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• youtube stream from ffmpeg is buffering

  1er juin 2020, par Bartonsen

  I'm using ffmpeg running on a Raspberry Pi 3B with 1GB RAM to stream live video on youtube.
In the beginning the audio+video stream is excellent, but after some minutes I see error messages in YT studio, and video starts buffering.
After some more time (could be 30 mins or 1 hr) the youtube stream is gone, although ffmpeg is still running.

  



  ffmpeg is configured like this :

  



  ./configure --arch=armel --target-os=linux --enable-gpl --enable-libx264 --enable-nonfree --enable-omx-rpi


  



  Running ffmpeg :

  



  pi@raspberrypi:~ $ ffmpeg -thread_queue_size 512 -rtsp_transport udp -i "rtsp://10.x.x.x:554/user=user&password=password&channel=1&stream=0.sdp?real_stream" -c:v copy -c:a aac -f flv rtmp://a.rtmp.youtube.com/live2/mykey
ffmpeg version git-2020-05-01-3c740f2 Copyright (c) 2000-2020 the FFmpeg developers
  built with gcc 6.3.0 (Raspbian 6.3.0-18+rpi1+deb9u1) 20170516
  configuration: --arch=armel --target-os=linux --enable-gpl --enable-libx264 --enable-nonfree --enable-omx-rpi
  libavutil      56. 43.100 / 56. 43.100
  libavcodec     58. 82.100 / 58. 82.100
  libavformat    58. 42.102 / 58. 42.102
  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
[udp @ 0x3a8c370] attempted to set receive buffer to size 393216 but it only ended up set as 327680
[udp @ 0x3a9ea80] attempted to set receive buffer to size 393216 but it only ended up set as 327680
[udp @ 0x3a8c3e0] attempted to set receive buffer to size 393216 but it only ended up set as 327680
[udp @ 0x3abf4d0] attempted to set receive buffer to size 393216 but it only ended up set as 327680
Guessed Channel Layout for Input Stream #0.1 : mono
Input #0, rtsp, from 'rtsp://10.x.x.x:554/user=user&password=password&channel=1&stream=0.sdp?real_stream':
  Metadata:
    title           : RTSP Session
  Duration: N/A, start: 0.000000, bitrate: N/A
    Stream #0:0: Video: h264 (Main), yuvj420p(pc, bt709, progressive), 1920x1080, 20 fps, 20 tbr, 90k tbn, 180k tbc
    Stream #0:1: Audio: pcm_alaw, 8000 Hz, mono, s16, 64 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (copy)
  Stream #0:1 -> #0:1 (pcm_alaw (native) -> aac (native))
Press [q] to stop, [?] for help
[aac @ 0x3ae9f00] Too many bits 8832.000000 > 6144 per frame requested, clamping to max
Output #0, flv, to 'rtmp://a.rtmp.youtube.com/live2/mykey':
  Metadata:
    title           : RTSP Session
    encoder         : Lavf58.42.102
    Stream #0:0: Video: h264 (Main) ([7][0][0][0] / 0x0007), yuvj420p(pc, bt709, progressive), 1920x1080, q=2-31, 20 fps, 20 tbr, 1k tbn, 90k tbc
    Stream #0:1: Audio: aac (LC) ([10][0][0][0] / 0x000A), 8000 Hz, mono, fltp, 48 kb/s
    Metadata:
      encoder         : Lavc58.82.100 aac


  



  In youtube studio I see this :

  



  19:36 Good transmission. The quality is excellent.
19:39 Suggestion: The current sampling rate is 0. Recommended sampling rates are 44.1 kHz and 48 kHz.
19:39 Suggestion: The current bitrate (0) of the audio stream is lower than the recommended bitrate. We recommend using a 128 Kbps bitrate for the audio stream.
19:39 Warning: The current bit rate (1974.24 kbps) is lower than the recommended bit rate. We recommend using a 4500 Kbps bitrate for the stream.
19:41 Suggestion: The current sampling rate is 0. Recommended sampling rates are 44.1 kHz and 48 kHz.
19:41 Suggestion: The current bitrate (0) of the audio stream is lower than the recommended bitrate. We recommend using a 128 Kbps bitrate for the audio stream.
19:41 Warning: The current bit rate (2151.41 kbps) is lower than the recommended bit rate. We recommend using a 4500 Kbps bitrate for the stream.
19:43 Suggestion: The current sampling rate is 0. Recommended sampling rates are 44.1 kHz and 48 kHz.
19:43 Suggestion: The current bitrate (0) of the audio stream is lower than the recommended bitrate. We recommend using a 128 Kbps bitrate for the audio stream.
19:45 Suggestion: The current sampling rate is 0. Recommended sampling rates are 44.1 kHz and 48 kHz.
19:45 Suggestion: The current bitrate (0) of the audio stream is lower than the recommended bitrate. We recommend using a 128 Kbps bitrate for the audio stream.
19:45 Warning: The current bit rate (1737.61 kbps) is lower than the recommended bit rate. We recommend using a 4500 Kbps bitrate for the stream.
...
19:54: Error: YouTube does not receive enough video to maintain consistent streaming. Viewers will therefore experience buffering.


