Chiebot-Mirror
/
FFmpeg
mirror of https://github.com/FFmpeg/FFmpeg.git
8
0
Fork 0
Code Issues Projects Releases Wiki Activity
You can not select more than 25 topics Topics must start with a letter or number, can include dashes ('-') and can be up to 35 characters long.
27546 Commits
37 Branches
400 Tags
465 MiB
 
 
 
 
Tag: Branch: Tree: 3c90abf40f
Branches Tags
${ item.name }
Create tag ${ searchTerm }
Create branch ${ searchTerm }
from '3c90abf40f'
${ noResults }
FFmpeg/libavcodec/mpegaudiodec.c

2388 lines
73 KiB
Raw Blame History

/*
* 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);
av_dlog(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);
av_dlog(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;
av_dlog(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++) {
bit_alloc[ch][i] = get_bits(&s->gb, bit_alloc_bits);
}
j += 1 << bit_alloc_bits;
}
for(i=bound;i<sblimit;i++) {
bit_alloc_bits = alloc_table[j];
v = get_bits(&s->gb, bit_alloc_bits);
bit_alloc[0][i] = v;
bit_alloc[1][i] = v;
j += 1 << bit_alloc_bits;
}
/* scale codes */
for(i=0;i<sblimit;i++) {
for(ch=0;ch<s->nb_channels;ch++) {
if (bit_alloc[ch][i])
scale_code[ch][i] = get_bits(&s->gb, 2);
}
}
/* scale factors */
for(i=0;i<sblimit;i++) {
for(ch=0;ch<s->nb_channels;ch++) {
if (bit_alloc[ch][i]) {
sf = scale_factors[ch][i];
switch(scale_code[ch][i]) {
default:
case 0:
sf[0] = get_bits(&s->gb, 6);
sf[1] = get_bits(&s->gb, 6);
sf[2] = get_bits(&s->gb, 6);
break;
case 2:
sf[0] = get_bits(&s->gb, 6);
sf[1] = sf[0];
sf[2] = sf[0];
break;
case 1:
sf[0] = get_bits(&s->gb, 6);
sf[2] = get_bits(&s->gb, 6);
sf[1] = sf[0];
break;
case 3:
sf[0] = get_bits(&s->gb, 6);
sf[2] = get_bits(&s->gb, 6);
sf[1] = sf[2];
break;
}
}
}
}
/* samples */
for(k=0;k<3;k++) {
for(l=0;l<12;l+=3) {
j = 0;
for(i=0;i<bound;i++) {
bit_alloc_bits = alloc_table[j];
for(ch=0;ch<s->nb_channels;ch++) {
b = bit_alloc[ch][i];
if (b) {
scale = scale_factors[ch][i][k];
qindex = alloc_table[j+b];
bits = ff_mpa_quant_bits[qindex];
if (bits < 0) {
int v2;
/* 3 values at the same time */
v = get_bits(&s->gb, -bits);
v2 = division_tabs[qindex][v];
steps = ff_mpa_quant_steps[qindex];
s->sb_samples[ch][k * 12 + l + 0][i] =
l2_unscale_group(steps, v2 & 15, scale);
s->sb_samples[ch][k * 12 + l + 1][i] =
l2_unscale_group(steps, (v2 >> 4) & 15, scale);
s->sb_samples[ch][k * 12 + l + 2][i] =
l2_unscale_group(steps, v2 >> 8 , scale);
} else {
for(m=0;m<3;m++) {
v = get_bits(&s->gb, bits);
v = l1_unscale(bits - 1, v, scale);
s->sb_samples[ch][k * 12 + l + m][i] = v;
}
}
} else {
s->sb_samples[ch][k * 12 + l + 0][i] = 0;
s->sb_samples[ch][k * 12 + l + 1][i] = 0;
s->sb_samples[ch][k * 12 + l + 2][i] = 0;
}
}
/* next subband in alloc table */
j += 1 << bit_alloc_bits;
}
/* XXX: find a way to avoid this duplication of code */
for(i=bound;i<sblimit;i++) {
bit_alloc_bits = alloc_table[j];
b = bit_alloc[0][i];
if (b) {
int mant, scale0, scale1;
scale0 = scale_factors[0][i][k];
scale1 = scale_factors[1][i][k];
qindex = alloc_table[j+b];
bits = ff_mpa_quant_bits[qindex];
if (bits < 0) {
/* 3 values at the same time */
v = get_bits(&s->gb, -bits);
steps = ff_mpa_quant_steps[qindex];
mant = v % steps;
v = v / steps;
s->sb_samples[0][k * 12 + l + 0][i] =
l2_unscale_group(steps, mant, scale0);
s->sb_samples[1][k * 12 + l + 0][i] =
l2_unscale_group(steps, mant, scale1);
mant = v % steps;
v = v / steps;
s->sb_samples[0][k * 12 + l + 1][i] =
l2_unscale_group(steps, mant, scale0);
s->sb_samples[1][k * 12 + l + 1][i] =
l2_unscale_group(steps, mant, scale1);
s->sb_samples[0][k * 12 + l + 2][i] =
l2_unscale_group(steps, v, scale0);
s->sb_samples[1][k * 12 + l + 2][i] =
l2_unscale_group(steps, v, scale1);
} else {
for(m=0;m<3;m++) {
mant = get_bits(&s->gb, bits);
s->sb_samples[0][k * 12 + l + m][i] =
l1_unscale(bits - 1, mant, scale0);
s->sb_samples[1][k * 12 + l + m][i] =
l1_unscale(bits - 1, mant, scale1);
}
}
} else {
s->sb_samples[0][k * 12 + l + 0][i] = 0;
s->sb_samples[0][k * 12 + l + 1][i] = 0;
s->sb_samples[0][k * 12 + l + 2][i] = 0;
s->sb_samples[1][k * 12 + l + 0][i] = 0;
s->sb_samples[1][k * 12 + l + 1][i] = 0;
