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• Streaming UDP packets to two different ports (for video and audio). Video works fine, but the audio does not show

  8 avril 2018, par Winston Chen

  I am taking a rtsp stream, split the video and audio out, and stream them to two different ports respectively using gstreamer so that my ffserver would be able to display the stream on my browser.

  My gstreamer pipeline :

  gst-launch-1.0 -v rtspsrc location=rtsp://wowzaec2demo.streamlock.net/vod/mp4:BigBuckBunny_115k.mov latency=300 timeout=0 drop-on-latency=true rtp-blocksize=4096 name=rtsp_source ! \
  
    queue ! capsfilter caps="application/x-rtp,media=video" ! rtph264depay ! h264parse ! rtph264pay config-interval=1 pt=96 ! udpsink host=127.0.0.1 port=9527 rtsp_source. ! \
  
    queue ! rtpmp4apay pt=97 ! udpsink host=127.0.0.1 port=9327

  Here comes the sdp and my ffmpeg commnad :

  m=video 9527 RTP/AVP 96
  
  a=rtpmap:96 H264/90000
  
  c=IN IP4 127.0.0.1
  
  
  
  m=audio 9327 RTP/AVP 97
  
  a=rtpmap:97 mpeg4-generic/48000/6
  
  c=IN IP4 127.0.0.1

  ffmpeg -protocol_whitelist "file,tcp,rtp,udp" -i ~/test.sdp -max_muxing_queue_size 1024 http://localhost:8090/feed1.ffm

  And finally, this is my ffserver config (the important part) :

  <feed>               # This is the input feed where FFmpeg will send
  
     File ./feed1.ffm            # video stream.
  
     FileMaxSize 1GB             # Maximum file size for buffering video
  
     ACL allow 127.0.0.1         # Allowed IPs
  
  </feed>
  
  
  
  <stream>              # Output stream URL definition
  
     Feed feed1.ffm              # Feed from which to receive video
  
     Format webm
  
  
  
     # NoDefaults
  
     # NoAudio
  
  
  
     # Audio settings
  
     AudioCodec vorbis
  
     AudioBitRate 64             # Audio bitrate
  
  
  
     # Video settings
  
     VideoCodec libvpx
  
     VideoSize 240x160           # Video resolution
  
     VideoFrameRate 10           # Video FPS
  
     AVOptionVideo flags +global_header  # Parameters passed to encoder
  
                                         # (same as ffmpeg command-line parameters)
  
  
  
     PreRoll 0
  
     StartSendOnKey
  
     VideoGopSize 12
  
     VideoBitRate 256
  
  </stream>

  The thing is that if I take away the audio part and apply NoAudio, the video streams fine. However, I could not get the audio to work. Am I doing anything wrong ?

• FFMpeg/FFServer P2P streaming between embedded device and smartphone

  25 avril 2016, par Bogdan Emil Mariesan

  I’ve been playing for the last couple of days with FFMpeg and FFServer as I am considering them as candidates for livestreaming between an embedded device(with an integrated HD camera) and various clients(smartphones).

  I’ve managed to achieve the stream using with the following config for FFServer :

  HTTPPort 1234
  
  RTSPPort 1235
  
  MaxHTTPConnections 2000
  
  MaxClients 1000
  
  MaxBandwidth 512000             # Maximum bandwidth per client
  
                                 # set this high enough to exceed stream bitrate
  
  CustomLog -
  
  NoDaemon                       # Remove this if you want FFserver to daemonize after start
  
  <feed>               # This is the input feed where FFmpeg will send
  
    File /tmp/feed.ffm            # video stream.
  
    FileMaxSize 512K
  
  </feed>
  
  <stream>                      # Output stream URL definition
  
     Feed feed.ffm              # Feed from which to receive video
  
     Format rtp
  
     # Video settings
  
     VideoCodec libvpx
  
     VideoSize 720x576           # Video resolution
  
     VideoFrameRate 60           # Video FPS
  
     AVOptionVideo flags +global_header  # Parameters passed to encoder
  
                                         # (same as ffmpeg command-line parameters)
  
     AVOptionVideo cpu-used 0
  
     AVOptionVideo qmin 10
  
     AVOptionVideo qmax 42
  
     AVOptionVideo quality good
  
     AVOptionAudio flags +global_header
  
     PreRoll 15
  
     StartSendOnKey
  
     VideoBitRate 400            # Video bitrate
  
     NoAudio
  
  </stream>

  And the following FFMpeg command to send the stream to FFServer :

  ffmpeg -rtbufsize 2100M -f dshow -i video="Integrated Camera" -vcodec libx264 http://127.0.0.1:1234/feed.ffm

  I also have a simple Android client that plays the RTSP stream using the following URL :

  rtsp://mylocalnetworkip:1235/test.h264

  But now I am trying to achieve a P2P connection between the embedded device and a smartphone. This has to be over the internet(not in the LAN) and capable to achieve UDP hole punching (such as Skype does for p2p video-calling).

  1. Is this achievable with ffmpeg alone ?
  2. Can ffmpeg integrate with a Stun/Turn server such as Coturn to bypass symmetric NATs ?

• aacenc : Move local encoder specific tables to a separate file

  29 juillet 2015, par Rostislav Pehlivanov

  aacenc : Move local encoder specific tables to a separate file
  This commit moves any tables specific to the encoder from aacenc
  
  and aaccoder to a separate file called ’aacenctab.c/.h’.
  
  This was done as a clean up attempt as the encoder was filled with
  
  tables pasted in between functions which made it confusing to follow
  
  and track where each table and definition had been used.
  
  This commit solves this by simply exporting the smaller tables out to
  
  the aacenctab.h while the larger ones are compiled using aacenctab.c
  
  and are referenced from the header file.
  Signed-off-by : Claudio Freire <klaussfreire@gmail.com>
  

  • [DH] libavcodec/Makefile
  • [DH] libavcodec/aaccoder.c
  • [DH] libavcodec/aacenc.c
  • [DH] libavcodec/aacenctab.c
  • [DH] libavcodec/aacenctab.h