketplace. It's the best place to buy or sell services for just about anything Open Source. http://ad.doubleclick.net/clk;164216239;13503038;w?http://sf.net/marketplace ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] 2 loopback device
Carlos Hernandez wrote: I got a loopback device by doing modproble snd-aloop. But now, I need a second loopback device. How can I up that second device? do I need to modprobe with some parameter? pcm_devs=2 HTH Clemens - Check out the new SourceForge.net Marketplace. It's the best place to buy or sell services for just about anything Open Source. http://ad.doubleclick.net/clk;164216239;13503038;w?http://sf.net/marketplace ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] PXA270 + UCB1400 no sound
Kim wrote: $ cat sounds/alsa/Front_Center.wav /dev/dsp This device doesn't support .wav files; the sample rate will be wrong. Better try aplay. Anyway, you should get _some_ sound this way ... == ./card0/oss_mixer == VOLUME Master 0 ... if the volume is raised and unmuted. HTH Clemens - SF.Net email is sponsored by: Check out the new SourceForge.net Marketplace. It's the best place to buy or sell services for just about anything Open Source. http://ad.doubleclick.net/clk;164216239;13503038;w?http://sf.net/marketplace ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Suzuki Hp-55 dijital piano with ALSA USB MIDI
S.Çağlar Onur wrote: I'm using ALSA 1.0.15 on 2.6.18. The MIDI device is detected and the needed modules hotplugs itself as soon as i plug the cable, but I'm not getting any data via the raw MIDI port or sequencer ports. Here is the output from dmesg when plugging the device: usb 2-1: new low speed USB device using uhci_hcd and address 7 The USB Audio specification does not allow low speed USB MIDI devices because MIDI transfer must use USB bulk transers, but bulk transfers do not exist in low speed mode. The 1.0.15 driver has a workaround to change the transfer type to interrupt transfers, ... ALSA usb/usbmidi.c:170: usb_submit_urb: -22 ... but it seems your kernel does not allow interrupt transfers on bulk endpoints. Try a different kernel. HTH Clemens - SF.Net email is sponsored by: Check out the new SourceForge.net Marketplace. It's the best place to buy or sell services for just about anything Open Source. http://ad.doubleclick.net/clk;164216239;13503038;w?http://sf.net/marketplace ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] alsamixer: function snd_mixer_load failed: Inappropriate ioctl for device
Salatiel Filho wrote: On 12/13/07, Clemens Ladisch [EMAIL PROTECTED] wrote: Salatiel Filho wrote: I am having a big trouble to make sound work on my ARM machine. alsamixer: function snd_mixer_load failed: Inappropriate ioctl for device Make sure that the kernel and the userspace tools are compiled with exactly the same architecture and other options that can affect structure sizes. They are. I compiled both. The size of the respective parameter structures is encoded in the IOCTL numbers. Since it is the snd_mixer_load() function that failed, earlier ioctl calls succeeded, so it's very likely that the reason that SNDRV_CTL_IOCTL_ELEM_READ is unknown is that the size of struct snd_ctl_elem_value is different. Please put a printf(struct size: %u\n, sizeof(struct snd_ctl_elem_value)); (or printk) into both alsamixer and some kernel code (for example, snd_ctl_ioctl() in core/control.c). Regards, Clemens - SF.Net email is sponsored by: Check out the new SourceForge.net Marketplace. It's the best place to buy or sell services for just about anything Open Source. http://ad.doubleclick.net/clk;164216239;13503038;w?http://sf.net/marketplace ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] alsamixer: function snd_mixer_load failed: Inappropriate ioctl for device
Salatiel Filho wrote: I am having a big trouble to make sound work on my ARM machine. alsamixer: function snd_mixer_load failed: Inappropriate ioctl for device Make sure that the kernel and the userspace tools are compiled with exactly the same architecture and other options that can affect structure sizes. HTH Clemens - SF.Net email is sponsored by: Check out the new SourceForge.net Marketplace. It's the best place to buy or sell services for just about anything Open Source. http://ad.doubleclick.net/clk;164216239;13503038;w?http://sf.net/marketplace ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] sharing an input device?
Giuliano Pochini wrote: william estrada [EMAIL PROTECTED] wrote: Is it possible to share an audio device for input? In other words, use the same device for two running programs? If audio device == ALSA card, yes you can if the card has more than one ALSA device (e.g. line-in and mic). If you mean one ALSA card with one ALSA device, you can only if the hardware and the driver do provide this capability, but the two apps must use different subdevices and thus different sources. Two apps cannot open the same card:device:subdevice at the same time. They can when using the dsnoop plugin. The configuration files for most cards already use dsnoop for the default capture device. Regards, Clemens - SF.Net email is sponsored by: Check out the new SourceForge.net Marketplace. It's the best place to buy or sell services for just about anything Open Source. http://sourceforge.net/services/buy/index.php ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] LADSPA plugin systemwide
AM wrote:
I am trying to use a LADSPA plugin (the multiband equalizer - 1197)
system-wide via the /etc/asound.conf file. I want to use my second
sound card for the output.
...
Then I tried the following which gives me an error with aplay;
defaults.pcm.rate_converter samplerate_best
pcm.ladspa {
# pcm
type ladspa
slave.pcm plughw:1,0;
rate 48000;
path /usr/lib/ladspa;
plugins [
{
# label mbeq_1197
id 1197
input {
controls [ 6 2 0 -2 -3 -3 -4 -5 -4 -3 0 3 6 9 13 768 ]
}
}
]
}
#pcm.!default {
pcm.multi_eq {
type plug
slave.pcm ladspa
}
[EMAIL PROTECTED]:~$ aplay -Dplug:multi_eq /music/one.mp3
ALSA lib pcm_ladspa.c:1771:(_snd_pcm_ladspa_open) Unknown field rate
The ladspa plugin does not have a rate field; remove it.
HTH
Clemens
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Re: [Alsa-user] (no subject)
Victor Hahn wrote: I'm trying to set up ALSA to route all audio output to jackd using ALSA's JACK plugin. I set up /etc/asound.conf as described here: http://alsa.opensrc.org/index.php/Jack_(plugin) Unfortunately it doesn't work this way, I won't get any sound out of ALSA any more. Trying to play some audio with mplayer for example just gives me: alsa-init: format s16le are not supported by hardware, trying default alsa-init: unable to set format: Invalid argument The plug plugin in the default definition should automatically convert from S16_LE to Jack's floating-point format. Is there an ALSA device name configured for mplayer? Does playback with aplay work? Regards, Clemens - SF.Net email is sponsored by: Check out the new SourceForge.net Marketplace. It's the best place to buy or sell services for just about anything Open Source. http://sourceforge.net/services/buy/index.php ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] best card for bitperfect SPDIF I/O with external clock sync ?
Paolo Saggese wrote: On Thursday 22 November 2007 15:25, you wrote: This is a valid wish, isn't it? And at least the M-Audio Audiphile cards can sync themselves to the S/PDIF In clock. They can, but the word clock is used only as a sample clock. It will not reduce the amount of jitter of the card's S/PDIF output. that's not a concern! Let' try to explain it better: it's all about avoiding the use of the SPDIF reconstructed clock altogether (on the DAC side). The S/PDIF standard specifies that the clock must be reconstructed from the signal itself. If I can slave the SPDIF clock coming into the external DAC to the same DAC local clock, I can simply ignore the incoming SPDIF clock and just use the local one without any need for reclocking, PLLs, etc! Yes, _if_ you can. http://www.audiocraftersguild.com/AandE/npt.on.jitter2.htm says: | A few companies which make both transports and external DACs have | implemented schemes in which the S/PDIF signal is supplemented with a | second line carrying the master clock back from the external DAC to | the transport. In this way the DAC's crystal becomes the master rather | than the transport and the problems of recovering a spectrally pure | clock are eliminated. No standards for this type of implementation | exist. In reference #4 Dr. Hawksford calls for the clock signal to be | transmitted on a second S/PDIF line, I know of no actual product which | implements this scheme. Sony (in one product) and Arcam send the | actual clock, Linn argues this leads to RFI problems and so they send | a DC servo voltage which controls a VCOX in the transport. So it seems you wouldn't be able to find a sound card that does what you want. If all your audio data comes from the PC anyway, you might want to drop S/PDIF and use a sound card with a break-out box. The I2S bus used to control the DACs in those devices has a separate master clock line. Regards, Clemens - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2005. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] best card for bitperfect SPDIF I/O with external clock sync ?
