Re: [Alsa-user] Layla24-Cardbus

2006-03-05 Thread Lee Revell
[re-added alsa-user to CC list]

On Sun, 2006-03-05 at 21:13 +0100, pw.marcus wrote:
> Driver Info #0:
>  Driver Status: maestro is not active
>  Driver Activation Cmd: "modprobe maestro"
>  Driver Info #1:
>  Driver Info: snd-es1968
>  Config Status: cfg=new, avail=yes, need=no, active=unknown
> 
> 23: PCI 600.0: 0480 Multimedia controller
>  [Created at pci.244]
>  Unique ID: vTuk.WgIgdYU+GZ6
>  Parent ID: +6Nb.IR44dvWvxb8
>  SysFS ID: /devices/pci:00/:00:04.1/:06:00.0
>  SysFS BusID: :06:00.0
>  Hardware Class: unknown
>  Model: "Echo Digital Audio Multimedia controller"
>  Hotplug: CardBus
>  Vendor: pci 0x1057 "Motorola"
>  Device: pci 0x3410
>  SubVendor: pci 0xecc0 "Echo Digital Audio Corporation"
>  SubDevice: pci 0x0060
>  Revision: 0x01
>  Memory Range: 0x10c0-0x10cf (rw,non-prefetchable,disabled)
>  IRQ: 11 (334839 events)
>  Config Status: cfg=new, avail=yes, need=no, active=unknown
>  Attached to: #14 (CardBus bridge)
> 

It appears that no driver is being loaded for the EchoAudio card.

What happens if you run "modprobe snd-layla24" as root?

You may just need a newer ALSA version.

Lee



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Re: [Alsa-user] Edirol DA-2496

2006-03-05 Thread Lee Revell
On Mon, 2006-03-06 at 11:30 +1100, Nathan Creek wrote:
> Lee Revell wrote:
> > It's impossible to tell how difficult it will be to develop a driver
> > until someone tries it.  If you're looking for a guarantee that it will
> > be possible to support the card, you won't get one.
> 
> Thanks Lee. I'm not looking for a guarantee. I'm just looking for more 
> information about what would be involved so that I can decide weather 
> it's worth taking a chance on getting the card.
> 
> Let's say I went and bought the card today, bought it home, and plugged 
> it in. What would be the next steps for me to get ALSA support developed 
> for it?
> 
> Would I just need to run programs (lspci?) to gather the needed 
> information?

This will probably not be enough.

>  or would I need to physically open the unit and somehow 
> track which pins on the IC's are connected to what? What tools would I 
> need to do this? I'm a software engineer and have limited electrical 
> engineering knowledge, but I know my way around a multimeter. Would that 
> be enough?
> 

Any or all of the above.  You can search the ALSA bug tracker and
mailing list archives to see what has been required in the past and how
it turned out.

> Would I need to get information from the manufacturer? If so what 
> information would I ask for?

It's impossible to say because no one has tried it yet.  It's possible
you could get the card and then none of the ALSA developers will have
time to help and you'll have to do all the work yourself.

It's also possible that you'll end up with limited support (some
features not working).

Lee



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Re: [Alsa-user] Layla24-Cardbus

2006-03-06 Thread Lee Revell
On Mon, 2006-03-06 at 16:13 +0100, pw.marcus wrote:
> 1. Were could I find an asound.rc exemple for the Layla24 ?

What makes you think you need one?

>2. How is it possible to see the ADAT inputs in jack control (jack
>   control only wants to start up with 8 in/8 out) ? 

No idea



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Re: [Alsa-user] Trouble setting up ALSA on Asus K8U-X motherboard

2006-03-06 Thread Lee Revell
On Mon, 2006-03-06 at 08:35 -0300, Eduardo M KALINOWSKI wrote:
> I have an AMD64 system on an Asus K8U-X motherboard. However, I
> haven't had success using ALSA and I have no sound at the moment.
> 
> The sound card is identified this way:
> 
> # lspci | grep audio
> :00:04.0 Multimedia audio controller: ALi Corporation M5455 PCI
> AC-Link Controller Audio Device (rev 20)

Maybe alsaconf is getting the driver wrong, does it work with
snd-ali5451?

Lee



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Re: [Alsa-user] arecord under 2.6.15.4-rt17 ->overruns...

2006-03-07 Thread Lee Revell
On Tue, 2006-03-07 at 19:30 +0100, Guennadi Liakhovetski wrote:
> Ok, my mistake. The quirk checks the southbridge revision, but configures 
> byte 0x76 on the host (north) bridge, where it does get successfully set 
> and remains set. But it doesn't seem to help. Still, at least the comment 
> is wrong - it contradicts the code. Who is the author of that quirk? Any 
> comments whether my problem seems similar to what others observed with 
> this chipset?
> 

I doubt that issue has anything to do with your problem.

> And my audio still doesn't work properly... 

Yes you've mentioned that several times.

Lee



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Re: [Alsa-user] arecord under 2.6.15.4-rt17 ->overruns...

2006-03-07 Thread Lee Revell
On Tue, 2006-03-07 at 20:00 +0100, Guennadi Liakhovetski wrote:
> On Tue, 7 Mar 2006, Lee Revell wrote:
> 
> > On Tue, 2006-03-07 at 19:30 +0100, Guennadi Liakhovetski wrote:
> > 
> > > And my audio still doesn't work properly... 
> > 
> > Yes you've mentioned that several times.
> 
> Ok, sorry, I just wanted to come to some resolution - either a fix or an 
> agreement that it's unfixable. So, if there are no further ideas, I'll 
> just open an entry on bugzilla and be done with it.
> 
> Thanks for trying to help me, everyone, sorry, if I abused your help in 
> any way:-)

Does the OSS driver have the same problem?

Lee



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Re: [Alsa-user] intel8x0 & mpu_port

2006-03-07 Thread Lee Revell
On Tue, 2006-03-07 at 22:37 +0100, Frog wrote:
> Hello,
> 
> I'm trying to setup midi on my motherboard Asus A7N8X Deluxe (chipset 
> nforce2). I use a Debian etch distribution with the kernel 2.6.15.
> 

What gave you the idea that snd-intel8x0 has an mpu_port parameter?

>From Documentation/sound/alsa/ALSA-Configuration.txt:

  Module snd-intel8x0
  ---

Module for AC'97 motherboards from Intel and compatibles.
* Intel i810/810E, i815, i820, i830, i84x, MX440
ICH5, ICH6, ICH7, ESB2
* SiS 7012 (SiS 735)
* NVidia NForce, NForce2, NForce3, MCP04, CK804
 CK8, CK8S, MCP501
* AMD AMD768, AMD8111 
* ALi m5455

ac97_clock- AC'97 codec clock base (0 = auto-detect)
ac97_quirk- AC'97 workaround for strange hardware
See "AC97 Quirk Option" section below.
buggy_irq - Enable workaround for buggy interrupts on some 
motherboards (default yes on nForce chips,
otherwise off)
buggy_semaphore - Enable workaround for hardwares with buggy
semaphores (e.g. on some ASUS laptops)
(default off)

This module supports one chip and autoprobe.

Note: the latest driver supports auto-detection of chip clock.
if you still encounter too fast playback, specify the clock
explicitly via the module option "ac97_clock=41194".

Joystick/MIDI ports are not supported by this driver.  If your 
motherboard has these devices, use the ns558 or snd-mpu401
modules, respectively.

Lee



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Re: [Alsa-user] Soundblaster Live!

2006-03-08 Thread Lee Revell
On Wed, 2006-03-08 at 11:43 +0200, Милен Панков wrote:
> But there appears there is no an overall master volume.
> I read everything I found in the mailing list archives and it seems
> this 
> question is asked many times, but I found no answer given.
> How can I control the overall volume of the card no meter if I play a 
> stereo or 6-channel sound? Is there a way to do this from the drivers?
> And 
> if not is there a possibility to do this in some other way? I was
> thinking 
> of making a softvolume in asound.conf called for example
> 'GrandVolume' 
> where to attach as slaves all the controls mentioned above - for
> stereo and 
> for 6-channel sound. I was trying for some time to make such a thing,
> but 
> my understandings of the syntax of the file seems not to be so good 
> althought I read several times the explanations... Can somebody help
> me in 
> implementing this if it is possible?
> 

This is a very old problem.  There's no easy fix.

Lee



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Re: [Alsa-user] Soundblaster Live!

2006-03-08 Thread Lee Revell
On Wed, 2006-03-08 at 11:43 +0200, Милен Панков wrote:
> The second ting about the card - it does not have an analog CD-IN
> input. It 
> only has a CD_SPDIF input. I've connected this to my DVD-Rom digital 
> output, but can not make an audio cd play at all. I've tried putting
> every 
> one of these options mentioned in the INSTALL: 

Use a player that extracts the audio digitally.  Analog CD connections
are going away.

Lee



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Re: [Alsa-user] Roland UA-100

2006-03-08 Thread Lee Revell
On Wed, 2006-03-08 at 08:52 -0600, Davide Fossati wrote:
> Hi,
> I'm trying to use my Roland/Edirol UA-100 under Fedora Core 4 with ALSA
> 1.0.10. I'm having the same problems that were posted to this list by
> another user a few weeks ago (title: "No sound with Roland UA-100").
> Here is what happens:

Upgrade to the latest 2.6.15.x kernel



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Re: [Alsa-user] Debian Sarge Kernel 2.6.^5-i686 and soundblaster 24 Bit Live doesn`t work/NO SOUND!

2006-03-08 Thread Lee Revell
On Wed, 2006-03-08 at 16:27 +0100, niels jende wrote:
> hi there!
> 
> And a modprobe trial tells me that the module CA0106 couldn`t be found! 
> Now, i think, it`s getting quite "funny"!
> Here is the CA0106 directory:

Try a newer kernel, Sarge might just be too old to support this device.



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Re: [Alsa-user] Debian Sarge Kernel 2.6.^5-i686 and soundblaster 24 Bit Live doesn`t work/NO SOUND!

2006-03-08 Thread Lee Revell
On Wed, 2006-03-08 at 19:12 +0100, niels jende wrote:
> Lee Revell schrieb:
> > On Wed, 2006-03-08 at 16:27 +0100, niels jende wrote:
> >   
> >> hi there!
> >>
> >> And a modprobe trial tells me that the module CA0106 couldn`t be found! 
> >> Now, i think, it`s getting quite "funny"!
> >> Here is the CA0106 directory:
> >> 
> >
> > Try a newer kernel, Sarge might just be too old to support this device.
> >
> >
> >   
> I have the latest Kernel, AFAIK, 2.6.15-i686!

Do you also have the latest libasound2?

$ apt-cache policy libasound2
libasound2:
  Installed: 1.0.10-2ubuntu1
  Candidate: 1.0.10-2ubuntu1

Version 1.0.10 or higher is required for kernel 2.6.15.

Lee



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Re: [Alsa-user] Soundblaster Live!

