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• How can I get start time of rtsp-sesson via ffmpeg (C++) ? start_time_realtime always equal -9223372036854775808

  5 août 2019, par chuchuchu

  I’m trying to get a frame by rtsp and calculate its real-world timestamp. I previously used Live555 for this (presentationTime).

  As far as I understand, ffmpeg does not provide such functionality, but provides the ability to read the relative time of each frame and the start time of the stream. In my case, the frame timestamps (pts) works correctly, but the stream start time (start_time_realtime) is always -9223372036854775808.

  I’m trying to use simple example from this Q : https://stackoverflow.com/a/11054652/5355846

  Value does not change. regardless of the position in the code

  int main(int argc, char** argv) {
  
      // Open the initial context variables that are needed
  
      SwsContext *img_convert_ctx;
  
      AVFormatContext* format_ctx = avformat_alloc_context();
  
      AVCodecContext* codec_ctx = NULL;
  
      int video_stream_index;
  
  
  
      // Register everything
  
      av_register_all();
  
      avformat_network_init();
  
  
  
      //open RTSP
  
      if (avformat_open_input(&format_ctx, "path_to_rtsp_stream",
  
                              NULL, NULL) != 0) {
  
          return EXIT_FAILURE;
  
      }
  
      ...
  
  }

  while (av_read_frame(format_ctx, &packet) >= 0 && cnt < 1000) { //read ~ 1000 frames
  
  
  
          //// here!
  
          std::cout<< " ***** "
  
          << std::to_string(format_ctx->start_time_realtime) 
  
          << " | "<start_time
  
          << " | "<best_effort_timestamp;
  
  
  
  ...
  
  }

  ***** -9223372036854775808 | 0 | 4120 | 40801 Frame : 103

  What am I doing wrong ?

• avcodec/ac3_parser : add avpriv_ac3_parse_header2() and use it in libavcodec

  1er mars 2014, par Michael Niedermayer

  avcodec/ac3_parser : add avpriv_ac3_parse_header2() and use it in libavcodec
  The new function has the ability to allocate the structure, allowing it to grow
  
  without needing major bumps
  Signed-off-by : Michael Niedermayer <michaelni@gmx.at>
  

  • [DH] libavcodec/ac3_parser.c
  • [DH] libavcodec/ac3_parser.h
  • [DH] libavcodec/ac3dec.c

• FFmpeg support for repeating last audiosample and videoframe when input stream stops or delivers corrupt data ?

  14 novembre 2017, par Vemund Kvam

  Our goal is to deliver on-the-fly packaged Live HLS and Mpeg DASH from a mpeg-ts multicast source.

  We use ffmpeg to transcode/mux the mpeg-ts input to ISMV (smooth streaming format) and transfer via http to Unified Origin running Apache.

  What we want to achieve is a constant ISMV stream to Unified Origin regardless of input stability.

  We have tried separating decoding and encoding by having two FFMPEG processes :

  1. Decodes "rawvideo" and "s16be" (audio) and sends to two python developed buffers that have the ability to repeat the last frame/sample if no input is provided.
  2. Uses the two buffer as input (over tcp) and encode/mux to ISMV.

  This solution seems to solve the stability issue, but synchronising a/v and knowing when to provide repeated frames instead of live frames seems to be impossible. We have so far assumed that ffmpeg requires a continuous input stream to produce any output if the input is not or can not be provided, output stops.

  Does FFmpeg have built in functionality for repeating a/v when the input stream either stops delivering data or delivers corrupt data ?

  Is there any industry standard way to deal with unstable input ?