I had done some search on the internet about using select(2) with alsa and
found the function
snd_pcm_file_descriptor( *snd_pcm_handle_t).
However when I try to test it with a simple program, I got compliation error
of fundefined reference.
I also try to lokk it up in the library documentation bu
At Tue, 6 Jan 2004 22:38:28 +0100,
Florian Schmidt wrote:
>
> On Tue, 6 Jan 2004 11:05:52 -0600 (CST)
> David Lloyd <[EMAIL PROTECTED]> wrote:
>
> > This patch works great - I'm now able to use TeamSpeak (which opens
> > capture and playback through oss) at the same time as xmms.
> >
>
> Cool.
On Wed, 07 Jan 2004 12:01:02 +0100
Takashi Iwai <[EMAIL PROTECTED]> wrote:
> as written in my previous mail, it was a quick hack. and there is a
> better approach.
>
> the attached patch will add a new plugin, asym, which defines
> different slave pcms for playback and capture streams.
> for exa
Alex Lau wrote:
>I had done some search on the internet about using select(2) with
>alsa and found the function snd_pcm_file_descriptor(
>*snd_pcm_handle_t). However when I try to test it with a simple
>program, I got compliation error of fundefined reference. I also try
>to lokk it up in the libr
Takashi Iwai wrote:
as written in my previous mail, it was a quick hack. and there is a
better approach.
I'd prefer the following approach:
pcmp.duplex {
type dmix
...
}
pcmc.duplex {
type dsnoop
...
}
with the following policy:
- if capture is specified pcmc and
On Wed, 7 Jan 2004 13:05:30 +0100
Florian Schmidt <[EMAIL PROTECTED]> wrote:
> I think of this plugin as making a full duplex pcm out of two half
> duplex pcm. So why not call the plugin "duplexer"?
Btw: This plugin reminds me somewhat of the "multi" plugin since it,
too, combines several pcm's
Title: Message
Hi,
Sorry if this has
been mentioned before,
I tried 1.0.0rc1
with 2.4.19 kernel on my VIA 8233 onboard sound.
Seems to work fine
mostly, except aRTS (KDE) when starts up generates a horrible squealy noise and
barfs out with a CPU overload message after about 10/15 seconds or
At Wed, 07 Jan 2004 13:45:05 +0100,
Abramo Bagnara wrote:
>
> Takashi Iwai wrote:
>
> > as written in my previous mail, it was a quick hack. and there is a
> > better approach.
>
> I'd prefer the following approach:
>
> pcmp.duplex {
> type dmix
> ...
> }
>
> pcmc.duplex {
>
At Wed, 7 Jan 2004 14:06:22 +0100,
Florian Schmidt wrote:
>
> On Wed, 7 Jan 2004 13:05:30 +0100
> Florian Schmidt <[EMAIL PROTECTED]> wrote:
>
> > I think of this plugin as making a full duplex pcm out of two half
> > duplex pcm. So why not call the plugin "duplexer"?
"duplex" was my first idea,
Dear ALSA developers,
i am sorry to disturb you with my own problem, but i would like to ask
you for advice with confirming the number of samples played.
There is the description of the problem:
We have been developping speech synthesis system at our university ans
naturally we need to play the
> Hi, I'm having some problems getting the alsa input into Jack working
> correctly.
>
> The soundcard is an RME96/8 PAD, Alsa from Kernel 2.6 pre9, and Jack
version
> 0.91.1 - I'm firing up Jackd with:
>
> jackd -R -t 2000 -d alsa -d rme96 -r 44100 -p 1024 -n 8 -m -zs -H -M &
>
> ...and I have a
Hi
Sorry about the HTML formatting. :(
Should be fixed now.
Nobody likes all that crap in their mail digests!
Regards,
Ralph
-
We have moved! Tribal Data Solutions (formerly Trackit Systems) has moved to bigger
and better premises at Hawthorn Park, which is two minutes from the junctio
Takashi Iwai wrote:
At Wed, 07 Jan 2004 13:45:05 +0100,
Abramo Bagnara wrote:
Takashi Iwai wrote:
as written in my previous mail, it was a quick hack. and there is a
better approach.
I'd prefer the following approach:
pcmp.duplex {
type dmix
...
}
pcmc.duplex {
type dsno
At Wed, 07 Jan 2004 15:39:14 +0100,
Abramo Bagnara wrote:
>
> Takashi Iwai wrote:
> > At Wed, 07 Jan 2004 13:45:05 +0100,
> > Abramo Bagnara wrote:
> >
> >>Takashi Iwai wrote:
> >>
> >>
> >>>as written in my previous mail, it was a quick hack. and there is a
> >>>better approach.
> >>
> >>I'd pr
Takashi Iwai wrote:
At Wed, 07 Jan 2004 15:39:14 +0100,
Abramo Bagnara wrote:
I don't see your point, can you show me an example of what you mean?
AFAICS the only code that need to be changed is the PCM definition lookup.
there are two parts to be modified, snd_pcm_open_noupdate() and
snd_pcm_sl
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On 07 Jan 2004 14:22 p z wrote :
> Hi,
>
> And maybe both coaxial and optical inputs are on one EMU10k1 input
> marked as "IEC958 Optical".
> This is maybe same for Audigy. Can some one verify this ???
>
YES - this is true !!
So the "IEC958 Coax
At Wed, 07 Jan 2004 16:15:40 +0100,
Abramo Bagnara wrote:
>
> Takashi Iwai wrote:
> > At Wed, 07 Jan 2004 15:39:14 +0100,
> > Abramo Bagnara wrote:
> >
> >>
> >>I don't see your point, can you show me an example of what you mean?
