[pulseaudio-discuss] GSoC ideas

[Arun Raghavan]

On Fri, 2013-04-26 at 22:05 +0200, Thomas Martitz wrote:
> Am 26.04.2013 14:54, schrieb David Henningsson:
> >>>
> >>> My ideas for a gsoc application:
> >>> - Fix network sinks. Try to move a stream to network sink and back
> >>> moments later it will run into problems.
> >>>      e.g. mplayer just stop playing and hang. My job would be
> >>> additional testing and fixing upcoming bugs in pulseaudio.
> >>
> >> Making module-tunnel-sink reliable would be very welcome. Estimating the
> >> amount of work is hard, though, when you don't know what exactly are the
> >> root causes for the bugs, which makes writing the project plan hard too.
> >
> > I'd like to see a rewrite of module-tunnel-sink to use the libpulse 
> > API instead of doing the protocol stuff directly.
> >
> > I also think that wifi + TCP + low latency is a very hard thing to 
> > achieve reliably. The question is if it is possible at all, and if 
> > not, what the options are. Arun didn't seem very happy about improving 
> > RTP support in PulseAudio. 
> 
> 
> How could you possibly be unhappy about "improving RTP support"? Right 
> now it's basically unusable (for me, anyway) due to this problem/bug: 
> https://bugs.freedesktop.org/show_bug.cgi?id=44777

I'd be delighted to have better RTP support. But I think it is somewhat
misguided to try to maintain a mature RTP stack within PulseAudio. Full
post is at:

http://lists.freedesktop.org/archives/pulseaudio-discuss/2013-March/016667.html

Summary of that: it makes much more sense to reuse something mature like
GStreamer's RTP elements for a large number of reasons such as active
maintenance and easy support for compressed output.

-- Arun