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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.

• Ffmpeg hangs while transcoding HDHomerun Prime on channel change

  23 octobre 2017, par devo1929

  I’m using ffmpeg to transcode a live stream from a HDHomerun Prime. Everything is working beautifully. However, one ability that I would like, if possible, is the ability to change the channel on the HDHomerun without having to stop and restart the ffmpeg transcoding process.

  I start the ffmpeg process to begin reading the UDP feed from the HDHomerun. It writes the stream to a series of *.ts files with a m3u8 playlist.

  The second I use hdhomerun_config to change the channel on the device, ffmpeg immediately reports the following and hangs :

  [mpegts @ 0000018b4e05be60] New video stream 0:3 at pos:295211888 and DTS:40884.7s=108 drop=0 speed=1.02x
[mpegts @ 0000018b4e05be60] New audio stream 0:5 at pos:295279568 and DTS:40884.4s
frame= 4488 fps= 29 q=23.0 q=27.0 q=23.0 size=N/A time=00:02:28.94 bitrate=N/A dup=108 drop=0 speed=0.959x

  The command I am using to launch ffmpeg is :

  ffmpeg.exe -t 03:00:00 -i "udp://192.168.1.150:5000?fifo_size=1000000&overrun_nonfatal=1" -vf yadif=0:-1:1 -y -threads 4 -c:v libx264 -s 1280x720 -r 30 -b:v 4500k -force_key_frames expr:gte(t,n_forced*2) -profile:v high -preset fast -x264opts level=41 -c:a libfdk_aac -b:a 96k -ac 2 -hls_time 10 -hls_list_size 6 -hls_wrap 6 -hls_base_url /stream/ -hls_flags temp_file -hls_playlist_type event "C:\temp\streams\4500-stream.m3u8"

  Is there a specific command that I can pass to allow ffmpeg to "recover" from this ? Or, is there an arg that prevents this hang in the first place ? I’m using the latest version v3.4 of ffmpeg cross-compiled for Windows from Ubuntu. The issue also occurred using the stable version v3.4 of ffmpeg for Windows from ffmpeg.org.

  Edit :

  A new discovery to the issue, but still not yet resolved :
  If I change to the channel BACK to the original channel, Ffmpeg is able to continue writing the stream.

  Example : I start on channel X. Ffmpeg is recording to a file. I change to channel Y. Ffmpeg outputs a message similar to the one posted above and "hangs." I change back to channel X and ffmpeg picks up where it left off, no problem.

• Live streaming doesn't work

  18 octobre 2017, par John Smith

  I’m using FFmpeg to capture my screen :

  ffmpeg -f dshow -i video="UScreenCapture" -r 5 -s 640x480 -acodec libmp3lame -ac 1 -vcodec mpeg 4 -vtag divx -q 10 -f mpegts tcp://127.0.0.1:1234

  so let it stream to somewhere. The accepter script :

   error_reporting(E_ALL); /* Allow the script to hang around waiting for connections. */
  
   set_time_limit(30); /* Turn on implicit output flushing so we see what we're getting as it comes in. */
  
   ob_implicit_flush();
  
  
  
  
  
  $address = '127.0.0.1';
  
  $port = 1234;
  
  $outfile = dirname(__FILE__)."/output.flv";
  
  $ofp = fopen($outfile, 'wb');
  
  
  
   if (($sock = socket_create(AF_INET, SOCK_STREAM, SOL_TCP)) === false) { echo "socket_create() failed: reason: " . socket_strerror(socket_last_error()) . "\n"; sleep (5); die; }
  
   if (socket_bind($sock, $address, $port) === false) { echo "socket_bind() failed: reason: " . socket_strerror(socket_last_error($sock)) . "\n"; sleep (5); die; }
  
   if (socket_listen($sock, 5) === false) { echo "socket_listen() failed: reason: " . socket_strerror(socket_last_error($sock)) . "\n"; sleep (5); die; }
  
   if (($msgsock = socket_accept($sock)) === false) { echo "socket_accept() failed: reason: " . socket_strerror(socket_last_error($sock)) . "\n"; sleep (5); break; }
  
   do {
  
      $a = '';
  
      socket_recv ($msgsock, $a, 65536, MSG_WAITALL);
  
      fwrite ($ofp, $a);
  
      //echo strlen($a)."\r\n";
  
   } while (true);

  it seems to save the stuff to the disk OK. Now here comes the html :

  I dont really know how to do this, but based on an example :

  <video src="/output.flv"></video>

  but it doesn’t do anything. And if I want to stream the live incoming stuff, then what’s the matter ?