Re: [Alsa-user] Duplexing (?) Audiophile 2496

2006-09-16 Thread Lee Revell
On Sat, 2006-09-16 at 12:05 -0500, Bob van der Poel wrote:
> I recently installed a M-Audio Audiophile 2496 in machine running Mandriva 
> 2006.
> 
> All, including MIDI, runs just fine. But, it appears that I can only listen 
> to a single stream at one time.
> 
> I _think_ that it has something to do with the setup of either the 
> /etc/asound.state or the .asoundrc files.
> I have copied/pasted some .asoundrc examples from this mailing list, but find 
> no difference at all.
> 
> My alsa version is 1.0.9b.

ICE1712 has no hardware mixing, so this is the expected behavior with
that ALSA version.

Upgrade to a recent ALSA and software mixing (dmix) will be
automatically enabled for this card, but you'll have better results just
using JACK.

Lee


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Re: [Alsa-user] Duplexing (?) Audiophile 2496

2006-09-16 Thread Lee Revell
On Sat, 2006-09-16 at 17:52 -0500, Bob van der Poel wrote:
> Thanks, Lee. I'll see if I can get JACK running.
> 
> But, if there is no hardware mixing ... then what is the point of all those 
> neat little VUE meters in envy24control?
> 

The term "mixing" is overloaded.  The device does have a hardware mixer
that allows one to set the levels of various input and output channels.
What it does not have is the ability to mix multiple audio streams from
different applications.

Lee


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Re: [Alsa-user] Duplexing (?) Audiophile 2496

2006-09-16 Thread Lee Revell
On Sat, 2006-09-16 at 19:15 -0500, Bob van der Poel wrote:
> So, it appears that jack is running. But, when I try to run audio I
> get "busy" error messages. The
> most clear comes from mpg123:
> 
> mpg123 *Yell*
> High Performance MPEG 1.0/2.0/2.5 Audio Player for Layer 1, 2 and 3.
> Version 0.59r (1999/Jun/15). Written and copyrights by Michael Hipp.
> Uses code from various people. See 'README' for more!
> THIS SOFTWARE COMES WITH ABSOLUTELY NO WARRANTY! USE AT YOUR OWN RISK!
> /dev/dsp: Device or resource busy
> /dev/dsp: Device or resource busy
> /dev/dsp: Device or resource busy
> /dev/dsp: Device or resource busy
> /dev/dsp: Device or resource busy
> /dev/dsp: Device or resource busy
> /dev/dsp: Device or resource busy
> audio: Device or resource busy
> 

OSS apps like mpg123 don't work with ALSA's software mixing, unless you
run them with the aoss wrapper.

If you run JACK all your applications must be JACK capable.  mpg123 is
not.

> Your help on all this IS appreciated. I just don't get what's going
> on.
> 

The designers of this device assumed that mixing of multiple PCM streams
output by applications would be handled in software (like Windows, JACK,
and recent versions of ALSA do).

Lee




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Re: [Alsa-user] Audio drop-out with ac3 on iec958 during heavy SATA io

2006-09-19 Thread Lee Revell
On Tue, 2006-09-19 at 18:13 +0200, Takashi Iwai wrote:
> Well, you've seen the xrun detection there, so it's basically the
> problem of other parts, such as disk contoller driver.  That is,
> something else takes too long time, and the irq for sound can't be
> handled at the right time.
> 
> If the preempt option doesn't solve the problem, I'd recommend you to
> consult on LKML.
> 

Does this help?

# echo 64 > /sys/block/hdc/queue/max_sectors_kb

Replace hdc with sda or whatever the kernel calls your disk and repeat
for all disks.

It's still a bug IMHO if this is needed, but might make your system
usable until the SATA maintainers fix it.

Lee




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Re: [Alsa-user] What are all those devices?

2006-09-19 Thread Lee Revell
On Tue, 2006-09-19 at 21:58 +0100, Mario Rossi wrote:
> Hi,
> 
> I would like to know what is the second device Intel ICH - IEC958.
> If I try to play via it no sound is emitted
> 
> aplay -D plughw:0,4 somefile.wav
> 
> nothing happens.
> 
> The first device is the actual device, while the third is the modem,
> what is the second?
> 
> card 0: I82801DBICH4 [Intel 82801DB-ICH4], device 4: Intel ICH -
> IEC958 [Intel 82801DB-ICH4 - IEC958]
  ^^

IEC958.  AKA S/PDIF.

> the same for the capture

1 and 2 are MIC and MIC2 (just as the description says).  3 is some kind
of ADC, but the driver has no way of knowing what it's connected to, so
just calls it ADC.

Lee


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Re: [Alsa-user] C-Media USB soundcard (usb id 0d8c:000c) unable to get to work.

2006-09-19 Thread Lee Revell
On Tue, 2006-09-19 at 14:48 -0700, Jose Gilberto Torres wrote:
> aplay -D hw:1,0 ~/GNUstep/Library/WindowMaker/Sounds/unhide.wav
> Playing WAVE 
> '/home/torresj/GNUstep/Library/WindowMaker/Sounds/unhide.wav' :
> Unsigned 
> 8 bit, Rate 22050 Hz, Mono
> aplay: set_params:901: Sample format non available
> 

Try plughw:1,0.

> I was unable to get any of my media players (xmms, mplayer, etc) to
> work 
> either.   I'm running Fedora Core 5.  Any suggestions?
> 

Both of those have buggy ALSA support.

Lee


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Re: [Alsa-user] switch between two soundcards

2006-09-19 Thread Lee Revell
On Wed, 2006-09-20 at 00:18 +0200, Jürgen Herrmann wrote:
> 
> my goal is that applications can always use alsa hw:0:0 without
> reconfiguration, maybe this could even be automated via hotplug
> when (dis)connecting the usb device? 

Applications shouldn't use hw:0,0 anyway, it only supports one app
playing at a time, and will fail if the sample rate and format don't
exactly match the device's capabilities.

The correct solution is for apps to use the ALSA device "default", and
for a userspace application (like gnome-sound-properties) to change the
"default" device when sound cards are plugged or unplugged or when the
user requests it.

There's no way to have apps that are playing audio when the device
changes to seamlessly switch to the other device though.

Lee


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Re: [Alsa-user] What are all those devices?

2006-09-20 Thread Lee Revell
On Wed, 2006-09-20 at 22:04 +0100, Mario Rossi wrote:
> Thanks for your answer, but I don't know much about digital audio component.
> On my laptop there are
> 
> 1) integrated MIC (is it MIC1?)
> 2) integrated speakers
> 3) port for external MIC (is it MIC2?)
> 4) port for external speakers
> 
> I've tried to record and play through the devices to guess what they
> correspond to, but I had no luck.
> I'm curious to know when to use them and what to use them for.

No idea.  There's no way to tell how they're really wired up.  They may
not be connected to anything - ALSA has no way of knowing.

Does your mic work?

Lee


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Re: [Alsa-user] strange period size behaviour

2006-09-20 Thread Lee Revell
On Thu, 2006-09-21 at 03:24 +0200, Tomas Carnecky wrote:
> Without a .asoundrc, alsaplayer gives this: PERIOD_TIME: (21333 21334)
> I can play several sounds at the same time through dmix.
> 
> But, with a .asounrc that has the following:
> 
> cards.ICE1724.pcm.dmix.period_time 1024
> 
> (note that 1024 is the default value for dmix, so that shouldn't change 
> anything), alsaplayer gives: PERIOD_TIME: (41 2048000], and I can play 
> only one sound, alsaplayer fails with: snd_pcm_open: Device or resource 
> busy (default)
> 

You are mixing up period_time (which is in microseconds) with
period_size (which is in frames).  period_time defaults to 125000.

Why do you want to use an .asoundrc at all?

Lee


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Re: [Alsa-user] strange period size behaviour

2006-09-21 Thread Lee Revell
On Thu, 2006-09-21 at 08:55 +0200, Tomas Carnecky wrote:
> Lee Revell wrote:
> > You are mixing up period_time (which is in microseconds) with
> > period_size (which is in frames).  period_time defaults to 125000.
> > 
> > Why do you want to use an .asoundrc at all?
> > 
> 
> I want to set period size or time to a value so that the buffer is 16kb 
> - because that's what the application expects. For now I simply made the 
> buffer slightly bigger (18kb) and 'force' the application to use all of 
> it, but I hear periodic artifacts and I think it has something to do 
> with the buffer size since the pauses between them increase 
> proportionally with the buffer size.

Well, you won't ever get true "zero-copy" with this device, because the
hardware only supports S32_LE - software format conversion will be
required.

Lee


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Re: [Alsa-user] [Alsa-devel] USB card @48kHz only: at what layer to resample?

2006-09-24 Thread Lee Revell
On Sun, 2006-09-24 at 18:41 +0200, [EMAIL PROTECTED] wrote:
> Thanks Markus and all for the replies.
> 
> My problem is that STK does not seem able to see the alsa plugins.
> 
> In other words, when one uses -say- aplay it is possible to specify what 
> device or plugin to use with the -D option, for example:
> -D plug:duplex
> as you suggest in the example.
> 
> In the STK api for alsa I can only select one of the audio cards, in my case 
> I see the following 4:
> 
> hw:I82801CAICH3,0
> hw:I82801CAICH3,1
> hw:U0x4710x311,0
> hw:Audio,0
> 
> The first two seem to be related to the built-in laptop sound card (intel 
> chip based) (why 2 entries???)

Probably analog and SPDIF.

> , the third one is probably the mic input of the webcam and the last one is 
> my terratec.
> I can only select one of these by number (i.e. I can select 1, 2, 3 or 4)
> 

This is an incorrect use of the ALSA API by STK (It doesn't help that
ALSA has no good device enumeration API).  It should allow you to choose
plughw:* devices, but it only lists hw:* devices.  This means that only
sample rates supported by the hardware can be chosen.

> If I try to open the last one at 44.1kHz I get: 
> RtApi: unable to open specified device(s) with given stream parameters:
> RtApiAlsa: error setting sample rate (44100) on device (hw:Audio,0): 
> Invalid argument.
> 
> If I open it at 48kHz it is fine.
> 
> In general, from the discussion I understand that the best (but probably long 
> term for me) solution is to change my code to work at 48kHz, but for now it's 
> really important to get the old code to work. 
> 

You just need to patch STK to allow plughw devices to be used, or give
up and run at 48Khz.  There's no other solution.

