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
--
CONFIDENTIALITY CAUTION
*** DISCLAIMER ***
This e-mail contains public information intended for any subscriber of
this mailing list and for anybody else who bothers to read it; it will
be copied, disclosed and distributed to the public. If you think you
are not the intended recipient, please commit suicide immediately.
These terms apply also to any e-mails quoted in, referenced from, or
answering this e-mail, and supersede any disclaimers in those e-mails.
Additionally, disclaimers in those e-mails will incur legal processing
fees of $42 per line; you have agreed to this by reading this
disclaimer.
*** END OF DISCLAIMER ***
-
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
[Alsa-user] asound.conf
Hi, I started to play with ALSA 2 days ago and Im trying to figure out how
ALSA works and how to add PCM devices in the configuration file.
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
}
ctl.jcb-in-1 {
type hw
card 0
}
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?
Also is there any good documentation about how the asound.conf file works? I
found some stuff on the web but it rather sucks.
Thank you!
CONFIDENTIALITY CAUTION
This e-mail and any attachments may be confidential or legally privileged. If
you received this message in error or are not the intended recipient, you
should destroy the e-mail message and any attachments or copies, and you are
prohibited from retaining, distributing, disclosing or using any information
contained herein. Please inform us of the erroneous delivery by return e-mail.
Thank you for your cooperation.
DOCUMENT CONFIDENTIEL
Le présent courriel et tout fichier joint à celui-ci peuvent contenir des
renseignements confidentiels ou privilégiés. Si cet envoi ne s'adresse pas à
vous ou si vous l'avez reçu par erreur, vous devez l'effacer. Vous ne pouvez
conserver, distribuer, communiquer ou utiliser les renseignements qu'il
contient. Nous vous prions de nous signaler l'erreur par courriel. Merci de
votre collaboration.
-
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
[Alsa-user] asound.conf on embedded
Hello all, I'm developing an embedded system and I would like to do some sound stuff. The target has 3 inputs and 4 outputs. Now I want mix 2 inputs with a wav-file and play back the result at 2 outputs. The other input should be playback at the to other outputs. The driver handles all channels as mono. I use the AD1939 and my soundcard doesn't support harware-mixing. I tried to work with the dsnoop and the dmix plugins, but my result is not correctly . Maybe I have to work with ttables and slaves, too? Have someone a cause for thought to my asound.conf (or an example)? Kind regards, Miriam - 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
[Alsa-user] asound.conf for via82xx
I can't figure it out how to get 5.1 sound working on my system. Could please somebody send me his asound.conf? Thanks, Dennis --- Using Tomcat but need to do more? Need to support web services, security? Get stuff done quickly with pre-integrated technology to make your job easier Download IBM WebSphere Application Server v.1.0.1 based on Apache Geronimo http://sel.as-us.falkag.net/sel?cmd=lnkkid=120709bid=263057dat=121642 ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] asound.conf for via82xx
On Fri, 2006-04-21 at 09:59 +0200, Dennis Heuer wrote: I can't figure it out how to get 5.1 sound working on my system. Could please somebody send me his asound.conf? You don't need an asound.conf for this, just use the plug:surround51 device --- Using Tomcat but need to do more? Need to support web services, security? Get stuff done quickly with pre-integrated technology to make your job easier Download IBM WebSphere Application Server v.1.0.1 based on Apache Geronimo http://sel.as-us.falkag.net/sel?cmd=lnkkid=120709bid=263057dat=121642 ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] asound.conf for via82xx
Please again for a beginner. Do you mean with aplay? I don't use it, it doesn't play ogg files, and xine doesn't use it too. I need out the box 5.1 support. Dennis On Fri, 21 Apr 2006 10:53:24 -0400 Lee Revell [EMAIL PROTECTED] wrote: On Fri, 2006-04-21 at 09:59 +0200, Dennis Heuer wrote: I can't figure it out how to get 5.1 sound working on my system. Could please somebody send me his asound.conf? You don't need an asound.conf for this, just use the plug:surround51 device --- Using Tomcat but need to do more? Need to support web services, security? Get stuff done quickly with pre-integrated technology to make your job easier Download IBM WebSphere Application Server v.1.0.1 based on Apache Geronimo http://sel.as-us.falkag.net/sel?cmd=lnkkid=120709bid=263057dat=121642 ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user --- Using Tomcat but need to do more? Need to support web services, security? Get stuff done quickly with pre-integrated technology to make your job easier Download IBM WebSphere Application Server v.1.0.1 based on Apache Geronimo http://sel.as-us.falkag.net/sel?cmd=lnkkid=120709bid=263057dat=121642 ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] asound.conf for via82xx
On Fri, 21 Apr 2006 11:31:19 -0400
Lee Revell [EMAIL PROTECTED] wrote:
Then configure xine to use the plug:surround51. It should have a dialog
to let you select the sound device.
