Problem Streaming Live Audio Using RTP and UDPSink

[Tiago Katcipis]

On Mon, Jul 18, 2011 at 3:23 PM, William Metcalf <wmetcalf at niftytv.com>wrote:

> I am trying to capture live audio from a capture card in a gstreamer
> application, encode it into AAC format, and then send the audio over a
> network using UDPSink and RTP.  My method of accomplishing this task works
> almost perfectly, except that occasionally the audio will become very jumpy
> for a few seconds and then return to normal, and then after a few seconds it
> will get jumpy, etc.  I am not getting any errors when I play the audio, so
> I am assuming that it must be some property I am not setting correctly, or
> maybe there is an element I am missing which can help solve the problem.  My
> pipelines are as follows:
>
> Server Pipeline: appsrc max-bytes=8000 is-live=true typefind=true !
> audioparse ! faac bitrate=320000 ! rtpmp4apay host=192.168.42.68 port=52222
>
> Client Pipeline: udpsrc port=52222 ! "application/x-rtp,media=(**
> string)audio,clock-rate=(int)**44100,encoding-name=(string)**MP4A-LATM,payload=(int)96"
> ! rtpmp4adepay ! "audio/mpeg,mpegversion=(int)**
> 4,channels=(int)2,rate=(int)**44100" ! faad ! autoaudiosink
>
> I am very close to having this work, so any help that anyone can provide
> will be greatly appreciated!
>

Try:

udpsrc port=52222 ! "application/x-rtp,media=(stri
ng)audio,clock-rate=(int)44100,encoding-name=(string)MP4A-LATM,payload=(int)96"
! gstrtpjitterbuffer ! rtpmp4adepay !
"audio/mpeg,mpegversion=(int)4,channels=(int)2,rate=(int)44100" ! faad !
autoaudiosink

and consider using the whole gstrtpbin when dealing with RTP streams.

Hope this helps.

Best regards,
Katcipis

>
> William
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