On Thu, 14 May 2009 07:39:14 -0700
Grant <emailgr...@gmail.com> wrote:
> >> >> 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
>
(For no good reason ?) try as root, play something long, increase runtime
priority while playing.
Of course, I am not sure, I'm just suggesting to check whether priority
is the issue.
By the way, 96KHz sampling rate is a pretty high load for the system.
Regards.
Sergei.
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