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FFmpeg/libavcodec/dca_lbr.c
Michael Niedermayer bfea054a75 avcodec/dca_lbr: Fix some error codes and error passing 
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2020-02-03 00:11:18 +01:00

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/*
* Copyright (C) 2016 foo86
*
* 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
*/
#define BITSTREAM_READER_LE
#include "libavutil/channel_layout.h"
#include "dcadec.h"
#include "dcadata.h"
#include "dcahuff.h"
#include "dca_syncwords.h"
#include "bytestream.h"
#define AMP_MAX 56
enum LBRFlags {
LBR_FLAG_24_BIT = 0x01,
LBR_FLAG_LFE_PRESENT = 0x02,
LBR_FLAG_BAND_LIMIT_2_3 = 0x04,
LBR_FLAG_BAND_LIMIT_1_2 = 0x08,
LBR_FLAG_BAND_LIMIT_1_3 = 0x0c,
LBR_FLAG_BAND_LIMIT_1_4 = 0x10,
LBR_FLAG_BAND_LIMIT_1_8 = 0x18,
LBR_FLAG_BAND_LIMIT_NONE = 0x14,
LBR_FLAG_BAND_LIMIT_MASK = 0x1c,
LBR_FLAG_DMIX_STEREO = 0x20,
LBR_FLAG_DMIX_MULTI_CH = 0x40
};
enum LBRChunkTypes {
LBR_CHUNK_NULL = 0x00,
LBR_CHUNK_PAD = 0x01,
LBR_CHUNK_FRAME = 0x04,
LBR_CHUNK_FRAME_NO_CSUM = 0x06,
LBR_CHUNK_LFE = 0x0a,
LBR_CHUNK_ECS = 0x0b,
LBR_CHUNK_RESERVED_1 = 0x0c,
LBR_CHUNK_RESERVED_2 = 0x0d,
LBR_CHUNK_SCF = 0x0e,
LBR_CHUNK_TONAL = 0x10,
LBR_CHUNK_TONAL_GRP_1 = 0x11,
LBR_CHUNK_TONAL_GRP_2 = 0x12,
LBR_CHUNK_TONAL_GRP_3 = 0x13,
LBR_CHUNK_TONAL_GRP_4 = 0x14,
LBR_CHUNK_TONAL_GRP_5 = 0x15,
LBR_CHUNK_TONAL_SCF = 0x16,
LBR_CHUNK_TONAL_SCF_GRP_1 = 0x17,
LBR_CHUNK_TONAL_SCF_GRP_2 = 0x18,
LBR_CHUNK_TONAL_SCF_GRP_3 = 0x19,
LBR_CHUNK_TONAL_SCF_GRP_4 = 0x1a,
LBR_CHUNK_TONAL_SCF_GRP_5 = 0x1b,
LBR_CHUNK_RES_GRID_LR = 0x30,
LBR_CHUNK_RES_GRID_LR_LAST = 0x3f,
LBR_CHUNK_RES_GRID_HR = 0x40,
LBR_CHUNK_RES_GRID_HR_LAST = 0x4f,
LBR_CHUNK_RES_TS_1 = 0x50,
LBR_CHUNK_RES_TS_1_LAST = 0x5f,
LBR_CHUNK_RES_TS_2 = 0x60,
LBR_CHUNK_RES_TS_2_LAST = 0x6f,
LBR_CHUNK_EXTENSION = 0x7f
};
typedef struct LBRChunk {
int id, len;
const uint8_t *data;
} LBRChunk;
static const int8_t channel_reorder_nolfe[7][5] = {
{ 0, -1, -1, -1, -1 }, // C
{ 0, 1, -1, -1, -1 }, // LR
{ 0, 1, 2, -1, -1 }, // LR C
{ 0, 1, -1, -1, -1 }, // LsRs
{ 1, 2, 0, -1, -1 }, // LsRs C
{ 0, 1, 2, 3, -1 }, // LR LsRs
{ 0, 1, 3, 4, 2 }, // LR LsRs C
};
static const int8_t channel_reorder_lfe[7][5] = {
{ 0, -1, -1, -1, -1 }, // C
{ 0, 1, -1, -1, -1 }, // LR
{ 0, 1, 2, -1, -1 }, // LR C
{ 1, 2, -1, -1, -1 }, // LsRs
{ 2, 3, 0, -1, -1 }, // LsRs C
{ 0, 1, 3, 4, -1 }, // LR LsRs
{ 0, 1, 4, 5, 2 }, // LR LsRs C
};
static const uint8_t lfe_index[7] = {
1, 2, 3, 0, 1, 2, 3
};
static const uint8_t channel_counts[7] = {
1, 2, 3, 2, 3, 4, 5
};
static const uint16_t channel_layouts[7] = {
AV_CH_LAYOUT_MONO,
AV_CH_LAYOUT_STEREO,
AV_CH_LAYOUT_SURROUND,
AV_CH_SIDE_LEFT | AV_CH_SIDE_RIGHT,
AV_CH_FRONT_CENTER | AV_CH_SIDE_LEFT | AV_CH_SIDE_RIGHT,
AV_CH_LAYOUT_2_2,
AV_CH_LAYOUT_5POINT0
};
static float cos_tab[256];
static float lpc_tab[16];
static av_cold void init_tables(void)
{
static int initialized;
int i;
if (initialized)
return;
for (i = 0; i < 256; i++)
cos_tab[i] = cos(M_PI * i / 128);
for (i = 0; i < 16; i++)
lpc_tab[i] = sin((i - 8) * (M_PI / ((i < 8) ? 17 : 15)));
initialized = 1;
}
static int parse_lfe_24(DCALbrDecoder *s)
{
int step_max = FF_ARRAY_ELEMS(ff_dca_lfe_step_size_24) - 1;
int i, ps, si, code, step_i;
float step, value, delta;
ps = get_bits(&s->gb, 24);
si = ps >> 23;
value = (((ps & 0x7fffff) ^ -si) + si) * (1.0f / 0x7fffff);
step_i = get_bits(&s->gb, 8);
if (step_i > step_max) {
av_log(s->avctx, AV_LOG_ERROR, "Invalid LFE step size index\n");
return AVERROR_INVALIDDATA;
}
step = ff_dca_lfe_step_size_24[step_i];
for (i = 0; i < 64; i++) {
code = get_bits(&s->gb, 6);
delta = step * 0.03125f;
if (code & 16)
delta += step;
if (code & 8)
delta += step * 0.5f;
if (code & 4)
delta += step * 0.25f;
if (code & 2)
delta += step * 0.125f;
if (code & 1)
delta += step * 0.0625f;
if (code & 32) {
value -= delta;
if (value < -3.0f)
value = -3.0f;
} else {
value += delta;
if (value > 3.0f)
value = 3.0f;
}
step_i += ff_dca_lfe_delta_index_24[code & 31];
step_i = av_clip(step_i, 0, step_max);
step = ff_dca_lfe_step_size_24[step_i];
s->lfe_data[i] = value * s->lfe_scale;
}
return 0;
}
static int parse_lfe_16(DCALbrDecoder *s)
{
int step_max = FF_ARRAY_ELEMS(ff_dca_lfe_step_size_16) - 1;
int i, ps, si, code, step_i;
float step, value, delta;
ps = get_bits(&s->gb, 16);
si = ps >> 15;
value = (((ps & 0x7fff) ^ -si) + si) * (1.0f / 0x7fff);
step_i = get_bits(&s->gb, 8);
if (step_i > step_max) {
