Realtime audio over UDP - Low quality

[Nicolas Dufresne]

Le 4 nov. 2016 1:42 PM, "David Ventura" <davidventura27 at gmail.com> a écrit :
>
> Hi. I'm trying to get audio (voice only) over udp on local network.
>
> First: Listening to the mic directly through alsa gives me great quality;
there's no overdrive when shouting/talking WAY too close to the mic.
>
> Now, when using the following pipeline the audio is .. ok. The quality is
not 'good'. It's good enough, but there's a clear quality drop from
directly listening to the mic. There's overdrive when there's shouting, etc.

That is likely the side effect of using mu-law compression. It should sound
like weird phone at best.

Maybe your use case allow using raw pcm (l16pay) or opus codec if
compression matter (you can even turn on music mode on opus, it gives great
results).

>
>
> Producer:
>
> gst-launch-1.0  alsasrc slave-method=resample do-timestamp=true !
audioconvert ! audioresample ! mulawenc ! rtppcmupay ! multiudpsink clients=
192.168.2.120:5001
>
> Consumer:
>
>
> 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
>
> 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.
>
> --
> Stack is the new term for "I have no idea what I'm actually using".
>
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>
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