Xine seems to do it. At least if I unmute spread and choose shared
for surround in alsamixer. The quality is not impressing though (It
sounds like the both front channels are spreaded over to the other
channels. The subwoofer rarely creates at least the smallest woof. Some
music, mainly folk, is played very clear but from distance, and the rest
seems to miss some frequency-ranges.)
Please tell us exactly what you are trying to do. Are you using actual
5.1 sources like DVDs, or do you want to play stereo sources and have
them upmixed to 5.1?
Lee
I tried both. There's already an unanswered mail from me here in the
archive. It's called 5.1 not working well with via82xx and sblive 5.1.
I also tried examples and demo files (ac3, wav) from the net with the
mentioned programs (xine, aplay, mpg321). The asound.conf examples
mentioned in the docs in the alsa-tools/alsa-utils-packages
don't work too. For example:
pcm.upmix51 {
type upmix
slave.pcm upmix51
channels 6
}
doesn't change anything--at least if I can trust my ears and when I
look into alsamixer. And:
pcm.!default plug:upmix51
gets rejected because the plug is not valid for pcm default:
ALSA lib pcm.c:2017:(snd_pcm_open_conf) Invalid type for PCM default
definition (id: default, value: plug:upmix51)
Really, alsa is anything else than intuitive. Alsamixer doesn't reflect
changes to the configuration--not even by renaming the channels. You
should start to think about profiles and use-cases and distribute good
settings and easy setups for the card types.
Dennis
---
Using Tomcat but need to do more? Need to support web services, security?
Get stuff done quickly with pre-integrated technology to make your job easier
Download IBM WebSphere Application Server v.1.0.1 based on Apache Geronimo
http://sel.as-us.falkag.net/sel?cmd=lnkkid=120709bid=263057dat=121642
___
Alsa-user mailing list
Alsa-user@lists.sourceforge.net
https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] asound.conf for via82xx
On Sat, Apr 22, 2006 at 05:17:38AM +0200, Dennis Heuer wrote: Really, alsa is anything else than intuitive. Alsamixer doesn't reflect changes to the configuration--not even by renaming the channels. You should start to think about profiles and use-cases and distribute good settings and easy setups for the card types. Dennis Hi Dennis, The VIA82xx is not exactly high-end hardware, AFAIK. And there is the problem that not all manufacturers provide full information needed for those who are willing to write the low-level drivers for Linux/Alsa. The low-level drivers, you must understand, are specific to each sound card. Their quality (or lack) for certain cards does not define the quality of ALSA. Maybe I am oversensitive, but I think I read a tone of expectation or judgment into your words. You have to remember that help from the mailing list is a gift. The drivers are written by those who have time, inclination and/or corporate sponsorship. I happen to have a VIA82xx on my motherboard. It works fine for stereo with no special options. Some people are reporting particular options for the VIA82xx driver here: http://www.alsa-project.org/alsa-doc/doc-php/notes-full.php There is a 'dxs_support=4' option and a 'ac97_quirk' option that you can try. Much of computer twigging (and life!) is trial-and-error. Otherwise, if you need the best possible drivers and support available for 5.1 surround under VIA82xx, you may have to go over to the dark side. If you want great sound under Linux, welcome! So do we!! You might want to choose specific hardware for that purpose. You may need to search through a few web pages to find something for your application/budget. This article from 2003 recommends SoundBlaster Live! and SB16PCI cards as cheap and well supported. http://www.findarticles.com/p/articles/mi_zdext/is_200308/ai_ziff49207#continue Good luck to you. Be polite and you'll be surprised how much help you get. -- Joel Roth --- Using Tomcat but need to do more? Need to support web services, security? Get stuff done quickly with pre-integrated technology to make your job easier Download IBM WebSphere Application Server v.1.0.1 based on Apache Geronimo http://sel.as-us.falkag.net/sel?cmd=lnkkid=120709bid=263057dat=121642 ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] asound.conf for via82xx
Hello Joel. Am not shure if you are *too* oversensitive. Otherwise you wouldn't have written this piece. I already answered to a lot of it. For example, the dxs and ac97_quirks options. Also, I'm talking about 5.1 and not about stereo. Third, you seem to misuse valid arguments for the wrong case: arguing for your attitude. Nothing of what you wrote argues against what I've written in the exclamated paragraph. The alsa product strategy is independent from low-level issues like you mentioned them. It is mainly a matter of how the developers look at their baby (if it's for geeks and experts or a default solution for out-of-the-box audio services). Remember that alsa is the preferred solution for linux and that OSS is expected to be dropped. There is some responsibility you've taken over with this decision. If you refuse to carry this responsibility, as you seem to argue with this mailinglist being a 'gift' (am