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ffmpeg-rockchip/libavcodec/mpegaudiodec.c
Michael Niedermayer 0aa095483d Merge commit '6fee1b90ce3bf4fbdfde7016e0890057c9000487' 
* commit '6fee1b90ce3bf4fbdfde7016e0890057c9000487':
  avcodec: Add av_cold attributes to init functions missing them

Conflicts:
	libavcodec/aacpsy.c
	libavcodec/atrac3.c
	libavcodec/dvdsubdec.c
	libavcodec/ffv1.c
	libavcodec/ffv1enc.c
	libavcodec/h261enc.c
	libavcodec/h264_parser.c
	libavcodec/h264dsp.c
	libavcodec/h264pred.c
	libavcodec/libschroedingerenc.c
	libavcodec/libxvid_rc.c
	libavcodec/mpeg12.c
	libavcodec/mpeg12enc.c
	libavcodec/proresdsp.c
	libavcodec/rangecoder.c
	libavcodec/videodsp.c
	libavcodec/x86/proresdsp_init.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2013-05-05 11:34:29 +02:00

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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 "libavutil/attributes.h"
#include "libavutil/avassert.h"
#include "libavutil/channel_layout.h"
#include "libavutil/float_dsp.h"
#include "libavutil/libm.h"
#include "avcodec.h"
#include "get_bits.h"
#include "internal.h"
#include "mathops.h"
#include "mpegaudiodsp.h"
/*
* TODO:
* - test lsf / mpeg25 extensively.
*/
#include "mpegaudio.h"
#include "mpegaudiodecheader.h"
#define BACKSTEP_SIZE 512
#define EXTRABYTES 24
#define LAST_BUF_SIZE 2 * BACKSTEP_SIZE + EXTRABYTES
/* layer 3 "granule" */
typedef struct GranuleDef {
uint8_t scfsi;
int part2_3_length;
int big_values;
int global_gain;
int scalefac_compress;
uint8_t block_type;
uint8_t switch_point;
int table_select[3];
int subblock_gain[3];
uint8_t scalefac_scale;
uint8_t count1table_select;
int region_size[3]; /* number of huffman codes in each region */
int preflag;
int short_start, long_end; /* long/short band indexes */
uint8_t scale_factors[40];
DECLARE_ALIGNED(16, INTFLOAT, sb_hybrid)[SBLIMIT * 18]; /* 576 samples */
} GranuleDef;
typedef struct MPADecodeContext {
MPA_DECODE_HEADER
uint8_t last_buf[LAST_BUF_SIZE];
int last_buf_size;
/* next header (used in free format parsing) */
uint32_t free_format_next_header;
GetBitContext gb;
GetBitContext in_gb;
DECLARE_ALIGNED(32, MPA_INT, synth_buf)[MPA_MAX_CHANNELS][512 * 2];
int synth_buf_offset[MPA_MAX_CHANNELS];
DECLARE_ALIGNED(32, INTFLOAT, sb_samples)[MPA_MAX_CHANNELS][36][SBLIMIT];
INTFLOAT mdct_buf[MPA_MAX_CHANNELS][SBLIMIT * 18]; /* previous samples, for layer 3 MDCT */
GranuleDef granules[2][2]; /* Used in Layer 3 */
int adu_mode; ///< 0 for standard mp3, 1 for adu formatted mp3
int dither_state;
int err_recognition;
AVCodecContext* avctx;
MPADSPContext mpadsp;
AVFloatDSPContext fdsp;
AVFrame *frame;
} MPADecodeContext;
#if CONFIG_FLOAT
# define SHR(a,b) ((a)*(1.0f/(1<<(b))))
# 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
# define OUT_FMT AV_SAMPLE_FMT_FLT
# define OUT_FMT_P AV_SAMPLE_FMT_FLTP
#else
# define SHR(a,b) ((a)>>(b))
/* WARNING: only correct for positive 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 ## _fixed
# define OUT_FMT AV_SAMPLE_FMT_S16
# define OUT_FMT_P AV_SAMPLE_FMT_S16P
#endif
/****************/
#define HEADER_SIZE 4
#include "mpegaudiodata.h"
#include "mpegaudiodectab.h"
/* 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 INTFLOAT csa_table[8][4];
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 */
};
/**
* Convert region offsets to region sizes and truncate
* size to big_values.
