From c9a13a289d0e1607387854127476813a1ee3d34b Mon Sep 17 00:00:00 2001 From: Anton Khirnov Date: Mon, 28 Oct 2013 07:27:35 +0100 Subject: lavc: remove old unused audio conversion functions. --- libavcodec/Makefile | 1 - libavcodec/audioconvert.c | 116 ---------------------------------------------- libavcodec/audioconvert.h | 70 ---------------------------- 3 files changed, 187 deletions(-) delete mode 100644 libavcodec/audioconvert.c delete mode 100644 libavcodec/audioconvert.h diff --git a/libavcodec/Makefile b/libavcodec/Makefile index 8e0d60d18f..6f80a9e6ba 100644 --- a/libavcodec/Makefile +++ b/libavcodec/Makefile @@ -11,7 +11,6 @@ HEADERS = avcodec.h \ xvmc.h \ OBJS = allcodecs.o \ - audioconvert.o \ avpacket.o \ avpicture.o \ bitstream.o \ diff --git a/libavcodec/audioconvert.c b/libavcodec/audioconvert.c deleted file mode 100644 index 3714de78f1..0000000000 --- a/libavcodec/audioconvert.c +++ /dev/null @@ -1,116 +0,0 @@ -/* - * audio conversion - * Copyright (c) 2006 Michael Niedermayer - * - * 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 - */ - -#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; chout_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; -} diff --git a/libavcodec/audioconvert.h b/libavcodec/audioconvert.h deleted file mode 100644 index 7d76fd6879..0000000000 --- a/libavcodec/audioconvert.h +++ /dev/null @@ -1,70 +0,0 @@ -/* - * audio conversion - * Copyright (c) 2006 Michael Niedermayer - * 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 */ -- cgit v1.2.3