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ffmpeg-bb/libavcodec/aac/aacdec_usac.c
Lynne 3b8fe34a30
aacdec_usac: use RefStruct to track unfinished extension buffers 
Extensions in AAC USAC can be stored across multiple frames (mainly to keep CBR compliance).
This means that we need to reallocate a buffer when new data is received, accumulate the bitstream data,
and so on until the end of extension flag is signalled and the extension can be decoded.

This is made more complicated by the way in which the AAC channel layout switching is performed.
After decades of evolution, our AAC decoder evolved to double-buffer its entire configuration.
All changes are buffered, verified, and applied, on a per-frame basis if required, in often
random order.

Since we allocate the extension data on heap, this means that if configuration is applied,
in order to avoid double-freeing, we have to keep track of what we've allocated.

It should be noted that extensions which are spread in multiple frames are generally rare,
so an optimization to introduce av_refstruct_realloc() wouldn't generally be useful across the codebase.
Therefore, a copy is good enough for now.

Thanks to Michael Niedermayer for additional fixing.

Fixes: double free
Fixes: 393523547/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_AAC_LATM_fuzzer-6740617236905984

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
(cherry picked from commit c05fc27dd33b361eb0105157ab7d3a01c2ffa782)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2025-08-19 17:56:52 +02:00

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/*
* 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_usac.h"
#include "aacdec_tab.h"
#include "aacdec_lpd.h"
#include "aacdec_ac.h"
#include "libavcodec/aacsbr.h"
#include "libavcodec/aactab.h"
#include "libavcodec/mpeg4audio.h"
#include "libavcodec/unary.h"
#include "libavutil/mem.h"
#include "libavutil/refstruct.h"
/* Number of scalefactor bands per complex prediction band, equal to 2. */
#define SFB_PER_PRED_BAND 2
static inline uint32_t get_escaped_value(GetBitContext *gb, int nb1, int nb2, int nb3)
{
uint32_t val = get_bits(gb, nb1), val2;
if (val < ((1 << nb1) - 1))
return val;
val += val2 = get_bits(gb, nb2);
if (nb3 && (val2 == ((1 << nb2) - 1)))
val += get_bits(gb, nb3);
return val;
}
/* ISO/IEC 23003-3, Table 74 — bsOutputChannelPos */
static const enum AVChannel usac_ch_pos_to_av[64] = {
[0] = AV_CHAN_FRONT_LEFT,
[1] = AV_CHAN_FRONT_RIGHT,
[2] = AV_CHAN_FRONT_CENTER,
[3] = AV_CHAN_LOW_FREQUENCY,
[4] = AV_CHAN_SIDE_LEFT, // +110 degrees, Ls|LS|kAudioChannelLabel_LeftSurround
[5] = AV_CHAN_SIDE_RIGHT, // -110 degrees, Rs|RS|kAudioChannelLabel_RightSurround
[6] = AV_CHAN_FRONT_LEFT_OF_CENTER,
[7] = AV_CHAN_FRONT_RIGHT_OF_CENTER,
[8] = AV_CHAN_BACK_LEFT, // +135 degrees, Lsr|BL|kAudioChannelLabel_RearSurroundLeft
[9] = AV_CHAN_BACK_RIGHT, // -135 degrees, Rsr|BR|kAudioChannelLabel_RearSurroundRight
[10] = AV_CHAN_BACK_CENTER,
[11] = AV_CHAN_SURROUND_DIRECT_LEFT,
[12] = AV_CHAN_SURROUND_DIRECT_RIGHT,
[13] = AV_CHAN_SIDE_SURROUND_LEFT, // +90 degrees, Lss|SL|kAudioChannelLabel_LeftSideSurround