  



  What is the problem ? How can I get rid of the buffering ?
I've also tried the below two commands, but found the output to be worse...

  



  ffmpeg -i rtsp://... -c:v libx264  -b:v 4000k -maxrate 4000k -bufsize 8000k -g 40 -preset ultrafast -vf format=yuv420p -c:a aac -f flv output
ffmpeg -i rtsp://... -c:v h264_omx -b:v 4000k -maxrate 4000k -bufsize 8000k -g 40 -preset ultrafast -vf format=yuv420p -c:a aac -f flv output


  


• Recording voice using HTML5 and processing it with ffmpeg

  22 mars 2015, par user3789242

  I need to use ffmpeg in my javascript/HTML5 project which allows the user to select the format he wants the audio to open with.I don’t know anything about ffmpeg and I’ve been doing lots of research I don’t know how to use it in my project. I found an example https://github.com/sopel39/audioconverter.js but the problem how can I install the ffmpeg.js which is 8 mg to m project. please if someone can help me I’ll be very thankfull
  here is my full code :

  the javascript page :

  // variables
  
  var leftchannel = [];
  
  var rightchannel = [];
  
  var recorder = null;
  
  var recording = false;
  
  var recordingLength = 0;
  
  var volume = null;
  
  var audioInput = null;
  
  var sampleRate = 44100;
  
  var audioContext = null;
  
  var context = null;
  
  var outputString;
  
  
  
  
  
  
  
  if (!navigator.getUserMedia)
  
  navigator.getUserMedia = navigator.getUserMedia ||
  
                           navigator.webkitGetUserMedia ||
  
                           navigator.mozGetUserMedia || 
  
                           navigator.msGetUserMedia;
  
  
  
  if (navigator.getUserMedia){
  
  navigator.getUserMedia({audio:true}, success, function(e) {
  
  alert('Error capturing audio.');
  
  });
  
  } else alert('getUserMedia not supported in this browser.');
  
  
  
  
  
  
  
  function getVal(value)
  
    {
  
  
  
  // if R is pressed, we start recording
  
  if ( value == "record"){
  
      recording = true;
  
      // reset the buffers for the new recording
  
      leftchannel.length = rightchannel.length = 0;
  
      recordingLength = 0;
  
      document.getElementById('output').innerHTML="Recording now...";
  
  
  
   // if S is pressed, we stop the recording and package the WAV file
  
   } else if ( value == "stop" ){
  
  
  
      // we stop recording
  
      recording = false;
  
      document.getElementById('output').innerHTML="Building wav file...";
  
  
  
      // we flat the left and right channels down
  
      var leftBuffer = mergeBuffers ( leftchannel, recordingLength );
  
      var rightBuffer = mergeBuffers ( rightchannel, recordingLength );
  
      // we interleave both channels together
  
      var interleaved = interleave ( leftBuffer, rightBuffer );
  
  
  
  
  
  
  
      var buffer = new ArrayBuffer(44 + interleaved.length * 2);
  
      var view = new DataView(buffer);
  
  
  
      // RIFF chunk descriptor
  
      writeUTFBytes(view, 0, 'RIFF');
  
      view.setUint32(4, 44 + interleaved.length * 2, true);
  
      writeUTFBytes(view, 8, 'WAVE');
  
      // FMT sub-chunk
  
      writeUTFBytes(view, 12, 'fmt ');
  
      view.setUint32(16, 16, true);
  
      view.setUint16(20, 1, true);
  
      // stereo (2 channels)
  
      view.setUint16(22, 2, true);
  
      view.setUint32(24, sampleRate, true);
  
      view.setUint32(28, sampleRate * 4, true);
  
      view.setUint16(32, 4, true);
  
      view.setUint16(34, 16, true);
  
      // data sub-chunk
  
      writeUTFBytes(view, 36, 'data');
  
      view.setUint32(40, interleaved.length * 2, true);
  
  
  
  
  
      var lng = interleaved.length;
  
      var index = 44;
  
      var volume = 1;
  
      for (var i = 0; i < lng; i++){
  
          view.setInt16(index, interleaved[i] * (0x7FFF * volume), true);
  
          index += 2;
  
      }
  
  
  
      var blob = new Blob ( [ view ], { type : 'audio/wav' } );
  
  
  
      // let's save it locally
  
  
  
      document.getElementById('output').innerHTML='Handing off the file now...';
  
      var url = (window.URL || window.webkitURL).createObjectURL(blob);
  
  
  
      var li = document.createElement('li');
  
      var au = document.createElement('audio');
  
      var hf = document.createElement('a');
  
