[gst-devel] AC3 audio on gstreamer + rtp

[Sergio Belmar Argudo]

I'm trying to send audio form a mp3 file with gstreamer encoded with faac on RTP.


The problem is that I can't play it with any player (I like to play it with mplayer, vlc, etc. using a SDP file)



I'd tested it with a mp3 file and the following command:



gst-launch-0.10 -v filesrc location=/share/sample.mp3 ! flump3dec !
queue max-size-buffers=0 max-size-time=0 ! ffenc_libfaac ! queue
max-size-buffers=0 max-size-time=0 ! queue max-size-buffers=0
max-size-time=0 ! rtpmp4apay ! udpsink host=192.168.1.168 port=6060


and I have this SDP:



v=0

o=- 116704633 116704633 IN IP4 127.0.0.1

s=GStreamer

t=0 0

a=range:npt=now-

a=control:trackID=1

c=IN IP4 192.168.1.168

m=audio 6060 RTP/AVP 96

a=rtpmap:96 MP4A-LATM/48000/2


But there is no sound in mplayer or vlc.

I also tried faac and ffenc_libfaac to encoding, rptmpapay and rtpmpa4pay to packetise, MP4A-LATM and mpeg-generic in SDP file..., but nothing.


Has Anybody any idea?

Has Anybody tried anything like that?

Is it a problem only with audio encoding or SDP file config or is there a bug in any part?



Thanks



      
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