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FFmpeg/libavcodec/ac3dec.c

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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 "libavutil/crc.h"
#include "libavutil/random.h"
#include "avcodec.h"
#include "ac3_parser.h"
#include "bitstream.h"
#include "dsputil.h"
#include "ac3dec.h"
#include "ac3dec_data.h"
/** Maximum possible frame size when the specification limit is ignored */
#define AC3_MAX_FRAME_SIZE 21695
/** table for grouping exponents */
static uint8_t exp_ungroup_tab[128][3];
/** tables for ungrouping mantissas */
static int b1_mantissas[32][3];
static int b2_mantissas[128][3];
static int b3_mantissas[8];
static int b4_mantissas[128][2];
static int b5_mantissas[16];
/**
* Quantization table: levels for symmetric. bits for asymmetric.
* reference: Table 7.18 Mapping of bap to Quantizer
*/
static const uint8_t quantization_tab[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 dynamic_range_tab[256];
/** Adjustments in dB gain */
#define LEVEL_PLUS_3DB 1.4142135623730950
#define LEVEL_PLUS_1POINT5DB 1.1892071150027209
#define LEVEL_MINUS_1POINT5DB 0.8408964152537145
#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[9] = {
LEVEL_PLUS_3DB,
LEVEL_PLUS_1POINT5DB,
LEVEL_ONE,
LEVEL_MINUS_1POINT5DB,
LEVEL_MINUS_3DB,
LEVEL_MINUS_4POINT5DB,
LEVEL_MINUS_6DB,
LEVEL_ZERO,
LEVEL_MINUS_9DB
};
/**
* Table for center mix levels
* reference: Section 5.4.2.4 cmixlev
*/
static const uint8_t center_levels[4] = { 4, 5, 6, 5 };
/**
* Table for surround mix levels
* reference: Section 5.4.2.5 surmixlev
*/
static const uint8_t surround_levels[4] = { 4, 6, 7, 6 };
/**
* 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] = {
{ { 2, 7 }, { 7, 2 }, },
{ { 4, 4 }, },
{ { 2, 7 }, { 7, 2 }, },
{ { 2, 7 }, { 5, 5 }, { 7, 2 }, },
{ { 2, 7 }, { 7, 2 }, { 6, 6 }, },
{ { 2, 7 }, { 5, 5 }, { 7, 2 }, { 8, 8 }, },
{ { 2, 7 }, { 7, 2 }, { 6, 7 }, { 7, 6 }, },
{ { 2, 7 }, { 5, 5 }, { 7, 2 }, { 6, 7 }, { 7, 6 }, },
};
/**
* Symmetrical Dequantization
* reference: Section 7.3.3 Expansion of Mantissas for Symmetrical Quantization
* Tables 7.19 to 7.23
*/
static inline int
symmetric_dequant(int code, int levels)
{
return ((code - (levels >> 1)) << 24) / levels;
}
/*
* Initialize tables at runtime.
*/
static av_cold 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;
dynamic_range_tab[i] = powf(2.0f, v) * ((i & 0x1F) | 0x20);
}
/* 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 av_cold int ac3_decode_init(AVCodecContext *avctx)
{
AC3DecodeContext *s = avctx->priv_data;
s->avctx = avctx;
ac3_common_init();
ac3_tables_init();
ff_mdct_init(&s->imdct_256, 8, 1);
ff_mdct_init(&s->imdct_512, 9, 1);
ff_kbd_window_init(s->window, 5.0, 256);
dsputil_init(&s->dsp, avctx);
av_init_random(0, &s->dith_state);
/* set bias values for float to int16 conversion */
if(s->dsp.float_to_int16 == ff_float_to_int16_c) {
s->add_bias = 385.0f;
s->mul_bias = 1.0f;
} else {
s->add_bias = 0.0f;
s->mul_bias = 32767.0f;
}
/* allow downmixing to stereo or mono */
if (avctx->channels > 0 && avctx->request_channels > 0 &&
avctx->request_channels < avctx->channels &&
avctx->request_channels <= 2) {
avctx->channels = avctx->request_channels;
}
s->downmixed = 1;
/* allocate context input buffer */
if (avctx->error_resilience >= FF_ER_CAREFUL) {
s->input_buffer = av_mallocz(AC3_MAX_FRAME_SIZE + FF_INPUT_BUFFER_PADDING_SIZE);
