virtualenv/FFmpeg
1
0
Fork 0
mirror of https://github.com/FFmpeg/FFmpeg.git synced 2024-12-07 11:13:41 +02:00
Code Issues Releases Activity
fecfdf6e72
FFmpeg/libavcodec/ac3dec.c
Michael Niedermayer e3e09f2bad Merge remote-tracking branch 'qatar/master' 
* qatar/master:
  os_support: Choose between direct.h and io.h using a configure check
  os_support: Include io.h instead of direct.h on mingw32ce
  x86: ac3dsp: Only refer to the ac3_downmix_sse symbol if it has been declared
  swscale: Remove two bogus asserts
  ac3: move ac3_downmix() from dsputil to ac3dsp
  lavr/audio_mix_matrix: acknowledge the existence of LFE2.
  mlp_parser: avoid mapping multiple disctinct TrueHD channels to the same Libav channel.
  lavu/audioconvert: add a second low frequency channel.

Conflicts:
	doc/APIchanges
	libavcodec/ac3dsp.c
	libavcodec/ac3dsp.h
	libavcodec/mlp_parser.c
	libavutil/audioconvert.c
	libavutil/version.h

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2012-09-13 15:35:50 +02:00

1489 lines
54 KiB
C
Raw Blame History

/*
* AC-3 Audio Decoder
* This code was developed as part of Google Summer of Code 2006.
* E-AC-3 support was added as part of Google Summer of Code 2007.
*
* Copyright (c) 2006 Kartikey Mahendra BHATT (bhattkm at gmail dot com)
* Copyright (c) 2007-2008 Bartlomiej Wolowiec <bartek.wolowiec@gmail.com>
* Copyright (c) 2007 Justin Ruggles <justin.ruggles@gmail.com>
*
* 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
*/
#include <stdio.h>
#include <stddef.h>
#include <math.h>
#include <string.h>
#include "libavutil/crc.h"
#include "libavutil/opt.h"
#include "internal.h"
#include "aac_ac3_parser.h"
#include "ac3_parser.h"
#include "ac3dec.h"
#include "ac3dec_data.h"
#include "kbdwin.h"
/**
* table for ungrouping 3 values in 7 bits.
* used for exponents and bap=2 mantissas
*/
static uint8_t ungroup_3_in_7_bits_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 */
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 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 table for ungrouping 3 values in 7 bits
reference: Section 7.1.3 Exponent Decoding */
for (i = 0; i < 128; i++) {
ungroup_3_in_7_bits_tab[i][0] = i / 25;
ungroup_3_in_7_bits_tab[i][1] = (i % 25) / 5;
ungroup_3_in_7_bits_tab[i][2] = (i % 25) % 5;
}
/* 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(ff_ac3_ungroup_3_in_5_bits_tab[i][0], 3);
b1_mantissas[i][1] = symmetric_dequant(ff_ac3_ungroup_3_in_5_bits_tab[i][1], 3);
b1_mantissas[i][2] = symmetric_dequant(ff_ac3_ungroup_3_in_5_bits_tab[i][2], 3);
}
for (i = 0; i < 128; i++) {
/* bap=2 mantissas */
b2_mantissas[i][0] = symmetric_dequant(ungroup_3_in_7_bits_tab[i][0], 5);
b2_mantissas[i][1] = symmetric_dequant(ungroup_3_in_7_bits_tab[i][1], 5);
b2_mantissas[i][2] = symmetric_dequant(ungroup_3_in_7_bits_tab[i][2], 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);
}
}
/**
* AVCodec initialization
*/
static av_cold int ac3_decode_init(AVCodecContext *avctx)
{
AC3DecodeContext *s = avctx->priv_data;
s->avctx = avctx;
ff_ac3_common_init();
ac3_tables_init();
ff_mdct_init(&s->imdct_256, 8, 1, 1.0);
ff_mdct_init(&s->imdct_512, 9, 1, 1.0);
ff_kbd_window_init(s->window, 5.0, 256);
ff_dsputil_init(&s->dsp, avctx);
ff_ac3dsp_init(&s->ac3dsp, avctx->flags & CODEC_FLAG_BITEXACT);
ff_fmt_convert_init(&s->fmt_conv, avctx);
av_lfg_init(&s->dith_state, 0);
/* set scale value for float to int16 conversion */
if (avctx->request_sample_fmt == AV_SAMPLE_FMT_FLT) {
s->mul_bias = 1.0f;
avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
} else {
s->mul_bias = 32767.0f;
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
}
/* 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;
avcodec_get_frame_defaults(&s->frame);
avctx->coded_frame = &s->frame;
return 0;
}
/**
* Parse the 'sync info' and 'bit stream info' from the AC-3 bitstream.
