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mpegaudiodec.c
2389 lines (2124 loc) · 72.8 KB
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mpegaudiodec.c
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/*
* MPEG Audio decoder
* Copyright (c) 2001, 2002 Fabrice Bellard
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* MPEG Audio decoder.
*/
#include "avcodec.h"
#include "get_bits.h"
#include "dsputil.h"
/*
* TODO:
* - in low precision mode, use more 16 bit multiplies in synth filter
* - test lsf / mpeg25 extensively.
*/
#include "mpegaudio.h"
#include "mpegaudiodecheader.h"
#include "mathops.h"
#if CONFIG_FLOAT
# define SHR(a,b) ((a)*(1.0f/(1<<(b))))
# define compute_antialias compute_antialias_float
# define FIXR_OLD(a) ((int)((a) * FRAC_ONE + 0.5))
# define FIXR(x) ((float)(x))
# define FIXHR(x) ((float)(x))
# define MULH3(x, y, s) ((s)*(y)*(x))
# define MULLx(x, y, s) ((y)*(x))
# define RENAME(a) a ## _float
#else
# define SHR(a,b) ((a)>>(b))
# define compute_antialias compute_antialias_integer
/* WARNING: only correct for posititive numbers */
# define FIXR_OLD(a) ((int)((a) * FRAC_ONE + 0.5))
# define FIXR(a) ((int)((a) * FRAC_ONE + 0.5))
# define FIXHR(a) ((int)((a) * (1LL<<32) + 0.5))
# define MULH3(x, y, s) MULH((s)*(x), y)
# define MULLx(x, y, s) MULL(x,y,s)
# define RENAME(a) a
#endif
/****************/
#define HEADER_SIZE 4
#include "mpegaudiodata.h"
#include "mpegaudiodectab.h"
#if CONFIG_FLOAT
# include "fft.h"
#else
# include "dct32.c"
#endif
static void compute_antialias(MPADecodeContext *s, GranuleDef *g);
static void apply_window_mp3_c(MPA_INT *synth_buf, MPA_INT *window,
int *dither_state, OUT_INT *samples, int incr);
/* vlc structure for decoding layer 3 huffman tables */
static VLC huff_vlc[16];
static VLC_TYPE huff_vlc_tables[
0+128+128+128+130+128+154+166+
142+204+190+170+542+460+662+414
][2];
static const int huff_vlc_tables_sizes[16] = {
0, 128, 128, 128, 130, 128, 154, 166,
142, 204, 190, 170, 542, 460, 662, 414
};
static VLC huff_quad_vlc[2];
static VLC_TYPE huff_quad_vlc_tables[128+16][2];
static const int huff_quad_vlc_tables_sizes[2] = {
128, 16
};
/* computed from band_size_long */
static uint16_t band_index_long[9][23];
#include "mpegaudio_tablegen.h"
/* intensity stereo coef table */
static INTFLOAT is_table[2][16];
static INTFLOAT is_table_lsf[2][2][16];
static int32_t csa_table[8][4];
static float csa_table_float[8][4];
static INTFLOAT mdct_win[8][36];
static int16_t division_tab3[1<<6 ];
static int16_t division_tab5[1<<8 ];
static int16_t division_tab9[1<<11];
static int16_t * const division_tabs[4] = {
division_tab3, division_tab5, NULL, division_tab9
};
/* lower 2 bits: modulo 3, higher bits: shift */
static uint16_t scale_factor_modshift[64];
/* [i][j]: 2^(-j/3) * FRAC_ONE * 2^(i+2) / (2^(i+2) - 1) */
static int32_t scale_factor_mult[15][3];
/* mult table for layer 2 group quantization */
#define SCALE_GEN(v) \
{ FIXR_OLD(1.0 * (v)), FIXR_OLD(0.7937005259 * (v)), FIXR_OLD(0.6299605249 * (v)) }
static const int32_t scale_factor_mult2[3][3] = {
SCALE_GEN(4.0 / 3.0), /* 3 steps */
SCALE_GEN(4.0 / 5.0), /* 5 steps */
SCALE_GEN(4.0 / 9.0), /* 9 steps */
};
DECLARE_ALIGNED(16, MPA_INT, RENAME(ff_mpa_synth_window))[512+256];
/**
* Convert region offsets to region sizes and truncate
* size to big_values.
