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• How to restream multicast stream with ffmpeg

  26 octobre 2020, par verb

  I am new to ffmpeg and need to restream multicast and scale it. Tried different parameters and i have managed to restream and scale but it always appear some pat,pmt or pcr error and som interuptions in the stream appear.The input stream is cbr 14Mbit and i try to set the bitrate as 6Mbit please check my config and if you notice something wrong let me know :

  


  


  ffmpeg -re -i "udp ://@238.252.250.9:5000 ?overrun_nonfatal=1&fifo_size=1000000&bitrate=70000000&pkt_size=188" -map 0:0 -map 0:2 -b:v 3000k -minrate 3000k -maxrate 4000k -bufsize 8000K -pcr_period 20 -flush_packets 0 -tune zerolatency -preset ultrafast -threads 2 -c:a copy -qmax 12 -f mpegts -muxrate 6M "udp ://@239.253.251.13:5505 ?pkt_size=188&overrun_nonfatal=1&localaddr=10.253.251.66&bitrate=6000000"

  


  


  here is the input stream :

  


  Input #0, mpegts, from 'udp://@238.252.250.9:5000':
  Duration: N/A, start: 46612.831967, bitrate: N/A
  Program 2002 
    Metadata:
      service_name    : RT Doc HD
      service_provider: GLOBECAST
    Stream #0:0[0x7e5]: Video: h264 (High) ([27][0][0][0] / 0x001B), yuv420p(tv, bt709, top first), 1920x1080 [SAR 1:1 DAR 16:9], 25 fps, 50 tbr, 90k tbn, 50 tbc
    Stream #0:1[0x7e6](eng): Audio: ac3 (AC-3 / 0x332D4341), 48000 Hz, stereo, fltp, 192 kb/s
    Stream #0:2[0x7e7](eng): Audio: mp2 ([3][0][0][0] / 0x0003), 48000 Hz, stereo, fltp, 192 kb/s


  


  I don't understand all parameters especially the parameters concerning input/output udp stream so please help me to solve the correct command.

  


• how to send the input data to FFMPEG from a C# program

  18 octobre 2020, par jstuardo

  I need to send a binary stream to FFMPEG so that it sends to an RTMP server.

  


  I did it in a nodejs script using socket.io library and in Linux. It works perfectly.

  


  I need to do the same, but in a Windows Forms application using C#.

  


  This is how I run the ffmpeg.exe application :

  


          _currentProcess = new Process();
        _currentProcess.StartInfo.FileName = _ffmpegExe;
        _currentProcess.StartInfo.Arguments = BuildOptions(framesPerSecond, audioBitRate, audioEncoding, rtmpServer);
        _currentProcess.StartInfo.UseShellExecute = false;
        _currentProcess.StartInfo.CreateNoWindow = true;
        _currentProcess.StartInfo.RedirectStandardInput = true;
        _currentProcess.StartInfo.RedirectStandardError = true;
        _currentProcess.ErrorDataReceived += CurrentProcess_ErrorDataReceived;
        _currentProcess.Start();
        _currentProcess.BeginErrorReadLine();


  


  BuildOptions method is defined this way :

  


      private string BuildOptions(int framesPerSecond, int audioBitRate, string audioEncoding, string rtmpServer)
    {
        string options;
        if (framesPerSecond == 1)
        {
            options = $"-i - -c:v libx264 -preset ultrafast -tune zerolatency -r 1 -g 2 -keyint_min 2 -x264opts keyint=2 -crf 25 -pix_fmt yuv420p -profile:v baseline -level 3 -c:a aac -b:a {audioEncoding} -ar {audioBitRate}-f flv {rtmpServer}";
        }
        else if (framesPerSecond == 15)
        {
            options = $"-i - -c:v libx264 -preset ultrafast -tune zerolatency max_muxing_queue_size 1000 -bufsize 5000 -r 15 -g 30 -keyint_min 30 -x264opts keyint=30 -crf 25 -pix_fmt yuv420p -profile:v baseline -level 3 -c:a aac -b:a {audioEncoding} -ar {audioBitRate} -f flv {rtmpServer}";
        }
        else
        {
            options = $"-i - -c:v libx264 -preset ultrafast -tune zerolatency -c:a aac -ar {audioBitRate} -b:a {audioEncoding} -bufsize 5000 -f flv {rtmpServer}";
        }

        return options;
    }


  