> >>
> >>AFAICS the only code that need to be changed is the PCM de
At Wed, 7 Jan 2004 16:29:48 +0100,
Eckhard Jokisch wrote:
>
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> On 07 Jan 2004 14:22 p z wrote :
> > Hi,
> >
> > And maybe both coaxial and optical inputs are on one EMU10k1 input
> > marked as "IEC958 Optical".
> > This is maybe same for Au
Takashi Iwai wrote:
if i understand the code correctly, for defining a pcmp or pcmc as a
slave pcm, such as,
pcm.foo {
type plug
slave {
pcmp bar_playback
pcmc bar_capture
}
}
snd_pcm_s
Will do!
Also saw some possibly related posts related to the sample rate
converison in OSS support.
I'll try some different sample rates also.
I might have the same problem as here:
http://www.mail-archive.com/[EMAIL PROTECTED]/msg10346.ht
ml
Am running currently on 0.9.2 and this works fine.
M
Hi,
Try Threaded OSS (toss) instead of OSS driver in arts.
Peter Zubaj
http://www.logofun.pobox.sk - urobte radost svojmu telefonu
---
This SF.net email is sponsored by: Perforce Software.
Perforce is t
At Wed, 07 Jan 2004 16:47:53 +0100,
Abramo Bagnara wrote:
>
> Takashi Iwai wrote:
> >
> >
> > if i understand the code correctly, for defining a pcmp or pcmc as a
> > slave pcm, such as,
> >
> > pcm.foo {
> > type plug
> > slave {
> > pcmp bar_pla
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
>
> - "IEC958 Optical Playback Volume" up, "IEC958 Coaxial Playback
> Volume" down
> -> digital-input (from live-drive) to analog out ?
If you mean the output on the rear side of the computer (the green jack )
that's what I did.
>
> - in the a
At Wed, 7 Jan 2004 17:20:27 +0100,
Eckhard Jokisch wrote:
>
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> >
> > - "IEC958 Optical Playback Volume" up, "IEC958 Coaxial Playback
> > Volume" down
> > -> digital-input (from live-drive) to analog out ?
> If you mean the output on the rea
Takashi Iwai wrote:
pcm.foo {
type plug
slave {
pcm bar
}
}
pcmp.bar bar_playback
pcmc.bar bar_capture
i think it's a big restriction.
As a matter of personal preference I very much prefer this latte
At Tue, 06 Jan 2004 12:23:35 +0100,
Dominik 'Aeneas' Schnitzer wrote:
>
> On Mon, 2004-01-05 at 17:51, Takashi Iwai wrote:
> > "Wave XXX Volume" controls should be zero for multi-channel outputs
> > indeed, because these are volumes of signals duplicated from the front
> > channel to surround/cent
At Wed, 07 Jan 2004 17:58:10 +0100,
Abramo Bagnara wrote:
>
> Takashi Iwai wrote:
> >>>
> >>> pcm.foo {
> >>> type plug
> >>> slave {
> >>> pcm bar
> >>> }
> >>> }
> >>> pcmp.bar bar_playback
> >>> pcmc.bar bar_capture
> >>>
> >>>i think it's
I am seeing strange things happen while using the hdsp and dmix plugin
and aplay as a test app.
JACK works perfectly.
I can get sound from aplay if I use this commandline:
aplay -f cd -D plughw audio.wav
Playing WAVE 'audio.wav' : Signed 16 bit Little Endian, Rate 44100 Hz,
Stereo
underrun!!!
On Mit, 2004-01-07 at 18:07, Takashi Iwai wrote:
> the switch of "SB Live Analog/Digital" can be removed if it works for
> you. it was set to make sure the center "analog" jack works.
> (in the recent models, the digital out is shared with the center/lfe
> jack.)
> with my emu10k1 board, i don't
At Wed, 07 Jan 2004 18:50:42 +0100,
Dominik 'Aeneas' Schnitzer wrote:
>
> On Mit, 2004-01-07 at 18:07, Takashi Iwai wrote:
> > the switch of "SB Live Analog/Digital" can be removed if it works for
> > you. it was set to make sure the center "analog" jack works.
> > (in the recent models, the dig
On Wed, 7 Jan 2004, Takashi Iwai wrote:
> At Wed, 7 Jan 2004 17:20:27 +0100,
> Eckhard Jokisch wrote:
> >
> > -BEGIN PGP SIGNED MESSAGE-
> > Hash: SHA1
> >
> > >
> > > - "IEC958 Optical Playback Volume" up, "IEC958 Coaxial Playback
> > > Volume" down
> > > -> digital-input (from live
I had several issues getting alsa to compile and to load correctly,
after some debugging and lots of coffee I've been able to fix those.
Included is a unified diff so you all can enjoy the work in case you
have the same error.
Kind regards,
Alexander Maassen
diff -burd alsa-driver-1.0.0rc2/acore
I have two warnings with 2.2. The first one is still not fixed:
pci_compat_22.c: In function snd_pci_compat_release_region':
pci_compat_22.c:384: warning: implicit declaration of function
release_mem_region'
... and the second one needs this fix:
--- pcm_native.c~ Thu Jan 1 22:06:44 20
Hi,
I'm cross posting this from OpenAL, since it may have some relevance to
ALSA.
As some already might know, i'm designing a ALSA - OpenAL interface for
advanced hardware feature support.
As far as i understood, they say that the would prefer to handle any
kind of resource managing and moire sp
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