1
0
mirror of https://github.com/FFmpeg/FFmpeg.git synced 2024-12-02 03:06:28 +02:00
FFmpeg/libavcodec/audioconvert.c
Michael Niedermayer 23694e27f0 Drop deprecated SAMPLE_FMT_S24.
Originally committed as revision 15264 to svn://svn.ffmpeg.org/ffmpeg/trunk
2008-09-08 15:24:16 +00:00

155 lines
5.8 KiB
C

/*
* audio conversion
* Copyright (c) 2006 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
*/
/**
* @file audioconvert.c
* audio conversion
* @author Michael Niedermayer <michaelni@gmx.at>
*/
#include "avcodec.h"
#include "audioconvert.h"
typedef struct SampleFmtInfo {
const char *name;
int bits;
} SampleFmtInfo;
/** this table gives more information about formats */
static const SampleFmtInfo sample_fmt_info[SAMPLE_FMT_NB] = {
[SAMPLE_FMT_U8] = { .name = "u8", .bits = 8 },
[SAMPLE_FMT_S16] = { .name = "s16", .bits = 16 },
[SAMPLE_FMT_S32] = { .name = "s32", .bits = 32 },
[SAMPLE_FMT_FLT] = { .name = "flt", .bits = 32 },
[SAMPLE_FMT_DBL] = { .name = "dbl", .bits = 64 },
};
const char *avcodec_get_sample_fmt_name(int sample_fmt)
{
if (sample_fmt < 0 || sample_fmt >= SAMPLE_FMT_NB)
return NULL;
return sample_fmt_info[sample_fmt].name;
}
enum SampleFormat avcodec_get_sample_fmt(const char* name)
{
int i;
for (i=0; i < SAMPLE_FMT_NB; i++)
if (!strcmp(sample_fmt_info[i].name, name))
return i;
return SAMPLE_FMT_NONE;
}
void avcodec_sample_fmt_string (char *buf, int buf_size, int sample_fmt)
{
/* print header */
if (sample_fmt < 0)
snprintf (buf, buf_size, "name " " depth");
else if (sample_fmt < SAMPLE_FMT_NB) {
SampleFmtInfo info= sample_fmt_info[sample_fmt];
snprintf (buf, buf_size, "%-6s" " %2d ", info.name, info.bits);
}
}
struct AVAudioConvert {
int in_channels, out_channels;
int fmt_pair;
};
AVAudioConvert *av_audio_convert_alloc(enum SampleFormat out_fmt, int out_channels,
enum SampleFormat in_fmt, int in_channels,
const const float *matrix, int flags)
{
AVAudioConvert *ctx;
if (in_channels!=out_channels)
return NULL; /* FIXME: not supported */
ctx = av_malloc(sizeof(AVAudioConvert));
if (!ctx)
return NULL;
ctx->in_channels = in_channels;
ctx->out_channels = out_channels;
ctx->fmt_pair = out_fmt + SAMPLE_FMT_NB*in_fmt;
return ctx;
}
void av_audio_convert_free(AVAudioConvert *ctx)
{
av_free(ctx);
}
int av_audio_convert(AVAudioConvert *ctx,
void * const out[6], const int out_stride[6],
const void * const in[6], const int in_stride[6], int len)
{
int ch;
//FIXME optimize common cases
for(ch=0; ch<ctx->out_channels; ch++){
const int is= in_stride[ch];
const int os= out_stride[ch];
uint8_t *pi= in[ch];
uint8_t *po= out[ch];
uint8_t *end= po + os*len;
if(!out[ch])
continue;
#define CONV(ofmt, otype, ifmt, expr)\
if(ctx->fmt_pair == ofmt + SAMPLE_FMT_NB*ifmt){\
do{\
*(otype*)po = expr; pi += is; po += os;\
}while(po < end);\
}
//FIXME put things below under ifdefs so we do not waste space for cases no codec will need
//FIXME rounding and clipping ?
CONV(SAMPLE_FMT_U8 , uint8_t, SAMPLE_FMT_U8 , *(uint8_t*)pi)
else CONV(SAMPLE_FMT_S16, int16_t, SAMPLE_FMT_U8 , (*(uint8_t*)pi - 0x80)<<8)
else CONV(SAMPLE_FMT_S32, int32_t, SAMPLE_FMT_U8 , (*(uint8_t*)pi - 0x80)<<24)
else CONV(SAMPLE_FMT_FLT, float , SAMPLE_FMT_U8 , (*(uint8_t*)pi - 0x80)*(1.0 / (1<<7)))
else CONV(SAMPLE_FMT_DBL, double , SAMPLE_FMT_U8 , (*(uint8_t*)pi - 0x80)*(1.0 / (1<<7)))
else CONV(SAMPLE_FMT_U8 , uint8_t, SAMPLE_FMT_S16, (*(int16_t*)pi>>8) + 0x80)
else CONV(SAMPLE_FMT_S16, int16_t, SAMPLE_FMT_S16, *(int16_t*)pi)
else CONV(SAMPLE_FMT_S32, int32_t, SAMPLE_FMT_S16, *(int16_t*)pi<<16)
else CONV(SAMPLE_FMT_FLT, float , SAMPLE_FMT_S16, *(int16_t*)pi*(1.0 / (1<<15)))
else CONV(SAMPLE_FMT_DBL, double , SAMPLE_FMT_S16, *(int16_t*)pi*(1.0 / (1<<15)))
else CONV(SAMPLE_FMT_U8 , uint8_t, SAMPLE_FMT_S32, (*(int32_t*)pi>>24) + 0x80)
else CONV(SAMPLE_FMT_S16, int16_t, SAMPLE_FMT_S32, *(int32_t*)pi>>16)
else CONV(SAMPLE_FMT_S32, int32_t, SAMPLE_FMT_S32, *(int32_t*)pi)
else CONV(SAMPLE_FMT_FLT, float , SAMPLE_FMT_S32, *(int32_t*)pi*(1.0 / (1<<31)))
else CONV(SAMPLE_FMT_DBL, double , SAMPLE_FMT_S32, *(int32_t*)pi*(1.0 / (1<<31)))
else CONV(SAMPLE_FMT_U8 , uint8_t, SAMPLE_FMT_FLT, lrintf(*(float*)pi * (1<<7)) + 0x80)
else CONV(SAMPLE_FMT_S16, int16_t, SAMPLE_FMT_FLT, lrintf(*(float*)pi * (1<<15)))
else CONV(SAMPLE_FMT_S32, int32_t, SAMPLE_FMT_FLT, lrintf(*(float*)pi * (1<<31)))
else CONV(SAMPLE_FMT_FLT, float , SAMPLE_FMT_FLT, *(float*)pi)
else CONV(SAMPLE_FMT_DBL, double , SAMPLE_FMT_FLT, *(float*)pi)
else CONV(SAMPLE_FMT_U8 , uint8_t, SAMPLE_FMT_DBL, lrint(*(double*)pi * (1<<7)) + 0x80)
else CONV(SAMPLE_FMT_S16, int16_t, SAMPLE_FMT_DBL, lrint(*(double*)pi * (1<<15)))
else CONV(SAMPLE_FMT_S32, int32_t, SAMPLE_FMT_DBL, lrint(*(double*)pi * (1<<31)))
else CONV(SAMPLE_FMT_FLT, float , SAMPLE_FMT_DBL, *(double*)pi)
else CONV(SAMPLE_FMT_DBL, double , SAMPLE_FMT_DBL, *(double*)pi)
else return -1;
}
return 0;
}