2006-03-08 Thread Lee Revell
On Wed, 2006-03-08 at 21:02 +0200, Милен Панков wrote:
> Lee Revell написа:
> > On Wed, 2006-03-08 at 11:43 +0200, Милен Панков wrote:
> >> But there appears there is no an overall master volume.
> >> I read everything I found in the mailing list archives and it seems
> >> this 
> >> question is asked many times, but I found no answer given.
> >> How can I control the overall volume of the card no meter if I play a 
> >> stereo or 6-channel sound? Is there a way to do this from the drivers?
> >> And 
> >> if not is there a possibility to do this in some other way? I was
> >> thinking 
> >> of making a softvolume in asound.conf called for example
> >> 'GrandVolume' 
> >> where to attach as slaves all the controls mentioned above - for
> >> stereo and 
> >> for 6-channel sound. I was trying for some time to make such a thing,
> >> but 
> >> my understandings of the syntax of the file seems not to be so good 
> >> althought I read several times the explanations... Can somebody help
> >> me in 
> >> implementing this if it is possible?
> >>
> > 
> > This is a very old problem.  There's no easy fix.
> > 
> > Lee
> > 
> 
> Ok.
> But as I understand there is some possibility to fix this, although not 
> an easy one...
> Where to look for some documentation on that, what to try? Am I on the 
> right way trying to find a solution with software mixer as I continue 
> trying with these attempts.
> Thanks,
> Milen

Unless you can read and understand the emu10k1 driver code, you will
have a very hard time.  And if you can read and understand it you know
what to do. 

The best workaround for the time being (if you can't simply reduce the
volume in the app) is to just write a script that implements "Master
volume" by controlling Front, Rear, Center/LFE at the same time.

You can probably find such a script in the list archive.

Lee



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Re: [Alsa-user] Soundblaster Live!

2006-03-08 Thread Lee Revell
On Wed, 2006-03-08 at 20:38 +0100, Christian Härtwig wrote:
> I have discussed this topic for a very long time on this mailing list, too. 
> And guess what: a "good" (where good means pretty easy to realize) fix *is* 
> available. I have attached the patch file that does the job. You need to 
> recompile alsa drivers that for. with the Patch included you will have a new 
> volume slider that indeed sets the overall volume for all channels. Since i 
> am no developer i cant tell you exactly why this patch is not included in the 
> drivers, but for emu10k1 and sb live i have found out it works great. i am 
> using it with alsa-driver and -lib version 1.0.10 and have had no *real* 
> issues yet. unfortunately i know people reporting that this patch causes the 
> overall output volume to be a bit quiter than normal, but i cannot confirm. 
> Thanks again to whoever has send that patch to me. 

The patch creates a control "Master Digital Playback Volume" - is this
control also effective in analog mode?

Lee



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Re: [Alsa-user] Soundblaster Live!

2006-03-08 Thread Lee Revell
On Wed, 2006-03-08 at 22:42 +0200, Милен Панков wrote:
> Christian Härtwig написа:
>   > Hi There!
> > 
> > I have discussed this topic for a very long time on this mailing list, too. 
> > And guess what: a "good" (where good means pretty easy to realize) fix *is* 
> > available. I have attached the patch file that does the job. You need to 
> > recompile alsa drivers that for. with the Patch included you will have a 
> > new 
> > volume slider that indeed sets the overall volume for all channels. Since i 
> > am no developer i cant tell you exactly why this patch is not included in 
> > the 
> > drivers, but for emu10k1 and sb live i have found out it works great. i am 
> > using it with alsa-driver and -lib version 1.0.10 and have had no *real* 
> > issues yet. unfortunately i know people reporting that this patch causes 
> > the 
> > overall output volume to be a bit quiter than normal, but i cannot confirm. 
> > Thanks again to whoever has send that patch to me.
> > 
> > Greetz
> > Christian Haertwig
> > Alsa User :)
> > 
> 
> It works! in analog mode.
> Just that when I play a 2-channel stereo my left front speaker is muted 
> (other 5 working and being controlled form the Master Digital) and when 
> playing 6-channel audio the rear left speaker is muted (other 5 again 
> working and being controlled form the Master Digital)...
> Pity
> May be this has to do something with these mono statements in the patch.
> I'll give it a try with alsa 1.0.10 and give some feedback.
> 

No, that's not it - all the mono statements do is make the control a
single slider instead of a left/right slider.

It's probably incorrect GPR usage by the patch.

Lee



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Re: [Alsa-user] Soundblaster Live!

2006-03-08 Thread Lee Revell
On Wed, 2006-03-08 at 22:42 +0200, Милен Панков wrote:
> Just that when I play a 2-channel stereo my left front speaker is
> muted 
> (other 5 working and being controlled form the Master Digital) and
> when 
> playing 6-channel audio the rear left speaker is muted (other 5 again 
> working and being controlled form the Master Digital)...
> Pity 

Try deleting the master switch and leaving the volume control, does that
change the behavior?



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Re: [Alsa-user] intel8x0 & mpu_port

2006-03-08 Thread Lee Revell
On Wed, 2006-03-08 at 22:14 +0100, Frog wrote:
> I've read about this options in the ALSA snd-intel8x0 page :
> http://www.alsa-project.org/alsa-doc/doc-php/template.php?module=intel8x0
> 
> in the "module options" section and in the "users contributed notes"
> 
> also in some other pages on the web.
> 
> The official page is wrong ?

The documentatiion is authoritative.  That page is out of date.

Lee



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Re: [Alsa-user] Soundblaster Live!

2006-03-08 Thread Lee Revell
On Wed, 2006-03-08 at 23:40 +0200, Милен Панков wrote:
> Than it was the unknown card. Now I uninstalled 1.0.11rc3 and
> installed 
> 1.0.10, but not the slackware packages, but compiled my own and the
> card 
> is still unknown.
> 

Yes, it's a known issue but this has nothing to do with your problem.

Lee



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Re: [Alsa-user] Soundblaster Live!

2006-03-08 Thread Lee Revell
On Wed, 2006-03-08 at 23:06 +0100, Christian Härtwig wrote:
> This is strange, i only have analog speakers connected, and playback always 
> works properly: 2 channel playback full stereo or enhanced stereo by the 
> soundcard and 6 channel output works as well ... i have tested this using 
> speaker-test, jack (connectors to each output) and xine (with its different 
> speaker arrangement setups) - i had no issues at all
> 

This is with the master volume patch applied?

Lee



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Re: [Alsa-user] Soundblaster Live!

2006-03-08 Thread Lee Revell
On Thu, 2006-03-09 at 00:39 +0200, Милен Панков wrote:
> Lee Revell написа:
> > On Wed, 2006-03-08 at 22:42 +0200, Милен Панков wrote:
> >> Christian Härtwig написа:
> >>   > Hi There!
> >>> I have discussed this topic for a very long time on this mailing list, 
> >>> too. 
> >>> And guess what: a "good" (where good means pretty easy to realize) fix 
> >>> *is* 
> >>> available. I have attached the patch file that does the job. You need to 
> >>> recompile alsa drivers that for. with the Patch included you will have a 
> >>> new 
> >>> volume slider that indeed sets the overall volume for all channels. Since 
> >>> i 
> >>> am no developer i cant tell you exactly why this patch is not included in 
> >>> the 
> >>> drivers, but for emu10k1 and sb live i have found out it works great. i 
> >>> am 
> >>> using it with alsa-driver and -lib version 1.0.10 and have had no *real* 
> >>> issues yet. unfortunately i know people reporting that this patch causes 
> >>> the 
> >>> overall output volume to be a bit quiter than normal, but i cannot 
> >>> confirm. 
> >>> Thanks again to whoever has send that patch to me.
> >>>
> >>> Greetz
> >>> Christian Haertwig
> >>> Alsa User :)
> >>>
> >> It works! in analog mode.
> >> Just that when I play a 2-channel stereo my left front speaker is muted 
> >> (other 5 working and being controlled form the Master Digital) and when 
> >> playing 6-channel audio the rear left speaker is muted (other 5 again 
> >> working and being controlled form the Master Digital)...
> >> Pity
> >> May be this has to do something with these mono statements in the patch.
> >> I'll give it a try with alsa 1.0.10 and give some feedback.
> >>
> > 
> > No, that's not it - all the mono statements do is make the control a
> > single slider instead of a left/right slider.
> > 
> > It's probably incorrect GPR usage by the patch.
> > 
> > Lee
> > 
> 
> Yes it was.
> 
> Leaving '#define SND_EMU10K1_GPR_CONTROLS' to 44 and not changing it to 
> 46 works for me
> Every channel is working in every mode and the Digital master controls 
> everything.
> I have to wait till tomorrow to test it more accurate, because now here 
> is almost one at the middle of the night and I can't switch the volumes 
> too hi.
> 
> Thanks everybody who answered and helped.
> Milen
> 

That's great.  It would be nice if you could regression test it more.

Did you test analog and digital mode?

Lee



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Re: [Alsa-user] Soundblaster Live!

2006-03-08 Thread Lee Revell
On Thu, 2006-03-09 at 01:12 +0200, Милен Панков wrote:
> Lee Revell написа:
> 
> >>> It's probably incorrect GPR usage by the patch.
> >>>
> >>> Lee
> >>>
> >> Yes it was.
> >>
> >> Leaving '#define SND_EMU10K1_GPR_CONTROLS' to 44 and not changing it to 
> >> 46 works for me
> >> Every channel is working in every mode and the Digital master controls 
> >> everything.
> >> I have to wait till tomorrow to test it more accurate, because now here 
> >> is almost one at the middle of the night and I can't switch the volumes 
> >> too hi.
> >>
> >> Thanks everybody who answered and helped.
> >> Milen
> >>
> > 
> > That's great.  It would be nice if you could regression test it more.
> > Did you test analog and digital mode?
> > 
> > Lee
> > 
> 
> I'll test it more accurate and write some feedback to the list.
> I don't have digital speaker system. I can test it only with analog output.
> So far I tested 2-channel and 6-channel audio, cd audio and aux in from 
> a tv-tuner. Everything seems to work and everything is controlled from 
> the 'Digital Master'.

OK.

Is it called "Digital Master" for any good reason?  Simply "Master"
could be better...

Lee



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Re: [Alsa-user] Soundblaster Live!

2006-03-08 Thread Lee Revell
On Thu, 2006-03-09 at 01:12 +0200, Милен Панков wrote:
> 
> I'll test it more accurate and write some feedback to the list.
> I don't have digital speaker system. I can test it only with analog output.
> So far I tested 2-channel and 6-channel audio, cd audio and aux in from 
> a tv-tuner. Everything seems to work and everything is controlled from 
> the 'Digital Master'. 

Can you attach the corrected patch to this (ancient) bug report so it
can be tested by more users.  Make sure to credit the peson you got the
patch from:

https://bugtrack.alsa-project.org/alsa-bug/view.php?id=172

Just FYI, this small patch was not all that was needed to fix the
original problem - there were some larger changes made recently to the
driver that enabled this to work (and broke backwards compatibility).
Otherwise it would have been fixed long ago...

Lee



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Re: [Alsa-user] Soundblaster Live!

2006-03-09 Thread Lee Revell
On Thu, 2006-03-09 at 11:58 +0200, Милен Панков wrote:
> I don't know. For sure it is possible to rename it, but would't Master
> conflict with the original Master in the mixer? 

The original Master only controls the front speakers right?  So I think
it should be renamed to "Front" and the new control becomes "Master".

Lee



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Re: [Alsa-user] snd_usb_audio alias cmedia, SPDIF passtrough

2006-03-09 Thread Lee Revell
On Thu, 2006-03-09 at 23:33 +0100, Denny Schierz wrote:
> hi,
> 
> i trying to get AC3 passthrough working under Ubuntu
> (2.6.12-10-powerpc). Digital output works, but i don't get any video
> player to working with AC3. The USB soundcard is a cheap cmedia and i
> need it only for SPDIF. My Speakers is a Creative DTT2500 soundset.

I'm not sure that AC3 passthrough for USB devices is supported.

Lee



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Re: [Alsa-user] arecord under 2.6.15.4-rt17 ->overruns...

