Hello All! This is a fairly lengthy question. Please bear with me. I am using gstreamer to implement voice conference between two systems. One system is Fedora Core 9 PC (gstreamer 0.10.20 binaries)
and the other is
an ARM core based Multimedia development board (TI Davinci DVEVM board,
with MontaVista 2.6.10 kernel). I am using gstreamer to stream and playback on both sides. The problem that I am facing is as below: If I start the the voice conference both ways, the voice play back at board does not work. If I capture, encode, and stream from PC to the board and play back at board, It Works. If I capture, encode, and stream from board to the PC, and play back at PC, It Works. If I start both (by starting play back at the board, once this works fine,
start the capture, encode and stream on the board. Correspondingly on the PC side also.)
In this scenario, I am not able to hear anything on the PC side. If I reverse the scenario (first capture and streaam at board and later
play back at board) I am not able to here voice on the board. Any idea about what could be going wrong in this kind of use case? The pipelines that I am using at the board and on the PC are as follows: At Board: --------- Playback pipeline ------------------------------ export STREAM_HOST=172.22.64.225 gst-launch-0.10 -m -v gstrtpbin name=rtpbin \ udpsrc port=9011 \ caps="application/x-rtp,media=(string)audio,encoding-name=(string)PCMA,clock-rate=(int)8000" ! \ rtpbin.recv_rtp_sink_5 rtpbin. ! \ rtppcmadepay ! alawdec ! osssink sync=false \ udpsrc port=9012 ! rtpbin.recv_rtcp_sink_5 \ rtpbin.send_rtcp_src_5 ! udpsink host="$STREAM_HOST" port=9013 sync=false async=false -t
Stream pipeline ------------------------------ export PLAY_HOST=172.22.64.225 # FC9 PC gst-launch-0.10 -m -v --gst-debug-level=0 \ --gst-debug=GST_CAT=0:GST_ELEMENT_*=0:GST_CALL_TRACE=0:GST_ELEMENT_PADS=0:GST_MESSAGE=0:GST_PIPELINE=0:GST_SCHEDULING=0:\ GST_STATES=0:GST_SIGNAL=0:GST_BUS=0,osssrc:0,gstrtpbin:0 \ gstrtpbin name=rtpbin \ osssrc device="/dev/dsp" ! \ capsfilter caps="audio/x-raw-int,channels=2,endianness=1234,width=16,signed=true,rate=8000" ! \ deinterleave name=de de.src0 ! \ alawenc ! \ rtppcmapay ! \ capsfilter caps="application/x-rtp,media=(string)audio,clock-rate=(int)8000,encoding-name=(string)PCMA" ! \ rtpbin.send_rtp_sink_0 rtpbin.send_rtp_src_0 ! \ udpsink host="$PLAY_HOST" port=9000 auto-multicast=false \ rtpbin.send_rtcp_src_0 ! udpsink host="$PLAY_HOST" port=9001 auto-multicast=false sync=false async=false -t \ udpsrc port=9002 ! rtpbin.recv_rtcp_sink_0
------------------------------ At PC: ------- Playback pipeline ------------------------------ export STREAM_HOST="172.22.67.149" #ARM Board gst-launch-0.10 -m -v gstrtpbin name=rtpbin latency=1000 \ udpsrc port=9000 \ caps="application/x-rtp,media=(string)audio,encoding-name=(string)PCMA,clock-rate=(int)8000" ! \ rtpbin.recv_rtp_sink_0 rtpbin. ! \ rtppcmadepay ! alawdec ! osssink sync=false \ udpsrc port=9001 ! rtpbin.recv_rtcp_sink_0 \ rtpbin.send_rtcp_src_0 ! udpsink host="$STREAM_HOST" port=9002 sync=false async=false -t Stream pipeline ------------------------------ export PLAY_HOST="172.22.67.149" # ARM board gst-launch -m -v --gst-debug-level=0 --gst-debug-no-color \ --gst-debug=GST_CAT=0:GST_ELEMENT_*=0,gstrtpbin:0,osssrc:0,audiosrc:0,baseaudiosrc:0,alawenc:0 \ gstrtpbin name=rtpbin latency=0 \ osssrc device="/dev/dsp" do-timestamp=true ! \ capsfilter caps="audio/x-raw-int,width=16,depth=16,endianess=1234,rate=8000,channels=1" ! \ alawenc ! \ rtppcmapay ! \ rtpbin.send_rtp_sink_3 rtpbin.send_rtp_src_3 ! \ udpsink host="$PLAY_HOST" port=9011 auto-multicast=false \ rtpbin.send_rtcp_src_3 ! udpsink host="$PLAY_HOST" port=9012 auto-multicast=false sync=false async=false -t \ udpsrc port=9013 ! rtpbin.recv_rtcp_sink_3
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