> Does this make any sense?
> Any suggestions?
> 
> Thanks,
> Enrico
> 
> -- Initial Header ---
> 
> >From  : "Markus Rechberger" [EMAIL PROTECTED]
> To  : "[EMAIL PROTECTED]" [EMAIL PROTECTED]
> Cc  : "alsa-user" alsa-user@lists.sourceforge.net,"alsa-devel" 
> alsa-devel@lists.sourceforge.net
> Date  : Sun, 24 Sep 2006 16:32:23 +0200
> Subject : Re: [Alsa-devel] USB card @48kHz only: at what layer to resample?
> 
> 
> 
> 
> 
> 
> 
> > Hi,
> > 
> > you could use the alsa rate plugin.
> > 
> > I used following .asoundrc for testing my driver:
> > 
> > pcm.au600playback {
> > type dmix
> > ipc_key 1024
> > ipc_perm 0777
> > slave {
> > pcm "hw:2,0"
> > period_time 0
> > period_size 96
> > buffer_size 1920
> > format S16_LE
> > periods 4
> > rate 9600
> > }
> > }
> > 
> > pcm.au600capture {
> > type rate
> > slave {
> > pcm "hw:2,1"
> > format S16_LE
> > rate 9600
> > }
> > }
> > pcm.duplex {
> > type asym
> > playback.pcm "au600playback"
> > capture.pcm "au600capture"
> > }
> > 
> > aplay -D plug:duplex 
> > 
> > cheers,
> > Markus
> > 
> > On 9/24/06, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
> > > Dear All,
> > >
> > > Apologies in advance if I am asking a stupid question, I am quite new with
> > > alsa.  I posting to both lists because I am not sure about handling this
> > > issue at the application development level or at the alsa driver
> > > configuration level.
> > >
> > > I am developing an audio application using the CCRMA synthesis tool-kit
> > > (STK) in Fedora Core4 using alsa.  I recently bought a Terratec Aureon 5.1
> > > USB MKII card, because I needed to have better sound quality than what
> > > offered by my laptop.
> > >
> > > My understanding is that the usb card only works at 48kHz -- is this
> > > correct?
> > > I manage to get aplay to work using the following plugin in 
> > > /etc/asound.conf
> > > pcm.!default {
> > > type plug
> > > slave {
> > > pcm "hw:2,0"
> > > rate 48000
> > > }
> > > }
> > >
> > > However, in SKT I can enumerate the audio cards on my machine, but this 
> > > does
> > > not give me alsa plugins, only the audio cards.  If I try to open the 
> > > Aureon
> > > at 44.1kHz, I get an error saying rate not supported.
> > >
> > > Is there any way to have alsa handle the rate conversion for me?  
> > > Shouldn't
> > > or couldn't the driver take care of this?  (Was it my mistake in 
> > > installing
> > > the driver? I can post my /etc/modprobe.conf if it helps)
> > > Or is there any way to get STK to see the plugin?
> > >
> > > Thank you in advance,
> > > Enrico
> > >
> > >
> > >
> > >
> > >
> > > -
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Re: [Alsa-user] [Alsa-devel] USB card @48kHz only: at what layer to resample?

2006-09-24 Thread Lee Revell
On Sun, 2006-09-24 at 08:12 -0700, Bill Unruh wrote:
> This is audio card insanity. All CDs are at 44100 but for some reason
> many card manufacturers designed their cards to run at 48000. This is
> a real problem since this requires rate conversion. Rate conversion is
> in general a very very noisy operation. In particular simple linear
> interpolation, which is quick, is also really really terrible for
> sound quality.
> 

DVDs use 48000.  But I seem to recall vendors shipping 48Khz-only cards
before playing DVD on a computer was common...

Lee


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Re: [Alsa-user] /dev/dsp not found with kernel 2.6.16.29

2006-09-24 Thread Lee Revell
On Sun, 2006-09-24 at 21:37 +0200, [EMAIL PROTECTED]
wrote:
> Hallo alsa-users,
> 
> Currently I'm running alsa driver 1.0.12 with kernel 2.6.16.27.
> This afternoon I tried to use kernel 2.6.16.29. I used 'make
> oldconfig' with the config of kernel 2.6.16.27.
> That made alsa complain that /dev/dsp could not be found.
> But it *does* exist: /dev/dsp -> sound/dsp, permissions 664,
> owned by root:audio.
> What might be the problem?

(Well, aside from the fact that apps shouldn't be using /dev/dsp in
2006...)

Does "aplay file.wav" work?

Lee




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Re: [Alsa-user] /dev/dsp not found with kernel 2.6.16.29

2006-09-25 Thread Lee Revell
On Mon, 2006-09-25 at 11:56 +0200, [EMAIL PROTECTED]
wrote:
>  LR> Does "aplay file.wav" work?
> 
> That produces the following:
> 
> === import ===
> ALSA lib confmisc.c:670:(snd_func_card_driver) cannot find card '0'
> ALSA lib conf.c:3479:(_snd_config_evaluate) function snd_func_card_driver
> returned error: No such device
> ALSA lib confmisc.c:391:(snd_func_concat) error evaluating strings
> ALSA lib conf.c:3479:(_snd_config_evaluate) function snd_func_concat
> returned error: No such device
> ALSA lib confmisc.c:1070:(snd_func_refer) error evaluating name
> ALSA lib conf.c:3479:(_snd_config_evaluate) function snd_func_refer returned
> error: No such device
> ALSA lib conf.c:3947:(snd_config_expand) Evaluate error: No such device
> ALSA lib pcm.c:2143:(snd_pcm_open_noupdate) Unknown PCM default
> aplay: main:550: audio open error: No such device
> === tropmi ===
> 

This is probably a udev bug.  Try updating to the latest udev package
your distro offers.

Lee


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Re: [Alsa-user] Pb with simultaneous SATA and ALSA I/O

2006-09-25 Thread Lee Revell
On Mon, 2006-09-25 at 10:38 -0400, Lennart Sorensen wrote:
> Well i agree with the suggestion of trying a different PCI slot for
> the sb live.  There is so much onboard stuff sharing interrupts on
> those boards that you might have problems because of that.  Creative
> cards are not very good at dealing with anything other than ideal
> conditions from what I have gathered over the years.  The manual for
> the board will tell you which IRQ goes to which slot, and I guess you
> want to avoid using a slot that shares with the SATA controller. 

It might not be interrupt related, it could be DMA starvation.  This has
been observed with some SATA controllers while testing the -rt patches.
The symptom is that the latency traces show the machine going in "slow
motion".

Dominique: try the -rt kernel, enable latency tracing and post the
output.

Lee


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Re: [Alsa-user] loading snd-ca0106 fails for Audigy Live! 7.1 on Debian

2006-09-26 Thread Lee Revell
On Tue, 2006-09-26 at 11:50 +0200, Till Wimmer wrote:
> i bought a CS Audigy Live! 7.1 and put it in my Debian Sarge
> 2.6.8-3-68 
> box...
> 
> The card is detected by alsaconfig, but the kernel module doesn't
> load 
> correctly. 

You need a much newer kernel than 2.6.8 for a card this new.

Try something recent like 2.6.18

Lee


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Re: [Alsa-user] Pb with simultaneous SATA and ALSA I/O

2006-09-26 Thread Lee Revell
On Tue, 2006-09-26 at 16:37 +0200, Dominique Dumont wrote:
> Lee Revell <[EMAIL PROTECTED]> writes:
> 
> > Dominique: try the -rt kernel, enable latency tracing and post the
> > output.
> 
> Should I use 2.6.17 + rt to stay closer to my current setup or should I get 
> 2.6.18 + rt ?
> 

I would say 2.6.18 + -rt - it's not as if 2.6.17 could be fixed at this
point.

Lee


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Re: [Alsa-user] RME Digi96/8 PST aplay: Channels count not available

2006-09-26 Thread Lee Revell
On Tue, 2006-09-26 at 11:15 +0200, Ismael Valladolid Torres wrote:
> Jack works nicely through this card although I'm still not being able
> to output sound via S/PDIF.
> 
> So to test S/PDIF I tried a simple aplay, however it's failing
> strangely:
> 
> $ aplay -D hw:1,0 mlk.wav
> Playing WAVE 'mlk.wav' : Signed 16 bit Little Endian, Rate 22050 Hz, Mono
> aplay: set_params:906: Channels count non available
> 

Use plughw:1,0

> The same using this .asoundrc suggested in AlsaOpensrcOrg:
> 
> $ cat .asoundrc
> pcm.rme_stereo {
> type hw
> card 1
> }
> ctl.rme_stereo {
> type hw
> card 1
> }
> pcm.rme_adat {
> type hw
> card 1
> device 1
> }
> ctl.rme_adat {
> type hw
> card 1
> }

Delete the .asoundrc, it's useless.

Lee


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Re: [Alsa-user] hda-intel on Thinkpad T60p fails to load with unable to handle kernel paging request

2006-09-26 Thread Lee Revell
On Tue, 2006-09-26 at 11:02 +0200, Eivind Tagseth wrote:
> Sep 26 09:09:10 [kernel] EIP:0060:[]Tainted: GF

Why did you forcibly load a module (insmod -f)?

Lee


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Re: [Alsa-user] can't seem to get ac3 passthrough working with ymfpci

2006-09-28 Thread Lee Revell
On Thu, 2006-09-28 at 14:51 -0400, Brian J. Murrell wrote:
> I've been trying and trying and I just can't seem to get DD/DTS/AC3 to
> work.  My receiver receives PCM just fine so at least I know the
> physical bits are working.
> 
> I have tried both mplayer and ac3dec to drive the card into decoding AC3
> but no luck.  I believe I have the mixer settings correct.  In fact I
> think I have everything unmuted and at somewhere in the 2/3 volume
> range.  I have IEC958 Loop disabled/muted and Direct Recording Source
> set to IEC958.
> 

What hardware?  I think AC3 passthrough cannot work on Nvidia HDA
chipsets doe to proprietary undocumented hardware implementation.

Can you try the latest ALSA release?

Lee



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Re: [Alsa-user] can't seem to get ac3 passthrough working with ymfpci

2006-09-28 Thread Lee Revell
On Thu, 2006-09-28 at 15:17 -0400, Brian J. Murrell wrote:
> What is the latest?  I am using
> 
> Advanced Linux Sound Architecture Driver Version 1.0.10. 

1.0.12 is the latest stable release, 1.0.13-rc? the latest development
release.

So you use the ymfpci driver?

Clemens, don't you have this hardware?  Should it work?

Lee


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Re: [Alsa-user] set_params:901: Sample format non available

2006-09-30 Thread Lee Revell
On Sat, 2006-09-30 at 11:07 +0200, Udo van den Heuvel wrote:
> This stuff worked under Windoze.
> How can I fix this?
> Am I doing something wrong?

Yes.  Use plughw:1 rather than hw:1.  Or just use default:1 for
automatic rate, format, channel conversion and dmix.

Lee


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Re: [Alsa-user] Drop outs under I/O load with Terratec EWX 24/96

2006-09-30 Thread Lee Revell
On Sat, 2006-09-30 at 15:21 +0200, Dominique Dumont wrote:
> Jascha Berberich <[EMAIL PROTECTED]> writes:
> 
> > While there is I/O load on the PCI bus on my mainboard (Asus A7N8X-E
> > deluxe) I got heavy drop outs (xruns) and the sound seems to lower
> > its speed (as an walkman with an empty battery), regardless by witch
> > device the load is produced (the most ugliest drop outs I had with
> > load produced by my S-ATA Seagate harddrives, but also when copying
> > a CD on the harddrives or when having heavy network traffic).
> 
> Uh oh, this looks like the problem I have with my a7n8x deluxe board +
> sata IO + drop-out .
> 
> See the threads
> http://www.mail-archive.com/alsa-user@lists.sourceforge.net/msg17399.html
> http://www.mail-archive.com/alsa-user@lists.sourceforge.net/msg17641.html
> 
> I'm still digging the issue. 
> 
> Lee Revell asked me to "try the -rt kernel, enable latency tracing and
> post the output." So far, I've can't boot this rt kernel (Still trying
> though)
> 

What are the symptoms?  -rt seems to work for everyone on linux-kernel
but no one on this list.  Can you provide the boot messages?
Your .config?

You've tested the latest 2.6.18 -rt patch?

> If you could get these traces on your side, it would be helpfull.
> 
> Thanks
> 
> 
> 
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Re: [Alsa-user] Drop outs under I/O load with Terratec EWX 24/96

2006-09-30 Thread Lee Revell
On Sat, 2006-09-30 at 18:39 +0200, Dominique Dumont wrote:
> Lee Revell <[EMAIL PROTECTED]> writes:
> 
> >> Lee Revell asked me to "try the -rt kernel, enable latency tracing and
> >> post the output." So far, I've can't boot this rt kernel (Still trying
> >> though)
> >> 
> >
> > What are the symptoms?  -rt seems to work for everyone on linux-kernel
> > but no one on this list.  Can you provide the boot messages?
> 
> I get something like a ooops backtrace involving irq. I did not find
> any trace of this oops in /var/log so the boot message is lost. 
> 

Can you take a picture of the screen with a digital camera and post it
somewhere?

