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