* GetBitContext within AC3DecodeContext must point to
* the start of the synchronized AC-3 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 AC-3 or E-AC-3 frame header
*/
static int parse_frame_header(AC3DecodeContext *s)
{
AC3HeaderInfo hdr;
int err;
err = avpriv_ac3_parse_header(&s->gbc, &hdr);
if (err)
return err;
/* get decoding parameters from header info */
s->bit_alloc_params.sr_code = hdr.sr_code;
s->bitstream_mode = hdr.bitstream_mode;
s->channel_mode = hdr.channel_mode;
s->channel_layout = hdr.channel_layout;
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;
}
if (hdr.bitstream_id <= 10) {
s->eac3 = 0;
s->snr_offset_strategy = 2;
s->block_switch_syntax = 1;
s->dither_flag_syntax = 1;
s->bit_allocation_syntax = 1;
s->fast_gain_syntax = 0;
s->first_cpl_leak = 0;
s->dba_syntax = 1;
s->skip_syntax = 1;
memset(s->channel_uses_aht, 0, sizeof(s->channel_uses_aht));
return ac3_parse_header(s);
} else if (CONFIG_EAC3_DECODER) {
s->eac3 = 1;
return ff_eac3_parse_header(s);
} else {
av_log(s->avctx, AV_LOG_ERROR, "E-AC-3 support not compiled in\n");
return -1;
}
}
/**
* 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[s-> center_mix_level];
float smix = gain_levels[s->surround_mix_level];
float norm0, norm1;
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;
}
/* renormalize */
norm0 = norm1 = 0.0;
for (i = 0; i < s->fbw_channels; i++) {
norm0 += s->downmix_coeffs[i][0];
norm1 += s->downmix_coeffs[i][1];
}
norm0 = 1.0f / norm0;
norm1 = 1.0f / norm1;
for (i = 0; i < s->fbw_channels; i++) {
s->downmix_coeffs[i][0] *= norm0;
s->downmix_coeffs[i][1] *= norm1;
}
if (s->output_mode == AC3_CHMODE_MONO) {
for (i = 0; i < s->fbw_channels; i++)
s->downmix_coeffs[i][0] = (s->downmix_coeffs[i][0] +
s->downmix_coeffs[i][1]) * LEVEL_MINUS_3DB;
}
}
/**
* Decode the grouped exponents according to exponent strategy.
* reference: Section 7.1.3 Exponent Decoding
*/
static int 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++] = ungroup_3_in_7_bits_tab[expacc][0];
dexp[i++] = ungroup_3_in_7_bits_tab[expacc][1];
dexp[i++] = ungroup_3_in_7_bits_tab[expacc][2];
}
/* convert to absolute exps and expand groups */
prevexp = absexp;
for (i = 0, j = 0; i < ngrps * 3; i++) {
prevexp += dexp[i] - 2;
if (prevexp > 24U)
return -1;
switch (group_size) {
case 4: dexps[j++] = prevexp;
dexps[j++] = prevexp;
case 2: dexps[j++] = prevexp;
case 1: dexps[j++] = prevexp;
}
}
return 0;
}
/**
* 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 calc_transform_coeffs_cpl(AC3DecodeContext *s)
{
int bin, band, ch;
bin = s->start_freq[CPL_CH];
for (band = 0; band < s->num_cpl_bands; band++) {
int band_start = bin;
int band_end = bin + s->cpl_band_sizes[band];
for (ch = 1; ch <= s->fbw_channels; ch++) {
if (s->channel_in_cpl[ch]) {
int cpl_coord = s->cpl_coords[ch][band] << 5;
for (bin = band_start; bin < band_end; bin++) {
s->fixed_coeffs[ch][bin] =
MULH(s->fixed_coeffs[CPL_CH][bin] << 4, cpl_coord);
}
if (ch == 2 && s->phase_flags[band]) {
for (bin = band_start; bin < band_end; bin++)
s->fixed_coeffs[2][bin] = -s->fixed_coeffs[2][bin];
}
}
}
bin = band_end;
}
}
/**
* Grouped mantissas for 3-level 5-level and 11-level quantization
*/
typedef struct {
int b1_mant[2];
int b2_mant[2];
int b4_mant;
int b1;
int b2;
int b4;
} mant_groups;
/**
* Decode the transform coefficients for a particular channel
* reference: Section 7.3 Quantization and Decoding of Mantissas
*/
static void ac3_decode_transform_coeffs_ch(AC3DecodeContext *s, int ch_index, mant_groups *m)
{
int start_freq = s->start_freq[ch_index];
int end_freq = s->end_freq[ch_index];
uint8_t *baps = s->bap[ch_index];
int8_t *exps = s->dexps[ch_index];
int *coeffs = s->fixed_coeffs[ch_index];
int dither = (ch_index == CPL_CH) || s->dither_flag[ch_index];
GetBitContext *gbc = &s->gbc;
int freq;
for (freq = start_freq; freq < end_freq; freq++) {
