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author | Michael Niedermayer <michaelni@gmx.at> | 2006-08-19 20:22:57 +0000 |
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committer | Michael Niedermayer <michaelni@gmx.at> | 2006-08-19 20:22:57 +0000 |
commit | 0eb6817d9821bf67f33ccfe9b427cf736b95881e (patch) | |
tree | 30d8acdcc5cadb03d9ca8b9dcf43e18b8a5279bf | |
parent | c52e13f1a154929808508def4ba9d3ba0c7a6fa1 (diff) | |
download | ffmpeg-0eb6817d9821bf67f33ccfe9b427cf736b95881e.tar.gz |
audio format conversion
untested and unused
Originally committed as revision 6029 to svn://svn.ffmpeg.org/ffmpeg/trunk
-rw-r--r-- | libavcodec/Makefile | 1 | ||||
-rw-r--r-- | libavcodec/audioconvert.c | 77 |
2 files changed, 78 insertions, 0 deletions
diff --git a/libavcodec/Makefile b/libavcodec/Makefile index 89f4c402c2..91a0d5daf5 100644 --- a/libavcodec/Makefile +++ b/libavcodec/Makefile @@ -19,6 +19,7 @@ OBJS= bitstream.o utils.o allcodecs.o \ vp3dsp.o h264idct.o rangecoder.o pnm.o h263.o msmpeg4.o h263dec.o \ opt.o \ bitstream_filter.o \ + audioconvert.o \ HEADERS = avcodec.h diff --git a/libavcodec/audioconvert.c b/libavcodec/audioconvert.c new file mode 100644 index 0000000000..56d351c50e --- /dev/null +++ b/libavcodec/audioconvert.c @@ -0,0 +1,77 @@ +/* + * audio conversation + * Copyright (c) 2006 Michael Niedermayer <michaelni@gmx.at> + * + * This library 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 of the License, or (at your option) any later version. + * + * This library 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 this library; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + * + */ + +/** + * @file audioconvert.c + * audio conversation + * @author Michael Niedermayer <michaelni@gmx.at> + */ + +#include "avcodec.h" + +int av_audio_convert(void *maybe_dspcontext_or_something_av_convert_specific, + void *out[6], int out_stride[6], enum SampleFormat out_fmt, + void * in[6], int in_stride[6], enum SampleFormat in_fmt, int len){ + int ch; + const int isize= FFMIN( in_fmt+1, 4); + const int osize= FFMIN(out_fmt+1, 4); + const int fmt_pair= out_fmt + 5*in_fmt; + + //FIXME optimize common cases + + for(ch=0; ch<6; ch++){ + const int is= in_stride[ch] * isize; + const int os= out_stride[ch] * osize; + uint8_t *pi= in[ch]; + uint8_t *po= out[ch]; + uint8_t *end= po + os; + if(!out[ch]) + continue; + +#define CONV(ofmt, otype, ifmt, expr)\ +if(fmt_pair == ofmt + 5*ifmt){\ + do{\ + *(otype*)po = expr; pi += is; po += os;\ + }while(po < end);\ +} + +//FIXME put things below under ifdefs so we dont waste space for cases no codec will need +//FIXME rounding and cliping ? + + CONV(SAMPLE_FMT_U8 , uint8_t, SAMPLE_FMT_U8 , *(uint8_t*)pi) + else CONV(SAMPLE_FMT_S16, int16_t, SAMPLE_FMT_U8 , (*(uint8_t*)pi - 0x80)<<8) + else CONV(SAMPLE_FMT_S32, int32_t, SAMPLE_FMT_U8 , (*(uint8_t*)pi - 0x80)<<24) + else CONV(SAMPLE_FMT_FLT, float , SAMPLE_FMT_U8 , (*(uint8_t*)pi - 0x80)*(1.0 / (1<<7))) + else CONV(SAMPLE_FMT_U8 , uint8_t, SAMPLE_FMT_S16, (*(int16_t*)pi>>8) + 0x80) + else CONV(SAMPLE_FMT_S16, int16_t, SAMPLE_FMT_S16, *(int16_t*)pi) + else CONV(SAMPLE_FMT_S32, int32_t, SAMPLE_FMT_S16, *(int16_t*)pi<<16) + else CONV(SAMPLE_FMT_FLT, float , SAMPLE_FMT_S16, *(int16_t*)pi*(1.0 / (1<<15))) + else CONV(SAMPLE_FMT_U8 , uint8_t, SAMPLE_FMT_S32, (*(int32_t*)pi>>24) + 0x80) + else CONV(SAMPLE_FMT_S16, int16_t, SAMPLE_FMT_S32, *(int32_t*)pi>>16) + else CONV(SAMPLE_FMT_S32, int32_t, SAMPLE_FMT_S32, *(int32_t*)pi) + else CONV(SAMPLE_FMT_FLT, float , SAMPLE_FMT_S32, *(int32_t*)pi*(1.0 / (1<<31))) + else CONV(SAMPLE_FMT_U8 , uint8_t, SAMPLE_FMT_FLT, lrintf(*(float*)pi * (1<<7)) + 0x80) + else CONV(SAMPLE_FMT_S16, int16_t, SAMPLE_FMT_FLT, lrintf(*(float*)pi * (1<<15))) + else CONV(SAMPLE_FMT_S32, int32_t, SAMPLE_FMT_FLT, lrintf(*(float*)pi * (1<<31))) + else CONV(SAMPLE_FMT_FLT, float , SAMPLE_FMT_FLT, *(float*)pi) + else return -1; + } + return 0; +} |