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-rw-r--r--mp3lib/decod386.c253
1 files changed, 0 insertions, 253 deletions
diff --git a/mp3lib/decod386.c b/mp3lib/decod386.c
deleted file mode 100644
index 82657aedc3..0000000000
--- a/mp3lib/decod386.c
+++ /dev/null
@@ -1,253 +0,0 @@
-/*
- * Modified for use with MPlayer, for details see the changelog at
- * http://svn.mplayerhq.hu/mplayer/trunk/
- * $Id$
- */
-
-/*
- * Mpeg Layer-1,2,3 audio decoder
- * ------------------------------
- * copyright (c) 1995,1996,1997 by Michael Hipp, All rights reserved.
- * See also 'README'
- *
- * slighlty optimized for machines without autoincrement/decrement.
- * The performance is highly compiler dependend. Maybe
- * the decode.c version for 'normal' processor may be faster
- * even for Intel processors.
- */
-
-
-#include "config.h"
-
-#if 0
- /* old WRITE_SAMPLE */
- /* is portable */
-#define WRITE_SAMPLE(samples,sum,clip) { \
- if( (sum) > 32767.0) { *(samples) = 0x7fff; (clip)++; } \
- else if( (sum) < -32768.0) { *(samples) = -0x8000; (clip)++; }\
- else { *(samples) = sum; } \
-}
-#else
- /* new WRITE_SAMPLE */
-
-/*
- * should be the same as the "old WRITE_SAMPLE" macro above, but uses
- * some tricks to avoid double->int conversions and floating point compares.
- *
- * Here's how it works:
- * ((((65536.0 * 65536.0 * 16)+(65536.0 * 0.5))* 65536.0)) is
- * 0x0010000080000000LL in hex. It computes 0x0010000080000000LL + sum
- * as a double IEEE fp value and extracts the low-order 32-bits from the
- * IEEE fp representation stored in memory. The 2^56 bit in the constant
- * is intended to force the bits of "sum" into the least significant bits
- * of the double mantissa. After an integer substraction of 0x80000000
- * we have the original double value "sum" converted to an 32-bit int value.
- *
- * (Is that really faster than the clean and simple old version of the macro?)
- */
-
-/*
- * On a SPARC cpu, we fetch the low-order 32-bit from the second 32-bit
- * word of the double fp value stored in memory. On an x86 cpu, we fetch it
- * from the first 32-bit word.
- * I'm not sure if the HAVE_BIGENDIAN feature test covers all possible memory
- * layouts of double floating point values an all cpu architectures. If
- * it doesn't work for you, just enable the "old WRITE_SAMPLE" macro.
- */
-#if HAVE_BIGENDIAN
-#define MANTISSA_OFFSET 1
-#else
-#define MANTISSA_OFFSET 0
-#endif
-
- /* sizeof(int) == 4 */
-#define WRITE_SAMPLE(samples,sum,clip) { \
- union { double dtemp; int itemp[2]; } u; int v; \
- u.dtemp = ((((65536.0 * 65536.0 * 16)+(65536.0 * 0.5))* 65536.0)) + (sum);\
- v = u.itemp[MANTISSA_OFFSET] - 0x80000000; \
- if( v > 32767) { *(samples) = 0x7fff; (clip)++; } \
- else if( v < -32768) { *(samples) = -0x8000; (clip)++; } \
- else { *(samples) = v; } \
-}
-#endif
-
-
-/*
-#define WRITE_SAMPLE(samples,sum,clip) { \
- double dtemp; int v; \
- dtemp = ((((65536.0 * 65536.0 * 16)+(65536.0 * 0.5))* 65536.0)) + (sum);\
- v = ((*(int *)&dtemp) - 0x80000000); \
- if( v > 32767) { *(samples) = 0x7fff; (clip)++; } \
- else if( v < -32768) { *(samples) = -0x8000; (clip)++; } \
- else { *(samples) = v; } \
-}
-*/
-
-static int synth_1to1(real *bandPtr,int channel,unsigned char *out,int *pnt);
-
-static int synth_1to1_mono2stereo(real *bandPtr,unsigned char *samples,int *pnt)
-{
- int i,ret;
-
- ret = synth_1to1(bandPtr,0,samples,pnt);
- samples = samples + *pnt - 128;
-
- for(i=0;i<32;i++) {
- ((short *)samples)[1] = ((short *)samples)[0];
- samples+=4;
- }
-
- return ret;
-}
-
-static synth_func_t synth_func;
-
-#if HAVE_ALTIVEC
-#define dct64_base(a,b,c) if(gCpuCaps.hasAltiVec) dct64_altivec(a,b,c); else dct64(a,b,c)
-#else /* HAVE_ALTIVEC */
-#define dct64_base(a,b,c) dct64(a,b,c)
-#endif /* HAVE_ALTIVEC */
-
-static int synth_1to1(real *bandPtr,int channel,unsigned char *out,int *pnt)
-{
- static real buffs[2][2][0x110];
- static const int step = 2;
- static int bo = 1;
- short *samples = (short *) (out + *pnt);
- real *b0,(*buf)[0x110];
- int clip = 0;
- int bo1;
-
- *pnt += 128;
-
-/* optimized for x86 */
-#if ARCH_X86
- if ( synth_func )
- {
-// printf("Calling %p, bandPtr=%p channel=%d samples=%p\n",synth_func,bandPtr,channel,samples);
- // FIXME: synth_func() may destroy EBP, don't rely on stack contents!!!