av_log(s->avctx, AV_LOG_ERROR, "Invalid LFE step size index\n");
return AVERROR_INVALIDDATA;
}
step = ff_dca_lfe_step_size_16[step_i];
for (i = 0; i < 64; i++) {
code = get_bits(&s->gb, 4);
delta = step * 0.125f;
if (code & 4)
delta += step;
if (code & 2)
delta += step * 0.5f;
if (code & 1)
delta += step * 0.25f;
if (code & 8) {
value -= delta;
if (value < -3.0f)
value = -3.0f;
} else {
value += delta;
if (value > 3.0f)
value = 3.0f;
}
step_i += ff_dca_lfe_delta_index_16[code & 7];
step_i = av_clip(step_i, 0, step_max);
step = ff_dca_lfe_step_size_16[step_i];
s->lfe_data[i] = value * s->lfe_scale;
}
return 0;
}
static int parse_lfe_chunk(DCALbrDecoder *s, LBRChunk *chunk)
{
int ret;
if (!(s->flags & LBR_FLAG_LFE_PRESENT))
return 0;
if (!chunk->len)
return 0;
ret = init_get_bits8(&s->gb, chunk->data, chunk->len);
if (ret < 0)
return ret;
// Determine bit depth from chunk size
if (chunk->len >= 52)
return parse_lfe_24(s);
if (chunk->len >= 35)
return parse_lfe_16(s);
av_log(s->avctx, AV_LOG_ERROR, "LFE chunk too short\n");
return AVERROR_INVALIDDATA;
}
static inline int parse_vlc(GetBitContext *s, VLC *vlc, int max_depth)
{
int v = get_vlc2(s, vlc->table, vlc->bits, max_depth);
if (v > 0)
return v - 1;
// Rare value
return get_bits(s, get_bits(s, 3) + 1);
}
static int parse_tonal(DCALbrDecoder *s, int group)
{
unsigned int amp[DCA_LBR_CHANNELS_TOTAL];
unsigned int phs[DCA_LBR_CHANNELS_TOTAL];
unsigned int diff, main_amp, shift;
int sf, sf_idx, ch, main_ch, freq;
int ch_nbits = av_ceil_log2(s->nchannels_total);
// Parse subframes for this group
for (sf = 0; sf < 1 << group; sf += diff ? 8 : 1) {
sf_idx = ((s->framenum << group) + sf) & 31;
s->tonal_bounds[group][sf_idx][0] = s->ntones;
// Parse tones for this subframe
for (freq = 1;; freq++) {
if (get_bits_left(&s->gb) < 1) {
av_log(s->avctx, AV_LOG_ERROR, "Tonal group chunk too short\n");
return AVERROR_INVALIDDATA;
}
diff = parse_vlc(&s->gb, &ff_dca_vlc_tnl_grp[group], 2);
if (diff >= FF_ARRAY_ELEMS(ff_dca_fst_amp)) {
av_log(s->avctx, AV_LOG_ERROR, "Invalid tonal frequency diff\n");
return AVERROR_INVALIDDATA;
}
diff = get_bitsz(&s->gb, diff >> 2) + ff_dca_fst_amp[diff];
if (diff <= 1)
break; // End of subframe
freq += diff - 2;
if (freq >> (5 - group) > s->nsubbands * 4 - 6) {
av_log(s->avctx, AV_LOG_ERROR, "Invalid spectral line offset\n");
return AVERROR_INVALIDDATA;
}
// Main channel
main_ch = get_bitsz(&s->gb, ch_nbits);
main_amp = parse_vlc(&s->gb, &ff_dca_vlc_tnl_scf, 2)
+ s->tonal_scf[ff_dca_freq_to_sb[freq >> (7 - group)]]
+ s->limited_range - 2;
amp[main_ch] = main_amp < AMP_MAX ? main_amp : 0;
phs[main_ch] = get_bits(&s->gb, 3);
// Secondary channels
for (ch = 0; ch < s->nchannels_total; ch++) {
if (ch == main_ch)
continue;
if (get_bits1(&s->gb)) {
amp[ch] = amp[main_ch] - parse_vlc(&s->gb, &ff_dca_vlc_damp, 1);
phs[ch] = phs[main_ch] - parse_vlc(&s->gb, &ff_dca_vlc_dph, 1);
} else {
amp[ch] = 0;
phs[ch] = 0;
}
}
if (amp[main_ch]) {
// Allocate new tone
DCALbrTone *t = &s->tones[s->ntones];
s->ntones = (s->ntones + 1) & (DCA_LBR_TONES - 1);
t->x_freq = freq >> (5 - group);
t->f_delt = (freq & ((1 << (5 - group)) - 1)) << group;
t->ph_rot = 256 - (t->x_freq & 1) * 128 - t->f_delt * 4;
shift = ff_dca_ph0_shift[(t->x_freq & 3) * 2 + (freq & 1)]
- ((t->ph_rot << (5 - group)) - t->ph_rot);
for (ch = 0; ch < s->nchannels; ch++) {
t->amp[ch] = amp[ch] < AMP_MAX ? amp[ch] : 0;
t->phs[ch] = 128 - phs[ch] * 32 + shift;
}
}
}
s->tonal_bounds[group][sf_idx][1] = s->ntones;
}
return 0;
}
static int parse_tonal_chunk(DCALbrDecoder *s, LBRChunk *chunk)
{
int sb, group, ret;
if (!chunk->len)
return 0;
ret = init_get_bits8(&s->gb, chunk->data, chunk->len);
if (ret < 0)
return ret;
// Scale factors
if (chunk->id == LBR_CHUNK_SCF || chunk->id == LBR_CHUNK_TONAL_SCF) {
if (get_bits_left(&s->gb) < 36) {
av_log(s->avctx, AV_LOG_ERROR, "Tonal scale factor chunk too short\n");
return AVERROR_INVALIDDATA;
}
for (sb = 0; sb < 6; sb++)
s->tonal_scf[sb] = get_bits(&s->gb, 6);
}
// Tonal groups
if (chunk->id == LBR_CHUNK_TONAL || chunk->id == LBR_CHUNK_TONAL_SCF)
for (group = 0; group < 5; group++) {
ret = parse_tonal(s, group);
if (ret < 0)
return ret;
}
return 0;
}
static int parse_tonal_group(DCALbrDecoder *s, LBRChunk *chunk)
{
int ret;
if (!chunk->len)
return 0;
ret = init_get_bits8(&s->gb, chunk->data, chunk->len);
if (ret < 0)
return ret;
return parse_tonal(s, chunk->id);
}
/**
* Check point to ensure that enough bits are left. Aborts decoding
* by skipping to the end of chunk otherwise.
