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• 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

• Starnge behavior of ffmpeg while opening rtsp with .sdp

  7 avril 2014, par Pawel Rutka

  I have problem with reading rtsp stream(.sdp) over ffmpeg - is here anyone who is able to help me or have something incommon with this kind of troubles ?

  url stream :

  rtsp://192.168.1.7/moja.sdp

  Here is a log with -loglevel debug , why is there 127.0.0.1 ?

  [rtsp @ 0x9fef0a0] SDP:
  
   v=0
  
   o=- 1 1 IN IP4 127.0.0.1
  
   s=IPCam
  
   c=IN IP4 0.0.0.0
  
   t=0 0
  
   a=type:broadcast
  
   m=video 0 RTP/AVP 96
  
   a=rtpmap:96 MP4V-ES/90000
  
   a=fmtp:96 profile-level-id=1;config=000001B001000001B58913000001000000012000C488800F514043C1463F
  
   a=control:track0
  
   m=audio 0 RTP/AVP 8
  
   a=rtpmap:8 PCMA/8000
  
   a=control:track1
  
  
  
  [rtsp @ 0x9fef0a0] video codec set to: mpeg4
  
  [rtsp @ 0x9fef0a0] audio codec set to: pcm_alaw
  
  [rtsp @ 0x9fef0a0] audio samplerate set to: 8000
  
  [rtsp @ 0x9fef0a0] audio channels set to: 1
  
  [rtsp @ 0x9fef0a0] hello state=0
  
  [rtsp @ 0x9fef0a0] UDP timeout, retrying with TCP
  
  [rtsp @ 0x9fef0a0] method PAUSE failed: 501 Not Implemented

• tests/fate-run : Always overwrite output files for md5 tests

  20 septembre 2020, par Andreas Rheinhardt

  tests/fate-run : Always overwrite output files for md5 tests
  Otherwise the result of such tests will not accurately reflect the
  
  current state.
  Reviewed-by : Jan Ekström <jeebjp@gmail.com>
  
  Signed-off-by : Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
  

  • [DH] tests/fate-run.sh