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bfcf690c68
FFmpeg/libavcodec/ac3dec.c
Justin Ruggles bfcf690c68 cosmetics: rename variables. tbl->tab for consistency with other AC3 tables 
Originally committed as revision 10494 to svn://svn.ffmpeg.org/ffmpeg/trunk
2007-09-15 00:19:21 +00:00

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/*
* AC-3 Audio Decoder
* This code is developed as part of Google Summer of Code 2006 Program.
*
* Copyright (c) 2006 Kartikey Mahendra BHATT (bhattkm at gmail dot com).
* Copyright (c) 2007 Justin Ruggles
*
* Portions of this code are derived from liba52
* http://liba52.sourceforge.net
* Copyright (C) 2000-2003 Michel Lespinasse <walken@zoy.org>
* Copyright (C) 1999-2000 Aaron Holtzman <aholtzma@ess.engr.uvic.ca>
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU General Public
* License as published by the Free Software Foundation; either
* version 2 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
* General Public License for more details.
*
* You should have received a copy of the GNU 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
*/
#include <stdio.h>
#include <stddef.h>
#include <math.h>
#include <string.h>
#include "avcodec.h"
#include "ac3_parser.h"
#include "bitstream.h"
#include "dsputil.h"
#include "random.h"
/**
* Table of bin locations for rematrixing bands
* reference: Section 7.5.2 Rematrixing : Frequency Band Definitions
*/
static const uint8_t rematrix_band_tab[5] = { 13, 25, 37, 61, 253 };
/**
* table for exponent to scale_factor mapping
* scale_factors[i] = 2 ^ -i
*/
static float scale_factors[25];
/** table for grouping exponents */
static uint8_t exp_ungroup_tab[128][3];
/** tables for ungrouping mantissas */
static float b1_mantissas[32][3];
static float b2_mantissas[128][3];
static float b3_mantissas[8];
static float b4_mantissas[128][2];
static float b5_mantissas[16];
/**
* Quantization table: levels for symmetric. bits for asymmetric.
* reference: Table 7.18 Mapping of bap to Quantizer
*/
static const uint8_t qntztab[16] = {
0, 3, 5, 7, 11, 15,
5, 6, 7, 8, 9, 10, 11, 12, 14, 16
};
/** dynamic range table. converts codes to scale factors. */
static float dynrng_tab[256];
/** dialogue normalization table */
static float dialnorm_tab[32];
/** Adjustments in dB gain */
#define LEVEL_MINUS_3DB 0.7071067811865476
#define LEVEL_MINUS_4POINT5DB 0.5946035575013605
#define LEVEL_MINUS_6DB 0.5000000000000000
#define LEVEL_MINUS_9DB 0.3535533905932738
#define LEVEL_ZERO 0.0000000000000000
#define LEVEL_ONE 1.0000000000000000
static const float gain_levels[6] = {
LEVEL_ZERO,
LEVEL_ONE,
LEVEL_MINUS_3DB,
LEVEL_MINUS_4POINT5DB,
LEVEL_MINUS_6DB,
LEVEL_MINUS_9DB
};
/**
* Table for center mix levels
* reference: Section 5.4.2.4 cmixlev
*/
static const uint8_t clevs[4] = { 2, 3, 4, 3 };
/**
* Table for surround mix levels
* reference: Section 5.4.2.5 surmixlev
*/
static const uint8_t slevs[4] = { 2, 4, 0, 4 };
/**
* Table for default stereo downmixing coefficients
* reference: Section 7.8.2 Downmixing Into Two Channels
*/
static const uint8_t ac3_default_coeffs[8][5][2] = {
{ { 1, 0 }, { 0, 1 }, },
{ { 2, 2 }, },
{ { 1, 0 }, { 0, 1 }, },
{ { 1, 0 }, { 3, 3 }, { 0, 1 }, },
{ { 1, 0 }, { 0, 1 }, { 4, 4 }, },
{ { 1, 0 }, { 3, 3 }, { 0, 1 }, { 5, 5 }, },
{ { 1, 0 }, { 0, 1 }, { 4, 0 }, { 0, 4 }, },
{ { 1, 0 }, { 3, 3 }, { 0, 1 }, { 4, 0 }, { 0, 4 }, },
};
/* override ac3.h to include coupling channel */
#undef AC3_MAX_CHANNELS
#define AC3_MAX_CHANNELS 7
#define CPL_CH 0
#define AC3_OUTPUT_LFEON 8
typedef struct {
int acmod; ///< audio coding mode
int dsurmod; ///< dolby surround mode
int blksw[AC3_MAX_CHANNELS]; ///< block switch flags
int dithflag[AC3_MAX_CHANNELS]; ///< dither flags
int dither_all; ///< true if all channels are dithered
int cplinu; ///< coupling in use
int chincpl[AC3_MAX_CHANNELS]; ///< channel in coupling