Paolo Saggese wrote: Of course I plan to use the PC only to provide a bitperfect (exact copy of the original media, normally CD) digital stream to an external DAC. As you probably know better than me, the one major known problem when you strive for the highest possible quality in digital audio reproduction is jitter... and the best (if not only) way to really minimize it is to use a good, clean and stable clock close to the DAC chip, slaving everything else to that one. Thus, what I would need to do would be to slave the sound card SPDIF output clock to the external DAC clock i.e. to make this one become the master clock for the whole digital audio stream. An SPDIF input _always_ derives its clock from its signal. Besides, the clock for the actual DAC has to be a multiple of the bit clock anyway, so there must be a PLL to derive the DAC's clock from the input signual, i.e., the amount of DAC clock jitter depends more on the PLL implementation than on the input signal quality. AFAIK, one possible way to do this is to set up a fake SPDIF output from the external DAC and connect it to an input of the sound card whose SPDIF output goes to the DAC for conversion. Of course the sound card must be able to slave (synchronize) its SPDIF output clock with the one coming from its SPDIF input. There are sound cards that have a word clock input (M-Audio Delta and others based on the ICE1712 chip), but a word clock is only used to force multiple devices to run at the exactly same frequency, i.e., to prevent their clocks from drifting apart. This clock can _not_ be used as a bit clock for digital signals. Even if it were possible, the extra SPDIF connection would introduce additional jitter, thus the SPDIF signal received by the DAC would have more jitter, compared to the case where the sound card uses its own clock for the SPDIF signal. In other words, the sound card's internal clock is of higher quality than any clock that has to be received from an external device. Thus, I would need a sound card which must be: * cabable of bitperfect (pass through) operation at CD standard 16bit/44.1KHz (as well as, possibly, also at higher resolutions and sample rates such as 16/48, 24/48, 24/96 and 24/192). Any cheap CMI8738-based card can do 16 or 24 bits at 44.1 or 48 kHz. Cards based on the CMI8768 chip (and some later 8738 models) or based on the ICE1712 chip support 24/96. Regards, Clemens - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2005. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] best card for bitperfect SPDIF I/O with external clock sync ?
Rene Herman wrote: On 22-11-07 11:10, Clemens Ladisch wrote: An SPDIF input _always_ derives its clock from its signal. Besides, the clock for the actual DAC has to be a multiple of the bit clock anyway, so there must be a PLL to derive the DAC's clock from the input signual, i.e., the amount of DAC clock jitter depends more on the PLL implementation than on the input signal quality. I don't quite understand your reply. He isn't asking to sync the S/PDIF in to the signal, he's asking to sync the S/PDIFF _out_ to the S/PDIF in. Ie, to have a dummy S/PDIF connection DAC -- Card S/PDIF In that exists only to supply a clock to the card, to which Card S/PDIF Out -- DAC will then be synced. This is a valid wish, isn't it? And at least the M-Audio Audiphile cards can sync themselves to the S/PDIF In clock. They can, but the word clock is used only as a sample clock. It will not reduce the amount of jitter of the card's S/PDIF output. Regards, Clemens - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2005. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] best card for bitperfect SPDIF I/O with external clock sync ?
Rene Herman wrote: On 21-11-07 20:33, Paolo Saggese wrote: Thus, I would need a sound card which must be: * cabable of bitperfect (pass through) operation at CD standard 16bit/44.1KHz (as well as, possibly, also at higher resolutions and sample rates such as 16/48, 24/48, 24/96 and 24/192). 16/44.1 is basic enough that any card will be able to operate its S/PDIF output at it. Most cards support only 48 kHz output. This is either a consequence of being a wavetable card that has to mix all data together at a common frequency, or the low-budget design of the AC'97 and HDA standards. Regards, Clemens - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2005. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] UA-4FX problems
Isaac wrote: I'm having a lot of trouble with the Edirol UA-4FX USB/MIDI soundcard in Ubuntu Gutsy. I saw the UA-3FX listed and hoped that the 4FX would also work... aplay -l List of PLAYBACK Hardware Devices card 1: UA4FX [UA-4FX], device 0: USB Audio [USB Audio] It's there. However, typing alsamixer just sends an error message that it can't load because there isn't a default soundcard! The default sound card would be that at index 0. The UA-4FX is loaded at index 1. Try using the sound card configuration utility of your distribution to configure the order of sound cards. HTH Clemens - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2005. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] default to card other than card 0 - asoundrc
James Shatto wrote: If I want the default alsa device to be something other than card 0, is there a simple entry I can put in asoundrc to make that happen? defaults.pcm.card 42 HTH Clemens - This SF.net email is sponsored by: Splunk Inc. Still grepping through log files to find problems? Stop. Now Search log events and configuration files using AJAX and a browser. Download your FREE copy of Splunk now http://get.splunk.com/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Default pcm device question
Ramiro Barreiro wrote: I have an Edirol UA-25 USB soundcard, which is my default and only soundcard: If there is no .asoundrc neither /etc/asound.conf aplay segfaults: [EMAIL PROTECTED]:~$ aplay test.wav Playing WAVE 'test.wav' : Unsigned 8 bit, Rate 22050 Hz, Mono Segmentation fault However, specifying the device does work: [EMAIL PROTECTED]:~$ aplay -D plughw:UA25 test.wav Playing WAVE 'test.wav' : Unsigned 8 bit, Rate 22050 Hz, Mono If you're using alsa-lib 1.0.14, this may be a bug in the dmix plugin. In this case, please try alsa-lib 1.0.15. HTH Clemens - This SF.net email is sponsored by: Splunk Inc. Still grepping through log files to find problems? Stop. Now Search log events and configuration files using AJAX and a browser. Download your FREE copy of Splunk now http://get.splunk.com/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Unsupported soundcard intel8x0 problem
Huynh Phuoc Tai wrote: Clemens Ladisch [EMAIL PROTECTED] wrote: Does the file /home/hptai/source/KZM/4.5/linux-2.6.16-alsa/include/linux/version.h exist? This file exists. #define UTS_RELEASE 2.6.16.19-kzm-lttng-0.5.69byComputex-smp #define LINUX_VERSION_CODE 132624 #define KERNEL_VERSION(a,b,c) (((a) 16) + ((b) 8) + (c)) Strange. Please show the contents of the config.log file. Regards, Clemens - This SF.net email is sponsored by: Splunk Inc. Still grepping through log files to find problems? Stop. Now Search log events and configuration files using AJAX and a browser. Download your FREE copy of Splunk now http://get.splunk.com/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Independent Simultaneous playback over 5 channels
Allen Kennedy wrote:
On 10/25/07, Clemens Ladisch [EMAIL PROTECTED] wrote:
Try this:
pcm.channel {
@args [ CHANNEL ]
@args.CHANNEL {
type integer
}
type plug
slave.pcm {
type dshare
ipc_key 220057 # some random but unique number
slave.pcm hw:0# the output device
slave.channels 6
bindings {
0 $CHANNEL
}
}
}
...
One issue though, it appears as though your solution would only
route a single channel of the input wave file.
The plug plugin should automatically downmix multichannel data before
sending it to the dshare plugin.
