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.
https://launchpad.net/~diwic
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