int phsflginu; ///< phase flags in use
int cplbndstrc[18]; ///< coupling band structure
int rematstr; ///< rematrixing strategy
int nrematbnd; ///< number of rematrixing bands
int rematflg[4]; ///< rematrixing flags
int expstr[AC3_MAX_CHANNELS]; ///< exponent strategies
int snroffst[AC3_MAX_CHANNELS]; ///< signal-to-noise ratio offsets
int fgain[AC3_MAX_CHANNELS]; ///< fast gain values (signal-to-mask ratio)
int deltbae[AC3_MAX_CHANNELS]; ///< delta bit allocation exists
int deltnseg[AC3_MAX_CHANNELS]; ///< number of delta segments
uint8_t deltoffst[AC3_MAX_CHANNELS][8]; ///< delta segment offsets
uint8_t deltlen[AC3_MAX_CHANNELS][8]; ///< delta segment lengths
uint8_t deltba[AC3_MAX_CHANNELS][8]; ///< delta values for each segment
int sampling_rate; ///< sample frequency, in Hz
int bit_rate; ///< stream bit rate, in bits-per-second
int frame_size; ///< current frame size, in bytes
int nchans; ///< number of total channels
int nfchans; ///< number of full-bandwidth channels
int lfeon; ///< lfe channel in use
int lfe_ch; ///< index of LFE channel
int output_mode; ///< output channel configuration
int out_channels; ///< number of output channels
float downmix_coeffs[AC3_MAX_CHANNELS][2]; ///< stereo downmix coefficients
float dialnorm[2]; ///< dialogue normalization
float dynrng[2]; ///< dynamic range
float cplco[AC3_MAX_CHANNELS][18]; ///< coupling coordinates
int ncplbnd; ///< number of coupling bands
int ncplsubnd; ///< number of coupling sub bands
int startmant[AC3_MAX_CHANNELS]; ///< start frequency bin
int endmant[AC3_MAX_CHANNELS]; ///< end frequency bin
AC3BitAllocParameters bit_alloc_params; ///< bit allocation parameters
int8_t dexps[AC3_MAX_CHANNELS][256]; ///< decoded exponents
uint8_t bap[AC3_MAX_CHANNELS][256]; ///< bit allocation pointers
int16_t psd[AC3_MAX_CHANNELS][256]; ///< scaled exponents
int16_t bndpsd[AC3_MAX_CHANNELS][50]; ///< interpolated exponents
int16_t mask[AC3_MAX_CHANNELS][50]; ///< masking curve values
DECLARE_ALIGNED_16(float, transform_coeffs[AC3_MAX_CHANNELS][256]); ///< transform coefficients
/* For IMDCT. */
MDCTContext imdct_512; ///< for 512 sample IMDCT
MDCTContext imdct_256; ///< for 256 sample IMDCT
DSPContext dsp; ///< for optimization
float add_bias; ///< offset for float_to_int16 conversion
float mul_bias; ///< scaling for float_to_int16 conversion
DECLARE_ALIGNED_16(float, output[AC3_MAX_CHANNELS-1][256]); ///< output after imdct transform and windowing
DECLARE_ALIGNED_16(short, int_output[AC3_MAX_CHANNELS-1][256]); ///< final 16-bit integer output
DECLARE_ALIGNED_16(float, delay[AC3_MAX_CHANNELS-1][256]); ///< delay - added to the next block
DECLARE_ALIGNED_16(float, tmp_imdct[256]); ///< temporary storage for imdct transform
DECLARE_ALIGNED_16(float, tmp_output[512]); ///< temporary storage for output before windowing
DECLARE_ALIGNED_16(float, window[256]); ///< window coefficients
/* Miscellaneous. */
GetBitContext gb; ///< bitstream reader
AVRandomState dith_state; ///< for dither generation
AVCodecContext *avctx; ///< parent context
} AC3DecodeContext;
/**
* Generate a Kaiser-Bessel Derived Window.
*/
static void ac3_window_init(float *window)
{
int i, j;
double sum = 0.0, bessel, tmp;
double local_window[256];
double alpha2 = (5.0 * M_PI / 256.0) * (5.0 * M_PI / 256.0);
for (i = 0; i < 256; i++) {
tmp = i * (256 - i) * alpha2;
bessel = 1.0;
for (j = 100; j > 0; j--) /* default to 100 iterations */
bessel = bessel * tmp / (j * j) + 1;
sum += bessel;
local_window[i] = sum;
}
sum++;
for (i = 0; i < 256; i++)
window[i] = sqrt(local_window[i] / sum);
}
/**
* Symmetrical Dequantization
* reference: Section 7.3.3 Expansion of Mantissas for Symmetrical Quantization
* Tables 7.19 to 7.23
*/
static inline float
symmetric_dequant(int code, int levels)
{
return (code - (levels >> 1)) * (2.0f / levels);
}
/*
* Initialize tables at runtime.