*/
static int ensure_bits(GetBitContext *s, int n)
{
int left = get_bits_left(s);
if (left < 0)
return AVERROR_INVALIDDATA;
if (left < n) {
skip_bits_long(s, left);
return 1;
}
return 0;
}
static int parse_scale_factors(DCALbrDecoder *s, uint8_t *scf)
{
int i, sf, prev, next, dist;
// Truncated scale factors remain zero
if (ensure_bits(&s->gb, 20))
return 0;
// Initial scale factor
prev = parse_vlc(&s->gb, &ff_dca_vlc_fst_rsd_amp, 2);
for (sf = 0; sf < 7; sf += dist) {
scf[sf] = prev; // Store previous value
if (ensure_bits(&s->gb, 20))
return 0;
// Interpolation distance
dist = parse_vlc(&s->gb, &ff_dca_vlc_rsd_apprx, 1) + 1;
if (dist > 7 - sf) {
av_log(s->avctx, AV_LOG_ERROR, "Invalid scale factor distance\n");
return AVERROR_INVALIDDATA;
}
if (ensure_bits(&s->gb, 20))
return 0;
// Final interpolation point
next = parse_vlc(&s->gb, &ff_dca_vlc_rsd_amp, 2);
if (next & 1)
next = prev + ((next + 1) >> 1);
else
next = prev - ( next >> 1);
// Interpolate
switch (dist) {
case 2:
if (next > prev)
scf[sf + 1] = prev + ((next - prev) >> 1);
else
scf[sf + 1] = prev - ((prev - next) >> 1);
break;
case 4:
if (next > prev) {
scf[sf + 1] = prev + ( (next - prev) >> 2);
scf[sf + 2] = prev + ( (next - prev) >> 1);
scf[sf + 3] = prev + (((next - prev) * 3) >> 2);
} else {
scf[sf + 1] = prev - ( (prev - next) >> 2);
scf[sf + 2] = prev - ( (prev - next) >> 1);
scf[sf + 3] = prev - (((prev - next) * 3) >> 2);
}
break;
default:
for (i = 1; i < dist; i++)
scf[sf + i] = prev + (next - prev) * i / dist;
break;
}
prev = next;
}
scf[sf] = next; // Store final value
return 0;
}
static int parse_st_code(GetBitContext *s, int min_v)
{
unsigned int v = parse_vlc(s, &ff_dca_vlc_st_grid, 2) + min_v;
if (v & 1)
v = 16 + (v >> 1);
else
v = 16 - (v >> 1);
if (v >= FF_ARRAY_ELEMS(ff_dca_st_coeff))
v = 16;
return v;
}
static int parse_grid_1_chunk(DCALbrDecoder *s, LBRChunk *chunk, int ch1, int ch2)
{
int ch, sb, sf, nsubbands, ret;
if (!chunk->len)
return 0;
ret = init_get_bits8(&s->gb, chunk->data, chunk->len);
if (ret < 0)
return ret;
// Scale factors
nsubbands = ff_dca_scf_to_grid_1[s->nsubbands - 1] + 1;
for (sb = 2; sb < nsubbands; sb++) {
ret = parse_scale_factors(s, s->grid_1_scf[ch1][sb]);
if (ret < 0)
return ret;
if (ch1 != ch2 && ff_dca_grid_1_to_scf[sb] < s->min_mono_subband) {
ret = parse_scale_factors(s, s->grid_1_scf[ch2][sb]);
if (ret < 0)
return ret;
}
}
if (get_bits_left(&s->gb) < 1)
return 0; // Should not happen, but a sample exists that proves otherwise
// Average values for third grid
for (sb = 0; sb < s->nsubbands - 4; sb++) {
s->grid_3_avg[ch1][sb] = parse_vlc(&s->gb, &ff_dca_vlc_avg_g3, 2) - 16;
if (ch1 != ch2) {
if (sb + 4 < s->min_mono_subband)
s->grid_3_avg[ch2][sb] = parse_vlc(&s->gb, &ff_dca_vlc_avg_g3, 2) - 16;
else
s->grid_3_avg[ch2][sb] = s->grid_3_avg[ch1][sb];
}
}
if (get_bits_left(&s->gb) < 0) {
av_log(s->avctx, AV_LOG_ERROR, "First grid chunk too short\n");
return AVERROR_INVALIDDATA;
}
// Stereo image for partial mono mode
if (ch1 != ch2) {
int min_v[2];
if (ensure_bits(&s->gb, 8))
return 0;
min_v[0] = get_bits(&s->gb, 4);
min_v[1] = get_bits(&s->gb, 4);
nsubbands = (s->nsubbands - s->min_mono_subband + 3) / 4;
for (sb = 0; sb < nsubbands; sb++)
for (ch = ch1; ch <= ch2; ch++)
for (sf = 1; sf <= 4; sf++)
s->part_stereo[ch][sb][sf] = parse_st_code(&s->gb, min_v[ch - ch1]);
if (get_bits_left(&s->gb) >= 0)
s->part_stereo_pres |= 1 << ch1;
}
// Low resolution spatial information is not decoded
return 0;
}
static int parse_grid_1_sec_ch(DCALbrDecoder *s, int ch2)
{
int sb, nsubbands, ret;
// Scale factors
nsubbands = ff_dca_scf_to_grid_1[s->nsubbands - 1] + 1;
for (sb = 2; sb < nsubbands; sb++) {
if (ff_dca_grid_1_to_scf[sb] >= s->min_mono_subband) {
ret = parse_scale_factors(s, s->grid_1_scf[ch2][sb]);
if (ret < 0)
return ret;
}
}
// Average values for third grid
for (sb = 0; sb < s->nsubbands - 4; sb++) {
if (sb + 4 >= s->min_mono_subband) {
if (ensure_bits(&s->gb, 20))
return 0;
s->grid_3_avg[ch2][sb] = parse_vlc(&s->gb, &ff_dca_vlc_avg_g3, 2) - 16;
}
}
return 0;
}
static void parse_grid_3(DCALbrDecoder *s, int ch1, int ch2, int sb, int flag)
{
int i, ch;
for (ch = ch1; ch <= ch2; ch++) {
if ((ch != ch1 && sb + 4 >= s->min_mono_subband) != flag)
continue;
if (s->grid_3_pres[ch] & (1U << sb))
continue; // Already parsed
for (i = 0; i < 8; i++) {
if (ensure_bits(&s->gb, 20))
return;
s->grid_3_scf[ch][sb][i] = parse_vlc(&s->gb, &ff_dca_vlc_grid_3, 2) - 16;
}
// Flag scale factors for this subband parsed
s->grid_3_pres[ch] |= 1U << sb;
}
}
static float lbr_rand(DCALbrDecoder *s, int sb)
{
s->lbr_rand = 1103515245U * s->lbr_rand + 12345U;
return s->lbr_rand * s->sb_scf[sb];
}
/**
* Parse time samples for one subband, filling truncated samples with randomness
*/
static void parse_ch(DCALbrDecoder *s, int ch, int sb, int quant_level, int flag)
{
float *samples = s->time_samples[ch][sb];
int i, j, code, nblocks, coding_method;
if (ensure_bits(&s->gb, 20))
return; // Too few bits left
coding_method = get_bits1(&s->gb);
switch (quant_level) {
case 1:
nblocks = FFMIN(get_bits_left(&s->gb) / 8, DCA_LBR_TIME_SAMPLES / 8);
for (i = 0; i < nblocks; i++, samples += 8) {
code = get_bits(&s->gb, 8);
for (j = 0; j < 8; j++)
samples[j] = ff_dca_rsd_level_2a[(code >> j) & 1];
}
i = nblocks * 8;
break;
case 2:
if (coding_method) {
for (i = 0; i < DCA_LBR_TIME_SAMPLES && get_bits_left(&s->gb) >= 2; i++) {
if (get_bits1(&s->gb))
samples[i] = ff_dca_rsd_level_2b[get_bits1(&s->gb)];
else
samples[i] = 0;
}
} else {
nblocks = FFMIN(get_bits_left(&s->gb) / 8, (DCA_LBR_TIME_SAMPLES + 4) / 5);
for (i = 0; i < nblocks; i++, samples += 5) {
code = ff_dca_rsd_pack_5_in_8[get_bits(&s->gb, 8)];
for (j = 0; j < 5; j++)
samples[j] = ff_dca_rsd_level_3[(code >> j * 2) & 3];
}
i = nblocks * 5;
}
break;
case 3:
nblocks = FFMIN(get_bits_left(&s->gb) / 7, (DCA_LBR_TIME_SAMPLES + 2) / 3);
for (i = 0; i < nblocks; i++, samples += 3) {
code = get_bits(&s->gb, 7);
for (j = 0; j < 3; j++)
samples[j] = ff_dca_rsd_level_5[ff_dca_rsd_pack_3_in_7[code][j]];
}
i = nblocks * 3;
break;
case 4:
