Re: [Alsa-user] ESI U46DJ

2006-09-14 Thread Lukas Zapletal
Thank you very much, I will order it.

But I have found that Ego Systems Inc = ESI and Audiotrak. I have sold
Audiotrak MAYA 44USB a week ago because ALSA was not able to detect
its mixer properly. I worry if this soundcard (produced by the same
vendor) have the same problem.

I will try it and report here. Thanks.

2006/9/14, Lukas Zapletal <[EMAIL PROTECTED]>:
> Thank you very much, I will order it.
>
> But I have found that Ego Systems Inc = ESI and Audiotrak. I have sold
> Audiotrak MAYA 44USB a week ago because ALSA was not able to detect
> its mixer properly. I worry if this soundcard (produced by the same
> vendor) have the same problem.
>
> I will try it and report here. Thanks.
>
> 2006/9/13, Thomas Kuther <[EMAIL PROTECTED]>:
> > On Tue, 12 Sep 2006 20:51:17 +0200
> > "Lukas Zapletal" <[EMAIL PROTECTED]> wrote:
> >
> > > Hello,
> > >
> > > do this external USB soundcard work in ALSA? I cannot find any "ESI"
> > > soundcard in the supported cards matrix.
> > >
> >
> > It's "Ego Systems Inc" (EgoSys)
> >
> > HTH
> > Tom
> >
> >
> > -
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> >
> >
> >
>
>
> --
> Lukas Zapletal
> http://lukas.zapletalovi.com
>


-- 
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[Alsa-user] what condition is aserver used in?

2006-09-14 Thread littertiger
hi,group
  when I play music using alsa dmix, ps -A cannot list aserver process.
  what conditions is aserver used in?  

BEST RIGARDS
littertiger 



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Re: [Alsa-user] Is the RME Digi9652 supported ? [signed]

2006-09-14 Thread dtamm

 Original-Nachricht 
Datum: Tue, 05 Sep 2006 13:39:06 +0200
Von: Friedrich_Ewaldt <[EMAIL PROTECTED]>
An: Eduard van den Bongard <[EMAIL PROTECTED]>
Betreff: Re: [Alsa-user] Is the RME Digi9652 supported ? [signed]

> on 09/05/2006 12:47 PM, Eduard van den Bongard wrote:
> > Hi,
> > 
> > Good to hear, I will install Suse - do you think the latest version will
> > also work, or should I stick with 9.x - 
> I don't know. 9.3 works here, as did suse 9.0, some old redhat...
> I don't see any distro specific problems if your soundcard already works 
> for stereo plaback. This means the distro currently installed should 
> also work.
> 
> on the box and will try these
> > configs. I'll let you know whether it is working or not. If it's working
> > I'll create a how-to and publish that.
> > 
> > Thanks and Cheers,
> > Eduard
> > 


Hi,

I managed after returning to SuSE that there is at least stereo putput again.
Did anyone succeed in setting up a working .asoundrc for sourround71 oder 
surround51?
I would very much like to have a glimpse at those HOWTOs you mentioned.

Thanks in advance!
Dominik
-- 
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Re: [Alsa-user] emu10k1x inexplicably with no sound

2006-09-14 Thread Paul Moss
I have been trying to always remember to check the mixer whenever I
recompiled or reran any config scripts.  Every playback control is
turned up.

