[gst-devel] convert aac rtp stream to amr stream

[Terry Leung]

Hi all,

I am trying to convert a aac rtp stream to a amr(nb) stream

Here is the pipleine i am using

generate aac stream from a 3gp file:
gst-launch -m filesrc location=test-aac.3gp ! qtdemux ! rtpmp4gpay !
udpsink  host=172.20.122.9 port=19790

receive and convert the stream:
gst-launch -m udpsrc port=19790 !  rtpmp4gdepay ! faad ! queue !
amrnbenc ! rtpamrpay pt=98 ! udpsink  host=172.20.122.23 port=2006

I have tried to capture those packet, I can see that there is acc rtp
packet generated by the first pipleine but it dont have packet come
out from the second piple
Also, I am not sure if it is related to the udp checksum
The udp checksum generated by the first pipeline is wrong in wire shark
I also attach the capture file


I also have another two pipeline doing similar job but it is for g711 to amr
those two pipeline work gracefully

gst-launch -m filesrc location=test3.wav ! wavparse ! audioconvert !
audioresample !alawenc !  rtppcmapay ! udpsink  host=172.20.122.9
port=19790

gst-launch -m udpsrc port=19790 !  rtppcmadepay ! alawdec ! queue !
amrnbenc ! rtpamrpay pt=98 ! udpsink  host=172.20.122.23 port=2006

The difference from the first pipeline is that the udpsink of the
second piple change its state to playing when rtp stream arrive from
the udpsrc

Anyone have idea to solve this?

Terry
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