> > Your .config?
> 
> See below
> 

I'll try to build this.

> 
> > You've tested the latest 2.6.18 -rt patch?
> 
> Yes. rt2 patch
> 

-rt5 is the latest

Lee


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Re: [Alsa-user] Drop outs under I/O load with Terratec EWX 24/96

2006-09-30 Thread Lee Revell
On Sat, 2006-09-30 at 18:39 +0200, Dominique Dumont wrote:
> Lee Revell <[EMAIL PROTECTED]> writes:
> 
> >> Lee Revell asked me to "try the -rt kernel, enable latency tracing and
> >> post the output." So far, I've can't boot this rt kernel (Still trying
> >> though)
> >> 
> >
> > What are the symptoms?  -rt seems to work for everyone on linux-kernel
> > but no one on this list.  Can you provide the boot messages?
> 
> I get something like a ooops backtrace involving irq. I did not find
> any trace of this oops in /var/log so the boot message is lost. 

Well, my 2.8.18-rt3 system Oopsed while building that kernel, so I have
to debug that first ;-)

Lee


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Re: [Alsa-user] Drop outs under I/O load with Terratec EWX 24/96

2006-10-02 Thread Lee Revell
On Mon, 2006-10-02 at 13:43 +0200, Jascha Berberich wrote:
> Uhm... I have searched in your threads but couldn't find it. Would be
> nice if somebody could tell me wow to trace the latency :)
> 

Search LKML or LAU archive, this has been explained many times.


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Re: [Alsa-user] alsa sound problem

2006-10-02 Thread Lee Revell
On Mon, 2006-10-02 at 14:00 +0300, Martin Ivanov wrote:
> Hello!
> I am running Slackware Linux 10.2 with kernel 2.4.31
> My question is as follows:
> When I listen to music for example in xine or any other player, no other 
> sound application is able to run, for example if I am listening to music and 
> someone try to call me in skype, he can not do that. I have noticed that at X 
> shoutdown alsa complains about being unable to open slave.
> In windows one can play several sound applications simultaneously. I do not 
> believe it is impossible in Slackware.
> I would be thankful if you suggest me some solution.
> I thank you in advance.
> 

Your ALSA version is probably too old.  This should work by default in
ALSA 1.0.9 or later.

> Regards,
> Martin
> 
> -
> vbox7.com - Забавни видео клипове!
> 
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Re: [Alsa-user] alsa sound problem

2006-10-02 Thread Lee Revell
On Mon, 2006-10-02 at 10:20 -0700, Bob wrote:
> 
> Lee Revell wrote:
> > On Mon, 2006-10-02 at 14:00 +0300, Martin Ivanov wrote:
> >> Hello!
> >> I am running Slackware Linux 10.2 with kernel 2.4.31
> >> My question is as follows:
> >> When I listen to music for example in xine or any other player, no other 
> >> sound application is able to run, for example if I am listening to music 
> >> and someone try to call me in skype, he can not do that. I have noticed 
> >> that at X shoutdown alsa complains about being unable to open slave.
> >> In windows one can play several sound applications simultaneously. I do 
> >> not believe it is impossible in Slackware.
> >> I would be thankful if you suggest me some solution.
> >> I thank you in advance.
> >>
> > 
> > Your ALSA version is probably too old.  This should work by default in
> > ALSA 1.0.9 or later.
> 
> I _thought_ I'd asked the same question in regard to my newly installed 
> soundcard and was told about all the different programs which don't play 
> nicely with ALSA. However, I am running ALSA 1.0.9b and I WAS getting 
> multiple input via my cheap builtin motherboard soundcard. But, with the 
> new card (maudio audiophile 2496) I get only one application at a time. 
> Just like the OP on this thread.
> 
> I think there is a minor config issue, but damned if I can figure out what.

You need a newer version than 1.0.9b to get automatic mixing with that
card.

Lee



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Re: [Alsa-user] alsa sound problem

2006-10-02 Thread Lee Revell
On Mon, 2006-10-02 at 10:40 -0700, Bob wrote:
> Okay. But since the OP did not mention a specific sound card I assumed
> that you meant that mixing didn't work until 1.0.9 ... and that it did
> on 1.0.9 and later. I guess the part that I don't get is why the card 
> would make a difference ... doesn't alsa just take multiple inputs
> from the sound port and mix them and then output to the card? After
> all, isn't that what a software abstraction layer is for? 

The card matters because some devices handle this mixing themselves, and
it would be inefficient for ALSA to do it also in software.  So ALSA
needs to know which cards lack hardware mixing support.

This was enabled for most cards in 1.0.9 but the ICE1712 and ICE1724
were overlooked.

Lee




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Re: [Alsa-user] configuring ALSA driver to avoid invertion

2006-10-03 Thread Lee Revell
On Tue, 2006-10-03 at 06:41 +0400, Andrew Gaydenko wrote:
> Sergei,
> 
> You are right, "as binary signal value increases analog output
> value decreases". As for reference, I'm an audio DIY-er, almost all
> signal chain is made by my hand :-)
> 
> LADSPA is a more high level. ALSA driver configuring is much more
> appreciated.
> 

I don't think it's possible without a LADSPA plugin.  ALSA driver does
not touch the audio samples.  It just sets up DMA between the card and
the PC.

> 
> Andrew
> 
> === On Tuesday 03 October 2006 06:29, you wrote: ===
> 
> But what do you use as reference ?
> 
> Or, you mean, as binary signal value increases analog output
> value decreases ?
> 
> If so, AFAIR there is LADSPA inverting plugin, or it can be easily
> written.
> 
> But, anyway, why do you care ? That is, are there analog outputs which
> are not inverted, and you need to use the with line out, so you need
> all of them to be in the same polarity ?
> 
> --Sergei.
> 
> 
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Re: [Alsa-user] Some questions about the Audigy 2 NX

2006-10-03 Thread Lee Revell
On Tue, 2006-10-03 at 20:25 +0200, Evgeni Golov wrote:
> I though, dmix is not needed anymore (or is setup'ed automagicaly)?
> 

Not for USB audio.  There are too many different devices for a single
dmix config to work with them all.  You'll have to create your own.

Lee


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Re: [Alsa-user] Problems with snd hda intel: works only part of the time

2006-10-05 Thread Lee Revell
On Thu, 2006-10-05 at 20:42 +0200, Dominique Michel wrote:
> It can be the case if you have an USB sound card as a webcam. The
> snd_usb_audio will be loaded by hotplug before at alsasound load the
> alsa
> modules, and the regular sound card will be the second sound card.
> 
> If it is the case, you have to add
> "snd-usb-audio" (without the double quotes) in /etc/udev/blacklist and
> alsasound
> will work as expected.
> 
> If it not solve the problem, what is the output of
> 
> lsmod|grep snd
> 

Blacklisting snd-usb-audio would be a pretty heavy handed approach.
Applications should not rely on any specific card ordering.

Lee


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Re: [Alsa-user] Pb with simultaneous SATA and ALSA I/O

2006-10-05 Thread Lee Revell
On Thu, 2006-10-05 at 22:42 +0200, Dominique Dumont wrote:
> "Francesco Peeters" <[EMAIL PROTECTED]> writes:
> 
> > Have you tried using a different slot for the SB Live?
> 
> Yes. No change at all. (Sorry for the delay).

This is going to be a problem with the SATA driver not ALSA.  I've heard
that some motherboards do evil stuff like implementing legacy drive
access modes using SMM which would cause dropouts without xruns
reported.

Please report it on LKML.

Lee


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Re: [Alsa-user] Pb with simultaneous SATA and ALSA I/O

2006-10-05 Thread Lee Revell
On Thu, 2006-10-05 at 22:52 +0100, Alan Cox wrote:
> Ar Iau, 2006-10-05 am 16:45 -0400, ysgrifennodd Lee Revell:
> > I've heard that some motherboards do evil stuff like implementing legacy 
> > drive
> > access modes using SMM which would cause dropouts without xruns
> > reported.
> 
> They don't. SATA causes audio dropouts on some systems because its fast
> enough to starve the audio device of regular enough access to the PCI
> bus. If that is a problem the audio device should be tuning PCI
> latencies
> 

OK.  In fact the Windows driver and IIRC the OSS driver do tune PCI
latencies.  But that can't be the problem here if analog playback is
unaffected.  Sounds like electrical noise could be the issue...

Lee


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Re: [Alsa-user] Pb with simultaneous SATA and ALSA I/O

2006-10-05 Thread Lee Revell
On Thu, 2006-10-05 at 23:01 +0200, Dominique Dumont wrote:
> Lee Revell <[EMAIL PROTECTED]> writes:
> 
> > This is going to be a problem with the SATA driver not ALSA.  I've heard
> > that some motherboards do evil stuff like implementing legacy drive
> > access modes using SMM which would cause dropouts without xruns
> > reported.
> 
> Why do I hear distortion with PCM on SPDIF output and no distortion at
> all on analog front output ? (both with the SB Live card)
> 
> > Please report it on LKML.
> 
> Will do. (although your mail and this mail are cc'ed to LKLM)

Did you ever try the latency tracer?  (See LKML archives for
instructions)

Lee


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Re: [Alsa-user] Pb with simultaneous SATA and ALSA I/O

2006-10-05 Thread Lee Revell
On Thu, 2006-10-05 at 23:01 +0200, Dominique Dumont wrote:
> Lee Revell <[EMAIL PROTECTED]> writes:
> 
> > This is going to be a problem with the SATA driver not ALSA.  I've heard
> > that some motherboards do evil stuff like implementing legacy drive
> > access modes using SMM which would cause dropouts without xruns
> > reported.
> 
> Why do I hear distortion with PCM on SPDIF output and no distortion at
> all on analog front output ? (both with the SB Live card)

No idea.  Maybe SPDIF playback is more sensitive to timing glitches.
But the problem must be in the SATA driver not the emu10k1 driver.

Lee


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Re: [Alsa-user] Problems with snd hda intel: works only part of the time

2006-10-05 Thread Lee Revell
On Thu, 2006-10-05 at 14:33 -0700, Bill Unruh wrote:
> On Thu, 5 Oct 2006, Lee Revell wrote:
> 
> > On Thu, 2006-10-05 at 20:42 +0200, Dominique Michel wrote:
> >> It can be the case if you have an USB sound card as a webcam. The
> >> snd_usb_audio will be loaded by hotplug before at alsasound load the
> >> alsa
> >> modules, and the regular sound card will be the second sound card.
> >>
> >> If it is the case, you have to add
> >> "snd-usb-audio" (without the double quotes) in /etc/udev/blacklist and
> >> alsasound
> >> will work as expected.
> >>
> >> If it not solve the problem, what is the output of
> >>
> >> lsmod|grep snd
> >>
> >
> > Blacklisting snd-usb-audio would be a pretty heavy handed approach.
> > Applications should not rely on any specific card ordering.
> 
> And people should not speed on the highways.
> 

But what if people actually have USB audio devices?  There must be a
better way...