*/
static void ff_region_offset2size(GranuleDef *g){
int i, k, j=0;
g->region_size[2] = (576 / 2);
for(i=0;i<3;i++) {
k = FFMIN(g->region_size[i], g->big_values);
g->region_size[i] = k - j;
j = k;
}
}
static void ff_init_short_region(MPADecodeContext *s, GranuleDef *g){
if (g->block_type == 2)
g->region_size[0] = (36 / 2);
else {
if (s->sample_rate_index <= 2)
g->region_size[0] = (36 / 2);
else if (s->sample_rate_index != 8)
g->region_size[0] = (54 / 2);
else
g->region_size[0] = (108 / 2);
}
g->region_size[1] = (576 / 2);
}
static void ff_init_long_region(MPADecodeContext *s, GranuleDef *g, int ra1, int ra2){
int l;
g->region_size[0] =
band_index_long[s->sample_rate_index][ra1 + 1] >> 1;
/* should not overflow */
l = FFMIN(ra1 + ra2 + 2, 22);
g->region_size[1] =
band_index_long[s->sample_rate_index][l] >> 1;
}
static void ff_compute_band_indexes(MPADecodeContext *s, GranuleDef *g){
if (g->block_type == 2) {
if (g->switch_point) {
/* if switched mode, we handle the 36 first samples as
long blocks. For 8000Hz, we handle the 48 first
exponents as long blocks (XXX: check this!) */
if (s->sample_rate_index <= 2)
g->long_end = 8;
else if (s->sample_rate_index != 8)
g->long_end = 6;
else
g->long_end = 4; /* 8000 Hz */
g->short_start = 2 + (s->sample_rate_index != 8);
} else {
g->long_end = 0;
g->short_start = 0;
}
} else {
g->short_start = 13;
g->long_end = 22;
}
}
/* layer 1 unscaling */
/* n = number of bits of the mantissa minus 1 */
static inline int l1_unscale(int n, int mant, int scale_factor)
{
int shift, mod;
int64_t val;
shift = scale_factor_modshift[scale_factor];
mod = shift & 3;
shift >>= 2;
val = MUL64(mant + (-1 << n) + 1, scale_factor_mult[n-1][mod]);
shift += n;
/* NOTE: at this point, 1 <= shift >= 21 + 15 */
return (int)((val + (1LL << (shift - 1))) >> shift);
}
static inline int l2_unscale_group(int steps, int mant, int scale_factor)
{
int shift, mod, val;
shift = scale_factor_modshift[scale_factor];
mod = shift & 3;
shift >>= 2;
val = (mant - (steps >> 1)) * scale_factor_mult2[steps >> 2][mod];
/* NOTE: at this point, 0 <= shift <= 21 */
if (shift > 0)
val = (val + (1 << (shift - 1))) >> shift;
return val;
}
/* compute value^(4/3) * 2^(exponent/4). It normalized to FRAC_BITS */
static inline int l3_unscale(int value, int exponent)
{
unsigned int m;
int e;
e = table_4_3_exp [4*value + (exponent&3)];
m = table_4_3_value[4*value + (exponent&3)];
e -= (exponent >> 2);
assert(e>=1);
if (e > 31)
return 0;
m = (m + (1 << (e-1))) >> e;
return m;
}
/* all integer n^(4/3) computation code */
#define DEV_ORDER 13
#define POW_FRAC_BITS 24
#define POW_FRAC_ONE (1 << POW_FRAC_BITS)
#define POW_FIX(a) ((int)((a) * POW_FRAC_ONE))
#define POW_MULL(a,b) (((int64_t)(a) * (int64_t)(b)) >> POW_FRAC_BITS)
static int dev_4_3_coefs[DEV_ORDER];