2006-03-09 Thread Lee Revell
On Thu, 2006-03-09 at 23:41 +0100, Guennadi Liakhovetski wrote:
> On Tue, 7 Mar 2006, Lee Revell wrote:
> 
> > Does the OSS driver have the same problem?
> 
> Surprise - I was not able to reproduce the problem with oss. Whereas with 
> alsa I was able to lock my PC again. What I do - just load respective 
> drivers and either "jackd -v -d alsa" or "jackd -v -d oss". And then just 
> put some load in the background + try to start ardour... With alsa I 
> wasn't even able to start it. With oss it did run, and no xruns reported 
> from jackd. Normal non-rt kernel. jackd started without --realtime.
> 
> Ouch
> 

OK, please file a report in the ALSA bug tracker against this driver.

Adrian, please add VIA to your list of OSS drivers that need to remain in the 
kernel.

Lee



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Re: [Alsa-user] Soundblaster Live!

2006-03-10 Thread Lee Revell
On Fri, 2006-03-10 at 20:11 +0200, Милен Панков wrote:
> Just that in the above posting I was mistaken for one thing - The 
> 'Digital Master' does NOT control the 'AUX-IN', It is controlled from 
> 'Master'. But I think this is normal as the AUX-IN is an analog
> output. 

OK, now we just need a sane name for these controls because "Digital
Master" won't fly.

Lee



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Re: [Alsa-user] [SOLVED] Trouble setting up ALSA on Asus K8U-X motherboard

2006-03-11 Thread Lee Revell
On Sat, 2006-03-11 at 09:10 -0300, Eduardo M KALINOWSKI wrote:
> Perhaps alsaconf should automatically add this option to the modules
> configuration files when this specific card is found. I guess I should
> file a bug report against that, shoudn't I? 

Yes, please do, including lspci output for your card.

Lee



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Re: [Alsa-user] /proc/asound/pcm shows 2 "wrong" additional outputs

2006-03-11 Thread Lee Revell
On Sun, 2006-03-12 at 01:31 +, [EMAIL PROTECTED] wrote:
> a) Does this mean that the last two entries in the above file are
> "wrong"? (in so far as they indicate audio outputs that I don't
> really have?)
> 

Yes - without the full specs to every device that ALSA supports it's
impossible to get this right 100% of the time.  Often it means the
devices are present in your soundcard but the outputs simply aren't
hooked up to anything.

> b) I think that the kernel module for my soundcard is snd_cs46xx.
> Is there anything I can do to "remove" the last two entries in the 
> above file? ("modinfo snd-cs46xx" doesn't appear to contain
> anything appropriate) 

No.

Lee



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Re: [Alsa-user] amd 64 utils compile error for 1.0.11.rc2

2006-03-12 Thread Lee Revell
On Sun, 2006-03-12 at 08:57 -0500, Jerry Geis wrote:
> I did ./configure; make ; make install for
> drivers and libs on centsos 4.2.
> 
> When I do ./configure and make for alsa-utils I get the following erros.
> I think I did things right. Is there a problem with alsa-utils?

It sounds like your alsa-lib version is too old.

Lee



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Re: [Alsa-user] [Q] on-board ISA card higher quality sound than VIA686?

2006-03-12 Thread Lee Revell
On Sun, 2006-03-12 at 21:19 +0100, Guennadi Liakhovetski wrote:
> Hi
> 
> Just a practical curiousity question - it seems to me that audio recorded 
> from the line-in of an on-board es18xx ISA card has higher quality than on 
> an on-board too PCI VIA686? Is it expected, and is there an explanation? 
> Or just an accidental better electrical match with the audio-source - it's 
> a headphone output of an md-player, but the disc and volume were the same.
> The software (apart from the driver, of course) is roughly the same - 
> Debian Sarge, 2.6.15.4 kernel.

I'm not surprised, vendors are under incredible pressure to "weed out
inefficiencies" which in real life means "cutting corners".  Hardware
quality declines every year.

The rationalization is that most people can't hear the difference and
will only use the mic for VOIP anyway.

Lee



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Re: [Alsa-user] Audiotrak Maya 44 USB Soundcard

2006-03-12 Thread Lee Revell
On Sun, 2006-03-12 at 22:31 +0100, Hiram Abiff wrote:
> Hi!
> 
> I've lost a lot of time and patience trying to set up my
> Maya 44USB soundcard, the guys from the Linux-Audio-Users
> mailing list tried to help me, but it turned out to be
> a bug in ALSA.

What is the output of "lsmod" and "dmesg"?

Lee



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Re: [Alsa-user] amd 64 utils compile error for 1.0.11.rc2

2006-03-12 Thread Lee Revell
On Sun, 2006-03-12 at 17:08 -0500, Jerry Geis wrote:
> Lee,
> 
> I just tried it again. cd into alsa-lib-1.0.11rc3. I did ./configure
> make clean; make; make install.
> 
> The I cd into th alsa-utils-1.0.11.rc2 and do ./configure; make clean;
> make
> and get these errors.

Sorry, no idea.  Your build environment is not correct but I can't tell
what the problem is.  Maybe you have old alsa-lib still around.

Lee



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Re: [Alsa-user] amd 64 utils compile error for 1.0.11.rc2

2006-03-12 Thread Lee Revell
On Sun, 2006-03-12 at 17:15 -0500, Jerry Geis wrote:
> > Lee,
> >
> > I just tried it again. cd into alsa-lib-1.0.11rc3. I did ./configure
> > make clean; make; make install.
> >
> > The I cd into th alsa-utils-1.0.11.rc2 and do ./configure; make clean; 
> > make
> > and get these errors.
> >

Blank message - ?



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Re: [Alsa-user] amd 64 utils compile error for 1.0.11.rc2

2006-03-12 Thread Lee Revell
On Sun, 2006-03-12 at 17:59 -0500, Jerry Geis wrote:
> Lee,
> 
> My include files look like it's there...
> [EMAIL PROTECTED] alsa-utils-1.0.11rc2]# grep snd_ctl_elem_info_is_user 
> /usr/include/*/*.h
> /usr/include/alsa/control.h:int snd_ctl_elem_info_is_user(const 
> snd_ctl_elem_info_t *obj);
> 
> This is a compile time problem at this point I think.
> Why would it not finding the include file? Everything looks right?
> 

I don't know.  It works perfectly for me, and it looks like the same
build commands:

Making all in alsactl
make[1]: Entering directory `/smb/rlrevell/cvs/alsa/alsa-utils/alsactl'
if gcc -DHAVE_CONFIG_H -I. -I. -I../include -g -O2 -MT alsactl.o -MD
-MP -MF ".deps/alsactl.Tpo" -c -o alsactl.o alsactl.c; \
then mv -f ".deps/alsactl.Tpo" ".deps/alsactl.Po"; else rm -f
".deps/alsactl.Tpo"; exit 1; fi
if gcc -DHAVE_CONFIG_H -I. -I. -I../include -g -O2 -MT state.o -MD
-MP -MF ".deps/state.Tpo" -c -o state.o state.c; \
then mv -f ".deps/state.Tpo" ".deps/state.Po"; else rm -f
".deps/state.Tpo"; exit 1; fi
if gcc -DHAVE_CONFIG_H -I. -I. -I../include -g -O2 -MT power.o -MD
-MP -MF ".deps/power.Tpo" -c -o power.o power.c; \
then mv -f ".deps/power.Tpo" ".deps/power.Po"; else rm -f
".deps/power.Tpo"; exit 1; fi
if gcc -DHAVE_CONFIG_H -I. -I. -I../include -g -O2 -MT names.o -MD
-MP -MF ".deps/names.Tpo" -c -o names.o names.c; \
then mv -f ".deps/names.Tpo" ".deps/names.Po"; else rm -f
".deps/names.Tpo"; exit 1; fi
gcc  -g -O2   -o alsactl  alsactl.o state.o power.o names.o  -lasound
-lm -ldl -lpthread
make[1]: Leaving directory `/smb/rlrevell/cvs/alsa/alsa-utils/alsactl'

Maybe you can try with a clean source tree?

What distro?

Have you customized your build environment at all (CFLAGS, etc)?

Can you try with CVS rather than 1.0.11-rc*?

Is it possible you have old alsa-lib headers in /usr/local?

Lee



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Re: [Alsa-user] [Q] on-board ISA card higher quality sound than VIA686?

2006-03-12 Thread Lee Revell
On Mon, 2006-03-13 at 00:42 +0100, Guennadi Liakhovetski wrote:
> On Sun, 12 Mar 2006, Lee Revell wrote:
> 
> > On Sun, 2006-03-12 at 21:19 +0100, Guennadi Liakhovetski wrote:
> > > Hi
> > > 
> > > Just a practical curiousity question - it seems to me that audio recorded 
> > > from the line-in of an on-board es18xx ISA card has higher quality than 
> > > on 
> > > an on-board too PCI VIA686? Is it expected, and is there an explanation? 
> > > Or just an accidental better electrical match with the audio-source - 
> > > it's 
> > > a headphone output of an md-player, but the disc and volume were the same.
> > > The software (apart from the driver, of course) is roughly the same - 
> > > Debian Sarge, 2.6.15.4 kernel.
> > 
> > I'm not surprised, vendors are under incredible pressure to "weed out
> > inefficiencies" which in real life means "cutting corners".  Hardware
> > quality declines every year.
> > 
> > The rationalization is that most people can't hear the difference and
> > will only use the mic for VOIP anyway.
> 
> Well, _this_ difference everybody would here, but, well...

I guess you are referring to recording with the OSS driver, as you said
that capture with ALSA is unusable with this device?

Any chance you can try to verify the bug with Windows? ;-)

Lee 



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Re: [Alsa-user] Audiotrak Maya 44 USB Soundcard

2006-03-13 Thread Lee Revell
On Mon, 2006-03-13 at 19:50 +0100, Hiram Abiff wrote:
> > I checkveed dmesg, this message appears after each aplay's
> > unsuccesful attempt:
> >
> > cannot submit datapipe for urb 0, err = -38
> >
> > What is this "urb 0"?
> 
> I found this in the LAU archive from , pertaining
> to this error message:
> 
> "> > > ALSA: could not start playback (Broken pipe)
> >
> > kernel: cannot submit datapipe for urb 0, err = -28
> 
> 28 = ENOSPC: "This request would overcommit the usb bandwidth reserved
> for periodic transfers (interrupt, isochronous)."
> 
> This doesn't seem to be quite correct for two 16-bit stereo streams.
> Which kernel and host controller driver are you using?
> Probably you should just deactivate bandwith checking in the kernel
> config. "
> 
> This happened after Clemens advised Chris Eckert to try loading
> the snd-usb-audio with "nrpacks=1" and setting the period in jack
> to 3. But, I didn;t pass this argument with the snd-usb-audio
> and didn't set jack to use 3 periods. Would deactivating
> bandwidht checking work for me too, then?

Error -38 is not the same as error -28.  If you have bandwidth checking
enabled then certainly disable it (it's well known that it's broken).

Is this device connected through a hub?

Lee



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Re: [Alsa-user] Audiotrak Maya 44 USB Soundcard

2006-03-13 Thread Lee Revell
On Mon, 2006-03-13 at 20:53 +0100, Hiram Abiff wrote:
> > "alsamixer -c 2 -V capture" but it says " No mixer elems found".
> > When I run "alsamixer -c 2" I get 3 sliders for playback, although
> > Maya has 2 stereo outputs???!! I discovered that the third actually
> > controls one of the input channels, but driving it to 100 still produces low
> > volume audio. When I use my laptop's soundcard input the captured
> > volume is much much louder.
> 
> I ran  amixer -c 2
> Simple mixer control 'Speaker',0
>   Capabilities: pvolume pswitch pswitch-joined
>   Playback channels: Front Left - Front Right
>   Limits: Playback 0 - 255
>   Front Left: Playback 255 [100%] [on]
>   Front Right: Playback 255 [100%] [on]
> Simple mixer control 'Speaker',1
>   Capabilities: pvolume pswitch pswitch-joined
>   Playback channels: Front Left - Front Right
>   Limits: Playback 0 - 255
>   Front Left: Playback 255 [100%] [on]
>   Front Right: Playback 255 [100%] [on]
> Simple mixer control 'Speaker',2
>   Capabilities: pvolume pswitch pswitch-joined
>   Playback channels: Front Left - Front Right
>   Limits: Playback 0 - 255
>   Front Left: Playback 255 [100%] [on]
>   Front Right: Playback 255 [100%] [on]
> 
> For some reason I see 3 Speaker entries only in the playback section
> and no capture entries.
> 

Many USB audio devices don't have (much of) a software controllable
mixer, the only mixer is controlled by the knobs on the device.