*/
static void 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 init_short_region(MPADecodeContext *s, GranuleDef *g)
{
if (g->block_type == 2) {
if (s->sample_rate_index != 8)
g->region_size[0] = (36 / 2);
else
g->region_size[0] = (72 / 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 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 compute_band_indexes(MPADecodeContext *s, GranuleDef *g)
{
if (g->block_type == 2) {
if (g->switch_point) {
if(s->sample_rate_index == 8)
avpriv_request_sample(s->avctx, "switch point in 8khz");
/* if switched mode, we handle the 36 first samples as
long blocks. For 8000Hz, we handle the 72 first
exponents as long blocks */
if (s->sample_rate_index <= 2)
g->long_end = 8;
else
g->long_end = 6;
g->short_start = 3;
} 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;
#ifdef DEBUG
if(e < 1)
av_log(NULL, AV_LOG_WARNING, "l3_unscale: e is %d\n", e);
#endif
if (e > 31)
return 0;
m = (m + (1 << (e - 1))) >> e;
return m;
}
static av_cold void decode_init_static(void)
{
int i, j, k;
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(NULL, "%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] = { 0 };
uint16_t tmp_codes[512] = { 0 };
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];
}
av_assert0(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];
}
av_assert0(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 */
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 = exp2(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(NULL, "is_table_lsf %d %d: %f %f\n",
i, j, (float) is_table_lsf[j][0][i],
(float) 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;
#if !CONFIG_FLOAT
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);
#else
csa_table[i][0] = cs;
csa_table[i][1] = ca;
csa_table[i][2] = ca + cs;
csa_table[i][3] = ca - cs;
#endif
}
}
static av_cold int decode_init(AVCodecContext * avctx)
{
static int initialized_tables = 0;
MPADecodeContext *s = avctx->priv_data;
if (!initialized_tables) {
decode_init_static();
initialized_tables = 1;
}
s->avctx = avctx;
avpriv_float_dsp_init(&s->fdsp, avctx->flags & CODEC_FLAG_BITEXACT);
ff_mpadsp_init(&s->mpadsp);
if (avctx->request_sample_fmt == OUT_FMT &&
avctx->codec_id != AV_CODEC_ID_MP3ON4)
avctx->sample_fmt = OUT_FMT;
else
avctx->sample_fmt = OUT_FMT_P;
s->err_recognition = avctx->err_recognition;
if (avctx->codec_id == AV_CODEC_ID_MP3ADU)
s->adu_mode = 1;
return 0;
}
#define C3 FIXHR(0.86602540378443864676/2)
#define C4 FIXHR(0.70710678118654752439/2) //0.5 / cos(pi*(9)/36)
#define C5 FIXHR(0.51763809020504152469/2) //0.5 / cos(pi*(5)/36)
#define C6 FIXHR(1.93185165257813657349/4) //0.5 / cos(pi*(15)/36)
/* 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, C4, 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, C5, 1);
out[ 8] =
out[ 9] = in4 + in1;
out[ 2] =
out[ 3] = in4 - in1;
in0 -= in2;
in5 = MULH3(in5 - in3, C6, 2);
out[ 0] =
out[ 5] = in0 - in5;
out[ 6] =
out[11] = in0 + in5;
}
/* 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)
{
return n ? get_bits(s, n) : 0;
}
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;
av_assert2((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){
switch_buffer(s, &pos, &end_pos, &end_pos2);
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->err_recognition & (AV_EF_BITSTREAM|AV_EF_COMPLIANT))
s_index=0;
break;
}
switch_buffer(s, &pos, &end_pos, &end_pos2);
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);
if (bits_left < 0 && (s->err_recognition & (AV_EF_BUFFER|AV_EF_COMPLIANT))) {
av_log(s->avctx, AV_LOG_ERROR, "bits_left=%d\n", bits_left);
s_index=0;
} else if (bits_left > 0 && (s->err_recognition & (AV_EF_BUFFER|AV_EF_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 + 72;
