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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.

• fate : Add test for APV 422-10 profile

  27 avril, par Mark Thompson

  fate : Add test for APV 422-10 profile
  Bitstream generated using the reference encoder, then edited to fix the
  
  colour description and an extra metadata block added.  FFmpeg decoder
  
  output is identical to the reference decoder output.
  The content used is the first three frames of "Waterfall" from the SVT
  
  Open Content Video Test Suite 2022.  This is copyright Sveriges
  
  Television AB and is used under the Creative Commons Attribution 4.0
  
  International License.
  

  • [DH] tests/Makefile
  • [DH] tests/fate/apv.mak
  • [DH] tests/ref/fate/apv-422-10

• ffmpeg mp4 to hls transcode fails on Azure Windows Server Image [closed]

  13 février, par James Weatherhead

  I am having an issue with transcoding an mp4 (any mp4) to hls. If I run the command below on my Windows 11 machine it works fine. If I run it on a clean Windows 11 VM, it works fine. If I run it on a clean Windows 2022 Server VM it works fine. If I create a Windows 2022 server image in Azure and run it... it fails (see error below)

  


  I feel like something that FFMPEG needs is missing from the default server image in Azure. Has anyone experienced this issue and found a way to solve it ?

  


  The command I am using :

  


  ffmpeg.exe -i LowPriVMs-1.mp4 -filter_complex "[0:v]split=1[v1] ; [v1]scale=w=1920:h=1080[v1out] ; " -map "[v1out]" -c:v:0 mpeg4 -b:v:0 5000k -maxrate:v:0 5350k -bufsize:v:0 7500k -map a:0 -c:a aac -b:a:0 192k -ac 2 -f hls -hls_time 10 -hls_playlist_type vod -hls_flags independent_segments -hls_segment_type mpegts -hls_segment_filename stream_%v/data%03d.ts -master_pl_name master.m3u8 -var_stream_map "v:0,a:0" stream_%v/playlist.m3u8

  


  The error returned by FFMPEG is :

  


  [out#0/hls @ 00000204c6aded40] Could not write header (incorrect codec parameters ?) : Invalid argument

  


  Has anyone else encountered this and found a solution ? I wondered if it is required to add a Feature Pack or some such dependency to Windows ?

  


  Thanks
James