for (i = 0; i < DCA_LBR_TIME_SAMPLES && get_bits_left(&s->gb) >= 6; i++)
samples[i] = ff_dca_rsd_level_8[get_vlc2(&s->gb, ff_dca_vlc_rsd.table, 6, 1)];
break;
case 5:
nblocks = FFMIN(get_bits_left(&s->gb) / 4, DCA_LBR_TIME_SAMPLES);
for (i = 0; i < nblocks; i++)
samples[i] = ff_dca_rsd_level_16[get_bits(&s->gb, 4)];
break;
default:
av_assert0(0);
}
if (flag && get_bits_left(&s->gb) < 20)
return; // Skip incomplete mono subband
for (; i < DCA_LBR_TIME_SAMPLES; i++)
s->time_samples[ch][sb][i] = lbr_rand(s, sb);
s->ch_pres[ch] |= 1U << sb;
}
static int parse_ts(DCALbrDecoder *s, int ch1, int ch2,
int start_sb, int end_sb, int flag)
{
int sb, sb_g3, sb_reorder, quant_level;
for (sb = start_sb; sb < end_sb; sb++) {
// Subband number before reordering
if (sb < 6) {
sb_reorder = sb;
} else if (flag && sb < s->max_mono_subband) {
sb_reorder = s->sb_indices[sb];
} else {
if (ensure_bits(&s->gb, 28))
break;
sb_reorder = get_bits(&s->gb, s->limited_range + 3);
if (sb_reorder < 6)
sb_reorder = 6;
s->sb_indices[sb] = sb_reorder;
}
if (sb_reorder >= s->nsubbands)
return AVERROR_INVALIDDATA;
// Third grid scale factors
if (sb == 12) {
for (sb_g3 = 0; sb_g3 < s->g3_avg_only_start_sb - 4; sb_g3++)
parse_grid_3(s, ch1, ch2, sb_g3, flag);
} else if (sb < 12 && sb_reorder >= 4) {
parse_grid_3(s, ch1, ch2, sb_reorder - 4, flag);
}
// Secondary channel flags
if (ch1 != ch2) {
if (ensure_bits(&s->gb, 20))
break;
if (!flag || sb_reorder >= s->max_mono_subband)
s->sec_ch_sbms[ch1 / 2][sb_reorder] = get_bits(&s->gb, 8);
if (flag && sb_reorder >= s->min_mono_subband)
s->sec_ch_lrms[ch1 / 2][sb_reorder] = get_bits(&s->gb, 8);
}
quant_level = s->quant_levels[ch1 / 2][sb];
if (!quant_level)
return AVERROR_INVALIDDATA;
// Time samples for one or both channels
if (sb < s->max_mono_subband && sb_reorder >= s->min_mono_subband) {
if (!flag)
parse_ch(s, ch1, sb_reorder, quant_level, 0);
else if (ch1 != ch2)
parse_ch(s, ch2, sb_reorder, quant_level, 1);
} else {
parse_ch(s, ch1, sb_reorder, quant_level, 0);
if (ch1 != ch2)
parse_ch(s, ch2, sb_reorder, quant_level, 0);
}
}
return 0;
}
/**
* Convert from reflection coefficients to direct form coefficients
*/
static void convert_lpc(float *coeff, const int *codes)
{
int i, j;
for (i = 0; i < 8; i++) {
float rc = lpc_tab[codes[i]];
for (j = 0; j < (i + 1) / 2; j++) {
float tmp1 = coeff[ j ];
float tmp2 = coeff[i - j - 1];
coeff[ j ] = tmp1 + rc * tmp2;
coeff[i - j - 1] = tmp2 + rc * tmp1;
}
coeff[i] = rc;
}
}
static int parse_lpc(DCALbrDecoder *s, int ch1, int ch2, int start_sb, int end_sb)
{
int f = s->framenum & 1;
int i, sb, ch, codes[16];
// First two subbands have two sets of coefficients, third subband has one
for (sb = start_sb; sb < end_sb; sb++) {
int ncodes = 8 * (1 + (sb < 2));
for (ch = ch1; ch <= ch2; ch++) {
if (ensure_bits(&s->gb, 4 * ncodes))
return 0;
for (i = 0; i < ncodes; i++)
codes[i] = get_bits(&s->gb, 4);
for (i = 0; i < ncodes / 8; i++)
convert_lpc(s->lpc_coeff[f][ch][sb][i], &codes[i * 8]);
}
}
return 0;
}
static int parse_high_res_grid(DCALbrDecoder *s, LBRChunk *chunk, int ch1, int ch2)
{
int quant_levels[DCA_LBR_SUBBANDS];
int sb, ch, ol, st, max_sb, profile, ret;
if (!chunk->len)
return 0;
ret = init_get_bits8(&s->gb, chunk->data, chunk->len);
if (ret < 0)
return ret;
// Quantizer profile
profile = get_bits(&s->gb, 8);
// Overall level
ol = (profile >> 3) & 7;
// Steepness
st = profile >> 6;
// Max energy subband
max_sb = profile & 7;
// Calculate quantization levels
for (sb = 0; sb < s->nsubbands; sb++) {
int f = sb * s->limited_rate / s->nsubbands;
int a = 18000 / (12 * f / 1000 + 100 + 40 * st) + 20 * ol;
if (a <= 95)
quant_levels[sb] = 1;
else if (a <= 140)
quant_levels[sb] = 2;
else if (a <= 180)
quant_levels[sb] = 3;
else if (a <= 230)
quant_levels[sb] = 4;
else
quant_levels[sb] = 5;
}
// Reorder quantization levels for lower subbands
for (sb = 0; sb < 8; sb++)
s->quant_levels[ch1 / 2][sb] = quant_levels[ff_dca_sb_reorder[max_sb][sb]];
for (; sb < s->nsubbands; sb++)
s->quant_levels[ch1 / 2][sb] = quant_levels[sb];
// LPC for the first two subbands
ret = parse_lpc(s, ch1, ch2, 0, 2);
if (ret < 0)
return ret;
// Time-samples for the first two subbands of main channel
ret = parse_ts(s, ch1, ch2, 0, 2, 0);
if (ret < 0)
return ret;
// First two bands of the first grid
for (sb = 0; sb < 2; sb++)
for (ch = ch1; ch <= ch2; ch++)
if ((ret = parse_scale_factors(s, s->grid_1_scf[ch][sb])) < 0)
return ret;
return 0;
}
static int parse_grid_2(DCALbrDecoder *s, int ch1, int ch2,
int start_sb, int end_sb, int flag)
{
int i, j, sb, ch, nsubbands;
nsubbands = ff_dca_scf_to_grid_2[s->nsubbands - 1] + 1;
if (end_sb > nsubbands)
end_sb = nsubbands;
for (sb = start_sb; sb < end_sb; sb++) {
for (ch = ch1; ch <= ch2; ch++) {
uint8_t *g2_scf = s->grid_2_scf[ch][sb];
if ((ch != ch1 && ff_dca_grid_2_to_scf[sb] >= s->min_mono_subband) != flag) {
if (!flag)
memcpy(g2_scf, s->grid_2_scf[ch1][sb], 64);
continue;
}
// Scale factors in groups of 8
for (i = 0; i < 8; i++, g2_scf += 8) {
if (get_bits_left(&s->gb) < 1) {
memset(g2_scf, 0, 64 - i * 8);
break;
}
// Bit indicating if whole group has zero values
if (get_bits1(&s->gb)) {
for (j = 0; j < 8; j++) {
if (ensure_bits(&s->gb, 20))
break;
g2_scf[j] = parse_vlc(&s->gb, &ff_dca_vlc_grid_2, 2);
}
} else {
memset(g2_scf, 0, 8);
}
}
}
}
return 0;
}
static int parse_ts1_chunk(DCALbrDecoder *s, LBRChunk *chunk, int ch1, int ch2)
{
int ret;
if (!chunk->len)
return 0;
if ((ret = init_get_bits8(&s->gb, chunk->data, chunk->len)) < 0)
return ret;
if ((ret = parse_lpc(s, ch1, ch2, 2, 3)) < 0)
return ret;
if ((ret = parse_ts(s, ch1, ch2, 2, 4, 0)) < 0)
return ret;
if ((ret = parse_grid_2(s, ch1, ch2, 0, 1, 0)) < 0)
return ret;
if ((ret = parse_ts(s, ch1, ch2, 4, 6, 0)) < 0)
return ret;
return 0;
}
static int parse_ts2_chunk(DCALbrDecoder *s, LBRChunk *chunk, int ch1, int ch2)
{
int ret;
if (!chunk->len)
return 0;
if ((ret = init_get_bits8(&s->gb, chunk->data, chunk->len)) < 0)
return ret;
if ((ret = parse_grid_2(s, ch1, ch2, 1, 3, 0)) < 0)
return ret;
if ((ret = parse_ts(s, ch1, ch2, 6, s->max_mono_subband, 0)) < 0)
return ret;
