Re: Quality resampling code for libasound

[Jean-Marc Valin <jean-marc.valin@usherbrooke.ca>]

>>> That's one part making alsa-lib rate plugin so complicated.
>>> Anyways, the resampling algorithm plugin doesn't take care much about
>>> it, so far.
>> Are you telling me people use the high-quality sinc in libsamplerate
>> just to have the plugin drop samples once in a while? Can we just have a
>> resampler that actually produces valid output once and for all? I guess
>> it's another good reason to go directly with a non-plugin resampler.
> 
> No, I mean that there are many black magics in the alsa rate plugin to
> calculate the sampling buffer size constant so that the resampler side
> doesn't take care of it so much.  And this has nothing to do with the
> fact whether it's an external plugin or not.

OK, so plugin or not, how do you actually make sure you process all
samples and not just the ones that are the most convenient? I can't
believe I'm the only one here who actually cares about having correct
audio coming in/out the soundcard.

> I understand that you would like to put that to alsa-lib.  But, it
> would reduce neither code size and complication.  So let's prove the
> quality and performance first as a plugin, then replace the default
> one later.

Well, in terms of quality, Erik was kind enough to produce these graphs
of a chirp being down-sampled from 96 kHz to 44.1 kHz:
http://www.mega-nerd.com/tmp/speex_resampled_q10.png
http://www.mega-nerd.com/tmp/speex_resampled_q5.png
It should be pretty clear that the main limitation will be the 16-bit
resolution (the aliasing lines are all below -120 dB). That, and of
course the "discarding every N sample" problem, which is really bad.

As far as performance goes, the code is really fast in the general case
(faster than libsamplerate according to the tests I made) and even 4x
faster than that when doing simple conversions (integer ratio of simple
fraction). BTW, have you tested the plugin I sent yet? In the worst
case, the default can be set to the lower quality resampling modes,
which are still way better than linear resampling and yet take nearly no
CPU.

> The actual problem isn't that.  It's the restriction of "period" model
> that the current ALSA PCM is based on.

And what's the solution to that one. What do you currently do with the
linear resampler anyway? Drop samples again?

	Jean-Marc

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