1. 03 Dec, 2013 1 commit
  2. 23 Aug, 2013 1 commit
  3. 04 Jan, 2013 1 commit
    • Alexander Strasser's avatar
      lswr: Improve default resampler's default parameters · ac25b31e
      Alexander Strasser authored
      After making some blind tests on a small collection of music
      samples for home usage. It turned out that the default cutoff
      was too low.
      
      The impact of filter_size was not clearly distinguishable (the
      results were on the edge) with the music samples but turned out
      to be clearly audible in some synthetic samples.
      
      Thanks to Daniel for helping out with the listening tests.
      Signed-off-by: 's avatarAlexander Strasser <eclipse7@gmx.net>
      ac25b31e
  4. 20 Dec, 2012 1 commit
  5. 02 Dec, 2012 1 commit
  6. 29 May, 2012 1 commit
  7. 18 May, 2012 1 commit
  8. 17 May, 2012 3 commits
    • Michael Niedermayer's avatar
      aresample: add code to flush the internal swr buffer. · 847943bc
      Michael Niedermayer authored
      Inspired-by code from af_resample.c written by Anton Khirnov
      Signed-off-by: 's avatarMichael Niedermayer <michaelni@gmx.at>
      847943bc
    • Mans Rullgard's avatar
      pcmenc: set correct bitrate value · 7d7b40f4
      Mans Rullgard authored
      This fixes a bogus bitrate value in the header of WAV files with
      alaw/ulaw audio.
      Signed-off-by: 's avatarMans Rullgard <mans@mansr.com>
      7d7b40f4
    • 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
  9. 20 Apr, 2012 3 commits
  10. 10 Apr, 2012 1 commit
    • Justin Ruggles's avatar
      avconv: use default channel layouts when they are unknown · d3c59d50
      Justin Ruggles authored
      If either input or output layout is known and the channel counts match,
      use the known layout for both. Otherwise choose the default layout based on
      av_get_default_channel_layout().
      
      Changed some FATE references due to some WAVE files now having a non-zero
      channel mask.
      d3c59d50
  11. 08 Apr, 2012 2 commits
  12. 07 Apr, 2012 4 commits
  13. 20 Mar, 2012 1 commit
  14. 17 Mar, 2012 1 commit
  15. 03 Mar, 2012 1 commit
    • Justin Ruggles's avatar
      wmaenc: fix m/s stereo encoding for the first frame · 51ddf35c
      Justin Ruggles authored
      We need to set ms_stereo in encode_init() in order to avoid incorrectly
      encoding the first frame as non-m/s while flagging it as m/s. Fixes an
      uncomfortable pop in the left channel at the start of playback.
      
      CC:libav-stable@libav.org
      51ddf35c
  16. 02 Mar, 2012 1 commit
    • Martin Storsjö's avatar
      g722: Fix the QMF scaling · b087ce2b
      Martin Storsjö authored
      This fixes clipping if the encoder input used the full 16 bit
      input range (samples with a magnitude below 16383 worked fine).
      The filtered subband samples should be 15 bit maximum, while
      the code earlier produced them scaled to 16 bit.
      
      This makes the decoder output have double the magnitude
      compared to before.
      
      The spec reference samples doesn't test the QMF at all, which
      was why this part slipped past initially.
      Signed-off-by: 's avatarMartin Storsjö <martin@martin.st>
      b087ce2b
  17. 20 Feb, 2012 1 commit
    • Justin Ruggles's avatar
      adpcmenc: Use correct frame_size for Yamaha ADPCM. · 770a5c6d
      Justin Ruggles authored
      Output packet size should match avctx->block_align. The target output packet
      size is 1024 bytes.
      Before:
      mono   - 1024 samples -> 512 bytes
      stereo - 2048 samples -> 2048 bytes
      After:
      mono   - 2048 samples -> 1024 bytes
      stereo - 1024 samples -> 1024 bytes
      770a5c6d
  18. 11 Feb, 2012 1 commit
  19. 02 Feb, 2012 1 commit
  20. 23 Jan, 2012 1 commit
  21. 22 Jan, 2012 1 commit
  22. 10 Jan, 2012 1 commit
  23. 07 Jan, 2012 1 commit
  24. 03 Jan, 2012 1 commit
  25. 16 Dec, 2011 1 commit
  26. 15 Dec, 2011 1 commit
  27. 09 Dec, 2011 1 commit
  28. 07 Dec, 2011 1 commit
  29. 05 Dec, 2011 1 commit
  30. 01 Dec, 2011 1 commit
  31. 29 Nov, 2011 1 commit
  32. 02 Nov, 2011 1 commit