int bap = baps[freq];
int mantissa;
switch (bap) {
case 0:
if (dither)
mantissa = (av_lfg_get(&s->dith_state) & 0x7FFFFF) - 0x400000;
else
mantissa = 0;
break;
case 1:
if (m->b1) {
m->b1--;
mantissa = m->b1_mant[m->b1];
} else {
int bits = get_bits(gbc, 5);
mantissa = b1_mantissas[bits][0];
m->b1_mant[1] = b1_mantissas[bits][1];
m->b1_mant[0] = b1_mantissas[bits][2];
m->b1 = 2;
}
break;
case 2:
if (m->b2) {
m->b2--;
mantissa = m->b2_mant[m->b2];
} else {
int bits = get_bits(gbc, 7);
mantissa = b2_mantissas[bits][0];
m->b2_mant[1] = b2_mantissas[bits][1];
m->b2_mant[0] = b2_mantissas[bits][2];
m->b2 = 2;
}
break;
case 3:
mantissa = b3_mantissas[get_bits(gbc, 3)];
break;
case 4:
if (m->b4) {
m->b4 = 0;
mantissa = m->b4_mant;
} else {
int bits = get_bits(gbc, 7);
mantissa = b4_mantissas[bits][0];
m->b4_mant = b4_mantissas[bits][1];
m->b4 = 1;
}
break;
case 5:
mantissa = b5_mantissas[get_bits(gbc, 4)];
break;
default: /* 6 to 15 */
/* Shift mantissa and sign-extend it. */
mantissa = get_sbits(gbc, quantization_tab[bap]);
mantissa <<= 24 - quantization_tab[bap];
break;
}
coeffs[freq] = mantissa >> exps[freq];
}
}
/**
* Remove random dithering from coupling range coefficients with zero-bit
* mantissas for coupled channels which do not use dithering.
* reference: Section 7.3.4 Dither for Zero Bit Mantissas (bap=0)
*/
static void remove_dithering(AC3DecodeContext *s) {
int ch, i;
for (ch = 1; ch <= s->fbw_channels; ch++) {
if (!s->dither_flag[ch] && s->channel_in_cpl[ch]) {
for (i = s->start_freq[CPL_CH]; i < s->end_freq[CPL_CH]; i++) {
if (!s->bap[CPL_CH][i])
s->fixed_coeffs[ch][i] = 0;
}
}
}
}
static void decode_transform_coeffs_ch(AC3DecodeContext *s, int blk, int ch,
mant_groups *m)
{
if (!s->channel_uses_aht[ch]) {
ac3_decode_transform_coeffs_ch(s, ch, m);
} else {
/* if AHT is used, mantissas for all blocks are encoded in the first
block of the frame. */
int bin;
if (!blk && CONFIG_EAC3_DECODER)
ff_eac3_decode_transform_coeffs_aht_ch(s, ch);
for (bin = s->start_freq[ch]; bin < s->end_freq[ch]; bin++) {
s->fixed_coeffs[ch][bin] = s->pre_mantissa[ch][bin][blk] >> s->dexps[ch][bin];
}
}
}
/**
* Decode the transform coefficients.
*/
static void decode_transform_coeffs(AC3DecodeContext *s, int blk)
{
int ch, end;
int got_cplchan = 0;
mant_groups m;
m.b1 = m.b2 = m.b4 = 0;
for (ch = 1; ch <= s->channels; ch++) {
/* transform coefficients for full-bandwidth channel */
decode_transform_coeffs_ch(s, blk, 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) {
decode_transform_coeffs_ch(s, blk, CPL_CH, &m);
calc_transform_coeffs_cpl(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);
}
/* zero the dithered coefficients for appropriate channels */
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;
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++) {
int tmp0 = s->fixed_coeffs[1][i];
s->fixed_coeffs[1][i] += s->fixed_coeffs[2][i];
s->fixed_coeffs[2][i] = tmp0 - s->fixed_coeffs[2][i];
}
}
}
}
/**
* 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]) {
int i;
float *x = s->tmp_output + 128;
for (i = 0; i < 128; i++)
x[i] = s->transform_coeffs[ch][2 * i];
s->imdct_256.imdct_half(&s->imdct_256, s->tmp_output, x);
s->dsp.vector_fmul_window(s->output[ch - 1], s->delay[ch - 1],
s->tmp_output, s->window, 128);
for (i = 0; i < 128; i++)
x[i] = s->transform_coeffs[ch][2 * i + 1];
s->imdct_256.imdct_half(&s->imdct_256, s->delay[ch - 1], x);
} else {
s->imdct_512.imdct_half(&s->imdct_512, s->tmp_output, s->transform_coeffs[ch]);
s->dsp.vector_fmul_window(s->output[ch - 1], s->delay[ch - 1],
s->tmp_output, s->window, 128);
memcpy(s->delay[ch - 1], s->tmp_output + 128, 128 * sizeof(float));
}
}
}
/**
* 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;
}
}
/**
* Decode band structure for coupling, spectral extension, or enhanced coupling.
