Thank you so much for the informative and detailed response!
Clemens Ladisch wrote:
> This plugin will not do resampling.
It's a relief to hear that plug, in this situation, will not cause resampling.
>> Specifically, does sample rate conversion still occur anywhere in the
>> signal chain even t
Vincent Yu wrote:
> Will any sample rate conversions occur with this .asoundrc config?
>
> pcm_slave.force_24_bit_no_rate_convert {
> pcm "hw:0,0"
> format S24_3LE # Or other 24-bit format
> rate "unchanged" # Necessary?
> }
>
> pcm.my_new_default {
> type plug
>
Hello all,
I've been scratching my head over this for a few hours, looking
through forums, docs, and finally the alsa-lib git repository's C code
to try to find the answer to my question, but, alas, I feel that I may
need to ask for your help.
Will any sample rate conversions occur with this .aso
Hi there,
I would like to create a TDM stream out. Is it possible to create a
virtual multichannel card which interleaves its outputs to a TDM stream ?
Similar for the input stream ? From a TDM stream to multichannel ?
thanks
Matt
Dmitri Seletski wrote:
> I guess if I resample everything to higher rate, applications who
> suffer from lag will have issues? I think i did it a few months ago,
> but I noticed lag in sound if I used re-sampling.
Try using PulseAudio instead of ALSA's resampler.
Regards,
Clemens
--
On Mon, 23 Mar 2015 09:06:15 +0100
Clemens Ladisch wrote:
> Dmitri Seletski wrote:
> > I read somewhere, that it's impossible to have dual rate PCM's
> > working at the same time.(I tried such approach and failed.)
>
> A DAC outputs a stream at a single rate. So when multiple streams are
> mixe
Dmitri Seletski wrote:
> I read somewhere, that it's impossible to have dual rate PCM's working
> at the same time.(I tried such approach and failed.)
A DAC outputs a stream at a single rate. So when multiple streams are
mixed together, they must have a common rate.
> So does DMIX allows for sev
Hello.
I have 2 questions, but I shall ask second in a separate email.
I have PCI-E Oxygen based card(Asus Xonair DX)
It supports rate up to 192000.
I noticed most Steam games don't work with it with Rate above 48000. I
guess this is unavoidable.
A year ago or so I created config where I created
> On Tue, Oct 29, 2013 at 04:38:00PM -0800, rogerx@gmail.com wrote:
>> On Tue, Oct 29, 2013 at 04:11:38PM -0800, Roger wrote:
>>> On Tue, Oct 29, 2013 at 08:01:35AM +0100, Clemens Ladisch wrote:
>>>Roger wrote:
On Mon, Oct 28, 2013 at 12:53:34PM +0100, Clemens Ladisch wrote:
> Roger wr
> On Tue, Oct 29, 2013 at 04:11:38PM -0800, Roger wrote:
>> On Tue, Oct 29, 2013 at 08:01:35AM +0100, Clemens Ladisch wrote:
>>Roger wrote:
>>> On Mon, Oct 28, 2013 at 12:53:34PM +0100, Clemens Ladisch wrote:
Roger wrote:
> Trying to reduce my left and right channels to 0.5, but am noticin
> On Tue, Oct 29, 2013 at 08:01:35AM +0100, Clemens Ladisch wrote:
>Roger wrote:
>> On Mon, Oct 28, 2013 at 12:53:34PM +0100, Clemens Ladisch wrote:
>>> Roger wrote:
Trying to reduce my left and right channels to 0.5, but am noticing ALSA is
only reading values specified as floating as "0
Roger wrote:
> On Mon, Oct 28, 2013 at 12:53:34PM +0100, Clemens Ladisch wrote:
>> Roger wrote:
>>> Trying to reduce my left and right channels to 0.5, but am noticing ALSA is
>>> only reading values specified as floating as "0".
>>>
>>> The below should reduce the volume slightly on channels left
> On Mon, Oct 28, 2013 at 12:53:34PM +0100, Clemens Ladisch wrote:
>Roger wrote:
>> Trying to reduce my left and right channels to 0.5, but am noticing ALSA is
>> only reading values specified as floating as "0".
