mirror of
https://github.com/mpv-player/mpv
synced 2024-12-09 08:29:42 +00:00
0c3d542dc7
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@28374 b3059339-0415-0410-9bf9-f77b7e298cf2
254 lines
6.7 KiB
C
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
|