[pulseaudio-discuss] GSoC: Call for project ideas

David Henningsson david.henningsson at canonical.com
Sun Mar 24 23:13:24 PDT 2013

On 03/23/2013 12:11 PM, Toby Smithe wrote:
> Hi,
> I have a fairly free summer coming up, and thought it would be nice to
> participate in GSoC. For a while I've been interested in PulseAudio, and
> I have an idea for a project. I wonder if you might say whether you
> think it plausible.
> I use PulseAudio's native protocol streaming quite a lot, and I've
> noticed that it seems quite rudimentary. I read the code a couple of
> releases back, and it seems just to stream uncompressed PCM over
> TCP. With a wireless connection and multi-channel audio, this quickly
> becomes impractical, with drops and latency problems. A while ago, I
> looked into implementing Opus compression for the network streams, but
> never had a chance. I think Opus would make the ideal codec because it
> is very flexible, recently ratified as an Internet standard, and can be
> remarkably lightweight (according to the official benchmarks).
> In doing these network audio, I might also be able to move on to
> auxiliary tasks like improving the GUI tools for this use-case.
> Do you think this might work?

I think it sounds interesting. Networked audio (and its latency) is 
indeed something people complain about every now and then.

But Opus is just a codec, right? Or does it also specify how it is 
actually transferred over the network (UDP/TCP etc)?

Are you planning to extend our RTP module with Opus support?

In short; Opus might be one piece of the puzzle to get reliable 
low-latency streaming, but could you also outline how you think about 
the network stack part?

David Henningsson, Canonical Ltd.

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