if (ch1 != ch2) {
if ((ret = parse_grid_1_sec_ch(s, ch2)) < 0)
return ret;
if ((ret = parse_grid_2(s, ch1, ch2, 0, 3, 1)) < 0)
return ret;
}
if ((ret = parse_ts(s, ch1, ch2, s->min_mono_subband, s->nsubbands, 1)) < 0)
return ret;
return 0;
}
static int init_sample_rate(DCALbrDecoder *s)
{
double scale = (-1.0 / (1 << 17)) * sqrt(1 << (2 - s->limited_range));
int i, br_per_ch = s->bit_rate_scaled / s->nchannels_total;
int ret;
ff_mdct_end(&s->imdct);
ret = ff_mdct_init(&s->imdct, s->freq_range + 6, 1, scale);
if (ret < 0)
return ret;
for (i = 0; i < 32 << s->freq_range; i++)
s->window[i] = ff_dca_long_window[i << (2 - s->freq_range)];
if (br_per_ch < 14000)
scale = 0.85;
else if (br_per_ch < 32000)
scale = (br_per_ch - 14000) * (1.0 / 120000) + 0.85;
else
scale = 1.0;
scale *= 1.0 / INT_MAX;
for (i = 0; i < s->nsubbands; i++) {
if (i < 2)
s->sb_scf[i] = 0; // The first two subbands are always zero
else if (i < 5)
s->sb_scf[i] = (i - 1) * 0.25 * 0.785 * scale;
else
s->sb_scf[i] = 0.785 * scale;
}
s->lfe_scale = (16 << s->freq_range) * 0.0000078265894;
return 0;
}
static int alloc_sample_buffer(DCALbrDecoder *s)
{
// Reserve space for history and padding
int nchsamples = DCA_LBR_TIME_SAMPLES + DCA_LBR_TIME_HISTORY * 2;
int nsamples = nchsamples * s->nchannels * s->nsubbands;
int ch, sb;
float *ptr;
// Reallocate time sample buffer
av_fast_mallocz(&s->ts_buffer, &s->ts_size, nsamples * sizeof(float));
if (!s->ts_buffer)
return AVERROR(ENOMEM);
ptr = s->ts_buffer + DCA_LBR_TIME_HISTORY;
for (ch = 0; ch < s->nchannels; ch++) {
for (sb = 0; sb < s->nsubbands; sb++) {
s->time_samples[ch][sb] = ptr;
ptr += nchsamples;
}
}
return 0;
}
static int parse_decoder_init(DCALbrDecoder *s, GetByteContext *gb)
{
int old_rate = s->sample_rate;
int old_band_limit = s->band_limit;
int old_nchannels = s->nchannels;
int version, bit_rate_hi;
unsigned int sr_code;
// Sample rate of LBR audio
sr_code = bytestream2_get_byte(gb);
if (sr_code >= FF_ARRAY_ELEMS(ff_dca_sampling_freqs)) {
av_log(s->avctx, AV_LOG_ERROR, "Invalid LBR sample rate\n");
return AVERROR_INVALIDDATA;
}
s->sample_rate = ff_dca_sampling_freqs[sr_code];
if (s->sample_rate > 48000) {
avpriv_report_missing_feature(s->avctx, "%d Hz LBR sample rate", s->sample_rate);
return AVERROR_PATCHWELCOME;
}
// LBR speaker mask
s->ch_mask = bytestream2_get_le16(gb);
if (!(s->ch_mask & 0x7)) {
avpriv_report_missing_feature(s->avctx, "LBR channel mask %#x", s->ch_mask);
return AVERROR_PATCHWELCOME;
}
if ((s->ch_mask & 0xfff0) && !(s->warned & 1)) {
avpriv_report_missing_feature(s->avctx, "LBR channel mask %#x", s->ch_mask);
s->warned |= 1;
}
// LBR bitstream version
version = bytestream2_get_le16(gb);
if ((version & 0xff00) != 0x0800) {
avpriv_report_missing_feature(s->avctx, "LBR stream version %#x", version);
return AVERROR_PATCHWELCOME;
}
// Flags for LBR decoder initialization
s->flags = bytestream2_get_byte(gb);
if (s->flags & LBR_FLAG_DMIX_MULTI_CH) {
avpriv_report_missing_feature(s->avctx, "LBR multi-channel downmix");
return AVERROR_PATCHWELCOME;
}
if ((s->flags & LBR_FLAG_LFE_PRESENT) && s->sample_rate != 48000) {
if (!(s->warned & 2)) {
avpriv_report_missing_feature(s->avctx, "%d Hz LFE interpolation", s->sample_rate);
s->warned |= 2;
}
s->flags &= ~LBR_FLAG_LFE_PRESENT;
}
// Most significant bit rate nibbles
bit_rate_hi = bytestream2_get_byte(gb);
// Least significant original bit rate word
s->bit_rate_orig = bytestream2_get_le16(gb) | ((bit_rate_hi & 0x0F) << 16);
// Least significant scaled bit rate word
s->bit_rate_scaled = bytestream2_get_le16(gb) | ((bit_rate_hi & 0xF0) << 12);
// Setup number of fullband channels
s->nchannels_total = ff_dca_count_chs_for_mask(s->ch_mask & ~DCA_SPEAKER_PAIR_LFE1);
s->nchannels = FFMIN(s->nchannels_total, DCA_LBR_CHANNELS);
// Setup band limit
switch (s->flags & LBR_FLAG_BAND_LIMIT_MASK) {
case LBR_FLAG_BAND_LIMIT_NONE:
s->band_limit = 0;
break;
case LBR_FLAG_BAND_LIMIT_1_2:
s->band_limit = 1;
break;
case LBR_FLAG_BAND_LIMIT_1_4:
s->band_limit = 2;
break;
default:
avpriv_report_missing_feature(s->avctx, "LBR band limit %#x", s->flags & LBR_FLAG_BAND_LIMIT_MASK);
return AVERROR_PATCHWELCOME;
}
// Setup frequency range
s->freq_range = ff_dca_freq_ranges[sr_code];
// Setup resolution profile
if (s->bit_rate_orig >= 44000 * (s->nchannels_total + 2))
s->res_profile = 2;
else if (s->bit_rate_orig >= 25000 * (s->nchannels_total + 2))
s->res_profile = 1;
else
s->res_profile = 0;
// Setup limited sample rate, number of subbands, etc
s->limited_rate = s->sample_rate >> s->band_limit;
s->limited_range = s->freq_range - s->band_limit;
if (s->limited_range < 0) {
av_log(s->avctx, AV_LOG_ERROR, "Invalid LBR band limit for frequency range\n");
return AVERROR_INVALIDDATA;
}
s->nsubbands = 8 << s->limited_range;
s->g3_avg_only_start_sb = s->nsubbands * ff_dca_avg_g3_freqs[s->res_profile] / (s->limited_rate / 2);
if (s->g3_avg_only_start_sb > s->nsubbands)
s->g3_avg_only_start_sb = s->nsubbands;
s->min_mono_subband = s->nsubbands * 2000 / (s->limited_rate / 2);
if (s->min_mono_subband > s->nsubbands)
s->min_mono_subband = s->nsubbands;
s->max_mono_subband = s->nsubbands * 14000 / (s->limited_rate / 2);
if (s->max_mono_subband > s->nsubbands)
s->max_mono_subband = s->nsubbands;
// Handle change of sample rate
if ((old_rate != s->sample_rate || old_band_limit != s->band_limit) && init_sample_rate(s) < 0)
return AVERROR(ENOMEM);
// Setup stereo downmix
if (s->flags & LBR_FLAG_DMIX_STEREO) {
DCAContext *dca = s->avctx->priv_data;
if (s->nchannels_total < 3 || s->nchannels_total > DCA_LBR_CHANNELS_TOTAL - 2) {
av_log(s->avctx, AV_LOG_ERROR, "Invalid number of channels for LBR stereo downmix\n");
return AVERROR_INVALIDDATA;
}
// This decoder doesn't support ECS chunk
if (dca->request_channel_layout != DCA_SPEAKER_LAYOUT_STEREO && !(s->warned & 4)) {
avpriv_report_missing_feature(s->avctx, "Embedded LBR stereo downmix");
s->warned |= 4;
}
// Account for extra downmixed channel pair
s->nchannels_total += 2;
s->nchannels = 2;
s->ch_mask = DCA_SPEAKER_PAIR_LR;
s->flags &= ~LBR_FLAG_LFE_PRESENT;
}
// Handle change of sample rate or number of channels
