[pulseaudio-discuss] GSoC Proposal: Resampling Improvements

[Damir Jelić]

Hi.

For those who don't know, I'm Damir Jelić from Croatia and I would like to
work on the "Resampling Improvements" project during this year's Summer of
Code.

Last year I participated and finished the "Latency offset" project. After
the summer I tried to become a regular contributor. For those who are
interested, all my contributions can be seen in the git log [1].

I know I said that last year should be my final year as a student but I
prolonged my studies for a year and I'm still eligible for this year's
Summer of Code.

I picked this project because it's a part of pulseaudio that I consider
quite interesting and didn't had the chance to touch yet.

I have some DSP background (although more from control theory than from
digital audio) and also I have some assembly background, but not from any
modern instruction set (z80), which should make me a good candidate for this
project.

What I would like to do over the summer:
    - enable resampling with libav
    - write some test coverage for the libav resampling method
    - deprecate the ffmpeg resample method
    - update the speex resample method

The ideas page mentions also libresample, if I'm not mistaken pulseaudio
already supports resampling via libresample and I'm not sure what would I
need to do about this.

Speex has been obsoleted [2] and the resampler seems to had been moved to
opus-tools. There are some interesting commits [3] inside this repo and I think
it would be nice to update our speex resample method although I'm not sure
if it's ok to drop speex and replace it with opus-tools just like that.

Suggestions and comments are welcome.

Thanks, poljar.

[1] http://cgit.freedesktop.org/pulseaudio/pulseaudio/log/?qt=author&q=poljar
[2] http://www.speex.org/
[3] http://git.xiph.org/?p=opus-tools.git;a=history;f=src/resample.c;h=d957090b2910c212abba567b0ec2f52f2beadb4a;hb=HEAD