if (!s->input_buffer)
return AVERROR_NOMEM;
}
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 *s)
{
GetBitContext *gbc = &s->gbc;
int i;
/* read the rest of the bsi. read twice for dual mono mode. */
i = !(s->channel_mode);
do {
skip_bits(gbc, 5); // skip dialog normalization
if (get_bits1(gbc))
skip_bits(gbc, 8); //skip compression
if (get_bits1(gbc))
skip_bits(gbc, 8); //skip language code
if (get_bits1(gbc))
skip_bits(gbc, 7); //skip audio production information
} while (i--);
skip_bits(gbc, 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(gbc))
skip_bits(gbc, 14); //skip timecode1 / xbsi1
if (get_bits1(gbc))
skip_bits(gbc, 14); //skip timecode2 / xbsi2
/* skip additional bitstream info */
if (get_bits1(gbc)) {
i = get_bits(gbc, 6);
do {
skip_bits(gbc, 8);
} while(i--);
}
return 0;
}
/**
* Common function to parse AC3 or E-AC3 frame header
*/
static int parse_frame_header(AC3DecodeContext *s)
{
AC3HeaderInfo hdr;
GetBitContext *gbc = &s->gbc;
int err;
err = ff_ac3_parse_header(gbc, &hdr);
if(err)
return err;
if(hdr.bitstream_id > 10)
return AC3_PARSE_ERROR_BSID;
/* get decoding parameters from header info */
s->bit_alloc_params.sr_code = hdr.sr_code;
s->channel_mode = hdr.channel_mode;
s->lfe_on = hdr.lfe_on;
s->bit_alloc_params.sr_shift = hdr.sr_shift;
s->sample_rate = hdr.sample_rate;
s->bit_rate = hdr.bit_rate;
s->channels = hdr.channels;
s->fbw_channels = s->channels - s->lfe_on;
s->lfe_ch = s->fbw_channels + 1;
s->frame_size = hdr.frame_size;
s->center_mix_level = hdr.center_mix_level;
s->surround_mix_level = hdr.surround_mix_level;
s->num_blocks = hdr.num_blocks;
s->frame_type = hdr.frame_type;
s->substreamid = hdr.substreamid;
if(s->lfe_on) {
s->start_freq[s->lfe_ch] = 0;
s->end_freq[s->lfe_ch] = 7;
s->num_exp_groups[s->lfe_ch] = 2;
s->channel_in_cpl[s->lfe_ch] = 0;
}
return ac3_parse_header(s);
}
/**
* Set stereo downmixing coefficients based on frame header info.
* reference: Section 7.8.2 Downmixing Into Two Channels
*/
static void set_downmix_coeffs(AC3DecodeContext *s)
{
int i;
float cmix = gain_levels[center_levels[s->center_mix_level]];
float smix = gain_levels[surround_levels[s->surround_mix_level]];
for(i=0; i<s->fbw_channels; i++) {
s->downmix_coeffs[i][0] = gain_levels[ac3_default_coeffs[s->channel_mode][i][0]];
s->downmix_coeffs[i][1] = gain_levels[ac3_default_coeffs[s->channel_mode][i][1]];
}
if(s->channel_mode > 1 && s->channel_mode & 1) {
s->downmix_coeffs[1][0] = s->downmix_coeffs[1][1] = cmix;
}
if(s->channel_mode == AC3_CHMODE_2F1R || s->channel_mode == AC3_CHMODE_3F1R) {
int nf = s->channel_mode - 2;
s->downmix_coeffs[nf][0] = s->downmix_coeffs[nf][1] = smix * LEVEL_MINUS_3DB;
}
if(s->channel_mode == AC3_CHMODE_2F2R || s->channel_mode == AC3_CHMODE_3F2R) {
int nf = s->channel_mode - 4;
s->downmix_coeffs[nf][0] = s->downmix_coeffs[nf+1][1] = smix;
}
/* calculate adjustment needed for each channel to avoid clipping */
s->downmix_coeff_adjust[0] = s->downmix_coeff_adjust[1] = 0.0f;
for(i=0; i<s->fbw_channels; i++) {
s->downmix_coeff_adjust[0] += s->downmix_coeffs[i][0];
s->downmix_coeff_adjust[1] += s->downmix_coeffs[i][1];
}
s->downmix_coeff_adjust[0] = 1.0f / s->downmix_coeff_adjust[0];
s->downmix_coeff_adjust[1] = 1.0f / s->downmix_coeff_adjust[1];
}
/**
* Decode the grouped exponents according to exponent strategy.
