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

[Aurelien Grimaud]

Hi Terry, 

Is your receiver actually receiving stream ?


You have to specify caps for faad to decode stream.


Use gst-launch -v on your sender to get "application/x-rtp" caps used
for aac and report them as a caps filter in your receiver.


gst-launch -vm udpsrc port=19790 ! "application/x-rtp, media=audio,
clock-rate=8000, encoding-name=MPEG4-GENERIC, ....." ! rtpmp4gdepay !
faad ! fakesink





Also, your aac maybe stereo and amrnbenc only supports mono, so you
have to use deinterleave.



Aurelien

--- En date de : Lun 25.8.08, Terry Leung <terry83 at gmail.com> a écrit :
De: Terry Leung <terry83 at gmail.com>
Objet: [gst-devel] convert aac rtp stream to amr stream
À: gstreamer-devel at lists.sourceforge.net
Date: Lundi 25 Août 2008, 6h05

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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