s->sb_samples[1][k * 12 + l + 2][i] = 0;
}
/* next subband in alloc table */
j += 1 << bit_alloc_bits;
}
/* fill remaining samples to zero */
for(i=sblimit;i<SBLIMIT;i++) {
for(ch=0;ch<s->nb_channels;ch++) {
s->sb_samples[ch][k * 12 + l + 0][i] = 0;
s->sb_samples[ch][k * 12 + l + 1][i] = 0;
s->sb_samples[ch][k * 12 + l + 2][i] = 0;
}
}
}
}
return 3 * 12;
}
#define SPLIT(dst,sf,n)\
if(n==3){\
int m= (sf*171)>>9;\
dst= sf - 3*m;\
sf=m;\
}else if(n==4){\
dst= sf&3;\
sf>>=2;\
}else if(n==5){\
int m= (sf*205)>>10;\
dst= sf - 5*m;\
sf=m;\
}else if(n==6){\
int m= (sf*171)>>10;\
dst= sf - 6*m;\
sf=m;\
}else{\
dst=0;\
}
static av_always_inline void lsf_sf_expand(int *slen,
int sf, int n1, int n2, int n3)
{
SPLIT(slen[3], sf, n3)
SPLIT(slen[2], sf, n2)
SPLIT(slen[1], sf, n1)
slen[0] = sf;
}
static void exponents_from_scale_factors(MPADecodeContext *s,
GranuleDef *g,
int16_t *exponents)
{
const uint8_t *bstab, *pretab;
int len, i, j, k, l, v0, shift, gain, gains[3];
int16_t *exp_ptr;
exp_ptr = exponents;
gain = g->global_gain - 210;
shift = g->scalefac_scale + 1;
bstab = band_size_long[s->sample_rate_index];
pretab = mpa_pretab[g->preflag];
for(i=0;i<g->long_end;i++) {
v0 = gain - ((g->scale_factors[i] + pretab[i]) << shift) + 400;
len = bstab[i];
for(j=len;j>0;j--)
*exp_ptr++ = v0;
}
if (g->short_start < 13) {
bstab = band_size_short[s->sample_rate_index];
gains[0] = gain - (g->subblock_gain[0] << 3);
gains[1] = gain - (g->subblock_gain[1] << 3);
gains[2] = gain - (g->subblock_gain[2] << 3);
k = g->long_end;
for(i=g->short_start;i<13;i++) {
len = bstab[i];
for(l=0;l<3;l++) {
v0 = gains[l] - (g->scale_factors[k++] << shift) + 400;
for(j=len;j>0;j--)
*exp_ptr++ = v0;
}
}
}
}
/* handle n = 0 too */
static inline int get_bitsz(GetBitContext *s, int n)
{
if (n == 0)
return 0;
else
return get_bits(s, n);
}
static void switch_buffer(MPADecodeContext *s, int *pos, int *end_pos, int *end_pos2){
if(s->in_gb.buffer && *pos >= s->gb.size_in_bits){
s->gb= s->in_gb;
s->in_gb.buffer=NULL;
assert((get_bits_count(&s->gb) & 7) == 0);
skip_bits_long(&s->gb, *pos - *end_pos);
*end_pos2=
*end_pos= *end_pos2 + get_bits_count(&s->gb) - *pos;
*pos= get_bits_count(&s->gb);
}
}
/* Following is a optimized code for
INTFLOAT v = *src
if(get_bits1(&s->gb))
v = -v;
*dst = v;
*/
#if CONFIG_FLOAT
#define READ_FLIP_SIGN(dst,src)\
v = AV_RN32A(src) ^ (get_bits1(&s->gb)<<31);\
AV_WN32A(dst, v);
#else
#define READ_FLIP_SIGN(dst,src)\
v= -get_bits1(&s->gb);\
*(dst) = (*(src) ^ v) - v;
#endif
static int huffman_decode(MPADecodeContext *s, GranuleDef *g,
int16_t *exponents, int end_pos2)
{
int s_index;
int i;
int last_pos, bits_left;
VLC *vlc;
int end_pos= FFMIN(end_pos2, s->gb.size_in_bits);
/* low frequencies (called big values) */
s_index = 0;
for(i=0;i<3;i++) {
int j, k, l, linbits;
j = g->region_size[i];
if (j == 0)
continue;
/* select vlc table */
k = g->table_select[i];
l = mpa_huff_data[k][0];
linbits = mpa_huff_data[k][1];
vlc = &huff_vlc[l];
if(!l){
memset(&g->sb_hybrid[s_index], 0, sizeof(*g->sb_hybrid)*2*j);
s_index += 2*j;
continue;
}
/* read huffcode and compute each couple */
for(;j>0;j--) {
int exponent, x, y;
int v;
int pos= get_bits_count(&s->gb);
if (pos >= end_pos){
// av_log(NULL, AV_LOG_ERROR, "pos: %d %d %d %d\n", pos, end_pos, end_pos2, s_index);
switch_buffer(s, &pos, &end_pos, &end_pos2);
// av_log(NULL, AV_LOG_ERROR, "new pos: %d %d\n", pos, end_pos);
if(pos >= end_pos)
break;
}
y = get_vlc2(&s->gb, vlc->table, 7, 3);
if(!y){
g->sb_hybrid[s_index ] =
g->sb_hybrid[s_index+1] = 0;
s_index += 2;
continue;
}
exponent= exponents[s_index];
av_dlog(s->avctx, "region=%d n=%d x=%d y=%d exp=%d\n",
i, g->region_size[i] - j, x, y, exponent);
if(y&16){
x = y >> 5;
y = y & 0x0f;
if (x < 15){
READ_FLIP_SIGN(g->sb_hybrid+s_index, RENAME(expval_table)[ exponent ]+x)
}else{
x += get_bitsz(&s->gb, linbits);
v = l3_unscale(x, exponent);
if (get_bits1(&s->gb))
v = -v;
g->sb_hybrid[s_index] = v;
}
if (y < 15){
READ_FLIP_SIGN(g->sb_hybrid+s_index+1, RENAME(expval_table)[ exponent ]+y)
}else{
y += get_bitsz(&s->gb, linbits);
v = l3_unscale(y, exponent);
if (get_bits1(&s->gb))
v = -v;
