Re: alsa, oss and jack

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On Sat, 30 Dec 2006, Dominique Michel wrote:

> Hi all,
>



>
> A sub question is about the rate_convert plugin on
> http://www.alsa-project.org/alsa-doc/doc-php/asoundrc.php
> My audigy card use 48kHz as internal sampling rate and a 16 bits internal
> format. Will a rate_convert plugin be necessary in order to increase the
> performances of the dmix plugin?
rate conversion will always decrease the quality of the sound. HOwever if
you are mixing then all rates must be the same. Also some sound cards (eg
the Creative line) only do 48000 but CDs are all at 44100. Thus rate
conversion is absolutely required. 
Decent rate conversion is very hard to do on the fly. The standard on the
fly is linear inperpolation which isi fast and attrocious.

>
> The .asoundrc I used for my test was as follow:
>
> pcm.oss {
>    type oss
>    device /dev/dsp
>    }
>
> pcm.jack {
>    type jack
>    playback_ports {
>        0 alsa_pcm:playback_1
>        1 alsa_pcm:playback_2
>        }
>    capture_ports {
>        0 alsa_pcm:capture_1
>        1 alsa_pcm:capture_2
>        }
> }
>
> pcm.!default {
>        type plug
>        slave.pcm "dmixer"
> }
>
> pcm.dmixer  {
>        type dmix
>        ipc_key 1024
>        slave {
>            pcm "hw:0,0"
>            period_time 0
>            period_size 1024
>            buffer_size 4096
>            rate 44100
>        }
>        bindings {
>            0 0
>            1 1
>        }
>
> pcm.dsp0
> {
>    type plug
>    slave.pcm "dmixer"
> }
>
> ctl.dmixer0 {
>        type hw
>        card 0
> }
>
> Another issue is at the oss applications as audacity was not working with this
> file but only complaining about no sound device.

You must have teh alsa oss emulation set up. (look to see if the files
/dev/dsp* exist.)

>
> Best,
> Dominique
>
>
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