Lee



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Re: [Alsa-user] Audiotrak Maya 44 USB Soundcard

2006-03-13 Thread Lee Revell
On Mon, 2006-03-13 at 20:41 +0100, Hiram Abiff wrote:
> I can get output from muse no prob and from other music making apps.
> I tried running xmms but it complains and cannot start:
> 
> Input has 2 channels, soundcard uses 4 channels
> No conversion is available 

XMMS bug, it wants to use the hw:x device even if the format is
unsupported.  You have to manually configure it to use plughw:x

Clemens, it seems like some of the possible USB error messages (like
this one) are only possible if a hub is involved, maybe you could add
this to your human readable error codes?

Lee



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Re: [Alsa-user] Roland UA-100

2006-03-13 Thread Lee Revell
On Mon, 2006-03-13 at 16:39 -0600, Davide Fossati wrote:
> I tried with a short (10 ms) wave file. It hangs forever.
> 

Does anything appear in the output of "dmesg" when you try this?

Lee



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Re: [Alsa-user] Soundblaster Live!

2006-03-14 Thread Lee Revell
On Thu, 2006-03-09 at 18:59 +0100, Mikael Magnusson wrote:
> I'm the author of the patch, which I also have posted on alsa-devel
> some time ago. I have another patch that renames the controls, but I'm
> not sure it's a good idea to call it Master. It doesn't control the
> analog mixer, only the pcm output from the Emu10k1 synthesizer. The
> analog input(s) that are captured are routed through the Emu10k1 and
> can be controlled with "Master Digital". 

Well it definitely can't be called "Master Digital" as this implies that
it controls the level of SPDIF output.

https://bugtrack.alsa-project.org/alsa-bug/view.php?id=154

"I'd suggest to rename the ac97 Master to Front, and the new Master
Digital to a real Master.  Also, for avoid regressions (no sound after
update), make the default value of Front full and unmuted."

Can you respin the patch according to tiwais suggestion and post it to
alsa-devel, so this can be fixed for the 1.0.11 release?

Lee



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Re: [linux-audio-user] Re: [Alsa-user] VxPocket and hotplug - need help getting it working

2006-03-14 Thread Lee Revell
On Tue, 2006-03-14 at 15:52 +0100, Asbjørn Sæbø wrote:
> (I am cc-ing the linux audio user list, in the hope of broadening 
> the audience and reaching someone able to help me sort this out.)
> 
> 
> On Tue, Mar 14, 2006 at 12:02:56PM +0100, Asbjørn Sæbø wrote:
> > I am trying to get a Digigram VxPocket sound card working, and really
> > need some advice on how to proceed.
> > [Rest of previous post far below.]
> 
> In addition to what I have done before (see below), I have now created a 
> /etc/pcmcia/vxpocket.conf according to the description at the bottom of 
> the alsa-project page for the vxpocket card.  And I have added the 
> necessary lines to /etc/modutils/alsa-base, and run update-modules.

IIRC these require firmware, is alsa-firmware installed?

Any chance you can try a recent distro?  Kernel 2.6.12 is a year old...

Lee



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Re: [Alsa-user] Roland UA-100

2006-03-14 Thread Lee Revell
On Tue, 2006-03-14 at 13:18 -0600, Davide Fossati wrote:
> On Tue, 2006-03-14 at 08:44 +0100, Clemens Ladisch wrote:
> > Davide Fossati wrote:
> > > On Mon, 2006-03-13 at 16:41 +0100, Clemens Ladisch wrote:
> > > > When you try to play a short file, will it play nothing for a short
> > > > time, or will it hand forever?
> > > 
> > > I tried with a short (10 ms) wave file. It hangs forever.
> > 
> > ALSA should automatically abort when nothing happens for ten seconds.
> > Is this "forever" longer than that?
> > 
> > Is it possible that some other program is using the device?
> > ("lsof /dev/dsp* /dev/snd/*")
> 
> I tried, nobody else is using it.

And the device is not connected through a hub?

Other USB devices work on the same port?

Have you customized the kernel config?

Lee



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Re: [Alsa-user] Roland UA-100

2006-03-14 Thread Lee Revell
On Tue, 2006-03-14 at 13:48 -0600, Davide Fossati wrote:
> > And the device is not connected through a hub?
> 
> No
> 
> > Other USB devices work on the same port?
> 
> There is a USB mouse physically connected next to the UA-100. Is it
> considered the same port?
> However, when I tried on my laptop there was no other USB devices at
> all.
> 

If you have the same problem on multiple machine this is not the issue.

> > Have you customized the kernel config?
> 
> Yes. I added preemption and NTFS. Should I do something else?
> 

No, it should work.  You made sure that the OSS USB audio and midi
drivers are not enabled?

Lee



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Re: [Alsa-user] Roland UA-100

2006-03-14 Thread Lee Revell
On Tue, 2006-03-14 at 14:49 -0600, Davide Fossati wrote:
> OSS is disabled by default in the kernel, and I did not enable it.
> 
> Is there any diagnostic tool I can use to monitor the packets sent
> over
> the USB channel? Could that help figure out what's wrong? What other
> kind of test should I try? 

That does not guarantee the OSS audio and MIDI drivers are disabled, you
have to make sure they are disabled in the kernel config, or blacklist
them.

If it's a fully modular kernel then just check the output of "lsmod".

Sorry I am out of ideas.

Lee



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Re: [Alsa-user] Does alsa support the Intel 82801DB/DBL/DBM

2006-03-15 Thread Lee Revell
On Tue, 2006-03-14 at 22:34 -0800, BassPlayer wrote:
> Hello,
> 
> I seem to be having problems with my Intel Corporation 82801DB/DBL/DBM
> chip onmy IBM T42.
> 
> Is that supported? I've tried 1.0.10.
> 
> # cat /proc/asound/version
> Advanced Linux Sound Architecture Driver Version 1.0.10.
> Compiled on Mar 14 2006 for kernel 2.4.31.

Try:

 - a newer ALSA version like 1.0.11-rc3
 - a 2.6 kernel
 - explaining exactly what "doesn't work"



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Re: [Alsa-user] Does alsa support the Intel 82801DB/DBL/DBM

2006-03-15 Thread Lee Revell
On Tue, 2006-03-14 at 22:34 -0800, BassPlayer wrote:
> Hello,
> 
> I seem to be having problems with my Intel Corporation 82801DB/DBL/DBM
> chip onmy IBM T42.
> 
> Is that supported? I've tried 1.0.10.

Please disregard my message, I missed Clemens' reply.

Lee



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Re: [linux-audio-user] Re: [Alsa-user] VxPocket and hotplug - need help getting it working

2006-03-15 Thread Lee Revell
On Wed, 2006-03-15 at 09:13 +0100, Asbjørn Sæbø wrote:
> I did try the most recent live CD for Ubuntu Dapper Drake, but that
> one 
> did not find my card out of the box either.
> (And I have previously tried Planet CCRMA, without any luck.) 

Last time I tried Ubuntu with an RME device I was disappointed to find
that they don't make alsa-firmware available at all - I eventually gave
up and installed it from the ALSA site.

Lee



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Re: [Alsa-user] Soundblaster Live!

2006-03-15 Thread Lee Revell
On Wed, 2006-03-15 at 21:35 +0100, Christian Härtwig wrote:
> That sounds great! Which version of alsa-driver does this patch apply
> to? I want to test it myself very soon. 

Any patch you see on this list that doesn't specify a version should be
against CVS.  But it might work against whatever version you have (this
code has not changed in a while).

Lee



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Re: [Alsa-user] VxPocket and hotplug - need help getting it working

2006-03-17 Thread Lee Revell
On Fri, 2006-03-17 at 09:33 +0100, Asbjørn Sæbø wrote:
> On Wed, Mar 15, 2006 at 03:16:02PM +0100, Takashi Iwai wrote:
> > At Wed, 15 Mar 2006 15:03:13 +0100,
> > [...]
> 
> > > OK, I have installed DeMuDi 1.3.0rc1, which has kernel 2.6.14, and alsa 
> > > version 1.0.10.
> > 
> > Still way too old to debug...
> 
> What is too old, the kernel, alsa or both?

Both.  Is there some way you can try the latest 2.6.16-rc6 kernel or
2.6.15.x?

Lee



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Re: [Alsa-user] No sound with Intel ICH7

2006-03-17 Thread Lee Revell
On Fri, 2006-03-17 at 16:00 +0100, Torbjørn Lium wrote:
> I get to load the modules and everything looks fine, but I can't
> produce 
> any sound at all. Nothing from the notebooks internal speakers,
> nothing 
> when I have my headset connected to the SPDIF output. No fun at all.
> And yes, I have unmuted all the channels and turned the volume up,
> but 
> still no audio.
> If it's any help at all, I'm using alsa driver,lib and utils version 
> 1.0.10 compiled from source on Slackware 10.2.
> 

You need to use ALSA CVS, or wait for the 1.0.11 release.  This device
did not work until recently.

Lee



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Re: [Alsa-user] No sound with Intel ICH7

2006-03-17 Thread Lee Revell
On Fri, 2006-03-17 at 19:12 +0100, Torbjørn Lium wrote:
> 
> I did try the cvs driver yesterday, but I must have made some mistake 
> somewhere.
> I just downloaded and installed the cvs drivers, libs and utils some
> ten 
> minutes ago and suddenly everything worked. I must have failed to
> remove 
> the old alsadrivers completely.
> Thanks a lot Lee! 

No you did it right, I think the bug was just fixed today or yesterday.

Lee



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Re: [Alsa-user] VxPocket and hotplug - need help getting it working

2006-03-20 Thread Lee Revell
On Mon, 2006-03-20 at 11:24 +0100, Asbjørn Sæbø wrote:
> On Fri, Mar 17, 2006 at 12:17:29PM -0500, Lee Revell wrote:
> > On Fri, 2006-03-17 at 09:33 +0100, Asbjørn Sæbø wrote:
> > > On Wed, Mar 15, 2006 at 03:16:02PM +0100, Takashi Iwai wrote:
> > > > At Wed, 15 Mar 2006 15:03:13 +0100,
> > > > [...]
> > > 
> > > > > OK, I have installed DeMuDi 1.3.0rc1, which has kernel 2.6.14, and 
> > > > > alsa 
> > > > > version 1.0.10.
> > > > 
> > > > Still way too old to debug...
> > > 
> > > What is too old, the kernel, alsa or both?
> > 
> > Both.  Is there some way you can try the latest 2.6.16-rc6 kernel or
> > 2.6.15.x?
> 
> I can try.  But those newer kernels don't use hotplug, they use udev (I 
> think).  Will that be an obstacle?

It should work, but I have no idea how well Demudi plays with newer
kernels.

Lee



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Re: [Alsa-user] SB 512 EMU10K1!

