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mpv/mp3lib/layer2.c
zuxy 739f79a5ff Align output pointer so that we can use movaps instead of movups in dct64_sse;
1.5% faster decode.


git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@23484 b3059339-0415-0410-9bf9-f77b7e298cf2
2007-06-06 05:13:13 +00:00

323 lines
9.6 KiB
C

/*
* Modified for use with MPlayer, for details see the changelog at
* http://svn.mplayerhq.hu/mplayer/trunk/
* $Id$
*/
/*
* Mpeg Layer-2 audio decoder
* --------------------------
* copyright (c) 1995 by Michael Hipp, All rights reserved. See also 'README'
*
*/
//#include "mpg123.h"
#include "l2tables.h"
static int grp_3tab[32 * 3] = { 0, }; /* used: 27 */
static int grp_5tab[128 * 3] = { 0, }; /* used: 125 */
static int grp_9tab[1024 * 3] = { 0, }; /* used: 729 */
static real muls[27][64]; /* also used by layer 1 */
static void init_layer2(void)
{
static double mulmul[27] = {
0.0 , -2.0/3.0 , 2.0/3.0 ,
2.0/7.0 , 2.0/15.0 , 2.0/31.0, 2.0/63.0 , 2.0/127.0 , 2.0/255.0 ,
2.0/511.0 , 2.0/1023.0 , 2.0/2047.0 , 2.0/4095.0 , 2.0/8191.0 ,
2.0/16383.0 , 2.0/32767.0 , 2.0/65535.0 ,
-4.0/5.0 , -2.0/5.0 , 2.0/5.0, 4.0/5.0 ,
-8.0/9.0 , -4.0/9.0 , -2.0/9.0 , 2.0/9.0 , 4.0/9.0 , 8.0/9.0 };
static int base[3][9] = {
{ 1 , 0, 2 , } ,
{ 17, 18, 0 , 19, 20 , } ,
{ 21, 1, 22, 23, 0, 24, 25, 2, 26 } };
int i,j,k,l,len;
real *table;
static int tablen[3] = { 3 , 5 , 9 };
static int *itable,*tables[3] = { grp_3tab , grp_5tab , grp_9tab };
for(i=0;i<3;i++)
{
itable = tables[i];
len = tablen[i];
for(j=0;j<len;j++)
for(k=0;k<len;k++)
for(l=0;l<len;l++)
{
*itable++ = base[i][l];
*itable++ = base[i][k];
*itable++ = base[i][j];
}
}
for(k=0;k<27;k++)
{
double m=mulmul[k];
table = muls[k];
if(_has_mmx)
{
for(j=3,i=0;i<63;i++,j--)
*table++ = 16384 * m * pow(2.0,(double) j / 3.0);
}
else
for(j=3,i=0;i<63;i++,j--)
{
*table++ = m * pow(2.0,(double) j / 3.0);
}
*table++ = 0.0;
}
}
static void II_step_one(unsigned int *bit_alloc,int *scale,struct frame *fr)
{
int stereo = fr->stereo-1;
int sblimit = fr->II_sblimit;
int jsbound = fr->jsbound;
int sblimit2 = fr->II_sblimit<<stereo;
struct al_table *alloc1 = fr->alloc;
int i;
static unsigned int scfsi_buf[64];
unsigned int *scfsi,*bita;
int sc,step;
bita = bit_alloc;
if(stereo)
{
for (i=jsbound;i>0;i--,alloc1+=(1<<step))
{
*bita++ = (char) getbits(step=alloc1->bits);
*bita++ = (char) getbits(step);
}
for (i=sblimit-jsbound;i>0;i--,alloc1+=(1<<step))
{
bita[0] = (char) getbits(step=alloc1->bits);
bita[1] = bita[0];
bita+=2;
}
bita = bit_alloc;
scfsi=scfsi_buf;
for (i=sblimit2;i>0;i--)
if (*bita++)
*scfsi++ = (char) getbits_fast(2);
}
else /* mono */
{
for (i=sblimit;i>0;i--,alloc1+=(1<<step))
*bita++ = (char) getbits(step=alloc1->bits);
bita = bit_alloc;
scfsi=scfsi_buf;
for (i=sblimit;i>0;i--)
if (*bita++)
*scfsi++ = (char) getbits_fast(2);
}
bita = bit_alloc;
scfsi=scfsi_buf;
for (i=sblimit2;i>0;i--)
if (*bita++)
switch (*scfsi++)