- return (*synth_func)( bandPtr,channel,samples);
- }
-#endif
- if(!channel) { /* channel=0 */
- bo--;
- bo &= 0xf;
- buf = buffs[0];
- }
- else {
- samples++;
- buf = buffs[1];
- }
-
- if(bo & 0x1) {
- b0 = buf[0];
- bo1 = bo;
- dct64_base(buf[1]+((bo+1)&0xf),buf[0]+bo,bandPtr);
- }
- else {
- b0 = buf[1];
- bo1 = bo+1;
- dct64_base(buf[0]+bo,buf[1]+bo+1,bandPtr);
- }
-
- {
- register int j;
- real *window = mp3lib_decwin + 16 - bo1;
-
- for (j=16;j;j--,b0+=0x10,window+=0x20,samples+=step)
- {
- real sum;
- sum = window[0x0] * b0[0x0];
- sum -= window[0x1] * b0[0x1];
- sum += window[0x2] * b0[0x2];
- sum -= window[0x3] * b0[0x3];
- sum += window[0x4] * b0[0x4];
- sum -= window[0x5] * b0[0x5];
- sum += window[0x6] * b0[0x6];
- sum -= window[0x7] * b0[0x7];
- sum += window[0x8] * b0[0x8];
- sum -= window[0x9] * b0[0x9];
- sum += window[0xA] * b0[0xA];
- sum -= window[0xB] * b0[0xB];
- sum += window[0xC] * b0[0xC];
- sum -= window[0xD] * b0[0xD];
- sum += window[0xE] * b0[0xE];
- sum -= window[0xF] * b0[0xF];
-
- WRITE_SAMPLE(samples,sum,clip);
- }
-
- {
- real sum;
- sum = window[0x0] * b0[0x0];
- sum += window[0x2] * b0[0x2];
- sum += window[0x4] * b0[0x4];
- sum += window[0x6] * b0[0x6];
- sum += window[0x8] * b0[0x8];
- sum += window[0xA] * b0[0xA];
- sum += window[0xC] * b0[0xC];
- sum += window[0xE] * b0[0xE];
- WRITE_SAMPLE(samples,sum,clip);
- b0-=0x10,window-=0x20,samples+=step;
- }
- window += bo1<<1;
-
- for (j=15;j;j--,b0-=0x10,window-=0x20,samples+=step)
- {
- real sum;
- sum = -window[-0x1] * b0[0x0];
- sum -= window[-0x2] * b0[0x1];
- sum -= window[-0x3] * b0[0x2];
- sum -= window[-0x4] * b0[0x3];
- sum -= window[-0x5] * b0[0x4];
- sum -= window[-0x6] * b0[0x5];
- sum -= window[-0x7] * b0[0x6];
- sum -= window[-0x8] * b0[0x7];
- sum -= window[-0x9] * b0[0x8];
- sum -= window[-0xA] * b0[0x9];
- sum -= window[-0xB] * b0[0xA];
- sum -= window[-0xC] * b0[0xB];
- sum -= window[-0xD] * b0[0xC];
- sum -= window[-0xE] * b0[0xD];
- sum -= window[-0xF] * b0[0xE];
- sum -= window[-0x0] * b0[0xF];
-
- WRITE_SAMPLE(samples,sum,clip);
- }
- }
-
- return clip;
-
-}
-
-#ifdef CONFIG_FAKE_MONO
-static int synth_1to1_l(real *bandPtr,int channel,unsigned char *out,int *pnt)
-{
- int i,ret;
-
- ret = synth_1to1(bandPtr,channel,out,pnt);
- out = out + *pnt - 128;
-
- for(i=0;i<32;i++) {
- ((short *)out)[1] = ((short *)out)[0];
- out+=4;
- }
-
- return ret;
-}
-
-static int synth_1to1_r(real *bandPtr,int channel,unsigned char *out,int *pnt)
-{
- int i,ret;
-
- ret = synth_1to1(bandPtr,channel,out,pnt);
- out = out + *pnt - 128;
-
- for(i=0;i<32;i++) {
- ((short *)out)[0] = ((short *)out)[1];
- out+=4;
- }
-
- return ret;
-}
-#endif
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