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
36e27c87e7
FFmpeg/libavcodec/dca_lbr.c
foo86 d1f558b362 avcodec/dca: require checked bitstream reader 
Remove half-working attempt at supporting unchecked bitstream reader by
always copying input data into intermediate buffer with large amount of
padding at the end.

Convert LBR decoder to checked bitstream reader. Convert
dcadec_decode_frame() to parse input data directly if possible.

Signed-off-by: James Almer <jamrial@gmail.com>
2016-05-31 11:45:48 -03:00

1820 lines
55 KiB
C
Raw Blame History

/*
* 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 LBRHeader {
LBR_HEADER_SYNC_ONLY = 1,
LBR_HEADER_DECODER_INIT = 2
};
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 -1;
}
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 -1;
}
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)
{
if (!(s->flags & LBR_FLAG_LFE_PRESENT))
return 0;
if (!chunk->len)
return 0;
if (init_get_bits8(&s->gb, chunk->data, chunk->len) < 0)
return -1;
// 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 -1;
}
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 -1;
}
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 -1;
}
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 - 5) {
av_log(s->avctx, AV_LOG_ERROR, "Invalid spectral line offset\n");
return -1;
}
// 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;
if (!chunk->len)
return 0;
if (init_get_bits8(&s->gb, chunk->data, chunk->len) < 0)
return -1;
// 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 -1;
}
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++)
if (parse_tonal(s, group) < 0)
return -1;
return 0;
}
static int parse_tonal_group(DCALbrDecoder *s, LBRChunk *chunk)
{
if (!chunk->len)
return 0;
if (init_get_bits8(&s->gb, chunk->data, chunk->len) < 0)
return -1;
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 -1;
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 -1;
}
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;
if (!chunk->len)
return 0;
if (init_get_bits8(&s->gb, chunk->data, chunk->len) < 0)
return -1;
// Scale factors
nsubbands = ff_dca_scf_to_grid_1[s->nsubbands - 1] + 1;
for (sb = 2; sb < nsubbands; sb++) {
if (parse_scale_factors(s, s->grid_1_scf[ch1][sb]) < 0)
return -1;
if (ch1 != ch2 && ff_dca_grid_1_to_scf[sb] < s->min_mono_subband
&& parse_scale_factors(s, s->grid_1_scf[ch2][sb]) < 0)
return -1;
}
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 -1;
}
// 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;
// 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
&& parse_scale_factors(s, s->grid_1_scf[ch2][sb]) < 0)
return -1;
}
// 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 -1;
// 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 -1;
// 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;
if (!chunk->len)
return 0;
if (init_get_bits8(&s->gb, chunk->data, chunk->len) < 0)
return -1;
// 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
if (parse_lpc(s, ch1, ch2, 0, 2) < 0)
return -1;
// Time-samples for the first two subbands of main channel
if (parse_ts(s, ch1, ch2, 0, 2, 0) < 0)
return -1;
// First two bands of the first grid
for (sb = 0; sb < 2; sb++)
for (ch = ch1; ch <= ch2; ch++)
if (parse_scale_factors(s, s->grid_1_scf[ch][sb]) < 0)
return -1;
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)
{
if (!chunk->len)
return 0;
if (init_get_bits8(&s->gb, chunk->data, chunk->len) < 0)
return -1;
if (parse_lpc(s, ch1, ch2, 2, 3) < 0)
return -1;
if (parse_ts(s, ch1, ch2, 2, 4, 0) < 0)
return -1;
if (parse_grid_2(s, ch1, ch2, 0, 1, 0) < 0)
return -1;
if (parse_ts(s, ch1, ch2, 4, 6, 0) < 0)
return -1;
return 0;
}
static int parse_ts2_chunk(DCALbrDecoder *s, LBRChunk *chunk, int ch1, int ch2)
{
if (!chunk->len)
return 0;
if (init_get_bits8(&s->gb, chunk->data, chunk->len) < 0)
return -1;
if (parse_grid_2(s, ch1, ch2, 1, 3, 0) < 0)
return -1;
if (parse_ts(s, ch1, ch2, 6, s->max_mono_subband, 0) < 0)
return -1;
if (ch1 != ch2) {
if (parse_grid_1_sec_ch(s, ch2) < 0)
return -1;
if (parse_grid_2(s, ch1, ch2, 0, 3, 1) < 0)
return -1;
}
if (parse_ts(s, ch1, ch2, s->min_mono_subband, s->nsubbands, 1) < 0)
return -1;
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;
ff_mdct_end(&s->imdct);
if (ff_mdct_init(&s->imdct, s->freq_range + 6, 1, scale) < 0)
return -1;
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 -1;
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 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 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 -1;
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);
}
Reference in New Issue View Git Blame Copy Permalink