What is the output of aplay -v -D channel:0 somestereo.wav?
slave.pcm channel: +$CHANNEL # can't figure out how to concat
That line would have to be written like this:
slave.pcm { @func concat strings [ channel: $CHANNEL ] }
Regards,
Clemens
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Re: [Alsa-user] Unsupported soundcard intel8x0 problem
Huynh Phuoc Tai wrote: ... checking for directory with kernel source... /home/hptai/source/KZM/4.5/linux-2.6.16-alsa checking for directory with kernel build... /home/hptai/source/KZM/4.5/linux-2.6.16-alsa checking for kernel version... 0.0.0 That directory contains only the source files but not the configuration. Make sure that the .config file contains the configuration of the target kernel, and that at least make prepare has been run. HTH Clemens - This SF.net email is sponsored by: Splunk Inc. Still grepping through log files to find problems? Stop. Now Search log events and configuration files using AJAX and a browser. Download your FREE copy of Splunk now http://get.splunk.com/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Independent Simultaneous playback over 5 channels
Allen Kennedy wrote:
I'm trying to get an asound.conf file set up that will allow me
to play 5 different wav files over separate channels
simultaneously, with a command like this:
#aplay -D channel1 1.wav aplay -D channel2 2.wav
aplay -D channel3 3.wav aplay -D channel4 4.wav
aplay -D channel5 5.wav
Try this:
pcm.channel {
@args [ CHANNEL ]
@args.CHANNEL {
type integer
}
type plug
slave.pcm {
type dshare
ipc_key 220057 # some random but unique number
slave.pcm hw:0# the output device
slave.channels 6
bindings {
0 $CHANNEL
}
}
}
Then you can use devices channel:0 through channel:5.
HTH
Clemens
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Re: [Alsa-user] Unsupported soundcard intel8x0 problem
(please don't top-post) Huynh Phuoc Tai wrote: Clemens Ladisch [EMAIL PROTECTED] wrote: Huynh Phuoc Tai wrote: checking for directory with kernel source... /home/hptai/source/KZM/4.5/linux-2.6.16-alsa checking for kernel version... 0.0.0 That directory contains only the source files but not the configuration. I had done make menuconfig and make before installed alsa-driver. Does the file /home/hptai/source/KZM/4.5/linux-2.6.16-alsa/include/linux/version.h exist? Regards, Clemens - This SF.net email is sponsored by: Splunk Inc. Still grepping through log files to find problems? Stop. Now Search log events and configuration files using AJAX and a browser. Download your FREE copy of Splunk now http://get.splunk.com/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Need some assistance with a loopback configuration (willing to pay)
VoIP carrier wrote: I need to setup a configuration that will allow me to route audio between two virtual devices. For example, one application using ALSA will use virtual device A's input and output, and another will use virtual device B's. The output from A will route to the input of B, and the input of A to the output of B. So far, I've tried: -Using snd_dummy and JACK, but there seem to be some quirks and I haven't gotten anything to work correctly. snd_dummy throws away all data sent to it. To get this to work, the applications would have to support input from Jack. -snd_aloop, but I haven't gotten very far as it keeps segfaulting. Are you using ALSA 1.0.14 or later? Regards, Clemens - This SF.net email is sponsored by: Splunk Inc. Still grepping through log files to find problems? Stop. Now Search log events and configuration files using AJAX and a browser. Download your FREE copy of Splunk now http://get.splunk.com/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Soundblaster Connect supported?
Georg Mainik wrote: The web site mentions a driver, so it's likely that the standard driver, which assumes a USB audio class-compliant device, does not work. I still could not find it. Are we talking about the same sites? I meant Creative's web site. http://www.alsa-project.org/main/index.php/Matrix:Vendor-Creative_Labs I added it as unknown. It's possible that only small changes to the driver are needed to get it working. Where can I find out whether there are people working on this device? Here. Nobody is working on it. Nothing will happen unless someone buys one and sends some information (the output of lsusb -v for this device) to the alsa-devel list. Regards, Clemens - This SF.net email is sponsored by: Splunk Inc. Still grepping through log files to find problems? Stop. Now Search log events and configuration files using AJAX and a browser. Download your FREE copy of Splunk now http://get.splunk.com/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Can't get any output from my java program
Edmundo Carmona wrote: I can get the line (Java Sound API) and send it the buffer of information (one second) that I want to listen... however I'm not able to listen to a single beep from the speakers. ... This is the mixer java is using: ICH5 [plughw:0,4]: Direct Audio Device: Intel ICH5, Intel ICH - IEC958, Intel ICH5 - IEC958 This is the SPDIF output. Use default instead of plughw:0,4. HTH Clemens - This SF.net email is sponsored by: Splunk Inc. Still grepping through log files to find problems? Stop. Now Search log events and configuration files using AJAX and a browser. Download your FREE copy of Splunk now http://get.splunk.com/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] SPDIF AC3 Passthrough please help before I need the padded wagon!!!
Another Sillyname wrote: I have three seperate machines outputting to a Sony STR-DB940 AC3 Amp via SPDIF (two optical one RCA). I can get all of them to output as standard ALSA devices from Fedora 7 setups and no specific asound.conf or asound.rc files however I cannot for the life of me get AC3 passthrough to work on any of them (using either VLC or MPLAYER). card 0: CK804 [NVidia CK804], device 0: Intel ICH [NVidia CK804] AC3 passthrough is not supported on nVidia AC'97 controllers if the SPDIF output is not handled by the AC'97 codec. The controller chip probably supports it, but we have no documentation. What hardware are you using on the other two machines? Regards, Clemens - This SF.net email is sponsored by: Splunk Inc. Still grepping through log files to find problems? Stop. Now Search log events and configuration files using AJAX and a browser. Download your FREE copy of Splunk now http://get.splunk.com/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] midi port does not work
Alexander Saydakov wrote: This port shows garbage half of the time. Sometimes it shows right note on/off events, but sometimes complete garbage events like channel aftertouch (keyboard does not have any), resets or other events unrelated to the keys I push. Here I press the A note in the middle octave several times: $ amidi --dump 98 45 F4 90 14 F0 D8 51 E9 D8 45 00 90 45 50 90 45 00 90 D4 F4 90 45 00 About half the bytes are correct, but the others are garbled. It's possible that the port picks up noise from somewhere, or that there is some mechanical problem with the connector. Regards, Clemens - This SF.net email is sponsored by: Splunk Inc. Still grepping through log files to find problems? Stop. Now Search log events and configuration files using AJAX and a browser. Download your FREE copy of Splunk now http://get.splunk.com/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] SPDIF AC3 Passthrough please help before I need the padded wagon!!!
Another Sillyname wrote: Thanks for the response, so does that mean there is a way to get AC3 supported on the chipset using a different setup/configuration? No. card 0: I82801DBICH4 [Intel 82801DB-ICH4], device 4: Intel ICH - IEC958 [Intel 82801DB-ICH4 - IEC958] card 0: Intel [HDA Intel], device 1: AD198x Digital [AD198x Digital] These should work. What version of mplayer are you using? Regards, Clemens - This SF.net email is sponsored by: Splunk Inc. Still grepping through log files to find problems? Stop. Now Search log events and configuration files using AJAX and a browser. Download your FREE copy of Splunk now http://get.splunk.com/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Soundblaster Connect supported?
Georg Mainik wrote: Hi, I am trying to find out whether I would get the Soundblaster Connect (external USB card) working. The web site mentions a driver, so it's likely that the standard driver, which assumes a USB audio class-compliant device, does not work. It's possible that only small changes to the driver are needed to get it working. HTH Clemens - This SF.net email is sponsored by: Splunk Inc. Still grepping through log files to find problems? Stop. Now Search log events and configuration files using AJAX and a browser. Download your FREE copy of Splunk now http://get.splunk.com/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Midi pitchbend sends incorrect message.
