mirror of
https://github.com/FFmpeg/FFmpeg.git
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cae8f469fe
* commit '38282149b6ce8f4b8361e3b84542ba9aa8a1f32f': ppc: More consistent arch initialization Conflicts: libavcodec/fft.h libavcodec/mpegaudiodsp.c libavcodec/mpegaudiodsp.h Merged-by: Michael Niedermayer <michaelni@gmx.at>
122 lines
3.7 KiB
C
122 lines
3.7 KiB
C
/*
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* Format Conversion Utils
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* Copyright (c) 2000, 2001 Fabrice Bellard
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* Copyright (c) 2002-2004 Michael Niedermayer <michaelni@gmx.at>
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include "avcodec.h"
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#include "fmtconvert.h"
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#include "libavutil/common.h"
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static void int32_to_float_fmul_scalar_c(float *dst, const int *src, float mul, int len){
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int i;
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for(i=0; i<len; i++)
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dst[i] = src[i] * mul;
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}
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static av_always_inline int float_to_int16_one(const float *src){
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return av_clip_int16(lrintf(*src));
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}
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static void float_to_int16_c(int16_t *dst, const float *src, long len)
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{
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int i;
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for(i=0; i<len; i++)
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dst[i] = float_to_int16_one(src+i);
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}
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static void float_to_int16_interleave_c(int16_t *dst, const float **src,
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long len, int channels)
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{
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int i,j,c;
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if(channels==2){
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for(i=0; i<len; i++){
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dst[2*i] = float_to_int16_one(src[0]+i);
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dst[2*i+1] = float_to_int16_one(src[1]+i);
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}
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}else{
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for(c=0; c<channels; c++)
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for(i=0, j=c; i<len; i++, j+=channels)
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dst[j] = float_to_int16_one(src[c]+i);
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}
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}
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void ff_float_interleave_c(float *dst, const float **src, unsigned int len,
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int channels)
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{
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int j, c;
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unsigned int i;
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if (channels == 2) {
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for (i = 0; i < len; i++) {
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dst[2*i] = src[0][i];
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dst[2*i+1] = src[1][i];
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}
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} else if (channels == 1 && len < INT_MAX / sizeof(float)) {
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memcpy(dst, src[0], len * sizeof(float));
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} else {
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for (c = 0; c < channels; c++)
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for (i = 0, j = c; i < len; i++, j += channels)
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dst[j] = src[c][i];
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}
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}
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av_cold void ff_fmt_convert_init(FmtConvertContext *c, AVCodecContext *avctx)
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{
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c->int32_to_float_fmul_scalar = int32_to_float_fmul_scalar_c;
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c->float_to_int16 = float_to_int16_c;
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c->float_to_int16_interleave = float_to_int16_interleave_c;
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c->float_interleave = ff_float_interleave_c;
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if (ARCH_ARM) ff_fmt_convert_init_arm(c, avctx);
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if (ARCH_PPC) ff_fmt_convert_init_ppc(c, avctx);
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if (ARCH_X86) ff_fmt_convert_init_x86(c, avctx);
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if (HAVE_MIPSFPU) ff_fmt_convert_init_mips(c);
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}
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/* ffdshow custom code */
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void float_interleave(float *dst, const float **src, long len, int channels)
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{
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int i,j,c;
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if(channels==2){
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for(i=0; i<len; i++){
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dst[2*i] = src[0][i] / 32768.0f;
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dst[2*i+1] = src[1][i] / 32768.0f;
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}
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}else{
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for(c=0; c<channels; c++)
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for(i=0, j=c; i<len; i++, j+=channels)
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dst[j] = src[c][i] / 32768.0f;
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}
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}
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void float_interleave_noscale(float *dst, const float **src, long len, int channels)
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{
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int i,j,c;
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if(channels==2){
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for(i=0; i<len; i++){
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dst[2*i] = src[0][i];
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dst[2*i+1] = src[1][i];
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}
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}else{
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for(c=0; c<channels; c++)
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for(i=0, j=c; i<len; i++, j+=channels)
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dst[j] = src[c][i];
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}
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}
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