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mirror of https://github.com/FFmpeg/FFmpeg.git synced 2024-11-21 10:55:51 +02:00
FFmpeg/libavcodec/aac/aacdec_lpd.c
Lynne eee5fa0808
aacdec: add a decoder for AAC USAC (xHE-AAC)
This commit adds a decoder for the frequency-domain part of USAC.

What works:
 - Mono
 - Stereo (no prediction)
 - Stereo (mid/side coding)
 - Stereo (complex prediction)

What's left:
 - SBR
 - Speech coding

Known issues:
 - Desync with certain sequences
 - Preroll crossover missing (shouldn't matter, bitrate adaptation only)
2024-06-02 18:34:45 +02:00

199 lines
5.2 KiB
C

/*
* Copyright (c) 2024 Lynne <dev@lynne.ee>
*
* 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 "aacdec_lpd.h"
#include "aacdec_usac.h"
#include "libavcodec/unary.h"
const uint8_t ff_aac_lpd_mode_tab[32][4] = {
{ 0, 0, 0, 0 },
{ 1, 0, 0, 0 },
{ 0, 1, 0, 0 },
{ 1, 1, 0, 0 },
{ 0, 0, 1, 0 },
{ 1, 0, 1, 0 },
{ 0, 1, 1, 0 },
{ 1, 1, 1, 0 },
{ 0, 0, 0, 1 },
{ 1, 0, 0, 1 },
{ 0, 1, 0, 1 },
{ 1, 1, 0, 1 },
{ 0, 0, 1, 1 },
{ 1, 0, 1, 1 },
{ 0, 1, 1, 1 },
{ 1, 1, 1, 1 },
{ 2, 2, 0, 0 },
{ 2, 2, 1, 0 },
{ 2, 2, 0, 1 },
{ 2, 2, 1, 1 },
{ 0, 0, 2, 2 },
{ 1, 0, 2, 2 },
{ 0, 1, 2, 2 },
{ 1, 1, 2, 2 },
{ 2, 2, 2, 2 },
{ 3, 3, 3, 3 },
/* Larger values are reserved, but permit them for resilience */
{ 0, 0, 0, 0 },
{ 0, 0, 0, 0 },
{ 0, 0, 0, 0 },
{ 0, 0, 0, 0 },
{ 0, 0, 0, 0 },
{ 0, 0, 0, 0 },
};
static void parse_qn(GetBitContext *gb, int *qn, int nk_mode, int no_qn)
{
if (nk_mode == 1) {
for (int k = 0; k < no_qn; k++) {
qn[k] = get_unary(gb, 0, INT32_MAX); // TODO: find proper ranges
if (qn[k])
qn[k]++;
}
return;
}
for (int k = 0; k < no_qn; k++)
qn[k] = get_bits(gb, 2) + 2;
if (nk_mode == 2) {
for (int k = 0; k < no_qn; k++) {
if (qn[k] > 4) {
qn[k] = get_unary(gb, 0, INT32_MAX);;
if (qn[k])
qn[k] += 4;
}
}
return;
}
for (int k = 0; k < no_qn; k++) {
if (qn[k] > 4) {
int qn_ext = get_unary(gb, 0, INT32_MAX);;
switch (qn_ext) {
case 0: qn[k] = 5; break;
case 1: qn[k] = 6; break;
case 2: qn[k] = 0; break;
default: qn[k] = qn_ext + 4; break;
}
}
}
}
static int parse_codebook_idx(GetBitContext *gb, uint32_t *kv,
int nk_mode, int no_qn)
{
int idx, n, nk;
int qn[2];
parse_qn(gb, qn, nk_mode, no_qn);
for (int k = 0; k < no_qn; k++) {
if (qn[k] > 4) {
nk = (qn[k] - 3) / 2;
n = qn[k] - nk*2;
} else {
nk = 0;
n = qn[k];
}
}
idx = get_bits(gb, 4*n);
if (nk > 0)
for (int i = 0; i < 8; i++)
kv[i] = get_bits(gb, nk);
return 0;
}
int ff_aac_parse_fac_data(AACUsacElemData *ce, GetBitContext *gb,
int use_gain, int len)
{
int ret;
if (use_gain)
ce->fac.gain = get_bits(gb, 7);
for (int i = 0; i < len/8; i++) {
ret = parse_codebook_idx(gb, ce->fac.kv[i], 1, 1);
if (ret < 0)
return ret;
}
return 0;
}
int ff_aac_ldp_parse_channel_stream(AACDecContext *ac, AACUSACConfig *usac,
AACUsacElemData *ce, GetBitContext *gb)
{
int k;
const uint8_t *mod;
int first_ldp_flag;
int first_tcx_flag;
ce->ldp.acelp_core_mode = get_bits(gb, 3);
ce->ldp.lpd_mode = get_bits(gb, 5);
ce->ldp.bpf_control_info = get_bits1(gb);
ce->ldp.core_mode_last = get_bits1(gb);
ce->ldp.fac_data_present = get_bits1(gb);
mod = ff_aac_lpd_mode_tab[ce->ldp.lpd_mode];
first_ldp_flag = !ce->ldp.core_mode_last;
first_tcx_flag = 1;
if (first_ldp_flag)
ce->ldp.last_lpd_mode = -1; /* last_ldp_mode is a **STATEFUL** value */
k = 0;
while (k < 0) {
if (!k) {
if (ce->ldp.core_mode_last && ce->ldp.fac_data_present)
ff_aac_parse_fac_data(ce, gb, 0, usac->core_frame_len/8);
} else {
if (!ce->ldp.last_lpd_mode && mod[k] > 0 ||
ce->ldp.last_lpd_mode && !mod[k])
ff_aac_parse_fac_data(ce, gb, 0, usac->core_frame_len/8);
}
if (!mod[k]) {
// parse_acelp_coding();
ce->ldp.last_lpd_mode = 0;
k++;
} else {
// parse_tcx_coding();
ce->ldp.last_lpd_mode = mod[k];
k += (1 << (mod[k] - 1));
first_tcx_flag = 0;
}
}
// parse_lpc_data(first_lpd_flag);
if (!ce->ldp.core_mode_last && ce->ldp.fac_data_present) {
uint16_t len_8 = usac->core_frame_len / 8;
uint16_t len_16 = usac->core_frame_len / 16;
uint16_t fac_len = get_bits1(gb) /* short_fac_flag */ ? len_8 : len_16;
int ret = ff_aac_parse_fac_data(ce, gb, 1, fac_len);
if (ret < 0)
return ret;
}
return 0;
}