* The band structure defines how many subbands are in each band. For each
* subband in the range, 1 means it is combined with the previous band, and 0
* means that it starts a new band.
*
* @param[in] gbc bit reader context
* @param[in] blk block number
* @param[in] eac3 flag to indicate E-AC-3
* @param[in] ecpl flag to indicate enhanced coupling
* @param[in] start_subband subband number for start of range
* @param[in] end_subband subband number for end of range
* @param[in] default_band_struct default band structure table
* @param[out] num_bands number of bands (optionally NULL)
* @param[out] band_sizes array containing the number of bins in each band (optionally NULL)
*/
static void decode_band_structure(GetBitContext *gbc, int blk, int eac3,
int ecpl, int start_subband, int end_subband,
const uint8_t *default_band_struct,
int *num_bands, uint8_t *band_sizes)
{
int subbnd, bnd, n_subbands, n_bands=0;
uint8_t bnd_sz[22];
uint8_t coded_band_struct[22];
const uint8_t *band_struct;
n_subbands = end_subband - start_subband;
/* decode band structure from bitstream or use default */
if (!eac3 || get_bits1(gbc)) {
for (subbnd = 0; subbnd < n_subbands - 1; subbnd++) {
coded_band_struct[subbnd] = get_bits1(gbc);
}
band_struct = coded_band_struct;
} else if (!blk) {
band_struct = &default_band_struct[start_subband+1];
} else {
/* no change in band structure */
return;
}
/* calculate number of bands and band sizes based on band structure.
note that the first 4 subbands in enhanced coupling span only 6 bins
instead of 12. */
if (num_bands || band_sizes ) {
n_bands = n_subbands;
bnd_sz[0] = ecpl ? 6 : 12;
for (bnd = 0, subbnd = 1; subbnd < n_subbands; subbnd++) {
int subbnd_size = (ecpl && subbnd < 4) ? 6 : 12;
if (band_struct[subbnd - 1]) {
n_bands--;
bnd_sz[bnd] += subbnd_size;
} else {
bnd_sz[++bnd] = subbnd_size;
}
}
}
/* set optional output params */
if (num_bands)
*num_bands = n_bands;
if (band_sizes)
memcpy(band_sizes, bnd_sz, n_bands);
}
/**
* Decode a single audio block from the AC-3 bitstream.
*/
static int decode_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] = { 0 };
/* block switch flags */
different_transforms = 0;
if (s->block_switch_syntax) {
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 */
if (s->dither_flag_syntax) {
for (ch = 1; ch <= fbw_channels; ch++) {
s->dither_flag[ch] = get_bits1(gbc);
}
}
/* 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->drc_scale) + 1.0;
} else if (blk == 0) {
s->dynamic_range[i] = 1.0f;
}
} while (i--);
/* spectral extension strategy */
if (s->eac3 && (!blk || get_bits1(gbc))) {
s->spx_in_use = get_bits1(gbc);
if (s->spx_in_use) {
int dst_start_freq, dst_end_freq, src_start_freq,
start_subband, end_subband;
/* determine which channels use spx */
if (s->channel_mode == AC3_CHMODE_MONO) {
s->channel_uses_spx[1] = 1;
} else {
for (ch = 1; ch <= fbw_channels; ch++)
s->channel_uses_spx[ch] = get_bits1(gbc);
}
/* get the frequency bins of the spx copy region and the spx start
and end subbands */
dst_start_freq = get_bits(gbc, 2);
start_subband = get_bits(gbc, 3) + 2;
if (start_subband > 7)
start_subband += start_subband - 7;
end_subband = get_bits(gbc, 3) + 5;
if (end_subband > 7)
end_subband += end_subband - 7;
dst_start_freq = dst_start_freq * 12 + 25;
src_start_freq = start_subband * 12 + 25;
dst_end_freq = end_subband * 12 + 25;
/* check validity of spx ranges */
if (start_subband >= end_subband) {
av_log(s->avctx, AV_LOG_ERROR, "invalid spectral extension "
"range (%d >= %d)\n", start_subband, end_subband);
return -1;
}
if (dst_start_freq >= src_start_freq) {
av_log(s->avctx, AV_LOG_ERROR, "invalid spectral extension "
"copy start bin (%d >= %d)\n", dst_start_freq, src_start_freq);
return -1;
}
s->spx_dst_start_freq = dst_start_freq;
