Realtime audio over UDP - Low quality

[Tim Müller]

On Fri, 2016-11-04 at 14:28 -0300, David Ventura wrote:

Hi,

> gst-launch-1.0 -q udpsrc port=5001 caps="application/x-rtp" do-
> timestamp=true ! rtppcmudepay ! mulawdec 
> ! audioconvert ! audioresample ! audio/x-
> raw,format=S16LE,layout=interleaved, rate=44100, channels=2 ! fdsink
> fd=1 sync=true 

fdsink? I think you also want an rtpjitterbuffer after udpsrc (set
latency property, default latency is quite high), and then perhaps try 

 .. ! audiorate ! wavenc ! filesink location=foo.wav

for starters.

> What's happening? Is this the recommended way to do audio over udp?
> I have gstreamer 1.4.4 on the producer and 1.9.90 on the consumer.

https://cgit.freedesktop.org/gstreamer/gst-plugins-good/tree/tests/exam
ples/rtp

has some examples for various things.

Cheers
 -Tim

-- 
Tim Müller, Centricular Ltd - http://www.centricular.com
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