g->sb_hybrid[s_index+1] = v;
}
}else{
x = y >> 5;
y = y & 0x0f;
x += y;
if (x < 15){
READ_FLIP_SIGN(g->sb_hybrid+s_index+!!y, RENAME(expval_table)[ exponent ]+x)
}else{
x += get_bitsz(&s->gb, linbits);
v = l3_unscale(x, exponent);
if (get_bits1(&s->gb))
v = -v;
g->sb_hybrid[s_index+!!y] = v;
}
g->sb_hybrid[s_index+ !y] = 0;
}
s_index+=2;
}
}
/* high frequencies */
vlc = &huff_quad_vlc[g->count1table_select];
last_pos=0;
while (s_index <= 572) {
int pos, code;
pos = get_bits_count(&s->gb);
if (pos >= end_pos) {
if (pos > end_pos2 && last_pos){
/* some encoders generate an incorrect size for this
part. We must go back into the data */
s_index -= 4;
skip_bits_long(&s->gb, last_pos - pos);
av_log(s->avctx, AV_LOG_INFO, "overread, skip %d enddists: %d %d\n", last_pos - pos, end_pos-pos, end_pos2-pos);
if(s->error_recognition >= FF_ER_COMPLIANT)
s_index=0;
break;
}
// av_log(NULL, AV_LOG_ERROR, "pos2: %d %d %d %d\n", pos, end_pos, end_pos2, s_index);
switch_buffer(s, &pos, &end_pos, &end_pos2);
// av_log(NULL, AV_LOG_ERROR, "new pos2: %d %d %d\n", pos, end_pos, s_index);
if(pos >= end_pos)
break;
}
last_pos= pos;
code = get_vlc2(&s->gb, vlc->table, vlc->bits, 1);
av_dlog(s->avctx, "t=%d code=%d\n", g->count1table_select, code);
g->sb_hybrid[s_index+0]=
g->sb_hybrid[s_index+1]=
g->sb_hybrid[s_index+2]=
g->sb_hybrid[s_index+3]= 0;
while(code){
static const int idxtab[16]={3,3,2,2,1,1,1,1,0,0,0,0,0,0,0,0};
int v;
int pos= s_index+idxtab[code];
code ^= 8>>idxtab[code];
READ_FLIP_SIGN(g->sb_hybrid+pos, RENAME(exp_table)+exponents[pos])
}
s_index+=4;
}
/* skip extension bits */
bits_left = end_pos2 - get_bits_count(&s->gb);
//av_log(NULL, AV_LOG_ERROR, "left:%d buf:%p\n", bits_left, s->in_gb.buffer);
if (bits_left < 0 && s->error_recognition >= FF_ER_COMPLIANT) {
av_log(s->avctx, AV_LOG_ERROR, "bits_left=%d\n", bits_left);
s_index=0;
}else if(bits_left > 0 && s->error_recognition >= FF_ER_AGGRESSIVE){
av_log(s->avctx, AV_LOG_ERROR, "bits_left=%d\n", bits_left);
s_index=0;
}
memset(&g->sb_hybrid[s_index], 0, sizeof(*g->sb_hybrid)*(576 - s_index));
skip_bits_long(&s->gb, bits_left);
i= get_bits_count(&s->gb);
switch_buffer(s, &i, &end_pos, &end_pos2);
return 0;
}
/* Reorder short blocks from bitstream order to interleaved order. It
would be faster to do it in parsing, but the code would be far more
complicated */
static void reorder_block(MPADecodeContext *s, GranuleDef *g)
{
int i, j, len;
INTFLOAT *ptr, *dst, *ptr1;
INTFLOAT tmp[576];
if (g->block_type != 2)
return;
if (g->switch_point) {
if (s->sample_rate_index != 8) {
ptr = g->sb_hybrid + 36;
} else {
ptr = g->sb_hybrid + 48;
}
} else {
ptr = g->sb_hybrid;
}
for(i=g->short_start;i<13;i++) {
len = band_size_short[s->sample_rate_index][i];
ptr1 = ptr;
dst = tmp;
for(j=len;j>0;j--) {
*dst++ = ptr[0*len];
*dst++ = ptr[1*len];
*dst++ = ptr[2*len];
ptr++;
}
ptr+=2*len;
memcpy(ptr1, tmp, len * 3 * sizeof(*ptr1));
}
}
#define ISQRT2 FIXR(0.70710678118654752440)
static void compute_stereo(MPADecodeContext *s,
GranuleDef *g0, GranuleDef *g1)
{
int i, j, k, l;
int sf_max, sf, len, non_zero_found;
INTFLOAT (*is_tab)[16], *tab0, *tab1, tmp0, tmp1, v1, v2;
int non_zero_found_short[3];
/* intensity stereo */
if (s->mode_ext & MODE_EXT_I_STEREO) {
if (!s->lsf) {
is_tab = is_table;
sf_max = 7;
} else {
is_tab = is_table_lsf[g1->scalefac_compress & 1];
sf_max = 16;
}
tab0 = g0->sb_hybrid + 576;
tab1 = g1->sb_hybrid + 576;
non_zero_found_short[0] = 0;
non_zero_found_short[1] = 0;
non_zero_found_short[2] = 0;
k = (13 - g1->short_start) * 3 + g1->long_end - 3;
for(i = 12;i >= g1->short_start;i--) {
/* for last band, use previous scale factor */
if (i != 11)
k -= 3;
len = band_size_short[s->sample_rate_index][i];
for(l=2;l>=0;l--) {
tab0 -= len;
tab1 -= len;
if (!non_zero_found_short[l]) {
/* test if non zero band. if so, stop doing i-stereo */
for(j=0;j<len;j++) {
if (tab1[j] != 0) {
non_zero_found_short[l] = 1;
goto found1;
}
}
sf = g1->scale_factors[k + l];
if (sf >= sf_max)
goto found1;
v1 = is_tab[0][sf];
v2 = is_tab[1][sf];
for(j=0;j<len;j++) {
tmp0 = tab0[j];
tab0[j] = MULLx(tmp0, v1, FRAC_BITS);