s->spx_src_start_freq = src_start_freq;
s->spx_dst_end_freq = dst_end_freq;
decode_band_structure(gbc, blk, s->eac3, 0,
start_subband, end_subband,
ff_eac3_default_spx_band_struct,
&s->num_spx_bands,
s->spx_band_sizes);
} else {
for (ch = 1; ch <= fbw_channels; ch++) {
s->channel_uses_spx[ch] = 0;
s->first_spx_coords[ch] = 1;
}
}
}
/* spectral extension coordinates */
if (s->spx_in_use) {
for (ch = 1; ch <= fbw_channels; ch++) {
if (s->channel_uses_spx[ch]) {
if (s->first_spx_coords[ch] || get_bits1(gbc)) {
float spx_blend;
int bin, master_spx_coord;
s->first_spx_coords[ch] = 0;
spx_blend = get_bits(gbc, 5) * (1.0f/32);
master_spx_coord = get_bits(gbc, 2) * 3;
bin = s->spx_src_start_freq;
for (bnd = 0; bnd < s->num_spx_bands; bnd++) {
int bandsize;
int spx_coord_exp, spx_coord_mant;
float nratio, sblend, nblend, spx_coord;
/* calculate blending factors */
bandsize = s->spx_band_sizes[bnd];
nratio = ((float)((bin + (bandsize >> 1))) / s->spx_dst_end_freq) - spx_blend;
nratio = av_clipf(nratio, 0.0f, 1.0f);
nblend = sqrtf(3.0f * nratio); // noise is scaled by sqrt(3)
// to give unity variance
sblend = sqrtf(1.0f - nratio);
bin += bandsize;
/* decode spx coordinates */
spx_coord_exp = get_bits(gbc, 4);
spx_coord_mant = get_bits(gbc, 2);
if (spx_coord_exp == 15) spx_coord_mant <<= 1;
else spx_coord_mant += 4;
spx_coord_mant <<= (25 - spx_coord_exp - master_spx_coord);
spx_coord = spx_coord_mant * (1.0f / (1 << 23));
/* multiply noise and signal blending factors by spx coordinate */
s->spx_noise_blend [ch][bnd] = nblend * spx_coord;
s->spx_signal_blend[ch][bnd] = sblend * spx_coord;
}
}
} else {
s->first_spx_coords[ch] = 1;
}
}
}
/* coupling strategy */
if (s->eac3 ? s->cpl_strategy_exists[blk] : get_bits1(gbc)) {
memset(bit_alloc_stages, 3, AC3_MAX_CHANNELS);
if (!s->eac3)
s->cpl_in_use[blk] = get_bits1(gbc);
if (s->cpl_in_use[blk]) {
/* coupling in use */
int cpl_start_subband, cpl_end_subband;
if (channel_mode < AC3_CHMODE_STEREO) {
av_log(s->avctx, AV_LOG_ERROR, "coupling not allowed in mono or dual-mono\n");
return -1;
}
/* check for enhanced coupling */
if (s->eac3 && get_bits1(gbc)) {
/* TODO: parse enhanced coupling strategy info */
av_log_missing_feature(s->avctx, "Enhanced coupling", 1);
return -1;
}
/* determine which channels are coupled */
if (s->eac3 && s->channel_mode == AC3_CHMODE_STEREO) {
s->channel_in_cpl[1] = 1;
s->channel_in_cpl[2] = 1;
} else {
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 */
cpl_start_subband = get_bits(gbc, 4);
cpl_end_subband = s->spx_in_use ? (s->spx_src_start_freq - 37) / 12 :
get_bits(gbc, 4) + 3;
if (cpl_start_subband >= cpl_end_subband) {
av_log(s->avctx, AV_LOG_ERROR, "invalid coupling range (%d >= %d)\n",
cpl_start_subband, cpl_end_subband);
return -1;
}
s->start_freq[CPL_CH] = cpl_start_subband * 12 + 37;
s->end_freq[CPL_CH] = cpl_end_subband * 12 + 37;
decode_band_structure(gbc, blk, s->eac3, 0, cpl_start_subband,
cpl_end_subband,
ff_eac3_default_cpl_band_struct,
&s->num_cpl_bands, s->cpl_band_sizes);
} else {
/* coupling not in use */
for (ch = 1; ch <= fbw_channels; ch++) {
s->channel_in_cpl[ch] = 0;
s->first_cpl_coords[ch] = 1;
}
s->first_cpl_leak = s->eac3;
s->phase_flags_in_use = 0;
}
} else if (!s->eac3) {
if (!blk) {
av_log(s->avctx, AV_LOG_ERROR, "new coupling strategy must "
"be present in block 0\n");
return -1;
} else {
s->cpl_in_use[blk] = s->cpl_in_use[blk-1];
}
}
cpl_in_use = s->cpl_in_use[blk];
/* 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 ((s->eac3 && s->first_cpl_coords[ch]) || get_bits1(gbc)) {
int master_cpl_coord, cpl_coord_exp, cpl_coord_mant;
s->first_cpl_coords[ch] = 0;
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;
}
} else {
/* channel not in coupling */