} 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 */
#if CONFIG_FLOAT
s->fdsp.butterflies_float(g0->sb_hybrid, g1->sb_hybrid, 576);
#else
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;
}
#endif
}
}
#if CONFIG_FLOAT
#if HAVE_MIPSFPU
# include "mips/compute_antialias_float.h"
#endif /* HAVE_MIPSFPU */
#else
#if HAVE_MIPSDSPR1
# include "mips/compute_antialias_fixed.h"
#endif /* HAVE_MIPSDSPR1 */
#endif /* CONFIG_FLOAT */
#ifndef compute_antialias
#if CONFIG_FLOAT
#define AA(j) do { \
float tmp0 = ptr[-1-j]; \
float tmp1 = ptr[ j]; \
ptr[-1-j] = tmp0 * csa_table[j][0] - tmp1 * csa_table[j][1]; \
ptr[ j] = tmp0 * csa_table[j][1] + tmp1 * csa_table[j][0]; \
} while (0)
#else
#define AA(j) do { \
int tmp0 = ptr[-1-j]; \
int tmp1 = ptr[ j]; \
int tmp2 = MULH(tmp0 + tmp1, csa_table[j][0]); \
ptr[-1-j] = 4 * (tmp2 - MULH(tmp1, csa_table[j][2])); \
ptr[ j] = 4 * (tmp2 + MULH(tmp0, csa_table[j][3])); \
} while (0)
#endif
static void compute_antialias(MPADecodeContext *s, GranuleDef *g)
{
INTFLOAT *ptr;
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--) {
AA(0);
AA(1);
AA(2);
AA(3);
AA(4);
AA(5);
AA(6);
AA(7);
ptr += 18;
}
}
#endif /* compute_antialias */
static void compute_imdct(MPADecodeContext *s, GranuleDef *g,
INTFLOAT *sb_samples, INTFLOAT *mdct_buf)
{
INTFLOAT *win, *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;
}
s->mpadsp.RENAME(imdct36_blocks)(sb_samples, mdct_buf, g->sb_hybrid,
mdct_long_end, g->switch_point,
g->block_type);
buf = mdct_buf + 4*18*(mdct_long_end >> 2) + (mdct_long_end & 3);
ptr = g->sb_hybrid + 18 * mdct_long_end;
for (j = mdct_long_end; j < sblimit; j++) {
/* select frequency inversion */
win = RENAME(ff_mdct_win)[2 + (4 & -(j & 1))];
out_ptr = sb_samples + j;
for (i = 0; i < 6; i++) {
*out_ptr = buf[4*i];
out_ptr += SBLIMIT;
}
imdct12(out2, ptr + 0);
for (i = 0; i < 6; i++) {
*out_ptr = MULH3(out2[i ], win[i ], 1) + buf[4*(i + 6*1)];
buf[4*(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[4*(i + 6*2)];
buf[4*(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[4*(i + 6*0)] = MULH3(out2[i ], win[i ], 1) + buf[4*(i + 6*0)];
buf[4*(i + 6*1)] = MULH3(out2[i + 6], win[i + 6], 1);
buf[4*(i + 6*2)] = 0;
}
ptr += 18;
buf += (j&3) != 3 ? 1 : (4*18-3);
}
/* zero bands */
for (j = sblimit; j < SBLIMIT; j++) {
/* overlap */
out_ptr = sb_samples + j;
for (i = 0; i < 18; i++) {
*out_ptr = buf[4*i];
buf[4*i] = 0;
out_ptr += SBLIMIT;
}
buf += (j&3) != 3 ? 1 : (4*18-3);
}
}
/* main layer3 decoding function */
static int mp_decode_layer3(MPADecodeContext *s)
{
int nb_granules, main_data_begin;
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);
skip_bits(&s->gb, s->nb_channels);
nb_granules = 1;
} else {
main_data_begin = get_bits(&s->gb, 9);
if (s->nb_channels == 2)
skip_bits(&s->gb, 3);
else
skip_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 AVERROR_INVALIDDATA;
}
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 AVERROR_INVALIDDATA;
}
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);
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);
init_long_region(s, g, region_address1, region_address2);
}
region_offset2size(g);
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) {
int skip;
const uint8_t *ptr = s->gb.buffer + (get_bits_count(&s->gb)>>3);
int extrasize = av_clip(get_bits_left(&s->gb) >> 3, 0, EXTRABYTES);
av_assert1((get_bits_count(&s->gb) & 7) == 0);
/* now we get bits from the main_data_begin offset */
av_dlog(s->avctx, "seekback:%d, lastbuf:%d\n",
main_data_begin, s->last_buf_size);
memcpy(s->last_buf + s->last_buf_size, ptr, extrasize);
s->in_gb = s->gb;
init_get_bits(&s->gb, s->last_buf, s->last_buf_size*8);