tab1[j] = MULLx(tmp0, v2, FRAC_BITS);
}
} else {
found1:
if (s->mode_ext & MODE_EXT_MS_STEREO) {
/* lower part of the spectrum : do ms stereo
if enabled */
for(j=0;j<len;j++) {
tmp0 = tab0[j];
tmp1 = tab1[j];
tab0[j] = MULLx(tmp0 + tmp1, ISQRT2, FRAC_BITS);
tab1[j] = MULLx(tmp0 - tmp1, ISQRT2, FRAC_BITS);
}
}
}
}
}
non_zero_found = non_zero_found_short[0] |
non_zero_found_short[1] |
non_zero_found_short[2];
for(i = g1->long_end - 1;i >= 0;i--) {
len = band_size_long[s->sample_rate_index][i];
tab0 -= len;
tab1 -= len;
/* test if non zero band. if so, stop doing i-stereo */
if (!non_zero_found) {
for(j=0;j<len;j++) {
if (tab1[j] != 0) {
non_zero_found = 1;
goto found2;
}
}
/* for last band, use previous scale factor */
k = (i == 21) ? 20 : i;
sf = g1->scale_factors[k];
if (sf >= sf_max)
goto found2;
v1 = is_tab[0][sf];
v2 = is_tab[1][sf];
for(j=0;j<len;j++) {
tmp0 = tab0[j];
tab0[j] = MULLx(tmp0, v1, FRAC_BITS);
tab1[j] = MULLx(tmp0, v2, FRAC_BITS);
}
} else {
found2:
if (s->mode_ext & MODE_EXT_MS_STEREO) {
/* lower part of the spectrum : do ms stereo
if enabled */
for(j=0;j<len;j++) {
tmp0 = tab0[j];
tmp1 = tab1[j];
tab0[j] = MULLx(tmp0 + tmp1, ISQRT2, FRAC_BITS);
tab1[j] = MULLx(tmp0 - tmp1, ISQRT2, FRAC_BITS);
}
}
}
}
} else if (s->mode_ext & MODE_EXT_MS_STEREO) {
/* ms stereo ONLY */
/* NOTE: the 1/sqrt(2) normalization factor is included in the
global gain */
tab0 = g0->sb_hybrid;
tab1 = g1->sb_hybrid;
for(i=0;i<576;i++) {
tmp0 = tab0[i];
tmp1 = tab1[i];
tab0[i] = tmp0 + tmp1;
tab1[i] = tmp0 - tmp1;
}
}
}
#if !CONFIG_FLOAT
static void compute_antialias_integer(MPADecodeContext *s,
GranuleDef *g)
{
int32_t *ptr, *csa;
int n, i;
/* we antialias only "long" bands */
if (g->block_type == 2) {
if (!g->switch_point)
return;
/* XXX: check this for 8000Hz case */
n = 1;
} else {
n = SBLIMIT - 1;
}
ptr = g->sb_hybrid + 18;
for(i = n;i > 0;i--) {
int tmp0, tmp1, tmp2;
csa = &csa_table[0][0];
#define INT_AA(j) \
tmp0 = ptr[-1-j];\
tmp1 = ptr[ j];\
tmp2= MULH(tmp0 + tmp1, csa[0+4*j]);\
ptr[-1-j] = 4*(tmp2 - MULH(tmp1, csa[2+4*j]));\
ptr[ j] = 4*(tmp2 + MULH(tmp0, csa[3+4*j]));
INT_AA(0)
INT_AA(1)
INT_AA(2)
INT_AA(3)
INT_AA(4)
INT_AA(5)
INT_AA(6)
INT_AA(7)
ptr += 18;
}
}
#endif
static void compute_imdct(MPADecodeContext *s,
GranuleDef *g,
INTFLOAT *sb_samples,
INTFLOAT *mdct_buf)
{
INTFLOAT *win, *win1, *out_ptr, *ptr, *buf, *ptr1;
INTFLOAT out2[12];
int i, j, mdct_long_end, sblimit;
/* find last non zero block */
ptr = g->sb_hybrid + 576;
ptr1 = g->sb_hybrid + 2 * 18;
while (ptr >= ptr1) {
int32_t *p;
ptr -= 6;
p= (int32_t*)ptr;
if(p[0] | p[1] | p[2] | p[3] | p[4] | p[5])
break;
}
sblimit = ((ptr - g->sb_hybrid) / 18) + 1;
if (g->block_type == 2) {
/* XXX: check for 8000 Hz */
if (g->switch_point)
mdct_long_end = 2;
else
mdct_long_end = 0;
} else {
mdct_long_end = sblimit;
}
buf = mdct_buf;
ptr = g->sb_hybrid;
for(j=0;j<mdct_long_end;j++) {
/* apply window & overlap with previous buffer */
out_ptr = sb_samples + j;
/* select window */
if (g->switch_point && j < 2)
win1 = mdct_win[0];
else
win1 = mdct_win[g->block_type];
/* select frequency inversion */
win = win1 + ((4 * 36) & -(j & 1));
imdct36(out_ptr, buf, ptr, win);
out_ptr += 18*SBLIMIT;
ptr += 18;
buf += 18;
}
for(j=mdct_long_end;j<sblimit;j++) {
/* select frequency inversion */
win = mdct_win[2] + ((4 * 36) & -(j & 1));
out_ptr = sb_samples + j;
for(i=0; i<6; i++){
*out_ptr = buf[i];
out_ptr += SBLIMIT;
}
imdct12(out2, ptr + 0);
for(i=0;i<6;i++) {
*out_ptr = MULH3(out2[i ], win[i ], 1) + buf[i + 6*1];
buf[i + 6*2] = MULH3(out2[i + 6], win[i + 6], 1);
out_ptr += SBLIMIT;
}
imdct12(out2, ptr + 1);
for(i=0;i<6;i++) {
*out_ptr = MULH3(out2[i ], win[i ], 1) + buf[i + 6*2];
buf[i + 6*0] = MULH3(out2[i + 6], win[i + 6], 1);
out_ptr += SBLIMIT;
}
imdct12(out2, ptr + 2);
for(i=0;i<6;i++) {
buf[i + 6*0] = MULH3(out2[i ], win[i ], 1) + buf[i + 6*0];
buf[i + 6*1] = MULH3(out2[i + 6], win[i + 6], 1);
buf[i + 6*2] = 0;
}
ptr += 18;
buf += 18;
}
/* zero bands */
for(j=sblimit;j<SBLIMIT;j++) {
/* overlap */
out_ptr = sb_samples + j;
for(i=0;i<18;i++) {
*out_ptr = buf[i];
buf[i] = 0;
out_ptr += SBLIMIT;
}