s->first_cpl_coords[ch] = 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 ((s->eac3 && !blk) || 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);
} else if (s->spx_in_use && s->spx_src_start_freq <= 61) {
s->num_rematrixing_bands--;
}
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_WARNING, "Warning: "
"new rematrixing strategy not present in block 0\n");
s->num_rematrixing_bands = 0;
}
}
/* exponent strategies for each channel */
for (ch = !cpl_in_use; ch <= s->channels; ch++) {
if (!s->eac3)
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 if (s->channel_uses_spx[ch])
s->end_freq[ch] = s->spx_src_start_freq;
else {
int bandwidth_code = get_bits(gbc, 6);
if (bandwidth_code > 60) {
av_log(s->avctx, AV_LOG_ERROR, "bandwidth code = %d > 60\n", 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;
if (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])) {
av_log(s->avctx, AV_LOG_ERROR, "exponent out-of-range\n");
return -1;
}
if (ch != CPL_CH && ch != s->lfe_ch)
skip_bits(gbc, 2); /* skip gainrng */
}
}
/* bit allocation information */
if (s->bit_allocation_syntax) {
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 (!s->eac3 || !blk) {
if (s->snr_offset_strategy && get_bits1(gbc)) {
int snr = 0;
int csnr;
csnr = (get_bits(gbc, 6) - 15) << 4;
for (i = ch = !cpl_in_use; ch <= s->channels; ch++) {
/* snr offset */
if (ch == i || s->snr_offset_strategy == 2)
snr = (csnr + get_bits(gbc, 4)) << 2;
/* run at least last bit allocation stage if snr offset changes */
if (blk && s->snr_offset[ch] != snr) {
bit_alloc_stages[ch] = FFMAX(bit_alloc_stages[ch], 1);
}
s->snr_offset[ch] = snr;
/* fast gain (normal AC-3 only) */
if (!s->eac3) {
int prev = s->fast_gain[ch];
s->fast_gain[ch] = ff_ac3_fast_gain_tab[get_bits(gbc, 3)];
/* run last 2 bit allocation stages if fast gain changes */
if (blk && prev != s->fast_gain[ch])
bit_alloc_stages[ch] = FFMAX(bit_alloc_stages[ch], 2);
}
}
} else if (!s->eac3 && !blk) {
av_log(s->avctx, AV_LOG_ERROR, "new snr offsets must be present in block 0\n");
return -1;
}
}
/* fast gain (E-AC-3 only) */
if (s->fast_gain_syntax && get_bits1(gbc)) {
for (ch = !cpl_in_use; ch <= s->channels; ch++) {
int prev = s->fast_gain[ch];
s->fast_gain[ch] = ff_ac3_fast_gain_tab[get_bits(gbc, 3)];
/* run last 2 bit allocation stages if fast gain changes */
if (blk && prev != s->fast_gain[ch])
bit_alloc_stages[ch] = FFMAX(bit_alloc_stages[ch], 2);
}
} else if (s->eac3 && !blk) {
for (ch = !cpl_in_use; ch <= s->channels; ch++)
s->fast_gain[ch] = ff_ac3_fast_gain_tab[4];
}
/* E-AC-3 to AC-3 converter SNR offset */
if (s->frame_type == EAC3_FRAME_TYPE_INDEPENDENT && get_bits1(gbc)) {
skip_bits(gbc, 10); // skip converter snr offset
}
/* coupling leak information */
if (cpl_in_use) {
if (s->first_cpl_leak || get_bits1(gbc)) {
int fl = get_bits(gbc, 3);
int sl = get_bits(gbc, 3);
/* run last 2 bit allocation stages for coupling channel if
coupling leak changes */
if (blk && (fl != s->bit_alloc_params.cpl_fast_leak ||
sl != s->bit_alloc_params.cpl_slow_leak)) {
bit_alloc_stages[CPL_CH] = FFMAX(bit_alloc_stages[CPL_CH], 2);
}
s->bit_alloc_params.cpl_fast_leak = fl;
s->bit_alloc_params.cpl_slow_leak = sl;
} else if (!s->eac3 && !blk) {
av_log(s->avctx, AV_LOG_ERROR, "new coupling leak info must "
"be present in block 0\n");
return -1;
}
s->first_cpl_leak = 0;
}
/* delta bit allocation information */
if (s->dba_syntax && 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) + 1;
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 */
if (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])) {
av_log(s->avctx, AV_LOG_ERROR, "error in bit allocation\n");
return -1;
}
}
if (bit_alloc_stages[ch] > 0) {
/* Compute bit allocation */
const uint8_t *bap_tab = s->channel_uses_aht[ch] ?