really not shure, because one doesn't get usable response), the linux guys should rethink their decision and drop alsa from the default branch. So much for the flaming. Now please rethink my paragraph because it contains some truth valid for all serious IT-projects. Dennis On Sat, 22 Apr 2006 13:34:29 +0900 Joel Roth [EMAIL PROTECTED] wrote: On Sat, Apr 22, 2006 at 05:17:38AM +0200, Dennis Heuer wrote: Really, alsa is anything else than intuitive. Alsamixer doesn't reflect changes to the configuration--not even by renaming the channels. You should start to think about profiles and use-cases and distribute good settings and easy setups for the card types. Dennis Hi Dennis, The VIA82xx is not exactly high-end hardware, AFAIK. And there is the problem that not all manufacturers provide full information needed for those who are willing to write the low-level drivers for Linux/Alsa. The low-level drivers, you must understand, are specific to each sound card. Their quality (or lack) for certain cards does not define the quality of ALSA. Maybe I am oversensitive, but I think I read a tone of expectation or judgment into your words. You have to remember that help from the mailing list is a gift. The drivers are written by those who have time, inclination and/or corporate sponsorship. I happen to have a VIA82xx on my motherboard. It works fine for stereo with no special options. Some people are reporting particular options for the VIA82xx driver here: http://www.alsa-project.org/alsa-doc/doc-php/notes-full.php There is a 'dxs_support=4' option and a 'ac97_quirk' option that you can try. Much of computer twigging (and life!) is trial-and-error. Otherwise, if you need the best possible drivers and support available for 5.1 surround under VIA82xx, you may have to go over to the dark side. If you want great sound under Linux, welcome! So do we!! You might want to choose specific hardware for that purpose. You may need to search through a few web pages to find something for your application/budget. This article from 2003 recommends SoundBlaster Live! and SB16PCI cards as cheap and well supported. http://www.findarticles.com/p/articles/mi_zdext/is_200308/ai_ziff49207#continue Good luck to you. Be polite and you'll be surprised how much help you get. -- Joel Roth --- Using Tomcat but need to do more? Need to support web services, security? Get stuff done quickly with pre-integrated technology to make your job easier Download IBM WebSphere Application Server v.1.0.1 based on Apache Geronimo http://sel.as-us.falkag.net/sel?cmd=lnkkid=120709bid=263057dat=121642 ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user --- Using Tomcat but need to do more? Need to support web services, security? Get stuff done quickly with pre-integrated technology to make your job easier Download IBM WebSphere Application Server v.1.0.1 based on Apache Geronimo http://sel.as-us.falkag.net/sel?cmd=lnkkid=120709bid=263057dat=121642 ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
[Alsa-user] asound.conf and mixing for the aureon space
Hi,
I have an Auron Space card 7.1 witch uses the ice_1724 driver.
Sound is working with most application, but now I wanted
all 8 channels to work, so searched around and found a config
on this list. I get sound from all 8 channels. Now I wanted applications
to produce sound simultaniously and understood that the dmix plug
is just made for that.
Thus I tried to learn how plugins worked and how to set them in
asound.conf, but I just can't grasp the logic of it. It's not comprehensive.
The documents I try to read jump in at a technical level.
I thought I just give it a shot and try, here is the asound.conf:
pcm.surround71 {
type plug
slave {
pcm hw:0,0
channels 8
}
route_policy duplicate
ttable.0.0 1
ttable.1.1 1
ttable.0.2 1
ttable.1.3 1
ttable.0.4 1
ttable.1.5 1
ttable.0.6 1
ttable.1.7 1
}
pcm_slave.ice1724_S32_LE {
pcm surround71;
format S32_LE;
}
pcm.convert {
type plug;
slave ice1724_S32_LE;
}
#-
pcm.!default {
type plug
slave.pcm dmixer
}
pcm.dmixer {
type dmix
ipc_key 1024
slave {
slave.pcm convert
period_time 0
period_size 1024
buffer_size 4096
rate 44100
}
bindings {
0 0
1 1
}
}
ctl.dmixer {
type hw
card 0
}
My understanding from this config is that alsa aware applications always
write to the default device when started without any parameters. And that
the default device is hard coded in ALSA lib. But when preceding a virtual
device (in asound.conf) with an ! this gets overidden and the applications
use the virtual one. In my situations this is a dmix plug (or if i'am
correct,
the slave pcm of the !default pcm is the mixer?). So my logic tells me
that sound gets mixed and passed to slave pcm convert, witch will call
slave pcm ice1724_S32_LE, that will call pcm surround71 that finally
will
pass the audio stream to all 8 channels. Am I correct? Guess not because
there is no noise with this config.
It works with one of the to config blocks (dmix or surround71), only
the dmix gives me crackling sound, ticking. Because of the buffersize?
Anyhow, I gave up and so my question to the more technical users that
visit this list. What am I doing wrong here (or tell me that I messing
things up and should be better of with Windows(tm)).