#if 0 /* unused */
static int pow_mult3[3] = {
POW_FIX(1.0),
POW_FIX(1.25992104989487316476),
POW_FIX(1.58740105196819947474),
};
#endif
static av_cold void int_pow_init(void)
{
int i, a;
a = POW_FIX(1.0);
for(i=0;i<DEV_ORDER;i++) {
a = POW_MULL(a, POW_FIX(4.0 / 3.0) - i * POW_FIX(1.0)) / (i + 1);
dev_4_3_coefs[i] = a;
}
}
#if 0 /* unused, remove? */
/* return the mantissa and the binary exponent */
static int int_pow(int i, int *exp_ptr)
{
int e, er, eq, j;
int a, a1;
/* renormalize */
a = i;
e = POW_FRAC_BITS;
while (a < (1 << (POW_FRAC_BITS - 1))) {
a = a << 1;
e--;
}
a -= (1 << POW_FRAC_BITS);
a1 = 0;
for(j = DEV_ORDER - 1; j >= 0; j--)
a1 = POW_MULL(a, dev_4_3_coefs[j] + a1);
a = (1 << POW_FRAC_BITS) + a1;
/* exponent compute (exact) */
e = e * 4;
er = e % 3;
eq = e / 3;
a = POW_MULL(a, pow_mult3[er]);
while (a >= 2 * POW_FRAC_ONE) {
a = a >> 1;
eq++;
}
/* convert to float */
while (a < POW_FRAC_ONE) {
a = a << 1;
eq--;
}
/* now POW_FRAC_ONE <= a < 2 * POW_FRAC_ONE */
#if POW_FRAC_BITS > FRAC_BITS
a = (a + (1 << (POW_FRAC_BITS - FRAC_BITS - 1))) >> (POW_FRAC_BITS - FRAC_BITS);
/* correct overflow */
if (a >= 2 * (1 << FRAC_BITS)) {
a = a >> 1;
eq++;
}
#endif
*exp_ptr = eq;
return a;
}
#endif
static av_cold int decode_init(AVCodecContext * avctx)
{
MPADecodeContext *s = avctx->priv_data;
static int init=0;
int i, j, k;
s->avctx = avctx;
s->apply_window_mp3 = apply_window_mp3_c;
#if HAVE_MMX && CONFIG_FLOAT
ff_mpegaudiodec_init_mmx(s);
#endif
#if CONFIG_FLOAT
ff_dct_init(&s->dct, 5, DCT_II);
#endif
if (HAVE_ALTIVEC && CONFIG_FLOAT) ff_mpegaudiodec_init_altivec(s);
avctx->sample_fmt= OUT_FMT;
s->error_recognition= avctx->error_recognition;
if (!init && !avctx->parse_only) {
int offset;
/* scale factors table for layer 1/2 */
for(i=0;i<64;i++) {
int shift, mod;
/* 1.0 (i = 3) is normalized to 2 ^ FRAC_BITS */
shift = (i / 3);
mod = i % 3;
scale_factor_modshift[i] = mod | (shift << 2);
}
/* scale factor multiply for layer 1 */
for(i=0;i<15;i++) {
int n, norm;
n = i + 2;
norm = ((INT64_C(1) << n) * FRAC_ONE) / ((1 << n) - 1);
scale_factor_mult[i][0] = MULLx(norm, FIXR(1.0 * 2.0), FRAC_BITS);
scale_factor_mult[i][1] = MULLx(norm, FIXR(0.7937005259 * 2.0), FRAC_BITS);
scale_factor_mult[i][2] = MULLx(norm, FIXR(0.6299605249 * 2.0), FRAC_BITS);
dprintf(avctx, "%d: norm=%x s=%x %x %x\n",
i, norm,
scale_factor_mult[i][0],
scale_factor_mult[i][1],
scale_factor_mult[i][2]);
}
RENAME(ff_mpa_synth_init)(RENAME(ff_mpa_synth_window));
/* huffman decode tables */
offset = 0;
for(i=1;i<16;i++) {
const HuffTable *h = &mpa_huff_tables[i];
int xsize, x, y;
uint8_t tmp_bits [512];
uint16_t tmp_codes[512];
memset(tmp_bits , 0, sizeof(tmp_bits ));
memset(tmp_codes, 0, sizeof(tmp_codes));
xsize = h->xsize;
j = 0;
for(x=0;x<xsize;x++) {