* reference: Section 7.1.3 Exponent Decoding
*/
static void decode_exponents(GetBitContext *gbc, int exp_strategy, int ngrps,
uint8_t absexp, int8_t *dexps)
{
int i, j, grp, group_size;
int dexp[256];
int expacc, prevexp;
/* unpack groups */
group_size = exp_strategy + (exp_strategy == EXP_D45);
for(grp=0,i=0; grp<ngrps; grp++) {
expacc = get_bits(gbc, 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<group_size; j++) {
dexps[(i*group_size)+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 *s)
{
int i, j, ch, bnd, subbnd;
subbnd = -1;
i = s->start_freq[CPL_CH];
for(bnd=0; bnd<s->num_cpl_bands; bnd++) {
do {
subbnd++;
for(j=0; j<12; j++) {
for(ch=1; ch<=s->fbw_channels; ch++) {
if(s->channel_in_cpl[ch]) {
s->fixed_coeffs[ch][i] = ((int64_t)s->fixed_coeffs[CPL_CH][i] * (int64_t)s->cpl_coords[ch][bnd]) >> 23;
if (ch == 2 && s->phase_flags[bnd])
s->fixed_coeffs[ch][i] = -s->fixed_coeffs[ch][i];
}
}
i++;
}
} while(s->cpl_band_struct[subbnd]);
}
}
/**
* Grouped mantissas for 3-level 5-level and 11-level quantization
*/
typedef struct {
int b1_mant[3];
int b2_mant[3];
int 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 void get_transform_coeffs_ch(AC3DecodeContext *s, int ch_index, mant_groups *m)
{
GetBitContext *gbc = &s->gbc;
int i, gcode, tbap, start, end;
uint8_t *exps;
uint8_t *bap;
int *coeffs;
exps = s->dexps[ch_index];
bap = s->bap[ch_index];
coeffs = s->fixed_coeffs[ch_index];
start = s->start_freq[ch_index];
end = s->end_freq[ch_index];
for (i = start; i < end; i++) {
tbap = bap[i];
switch (tbap) {
case 0:
coeffs[i] = (av_random(&s->dith_state) & 0x7FFFFF) - 4194304;
break;
case 1:
if(m->b1ptr > 2) {
gcode = get_bits(gbc, 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(gbc, 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(gbc, 3)];
break;
case 4:
if(m->b4ptr > 1) {
gcode = get_bits(gbc, 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(gbc, 4)];
break;
default: {
/* asymmetric dequantization */
int qlevel = quantization_tab[tbap];
coeffs[i] = get_sbits(gbc, qlevel) << (24 - qlevel);
break;
}
}
coeffs[i] >>= exps[i];
}
}
/**
* 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 *s) {
int ch, i;
int end=0;
int *coeffs;
uint8_t *bap;
for(ch=1; ch<=s->fbw_channels; ch++) {
if(!s->dither_flag[ch]) {
coeffs = s->fixed_coeffs[ch];
bap = s->bap[ch];
if(s->channel_in_cpl[ch])
end = s->start_freq[CPL_CH];
else
end = s->end_freq[ch];
for(i=0; i<end; i++) {
if(!bap[i])
coeffs[i] = 0;
}
if(s->channel_in_cpl[ch]) {
bap = s->bap[CPL_CH];
for(; i<s->end_freq[CPL_CH]; i++) {
if(!bap[i])
coeffs[i] = 0;
}
}
}
}
}
/**
* Get the transform coefficients.
*/
static void get_transform_coeffs(AC3DecodeContext *s)
{
int ch, end;
int got_cplchan = 0;
mant_groups m;
m.b1ptr = m.b2ptr = m.b4ptr = 3;
for (ch = 1; ch <= s->channels; ch++) {
/* transform coefficients for full-bandwidth channel */
get_transform_coeffs_ch(s, ch, &m);
/* tranform coefficients for coupling channel come right after the
coefficients for the first coupled channel*/
if (s->channel_in_cpl[ch]) {
if (!got_cplchan) {
get_transform_coeffs_ch(s, CPL_CH, &m);
uncouple_channels(s);
got_cplchan = 1;
}
end = s->end_freq[CPL_CH];
} else {
end = s->end_freq[ch];
}
do
s->fixed_coeffs[ch][end] = 0;
while(++end < 256);
}
/* if any channel doesn't use dithering, zero appropriate coefficients */
if(!s->dither_all)
remove_dithering(s);
}
/**
* Stereo rematrixing.
