1. 07 Aug, 2013 1 commit
  2. 30 Apr, 2013 1 commit
    • Anton Khirnov's avatar
      avconv: make -t insert trim/atrim filters. · a83c0da5
      Anton Khirnov authored
      This makes -t sample-accurate for audio and will allow further
      simplication in the future.
      
      Most of the FATE changes are due to audio now being sample accurate. In
      some cases a video frame was incorrectly passed with the old code, while
      its was over the limit.
      a83c0da5
  3. 23 Dec, 2012 1 commit
  4. 27 Sep, 2012 1 commit
  5. 17 May, 2012 1 commit
    • Anton Khirnov's avatar
      ffmpeg: add support for audio filters. · fc49f22c
      Anton Khirnov authored
      Some of the FATE changes are due to off-by-one different rounding being used
      (lrintf vs av_rescale_q).
      Some fate changes are due to 1 audio frame less being encoded (the new variant seems
      matching what qatar does and according to ffprobe its closer to the requested duration)
      the mapchan feature sadly is lost in this commit because it depends on resampling
      being done in ffmpeg.c which is now moved completely into the av filter layer
      -async is broken after this commit, this will be fixed in subsequent commits
      the new filter reconfiguration system is flawed and will drop a frame on each
      parameter change which is why the nelly moser checksums need updating.
      
      Conflicts:
      
      	ffmpeg.c
      	tests/ref/fate/smjpeg
      fc49f22c
  6. 20 Mar, 2012 1 commit
  7. 07 Feb, 2012 1 commit
    • Anton Khirnov's avatar
      avconv: rework -t handling for encoding. · 1270e12e
      Anton Khirnov authored
      Current code compares the desired recording time with InputStream.pts,
      which has a very unclear meaning. Change the code to use actual
      timestamps of the frames passed to the encoder.
      
      In several tests, one less frame is encoded, which is more correct.
      
      In the idroq test one more frame is encoded, which is again more
      correct.
      
      Behavior with stream copy should be unchanged.
      1270e12e
  8. 12 Dec, 2011 1 commit
  9. 26 Apr, 2011 1 commit
  10. 04 Apr, 2011 1 commit
  11. 03 Apr, 2011 1 commit
  12. 23 Mar, 2011 1 commit
  13. 14 Mar, 2011 1 commit
    • Justin's avatar
      ac3enc: do not right-shift fixed-point coefficients in the final MDCT stage. · 323e6fea
      Justin authored
      This increases the accuracy of coefficients, leading to improved quality.
      Rescaling of the coefficients to full 25-bit accuracy is done rather than
      offsetting the exponent values. This requires coefficient scaling to be done
      before determining the rematrixing strategy. Also, the rematrixing strategy
      calculation must use 64-bit math to prevent overflow due to the higher
      precision coefficients.
      323e6fea
  14. 16 Feb, 2011 1 commit
  15. 15 Feb, 2011 1 commit
  16. 04 Feb, 2011 1 commit
  17. 02 Feb, 2011 1 commit
  18. 29 Dec, 2010 1 commit
    • Justin Ruggles's avatar
      Change the default dB-per-bit code from 2 to 3. · ec44dd5f
      Justin Ruggles authored
      This gives slightly better quality in PEAQ tests.
      Code 3 gives a dBpb value of 2816 = -132dB (128 psd units = -6dB), which
      corresponds to 22 bits. Since the exponents have an offset applied, the
      16-bit source looks like 24-bit source to the bit allocation routine.
      So using dBpb code=3 is a closer match to the exponent range.
      
      Regression test refs updated for acodec-ac3, lavf-rm, and seek-ac3_rm.
      
      Originally committed as revision 26144 to svn://svn.ffmpeg.org/ffmpeg/trunk
      ec44dd5f
  19. 21 Dec, 2010 1 commit
  20. 14 Dec, 2010 1 commit
  21. 02 Mar, 2010 1 commit
  22. 16 Jan, 2010 1 commit