*/
static void ac3_tables_init(void)
{
int i;
/* generate grouped mantissa tables
reference: Section 7.3.5 Ungrouping of Mantissas */
for(i=0; i<32; i++) {
/* bap=1 mantissas */
b1_mantissas[i][0] = symmetric_dequant( i / 9 , 3);
b1_mantissas[i][1] = symmetric_dequant((i % 9) / 3, 3);
b1_mantissas[i][2] = symmetric_dequant((i % 9) % 3, 3);
}
for(i=0; i<128; i++) {
/* bap=2 mantissas */
b2_mantissas[i][0] = symmetric_dequant( i / 25 , 5);
b2_mantissas[i][1] = symmetric_dequant((i % 25) / 5, 5);
b2_mantissas[i][2] = symmetric_dequant((i % 25) % 5, 5);
/* bap=4 mantissas */
b4_mantissas[i][0] = symmetric_dequant(i / 11, 11);
b4_mantissas[i][1] = symmetric_dequant(i % 11, 11);
}
/* generate ungrouped mantissa tables
reference: Tables 7.21 and 7.23 */
for(i=0; i<7; i++) {
/* bap=3 mantissas */
b3_mantissas[i] = symmetric_dequant(i, 7);
}
for(i=0; i<15; i++) {
/* bap=5 mantissas */
b5_mantissas[i] = symmetric_dequant(i, 15);
}
/* generate dynamic range table
reference: Section 7.7.1 Dynamic Range Control */
for(i=0; i<256; i++) {
int v = (i >> 5) - ((i >> 7) << 3) - 5;
dynrng_tab[i] = powf(2.0f, v) * ((i & 0x1F) | 0x20);
}
/* generate dialogue normalization table
references: Section 5.4.2.8 dialnorm
Section 7.6 Dialogue Normalization */
for(i=1; i<32; i++) {
dialnorm_tab[i] = expf((i-31) * M_LN10 / 20.0f);
}
dialnorm_tab[0] = dialnorm_tab[31];
/* generate scale factors for exponents and asymmetrical dequantization
reference: Section 7.3.2 Expansion of Mantissas for Asymmetric Quantization */
for (i = 0; i < 25; i++)
scale_factors[i] = pow(2.0, -i);
/* generate exponent tables
reference: Section 7.1.3 Exponent Decoding */
for(i=0; i<128; i++) {
exp_ungroup_tab[i][0] = i / 25;
exp_ungroup_tab[i][1] = (i % 25) / 5;
exp_ungroup_tab[i][2] = (i % 25) % 5;
}
}
/**
* AVCodec initialization
*/
static int ac3_decode_init(AVCodecContext *avctx)
{
AC3DecodeContext *ctx = avctx->priv_data;
ctx->avctx = avctx;
ac3_common_init();
ac3_tables_init();
ff_mdct_init(&ctx->imdct_256, 8, 1);
ff_mdct_init(&ctx->imdct_512, 9, 1);
ac3_window_init(ctx->window);
dsputil_init(&ctx->dsp, avctx);
av_init_random(0, &ctx->dith_state);
/* set bias values for float to int16 conversion */
if(ctx->dsp.float_to_int16 == ff_float_to_int16_c) {
ctx->add_bias = 385.0f;
ctx->mul_bias = 1.0f;
} else {
ctx->add_bias = 0.0f;
ctx->mul_bias = 32767.0f;
}
return 0;
}
/**
* Parse the 'sync info' and 'bit stream info' from the AC-3 bitstream.
* GetBitContext within AC3DecodeContext must point to
* start of the synchronized ac3 bitstream.
*/
static int ac3_parse_header(AC3DecodeContext *ctx)
{
AC3HeaderInfo hdr;
GetBitContext *gb = &ctx->gb;
float cmixlev, surmixlev;
int err, i;
err = ff_ac3_parse_header(gb->buffer, &hdr);
if(err)
return err;
/* get decoding parameters from header info */
ctx->bit_alloc_params.fscod = hdr.fscod;
ctx->acmod = hdr.acmod;
cmixlev = gain_levels[clevs[hdr.cmixlev]];
surmixlev = gain_levels[slevs[hdr.surmixlev]];
ctx->dsurmod = hdr.dsurmod;
ctx->lfeon = hdr.lfeon;
ctx->bit_alloc_params.halfratecod = hdr.halfratecod;
ctx->sampling_rate = hdr.sample_rate;
ctx->bit_rate = hdr.bit_rate;
ctx->nchans = hdr.channels;
ctx->nfchans = ctx->nchans - ctx->lfeon;
ctx->lfe_ch = ctx->nfchans + 1;
ctx->frame_size = hdr.frame_size;
/* set default output to all source channels */
ctx->out_channels = ctx->nchans;
ctx->output_mode = ctx->acmod;
if(ctx->lfeon)
ctx->output_mode |= AC3_OUTPUT_LFEON;
/* skip over portion of header which has already been read */
skip_bits(gb, 16); // skip the sync_word
skip_bits(gb, 16); // skip crc1
skip_bits(gb, 8); // skip fscod and frmsizecod
skip_bits(gb, 11); // skip bsid, bsmod, and acmod
if(ctx->acmod == AC3_ACMOD_STEREO) {
skip_bits(gb, 2); // skip dsurmod
} else {
if((ctx->acmod & 1) && ctx->acmod != AC3_ACMOD_MONO)
skip_bits(gb, 2); // skip cmixlev
if(ctx->acmod & 4)
skip_bits(gb, 2); // skip surmixlev
}
skip_bits1(gb); // skip lfeon
/* read the rest of the bsi. read twice for dual mono mode. */
i = !(ctx->acmod);
do {
ctx->dialnorm[i] = dialnorm_tab[get_bits(gb, 5)]; // dialogue normalization
if (get_bits1(gb))
skip_bits(gb, 8); //skip compression
if (get_bits1(gb))
skip_bits(gb, 8); //skip language code
if (get_bits1(gb))
skip_bits(gb, 7); //skip audio production information
} while (i--);
skip_bits(gb, 2); //skip copyright bit and original bitstream bit
/* skip the timecodes (or extra bitstream information for Alternate Syntax)
TODO: read & use the xbsi1 downmix levels */
if (get_bits1(gb))
skip_bits(gb, 14); //skip timecode1 / xbsi1
if (get_bits1(gb))
skip_bits(gb, 14); //skip timecode2 / xbsi2
/* skip additional bitstream info */
if (get_bits1(gb)) {
i = get_bits(gb, 6);
do {
skip_bits(gb, 8);
} while(i--);
}
/* set stereo downmixing coefficients
reference: Section 7.8.2 Downmixing Into Two Channels */
for(i=0; i<ctx->nfchans; i++) {
ctx->downmix_coeffs[i][0] = gain_levels[ac3_default_coeffs[ctx->acmod][i][0]];
ctx->downmix_coeffs[i][1] = gain_levels[ac3_default_coeffs[ctx->acmod][i][1]];
}
if(ctx->acmod > 1 && ctx->acmod & 1) {
ctx->downmix_coeffs[1][0] = ctx->downmix_coeffs[1][1] = cmixlev;
}
if(ctx->acmod == AC3_ACMOD_2F1R || ctx->acmod == AC3_ACMOD_3F1R) {
int nf = ctx->acmod - 2;
ctx->downmix_coeffs[nf][0] = ctx->downmix_coeffs[nf][1] = surmixlev * LEVEL_MINUS_3DB;
}
if(ctx->acmod == AC3_ACMOD_2F2R || ctx->acmod == AC3_ACMOD_3F2R) {
int nf = ctx->acmod - 4;
ctx->downmix_coeffs[nf][0] = ctx->downmix_coeffs[nf+1][1] = surmixlev;
}
return 0;
}
/**
* Decode the grouped exponents according to exponent strategy.