2006-03-20 Thread Lee Revell
On Mon, 2006-03-20 at 21:49 +0100, Benjamin Hansmann wrote:
> Hi,
> 
> up to know I used the EMU10K1 OSS for using my SB 512. I recompiled a
> new kernel (2.6.16) a few times and I got no sound to work.
> 
> Does anybody have any suggestions?
> 
> See attachment (kernel .config) for details of my kernel config.
> 
> Thanks a lot, help would be apreciated.

What is the output of "amixer"?

Lee



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Re: [Alsa-user] FW: Ice 1712 problems

2006-03-20 Thread Lee Revell
On Mon, 2006-03-20 at 16:04 -0700, Bobby Acha wrote:
> 
> Hi all,
> 
> 
> 
>   Here is my problem.  I have set up a Delta 44 (ice1712) soundcard.
> It is functioning, but I am having great difficulty in setting up the
> ability to record or play from multiple pairs of channels at the same
> time.  I also am having a hard time playing directly to the hw:0,0
> device.  I am however able to use plughw:0,0.

Upgrade to ALSA 1.0.11-rc3 (or replace
your /usr/share/alsa/cards/ICE1712.conf with the one from 1.0.11-rc3)
and lose the .asoundrc and this will Just Work.

Lee



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RE: [Alsa-user] FW: Ice 1712 problems

2006-03-20 Thread Lee Revell
On Mon, 2006-03-20 at 16:31 -0700, Bobby Acha wrote:
> ok,  I am currently compiling 1.0.11 rc3.  Do I still not need
> a .asoundrc to tell it which of the 4 channels I want to record?
> 

Actually I think I misread your message, you don't need to upgade
(sorry).

See my response to your other message



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Re: [Alsa-user] FW: Ice 1712 problems

2006-03-20 Thread Lee Revell
On Mon, 2006-03-20 at 16:04 -0700, Bobby Acha wrote:
> 
> Hi all,
> 
> 
> 
>   Here is my problem.  I have set up a Delta 44 (ice1712) soundcard.
> It is functioning, but I am having great difficulty in setting up the
> ability to record or play from multiple pairs of channels at the same
> time.  I also am having a hard time playing directly to the hw:0,0
> device.  I am however able to use plughw:0,0.
> 
> 
> 
> [EMAIL PROTECTED] ~]# aplay test.wav
> 
> Playing WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 44100 Hz,
> Stereo Aborted by signal Interrupt...
> 
> 
> 
> 
> 
> [EMAIL PROTECTED] ~]# aplay -D hw:0,0 test.wav Playing WAVE
> 'test.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo
> 
> aplay: set_params:896: Sample format non available
> 
This won't work as you are trying to play an S16_LE .wav file to the hw
device which only supports S32_LE.  You must use plughw to play
normal .wav files on this device.
> 
> 
> [EMAIL PROTECTED] ~]# arecord -D ch12 test.wav ALSA lib
> pcm_dshare.c:613:(snd_pcm_dshare_open) The dshare plugin supports only
> playback stream
> 
> arecord: main:544: audio open error: Invalid argument
> 
I think you want dsnoop not dshare here.
> 
>  [EMAIL PROTECTED] ~]# arecord -D hw:0,0 test.wav Recording WAVE
> 'test.wav' : Unsigned 8 bit, Rate 8000 Hz, Mono
> 
> arecord: set_params:896: Sample format non available
> 
See above - you are trying to capture 8 bit audio from the hw: device
which only supports S32_LE.  Plughw should work.

> Following is my asound.conf file.  I have checked sing aplay and
> arecord with the -l flag, and the conf file is being used.
> 
> 
> 
> 
> 
> pcm_slave.delta {
> 
>   pcm "hw:0,0"
> 
>   channels 4
> 
>   rate 44100
> 
>   buffer_size 4096
> 
>   period_size 2048
> 
>   periods 0
> 
>   period_time 0
> 
> }
> 
> 
> 
> pcm.ch12 {
> 
>   type dshare
> 
>   ipc_key 47110815
> 
>   slave delta
> 
>   bindings.0 0
> 
>   bindings.1 1
> 
> }
> 
> 
> 
> pcm.ch34 {
> 
>   type dshare
> 
>   ipc_key 47110815
> 
>   slave delta
> 
>   bindings.0 2
> 
>   bindings.1 3
> 
> }
> 
> 
> 
> 
> 
> I have also tried using the pcm as "plughw:0,0". 
> 
> 
> 
> 
> 
> [EMAIL PROTECTED] ~]# aplay -D ch12 test.wav ALSA lib
> pcm_dshare.c:687:(snd_pcm_dshare_open) dshare plugin can be only
> connected to hw plugin Segmentation fault
> 
> 
> 
> Any help would be greatly appreciated.
> 
> 
> 
> Thanks,
> 
> 
> 
> Bobby
> 
> 
> 
> Senior Systems Engineer
> 
> Colorado Public Radio
> 
> Englewood, CO 80012
> 
> 303 871 9191 x 252
> 
> 
> 
> 



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RE: [Alsa-user] FW: Ice 1712 problems

2006-03-20 Thread Lee Revell
On Mon, 2006-03-20 at 17:22 -0700, Bobby Acha wrote:
> I am able to play and record using plughw:0,0.  My main issue is that
> I am only able to record the first 2 channels with this solution.  I
> need to record all 4 input channels and I have no idea how to do this.
> I figured I would play with the .asoundrc file and that has led to
> nothing but more confusion for me.  How do I specify recording from
> channels 3&4 using the plughw solution?
> 
I would just use JACK, it's MUCH easier than all this .asoundrc stuff.

> Thanks
> 
> -Bobby
> 
> Senior Systems Engineer
> Colorado Public Radio
> Englewood, CO 80112
> 303 871 9191 x252
> 
> 
> 
> 
> -Original Message-
> From: Lee Revell [mailto:[EMAIL PROTECTED]
> Sent: Mon 3/20/2006 4:49 PM
> To: Bobby Acha
> Cc: alsa-user@lists.sourceforge.net
> Subject: Re: [Alsa-user] FW: Ice 1712 problems
> 
> On Mon, 2006-03-20 at 16:04 -0700, Bobby Acha wrote:
> >
> > Hi all,
> >
> >
> >
> >   Here is my problem.  I have set up a Delta 44 (ice1712) soundcard.
> > It is functioning, but I am having great difficulty in setting up
> the
> > ability to record or play from multiple pairs of channels at the
> same
> > time.  I also am having a hard time playing directly to the hw:0,0
> > device.  I am however able to use plughw:0,0.
> >
> >
> >
> > [EMAIL PROTECTED] ~]# aplay test.wav
> >
> > Playing WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 44100
> Hz,
> > Stereo Aborted by signal Interrupt...
> >
> >
> >
> >
> >
> > [EMAIL PROTECTED] ~]# aplay -D hw:0,0 test.wav Playing WAVE
> > 'test.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo
> >
> > aplay: set_params:896: Sample format non available
> >
> This won't work as you are trying to play an S16_LE .wav file to the
> hw
> device which only supports S32_LE.  You must use plughw to play
> normal .wav files on this device.
> >
> >
> > [EMAIL PROTECTED] ~]# arecord -D ch12 test.wav ALSA lib
> > pcm_dshare.c:613:(snd_pcm_dshare_open) The dshare plugin supports
> only
> > playback stream
> >
> > arecord: main:544: audio open error: Invalid argument
> >
> I think you want dsnoop not dshare here.
> >
> >  [EMAIL PROTECTED] ~]# arecord -D hw:0,0 test.wav Recording WAVE
> > 'test.wav' : Unsigned 8 bit, Rate 8000 Hz, Mono
> >
> > arecord: set_params:896: Sample format non available
> >
> See above - you are trying to capture 8 bit audio from the hw: device
> which only supports S32_LE.  Plughw should work.
> 
> > Following is my asound.conf file.  I have checked sing aplay and
> > arecord with the -l flag, and the conf file is being used.
> >
> >
> >
> >
> >
> > pcm_slave.delta {
> >
> >   pcm "hw:0,0"
> >
> >   channels 4
> >
> >   rate 44100
> >
> >   buffer_size 4096
> >
> >   period_size 2048
> >
> >   periods 0
> >
> >   period_time 0
> >
> > }
> >
> >
> >
> > pcm.ch12 {
> >
> >   type dshare
> >
> >   ipc_key 47110815
> >
> >   slave delta
> >
> >   bindings.0 0
> >
> >   bindings.1 1
> >
> > }
> >
> >
> >
> > pcm.ch34 {
> >
> >   type dshare
> >
> >   ipc_key 47110815
> >
> >   slave delta
> >
> >   bindings.0 2
> >
> >   bindings.1 3
> >
> > }
> >
> >
> >
> >
> >
> > I have also tried using the pcm as "plughw:0,0".
> >
> >
> >
> >
> >
> > [EMAIL PROTECTED] ~]# aplay -D ch12 test.wav ALSA lib
> > pcm_dshare.c:687:(snd_pcm_dshare_open) dshare plugin can be only
> > connected to hw plugin Segmentation fault
> >
> >
> >
> > Any help would be greatly appreciated.
> >
> >
> >
> > Thanks,
> >
> >
> >
> > Bobby
> >
> >
> >
> > Senior Systems Engineer
> >
> > Colorado Public Radio
> >
> > Englewood, CO 80012
> >
> > 303 871 9191 x 252
> >
> >
> >
> >
> 
> 
> 
> 



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[Alsa-user] Skype

2006-03-21 Thread Lee Revell
I am afraid the well known Skype+aoss problems may be unfixable from the
ALSA side. It appears that the Skype binary is self modifying and takes
extensive measures to make sure it is not being run in a debugger or
emulator. This almost certainly would eliminate any LD_PRELOAD hacks
from working.

http://www.secdev.org/conf/skype_BHEU06.handout.pdf [^]

I believe the only solution to get Skype working with dmix etc. is for
the Skype developers to add proper native ALSA support.

Lee



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Re: [Alsa-user] Skype

2006-03-21 Thread Lee Revell
On Wed, 2006-03-22 at 03:40 +0200, Sergei Steshenko wrote:
> 
> But it used to work with 'aoss'. 

Yes, it works now on some hardware.  What Skype/aoss versions and driver
was this, and what was the version where it stopped working?

They might have just changed the binary to detect aoss and fail.

Lee



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Re: [Alsa-user] Skype

2006-03-21 Thread Lee Revell
On Wed, 2006-03-22 at 08:15 +0200, Sergei Steshenko wrote:
> There aren't that many bug reports of mine, it says something like
> 'aoss + skype'
> and it's about a lot of error messages regarding inability to control
> capture gain.
> 

That's a different issue, I was referring to it not working at all on
many systems.

Lee



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Re: [Alsa-user] more usb-audio devices

2006-03-22 Thread Lee Revell
On Wed, 2006-03-22 at 17:47 +, Attila Egri-Nagy wrote:
> hello,
> 
> Is it possible to have more usb-audio devices connected to one machine?
> For example two edirol ua-25 connected to jack in order to have 4
> channels for recording.
> 
> I guess it can be done without any problem, but there might be some
> constraints...

Not on the same bus at least with 1.1 (as USB bandwidth limits the
number of channels of audio), and the only 2.0 device supported sounds
terrible (Audigy2 NX).  You could do it with 2 USB buses, 2 jackd's and
jack_diplomat.