if (old_rate != s->sample_rate
|| old_band_limit != s->band_limit
|| old_nchannels != s->nchannels) {
if (alloc_sample_buffer(s) < 0)
return AVERROR(ENOMEM);
ff_dca_lbr_flush(s);
}
return 0;
}
int ff_dca_lbr_parse(DCALbrDecoder *s, uint8_t *data, DCAExssAsset *asset)
{
struct {
LBRChunk lfe;
LBRChunk tonal;
LBRChunk tonal_grp[5];
LBRChunk grid1[DCA_LBR_CHANNELS / 2];
LBRChunk hr_grid[DCA_LBR_CHANNELS / 2];
LBRChunk ts1[DCA_LBR_CHANNELS / 2];
LBRChunk ts2[DCA_LBR_CHANNELS / 2];
} chunk = { {0} };
GetByteContext gb;
int i, ch, sb, sf, ret, group, chunk_id, chunk_len;
bytestream2_init(&gb, data + asset->lbr_offset, asset->lbr_size);
// LBR sync word
if (bytestream2_get_be32(&gb) != DCA_SYNCWORD_LBR) {
av_log(s->avctx, AV_LOG_ERROR, "Invalid LBR sync word\n");
return AVERROR_INVALIDDATA;
}
// LBR header type
switch (bytestream2_get_byte(&gb)) {
case DCA_LBR_HEADER_SYNC_ONLY:
if (!s->sample_rate) {
av_log(s->avctx, AV_LOG_ERROR, "LBR decoder not initialized\n");
return AVERROR_INVALIDDATA;
}
break;
case DCA_LBR_HEADER_DECODER_INIT:
if ((ret = parse_decoder_init(s, &gb)) < 0) {
s->sample_rate = 0;
return ret;
}
break;
default:
av_log(s->avctx, AV_LOG_ERROR, "Invalid LBR header type\n");
return AVERROR_INVALIDDATA;
}
// LBR frame chunk header
chunk_id = bytestream2_get_byte(&gb);
chunk_len = (chunk_id & 0x80) ? bytestream2_get_be16(&gb) : bytestream2_get_byte(&gb);
if (chunk_len > bytestream2_get_bytes_left(&gb)) {
chunk_len = bytestream2_get_bytes_left(&gb);
av_log(s->avctx, AV_LOG_WARNING, "LBR frame chunk was truncated\n");
if (s->avctx->err_recognition & AV_EF_EXPLODE)
return AVERROR_INVALIDDATA;
}
bytestream2_init(&gb, gb.buffer, chunk_len);
switch (chunk_id & 0x7f) {
case LBR_CHUNK_FRAME:
if (s->avctx->err_recognition & (AV_EF_CRCCHECK | AV_EF_CAREFUL)) {
int checksum = bytestream2_get_be16(&gb);
uint16_t res = chunk_id;
res += (chunk_len >> 8) & 0xff;
res += chunk_len & 0xff;
for (i = 0; i < chunk_len - 2; i++)
res += gb.buffer[i];
if (checksum != res) {
av_log(s->avctx, AV_LOG_WARNING, "Invalid LBR checksum\n");
if (s->avctx->err_recognition & AV_EF_EXPLODE)
return AVERROR_INVALIDDATA;
}
} else {
bytestream2_skip(&gb, 2);
}
break;
case LBR_CHUNK_FRAME_NO_CSUM:
break;
default:
av_log(s->avctx, AV_LOG_ERROR, "Invalid LBR frame chunk ID\n");
return AVERROR_INVALIDDATA;
}
// Clear current frame
memset(s->quant_levels, 0, sizeof(s->quant_levels));
memset(s->sb_indices, 0xff, sizeof(s->sb_indices));
memset(s->sec_ch_sbms, 0, sizeof(s->sec_ch_sbms));
memset(s->sec_ch_lrms, 0, sizeof(s->sec_ch_lrms));
memset(s->ch_pres, 0, sizeof(s->ch_pres));
memset(s->grid_1_scf, 0, sizeof(s->grid_1_scf));
memset(s->grid_2_scf, 0, sizeof(s->grid_2_scf));
memset(s->grid_3_avg, 0, sizeof(s->grid_3_avg));
memset(s->grid_3_scf, 0, sizeof(s->grid_3_scf));
memset(s->grid_3_pres, 0, sizeof(s->grid_3_pres));
memset(s->tonal_scf, 0, sizeof(s->tonal_scf));
memset(s->lfe_data, 0, sizeof(s->lfe_data));
s->part_stereo_pres = 0;
s->framenum = (s->framenum + 1) & 31;
for (ch = 0; ch < s->nchannels; ch++) {
for (sb = 0; sb < s->nsubbands / 4; sb++) {
s->part_stereo[ch][sb][0] = s->part_stereo[ch][sb][4];
s->part_stereo[ch][sb][4] = 16;
}
}
memset(s->lpc_coeff[s->framenum & 1], 0, sizeof(s->lpc_coeff[0]));
for (group = 0; group < 5; group++) {
for (sf = 0; sf < 1 << group; sf++) {
int sf_idx = ((s->framenum << group) + sf) & 31;
s->tonal_bounds[group][sf_idx][0] =
s->tonal_bounds[group][sf_idx][1] = s->ntones;
}
}
// Parse chunk headers
while (bytestream2_get_bytes_left(&gb) > 0) {
chunk_id = bytestream2_get_byte(&gb);
chunk_len = (chunk_id & 0x80) ? bytestream2_get_be16(&gb) : bytestream2_get_byte(&gb);
chunk_id &= 0x7f;
if (chunk_len > bytestream2_get_bytes_left(&gb)) {
chunk_len = bytestream2_get_bytes_left(&gb);
av_log(s->avctx, AV_LOG_WARNING, "LBR chunk %#x was truncated\n", chunk_id);
if (s->avctx->err_recognition & AV_EF_EXPLODE)
return AVERROR_INVALIDDATA;
}
switch (chunk_id) {
case LBR_CHUNK_LFE:
chunk.lfe.len = chunk_len;
chunk.lfe.data = gb.buffer;
break;
case LBR_CHUNK_SCF:
case LBR_CHUNK_TONAL:
case LBR_CHUNK_TONAL_SCF:
chunk.tonal.id = chunk_id;
chunk.tonal.len = chunk_len;
chunk.tonal.data = gb.buffer;
break;
case LBR_CHUNK_TONAL_GRP_1:
case LBR_CHUNK_TONAL_GRP_2:
case LBR_CHUNK_TONAL_GRP_3:
case LBR_CHUNK_TONAL_GRP_4:
case LBR_CHUNK_TONAL_GRP_5:
i = LBR_CHUNK_TONAL_GRP_5 - chunk_id;
chunk.tonal_grp[i].id = i;
chunk.tonal_grp[i].len = chunk_len;
chunk.tonal_grp[i].data = gb.buffer;
break;
case LBR_CHUNK_TONAL_SCF_GRP_1:
case LBR_CHUNK_TONAL_SCF_GRP_2:
case LBR_CHUNK_TONAL_SCF_GRP_3:
case LBR_CHUNK_TONAL_SCF_GRP_4:
case LBR_CHUNK_TONAL_SCF_GRP_5:
i = LBR_CHUNK_TONAL_SCF_GRP_5 - chunk_id;
chunk.tonal_grp[i].id = i;
chunk.tonal_grp[i].len = chunk_len;
chunk.tonal_grp[i].data = gb.buffer;
break;
case LBR_CHUNK_RES_GRID_LR:
case LBR_CHUNK_RES_GRID_LR + 1:
case LBR_CHUNK_RES_GRID_LR + 2:
i = chunk_id - LBR_CHUNK_RES_GRID_LR;
chunk.grid1[i].len = chunk_len;
chunk.grid1[i].data = gb.buffer;
break;
case LBR_CHUNK_RES_GRID_HR:
case LBR_CHUNK_RES_GRID_HR + 1:
case LBR_CHUNK_RES_GRID_HR + 2:
i = chunk_id - LBR_CHUNK_RES_GRID_HR;
chunk.hr_grid[i].len = chunk_len;
chunk.hr_grid[i].data = gb.buffer;
break;
case LBR_CHUNK_RES_TS_1:
case LBR_CHUNK_RES_TS_1 + 1:
case LBR_CHUNK_RES_TS_1 + 2:
i = chunk_id - LBR_CHUNK_RES_TS_1;
chunk.ts1[i].len = chunk_len;
chunk.ts1[i].data = gb.buffer;
break;
case LBR_CHUNK_RES_TS_2:
case LBR_CHUNK_RES_TS_2 + 1:
case LBR_CHUNK_RES_TS_2 + 2:
i = chunk_id - LBR_CHUNK_RES_TS_2;
chunk.ts2[i].len = chunk_len;
chunk.ts2[i].data = gb.buffer;
break;
}
bytestream2_skip(&gb, chunk_len);
}
// Parse the chunks
ret = parse_lfe_chunk(s, &chunk.lfe);
ret |= parse_tonal_chunk(s, &chunk.tonal);
for (i = 0; i < 5; i++)
ret |= parse_tonal_group(s, &chunk.tonal_grp[i]);
for (i = 0; i < (s->nchannels + 1) / 2; i++) {
int ch1 = i * 2;
int ch2 = FFMIN(ch1 + 1, s->nchannels - 1);
if (parse_grid_1_chunk (s, &chunk.grid1 [i], ch1, ch2) < 0 ||
parse_high_res_grid(s, &chunk.hr_grid[i], ch1, ch2) < 0) {
ret = -1;
continue;
}
// TS chunks depend on both grids. TS_2 depends on TS_1.