* reference: Section 7.1.3 Exponent Decoding
*/
static void decode_exponents(GetBitContext *gb, int expstr, int ngrps,
uint8_t absexp, int8_t *dexps)
{
int i, j, grp, grpsize;
int dexp[256];
int expacc, prevexp;
/* unpack groups */
grpsize = expstr + (expstr == EXP_D45);
for(grp=0,i=0; grp<ngrps; grp++) {
expacc = get_bits(gb, 7);
dexp[i++] = exp_ungroup_tab[expacc][0];
dexp[i++] = exp_ungroup_tab[expacc][1];
dexp[i++] = exp_ungroup_tab[expacc][2];
}
/* convert to absolute exps and expand groups */
prevexp = absexp;
for(i=0; i<ngrps*3; i++) {
prevexp = av_clip(prevexp + dexp[i]-2, 0, 24);
for(j=0; j<grpsize; j++) {
dexps[(i*grpsize)+j] = prevexp;
}
}
}
/**
* Generate transform coefficients for each coupled channel in the coupling
* range using the coupling coefficients and coupling coordinates.
* reference: Section 7.4.3 Coupling Coordinate Format
*/
static void uncouple_channels(AC3DecodeContext *ctx)
{
int i, j, ch, bnd, subbnd;
subbnd = -1;
i = ctx->startmant[CPL_CH];
for(bnd=0; bnd<ctx->ncplbnd; bnd++) {
do {
subbnd++;
for(j=0; j<12; j++) {
for(ch=1; ch<=ctx->nfchans; ch++) {
if(ctx->chincpl[ch])
ctx->transform_coeffs[ch][i] = ctx->transform_coeffs[CPL_CH][i] * ctx->cplco[ch][bnd] * 8.0f;
}
i++;
}
} while(ctx->cplbndstrc[subbnd]);
}
}
/**
* Grouped mantissas for 3-level 5-level and 11-level quantization
*/
typedef struct {
float b1_mant[3];
float b2_mant[3];
float b4_mant[2];
int b1ptr;
int b2ptr;
int b4ptr;
} mant_groups;
/**
* Get the transform coefficients for a particular channel
* reference: Section 7.3 Quantization and Decoding of Mantissas
*/
static int get_transform_coeffs_ch(AC3DecodeContext *ctx, int ch_index, mant_groups *m)
{
GetBitContext *gb = &ctx->gb;
int i, gcode, tbap, start, end;
uint8_t *exps;
uint8_t *bap;
float *coeffs;
exps = ctx->dexps[ch_index];
bap = ctx->bap[ch_index];
coeffs = ctx->transform_coeffs[ch_index];
start = ctx->startmant[ch_index];
end = ctx->endmant[ch_index];
for (i = start; i < end; i++) {
tbap = bap[i];
switch (tbap) {
case 0:
coeffs[i] = ((av_random(&ctx->dith_state) & 0xFFFF) * LEVEL_MINUS_3DB) / 32768.0f;
break;
case 1:
if(m->b1ptr > 2) {
gcode = get_bits(gb, 5);
m->b1_mant[0] = b1_mantissas[gcode][0];
m->b1_mant[1] = b1_mantissas[gcode][1];
m->b1_mant[2] = b1_mantissas[gcode][2];
m->b1ptr = 0;
}
coeffs[i] = m->b1_mant[m->b1ptr++];
break;
case 2:
if(m->b2ptr > 2) {
gcode = get_bits(gb, 7);
m->b2_mant[0] = b2_mantissas[gcode][0];
m->b2_mant[1] = b2_mantissas[gcode][1];
m->b2_mant[2] = b2_mantissas[gcode][2];
m->b2ptr = 0;
}
coeffs[i] = m->b2_mant[m->b2ptr++];
break;
case 3:
coeffs[i] = b3_mantissas[get_bits(gb, 3)];
break;
case 4:
if(m->b4ptr > 1) {
gcode = get_bits(gb, 7);
m->b4_mant[0] = b4_mantissas[gcode][0];
m->b4_mant[1] = b4_mantissas[gcode][1];
m->b4ptr = 0;
}
coeffs[i] = m->b4_mant[m->b4ptr++];
break;
case 5:
coeffs[i] = b5_mantissas[get_bits(gb, 4)];
break;
default:
/* asymmetric dequantization */
coeffs[i] = get_sbits(gb, qntztab[tbap]) * scale_factors[qntztab[tbap]-1];
break;
}
coeffs[i] *= scale_factors[exps[i]];
}
return 0;
}
/**
* Remove random dithering from coefficients with zero-bit mantissas
* reference: Section 7.3.4 Dither for Zero Bit Mantissas (bap=0)
*/
static void remove_dithering(AC3DecodeContext *ctx) {
int ch, i;
int end=0;
float *coeffs;
uint8_t *bap;
for(ch=1; ch<=ctx->nfchans; ch++) {
if(!ctx->dithflag[ch]) {
coeffs = ctx->transform_coeffs[ch];
bap = ctx->bap[ch];
if(ctx->chincpl[ch])
end = ctx->startmant[CPL_CH];
else
end = ctx->endmant[ch];
for(i=0; i<end; i++) {
if(bap[i] == 0)
coeffs[i] = 0.0f;
}
if(ctx->chincpl[ch]) {
bap = ctx->bap[CPL_CH];
for(; i<ctx->endmant[CPL_CH]; i++) {
if(bap[i] == 0)
coeffs[i] = 0.0f;
}
}
}
}
}
/**
* Get the transform coefficients.
