Commit Graph

9 Commits

Author SHA1 Message Date
Michael Niedermayer
bd4ebbbbed Merge remote-tracking branch 'qatar/master'
* qatar/master:
  proresdsp: fix function prototypes.
  prores-idct: fix overflow in c code.
  fate: update prores-alpha ref after changing pix_fmt to yuv444p10le
  prores: add missing feature warning for alpha
  mov: 10l: Terminate string with 0 not '0'
  mov: Prevent illegal writes when chapter titles are very short.
  prores: add appropriate -fix_fmt parameter to FATE command
  riff: always generate a proper WAVEFORMATEX structure in ff_put_wav_header
  lavc: add a flag-based error_recognition field to AVCodecContext and deprecate non-flag-based ER field
  lavc: rename deprecation symbol FF_API_VERY_AGGRESSIVE to FF_API_ER

Conflicts:
	libavcodec/avcodec.h
	libavformat/mov.c
	tests/fate/prores.mak
	tests/ref/acodec/g726
	tests/ref/fate/prores-alpha

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2011-10-14 22:24:00 +02:00
John Brooks
2c4e08d893 riff: always generate a proper WAVEFORMATEX structure in ff_put_wav_header
The cbSize field should be included in all cases, even with PCM where
its value is ignored.

Fixes encoding PCM audio in Matroska for some players which insist on
a full WAVEFORMATEX structure for A_MS/ACM audio.

Since fate uses wav files for the audio test a larger number of tests
has changed checksums or shifted positions due to the 2 byte longer
wave header.

Signed-off-by: Janne Grunau <janne-libav@jannau.net>
2011-10-14 13:28:58 +02:00
Michael Niedermayer
8593b743a8 rematrix: dont use floats for int16 code.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2011-09-19 09:25:50 +02:00
Michael Niedermayer
b5875b9111 Add libswresample.
Similar to libswscale this does resampling and format convertion, just for audio
instead of video.
changing sampling rate, sample formats, channel layouts and sample packing all
in one with a very simple public interface.

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2011-09-19 07:04:17 +02:00
Justin Ruggles
c7d89948a3 Set a constant frame size for encoding G.726 audio.
Originally committed as revision 25107 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-09-11 19:52:09 +00:00
Måns Rullgård
c43d77c163 tiny_psnr: skip wav headers on input files
The byte count printed excludes the header, and offsets are applied
after the the headers are skipped.

Reference files updated to reflect new output.  Some stddev/psnr values
have changed slightly due to headers no longer being compared.

Originally committed as revision 24143 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-07-09 16:06:05 +00:00
Vitor Sessak
cb0067ec25 tiny_psnr: print max absolute difference between files
Regression test reference updates are due to the extra output
from tiny_psnr.

Patch by Vitor Sessak

Originally committed as revision 24132 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-07-09 00:40:37 +00:00
Måns Rullgård
cc3e2472f3 Place regression test output files in subdirs per family
Originally committed as revision 22155 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-03-02 21:41:52 +00:00
Måns Rullgård
c676895fd9 Separate audio-only tests so they are only run once
Originally committed as revision 21556 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-01-30 21:47:13 +00:00