[EMAIL PROTECTED] wrote: I am using an m-audio Oxygen keyboard. This has USB and a normal midi output. The pitchbend does not work properly. The pitchbend message for the default (middle position) should should be data bytes 0 and 64. Monitoring the pitch bend using the java API I see data bytes 0 and 0. Regardless of using USB or the standard midi plug. Running under windows works correctly so it is not a hardware problem. Please show the output of aseqdump or amidi --dump for such a pitch bend message. Regards, Clemens - This SF.net email is sponsored by: Splunk Inc. Still grepping through log files to find problems? Stop. Now Search log events and configuration files using AJAX and a browser. Download your FREE copy of Splunk now http://get.splunk.com/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] midi port does not work
Alexander Saydakov wrote: I am trying to hook up a MIDI keyboard. Here is my configuration: Asus P4P800-E motherboard with on-board sound and MIDI/game port. MIDI port is configured 0x330 irq 5 in BIOS setup ... $ aseqdump -l PortClient name Port name ... 20:0MPU-401 UART MPU-401 UART MIDI $ aseqdump -p 20:0 Cannot connect from port 20:0 - Input/output error # tail /var/log/messages ...mpu401_uart.c:263: cmd: 0x3f failed at 0x330 (status = 0x80, data = 0x90) This indicates that the driver was not able to switch the device into UART mode. It's possible that there was some data on the MIDI port that prevented the driver from reading the port's status message; please try running aseqdump after rebooting, before pressing any key on the MIDI keyboard. HTH Clemens - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2005. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Is it possible to get dmix working with an 8bits USB sound card?
David Bourgeois wrote:
Can you tell me which version of alsa-lib you're using? I tried the
stable/development/HG versions but your patch didn't apply straightaway as
the line numbers differ.
It should apply to the Hg version.
I patched it manually and had to add SND_PCM_FORMAT_U8 to
dmix_supported_format
Oops. But this value should be added to generic_dmix_supported_format.
Now the sounds get through but there's a loop somewhere. If I play a song,
it sounds like each second the song is started again and mixed over the
previous one, so after 5 seconds it quickly saturates, playing
simultanously 5 times the same song with different starting times.
It tests for a signed zero sample instead of an unsigned one. In the
generic_mix_areas_u8 function, change the line
if (!*dst) {
to
if (*dst == 0x80) {
HTH
Clemens
-
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Re: [Alsa-user] Laptop problem
Per Andersson wrote: when I try to use sound on my laptop it works fine on the console, but when I start X11 and use programs like sox or aplay the system hangs badly ( i need to remove the battery to get going again ). Same thing if sound is playing when i start X11. The NM256 chip uses the same memory for graphics and sound and isn't properly documented. Make sure that the snd-nm256 module is loaded before starting X, and playing sounds while starting X might be a bad idea. (Siemens 510 Scenic + Slackware 11.0 + alsa ) Slackware 11.0 should come with ALSA 1.0.11 which should contain the latest NM256-related bugfixes, but you might try to compile the latest alsa-driver anyway. HTH Clemens - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2005. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Is it possible to get dmix working with an 8bits USB sound card?
David Bourgeois wrote:
Or can I imagine patching dmix to add support for 8 bits
Please try this patch.
Index: alsa/alsa-lib/src/pcm/pcm_dmix.c
===
--- alsa.orig/alsa-lib/src/pcm/pcm_dmix.c 2007-07-07 11:07:05.0
+0200
+++ alsa/alsa-lib/src/pcm/pcm_dmix.c2007-09-23 12:32:44.0 +0200
@@ -219,7 +219,7 @@ static void mix_areas(snd_pcm_direct_t *
sum = dmix-u.dmix.sum_buffer + channels * dst_ofs +
chn;
dmix-u.dmix.mix_areas2(size, dst, src, sum, dst_step,
src_step, channels * sizeof(signed int));
}
- } else { /* SND_PCM_FORMAT_S24_3LE */
+ } else if (dmix-shmptr-s.format == SND_PCM_FORMAT_S24_3LE) {
unsigned char *src;
volatile unsigned char *dst;
if (dmix-interleaved) {
@@ -245,6 +245,32 @@ static void mix_areas(snd_pcm_direct_t *
sum = dmix-u.dmix.sum_buffer + channels * dst_ofs +
chn;
dmix-u.dmix.mix_areas3(size, dst, src, sum, dst_step,
src_step, channels * sizeof(signed int));
}
+ } else { /* SND_PCM_FORMAT_U8 */
+ unsigned char *src;
+ volatile unsigned char *dst;
+ if (dmix-interleaved) {
+ /*
+* process all areas in one loop
+* it optimizes the memory accesses for this case
+*/
+ dmix-u.dmix.mix_areas_u8(size * channels,
+ ((unsigned char *)dst_areas[0].addr) +
dst_ofs * channels,
+ ((unsigned char *)src_areas[0].addr) +
src_ofs * channels,
+ dmix-u.dmix.sum_buffer + (dst_ofs *
channels),
+ 1, 1, sizeof(signed int));
+ return;
+ }
+ for (chn = 0; chn channels; chn++) {
+ dchn = dmix-bindings ? dmix-bindings[chn] : chn;
+ if (dchn = dmix-shmptr-s.channels)
+ continue;
+ src_step = src_areas[chn].step / 8;
+ dst_step = dst_areas[dchn].step / 8;
+ src = (unsigned char *)(((char *)src_areas[chn].addr +
src_areas[chn].first / 8) + (src_ofs * src_step));
+ dst = (unsigned char *)(((char *)dst_areas[dchn].addr +
dst_areas[dchn].first / 8) + (dst_ofs * dst_step));
+ sum = dmix-u.dmix.sum_buffer + channels * dst_ofs +
chn;
+ dmix-u.dmix.mix_areas_u8(size, dst, src, sum,
dst_step, src_step, channels * sizeof(signed int));
+ }
}
}
Index: alsa/alsa-lib/src/pcm/pcm_direct.c
===
--- alsa.orig/alsa-lib/src/pcm/pcm_direct.c 2007-05-24 20:41:40.0
+0200
+++ alsa/alsa-lib/src/pcm/pcm_direct.c 2007-09-23 12:34:10.0 +0200
@@ -883,6 +883,7 @@ int snd_pcm_direct_initialize_slave(snd_
SND_PCM_FORMAT_S16,
SND_PCM_FORMAT_S16 ^ SND_PCM_FORMAT_S16_LE ^
SND_PCM_FORMAT_S16_BE,
SND_PCM_FORMAT_S24_3LE,
+ SND_PCM_FORMAT_U8,
};
snd_pcm_format_t format;
unsigned int i;
Index: alsa/alsa-lib/src/pcm/pcm_direct.h
===
--- alsa.orig/alsa-lib/src/pcm/pcm_direct.h 2007-04-07 09:19:48.0
+0200
+++ alsa/alsa-lib/src/pcm/pcm_direct.h 2007-09-23 12:26:17.0 +0200
@@ -39,6 +39,11 @@ typedef void (mix_areas3_t)(unsigned int
volatile signed int *sum, size_t dst_step,
size_t src_step, size_t sum_step);
+typedef void (mix_areas_u8_t)(unsigned int size,
+ volatile unsigned char *dst, unsigned char *src,
+ volatile signed int *sum, size_t dst_step,
+ size_t src_step, size_t sum_step);
+
struct slave_params {
snd_pcm_format_t format;
int rate;
@@ -151,6 +156,7 @@ struct snd_pcm_direct {
mix_areas1_t *mix_areas1;
mix_areas2_t *mix_areas2;
mix_areas3_t *mix_areas3;
+ mix_areas_u8_t *mix_areas_u8;
} dmix;
struct {
} dsnoop;
Index: alsa/alsa-lib/src/pcm/pcm_dmix_generic.c
===
--- alsa.orig/alsa-lib/src/pcm/pcm_dmix_generic.c 2007-07-07
11:07:05.0 +0200
+++ alsa/alsa-lib/src/pcm/pcm_dmix_generic.c2007-09-23 12:40:52.0
+0200
@@ -277,6 +277,32 @@ static void generic_mix_areas3(unsigned
}
}
+static
Re: [Alsa-user] Is it possible to get dmix working with an 8bits USB sound card?
