Re: [Alsa-user] choice of a new soundcard

2008-10-28 Thread Clemens Ladisch
Vedran Miletić wrote:
 http://www.via.com.tw/en/products/audio/controllers/envy24/
 Claims that it has 20-channel, 26-bit wide built in mixer. What does
 that mean?

The chip plays 10 channels in parallel (8 analog, 2 digital) and records
12 channels in parallel (8 analog, 2 digital, 2 monitoring).  The two
monitoring channels result from mixing together the other 20 channels.

This is not the usual hardware mixing scenario because we don't have
several independent stereo streams, and the mixing result does not go
to the speaker outputs.


Regards,
Clemens

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Re: [Alsa-user] choice of a new soundcard

2008-10-28 Thread Vedran Miletić
Thanks for clarification. There was a post on a forum (can't remember
where) that claimed that Envy24 supported HW mixing, but not in ALSA.
So, that means that ALSA already fully supports Envy24 or not?

2008/10/28 Clemens Ladisch [EMAIL PROTECTED]:
 Vedran Miletić wrote:
 http://www.via.com.tw/en/products/audio/controllers/envy24/
 Claims that it has 20-channel, 26-bit wide built in mixer. What does
 that mean?

 The chip plays 10 channels in parallel (8 analog, 2 digital) and records
 12 channels in parallel (8 analog, 2 digital, 2 monitoring).  The two
 monitoring channels result from mixing together the other 20 channels.

 This is not the usual hardware mixing scenario because we don't have
 several independent stereo streams, and the mixing result does not go
 to the speaker outputs.


 Regards,
 Clemens




-- 
Vedran Miletić
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Re: [Alsa-user] choice of a new soundcard

2008-10-28 Thread Takashi Iwai
At Tue, 28 Oct 2008 09:44:19 +0100,
=?UTF-8?Q?Vedran_Mileti=C4=87?= wrote:
 
 Thanks for clarification. There was a post on a forum (can't remember
 where) that claimed that Envy24 supported HW mixing, but not in ALSA.
 So, that means that ALSA already fully supports Envy24 or not?

Yep.  Simply it's a different meaning of hw-mixing on envy24.


Takashi

 2008/10/28 Clemens Ladisch [EMAIL PROTECTED]:
  Vedran Miletić wrote:
  http://www.via.com.tw/en/products/audio/controllers/envy24/
  Claims that it has 20-channel, 26-bit wide built in mixer. What does
  that mean?
 
  The chip plays 10 channels in parallel (8 analog, 2 digital) and records
  12 channels in parallel (8 analog, 2 digital, 2 monitoring).  The two
  monitoring channels result from mixing together the other 20 channels.
 
  This is not the usual hardware mixing scenario because we don't have
  several independent stereo streams, and the mixing result does not go
  to the speaker outputs.
 
 
  Regards,
  Clemens
 
 
 
 
 -- 
 Vedran Miletić
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[Alsa-user] Digital (spdif) and Analog capture on AU8830

2008-10-28 Thread Jacob Joseph
Hi all.

Disclaimer: This question is surely similar to many posted before here,
and to solutions suggested on the alsa pages.  Alas, I've had no
success with what I've found over the past few days.

Goal: 
I'm running an AU8830 device connected by coax spdif to a surround
receiver.  I'd like also to be able to capture a copy of the output
with arecord so that I may perform realtime analysis irrespective of
the application playing.  Unfortunately, playing on the spdif output
(hw:0,1), results in no sound in the capture.  Playing on hw:0,0 does
allow me to capture as I expect, but the spdif ouput does not work
correctly.

Basically, my tradeoffs are (spdif with distortion  capture) or (spdif
 no capture).  I'd appreciate any help resolving this.

I am playing mostly stereo music, so I'd accept tradeoffs that loose
proper surround support.  I would, however, very much like to not alter
the PCM data from a playing WAV to the stereo.  In particular, I don't
want to alter volumes, or equalize, and don't really need sample rate
conversion.

What I've tried:
To be sure, the spdif output does work with the following.  Capturing
from Mix results in no sound in the recording.
pcm.!default {
type plug
slave.pcm hw:0,1
}

I then tried using adding a plug with 'route_policy copy'.  No luck.
pcm.copy {
  type plug
  slave.pcm hw
  route_policy copy
}

Then, I tried outputting to hw:0,0 using the above pcm.copy.  This does
record the playing sound, and the spdif output does produce sound.
However, the spdif output now contains a great deal of distortion.
pcm.!default {
type plug
slave.pcm hw:0,0
}

I've also tested using the 'asym' type for pcm, and dsnooper for the
capture, but it remains true that I can capture only when playing to
hw:0,0, which yields distorted sound from the speakers.  None of these
combinations succeed.
pcm.!default {
type asym
playback.pcm nodmix
#playback.pcm both
#capture.pcm dsnooper
capture.pcm copy
}
pcm.nodmix {
type plug
#slave.pcm hw:0,0
slave.pcm hw:0,1
#bindings {
#   0 0
#   1 1
#}
}
pcm.dsnooper {
type dsnoop
ipc_key 2048
slave.pcm hw
}

It doesn't seem possible to have either dsnoop or copy use slave.pcm
hw:0,1.  arecord fails to start recording when this is the case.

Finally, I tried routing sound to both hw:0,0 and hw:0,1 using the
'multi' type.  I'm not sure I have the bindings right
(documentation seems sparse), but I know this produces distored sound
from the spdif, and allows capture:
pcm.both {
type multi
slaves.a.pcm hw:0,0
slaves.a.channels 2
slaves.b.pcm hw:0,1
slaves.b.channels 2

bindings.0.slave a
bindings.0.channel 0
bindings.1.slave a
bindings.1.channel 1
bindings.0.slave b
bindings.0.channel 0
bindings.1.slave b
bindings.1.channel 1
}


Also, is there documentation for the 'ttable' or 'bindings' directives
so often used on the alsa wiki pages?

Thanks in advance for the help!

~Jacob

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[Alsa-user] Definition of a frame

2008-10-28 Thread Bankim Bhavsar
I want to clarify my understanding of the definition of frame in Alsa lingo.

For a stereo DAC channel with 16-bit samples, number of bytes per
frame = 4 bytes, correct?
basically L  R samples each 16 bit, 16 * 2 = 32 bits = 4 bytes.

Reason being alsa-lib APIs use number of frames instead of bytes.

Thanks,
Bankim

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