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mpv/mp3lib/decod386.c
diego 0c3d542dc7 WORDS_BIGENDIAN is defined/undefined, not 0/1.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@28374 b3059339-0415-0410-9bf9-f77b7e298cf2
2009-01-26 09:56:27 +00:00

254 lines
6.7 KiB
C

/*
* 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 WORDS_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.
*/
#ifdef WORDS_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