>>
>> The below should reduce the volume slightly on channels left (0) and right
>> (
Roger wrote:
> Trying to reduce my left and right channels to 0.5, but am noticing ALSA is
> only reading values specified as floating as "0".
>
> The below should reduce the volume slightly on channels left (0) and right
> (0),
> but it's obvious I'm getting the channel set to zero (0) or off.
>
Trying to reduce my left and right channels to 0.5, but am noticing ALSA is
only reading values specified as floating as "0". (Tried the old syntax as
well.)
So ttable only see integer (INT) values for the third field now? Either 0 or 1
only?
The below should reduce the volume slightly on ch
- Original Message -
From: Clemens Ladisch
Sent: 08/14/12 01:16 PM
To: Dagg Stompler
Subject: Re: [Alsa-user] .asoundrc affects only one card.
Dagg Stompler wrote: > pcm.!default { > type hw > card 0 > } > > ctl.!default
{ > type hw > card 0 > } Replace thi
Dagg Stompler wrote:
> pcm.!default {
> type hw
> card 0
> }
>
> ctl.!default {
> type hw
> card 0
> }
Replace this with:
defaults.pcm.card 0
defaults.ctl.card 0
Regards,
Clemens
--
Live
Hello,
I have a multiseat setup which is missing the separated sound feature, for
that I've bought a usb sound card.
now I have two cards, the O\B one and the usb one.
both cards do work (tested them using mplayer) so I've went to devise a way to
try and assign them, the best idea that I've
Hi all,
I am having a problem that whenever I startup skype, it is much much louder
than my other applications and it almost deafens me. Skype on linux does not
have its own volume control; so I was wondering if there were a simple way to
use my asoundrc to create a virtual output device or pl
Juha Heinanen wrote:
> to define in .asoundrc default ALSA device separately for capture and
> playback. is that possible?
pcm.whatever {
type asym
playback.pcm "plug:bluetooth"
capture.pcm "noise_generator"
}
Regards,
Clemens
---
Clemens Ladisch writes:
> Juha Heinanen wrote:
> > both speaker and mic work fine if i choose ALSA:default.
> > what i would like to achieve is that i can use some other device than
> > default for mic
>
> "default:x", where x is the card number or ID.
sorry that i didn't explain the problem cle
Juha Heinanen wrote:
> both speaker and mic work fine if i choose ALSA:default.
> what i would like to achieve is that i can use some other device than
> default for mic
"default:x", where x is the card number or ID.
Regards,
Clemens
-
Clemens Ladisch writes:
> Don't these applications allow to set the sample rate/format?
clemens,
thanks for your reply.
in one application i can only choose driver/device. the choices for
speaker and mic are:
ALSA:default
ALSA:plughw:0,0
ALSA:other device
if i choose the last option, i can w
Juha Heinanen wrote:
> this kind of arecord command produces good quality output file on my
> laptop:
>
> arecord -D plughw:0,0 -f S32_LE -c 2 /tmp/test-mic.wav
>
> how should my .asoundrc look like in order to get the same effect
> without -D, -f and -c parameters?
This is not possible; for histo
this kind of arecord command produces good quality output file on my
laptop:
arecord -D plughw:0,0 -f S32_LE -c 2 /tmp/test-mic.wav
how should my .asoundrc look like in order to get the same effect
without -D, -f and -c parameters?
i have tried:
pcm.internal
{
type hw
card 0
format
Hi all,
I am having the following ALSA configuration in my .asoundrc file:
pcm_slave.custom_slave {
pcm "hw:0,0"
rate 8000
channels 1
rate 8000
period_time 2
buffer_time 8
}
pcm.custom_pcm {
type plug
slave custom_slave
}
I get an error messa
Le Mon, 21 Dec 2009 16:25:59 +0200,
Iwan Ferreira a écrit :
> >
> > Hi!
> >
> > I'm trying to add a softvol control to act as a master control for all the
> > channels on my sound card. I have a M-Audio Revolution 7.1 soundcard.
> > ALSAMIXER shows PCM, center, LFE, rear and side mixers, but no m
>
> Hi!
