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• fate/apng : add test for apng decoding

  2 juillet 2016, par Martin Vignali

  fate/apng : add test for apng decoding
  Signed-off-by : Michael Niedermayer <michael@niedermayer.cc>
  

  • [DH] tests/Makefile
  • [DH] tests/fate/apng.mak
  • [DH] tests/ref/fate/apng-clock
  • [DH] tests/ref/fate/apng-osample

• Errors when streaming h264 video from gstreamer to ffmpeg

  14 juin 2016, par Michael Ng

  Hi I am trying to receive a udp/rtp stream with ffmpeg in the client side but is having trouble.

  Server side pipeline :

   gst-launch-1.0 -v filesrc location=video2.mp4 ! decodebin ! x264enc  !
  
   rtph264pay ! udpsink host=127.0.0.1 port=5006

  On the client side, I can play the video with the following pipeline :

  gst-launch-1.0 -e udpsrc uri=udp://0.0.0.0:5006 ! application/x-rtp, clock-rate=90000, payload=96 ! rtph264depay ! decodebin ! autovideosink

  However, since I want to convert the stream into a rtsp/http stream, I tried to receive the rtp stream with ffmpeg and perform something like :

  ffmpeg -i udp://127.0.0.1:5006 -acodec copy -vcodec copy http://localhost:8090/feed1.ffm

  But before doing that, I was testing this approach by saving the stream into a mp4 file with :

  ffmpeg -f h264 -i udp://127.0.0.1:5006 -strict -2 -f mp4 stream.mp4

  But this did not work, it gave me the following error :

  missing picture in access unit with size 15019525 [h264 @ 0x11d5100]
  no frame ! [h264 @ 0x11f06c0] decoding for stream 0 failed [h264 @
  0x11f06c0] Could not find codec parameters for stream 0 (Video : h264) :
  unspecified size Consider increasing the value for the
  ’analyzeduration’ and ’probesize’ options [h264 @ 0x11f06c0]
  Estimating duration from bitrate, this may be inaccurate
  udp ://127.0.0.1:5006 : could not find codec parameters

  Have anyone tried such approach before or experienced similar problem, I would like to get some direction on how to solve it. Thanks !

• FFMpeg Muxer Internal Buffering

  7 mars 2016, par rkrishnan2012

  I am using libFFmpeg to mux h264 and aac packets into an flv container and streaming that via rtmp.

  I want to be able to detect when frames are being sent to the server or the size of the internal buffer, in order to detect the network lag and apply variable bit-rate. However, when I clock the time it takes for av_write_frame, it is much too fast for it to be syncronous (so we can’t use this to detect network speed).

  I already tried calling av_write_frame with a null parameter (to flush data) every frame, and also directly calling avio_flush but the bitrate is still too high. So, I am wondering if there is another buffer that I am missing (such as a socket-level buffer).

  I also tried to print the AVIOContext->buf_end, buffer_size, and buf_ptr values but none of those are indicative of my network speed getting throttled (when I limit the bandwidth on my router, I would expect the buffer size to increase).

  tl ;dr - I would like to know how to properly detect packets being sent to the server, or the network speed lagging compared to my av_write_frame calls.

  If it makes a difference, this is being run on an arm-v7a Android device.