[Alsa-user] Alsa compilation - lasound

2008-01-11 Thread Felipe Uderman
Hello,

I have successfully  ported the alsa library to the arm platform. But I
don't know what are the binaries of alsa, and where they are located. I have
compiled alsa because I need to cross compile a program that depends on
alsa, it is a softphone. The compilation of this program make reference to
-lasound, and fails because it can't find it. I assumed that I would have as
output of the compilation of alsa a few files, with lasound among then. Can
anyone please help me? Please let me know if you need more information.

Regards,

Felipe
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[Alsa-user] MCP67 audio and alsa 1.0.15

2008-01-11 Thread Jerry Geis
I just got a new HP laptop. Nvidia based. It has MCP67 components from 
lspci.

I installed alsa 1.0.15 and it does not find any sound cards.
I have install alsa a number of times so I am not new to that.
I thought there was supposed to be support for MCP67 in 1.0.15?

Any ideas?

Jerry

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Re: [Alsa-user] ALSA support for toslink

2008-01-11 Thread Clemens Ladisch
Michelle Dupuis wrote:
 My MB supports both toslink and SPDIF - from what I've read toslink has a
 much higher bandwidth.  Will the ALSA driver support moving more data
 through toslink?

It's very likely that in your case, Toslink means optical as opposed
to the usual electrical coaxial SPDIF connector.  The data at both
connectors will be exactly the same.


Regards,
Clemens

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Re: [Alsa-user] MCP67 audio and alsa 1.0.15

2008-01-11 Thread James Shatto
 I just got a new HP laptop. Nvidia based. It has MCP67 components from 
 lspci.
 
 I installed alsa 1.0.15 and it does not find any sound cards.

# modprobe snd-hda-intel

At least that's the one for my MCP61.  Does the card show up in lspci?

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Re: [Alsa-user] Alsa compilation - lasound

2008-01-11 Thread Clemens Ladisch
Felipe Uderman wrote:
 I have successfully  ported the alsa library to the arm platform.

It shouldn't need porting.  What did you have to change?

 But I don't know what are the binaries of alsa, and where they are
 located.

make install installs the binaries and the configuration files to the
default location (/usr/lib and /usr/share/alsa).  If you have cross-
compiled alsa-lib, you usually use make DESTDIR=/somewhere to put the
files below the root of the target machine's file system.


HTH
Clemens

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[Alsa-user] Intel ICH8 82801H (rev3) support

2008-01-11 Thread Vasil Kolev
Hi,
I have a laptop with
00:1b.0 Audio device: Intel Corporation 82801H (ICH8 Family) HD Audio
Controller (rev 03)
00:1b.0 0403: 8086:284b (rev 03)

I'm running ubuntu hardy with the latest ubuntu kernel (2.6.24-3) and
this card was supposed to work with it, but what I get is
hda_codec: Unknown model for ALC268, trying auto-probe from BIOS...
in dmesg. I have the mixer, but no audio output whatsoever, and seems
that this autoprobe doesn't work right :) I went and looked in different
solutions in google, and found some, but what it boiled down to was that
at patch_realtek.c, there's
/* ALC268 models */
enum {
ALC268_3ST,
ALC268_TOSHIBA,
ALC268_ACER,
ALC268_AUTO,
ALC268_MODEL_LAST /* last tag */
};
and that message shows up when the model I get is not matched in the
list, and what else I've seen online I see that the people use the
revision 2 of the hardware, and this is revision 3. It's the same in the
kernel and in the mercurial repository, so from here on - what can I do
to try to make it work?


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[Alsa-user] Software volume mixer for S/PDIF output

2008-01-11 Thread Johannes Bauer
Hello list,

I have recently switched to a 5.1 Dolby Digital system and was surprised
how nicely everything worked with ALSA. There is however one problem
which I have - read through a lot of forums, but did not find a
satisfying answer.

I'm using my soundcards (Terratec Aureon 5.1 = C-Media Electronics Inc
CM8738) optical output to interface my amplifier. The optical output
logically always uses maximum output so no data is lost. This makes
perfect sense to me.

