Re: Quality resampling code for libasound

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At Tue, 20 Feb 2007 00:38:16 +1100,
Jean-Marc Valin wrote:
> 
> >> What?? I hope you're joking here. It's one thing if you don't want my
> >> code for a reason or another, but being happy to keep things as they are
> >> currently is just insane. What libasound does is not resampling, it's
> >> audio mutilation, plain and simple. It's not an audiophile thing we're
> >> talking about. The aliasing is so bad people notice it even with speech
> >> (which usually tolerates a lot of abuse) and think it's an application
> >> bug (e.g. http://www.mail-archive.com/ekiga-list@xxxxxxxxx/msg01316.html
> >> ). This is only getting worse as people get new machines with HDA-based
> >> cards that only do 44.1/48 kHz, so it'll become even more embarrassing.
> >> Do you expect most Linux users who use a soundcard to know that they
> >> need a plugin to have their cards actually reproduce sounds properly?
> > 
> > Yes.  Of course, we can do eventually merge, but let's see the result
> > at first.  I don't want to be a politician who promisses the vapor
> > things.
> 
> As I said in my original post, I'm quite willing to help whoever is
> interested in integrating this code in libasound -- but at this point,
> I'm not planning on writing the plugin itself. Any taker?

Well, did you take a look at alsa-plugins/rate/rate_samplerate.c?
That's _much_ easier than rewrting alsa-lib pcm_rate.c.  This is
another reason I recommended the external plugin as a first attempt.
(And we can change the default thing later on, for example, after
 checking whether it gets no performance drawback.)

> > That's why I asked you kick off the patch first...
> 
> Patch attached. This is based on my (very little) understanding of the
> sample conversion code, so it could also be just dead wrong.

I'll check it later.  Thanks!


Takashi

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