Lee


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Re: [Alsa-user] Problems with snd hda intel: works only part of the time

2006-10-05 Thread Lee Revell
On Thu, 2006-10-05 at 16:19 -0700, Bill Unruh wrote:
> On Thu, 5 Oct 2006, Lee Revell wrote:
> 
> > On Thu, 2006-10-05 at 14:33 -0700, Bill Unruh wrote:
> >> On Thu, 5 Oct 2006, Lee Revell wrote:
> >>
> >>> On Thu, 2006-10-05 at 20:42 +0200, Dominique Michel wrote:
> >>>> It can be the case if you have an USB sound card as a webcam. The
> >>>> snd_usb_audio will be loaded by hotplug before at alsasound load the
> >>>> alsa
> >>>> modules, and the regular sound card will be the second sound card.
> >>>>
> >>>> If it is the case, you have to add
> >>>> "snd-usb-audio" (without the double quotes) in /etc/udev/blacklist and
> >>>> alsasound
> >>>> will work as expected.
> >>>>
> >>>> If it not solve the problem, what is the output of
> >>>>
> >>>> lsmod|grep snd
> >>>>
> >>>
> >>> Blacklisting snd-usb-audio would be a pretty heavy handed approach.
> >>> Applications should not rely on any specific card ordering.
> >>
> >> And people should not speed on the highways.
> >>
> >
> > But what if people actually have USB audio devices?  There must be a
> > better way...
> >
> 
> Yes, it is a problem with Linux that one cannot specify the order in which
> Linux specifies things like soundcards, disk drives, ethernet cards,
> It tends to depend on the random events during bootup.
> 

This is not a problem with Linux.  ALSA supports addressing devices by
name.  Applications that rely on a specific card order are broken.

Fix the apps, problem solved.

Lee


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Re: [Alsa-user] Problems with snd hda intel: works only part of the time

2006-10-06 Thread Lee Revell
On Fri, 2006-10-06 at 21:37 +0200, Dominique Michel wrote:
> options snd-snd-usb-audio index=1

[...]

> alias snd-card-1 snd-usb-audio
> options snd-snd-usb-audio index=1 

snd-snd-usb-audio is wrong, it should be snd-usb-audio.

Did you create these files by hand?  Or is it a Gentoo bug?

Lee


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Re: [Alsa-user] Problems with snd hda intel: works only part of the time

2006-10-06 Thread Lee Revell
On Fri, 2006-10-06 at 22:29 +0200, Sergei Steshenko wrote:
> I think that we, end users, want the following:
> 
> * we want to be able to disconnect/reconnect external cards;
> * we want to be able to use (crappy) internal audio chips for things
> like
> Internet telephony;
> * whenever we disconnect/reconnect external cards, we do NOT want to
> adjust
> ALSA configuration files - especially taking into account difficulty
> creating
> them.
> 
> So, the mechanism should allow to assign to cards indices once and for
> all - regardless whether the card is present or not. 

Please make sure to file bug reports against any application that relies
on a fixed device order.

Also, it's not necessary to manage the default card via module
parameters.  Look at gnome-sound-properties, which sets the default card
by changing the definition in the user's ALSA config file.

Module indexes are a short-term band aid, they should definitely not be
considered a real solution.

Lee


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Re: [Alsa-user] Problems with snd hda intel: works only part of the time

2006-10-07 Thread Lee Revell
On Sat, 2006-10-07 at 10:24 -0500, Peter Hartman wrote:
> I'm glad your issue was resolved, but if you check the start of this
> thread, it was my original issue, which is still unresolved, that
> started this all off!  :) Any insights into the issue?
> 

It must be a driver bug.  You're not doing anything wrong.  Please file
a report in the ALSA bug tracker (search for existing ones first).

Lee



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Re: [Alsa-user] Pb with simultaneous SATA and ALSA I/O

2006-10-07 Thread Lee Revell
On Sat, 2006-10-07 at 14:03 +0200, Dominique Dumont wrote:
> Lee Revell <[EMAIL PROTECTED]> writes:
> 
> > Did you ever try the latency tracer?  (See LKML archives for
> > instructions)
> 
> Yes I did (this time better than 2 or 3 days ago :-/ ).
> 
> - I compiled and booted 2.6.28-rt5 with latency tracer enabled
> - I verified that I got latency trace enabled (seen trace in kern.log)
> 
> I only got some traces after running this (detail added for the sake
> of other newbies like me) :
> 
>gandalf:/proc/sys/kernel# echo 0 > preempt_max_latency
> 
> Here's the max latency trace I got (note that I got a similar trace
> *before* running the test, so I'd say it's unrelated to the AC3
> drop-out problem.):

I think it must be an electrical noise issue.

Lee


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Re: [Alsa-user] PCM distorsion snapshots from SATA/ALSA conflict (was: Pb with simultaneous SATA and ALSA I/O)

2006-10-08 Thread Lee Revell
(it's probably a bad idea to attach .jpgs to a LKML post - please post
them on the web and send a URL)

On Sun, 2006-10-08 at 12:38 +0200, Dominique Dumont wrote:
> Anyway, the oscilloscope shows that:
> - a spike occurs every 330 useconds (about 3kHz)
>   (note: 330us is 15.85 times the period of the 48KHz spdif stream) 
> - the spike level roughly matches the level of the sine waves 330
>   useconds sooner
> 

You seem to be using a period size of 256 samples - any change if you
use a larger period size?

Any change if you use setpci to increase LATENCY_TIMER for the SBLive!
card?

Lee


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Re: [Alsa-user] Problems with Creative SoundBlaster Extigy in Ubuntu (Dapper Drake)

2006-10-08 Thread Lee Revell
On Sun, 2006-10-08 at 19:10 +0200, Stefan Nagy wrote:
> I tested the soundcard in Window$ and there it workes... I suppose my
> problem has something to do with some Ubuntu-Update because I didn't
> do
> anything else in the past days. 

Did you not change anything, or did you update Ubuntu?

If you did update Ubuntu this should be filed in the Ubuntu bug tracker.

Lee


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Re: [Alsa-user] Problem with an ice1712 soundcard on recording

2006-10-11 Thread Lee Revell
On Wed, 2006-10-11 at 22:02 +0100, [EMAIL PROTECTED] wrote:
> Hi,
> 
>   I just tried arecord with my m-audio 1010LT
> 
> arecord x.wav
> 
>   On playback it sounds like it is repeating chunks of the sound then 
> skipping to the next chunk.
> 
> 
> arecord -f cd x.wav
> 
>   Nothing on playback.
> 
>   I have had success recording/playback using the javax.sound.sampled API.
> 
>   Do we need somethging in .asoundrc ?  This file is a mystry to me.
> What is ttable ?

"arecord file.wav" with no arguments records 8 bit mono at 8Khz.  Does
"arecord -f dat file.wav" sound OK?

Lee


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Re: [Alsa-user] Problem with an ice1712 soundcard on recording

2006-10-12 Thread Lee Revell
On Thu, 2006-10-12 at 08:32 +0100, [EMAIL PROTECTED] wrote:
> Quoting Lee Revell <[EMAIL PROTECTED]>:
> 
> > On Wed, 2006-10-11 at 22:02 +0100, [EMAIL PROTECTED] wrote:
> >> Hi,
> >>
> >>   I just tried arecord with my m-audio 1010LT
> >>
> >> arecord x.wav
> >>
> >>   On playback it sounds like it is repeating chunks of the sound then
> >> skipping to the next chunk.
> >>
> >>
> >> arecord -f cd x.wav
> >>
> >>   Nothing on playback.
> >>
> >>   I have had success recording/playback using the javax.sound.sampled API.
> >>
> >>   Do we need somethging in .asoundrc ?  This file is a mystry to me.
> >> What is ttable ?
> >
> > "arecord file.wav" with no arguments records 8 bit mono at 8Khz.  Does
> > "arecord -f dat file.wav" sound OK?
> 
> 
> 
> arecord x.wav (8KhZ low is broken with repeating chunks)
> 
> arecord -f dat x.wav is silent
> 
> arecord -r 441000 x.wav  is not broken but distorted more then 8 bit 
> quantization.
> 

I guess you mean 44100?

Anyway that last command will record 8 bit audio at 44100Hz which still
won't sound good.

Try arecord -r 44100 -f S16_LE

> 
> cheers Paul.
> 
> 


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Re: [Alsa-user] Alsa on RedHat EL4 With Creative SoundBlaster live! value 5.1 card

2006-10-12 Thread Lee Revell
On Fri, 2006-10-13 at 00:49 +0800, Jason wrote:
> Hello,list,
>I install one creative soundblaster live! value 5.1 card, my os
> is RedHat EL4(AS4) Update4,it's kenel is 2.6.9-42EL, I used alsa
> 1.0.13rc2 ,I configured alsa-driver with option --with-cards=emu10k1x
> and make,make install. Then I configured,maked,make installed alsa-lib
> and alsa-util.After that,system detected that card is Creative Labs
> [SoundBlaster Live! Value] emu10k1x,and it's driver is emu10k1x.But I
> run system-config-soundcard,no sound out put but one error
> message":aplay pcm_write:1258 write error,input/output error".I'm
> confused with that,everything seems ok.
> Any help is appreciated.

Wrong driver.  Use emu10k1 not emu10k1x.


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Re: [Alsa-user] Alsa on RedHat EL4 With Creative SoundBlaster live! value 5.1 card

2006-10-12 Thread Lee Revell
On Fri, 2006-10-13 at 01:04 +0800, Jason wrote:
> But before I use RedHat EL4(AS4) update4,I use RedHat EL4(AS4) ,not
> update4,OS load emu10k1,but same thing occured.BTW,how to change
> driver?
> 

No idea, 2.6.9 is a very old kernel.  Ask RedHat why it does not work.

Just build ALSA with --with-cards=emu10k1x rather than
--with-cards=emu10k1.

Lee

> Thanks for advice!
> JASON
> 
> 2006/10/13, Lee Revell <[EMAIL PROTECTED]>:
> > On Fri, 2006-10-13 at 00:49 +0800, Jason wrote:
> > > Hello,list,
> > >I install one creative soundblaster live! value 5.1 card, my os
> > > is RedHat EL4(AS4) Update4,it's kenel is 2.6.9-42EL, I used alsa
> > > 1.0.13rc2 ,I configured alsa-driver with option --with-cards=emu10k1x
> > > and make,make install. Then I configured,maked,make installed alsa-lib
> > > and alsa-util.After that,system detected that card is Creative Labs
> > > [SoundBlaster Live! Value] emu10k1x,and it's driver is emu10k1x.But I
> > > run system-config-soundcard,no sound out put but one error
> > > message":aplay pcm_write:1258 write error,input/output error".I'm
> > > confused with that,everything seems ok.
> > > Any help is appreciated.
> >
> > Wrong driver.  Use emu10k1 not emu10k1x.
> >
> >
> 
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Re: [Alsa-user] Surround sound via optical spdif out

2006-10-13 Thread Lee Revell
On Fri, 2006-10-13 at 04:10 -0400, Nathan A. Smith wrote:
> Unfortunately, I still am unable to get surround sound to work.  The
> way we were testing this is 
> with "speaker-test -Dplug:iec958 -c6" (note:  iec958 and spdif are the
> same for this motherboard).
> I am getting sound from my left and right speakers, but nothing 
> from anything else.   

SPDIF only supports 2 channels unless the signal is AC3-encoded.

