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• Sound in videos is full of static

  15 octobre 2017, par Shawn Blakesley

  I’m trying to play sound from an FFMpegFrameGrabber by getting the Frame and sending the audio samples to a SourceDataLine. Here’s what I have so far :

  Creating the SourceDataLine :

  int channels = _grabber.getAudioChannels();
  
  int format = _grabber.getSampleFormat();
  
  AudioFormat fmt = new AudioFormat(_grabber.getSampleRate(), format, channels, true, true);
  
  _sourceDataLine=(SourceDataLine)AudioSystem.getLine(new DataLine.Info(SourceDataLine.class, fmt));
  
  _sourceDataLine.open(fmt);
  
  _sourceDataLine.start();

  Attempting to play sound (images are handled in the else block) :

  org.bytedeco.javacv.Frame f = _grabber.grabFrame();
  
  
  
  if (f.samples != null && f.samples.length > 0)
  
  {
  
      byte[] bytes = new byte[4096];
  
      for (Buffer buffer : f.samples)
  
      {
  
          FloatBuffer floatBuffer = (FloatBuffer) buffer;
  
          ByteBuffer byteBuffer = ByteBuffer.allocate(floatBuffer.capacity() * 4);
  
          byteBuffer.asFloatBuffer().put(floatBuffer);
  
          byteBuffer.rewind();
  
          byteBuffer.get(bytes);
  
          _sourceDataLine.write(bytes, 0, bytes.length);
  
      }
  
  }

  (Note : I tried a few different versions of this and they all have static. The versions I tried included combining the buffers into one large buffer, only trying to play one sample instead of each channel, and changing the audio format to many different permutations.)

  The problem is the sound is full of static, and almost completely unintelligible. This is my first time doing any audio programming, so I’m sure I’m doing something completely ridiculous.

  I appreciate any help. Thank you.

  EDIT

  In response to Radiodef, I tried a number of AudioFormats, and I couldn’t find one that worked for PCM_FLOAT. I found an example that used this :

  fmt = new AudioFormat(AudioFormat.Encoding.PCM_FLOAT, _grabber.getSampleRate(), format, channels, channels, _grabber.getSampleRate(), true);

  Note : I tried a few different values for the framesize from examples : channels * format / 8, channels * 8 with a hardcoded samplerate of 64, channels * 4 with a hardcoded samplerate of 32, and any combinations of those

  But it give me this exception :

  java.lang.IllegalArgumentException: No line matching interface SourceDataLine supporting format PCM_FLOAT 44100.0 Hz, 8 bit, stereo, 2 bytes/frame,  is supported.
  
      at javax.sound.sampled.AudioSystem.getLine(Unknown Source)
  
      at com.enplug.player.video.Video.<init>(Video.java:52) &lt;- where I get the SourceDataLine
  
      ...
  
  </init>

  EDIT 2

  Sorry for the delay. I appreciate all the help Radiodef.

  Here is some output from the FFMpegGrabber that is automatically output.

  Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'C:\Users\Shawn\AppData\Roaming\Enplug Display\Download\Resource\c7cb496d-96ea-4be8-a238-5ffd50955a3e.mp4':
  
    Metadata:
  
      major_brand     : qt
  
      minor_version   : 0
  
      compatible_brands: qt
  
      creation_time   : 2014-10-02 07:14:38
  
    Duration: 00:00:31.13, start: 0.000000, bitrate: 2412 kb/s
  
      Stream #0:0(und): Audio: aac (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 246 kb/s (default)
  
      Metadata:
  
        creation_time   : 2014-10-02 07:14:38
  
        handler_name    : Core Media Data Handler
  
      Stream #0:1(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p(tv, bt709), 1280x720, 2157 kb/s, 30 fps, 30 tbr, 600 tbn, 1200 tbc (default)
  
      Metadata:
  
        creation_time   : 2014-10-02 07:14:38
  
        handler_name    : Core Media Data Handler
  
        encoder         : H.264

  I have two videos I’m testing with, and the first one (which is the one in the example above) has the following :

  Bit rate: 247 kbps
  
  Channels: 2 (stereo)
  
  Audio sample rate: 44 kHz

  And the second is :

  Bit rate: 161 kbps
  
  Channels: 2 (stereo)
  
  Audio sample rate: 48 kHz

  They’re both mp4s, and I can provide any details about the video itself if needed.

  As for the library, yeah I’m pretty locked into JavaCV. We already have videos running without sound, but we’re now trying to add sound to our program.

  When I run the sample program from your JSR link I get :

