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• Need a workaround for error in xuggler todo with vorbis

  25 septembre 2012, par Chris

  I have been at this for days, I am using xuggler, the only version i can use is 3.4 because upto 4.4 writing to outputstream directly causes jvm crash.

  I need to encode audio using vorbis, however, when i run code which works with any other audio codec I get

  2012-09-25 13:54:16,578 [Thread-4] ERROR org.ffmpeg - [libvorbis @ 0308a010] libvorbis : buffer overflow.
  2012-09-25 13:54:16,578 [Thread-4] WARN com.xuggle.xuggler - error : avcodec_encode_audio failed (../../../../../../../csrc/com/xuggle/xuggler/StreamCoder.cpp:1586)

  I think a workaround for found for this is in later versions of StreamCoder.cpp

  if (mCodecContext->codec->id == CODEC_ID_FLAC
  
            || mCodecContext->codec->id == CODEC_ID_VORBIS)
  
    {
  
      // FLAC & VORBIS audio for some reason gives an error if your output buffer isn't
  
      // over double the frame size, so we fake it here.  This could be further optimized
  
      // to only require an exact number, but this math is simpler and will always
  
      // be large enough.
  
      bufferSize = (64 + getAudioFrameSize() * (bytesPerSample + 1)) * 2;
  
    }

  But if I use the following code to make the IBuffer,

      int bufferSize = aCoder.getAudioFrameSize() * (aCoder.getBitRate());
  
      final byte[] audioBuf = new byte[bufferSize];
  
      //read audio bytes into audiobuf....
  
  
  
      buf = IBuffer.make(null, audioBuf, 0, (64 + aCoder.getAudioFrameSize() *    (aCoder.getBitRate() + 1)) * 2);
  
      System.out.println("afterbufmake");
  
      smp = IAudioSamples.make(buf,1,IAudioSamples.Format.FMT_S16);
  
              if (smp == null) {
  
                    System.out.println("smp == null");
  
                 // return;
  
              }

  smp always ends up null.

  Can anyone think of what might be happening with buffer overflow for just Vorbis and a workaround as I have only 3.4 version of Xuggler that works for what I need which is writing to outputstream.

  Thank you.

• FFMPEG Segmenting skips when m3u8 changes file

  25 septembre 2012, par user792164

  I am attempting to segment a large video file in to segments. When streamed (even locally) by opening the m3u8 file it will jump forward in time by some period of time less than 1 second.

  The following commands are executed :

  First mp4 —> ts :

  ffmpeg -i input_file.mp4 -bsf:v h264_mp4toannexb -acodec libfaac -vcodec libx264 -f mpegts -threads 0 output.ts

  Then I split using :

  ffmpeg -i output.ts -vcodec copy -acodec copy -map 0 -f segment -segment_time 30 -segment_list output.m3u8 -segment_list_type m3u8 -segment_format mpegts output%03d.ts 

  Note : Changing segment time has no effect on issue.

  Generated manifest :

  #EXTM3U
  
  #EXT-X-VERSION:4
  
  #EXTINF:30.947578,
  
  output000.ts
  
  #EXTINF:30.155111,
  
  output001.ts
  
  ...
  
  #EXTINF:24.023989,
  
  output082.ts
  
  #EXT-X-TARGETDURATION:37
  
  #EXT-X-ENDLIST

  ---

  Meta Data :

  $> ffmpeg -version
  
  ffmpeg version git-2012-08-19-a93c221
  
  built on Aug 19 2012 13:18:58 with gcc 4.4.5 (Debian 4.4.5-8)
  
  configuration: --prefix=/usr --bindir=/usr/bin --datadir=/usr/share/ffmpeg --incdir=/usr/include/ffmpeg --libdir=/usr/lib/ --mandir=/usr/share/man --extra-cflags='-O3 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions \ -fstack-protector --param=ssp-buffer-size=4 -mtune=generic' --enable-gpl --enable-shared --enable-nonfree --enable-version3 --enable-libfaac --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-librtmp --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-pthreads --enable-libxvid --enable-postproc --enable-libgsm --enable-libspeex --enable-avfilter --disable-decoder=libdirac --enable-libfreetype --enable-libschroedinger --disable-decoder=libschroedinger --enable-libopenjpeg --disable-ffplay --disable-ffserver
  
  libavutil      51. 70.100 / 51. 70.100
  
  libavcodec     54. 53.100 / 54. 53.100
  
  libavformat    54. 25.104 / 54. 25.104
  
  libavdevice    54.  2.100 / 54.  2.100
  
  libavfilter     3. 11.101 /  3. 11.101
  
  libswscale      2.  1.101 /  2.  1.101
  
  libswresample   0. 15.100 /  0. 15.100
  
  libpostproc    52.  0.100 / 52.  0.100

  -

  $> ffprobe input_file.mp4
  
  
  
  Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'input_file.mp4':
  
  Metadata:
  
  major_brand     : isom
  
  minor_version   : 1
  
  compatible_brands: isom
  
  creation_time   : 2011-09-08 11:43:25
  
  Duration: 00:41:31.00, start: 0.000000, bitrate: 1146 kb/s
  
  Stream #0.0(und): Video: h264 (High), yuv420p, 720x404 [PAR 1:1 DAR 180:101], 1015 kb/s,    23.98 fps, 23.98 tbr, 24k tbn, 47.95 tbc
  
  Metadata:
  
    creation_time   : 2011-09-08 11:43:25
  
  Stream #0.1(und): Audio: aac, 48000 Hz, stereo, s16, 124 kb/s
  
  Metadata:
  
    creation_time   : 2011-09-08 11:43:25

  -

  $> ffprobe output_file.ts
  
  Input #0, mpegts, from 'output_file.ts':
  
  Duration: 00:41:30.98, start: 1.400000, bitrate: 807 kb/s
  
  Program 1 
  
  Metadata:
  
    service_name    : Service01
  
    service_provider: FFmpeg
  
  Stream #0.0[0x100]: Video: h264 (High), yuv420p, 720x404 [PAR 1:1 DAR 180:101], 23.98 fps, 23.98 tbr, 90k tbn, 47.95 tbc
  
  Stream #0.1[0x101](und): Audio: aac, 48000 Hz, stereo, s16, 141 kb/

  ---

  Is it possible to remove this jump, if so, what encoding parameters or incorrect assumptions have I made ? Thanks.

• Decode mp4 to raw frames windows c++

  27 septembre 2012, par p.streef

  I'm trying to decode an MP4 file i've created using mp4creator library. I can't find a ffmpeg or libav example that actually works with the libraries, because the code is incompatible with the examples.

  I need a simple way of reading the mp4 file and decoding it into raw audio and video frames. In needs to work on windows too.

  If someone can point me towards a working and compatible ffmpeg/libav library + example source file that could work too.

  If i could get the library version which was used in this tutorial that would work, since this guy seems to be the only one who wrote a decent tutorial on FFMPEG (albeit on some ancient version)
  http://dranger.com/ffmpeg/

  edit :

  In the latest version of ffmpeg i found the example named : filtering_video.c
  using a patch found : http://ffmpeg.org/pipermail/ffmpeg-devel/2012-June/126587.html I have gotten the program to do "something" It looks to be working so I will investigate further.