Merge remote-tracking branch 'qatar/master'

* qatar/master:
  Fix compilation of iirfilter-test.
  libx264: handle closed GOP codec flag
  lavf: remove duplicate assignment in avformat_alloc_context.
  lavf: use designated initializers for AVClasses.
  flvdec: clenup debug code
  asfdec: fix possible overread on broken files.
  asfdec: do not fall back to binary/generic search
  asfdec: reindent after previous commit c7bd5ed
  asfdec: fallback to binary search internally
  mpegaudio: add _fixed suffix to some names
  Modify x86util.asm to ease transitioning to 10-bit H.264 assembly.
  dct: build dct32 as separate object files
  qdm2: include correct header for rdft

Conflicts:
	ffpresets/libx264-fast.ffpreset
	ffpresets/libx264-fast_firstpass.ffpreset
	ffpresets/libx264-faster.ffpreset
	ffpresets/libx264-faster_firstpass.ffpreset
	ffpresets/libx264-medium.ffpreset
	ffpresets/libx264-medium_firstpass.ffpreset
	ffpresets/libx264-placebo.ffpreset
	ffpresets/libx264-placebo_firstpass.ffpreset
	ffpresets/libx264-slow.ffpreset
	ffpresets/libx264-slow_firstpass.ffpreset
	ffpresets/libx264-slower.ffpreset
	ffpresets/libx264-slower_firstpass.ffpreset
	ffpresets/libx264-superfast.ffpreset
	ffpresets/libx264-superfast_firstpass.ffpreset
	ffpresets/libx264-ultrafast.ffpreset
	ffpresets/libx264-ultrafast_firstpass.ffpreset
	ffpresets/libx264-veryfast.ffpreset
	ffpresets/libx264-veryfast_firstpass.ffpreset
	ffpresets/libx264-veryslow.ffpreset
	ffpresets/libx264-veryslow_firstpass.ffpreset
	libavformat/flvdec.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
This commit is contained in:
Michael Niedermayer
2011-05-18 05:42:42 +02:00
29 changed files with 299 additions and 171 deletions

View File

@@ -29,6 +29,7 @@
#include "get_bits.h"
#include "dsputil.h"
#include "mathops.h"
#include "dct32.h"
/*
* TODO:
@@ -57,7 +58,7 @@
# define FIXHR(a) ((int)((a) * (1LL<<32) + 0.5))
# define MULH3(x, y, s) MULH((s)*(x), y)
# define MULLx(x, y, s) MULL(x,y,s)
# define RENAME(a) a
# define RENAME(a) a ## _fixed
# define OUT_FMT AV_SAMPLE_FMT_S16
#endif
@@ -68,12 +69,6 @@
#include "mpegaudiodata.h"
#include "mpegaudiodectab.h"
#if CONFIG_FLOAT
# include "fft.h"
#else
# include "dct32.c"
#endif
static void compute_antialias(MPADecodeContext *s, GranuleDef *g);
static void apply_window_mp3_c(MPA_INT *synth_buf, MPA_INT *window,
int *dither_state, OUT_INT *samples, int incr);
@@ -626,7 +621,7 @@ static void apply_window_mp3_c(MPA_INT *synth_buf, MPA_INT *window,
32 samples. */
/* XXX: optimize by avoiding ring buffer usage */
#if !CONFIG_FLOAT
void ff_mpa_synth_filter(MPA_INT *synth_buf_ptr, int *synth_buf_offset,
void ff_mpa_synth_filter_fixed(MPA_INT *synth_buf_ptr, int *synth_buf_offset,
MPA_INT *window, int *dither_state,
OUT_INT *samples, int incr,
INTFLOAT sb_samples[SBLIMIT])
@@ -637,7 +632,7 @@ void ff_mpa_synth_filter(MPA_INT *synth_buf_ptr, int *synth_buf_offset,
offset = *synth_buf_offset;
synth_buf = synth_buf_ptr + offset;
dct32(synth_buf, sb_samples);
ff_dct32_fixed(synth_buf, sb_samples);
apply_window_mp3_c(synth_buf, window, dither_state, samples, incr);
offset = (offset - 32) & 511;