buf += 18;
}
}
/* main layer3 decoding function */
static int mp_decode_layer3(MPADecodeContext *s)
{
int nb_granules, main_data_begin, private_bits;
int gr, ch, blocksplit_flag, i, j, k, n, bits_pos;
GranuleDef *g;
int16_t exponents[576]; //FIXME try INTFLOAT
/* read side info */
if (s->lsf) {
main_data_begin = get_bits(&s->gb, 8);
private_bits = get_bits(&s->gb, s->nb_channels);
nb_granules = 1;
} else {
main_data_begin = get_bits(&s->gb, 9);
if (s->nb_channels == 2)
private_bits = get_bits(&s->gb, 3);
else
private_bits = get_bits(&s->gb, 5);
nb_granules = 2;
for(ch=0;ch<s->nb_channels;ch++) {
s->granules[ch][0].scfsi = 0;/* all scale factors are transmitted */
s->granules[ch][1].scfsi = get_bits(&s->gb, 4);
}
}
for(gr=0;gr<nb_granules;gr++) {
for(ch=0;ch<s->nb_channels;ch++) {
av_dlog(s->avctx, "gr=%d ch=%d: side_info\n", gr, ch);
g = &s->granules[ch][gr];
g->part2_3_length = get_bits(&s->gb, 12);
g->big_values = get_bits(&s->gb, 9);
if(g->big_values > 288){
av_log(s->avctx, AV_LOG_ERROR, "big_values too big\n");
return -1;
}
g->global_gain = get_bits(&s->gb, 8);
/* if MS stereo only is selected, we precompute the
1/sqrt(2) renormalization factor */
if ((s->mode_ext & (MODE_EXT_MS_STEREO | MODE_EXT_I_STEREO)) ==
MODE_EXT_MS_STEREO)
g->global_gain -= 2;
if (s->lsf)
g->scalefac_compress = get_bits(&s->gb, 9);
else
g->scalefac_compress = get_bits(&s->gb, 4);
blocksplit_flag = get_bits1(&s->gb);
if (blocksplit_flag) {
g->block_type = get_bits(&s->gb, 2);
if (g->block_type == 0){
av_log(s->avctx, AV_LOG_ERROR, "invalid block type\n");
return -1;
}
g->switch_point = get_bits1(&s->gb);
for(i=0;i<2;i++)
g->table_select[i] = get_bits(&s->gb, 5);
for(i=0;i<3;i++)
g->subblock_gain[i] = get_bits(&s->gb, 3);
ff_init_short_region(s, g);
} else {
int region_address1, region_address2;
g->block_type = 0;
g->switch_point = 0;
for(i=0;i<3;i++)
g->table_select[i] = get_bits(&s->gb, 5);
/* compute huffman coded region sizes */
region_address1 = get_bits(&s->gb, 4);
region_address2 = get_bits(&s->gb, 3);
av_dlog(s->avctx, "region1=%d region2=%d\n",
region_address1, region_address2);
ff_init_long_region(s, g, region_address1, region_address2);
}
ff_region_offset2size(g);
ff_compute_band_indexes(s, g);
g->preflag = 0;
if (!s->lsf)
g->preflag = get_bits1(&s->gb);
g->scalefac_scale = get_bits1(&s->gb);
g->count1table_select = get_bits1(&s->gb);
av_dlog(s->avctx, "block_type=%d switch_point=%d\n",
g->block_type, g->switch_point);
}
}
if (!s->adu_mode) {
const uint8_t *ptr = s->gb.buffer + (get_bits_count(&s->gb)>>3);
assert((get_bits_count(&s->gb) & 7) == 0);
/* now we get bits from the main_data_begin offset */
av_dlog(s->avctx, "seekback: %d\n", main_data_begin);
//av_log(NULL, AV_LOG_ERROR, "backstep:%d, lastbuf:%d\n", main_data_begin, s->last_buf_size);
memcpy(s->last_buf + s->last_buf_size, ptr, EXTRABYTES);
s->in_gb= s->gb;
init_get_bits(&s->gb, s->last_buf, s->last_buf_size*8);
skip_bits_long(&s->gb, 8*(s->last_buf_size - main_data_begin));
}
for(gr=0;gr<nb_granules;gr++) {
for(ch=0;ch<s->nb_channels;ch++) {
g = &s->granules[ch][gr];
if(get_bits_count(&s->gb)<0){
av_log(s->avctx, AV_LOG_DEBUG, "mdb:%d, lastbuf:%d skipping granule %d\n",
main_data_begin, s->last_buf_size, gr);
skip_bits_long(&s->gb, g->part2_3_length);
memset(g->sb_hybrid, 0, sizeof(g->sb_hybrid));
if(get_bits_count(&s->gb) >= s->gb.size_in_bits && s->in_gb.buffer){
skip_bits_long(&s->in_gb, get_bits_count(&s->gb) - s->gb.size_in_bits);
s->gb= s->in_gb;
s->in_gb.buffer=NULL;
}
continue;
}
bits_pos = get_bits_count(&s->gb);
if (!s->lsf) {
uint8_t *sc;
int slen, slen1, slen2;
/* MPEG1 scale factors */
slen1 = slen_table[0][g->scalefac_compress];
slen2 = slen_table[1][g->scalefac_compress];
av_dlog(s->avctx, "slen1=%d slen2=%d\n", slen1, slen2);
if (g->block_type == 2) {
n = g->switch_point ? 17 : 18;
j = 0;
if(slen1){
for(i=0;i<n;i++)
g->scale_factors[j++] = get_bits(&s->gb, slen1);
}else{
for(i=0;i<n;i++)
g->scale_factors[j++] = 0;
}
if(slen2){
for(i=0;i<18;i++)
g->scale_factors[j++] = get_bits(&s->gb, slen2);
for(i=0;i<3;i++)
g->scale_factors[j++] = 0;
}else{
for(i=0;i<21;i++)
