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• Stream real-time (video+audio) via WebRTC (TCP) with chromakey && webm, best practices - how ? [on hold]

  18 octobre 2018, par Kirill K

  Please tell us about the best practics in your opinion for the case described below.
  Are there any hardware solutions for this case ?

  I want to get the real-time stream from the ip camera, overlay a chromakey, transcode into the necessary codecs (VP8 + opus), and distribute stream via webrtc over tcp to many users with some kind of authentication, for example, via a dynamic token.
  The delay from real time should be minimal.
  The solution should be stable (do not fall after 1 hour or 24 hours).

  Now I have decided on such a solution, but the time costs are too high (delay from real-time strem), maybe there you will offer be a more elegant solution :

  1. IP Camera (h264 + aac)
  2. FFmpeg (transcoding to VP8\OPUS + chromakey)
  3. FFserver (pack to rtp (for webcallserver))
  4. WebCallServer (WebRTC)

  I tried the following solutions :

  • Flussonic - missing WebRTC via TCP
  • Wowza (written in java) - crashes, support could not say the exact reasons, talked for more than 2 months, tested on different servers in different Data Centres

• Cut a video in between key frames without re-encoding the full video using ffpeg ?

  1er septembre 2020, par bguiz

  I would like to cut a video at the beginning at any particular timestamp, and it need to be precise, so the nearest key frame is not good enough.

  


  Also, these videos are rather long - an hour or longer - so I would like to avoid re-encoding this altogether if possible, or otherwise only re-encode a minimal fraction of the total duration. Thus, would like to maximise the use of -vcodec copy.

  


  How can I accomplish this using ffmpeg ?

  


  NOTE : See scenario, and my own rough idea for a possible solution below.

  


  ---

  


  Scenario :

  


  

  • Original video

    

    • Length of 1:00:00

    


    • Has a key frame every 10s

    


    


  


  • Desired cut :

    

    • From 0:01:35 through till the end

    


    


  


  • Attempt #1 :

    

    • Using -ss 0:01:35 -i blah.mp4 -vcodec copy, what results is a file where :

    


    • audio starts at 0:01:30

    


    • video also starts at 0:01:30

    


    • this starts both the audio and the video too early

    


    


  


  • using -i blah.mp4 -ss 0:01:35 -vcodec copy, what results is a file where :

    

    • audio starts at 0:01:35,

    


    • but the video is blank/ black for the first 5 seconds,

      

      • until 0:01:40, when the video starts

      


      


    


    • this starts the audio on time,
but the video starts too late

    


    


  


  


  ---

  


  Rough idea

  


  

  • (1) cut 0:01:30 to 0:01:40

    

    • re-encode this to have new key frames,
including one at the target time of 0:01:35

    


    • then cut this to get the 5 seconds from 0:01:35 through 0:01:40

    


    


  


  • (2) cut 0:01:40 through till the end

    

    • without re-encoding, using -vcodec copy

    


    


  


  • (3) ffmpeg concat the first short clip (the 5 second one)
with the second long clip

  


  


  I know/ can work out the commands for (2) and (3), but am unsure about what commands are needed for (1).

  


• How to open a remote radio stream with ffmpeg's `avformat_open_input` without segfault ?

  19 août 2020, par Keyboard embossed forhead

  I'm at the beginning stage of writing a small app to stream internet radio stations. For the moment I'm just trying to get the detected info of the input stream. Whilst I am successful in getting all the stream's details via the the command line tool (ffmpeg -i ${URL}), calling the library's avformat_open_input(...) method call results in a SEGFAULT (a stack overflow to be precise when checked in valgrind).

  


  Passing a local file url works fine though in both the command line utility and the library call.

  


  Here's a minimal example :

  


  int test() {
    const char      * station_url = "http://stream.srg-ssr.ch/m/rsc_de/aacp_96";
    const char      * test_file   = "test.mp3"; //works
    AVFormatContext * av_ctx      = avformat_alloc_context();
    int               ret         = 0;

    avformat_network_init();

    if( ( ret = avformat_open_input( &av_ctx, station_url, NULL, NULL ) ) < 0 ) { //SEGFAULT   
        printf( "Could not open file '%s': %i", station_url, ret );
        return -1;
    }

    printf( "Format %s, duration %ld us", av_ctx->iformat->long_name, av_ctx->duration );

    avformat_network_deinit();
    return 0;
}


  


  If anyone with experience in dealing with acquiring remote streams using ffmpeg libraries in C has some insights I'd be grateful. Thanks in advance.

  


  I'm using ffmpeg v4.3.1 on Linux.