I would appreciate a human understandable explenation on how this
works (the big picture would be fine).
Thanks in advance,
rel
_
MSN Zoeken helpt je om de gekste dingen te vinden! http://search.msn.nl
---
This SF.Net email is sponsored by: IBM Linux Tutorials
Free Linux tutorial presented by Daniel Robbins, President and CEO of
GenToo technologies. Learn everything from fundamentals to system
administration.http://ads.osdn.com/?ad_id=1470alloc_id=3638op=click
___
Alsa-user mailing list
[EMAIL PROTECTED]
https://lists.sourceforge.net/lists/listinfo/alsa-user
[Alsa-user] asound.conf for RME digi96/8
Hi. I have to use a RME Digi96/8 PTS. Right now I can use device 0, but when I try to use device 1 (Digi96 ADAT) i got the following error: # aplay -D hw:0,1 dlg1.wav Playing WAVE 'dlg1.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo aplay: set_params:791: Channels count non available What's wrong. How do I set up the ADAT output on my card?? Best regards Jesper Krogh Aalborg University Dept. of Acoustics --- This sf.net email is sponsored by:ThinkGeek Welcome to geek heaven. http://thinkgeek.com/sf ___ Alsa-user mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] asound.conf for RME digi96/8
On Sun, 10 Nov 2002, Jesper Krogh Christensen wrote: Hi. I have to use a RME Digi96/8 PTS. Right now I can use device 0, but when I try to use device 1 (Digi96 ADAT) i got the following error: # aplay -D hw:0,1 dlg1.wav Playing WAVE 'dlg1.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo aplay: set_params:791: Channels count non available What's wrong. How do I set up the ADAT output on my card?? The DAT device is probably expecting a stream with 8 channel. Use '-D plughw:0,1' to allow automatic conversion from 2 channel stream to 8 channels. Jaroslav - Jaroslav Kysela [EMAIL PROTECTED] Linux Kernel Sound Maintainer ALSA Project, SuSE Labs --- This sf.net email is sponsored by:ThinkGeek Welcome to geek heaven. http://thinkgeek.com/sf ___ Alsa-user mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] asound.conf for RME digi96/8
Thanks - it solved my problem :-) /Jesper On Sun, 10 Nov 2002, Jaroslav Kysela wrote: On Sun, 10 Nov 2002, Jesper Krogh Christensen wrote: Hi. I have to use a RME Digi96/8 PTS. Right now I can use device 0, but when I try to use device 1 (Digi96 ADAT) i got the following error: # aplay -D hw:0,1 dlg1.wav Playing WAVE 'dlg1.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo aplay: set_params:791: Channels count non available What's wrong. How do I set up the ADAT output on my card?? The DAT device is probably expecting a stream with 8 channel. Use '-D plughw:0,1' to allow automatic conversion from 2 channel stream to 8 channels. Jaroslav - Jaroslav Kysela [EMAIL PROTECTED] Linux Kernel Sound Maintainer ALSA Project, SuSE Labs --- This sf.net email is sponsored by:ThinkGeek Welcome to geek heaven. http://thinkgeek.com/sf ___ Alsa-user mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-user --- This sf.net email is sponsored by:ThinkGeek Welcome to geek heaven. http://thinkgeek.com/sf ___ Alsa-user mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-user
[Alsa-user] asound.conf pcm.tee, anyone know how to use it
Please someone correct me if I am mistaken. Can the pcm.tee option in alsa.con (/usr/share/alsa/alsa.conf) be setup so that the output of a program that uses alsa natively be monitored by connecting to this tee ?? My primary app (eXtace) needs this sort of functionality, The old ALSA 0.5.x called it loopback, can this do the same? Can anyone offer tips for someone who really doesn't understand the asound.conf syntax to possibly get it to work? Thanks for any input... = Dave J. Andruczyk ERP __ Do You Yahoo!? LAUNCH - Your Yahoo! Music Experience http://launch.yahoo.com ___ Have big pipes? SourceForge.net is looking for download mirrors. We supply the hardware. You get the recognition. Email Us: [EMAIL PROTECTED] ___ Alsa-user mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-user
[Alsa-user] asound.conf
hi i just installed alsa 0.9.0 i got a delta44 and sblive platinum. when i try and do anything i get the error asound.conf may be old or corrupted ive looked at the file and decided it looks to hard to hack. is there a tool to generate this file or anywhere i can get help on building one??? -cheers eddy = 1.4 ATHLON GA7-DXR Gigbyte M/Board 512MB Crucial DDR NVidia Ge-Force 3. Linux Debian : http://www.debian.com Linux Mandrake : http://www.mandrake.com __ Do You Yahoo!? Everything you'll ever need on one web page from News and Sport to Email and Music Charts http://uk.my.yahoo.com ___ Alsa-user mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-user