* reference: Section 7.5.4 Rematrixing : Decoding Technique
*/
static void do_rematrixing(AC3DecodeContext *s)
{
int bnd, i;
int end, bndend;
int tmp0, tmp1;
end = FFMIN(s->end_freq[1], s->end_freq[2]);
for(bnd=0; bnd<s->num_rematrixing_bands; bnd++) {
if(s->rematrixing_flags[bnd]) {
bndend = FFMIN(end, ff_ac3_rematrix_band_tab[bnd+1]);
for(i=ff_ac3_rematrix_band_tab[bnd]; i<bndend; i++) {
tmp0 = s->fixed_coeffs[1][i];
tmp1 = s->fixed_coeffs[2][i];
s->fixed_coeffs[1][i] = tmp0 + tmp1;
s->fixed_coeffs[2][i] = tmp0 - tmp1;
}
}
}
}
/**
* Perform the 256-point IMDCT
*/
static void do_imdct_256(AC3DecodeContext *s, int chindex)
{
int i, k;
DECLARE_ALIGNED_16(float, x[128]);
FFTComplex z[2][64];
float *o_ptr = s->tmp_output;
for(i=0; i<2; i++) {
/* de-interleave coefficients */
for(k=0; k<128; k++) {
x[k] = s->transform_coeffs[chindex][2*k+i];
}
/* run standard IMDCT */
s->imdct_256.fft.imdct_calc(&s->imdct_256, o_ptr, x, s->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 *s, int channels)
{
int ch;
for (ch=1; ch<=channels; ch++) {
if (s->block_switch[ch]) {
do_imdct_256(s, ch);
} else {
s->imdct_512.fft.imdct_calc(&s->imdct_512, s->tmp_output,
s->transform_coeffs[ch], s->tmp_imdct);
}
/* For the first half of the block, apply the window, add the delay
from the previous block, and send to output */
s->dsp.vector_fmul_add_add(s->output[ch-1], s->tmp_output,
s->window, s->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 */
s->dsp.vector_fmul_reverse(s->delay[ch-1], s->tmp_output+256,
s->window, 256);
}
}
/**
* Downmix the output to mono or stereo.
*/
static void ac3_downmix(AC3DecodeContext *s,
float samples[AC3_MAX_CHANNELS][256], int ch_offset)
{
int i, j;
float v0, v1;
for(i=0; i<256; i++) {
v0 = v1 = 0.0f;
for(j=0; j<s->fbw_channels; j++) {
v0 += samples[j+ch_offset][i] * s->downmix_coeffs[j][0];
v1 += samples[j+ch_offset][i] * s->downmix_coeffs[j][1];
}
v0 *= s->downmix_coeff_adjust[0];
v1 *= s->downmix_coeff_adjust[1];
if(s->output_mode == AC3_CHMODE_MONO) {
samples[ch_offset][i] = (v0 + v1) * LEVEL_MINUS_3DB;
} else if(s->output_mode == AC3_CHMODE_STEREO) {
samples[ ch_offset][i] = v0;
samples[1+ch_offset][i] = v1;
}
}
}
/**
* Upmix delay samples from stereo to original channel layout.
*/
static void ac3_upmix_delay(AC3DecodeContext *s)
{
int channel_data_size = sizeof(s->delay[0]);
switch(s->channel_mode) {
case AC3_CHMODE_DUALMONO:
case AC3_CHMODE_STEREO:
/* upmix mono to stereo */
memcpy(s->delay[1], s->delay[0], channel_data_size);
break;
case AC3_CHMODE_2F2R:
memset(s->delay[3], 0, channel_data_size);
case AC3_CHMODE_2F1R:
memset(s->delay[2], 0, channel_data_size);
break;
case AC3_CHMODE_3F2R:
memset(s->delay[4], 0, channel_data_size);
case AC3_CHMODE_3F1R:
memset(s->delay[3], 0, channel_data_size);
case AC3_CHMODE_3F:
memcpy(s->delay[2], s->delay[1], channel_data_size);
memset(s->delay[1], 0, channel_data_size);
break;
}
}
/**
* Parse an audio block from AC-3 bitstream.
