Realtime audio over UDP - Low quality

[David Ventura]

Hi Tim,

I'm using fdsink because I'm feeding the output to another program.

Using rtpjitterbuffer seems to have helped a lot.

Testing server-alsasrc-PCMA.sh and client-PCMA.sh seems to reduce 'noise' a
lot (currently using alsasrc with the input OFF and I hear a little
crackling.. a lot less than using my current method). The problem is the
receiver doesn't survive a producer restart, in our setup the producer
starts/stops at any time

Is this solvable?


Also, what's with the crackling? Running

>gst-launch-1.0 alsasrc ! audiorate ! wavenc ! filesink location=foo.wav

I hear a little hissing in the background, but that's it, no crackling at
all.

David

On 4 November 2016 at 15:04, Tim Müller <tim at centricular.com> wrote:
>
> 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/examples/rtp
>
> has some examples for various things.
>
> Cheers
> -Tim
>
> --
> Tim Müller, Centricular Ltd - http://www.centricular.com
>
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>



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