*/
static int get_transform_coeffs(AC3DecodeContext * ctx)
{
int ch, end;
int got_cplchan = 0;
mant_groups m;
m.b1ptr = m.b2ptr = m.b4ptr = 3;
for (ch = 1; ch <= ctx->nchans; ch++) {
/* transform coefficients for full-bandwidth channel */
if (get_transform_coeffs_ch(ctx, ch, &m))
return -1;
/* tranform coefficients for coupling channel come right after the
coefficients for the first coupled channel*/
if (ctx->chincpl[ch]) {
if (!got_cplchan) {
if (get_transform_coeffs_ch(ctx, CPL_CH, &m)) {
av_log(ctx->avctx, AV_LOG_ERROR, "error in decoupling channels\n");
return -1;
}
uncouple_channels(ctx);
got_cplchan = 1;
}
end = ctx->endmant[CPL_CH];
} else {
end = ctx->endmant[ch];
}
do
ctx->transform_coeffs[ch][end] = 0;
while(++end < 256);
}
/* if any channel doesn't use dithering, zero appropriate coefficients */
if(!ctx->dither_all)
remove_dithering(ctx);
return 0;
}
/**
* Stereo rematrixing.
* reference: Section 7.5.4 Rematrixing : Decoding Technique
*/
static void do_rematrixing(AC3DecodeContext *ctx)
{
int bnd, i;
int end, bndend;
float tmp0, tmp1;
end = FFMIN(ctx->endmant[1], ctx->endmant[2]);
for(bnd=0; bnd<ctx->nrematbnd; bnd++) {
if(ctx->rematflg[bnd]) {
bndend = FFMIN(end, rematrix_band_tab[bnd+1]);
for(i=rematrix_band_tab[bnd]; i<bndend; i++) {
tmp0 = ctx->transform_coeffs[1][i];
tmp1 = ctx->transform_coeffs[2][i];
ctx->transform_coeffs[1][i] = tmp0 + tmp1;
ctx->transform_coeffs[2][i] = tmp0 - tmp1;
}
}
}
}
/**
* Perform the 256-point IMDCT
*/
static void do_imdct_256(AC3DecodeContext *ctx, int chindex)
{
int i, k;
DECLARE_ALIGNED_16(float, x[128]);
FFTComplex z[2][64];
float *o_ptr = ctx->tmp_output;
for(i=0; i<2; i++) {
/* de-interleave coefficients */
for(k=0; k<128; k++) {
x[k] = ctx->transform_coeffs[chindex][2*k+i];
}
/* run standard IMDCT */
ctx->imdct_256.fft.imdct_calc(&ctx->imdct_256, o_ptr, x, ctx->tmp_imdct);
/* reverse the post-rotation & reordering from standard IMDCT */
for(k=0; k<32; k++) {
z[i][32+k].re = -o_ptr[128+2*k];
z[i][32+k].im = -o_ptr[2*k];
z[i][31-k].re = o_ptr[2*k+1];
z[i][31-k].im = o_ptr[128+2*k+1];
}
}
/* apply AC-3 post-rotation & reordering */
for(k=0; k<64; k++) {
o_ptr[ 2*k ] = -z[0][ k].im;
o_ptr[ 2*k+1] = z[0][63-k].re;
o_ptr[128+2*k ] = -z[0][ k].re;
o_ptr[128+2*k+1] = z[0][63-k].im;
o_ptr[256+2*k ] = -z[1][ k].re;
o_ptr[256+2*k+1] = z[1][63-k].im;
o_ptr[384+2*k ] = z[1][ k].im;
o_ptr[384+2*k+1] = -z[1][63-k].re;
}
}
/**
* Inverse MDCT Transform.
* Convert frequency domain coefficients to time-domain audio samples.
* reference: Section 7.9.4 Transformation Equations
*/
static inline void do_imdct(AC3DecodeContext *ctx)
{
int ch;
int nchans;
/* Don't perform the IMDCT on the LFE channel unless it's used in the output */
nchans = ctx->nfchans;
if(ctx->output_mode & AC3_OUTPUT_LFEON)
nchans++;
for (ch=1; ch<=nchans; ch++) {
if (ctx->blksw[ch]) {
do_imdct_256(ctx, ch);
} else {
ctx->imdct_512.fft.imdct_calc(&ctx->imdct_512, ctx->tmp_output,
ctx->transform_coeffs[ch],
ctx->tmp_imdct);
}
/* For the first half of the block, apply the window, add the delay
from the previous block, and send to output */
ctx->dsp.vector_fmul_add_add(ctx->output[ch-1], ctx->tmp_output,
ctx->window, ctx->delay[ch-1], 0, 256, 1);
/* For the second half of the block, apply the window and store the
samples to delay, to be combined with the next block */
ctx->dsp.vector_fmul_reverse(ctx->delay[ch-1], ctx->tmp_output+256,
ctx->window, 256);
}
}
/**
* Downmix the output to mono or stereo.
