Problems trying to capture RTSP stream

[Gary Thomas]

I'm trying to be able to both play (to my desktop) and record
an RTSP stream.  My source RTSP stream looks like this:
   rtsp-server "(v4l2src device=/dev/video1 queue-size=16  \
                   ! video/x-raw-yuv,format=(fourcc)UYVY,width=720,height=480,framerate=30/1 \
                   ! videoscale \
                   ! video/x-raw-yuv,format=(fourcc)UYVY,width=320,height=240 \
                   ! ffmpegcolorspace \
                   ! video/x-raw-yuv,format=(fourcc)NV12 \
                   ! queue ! ducatih264enc ! rtph264pay name=pay0 pt=96  \
                 alsasrc device=hw:0,1 ! queue ! audioconvert ! ffenc_mp2 ! rtpmpapay name=pay1 pt=97
   )"
n.b. the 'ducatih264enc' element is a hardware H.264 encoder.

This command works well for playing the audio+video stream:
   gst-launch rtspsrc location=rtsp://192.168.1.136:8554/test name=src \
     src. ! queue ! rtph264depay ! h264parse ! ffdec_h264 ! xvimagesink \
     src. ! queue ! rtpmpadepay ! mpegaudioparse ! flump3dec ! audioconvert ! alsasink

However, when I try to record the video as well using this pipeline:
   gst-launch -e rtspsrc location=rtsp://192.168.1.136:8554/test name=src \
     src. ! tee name=v-t \
       v-t. ! queue ! rtph264depay ! h264parse ! ffdec_h264 ! xvimagesink \
       v-t. ! queue ! rtph264depay ! h264parse ! mp4mux ! filesink location=/tmp/video0.mp4 \
     src. ! queue ! rtpmpadepay ! mpegaudioparse ! flump3dec ! audioconvert ! alsasink

it fails with this message (a bit reformatted for clarity):
   <mp4mux0> pad video_00 refused renegotiation
         to video/x-h264, stream-format=(string)avc, alignment=(string)au, width=(int)320, height=(int)240,
           parsed=(boolean)true, codec_data=(buffer)01640028ffe1000927640028acca8141f901000528de01ae2c
         from video/x-h264, stream-format=(string)avc, alignment=(string)au, width=(int)320, height=(int)240,
           parsed=(boolean)true, codec_data=(buffer)01640028ffe1000a27640028acca8141f90001000628de01ae2c0

I can run this command on my server machine (where the camera is) and record
just fine:
   gst-launch -e mp4mux name=mux ! filesink location=/dcim/Camera/2013-01-31.14:35:25.mp4 \
     v4l2src device=/dev/video1 queue-size=16  \
         ! video/x-raw-yuv,format='(fourcc)UYVY',width=720,height=480,framerate=30/1 \
         ! ffmpegcolorspace \
         ! video/x-raw-yuv,format='(fourcc)NV12' \
         ! queue ! ducatih264enc ! h264parse ! mux.  \
     alsasrc device=hw:0,1 ! queue ! audioconvert ! ffenc_alac ! mux.

Since that pipeline works, I thought maybe the problem is that the mux wasn't
being fed both audio and video streams so I changed my original pipeline to be:
   gst-launch -e rtspsrc location=rtsp://192.168.1.136:8554/test name=src \
     mp4mux name=av-mux ! filesink location=/tmp/video0.mp4 \
     src. ! tee name=v-t \
       v-t. ! queue ! rtph264depay ! h264parse ! ffdec_h264 ! xvimagesink \
       v-t. ! queue ! rtph264depay ! h264parse ! av-mux. \
     src. ! tee name=a-t \
       a-t. ! queue ! rtpmpadepay ! mpegaudioparse ! flump3dec ! audioconvert ! alsasink \
       a-t. ! queue ! audioconvert ! ffenc_alac ! av-mux.

However this only yields an "Internal dataflow error" without any supporting information.
I turned on the debugging - the log is at http://www.mlbassoc.com/misc/gst.log

Any ideas what I'm doing wrong and how to make this process work?  Maybe I have the
tee(s) in the wrong place or similar structural errors?

Once I get this going, I'm going to want to be able to selectively pick and choose
how this pipeline gets built.  Of course I'll be creating the pipeline in Python
and I need to figure out how to add the video parts only if the source stream
has video (it can have audio, video, or both), maybe recording locally, maybe
not, etc.  Any pointers on how to be able to do that would also be appreciated.

Thanks

-- 
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Gary Thomas                 |  Consulting for the
MLB Associates              |    Embedded world
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