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• ffmpeg : `ffmpeg -i "/video.mp4" -i "/audio.m4a" -c copy -map 0:v:0 -map 1:a:0 -shortest "/nu.mp4"` truncates, how to loop audio to match videos ? [closed]

  18 avril 2024, par Swudu Susuwu

  This is with "FFmpeg Media Encoder" from Google Store (Linux-based Android OS), but it has all the commands of ffmpeg for normal Linux.

  


  -shortest truncates the video to match the audio, and -longest has the last half of the video not have audio (for videos twice as long as audio,)

  


  what to use to loop audio (to match length of video with this) ?

  


  Video length is 15:02, so used ffmpeg -i "/audio.m4a" -c copy -map 0:a:0 "/audionew.m4a"-t 15:02 -stream_loop -1`, but got errors.

  


• WebM Video Codec in Skype 5.0 Group Video

  9 novembre 2010, par noreply@blogger.com (John Luther)

  Our friends at Skype recently released Skype 5.0 for Windows and Mac, which features a beta group video feature using VP8, the video codec in WebM. In the first half of 2010, approximately 40% of Skype-to-Skype calls were video calls and we’re excited that in the future many of those calls will be made using VP8.

  To check out the new group video feature, simply download the Skype Software.

  Note : For group video to work, everyone on the call needs the new Skype for Windows or Mac, a webcam and a broadband connection.

  For more info about Skype group video, visit the Skype website.

  

• How to write a video stream containing B-frame and no DTS to a MP4 container ?

  14 février 2020, par SteveH

  I want to save a h264 video stream received from a RTSP source to a MP4 container.
  Not like other questions asked on SO, here the challenges I face are :

  • The stream contains B frames.

  • The stream has only PTS given by the RTP/RTCP.

  Here is the code I did

  //  ffmpeg
  
      pkt->data = ..;
  
      pkt->size = ..;
  
      pkt->flags = bKeyFrame? AV_PKT_FLAG_KEY : 0;    
  
      pkt->dts = AV_NOPTS_VALUE;
  
      pkt->pts = PTS;
  
  
  
      // PTS is based on epoch microseconds so I ignored re-scaling.
  
      //av_packet_rescale_ts(pkt, { 1, AV_TIME_BASE }, muxTimebase);
  
  
  
      auto ret = av_interleaved_write_frame(m_pAVFormatCtx, pkt);

  I received a lot of error messages like this :
  "Application provided invalid, non monotonically increasing dts to muxer ...".

  Result : the mp4 file is playable via VLC but the FPS is just a half of the original FPS and the video duration is incorrect (VLC shows a weird number).

  So how do I set correct DTS and PTS before sending to the container ?

  Update :
  I have tried some changes, though not successfully yet, I found that the reason of the frame rate drop is due to the muxer discards frames having incorrect DTS.
  Additionally, if I set start of PTS and DTS value too big, some players like VLC has to delay some time before showing video.