*/
static int ac3_parse_audio_block(AC3DecodeContext *s, int blk)
{
int fbw_channels = s->fbw_channels;
int channel_mode = s->channel_mode;
int i, bnd, seg, ch;
int different_transforms;
int downmix_output;
int cpl_in_use;
GetBitContext *gbc = &s->gbc;
uint8_t bit_alloc_stages[AC3_MAX_CHANNELS];
memset(bit_alloc_stages, 0, AC3_MAX_CHANNELS);
/* block switch flags */
different_transforms = 0;
for (ch = 1; ch <= fbw_channels; ch++) {
s->block_switch[ch] = get_bits1(gbc);
if(ch > 1 && s->block_switch[ch] != s->block_switch[1])
different_transforms = 1;
}
/* dithering flags */
s->dither_all = 1;
for (ch = 1; ch <= fbw_channels; ch++) {
s->dither_flag[ch] = get_bits1(gbc);
if(!s->dither_flag[ch])
s->dither_all = 0;
}
/* dynamic range */
i = !(s->channel_mode);
do {
if(get_bits1(gbc)) {
s->dynamic_range[i] = ((dynamic_range_tab[get_bits(gbc, 8)]-1.0) *
s->avctx->drc_scale)+1.0;
} else if(blk == 0) {
s->dynamic_range[i] = 1.0f;
}
} while(i--);
/* coupling strategy */
if (get_bits1(gbc)) {
memset(bit_alloc_stages, 3, AC3_MAX_CHANNELS);
cpl_in_use = get_bits1(gbc);
if (cpl_in_use) {
/* coupling in use */
int cpl_begin_freq, cpl_end_freq;
if (channel_mode < AC3_CHMODE_STEREO) {
av_log(s->avctx, AV_LOG_ERROR, "coupling not allowed in mono or dual-mono\n");
return -1;
}
/* determine which channels are coupled */
for (ch = 1; ch <= fbw_channels; ch++)
s->channel_in_cpl[ch] = get_bits1(gbc);
/* phase flags in use */
if (channel_mode == AC3_CHMODE_STEREO)
s->phase_flags_in_use = get_bits1(gbc);
/* coupling frequency range and band structure */
cpl_begin_freq = get_bits(gbc, 4);
cpl_end_freq = get_bits(gbc, 4);
if (3 + cpl_end_freq - cpl_begin_freq < 0) {
av_log(s->avctx, AV_LOG_ERROR, "3+cplendf = %d < cplbegf = %d\n", 3+cpl_end_freq, cpl_begin_freq);
return -1;
}
s->num_cpl_bands = s->num_cpl_subbands = 3 + cpl_end_freq - cpl_begin_freq;
s->start_freq[CPL_CH] = cpl_begin_freq * 12 + 37;
s->end_freq[CPL_CH] = cpl_end_freq * 12 + 73;
for (bnd = 0; bnd < s->num_cpl_subbands - 1; bnd++) {
if (get_bits1(gbc)) {
s->cpl_band_struct[bnd] = 1;
s->num_cpl_bands--;
}
}
s->cpl_band_struct[s->num_cpl_subbands-1] = 0;
} else {
/* coupling not in use */
for (ch = 1; ch <= fbw_channels; ch++)
s->channel_in_cpl[ch] = 0;
}
} else if (!blk) {
av_log(s->avctx, AV_LOG_ERROR, "new coupling strategy must be present in block 0\n");
return -1;
} else {
cpl_in_use = s->cpl_in_use[blk-1];
}
s->cpl_in_use[blk] = cpl_in_use;
/* coupling coordinates */
if (cpl_in_use) {
int cpl_coords_exist = 0;
for (ch = 1; ch <= fbw_channels; ch++) {
if (s->channel_in_cpl[ch]) {
if (get_bits1(gbc)) {
int master_cpl_coord, cpl_coord_exp, cpl_coord_mant;
cpl_coords_exist = 1;
master_cpl_coord = 3 * get_bits(gbc, 2);
for (bnd = 0; bnd < s->num_cpl_bands; bnd++) {
cpl_coord_exp = get_bits(gbc, 4);
cpl_coord_mant = get_bits(gbc, 4);
if (cpl_coord_exp == 15)
s->cpl_coords[ch][bnd] = cpl_coord_mant << 22;
else
s->cpl_coords[ch][bnd] = (cpl_coord_mant + 16) << 21;
s->cpl_coords[ch][bnd] >>= (cpl_coord_exp + master_cpl_coord);
}
} else if (!blk) {
av_log(s->avctx, AV_LOG_ERROR, "new coupling coordinates must be present in block 0\n");
return -1;
}
}
}
/* phase flags */
if (channel_mode == AC3_CHMODE_STEREO && cpl_coords_exist) {
for (bnd = 0; bnd < s->num_cpl_bands; bnd++) {
s->phase_flags[bnd] = s->phase_flags_in_use? get_bits1(gbc) : 0;
}
}
}
/* stereo rematrixing strategy and band structure */
if (channel_mode == AC3_CHMODE_STEREO) {