# amixer
Simple mixer control 'Master',0
  Capabilities: pvolume pswitch pswitch-joined
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 31
  Mono:
  Front Left: Playback 24 [77%] [on]
  Front Right: Playback 24 [77%] [on]
Simple mixer control 'Master Mono',0
  Capabilities: pvolume pvolume-joined pswitch pswitch-joined
  Playback channels: Mono
  Limits: Playback 0 - 31
  Mono: Playback 24 [77%] [on]
Simple mixer control '3D Control - Switch',0
  Capabilities: pswitch pswitch-joined
  Playback channels: Mono
  Mono: Playback [on]
Simple mixer control '3D Control Sigmatel - Depth',0
  Capabilities: volume volume-joined
  Playback channels: Mono
  Capture channels: Mono
  Limits: 0 - 3
  Mono: 3 [100%]
Simple mixer control '3D Control Sigmatel - Rear Depth',0
  Capabilities: volume volume-joined
  Playback channels: Mono
  Capture channels: Mono
  Limits: 0 - 3
  Mono: 3 [100%]
Simple mixer control 'PCM',0
  Capabilities: pvolume pswitch pswitch-joined
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 31
  Mono:
  Front Left: Playback 24 [77%] [on]
  Front Right: Playback 24 [77%] [on]
Simple mixer control 'PCM Out Path & Mute',0
  Capabilities: enum
  Items: 'pre 3D' 'post 3D'
  Item0: 'pre 3D'
Simple mixer control 'Surround',0
  Capabilities: pvolume pswitch
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 31
  Mono:
  Front Left: Playback 26 [84%] [on]
  Front Right: Playback 26 [84%] [on]
Simple mixer control 'Line',0
  Capabilities: pvolume pswitch pswitch-joined cswitch cswitch-exclusive
  Capture exclusive group: 0
  Playback channels: Front Left - Front Right
  Capture channels: Front Left - Front Right
  Limits: Playback 0 - 31
  Front Left: Playback 26 [84%] [on] Capture [off]
  Front Right: Playback 26 [84%] [on] Capture [off]
Simple mixer control 'CD',0
  Capabilities: pvolume pswitch pswitch-joined cswitch cswitch-exclusive
  Capture exclusive group: 0
  Playback channels: Front Left - Front Right
  Capture channels: Front Left - Front Right
  Limits: Playback 0 - 31
  Front Left: Playback 26 [84%] [on] Capture [off]
  Front Right: Playback 26 [84%] [on] Capture [off]
Simple mixer control 'Mic',0
  Capabilities: pvolume pvolume-joined pswitch pswitch-joined cswitch
cswitch-exclusive
  Capture exclusive group: 0
  Playback channels: Mono
  Capture channels: Front Left - Front Right
  Limits: Playback 0 - 31
  Mono: Playback 25 [81%] [on]
  Front Left: Capture [on]
  Front Right: Capture [on]
Simple mixer control 'Mic Boost (+20dB)',0
  Capabilities: pswitch pswitch-joined
  Playback channels: Mono
  Mono: Playback [on]
Simple mixer control 'Mic Select',0
  Capabilities: enum
  Items: 'Mic1' 'Mic2'
  Item0: 'Mic1'
Simple mixer control 'Video',0
  Capabilities: pvolume pswitch pswitch-joined cswitch cswitch-exclusive
  Capture exclusive group: 0
  Playback channels: Front Left - Front Right
  Capture channels: Front Left - Front Right
  Limits: Playback 0 - 31
  Front Left: Playback 25 [81%] [on] Capture [off]
  Front Right: Playback 25 [81%] [on] Capture [off]
Simple mixer control 'Phone',0
  Capabilities: pvolume pvolume-joined pswitch pswitch-joined cswitch
cswitch-exclusive
  Capture exclusive group: 0
  Playback channels: Mono
  Capture channels: Front Left - Front Right
  Limits: Playback 0 - 31
  Mono: Playback 12 [39%] [on]
  Front Left: Capture [off]
  Front Right: Capture [off]
Simple mixer control 'PC Speaker',0
  Capabilities: pvolume pvolume-joined pswitch pswitch-joined
  Playback channels: Mono
  Limits: Playback 0 - 15
  Mono: Playback 12 [80%] [on]
Simple mixer control 'Aux',0
  Capabilities: pvolume pswitch pswitch-joined cswitch cswitch-exclusive
  Capture exclusive group: 0
  Playback channels: Front Left - Front Right
  Capture channels: Front Left - Front Right
  Limits: Playback 0 - 31
  Front Left: Playback 24 [77%] [on] Capture [off]
  Front Right: Playback 24 [77%] [on] Capture [off]
Simple mixer control 'Mono Output Select',0
  Capabilities: enum
  Items: 'Mix' 'Mic'
  Item0: 'Mix'
Simple mixer control 'Capture',0
  Capabilities: cvolume cswitch cswitch-joined
  Capture channels: Front Left - Front Right
  Limits: Capture 0 - 15
  Front Left: Capture 0 [0%] [on]
  Front Right: Capture 0 [0%] [on]
Simple mixer control 'Mix',0
  Capabilities: cswitch cswitch-exclusive
  Capture exclusive group: 0
  Capture channels: Front Left - Front Right
  Front Left: Capture [off]
  Front Right: Capture [off]
Simple mixer control 'Mix Mono',0
  Capabilities: cswitch cswitch-exclusive
  Capture exclusive group: 0
  Capture channels: Front Left - Front Right
  Front Left: Capture [off]
  Front Right: Capture [off]
Simple mixer control 'Analog/Digital Output Jack',0
  Capabilities: pswitch pswitch-joined
  Playback channels: Mono
  Mono: Playback [on]
Simple mixer control 'External Ampl

