Subject: Re: [linux-audio-dev] low latency + mp3
From: brian redfern (nutmeggatron_AT_yahoo.com)
Date: Mon Oct 22 2001 - 20:27:28 EEST
Check out terminatorX, its a dj - style gui interface
to mpg123, and you have user created buffers, so you
can run mpg123 with 4 buffers if you want.
--- Andy Wingo <apwingo_AT_eos.ncsu.edu> wrote:
> Hi David,
>
> On Mon, 22 Oct 2001, David Burrows wrote:
> > My question is this. How would one go about
> modifying say mpg123, so that
> > it uses 3x256 (or 3x128) byte buffering? I've ran
> the source through
> > doxygen, and started to have a brief look into it
> but I am wondering if
> > someone has already achieved this or could give a
> description of where to
> > start.
>
> The question becomes, why? Latency is important when
> receiving audio processing
> it, and resending it somewhere else. It is also
> important when composing on the
> computer. But playing back mp3s? What does having 2
> ms buffers for mp3 playback
> get you? I'll have to admit I don't quite
> understand.
>
> > The bonus question is about pitch control. I
> understand that this can be
> > achieved by simply changing the sampling rate,
> however, I'm wondering if
> > anyone has knowledge of fast or high quality
> resampling algorithms?
>
> Of course, I'm biased, having worked with GStreamer
> a bit. We have a couple of
> mp3 decoders (mpg123 and mad, which is imho better),
> a resampler (called 'speed'
> - just a linear interpolator), and an alsa sink that
> you can set to the desired
> period size and number. So if you want to play
> around with this sort of thing,
> I'm sure that there are other tool kits as well. Be
> sure to use CVS gstreamer if
> at all, at least until our next release. Also, the
> code is on sourceforge, so
> you can browse it directly: gstreamer.net/cvs/
>
> Good luck, but consider your needs first.
>
> wingo.
=====
This message was
sent by
Brian W. Redfern
Have a nice day :=)
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