David Bourgeois wrote: I have a USB sound card that only accepts unsigned 8 bits format (a tux droid). That works fine with plug but I can't get dmix working as SND_PCM_FORMAT_U8 is not in the supported formats of dmix. Is it possible to do the mix in another format (S16_LE) then have plug do the resampling to 8bits? No, dmix requires access to the hardware buffer. Otherwise could I do that with Jack? If all your application use Jack. Or can I imagine patching dmix to add support for 8 bits (maybe dreaming here ;-) ) Nobody prevents you from imagining or dreaming. :-) 8-bit support shouldn't be too difficult. Regards, Clemens - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2005. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] No sound from new system
Daniel D Jones wrote: Everything is turned up - nothing muted. Probably some other control isn't set correctly. Please show the output of amixer contents. Regards, Clemens - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2005. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] cmipci: Digital input doesn't work
Bernd Eggink wrote: I can't get my AudioExel/MD-Mate to record from the spdif input. How exactly are you trying to record? Are you using the spdif device? Even after editing /etc/asound.state and setting 'IEC958 In Select' to 'true' (and calling 'alsactl restore'), the corresponding control in alsamixer is off, and remains off, whatever I do. That control allows to switch between the to SPDIF inputs pins of the chip. Regards, Clemens - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2005. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] SB Live 24 bit External
michael norman wrote: I have on of these running on OpenSUSE 10.2 using the snd-usb-audio modile as provide by SUSE. The only function available as far as Kmiix or Alsamixer seems to be PCM volume. Then that is probably the only mixer control implemented by the device. Please show the output of lsusb -v for this device. Regards, Clemens - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2005. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] emu 0404 USB support
Aleksander Kamenik wrote: Neither the E-MU 0404 or 0202 USB sound cards are supported in Linux. I have samples, but work on a driver has not yet started. They are not usb-audio standards compliant, so a new driver will have to be written for them. Any chance work on this driver been done? Also, by a new driver James means a separate module next to snd-emu10k1 will have to be created just for EMU USB soundcards? Probably. Please show the output of lsusb -v for this device. Regards, Clemens - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2005. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Mixer problem with cmipci
Jan-Benedict Glaw wrote: Fired up an alsaplayer to play some FLAC. Switched on Four Channel Mode for a test, which nicely adds sound to the rear left/right outputs. After the FLAC ended, I fired up speaker-test again (and of course switched off Four Channel Mode.) This time, it's different than before: front center/LFE, front left/right and side left/right are okay, but rear right/left are highly distorted. It appears that the last sample played in stereo mode gets added to all rear samples. The distortion does not appear when the last stereo file ended with silence. I added a workaround to the driver to play a few silence samples after a stream has been stopped. Get the Hg version, or wait for alsa-driver-1.0.15rc3. HTH Clemens - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2005. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] cmipci: Digital input doesn't work
Bernd Eggink wrote: Clemens Ladisch schrieb: Bernd Eggink wrote: I can't get my AudioExel/MD-Mate to record from the spdif input. How exactly are you trying to record? Are you using the spdif device? I had been using hw:0,2, but spdif doesn't work either. The command arecord -f cd -D spdif aha.wav results, after some seconds, in the error message arecord: pcm_read:1346: read error: Input/output error leaving a 44 byte (i.e., empty) wav file. Does that message mean anything to you? In this case it means that the card tried to record from a digital input, but that there was not any incoming data. Try to switch to the other input, if possible, and/or try the invert switch. HTH Clemens - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2005. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] C-Media Electronics Inc Unknown device 8788
Bill Unruh wrote: I do not know if they have fixed the old problem in the make file. In 14.0 the script cleans out all of the snd-*.o files from /lib/modules/version/kernel but not the .k0 files which are now more usual in kernel module trees. The current version removes all snd*.*o files. Regards, Clemens - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2005. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] C-Media Electronics Inc Unknown device 8788
leegold wrote: [EMAIL PROTECTED]:~$ sudo modprobe snd-cmi8788 FATAL: Module snd_cmi8788 not found. Here's the output form the install, maybe you'll see an error? http://ljg.netfirms.com/s1.html $ sudo ./configure --with-cards=hda-intel This tells the configure script that no driver except snd-hda-intel is needed. To compile only snd-cmi8788, use --with-cards=cmi8788, or just drop the option to compile all drivers. For the last command, you can use make install-modules instead of make install because you only want the new modules. Except for the last make install-modules, you don't need to use sudo. HTH Clemens - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2005. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Basic sound card that supports hardware mixing?
James Pearson wrote: Can anyone recommend a 'basic' PCI sound card that supports hardware mixing? Any card for which the snd-emu10k1, snd-cs46xx or snd-ymfpci drivers are used, i.e., most SB Live! models and cards based on CS46xx or YMF7x4 chips. None of these cards are manufactured anymore, but you should be able to buy them used. HTH Clemens - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2005. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] debugging strange usb audio behaviour.
Jonathan Stowe wrote: Changing the period to 2048 fixes the problem as you suggest, but I'm not quite sure why and who if at all to report the bug to - does a jack client need to take into account the buffer size? Yes, jackd tells the clients what buffer size to use. Regards, Clemens - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2005. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Basic sound card that supports hardware mixing?
James Pearson wrote: Do you happen known what card make/modules use the CS46xx or YMF7x4 chips? CS46xx: BlackGold II 5.1 Hercules Game Theater Hercules Gamesurround Fortissimo III Terratec DMX XFire 1024 Turtle Beach Santa Cruz VideoLogic Sonic Fury My YMF754 card is a Hoontech Soundtrack Digital XG. Many card names mention a chip name (YMF744/YMF754/YMF754B) or Yamaha DS-XG or DS-1. HTH Clemens - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2005. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] C-Media Electronics Inc Unknown device 8788
leegold wrote: On Wed, 12 Sep 2007 18:38:23 +0200, Clemens Ladisch [EMAIL PROTECTED] said: You didn't say that you're using the latest driver version, so please download the alsa-driver-1.0.15rcX package from http://www.alsa-project.org/, unpack it, go into the alsa-driver-1.0.5rcX directory, run ./configure, run make, then, as root, run make install-modules. (This requires that the kernel source or at least the kernel headers package of your distribution is installed.) Ubuntu 7.04: I installed *exactly* according to: https://help.ubuntu.com/community/HdaIntelSoundHowto but: [EMAIL PROTECTED]:~$ aplay --version aplay: version 1.0.15rc1 by Jaroslav Kysela [EMAIL PROTECTED] This indicates that the alsa-utils package is installed. [EMAIL PROTECTED]:~$ aplay -l aplay: device_list:205: no soundcards found... [EMAIL PROTECTED]:~$ sudo modprobe snd-cmi8788 FATAL: Module snd_cmi8788 not found. The alsa-driver package isn't. Was there any error message when running configure or make? (If so, it should be in the last few lines of output.) Additionally many documents say to look at: /proc/asound/cards But I have no asound dir. This means that no sound driver is currently loaded. Regards, Clemens - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2005. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Diamond50 7.1 audio card at Staples
Rene Herman wrote: On 09/12/2007 05:39 PM, Clemens Ladisch wrote: Rene Herman wrote: Latency-wise, I expect softvol is a non-starter always but if this is not a concern, I don't know. Nothing needs to be scheduled separately; there shouldn't be any noticeable latency. Well, there _has_ to be a difference between (at least in principle) being able to DMA from application to card and needing to process each individual sample, no? Yes, but the application processes each individual sample anyway, and the processing can be done synchronously when the application uses an ALSA function to write to the buffer or to tell ALSA that it has done writing to the DMA buffer. The actual time needed to multiply each sample by some value is negligible. Regards, Clemens - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2005. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Diamond50 7.1 audio card at Staples
Gene Heskett wrote: On Wednesday 12 September 2007, Clemens Ladisch wrote: The ADC is still as crappy as in earlier CMI chips, but the CMI8768's DACs have been improved and are as good as other good 16-bit sound cards. 16 bit. Figures. at a 30 dollar bill, one doesn't get much monotonicity. The 24 bit blurb on the box might mean it outputs 24 bits, but one would be ahead to just throw away the LSByte be done with it. I wrote 16 bit because that's all that you would be able to get with the current driver; C-Media's Windows driver doesn't support 24 bits either. I don't know why, there's evidence that the hardware does support 24 bits. (I'm currently trying to find the register bit(s) that would enable it.) The SPDIF output, however, _does_ support 24 bits, as long as you don't use Windows. Regards, Clemens - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2005. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] debugging strange usb audio behaviour.
