diff options
author | James Almer <jamrial@gmail.com> | 2017-10-21 23:16:13 -0300 |
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committer | James Almer <jamrial@gmail.com> | 2017-10-21 23:16:13 -0300 |
commit | 43befa68263be9db116997226e5147d88aba008d (patch) | |
tree | 70494f4334e049bf2c6b8a296379bbd74b4ad0ec | |
parent | 8f483108b503fa03ed5e956e25df4cb899171df5 (diff) | |
download | ffmpeg-43befa68263be9db116997226e5147d88aba008d.tar.gz |
avcodec: Drop deprecated audio convert API
Deprecated in 10/2013.
-rw-r--r-- | libavcodec/Makefile | 1 | ||||
-rw-r--r-- | libavcodec/audioconvert.c | 120 | ||||
-rw-r--r-- | libavcodec/audioconvert.h | 86 | ||||
-rw-r--r-- | libavcodec/version.h | 3 |
4 files changed, 0 insertions, 210 deletions
diff --git a/libavcodec/Makefile b/libavcodec/Makefile index 651348972e..4424e749ce 100644 --- a/libavcodec/Makefile +++ b/libavcodec/Makefile @@ -20,7 +20,6 @@ HEADERS = avcodec.h \ xvmc.h \ OBJS = allcodecs.o \ - audioconvert.o \ avdct.o \ avpacket.o \ avpicture.o \ diff --git a/libavcodec/audioconvert.c b/libavcodec/audioconvert.c deleted file mode 100644 index 5e46fae2df..0000000000 --- a/libavcodec/audioconvert.c +++ /dev/null @@ -1,120 +0,0 @@ -/* - * 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 - * 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" - -#if FF_API_AUDIO_CONVERT - -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; -} - -#endif /* FF_API_AUDIO_CONVERT */ diff --git a/libavcodec/audioconvert.h b/libavcodec/audioconvert.h deleted file mode 100644 index 996c3f37ad..0000000000 --- a/libavcodec/audioconvert.h +++ /dev/null @@ -1,86 +0,0 @@ -/* - * audio conversion - * Copyright (c) 2006 Michael Niedermayer <michaelni@gmx.at> - * Copyright (c) 2008 Peter Ross - * - * 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 - */ - -#ifndef AVCODEC_AUDIOCONVERT_H -#define AVCODEC_AUDIOCONVERT_H - -#include "version.h" - -/** - * @file - * Audio format conversion routines - * This interface is deprecated and will be dropped in a future - * version. You should use the libswresample library instead. - */ - -#if FF_API_AUDIO_CONVERT - -#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 - * @deprecated See libswresample - */ - -attribute_deprecated -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 - * @deprecated See libswresample - */ - -attribute_deprecated -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) - * @deprecated See libswresample - */ - -attribute_deprecated -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 /* FF_API_AUDIO_CONVERT */ - -#endif /* AVCODEC_AUDIOCONVERT_H */ diff --git a/libavcodec/version.h b/libavcodec/version.h index acbc61d757..1431d94d76 100644 --- a/libavcodec/version.h +++ b/libavcodec/version.h @@ -54,9 +54,6 @@ #ifndef FF_API_VIMA_DECODER #define FF_API_VIMA_DECODER (LIBAVCODEC_VERSION_MAJOR < 58) #endif -#ifndef FF_API_AUDIO_CONVERT -#define FF_API_AUDIO_CONVERT (LIBAVCODEC_VERSION_MAJOR < 58) -#endif #ifndef FF_API_LOWRES #define FF_API_LOWRES (LIBAVCODEC_VERSION_MAJOR < 59) #endif |