*/
static void ac3_downmix(float samples[AC3_MAX_CHANNELS][256], int nfchans,
int output_mode, float coef[AC3_MAX_CHANNELS][2])
{
int i, j;
float v0, v1, s0, s1;
for(i=0; i<256; i++) {
v0 = v1 = s0 = s1 = 0.0f;
for(j=0; j<nfchans; j++) {
v0 += samples[j][i] * coef[j][0];
v1 += samples[j][i] * coef[j][1];
s0 += coef[j][0];
s1 += coef[j][1];
}
v0 /= s0;
v1 /= s1;
if(output_mode == AC3_ACMOD_MONO) {
samples[0][i] = (v0 + v1) * LEVEL_MINUS_3DB;
} else if(output_mode == AC3_ACMOD_STEREO) {
samples[0][i] = v0;
samples[1][i] = v1;
}
}
}
/**
* Parse an audio block from AC-3 bitstream.
*/
static int ac3_parse_audio_block(AC3DecodeContext *ctx, int blk)
{
int nfchans = ctx->nfchans;
int acmod = ctx->acmod;
int i, bnd, seg, ch;
GetBitContext *gb = &ctx->gb;
uint8_t bit_alloc_stages[AC3_MAX_CHANNELS];
memset(bit_alloc_stages, 0, AC3_MAX_CHANNELS);
/* block switch flags */
for (ch = 1; ch <= nfchans; ch++)
ctx->blksw[ch] = get_bits1(gb);
/* dithering flags */
ctx->dither_all = 1;
for (ch = 1; ch <= nfchans; ch++) {
ctx->dithflag[ch] = get_bits1(gb);
if(!ctx->dithflag[ch])
ctx->dither_all = 0;
}
/* dynamic range */
i = !(ctx->acmod);
do {
if(get_bits1(gb)) {
ctx->dynrng[i] = dynrng_tab[get_bits(gb, 8)];
} else if(blk == 0) {
ctx->dynrng[i] = 1.0f;
}
} while(i--);
/* coupling strategy */
if (get_bits1(gb)) {
memset(bit_alloc_stages, 3, AC3_MAX_CHANNELS);
ctx->cplinu = get_bits1(gb);
if (ctx->cplinu) {
/* coupling in use */
int cplbegf, cplendf;
/* determine which channels are coupled */
for (ch = 1; ch <= nfchans; ch++)
ctx->chincpl[ch] = get_bits1(gb);
/* phase flags in use */
if (acmod == AC3_ACMOD_STEREO)
ctx->phsflginu = get_bits1(gb);
/* coupling frequency range and band structure */
cplbegf = get_bits(gb, 4);
cplendf = get_bits(gb, 4);
if (3 + cplendf - cplbegf < 0) {
av_log(ctx->avctx, AV_LOG_ERROR, "cplendf = %d < cplbegf = %d\n", cplendf, cplbegf);
return -1;
}
ctx->ncplbnd = ctx->ncplsubnd = 3 + cplendf - cplbegf;
ctx->startmant[CPL_CH] = cplbegf * 12 + 37;
ctx->endmant[CPL_CH] = cplendf * 12 + 73;
for (bnd = 0; bnd < ctx->ncplsubnd - 1; bnd++) {
if (get_bits1(gb)) {
ctx->cplbndstrc[bnd] = 1;
ctx->ncplbnd--;
}
}
} else {
/* coupling not in use */
for (ch = 1; ch <= nfchans; ch++)
ctx->chincpl[ch] = 0;
}
}
/* coupling coordinates */
if (ctx->cplinu) {
int cplcoe = 0;
for (ch = 1; ch <= nfchans; ch++) {
if (ctx->chincpl[ch]) {
if (get_bits1(gb)) {
int mstrcplco, cplcoexp, cplcomant;
cplcoe = 1;
mstrcplco = 3 * get_bits(gb, 2);
for (bnd = 0; bnd < ctx->ncplbnd; bnd++) {
cplcoexp = get_bits(gb, 4);
cplcomant = get_bits(gb, 4);
if (cplcoexp == 15)
ctx->cplco[ch][bnd] = cplcomant / 16.0f;
else
ctx->cplco[ch][bnd] = (cplcomant + 16.0f) / 32.0f;
ctx->cplco[ch][bnd] *= scale_factors[cplcoexp + mstrcplco];
}
}
}
}
/* phase flags */
if (acmod == AC3_ACMOD_STEREO && ctx->phsflginu && cplcoe) {
for (bnd = 0; bnd < ctx->ncplbnd; bnd++) {
if (get_bits1(gb))
ctx->cplco[2][bnd] = -ctx->cplco[2][bnd];
}
}
}
/* stereo rematrixing strategy and band structure */
if (acmod == AC3_ACMOD_STEREO) {
ctx->rematstr = get_bits1(gb);
if (ctx->rematstr) {
ctx->nrematbnd = 4;
if(ctx->cplinu && ctx->startmant[CPL_CH] <= 61)
ctx->nrematbnd -= 1 + (ctx->startmant[CPL_CH] == 37);
for(bnd=0; bnd<ctx->nrematbnd; bnd++)
ctx->rematflg[bnd] = get_bits1(gb);
}
}
/* exponent strategies for each channel */
ctx->expstr[CPL_CH] = EXP_REUSE;
ctx->expstr[ctx->lfe_ch] = EXP_REUSE;
for (ch = !ctx->cplinu; ch <= ctx->nchans; ch++) {
if(ch == ctx->lfe_ch)
ctx->expstr[ch] = get_bits(gb, 1);
else
ctx->expstr[ch] = get_bits(gb, 2);
if(ctx->expstr[ch] != EXP_REUSE)
bit_alloc_stages[ch] = 3;
}
/* channel bandwidth */
for (ch = 1; ch <= nfchans; ch++) {