#if !UNCHECKED_BITSTREAM_READER
s->gb.size_in_bits_plus8 += FFMAX(extrasize, LAST_BUF_SIZE - s->last_buf_size) * 8;
#endif
s->last_buf_size <<= 3;
for (gr = 0; gr < nb_granules && (s->last_buf_size >> 3) < main_data_begin; gr++) {
for (ch = 0; ch < s->nb_channels; ch++) {
g = &s->granules[ch][gr];
s->last_buf_size += g->part2_3_length;
memset(g->sb_hybrid, 0, sizeof(g->sb_hybrid));
compute_imdct(s, g, &s->sb_samples[ch][18 * gr][0], s->mdct_buf[ch]);
}
}
skip = s->last_buf_size - 8 * main_data_begin;
if (skip >= s->gb.size_in_bits && s->in_gb.buffer) {
skip_bits_long(&s->in_gb, skip - s->gb.size_in_bits);
s->gb = s->in_gb;
s->in_gb.buffer = NULL;
} else {
skip_bits_long(&s->gb, skip);
}
} else {
gr = 0;
}
for (; gr < nb_granules; gr++) {
for (ch = 0; ch < s->nb_channels; ch++) {
g = &s->granules[ch][gr];
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->mode == MPA_JSTEREO)
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, ret;
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);
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);
av_assert1((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);
}
av_assert1(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;
}
if(nb_frames < 0)
return nb_frames;
/* get output buffer */
if (!samples) {
av_assert0(s->frame != NULL);
s->frame->nb_samples = s->avctx->frame_size;
if ((ret = ff_get_buffer(s->avctx, s->frame, 0)) < 0)
return ret;
samples = (OUT_INT **)s->frame->extended_data;
}
/* apply the synthesis filter */
for (ch = 0; ch < s->nb_channels; ch++) {
int sample_stride;
if (s->avctx->sample_fmt == OUT_FMT_P) {
samples_ptr = samples[ch];
sample_stride = 1;
} else {
samples_ptr = samples[0] + ch;
sample_stride = s->nb_channels;
}
for (i = 0; i < nb_frames; i++) {
RENAME(ff_mpa_synth_filter)(&s->mpadsp, s->synth_buf[ch],
&(s->synth_buf_offset[ch]),
RENAME(ff_mpa_synth_window),
&s->dither_state, samples_ptr,
sample_stride, s->sb_samples[ch][i]);
samples_ptr += 32 * sample_stride;
}
}
return nb_frames * 32 * sizeof(OUT_INT) * s->nb_channels;
}
static int decode_frame(AVCodecContext * avctx, void *data, int *got_frame_ptr,
AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
MPADecodeContext *s = avctx->priv_data;
uint32_t header;
int ret;
while(buf_size && !*buf){
buf++;
buf_size--;
}
if (buf_size < HEADER_SIZE)
return AVERROR_INVALIDDATA;
header = AV_RB32(buf);
if (header>>8 == AV_RB32("TAG")>>8) {
av_log(avctx, AV_LOG_DEBUG, "discarding ID3 tag\n");
return buf_size;
}
if (ff_mpa_check_header(header) < 0) {
av_log(avctx, AV_LOG_ERROR, "Header missing\n");
return AVERROR_INVALIDDATA;
}
if (avpriv_mpegaudio_decode_header((MPADecodeHeader *)s, header) == 1) {
/* free format: prepare to compute frame size */
s->frame_size = -1;
return AVERROR_INVALIDDATA;
}
/* update codec info */
avctx->channels = s->nb_channels;
avctx->channel_layout = s->nb_channels == 1 ? AV_CH_LAYOUT_MONO : AV_CH_LAYOUT_STEREO;
if (!avctx->bit_rate)
avctx->bit_rate = s->bit_rate;
if (s->frame_size <= 0 || s->frame_size > buf_size) {
av_log(avctx, AV_LOG_ERROR, "incomplete frame\n");
return AVERROR_INVALIDDATA;
} else if (s->frame_size < buf_size) {
av_log(avctx, AV_LOG_DEBUG, "incorrect frame size - multiple frames in buffer?\n");
buf_size= s->frame_size;
}
s->frame = data;
ret = mp_decode_frame(s, NULL, buf, buf_size);
if (ret >= 0) {
s->frame->nb_samples = avctx->frame_size;
*got_frame_ptr = 1;
avctx->sample_rate = s->sample_rate;
//FIXME maybe move the other codec info stuff from above here too
} else {
av_log(avctx, AV_LOG_ERROR, "Error while decoding MPEG audio frame.\n");
/* Only return an error if the bad frame makes up the whole packet or
* the error is related to buffer management.