[14] = AV_CHAN_SIDE_SURROUND_RIGHT, // -90 degrees, Rss|SR|kAudioChannelLabel_RightSideSurround
[15] = AV_CHAN_WIDE_LEFT, // +60 degrees, Lw|FLw|kAudioChannelLabel_LeftWide
[16] = AV_CHAN_WIDE_RIGHT, // -60 degrees, Rw|FRw|kAudioChannelLabel_RightWide
[17] = AV_CHAN_TOP_FRONT_LEFT,
[18] = AV_CHAN_TOP_FRONT_RIGHT,
[19] = AV_CHAN_TOP_FRONT_CENTER,
[20] = AV_CHAN_TOP_BACK_LEFT,
[21] = AV_CHAN_TOP_BACK_RIGHT,
[22] = AV_CHAN_TOP_BACK_CENTER,
[23] = AV_CHAN_TOP_SIDE_LEFT,
[24] = AV_CHAN_TOP_SIDE_RIGHT,
[25] = AV_CHAN_TOP_CENTER,
[26] = AV_CHAN_LOW_FREQUENCY_2,
[27] = AV_CHAN_BOTTOM_FRONT_LEFT,
[28] = AV_CHAN_BOTTOM_FRONT_RIGHT,
[29] = AV_CHAN_BOTTOM_FRONT_CENTER,
[30] = AV_CHAN_TOP_SURROUND_LEFT, ///< +110 degrees, Lvs, TpLS
[31] = AV_CHAN_TOP_SURROUND_RIGHT, ///< -110 degrees, Rvs, TpRS
};
static int decode_loudness_info(AACDecContext *ac, AACUSACLoudnessInfo *info,
GetBitContext *gb)
{
info->drc_set_id = get_bits(gb, 6);
info->downmix_id = get_bits(gb, 7);
if ((info->sample_peak.present = get_bits1(gb))) /* samplePeakLevelPresent */
info->sample_peak.lvl = get_bits(gb, 12);
if ((info->true_peak.present = get_bits1(gb))) { /* truePeakLevelPresent */
info->true_peak.lvl = get_bits(gb, 12);
info->true_peak.measurement = get_bits(gb, 4);
info->true_peak.reliability = get_bits(gb, 2);
}
info->nb_measurements = get_bits(gb, 4);
for (int i = 0; i < info->nb_measurements; i++) {
info->measurements[i].method_def = get_bits(gb, 4);
info->measurements[i].method_val = get_unary(gb, 0, 8);
info->measurements[i].measurement = get_bits(gb, 4);
info->measurements[i].reliability = get_bits(gb, 2);
}
return 0;
}
static int decode_loudness_set(AACDecContext *ac, AACUSACConfig *usac,
GetBitContext *gb)
{
int ret;
usac->loudness.nb_album = get_bits(gb, 6); /* loudnessInfoAlbumCount */
usac->loudness.nb_info = get_bits(gb, 6); /* loudnessInfoCount */
for (int i = 0; i < usac->loudness.nb_album; i++) {
ret = decode_loudness_info(ac, &usac->loudness.album_info[i], gb);
if (ret < 0)
return ret;
}
for (int i = 0; i < usac->loudness.nb_info; i++) {
ret = decode_loudness_info(ac, &usac->loudness.info[i], gb);
if (ret < 0)
return ret;
}
if (get_bits1(gb)) { /* loudnessInfoSetExtPresent */
enum AACUSACLoudnessExt type;
while ((type = get_bits(gb, 4)) != UNIDRCLOUDEXT_TERM) {
uint8_t size_bits = get_bits(gb, 4) + 4;
uint8_t bit_size = get_bits(gb, size_bits) + 1;
switch (type) {
case UNIDRCLOUDEXT_EQ:
avpriv_report_missing_feature(ac->avctx, "loudnessInfoV1");
return AVERROR_PATCHWELCOME;
default:
for (int i = 0; i < bit_size; i++)
skip_bits1(gb);
}
}
}
return 0;
}
static int decode_usac_sbr_data(AACDecContext *ac,
AACUsacElemConfig *e, GetBitContext *gb)
{
uint8_t header_extra1;
uint8_t header_extra2;
e->sbr.harmonic_sbr = get_bits1(gb); /* harmonicSBR */
e->sbr.bs_intertes = get_bits1(gb); /* bs_interTes */
e->sbr.bs_pvc = get_bits1(gb); /* bs_pvc */
if (e->sbr.harmonic_sbr || e->sbr.bs_intertes || e->sbr.bs_pvc) {
avpriv_report_missing_feature(ac->avctx, "AAC USAC eSBR");
return AVERROR_PATCHWELCOME;
}
e->sbr.dflt.start_freq = get_bits(gb, 4); /* dflt_start_freq */
e->sbr.dflt.stop_freq = get_bits(gb, 4); /* dflt_stop_freq */