>
> I'm trying to add a softvol control to act as a master control for all the
> channels on my sound card. I have a M-Audio Revolution 7.1 soundcard.
> ALSAMIXER shows PCM, center, LFE, rear and side mixers, but no master mixer
> to control all the channels. I'm currently running Ubuntu 9.
Tim Blechmann klingt.org> writes:
>
> hi all,
>
> trying to combine two devices via an asoundrc file to one virtual audio
> interface, i run into an issue, basically getting a huge number of xruns.
>
> the devices are:
> digigram lx6464es, period sizes 32 to 512, 2 to 5 periods, interleaved
>
hi all,
trying to combine two devices via an asoundrc file to one virtual audio
interface, i run into an issue, basically getting a huge number of xruns.
the devices are:
digigram lx6464es, period sizes 32 to 512, 2 to 5 periods, interleaved
samples, 64 channels (1x ethersound)
rme 9652, period s
On Wed, 24 Jun 2009 18:54:50 -0700 (PDT)
"S. Aguinaga" wrote:
> Thank you Giuliano!!
>
> BTW: do you have a recommendation to test/check the maximum output level of
> the Layla3G. The specs say that is capable of 114dB( A weighted) ... I used
> alsamixer to turn up the volume all the way up
300mV(rms) and I would expect this
to be about 6.5-7.5Vrms (max).
// Thank you again
From: Giuliano Pochini
To: S. Aguinaga
Cc: alsa-user@lists.sourceforge.net
Sent: Wednesday, June 24, 2009 4:19:41 PM
Subject: Re: [Alsa-user] .asoundrc for layla3g &&am
On Tue, 23 Jun 2009 16:27:34 -0700 (PDT)
"S. Aguinaga" wrote:
> Hello,
>
> *A wave file plays on the built-in audio "aplay -v waveoutput.wave" without
> issues, but if I route this to my Layla3G card using the command:
> "aplay -v -D layla3g waveoutput.wave" this is the output/error messa
Hello,
*A wave file plays on the built-in audio "aplay -v waveoutput.wave" without
issues, but if I route this to my Layla3G card using the command:
"aplay -v -D layla3g waveoutput.wave" this is the output/error message:
Playing WAVE 'waveoutput.wave' : Signed 16 bit Little Endian, Ra
This might be useful for creating a particlar default device
no matter what cards are loaded or in what order...
% cat /usr/bin/asoundrc
#!/bin/sh
# asoundrc v0.1.0 20090101 ma...@renta.net GPLv3
#
# A simple script to create a particular default audio device regardless
# of what cards are loaded
Hi everyone
I read the documentation about the ~/.asoundrc and /etc/asound.conf
files on the ALSA project page. I did not have either of the two files
in my installation of Ubuntu 8.04. I created both files and rebooted my
system. After the reboot I tried aplay -L to find out if my ~/.asoundrc
Hi Pete!
On 2008.02.11 at 18:53:55 +0100, Pete wrote next:
> thanks for your answer. You're right: jack is simpler to configure. But I
> would like to make a configuration for users, wich is transparent so they
> just can use any alsa-application.
>
> So if anybody know's if there is an solut
On Feb 11, 2008 12:53 PM, Pete <[EMAIL PROTECTED]> wrote:
> Am Montag 11 Februar 2008 17:25 schrieb Lee Revell:
>
> > On Feb 10, 2008 5:08 PM, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
> > > I could use jack, but that's to complicated to handle and need's to much
> > > ressources.
> >
> > JACK i
Am Montag 11 Februar 2008 17:25 schrieb Lee Revell:
> On Feb 10, 2008 5:08 PM, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
> > I could use jack, but that's to complicated to handle and need's to much
> > ressources.
>
> JACK is much, much simpler than .asoundrc, and does not use any more
> resourc
On Feb 10, 2008 5:08 PM, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
> I could use jack, but that's to complicated to handle and need's to much
> ressources.
JACK is much, much simpler than .asoundrc, and does not use any more
resources. Use it.
Lee
Hi all,
I'm new on the list. Since I dont know what's usual here: I administrate the
network of an community radio station and dealing there with streaming and
recording. We are using 64studio mostly.