for(y=0;y<xsize;y++){
tmp_bits [(x << 5) | y | ((x&&y)<<4)]= h->bits [j ];
tmp_codes[(x << 5) | y | ((x&&y)<<4)]= h->codes[j++];
}
}
/* XXX: fail test */
huff_vlc[i].table = huff_vlc_tables+offset;
huff_vlc[i].table_allocated = huff_vlc_tables_sizes[i];
init_vlc(&huff_vlc[i], 7, 512,
tmp_bits, 1, 1, tmp_codes, 2, 2,
INIT_VLC_USE_NEW_STATIC);
offset += huff_vlc_tables_sizes[i];
}
assert(offset == FF_ARRAY_ELEMS(huff_vlc_tables));
offset = 0;
for(i=0;i<2;i++) {
huff_quad_vlc[i].table = huff_quad_vlc_tables+offset;
huff_quad_vlc[i].table_allocated = huff_quad_vlc_tables_sizes[i];
init_vlc(&huff_quad_vlc[i], i == 0 ? 7 : 4, 16,
mpa_quad_bits[i], 1, 1, mpa_quad_codes[i], 1, 1,
INIT_VLC_USE_NEW_STATIC);
offset += huff_quad_vlc_tables_sizes[i];
}
assert(offset == FF_ARRAY_ELEMS(huff_quad_vlc_tables));
for(i=0;i<9;i++) {
k = 0;
for(j=0;j<22;j++) {
band_index_long[i][j] = k;
k += band_size_long[i][j];
}
band_index_long[i][22] = k;
}
/* compute n ^ (4/3) and store it in mantissa/exp format */
int_pow_init();
mpegaudio_tableinit();
for (i = 0; i < 4; i++)
if (ff_mpa_quant_bits[i] < 0)
for (j = 0; j < (1<<(-ff_mpa_quant_bits[i]+1)); j++) {
int val1, val2, val3, steps;
int val = j;
steps = ff_mpa_quant_steps[i];
val1 = val % steps;
val /= steps;
val2 = val % steps;
val3 = val / steps;
division_tabs[i][j] = val1 + (val2 << 4) + (val3 << 8);
}
for(i=0;i<7;i++) {
float f;
INTFLOAT v;
if (i != 6) {
f = tan((double)i * M_PI / 12.0);
v = FIXR(f / (1.0 + f));
} else {
v = FIXR(1.0);
}
is_table[0][i] = v;
is_table[1][6 - i] = v;
}
/* invalid values */
for(i=7;i<16;i++)
is_table[0][i] = is_table[1][i] = 0.0;
for(i=0;i<16;i++) {
double f;
int e, k;
for(j=0;j<2;j++) {
e = -(j + 1) * ((i + 1) >> 1);
f = pow(2.0, e / 4.0);
k = i & 1;
is_table_lsf[j][k ^ 1][i] = FIXR(f);
is_table_lsf[j][k][i] = FIXR(1.0);
dprintf(avctx, "is_table_lsf %d %d: %x %x\n",
i, j, is_table_lsf[j][0][i], is_table_lsf[j][1][i]);
}
}
for(i=0;i<8;i++) {
float ci, cs, ca;
ci = ci_table[i];
cs = 1.0 / sqrt(1.0 + ci * ci);
ca = cs * ci;
csa_table[i][0] = FIXHR(cs/4);
csa_table[i][1] = FIXHR(ca/4);
csa_table[i][2] = FIXHR(ca/4) + FIXHR(cs/4);
csa_table[i][3] = FIXHR(ca/4) - FIXHR(cs/4);
csa_table_float[i][0] = cs;
csa_table_float[i][1] = ca;
csa_table_float[i][2] = ca + cs;
csa_table_float[i][3] = ca - cs;
}
/* compute mdct windows */
for(i=0;i<36;i++) {
for(j=0; j<4; j++){
double d;
if(j==2 && i%3 != 1)
continue;
d= sin(M_PI * (i + 0.5) / 36.0);
if(j==1){
if (i>=30) d= 0;
else if(i>=24) d= sin(M_PI * (i - 18 + 0.5) / 12.0);
else if(i>=18) d= 1;
}else if(j==3){
if (i< 6) d= 0;
else if(i< 12) d= sin(M_PI * (i - 6 + 0.5) / 12.0);
else if(i< 18) d= 1;
}
//merge last stage of imdct into the window coefficients
d*= 0.5 / cos(M_PI*(2*i + 19)/72);
if(j==2)
