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Merge branch 'master' into oldabi

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* master: (109 commits)
  libx264: fix open gop default. Please use -x264opts to force open gop This fixes Ticket268
  avfilter picture pool: double free hotfix
  mpegaudio_parser: be less picky on the start position
  ppc32: Fix movrel
  Replace usages of av_get_bits_per_sample_fmt() with av_get_bytes_per_sample().
  x86: cabac: fix register constraints for 32-bit mode
  cabac: move x86 asm to libavcodec/x86/cabac.h
  x86: h264: cast pointers to intptr_t rather than int
  x86: h264: remove hardcoded edi in decode_significance_8x8_x86()
  x86: h264: remove hardcoded esi in decode_significance[_8x8]_x86()
  x86: h264: remove hardcoded edx in decode_significance[_8x8]_x86()
  x86: h264: remove hardcoded eax in decode_significance[_8x8]_x86()
  x86: cabac: change 'a' constraint to 'r' in get_cabac_inline()
  x86: cabac: remove hardcoded esi in get_cabac_inline()
  x86: cabac: remove hardcoded edx in get_cabac_inline()
  x86: cabac: remove unused macro parameter
  x86: cabac: remove hardcoded ebx in inline asm
  x86: cabac: remove hardcoded struct offsets from inline asm
  cabac: remove inline asm under #if 0
  cabac: remove BRANCHLESS_CABAC_DECODER switch
  ...

Conflicts:
	cmdutils.c
	ffserver.c
	libavfilter/avfilter.h
	libavformat/avformat.h
	libavformat/utils.c
	libavformat/version.h
	libavutil/avutil.h

Merged-by: Michael Niedermayer <michaelni@gmx.at>
This commit is contained in:
Michael Niedermayer
2011-06-21 05:22:40 +02:00
parent 56629aa012 d39b33c63b
commit 779d7610c7
107 changed files with 2293 additions and 932 deletions
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4
libavcodec/resample.c
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@@ -239,8 +239,8 @@ ReSampleContext *av_audio_resample_init(int output_channels, int input_channels,
s->sample_fmt[0] = sample_fmt_in;
s->sample_fmt[1] = sample_fmt_out;
s->sample_size[0] = av_get_bits_per_sample_fmt(s->sample_fmt[0]) >> 3;
s->sample_size[1] = av_get_bits_per_sample_fmt(s->sample_fmt[1]) >> 3;
s->sample_size[0] = av_get_bytes_per_sample(s->sample_fmt[0]);
s->sample_size[1] = av_get_bytes_per_sample(s->sample_fmt[1]);
if (s->sample_fmt[0] != AV_SAMPLE_FMT_S16) {
if (!(s->convert_ctx[0] = av_audio_convert_alloc(AV_SAMPLE_FMT_S16, 1,