  PCM_UNSIGNED unknown sample rate, 8 bit, mono, 1 bytes/frame,
  
  PCM_SIGNED unknown sample rate, 8 bit, mono, 1 bytes/frame,
  
  PCM_SIGNED unknown sample rate, 16 bit, mono, 2 bytes/frame, little-endian
  
  PCM_SIGNED unknown sample rate, 16 bit, mono, 2 bytes/frame, big-endian
  
  PCM_UNSIGNED unknown sample rate, 8 bit, stereo, 2 bytes/frame,
  
  PCM_SIGNED unknown sample rate, 8 bit, stereo, 2 bytes/frame,
  
  PCM_SIGNED unknown sample rate, 16 bit, stereo, 4 bytes/frame, little-endian
  
  PCM_SIGNED unknown sample rate, 16 bit, stereo, 4 bytes/frame, big-endian
  
  PCM_UNSIGNED unknown sample rate, 8 bit, mono, 1 bytes/frame,
  
  PCM_SIGNED unknown sample rate, 8 bit, mono, 1 bytes/frame,
  
  PCM_SIGNED unknown sample rate, 16 bit, mono, 2 bytes/frame, little-endian
  
  PCM_SIGNED unknown sample rate, 16 bit, mono, 2 bytes/frame, big-endian
  
  PCM_UNSIGNED unknown sample rate, 8 bit, stereo, 2 bytes/frame,
  
  PCM_SIGNED unknown sample rate, 8 bit, stereo, 2 bytes/frame,
  
  PCM_SIGNED unknown sample rate, 16 bit, stereo, 4 bytes/frame, little-endian
  
  PCM_SIGNED unknown sample rate, 16 bit, stereo, 4 bytes/frame, big-endian
  
  PCM_UNSIGNED unknown sample rate, 8 bit, mono, 1 bytes/frame,
  
  PCM_SIGNED unknown sample rate, 8 bit, mono, 1 bytes/frame,
  
  PCM_SIGNED unknown sample rate, 16 bit, mono, 2 bytes/frame, little-endian
  
  PCM_SIGNED unknown sample rate, 16 bit, mono, 2 bytes/frame, big-endian
  
  PCM_UNSIGNED unknown sample rate, 8 bit, stereo, 2 bytes/frame,
  
  PCM_SIGNED unknown sample rate, 8 bit, stereo, 2 bytes/frame,
  
  PCM_SIGNED unknown sample rate, 16 bit, stereo, 4 bytes/frame, little-endian
  
  PCM_SIGNED unknown sample rate, 16 bit, stereo, 4 bytes/frame, big-endian
  
  PCM_UNSIGNED unknown sample rate, 8 bit, mono, 1 bytes/frame,
  
  PCM_SIGNED unknown sample rate, 8 bit, mono, 1 bytes/frame,
  
  PCM_SIGNED unknown sample rate, 16 bit, mono, 2 bytes/frame, little-endian
  
  PCM_SIGNED unknown sample rate, 16 bit, mono, 2 bytes/frame, big-endian
  
  PCM_UNSIGNED unknown sample rate, 8 bit, stereo, 2 bytes/frame,
  
  PCM_SIGNED unknown sample rate, 8 bit, stereo, 2 bytes/frame,
  
  PCM_SIGNED unknown sample rate, 16 bit, stereo, 4 bytes/frame, little-endian
  
  PCM_SIGNED unknown sample rate, 16 bit, stereo, 4 bytes/frame, big-endian

• swscale : more accurate DITHER_COPY macro for full and limited range

  6 octobre 2017, par Mateusz

  swscale : more accurate DITHER_COPY macro for full and limited range
  Signed-off-by : Michael Niedermayer <michael@niedermayer.cc>
  

  • [DH] libswscale/swscale_unscaled.c
  • [DH] tests/ref/vsynth/vsynth1-ffvhuff420p12
  • [DH] tests/ref/vsynth/vsynth1-vc2-420p10
  • [DH] tests/ref/vsynth/vsynth1-vc2-420p12
  • [DH] tests/ref/vsynth/vsynth2-ffvhuff420p12
  • [DH] tests/ref/vsynth/vsynth2-vc2-420p10
  • [DH] tests/ref/vsynth/vsynth2-vc2-420p12
  • [DH] tests/ref/vsynth/vsynth3-ffvhuff420p12
  • [DH] tests/ref/vsynth/vsynth_lena-ffvhuff420p12
  • [DH] tests/ref/vsynth/vsynth_lena-vc2-420p10
  • [DH] tests/ref/vsynth/vsynth_lena-vc2-420p12

• FFMPEG mp3 file segmentation on Mac OSx Error . Only one full mp3 file generated as segmented file

  2 novembre 2017, par iThirst

  What I am trying to achieve is to take an MP3 file as source file and generate multiple segments of approx 10s each from it. I am using node’s fluent ffmpeg inside node js application. Below code works absolutely fine and generates proper results on Linux/Ubuntu 16.04 server while fails with below error on mac osx. It ends by generating single .mp3 file and .m3u8 playlist file on mac osx while it generates multiple segmented mp3 files (each 11s) if executed on ubuntu server.

  Error Log: 
  
  [mp3 @ 0x7fc469010c00] Application provided invalid, non monotonically increasing dts to muxer in stream 0: 9223372036854422218 >= -9223372033278215478
  
  [segment @ 0x7fc469006a00] Packets poorly interleaved, failed to avoid negative timestamp -9223372033277846838 in stream 0.

  // Node js code to execute ffmpeg command is given below

  var command = ffmpeg(fs.createReadStream(directoryPath + fileName), {timeout : 900})
  
                      .inputFormat('mp3')
  
                      .inputOptions([                    
  
                          //'-re',
  
                          //'-i input.mp3',
  
                          //directoryPath +'albumart.jpg',
  
                          '-codec copy',
  
                          '-map 0',
  
                          '-f segment',
  
                          '-segment_list ' + directoryPath + fileName_noExtension + '.m3u8',
  
                          '-segment_list_flags +live',
  
                          '-segment_time 10',
  
                           directoryPath + 'out%03d.mp3'
  
                      ]);
  
   command.on('start',{}).on('stderror').on('end').output(directoryPath +  'Out.mp3')
  
          .run()

  //

  Any help is appreciated in this direction.