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• Using a HLS m3u8 or DASH mpd as ffmpeg input : seeking performance

  21 septembre 2020, par coder_uk

  I wonder if any FFMPEG wizards can help with this :

  


  I've seen examples (like FFmpeg code not working on http url for thumbnail extraction) of using a http URL as the input but so far only with an .mp4.

  


  If I were to instead use an ABR .m3u8/.mpd as the input (which, though a text file, does represent a video) ... is FFMPEG smart enough to work with it ? To parse it ? So ... if I gave it a 5 hour HLS VOD m3u8 as input (-i http...), and asked it for a frame at 4 hours in (-ss), would it only download that one 10s segment at the 4-hours point ? And so only need to download a small 10 second .ts file. Or does it download the whole thing ?

  


  Thanks.

  


• Why is the audio recording on Chrome missing duration

  16 août 2019, par Ivan Sedelkin

  I’m sending some user created audio to a server which later passes it to googles speech to text api for transcription. Everything works perfectly on firefox but when I try it on Chrome it doesn’t work. I then installed FFmpeg to check if the metadata somehow changed for the file on chrome and noticed that the duration on the file is tagged "N/A". This does not happen on firefox.

  This is the audio recorded from Chrome

  Input #0, matroska,webm, from 'PATH_TO_FILE':
  
    Metadata:
  
      encoder         : Chrome
  
    Duration: N/A, start: 0.000000, bitrate: N/A
  
      Stream #0:0(eng): Audio: opus, 48000 Hz, mono, fltp (default)

  This is the audio recorded from Firefox

  Input #0, ogg, from 'PATH_TO_FILE':
  
    Duration: 00:00:01.26, start: 0.000000, bitrate: 53 kb/s
  
      Stream #0:0: Audio: opus, 48000 Hz, mono, fltp
  
      Metadata:
  
        ENCODER         : Mozilla68.0

  The audio itself is recorded using the mediarecorder-api where the blob is later converted to base64-url and sent to my server.

  This is the code that I use to record the audio :

  navigator.mediaDevices
  
      .getUserMedia(
  
        // constraints - only audio needed for this app
  
        {
  
          audio: true
  
        }
  
      )
  
  
  
      // Success callback
  
      .then(function(stream) {
  
        console.log(navigator.mediaDevices.getSupportedConstraints());
  
        var mediaRecorder = new MediaRecorder(stream, { sampleRate: 44100 });
  
        var chunks = [];
  
        $(".rec-button")
  
          .mousedown(function() {
  
            console.log("rec start");
  
            $(".rec-button i").addClass("recStart");
  
            mediaRecorder.start();
  
            console.log(mediaRecorder.state);
  
            console.log("recorder started");
  
          })
  
          .mouseup(function() {
  
            console.log("rec end");
  
            $(".rec-button i").removeClass("recStart");
  
            mediaRecorder.stop();
  
            mediaRecorder.ondataavailable = function(e) {
  
              chunks.push(e.data);
  
              var blob = new Blob(chunks, { type: "audio/ogg; codecs=opus" });
  
              var player = document.getElementById("player");
  
              player.src = URL.createObjectURL(blob);
  
              chunks = [];
  
              var reader = new window.FileReader();
  
              reader.readAsDataURL(blob);
  
              reader.onloadend = function() {
  
                var base64 = reader.result;
  
                var audioArr = {
  
                  audio: base64
  
                };
  
                $.ajax({
  
                  url: "http://localhost:4242/api/1.0/post",
  
                  type: "POST",
  
                  contentType: "application/json",
  
                  dataType: "json",
  
                  data: JSON.stringify(audioArr),
  
                  success: function(response) {
  
                    console.log(response);
  
                  },
  
                  error: function(err) {
  
                    console.log(err);
  
                  }
  
                });
  
              };
  
              console.log(mediaRecorder.state);
  
              console.log("recorder stopped");
  
            };
  
          });
  
      })
  
  
  
      // Error callback
  
      .catch(function(err) {
  
        console.log("The following getUserMedia error occured: " + err);
  
      });

  My goal is that the audio file recorded from chrome has a duration so that the google api can transcribe it. If you guys have any way of overcoming this problem I would be grateful

• Does ffmpeg pass a file through if it already conforms to the given transmux options ?

  8 janvier 2021, par anonymous coward

  Assuming I have an MP3 file with this qualities :

  


  

  • mono

  


  • 44100 sample rate

  


  • 64k constant bitrate

  


  


  Question : If I provide that file to ffmpeg as input, using the command line arguments necessary to transmux it to an output file matching the aforementioned values, which of the following should I expect from the output result ? :

  


  

  • A worse quality version of the original file, because it's be re-compressed in a lossy way.

  


  • The same file, because ffmpeg didn't need to do anything.

  


  • Something else entirely...?

  


  


  ---

  


  Extra context :

  


  I'd like to batch transmux a set of provided audio files. But there's a chance that some of those files already conform to a given standard.

  


  I'm wondering if I need to check for those maximum values before handing the file to ffmpeg (to avoid unnecessary processing) or if that would be wasteful because ffmpeg is smart enough to handle that case ?