Jonathan Stowe wrote: I'm getting some strange behaviour between LinuxSampler, JACK and the USB audio driver and I'm not quite sure where to look to narrow it down further. What *appears* to be happening is that when LinuxSampler is being output to JACK and JACK is using a usb audio device (with or without an intervening client) then there is a weird arpeggiation of the notes being played which sounds like something is messing with the events in the ALSA sequencer, the additional notes are the fifth and octave of the original note it sounds like though I haven't tried it right across the keyboard, ( this happens whether the MIDI input is from e.g. a USB keyboard or aplaymidi.) Other clients (e.g. AlsaModularSynth ) in exactly the same configuration play fine as well as does ardour and so forth. LinuxSampler outputting to JACK when using the internal (hda-intel) audio does not exhibit this behaviour. LinuxSampler outputting directly to the usb-audio doesn't do it either. So it's _only_ the combination LinuxSampler+Jack+USB? I don't know how LinuxSampler could change its behaviour depending on Jacks output driver. The only possibility is that using another driver changes the period size, i.e., Jack's buffer size. What is the exact configuration when you start jackd with either driver? Regards, Clemens - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2005. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] OSS spdif by default - was - M-Audio Revolution 7.1 microphone level
Hal V. Engel wrote: On Tuesday 11 September 2007 02:32:59 Clemens Ladisch wrote: For example, to map /dev/dsp to hw:0,1, add the line options snd-pcm-oss dsp_map=2 Well, there is a typo; this should have been dsp_map=1. In /etc/modules.conf I changed alias /dev/dsp snd-pcm-oss to alias /dev/dsp snd-pcm-oss dsp_map=2 and it is now working. It's certainly not this line that made it work, because of the typo, and because options can only be specified in lines beginning with options. Regards, Clemens - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2005. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Diamond50 7.1 audio card at Staples
Rene Herman wrote: On 09/12/2007 03:12 PM, Gene Heskett wrote: Grepping though the driver shows it doesn't support PCM volume, meaning volume adjustment has to be in software (ALSA can do this) and that it records only at 44.1/48 if that's important to you. No wonder its quite affordable. That sounds rather crippled to me. It still has a master volume. You just cannot adjust the computer's sound relative to any of the analog inputs. Latency-wise, I expect softvol is a non-starter always but if this is not a concern, I don't know. Nothing needs to be scheduled separately; there shouldn't be any noticeable latency. As to the capture direction; well, capture isn't widely used. You don't need 192 kHz for recording, say, a VoIP conversation. The ADC is still as crappy as in earlier CMI chips, but the CMI8768's DACs have been improved and are as good as other good 16-bit sound cards. Regards, Clemens - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2005. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Serial Midi via snd-serial-u16550 not working
Denny Schierz wrote: FATAL: Error inserting snd_serial_u16550 (/lib/modules/2.6.22-gentoo-r2/kernel/sound/drivers/snd-serial-u16550.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error running install command for snd_serial_u16550 snd_serial_u16550: Unknown parameter `snd_port' Replace snd_port with port. The snd_ prefix was dropped from module parameter names several years ago; apparently, some old HOWTO still contains this. Make sure that snd_port doesn't appear in any of the configuration files. HTH Clemens - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2005. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] C-Media Electronics Inc Unknown device 8788
leegold wrote: $ sudo modprobe snd-CMI8788 FATAL: Module snd_CMI8788 not found. The name is snd-cmi8788, not snd-CMI8788. You didn't say that you're using the latest driver version, so please download the alsa-driver-1.0.15rcX package from http://www.alsa-project.org/, unpack it, go into the alsa-driver-1.0.5rcX directory, run ./configure, run make, then, as root, run make install-modules. (This requires that the kernel source or at least the kernel headers package of your distribution is installed.) HTH Clemens - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2005. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] OSS spdif by default - was - M-Audio Revolution 7.1 microphone level
Hal V. Engel wrote: How do I get output from OSS based applications to use spdif by default or at all? The default mapping of OSS devices to ALSA devices is as follows: /dev/dsp0 - hw:0,0 (first device on first card) /dev/adsp0 - hw:0,1 (second device on first card) /dev/dsp1 - hw:1,0 (first device on second card) /dev/adsp1 - hw:1,1 (second device on second card) ... (Often, /dev/dsp instead of /dev/dsp0 is used.) You can tell the OSS application to use /dev/adsp0 if the SPDIF output is the second device (this should be the case with the Revo7.1). If your application doesn't allow changing the device, or if you want to change the default for all applications, or if you want to map to a hw:X,2 device, you can change the mapping with the dsp_map and adsp_map options of the snd-pcm-oss module. These options specify, for each card, which subdevice (the number after the comma) to use. The default settings are dsp_map=0,0,0,0,0,0,0,0 and adsp_map=1,1,1,1,1,1,1,1, where each number is for one sound card. For example, to map /dev/dsp to hw:0,1, add the line options snd-pcm-oss dsp_map=2 to /etc/modprobe.conf (or whatever file contains ALSA module options). HTH Clemens - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2005. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] [alsa-devel] Disable MIDI port creation with ICE1724
Rene Herman wrote: On 09/08/2007 03:32 AM, James Roberts wrote: I need to disable the creation of the ICEnsemble ICE1724 midi device on my system. I am using a off-brand card (see below) and accessing the midi device crashes the system hard. Sounds like this may warrent a module parameter. Trivial, untested version below, but this needs a module maintainer to decide if there may be better ways to discriminate. It would be better to autodetect the card, i.e., add a new entry to card_tables[]. Does this card have some unique ID? (see /proc/asound/cardX/ice1724 and the output of lspci -v for this device) Regards, Clemens - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2005. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Fixed device number attribution
Lee Revell wrote: On 9/6/07, Clemens Ladisch [EMAIL PROTECTED] wrote: The snd-usb-audio has a pid parameter to specify that a device with a certain product ID should go to a specifc index. For example, options snd-usb-audio index=1,2 pid=0x1234,0x5678 would force the device with product ID 1234 to come first. What about multiple identical USB devices? Surely there is some kind of hardware GUID that can be used, otherwise no OS could handle this case... USB devices can have an optional serial number, but audio devices usually don't have one. Regards, Clemens - This SF.net email is sponsored by: Splunk Inc. Still grepping through log files to find problems? Stop. Now Search log events and configuration files using AJAX and a browser. Download your FREE copy of Splunk now http://get.splunk.com/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Fixed device number attribution
Rene Herman wrote: If you load the modules with a index=N parameter, this will fix the card at number N. Nornally, you do this by sticking options snd-foo index=0 options snd-bar index=1 lines in /etc/modprobe.conf (or /etc/modprobe.d/whatever). This alone doesn't work when multiple devices are handled by the same driver. The snd-usb-audio has a pid parameter to specify that a device with a certain product ID should go to a specifc index. For example, options snd-usb-audio index=1,2 pid=0x1234,0x5678 would force the device with product ID 1234 to come first. Vendor and product IDs can be listed with lsusb. HTH Clemens - This SF.net email is sponsored by: Splunk Inc. Still grepping through log files to find problems? Stop. Now Search log events and configuration files using AJAX and a browser. Download your FREE copy of Splunk now http://get.splunk.com/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] please test the beta CMI8788 driver
