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https://github.com/FFmpeg/FFmpeg.git
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lavc: remove old unused audio conversion functions.
This commit is contained in:
Anton Khirnov
parent
6c82c87dbb
commit
c9a13a289d
@ -11,7 +11,6 @@ HEADERS = avcodec.h \
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xvmc.h \
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OBJS = allcodecs.o \
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audioconvert.o \
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avpacket.o \
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avpicture.o \
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bitstream.o \
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@ -1,116 +0,0 @@
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/*
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* audio conversion
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* Copyright (c) 2006 Michael Niedermayer <michaelni@gmx.at>
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*
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* This file is part of Libav.
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*
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* Libav 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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* Libav 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 Libav; 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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/**
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* @file
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* audio conversion
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* @author Michael Niedermayer <michaelni@gmx.at>
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*/
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#include "libavutil/avstring.h"
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#include "libavutil/common.h"
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#include "libavutil/libm.h"
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#include "libavutil/samplefmt.h"
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#include "avcodec.h"
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#include "audioconvert.h"
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struct AVAudioConvert {
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int in_channels, out_channels;
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int fmt_pair;
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};
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AVAudioConvert *av_audio_convert_alloc(enum AVSampleFormat out_fmt, int out_channels,
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enum AVSampleFormat in_fmt, int in_channels,
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const float *matrix, int flags)
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{
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AVAudioConvert *ctx;
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if (in_channels!=out_channels)
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return NULL; /* FIXME: not supported */
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ctx = av_malloc(sizeof(AVAudioConvert));
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if (!ctx)
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return NULL;
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ctx->in_channels = in_channels;
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ctx->out_channels = out_channels;
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ctx->fmt_pair = out_fmt + AV_SAMPLE_FMT_NB*in_fmt;
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return ctx;
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}
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void av_audio_convert_free(AVAudioConvert *ctx)
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{
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av_free(ctx);
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}
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int av_audio_convert(AVAudioConvert *ctx,
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void * const out[6], const int out_stride[6],
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const void * const in[6], const int in_stride[6], int len)
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{
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int ch;
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//FIXME optimize common cases
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for(ch=0; ch<ctx->out_channels; ch++){
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const int is= in_stride[ch];
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const int os= out_stride[ch];
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const uint8_t *pi= in[ch];
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uint8_t *po= out[ch];
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uint8_t *end= po + os*len;
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if(!out[ch])
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continue;
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#define CONV(ofmt, otype, ifmt, expr)\
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if(ctx->fmt_pair == ofmt + AV_SAMPLE_FMT_NB*ifmt){\
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do{\
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*(otype*)po = expr; pi += is; po += os;\
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}while(po < end);\
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}
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//FIXME put things below under ifdefs so we do not waste space for cases no codec will need
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//FIXME rounding ?
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CONV(AV_SAMPLE_FMT_U8 , uint8_t, AV_SAMPLE_FMT_U8 , *(const uint8_t*)pi)
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else CONV(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80)<<8)
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else CONV(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80)<<24)
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else CONV(AV_SAMPLE_FMT_FLT, float , AV_SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80)*(1.0 / (1<<7)))
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else CONV(AV_SAMPLE_FMT_DBL, double , AV_SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80)*(1.0 / (1<<7)))
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else CONV(AV_SAMPLE_FMT_U8 , uint8_t, AV_SAMPLE_FMT_S16, (*(const int16_t*)pi>>8) + 0x80)
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else CONV(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_S16, *(const int16_t*)pi)
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else CONV(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_S16, *(const int16_t*)pi<<16)
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else CONV(AV_SAMPLE_FMT_FLT, float , AV_SAMPLE_FMT_S16, *(const int16_t*)pi*(1.0 / (1<<15)))
