Am 23.03.2013 12:11, schrieb Toby Smithe:
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 this is a great project idea. I've been suffering from the
problems you mentioned as well. However I think a lossless codec should
be considered. Lossy codecs might not be acceptable for lots of people
and a lossless codec might improve things sufficiently. But I can see a
lossy codec being useful also for congested networks, so it could be an
option.
Best regards.
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