[MPlayer-dev-eng] [PATCH] fix for -srate bug

[Ed Wildgoose]

>Ok, this solve part of my trouble. 
>The complete story is here.
>
>There are three mode of using the resample filter:
>
>1) mplayer 18010.asf -af resample=44100:0:0 (fast mode with distortion)
>2) mplayer 18010.asf -af resample=44100:0:1 (normal mode without distortion)
>3) mplayer 18010.asf -af resample=44100:0:2 (normal mode without distortion but with some noise)
>  
>

Aha, this might solve a problem I was just about to ask about:

I have my jack setup locked at 44100 for various reasons and so I am get 
mplayer to resample from 48K to 44.1K.  However, it sounds terrible and 
human voices often have a kind of "buzz" to them.  It's like a kind of 
ringing sound on certain words, probably just when they hit a certain 
freq of course.

 Can someone please provide a pointer to some docs on the meaning of the 
modes 2) and 3)? 

Also, I notice that the resampler seems to reduce the volume by perhaps 
4-6 db?  In my case that is rather unfortunate because I am struggling 
to keep the audio as loud as possible (passive volume control after the PC).

Would a patch to incorporate libsamplerate be acceptable?  We obviously 
don't want a ton of sample rate convertors, and I would expect it to be 
tricky to completely integrate libsamplerate (if we need to be able to 
compile it on funny architectures for example), but I could make it a 
compile time option to use the internal stuff or libsamplerate? 

Libsamplerate needs only about three function calls (start, stop and 
filter), and so it is trivial to integrate and the quality is absolutely 
first rate with about the same CPU requirements as the best mplayer 
convertor.  It has 5 different convertors though for those who have 
lower CPU requirements.

If someone can help with a few details about modifying config and make 
files then I would be very happy to submit such a patch?

Ed W