Lee



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Re: [Alsa-user] Compilation fails for alsa-1.0.11-rc4

2006-03-27 Thread Lee Revell
On Mon, 2006-03-27 at 22:24 +0530, Damodharan R wrote:
> Hi,
> I am a novice linux user. I have a Dell Inspiron 1705. It has a Intel
> 945 board with Sigmatel codec (STAC9200 as reported by alsa). I run
> Debian Unstable in my laptop. I installed alsa-1.0.10-3. But it gives
> silence output. I downloaded the latest alsa current release
> ( 1.0.11rc4) and tried to compile it. 
> 
> Configure options used,
> $ ./configure --with-debug=full
> --with-kernel=/usr/src/linux-headers-2.6.16-1-686-smp/ 

Do you have the full kernel sources installed or just the kernel
headers?  The full sources are required to compile ALSA.

> --with-cards=all -with-oss=yes
> checking for gcc... gcc
> checking for C compiler default output file name... a.out
> checking whether the C compiler works... yes
> checking whether we are cross compiling... no
> checking for suffix of executables...
> checking for suffix of object files... o
> checking whether we are using the GNU C compiler... yes 
> checking whether gcc accepts -g... yes
> checking for gcc option to accept ANSI C... none needed
> checking for ranlib... ranlib
> .
> .
> config.status: include/version.h is unchanged
> config.status: creating include/autoconf- extra.h
> config.status: include/autoconf-extra.h is unchanged
> Hacking autoconf.h...
> 
> $ make
> .
> .
> make -C /usr/src/linux-headers-2.6.16-1-686-smp/
> SUBDIRS=/root/alsa-driver-1.0.11rc4  modules
> make[1]: Entering directory `/usr/src/linux- headers-2.6.16-1-686-smp'
>   Building modules, stage 2.
>   MODPOST
> make[1]: Leaving directory `/usr/src/linux-headers-2.6.16-1-686-smp'
> utils/link-modules /root/alsa-driver-1.0.11rc4
> 
> ALSA modules were successfully compiled. 
> 
> It reports that modules are compiled succesfully. But make install
> fails
> 
> $ make install

make install must be run as root

> find /lib/modules/2.6.16-1-686-smp/kernel/sound -name 'snd*.*o' |
> xargs rm -f
> make[1]: Entering directory `/root/alsa- driver-1.0.11rc4/acore'
> mkdir -p /lib/modules/2.6.16-1-686-smp/kernel/sound/acore
> cp snd-hwdep.ko snd-page-alloc.ko snd-pcm.ko snd-rawmidi.ko
> snd-rtctimer.ko snd-timer.ko
> snd.ko /lib/modules/2.6.16-1-686-smp/kernel/sound/acore 
> cp: cannot stat `snd-hwdep.ko': No such file or directory
> cp: cannot stat `snd-page-alloc.ko': No such file or directory
> cp: cannot stat `snd-pcm.ko': No such file or directory
> cp: cannot stat `snd-rawmidi.ko ': No such file or directory
> cp: cannot stat `snd-rtctimer.ko': No such file or directory
> cp: cannot stat `snd-timer.ko': No such file or directory
> cp: cannot stat `snd.ko': No such file or directory
> make[1]: *** [modules_install] Error 1 
> make[1]: Leaving directory `/root/alsa-driver-1.0.11rc4/acore'
> make: *** [install-modules] Error 1
> 
> 
> The soundcar id listed as unknown device by lspci
> 
> :00:1b.0 0403: Intel Corp.: Unknown device 27d8 (rev 01) 
> 
> Can anyone give me hints to get my audio working.
> 
> Thanks in advance
> Damodharan



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Re: [Alsa-user] snd-bt87x

2006-03-27 Thread Lee Revell
On Mon, 2006-03-27 at 16:28 -0300, Paulo Cavalcanti wrote:
> Hi,
> 
> Alsa module snd-bt87x is broken since kernel 2.6.13.
> It works in kernel 2.6.12. After that, /dev/dsp2 and
> /dev/audio2 are no longer created, although the module is
> loaded with no error at all (it does not matter if "load_all"
> is being used or not).
> 
> My modprobe.cond (fedora 4) is below:
> 
> # Soundcard 2 
> alias snd-card-2 snd-bt87x 
> options snd-card-2 index=2 
> options snd-bt87x index=2 load_all 
> 
> Is there any plan to fix it, or has it been abandoned.
> I am using the loop-back cable to get sound from my pixelview 
> capture card for now.
> 
> Thanks,
> 
> /Paulo Roma.
> 

It is still broken with the latest 2.6.16 kernel?

Does it work with native ALSA apps (/dev/dsp* and /dev/audio* are the
legacy OSS emulation devices)?

What are the contents of /proc/asound/cards?  Is snd-pcm-oss loaded?

Lee



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Re: [Alsa-user] snd-bt87x

2006-03-27 Thread Lee Revell
( plain text is preferred over HTML on this list)

On Mon, 2006-03-27 at 18:18 -0300, Paulo Cavalcanti wrote:
> Hi, Lee
> 
> I did not test with 2.6.16. I am using Linux version
> 2.6.15-1.1833_FC4smp,
> from Fedora core 4.
> 
> What I know is that snd-bt87x works in kernel 2.6.12.
> It does not work in kernels, 2.6.13, 2.614, and 2.6.15.
> There is no entry in /proc/asound/cards for card 2 using these kernel
> versions, as you can see below, although all the pertinent modules are
> loaded.

Can you provide the "dmesg" output, and lspci -vn entry for the device?
It sounds like the OSS driver might be claiming the card.

Lee



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Re: [Alsa-user] ALSA: underrun, at least 0ms - fixed with an invalid asound.conf - I have searched around for an answer, honest

2006-03-27 Thread Lee Revell
On Mon, 2006-03-27 at 22:50 +0100, scott wrote:
> Thanks for the reply.  I take it that means something like "mpg321 -o 
> alsa -a plughw:0,0 ..."  I run that and I get:
> 
> Can't open libao driver with device plughw (is device in use?)
> 
> In case its important I dont have a /proc/asound/dev directory, could 
> that be a problem?

Do you have a sound server running, or another app that might be
blocking the sound device?

Lee



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Re: [Alsa-user] snd-bt87x

2006-03-27 Thread Lee Revell
On Mon, 2006-03-27 at 20:35 -0300, Paulo Cavalcanti wrote:
> There they go, Lee.
> 
> I still have kernel 2.6.12-1.1456_FC4smp installed.
> If I boot using it, the devices do appear (/dev/dsp2, /dev/audio2).
> 

bt878: AUDIO driver version 0.0.0 loaded
bt878: Bt878 AUDIO function found (0).

These messages are from the OSS driver.  Don't enable it, it conflicts
with the ALSA driver.

Lee




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Re: [Alsa-user] after alsaconf, no gnome events sounds

2006-03-27 Thread Lee Revell
On Mon, 2006-03-27 at 18:49 -0500, [EMAIL PROTECTED] wrote:
> One more detail: kernel is 2.6.8-2-k7.

That kernel is almost 2 years old, even if we found a bug it's way too
old to fix it.  Any chance you can try the latest kernel?

What is your alsa-lib (or libasound2 or whatever your distro decided to
call it) version?

Lee



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[Alsa-user] Re: hda-intel woes

2006-03-27 Thread Lee Revell
On Mon, 2006-03-27 at 20:22 -0500, Kyle Moffett wrote:
> Friedrich Göpel wrote:
> > This same message was sent to the alsa mailinglist 3 weeks ago, but  
> > it still seems to be waiting on being moderated, so I'm resending  
> > this here.
> 
> Hi!  Unfortunately messages tend to get lost on occasion due to the  
> sheer volume of email.  Sometimes you may have to resend a couple  
> times to get a response.  It also helps if you add relevant CC's from  
> the MAINTAINERS file (added to this message).  You can also file a  
> bug report on the ALSA bug tracker, I think the ALSA people tend to  
> be fairly good about keeping up with that; sometimes more so than the  
> mailing list.

First, try ALSA CVS - many hda-intel bugs have been fixed since
1.0.11-rc3.

Lee



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Re: [Alsa-user] snd-bt87x

2006-03-27 Thread Lee Revell
On Mon, 2006-03-27 at 23:54 -0300, Paulo Cavalcanti wrote:
> I think you mean btaudio.
> But it is not loaded.
> 
> lsmod shows nothing.
> 
> Where should I look?
> 

I may have been wrong about that.

Anyway, please provide the dmesg from the kernel that works.

It may be a udev problem - I know that new kernels can require a newer
udev version.  But I am not an expert on this or on Fedora.

Lee

> Thanks.
> 
> /Paulo Roma.
> 
> On 3/27/06, Lee Revell <[EMAIL PROTECTED]> wrote:
> On Mon, 2006-03-27 at 20:35 -0300, Paulo Cavalcanti wrote:
> > There they go, Lee.
> >
> > I still have kernel 2.6.12-1.1456_FC4smp installed.
> > If I boot using it, the devices do appear
> (/dev/dsp2, /dev/audio2). 
> >
> 
> bt878: AUDIO driver version 0.0.0 loaded
> bt878: Bt878 AUDIO function found (0).
> 
> These messages are from the OSS driver.  Don't enable it, it
> conflicts
> with the ALSA driver.
> 
> Lee
> 
> 
> 



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Re: [Alsa-user] after alsaconf, no gnome events sounds

2006-03-27 Thread Lee Revell
On Tue, 2006-03-28 at 00:21 -0500, [EMAIL PROTECTED] wrote:
> -Original Message-
> From: Lee Revell <[EMAIL PROTECTED]>
> To: [EMAIL PROTECTED]
> Cc: alsa-user@lists.sourceforge.net
> Sent: Mon, 27 Mar 2006 20:20:54 -0500
> Subject: Re: [Alsa-user] after alsaconf, no gnome events sounds
> 
>   On Mon, 2006-03-27 at 18:49 -0500, [EMAIL PROTECTED] wrote:
> > One more detail: kernel is 2.6.8-2-k7.
> 
> That kernel is almost 2 years old, even if we found a bug it's way too
> old to fix it.  Any chance you can try the latest kernel?
> 
> >The latest available from Debian is 2.6.8-3-k7. I'm using it now.
> 
> What is your alsa-lib (or libasound2 or whatever your distro decided to
> call it) version?
> 
> >It's really libasound2 and the version is 1.0.8-3.
> 
> >Since you mentioned it, I took a look in the files list of this package
> >and noticed that there are a few conf files, e.g. SI7018.conf.
> >I remember browsing some information on my motherboard and
> >found out something about SI7012 which has no file in the list.
> >Would it help in something?
> 
> > Thank you.
> 
> Lee
> 

It sounds like you needs to configure dmix for your card.  Later ALSA
versions do this by default.

Lee

> 
> ___
> Try the New Netscape Mail Today!
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Re: [Alsa-user] Compilation fails for alsa-1.0.11-rc4

2006-03-29 Thread Lee Revell
On Wed, 2006-03-29 at 21:40 +0530, Damodharan R wrote:
> 
> 
> On 29/03/06, Christoph Burger-Scheidlin <[EMAIL PROTECTED]> wrote:
> On Wednesday 29 March 2006 06:04, Damodharan R wrote:
> > Now I am able to play sound in totem. But when I open
> realplayer it says
> > device is used by another application. How to share the
> soundcard with more
> > than one program?
> 
> ALSA newer than 1.0.9 enables software mixing (=sharing) by
> default, so you
> don't need to configure anything. RealPlayer however uses OSS.
> To get it to
> play along, try aoss realplay.
> 
> Thanks for the info. aoss realplater worked fine. But I remember
> enablein oss support (--with-oss=yes) when compiling alsa-driver. Will
> it not take care of OSS applications.

No, it won't - the in-kernel OSS emulation cannot support software
mixing as mixing is implemented in userspace.  You must use aoss.