* If there is more data in the packet, just consume the bad frame
* instead of returning an error, which would discard the whole
* packet. */
*got_frame_ptr = 0;
if (buf_size == avpkt->size || ret != AVERROR_INVALIDDATA)
return ret;
}
s->frame_size = 0;
return buf_size;
}
static void mp_flush(MPADecodeContext *ctx)
{
memset(ctx->synth_buf, 0, sizeof(ctx->synth_buf));
ctx->last_buf_size = 0;
}
static void flush(AVCodecContext *avctx)
{
mp_flush(avctx->priv_data);
}
#if CONFIG_MP3ADU_DECODER || CONFIG_MP3ADUFLOAT_DECODER
static int decode_frame_adu(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
MPADecodeContext *s = avctx->priv_data;
uint32_t header;
int len, ret;
int av_unused out_size;
len = buf_size;
// Discard too short frames
if (buf_size < HEADER_SIZE) {
av_log(avctx, AV_LOG_ERROR, "Packet is too small\n");
return AVERROR_INVALIDDATA;
}
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
av_log(avctx, AV_LOG_ERROR, "Invalid frame header\n");
return AVERROR_INVALIDDATA;
}
avpriv_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;
s->frame_size = len;
s->frame = data;
ret = mp_decode_frame(s, NULL, buf, buf_size);
if (ret < 0) {
av_log(avctx, AV_LOG_ERROR, "Error while decoding MPEG audio frame.\n");
return ret;
}
*got_frame_ptr = 1;
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; ///< channel 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) */
/* number of mp3 decoder instances */
static const uint8_t mp3Frames[8] = { 0, 1, 1, 2, 3, 3, 4, 5 };
/* offsets into output buffer, assume output order is FL FR C LFE BL BR SL SR */
static const uint8_t chan_offset[8][5] = {
{ 0 },
{ 0 }, // C
{ 0 }, // FLR
{ 2, 0 }, // C FLR
{ 2, 0, 3 }, // C FLR BS
{ 2, 0, 3 }, // C FLR BLRS
{ 2, 0, 4, 3 }, // C FLR BLRS LFE
{ 2, 0, 6, 4, 3 }, // C FLR BLRS BLR LFE
};
/* mp3on4 channel layouts */
static const int16_t chan_layout[8] = {
0,
AV_CH_LAYOUT_MONO,
AV_CH_LAYOUT_STEREO,
AV_CH_LAYOUT_SURROUND,
AV_CH_LAYOUT_4POINT0,
AV_CH_LAYOUT_5POINT0,
AV_CH_LAYOUT_5POINT1,
AV_CH_LAYOUT_7POINT1
};
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 av_cold 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 AVERROR_INVALIDDATA;
}
avpriv_mpeg4audio_get_config(&cfg, avctx->extradata,
avctx->extradata_size * 8, 1);
if (!cfg.chan_config || cfg.chan_config > 7) {
av_log(avctx, AV_LOG_ERROR, "Invalid channel config number.\n");
return AVERROR_INVALIDDATA;
}
s->frames = mp3Frames[cfg.chan_config];
s->coff = chan_offset[cfg.chan_config];
avctx->channels = ff_mpeg4audio_channels[cfg.chan_config];
avctx->channel_layout = chan_layout[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));
if (!s->mp3decctx[0])
goto alloc_fail;
// 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));
if (!s->mp3decctx[i])
goto alloc_fail;
s->mp3decctx[i]->adu_mode = 1;
s->mp3decctx[i]->avctx = avctx;
s->mp3decctx[i]->mpadsp = s->mp3decctx[0]->mpadsp;
}
return 0;
alloc_fail:
decode_close_mp3on4(avctx);
return AVERROR(ENOMEM);
}
static void flush_mp3on4(AVCodecContext *avctx)
{
int i;
MP3On4DecodeContext *s = avctx->priv_data;
for (i = 0; i < s->frames; i++)
mp_flush(s->mp3decctx[i]);
}
static int decode_frame_mp3on4(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
AVFrame *frame = data;
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;
OUT_INT *outptr[2];
int fr, ch, ret;
/* get output buffer */
frame->nb_samples = MPA_FRAME_SIZE;
if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
return ret;
out_samples = (OUT_INT **)frame->extended_data;
// Discard too short frames