mdct_win[j][i/3] = FIXHR((d / (1<<5)));
else
mdct_win[j][i ] = FIXHR((d / (1<<5)));
}
}
/* NOTE: we do frequency inversion adter the MDCT by changing
the sign of the right window coefs */
for(j=0;j<4;j++) {
for(i=0;i<36;i+=2) {
mdct_win[j + 4][i] = mdct_win[j][i];
mdct_win[j + 4][i + 1] = -mdct_win[j][i + 1];
}
}
init = 1;
}
if (avctx->codec_id == CODEC_ID_MP3ADU)
s->adu_mode = 1;
return 0;
}
#if CONFIG_FLOAT
static inline float round_sample(float *sum)
{
float sum1=*sum;
*sum = 0;
return sum1;
}
/* signed 16x16 -> 32 multiply add accumulate */
#define MACS(rt, ra, rb) rt+=(ra)*(rb)
/* signed 16x16 -> 32 multiply */
#define MULS(ra, rb) ((ra)*(rb))
#define MLSS(rt, ra, rb) rt-=(ra)*(rb)
#elif FRAC_BITS <= 15
static inline int round_sample(int *sum)
{
int sum1;
sum1 = (*sum) >> OUT_SHIFT;
*sum &= (1<<OUT_SHIFT)-1;
return av_clip(sum1, OUT_MIN, OUT_MAX);
}
/* signed 16x16 -> 32 multiply add accumulate */
#define MACS(rt, ra, rb) MAC16(rt, ra, rb)
/* signed 16x16 -> 32 multiply */
#define MULS(ra, rb) MUL16(ra, rb)
#define MLSS(rt, ra, rb) MLS16(rt, ra, rb)
#else
static inline int round_sample(int64_t *sum)
{
int sum1;
sum1 = (int)((*sum) >> OUT_SHIFT);
*sum &= (1<<OUT_SHIFT)-1;
return av_clip(sum1, OUT_MIN, OUT_MAX);
}
# define MULS(ra, rb) MUL64(ra, rb)
# define MACS(rt, ra, rb) MAC64(rt, ra, rb)
# define MLSS(rt, ra, rb) MLS64(rt, ra, rb)
#endif
#define SUM8(op, sum, w, p) \
{ \
op(sum, (w)[0 * 64], (p)[0 * 64]); \
op(sum, (w)[1 * 64], (p)[1 * 64]); \
op(sum, (w)[2 * 64], (p)[2 * 64]); \
op(sum, (w)[3 * 64], (p)[3 * 64]); \
op(sum, (w)[4 * 64], (p)[4 * 64]); \
op(sum, (w)[5 * 64], (p)[5 * 64]); \
op(sum, (w)[6 * 64], (p)[6 * 64]); \
op(sum, (w)[7 * 64], (p)[7 * 64]); \
}
#define SUM8P2(sum1, op1, sum2, op2, w1, w2, p) \
{ \
INTFLOAT tmp;\
tmp = p[0 * 64];\
op1(sum1, (w1)[0 * 64], tmp);\
op2(sum2, (w2)[0 * 64], tmp);\
tmp = p[1 * 64];\
op1(sum1, (w1)[1 * 64], tmp);\
op2(sum2, (w2)[1 * 64], tmp);\
tmp = p[2 * 64];\
op1(sum1, (w1)[2 * 64], tmp);\
op2(sum2, (w2)[2 * 64], tmp);\
tmp = p[3 * 64];\
op1(sum1, (w1)[3 * 64], tmp);\
op2(sum2, (w2)[3 * 64], tmp);\
tmp = p[4 * 64];\
op1(sum1, (w1)[4 * 64], tmp);\
op2(sum2, (w2)[4 * 64], tmp);\
tmp = p[5 * 64];\
op1(sum1, (w1)[5 * 64], tmp);\
op2(sum2, (w2)[5 * 64], tmp);\
tmp = p[6 * 64];\
op1(sum1, (w1)[6 * 64], tmp);\
op2(sum2, (w2)[6 * 64], tmp);\
tmp = p[7 * 64];\
op1(sum1, (w1)[7 * 64], tmp);\
op2(sum2, (w2)[7 * 64], tmp);\
}
void av_cold RENAME(ff_mpa_synth_init)(MPA_INT *window)
{
int i, j;
/* max = 18760, max sum over all 16 coefs : 44736 */
for(i=0;i<257;i++) {
INTFLOAT v;
v = ff_mpa_enwindow[i];
#if CONFIG_FLOAT
v *= 1.0 / (1LL<<(16 + FRAC_BITS));
#elif WFRAC_BITS < 16
v = (v + (1 << (16 - WFRAC_BITS - 1))) >> (16 - WFRAC_BITS);
#endif
window[i] = v;
if ((i & 63) != 0)
v = -v;
if (i != 0)