ctx->startmant[ch] = 0;
if (ctx->expstr[ch] != EXP_REUSE) {
int prev = ctx->endmant[ch];
if (ctx->chincpl[ch])
ctx->endmant[ch] = ctx->startmant[CPL_CH];
else {
int chbwcod = get_bits(gb, 6);
if (chbwcod > 60) {
av_log(ctx->avctx, AV_LOG_ERROR, "chbwcod = %d > 60", chbwcod);
return -1;
}
ctx->endmant[ch] = chbwcod * 3 + 73;
}
if(blk > 0 && ctx->endmant[ch] != prev)
memset(bit_alloc_stages, 3, AC3_MAX_CHANNELS);
}
}
ctx->startmant[ctx->lfe_ch] = 0;
ctx->endmant[ctx->lfe_ch] = 7;
/* decode exponents for each channel */
for (ch = !ctx->cplinu; ch <= ctx->nchans; ch++) {
if (ctx->expstr[ch] != EXP_REUSE) {
int grpsize, ngrps;
grpsize = 3 << (ctx->expstr[ch] - 1);
if(ch == CPL_CH)
ngrps = (ctx->endmant[ch] - ctx->startmant[ch]) / grpsize;
else if(ch == ctx->lfe_ch)
ngrps = 2;
else
ngrps = (ctx->endmant[ch] + grpsize - 4) / grpsize;
ctx->dexps[ch][0] = get_bits(gb, 4) << !ch;
decode_exponents(gb, ctx->expstr[ch], ngrps, ctx->dexps[ch][0],
&ctx->dexps[ch][ctx->startmant[ch]+!!ch]);
if(ch != CPL_CH && ch != ctx->lfe_ch)
skip_bits(gb, 2); /* skip gainrng */
}
}
/* bit allocation information */
if (get_bits1(gb)) {
ctx->bit_alloc_params.sdecay = ff_sdecaytab[get_bits(gb, 2)] >> ctx->bit_alloc_params.halfratecod;
ctx->bit_alloc_params.fdecay = ff_fdecaytab[get_bits(gb, 2)] >> ctx->bit_alloc_params.halfratecod;
ctx->bit_alloc_params.sgain = ff_sgaintab[get_bits(gb, 2)];
ctx->bit_alloc_params.dbknee = ff_dbkneetab[get_bits(gb, 2)];
ctx->bit_alloc_params.floor = ff_floortab[get_bits(gb, 3)];
for(ch=!ctx->cplinu; ch<=ctx->nchans; ch++) {
bit_alloc_stages[ch] = FFMAX(bit_alloc_stages[ch], 2);
}
}
/* signal-to-noise ratio offsets and fast gains (signal-to-mask ratios) */
if (get_bits1(gb)) {
int csnr;
csnr = (get_bits(gb, 6) - 15) << 4;
for (ch = !ctx->cplinu; ch <= ctx->nchans; ch++) { /* snr offset and fast gain */
ctx->snroffst[ch] = (csnr + get_bits(gb, 4)) << 2;
ctx->fgain[ch] = ff_fgaintab[get_bits(gb, 3)];
}
memset(bit_alloc_stages, 3, AC3_MAX_CHANNELS);
}
/* coupling leak information */
if (ctx->cplinu && get_bits1(gb)) {
ctx->bit_alloc_params.cplfleak = get_bits(gb, 3);
ctx->bit_alloc_params.cplsleak = get_bits(gb, 3);
bit_alloc_stages[CPL_CH] = FFMAX(bit_alloc_stages[CPL_CH], 2);
}
/* delta bit allocation information */
if (get_bits1(gb)) {
/* delta bit allocation exists (strategy) */
for (ch = !ctx->cplinu; ch <= nfchans; ch++) {
ctx->deltbae[ch] = get_bits(gb, 2);
if (ctx->deltbae[ch] == DBA_RESERVED) {
av_log(ctx->avctx, AV_LOG_ERROR, "delta bit allocation strategy reserved\n");
return -1;
}
bit_alloc_stages[ch] = FFMAX(bit_alloc_stages[ch], 2);
}
/* channel delta offset, len and bit allocation */
for (ch = !ctx->cplinu; ch <= nfchans; ch++) {
if (ctx->deltbae[ch] == DBA_NEW) {
ctx->deltnseg[ch] = get_bits(gb, 3);
for (seg = 0; seg <= ctx->deltnseg[ch]; seg++) {
ctx->deltoffst[ch][seg] = get_bits(gb, 5);
ctx->deltlen[ch][seg] = get_bits(gb, 4);
ctx->deltba[ch][seg] = get_bits(gb, 3);
}
}
}
} else if(blk == 0) {
for(ch=0; ch<=ctx->nchans; ch++) {
ctx->deltbae[ch] = DBA_NONE;
}
}
/* Bit allocation */
for(ch=!ctx->cplinu; ch<=ctx->nchans; ch++) {
if(bit_alloc_stages[ch] > 2) {
/* Exponent mapping into PSD and PSD integration */
ff_ac3_bit_alloc_calc_psd(ctx->dexps[ch],
ctx->startmant[ch], ctx->endmant[ch],
ctx->psd[ch], ctx->bndpsd[ch]);
}
if(bit_alloc_stages[ch] > 1) {
/* Compute excitation function, Compute masking curve, and
Apply delta bit allocation */
ff_ac3_bit_alloc_calc_mask(&ctx->bit_alloc_params, ctx->bndpsd[ch],
ctx->startmant[ch], ctx->endmant[ch],
ctx->fgain[ch], (ch == ctx->lfe_ch),
ctx->deltbae[ch], ctx->deltnseg[ch],
ctx->deltoffst[ch], ctx->deltlen[ch],
