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• lavfi/loudnorm : add an internal libebur128 library

  15 octobre 2016, par Marton Balint

  lavfi/loudnorm : add an internal libebur128 library
  Also contains the following changes to the library :
  
   add ff_ prefix to functions
  
   remove cplusplus defines.
  
   add FF_ prefix to contants and some structs
  
   remove true peak calculation feature, since it uses its own resampler, and
  
    af_loudnorm does not need it.
  
   remove version info and some fprintf(stderr) functions
  
   convert to use av_malloc
  
   always use histogram mode for LRA calculation, otherwise LRA data is slowly
  
    consuming memory making af_loudnorm unfit for 24/7 operation. It also uses a
  
    BSD style linked list implementation which is probably not available on all
  
    platforms. So let’s just remove the classic mode which not uses histogram.
  
   add ff_thread_once for calculating static histogram tables
  
   convert some functions to void which cannot fail
  
   remove intrinsics and some unused headers
  
   add support for planar audio
  
   remove channel / sample rate changer function, in ffmpeg usually we simply
  
    alloc a new context
  
   convert some static variables to defines
  
   declare static histogram variables as aligned
  
   convert some initalizations to mallocz
  
   add window size parameter to init function and remove window size setter
  
    function
  
   convert return codes to AVERROR
  
   fix indentation
  Signed-off-by : Marton Balint <cus@passwd.hu>
  

  • [DH] Changelog
  • [DH] configure
  • [DH] doc/filters.texi
  • [DH] libavfilter/Makefile
  • [DH] libavfilter/af_loudnorm.c
  • [DH] libavfilter/ebur128.c
  • [DH] libavfilter/ebur128.h
  • [DH] libavfilter/version.h

• PSNR calculated with ffmpeg behaviours unexpected

  16 mars 2017, par muuh

  I’m trying to calculated the PSNR (Peak Signal to Noise Ratio) between to videos with ffmpeg.
  The basic command is : ffmpeg -i reference.avi -candidate.avi -filter_complex "psnr" out.avi.
  If I swap reference.avi and candidate.avi in the command, ffmpeg returns me a different PSNR value. The PSNR is utilizing the mean squared error between the frames, so the order should be irrelevant.

  According to this answer, this could be caused by different frame rates (or other mismatches in the video).

  The suggested code

  ffmpeg -i file1 -i file2 \
   -lavfi '[0:v]setpts=N[out0];[1:v]setpts=N[out1];[out0][out1]psnr' \
   -f null -v info -

  does not solve the issue.
  Further, in my case, candidate.avi are recoded copies of reference.avi(*). They have identical framerates and exactly same runtime. I also validated this with avprobe.

  Another possibility I think I can exclude are rounding errors. The results sometimes change from 49dB to 64dB [sic !] dependent on the order of input.

  How do I calculate the PRNU correctly with ffmpeg, and what is the reason for this behaviour ?

  (*) Recoding done for example with : ffmpeg -i reference.avi -c:v huffyuv candidate.avi

• Get waveform data from audio file using FFMPEG

  28 janvier 2017, par Lutando

  I am writing an application that needs to get the raw waveform data of an audio file so I can render it in an application (C#/.NET). I am using ffmpeg to offload this task but it looks like ffmpeg can only output the waveform data as a png or as a stream to gnuplot.

  I have looked at other libraries to do this (NAudio/CSCore) however they require windows/microsoft media foundation and since this app is going to be deployed to azure as a web app I can not use them.

  My strategy was to just read the waveform data from the png itself but this seems hacky and over the top. The ideal output would be a fix sampled series of peaks in an array where each value in the array is the peak value (ranging from 1-100 or something, like this for example).