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• how is time_base diffrent then framerate in ffmpeg ? [duplicate]

  29 juin 2019, par Gilgamesh22

  This question already has an answer here :

  • What is video timescale, timebase, or timestamp in ffmpeg ? [closed]
    
    1 answer
    

  I have a project in which I receive frames sequentially to encode a mp4-h264 video. I also receive the timestamp in milliseconds along side the frame to indicate where in the video to place the frame. Because of this I decided to have a time_base of 10000 to simplify the placement of the frame since the frame rate is variable with an average fps of 30. I later learned that this causes problems in some video players. I was wondering if there is a proper way to accomplish this. at the moment my current solution is to set the time_base to 30 and simply set the frame->pts to timestamp of the frame divide by (10000/30).

  Original Setup

      AVCodecContext *cctx;
  
      AVStream* stream;
  
  
  
      stream->time_base.num = 1;
  
      stream->time_base.den = 10000;
  
  
  
      stream->r_frame_rate.num = 30;
  
      stream->r_frame_rate.den = 1;
  
  
  
      cctx->time_base.num = 1;
  
      cctx->time_base.den = 10000;
  
  
  
      cctx->framerate.num = 30;
  
      cctx->framerate.den = 1;

  My Question

  what exactly is time_base because I always though time_base was simply used to extract location of the frame and was abstracted away during the encoding process. this is why i though you have to declare both a framerate and time_base. If it is simply a ffmpg abstraction why is a time_base with large gaps of missing frames cause this problem in only some players. does the time base and frame rate need to match

• wmv video ; packet fragment position invalid ; what does it mean and how to resolve it

  3 août 2021, par gufidown

  I am trying to do some work on a wmv video file. I'm using ffmpeg and ffprobe. I'm getting the message : "packet fragment position invalid 0,0 not in 0".

  


  I am not familiar with video processing, quite a newbie actually. I didn't find an explanation what this exactly means and no hints on how to possibly resolve that problem.

  


  I am able to play the video using VLC media player as well as other players.

  


• Muxing in audio to gstreamer RTMP stream kills both video and Audio

  1er avril 2015, par Adam

  I need some genius help here - I’m trying to set up a live stream for my upcoming wedding... and I have it ALMOST working - audio seems to be the problem.

  This is my setup

  • Raspberry Pi Model B+
  • Logitech C920 (with onboard h264 encoding that I am utilising)
  • on-camera (C920) microphone
  • USB wifi to iPhone 4G connection
  • gstreamer1.0
  • Amazon EC2 Wowza RTMP server

  I have it all set up, but as soon as I mux in the audio, the streams wont play by any player.

  What Works :
  - my gstreamer pipeline WITHOUT the audio muxed in
  - Wowza receives a consistent stream, no failures
  - The various Flash players / iOS / Android and VLC all play back the video

  What doesnt :
  - enabling audio in the mux (using the pipeline below)
  - BUT gstreamer doesnt complain
  - BUT Wowza receives a consistent stream, no failures
  - The various flash players fail to play both Audio and Video. some just display the first video frame
  - VLC plays 1 video frame, and about 100ms of audio, then stops

  Ideally I’d like the muxed audio/video FLV stored on the SD card too in case the network goes down - but if the ’tee’ needs to be sacrificed to make it work, so be it.

  This is my current FAILING pipeline - I assume there’s something really stupid in it because I know practically nothing about gstreamer.... The first frame loads in all the players (except iOS.. which never shows anything)

  # set camera resolution to 720p, and the data format to H264 (alternatives are YUV and JPG)
  
  v4l2-ctl --device=/dev/video0 --set-fmt-video=width=1280,height=720,pixelformat=1
  
  # set the frame rate
  
  v4l2-ctl --device=/dev/video0 --set-parm=10
  
  
  
  gst-launch-1.0 -v -e uvch264src initial-bitrate=300000 average-bitrate=300000 device=/dev/video0 name=src auto-start=true src.vidsrc \
  
                  ! queue \
  
                  ! video/x-h264,width=1280,height=720,framerate=10/1 \
  
                  ! h264parse \
  
                  ! flvmux streamable=true name=mux \
  
                  ! queue \
  
                  ! tee name=t \
  
                  ! queue \
  
                  ! filesink location=/home/pi/wedding.flv t. \
  
                  ! queue \
  
                  ! rtmpsink location='rtmp://wowzaserver/live/wedding live=1' >>/home/pi/wedding.log 2>&1

  Some of the things I can’t really afford to change at this late stage are the encapsulation (FLV) and wowza RTMP because I’ve built everything around that...

  Please Help !! Thanks !

  UPDATE

  Given that I am also saving the FLV file, I have found that if I use ffmpeg to send that FLV file (using audio copy, video copy) to the RTMP server, everything works (but obviously its not live) ! So I am now starting to believe this is a problem with the way Gstreamer encapsulates RTMP - and by putting ffmpeg in the middle it fixes it... but it’s not live of course.
  Is it possible to pipe my output to ffmpeg and using ffmpeg’s RTMP ?