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• Convert audio to MP3 : 4 frames left in the queue on closing

  21 août 2024, par Vololodymyr

  I am trying to convert AMR audio to MP3 :

  


          ByteArrayInputStream amrStream = new ByteArrayInputStream(amrData);
        ByteArrayOutputStream mp3Stream = new ByteArrayOutputStream();

        try (FFmpegFrameGrabber grabber = new FFmpegFrameGrabber(amrStream);
             FFmpegFrameRecorder recorder = new FFmpegFrameRecorder(mp3Stream, grabber.getAudioChannels())) {
            
            grabber.setFormat("amr");
            grabber.start();

            recorder.setFormat("mp3");
            int sampleRate = grabber.getSampleRate();

            recorder.setSampleRate( sampleRate < MIN_SAMPLE_RATE ? MIN_SAMPLE_RATE : sampleRate);
            recorder.setAudioChannels(grabber.getAudioChannels());

            // The AMR low bitrate sounds better then same bitrate on mp3, so we increase it
            int bitrate = grabber.getAudioBitrate();
            recorder.setAudioBitrate(bitrate < MIN_BITRATE ? MIN_BITRATE : bitrate);
            recorder.start();

            Frame frame;
            while ((frame = grabber.grab()) != null) {
                if (frame.samples != null) {
                    recorder.record(frame);
                }
            }
            recorder.flush();
            recorder.release();
            grabber.release();
            recorder.stop();
            grabber.stop();
        } catch (Exception e) {
            throw new IOException("Failed to convert AMR to MP3", e);
        }


  


  For some reason, the output mp3 file is always cut for half of a second.

  


  And in logs, I can see :
[libmp3lame @ 000002106e0b26c0] 4 frames left in the queue on closing

  


  Any ideas on how I can flush those 4 frames to my output steam ?
I tried flushing with Thread.sleep(), other order of release/stop calls, nothing works.

  


• jwplayer slow buffer video lag

  8 juillet 2015, par user281738

  I’m using Jwplayer for my mobile site... So i need mobile capability html5 player.. i choose jwplayer.

  I’m Converting Video with ffmpeg

  ffmpeg -i Input.mp4 -vcodec h264 -b 1400k -s 1280X720 -acodec libfaac -ac 2 -ab 128k -movflags faststart -y Output.mp4

  I’m Using Centos Apache server and so i installed H264 Streaming Module

  My videos are starting fast at first.. but in middle when my internet slows and buffering slows down.. At the same time video lags and sounds continues for few second ... at some point my video and audio sync well. This is happening because of slow buffer problem.. After whole video buffered when i replay..full video is playing well .
  I Used Some Android Phones Chrome Browser To Check .
  I need playing position to Show load Icon and wait if buffer slows.. not lag video

  I cant lower my video bitrate.. because of my competitors.. i want solution for my html5 player.. i have added my jwplayer code.

  <code class="echappe-js">&lt;script&gt;    jwplayer(&quot;myElement&quot;).setup({<br />
  
          width: &quot;100%&quot;,<br />
  
          skin: &quot;skin.xml&quot;,<br />
  
          autostart: true,<br />
  
  <br />
  
          sharing: {},<br />
  
        image: &quot;http://example.com/Mp4/screen.jpg&quot;,<br />
  
  <br />
  
        sources: [{<br />
  
          file: &quot;http://example.com/720.mp4&quot;,<br />
  
          label: &quot;720p&quot;,<br />
  
          &quot;default&quot;: &quot;true&quot;<br />
  
        },{<br />
  
          file: &quot;http://example.com/HD/hd.mp4&quot;,<br />
  
          label: &quot;HD&quot;<br />
  
        },{<br />
  
          file: &quot;http://example.com/Mp4/480x320.mp4&quot;,<br />
  
          label: &quot;480p&quot;<br />
  
        }],<br />
  
        startparam: &quot;starttime&quot;,<br />
  
          stretching: &quot;exactfit&quot;,<br />
  
          autostart: false,<br />
  
      });<br />
  
  <br />
  
  <br />
  
  <br />
  
  &lt;/script&gt;
  
  

• FFMPEG silencedetect output does not match Audacity Sound Finder

  12 avril 2017, par tngo

  I’m currently working on labeling sound sections in an audio file.

  When I ran my audio file through Audacity Sound Finder with the following settings this is what I get

  0.448980    0.498866    1
  
  0.698413    0.927891    2
  
  1.147392    1.426757    3
  
  1.566440    1.815873    4

  These are the first 4 labels for the audio file where the sounds were found to be louder than -12dB.

  However when I ran the same audio file through FFMPEG silencedetect filter with n=-12dB:d=0.1 this is the output

  [silencedetect @ 000000000272a8c0] silence_start: -0.00712018
  
  [silencedetect @ 000000000272a8c0] silence_end: 0.371519 | silence_duration: 0.378639
  
  [silencedetect @ 000000000272a8c0] silence_start: 0.457279
  
  [silencedetect @ 000000000272a8c0] silence_end: 0.650159 | silence_duration: 0.19288
  
  [silencedetect @ 000000000272a8c0] silence_start: 0.828798
  
  [silencedetect @ 000000000272a8c0] silence_end: 1.11456 | silence_duration: 0.28576
  
  [silencedetect @ 000000000272a8c0] silence_start: 1.38608
  
  [silencedetect @ 000000000272a8c0] silence_end: 1.8576 | silence_duration: 0.1
  
  [silencedetect @ 000000000272a8c0] silence_start: 2.31488

  For FFMPEG to detect sound I’m considering silence_end to be sound_start and silence_start to be sound_end. With that logic the output from FFMPEG actually is

  0.371519 0.457279 1
  
  0.650159 0.828798 2
  
  1.11456  1.38608  3
  
  1.8576   2.31488  4

  To me it seems like all the timestamps from FFMPEG are shifted to the left compare to Audacity Sound Finder. Is there something that I’m missing ?

  Any help would be much appreciated.
  Thanks,