header_extra1 = get_bits1(gb); /* dflt_header_extra1 */
header_extra2 = get_bits1(gb); /* dflt_header_extra2 */
e->sbr.dflt.freq_scale = 2;
e->sbr.dflt.alter_scale = 1;
e->sbr.dflt.noise_bands = 2;
if (header_extra1) {
e->sbr.dflt.freq_scale = get_bits(gb, 2); /* dflt_freq_scale */
e->sbr.dflt.alter_scale = get_bits1(gb); /* dflt_alter_scale */
e->sbr.dflt.noise_bands = get_bits(gb, 2); /* dflt_noise_bands */
}
e->sbr.dflt.limiter_bands = 2;
e->sbr.dflt.limiter_gains = 2;
e->sbr.dflt.interpol_freq = 1;
e->sbr.dflt.smoothing_mode = 1;
if (header_extra2) {
e->sbr.dflt.limiter_bands = get_bits(gb, 2); /* dflt_limiter_bands */
e->sbr.dflt.limiter_gains = get_bits(gb, 2); /* dflt_limiter_gains */
e->sbr.dflt.interpol_freq = get_bits1(gb); /* dflt_interpol_freq */
e->sbr.dflt.smoothing_mode = get_bits1(gb); /* dflt_smoothing_mode */
}
return 0;
}
static void decode_usac_element_core(AACUsacElemConfig *e,
GetBitContext *gb,
int sbr_ratio)
{
e->tw_mdct = get_bits1(gb); /* tw_mdct */
e->noise_fill = get_bits1(gb);
e->sbr.ratio = sbr_ratio;
}
static int decode_usac_element_pair(AACDecContext *ac,
AACUsacElemConfig *e, GetBitContext *gb)
{
e->stereo_config_index = 0;
if (e->sbr.ratio) {
int ret = decode_usac_sbr_data(ac, e, gb);
if (ret < 0)
return ret;
e->stereo_config_index = get_bits(gb, 2);
}
if (e->stereo_config_index) {
e->mps.freq_res = get_bits(gb, 3); /* bsFreqRes */
e->mps.fixed_gain = get_bits(gb, 3); /* bsFixedGainDMX */
e->mps.temp_shape_config = get_bits(gb, 2); /* bsTempShapeConfig */
e->mps.decorr_config = get_bits(gb, 2); /* bsDecorrConfig */
e->mps.high_rate_mode = get_bits1(gb); /* bsHighRateMode */
e->mps.phase_coding = get_bits1(gb); /* bsPhaseCoding */
if (get_bits1(gb)) /* bsOttBandsPhasePresent */
e->mps.otts_bands_phase = get_bits(gb, 5); /* bsOttBandsPhase */
e->mps.residual_coding = e->stereo_config_index >= 2; /* bsResidualCoding */
if (e->mps.residual_coding) {
e->mps.residual_bands = get_bits(gb, 5); /* bsResidualBands */
e->mps.pseudo_lr = get_bits1(gb); /* bsPseudoLr */
}
if (e->mps.temp_shape_config == 2)
e->mps.env_quant_mode = get_bits1(gb); /* bsEnvQuantMode */
}
return 0;
}
static int decode_usac_extension(AACDecContext *ac, AACUsacElemConfig *e,
GetBitContext *gb)
{
int len = 0, ext_config_len;
e->ext.type = get_escaped_value(gb, 4, 8, 16); /* usacExtElementType */
ext_config_len = get_escaped_value(gb, 4, 8, 16); /* usacExtElementConfigLength */
if (get_bits1(gb)) /* usacExtElementDefaultLengthPresent */
len = get_escaped_value(gb, 8, 16, 0) + 1;
e->ext.default_len = len;
e->ext.payload_frag = get_bits1(gb); /* usacExtElementPayloadFrag */
av_log(ac->avctx, AV_LOG_DEBUG, "Extension present: type %i, len %i\n",
e->ext.type, ext_config_len);
switch (e->ext.type) {
#if 0 /* Skip unsupported values */
case ID_EXT_ELE_MPEGS:
break;
case ID_EXT_ELE_SAOC:
break;
case ID_EXT_ELE_UNI_DRC:
break;
#endif
case ID_EXT_ELE_FILL:
break; /* This is what the spec does */
case ID_EXT_ELE_AUDIOPREROLL:
/* No configuration needed - fallthrough (len should be 0) */
default:
skip_bits(gb, 8*ext_config_len);
e->ext.type = ID_EXT_ELE_FILL;
break;
};
return 0;
}