  I am sending the data to the standard input this way :

  


      public void EncodeAndSend(byte[] data)
    {
        if (_currentProcess != null)
        {
            var streamWriter = _currentProcess.StandardInput;
            streamWriter.Write(Encoding.GetEncoding("ISO-8859-1").GetChars(data));
        }
    }


  


  And finally, this method is for receiving the standard error which receives the result from ffmpeg.exe :

  


      private void CurrentProcess_ErrorDataReceived(object sender, DataReceivedEventArgs e)
    {
        Console.WriteLine(e.Data);
    }


  


  When I run the application, this is shown in the console :

  


  ffmpeg version 4.3.1-2020-10-01-essentials_build-www.gyan.dev Copyright (c) 2000-2020 the FFmpeg developers
  built with gcc 10.2.0 (Rev3, Built by MSYS2 project)
  configuration: --enable-gpl --enable-version3 --enable-static --disable-w32threads --disable-autodetect --enable-fontconfig --enable-iconv --enable-gnutls --enable-libxml2 --enable-gmp --enable-lzma --enable-zlib --enable-libsrt --enable-libssh --enable-libzmq --enable-avisynth --enable-sdl2 --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxvid --enable-libaom --enable-libopenjpeg --enable-libvpx --enable-libass --enable-libfreetype --enable-libfribidi --enable-libvidstab --enable-libvmaf --enable-libzimg --enable-amf --enable-cuda-llvm --enable-cuvid --enable-ffnvcodec --enable-nvdec --enable-nvenc --enable-d3d11va --enable-dxva2 --enable-libmfx --enable-libgme --enable-libopenmpt --enable-libopencore-amrwb --enable-libmp3lame --enable-libtheora --enable-libvo-amrwbenc --enable-libgsm --enable-libopencore-amrnb --enable-libopus --enable-libspeex --enable-libvorbis --enable-librubberband
  libavutil      56. 51.100 / 56. 51.100
  libavcodec     58. 91.100 / 58. 91.100
  libavformat    58. 45.100 / 58. 45.100
  libavdevice    58. 10.100 / 58. 10.100
  libavfilter     7. 85.100 /  7. 85.100
  libswscale      5.  7.100 /  5.  7.100
  libswresample   3.  7.100 /  3.  7.100
  libpostproc    55.  7.100 / 55.  7.100
pipe:: Invalid data found when processing input


  


  If I change the EncodeAndSend method to be :

  


      public void EncodeAndSend(byte[] data)
    {
        if (_currentProcess != null)
        {
            var streamWriter = _currentProcess.StandardInput;
            streamWriter.Write(data);
        }
    }


  


  pipe:: Invalid data found when processing input error is not produced, but no more outputs are shown so it seems it is not working.

  


  What is wrong with this ? how can I send the data to the FFMPEG process ?

  


  Finally, I tell you that the binary stream comes from the camera by mean of MediaRecorder in a web page (the same used for my program in nodejs server, so that it is not the issue here)

  


• ffmpeg rtp-stream with gsm-codec

  15 octobre 2020, par Birgit

  I want to use ffmpeg for encoding and decoding gsm. I built ffmpeg with the --enable-libgsm option.

  


  I can now use the ffmpeg-command-line-tool to read gsm-encoded files, convert files to gsm, and also receive a gsm-encoded rtp stream.
So therefore I think the gsm-encoder and gsm-decoder are working properly.

  


  But for some reason I am not able to send and gsm-encoded rtp-stream.

  


  I tried the following comands :

  


  ffmpeg -re -i test.wav -c:a libgsm -f rtp rtp://127.0.0.1:5000

  


  ffmpeg -re -i test.wav -c:a gsm -f rtp rtp://127.0.0.1:5000

  


  I receive the error : Unsupported codec gsm. Could not write header for output file.

  


  I tried to use gdb to see what's going on. I think the problem is that in the file libavformat/rtpenc.c:49 gsm is not under the supported codecs. Does that mean it is not possible to use ffmpeg to create a gsm-encoded rtp-stream ? Is there a workaround, to overcome this issue ?

  


  I would appreciate any help and hints what I could try. :)