g->scale_factors[j++] = 0;
}
} else {
sc = s->granules[ch][0].scale_factors;
j = 0;
for(k=0;k<4;k++) {
n = (k == 0 ? 6 : 5);
if ((g->scfsi & (0x8 >> k)) == 0) {
slen = (k < 2) ? slen1 : slen2;
if(slen){
for(i=0;i<n;i++)
g->scale_factors[j++] = get_bits(&s->gb, slen);
}else{
for(i=0;i<n;i++)
g->scale_factors[j++] = 0;
}
} else {
/* simply copy from last granule */
for(i=0;i<n;i++) {
g->scale_factors[j] = sc[j];
j++;
}
}
}
g->scale_factors[j++] = 0;
}
} else {
int tindex, tindex2, slen[4], sl, sf;
/* LSF scale factors */
if (g->block_type == 2) {
tindex = g->switch_point ? 2 : 1;
} else {
tindex = 0;
}
sf = g->scalefac_compress;
if ((s->mode_ext & MODE_EXT_I_STEREO) && ch == 1) {
/* intensity stereo case */
sf >>= 1;
if (sf < 180) {
lsf_sf_expand(slen, sf, 6, 6, 0);
tindex2 = 3;
} else if (sf < 244) {
lsf_sf_expand(slen, sf - 180, 4, 4, 0);
tindex2 = 4;
} else {
lsf_sf_expand(slen, sf - 244, 3, 0, 0);
tindex2 = 5;
}
} else {
/* normal case */
if (sf < 400) {
lsf_sf_expand(slen, sf, 5, 4, 4);
tindex2 = 0;
} else if (sf < 500) {
lsf_sf_expand(slen, sf - 400, 5, 4, 0);
tindex2 = 1;
} else {
lsf_sf_expand(slen, sf - 500, 3, 0, 0);
tindex2 = 2;
g->preflag = 1;
}
}
j = 0;
for(k=0;k<4;k++) {
n = lsf_nsf_table[tindex2][tindex][k];
sl = slen[k];
if(sl){
for(i=0;i<n;i++)
g->scale_factors[j++] = get_bits(&s->gb, sl);
}else{
for(i=0;i<n;i++)
g->scale_factors[j++] = 0;
}
}
/* XXX: should compute exact size */
for(;j<40;j++)
g->scale_factors[j] = 0;
}
exponents_from_scale_factors(s, g, exponents);
/* read Huffman coded residue */
huffman_decode(s, g, exponents, bits_pos + g->part2_3_length);
} /* ch */
if (s->nb_channels == 2)
compute_stereo(s, &s->granules[0][gr], &s->granules[1][gr]);
for(ch=0;ch<s->nb_channels;ch++) {
g = &s->granules[ch][gr];
reorder_block(s, g);
compute_antialias(s, g);
compute_imdct(s, g, &s->sb_samples[ch][18 * gr][0], s->mdct_buf[ch]);
}
} /* gr */
if(get_bits_count(&s->gb)<0)
skip_bits_long(&s->gb, -get_bits_count(&s->gb));
return nb_granules * 18;
}
static int mp_decode_frame(MPADecodeContext *s,
OUT_INT *samples, const uint8_t *buf, int buf_size)
{
int i, nb_frames, ch;
OUT_INT *samples_ptr;
init_get_bits(&s->gb, buf + HEADER_SIZE, (buf_size - HEADER_SIZE)*8);
/* skip error protection field */
if (s->error_protection)
skip_bits(&s->gb, 16);
av_dlog(s->avctx, "frame %d:\n", s->frame_count);
switch(s->layer) {
case 1:
s->avctx->frame_size = 384;
nb_frames = mp_decode_layer1(s);
break;
case 2:
s->avctx->frame_size = 1152;
nb_frames = mp_decode_layer2(s);
break;
case 3:
s->avctx->frame_size = s->lsf ? 576 : 1152;
default:
nb_frames = mp_decode_layer3(s);
s->last_buf_size=0;
if(s->in_gb.buffer){
align_get_bits(&s->gb);
i= get_bits_left(&s->gb)>>3;
if(i >= 0 && i <= BACKSTEP_SIZE){
memmove(s->last_buf, s->gb.buffer + (get_bits_count(&s->gb)>>3), i);
s->last_buf_size=i;
}else
av_log(s->avctx, AV_LOG_ERROR, "invalid old backstep %d\n", i);
s->gb= s->in_gb;
s->in_gb.buffer= NULL;
}
align_get_bits(&s->gb);
assert((get_bits_count(&s->gb) & 7) == 0);
i= get_bits_left(&s->gb)>>3;
if(i<0 || i > BACKSTEP_SIZE || nb_frames<0){
if(i<0)
av_log(s->avctx, AV_LOG_ERROR, "invalid new backstep %d\n", i);
i= FFMIN(BACKSTEP_SIZE, buf_size - HEADER_SIZE);
}
assert(i <= buf_size - HEADER_SIZE && i>= 0);
memcpy(s->last_buf + s->last_buf_size, s->gb.buffer + buf_size - HEADER_SIZE - i, i);
s->last_buf_size += i;
break;
}
/* apply the synthesis filter */
for(ch=0;ch<s->nb_channels;ch++) {
samples_ptr = samples + ch;
for(i=0;i<nb_frames;i++) {
RENAME(ff_mpa_synth_filter)(
#if CONFIG_FLOAT
s,
#endif
s->synth_buf[ch], &(s->synth_buf_offset[ch]),
RENAME(ff_mpa_synth_window), &s->dither_state,
samples_ptr, s->nb_channels,
s->sb_samples[ch][i]);
samples_ptr += 32 * s->nb_channels;
}
}
return nb_frames * 32 * sizeof(OUT_INT) * s->nb_channels;
}
static int decode_frame(AVCodecContext * avctx,
void *data, int *data_size,
AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
MPADecodeContext *s = avctx->priv_data;
uint32_t header;
int out_size;
OUT_INT *out_samples = data;
if(buf_size < HEADER_SIZE)
return -1;
header = AV_RB32(buf);
if(ff_mpa_check_header(header) < 0){