int ff_aac_usac_reset_state(AACDecContext *ac, OutputConfiguration *oc)
{
AACUSACConfig *usac = &oc->usac;
int elem_id[3 /* SCE, CPE, LFE */] = { 0, 0, 0 };
ChannelElement *che;
enum RawDataBlockType type;
int id, ch;
/* Initialize state */
for (int i = 0; i < usac->nb_elems; i++) {
AACUsacElemConfig *e = &usac->elems[i];
if (e->type == ID_USAC_EXT)
continue;
switch (e->type) {
case ID_USAC_SCE:
ch = 1;
type = TYPE_SCE;
id = elem_id[0]++;
break;
case ID_USAC_CPE:
ch = 2;
type = TYPE_CPE;
id = elem_id[1]++;
break;
case ID_USAC_LFE:
ch = 1;
type = TYPE_LFE;
id = elem_id[2]++;
break;
}
che = ff_aac_get_che(ac, type, id);
if (che) {
AACUsacStereo *us = &che->us;
memset(us, 0, sizeof(*us));
if (e->sbr.ratio)
ff_aac_sbr_config_usac(ac, che, e);
for (int j = 0; j < ch; j++) {
SingleChannelElement *sce = &che->ch[ch];
AACUsacElemData *ue = &sce->ue;
memset(ue, 0, sizeof(*ue));
if (!ch)
ue->noise.seed = 0x3039;
else
che->ch[1].ue.noise.seed = 0x10932;
}
}
}
return 0;
}
/* UsacConfig */
int ff_aac_usac_config_decode(AACDecContext *ac, AVCodecContext *avctx,
GetBitContext *gb, OutputConfiguration *oc,
int channel_config)
{
int ret;
uint8_t freq_idx;
uint8_t channel_config_idx;
int nb_channels = 0;
int ratio_mult, ratio_dec;
int samplerate;
int sbr_ratio;
MPEG4AudioConfig *m4ac = &oc->m4ac;
AACUSACConfig *usac = &oc->usac;
int elem_id[3 /* SCE, CPE, LFE */];
int map_pos_set = 0;
uint8_t layout_map[MAX_ELEM_ID*4][3] = { 0 };
if (!ac)
return AVERROR_PATCHWELCOME;
memset(usac, 0, sizeof(*usac));
freq_idx = get_bits(gb, 5); /* usacSamplingFrequencyIndex */
if (freq_idx == 0x1f) {
samplerate = get_bits(gb, 24); /* usacSamplingFrequency */
} else {
samplerate = ff_aac_usac_samplerate[freq_idx];
if (samplerate < 0)
return AVERROR(EINVAL);
}
usac->core_sbr_frame_len_idx = get_bits(gb, 3); /* coreSbrFrameLengthIndex */
m4ac->frame_length_short = usac->core_sbr_frame_len_idx == 0 ||
usac->core_sbr_frame_len_idx == 2;
usac->core_frame_len = (usac->core_sbr_frame_len_idx == 0 ||
usac->core_sbr_frame_len_idx == 2) ? 768 : 1024;
sbr_ratio = usac->core_sbr_frame_len_idx == 2 ? 2 :
usac->core_sbr_frame_len_idx == 3 ? 3 :
usac->core_sbr_frame_len_idx == 4 ? 1 :
0;
if (sbr_ratio == 2) {
ratio_mult = 8;
ratio_dec = 3;
} else if (sbr_ratio == 3) {
ratio_mult = 2;
ratio_dec = 1;
} else if (sbr_ratio == 4) {
ratio_mult = 4;
ratio_dec = 1;
} else {
ratio_mult = 1;
ratio_dec = 1;
}
avctx->sample_rate = samplerate;
m4ac->ext_sample_rate = samplerate;
m4ac->sample_rate = (samplerate * ratio_dec) / ratio_mult;
m4ac->sampling_index = ff_aac_sample_rate_idx(m4ac->sample_rate);
m4ac->sbr = sbr_ratio > 0;
channel_config_idx = get_bits(gb, 5); /* channelConfigurationIndex */
if (!channel_config_idx) {
/* UsacChannelConfig() */
nb_channels = get_escaped_value(gb, 5, 8, 16); /* numOutChannels */
if (nb_channels > 64)
return AVERROR(EINVAL);
av_channel_layout_uninit(&ac->oc[1].ch_layout);
ret = av_channel_layout_custom_init(&ac->oc[1].ch_layout, nb_channels);
if (ret < 0)
return ret;
for (int i = 0; i < nb_channels; i++) {
AVChannelCustom *cm = &ac->oc[1].ch_layout.u.map[i];
cm->id = usac_ch_pos_to_av[get_bits(gb, 5)]; /* bsOutputChannelPos */