{
case 0:
*scale++ = getbits_fast(6);
*scale++ = getbits_fast(6);
*scale++ = getbits_fast(6);
break;
case 1 :
*scale++ = sc = getbits_fast(6);
*scale++ = sc;
*scale++ = getbits_fast(6);
break;
case 2:
*scale++ = sc = getbits_fast(6);
*scale++ = sc;
*scale++ = sc;
break;
default: /* case 3 */
*scale++ = getbits_fast(6);
*scale++ = sc = getbits_fast(6);
*scale++ = sc;
break;
}
}
static void II_step_two(unsigned int *bit_alloc,real fraction[2][4][SBLIMIT],int *scale,struct frame *fr,int x1)
{
int i,j,k,ba;
int stereo = fr->stereo;
int sblimit = fr->II_sblimit;
int jsbound = fr->jsbound;
struct al_table *alloc2,*alloc1 = fr->alloc;
unsigned int *bita=bit_alloc;
int d1,step;
for (i=0;i<jsbound;i++,alloc1+=(1<<step))
{
step = alloc1->bits;
for (j=0;j<stereo;j++)
{
if ( (ba=*bita++) )
{
k=(alloc2 = alloc1+ba)->bits;
if( (d1=alloc2->d) < 0)
{
real cm=muls[k][scale[x1]];
fraction[j][0][i] = ((real) ((int)getbits(k) + d1)) * cm;
fraction[j][1][i] = ((real) ((int)getbits(k) + d1)) * cm;
fraction[j][2][i] = ((real) ((int)getbits(k) + d1)) * cm;
}
else
{
static int *table[] = { 0,0,0,grp_3tab,0,grp_5tab,0,0,0,grp_9tab };
unsigned int idx,*tab,m=scale[x1];
idx = (unsigned int) getbits(k);
tab = (unsigned int *) (table[d1] + idx + idx + idx);
fraction[j][0][i] = muls[*tab++][m];
fraction[j][1][i] = muls[*tab++][m];
fraction[j][2][i] = muls[*tab][m];
}
scale+=3;
}
else
fraction[j][0][i] = fraction[j][1][i] = fraction[j][2][i] = 0.0;
}
}
for (i=jsbound;i<sblimit;i++,alloc1+=(1<<step))
{
step = alloc1->bits;
bita++; /* channel 1 and channel 2 bitalloc are the same */
if ( (ba=*bita++) )
{
k=(alloc2 = alloc1+ba)->bits;
if( (d1=alloc2->d) < 0)
{
real cm;
cm=muls[k][scale[x1+3]];
fraction[1][0][i] = (fraction[0][0][i] = (real) ((int)getbits(k) + d1) ) * cm;
fraction[1][1][i] = (fraction[0][1][i] = (real) ((int)getbits(k) + d1) ) * cm;
fraction[1][2][i] = (fraction[0][2][i] = (real) ((int)getbits(k) + d1) ) * cm;
cm=muls[k][scale[x1]];
fraction[0][0][i] *= cm; fraction[0][1][i] *= cm; fraction[0][2][i] *= cm;
}
else
{
static int *table[] = { 0,0,0,grp_3tab,0,grp_5tab,0,0,0,grp_9tab };
unsigned int idx,*tab,m1,m2;
m1 = scale[x1]; m2 = scale[x1+3];
idx = (unsigned int) getbits(k);
tab = (unsigned int *) (table[d1] + idx + idx + idx);
fraction[0][0][i] = muls[*tab][m1]; fraction[1][0][i] = muls[*tab++][m2];
fraction[0][1][i] = muls[*tab][m1]; fraction[1][1][i] = muls[*tab++][m2];
fraction[0][2][i] = muls[*tab][m1]; fraction[1][2][i] = muls[*tab][m2];
}
scale+=6;
}
else {
fraction[0][0][i] = fraction[0][1][i] = fraction[0][2][i] =
fraction[1][0][i] = fraction[1][1][i] = fraction[1][2][i] = 0.0;
}
/*
should we use individual scalefac for channel 2 or
is the current way the right one , where we just copy channel 1 to
channel 2 ??
The current 'strange' thing is, that we throw away the scalefac
values for the second channel ...!!
-> changed .. now we use the scalefac values of channel one !!
