[SCM] libav/experimental: audio format conversion untested and unused

siretart at users.alioth.debian.org siretart at users.alioth.debian.org
Sun Jun 30 15:50:05 UTC 2013


The following commit has been merged in the experimental branch:
commit 0eb6817d9821bf67f33ccfe9b427cf736b95881e
Author: Michael Niedermayer <michaelni at gmx.at>
Date:   Sat Aug 19 20:22:57 2006 +0000

    audio format conversion
    untested and unused
    
    Originally committed as revision 6029 to svn://svn.ffmpeg.org/ffmpeg/trunk

diff --git a/libavcodec/Makefile b/libavcodec/Makefile
index 89f4c40..91a0d5d 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 0000000..56d351c
--- /dev/null
+++ b/libavcodec/audioconvert.c
@@ -0,0 +1,77 @@
+/*
+ * audio conversation
+ * Copyright (c) 2006 Michael Niedermayer <michaelni at 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 at 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;
+}

-- 
Libav/FFmpeg packaging



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