I've the following problem: I want to stream via ices2 and redord the source
at the same time
Greetings,
I want to make a universal multi mapping. So far I have the following 4
channel version based on
http://www.alsa-project.org/alsa-doc/doc-php/asoundrc.php#jackplug
But it has some shortcomings I'd like to fix, don't know how
Q1: is there any way I can inherit all the @args [ CARD DE
On Mon, 20 Nov 2006 00:40:37 +0100
"J. Pauli" <[EMAIL PROTECTED]> wrote:
> Sergei Steshenko wrote:
> > Just curious - does LPF alone or HPF alone work ?
>
> Yes, sort of. You may need some time to get them working but for me they do.
>
> > Anyway, if you don't insist on .asoundrc implementation,
Sergei Steshenko wrote:
> Just curious - does LPF alone or HPF alone work ?
Yes, sort of. You may need some time to get them working but for me they do.
> Anyway, if you don't insist on .asoundrc implementation,
> you can use any jackd-capable LADSPA host and JACK.
>
> A well known jackd-capable
On Mon, 20 Nov 2006 00:15:23 +0100
"J. Pauli" <[EMAIL PROTECTED]> wrote:
> Hi Frédéric,
>
> I tried exactly the same some time ago. Unfortunately the outcome was
> the same: no sound on both amps. My guess was, that the front and rear
> "output thingy" mute each other when they got initialized, t
Hi Frédéric,
I tried exactly the same some time ago. Unfortunately the outcome was
the same: no sound on both amps. My guess was, that the front and rear
"output thingy" mute each other when they got initialized, that is one
could use either of them but not both. I don't know for sure but the
only
Hi there !
I'm trying to do biamplification (maybe triamplification) under Linux
(FC6 x86_64) with ALSA and an Audigy 1 ES.
I tried the following configuration :
-- Low-pass filter --- rear speakers --->
/
->-multi plugin
\
-- High-pa
hello!
i've here a m-audio 1010lt with an ice1712 chip.
on this card, there is no pcm mixer channel per default, but i think it
should be possible to add on in the .asoundrc file.
the problem is, that the documentation about the ctl.NAME is verry barely.
in the alsa wiki/documentation i only fin
Good morning everyone,
Ok, now here's some output showing that only 2 cards are recognized.
I've done this after removing ~/.asoundrc and restoring modules.conf to
original form.
Have to figure out why the third one doesn't work. I guess it's no
hardware problem, for the cards are almost unuse
Rolf Gassner wrote:
> (@admin: no need to review my mail anymore - I wrote it 2 days earlier
> with the same content, before subscribing to the list :-)
There is no moderator ...
> As I am trying to use 18 channels for wavefield synthesis, I bought 3
> cheap 5.1 soundcards with the C-Media CM87
Hi Rolf,
Rolf Gassner <[EMAIL PROTECTED]> writes:
> Unfortunately, this config-file doesn't work at all, the device "hw:1"
> doesn't seem to exist.
> Can anyone help me with the use of multiple soundcards for playback?
Well, I don't know much about the fancy things in the .asoundrc, but just
for
Hi,
(@admin: no need to review my mail anymore - I wrote it 2 days earlier
with the same content, before subscribing to the list :-)
As I am trying to use 18 channels for wavefield synthesis, I bought 3
cheap 5.1 soundcards with the C-Media CM8738 chip.
I need to configure ALSA to let me use
At Thu, 06 Apr 2006 11:46:19 -0400,
Lee Revell wrote:
>
> On Thu, 2006-04-06 at 10:35 +0300, Andras Lorincz wrote:
> > Hello,
> >
> > I'm using etch and I have installed alsa-base, alsa-oss and
> > alsa-utils. These device files are created at boot:
> >
> > /dev/dsp0 --> PCI card
> > /dev/dsp1
On Thu, 2006-04-06 at 10:35 +0300, Andras Lorincz wrote:
> Hello,
>
> I'm using etch and I have installed alsa-base, alsa-oss and
> alsa-utils. These device files are created at boot:
>
> /dev/dsp0 --> PCI card
> /dev/dsp1 --> Onboard card
> /dev/dsp2 --> TV tuner
>
> I'm using tvtime (which
Hello,
I'm using etch and I have installed alsa-base, alsa-oss and alsa-utils. These device files are created at boot:
/dev/dsp0 --> PCI card
/dev/dsp1 --> Onboard card
/dev/dsp2 --> TV tuner
I'm using tvtime (which is oss application). I usually use the PCI card but sometimes want to use the
On Thu, 2006-03-02 at 23:41 +, Haraldur Jóhannesson wrote:
> Is it possible config .asoundrc with mono dmix pcm
>
> So the following would play the wav file in only left speaker
> aplay -D testpcm monofile.wav
>
> and then config another pcm for the right speaker.