window[512 - i] = v;
}
// Needed for avoiding shuffles in ASM implementations
for(i=0; i < 8; i++)
for(j=0; j < 16; j++)
window[512+16*i+j] = window[64*i+32-j];
for(i=0; i < 8; i++)
for(j=0; j < 16; j++)
window[512+128+16*i+j] = window[64*i+48-j];
}
static void apply_window_mp3_c(MPA_INT *synth_buf, MPA_INT *window,
int *dither_state, OUT_INT *samples, int incr)
{
register const MPA_INT *w, *w2, *p;
int j;
OUT_INT *samples2;
#if CONFIG_FLOAT
float sum, sum2;
#elif FRAC_BITS <= 15
int sum, sum2;
#else
int64_t sum, sum2;
#endif
/* copy to avoid wrap */
memcpy(synth_buf + 512, synth_buf, 32 * sizeof(*synth_buf));
samples2 = samples + 31 * incr;
w = window;
w2 = window + 31;
sum = *dither_state;
p = synth_buf + 16;
SUM8(MACS, sum, w, p);
p = synth_buf + 48;
SUM8(MLSS, sum, w + 32, p);
*samples = round_sample(&sum);
samples += incr;
w++;
/* we calculate two samples at the same time to avoid one memory
access per two sample */
for(j=1;j<16;j++) {
sum2 = 0;
p = synth_buf + 16 + j;
SUM8P2(sum, MACS, sum2, MLSS, w, w2, p);
p = synth_buf + 48 - j;
SUM8P2(sum, MLSS, sum2, MLSS, w + 32, w2 + 32, p);
*samples = round_sample(&sum);
samples += incr;
sum += sum2;
*samples2 = round_sample(&sum);
samples2 -= incr;
w++;
w2--;
}
p = synth_buf + 32;
SUM8(MLSS, sum, w + 32, p);
*samples = round_sample(&sum);
*dither_state= sum;
}
/* 32 sub band synthesis filter. Input: 32 sub band samples, Output:
32 samples. */
/* XXX: optimize by avoiding ring buffer usage */
#if !CONFIG_FLOAT
void ff_mpa_synth_filter(MPA_INT *synth_buf_ptr, int *synth_buf_offset,
MPA_INT *window, int *dither_state,
OUT_INT *samples, int incr,
INTFLOAT sb_samples[SBLIMIT])
{
register MPA_INT *synth_buf;
int offset;
#if FRAC_BITS <= 15
int32_t tmp[32];
int j;
#endif
offset = *synth_buf_offset;
synth_buf = synth_buf_ptr + offset;
#if FRAC_BITS <= 15
dct32(tmp, sb_samples);
for(j=0;j<32;j++) {
/* NOTE: can cause a loss in precision if very high amplitude
sound */
synth_buf[j] = av_clip_int16(tmp[j]);
}
#else
dct32(synth_buf, sb_samples);
#endif
apply_window_mp3_c(synth_buf, window, dither_state, samples, incr);
offset = (offset - 32) & 511;
*synth_buf_offset = offset;
}
#endif
#define C3 FIXHR(0.86602540378443864676/2)
/* 0.5 / cos(pi*(2*i+1)/36) */
static const INTFLOAT icos36[9] = {
FIXR(0.50190991877167369479),
FIXR(0.51763809020504152469), //0
FIXR(0.55168895948124587824),
FIXR(0.61038729438072803416),
FIXR(0.70710678118654752439), //1
FIXR(0.87172339781054900991),
FIXR(1.18310079157624925896),
FIXR(1.93185165257813657349), //2
FIXR(5.73685662283492756461),
};
/* 0.5 / cos(pi*(2*i+1)/36) */
static const INTFLOAT icos36h[9] = {
FIXHR(0.50190991877167369479/2),
FIXHR(0.51763809020504152469/2), //0
FIXHR(0.55168895948124587824/2),
FIXHR(0.61038729438072803416/2),