Re: [Alsa-user] Is the RME Digi9652 supported ? [signed]

2006-09-14 Thread Eduard van den Bongard
Hi Dominik,

I have just installed 64studio, and there I used jackd to play a little bit
around. I was able to manage playing something to the outputs I liked, with
managing the outputs with qjackctl. I haven't yet tried different .asoundrc
files. I was not able to play any sound with Suse 10.1. I hope I have spare
time at the weekend to try the .asoundrc files I have.

Cheers,
Eduard



> Von: [EMAIL PROTECTED]
> Datum: Thu, 14 Sep 2006 14:21:01 +0200
> An: Friedrich_Ewaldt <[EMAIL PROTECTED]>, [EMAIL PROTECTED]
> Cc: alsa-user@lists.sourceforge.net
> Betreff: Re: [Alsa-user] Is the RME Digi9652 supported ? [signed]
> 
> 
>  Original-Nachricht 
> Datum: Tue, 05 Sep 2006 13:39:06 +0200
> Von: Friedrich_Ewaldt <[EMAIL PROTECTED]>
> An: Eduard van den Bongard <[EMAIL PROTECTED]>
> Betreff: Re: [Alsa-user] Is the RME Digi9652 supported ? [signed]
> 
>> on 09/05/2006 12:47 PM, Eduard van den Bongard wrote:
>>> Hi,
>>> 
>>> Good to hear, I will install Suse - do you think the latest version will
>>> also work, or should I stick with 9.x -
>> I don't know. 9.3 works here, as did suse 9.0, some old redhat...
>> I don't see any distro specific problems if your soundcard already works
>> for stereo plaback. This means the distro currently installed should
>> also work.
>> 
>> on the box and will try these
>>> configs. I'll let you know whether it is working or not. If it's working
>>> I'll create a how-to and publish that.
>>> 
>>> Thanks and Cheers,
>>> Eduard
>>> 
> 
> 
> Hi,
> 
> I managed after returning to SuSE that there is at least stereo putput again.
> Did anyone succeed in setting up a working .asoundrc for sourround71 oder
> surround51?
> I would very much like to have a glimpse at those HOWTOs you mentioned.
> 
> Thanks in advance!
> Dominik
> -- 
> "Feel free" - 10 GB Mailbox, 100 FreeSMS/Monat ...
> Jetzt GMX TopMail testen: http://www.gmx.net/de/go/topmail
> 


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[Alsa-user] Sound stops anomalous and goes on after some time

2006-09-14 Thread Henrik Niehaus
Hi,
i'm using an asus mainboard with a via 8237 sound chip and coaxial 
s/pdif out.
I'm trying to get the s/pdif working with oss-emulation and dmix. It 
works great and i'm absolutely satisfied with the setup except that the 
sound stops sometimes without any obvious reason. Sometimes it then goes 
on after a while, sometimes i have to restart the computer. I don't know 
if it is a problem with alsa, but maybe someone knows what's going 
wrong. Perhaps i'm using wrong values in the slave section in 
pcm.dmixer; i have copied them from another .asoundrc at the alsa homepage.