av_log(avctx, AV_LOG_ERROR, "Header missing\n");
return -1;
}
if (ff_mpegaudio_decode_header((MPADecodeHeader *)s, header) == 1) {
/* free format: prepare to compute frame size */
s->frame_size = -1;
return -1;
}
/* update codec info */
avctx->channels = s->nb_channels;
if (!avctx->bit_rate)
avctx->bit_rate = s->bit_rate;
avctx->sub_id = s->layer;
if(*data_size < 1152*avctx->channels*sizeof(OUT_INT))
return -1;
*data_size = 0;
if(s->frame_size<=0 || s->frame_size > buf_size){
av_log(avctx, AV_LOG_ERROR, "incomplete frame\n");
return -1;
}else if(s->frame_size < buf_size){
av_log(avctx, AV_LOG_ERROR, "incorrect frame size\n");
buf_size= s->frame_size;
}
out_size = mp_decode_frame(s, out_samples, buf, buf_size);
if(out_size>=0){
*data_size = out_size;
avctx->sample_rate = s->sample_rate;
//FIXME maybe move the other codec info stuff from above here too
}else
av_log(avctx, AV_LOG_DEBUG, "Error while decoding MPEG audio frame.\n"); //FIXME return -1 / but also return the number of bytes consumed
s->frame_size = 0;
return buf_size;
}
static void flush(AVCodecContext *avctx){
MPADecodeContext *s = avctx->priv_data;
memset(s->synth_buf, 0, sizeof(s->synth_buf));
s->last_buf_size= 0;
}
#if CONFIG_MP3ADU_DECODER || CONFIG_MP3ADUFLOAT_DECODER
static int decode_frame_adu(AVCodecContext * avctx,
void *data, int *data_size,
AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
MPADecodeContext *s = avctx->priv_data;
uint32_t header;
int len, out_size;
OUT_INT *out_samples = data;
len = buf_size;
// Discard too short frames
if (buf_size < HEADER_SIZE) {
*data_size = 0;
return buf_size;
}
if (len > MPA_MAX_CODED_FRAME_SIZE)
len = MPA_MAX_CODED_FRAME_SIZE;
// Get header and restore sync word
header = AV_RB32(buf) | 0xffe00000;
if (ff_mpa_check_header(header) < 0) { // Bad header, discard frame
*data_size = 0;
return buf_size;
}
ff_mpegaudio_decode_header((MPADecodeHeader *)s, header);
/* update codec info */
avctx->sample_rate = s->sample_rate;
avctx->channels = s->nb_channels;
if (!avctx->bit_rate)
avctx->bit_rate = s->bit_rate;
avctx->sub_id = s->layer;
s->frame_size = len;
if (avctx->parse_only) {
out_size = buf_size;
} else {
out_size = mp_decode_frame(s, out_samples, buf, buf_size);
}
*data_size = out_size;
return buf_size;
}
#endif /* CONFIG_MP3ADU_DECODER || CONFIG_MP3ADUFLOAT_DECODER */
#if CONFIG_MP3ON4_DECODER || CONFIG_MP3ON4FLOAT_DECODER
/**
* Context for MP3On4 decoder
*/
typedef struct MP3On4DecodeContext {
int frames; ///< number of mp3 frames per block (number of mp3 decoder instances)
int syncword; ///< syncword patch
const uint8_t *coff; ///< channels offsets in output buffer
MPADecodeContext *mp3decctx[5]; ///< MPADecodeContext for every decoder instance
} MP3On4DecodeContext;
#include "mpeg4audio.h"
/* Next 3 arrays are indexed by channel config number (passed via codecdata) */
static const uint8_t mp3Frames[8] = {0,1,1,2,3,3,4,5}; /* number of mp3 decoder instances */
/* offsets into output buffer, assume output order is FL FR BL BR C LFE */
static const uint8_t chan_offset[8][5] = {
{0},
{0}, // C
{0}, // FLR
{2,0}, // C FLR
{2,0,3}, // C FLR BS
{4,0,2}, // C FLR BLRS
{4,0,2,5}, // C FLR BLRS LFE
{4,0,2,6,5}, // C FLR BLRS BLR LFE
};
static int decode_init_mp3on4(AVCodecContext * avctx)
{
MP3On4DecodeContext *s = avctx->priv_data;
MPEG4AudioConfig cfg;
int i;
if ((avctx->extradata_size < 2) || (avctx->extradata == NULL)) {
av_log(avctx, AV_LOG_ERROR, "Codec extradata missing or too short.\n");
return -1;
}
ff_mpeg4audio_get_config(&cfg, avctx->extradata, avctx->extradata_size);
if (!cfg.chan_config || cfg.chan_config > 7) {
av_log(avctx, AV_LOG_ERROR, "Invalid channel config number.\n");
return -1;
}
s->frames = mp3Frames[cfg.chan_config];
s->coff = chan_offset[cfg.chan_config];
avctx->channels = ff_mpeg4audio_channels[cfg.chan_config];
if (cfg.sample_rate < 16000)
s->syncword = 0xffe00000;
else
s->syncword = 0xfff00000;
/* Init the first mp3 decoder in standard way, so that all tables get builded
* We replace avctx->priv_data with the context of the first decoder so that
* decode_init() does not have to be changed.