if (buf_size < HEADER_SIZE)
return AVERROR_INVALIDDATA;
avctx->bit_rate = 0;
ch = 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];
av_assert1(m);
if (fsize < HEADER_SIZE) {
av_log(avctx, AV_LOG_ERROR, "Frame size smaller than header size\n");
return AVERROR_INVALIDDATA;
}
header = (AV_RB32(buf) & 0x000fffff) | s->syncword; // patch header
if (ff_mpa_check_header(header) < 0) // Bad header, discard block
break;
avpriv_mpegaudio_decode_header((MPADecodeHeader *)m, header);
if (ch + m->nb_channels > avctx->channels || s->coff[fr] + m->nb_channels > avctx->channels) {
av_log(avctx, AV_LOG_ERROR, "frame channel count exceeds codec "
"channel count\n");
return AVERROR_INVALIDDATA;
}
ch += m->nb_channels;
outptr[0] = out_samples[s->coff[fr]];
if (m->nb_channels > 1)
outptr[1] = out_samples[s->coff[fr] + 1];
if ((ret = mp_decode_frame(m, outptr, buf, fsize)) < 0)
return ret;
out_size += ret;
buf += fsize;
len -= fsize;
avctx->bit_rate += m->bit_rate;
}
/* update codec info */
avctx->sample_rate = s->mp3decctx[0]->sample_rate;
frame->nb_samples = out_size / (avctx->channels * sizeof(OUT_INT));
*got_frame_ptr = 1;
return buf_size;
}
#endif /* CONFIG_MP3ON4_DECODER || CONFIG_MP3ON4FLOAT_DECODER */
#if !CONFIG_FLOAT
#if CONFIG_MP1_DECODER
AVCodec ff_mp1_decoder = {
.name = "mp1",
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_MP1,
.priv_data_size = sizeof(MPADecodeContext),
.init = decode_init,
.decode = decode_frame,
.capabilities = CODEC_CAP_DR1,
.flush = flush,
.long_name = NULL_IF_CONFIG_SMALL("MP1 (MPEG audio layer 1)"),
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
AV_SAMPLE_FMT_S16,
AV_SAMPLE_FMT_NONE },
};
#endif
#if CONFIG_MP2_DECODER
AVCodec ff_mp2_decoder = {
.name = "mp2",
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_MP2,
.priv_data_size = sizeof(MPADecodeContext),
.init = decode_init,
.decode = decode_frame,
.capabilities = CODEC_CAP_DR1,
.flush = flush,
.long_name = NULL_IF_CONFIG_SMALL("MP2 (MPEG audio layer 2)"),
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
AV_SAMPLE_FMT_S16,
AV_SAMPLE_FMT_NONE },
};
#endif
#if CONFIG_MP3_DECODER
AVCodec ff_mp3_decoder = {
.name = "mp3",
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_MP3,
.priv_data_size = sizeof(MPADecodeContext),
.init = decode_init,
.decode = decode_frame,
.capabilities = CODEC_CAP_DR1,
.flush = flush,
.long_name = NULL_IF_CONFIG_SMALL("MP3 (MPEG audio layer 3)"),
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
AV_SAMPLE_FMT_S16,
AV_SAMPLE_FMT_NONE },
};
#endif
#if CONFIG_MP3ADU_DECODER
AVCodec ff_mp3adu_decoder = {
.name = "mp3adu",
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_MP3ADU,
.priv_data_size = sizeof(MPADecodeContext),
.init = decode_init,
.decode = decode_frame_adu,
.capabilities = CODEC_CAP_DR1,
.flush = flush,
.long_name = NULL_IF_CONFIG_SMALL("ADU (Application Data Unit) MP3 (MPEG audio layer 3)"),
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
AV_SAMPLE_FMT_S16,
AV_SAMPLE_FMT_NONE },
};
#endif
#if CONFIG_MP3ON4_DECODER
AVCodec ff_mp3on4_decoder = {
.name = "mp3on4",
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_MP3ON4,
.priv_data_size = sizeof(MP3On4DecodeContext),
.init = decode_init_mp3on4,
.close = decode_close_mp3on4,
.decode = decode_frame_mp3on4,
.capabilities = CODEC_CAP_DR1,
.flush = flush_mp3on4,
.long_name = NULL_IF_CONFIG_SMALL("MP3onMP4"),
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
AV_SAMPLE_FMT_NONE },
};
#endif
#endif
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