Another thing is, that i have /dev/dsp, but programs using this device 
like audacity (or cat /dev/urandom > /dev/dsp) are mute.
Mixer levels are all about 75%

Any ideas? Thanks in advance,

henrik


Here are my .asoundrc and modules.conf (gentoo /etc/modules.d/alsa):

pcm.!default {
type plug
slave.pcm "dmixer"
}

pcm.dmixer {
type dmix
ipc_key 1024# must be unique!
ipc_perm 0660 # sound for everybody (at least in your group)
slave {
pcm "hw:0,0,3"# you cannot use a "plug" device here, darn.
rate 48000
period_time 0
period_size 1024  # must be power of 2
buffer_size 8192  # dito
}
}

pcm.dsp0 {
type plug
slave.pcm "dmixer"
}

ctl.mixer0 {
type hw
card 0
}



alias snd-card-0 snd-via82xx
alias sound-slot-0 snd-via82xx

alias /dev/mixer snd-mixer-oss
alias /dev/dsp snd-pcm-oss
alias /dev/midi snd-seq-oss



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Re: [Alsa-user] Sound stops anomalous and goes on after some time

2006-09-14 Thread Lee Revell
On Thu, 2006-09-14 at 15:52 +0200, Henrik Niehaus wrote:
> Another thing is, that i have /dev/dsp, but programs using this
> device 
> like audacity (or cat /dev/urandom > /dev/dsp) are mute.
> Mixer levels are all about 75%
> 

Don't use an .asoundrc - dmix should be enabled automatically with a
recent ALSA version.

Dmix cannot work with direct acess to /dev/dsp - you must run OSS apps
with the aoss wrapper.

Lee


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Re: [Alsa-user] Sound stops anomalous and goes on after some time

2006-09-14 Thread Henrik Niehaus
Lee Revell schrieb:
> On Thu, 2006-09-14 at 15:52 +0200, Henrik Niehaus wrote:
>   
>> Another thing is, that i have /dev/dsp, but programs using this
>> device 
>> like audacity (or cat /dev/urandom > /dev/dsp) are mute.
>> Mixer levels are all about 75%
>>
>> 
>
> Don't use an .asoundrc - dmix should be enabled automatically with a
> recent ALSA version.
>
> Dmix cannot work with direct acess to /dev/dsp - you must run OSS apps
> with the aoss wrapper.
>
> Lee
>
>
>   
to use no .asoundrc doesn't help me, because without it "s/pdif out" and 
dmix don't work.

the dsp issue: i'm using aoss. for example i'm running firefox with aoss 
to get flash output some sound. but audacity seems to run in an infinity 
loop (100% cpu), if it's started with aoss.

henrik

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[Alsa-user] i810 (ich5) and SPDIF problem - pls help

2006-09-14 Thread Warpme
Hi All,

I'm in the process of building HD ready Myth box.
Hw is ASUS P5P800 (i865 + i810 ICH5).
Software is KnoppMyth R5D1 (debain based distro integrated with mythtv)
Kernel is 2.6.17
Alsa is 1.0.12rc1 (reported via alsamixer) 

My problem is SPDIF out.

My Hw is working OK - it tested SPDIF out under Winblows.

Unfortunately under r5d1 I can't set-up SPDIF to work.

"aplay -l" reports that IEC958 in on card 0, dev 4.

Using
"aplay -D hw:0,4 test.wav" gives nothing (no any change, SPDIF is all 
the time dead).

File is OK, because "aplay -D hw:0,0 test.wav" (this is analog) is 
playing fine.

In alsamixer there is no any setting related to iec958 or spdif - so it looks 
really strange.

Maybe I have change something in modules.conf or
 
How can I debug and troubleshot this problem ?

many thanx in advance !







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Re: [Alsa-user] pcm.tee pluging

2006-09-14 Thread Mario Rossi
I've tried with

pcm.hardware {
type hw
card 0
device 0
}

pcm.file_raw {
type plug
slave {
  pcm "tee:hardware,/home/mario/a.raw"
}
}

so instead of the default (which in my case is dmix) I use the
hardware directly.
And it works: xmms, mplayer & ekiga.
Since I'm using alsa 1.0.11 rc and I've heard that dmix has got a bug,
I will try again after upgrading my system to 1.0.12.

cheers

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Re: [Alsa-user] call for help: owners of M-Audio Revolution 5.1 cards

2006-09-14 Thread Jochen Voss
Hi Sergei,

On Tue, Sep 12, 2006 at 04:10:02PM +0300, Sergei Steshenko wrote:
> > 1) The problem was with the capture volume control.  Left and right
> > input channel were virtually identical, but when I pull down the left
> > capture volume control, the right channel is muted instead.
> 
> Apparently, the controls are swapped, so ALSA developers have to look
> into this.