FIXHR(0.70710678118654752439/2), //1
FIXHR(0.87172339781054900991/2),
FIXHR(1.18310079157624925896/4),
FIXHR(1.93185165257813657349/4), //2
// FIXHR(5.73685662283492756461),
};
/* 12 points IMDCT. We compute it "by hand" by factorizing obvious
cases. */
static void imdct12(INTFLOAT *out, INTFLOAT *in)
{
INTFLOAT in0, in1, in2, in3, in4, in5, t1, t2;
in0= in[0*3];
in1= in[1*3] + in[0*3];
in2= in[2*3] + in[1*3];
in3= in[3*3] + in[2*3];
in4= in[4*3] + in[3*3];
in5= in[5*3] + in[4*3];
in5 += in3;
in3 += in1;
in2= MULH3(in2, C3, 2);
in3= MULH3(in3, C3, 4);
t1 = in0 - in4;
t2 = MULH3(in1 - in5, icos36h[4], 2);
out[ 7]=
out[10]= t1 + t2;
out[ 1]=
out[ 4]= t1 - t2;
in0 += SHR(in4, 1);
in4 = in0 + in2;
in5 += 2*in1;
in1 = MULH3(in5 + in3, icos36h[1], 1);
out[ 8]=
out[ 9]= in4 + in1;
out[ 2]=
out[ 3]= in4 - in1;
in0 -= in2;
in5 = MULH3(in5 - in3, icos36h[7], 2);
out[ 0]=
out[ 5]= in0 - in5;
out[ 6]=
out[11]= in0 + in5;
}
/* cos(pi*i/18) */
#define C1 FIXHR(0.98480775301220805936/2)
#define C2 FIXHR(0.93969262078590838405/2)
#define C3 FIXHR(0.86602540378443864676/2)
#define C4 FIXHR(0.76604444311897803520/2)
#define C5 FIXHR(0.64278760968653932632/2)
#define C6 FIXHR(0.5/2)
#define C7 FIXHR(0.34202014332566873304/2)
#define C8 FIXHR(0.17364817766693034885/2)
/* using Lee like decomposition followed by hand coded 9 points DCT */
static void imdct36(INTFLOAT *out, INTFLOAT *buf, INTFLOAT *in, INTFLOAT *win)
{
int i, j;
INTFLOAT t0, t1, t2, t3, s0, s1, s2, s3;
INTFLOAT tmp[18], *tmp1, *in1;
for(i=17;i>=1;i--)
in[i] += in[i-1];
for(i=17;i>=3;i-=2)
in[i] += in[i-2];
for(j=0;j<2;j++) {
tmp1 = tmp + j;
in1 = in + j;
t2 = in1[2*4] + in1[2*8] - in1[2*2];
t3 = in1[2*0] + SHR(in1[2*6],1);
t1 = in1[2*0] - in1[2*6];
tmp1[ 6] = t1 - SHR(t2,1);
tmp1[16] = t1 + t2;
t0 = MULH3(in1[2*2] + in1[2*4] , C2, 2);
t1 = MULH3(in1[2*4] - in1[2*8] , -2*C8, 1);
t2 = MULH3(in1[2*2] + in1[2*8] , -C4, 2);
tmp1[10] = t3 - t0 - t2;
tmp1[ 2] = t3 + t0 + t1;
tmp1[14] = t3 + t2 - t1;
tmp1[ 4] = MULH3(in1[2*5] + in1[2*7] - in1[2*1], -C3, 2);
t2 = MULH3(in1[2*1] + in1[2*5], C1, 2);
t3 = MULH3(in1[2*5] - in1[2*7], -2*C7, 1);
t0 = MULH3(in1[2*3], C3, 2);
t1 = MULH3(in1[2*1] + in1[2*7], -C5, 2);
tmp1[ 0] = t2 + t3 + t0;
tmp1[12] = t2 + t1 - t0;
tmp1[ 8] = t3 - t1 - t0;
}
i = 0;
for(j=0;j<4;j++) {
t0 = tmp[i];
t1 = tmp[i + 2];
s0 = t1 + t0;
s2 = t1 - t0;
t2 = tmp[i + 1];
t3 = tmp[i + 3];
s1 = MULH3(t3 + t2, icos36h[j], 2);
s3 = MULLx(t3 - t2, icos36[8 - j], FRAC_BITS);
t0 = s0 + s1;
t1 = s0 - s1;
out[(9 + j)*SBLIMIT] = MULH3(t1, win[9 + j], 1) + buf[9 + j];
out[(8 - j)*SBLIMIT] = MULH3(t1, win[8 - j], 1) + buf[8 - j];
buf[9 + j] = MULH3(t0, win[18 + 9 + j], 1);
buf[8 - j] = MULH3(t0, win[18 + 8 - j], 1);
t0 = s2 + s3;
t1 = s2 - s3;
out[(9 + 8 - j)*SBLIMIT] = MULH3(t1, win[9 + 8 - j], 1) + buf[9 + 8 - j];
