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• avcodec/prores_metadata_bsf : add smpte2084 format support

  4 novembre 2019, par Limin Wang

  avcodec/prores_metadata_bsf : add smpte2084 format support
  It's based on the following specs :
  
  RDD 36:2015 - SMPTE Registered Disclosure Doc - Apple ProRes Bitstream Syntax and Decoding Process
  Signed-off-by : Limin Wang <lance.lmwang@gmail.com>
  

  • [DH] doc/bitstream_filters.texi
  • [DH] libavcodec/prores_metadata_bsf.c

• mov : Implement parsing of the "HandlerName" from the MP4 HDLR atom

  3 février 2015, par Hendrik Leppkes

  mov : Implement parsing of the "HandlerName" from the MP4 HDLR atom
  This atom typically is used for a track title. The handler name is stored
  
  as a Pascal string in the QT specs (first byte is the length of the string),
  
  so do not export it.
  A second length check based on the first character is added to avoid
  
  overwriting an already specified handler_name (it happens with YouTube
  
  videos for instance, the handler_name get masked), or specifying an
  
  empty string metadata.
  The Pascal string fix and the second length check are written
  
  by Clément Bœsch <clement.boesch@smartjog.com>.
  Signed-off-by : Vittorio Giovara <vittorio.giovara@gmail.com>
  
  Signed-off-by : Luca Barbato <lu_zero@gentoo.org>
  

  • [DH] libavformat/mov.c

• Missing packets when transcoding using ffmpeg

  23 février 2021, par Adam Szmyd

  I have very weird issue with trans-coding opus to any other format using ffmpeg. I example it on transcoding to flac as this is what I'm currently using.

  


  So I have this one example opus file that after processing via ffmpeg kinda gets shorten like if the ffmpeg would drop some packets/data out of it.

  


  I wouldn't mind if ffmpeg was cleaning up some redundant stuff but in practice this thing is making my output file of multiple streams out of sync so at some point audio tracks gets overlapping each other.

  


  So I have this input input.opus file that's length is 00:00:05.78 and when i pass it through ffmpeg like this :

  


  $ ffmpeg -i input.opus -c flac output.flac
ffmpeg version 4.3.1 Copyright (c) 2000-2020 the FFmpeg developers
  built with gcc 10 (GCC)
  configuration: --prefix=/usr --bindir=/usr/bin --datadir=/usr/share/ffmpeg --docdir=/usr/share/doc/ffmpeg --incdir=/usr/include/ffmpeg --libdir=/usr/lib64 --mandir=/usr/share/man --arch=x86_64 --optflags='-O2 -flto=auto -ffat-lto-objects -fexceptions -g -grecord-gcc-switches -pipe -Wall -Werror=format-security -Wp,-D_FORTIFY_SOURCE=2 -Wp,-D_GLIBCXX_ASSERTIONS -specs=/usr/lib/rpm/redhat/redhat-hardened-cc1 -fstack-protector-strong -specs=/usr/lib/rpm/redhat/redhat-annobin-cc1 -m64 -mtune=generic -fasynchronous-unwind-tables -fstack-clash-protection' --extra-ldflags='-Wl,-z,relro -Wl,--as-needed -Wl,-z,now -specs=/usr/lib/rpm/redhat/redhat-hardened-ld ' --extra-cflags=' ' --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libvo-amrwbenc --enable-version3 --enable-bzlib --disable-crystalhd --enable-fontconfig --enable-frei0r --enable-gcrypt --enable-gnutls --enable-ladspa --enable-libaom --enable-libdav1d --enable-libass --enable-libbluray --enable-libcdio --enable-libdrm --enable-libjack --enable-libfreetype --enable-libfribidi --enable-libgsm --enable-liblensfun --enable-libmp3lame --enable-libmysofa --enable-nvenc --enable-openal --enable-opencl --enable-opengl --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librsvg --enable-librav1e --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libvorbis --enable-libv4l2 --enable-libvidstab --enable-libvmaf --enable-version3 --enable-vapoursynth --enable-libvpx --enable-vulkan --enable-libglslang --enable-libx264 --enable-libx265 --enable-libxvid --enable-libzimg --enable-libzvbi --enable-avfilter --enable-avresample --enable-libmodplug --enable-postproc --enable-pthreads --disable-static --enable-shared --enable-gpl --disable-debug --disable-stripping --shlibdir=/usr/lib64 --enable-lto --enable-libmfx --enable-runtime-cpudetect
  libavutil      56. 51.100 / 56. 51.100
  libavcodec     58. 91.100 / 58. 91.100
  libavformat    58. 45.100 / 58. 45.100
  libavdevice    58. 10.100 / 58. 10.100
  libavfilter     7. 85.100 /  7. 85.100
  libavresample   4.  0.  0 /  4.  0.  0
  libswscale      5.  7.100 /  5.  7.100
  libswresample   3.  7.100 /  3.  7.100
  libpostproc    55.  7.100 / 55.  7.100
Input #0, ogg, from 'input.opus':
  Duration: 00:00:05.78, start: -0.017500, bitrate: 48 kb/s
    Stream #0:0: Audio: opus, 48000 Hz, mono, fltp
    Metadata:
      DURATION        : 00:00:05.836000000
      encoder         : Lavc58.91.100 opus
Stream mapping:
  Stream #0:0 -> #0:0 (opus (native) -> flac (native))
Press [q] to stop, [?] for help
[flac @ 0x55f94c9c2f40] encoding as 24 bits-per-sample
Output #0, flac, to 'output.flac':
  Metadata:
    encoder         : Lavf58.45.100
    Stream #0:0: Audio: flac, 48000 Hz, mono, s32 (24 bit), 128 kb/s
    Metadata:
      DURATION        : 00:00:05.836000000
      encoder         : Lavc58.91.100 flac
size=     393kB time=00:00:05.79 bitrate= 555.7kbits/s speed= 373x    
video:0kB audio:385kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 2.102753%


  


  interestingly ffmpeg above shows that output should get 00:00:05.79 duration but when checking it with ffprobe, its shorter by 50ms :

  


  $ ffprobe -hide_banner output.flac
Input #0, flac, from 'output.flac':
  Metadata:
    encoder         : Lavf58.45.100
  Duration: 00:00:05.73, start: 0.000000, bitrate: 561 kb/s
    Stream #0:0: Audio: flac, 48000 Hz, mono, s32 (24 bit)


  


  It may seem silly small difference but I intentionally cut the file to be 5s long to keep troubleshooting easier. Source file is 30mins and I loose 1min out of it during that process so it is real.

  


  ffmpeg version 4.3.1

  


  How can i troubleshoot this ? When i try with -loglevel trace I noticed these few logs that looks "suspicious" :

  


  [opus @ 0x555a5abe6c40] skip 0 / discard 192 samples due to side data
[opus @ 0x555a5abe6c40] discard 192/960 samples


  


  But haven't found a way to stop it from discarding these samples (not even sure it is causing that..)

  


  I would appreciate any help or point troubleshooting direction.