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• How to find a "safe" point for -SS using FFMPEG to avoid breaking A/V sync ?

  16 juillet 2019, par user3144514

  I need to find a way to cut video using -SS without breaking the audio/video sync.

  Once in a while it works perfectly fine, but most of the time the audio falls slightly out of synch. Clearly there is something akin to a keyframe for audio, but I do not know how to find them.

  The problem happens when using -SS in conjunction with Any specification for a codec, including -c copy.

  For my purposes, I can’t lose quality so those are necessary.

  ffmpeg -i src.mp4 -ss 1:00 -t 30 -c copy result.mp4

  tends to break the sync.

  ffmpeg -i src.mpr -ss 1:00 -t 30 result.mp4

  produces matching a/v but with significant quality loss from using the default compressions.

  I need a way to find the times at which a cut will result in matching a/v.

• libavutil : add an FFT & MDCT implementation

  2 mai 2019, par Lynne

  libavutil : add an FFT & MDCT implementation
  This commit adds a new API to libavutil to allow for arbitrary transformations
  
  on various types of data.
  
  This is a partly new implementation, with the power of two transforms taken
  
  from libavcodec/fft_template, the 5 and 15-point FFT taken from mdct15, while
  
  the 3-point FFT was written from scratch.
  
  The (i)mdct folding code is taken from mdct15 as well, as the mdct_template
  
  code was somewhat old, messy and not easy to separate.
  A notable feature of this implementation is that it allows for 3xM and 5xM
  
  based transforms, where M is a power of two, e.g. 384, 640, 768, 1280, etc.
  
  AC-4 uses 3xM transforms while Siren uses 5xM transforms, so the code will
  
  allow for decoding of such streams.
  
  A non-exaustive list of supported sizes :
  
  4, 8, 12, 16, 20, 24, 32, 40, 48, 60, 64, 80, 96, 120, 128, 160, 192, 240,
  
  256, 320, 384, 480, 512, 640, 768, 960, 1024, 1280, 1536, 1920, 2048, 2560...
  The API was designed such that it allows for not only 1D transforms but also
  
  2D transforms of certain block sizes. This was partly on accident as the stride
  
  argument is required for Opus MDCTs, but can be used in the context of a 2D
  
  transform as well.
  
  Also, various data types would be implemented eventually as well, such as
  
  "double" and "int32_t".
  Some performance comparisons with libfftw3f (SIMD disabled for both) :
  
  120 :
  
    22353 decicycles in     fftwf_execute,     1024 runs,      0 skips
  
    21836 decicycles in compound_fft_15x8,     1024 runs,      0 skips
  128 :
  
    22003 decicycles in       fftwf_execute,   1024 runs,      0 skips
  
    23132 decicycles in monolithic_fft_ptwo,   1024 runs,      0 skips
  384 :
  
    75939 decicycles in      fftwf_execute,    1024 runs,      0 skips
  
    73973 decicycles in compound_fft_3x128,    1024 runs,      0 skips
  640 :
  
   104354 decicycles in       fftwf_execute,   1024 runs,      0 skips
  
   149518 decicycles in compound_fft_5x128,    1024 runs,      0 skips
  768 :
  
   109323 decicycles in      fftwf_execute,    1024 runs,      0 skips
  
   164096 decicycles in compound_fft_3x256,    1024 runs,      0 skips
  960 :
  
   186210 decicycles in      fftwf_execute,    1024 runs,      0 skips
  
   215256 decicycles in compound_fft_15x64,    1024 runs,      0 skips
  1024 :
  
   163464 decicycles in       fftwf_execute,   1024 runs,      0 skips
  
   199686 decicycles in monolithic_fft_ptwo,   1024 runs,      0 skips
  With SIMD we should be faster than fftw for 15xM transforms as our fft15 SIMD
  
  is around 2x faster than theirs, even if our ptwo SIMD is slightly slower.
  The goal is to remove the libavcodec/mdct15 code and deprecate the
  
  libavcodec/avfft interface once aarch64 and x86 SIMD code has been ported.
  
  New code throughout the project should use this API.
  The implementation passes fate when used in Opus, AAC and Vorbis, and the output
  
  is identical with ATRAC9 as well.
  

  • [DH] libavutil/Makefile
  • [DH] libavutil/tx.c
  • [DH] libavutil/tx.h

• Conversion of mp3 to flac results in file with longer duration

  18 juin 2019, par ThaDon

  I’ve noticed that when I convert an mp3 file to flac, the duration reported in the flac file will often differ from that of the source mp3 file. Mostly this difference is negligible and can be ignored (perhaps a fraction of a second).

  However, there are times when the timing is off by several seconds, and this causes my processing pipeline quite a bit of problem.

  For instance, take this podcast episode for example. If I run it through ffmpeg, I can see that it has a duration of :

  Duration: 00:52:38.39, start: 0.000000, bitrate: 128 kb/s

  If I then convert it to flac using the following command :

  ffmpeg -i startups-for-the-rest-of-us-448.mp3 -ac 1 -ar 16000 -f flac output.flac

  I can see that the duration of the flac file is :

  Duration: 00:52:45.65, start: 0.000000, bitrate: 133 kb/s

  Note there is an error message during conversion that states :

  [mp3 @ 0x7fffd16d6780] Header missing
  
  Error while decoding stream #0:0: Invalid data found when processing input

  Does the difference in duration have have to do with the bitrate difference ? When I listen to the file it sounds identical, I’m assuming the flac version must be ever so slightly slower as to gain the extra 7 seconds over the course of the podcast.