Lee


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Re: [Alsa-user] Fortissimo IV works great with 44.1khz material, not great at all in 48khz

2006-10-13 Thread Lee Revell
On Fri, 2006-10-13 at 15:46 +0400, Elie Morisse wrote:
> Hi,
> 
> I make my Fortissimo IV running by passing the model="ms300" option to the 
> snd-ice1724 module ( 
> https://bugtrack.alsa-project.org/alsa-bug/view.php?id=1556 ). It works 
> flawlessly on 44.1khz material, but horribly on 48khz, so *i can't watch 
> movies for now* :(.. I don't exactly know how to describe it, it's too fast, 
> and at the same time choppy on some players such as xine. I tryed everything 
> in xine and alsa mixer options, nothing helps, and also  GStreamer and 
> Mplayer do the same. Something weird about MPlayer is that it resamples even 
> 44.1khz sound to 48khz :
> 
> ==
> Opening audio decoder: [mp3lib] MPEG layer-2, layer-3
> AUDIO: 44100 Hz, 2 ch, s16le, 128.0 kbit/9.07% (ratio: 16000->176400)
> Selected audio codec: [mp3] afm: mp3lib (mp3lib MPEG layer-2, layer-3)
> ==
> alsa-init: using device default
> alsa: 48000 Hz/2 channels/4 bpf/65536 bytes buffer/Signed 16 bit Little Endian
> AO: [alsa] 48000Hz 2ch s16le (2 bytes per sample)
> 
> ..and plays ok if I pass "-af resample=44100:0:0" as option.
> What am I doing wrong ? Is there a workaround ? And.. will you ever support 
> Fortissimo IV ?

Maybe you need to use alsamixer to set the clock rate to 48000?

Lee


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Re: [Alsa-user] Problem with an ice1712 soundcard on recording

2006-10-13 Thread Lee Revell
On Fri, 2006-10-13 at 18:15 +0200, Udo van den Heuvel wrote:
> Lee Revell wrote:
> >> arecord -f dat x.wav is silent
> >>
> >> arecord -r 441000 x.wav  is not broken but distorted more then 8 bit 
> >> quantization.
> >>
> > 
> > I guess you mean 44100?
> > 
> > Anyway that last command will record 8 bit audio at 44100Hz which still
> > won't sound good.
> > 
> > Try arecord -r 44100 -f S16_LE
> 
> maybe add -c2 to record stereo: arecord -r 44100 -f S16_LE -c2 blah.wav
> Overe here mono works OK but stereo gives silence.
> 
> What is wrong/missing/needs configuration to make recording (from S/PDIF
> in my case) work? (M-Audio DiO2496)
> 

No idea.  Must be a driver bug.  Please file a report in ALSA bug
tracker.

Lee


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Re: [Alsa-user] Surround sound via optical spdif out

2006-10-13 Thread Lee Revell
On Fri, 2006-10-13 at 16:34 -0400, Nathan A. Smith wrote:
> On Fri, 2006-10-13 at 09:56 -0400, Lee Revell wrote:
> > On Fri, 2006-10-13 at 04:10 -0400, Nathan A. Smith wrote:
> > > Unfortunately, I still am unable to get surround sound to work.  The
> > > way we were testing this is 
> > > with "speaker-test -Dplug:iec958 -c6" (note:  iec958 and spdif are the
> > > same for this motherboard).
> > > I am getting sound from my left and right speakers, but nothing 
> > > from anything else.   
> > 
> > SPDIF only supports 2 channels unless the signal is AC3-encoded.
> Thanks for the reply Lee,
> 
> Does that mean I should do something like this:
> 
> speaker-test -Dplug:iec958 -c6 -w SURROUNDTEST_011212.wav
> 
> where the wav file is a ac3 encoded file?

I don't know if speaker-test will play AC3 encoded files.  Try xine or
mplayer.

Lee


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Re: [Alsa-user] Surround sound via optical spdif out

2006-10-13 Thread Lee Revell
On Fri, 2006-10-13 at 16:52 -0400, Nathan A. Smith wrote:
> On Fri, 2006-10-13 at 16:36 -0400, Lee Revell wrote:
> > On Fri, 2006-10-13 at 16:34 -0400, Nathan A. Smith wrote:
> > > On Fri, 2006-10-13 at 09:56 -0400, Lee Revell wrote:
> > > > On Fri, 2006-10-13 at 04:10 -0400, Nathan A. Smith wrote:
> > > > > Unfortunately, I still am unable to get surround sound to work.  The
> > > > > way we were testing this is 
> > > > > with "speaker-test -Dplug:iec958 -c6" (note:  iec958 and spdif are the
> > > > > same for this motherboard).
> > > > > I am getting sound from my left and right speakers, but nothing 
> > > > > from anything else.   
> > > > 
> > > > SPDIF only supports 2 channels unless the signal is AC3-encoded.
> > > Thanks for the reply Lee,
> > > 
> > > Does that mean I should do something like this:
> > > 
> > > speaker-test -Dplug:iec958 -c6 -w SURROUNDTEST_011212.wav
> > > 
> > > where the wav file is a ac3 encoded file?
> > 
> > I don't know if speaker-test will play AC3 encoded files.  Try xine or
> > mplayer.
> Damn,
> 
> No love at all  Sound from Left and right speakers, but nothing
> else.  How do I troubleshoot this?

Does the receiver indicate an AC3 signal rather than standard PCM?

IIRC AC3 passthrough might not work on nvidia chipsets due to a
proprietary hardware implementation, but it might have changed.

Lee


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Re: [Alsa-user] Surround sound via optical spdif out

2006-10-13 Thread Lee Revell
On Fri, 2006-10-13 at 17:39 -0400, Nathan A. Smith wrote:
> > Does the receiver indicate an AC3 signal rather than standard PCM?
> No, nothing visibile on my receiver. 

Do you mean the receiver has no way to display whether it's receiving
PCM or AC3, or that the indicator is not active?

Lee


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Re: [Alsa-user] Surround sound via optical spdif out

2006-10-13 Thread Lee Revell
On Fri, 2006-10-13 at 21:48 -0400, Nathan A. Smith wrote:
> BTW: using files from
> http://www.sr.se/cgi-bin/mall/index.asp?programid=2445
> 
> I ran "aplay -v ..."  and got the following,
> 
> 
> aplay -v SURROUNDTEST_DD_640.wav

Are you still testing AC3/DTS/etc passthrough?  aplay won't play those
files.

What happens if you test AC3 passthrough using xine or mplayer as
described in this article:

http://linuxgazette.net/118/knaggs.html

For example:

mplayer -ao alsa -ac hwac3 filename.ac3

Lee




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Re: [Alsa-user] Surround sound via optical spdif out

2006-10-13 Thread Lee Revell
On Fri, 2006-10-13 at 22:16 -0400, Nathan A. Smith wrote:
> Playing SURROUNDTEST_011212.wav.
> Audio file file format detected.
> ==
> Forced audio codec: hwac3
> Cannot find codec for audio format 0x2001.
> Read DOCS/HTML/en/codecs.html!
> ==
> Segmentation fault
> 
> remember: the test file is from 
> http://www.sr.se/cgi-bin/mall/index.asp?programid=2445
> which I found via a link out of an ALSA HOWTO
> 

I think that's a DTS/DD file.  Try this one:

http://downloads.lightspeed.cx/lynne/bjorn_lynne-secret_world_(surround_version).ac3

Lee


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Re: [Alsa-user] Surround sound via optical spdif out

2006-10-13 Thread Lee Revell
On Fri, 2006-10-13 at 22:52 -0400, Nathan A. Smith wrote:
> On Fri, 2006-10-13 at 22:39 -0400, Lee Revell wrote:
> > I think that's a DTS/DD file.  Try this one:
> > 
> > http://downloads.lightspeed.cx/lynne/bjorn_lynne-secret_world_(surround_version).ac3
> First off
> 
> Nice song -- I might have to look for more of him/her.
> 
> Which points out that the file actually worked (and it does sound really
> nice).
> 
> So I must ask -- DD/DTS/AC3 are all forms of encoding surround sound.
> Is it typical that only AC3 works on our systems?  Is AC3 the DVD
> standard (I didn't think so...)?  

I'm not really a home theater expert, but the DVD standard is AC3 which
is also known as DD.  DTS is an older, less common format.  So you
should be able to play DVDs in surround.

I'm not sure why the other file didn't work.  It looks like mplayer
tried and failed to guess the file type?

I *think* that if AC3 passthrough works then DTS must work - it's just a
matter of setting a non-audio bit on the stream.

Lee


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Re: [Alsa-user] Surround sound via optical spdif out

2006-10-14 Thread Lee Revell
On Sat, 2006-10-14 at 12:15 +0200, Dominique Dumont wrote:
> "Nathan A. Smith" <[EMAIL PROTECTED]> writes:
> 
> > Does that mean I should do something like this:
> >
> > speaker-test -Dplug:iec958 -c6 -w SURROUNDTEST_011212.wav
> >
> > where the wav file is a ac3 encoded file?
> 
> No you should use:
> 
>   ac3dec -C some_ac3_file.ac3

No, that will decode the AC3 file on the host in software, what Nathan
wants to get working is AC3 passthrough, where the receiver does the AC3
decoding.

Lee


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Re: [Alsa-user] Surround sound via optical spdif out

2006-10-14 Thread Lee Revell
On Sat, 2006-10-14 at 22:40 +0200, Dominique Dumont wrote:
> Lee Revell <[EMAIL PROTECTED]> writes:
> 
> >> No you should use:
> >> 
> >>   ac3dec -C some_ac3_file.ac3
> >
> > No, that will decode the AC3 file on the host in software, what Nathan
> > wants to get working is AC3 passthrough, where the receiver does the AC3
> > decoding.
> 
> Sorry, I must respecfully disagree. I've just tried this command:
> 
> $ ac3dec -C en.ac3 
> Using PCM device 'plug:iec958:{AES0 0x2 AES1 0x82 AES2 0x0 AES3 0x2}'
> AC3 Stream 48.0 KHz 384 kbps
> 
> And I can see the dolby digital AC3 indication on the display of my
> Yamaha DSP-A1 amplifier which means the passthrough is workign fine.

Sorry, I didn't realize that ac3dec supports passthrough with the -C
switch.

> (of course without taking into account the sata pb that's plaguing me
> ;-) )

I'm afraid that is a hardware problem and there's no software solution.

Lee


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Re: [Alsa-user] E-mu 1212m support in HG

2006-10-16 Thread Lee Revell
On Mon, 2006-10-16 at 17:33 +, Matt Bucknall wrote:
> Hi James,
> 
> I have tried as you have suggested, but still get the same error
> message! Any ideas?
> 
> Matt.
> 

The "Unsupported soundcard $FOO" message is actually quite misleading -
it just means that something went wrong such that the build is not
possible.  Often it's caused by not having the kernel sources installed
or not having the same compiler that was used to build the kernel
available.  (Someone should really fix that).

Please provide the complete output of "./hgcompile --with-cards=emu10k1"

Lee

> 
> - Original Message 
> From: James Courtier-Dutton <[EMAIL PROTECTED]>
> To: Matt Bucknall <[EMAIL PROTECTED]>
> Cc: alsa-user@lists.sourceforge.net
> Sent: Monday, 16 October, 2006 5:42:13 PM
> Subject: Re: [Alsa-user] E-mu 1212m support in HG
> 
> Matt Bucknall wrote:
> > Sorry for being dim, I've not installed ALSA before. I've downloaded
> the latest HG snapshot
> (http://suse.inode.at/projects/alsa/snapshot/driver/alsa-driver-hg20061014.tar.bz2)
>  and tried to configure with --use-cards=emu=emu10k1, but the script 
> complains saying 'error: Unsupported soundcard emu10k1'.
> > 
> > I've had a look around the 'net to get some info on the problem, but
> have still had no success.
> > 
> > Please can someone talk me through the procedure I need to get ALSA
> setup with the emu10k module (with 1212m support)?.. or point me to
> some equivalent information?
> > 
> > Thanks,
> > Matt.
> > 
> 
> Download mercurial (alsa called hg)
> 
> hg clone http://hg-mirror.alsa-project.org/alsa-driver alsa-driver
> hg clone http://hg-mirror.alsa-project.org/alsa-kernel alsa-kernel
> 
> cd alsa-driver
> ./hgcompile --with-cards=emu10k1
> make install
> 
> Do "lsmod" to check that the old snd-emu10k1 modules is not already
> loaded.
> Then do "modprobe snd-emu10k1" to load the new alsa module with the
> emu1212m support.
> 
> James
> 
> 
> 
> 
> Send instant messages to your online friends
> http://uk.messenger.yahoo.com 
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Re: [Alsa-user] E-mu 1212m support in HG

2006-10-16 Thread Lee Revell
On Mon, 2006-10-16 at 20:06 +0200, Sergei Steshenko wrote:
> Why the dependency on compiler is so strong ?
> 
> I can and do run on my system programs compiled by gcc 3.3.? and gcc
> 4.0.1, so why ALSA should be different ? 