  
  
      au.controls = true;
  
      au.src = url;
  
      hf.href = url;
  
      hf.download = 'audio_recording_' + new Date().getTime() + '.wav';
  
      hf.innerHTML = hf.download;
  
      li.appendChild(au);
  
      li.appendChild(hf);
  
      recordingList.appendChild(li);
  
  
  
  }
  
  }
  
  
  
  
  
  function success(e){
  
  
  
  audioContext = window.AudioContext || window.webkitAudioContext;
  
  context = new audioContext();
  
  
  
  
  
  volume = context.createGain();
  
  
  
  // creates an audio node from the microphone incoming stream(source)
  
  source = context.createMediaStreamSource(e);
  
  
  
  // connect the stream(source) to the gain node
  
  source.connect(volume);
  
  
  
  var bufferSize = 2048;
  
  
  
  recorder = context.createScriptProcessor(bufferSize, 2, 2);
  
  
  
  //node for the visualizer
  
  analyser = context.createAnalyser();
  
  analyser.smoothingTimeConstant = 0.3;
  
  analyser.fftSize = 512;
  
  
  
  splitter = context.createChannelSplitter();
  
  //when recording happens
  
  recorder.onaudioprocess = function(e){
  
  
  
      if (!recording) return;
  
      var left = e.inputBuffer.getChannelData (0);
  
      var right = e.inputBuffer.getChannelData (1);
  
  
  
      leftchannel.push (new Float32Array (left));
  
      rightchannel.push (new Float32Array (right));
  
      recordingLength += bufferSize;
  
  
  
      // get the average for the first channel
  
      var array =  new Uint8Array(analyser.frequencyBinCount);
  
      analyser.getByteFrequencyData(array);
  
  
  
      var c=document.getElementById("myCanvas");
  
      var ctx = c.getContext("2d");
  
      // clear the current state
  
      ctx.clearRect(0, 0, 1000, 325);
  
      var gradient = ctx.createLinearGradient(0,0,0,300);
  
      gradient.addColorStop(1,'#000000');
  
      gradient.addColorStop(0.75,'#ff0000');
  
      gradient.addColorStop(0.25,'#ffff00');
  
      gradient.addColorStop(0,'#ffffff');
  
      // set the fill style
  
      ctx.fillStyle=gradient;
  
      drawSpectrum(array);
  
      function drawSpectrum(array) {
  
          for ( var i = 0; i < (array.length); i++ ){
  
                  var value = array[i];
  
                  ctx.fillRect(i*5,325-value,3,325);
  
              } 
  
  
  
      }
  
  }
  
  
  
  function getAverageVolume(array) {
  
      var values = 0;
  
      var average;
  
  
  
      var length = array.length;
  
  
  
      // get all the frequency amplitudes
  
      for (var i = 0; i < length; i++) {
  
          values += array[i];
  
      }
  
  
  
      average = values / length;
  
      return average;
  
  }
  
  
  
      // we connect the recorder(node to destination(speakers))
  
      volume.connect(splitter);
  
      splitter.connect(analyser, 0, 0);
  
  
  
      analyser.connect(recorder);
  
      recorder.connect(context.destination);
  
  
  
  }
  
  
  
  
  
  
  
  
  
  function mergeBuffers(channelBuffer, recordingLength){
  
  var result = new Float32Array(recordingLength);
  
  var offset = 0;
  
  var lng = channelBuffer.length;
  
  for (var i = 0; i < lng; i++){
  
  var buffer = channelBuffer[i];
  
  result.set(buffer, offset);
  
  offset += buffer.length;
  
  }
  
      return result;
  
     }
  
  
  
  function interleave(leftChannel, rightChannel){
  
  var length = leftChannel.length + rightChannel.length;
  
  var result = new Float32Array(length);
  
  
  
  var inputIndex = 0;
  
  
  
  for (var index = 0; index < length; ){
  
   result[index++] = leftChannel[inputIndex];
  
   result[index++] = rightChannel[inputIndex];
  
   inputIndex++;
  
  }
  
  return result;
  
  }
  
  
  
  
  
  function writeUTFBytes(view, offset, string){ 
  
  var lng = string.length;
  
  for (var i = 0; i < lng; i++){
  
  
  
  view.setUint8(offset + i, string.charCodeAt(i));
  
  }
  
  }

  and here is the html code :

      
  
      
  
      
  
      <code class="echappe-js">&lt;script src=&quot;http://stackoverflow.com/feeds/tag/js/functions.js&quot;&gt;&lt;/script&gt;
      
  
      
  
  
  
  

• Revision 30148 : On pousse encore un peu plus loin la conf (toujours pas utilisable dans ...

  24 juillet 2009, par kent1@… — Log

  On pousse encore un peu plus loin la conf (toujours pas utilisable dans l’état)