if (!chunk.grid1[i].len || !chunk.hr_grid[i].len || !chunk.ts1[i].len)
continue;
if (parse_ts1_chunk(s, &chunk.ts1[i], ch1, ch2) < 0 ||
parse_ts2_chunk(s, &chunk.ts2[i], ch1, ch2) < 0) {
ret = -1;
continue;
}
}
if (ret < 0 && (s->avctx->err_recognition & AV_EF_EXPLODE))
return AVERROR_INVALIDDATA;
return 0;
}
/**
* Reconstruct high-frequency resolution grid from first and third grids
*/
static void decode_grid(DCALbrDecoder *s, int ch1, int ch2)
{
int i, ch, sb;
for (ch = ch1; ch <= ch2; ch++) {
for (sb = 0; sb < s->nsubbands; sb++) {
int g1_sb = ff_dca_scf_to_grid_1[sb];
uint8_t *g1_scf_a = s->grid_1_scf[ch][g1_sb ];
uint8_t *g1_scf_b = s->grid_1_scf[ch][g1_sb + 1];
int w1 = ff_dca_grid_1_weights[g1_sb ][sb];
int w2 = ff_dca_grid_1_weights[g1_sb + 1][sb];
uint8_t *hr_scf = s->high_res_scf[ch][sb];
if (sb < 4) {
for (i = 0; i < 8; i++) {
int scf = w1 * g1_scf_a[i] + w2 * g1_scf_b[i];
hr_scf[i] = scf >> 7;
}
} else {
int8_t *g3_scf = s->grid_3_scf[ch][sb - 4];
int g3_avg = s->grid_3_avg[ch][sb - 4];
for (i = 0; i < 8; i++) {
int scf = w1 * g1_scf_a[i] + w2 * g1_scf_b[i];
hr_scf[i] = (scf >> 7) - g3_avg - g3_scf[i];
}
}
}
}
}
/**
* Fill unallocated subbands with randomness
*/
static void random_ts(DCALbrDecoder *s, int ch1, int ch2)
{
int i, j, k, ch, sb;
for (ch = ch1; ch <= ch2; ch++) {
for (sb = 0; sb < s->nsubbands; sb++) {
float *samples = s->time_samples[ch][sb];
if (s->ch_pres[ch] & (1U << sb))
continue; // Skip allocated subband
if (sb < 2) {
// The first two subbands are always zero
memset(samples, 0, DCA_LBR_TIME_SAMPLES * sizeof(float));
} else if (sb < 10) {
for (i = 0; i < DCA_LBR_TIME_SAMPLES; i++)
samples[i] = lbr_rand(s, sb);
} else {
for (i = 0; i < DCA_LBR_TIME_SAMPLES / 8; i++, samples += 8) {
float accum[8] = { 0 };
// Modulate by subbands 2-5 in blocks of 8
for (k = 2; k < 6; k++) {
float *other = &s->time_samples[ch][k][i * 8];
for (j = 0; j < 8; j++)
accum[j] += fabs(other[j]);
}
for (j = 0; j < 8; j++)
samples[j] = (accum[j] * 0.25f + 0.5f) * lbr_rand(s, sb);
}
}
}
}
}
static void predict(float *samples, const float *coeff, int nsamples)
{
int i, j;
for (i = 0; i < nsamples; i++) {
float res = 0;
for (j = 0; j < 8; j++)
res += coeff[j] * samples[i - j - 1];
samples[i] -= res;
}
}
static void synth_lpc(DCALbrDecoder *s, int ch1, int ch2, int sb)
{
int f = s->framenum & 1;
int ch;
for (ch = ch1; ch <= ch2; ch++) {
float *samples = s->time_samples[ch][sb];
if (!(s->ch_pres[ch] & (1U << sb)))
continue;
if (sb < 2) {
predict(samples, s->lpc_coeff[f^1][ch][sb][1], 16);
predict(samples + 16, s->lpc_coeff[f ][ch][sb][0], 64);
predict(samples + 80, s->lpc_coeff[f ][ch][sb][1], 48);
} else {
predict(samples, s->lpc_coeff[f^1][ch][sb][0], 16);
predict(samples + 16, s->lpc_coeff[f ][ch][sb][0], 112);
}
}
}
static void filter_ts(DCALbrDecoder *s, int ch1, int ch2)
{
int i, j, sb, ch;
for (sb = 0; sb < s->nsubbands; sb++) {
// Scale factors
for (ch = ch1; ch <= ch2; ch++) {
float *samples = s->time_samples[ch][sb];
uint8_t *hr_scf = s->high_res_scf[ch][sb];
if (sb < 4) {
for (i = 0; i < DCA_LBR_TIME_SAMPLES / 16; i++, samples += 16) {
unsigned int scf = hr_scf[i];
if (scf > AMP_MAX)
scf = AMP_MAX;
for (j = 0; j < 16; j++)
samples[j] *= ff_dca_quant_amp[scf];
}
} else {
uint8_t *g2_scf = s->grid_2_scf[ch][ff_dca_scf_to_grid_2[sb]];
for (i = 0; i < DCA_LBR_TIME_SAMPLES / 2; i++, samples += 2) {
unsigned int scf = hr_scf[i / 8] - g2_scf[i];
if (scf > AMP_MAX)
scf = AMP_MAX;
samples[0] *= ff_dca_quant_amp[scf];
samples[1] *= ff_dca_quant_amp[scf];
}
}
}
// Mid-side stereo
if (ch1 != ch2) {
float *samples_l = s->time_samples[ch1][sb];
float *samples_r = s->time_samples[ch2][sb];
int ch2_pres = s->ch_pres[ch2] & (1U << sb);
for (i = 0; i < DCA_LBR_TIME_SAMPLES / 16; i++) {
int sbms = (s->sec_ch_sbms[ch1 / 2][sb] >> i) & 1;
int lrms = (s->sec_ch_lrms[ch1 / 2][sb] >> i) & 1;
if (sb >= s->min_mono_subband) {
if (lrms && ch2_pres) {
if (sbms) {
for (j = 0; j < 16; j++) {
float tmp = samples_l[j];
samples_l[j] = samples_r[j];
samples_r[j] = -tmp;
}
} else {
for (j = 0; j < 16; j++) {
float tmp = samples_l[j];
samples_l[j] = samples_r[j];
samples_r[j] = tmp;
}
}
} else if (!ch2_pres) {
if (sbms && (s->part_stereo_pres & (1 << ch1))) {
for (j = 0; j < 16; j++)
samples_r[j] = -samples_l[j];
} else {
for (j = 0; j < 16; j++)
samples_r[j] = samples_l[j];
}
}
} else if (sbms && ch2_pres) {
for (j = 0; j < 16; j++) {
float tmp = samples_l[j];
samples_l[j] = (tmp + samples_r[j]) * 0.5f;
samples_r[j] = (tmp - samples_r[j]) * 0.5f;
}
}
samples_l += 16;
samples_r += 16;
}
}
// Inverse prediction
if (sb < 3)
synth_lpc(s, ch1, ch2, sb);
}
}
/**
* Modulate by interpolated partial stereo coefficients
*/
static void decode_part_stereo(DCALbrDecoder *s, int ch1, int ch2)
{
int i, ch, sb, sf;
for (ch = ch1; ch <= ch2; ch++) {
for (sb = s->min_mono_subband; sb < s->nsubbands; sb++) {
uint8_t *pt_st = s->part_stereo[ch][(sb - s->min_mono_subband) / 4];
float *samples = s->time_samples[ch][sb];
if (s->ch_pres[ch2] & (1U << sb))
continue;
for (sf = 1; sf <= 4; sf++, samples += 32) {
float prev = ff_dca_st_coeff[pt_st[sf - 1]];
float next = ff_dca_st_coeff[pt_st[sf ]];
for (i = 0; i < 32; i++)
samples[i] *= (32 - i) * prev + i * next;
}
}
}
}
/**
* Synthesise tones in the given group for the given tonal subframe
*/
static void synth_tones(DCALbrDecoder *s, int ch, float *values,
int group, int group_sf, int synth_idx)
{
int i, start, count;
if (synth_idx < 0)
return;