ATT TynTyn wrote: Under Fedora 7 with the newest kernel, I get ~ 0.25 samples every ~ 0.75 sec. on the main output. That is, 0.25 s of sound at normal pitch, then 0.75 s of silence? Is it different with 2/4/6 channels (the -c parameter)? Regards, Clemens - This SF.net email is sponsored by: Splunk Inc. Still grepping through log files to find problems? Stop. Now Search log events and configuration files using AJAX and a browser. Download your FREE copy of Splunk now http://get.splunk.com/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Mixer problem with cmipci
Jan-Benedict Glaw wrote: On Thu, 2007-08-30 23:43:26 +0200, Jan-Benedict Glaw [EMAIL PROTECTED] wrote: Just out of interest, I ordered another, similar card (Ultron Octosound 7.1, lspci: C-Media Electronics Inc CM8738 (rev 10), /proc/asound/cards: C-Media PCI CMI8738-MC8 (model 68)) I have the same chip here. [...] After the FLAC ended, I fired up speaker-test again (and of course switched off Four Channel Mode.) This time, it's different than before: front center/LFE, front left/right and side left/right are okay, but rear right/left are highly distorted. I can reproduce this. Register 0x04 ~ Before: 0x07 CM_ASFC_SHIFT | 0x00 CM_DSFC_SHIFT | CM_BREQ After:0x07 CM_ASFC_SHIFT | 0x07 CM_DSFC_SHIFT | CM_BREQ (DAC Sample Frequency changed) 0 is the power-on default of 5512 Hz, 7 is 48 kHz. DAC and ADC aren't quite the correct names, both DMA channels can be used for either playback and capture, and the driver uses one for stereo and the other for multichannel playback. Does the FLAC file have a sample rate of 44.1 kHz or 48 kHz? Register 0x08 ~ Before: CM_CHB3D5C | CM_SPDIF_SELECT1 | 0x03 CM_CH1FMT_MASK | 0x00 CM_CH0FMT_SHIFT After:CM_CHB3D5C | CM_SPDIF_SELECT1 | 0x03 CM_CH1FMT_MASK | 0x03 CM_CH0FMT_SHIFT (Ch0 Format changed) 0 is 8-bit mono, 3 is 16-bit stereo (or more than two channels). Register 0x18 ~ Before: CM_TXVX | CM_FM_EN | CM_AC3EN2 | CM_VIDWPPRT | CM_ENCENTER | CM_FLINKOFF After:CM_TXVX | CM_FM_EN | CM_AC3EN2 | CM_VIDWPPRT | CM_SPDF_AC97 | CM_ENCENTER | CM_FLINKOFF (SPDIF/Out changed from 44.1kHz to 48kHz) My guess is that the change to SPDIF/Out frequency and DAC Sample Frequency are annoying at best, but irrelevant here. Am I probably right that the changed ch0 format is the cause of the distortion? I think it's the sample rate. It seems I can 'cure' the distortion by playing a 48 kHz file before running speaker-test. Can you confirm this? The chip has two sample rate registers (ASFC and DSFC), and these are internally mapped to three destinations (front+side+center DACs, rear DAC, capture ADC) depending on the state of a bunch of other bits. I guess the driver doesn't handle these bits correctly so that the rear sample rate is still the old value (although the two xSFC registers appear to have the correct value). Regards, Clemens - This SF.net email is sponsored by: Splunk Inc. Still grepping through log files to find problems? Stop. Now Search log events and configuration files using AJAX and a browser. Download your FREE copy of Splunk now http://get.splunk.com/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] XLR3 interfaces?
James Shatto wrote: Have a look at the Edirol UA-25. It is definitely supported. Thanks, that looks like Exactly what I'm looking for. Out of curiousity, since I've never used anything like these(yet), what else does it do? Regular sound card stuff, or just a recording interface. It does recording and playback. Like many other USB devices, it probably has few or no software-controllable mixer controls. What do you mean with regular sound cars stuff? Onboard midi, or just an interface? It has two MIDI jacks, input and output. What is the difference between onboard midi and interface? Multiple channel output with hardware mixing? USB devices don't do hardware mixing; sending all those streams over the bus would take more resources than mixing them in software. Regards, Clemens - This SF.net email is sponsored by: Splunk Inc. Still grepping through log files to find problems? Stop. Now Search log events and configuration files using AJAX and a browser. Download your FREE copy of Splunk now http://get.splunk.com/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] problems with dmix, a52 and upmix
Michael KAufmann wrote: Is there any way to make dmix work with a52 as slave? No. As it's currently designed, dmix requires a hw device as slave because it needs access to the sound card's DMA buffer and interrupt for timing purposes. Regards, Clemens - This SF.net email is sponsored by: Splunk Inc. Still grepping through log files to find problems? Stop. Now Search log events and configuration files using AJAX and a browser. Download your FREE copy of Splunk now http://get.splunk.com/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Mixer problem with cmipci
Jan-Benedict Glaw wrote: Register 0x04 ~ Before: 0x07 CM_ASFC_SHIFT | 0x00 CM_DSFC_SHIFT | CM_BREQ After:0x07 CM_ASFC_SHIFT | 0x07 CM_DSFC_SHIFT | CM_BREQ (DAC Sample Frequency changed) Register 0x08 ~ Before: CM_CHB3D5C | CM_SPDIF_SELECT1 | 0x03 CM_CH1FMT_MASK | 0x00 CM_CH0FMT_SHIFT After:CM_CHB3D5C | CM_SPDIF_SELECT1 | 0x03 CM_CH1FMT_MASK | 0x03 CM_CH0FMT_SHIFT (Ch0 Format changed) Register 0x18 ~ Before: CM_TXVX | CM_FM_EN | CM_AC3EN2 | CM_VIDWPPRT | CM_ENCENTER | CM_FLINKOFF After:CM_TXVX | CM_FM_EN | CM_AC3EN2 | CM_VIDWPPRT | CM_SPDF_AC97 | CM_ENCENTER | CM_FLINKOFF (SPDIF/Out changed from 44.1kHz to 48kHz) My guess is that the change to SPDIF/Out frequency and DAC Sample Frequency are annoying at best, but irrelevant here. Am I probably right that the changed ch0 format is the cause of the distortion? It could be any of the three changes. The FLAC file apparently is 44.1 kHz while speaker-test used 48 kHz by default, so I guess the driver doesn't correctly switch between these two frequencies. I'll do some testing over the weekend. Regards, Clemens - This SF.net email is sponsored by: Splunk Inc. Still grepping through log files to find problems? Stop. Now Search log events and configuration files using AJAX and a browser. Download your FREE copy of Splunk now http://get.splunk.com/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Mixer problem with cmipci
Jan-Benedict Glaw wrote: Just gave it a test in a Windows box. I did only test stereo output (I'm not familiar with windows and didn't find a test program like speaker-test) with this degenerated Media Player, but there was audible output, which didn't work under Linux, probably due to the stuck 'IEC958 In Monitor' mixer element. Well, now we'd need the source code of the Windows driver ... Regards, Clemens - This SF.net email is sponsored by: Splunk Inc. Still grepping through log files to find problems? Stop. Now Search log events and configuration files using AJAX and a browser. Download your FREE copy of Splunk now http://get.splunk.com/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] chipsets
paul blakeley wrote: I have installed linux on a number of PCs over the last year or so. Each PC having a different audio chipset. Recently I have installed linux on PCs with the ALC888 and CS4299 chipset. I was able to work out the CS4299 chipset basically downloading the chipset block diagram from the manufacturer. I am sure I mapped all mixer controls but I had to guess some mixer controls. The ALC888 is different though, looking at the chipset block diagram there are potentially 50 simple mixer controls. The alsa mixer has around half that. The initialization and configuration of HDA codecs should be described by a table in the BIOS, but the snd-hda-intel driver has workarounds for many motherboards. The ALC888 is handled by the ALC883-specific code. With this chipset I am having trouble capturing from the line-in and mic input. I can only capture from the line-in. And even with this the signal is distorted. Please try the 1.0.15rc1 driver, there have been many changes in the HDA code since 1.0.14. If the driver doesn't know your motherboard, it is possible that you have more success if you load the driver with one of the model options. (See ALSA-Configuration.txt for a list of supported ALC883 models.) HTH Clemens - This SF.net email is sponsored by: Splunk Inc. Still grepping through log files to find problems? Stop. Now Search log events and configuration files using AJAX and a browser. Download your FREE copy of Splunk now http://get.splunk.com/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Only one capture source??