However, it is sometimes inconvenient. When my phone rings and I want to
answer I'd like to make music almost completely silent by just some
mouseclicks - not standing up and turning the volume down in the
amplifier (no, it does not have a remote...).

Some programs (like XMMS) support it, but I would like to have a more
general solution which works with all programs in a unified way.

So my question is: does ALSA support software volume control? The loss
of quality (dynamics) is not so important to me in this case, I'd have
it almost always on 100% - but would sometimes like to quickly reduce
the volume. If it is supported, does it also work when playing back
direct AC3 streams?

If it can be done: how so?

Thanks in advance for your help,
Greetings,
Johannes

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Re: [Alsa-user] ALSA support for toslink

2008-01-11 Thread Michelle Dupuis
Can ALSA send audio to the analog ports AND the optical/toslink port at the
same time?  (Or is this a function of the motherboard)?

Thanks,
MD

 -Original Message-
 From: Clemens Ladisch [mailto:[EMAIL PROTECTED] 
 Sent: Friday, January 11, 2008 12:49 PM
 To: Michelle Dupuis
 Cc: ALSA Users List
 Subject: Re: [Alsa-user] ALSA support for toslink
 
 Michelle Dupuis wrote:
  My MB supports both toslink and SPDIF - from what I've read toslink 
  has a much higher bandwidth.  Will the ALSA driver support 
 moving more 
  data through toslink?
 
 It's very likely that in your case, Toslink means optical 
 as opposed to the usual electrical coaxial SPDIF connector.  
 The data at both connectors will be exactly the same.
 
 
 Regards,
 Clemens
 


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Re: [Alsa-user] ALSA support for toslink

2008-01-11 Thread Adam Nielsen
 Can ALSA send audio to the analog ports AND the optical/toslink port at the
 same time?  (Or is this a function of the motherboard)?

AFAIK most sound cards (especially onboard ones) only have one output
stream.  This same audio signal is sent to the analogue jacks, the coax
SPDIF and the optical SPDIF.

The hard part would be getting *different* audio coming out of each
connection - that would require a more advanced sound card.

Cheers,
Adam.


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Re: [Alsa-user] Software volume mixer for S/PDIF output

2008-01-11 Thread Adam Nielsen
 So my question is: does ALSA support software volume control? The loss
 of quality (dynamics) is not so important to me in this case, I'd have
 it almost always on 100% - but would sometimes like to quickly reduce
 the volume. If it is supported, does it also work when playing back
 direct AC3 streams?

 If it can be done: how so?

Yes, there is a softvol plugin you can attach to an ALSA device that
will create a new mixer control.  Anything played through though that
device will be adjusted according to the softvol mixer control.

I would've thought your card would already have a hardware mixer that
supports turning the volume down though, but perhaps not.

I imagine you would also need a program like xbindkeys that can run a
command when a key is pressed, for example so you can press the mute
button on your keyboard to run amixer and get it to set the volume to
a low level.

It won't work when playing AC3 streams, as the decoding is done inside
the amplifier - but presumably whatever program is playing the AC3
stream has a pause button ;-)

Cheers,
Adam.


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Re: [Alsa-user] plug refuses to play audio at normal sample rates

2008-01-11 Thread Adam Nielsen
 Well, as far as I know LADSPA - it's very difficult, if not impossible,
 to not detect sample rate correctly.
 
 Did you look into the plugin source ?

I've had a look through it, but I can't see where the code is that does
sample rate conversion.

 Can it be that ALSA calls the plugin initialization routine with
 wrong sample rate ?
 
 Can it be that ALSA calls the plugin initialization routine only once
 and not on each sample rate change ?

Since it works properly with unusual sampling rates (like 60kHz) I don't
think it's an ALSA issue.  It only fails when using a sampling rate
directly supported by the underlying hardware - in that case it still
needs to do resampling, but for whatever reason the audio is *not*
resampled and playback fails.

At some point there is a sample rate check, and instead of checking what
sample rates the next *plugin* in the chain supports, the code
incorrectly checks what sampling rates the *hardware* supports.  The
hardware supports four different sample rates, the plugin only supports
one - so sometimes the audio isn't being resampled when it should be.

Cheers,
Adam.


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