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