>
> Any ideas about how ca
Is it possible config .asoundrc with mono dmix pcm
So the following would play the wav file in only left speaker
aplay -D testpcm monofile.wav
and then config another pcm for the right speaker.
Any ideas about how can I do?
Best Regards
Halli
---
Hi,
I've got a RME 96/8 PAD, and I try to create a virtual
device with asoundrc, which would allowed me to have
both the stereo device and the adat one at the same
time in JACK.
He's my asoundrc:
pcm.rme_stereo {
type hw
card 0
device 0
}
ctl.rme_stereo {
type hw
Hallo,
Volker Hartmann hat gesagt: // Volker Hartmann wrote:
> can anyone post his .asoundrc for the Terratec DMX 6fire 24/94 with dmix?
the one at http://alsa.opensrc.org/?DmixPlugin could work.
ciao
--
Frank Barknecht _ __footils.org__
---
Hi,
can anyone post his .asoundrc for the Terratec DMX 6fire 24/94 with dmix?
CU Volker
---
This SF.Net email is sponsored by: IBM Linux Tutorials
Free Linux tutorial presented by Daniel Robbins, President and CEO of
GenToo technologies. Learn
On Tue, 27 Jan 2004, John W. Cocula wrote:
> Let me clarify: the real issue is *mixing* normal stereo out the L and R
> channels, while a mono channel is also coming out the center speaker. So,
> like dmix, but dmix only works with the first 2 channels?
Not really. Some hardware (like via82xx, i
Clemens, thanks very much for your advice. But it seems that the dshare
plugin can only slave to an actual hardware device:
---
$ aplay -Dcenter /tmp/foo.wav
ALSA lib pcm_dshare.c:578:(snd_pcm_dshare_open) dshare plugin can be only
connected to hw plugin
Segmentation fault
---
Trying another plu
John W. Cocula wrote:
> Let me clarify: the real issue is *mixing* normal stereo out the L and R
> channels, while a mono channel is also coming out the center speaker. So,
> like dmix, but dmix only works with the first 2 channels? Any solutions
> short of using JACK?
>
> The following fragment
Let me clarify: the real issue is *mixing* normal stereo out the L and R
channels, while a mono channel is also coming out the center speaker. So,
like dmix, but dmix only works with the first 2 channels? Any solutions
short of using JACK?
The following fragment works to drive only the center ch
Honestly, I've really been trying to figure out .asoundrc and plugins, but
you have to admit that the various docs can be terribly confusing.
I have an intel8x0 device, and I can pump a mono WAVE file out L and R
channels, no problem, as you might expect. But what is the magic
incantation to crea
Thus spake Scott Barnes on Thu, Jan 22, 2004 at 10:45:49AM CST
> On Thu, 2004-01-22 at 10:56, Lindsay Haisley wrote:
> > > > * If I could find some way to route gnome sounds to hw:0,1 then I could
> > > >swap channels and all would be OK, since I don't really need a per-app
> > > >volume c
On Thu, 2004-01-22 at 10:56, Lindsay Haisley wrote:
> > > * If I could find some way to route gnome sounds to hw:0,1 then I could
> > >swap channels and all would be OK, since I don't really need a per-app
> > >volume control for gnome event sounds other than the gnome alsa mixer.