out[( j)*SBLIMIT] = MULH3(t1, win[ j], 1) + buf[ j];
buf[9 + 8 - j] = MULH3(t0, win[18 + 9 + 8 - j], 1);
buf[ + j] = MULH3(t0, win[18 + j], 1);
i += 4;
}
s0 = tmp[16];
s1 = MULH3(tmp[17], icos36h[4], 2);
t0 = s0 + s1;
t1 = s0 - s1;
out[(9 + 4)*SBLIMIT] = MULH3(t1, win[9 + 4], 1) + buf[9 + 4];
out[(8 - 4)*SBLIMIT] = MULH3(t1, win[8 - 4], 1) + buf[8 - 4];
buf[9 + 4] = MULH3(t0, win[18 + 9 + 4], 1);
buf[8 - 4] = MULH3(t0, win[18 + 8 - 4], 1);
}
/* return the number of decoded frames */
static int mp_decode_layer1(MPADecodeContext *s)
{
int bound, i, v, n, ch, j, mant;
uint8_t allocation[MPA_MAX_CHANNELS][SBLIMIT];
uint8_t scale_factors[MPA_MAX_CHANNELS][SBLIMIT];
if (s->mode == MPA_JSTEREO)
bound = (s->mode_ext + 1) * 4;
else
bound = SBLIMIT;
/* allocation bits */
for(i=0;i<bound;i++) {
for(ch=0;ch<s->nb_channels;ch++) {
allocation[ch][i] = get_bits(&s->gb, 4);
}
}
for(i=bound;i<SBLIMIT;i++) {
allocation[0][i] = get_bits(&s->gb, 4);
}
/* scale factors */
for(i=0;i<bound;i++) {
for(ch=0;ch<s->nb_channels;ch++) {
if (allocation[ch][i])
scale_factors[ch][i] = get_bits(&s->gb, 6);
}
}
for(i=bound;i<SBLIMIT;i++) {
if (allocation[0][i]) {
scale_factors[0][i] = get_bits(&s->gb, 6);
scale_factors[1][i] = get_bits(&s->gb, 6);
}
}
/* compute samples */
for(j=0;j<12;j++) {
for(i=0;i<bound;i++) {
for(ch=0;ch<s->nb_channels;ch++) {
n = allocation[ch][i];
if (n) {
mant = get_bits(&s->gb, n + 1);
v = l1_unscale(n, mant, scale_factors[ch][i]);
} else {
v = 0;
}
s->sb_samples[ch][j][i] = v;
}
}
for(i=bound;i<SBLIMIT;i++) {
n = allocation[0][i];
if (n) {
mant = get_bits(&s->gb, n + 1);
v = l1_unscale(n, mant, scale_factors[0][i]);
s->sb_samples[0][j][i] = v;
v = l1_unscale(n, mant, scale_factors[1][i]);
s->sb_samples[1][j][i] = v;
} else {
s->sb_samples[0][j][i] = 0;
s->sb_samples[1][j][i] = 0;
}
}
}
return 12;
}
static int mp_decode_layer2(MPADecodeContext *s)
{
int sblimit; /* number of used subbands */
const unsigned char *alloc_table;
int table, bit_alloc_bits, i, j, ch, bound, v;
unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT];
unsigned char scale_code[MPA_MAX_CHANNELS][SBLIMIT];
unsigned char scale_factors[MPA_MAX_CHANNELS][SBLIMIT][3], *sf;
int scale, qindex, bits, steps, k, l, m, b;
/* select decoding table */
table = ff_mpa_l2_select_table(s->bit_rate / 1000, s->nb_channels,
s->sample_rate, s->lsf);
sblimit = ff_mpa_sblimit_table[table];
alloc_table = ff_mpa_alloc_tables[table];
if (s->mode == MPA_JSTEREO)
bound = (s->mode_ext + 1) * 4;
else
bound = sblimit;
dprintf(s->avctx, "bound=%d sblimit=%d\n", bound, sblimit);
/* sanity check */
if( bound > sblimit ) bound = sblimit;
/* parse bit allocation */
j = 0;
for(i=0;i<bound;i++) {
bit_alloc_bits = alloc_table[j];
for(ch=0;ch<s->nb_channels;ch++) {