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• Live audio using ffmpeg, javascript and nodejs

  8 novembre 2017, par klaus

  I am new to this thing. Please don’t hang me for the poor grammar. I am trying to create a proof of concept application which I will later extend. It does the following : We have a html page which asks for permission to use the microphone. We capture the microphone input and send it via websocket to a node js app.

  JS (Client) :

  var bufferSize = 4096;
  
  var socket = new WebSocket(URL);
  
  var myPCMProcessingNode = context.createScriptProcessor(bufferSize, 1, 1);
  
  myPCMProcessingNode.onaudioprocess = function(e) {
  
    var input = e.inputBuffer.getChannelData(0);
  
    socket.send(convertFloat32ToInt16(input));
  
  }
  
  
  
  function convertFloat32ToInt16(buffer) {
  
    l = buffer.length;
  
    buf = new Int16Array(l);
  
    while (l--) {
  
      buf[l] = Math.min(1, buffer[l])*0x7FFF;
  
    }
  
    return buf.buffer;
  
  }
  
  
  
  navigator.mediaDevices.getUserMedia({audio:true, video:false})
  
                                  .then(function(stream){
  
                                    var microphone = context.createMediaStreamSource(stream);
  
                                    microphone.connect(myPCMProcessingNode);
  
                                    myPCMProcessingNode.connect(context.destination);
  
                                  })
  
                                  .catch(function(e){});

  In the server we take each incoming buffer, run it through ffmpeg, and send what comes out of the std out to another device using the node js ’http’ POST. The device has a speaker. We are basically trying to create a 1 way audio link from the browser to the device.

  Node JS (Server) :

  var WebSocketServer = require('websocket').server;
  
  var http = require('http');
  
  var children = require('child_process');
  
  
  
  wsServer.on('request', function(request) {
  
    var connection = request.accept(null, request.origin);
  
    connection.on('message', function(message) {
  
      if (message.type === 'utf8') { /*NOP*/ }
  
      else if (message.type === 'binary') {
  
        ffm.stdin.write(message.binaryData);
  
      }
  
    });
  
    connection.on('close', function(reasonCode, description) {});
  
    connection.on('error', function(error) {});
  
  });
  
  
  
  var ffm = children.spawn(
  
      './ffmpeg.exe'
  
     ,'-stdin -f s16le -ar 48k -ac 2 -i pipe:0 -acodec pcm_u8 -ar 48000 -f aiff pipe:1'.split(' ')
  
  );
  
  
  
  ffm.on('exit',function(code,signal){});
  
  
  
  ffm.stdout.on('data', (data) => {
  
    req.write(data);
  
  });
  
  
  
  var options = {
  
    host: 'xxx.xxx.xxx.xxx',
  
    port: xxxx,
  
    path: '/path/to/service/on/device',
  
    method: 'POST',
  
    headers: {
  
     'Content-Type': 'application/octet-stream',
  
     'Content-Length': 0,
  
     'Authorization' : 'xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx',
  
     'Transfer-Encoding' : 'chunked',
  
     'Connection': 'keep-alive'
  
    }
  
  };
  
  
  
  var req = http.request(options, function(res) {});

  The device supports only continuous POST and only a couple of formats (ulaw, aiff, wav)

  This solution doesn’t seem to work. In the device speaker we only hear something like white noise.

  Also, I think I may have a problem with the buffer I am sending to the ffmpeg std in -> Tried to dump whatever comes out of the websocket to a .wav file then play it with VLC -> it plays everything in the record very fast -> 10 seconds of recording played in about 1 second.

  I am new to audio processing and have searched for about 3 days now for solutions on how to improve this and found nothing.

  I would ask from the community for 2 things :

  1. Is something wrong with my approach ? What more can I do to make this work ? I will post more details if required.

  2. If what I am doing is reinventing the wheel then I would like to know what other software / 3rd party service (like amazon or whatever) can accomplish the same thing.

  Thank you.

