Re: Higher quality dmix resampling

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>> >> I'm playing a video in miro and I get:
>> >>
>> >> # lsof|grep speex
>> >> miro.real  9019     user  mem       REG                8,3    108992
>> >> 28197654 /usr/lib64/libspeex.so.1.4.0
>> >>
>> >> Does this mean dmix is using speex?  If so, what else could be causing
>> >> my static problem?  I basically hear static whenever dmix is involved.
>> >>  If I have mpd resample with libsamplerate, I get no static.
>> >>
>> >> - Grant
>> >>
>> >
>> > Yes, you are using speex.
>>
>> I don't think my defaults.pcm.rate_converter is being obeyed.  I
>> switched from "speexrate_best" to "samplerate_best" and also tried
>> removing the definition entirely, but lsof still says whichever
>> program is playing audio is opening the speex file and not the
>> libsamplerate file.
>>
>> I also tried removing speex from the system and speex disappeared from
>> lsof, but the static remained.
>>
>> > I suggest first of all to temporarily leave 'miro' aside - it's a
>> > non-trivial piece of SW which might have its own quirks.
>> >
>> > I suggest to start from very basic 'aplay' with .wav files - just to
>> > make sure ALSA works OK.
>>
>> I can definitely confirm static with aplay .wav files that doesn't
>> exist in mpd.  If I don't have mpd bypass dmix I get static there too.
>>  Where should I go from here?
>>
>> # lsof|grep aplay
[snip]
>>
>> - Grant
>>
>>
>> > Then, say, 'mplayer' with .flac, .mp3.
>> >
>> > You can try to increase ALSA buffers size, but I do not remember how to
>> > do this, though I remember it was easy.
>> >
>> > Regards,
>> >  Sergei.
>>
>
> Then start from very basic things:
>
> 1) choose direct HW output;
> 2) choose sample rate supported by HW - if necessary, resample your
> input file by high quality stand-alone resampler;
> 3) also take care of number of bits if necessary;
> 4) start playing with ALSA buffer size.
>
> For resampling/format conversion you can use 'ecasound' or 'sox'.
>
> Disclaimer: I am not an ALSA developer, so my recommendation are from
> end user point of view.
>
> Regards,
>  Sergei.

I added this to /etc/asound.conf:

pcm.!default {
type plug
slave.pcm {
type dmix
ipc_key 1024
slave {
pcm "hw:0"
format S24_3LE
rate 96000
}
}
}

I can see that it works because the 96k LED lights up on the DAC, but
the static remains.  I've also tried it in combination with:

defaults.pcm.rate_converter "samplerate_best"

I also tried various values of buffer_size and it caused some skipping
but didn't affect the static at all.

- Grant

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