ff_eac3_hebap_tab : ff_ac3_bap_tab;
s->ac3dsp.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,
bap_tab, s->bap[ch]);
}
}
/* unused dummy data */
if (s->skip_syntax && 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. */
decode_transform_coeffs(s, blk);
/* TODO: generate enhanced coupling coordinates and uncouple */
/* 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[2 - ch];
} else {
gain *= s->dynamic_range[0];
}
s->fmt_conv.int32_to_float_fmul_scalar(s->transform_coeffs[ch],
s->fixed_coeffs[ch], gain, 256);
}
/* apply spectral extension to high frequency bins */
if (s->spx_in_use && CONFIG_EAC3_DECODER) {
ff_eac3_apply_spectral_extension(s);
}
/* 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) {
s->ac3dsp.downmix(s->output, s->downmix_coeffs,
s->out_channels, s->fbw_channels, 256);
}
} else {
if (downmix_output) {
s->ac3dsp.downmix(s->transform_coeffs + 1, s->downmix_coeffs,
s->out_channels, s->fbw_channels, 256);
}
if (downmix_output && !s->downmixed) {
s->downmixed = 1;
s->ac3dsp.downmix(s->delay, s->downmix_coeffs, s->out_channels,
s->fbw_channels, 128);
}
do_imdct(s, s->out_channels);
}
return 0;
}
/**
* Decode a single AC-3 frame.
*/
static int ac3_decode_frame(AVCodecContext * avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
AC3DecodeContext *s = avctx->priv_data;
float *out_samples_flt;
int16_t *out_samples_s16;
int blk, ch, err, ret;
const uint8_t *channel_map;
const float *output[AC3_MAX_CHANNELS];
/* copy input buffer to decoder context to avoid reading past the end
of the buffer, which can be caused by a damaged input stream. */
if (buf_size >= 2 && AV_RB16(buf) == 0x770B) {
// seems to be byte-swapped AC-3
int cnt = FFMIN(buf_size, AC3_FRAME_BUFFER_SIZE) >> 1;
s->dsp.bswap16_buf((uint16_t *)s->input_buffer, (const uint16_t *)buf, cnt);
} else
memcpy(s->input_buffer, buf, FFMIN(buf_size, AC3_FRAME_BUFFER_SIZE));
buf = s->input_buffer;
/* initialize the GetBitContext with the start of valid AC-3 Frame */
init_get_bits(&s->gbc, buf, buf_size * 8);
/* parse the syncinfo */
err = parse_frame_header(s);
if (err) {
switch (err) {
case AAC_AC3_PARSE_ERROR_SYNC:
av_log(avctx, AV_LOG_ERROR, "frame sync error\n");
return -1;
case AAC_AC3_PARSE_ERROR_BSID:
av_log(avctx, AV_LOG_ERROR, "invalid bitstream id\n");
break;
case AAC_AC3_PARSE_ERROR_SAMPLE_RATE:
av_log(avctx, AV_LOG_ERROR, "invalid sample rate\n");
break;
case AAC_AC3_PARSE_ERROR_FRAME_SIZE:
av_log(avctx, AV_LOG_ERROR, "invalid frame size\n");
break;
case AAC_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");
*got_frame_ptr = 0;
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;
}
} else {
/* 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 = AAC_AC3_PARSE_ERROR_FRAME_SIZE;
} else if (avctx->err_recognition & (AV_EF_CRCCHECK|AV_EF_CAREFUL)) {
/* check for crc mismatch */
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 = AAC_AC3_PARSE_ERROR_CRC;
}
}
}
/* 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;
s->channel_layout = avpriv_ac3_channel_layout_tab[s->output_mode];
}
avctx->channels = s->out_channels;
avctx->channel_layout = s->channel_layout;
s->loro_center_mix_level = gain_levels[s-> center_mix_level];
s->loro_surround_mix_level = gain_levels[s->surround_mix_level];
s->ltrt_center_mix_level = LEVEL_MINUS_3DB;
s->ltrt_surround_mix_level = LEVEL_MINUS_3DB;
/* 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;
}
if (avctx->channels != s->out_channels) {
av_log(avctx, AV_LOG_ERROR, "channel number mismatching on damaged frame\n");
return AVERROR_INVALIDDATA;
}