• Audio out of sync, direct capture device stream (Windows 10)

  3 mai 2020, par user3459555

  Using ffplay, the video stays in sync using this command :

  



  ffplay -f dshow -rtbufsize 702000k video="Cam Link"
ffplay version git-2020-05-01-39fb1e9 Copyright (c) 2003-2020 the FFmpeg developers
  built with gcc 9.3.1 (GCC) 20200328
  configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-fontconfig --enable-gnutls --enable-iconv --enable-libass --enable-libdav1d --enable-libbluray --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libsrt --enable-libtheora --enable-libtwolame --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libzimg --enable-lzma --enable-zlib --enable-gmp --enable-libvidstab --enable-libvmaf --enable-libvorbis --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex --enable-libxvid --enable-libaom --disable-w32threads --enable-libmfx --enable-ffnvcodec --enable-cuda-llvm --enable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth --enable-libopenmpt --enable-amf
  libavutil      56. 43.100 / 56. 43.100
  libavcodec     58. 82.100 / 58. 82.100
  libavformat    58. 42.101 / 58. 42.101
  libavdevice    58.  9.103 / 58.  9.103
  libavfilter     7. 80.100 /  7. 80.100
  libswscale      5.  6.101 /  5.  6.101
  libswresample   3.  6.100 /  3.  6.100
  libpostproc    55.  6.100 / 55.  6.100
Input #0, dshow, from 'video=Cam Link':vq=    0KB sq=    0B f=0/0
  Duration: N/A, start: 141954.961000, bitrate: N/A
    Stream #0:0: Video: rawvideo (YUY2 / 0x32595559), yuyv422, 1920x1080, 59.94 fps, 59.94 tbr, 10000k tbn, 10000k tbc
142992.53 M-V: -0.001 fd=   3 aq=    0KB vq=    0KB sq=    0B f=0/0


  



  Every controller button press stays in sync.

  



  The audio however :

  



  ffplay -f dshow audio="Digital Audio Interface (Cam Link)" -tune zerolatency
ffplay version git-2020-05-01-39fb1e9 Copyright (c) 2003-2020 the FFmpeg developers
  built with gcc 9.3.1 (GCC) 20200328
  configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-fontconfig --enable-gnutls --enable-iconv --enable-libass --enable-libdav1d --enable-libbluray --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libsrt --enable-libtheora --enable-libtwolame --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libzimg --enable-lzma --enable-zlib --enable-gmp --enable-libvidstab --enable-libvmaf --enable-libvorbis --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex --enable-libxvid --enable-libaom --disable-w32threads --enable-libmfx --enable-ffnvcodec --enable-cuda-llvm --enable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth --enable-libopenmpt --enable-amf
  libavutil      56. 43.100 / 56. 43.100
  libavcodec     58. 82.100 / 58. 82.100
  libavformat    58. 42.101 / 58. 42.101
  libavdevice    58.  9.103 / 58.  9.103
  libavfilter     7. 80.100 /  7. 80.100
  libswscale      5.  6.101 /  5.  6.101
  libswresample   3.  6.100 /  3.  6.100
  libpostproc    55.  6.100 / 55.  6.100
Input #0, dshow, from 'audio=Digital Audio Interface (Cam Link)':
  Duration: N/A, start: 143092.007000, bitrate: 1411 kb/s
    Stream #0:0: Audio: pcm_s16le, 44100 Hz, 2 channels, s16, 1411 kb/s
143103.21 M-A:  0.000 fd=   0 aq=    0KB vq=    0KB sq=    0B f=0/0


  



  Is always behined by about a full second.

  



  I'm not trying to record this, just trying to directly play from the Elgato Cam Link 1:1 output to my computer screen. When this is played in the Elgato Game Capture software, the video and audio are 1:1, no issues. So I know it's not the console or the capture device.

  


• USB webcam streaming in ARM board (i.MX6)

  5 septembre 2017, par Titus

  I want to streaming the camera via NETWORK. I have connected the USB webcam to i.MX6 board and want to stream in Ubuntu14.04/16.04 via network.

  Incidentally, I have installed the gstreamer and ffmpeg tools for this in i.MX6 board.

  Also I am able to stream USB webcam within Ubuntu14.04 PC using the following ffmpeg commands. But it’s also not working if I use ffplay in other Ubuntu16.04 and I am not sure why (both are same ffmpeg versions).

  ffmpeg -f v4l2 -i /dev/video0 -preset ultrafast -vcodec libx264 -tune zerolatency -b 900k -f mpegts udp://192.168.0.37:1234
  