Swapping them is easy, but it the change seems weird.  The purpose of
my mail was to get independent confirmation that the controls are
swapped.  Maybe I am only confused?

> However, if the card can't record stereo from microphone by design,
> it probably doesn't matter that much.
Depends on whether line-in is also affected.  I currently can't test.

> Does this card (Revo 5.1) support stereo microphone ? What does user
> manual say about it ?
It almost exclusively describes how to install the MS Windows driver :-(

> The microphone probably has two "heads", so tapping on one of them
> should produce loud sound in that channel and silent one in the other
> channel - is this the case ?
On some sound cards this is the case.  On the Revolution 5.1 it isn't.

> I mean, if the card is designed to provide power to electret microphone
> through the 3--terminal 3.5mm jack, it can't support stereo.

Do you have any reference for this?  Near the end of the wikipedia page
http://en.wikipedia.org/wiki/Minijack it reads

Many small video cameras, laptops, Minidisc recoders and other
consumer devices use a 3.5 mm microphone connector for attaching a
(mono/stereo) microphone to the system. These fall into three
categories:

* ...
* ...
* Devices (most consumer equipment) which use a "plug-in powered"
microphone: an electret microphone containing an internal FET
amplifier. These provide a good quality signal, in a very small
microphone. However, the internal FET requires a DC power supply,
which is provided as a bias voltage.

The expression "bias voltage", especially in connection with the
diagram on the page, makes me believe that there is no principal
reason that providing power forces mono recordings.

All the best,
Jochen
-- 
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[Alsa-user] Tascam us428 - inputs 3 and 4 don't work

2006-09-14 Thread gtrfree

Hi all, some infos:



suse 10.1 32bit

alsa 1.0.11-32

alsa-firmware 1.0.10-13

alsa-tools 1.0.11-16

jack 1.101.1 (new 'cause I read there's a better support for us428 in new 
versions)



I load the firmware with usx2yloader, then i run us428control. Everything seems 
fine: 1-2 ins, midi in and out, some Ardour controls work from the tascam...but 
no 'analog' nor 'digital' lights and, expecially, no sound from the 3-4 inputs. 
I tried in oss-mode, too and jack view all the 4 inputs but no way to use them 
in Audacity or Ardour.



It seems a bug (it's not in the bug-list), I've found just this:



http://www.nabble.com/-PATCH--Make-channels-3%2B4-of-US428-work-t243073.html#a679036



It's a post made the past year and seems to be a jack-patch but there's 
something about Alsa, too...sadly I'm a musician and I don't understand much 
about code! :-)

...and if this works with the new Alsa version.



In the Alsa 'changes from past version' section I did't find anything about the 
us428 in the last year and I wonder (if it's a bug) if it's fixed.

Ok, sorry for the long post (and for my ortography) but could someone explain 
me if it's possible to use ins 3-4 of my tascam, please? And how?



Thank you very much for every help.



Cheers.

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Re: [Alsa-user] snd-hda-intel problems

2006-09-14 Thread Andrew Mahone
I've done a little more experimenting. It does not reliably alternate
between the working and non-working states with module reloads, that
was a fluke. It seems to be random, and sadly it also seem to favor
the non-working state. I've compare the contents of /proc/asound in
both states, and nothing differs there - in particular, codec#0 has
exactly the same contents, whether the module load was "good" and
gives me sound, or is "bad" and renders my computer silent. For now, i
just restart the alsasound init script until my sound works.

The only difference I can find is that when my sound does *not* work,
the mixer has a simple mixer control labeled "SPDIF".  My laptop lacks
a physical SPDIF port, but perhaps when this happens, sound is being
directed to an (unconnected) SPDIF output on the codec?

In the "good" state, there is only the "Master" mixer control, and it
can adjust the volume and mute/unmute my sound. The sound is also
automatically redirected to the headphones if they are plugged in.

Another oddity: in the "bad" state, alsamixer shows a "Master" and a
"SPDIF" volume control. I can adjust and mute/unmute them
independently, but if i close and reopen alsamixer, they will both
match the state of whichever I adjusted last.

-- 
Andrew Mahone
andrew DOT mahone AT gmail DOT com

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