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• FFMPEG video compression php

  6 juin 2014, par Ravindra Kumar Soam

  I have compressed a video in php with the code, given below :

  $sourceFile = $filename;
  
  $targetFile    = $filename;
  
  $resolution    = '640x480';
  
  
  
  $exec_string   = "ffmpeg -i ".$sourceFile." -r 30 -s ".$resolution." ".$targetFile;
  
  exec($exec_string);

  video is compressed with desired resolution, but this video is playing properly on mobile site version but for web application it is not playing.
  

  Showing only black screen with video time duration, is somebody there we can help to fixed the same ?????

• Converting a binary stream to an mpegts stream

  22 décembre 2018, par John Kim

  I’m trying to create a livestream web app using NodeJS. The code I currently have emits a raw binary stream from the webcam on the client using socket IO and the node server receives this raw data. Using fluent-ffmpeg, I want to encode this binary stream into mpegts and send it to an RTMP server in real time, without creating any intermediary files. Could I somehow convert the binary stream into a webm stream and pipe that stream into an mpegts encoder in one ffmpeg command ?

  My relevant frontend client code :

  navigator.mediaDevices.getUserMedia(constraints).then(function(stream) {
  
      socket.emit('config_rtmpDestination',url);
  
      socket.emit('start','start');
  
      mediaRecorder = new MediaRecorder(stream);
  
      mediaRecorder.start(2000);
  
  
  
      mediaRecorder.onstop = function(e) {
  
          stream.stop();
  
      }
  
  
  
      mediaRecorder.ondataavailable = function(e) {
  
        socket.emit("binarystream",e.data);
  
      }
  
  }).catch(function(err) {
  
      console.log('The following error occured: ' + err);
  
      show_output('Local getUserMedia ERROR:'+err);
  
  });

  Relevant NodeJS server code :

  socket.on('binarystream',function(m){
  
      feedStream(m);
  
  });
  
  
  
  socket.on('start',function(m){
  
      ...
  
      var ops=[
  
          '-vcodec', socket._vcodec,'-i','-',
  
          '-c:v', 'libx264', '-preset', 'veryfast', '-tune', 'zerolatency',
  
          '-an', '-bufsize', '1000',
  
          '-f', 'mpegts', socket._rtmpDestination
  
      ];
  
      ffmpeg_process=spawn('ffmpeg', ops);
  
      feedStream=function(data){
  
          ffmpeg_process.stdin.write(data);
  
      }
  
      ...
  
  }

  The above code of course doesn’t work, I get these errors on ffmpeg :

  Error while decoding stream #0:1: Invalid data found when processing input
  
  [NULL @ 000001b15e67bd80] Invalid sync code 61f192.
  
  [libvpx @ 000001b15e6c5000] Failed to decode frame: Bitstream not supported by this decoder

  because I’m trying to convert raw binary data into mpegts.

• WebRTC : unsync audio after processing using ffmpeg (audio length is less than that of video)

  22 novembre 2013, par QuickSilver

  I am recording a video and using RecordRTC : WebRTC . After receiving the webm video and wav audio at server, I'm encoding it to a mp4 file using ffmpeg(executing shell command via php). But after encoding process, the audio is unsync with video (audio ends before video). How can I fix this ?

  I have noticed that the recorded audio is 1 sec less in length with video.

  js code is here

  record.onclick = function() {
  
      record.disabled = true;
  
      var video_constraints = {
  
          mandatory: {
  
              "minWidth": "320",
  
              "minHeight": "240",
  
              "minFrameRate": "24",
  
              "maxWidth": "320",
  
              "maxHeight": "240",
  
              "maxFrameRate": "24"
  
          },
  
          optional: []
  
      };
  
      navigator.getUserMedia({
  
          audio: true,
  
          video: video_constraints
  
      }, function(stream) {
  
          preview.src = window.URL.createObjectURL(stream);
  
          preview.play();
  
  
  
          // var legalBufferValues = [256, 512, 1024, 2048, 4096, 8192, 16384];
  
          // sample-rates in at least the range 22050 to 96000.
  
          recordAudio = RecordRTC(stream, {
  
              /* extra important, we need to set a big buffer when capturing audio and video at the same time*/
  
              bufferSize: 16384
  
              //sampleRate: 45000
  
          });
  
  
  
          recordVideo = RecordRTC(stream, {
  
              type: 'video'
  
          });
  
  
  
          recordVideo.startRecording();
  
          recordAudio.startRecording();
  
  
  
          stop.disabled = false;
  
          recording_flag = true;
  
          $("#divcounter").show();
  
          $("#second-step-title").text('Record your video');
  
          initCountdown();
  
          uploadStatus.video = false;
  
          uploadStatus.audio = false;
  
      });
  
  };

  ffmpeg command used is :

  ffmpeg -y -i 166890589.wav -i 166890589.webm -vcodec libx264 166890589.mp4

  Currently I'm adding an offset of -1 to ffmpeg, but i don't think it's right.

  ffmpeg -y -itsoffset -1 -i 166890589.wav -i 166890589.webm -vcodec libx264 166890589.mp4