Lee



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Re: [Alsa-user] Unknown PCM default

2006-03-29 Thread Lee Revell
On Wed, 2006-03-29 at 16:49 -0800, Carlos Munoz wrote:
> Hi All,
> 
> I'm getting the following error when using an alsa application:
> 
> ALSA lib pcm.c:2099:(snd_pcm_open_noupdate) Unknown PCM default
> Error opening PCM device default
> 

This is an incomplete driver - it sounds like the userspace component is
missing.  Please provide a pointer to the source.

> The output of the /proc files is:
> 
> # cat /proc/asound/cards
> 0 [siu-sh7343 ]: siu-sh7343 - Renesas sh7343 SIU
>  Renesas sh7343 SIU irq 108
> 

What driver is this?  I don't see it in the ALSA tree.  There should be
a file /usr/share/alsa/cards/SIU-SH7343.conf that defines the default
PCM.

> # cat /proc/asound/devices
>   1:   : sequencer
>  18: [0- 2]: digital audio playback
>  26: [0- 2]: digital audio capture
>  17: [0- 1]: digital audio playback
>  25: [0- 1]: digital audio capture
>  16: [0- 0]: digital audio playback
>  24: [0- 0]: digital audio capture
>   0: [0- 0]: ctl
>  33:   : timer
> 
> Reading /cat/sndstat produces an error:
> 
>  # cat /dev/sndstat
> cat: /dev/sndstat: No such device
> 

This is a legacy OSS interface and would only be available if
snd-pcm-oss is loaded.

> I searched for the solution to this problem but didn't find anything. If 
> anyone knows how to fix this problem please let me know.




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Re: [Alsa-user] Unknown PCM default

2006-03-29 Thread Lee Revell
On Wed, 2006-03-29 at 17:50 -0800, Carlos Munoz wrote:
> It is a new driver for the Renesas SH7343 Sound Interface Unit that I 
> wrote. 

This is a better question for the alsa-devel list than alsa-user.

Just create the config file that defines a default PCM, or use hw:0,0 or
plughw:0,0.

Lee



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Re: [Alsa-user] Unknown PCM default

2006-03-29 Thread Lee Revell
On Wed, 2006-03-29 at 17:50 -0800, Carlos Munoz wrote:
> It is a new driver for the Renesas SH7343 Sound Interface Unit that I 
> wrote. The product is a new cell phone with a LCD display that will play 
> movie clips and audio to headphones. It worked well on my test board 
> when I was using manually compiled alsa-lib and alsa-utils. Now I'm 
> testing the software image that will be release to the customer and I 
> get this errors. This image includes alsa libraries that were built by a 
> different process (not sure about the differences though). Since the 
> driver worked before, I think it might be a configuration problem... but 
> I can't say for sure.
> 

It sounds like alsa-lib is missing or incorrectly installed.
Do /usr/share/alsa/alsa.conf and /usr/share/alsa/pcm/default.conf exist?

Lee

> Thanks,
> 
> 
> Carlos



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Re: [Alsa-user] How to use Surroundsound with alsa??

2006-03-30 Thread Lee Revell
On Thu, 2006-03-30 at 21:52 +0200, Marcel Witte wrote:
> I have searched with google and found a lot of /etc/asoundrc...but noone of 
> them worked correctly.

Most of the .asoundrc examples people post on the net are garbage.
Check the list archives for some better examples.

> My dream is that i can use 5.1-sound with dmix and automatic upmix from 
> 2channel-sound to 6channel. Is this possible??

There's still no easy way to do this.  There is some work in progress
(dmix improvements and new upmixing plugins) to make it easier.

For now you'll have to choose between dmix and stereo->5.1 upmixing (I
have never seen a working .asoundrc that does both).

Lee




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Re: [Alsa-user] How do I boost my mic level?

2006-03-31 Thread Lee Revell
On Fri, 2006-03-31 at 09:25 -0500, Bill White wrote:
> Gizmo's alsa support
> only uses "hw:0", and is not selectable. 

That's completely broken (though unrelated to your problem), please
report it to the Gizmo developers as a bug.

For example this can't work on hardware that only supports 48000Hz or
weird formats like S32_LE.  I would say this has no chance of working on
about half the hardware out there.

If they refuse to let the user select the sound device, at least use
"default" vs. hw:0.

Lee



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Re: [Alsa-user] How to use Surroundsound with alsa??

2006-03-31 Thread Lee Revell
On Fri, 2006-03-31 at 19:10 +0200, Marcel Witte wrote:
> > Most of the .asoundrc examples people post on the net are garbage.
> > Check the list archives for some better examples.
> >   
> I've searched in the ml-archive but found no useable example. Can anyone 
> send me a good example for 5.1-Sound (not SPDIF) and dmix ??
> > There's still no easy way to do this.  There is some work in progress
> > (dmix improvements and new upmixing plugins) to make it easier.
> >
> > For now you'll have to choose between dmix and stereo->5.1 upmixing (I
> > have never seen a working .asoundrc that does both).
> >   
> But dmix and normal 5.1-surround is possible??

I don't think dmix and 5.1 surround are compatible now.

Lee



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Re: [Alsa-user] How do I boost my mic level?

2006-03-31 Thread Lee Revell
On Fri, 2006-03-31 at 12:56 -0500, Bill White wrote:
> Lee Revell wrote:
> > If they refuse to let the user select the sound device, at least use
> > "default" vs. hw:0.
> This may be what they are doing.  I don't know enough about Alsa to
> use the right terminology reliably.
> 
> Is there a book or a web site which has an explanation of Alsa for
> non-experts?  I'm a programmer, and I write compilers and debuggers
> for a living, but I don't know much about sound or sound hardware.
> I do find myself spending more time than I'd like on Alsa
> configuration and trying to understand sound configuration problems
> in general.
> 

There's no good high level overview, just extensive API docs and
annotated example source code (plus of course the source to any ALSA
app).

> Like, for example, is there a program which probes the Alsa device
> space, in some abstract sense of "space", and reports the entire set
> of devices and their capabilities?  Is that what the mixers do?
> When I run alsamixer on my SBLive! card I get about 40 switches and
> sliders, but I have no idea what they all do, or even how to find
> out.

Unfortunately in many cases this would require more info about the
hardware than the ALSA developers have, so trial and error is often
required.

The SBLive! mixer is documented in SB-Live-mixer.txt and
Audigy-mixer.txt.

Lee



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Re: [Alsa-user] divided multichannel

2006-03-31 Thread Lee Revell
On Fri, 2006-03-31 at 13:59 -0500, Gordon McLellan wrote:
> I searched google a little, and this list archives, and I have found
> this very question asked several times, but no one ever offered a
> response.   So I take it this must be harder than it sounds? 

Nope, just the opposite - it's too easy to bother explaining ;-)

Configure your apps to output to the "front", "rear", and "center_lfe"
devices to use them as independent stereo pairs.

Or use JACK which will expose each input and output (up to 16 of each,
obviously not connected on all boards) as an independent mono port.

Lee



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Re: [Alsa-user] How to use Surroundsound with alsa??

2006-04-01 Thread Lee Revell
On Sat, 2006-04-01 at 16:55 +0200, Marcel Witte wrote:
>  - with -dplug:dmix51 :
>  AlsaPlayer 0.99.76
>  (C) 1999-2003 Andy Lo A Foe <[EMAIL PROTECTED]> and others.
>  ALSA lib pcm.c:6648:(snd_pcm_slave_conf) Unknown field 
> variable_buffer_size

Something is not installed correctly.  What is the output of:

grep -rI variable_buffer /usr/share/alsa/

?

Are you sure the OSS driver is not loaded for your card?

Lee



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Re: [Alsa-user] How to use Surroundsound with alsa??

2006-04-01 Thread Lee Revell
On Sat, 2006-04-01 at 20:57 +0300, Sergei Steshenko wrote:
> Humans (end users) are supposed to answer this question by looking at
> the output of some Linux utilities.
> 

The distro's init scripts/modutils/hotplug/udev configuration is
supposed to make sure that this is not possible.  Users who don't use a
distro or compile everything from source are expected to know what they
are doing. 

Anyway this is rarely a problem, it's just to make sure.

> Can't ALSA programs look at the output of the same utilities (by
> invoking
> them) and tell users that their ALSA attempts are senseless in the
> presence
> of OSS ? 

This would be bloat.  The UNIX philosophy is not to prevent users from
doing stupid things as this would also prevent them from doing clever
things.

Lee



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Re: [Alsa-user] How to use Surroundsound with alsa??

2006-04-01 Thread Lee Revell
On Sat, 2006-04-01 at 16:55 +0200, Marcel Witte wrote:
> - with -Dplug:dmix51 :
>  speaker-test 1.0.11rc4
> 
>  Playback device is plug:dmix51
>  Stream parameters are 48000Hz, S16_LE, 6 channels
>  Using 16 octaves of pink noise
>  ALSA lib pcm.c:6648:(snd_pcm_slave_conf) Unknown field 
> variable_buffer_size 

I have been able to reproduce this exact problem here.  Investigating.

Lee



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Re: [Alsa-user] How to use Surroundsound with alsa??

2006-04-01 Thread Lee Revell
On Sat, 2006-04-01 at 12:23 -0800, Bill Unruh wrote:
> On Sat, 1 Apr 2006, Marcel Witte wrote:
> 
> >> Are you sure the OSS driver is not loaded for your card?
> >> 
> > How can I check this?? I'm using SuSE 10.0...
> 
> lsmod|grep oss
> iAnd look for something that does not start with snd-
> 

This won't work, OSS modules have names like "i810_audio".

Lee




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Re: [Alsa-user] How to use Surroundsound with alsa??

2006-04-01 Thread Lee Revell
On Sun, 2006-04-02 at 00:07 +0300, Sergei Steshenko wrote:
> It is not obvious to new users that both ALSA and OSS exist; Linux
> distros sometimes use OSS driver instead of ALSA one, so end user
> might miss this. 

But any sane distro will hide this from the user.  It's not worth the
trouble to check for it at every level.  I only asked about OSS to rule
out an exceptional case.

Lee



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Re: [Alsa-user] How to use Surroundsound with alsa??

2006-04-01 Thread Lee Revell
On Sat, 2006-04-01 at 21:34 +0200, Marcel Witte wrote:
> Lee Revell schrieb:
> > On Sat, 2006-04-01 at 16:55 +0200, Marcel Witte wrote:
> >   
> >>  - with -dplug:dmix51 :
> >>  AlsaPlayer 0.99.76
> >>  (C) 1999-2003 Andy Lo A Foe <[EMAIL PROTECTED]> and others.
> >>  ALSA lib pcm.c:6648:(snd_pcm_slave_conf) Unknown field 
> >> variable_buffer_size
> >> 
> >
> > Something is not installed correctly.  What is the output of:
> >
> > grep -rI variable_buffer /usr/share/alsa/
> >
> > ?
> >   
> /usr/share/alsa/pcm/dmix.conf:  variable_buffer_size {
> /usr/share/alsa/pcm/dmix.conf:  name 
> defaults.pcm.dmix_variable_buffer
> /usr/share/alsa/alsa.conf:defaults.pcm.dmix_variable_buffer true
> > Are you sure the OSS driver is not loaded for your card?
> >   
> How can I check this?? I'm using SuSE 10.0...

Never mind, this is an alsa-lib bug.  I'm having the exact same problem
and I can't figure it out - it looks like it should work.  You'll have
to wait for a fix.