if (get_bits1(gbc)) {
s->num_rematrixing_bands = 4;
if(cpl_in_use && s->start_freq[CPL_CH] <= 61)
s->num_rematrixing_bands -= 1 + (s->start_freq[CPL_CH] == 37);
for(bnd=0; bnd<s->num_rematrixing_bands; bnd++)
s->rematrixing_flags[bnd] = get_bits1(gbc);
} else if (!blk) {
av_log(s->avctx, AV_LOG_ERROR, "new rematrixing strategy must be present in block 0\n");
return -1;
}
}
/* exponent strategies for each channel */
s->exp_strategy[blk][CPL_CH] = EXP_REUSE;
s->exp_strategy[blk][s->lfe_ch] = EXP_REUSE;
for (ch = !cpl_in_use; ch <= s->channels; ch++) {
s->exp_strategy[blk][ch] = get_bits(gbc, 2 - (ch == s->lfe_ch));
if(s->exp_strategy[blk][ch] != EXP_REUSE)
bit_alloc_stages[ch] = 3;
}
/* channel bandwidth */
for (ch = 1; ch <= fbw_channels; ch++) {
s->start_freq[ch] = 0;
if (s->exp_strategy[blk][ch] != EXP_REUSE) {
int group_size;
int prev = s->end_freq[ch];
if (s->channel_in_cpl[ch])
s->end_freq[ch] = s->start_freq[CPL_CH];
else {
int bandwidth_code = get_bits(gbc, 6);
if (bandwidth_code > 60) {
av_log(s->avctx, AV_LOG_ERROR, "bandwidth code = %d > 60", bandwidth_code);
return -1;
}
s->end_freq[ch] = bandwidth_code * 3 + 73;
}
group_size = 3 << (s->exp_strategy[blk][ch] - 1);
s->num_exp_groups[ch] = (s->end_freq[ch]+group_size-4) / group_size;
if(blk > 0 && s->end_freq[ch] != prev)
memset(bit_alloc_stages, 3, AC3_MAX_CHANNELS);
}
}
if (cpl_in_use && s->exp_strategy[blk][CPL_CH] != EXP_REUSE) {
s->num_exp_groups[CPL_CH] = (s->end_freq[CPL_CH] - s->start_freq[CPL_CH]) /
(3 << (s->exp_strategy[blk][CPL_CH] - 1));
}
/* decode exponents for each channel */
for (ch = !cpl_in_use; ch <= s->channels; ch++) {
if (s->exp_strategy[blk][ch] != EXP_REUSE) {
s->dexps[ch][0] = get_bits(gbc, 4) << !ch;
decode_exponents(gbc, s->exp_strategy[blk][ch],
s->num_exp_groups[ch], s->dexps[ch][0],
&s->dexps[ch][s->start_freq[ch]+!!ch]);
if(ch != CPL_CH && ch != s->lfe_ch)
skip_bits(gbc, 2); /* skip gainrng */
}
}
/* bit allocation information */
if (get_bits1(gbc)) {
s->bit_alloc_params.slow_decay = ff_ac3_slow_decay_tab[get_bits(gbc, 2)] >> s->bit_alloc_params.sr_shift;
s->bit_alloc_params.fast_decay = ff_ac3_fast_decay_tab[get_bits(gbc, 2)] >> s->bit_alloc_params.sr_shift;
s->bit_alloc_params.slow_gain = ff_ac3_slow_gain_tab[get_bits(gbc, 2)];
s->bit_alloc_params.db_per_bit = ff_ac3_db_per_bit_tab[get_bits(gbc, 2)];
s->bit_alloc_params.floor = ff_ac3_floor_tab[get_bits(gbc, 3)];
for(ch=!cpl_in_use; ch<=s->channels; ch++)
bit_alloc_stages[ch] = FFMAX(bit_alloc_stages[ch], 2);
} else if (!blk) {
av_log(s->avctx, AV_LOG_ERROR, "new bit allocation info must be present in block 0\n");
return -1;
}
/* signal-to-noise ratio offsets and fast gains (signal-to-mask ratios) */
if (get_bits1(gbc)) {
int csnr;
csnr = (get_bits(gbc, 6) - 15) << 4;
for (ch = !cpl_in_use; ch <= s->channels; ch++) { /* snr offset and fast gain */
s->snr_offset[ch] = (csnr + get_bits(gbc, 4)) << 2;
s->fast_gain[ch] = ff_ac3_fast_gain_tab[get_bits(gbc, 3)];
}
memset(bit_alloc_stages, 3, AC3_MAX_CHANNELS);
} else if (!blk) {
av_log(s->avctx, AV_LOG_ERROR, "new snr offsets must be present in block 0\n");
return -1;
}
/* coupling leak information */
if (cpl_in_use) {
if (get_bits1(gbc)) {
s->bit_alloc_params.cpl_fast_leak = get_bits(gbc, 3);
s->bit_alloc_params.cpl_slow_leak = get_bits(gbc, 3);
bit_alloc_stages[CPL_CH] = FFMAX(bit_alloc_stages[CPL_CH], 2);
} else if (!blk) {