Kernel modules must be build by the same compiler as the kernel they
will be loaded into.  The userspace side of ALSA does not care.

Lee


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Re: [Alsa-user] SB Audigy SE (ca0106): no sound and no /dev/dsp

2006-10-16 Thread Lee Revell
On Tue, 2006-10-17 at 00:48 +, Matteo "Dargor" Modesti wrote:
> Now on Linux i can't hear any sound if i open, for example, an MP3
> file 
> with XMMS, and mpg123 tells me "Can't open /dev/dsp!".
> So i searched in /dev and saw that /dev/dsp didn't (and doesn't)
> exist!! 
> I searched in /dev/snd for it, but in that directory there are only 
> these files: 

You have no /dev/dsp because the snd_pcm_oss and snd_mixer_oss modules
are not loaded.

Lee




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Re: [Alsa-user] alsa buffering

2006-10-18 Thread Lee Revell
On Wed, 2006-10-18 at 18:51 +0930, Arthur Marsh wrote:
> [Please note that I run Debian unstable "Sid", so I don't know about
> the 
> availability of packages realtime-lsm and realtime-lsm-source in
> stable]
> 
> To enable realtime in your customised Debian 2.6.18 kernel, you need
> to 
> set under security options:
> 
> CONFIG_SECURITY_CAPABILITIES=m
> 

This will make no difference at all unless the app is specifically
written to use realtime scheduling.

I don't think this has anythign to do with why OSS apps sound good and
ALSA ones don't.  Maybe the app is not setting a big enough buffer.

Lee


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Re: [Alsa-user] volume keeps resetting at play

2006-10-19 Thread Lee Revell
On Thu, 2006-10-19 at 19:47 +0100, sonicx wrote:
> hello again,
> i still havent been able to fix the problem of the self-setting
> pcm_front channel. i have no idea where or how this happens. wasnt able
> to find anything logged about it, nor was i able to google a resolution
> or even find a hint in the soundcardlist. this is my soundcard:
> cat /proc/asound/cards
> 0 [SI7018 ]: SI7018 - SiS SI7018
>  SiS SI7018 PCI Audio at 0xd800, irq 11
> 1 [UART   ]: MPU-401 UART - MPU-401 UART
>  MPU-401 UART at 0x300, irq 10
> this didnt happen with a previous install of xubuntu, but i dont quite
> see the difference or what could be the problem as i now use ubuntu. if
> i open up a alsamixer i can see the volume go up everytime i play
> something, resetting just works for the playing song/movie, next title
> and it goes up again. may this be a gnome problem? i fiddled every
> option related to sound in gnome i could find but to no avail. 
> someone help me out please, the sound is just terrible this way
> (distorts when pcm_front sets itself to 100%), and i dont have any other
> music source then this machine. id gladly try out whatever tip you can
> give me. thanks,

Which application are you testing with?  Do you get the same results
using "aplay" to play audio?

It sounds like your application, not ALSA, is changing the PCM volume.

Lee


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Re: [Alsa-user] Surround sound via optical spdif out

2006-10-22 Thread Lee Revell
On Sun, 2006-10-22 at 14:18 +0200, Dominique Dumont wrote:
> Lee Revell <[EMAIL PROTECTED]> writes:
> 
> >> (of course without taking into account the sata pb that's plaguing me
> >> ;-) )
> >
> > I'm afraid that is a hardware problem and there's no software solution.
> 
> :-(
> 
> The only way to be sure is to check how this stuff works with
> Windoze. This is going to take a while as I'm quite windows challenged
> ;-)

It could work on Windows and still be a hardware problem, just because
Linux might drive the hardware in a different (but correct) way.  For
example some people have "singing capacitors" under Linux but not
Windows because of the higher timer IRQ frequency of Linux combined with
poorly shielded motherboard components.

Lee


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Re: [Alsa-user] Understanding ALSA Mixer

2006-10-23 Thread Lee Revell
On Mon, 2006-10-23 at 09:34 -0700, Bill Unruh wrote:
> On Mon, 23 Oct 2006, Mr. Shawn H. Corey wrote:
> 
> > Hi,
> >
> > I have been playing around with ALSA Mixer (actually alsamixergui) and
> > could someone explain some things to me?
> >
> > 1. Why does PCM have a volume control?
> 
> That, despite its name is teh main output volume control.
> 

No, it's not.  Master is the main output volume control.  PCM controls
the level of sound output from applications (as opposed to mic/line in,
cd, etc).  Windows calls it "Wave".

Lee


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Re: [Alsa-user] Understanding ALSA Mixer

2006-10-23 Thread Lee Revell
On Mon, 2006-10-23 at 13:22 -0400, Mr. Shawn H. Corey wrote:
> Lee Revell wrote:
> > No, it's not.  Master is the main output volume control.  PCM controls
> > the level of sound output from applications (as opposed to mic/line in,
> > cd, etc).  Windows calls it "Wave".
> 
> OK, why, if it's turn up too loud, does it produce crappy sound? If it
> was just clipping the sound, it would become harsher but I'm also
> getting a lot of static. I'm trying to figure out why it would do this.
> (Actually I'm trying to figure out the best value to set for it.)
> 

No idea.  Probably an incomplete or buggy driver.

Lee


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Re: [Alsa-user] Capturing audio with mencoder and alsa SOLVED!

2006-10-27 Thread Lee Revell
On Fri, 2006-10-27 at 12:10 -0700, Carl Weidling wrote:

> Well, the problem seems to be solved.  I was dealing with another
> mencoder problem and it was suggested I copy videodev.h and
> videodev2.h from my new kernel source into /usr/include/linux.  I did
> that and while I was at it I copied the new kernels include/sound over
> to /usr/include/sound and I then rebuilt lame (used by mencoder).  I
> don't know which of these things fixed the sound problem, but it now
> works.

It's a bug for userspace applications to include kernel headers.

Lee


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Re: [Alsa-user] my module is snd-intel8x0

2006-10-28 Thread Lee Revell
On Sat, 2006-10-28 at 02:16 -0700, sercan sen wrote:
> /lib/modules/2.6.9-34.EL.cern/source
> checking for directory with kernel build...
> /lib/modules/2.6.9-34.EL.cern/build 

You seem to be trying to build ALSA against a hacked up vendor kernel
from SUSE or RedHat.  Please contact the vendor for an ALSA update.

Lee


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Re: [Alsa-user] Duplicate sound in two sound cards

2006-10-29 Thread Lee Revell
On Sun, 2006-10-29 at 15:44 -0300, Salatiel Filho wrote:
> Hi guys, i have two sound stereo cards on my machine and i'd like to
> know if it is possible play the same sound in both at same time.
> I'm not talking about virtual 4 channels sound cards , i just want ,
> 
> 
> Channels : 
> Left on card 0 = left on card 1 
> right on card 0 = right on card 1
> 
> 
> Could anyone provide me a sample to put on my asoundrc ?

Cannot be done easily.  The two cards will consume data at different
rates as their clocks will not be synced.

You could set up 2 instances of JACK, run one for each card and connect
them with netjack or jack_diplomat, but that will only allow you to run
JACK-aware apps.

Lee


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Re: [Alsa-user] snd_ctl_open failed for default: No such device - my /dev/snd/controlC0 doesn't exist!

2006-11-07 Thread Lee Revell
On Tue, 2006-11-07 at 12:44 -0800, Troy McGuire wrote:
> My sound was working fine last night, I installed a game, but other than 
> that didn't change any settings/devices/configs on my system.  I've 
> tried logging out and rebooting, but it still seems like my sound card 
> doesn't exist anymore :(  Does anyone have good advice for me?
> 

The game installer must have hosed your /dev.  Try uninstalling it,
rebooting, then check if those /dev/snd/* files exist.

Lee


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Re: [Alsa-user] Help enabling 5.1

2006-11-07 Thread Lee Revell
On Tue, 2006-11-07 at 23:00 -0300, Salatiel Filho wrote:
> I'm trying to make my VIA8237 [REALTEK ALC655 REV 0] works in 5.1
> mode.
> I have found int "http://alsa.opensrc.org/index.php?page=AlsaTips"; 
> 
> Enabling 5+1 outputs on cards with line-out, mic-in and line-in jacks
> and it talks about my card and my board ,  but i do not have 
> Mic As Center/LFE
> Line-In As Surround
> in alsamixer. How can i make it work ? 

Please explain exactly how it does not work.

Did you try "speaker-test --channels 6 --device plug:surround51"?

Lee


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Re: [Alsa-user] No sound on HP dv2000 (dv2058ea) under SUSE 10.1

2006-11-08 Thread Lee Revell
On Wed, 2006-11-08 at 17:11 +, Anthony B. Coates (Alsa lists) wrote:
> Alsa has been upgraded to 1.0.11-32.

Try a newer version.  The current ALSA release is 1.0.13.


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Re: [Alsa-user] Help enabling 5.1

2006-11-08 Thread Lee Revell
On Wed, 2006-11-08 at 18:11 -0300, Salatiel Filho wrote:
> So , i tried 
> # speaker-test -Dplug:surround51 -c6 
> and again sound only on left and right channels.
> 

Please post the output of "amixer".


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Re: [Alsa-user] "PASS_THROUGH" is not working on ALC882.

2006-11-09 Thread Lee Revell
On Thu, 2006-11-09 at 08:14 +0100, Aleksandrov, Oleg wrote:
> When I change option "SPEAKER_SETUP" to "PASS_THROUGH" I get "Audio
> output unavailable. Device is busy. ()". Mplayer is also not able to
> use spdif/iec958. This way it was configured on my old PC ( AC97
> codec ). I tried to use different module options with no success. 

Another process like ESD or artsd or a web browser must be blocking the
sound card.  Your device must not support multiple access.

Lee


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Re: [Alsa-user] No sound on HP dv2000 (dv2058ea) under SUSE 10.1

2006-11-12 Thread Lee Revell
On Sun, 2006-11-12 at 12:04 +0100, steef wrote:
> Anthony B. Coates (Alsa lists) wrote:
> > OK, here's a follow-up.  Note first that I made a mistake when I
> > originally described my setup, as I had installed a pre-compiled version
> > of ALSA 1.0.13-0 (from PackMan).  I had an off-line discussion with Steve
> > Sinclair, and he told me that he had found it necessary to set 'model=z71v
> > position_fix=1 disable_mci=1' to the 'options sound-hda-intel ...' line in
> > '/etc/modprobe.d/sound' 
> 
> 
> > (in non-SUSE distributions, this may be
> > '/etc/modprobe.d/alsa-base')
> >   
> this last remark is *not* correct as far as it debian_etch and 
> slackware11 concerns.  combined with alsa 1.0.13 (compiled into the 
> kernel) and the chip snd-hada-intel  1986A (?)and a textfile < options 
> snd-hda-intel index=0 disable_msi=1 > in /etc/modprobe.d it makes the 
> driver work flawless on my machine (asus am2 motherboard, may 2006).
> 
> maybe this magic can make your  micro's) see beneath, work too??
> 

Unless you have the exact same codec, this is unlikely to help.

