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• sdp : Add an option for sending RTCP packets to the source of the last packets

  13 août 2013, par Martin Storsjö

  sdp : Add an option for sending RTCP packets to the source of the last packets
  An SDP description normally only contains the target IP address
  
  and port for the packets. This means that we don’t really have
  
  any clue where to send the RTCP RR packets - previously they’re
  
  sent to the destination IP written in the SDP (at the same port),
  
  which rarely is the actual peer. And if the source for the packets
  
  is on a different port than the destination, it’s never correct.
  With a new option, we can choose to send the packets to the
  
  address that the latest packet on each socket arrived from.
  
  — -
  
  Some may even argue that this should be the default - perhaps,
  
  but I’d rather keep it optional at first. Additionally, I’m not
  
  sure if sending RTCP RR directly back to the source is
  
  desireable for e.g. multicast.
  Signed-off-by : Martin Storsjö <martin@martin.st>
  

  • [DBH] libavformat/rtsp.c
  • [DBH] libavformat/rtsp.h

• Live555 fMaxSize and FFMPEG

  31 janvier 2015, par ALM865

  I have built a version of Live555 that uses FFMPEG to encode a video and stream it over RTSP.

  Basically it works but the RTSP stream is very jittery.

  I looked into it further and it turns out that Live555’s max buffer size (fMaxSize) is too small and Live555 is truncating the frame as shown below :

  /* This should never happen, but check anyway.. */
  
  if (newFrameSize > fMaxSize) {
  
    fFrameSize = fMaxSize;
  
    fNumTruncatedBytes = newFrameSize - fMaxSize;
  
  } else {
  
    fFrameSize = newFrameSize;
  
  }

  Now, I have almost no control over how big the packets are from FFMPEG, I can set the bitrate low but the quality is appauling and the packets are still too big !

  Basically FFMPEG decides how big each frame is here :

  int reti = avcodec_encode_video2(m_c, &pkt, m_frame, &got_packet);

  If pkt.size > fMaxSize then the frame will be truncated and Live555 will stuff up streaming the video, which is does ALL the time. Also FFMPEG sometimes decides to buffer frames so the packet could be more than one frame big.

  I can try and tell Live555 to up it’s buffer size but it ignores it completely :

  OutPacketBuffer::maxSize = 100000;

  Has anyone else got a solution to stream the encoded video correctly ? I have tried breaking the packets up and passing them to Live555 in smaller chunks but it doesn’t work, and Live555 brings down it’s fMaxSize if I send more packets.

  My code is here :

  https://dl.dropboxusercontent.com/u/15883001/Code.zip

  Some images of what is happening to the RTSP stream is here, as you can see in the higher detail images LIVE555 struggles to send the packets properly :

  https://dl.dropboxusercontent.com/u/15883001/vlcsnap-2013-12-12-09h34m30s225.zip

  In the black and white image, the frame size is 117000 bytes and is less than the max frame size 300000

  In the Iron coloured image, the frame size is 212000 bytes.

  In the rainbow coloured image, the frame size is 322000 bytes and is greater than the max frame size 300000 and is truncated resulting in what you see in the example image.

  Any help would be much appreciated

  Thanks

• FFMPEG - Streaming multiple video files to RTMP server [closed]

  17 janvier 2021, par JackCisco

  I am using Ubuntu 20.0.4

  



  What I want to achieve is using FFMPEG to stream many video files of different codecs and resolutions to my RTMP server one after the other. These files can be for example avi, mp4, mkv. All with seamless playback.

  



  So far I've managed to optimise the playback of a single video using the following command :

  



  ffmpeg -re -i video.mp4 -r 30 -preset medium -force_key_frames 'expr:gte(t,n_forced*4)' -g 120 -keyint_min 120 -acodec aac -vcodec libx264 -b:v 1M -b:a 192k -f flv rtmp://address/live/live


  



  Succeeding with that I then proceeded to attempt to use concat to read a list of files from a .txt file and play them one after the other, which looks like this :

  



  file '/pathtovideo/video1.mp4'
file '/pathtovideo/video2.avi'
file '/pathtovideo/video3.mkv'


  



  What I've ended up with below works well for the first file played, but there's a nice bit of pausing and freezing during playback before I start to see the next video, which has out of sync audio and video :

  



  ffmpeg -re -safe 0 -f h264 -f concat -segment_time_metadata 1 -i "videos.txt" -vf select=concatdec_select -af aselect=concatdec_select,aresample=async=1 -r 30 -preset medium -tune animation -force_key_frames 'expr:gte(t,n_forced*4)' -g 120 -keyint_min 120 -acodec aac -vcodec libx264 -b:v 1M -b:a 192k -f flv rtmp://address/live/live


  



  When the new video file starts, I notice the output of FFMPEG stops increasing the frames processed for about 5 seconds.

  



  Essentially the outcome is a poor transition between videos and lots of playback freezing on any proceeding video after the first.

  



  I'm a bit lost as to what to do to improve this, so any help is appreciated.