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• Sound in videos is full of static

  13 avril 2015, par banshee_walk_sly

  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

• Using ffmpeg to convert sound files for use in an android app

  10 janvier 2012, par stefs

  short : i'm trying to simply play a sound file converted with ffmpeg in my android app, but happen to have problems getting it to work.

  long : we have an iphone app and an android app doing the same thing, and i have to port the feature playing a sound on an user interaction. i have the source file in the aiff format, and tried to convert it to mp3 for android. but the app keeps crashing when it tries to load the file

  AssetFileDescriptor fileDescriptor = context.getResources().openRawResourceFd(resid);
  
  final MediaPlayer mp = new MediaPlayer();
  
  mp.setDataSource(fileDescriptor.getFileDescriptor(), fileDescriptor.getStartOffset(), fileDescriptor.getLength());
  
  fileDescriptor.close();
  
  mp.prepare();

  more specifically, mp.setDataSource crashes. some digging around led me to believe that something's wrong with the encoding. the sound file itself resides in res/raw.

  11-29 17:11:48.012: ERROR/SoundManager(15580): java.io.IOException: setDataSourceFD failed.: status=0x80000000
  
  11-29 17:11:48.012: ERROR/SoundManager(15580):     at android.media.MediaPlayer.setDataSource(Native Method)
  
  ...

  what i tried :

  • using a different mp3 that's already used with the same code in a different place. this works.
  • converted it to wav file. this didn't cause the app to crash, but it neither played a sound. that might be a different problem.
  • converted it to ogg ; crashed

  so, the the ffmpeg conversion parameters are as follows :

  $ ffmpeg -i click_24db.aif -f mp3 ~/foobar/wheel_click.mp3
  
  ffmpeg version 0.7.8, Copyright (c) 2000-2011 the FFmpeg developers
  
    built on Nov 24 2011 14:31:00 with gcc 4.2.1 (Apple Inc. build 5666) (dot 3)
  
    configuration: --prefix=/opt/local --enable-gpl --enable-postproc --enable-swscale --enable-avfilter --enable-libmp3lame --enable-libvorbis --enable-libtheora --enable-libdirac --enable-libschroedinger --enable-libopenjpeg --enable-libxvid --enable-libx264 --enable-libvpx --enable-libspeex --mandir=/opt/local/share/man --enable-shared --enable-pthreads --cc=/usr/bin/gcc-4.2 --arch=x86_64 --enable-yasm
  
    libavutil    50. 43. 0 / 50. 43. 0
  
    libavcodec   52.123. 0 / 52.123. 0
  
    libavformat  52.111. 0 / 52.111. 0
  
    libavdevice  52.  5. 0 / 52.  5. 0
  
    libavfilter   1. 80. 0 /  1. 80. 0
  
    libswscale    0. 14. 1 /  0. 14. 1
  
    libpostproc  51.  2. 0 / 51.  2. 0
  
  Input #0, aiff, from 'click_24db.aif':
  
    Duration: 00:00:00.01, start: 0.000000, bitrate: 1570 kb/s
  
      Stream #0.0: Audio: pcm_s16be, 44100 Hz, 2 channels, s16, 1411 kb/s
  
  Output #0, mp3, to '/Users/xyz/foobar/wheel_click.mp3':
  
    Metadata:
  
      TSSE            : Lavf52.111.0
  
      Stream #0.0: Audio: libmp3lame, 44100 Hz, 2 channels, s16, 64 kb/s
  
  Stream mapping:
  
    Stream #0.0 -> #0.0
  
  Press [q] to stop, [?] for help
  
  size=       1kB time=00:00:00.05 bitrate=  92.9kbits/s    
  
  video:0kB audio:0kB global headers:0kB muxing overhead 45.563549%

  the resulting file plays nice in itunes, does not play in vlc and crashes when loaded with the android.media.MediaPlayer (note : i first tried it with the SoundPool lib, with both mp3 and ogg, but that didn't work either).

  i also tried the following paramters, which didn't work :

  ffmpeg -i inputfile.aif -f mp3 -acodec libmp3lame -ab 192000 -ar 44100 outputfile.mp3

  i'm working on osx, built ffmpeg with macports today, android api level is 7 (google api, 2.1-update1). looking at the "supported formats" table on dev.android didn't indicate my file to be out of the spec, but i may be mistaken in that.

  i don't have the slightest clue regarding bitrates and so on, so could anybody please point me to the right combination of ffmpeg parameters to get a working mp3 for android ? i don't care if the resulting file would be mp3, ogg or 3gp or whatever.

• Extract the 0th frame of every second on a live video

  14 avril 2020, par geo-freak

  I have a command to extract the zero-th frame of every second. I got the command from here.

  



  ffmpeg -i input.ts -vf "select=between(mod(n\, 25)\, 0\, 0), setpts=N/24/TB" output-%04d.png


  



  But when I run the above command on live feed, it is extracting more than 100000 frames. The above command is not working on a live recording. Can anyone suggest or help me to extract the very first frame on a live recording ? Thanks in advance.

  



  P.S : For my testing I am running the above command on a tcr video.