start = s->tonal_bounds[group][group_sf][0];
count = (s->tonal_bounds[group][group_sf][1] - start) & (DCA_LBR_TONES - 1);
for (i = 0; i < count; i++) {
DCALbrTone *t = &s->tones[(start + i) & (DCA_LBR_TONES - 1)];
if (t->amp[ch]) {
float amp = ff_dca_synth_env[synth_idx] * ff_dca_quant_amp[t->amp[ch]];
float c = amp * cos_tab[(t->phs[ch] ) & 255];
float s = amp * cos_tab[(t->phs[ch] + 64) & 255];
const float *cf = ff_dca_corr_cf[t->f_delt];
int x_freq = t->x_freq;
switch (x_freq) {
case 0:
goto p0;
case 1:
values[3] += cf[0] * -s;
values[2] += cf[1] * c;
values[1] += cf[2] * s;
values[0] += cf[3] * -c;
goto p1;
case 2:
values[2] += cf[0] * -s;
values[1] += cf[1] * c;
values[0] += cf[2] * s;
goto p2;
case 3:
values[1] += cf[0] * -s;
values[0] += cf[1] * c;
goto p3;
case 4:
values[0] += cf[0] * -s;
goto p4;
}
values[x_freq - 5] += cf[ 0] * -s;
p4: values[x_freq - 4] += cf[ 1] * c;
p3: values[x_freq - 3] += cf[ 2] * s;
p2: values[x_freq - 2] += cf[ 3] * -c;
p1: values[x_freq - 1] += cf[ 4] * -s;
p0: values[x_freq ] += cf[ 5] * c;
values[x_freq + 1] += cf[ 6] * s;
values[x_freq + 2] += cf[ 7] * -c;
values[x_freq + 3] += cf[ 8] * -s;
values[x_freq + 4] += cf[ 9] * c;
values[x_freq + 5] += cf[10] * s;
}
t->phs[ch] += t->ph_rot;
}
}
/**
* Synthesise all tones in all groups for the given residual subframe
*/
static void base_func_synth(DCALbrDecoder *s, int ch, float *values, int sf)
{
int group;
// Tonal vs residual shift is 22 subframes
for (group = 0; group < 5; group++) {
int group_sf = (s->framenum << group) + ((sf - 22) >> (5 - group));
int synth_idx = ((((sf - 22) & 31) << group) & 31) + (1 << group) - 1;
synth_tones(s, ch, values, group, (group_sf - 1) & 31, 30 - synth_idx);
synth_tones(s, ch, values, group, (group_sf ) & 31, synth_idx);
}
}
static void transform_channel(DCALbrDecoder *s, int ch, float *output)
{
LOCAL_ALIGNED_32(float, values, [DCA_LBR_SUBBANDS ], [4]);
LOCAL_ALIGNED_32(float, result, [DCA_LBR_SUBBANDS * 2], [4]);
int sf, sb, nsubbands = s->nsubbands, noutsubbands = 8 << s->freq_range;
// Clear inactive subbands
if (nsubbands < noutsubbands)
memset(values[nsubbands], 0, (noutsubbands - nsubbands) * sizeof(values[0]));
for (sf = 0; sf < DCA_LBR_TIME_SAMPLES / 4; sf++) {
// Hybrid filterbank
s->dcadsp->lbr_bank(values, s->time_samples[ch],
ff_dca_bank_coeff, sf * 4, nsubbands);
base_func_synth(s, ch, values[0], sf);
s->imdct.imdct_calc(&s->imdct, result[0], values[0]);
// Long window and overlap-add
s->fdsp->vector_fmul_add(output, result[0], s->window,
s->history[ch], noutsubbands * 4);
s->fdsp->vector_fmul_reverse(s->history[ch], result[noutsubbands],
s->window, noutsubbands * 4);
output += noutsubbands * 4;
}
// Update history for LPC and forward MDCT
for (sb = 0; sb < nsubbands; sb++) {
float *samples = s->time_samples[ch][sb] - DCA_LBR_TIME_HISTORY;
memcpy(samples, samples + DCA_LBR_TIME_SAMPLES, DCA_LBR_TIME_HISTORY * sizeof(float));
}
}
int ff_dca_lbr_filter_frame(DCALbrDecoder *s, AVFrame *frame)
{
AVCodecContext *avctx = s->avctx;
int i, ret, nchannels, ch_conf = (s->ch_mask & 0x7) - 1;
const int8_t *reorder;
avctx->channel_layout = channel_layouts[ch_conf];
avctx->channels = nchannels = channel_counts[ch_conf];
avctx->sample_rate = s->sample_rate;
avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
avctx->bits_per_raw_sample = 0;
avctx->profile = FF_PROFILE_DTS_EXPRESS;
avctx->bit_rate = s->bit_rate_scaled;
if (s->flags & LBR_FLAG_LFE_PRESENT) {
avctx->channel_layout |= AV_CH_LOW_FREQUENCY;
avctx->channels++;
reorder = channel_reorder_lfe[ch_conf];
} else {
reorder = channel_reorder_nolfe[ch_conf];
}
frame->nb_samples = 1024 << s->freq_range;
if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
return ret;
// Filter fullband channels
for (i = 0; i < (s->nchannels + 1) / 2; i++) {
int ch1 = i * 2;
int ch2 = FFMIN(ch1 + 1, s->nchannels - 1);
decode_grid(s, ch1, ch2);
random_ts(s, ch1, ch2);
filter_ts(s, ch1, ch2);
if (ch1 != ch2 && (s->part_stereo_pres & (1 << ch1)))
decode_part_stereo(s, ch1, ch2);
if (ch1 < nchannels)
transform_channel(s, ch1, (float *)frame->extended_data[reorder[ch1]]);
if (ch1 != ch2 && ch2 < nchannels)
transform_channel(s, ch2, (float *)frame->extended_data[reorder[ch2]]);
}
// Interpolate LFE channel
if (s->flags & LBR_FLAG_LFE_PRESENT) {
s->dcadsp->lfe_iir((float *)frame->extended_data[lfe_index[ch_conf]],
s->lfe_data, ff_dca_lfe_iir,
s->lfe_history, 16 << s->freq_range);
}
if ((ret = ff_side_data_update_matrix_encoding(frame, AV_MATRIX_ENCODING_NONE)) < 0)
return ret;
return 0;
}
av_cold void ff_dca_lbr_flush(DCALbrDecoder *s)
{
int ch, sb;
if (!s->sample_rate)
return;
// Clear history
memset(s->part_stereo, 16, sizeof(s->part_stereo));
memset(s->lpc_coeff, 0, sizeof(s->lpc_coeff));
memset(s->history, 0, sizeof(s->history));
memset(s->tonal_bounds, 0, sizeof(s->tonal_bounds));
memset(s->lfe_history, 0, sizeof(s->lfe_history));
s->framenum = 0;
s->ntones = 0;
for (ch = 0; ch < s->nchannels; ch++) {
for (sb = 0; sb < s->nsubbands; sb++) {
float *samples = s->time_samples[ch][sb] - DCA_LBR_TIME_HISTORY;
memset(samples, 0, DCA_LBR_TIME_HISTORY * sizeof(float));
}
}
}
av_cold int ff_dca_lbr_init(DCALbrDecoder *s)
{
init_tables();
if (!(s->fdsp = avpriv_float_dsp_alloc(0)))
return AVERROR(ENOMEM);
s->lbr_rand = 1;
return 0;
}
av_cold void ff_dca_lbr_close(DCALbrDecoder *s)
{
s->sample_rate = 0;
av_freep(&s->ts_buffer);
s->ts_size = 0;
av_freep(&s->fdsp);
ff_mdct_end(&s->imdct);
}
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