[gst-devel] bidirectional audio/video streaming issue

[na]

Hello,

I have had success in streaming audio and video unidirectionally from a
custom application to a gstreamer pipeline.  Now, I am trying to stream
audio and video bidirectionally both to and from the gstreamer pipeline to a
custom application.  

I am using a file as a source and I want to encode the audio as PCMU and the
video as H.264 @ 320x200. The video I am trying to stream has a different
resolution and audio format than I want to use, hence the conversions. The
error I am seeing is shown below.

I have also attached the pipeline that I am using. I can use the same
pipeline minus the last 5 lines as a receiver without issue.

Any help is greatly appreciated.

Thanks for your time.

-Nadeem

---------
Error:

0:00:34.317748445 30473  0x8967590 WARN                 basesrc
gstbasesrc.c:2378:gst_base_src_loop:<udpsrc1> error: Internal data flow
error.
0:00:34.317799005 30473  0x8967590 WARN                 basesrc
gstbasesrc.c:2378:gst_base_src_loop:<udpsrc1> error: streaming task paused,
reason not-linked (-1)
ERROR: from element /GstPipeline:pipeline0/GstUDPSrc:udpsrc1: Internal data
flow error.
Additional debug info:
gstbasesrc.c(2378): gst_base_src_loop ():
/GstPipeline:pipeline0/GstUDPSrc:udpsrc1:
streaming task paused, reason not-linked (-1)
Execution ended after 33677594302 ns.
Setting pipeline to PAUSED ...
Setting pipeline to READY ...

--------------
My pipeline:

LATENCY=200
VIDEO_CAPS="application/x-rtp,media=(string)video,payload=(int)97,clock-rate=(int)90000,encoding-name=(string)H264"
AUDIO_CAPS="application/x-rtp,media=(string)audio,payload=(int)0,clock-rate=(int)8000,encoding-name=(string)PCMU"

AUDIO_IN="udpsrc caps=$AUDIO_CAPS port=5000"
AUDIO_RTCP_IN="udpsrc port=5001"
VIDEO_IN="udpsrc caps=$VIDEO_CAPS port=5002"
VIDEO_RTCP_IN="udpsrc port=5003"
AUDIO_OUT="udpsink port=6000"
AUDIO_RTCP_OUT="udpsink port=6001 host=$DEST sync=false async=false"
VIDEO_OUT="udpsink port=6002"
VIDEO_RTCP_OUT="udpsink port=6003 host=$DEST sync=false async=false"

VIDEO_PLAYOUT="rtph264depay ! ffdec_h264 ! ffmpegcolorspace ! videoscale !
ximagesink"
AUDIO_PLAYOUT="rtppcmudepay ! mulawdec ! audioconvert ! audioresample !
alsasink"

AV_OUT_SRC="filesrc location=hdv_0015_480p_384kbps_20fps.mp4 ! qtdemux
name=demux"
VIDEO_ENC="demux.video_00 ! queue ! ffdec_h264 ! ffmpegcolorspace !
videoscale ! video/x-raw-yuv,height=320,width=200 ! x264enc bitrate=200 !
rtph264pay"
AUDIO_ENC="demux.audio_00 ! queue ! faad ! audioconvert ! mulawenc !
rtppcmupay"

gst-launch -v --gst-debug-level=2 gstrtpbin name=rtpbin latency=$LATENCY \
     $VIDEO_IN ! rtpbin.recv_rtp_sink_0 \
       rtpbin. ! $VIDEO_PLAYOUT \
     $AUDIO_IN ! rtpbin.recv_rtp_sink_1 \
       rtpbin. ! $AUDIO_PLAYOUT \
     $VIDEO_RTCP_IN ! rtpbin.recv_rtcp_sink_0 \
       rtpbin.send_rtcp_src_0 ! $VIDEO_RTCP_OUT \
     $AUDIO_RTCP_IN ! rtpbin.recv_rtcp_sink_1 \
       rtpbin.send_rtcp_src_1 ! $AUDIO_RTCP_OUT \
     $AV_OUT_SRC \
       $VIDEO_ENC ! rtpbin.send_rtp_sink_0 \
          rtpbin.send_rtp_src_0 ! $VIDEO_OUT \
       $AUDIO_ENC ! rtpbin.send_rtp_sink_1 \
          rtpbin.send_rtp_src_1 ! $AUDIO_OUT

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