> >
> >
Lindsay Haisley wrote:
> The control interface for the front speakers is 'PCM',0, can be assigned to
> an alsa ctl with:
>
> ctl.main {
>type hw
>card 0
> }
>
> ... but I can find no way to link level control for the rear speakers
> defined in amxer as 'PCM',1 and 'Surround'1 with a ctl spe
Thus spake Clemens Ladisch on Thu, Jan 22, 2004 at 09:26:31AM CST
> Lindsay Haisley wrote:
> > The control interface for the front speakers is 'PCM',0, can be assigned to
> > an alsa ctl with:
> >
> > ctl.main {
> >type hw
> >card 0
> > }
> >
> > ... but I can find no way to link level cont
Here's what I want to do. I'm beginning to thing it isn't possible.
I am working with a single card with two stereo outputs, type cs46xx (Turtle
Beach Santa Cruz). I have the "front" speaker output going to a couple of
cheap monitor-mounted speakers and the "rear" speakers going to a nice amp
an
On Thu, Jan 08, 2004 at 03:39:07PM +0100, Jaroslav Kysela wrote:
> Basically, yes. We are able to mmap the DMA ring buffer to more
> applications at one time and mangle the stream parameters to satisfy
> user's requests. The nice thing is that the mangling is done on the user
> level, so the kerne
On Thu, 8 Jan 2004, Alfons Adriaensen wrote:
> On Thu, Jan 08, 2004 at 03:04:03PM +0100, Jaroslav Kysela wrote:
>
> > Oops. My fault. For capture is dsnoop plugin, of course (same syntax) the
> > dhare plugin is for playback only.
> >
> > We have the asym plugin in CVS now, so you can combine d
On Thu, Jan 08, 2004 at 03:04:03PM +0100, Jaroslav Kysela wrote:
> Oops. My fault. For capture is dsnoop plugin, of course (same syntax) the
> dhare plugin is for playback only.
>
> We have the asym plugin in CVS now, so you can combine dshare and dsnoop
> plugins for the full-duplex operation
On Thu, 8 Jan 2004, Alfons Adriaensen wrote:
> On Thu, Jan 08, 2004 at 02:34:50PM +0100, Jaroslav Kysela wrote:
>
> > You need to use the dshare plugin:
> >
> > pcm.in23 {
> > type dshare
> > ipc_key 321456 # any unique value
> > ipc_key_add_uid true
> > slav
On Thu, Jan 08, 2004 at 02:34:50PM +0100, Jaroslav Kysela wrote:
> You need to use the dshare plugin:
>
> pcm.in23 {
> type dshare
> ipc_key 321456 # any unique value
> ipc_key_add_uid true
> slave {
> pcm "hw:0,0"
> periods 0
On Thu, 8 Jan 2004, Julien Claassen wrote:
> Hi all!
> I want to perform a simple task. I have a delta1010lt soundcard. It has 10
> ins. I want to take two ins and combine them in one virtual device. Upto now I
> used the ttable. But I think it's not the best way of doing it. I did
> something l
Hi all!
I want to perform a simple task. I have a delta1010lt soundcard. It has 10
ins. I want to take two ins and combine them in one virtual device. Upto now I
used the ttable. But I think it's not the best way of doing it. I did
something like:
ttable.0.0 2
ttable.0.1 3
I'm not sure abou
On Tue, 9 Dec 2003, Greg Watson wrote:
> Hello all,
>
> I've been searching the docs on ~/.asoundrc and been unable to figure this
> out. I have a SB Audigy2 card and am trying to map three mixer controls into
> one. I would like Wave Center, Wave Surround, and PCM (which is left/right
> out
Hello all,
I've been searching the docs on ~/.asoundrc and been unable to figure this
out. I have a SB Audigy2 card and am trying to map three mixer controls into
one. I would like Wave Center, Wave Surround, and PCM (which is left/right
output) mapped together. So when PCM is changed, they
does anyone know how to make a virtual device in asoundrc that has 4 interleaved
channels. the hammerfall has noninterleaved channels, but most software expects a 4
channel stereo card with interleaving.
thanks
patrick
---
This SF.net email i
Hi,
The question is : How config alsa to have spdif used by xine and mplayer.
My suspicion: I have no controls IEC958 threw "amixer controls" ???
I just installed last dev release 1.0-pre1 (with alsa-driver-1.0-pre2).
