[mythtv-users] Multiple cards with different audio sample rates

Michael Lynch lynchmv at gmail.com
Fri Jun 2 19:55:41 UTC 2006


On 6/2/06, Michael T. Dean <mtdean at thirdcontact.com> wrote:
> On 06/02/06 00:09, Michael Lynch wrote:
>
> >I have 2 cards in my backend, one bttv based with audio cable from
> >line-out to line-in and one saa713x card that captures audio from the
> >pci bus.
> >
> >The bttv card needs to have a sample rate of 48000 or the audio gets
> >terrible (humming) and the saa713x card needs to have a sample rate of
> >32000.  Herein lies my problem.  With the audio sample rate set to
> >48000 for the 'Live TV' recording profile, the bttv audio is perfect,
> >but the saa713x audio turns to chipmunks.  If it change the 'Live TV'
> >audio sample rate to 32000 to accomodate the saa713x card, the bttv
> >card begins to hum.
> >
> >Being that both cards are 'Analog V4L capture card' card types when
> >setting up recording profiles I can not figure out what to do.  I have
> >tried setting up a custom recording profile, but since both cards are
> >'Analog V4L capture card' then the audio is still messed up.
> >
> >Any ideas how to set different recording profiles for 2 v4l based
> >cards with different audio needs?
> >
>
> The problem with 32kHz isn't your recording profile, it's your
> (playback) ALSA configuration.
> http://www.mythtv.org/wiki/index.php/DigitalSoundHowTo#Setting_up_ALSA.27s_.asoundrc.2C_Properly
>
> Fix that and you can play back the 32kHz audio without problems.
>
> Mike
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Does it matter if I'm not using the digital out?

The above question aside...if I understand the wiki page correctly, I
can pretty much drop in that config file with minor changes.

I need to verify the device number for the "analog-hw" and/or
"digital-hw" virtual devices, and change them accordingly in the
config.  Running aplay -l gives me card 0 with device 0 as ICH5 and
card 0 with device 4 as ICH5-IEC958.  That being said after I change
the settings in the config file should I change anything else?
Reading on the alsa-project web page it states that the
.asound/asound.conf files "allow extra functionality, such as routing
and sample-rate conversion, through the alsa-lib layer" - I'm guessing
this is how the captured audio gets 'fixed' so that my playback is
good no matter what.

If I'm missing the whole point, please let me know.


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