*/
}
if(sblimit > (fr->down_sample_sblimit) )
sblimit = fr->down_sample_sblimit;
for(i=sblimit;i<SBLIMIT;i++)
for (j=0;j<stereo;j++)
fraction[j][0][i] = fraction[j][1][i] = fraction[j][2][i] = 0.0;
}
static void II_select_table(struct frame *fr)
{
static int translate[9][2][16] =
{ { { 0,2,2,2,2,2,2,0,0,0,1,1,1,1,1,0 } , /*44.1 stereo*/
{ 0,2,2,0,0,0,1,1,1,1,1,1,1,1,1,0 } } , /*44.1 mono*/
{ { 0,2,2,2,2,2,2,0,0,0,0,0,0,0,0,0 } , /*48 stereo*/
{ 0,2,2,0,0,0,0,0,0,0,0,0,0,0,0,0 } } , /*48 mono*/
{ { 0,3,3,3,3,3,3,0,0,0,1,1,1,1,1,0 } , /*32 stereo*/
{ 0,3,3,0,0,0,1,1,1,1,1,1,1,1,1,0 } } , /*32 mono*/
{ { 2,2,2,2,2,2,2,3,3,3,3,3,3,3,3,0 } , /*22.05 stereo*/
{ 2,2,2,3,3,3,3,3,3,3,3,3,3,3,3,0 } } , /*22.05 mono*/
{ { 2,2,2,2,2,2,2,3,3,3,3,3,3,3,3,0 } , /*24 stereo*/
{ 2,2,2,3,3,3,3,3,3,3,3,3,3,3,3,0 } } , /*24 mono*/
{ { 2,2,2,2,2,2,2,3,3,3,3,3,3,3,3,0 } , /*16 stereo*/
{ 2,2,2,3,3,3,3,3,3,3,3,3,3,3,3,0 } } , /*16 mono*/
{ { 2,2,2,2,2,2,2,3,3,3,3,3,3,3,3,0 } , /*11.025 stereo*/
{ 2,2,2,3,3,3,3,3,3,3,3,3,3,3,3,0 } } , /*11.025 mono*/
{ { 2,2,2,2,2,2,2,3,3,3,3,3,3,3,3,0 } , /*12 stereo*/
{ 2,2,2,3,3,3,3,3,3,3,3,3,3,3,3,0 } } , /*12 mono*/
{ { 2,2,2,2,2,2,2,3,3,3,3,3,3,3,3,0 } , /*8 stereo*/
{ 2,2,2,3,3,3,3,3,3,3,3,3,3,3,3,0 } } /*8 mono*/
/* 0 48 64 96 128 192 256 384 */
/* 32 56 80 112 160 224 320 XX*/
};
int table,sblim;
static struct al_table *tables[5] =
{ alloc_0, alloc_1, alloc_2, alloc_3 , alloc_4 };
static int sblims[5] = { 27 , 30 , 8, 12 , 30 };
if(fr->lsf)
table = 4;
else
table = translate[fr->sampling_frequency][2-fr->stereo][fr->bitrate_index];
sblim = sblims[table];
fr->alloc = tables[table];
fr->II_sblimit = sblim;
}
static int do_layer2(struct frame *fr,int outmode)
{
int clip=0;
int i,j;
int stereo = fr->stereo;
DECLARE_ALIGNED(16, real, fraction[2][4][SBLIMIT]); /* pick_table clears unused subbands */
unsigned int bit_alloc[64];
int scale[192];
int single = fr->single;
II_select_table(fr);
fr->jsbound = (fr->mode == MPG_MD_JOINT_STEREO) ?
(fr->mode_ext<<2)+4 : fr->II_sblimit;
if(stereo == 1 || single == 3)
single = 0;
II_step_one(bit_alloc, scale, fr);
for (i=0;i<SCALE_BLOCK;i++)
{
II_step_two(bit_alloc,fraction,scale,fr,i>>2);
for (j=0;j<3;j++)
{
if(single >= 0)
{
clip += (fr->synth_mono) (fraction[single][j],pcm_sample,&pcm_point);
}
else {
int p1 = pcm_point;
clip += (fr->synth) (fraction[0][j],0,pcm_sample,&p1);
clip += (fr->synth) (fraction[1][j],1,pcm_sample,&pcm_point);
}
// if(pcm_point >= audiobufsize) audio_flush(outmode,ai);
}
}
return clip;
}