Daniel Porres wrote: [EMAIL PROTECTED]:/proc/asound/card0$ cat codec#0 Codec: Conexant CX20551 (Waikiki) ... Try loading the snd-hda-intel driver with the model=laptop-eapd option. (This model should be automatically used for Toshiba P100 laptops, but I guess you have a different one.) HTH Clemens - This SF.net email is sponsored by: Splunk Inc. Still grepping through log files to find problems? Stop. Now Search log events and configuration files using AJAX and a browser. Download your FREE copy of Splunk now http://get.splunk.com/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] snd-usb-audio and the Actiontec Internet Phone Wizard
TheOneKEA wrote: On 8/30/07, Clemens Ladisch [EMAIL PROTECTED] wrote: It does have an extra interface that is independent of the three audio interfaces. I guess the driver uses that one to send vendor-specific control messages. So what else could be tried? I would really like to get this device working in Linux, if at all possible. You could ask the manufacturer for programming information and/or the driver source code, but this is unlikely to have any result. You could reverse engineer the protocol used by the Windows driver, if you know how to do it. Regards, Clemens - This SF.net email is sponsored by: Splunk Inc. Still grepping through log files to find problems? Stop. Now Search log events and configuration files using AJAX and a browser. Download your FREE copy of Splunk now http://get.splunk.com/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Edirol UA-101 help.
Mu Zike!! wrote: I am new here. I upgraded my ALSA drivers about a month ago to get support for my Edirol soundcard. I would think it is not so well supported so I am offering for everything you could need to test it, or get information etc, for improving the driver for this card. The model is Edirol UA-101. What we'd need is developer time. Regards, Clemens - This SF.net email is sponsored by: Splunk Inc. Still grepping through log files to find problems? Stop. Now Search log events and configuration files using AJAX and a browser. Download your FREE copy of Splunk now http://get.splunk.com/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Best USB based XLR3 interface to buy?
James Shatto wrote: The few vendor:device numbers I've researched yielded no entries in pciids.sf.net. USB IDs are completely different from PCI IDs. There is a USB device list at http://www.qbik.ch/usb/devices/. HTH Clemens - This SF.net email is sponsored by: Splunk Inc. Still grepping through log files to find problems? Stop. Now Search log events and configuration files using AJAX and a browser. Download your FREE copy of Splunk now http://get.splunk.com/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] XLR3 interfaces?
James Shatto wrote: Is there a list of still produced USB based XLR3 microphone interfaces that actually work with linux/alsa? Tascam US-122L, US-144 I think the newer models are not supported. Digidesign mBox2 The mBox doesn't work. M-Audio Fast Track Pro Has issues. Lexicon Omega, Alpha, Lambda Alesis iO|2 SoundTech Lightsnake Unknown. They should work if they run on Windows or OS X without any separate driver, the magic words are class compliant. I'd like to do good quality (24 bit, 96kHz or better) recording on my laptop. I don't have firewire, and would like something portable. And need something that works (hopefully fully) in linux. 20Hz-20kHz response, minimum 96kHz stereo recording capabilities. Have a look at the Edirol UA-25. It is definitely supported. Preferably USB 2.0 based. You don't need high speed for 24 bits at 96 kHz, but full duplex won't work. There is no supported high speed device except the SB Audigy 2 NX. HTH Clemens - This SF.net email is sponsored by: Splunk Inc. Still grepping through log files to find problems? Stop. Now Search log events and configuration files using AJAX and a browser. Download your FREE copy of Splunk now http://get.splunk.com/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] alsa buffer size
paul blakeley wrote: I have successfully increased the amount of buffer by writing a new value into /proc/asound/cardX/pcm*/sub*/prealloc. For a mono .wav file I was able to successfully change this from 64 to 512. I was really happy with this. I presume this value is in the units of kBytes? Yes. And that this value needs to be 2 to the n. ie 16,32,64 etc? No, it can be any value supported by the driver. Dont know why but I couldn't change this to 512, do you? Because the driver's upper limit for the buffer size is 512 KB. Regards, Clemens - This SF.net email is sponsored by: Splunk Inc. Still grepping through log files to find problems? Stop. Now Search log events and configuration files using AJAX and a browser. Download your FREE copy of Splunk now http://get.splunk.com/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] alsa buffer size
paul blakeley wrote: I am writing an application that is generic enough to a number of different drivers. If thats possible! I am trying to achieve a 2 second of buffer for a number of different sample rates. This means that for a sample rate of 44.1KHz I require a buffer size of 88.2K frames For all the different types of sound cards that I have used I havent seen a buffer size 16K frames : ( I have noticed that various drivers have certain behaviour which makes writing a application that is generic for all drivers very difficult. For example for an intel chipset I have found that the driver was very versatile and allowed me to choose various period sizes for different buffer sizes. Whilst the driver for the CS4299 chipset, although allows a buffer size of upto 16K the period size has to be half that value. The CS4299 is just an AC'97 codec; the buffer constraints depend on the controller and/or the controller driver. Therefore this makes it impossible for me to use with the ALSA callback mechanism, which I need to use for the design to work! ALSA cannot guarantee that any specific buffer/period size combination is available on all sound cards. Your program must be able to work with any value. There are several sound cards that require exactly two periods per buffer. Why does your design need more? By the way how do I find out which driver is assigned to a chipset? /proc/asound/modules Can a driver be used on multiple chipsets? Yes, there are driver that can control several similar chipsets. Regards, Clemens - This SF.net email is sponsored by: Splunk Inc. Still grepping through log files to find problems? Stop. Now Search log events and configuration files using AJAX and a browser. Download your FREE copy of Splunk now http://get.splunk.com/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] snd-usb-audio and the Actiontec Internet Phone Wizard
TheOneKEA wrote: Was the lsusb output helpful in determining if the device does need a separate telephony driver? It does have an extra interface that is independent of the three audio interfaces. I guess the driver uses that one to send vendor-specific control messages. Regards, Clemens - This SF.net email is sponsored by: Splunk Inc. Still grepping through log files to find problems? Stop. Now Search log events and configuration files using AJAX and a browser. Download your FREE copy of Splunk now http://get.splunk.com/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Mixer problem with cmipci
Jan-Benedict Glaw wrote: On Tue, 2007-08-28 08:28:39 +0200, Clemens Ladisch [EMAIL PROTECTED] wrote: Jan-Benedict Glaw wrote: [EMAIL PROTECTED]:~$ amixer cset numid=35 off numid=35,iface=MIXER,name='IEC958 In Monitor' ; type=BOOLEAN,access=rw--,values=1 : values=on Seems it cannot change this control's value? Strange. Please show the contents of /proc/asounc/card0/cmipci. Since I just bought that card and never tested it somewhere else: Could this just be a hardware problem / defect? From a look at the driver's sources, it seems that this mixer element directly maps to the low bit of CM_REG_MIXER1. Indeed. (This is why I asked for the cmipci file and not the cards file, to see how the mixer control affects this bit in register 0x24.) Please unload the snd-cmipci module, then remove (or rename) the /etc/asound.state file, then reboot. This should reset the card without changing this mixer control. If it is still set (and stuck), then it looks like some hardware problem. (It would still be a good idea to test with the Windows driver.) Regards, Clemens - This SF.net email is sponsored by: Splunk Inc. Still grepping through log files to find problems? Stop. Now Search log events and configuration files using AJAX and a browser. Download your FREE copy of Splunk now http://get.splunk.com/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