av_log(s->avctx, AV_LOG_ERROR, "new coupling leak info must be present in block 0\n");
return -1;
}
}
/* delta bit allocation information */
if (get_bits1(gbc)) {
/* delta bit allocation exists (strategy) */
for (ch = !cpl_in_use; ch <= fbw_channels; ch++) {
s->dba_mode[ch] = get_bits(gbc, 2);
if (s->dba_mode[ch] == DBA_RESERVED) {
av_log(s->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 = !cpl_in_use; ch <= fbw_channels; ch++) {
if (s->dba_mode[ch] == DBA_NEW) {
s->dba_nsegs[ch] = get_bits(gbc, 3);
for (seg = 0; seg <= s->dba_nsegs[ch]; seg++) {
s->dba_offsets[ch][seg] = get_bits(gbc, 5);
s->dba_lengths[ch][seg] = get_bits(gbc, 4);
s->dba_values[ch][seg] = get_bits(gbc, 3);
}
/* run last 2 bit allocation stages if new dba values */
bit_alloc_stages[ch] = FFMAX(bit_alloc_stages[ch], 2);
}
}
} else if(blk == 0) {
for(ch=0; ch<=s->channels; ch++) {
s->dba_mode[ch] = DBA_NONE;
}
}
/* Bit allocation */
for(ch=!cpl_in_use; ch<=s->channels; ch++) {
if(bit_alloc_stages[ch] > 2) {
/* Exponent mapping into PSD and PSD integration */
ff_ac3_bit_alloc_calc_psd(s->dexps[ch],
s->start_freq[ch], s->end_freq[ch],
s->psd[ch], s->band_psd[ch]);
}
if(bit_alloc_stages[ch] > 1) {
/* Compute excitation function, Compute masking curve, and
Apply delta bit allocation */
ff_ac3_bit_alloc_calc_mask(&s->bit_alloc_params, s->band_psd[ch],
s->start_freq[ch], s->end_freq[ch],
s->fast_gain[ch], (ch == s->lfe_ch),
s->dba_mode[ch], s->dba_nsegs[ch],
s->dba_offsets[ch], s->dba_lengths[ch],
s->dba_values[ch], s->mask[ch]);
}
if(bit_alloc_stages[ch] > 0) {
/* Compute bit allocation */
ff_ac3_bit_alloc_calc_bap(s->mask[ch], s->psd[ch],
s->start_freq[ch], s->end_freq[ch],
s->snr_offset[ch],
s->bit_alloc_params.floor,
ff_ac3_bap_tab, s->bap[ch]);
}
}
/* unused dummy data */
if (get_bits1(gbc)) {
int skipl = get_bits(gbc, 9);
while(skipl--)
skip_bits(gbc, 8);
}
/* unpack the transform coefficients
this also uncouples channels if coupling is in use. */
get_transform_coeffs(s);
/* recover coefficients if rematrixing is in use */
if(s->channel_mode == AC3_CHMODE_STEREO)
do_rematrixing(s);
/* apply scaling to coefficients (headroom, dynrng) */
for(ch=1; ch<=s->channels; ch++) {
float gain = s->mul_bias / 4194304.0f;
if(s->channel_mode == AC3_CHMODE_DUALMONO) {
gain *= s->dynamic_range[ch-1];
} else {
gain *= s->dynamic_range[0];
}
for(i=0; i<256; i++) {
s->transform_coeffs[ch][i] = s->fixed_coeffs[ch][i] * gain;
}
}
/* downmix and MDCT. order depends on whether block switching is used for
any channel in this block. this is because coefficients for the long
and short transforms cannot be mixed. */
downmix_output = s->channels != s->out_channels &&
!((s->output_mode & AC3_OUTPUT_LFEON) &&
s->fbw_channels == s->out_channels);
if(different_transforms) {
/* the delay samples have already been downmixed, so we upmix the delay
samples in order to reconstruct all channels before downmixing. */
if(s->downmixed) {
s->downmixed = 0;
ac3_upmix_delay(s);
}
do_imdct(s, s->channels);
if(downmix_output) {
ac3_downmix(s, s->output, 0);
}
} else {
if(downmix_output) {
ac3_downmix(s, s->transform_coeffs, 1);
}
if(!s->downmixed) {
s->downmixed = 1;
ac3_downmix(s, s->delay, 0);
}
do_imdct(s, s->out_channels);
}
/* convert float to 16-bit integer */
for(ch=0; ch<s->out_channels; ch++) {
for(i=0; i<256; i++) {
s->output[ch][i] += s->add_bias;
}
s->dsp.float_to_int16(s->int_output[ch], s->output[ch], 256);
}
return 0;
}
/**
* Decode a single AC-3 frame.
