diff options
Diffstat (limited to 'libavcodec/audioconvert.c')
| -rw-r--r-- | libavcodec/audioconvert.c | 120 |
1 files changed, 120 insertions, 0 deletions
diff --git a/libavcodec/audioconvert.c b/libavcodec/audioconvert.c new file mode 100644 index 0000000000..5e46fae2df --- /dev/null +++ b/libavcodec/audioconvert.c @@ -0,0 +1,120 @@ +/* + * 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 */ |