* Other decoders will be initialized here copying data from the first context
*/
// Allocate zeroed memory for the first decoder context
s->mp3decctx[0] = av_mallocz(sizeof(MPADecodeContext));
// Put decoder context in place to make init_decode() happy
avctx->priv_data = s->mp3decctx[0];
decode_init(avctx);
// Restore mp3on4 context pointer
avctx->priv_data = s;
s->mp3decctx[0]->adu_mode = 1; // Set adu mode
/* Create a separate codec/context for each frame (first is already ok).
* Each frame is 1 or 2 channels - up to 5 frames allowed
*/
for (i = 1; i < s->frames; i++) {
s->mp3decctx[i] = av_mallocz(sizeof(MPADecodeContext));
s->mp3decctx[i]->adu_mode = 1;
s->mp3decctx[i]->avctx = avctx;
}
return 0;
}
static av_cold int decode_close_mp3on4(AVCodecContext * avctx)
{
MP3On4DecodeContext *s = avctx->priv_data;
int i;
for (i = 0; i < s->frames; i++)
av_free(s->mp3decctx[i]);
return 0;
}
static int decode_frame_mp3on4(AVCodecContext * avctx,
void *data, int *data_size,
AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
MP3On4DecodeContext *s = avctx->priv_data;
MPADecodeContext *m;
int fsize, len = buf_size, out_size = 0;
uint32_t header;
OUT_INT *out_samples = data;
OUT_INT decoded_buf[MPA_FRAME_SIZE * MPA_MAX_CHANNELS];
OUT_INT *outptr, *bp;
int fr, j, n;
if(*data_size < MPA_FRAME_SIZE * MPA_MAX_CHANNELS * s->frames * sizeof(OUT_INT))
return -1;
*data_size = 0;
// Discard too short frames
if (buf_size < HEADER_SIZE)
return -1;
// If only one decoder interleave is not needed
outptr = s->frames == 1 ? out_samples : decoded_buf;
avctx->bit_rate = 0;
for (fr = 0; fr < s->frames; fr++) {
fsize = AV_RB16(buf) >> 4;
fsize = FFMIN3(fsize, len, MPA_MAX_CODED_FRAME_SIZE);
m = s->mp3decctx[fr];
assert (m != NULL);
header = (AV_RB32(buf) & 0x000fffff) | s->syncword; // patch header
if (ff_mpa_check_header(header) < 0) // Bad header, discard block
break;
ff_mpegaudio_decode_header((MPADecodeHeader *)m, header);
out_size += mp_decode_frame(m, outptr, buf, fsize);
buf += fsize;
len -= fsize;
if(s->frames > 1) {
n = m->avctx->frame_size*m->nb_channels;
/* interleave output data */
bp = out_samples + s->coff[fr];
if(m->nb_channels == 1) {
for(j = 0; j < n; j++) {
*bp = decoded_buf[j];
bp += avctx->channels;
}
} else {
for(j = 0; j < n; j++) {
bp[0] = decoded_buf[j++];
bp[1] = decoded_buf[j];
bp += avctx->channels;
}
}
}
avctx->bit_rate += m->bit_rate;
}
/* update codec info */
avctx->sample_rate = s->mp3decctx[0]->sample_rate;
*data_size = out_size;
return buf_size;
}
#endif /* CONFIG_MP3ON4_DECODER || CONFIG_MP3ON4FLOAT_DECODER */
#if !CONFIG_FLOAT
#if CONFIG_MP1_DECODER
AVCodec ff_mp1_decoder =
{
"mp1",
AVMEDIA_TYPE_AUDIO,
CODEC_ID_MP1,
sizeof(MPADecodeContext),
decode_init,
NULL,
NULL,
decode_frame,
CODEC_CAP_PARSE_ONLY,
.flush= flush,
.long_name= NULL_IF_CONFIG_SMALL("MP1 (MPEG audio layer 1)"),
};
#endif
#if CONFIG_MP2_DECODER
AVCodec ff_mp2_decoder =
{
"mp2",
AVMEDIA_TYPE_AUDIO,
CODEC_ID_MP2,
sizeof(MPADecodeContext),
decode_init,
NULL,
NULL,
decode_frame,
CODEC_CAP_PARSE_ONLY,
.flush= flush,
.long_name= NULL_IF_CONFIG_SMALL("MP2 (MPEG audio layer 2)"),
};
#endif
#if CONFIG_MP3_DECODER
AVCodec ff_mp3_decoder =
{
"mp3",
AVMEDIA_TYPE_AUDIO,
CODEC_ID_MP3,
sizeof(MPADecodeContext),
decode_init,
NULL,
NULL,
decode_frame,
CODEC_CAP_PARSE_ONLY,
.flush= flush,
.long_name= NULL_IF_CONFIG_SMALL("MP3 (MPEG audio layer 3)"),
};
#endif
#if CONFIG_MP3ADU_DECODER
AVCodec ff_mp3adu_decoder =
{
"mp3adu",
AVMEDIA_TYPE_AUDIO,
CODEC_ID_MP3ADU,
sizeof(MPADecodeContext),
decode_init,
NULL,
NULL,
decode_frame_adu,
CODEC_CAP_PARSE_ONLY,
.flush= flush,
.long_name= NULL_IF_CONFIG_SMALL("ADU (Application Data Unit) MP3 (MPEG audio layer 3)"),
};
#endif
#if CONFIG_MP3ON4_DECODER
AVCodec ff_mp3on4_decoder =
{
"mp3on4",
AVMEDIA_TYPE_AUDIO,
CODEC_ID_MP3ON4,
sizeof(MP3On4DecodeContext),
decode_init_mp3on4,
NULL,
decode_close_mp3on4,
decode_frame_mp3on4,
.flush= flush,
.long_name= NULL_IF_CONFIG_SMALL("MP3onMP4"),
};
#endif
#endif