  
  
  ffplay udp://192.168.0.37:1234

  Ubuntu 16.06 : (NOT WORKING)

  tus@titus-PC:~/workdir$ ffplay udp://192.168.0.105:1234
  
  ffplay version 2.8.11-0ubuntu0.16.04.1 Copyright (c) 2003-2017 the FFmpeg developers
  
    built with gcc 5.4.0 (Ubuntu 5.4.0-6ubuntu1~16.04.4) 20160609
  
    configuration: --prefix=/usr --extra-version=0ubuntu0.16.04.1 --build-suffix=-ffmpeg --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --cc=cc --cxx=g++ --enable-gpl --enable-shared --disable-stripping --disable-decoder=libopenjpeg --disable-decoder=libschroedinger --enable-avresample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libmodplug --enable-libmp3lame --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-librtmp --enable-libschroedinger --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxvid --enable-libzvbi --enable-openal --enable-opengl --enable-x11grab --enable-libdc1394 --enable-libiec61883 --enable-libzmq --enable-frei0r --enable-libx264 --enable-libopencv
  
    libavutil      54. 31.100 / 54. 31.100
  
    libavcodec     56. 60.100 / 56. 60.100
  
    libavformat    56. 40.101 / 56. 40.101
  
    libavdevice    56.  4.100 / 56.  4.100
  
    libavfilter     5. 40.101 /  5. 40.101
  
    libavresample   2.  1.  0 /  2.  1.  0
  
    libswscale      3.  1.101 /  3.  1.101
  
    libswresample   1.  2.101 /  1.  2.101
  
    libpostproc    53.  3.100 / 53.  3.100
  
      nan    :  0.000 fd=   0 aq=    0KB vq=    0KB sq=    0B f=0/0  

  Ubuntu 14.04 : (WORKING)

  titus@titus-laptop:~$ 
  
  titus@titus-laptop:~$ ffplay udp://127.0.0.1:1234
  
  ffplay version 3.3.2 Copyright (c) 2003-2017 the FFmpeg developers
  
    built with gcc 4.8 (Ubuntu 4.8.4-2ubuntu1~14.04.3)
  
    configuration: --extra-libs=-ldl --prefix=/opt/ffmpeg --mandir=/usr/share/man --enable-avresample --disable-debug --enable-nonfree --enable-gpl --enable-version3 --enable-libopencore-amrnb --enable-libopencore-amrwb --disable-decoder=amrnb --disable-decoder=amrwb --enable-libpulse --enable-libfreetype --enable-gnutls --enable-libx264 --enable-libx265 --enable-libfdk-aac --enable-libvorbis --enable-libtheora --enable-libmp3lame --enable-libopus --enable-libvpx --enable-libspeex --enable-libass --enable-avisynth --enable-libsoxr --enable-libxvid --enable-libvidstab --enable-libwavpack --enable-nvenc
  
    libavutil      55. 58.100 / 55. 58.100
  
    libavcodec     57. 89.100 / 57. 89.100
  
    libavformat    57. 71.100 / 57. 71.100
  
    libavdevice    57.  6.100 / 57.  6.100
  
    libavfilter     6. 82.100 /  6. 82.100
  
    libavresample   3.  5.  0 /  3.  5.  0
  
    libswscale      4.  6.100 /  4.  6.100
  
    libswresample   2.  7.100 /  2.  7.100
  
    libpostproc    54.  5.100 / 54.  5.100
  
  [h264 @ 0xb0621660] non-existing PPS 0 referenced sq=    0B f=0/0   
  
      Last message repeated 1 times
  
  [h264 @ 0xb0621660] decode_slice_header error
  
  [h264 @ 0xb0621660] non-existing PPS 0 referenced
  
  [h264 @ 0xb0621660] decode_slice_header error
  
  [h264 @ 0xb0621660] non-existing PPS 0 referenced
  
  [h264 @ 0xb0621660] decode_slice_header error
  
  [h264 @ 0xb0621660] non-existing PPS 0 referenced
  
  [h264 @ 0xb0621660] decode_slice_header error
  
  [h264 @ 0xb0621660] no frame!
  
  [h264 @ 0xb0621660] non-existing PPS 0 referenced sq=    0B f=0/0   
  
  1751.47 M-V: -0.021 fd=   0 aq=    0KB vq=   11KB sq=    0B f=0/0   
  
  1751.63 M-V: -0.020 fd=   0 aq=    0KB vq=   11KB sq=    0B f=0/0   
  
  1751.80 M-V: -0.020 fd=   0 aq=    0KB vq=   11KB sq=    0B f=0/0 

  Finally I want to stream with different ARM boards. Am also not able to build ffplay command. Same issue with raspberry pi too. I am doing something wrong or misunderstood something here ?