[FFmpeg-cvslog] lavc: remove old unused audio conversion functions.
Anton Khirnov
git at videolan.org
Tue Oct 29 10:31:46 CET 2013
ffmpeg | branch: master | Anton Khirnov <anton at khirnov.net> | Mon Oct 28 07:27:35 2013 +0100| [c9a13a289d0e1607387854127476813a1ee3d34b] | committer: Anton Khirnov
lavc: remove old unused audio conversion functions.
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=c9a13a289d0e1607387854127476813a1ee3d34b
---
libavcodec/Makefile | 1 -
libavcodec/audioconvert.c | 116 ---------------------------------------------
libavcodec/audioconvert.h | 70 ---------------------------
3 files changed, 187 deletions(-)
diff --git a/libavcodec/Makefile b/libavcodec/Makefile
index 8e0d60d..6f80a9e 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 3714de7..0000000
--- a/libavcodec/audioconvert.c
+++ /dev/null
@@ -1,116 +0,0 @@
-/*
- * audio conversion
- * Copyright (c) 2006 Michael Niedermayer <michaelni at 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 at 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;
-}
diff --git a/libavcodec/audioconvert.h b/libavcodec/audioconvert.h
deleted file mode 100644
index 7d76fd6..0000000
--- a/libavcodec/audioconvert.h
+++ /dev/null
@@ -1,70 +0,0 @@
-/*
- * audio conversion
- * Copyright (c) 2006 Michael Niedermayer <michaelni at 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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