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mirror of https://git.videolan.org/git/ffmpeg.git synced 2024-09-07 00:20:09 +02:00
ffmpeg/libavcodec/fmtconvert.c
Justin Ruggles c73d99e672 Separate format conversion DSP functions from DSPContext.
This will be beneficial for use with the audio conversion API without
requiring it to depend on all of dsputil.

Signed-off-by: Mans Rullgard <mans@mansr.com>
2011-02-02 02:44:53 +00:00

69 lines
2.2 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);
}
}
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;
if (ARCH_ARM) ff_fmt_convert_init_arm(c, avctx);
if (ARCH_PPC) ff_fmt_convert_init_ppc(c, avctx);
if (HAVE_MMX) ff_fmt_convert_init_x86(c, avctx);
}