/* set audio service type based on bitstream mode for AC-3 */
avctx->audio_service_type = s->bitstream_mode;
if (s->bitstream_mode == 0x7 && s->channels > 1)
avctx->audio_service_type = AV_AUDIO_SERVICE_TYPE_KARAOKE;
/* get output buffer */
s->frame.nb_samples = s->num_blocks * 256;
if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
out_samples_flt = (float *)s->frame.data[0];
out_samples_s16 = (int16_t *)s->frame.data[0];
/* decode the audio blocks */
channel_map = ff_ac3_dec_channel_map[s->output_mode & ~AC3_OUTPUT_LFEON][s->lfe_on];
for (ch = 0; ch < s->out_channels; ch++)
output[ch] = s->output[channel_map[ch]];
for (blk = 0; blk < s->num_blocks; blk++) {
if (!err && decode_audio_block(s, blk)) {
av_log(avctx, AV_LOG_ERROR, "error decoding the audio block\n");
err = 1;
}
if (avctx->sample_fmt == AV_SAMPLE_FMT_FLT) {
s->fmt_conv.float_interleave(out_samples_flt, output, 256,
s->out_channels);
out_samples_flt += 256 * s->out_channels;
} else {
s->fmt_conv.float_to_int16_interleave(out_samples_s16, output, 256,
s->out_channels);
out_samples_s16 += 256 * s->out_channels;
}
}
s->frame.decode_error_flags = err ? FF_DECODE_ERROR_INVALID_BITSTREAM : 0;
*got_frame_ptr = 1;
*(AVFrame *)data = s->frame;
return FFMIN(buf_size, 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);
return 0;
}
#define OFFSET(x) offsetof(AC3DecodeContext, x)
#define PAR (AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM)
static const AVOption options[] = {
{ "drc_scale", "percentage of dynamic range compression to apply", OFFSET(drc_scale), AV_OPT_TYPE_FLOAT, {.dbl = 1.0}, 0.0, 1.0, PAR },
{"dmix_mode", "Preferred Stereo Downmix Mode", OFFSET(preferred_stereo_downmix), AV_OPT_TYPE_INT, {.i64 = -1 }, -1, 2, 0, "dmix_mode"},
{"ltrt_cmixlev", "Lt/Rt Center Mix Level", OFFSET(ltrt_center_mix_level), AV_OPT_TYPE_FLOAT, {.dbl = -1.0 }, -1.0, 2.0, 0},
{"ltrt_surmixlev", "Lt/Rt Surround Mix Level", OFFSET(ltrt_surround_mix_level), AV_OPT_TYPE_FLOAT, {.dbl = -1.0 }, -1.0, 2.0, 0},
{"loro_cmixlev", "Lo/Ro Center Mix Level", OFFSET(loro_center_mix_level), AV_OPT_TYPE_FLOAT, {.dbl = -1.0 }, -1.0, 2.0, 0},
{"loro_surmixlev", "Lo/Ro Surround Mix Level", OFFSET(loro_surround_mix_level), AV_OPT_TYPE_FLOAT, {.dbl = -1.0 }, -1.0, 2.0, 0},
{ NULL},
};
static const AVClass ac3_decoder_class = {
.class_name = "AC3 decoder",
.item_name = av_default_item_name,
.option = options,
.version = LIBAVUTIL_VERSION_INT,
};
AVCodec ff_ac3_decoder = {
.name = "ac3",
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_AC3,
.priv_data_size = sizeof (AC3DecodeContext),
.init = ac3_decode_init,
.close = ac3_decode_end,
.decode = ac3_decode_frame,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("ATSC A/52A (AC-3)"),
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLT,
AV_SAMPLE_FMT_S16,
AV_SAMPLE_FMT_NONE },
.priv_class = &ac3_decoder_class,
};
#if CONFIG_EAC3_DECODER
static const AVClass eac3_decoder_class = {
.class_name = "E-AC3 decoder",
.item_name = av_default_item_name,
.option = options,
.version = LIBAVUTIL_VERSION_INT,
};
AVCodec ff_eac3_decoder = {
.name = "eac3",
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_EAC3,
.priv_data_size = sizeof (AC3DecodeContext),
.init = ac3_decode_init,
.close = ac3_decode_end,
.decode = ac3_decode_frame,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("ATSC A/52B (AC-3, E-AC-3)"),
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLT,
AV_SAMPLE_FMT_S16,
AV_SAMPLE_FMT_NONE },
.priv_class = &eac3_decoder_class,
};
#endif
Reference in New Issue View Git Blame Copy Permalink