AW: [GST-DEVEL]Sending and receiving RTP audio

[Matthias Dodt]

Wesley!

You are GREAT! The reason for the messy sound was the missing "audioresample" plugin! If I insert it infront of directsoundsink it works perfectly! It is weired though that it works without the "audioresample" if I use UDP transfer only (without gstrtpbin). Another gstreamer lesson learned;-)

Thanks again!!

Best,

mat

-----Ursprüngliche Nachricht-----
Von: gstreamer-devel-bounces+mdodt=xion-medical.com at lists.freedesktop.org [mailto:gstreamer-devel-bounces+mdodt=xion-medical.com at lists.freedesktop.org] Im Auftrag von Wesley J. Miller
Gesendet: 08 September 2011 14:54
An: Discussion of the development of and with GStreamer
Betreff: Re: [GST-DEVEL]Sending and receiving RTP audio

Mat,

I had a problem a while back with terrible audio between two boxes using Linux GST.  There were two things that helped.

1.  Break the caps out of the receiver's udpsrc.  Instead code them as a caps filter / pipe stage.  No idea why this helped.

        udpsrc uri=udp://224.0.1.5:5002
        !
'caps=application/x-rtp,clock-rate=44100,encoding-name=L16,media=audio,chan
nels=1'
        ! rtpbin.recv_rtp_sink_0


2.  Since I was using alsasink or pulsesink, I was able to specify
sync-false.     Don't think you can do this with directsoundsink,

Also, from the online docs for directsoundsink:

Note that you should almost always use generic audio conversion elements like audioconvert and audioresample in front of an audiosink to make sure your pipeline works under all circumstances (those conversion elements will act in passthrough-mode if no conversion is necessary).




Wes


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