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• Merge commit ’e0f8be6413b6a8d334d6052e610af32935c310af’

  5 mai 2013, par Michael Niedermayer

  Merge commit ’e0f8be6413b6a8d334d6052e610af32935c310af’
  * commit ’e0f8be6413b6a8d334d6052e610af32935c310af’ :
  
    avformat : Add AVPROBE_SCORE_EXTENSION define and use where appropriate
  Conflicts :
  
  	libavformat/ac3dec.c
  
  	libavformat/avformat.h
  
  	libavformat/avs.c
  
  	libavformat/m4vdec.c
  
  	libavformat/mov.c
  
  	libavformat/mp3dec.c
  
  	libavformat/mpeg.c
  
  	libavformat/mpegvideodec.c
  
  	libavformat/psxstr.c
  
  	libavformat/pva.c
  
  	libavformat/utils.c
  Merged-by : Michael Niedermayer <michaelni@gmx.at>
  

  • [DH] libavformat/aacdec.c
  • [DH] libavformat/ac3dec.c
  • [DH] libavformat/avformat.h
  • [DH] libavformat/avidec.c
  • [DH] libavformat/avs.c
  • [DH] libavformat/cavsvideodec.c
  • [DH] libavformat/dtsdec.c
  • [DH] libavformat/flacdec.c
  • [DH] libavformat/h261dec.c
  • [DH] libavformat/h263dec.c
  • [DH] libavformat/h264dec.c
  • [DH] libavformat/idcin.c
  • [DH] libavformat/img2dec.c
  • [DH] libavformat/m4vdec.c
  • [DH] libavformat/matroskadec.c
  • [DH] libavformat/mm.c
  • [DH] libavformat/mov.c
  • [DH] libavformat/mp3dec.c
  • [DH] libavformat/mpc8.c
  • [DH] libavformat/mpeg.c
  • [DH] libavformat/mpegvideodec.c
  • [DH] libavformat/mtv.c
  • [DH] libavformat/nsvdec.c
  • [DH] libavformat/omadec.c
  • [DH] libavformat/psxstr.c
  • [DH] libavformat/pva.c
  • [DH] libavformat/rtsp.c
  • [DH] libavformat/sierravmd.c
  • [DH] libavformat/spdifdec.c
  • [DH] libavformat/takdec.c
  • [DH] libavformat/tta.c
  • [DH] libavformat/utils.c
  • [DH] libavformat/vc1test.c
  • [DH] libavformat/vqf.c
  • [DH] libavformat/westwood_aud.c
  • [DH] libavformat/xa.c

• How can I encode and segment audio files without having gaps (or audio pops) between segments when I reconstruct it ?

  16 mai 2013, par fenduru

  I'm working on a web application that requires streaming and synchronization of multiple audio files. For this, I am using the Web Audio API over HTML5 audio tags because of the importance of timing audio.

  Currently, I'm using FFMPEG's segmentation feature to encode and segment the audio files into smaller chunks. The reason I am segmenting them is so I can start streaming from the middle of the file instead of starting from the beginning (otherwise I would've just split the files using UNIX split, as shown here. The problem is that when I string the audio segments back together, I get an audio pop between segments.

  If I encode the segments using a PCM encoding (pcm_s24le) in a .wav file, the playback is seamless, which leads me to believe that the encoder is padding either the beginning or the end of the file. Since I will be dealing with many different audio files, using .wav would require far too much bandwidth.

  I'm looking to one of the following solutions to the problem :

  • How can I segment encoded audio files seamlessly,
  • How can I force an encoder to NOT pad audio frames using ffmpeg (or another utility), or
  • What is a better way to stream audio (starting at an arbitrary track time) without using an audio tag ?

  System Information

  • Custom node.js server
  • Upon upload of an audio file, node.js pipes the data into ffmpeg's encoder
  • Need to use HTML5 Web Audio API supported encoding
  • Server sends audio chunks 1 at a time through a WebSockets socket

  Thanks in advance. I've tried to be as clear as possible but if you need clarification I'd be more than willing to provide it.

• Convert Uncompressed AVI to JPEG2000 in MXF [closed]

  15 février 2013, par Nathan

  I'm looking to convert raw, uncompressed video (currently in AVI container) into lossless motion JPEG2000 encoded video in an MXF container. I've tried FFMPEG, but don't think it can do this.

  What's the best method for transcoding on Linux ?

  FFMPEG command tried :

  ffmpeg -i Glueck_Newsreel.avi -vcodec libopenjpeg -acodec copy ms0001.mxf

  Output :

  ffmpeg version 0.10.6 Copyright (c) 2000-2012 the FFmpeg developers
  
    built on Dec 11 2012 23:58:43 with gcc 4.4.6 20110731 (Red Hat 4.4.6-3)
  
    configuration: --prefix=/usr --bindir=/usr/bin --datadir=/usr/share/ffmpeg --incdir=/usr/include/ffmpeg --libdir=/usr/lib64 --mandir=/usr/share/man --arch=x86_64 --extra-cflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m64 -mtune=generic' --enable-bzlib --disable-crystalhd --enable-gnutls --enable-libass --enable-libcdio --enable-libcelt --enable-libdc1394 --disable-indev=jack --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-openal --enable-libopenjpeg --enable-libpulse --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libv4l2 --enable-libx264 --enable-libxvid --enable-x11grab--enable-avfilter --enable-postproc --enable-pthreads --disable-static --enable-shared --enable-gpl --disable-debug --disable-stripping --shlibdir=/usr/lib64 --enable-runtime-cpudetect
  
    libavutil      51. 35.100 / 51. 35.100
  
    libavcodec     53. 61.100 / 53. 61.100
  
    libavformat    53. 32.100 / 53. 32.100
  
    libavdevice    53.  4.100 / 53.  4.100
  
   libavfilter     2. 61.100 /  2. 61.100
  
   libswscale      2.  1.100 /  2.  1.100
  
   libswresample   0.  6.100 /  0.  6.100
  
    libpostproc    52.  0.100 / 52.  0.100
  
  Input #0, avi, from 'mp0001.avi':
  
    Duration: 00:09:45.11, start: 0.000000, bitrate: 226508 kb/s
  
      Stream #0:0: Video: v210 (v210 / 0x30313276), yuv422p10le, 720x486, 29.97 fp                                                                             s, 29.97 tbr, 29.97 tbn, 29.97 tbc
  
      Stream #0:1: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, 2 channels,s16, 1536 kb/s
  
  [buffer @ 0xbed800] w:720 h:486 pixfmt:yuv422p10le tb:1/1000000 sar:0/1 sws_param:
  
  [mxf @ 0xbef9a0] track 0: could not find essence container ul, codec not currently supported in container
  
  Output #0, mxf, to 'ms0001.mxf':
  
    Metadata:
  
      encoder         : Lavf53.32.100
  
      Stream #0:0: Video: j2k, yuv422p10le, 720x486, q=2-31, 200 kb/s, 29.97 tbn, 29.97 tbc
  
      Stream #0:1: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, 2 channels, 1536 kb/s
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (v210 -> libopenjpeg)
  
    Stream #0:1 -> #0:1 (copy)
  
  Could not write header for output file #0 (incorrect codec parameters ?)