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authorMichael Niedermayer <michaelni@gmx.at>2006-08-19 20:22:57 +0000
committerMichael Niedermayer <michaelni@gmx.at>2006-08-19 20:22:57 +0000
commit0eb6817d9821bf67f33ccfe9b427cf736b95881e (patch)
tree30d8acdcc5cadb03d9ca8b9dcf43e18b8a5279bf /libavcodec/audioconvert.c
parentc52e13f1a154929808508def4ba9d3ba0c7a6fa1 (diff)
downloadffmpeg-0eb6817d9821bf67f33ccfe9b427cf736b95881e.tar.gz
audio format conversion
untested and unused Originally committed as revision 6029 to svn://svn.ffmpeg.org/ffmpeg/trunk
Diffstat (limited to 'libavcodec/audioconvert.c')
-rw-r--r--libavcodec/audioconvert.c77
1 files changed, 77 insertions, 0 deletions
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;
+}