To the OP: Conexant codecs are poorly supported now because the vendor
refuses to release documentation.  Reverse engineering is ongoing.  Just
wait for new ALSA version.

Lee


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Re: [Alsa-user] /modprobe.d/sound config for audiophile usb

2006-11-13 Thread Lee Revell
On Mon, 2006-11-13 at 11:38 -0600, Tom Frei wrote:
> I'm running a 2.6.17.13 kernel on Slackware 11. I am trying to get a 
> M-Audio Audiophile USB to work.  No luck yet.  

You forgot to explain exactly how it "doesn't work".

Lee


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Re: [Alsa-user] Problems with ALSA + pulseaudio.

2006-11-14 Thread Lee Revell
On Mon, 2006-11-13 at 17:35 -0800, Avuton Olrich wrote:
> It's all described in detail in:
> http://pulseaudio.org/ticket/42
> 
> But to give a short for it, sometimes my wife clicks to start ALSA
> playing; sometimes it plays, sometimes it doesn't. Enabling dmix
> doesn't seem to help :/. They're pointing at ALSA;
> 
> load-module module-alsa-sink device=dmix sink_name=output
> 
> And sometimes when she clicks to start the music:
> Nov 13 15:56:52 tulip pulseaudio[25762]: socket-server.c:
> socket(PF_INET6): Address family not supported by protocol
> Nov 13 15:56:52 tulip pulseaudio[25762]: module.c: Failed to load
> module "module-alsa-sink" (argument: "device=hw:0
> sink_name=alsa_output.pci_10de_6a_alsa_playback_0"): init
> ialization failed.
> Nov 13 15:56:52 tulip pulseaudio[25762]: module-alsa-sink.c: Error
> opening PCM device hw:0: Device or resource busy
> Nov 13 15:56:52 tulip pulseaudio[25762]: alsa-util.c: Unable to attach
> to mixer dmix: No such file or directory
> Nov 13 15:56:52 tulip pulseaudio[25762]: alsa-util.c: device doesn't
> support 44100 Hz, changed to 48000 Hz.
> 
> 
> I'm really frusterated and was hoping for any help at all.

PulseAudio should use the ALSA device "default" (not "dmix" or "hw:0").
It's a bug if Pulse defaults to hw:0.

Something must be blocking the soundcard.  Next time you get this
problem post the output of "fuser /dev/dsp" and "fuser /dev/snd/*".

Lee


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Re: [Alsa-user] (snd_ctl_open_noupdate) Invalid CTL hw:0

2006-11-14 Thread Lee Revell
On Tue, 2006-11-14 at 14:53 -0800, Chuck Harrison wrote:
> Alsa n00b here starting to bring up alsa on an embedded platform:
> openWRT for the Broadcom mips processor used in wireless routers,
> I have a Netgear WGT634U. http://www.openwrt.org 

Probably a build error that causes aplay to look in the build host path
rather than the target path.  I recently had a similar problem bringing
up ALSA on an embedded PPC system.

Install strace on your target (it's very easy) and strace the failing
aplay process to see which open() call fails.

Lee


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Re: [Alsa-user] snd_hda_intel on Asus M2N32 WS Pro Asus MB

2006-11-15 Thread Lee Revell
On Wed, 2006-11-15 at 20:51 +0100, Mws wrote:
> hmm, starting x and kde brought up a freezed machine like i had
> sometime before.
> 
> then i took my kernel config again, 
> re-enabled msi 
> make 
> install
> reboot BUT with kernel cmd pci=nomsi 

Sounds like upstream kernel bug.  Apparently it's a known issue that
many machines hang without "pci=nomsi".  There's a thread on LKML now
about how to solve it.

Lee


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Re: [Alsa-user] alsaconf: setup of soundchip VIA 8237 fails - SuSE-Linux 9.2

2006-11-16 Thread Lee Revell
On Fri, 2006-11-17 at 02:21 +0100, Heinz Knoche wrote:
> 
> Clemens Ladisch schrieb:
> > Heinz Knoche wrote:
> > 
> >>Clemens Ladisch schrieb:
> >>
> >>>What are the contents of /proc/asound/cards?
> >>
> >>cat /proc/asound/cards
> >>0 [rev60  ]: VIA8233 - VIA 823x rev60
> >>   VIA 823x rev60 with CMI9761 at 0xd000, irq 3
> > 
> > 
> > This codec should be supported even by your old ALSA version.
> > 
> > What is the output of "amixer contents"?
> > 
> > 
> Sorry, it's somehow long ...
> 
> numid=1,iface=MIXER,name='Master Playback Switch'
>; type=BOOLEAN,access=rw---,values=1
>: values=off

You need to unmute this

> numid=20,iface=MIXER,name='PCM Playback Switch'
>; type=BOOLEAN,access=rw---,values=1
>: values=off

And this

> numid=7,iface=MIXER,name='Surround Playback Switch'
>; type=BOOLEAN,access=rw---,values=2
>: values=off,off

And this

> numid=3,iface=MIXER,name='Center Playback Switch'
>; type=BOOLEAN,access=rw---,values=1
>: values=off

And this

> numid=5,iface=MIXER,name='LFE Playback Switch'
>; type=BOOLEAN,access=rw---,values=1
>: values=off

And this

Lee


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Re: [Alsa-user] ALSA problem on Taihu board

2006-11-21 Thread Lee Revell
On Tue, 2006-11-21 at 19:38 -0500, David Cullen wrote:
> Dear Folks,
> 
> I am having a problem with ALSA on an AMCC Taihu board.  To keep
> things simple, I am just running "speaker-test".  The offending line
> of code is a call to mmap in pcm_mmap.c at line 368.  I have used
> KGDB to trace the call through the kernel, and I cannot figure out
> the problem.  The mmap call succeeds, but the address returned
> causes an OOPS when accessed.
> 
> Has anyone else had this type of problem with ALSA on PowerPC platforms?

Please post a link to your code.

Lee


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Re: [Alsa-user] Amarok, dmix, icecast

2006-11-26 Thread Lee Revell
On Sat, 2006-11-25 at 16:32 +0100, Emilio Scalise wrote:
> If you are using a recent alsa version (>1.0.11 I think, perhaps also
> 1.0.10, but I'm not sure..), you don't need to modify your asoundrc or
> asound.conf.
> Try to use default:1 as device. It should work out of the box.
> If not try to assign index 0 to the usb card (with module options) and
> use default device.
> 

Does not apply to USB devices - dmix must be configured manually.

Lee

> 2006/11/25, Denny Schierz <[EMAIL PROTECTED]>:
> > hi,
> >
> > i have two soundcards:
> >
> > ~$ cat /proc/asound/cards
> >  0 [SoundByLayout  ]: AppleOnbdAudio - SoundByLayout
> >   SoundByLayout
> >  1 [default]: USB-Audio - C-Media USB Headphone Set
> >   C-Media USB Headphone Set   at
> > usb-0001:10:1b.2-2.3, full speed
> >
> >
> >
> > i want to stream the sound from Amarok, so i need ices2 which grabs the
> > sound from the USB soundcard.
> >
> > in the ices2.config i have the following choice:
> >
> > 
> >   alsa
> >  44100
> >  2
> > plughw:hw:0,0
> > 
> >
> >   0
> >   test
> > 
> >
> > i red, that i need a dmix setup, so that Amarok and ices2 get working
> > together.
> >
> > does anybody has a setup like my one?
> >
> > The onboard soundcard is never used.
> >
> > cu denny
> >
> >
> > ==
> > USB Output
> >
> > Bus 002 Device 004: ID 0d8c:000c C-Media Electronics, Inc. Audio Adapter
> > Device Descriptor:
> >   bLength18
> >   bDescriptorType 1
> >   bcdUSB   1.10
> >   bDeviceClass0 (Defined at Interface level)
> >   bDeviceSubClass 0
> >   bDeviceProtocol 0
> >   bMaxPacketSize064
> >   idVendor   0x0d8c C-Media Electronics, Inc.
> >   idProduct  0x000c Audio Adapter
> >   bcdDevice1.00
> >   iManufacturer   0
> >   iProduct1
> >   iSerial 0
> >   bNumConfigurations  1
> >   Configuration Descriptor:
> > bLength 9
> > bDescriptorType 2
> > wTotalLength  253
> > bNumInterfaces  4
> > bConfigurationValue 1
> > iConfiguration  0
> > bmAttributes 0xa0
> >   (Bus Powered)
> >   Remote Wakeup
> > MaxPower  100mA
> > Interface Descriptor:
> >   bLength 9
> >   bDescriptorType 4
> >   bInterfaceNumber0
> >   bAlternateSetting   0
> >   bNumEndpoints   0
> >   bInterfaceClass 1 Audio
> >   bInterfaceSubClass  1 Control Device
> >   bInterfaceProtocol  0
> >   iInterface  0
> >   AudioControl Interface Descriptor:
> > bLength10
> > bDescriptorType36
> > bDescriptorSubtype  1 (HEADER)
> > bcdADC   1.00
> > wTotalLength  100
> > bInCollection   2
> > baInterfaceNr( 0)   1
> > baInterfaceNr( 1)   2
> >   AudioControl Interface Descriptor:
> > bLength12
> > bDescriptorType36
> > bDescriptorSubtype  2 (INPUT_TERMINAL)
> > bTerminalID 1
> > wTerminalType  0x0101 USB Streaming
> > bAssocTerminal  0
> > bNrChannels 2
> > wChannelConfig 0x0003
> >   Left Front (L)
> >   Right Front (R)
> > iChannelNames   0
> > iTerminal   0
> >   AudioControl Interface Descriptor:
> > bLength12
> > bDescriptorType36
> > bDescriptorSubtype  2 (INPUT_TERMINAL)
> > bTerminalID 2
> > wTerminalType  0x0201 Microphone
> > bAssocTerminal  0
> > bNrChannels 1
> > wChannelConfig 0x0001
> >   Left Front (L)
> > iChannelNames   0
> > iTerminal   0
> >   AudioControl Interface Descriptor:
> > bLength 9
> > bDescriptorType36
> > bDescriptorSubtype  3 (OUTPUT_TERMINAL)
> > bTerminalID 6
> > wTerminalType  0x0301 Speaker
> > bAssocTerminal  0
> > bSourceID   9
> > iTerminal   0
> >   AudioControl Interface Descriptor:
> > bLength 9
> > bDescriptorType36
> > bDescriptorSubtype  3 (OUTPUT_TERMINAL)
> > bTerminalID 7
> > wTerminalType  0x0101 USB Streaming
> > bAssocTerminal  0
> > bSourceID   8
> > iTerminal   0
> >   AudioControl Interface Descriptor:
> > bLength 7
> > bDescriptorType36
> > bDescriptorSubtype  5 (SELECTOR_U

Re: [Alsa-user] Amarok, dmix, icecast

2006-11-26 Thread Lee Revell
On Mon, 2006-11-27 at 06:47 +0100, Emilio Scalise wrote:
> I didn't know that I thought that was the same as other cards!
> 

It's trickier to enable dmix universally for USB devices.  I think
Clemens Ladisch was working on a solution?

Lee



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Re: [Alsa-user] Amarok, dmix, icecast

2006-11-28 Thread Lee Revell
On Tue, 2006-11-28 at 19:25 +0100, Denny Schierz wrote:
> hi,
> 
> Lee Revell schrieb:
> 
> > It's trickier to enable dmix universally for USB devices.  I think
> > Clemens Ladisch was working on a solution?
> 
> do you know a bit more about that setup? I have no idea, to let Amarok
> playing and ices2 grab the sound.
> 

Try creating an .asoundrc for dmix as described on the ALSA site.

