FFmpeg/libavcodec/fmtconvert.c
Michael Niedermayer 75a37b57a5 Merge remote-tracking branch 'qatar/master'
* qatar/master:
  APIchanges: fill in date and commit for request_sample_fmt
  Add floating-point sample format support to the ac3, eac3, dca, aac, and vorbis decoders.
  Add support for request_sample_format in ffmpeg and ffplay.
  Add APIchanges entry for request_sample_fmt.
  Add request_sample_fmt field to AVCodecContext.
  Add float_interleave() to FmtConvertContext with x86-optimized versions.
  Remove unused make variable SEEK_REFFILE
  fate: remove redundant aref and vref references
  fate: remove do_ffmpeg_nocheck function
  fate: do not collect -benchmark output
  mpegaudiodec: remove decode_end() function
  fate: run aref and vref as regular tests
  mpegaudio: sanitise compute_antialias_* names
  mpeg12: add slice-threading checks to slice-threading initializers.
  h264: copy pixel_shift between slice threading contexts.
  mdec: enable frame-level multithreading.
  mdec.c: fix overread.

Conflicts:
	libavcodec/aacdec.c
	libavcodec/ac3dec.c
	libavcodec/avcodec.h
	libavcodec/dca.c
	libavcodec/h264.c
	libavcodec/mdec.c
	libavcodec/mpeg12.c
	libavcodec/options.c
	libavcodec/version.h
	libavcodec/vorbisdec.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2011-05-19 06:00:31 +02:00

120 lines
3.6 KiB
C

/*
* Format Conversion Utils
* Copyright (c) 2000, 2001 Fabrice Bellard
* Copyright (c) 2002-2004 Michael Niedermayer <michaelni@gmx.at>
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "avcodec.h"
#include "fmtconvert.h"
static void int32_to_float_fmul_scalar_c(float *dst, const int *src, float mul, int len){
int i;
for(i=0; i<len; i++)
dst[i] = src[i] * mul;
}
static av_always_inline int float_to_int16_one(const float *src){
return av_clip_int16(lrintf(*src));
}
static void float_to_int16_c(int16_t *dst, const float *src, long len)
{
int i;
for(i=0; i<len; i++)
dst[i] = float_to_int16_one(src+i);
}
static void float_to_int16_interleave_c(int16_t *dst, const float **src,
long len, int channels)
{
int i,j,c;
if(channels==2){
for(i=0; i<len; i++){
dst[2*i] = float_to_int16_one(src[0]+i);
dst[2*i+1] = float_to_int16_one(src[1]+i);
}
}else{
for(c=0; c<channels; c++)
for(i=0, j=c; i<len; i++, j+=channels)
dst[j] = float_to_int16_one(src[c]+i);
}
}
void ff_float_interleave_c(float *dst, const float **src, unsigned int len,
int channels)
{
int j, c;
unsigned int i;
if (channels == 2) {
for (i = 0; i < len; i++) {
dst[2*i] = src[0][i];
dst[2*i+1] = src[1][i];
}
} else if (channels == 1 && len < INT_MAX / sizeof(float)) {
memcpy(dst, src[0], len * sizeof(float));
} else {
for (c = 0; c < channels; c++)
for (i = 0, j = c; i < len; i++, j += channels)
dst[j] = src[c][i];
}
}
av_cold void ff_fmt_convert_init(FmtConvertContext *c, AVCodecContext *avctx)
{
c->int32_to_float_fmul_scalar = int32_to_float_fmul_scalar_c;
c->float_to_int16 = float_to_int16_c;
c->float_to_int16_interleave = float_to_int16_interleave_c;
c->float_interleave = ff_float_interleave_c;
if (ARCH_ARM) ff_fmt_convert_init_arm(c, avctx);
if (HAVE_ALTIVEC) ff_fmt_convert_init_altivec(c, avctx);
if (HAVE_MMX) ff_fmt_convert_init_x86(c, avctx);
}
/* ffdshow custom code */
void float_interleave(float *dst, const float **src, long len, int channels)
{
int i,j,c;
if(channels==2){
for(i=0; i<len; i++){
dst[2*i] = src[0][i] / 32768.0f;
dst[2*i+1] = src[1][i] / 32768.0f;
}
}else{
for(c=0; c<channels; c++)
for(i=0, j=c; i<len; i++, j+=channels)
dst[j] = src[c][i] / 32768.0f;
}
}
void float_interleave_noscale(float *dst, const float **src, long len, int channels)
{
int i,j,c;
if(channels==2){
for(i=0; i<len; i++){
dst[2*i] = src[0][i];
dst[2*i+1] = src[1][i];
}
}else{
for(c=0; c<channels; c++)
for(i=0, j=c; i<len; i++, j+=channels)
dst[j] = src[c][i];
}
}