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ffmpeg.wasm-core
Commit c9a13a28 authored by Anton Khirnov's avatar Anton Khirnov
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lavc: remove old unused audio conversion functions.

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libavcodec/Makefile
View file @ c9a13a28
......@@ -11,7 +11,6 @@ HEADERS = avcodec.h \
xvmc.h \
OBJS = allcodecs.o \
audioconvert.o \
avpacket.o \
avpicture.o \
bitstream.o \
......
libavcodec/audioconvert.c deleted 100644 → 0
View file @ 6c82c87d
/*
* audio conversion
* Copyright (c) 2006 Michael Niedermayer <michaelni@gmx.at>
*
* This file is part of Libav.
*
* Libav 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.
*
* Libav 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 Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* audio conversion
* @author Michael Niedermayer <michaelni@gmx.at>
*/
#include "libavutil/avstring.h"
#include "libavutil/common.h"
#include "libavutil/libm.h"
#include "libavutil/samplefmt.h"
#include "avcodec.h"
#include "audioconvert.h"
struct AVAudioConvert {
int in_channels, out_channels;
int fmt_pair;
};
AVAudioConvert *av_audio_convert_alloc(enum AVSampleFormat out_fmt, int out_channels,
enum AVSampleFormat in_fmt, int in_channels,
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 + AV_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];
const 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 + AV_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 ?
CONV(AV_SAMPLE_FMT_U8 , uint8_t, AV_SAMPLE_FMT_U8 , *(const uint8_t*)pi)
else CONV(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80)<<8)
else CONV(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80)<<24)
else CONV(AV_SAMPLE_FMT_FLT, float , AV_SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80)*(1.0 / (1<<7)))
else CONV(AV_SAMPLE_FMT_DBL, double , AV_SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80)*(1.0 / (1<<7)))
else CONV(AV_SAMPLE_FMT_U8 , uint8_t, AV_SAMPLE_FMT_S16, (*(const int16_t*)pi>>8) + 0x80)
else CONV(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_S16, *(const int16_t*)pi)
else CONV(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_S16, *(const int16_t*)pi<<16)
else CONV(AV_SAMPLE_FMT_FLT, float , AV_SAMPLE_FMT_S16, *(const int16_t*)pi*(1.0 / (1<<15)))
else CONV(AV_SAMPLE_FMT_DBL, double , AV_SAMPLE_FMT_S16, *(const int16_t*)pi*(1.0 / (1<<15)))
else CONV(AV_SAMPLE_FMT_U8 , uint8_t, AV_SAMPLE_FMT_S32, (*(const int32_t*)pi>>24) + 0x80)
else CONV(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_S32, *(const int32_t*)pi>>16)
else CONV(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_S32, *(const int32_t*)pi)
else CONV(AV_SAMPLE_FMT_FLT, float , AV_SAMPLE_FMT_S32, *(const int32_t*)pi*(1.0 / (1U<<31)))
else CONV(AV_SAMPLE_FMT_DBL, double , AV_SAMPLE_FMT_S32, *(const int32_t*)pi*(1.0 / (1U<<31)))
else CONV(AV_SAMPLE_FMT_U8 , uint8_t, AV_SAMPLE_FMT_FLT, av_clip_uint8( lrintf(*(const float*)pi * (1<<7)) + 0x80))
else CONV(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, av_clip_int16( lrintf(*(const float*)pi * (1<<15))))
else CONV(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, av_clipl_int32(llrintf(*(const float*)pi * (1U<<31))))
else CONV(AV_SAMPLE_FMT_FLT, float , AV_SAMPLE_FMT_FLT, *(const float*)pi)
else CONV(AV_SAMPLE_FMT_DBL, double , AV_SAMPLE_FMT_FLT, *(const float*)pi)
else CONV(AV_SAMPLE_FMT_U8 , uint8_t, AV_SAMPLE_FMT_DBL, av_clip_uint8( lrint(*(const double*)pi * (1<<7)) + 0x80))
else CONV(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, av_clip_int16( lrint(*(const double*)pi * (1<<15))))
else CONV(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, av_clipl_int32(llrint(*(const double*)pi * (1U<<31))))
else CONV(AV_SAMPLE_FMT_FLT, float , AV_SAMPLE_FMT_DBL, *(const double*)pi)
else CONV(AV_SAMPLE_FMT_DBL, double , AV_SAMPLE_FMT_DBL, *(const double*)pi)
else return -1;
}
return 0;
}
libavcodec/audioconvert.h deleted 100644 → 0
View file @ 6c82c87d
/*
* audio conversion
* Copyright (c) 2006 Michael Niedermayer <michaelni@gmx.at>
* Copyright (c) 2008 Peter Ross
*
* This file is part of Libav.
*
* Libav 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.
*
* Libav 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 Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef AVCODEC_AUDIOCONVERT_H
#define AVCODEC_AUDIOCONVERT_H
/**
* @file
* Audio format conversion routines
*/
#include "libavutil/cpu.h"
#include "avcodec.h"
#include "libavutil/channel_layout.h"
struct AVAudioConvert;
typedef struct AVAudioConvert AVAudioConvert;
/**
* Create an audio sample format converter context
* @param out_fmt Output sample format
* @param out_channels Number of output channels
* @param in_fmt Input sample format
* @param in_channels Number of input channels
* @param[in] matrix Channel mixing matrix (of dimension in_channel*out_channels). Set to NULL to ignore.
* @param flags See AV_CPU_FLAG_xx
* @return NULL on error
*/
AVAudioConvert *av_audio_convert_alloc(enum AVSampleFormat out_fmt, int out_channels,
enum AVSampleFormat in_fmt, int in_channels,
const float *matrix, int flags);
/**
* Free audio sample format converter context
*/
void av_audio_convert_free(AVAudioConvert *ctx);
/**
* Convert between audio sample formats
* @param[in] out array of output buffers for each channel. set to NULL to ignore processing of the given channel.
* @param[in] out_stride distance between consecutive output samples (measured in bytes)
* @param[in] in array of input buffers for each channel
* @param[in] in_stride distance between consecutive input samples (measured in bytes)
* @param len length of audio frame size (measured in samples)
*/
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);
#endif /* AVCODEC_AUDIOCONVERT_H */
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