> cu denny
> 
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Re: [Alsa-user] Help!!! Help !! No sound on 82801G snd-hda-intel HP DV2000t laptop

2006-11-29 Thread Lee Revell
On Wed, 2006-11-29 at 12:30 -0500, sundru wrote:
> Hi thx for your response ,
> 
> snd-seq-oss ou mentioned old sound module is loaded , just curious on
> how i would get the new sound module 
> are these modules part of the kernel itself or they are separate ?
> 

When you compile ALSA from source you must remove all snd-* modules
using rmmod before loading the new ones.

Or just reboot.

> thx !! 
> Sundara
> 
> On 11/29/06, Clemens Ladisch <[EMAIL PROTECTED]> wrote:
> sundru wrote:
> > Just finished a complete compile of alsa 1.0.13 but when i
> try to insert
> > into kernel it errors out with
> >
> > FATAL: Error inserting snd_pcm_oss
> (/lib/modules/2.6.16.21-0.25-smp
> > /kernel/sound/acore/oss/snd-pcm-oss.ko): Unknown symbol in
> module, or
> > unknown parameter (see dmesg)
> >
> > snd_seq_oss: disagrees about version of symbol
> snd_info_register
> 
> This happens when some old sound module is still loaded. 
> 
> Unload all loaded snd* modules before loading the new ones,
> or just reboot.
> 
> 
> HTH
> Clemens
> 
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Re: [Alsa-user] fixed SRC for vdr-softdevice

2006-12-12 Thread Lee Revell
On Tue, 2006-12-12 at 11:44 +, Harald Milz wrote:
> http://www.alsa-project.org/alsa-doc/alsa-lib/pcm_plugins.html 
> 
> (This page is inconsistent about converter vs. rate_converter; using
> "converter" instead gives a similar error).  

What ALSA version?

Lee


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Re: [Alsa-user] VT8233/8237/snd_via82xx: arecord hangs at 44 bytes

2006-12-12 Thread Lee Revell
On Tue, 2006-12-12 at 16:58 +0100, Kiesel wrote:
> $ cat /proc/asound/version
> Advanced Linux Sound Architecture Driver Version 1.0.11rc4 (Wed Mar 22 
> 10:27:24 2006 UTC).

Try a recent ALSA release.

Lee


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Re: [Alsa-user] ALSA on FC6

2006-12-14 Thread Lee Revell
On Thu, 2006-12-14 at 16:36 -0800, Fernando Lopez-Lezcano wrote:
> I would not install from source. Rather you should try making rpm
> packages of the new version (not easy), so that whatever you do will
> match what fc is already doing. 
> 

Shouldn't it be as easy as:

rpmbuild -ba alsa-lib-1.x.x/utils/alsa-lib.spec

?

Cross compilation is indeed a nightmare, especially if you need to
support multiple autotools versions.  But native should be trivial...

Lee


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Re: [Alsa-user] Setting the Default Notebook Sound Card?

2006-12-17 Thread Lee Revell
On Sun, 2006-12-17 at 08:42 +, Q wrote:
> Hi I have a Toshiba Satellite A100-225 laptop and am running Kubuntu 6.10.
> 
> Anyway the thing is that my laptop has onboard sound which is an Intel 
> HD Audio Chip (which I believe is based on a Realtek design). The chip 
> for this card is called snd_hda_intel. However I also have an Audigy 2 
> ZS PCMCIA notebook card.  The driver for this card is called snd_emu10k1.
> 
> Originally with both cards active my Intel HDA card was being loaded as 
> the default card. This was relatively easy to resolve (though not as 
> easy as it should be) by following these guides:
> 

If you were using Gnome you could just use gnome-sound-properties to
select the default sound card.  KDE must have something similar...

Lee



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Re: [Alsa-user] No sound with ALSA on nVidia MCP51

2006-12-18 Thread Lee Revell
On Mon, 2006-12-18 at 18:50 +0200, naudefj wrote:
> Hi,
> 
> I'm unable to get sound from my newly acquired m2npv-vm mobo with ALSA 1.0.12 
> (Mandriva 2007).
> 
> The hardware is working as I can play music with GeeXBox 
> (http://geexbox.org). 
> I've also tried different kernel versions (up to 2.6.19). So, I can only 
> asume it's an ALSA problem. 
> 
> # cat /proc/asound/version
> Advanced Linux Sound Architecture Driver Version 1.0.12 (Mon Aug 28 13:46:17 
> UTC 2006).

What ALSA version did geexbox use?

With new hardware it's always best to use a recent ALSA version.  Try
1.0.14.

Lee


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Re: [Alsa-user] Audacity Line-In Recording with SB Audigy 2 ZS Doesn't Work

2006-12-18 Thread Lee Revell
On Mon, 2006-12-18 at 23:13 -0600, Dan Star wrote:
>   have a Compaq nx9010 laptop with built in Southbridge Audio.
> Because 
> it has a flaky head phone jack and the line-in recording quality was
> bad 
> I bought a SB Audigy 2 ZS PC Card. The SB seems to have disabled the 
> Southbridge because the notebook speakers don't work anymore.

No PCMCIA sound card will let you output to the internal speakers -
they're hard wired to the internal audio device.  This applies to every
OS.

>  But the 
> headphone output of the SB is great. The problem is that I use
> Audacity 
> 1.3 on Mandriva 2007 PWP KDE with all the updates as of 12/16/06 and 
> line-in recording does not work. Audacity states "Error while opening 
> sound device. Please check the input device settings and the project 
> sample rate."

You may need a newer ALSA version like 1.0.14 for that device.

If your distro does not provide up to date ALSA packages and you don't
want to risk breaking things you can try a live CD distro like the
latest Knoppix...

Lee


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Re: [Alsa-user] FC6 audio differences

2006-12-20 Thread Lee Revell
On Wed, 2006-12-20 at 10:33 -0500, Gene Heskett wrote:
> In an effort to make sure my dual audio setup continued to work, I 
> transfered my FC2's modprobe.conf to the FC6 version.  It should work 
> right?

No idea.

Doesn't FC6 provide a better interface to select the default sound card
than editing that file?

If you don't run legacy OSS apps, the cleanest solution is the way
gnome-sound-properties does it (define "default" PCM in
per-user .asoundrc)

Lee


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Re: [Alsa-user] FC6 audio differences

2006-12-20 Thread Lee Revell
On Wed, 2006-12-20 at 19:40 -0500, Gene Heskett wrote:
> On Wednesday 20 December 2006 16:26, Fernando Lopez-Lezcano wrote:
> >On Wed, 2006-12-20 at 10:33 -0500, Gene Heskett wrote:
> >> In an effort to make sure my dual audio setup continued to work, I
> >> transfered my FC2's modprobe.conf to the FC6 version.  It should work
> >> right?
> >
> >I don't know. In FC6 I would use the gui configurator.
> >In the menu "System" -> "Administration" -> "Soundcard Detection".
> >
> >It lets you select which card is the default and also which card is
> >first (something I think is new in fc6).
> >
> >-- Fernando
> 
> That is the same as calling system-config-soundcard from the cli.  And 
> that part works, its just that kde now thinks the motherboard nvidia 
> sound is the default.  Its the only sound it know about apparently.  I 
> want the emu10k1/audigy 2 to be the default.

Sounds like KDE may be configured to use OSS rather than ALSA?
Unfortunately I don't know how to check.  Must be in some KDE control
panel...

Lee


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Re: [Alsa-user] FC6 audio differences

2006-12-20 Thread Lee Revell
On Wed, 2006-12-20 at 21:16 -0500, Gene Heskett wrote:
> nope, everything BUT kde is working as near as I can tell, but it
> can't even find /dev/audio1 OR /dev/audio, I must leave that entry
> blank. And then the test sound comes out in the headphones plugged
> into the nvidia jacks. 

/dev/audio and /dev/audio1 are not ALSA devices.  Sounds like KDE is
using OSS.

Lee


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Re: [Alsa-user] FC6 audio differences

2006-12-21 Thread Lee Revell
On Thu, 2006-12-21 at 17:25 -0800, Fernando Lopez-Lezcano wrote:
> I just tried to test this. This is on a test fc6 install (x86_64), I
> logged in using kde, went to the "Control Center" then "Sound and
> Multimedia", then "Sound System", then the tab labeled "Hardware" and
> I finally selected "Advanced Linux Sound Architecture" from the
> plugin. It said "restarting audio system" or something like that and
> AFAICT sound is going to my delta66 card (and not to the mobo
> chipset), I can see the vu meters in envy24control move when the
> terminal bell should ring... 

What did the sound system default to?

Does this control center allow selecting the ALSA device to use?

Lee


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Re: [Alsa-user] system crash by MPU-401 after a short while of using it

2006-12-21 Thread Lee Revell
On Fri, 2006-12-22 at 03:09 +0100, Piotr Sawicki wrote:
> Hi, 
> I use opensuse 10.0 with kernel:
> # uname -a
> Linux linux 2.6.15-jad2-default #1 PREEMPT Sat Dec 24 00:00:07 CET 2005 i686 
> i686 i386 GNU/Linux

Any chance you can try a newer kernel, for example using a Knoppix CD?

Lee


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Re: [Alsa-user] How change soundcards?

2006-12-30 Thread Lee Revell
On Sat, 2006-12-23 at 16:46 +0100, Giuliano Pochini wrote:
> On Tue, 19 Dec 2006 21:23:11 -0600
> Dan Star <[EMAIL PROTECTED]> wrote:
> 
> > I have a built in Southbridge Chip in my nx9010 laptop and a Audigy 2 ZS 
> > PC Card.  How do I tell applications/alsa which card to use?
> 
> It depends on the application. It must have a way to choose the device.

If the app does not allow sound card selection, it should obey the
"default" device defined in the ALSA config.

If you use Gnome, System->Preferences->Sound and select the desired
device in the "Default sound card" control.

If you use KDE, I think you have to ask the KDE developers to implement
this functionality.

If the app is unaffected by the default sound card selection, please
file a bug against that application.

Lee


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Re: [Alsa-user] functionality of PCM slider in alsamixer for Realtek ALC883/HDA NVidia

2006-12-30 Thread Lee Revell
On Fri, 2006-12-29 at 17:24 +0200, Sergei Steshenko wrote:
> And what is the physical mechanism behind this ?
> 

Different for every device.

> What does the word "weird" mean in this context ? Are you sure it's a
> weird card and not a weird driver ?
> 

Most likely the problem is that the vendor did not release the required
docs to implement a proper driver.

You're right that the PCM volume control should affect any digital audio
originating on the PC.  Please file an ALSA bug report if it does not
work that way.

Lee


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Re: [Alsa-user] Swapped motherboard, CPU, etc. and Debian/Linux can't play audio/sound.

2006-12-30 Thread Lee Revell
On Mon, 2006-12-25 at 19:59 -0800, Bill Unruh wrote:
> I agree that the names are highly non-intuitive. for example on most
> cards, the input volume controls, like say the line in volume control,
> does not affect the volume of the signal delivered to the computer. It
> controls the volume that is delivered by the input to the output
> channel on the sound card. 

Are you referring to the line in playback control (F3 in alsamixer) or
the line in capture control (F4 in alsamixer)?  Line in playback control
should affect the volume delivered to the output of the soundcard; line
in capture should affect the volume of the signal delivered to the
computer.

Many mixer apps (kmix is the worst offender) cannot distinguish these.

Lee


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