Lee



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Re: [Alsa-user] capture problems

2006-04-04 Thread Lee Revell
On Wed, 2006-04-05 at 01:47 +0200, phaidros wrote:
> hi,
> 
> I'm using 1.0.10rc3 on thinkpad t41p with intel 810 / ac97.
> 1. when I plug input in the line-in, there is always sound in the
> speakers, its impossible to mute.
> 2. i cannot activate line-in capturing (gnome-mixer, alsamixer). all i
> try keeps the sound present [see 1.], but in recording app its around
> -50db .. 
> 
> any ideas?
> could this be some weired bios setting?
> or is it even a known alsa bug?

Please try the latest ALSA release, 1.0.11-rc4.

Lee



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Re: [Alsa-user] Re: Problems with SiS 7018 sound chipset (semaphore not ready)

2006-04-05 Thread Lee Revell
On Wed, 2006-04-05 at 17:00 +0200, Clemens Eisserer wrote:
> Hi again,
> 
> Does really nobody have an idea where that could come from.
> Just what I guess it looks like someone holds a semaphore locked (does
> this locking for registers happen inside of alsa?), maybe it does not
> get released since the driver still stucks in initialisation. Could
> that be?
> Any ideas how I could investigate this further or which useful
> informations could shed some more light into this issue?
> 
> Live's so boring without sound :-(
> 

Can you try to determine whether it was a hardware or kernel change that
introduced the problem?

Lee




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Re: [Alsa-user] Re: Problems with SiS 7018 sound chipset (semaphore not ready)

2006-04-05 Thread Lee Revell
On Wed, 2006-04-05 at 17:00 +0200, Clemens Eisserer wrote:
> Hi again,
> 
> Does really nobody have an idea where that could come from.
> Just what I guess it looks like someone holds a semaphore locked (does
> this locking for registers happen inside of alsa?), maybe it does not
> get released since the driver still stucks in initialisation. Could
> that be?
> Any ideas how I could investigate this further or which useful
> informations could shed some more light into this issue?
> 
> Live's so boring without sound :-(

Please try to identify the last kernel that worked.  If all that was
changed was hardware, it could be a hardware problem.

Lee



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Re: [Alsa-user] ALSA sees applications making the device/resource busy.

2006-04-05 Thread Lee Revell
On Wed, 2006-04-05 at 17:53 +0200, Tanim Islam wrote:
> Hi:
> 
> I have what I believe to be a generic problem on my Kubuntu breezy machine 
> (Ubuntu 5.10), and I was wondering if there was a transparent way to debug 
> ALSA. That is, is there a list of system log files that the ALSA daemon or 
> its equivalent outputs? Also, i get the error that the sound device is busy. 
> Is there a way to determine which application or service is hogging this 
> sound resource? For reference, here is my asoundrc file

If you have ALSA 1.0.9 or later multiple streams should work with
no .asoundrc.

Which device is giving you the error?

Lee



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Re: [Alsa-user] Re: Problems with SiS 7018 sound chipset (semaphore not ready)

2006-04-05 Thread Lee Revell
On Wed, 2006-04-05 at 13:46 -0400, Lee Revell wrote:

> Please try to identify the last kernel that worked.  If all that was
> changed was hardware, it could be a hardware problem.

Argh, sorry for the double reply, my mailer crashed (again) while
sending this...

Lee



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Re: [Alsa-user] .asoundrc and dsp device

2006-04-06 Thread Lee Revell
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 is oss application). I usually use the PCI
> card but sometimes want to use the onboard. So I tried this way: 
>  
>  - in .asoundrc introduced this:
>  
>pcm.dsp0 {
>  type hw
>  slave.pcm "hw:1,0"
>}
>  
>  - launch tvtime like this:
>   
>   aoss tvtime
>  
> But the sound continues to come from the PCI card. I know that it
> works by introducing this in .asoundrc:
>  
>   pcm.!default {
> type hw
> card 1
>   }
>  
> But shouldn't it work with the previous solution. Am I missing
> something here?

No, because aoss uses the "default" device, not "dsp0".

Isn't the easiest way to switch to set tvtime to output to /dev/dsp1?

Lee



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Re: [Alsa-user] on the fly routing

2006-04-06 Thread Lee Revell
On Thu, 2006-04-06 at 19:50 -0400, Gordon McLellan wrote:
> is it possible to (re)route an audio stream (source) such as mpg321
> "on the fly" to one or more sound cards?
> 
> by on the fly, I mean, without interrupting the source, ie ctrl-c and
> reissue the command with a different output device
> 
> I've designed an digitally controlled analog signal router, but it's
> looking like it'll cost $100 to actually build a few (pcb is very
> complex and costly to have fabricated in small quanity) ... so if I
> can do it in software, great!

You can do it with JACK, if the app has JACK support or you can get it
to work with oss2jack or the JACK ALSA plugin.

You have to use a solution like jack_diplomat if you want to use 2 cards
that aren't synced by word clock.

Lee



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Re: [Alsa-user] wrong modes for /dev/desp and /dev/mixer

2006-04-06 Thread Lee Revell
On Fri, 2006-04-07 at 07:33 +0200, Bernhard Kleine wrote:
> I had troubles recording from the microphone; after changing the modes
> 
> $ls -laF /dev/dsp
> crw-rw 1 root audio 14, 3 2006-04-06 21:47 /dev/dsp
> $ ls -laF /dev/mixer
> crw-rw 1 root audio 14, 0 2006-04-06 21:47 /dev/mixer
> 
> to 
> 
> chmod 666 /dev/dsp
> chmod 666 /dev/dsp
> 
> no further problem with recordings.
> 
> Since this 660 mode is the default, I have to adjust these settings
> after
> every reboot. I wonder where the default settings come from and wether
> I
> can change that to a more practical default. 
> 

Just add yourself to the "audio" group:

As root:

# gpasswd -a your_user_name audio

Then log out and back in.

Lee



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Re: [Alsa-user] sound in the Debian distro

2006-04-07 Thread Lee Revell
On Fri, 2006-04-07 at 08:26 -0500, Brian wrote:
> Having gotten the sound-application "Audacity" running well in 
> Mandrake, FC3, and SUSE, I decided to try getting sound running in 
> Debian.  I am using the version R4 (kernel 2.2. called "Woody"), and 
> during the install, used "tasksel" for the big-picture and "dselect" for 
> the fine-tuning parts. I saw the ALSA modules, and selected everything 
> that had "sound" or "ALSA" mentioned.  Although I want to get Audacity 
> with this distro, I had focused on getting basic "sound capabilities" 
> running first.
> 
>However, when the system started, right away it states "sound server 
> could not be started". It mentions something about "soundcore.o" not 
> being found. Running "sndconfig" does not find any "sound modules" at 
> all.

This is more of a Debian bug, not an ALSA issue, you would probably have
better luck getting an answer on a Debian specific mailing list.

Lee 



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Re: [Alsa-user] Attempts to influence sound card ordering: a report

2006-04-07 Thread Lee Revell
On Fri, 2006-04-07 at 11:27 -0400, Gene Heskett wrote:
> >In my case /etc/modules.conf is generated from
> >alsa-base and other files in /etc/modutils by running
> >update-modules.
> >
> >I made the following additions to /etc/modutils/alsa-base,
> >
> >options snd-via82xx index=0
> >options snd-ice1712 index=1
> >
> >run update-modules, and rebooted.
> > 

If you have multiple options lines then all but the first will be
ignored.

This is really more of a Debian bug than a general ALSA issue.

Lee



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Re: [Alsa-user] static

2006-04-07 Thread Lee Revell
On Fri, 2006-04-07 at 10:57 -0500, Jason Greene wrote:
> SB Live Platinum $15 (used)
> SB Live $15 (used) 

Either of these would be fine (as long as it's not the "SB Live! 24
bit")

Lee



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Re: [Alsa-user] Attempts to influence sound card ordering: a report

2006-04-07 Thread Lee Revell
On Fri, 2006-04-07 at 18:27 +0200, Matthias Koenig wrote:
> The correct Debian location for configuration files for new kernels
> is /etc/modprobe.d/.
> If you append the 2 option lines above to e.g. alsa-base it should
> work.
> You don't have to run update-modules, because it is obsolete.

Seems a bit silly that such a simple task requires editing obscure
config files, that don't have a stable interface, and that you had to
post to this list to figure it out.

Does anyone know if any distro has solved this problem cleanly?  I know
Ubuntu has a nice interface for setting the default soundcard for ALSA
apps, but I don't think it can control the loading of drivers (which
determines which card binds to /dev/dsp for example).

Lee



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Re: [Alsa-user] Attempts to influence sound card ordering: a report

2006-04-07 Thread Lee Revell
On Fri, 2006-04-07 at 10:16 -0700, Richard Fish wrote:
> On 4/7/06, Lee Revell <[EMAIL PROTECTED]> wrote:
> > Does anyone know if any distro has solved this problem cleanly?
> 
> Gentoo has /etc/modules.autoload.d/kernel-2.6, where you can place
> modules to be loaded on boot, and they are loaded in the same order
> they appear in the file.  The loading occurs almost immediately after
> /dev is setup.

By cleanly I mean a generic method preferably with a GUI that is not
distro specific (even a KDE/GNOME specific app would be an improvement
over a different method for each distro).

And it's not a kernel problem, the kernel already provides mechanisms to
control the loading of modules (the module options, hotplug, udev, etc).
There just needs to be a reasonable interface to it.

And BTW it's fine to use Reply-To-All for all alsa-user mail.

Lee



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Re: New Card (was Re: [Alsa-user] static)

2006-04-07 Thread Lee Revell
On Fri, 2006-04-07 at 13:05 -0500, Jason Greene wrote:
> OK, I put in the SB Live card. It has the emu10k1 chip on it
> 
> alsaconf still won't see the card.
> 
> Slackware 10.2 standard install alsa-driver-1.0.9b
> 
> Any Ideas what could be going on?
> 

Yes - don't load the "emu10k1" driver, it's an obsolete OSS driver.

You want to be loading "snd-emu10k1".

> 
> [EMAIL PROTECTED]:~#  lspci | grep audio
> 01:04.0 Multimedia audio controller: Creative Labs SB Live! EMU10k1
> (rev 04)
> 
> 
> [EMAIL PROTECTED]:~# lsmod
> Module  Size  Used byNot tainted
> emu10k159020   0  (unused)
> sound  56616   0  [emu10k1]
> ac97_codec 13556   0  [emu10k1]
> soundcore   3652   7  (autoclean) [emu10k1 sound]
> parport_pc 16196   1  (autoclean)
> lp  6628   0  (autoclean)
> parport25064   1  (autoclean) [parport_pc lp]
> ipt_LOG 3480   1  (autoclean)
> ipt_state568   1  (autoclean)
> ip_conntrack   19272   0  (autoclean) [ipt_state]
> iptable_filter  1772   1  (autoclean)
> ip_tables  12096   3  [ipt_LOG ipt_state iptable_filter]
> eepro100   19540   1
> mii 2400   0  [eepro100]
> usb-ohci   19496   0  (unused)
> usbcore62220   1  [usb-ohci]
> emu10k1-gp  1288   0  (unused)
> gameport1484   0  [emu10k1-gp]
> pcmcia_core41924   0
> aic7xxx   149688   2
> ide-scsi   10192   0
> 



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Re: [Alsa-user] Attempts to influence sound card ordering: a report

2006-04-07 Thread Lee Revell
On Fri, 2006-04-07 at 20:15 +0200, Matthias Koenig wrote:
> 
> What about hotpluggable devices which will just pop up at some random
> time?
> You can't just load module snd-xyz on boot time, if device xyz is not
> there 
> at that moment. 

I've asked about this on the Ubuntu lists, they are really at the
forefront of Linux desktop usability (they consider it a bug if a user
has to edit config files to get basic functionality to work).  Maybe
they'd be willing to put up a bounty for someone to fix this once and
for all.

Lee



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