[gst-devel] Re : convert aac rtp stream to amr stream
Aurelien Grimaud
gstelzz at yahoo.fr
Mon Aug 25 10:34:39 CEST 2008
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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