Technical data of my hardware and system configurations are at the end of t
Has anyone out there sucessfully configured
asoundrc for
multitrack recording with the SBlive series of
cards?
I have had a go but with no sucess as i don't
really
understand the jackplug capture scene.
A cd of fresh new tunes for anyone who can help
out.
i have been attempting to get my front channel to play
at the same time as my center_lfe channel, and i have
not been able to do it until now.
i wrote the following .asoundrc file, but it only
works if i do: aplay -D three test.wav...
and it does not work with the alsa-xmms plugin. i d
Doh! Found one just after posting. Sorry.
It's here in case anyone else is looking for it in the future.
(This card has more inputs then the 66 so I just deleted the extra
ones).
http://www.alsa-project.org/alsa-doc/doc-php/template.php3?company=Midiman&card=Delta+66&chip=Envy24&module=ice1712#no
Hi,
Can someone show me an example asoundrc for the delta 66 I can't get
mine working (everything works fine using oss emulation so I'm pretty
sure this is were the problem lies) at the moment mine looks like this.
pcm.via {
type hw
card 0
}
ctl.via {
type hw
card 0
}
pcm.ice1712
Hi All
I've been trying to configure an asound file for my Delta66 and Intel8x0
(SIS645DX chipset... yes it works with all functionality including MIDI)
cards.
I have built the file at the bottom based on examples. I would like to
convert it so that both cards are seen as one (purely for ease of
Hi,
is there a way to use the definitions from .asoundrc as a default
setup, including oss emulation, so that you can use ttable routings
or software mixing plugin with applications that only support oss?
thank you
---
This SF.net email is spo
Is there anyone with a working type share setup,
allowing to play soundfiles
simultaniously to different channel.
The documentation, tells me I can't use the same
channel twice, thats fine,
but it seems I can't even use different
channels.
The output is send to the correct channels, just
the
hi,
cause i didnt found any sufficent documentation again i started to wrote a
kind of asoundrc-minihowto. my plan is to put it on the alsa-wiki-page,
but i dont know where the (un)offical alsa-wiki-page is, its not linked
on alsa-project.org. otherwise i will put it 'statically' somwhere on the
Thanks to everyone for the responses. Very sorry for not being clear in the
previous message. I have the Delta 66 working well with beta10. (snd still
won't record for me via inputs 3+4, but that's another story+mailist)
I am however, still confused on a couple of points:
1. can the 'plughw:x,y'
Hi,
I have a rme96/8 card. It works well enough with the alsa drivers (0.9).
It has 2 devices, 0: the spdif output and 1: the ADAT 8 channel output.
What I would like to do is to set it up so it will always use the ADAT
output, routing stereo files to channels 1-2 or 3-4 etc of the ADAT
output.
On Thu, 27 Dec 2001, David Gerard Matthews Jr. wrote:
> Greetings all,
> About a week ago I posted a message to this list asking if anyone had
> an .asoundrc file for the M-Audio Delta 1010 that they could send me. I
> ended up rolling my own, with less than perfect success. When I run
>
Greetings all,
About a week ago I posted a message to this list asking if anyone had
an .asoundrc file for the M-Audio Delta 1010 that they could send me. I
ended up rolling my own, with less than perfect success. When I run
Ardour, I get a message telling me that I am using the "plug la
Greetings all,
I just became the proud owner of a shiny new Delta 1010, and was
wondering if anyone successfully using this card could email me their
.asoundrc file. Yes, I know I could do this myself, but I'm lazy.
Thanks a lot,
dgm
_
I was able to use my midiman Delta44 with ecasound under the
made up name 'midiman'. Here is my .asoundrc:
pcm.ice1712 {
type hw
card 0
device 0
}
pcm.midiman {
type plug
slave.pcm ice1712
}
Then I used -o:alsa,midiman in the ecasound command line.
This
Frank Barknecht wrote:
> One thing missing in the ALSA docs is a clear explanation of the
> asoundrc-system. A Howto for all those plughw, plug, front, surround5,
> you-name-it thingies would be nice. Any takers?
Paul Davis mentioned that he would like to but also said he doesn't have
the time.
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