ctx->deltba[ch], ctx->mask[ch]);
}
if(bit_alloc_stages[ch] > 0) {
/* Compute bit allocation */
ff_ac3_bit_alloc_calc_bap(ctx->mask[ch], ctx->psd[ch],
ctx->startmant[ch], ctx->endmant[ch],
ctx->snroffst[ch],
ctx->bit_alloc_params.floor,
ctx->bap[ch]);
}
}
/* unused dummy data */
if (get_bits1(gb)) {
int skipl = get_bits(gb, 9);
while(skipl--)
skip_bits(gb, 8);
}
/* unpack the transform coefficients
this also uncouples channels if coupling is in use. */
if (get_transform_coeffs(ctx)) {
av_log(ctx->avctx, AV_LOG_ERROR, "Error in routine get_transform_coeffs\n");
return -1;
}
/* recover coefficients if rematrixing is in use */
if(ctx->acmod == AC3_ACMOD_STEREO)
do_rematrixing(ctx);
/* apply scaling to coefficients (headroom, dialnorm, dynrng) */
for(ch=1; ch<=ctx->nchans; ch++) {
float gain = 2.0f * ctx->mul_bias;
if(ctx->acmod == AC3_ACMOD_DUALMONO) {
gain *= ctx->dialnorm[ch-1] * ctx->dynrng[ch-1];
} else {
gain *= ctx->dialnorm[0] * ctx->dynrng[0];
}
for(i=0; i<ctx->endmant[ch]; i++) {
ctx->transform_coeffs[ch][i] *= gain;
}
}
do_imdct(ctx);
/* downmix output if needed */
if(ctx->nchans != ctx->out_channels && !((ctx->output_mode & AC3_OUTPUT_LFEON) &&
ctx->nfchans == ctx->out_channels)) {
ac3_downmix(ctx->output, ctx->nfchans, ctx->output_mode,
ctx->downmix_coeffs);
}
/* convert float to 16-bit integer */
for(ch=0; ch<ctx->out_channels; ch++) {
for(i=0; i<256; i++) {
ctx->output[ch][i] += ctx->add_bias;
}
ctx->dsp.float_to_int16(ctx->int_output[ch], ctx->output[ch], 256);
}
return 0;
}
/**
* Decode a single AC-3 frame.
*/
static int ac3_decode_frame(AVCodecContext * avctx, void *data, int *data_size, uint8_t *buf, int buf_size)
{
AC3DecodeContext *ctx = (AC3DecodeContext *)avctx->priv_data;
int16_t *out_samples = (int16_t *)data;
int i, blk, ch;
/* initialize the GetBitContext with the start of valid AC-3 Frame */
init_get_bits(&ctx->gb, buf, buf_size * 8);
/* parse the syncinfo */
if (ac3_parse_header(ctx)) {
av_log(avctx, AV_LOG_ERROR, "\n");
*data_size = 0;
return buf_size;
}
avctx->sample_rate = ctx->sampling_rate;
avctx->bit_rate = ctx->bit_rate;
/* channel config */
ctx->out_channels = ctx->nchans;
if (avctx->channels == 0) {
avctx->channels = ctx->out_channels;
} else if(ctx->out_channels < avctx->channels) {
av_log(avctx, AV_LOG_ERROR, "Cannot upmix AC3 from %d to %d channels.\n",
ctx->out_channels, avctx->channels);
return -1;
}
if(avctx->channels == 2) {
ctx->output_mode = AC3_ACMOD_STEREO;
} else if(avctx->channels == 1) {
ctx->output_mode = AC3_ACMOD_MONO;
} else if(avctx->channels != ctx->out_channels) {
av_log(avctx, AV_LOG_ERROR, "Cannot downmix AC3 from %d to %d channels.\n",
ctx->out_channels, avctx->channels);
return -1;
}
ctx->out_channels = avctx->channels;
/* parse the audio blocks */
for (blk = 0; blk < NB_BLOCKS; blk++) {
if (ac3_parse_audio_block(ctx, blk)) {
av_log(avctx, AV_LOG_ERROR, "error parsing the audio block\n");
*data_size = 0;
return ctx->frame_size;
}
for (i = 0; i < 256; i++)
for (ch = 0; ch < ctx->out_channels; ch++)
*(out_samples++) = ctx->int_output[ch][i];
}
*data_size = NB_BLOCKS * 256 * avctx->channels * sizeof (int16_t);
return ctx->frame_size;
}
/**
* Uninitialize the AC-3 decoder.
*/
static int ac3_decode_end(AVCodecContext *avctx)
{
AC3DecodeContext *ctx = (AC3DecodeContext *)avctx->priv_data;
ff_mdct_end(&ctx->imdct_512);
ff_mdct_end(&ctx->imdct_256);
return 0;
}
AVCodec ac3_decoder = {
.name = "ac3",
.type = CODEC_TYPE_AUDIO,
.id = CODEC_ID_AC3,
.priv_data_size = sizeof (AC3DecodeContext),
.init = ac3_decode_init,
.close = ac3_decode_end,
.decode = ac3_decode_frame,
};
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