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else CONV(AV_SAMPLE_FMT_DBL, double , AV_SAMPLE_FMT_S16, *(const int16_t*)pi*(1.0 / (1<<15)))
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else CONV(AV_SAMPLE_FMT_U8 , uint8_t, AV_SAMPLE_FMT_S32, (*(const int32_t*)pi>>24) + 0x80)
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else CONV(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_S32, *(const int32_t*)pi>>16)
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else CONV(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_S32, *(const int32_t*)pi)
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else CONV(AV_SAMPLE_FMT_FLT, float , AV_SAMPLE_FMT_S32, *(const int32_t*)pi*(1.0 / (1U<<31)))
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else CONV(AV_SAMPLE_FMT_DBL, double , AV_SAMPLE_FMT_S32, *(const int32_t*)pi*(1.0 / (1U<<31)))
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else CONV(AV_SAMPLE_FMT_U8 , uint8_t, AV_SAMPLE_FMT_FLT, av_clip_uint8( lrintf(*(const float*)pi * (1<<7)) + 0x80))
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else CONV(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, av_clip_int16( lrintf(*(const float*)pi * (1<<15))))
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else CONV(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, av_clipl_int32(llrintf(*(const float*)pi * (1U<<31))))
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else CONV(AV_SAMPLE_FMT_FLT, float , AV_SAMPLE_FMT_FLT, *(const float*)pi)
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else CONV(AV_SAMPLE_FMT_DBL, double , AV_SAMPLE_FMT_FLT, *(const float*)pi)
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else CONV(AV_SAMPLE_FMT_U8 , uint8_t, AV_SAMPLE_FMT_DBL, av_clip_uint8( lrint(*(const double*)pi * (1<<7)) + 0x80))
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else CONV(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, av_clip_int16( lrint(*(const double*)pi * (1<<15))))
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else CONV(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, av_clipl_int32(llrint(*(const double*)pi * (1U<<31))))
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else CONV(AV_SAMPLE_FMT_FLT, float , AV_SAMPLE_FMT_DBL, *(const double*)pi)
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else CONV(AV_SAMPLE_FMT_DBL, double , AV_SAMPLE_FMT_DBL, *(const double*)pi)
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else return -1;
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}
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return 0;
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}
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@ -1,70 +0,0 @@
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/*
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* audio conversion
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* Copyright (c) 2006 Michael Niedermayer <michaelni@gmx.at>
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* Copyright (c) 2008 Peter Ross
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*
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* This file is part of Libav.
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*
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* Libav 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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* Libav 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 Libav; 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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#ifndef AVCODEC_AUDIOCONVERT_H
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#define AVCODEC_AUDIOCONVERT_H
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/**
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* @file
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* Audio format conversion routines
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*/
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#include "libavutil/cpu.h"
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#include "avcodec.h"
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#include "libavutil/channel_layout.h"
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struct AVAudioConvert;
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typedef struct AVAudioConvert AVAudioConvert;
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/**
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* Create an audio sample format converter context
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* @param out_fmt Output sample format
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* @param out_channels Number of output channels
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* @param in_fmt Input sample format
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* @param in_channels Number of input channels
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* @param[in] matrix Channel mixing matrix (of dimension in_channel*out_channels). Set to NULL to ignore.
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* @param flags See AV_CPU_FLAG_xx
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* @return NULL on error
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*/
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AVAudioConvert *av_audio_convert_alloc(enum AVSampleFormat out_fmt, int out_channels,
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enum AVSampleFormat in_fmt, int in_channels,
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const float *matrix, int flags);
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/**
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* Free audio sample format converter context
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*/
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void av_audio_convert_free(AVAudioConvert *ctx);
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/**
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* Convert between audio sample formats
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* @param[in] out array of output buffers for each channel. set to NULL to ignore processing of the given channel.
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* @param[in] out_stride distance between consecutive output samples (measured in bytes)
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* @param[in] in array of input buffers for each channel
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* @param[in] in_stride distance between consecutive input samples (measured in bytes)
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* @param len length of audio frame size (measured in samples)
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*/
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int av_audio_convert(AVAudioConvert *ctx,
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void * const out[6], const int out_stride[6],
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const void * const in[6], const int in_stride[6], int len);
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#endif /* AVCODEC_AUDIOCONVERT_H */
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