*/
static int ac3_decode_frame(AVCodecContext * avctx, void *data, int *data_size,
const uint8_t *buf, int buf_size)
{
AC3DecodeContext *s = avctx->priv_data;
int16_t *out_samples = (int16_t *)data;
int i, blk, ch, err;
/* initialize the GetBitContext with the start of valid AC-3 Frame */
if (s->input_buffer) {
/* copy input buffer to decoder context to avoid reading past the end
of the buffer, which can be caused by a damaged input stream. */
memcpy(s->input_buffer, buf, FFMIN(buf_size, AC3_MAX_FRAME_SIZE));
init_get_bits(&s->gbc, s->input_buffer, buf_size * 8);
} else {
init_get_bits(&s->gbc, buf, buf_size * 8);
}
/* parse the syncinfo */
*data_size = 0;
err = parse_frame_header(s);
/* check that reported frame size fits in input buffer */
if(s->frame_size > buf_size) {
av_log(avctx, AV_LOG_ERROR, "incomplete frame\n");
err = AC3_PARSE_ERROR_FRAME_SIZE;
}
/* check for crc mismatch */
if(err != AC3_PARSE_ERROR_FRAME_SIZE && avctx->error_resilience >= FF_ER_CAREFUL) {
if(av_crc(av_crc_get_table(AV_CRC_16_ANSI), 0, &buf[2], s->frame_size-2)) {
av_log(avctx, AV_LOG_ERROR, "frame CRC mismatch\n");
err = AC3_PARSE_ERROR_CRC;
}
}
if(err && err != AC3_PARSE_ERROR_CRC) {
switch(err) {
case AC3_PARSE_ERROR_SYNC:
av_log(avctx, AV_LOG_ERROR, "frame sync error\n");
return -1;
case AC3_PARSE_ERROR_BSID:
av_log(avctx, AV_LOG_ERROR, "invalid bitstream id\n");
break;
case AC3_PARSE_ERROR_SAMPLE_RATE:
av_log(avctx, AV_LOG_ERROR, "invalid sample rate\n");
break;
case AC3_PARSE_ERROR_FRAME_SIZE:
av_log(avctx, AV_LOG_ERROR, "invalid frame size\n");
break;
case AC3_PARSE_ERROR_FRAME_TYPE:
/* skip frame if CRC is ok. otherwise use error concealment. */
/* TODO: add support for substreams and dependent frames */
if(s->frame_type == EAC3_FRAME_TYPE_DEPENDENT || s->substreamid) {
av_log(avctx, AV_LOG_ERROR, "unsupported frame type : skipping frame\n");
return s->frame_size;
} else {
av_log(avctx, AV_LOG_ERROR, "invalid frame type\n");
}
break;
default:
av_log(avctx, AV_LOG_ERROR, "invalid header\n");
break;
}
}
/* if frame is ok, set audio parameters */
if (!err) {
avctx->sample_rate = s->sample_rate;
avctx->bit_rate = s->bit_rate;
/* channel config */
s->out_channels = s->channels;
s->output_mode = s->channel_mode;
if(s->lfe_on)
s->output_mode |= AC3_OUTPUT_LFEON;
if (avctx->request_channels > 0 && avctx->request_channels <= 2 &&
avctx->request_channels < s->channels) {
s->out_channels = avctx->request_channels;
s->output_mode = avctx->request_channels == 1 ? AC3_CHMODE_MONO : AC3_CHMODE_STEREO;
}
avctx->channels = s->out_channels;
/* set downmixing coefficients if needed */
if(s->channels != s->out_channels && !((s->output_mode & AC3_OUTPUT_LFEON) &&
s->fbw_channels == s->out_channels)) {
set_downmix_coeffs(s);
}
} else if (!s->out_channels) {
s->out_channels = avctx->channels;
if(s->out_channels < s->channels)
s->output_mode = s->out_channels == 1 ? AC3_CHMODE_MONO : AC3_CHMODE_STEREO;
}
/* parse the audio blocks */
for (blk = 0; blk < s->num_blocks; blk++) {
if (!err && ac3_parse_audio_block(s, blk)) {
av_log(avctx, AV_LOG_ERROR, "error parsing the audio block\n");
}
/* interleave output samples */
for (i = 0; i < 256; i++)
for (ch = 0; ch < s->out_channels; ch++)
*(out_samples++) = s->int_output[ch][i];
}
*data_size = s->num_blocks * 256 * avctx->channels * sizeof (int16_t);
return s->frame_size;
}
/**
* Uninitialize the AC-3 decoder.
*/
static av_cold int ac3_decode_end(AVCodecContext *avctx)
{
AC3DecodeContext *s = avctx->priv_data;
ff_mdct_end(&s->imdct_512);
ff_mdct_end(&s->imdct_256);
av_freep(&s->input_buffer);
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,
.long_name = "ATSC A/52 / AC-3",
};