}
ret = av_channel_layout_retype(&ac->oc[1].ch_layout,
AV_CHANNEL_ORDER_NATIVE,
AV_CHANNEL_LAYOUT_RETYPE_FLAG_CANONICAL);
if (ret < 0)
return ret;
ret = av_channel_layout_copy(&avctx->ch_layout, &ac->oc[1].ch_layout);
if (ret < 0)
return ret;
} else {
int nb_elements;
if ((ret = ff_aac_set_default_channel_config(ac, avctx, layout_map,
&nb_elements, channel_config_idx)))
return ret;
/* Fill in the number of expected channels */
for (int i = 0; i < nb_elements; i++)
nb_channels += layout_map[i][0] == TYPE_CPE ? 2 : 1;
map_pos_set = 1;
}
/* UsacDecoderConfig */
elem_id[0] = elem_id[1] = elem_id[2] = 0;
usac->nb_elems = get_escaped_value(gb, 4, 8, 16) + 1;
if (usac->nb_elems > 64) {
av_log(ac->avctx, AV_LOG_ERROR, "Too many elements: %i\n",
usac->nb_elems);
usac->nb_elems = 0;
return AVERROR(EINVAL);
}
for (int i = 0; i < usac->nb_elems; i++) {
int map_count = elem_id[0] + elem_id[1] + elem_id[2];
AACUsacElemConfig *e = &usac->elems[i];
memset(e, 0, sizeof(*e));
e->type = get_bits(gb, 2); /* usacElementType */
if (e->type != ID_USAC_EXT && (map_count + 1) > nb_channels) {
av_log(ac->avctx, AV_LOG_ERROR, "Too many channels for the channel "
"configuration\n");
usac->nb_elems = 0;
return AVERROR(EINVAL);
}
av_log(ac->avctx, AV_LOG_DEBUG, "Element present: idx %i, type %i\n",
i, e->type);
switch (e->type) {
case ID_USAC_SCE: /* SCE */
/* UsacCoreConfig */
decode_usac_element_core(e, gb, sbr_ratio);
if (e->sbr.ratio > 0) {
ret = decode_usac_sbr_data(ac, e, gb);
if (ret < 0)
return ret;
}
layout_map[map_count][0] = TYPE_SCE;
layout_map[map_count][1] = elem_id[0]++;
if (!map_pos_set)
layout_map[map_count][2] = AAC_CHANNEL_FRONT;
break;
case ID_USAC_CPE: /* UsacChannelPairElementConf */
/* UsacCoreConfig */
decode_usac_element_core(e, gb, sbr_ratio);
ret = decode_usac_element_pair(ac, e, gb);
if (ret < 0)
return ret;
layout_map[map_count][0] = TYPE_CPE;
layout_map[map_count][1] = elem_id[1]++;
if (!map_pos_set)
layout_map[map_count][2] = AAC_CHANNEL_FRONT;
break;
case ID_USAC_LFE: /* LFE */
/* LFE has no need for any configuration */
e->tw_mdct = 0;
e->noise_fill = 0;
layout_map[map_count][0] = TYPE_LFE;
layout_map[map_count][1] = elem_id[2]++;
if (!map_pos_set)
layout_map[map_count][2] = AAC_CHANNEL_LFE;
break;
case ID_USAC_EXT: /* EXT */
ret = decode_usac_extension(ac, e, gb);
if (ret < 0)
return ret;
break;
};
}
ret = ff_aac_output_configure(ac, layout_map, elem_id[0] + elem_id[1] + elem_id[2],
OC_GLOBAL_HDR, 0);
if (ret < 0) {
av_log(avctx, AV_LOG_ERROR, "Unable to parse channel config!\n");
usac->nb_elems = 0;
return ret;
}
if (get_bits1(gb)) { /* usacConfigExtensionPresent */
int invalid;
int nb_extensions = get_escaped_value(gb, 2, 4, 8) + 1; /* numConfigExtensions */
for (int i = 0; i < nb_extensions; i++) {
int type = get_escaped_value(gb, 4, 8, 16);
int len = get_escaped_value(gb, 4, 8, 16);
switch (type) {
case ID_CONFIG_EXT_LOUDNESS_INFO:
ret = decode_loudness_set(ac, usac, gb);
if (ret < 0)
return ret;
break;
case ID_CONFIG_EXT_STREAM_ID:
usac->stream_identifier = get_bits(gb, 16);
break;
case ID_CONFIG_EXT_FILL: /* fallthrough */
invalid = 0;
while (len--) {
if (get_bits(gb, 8) != 0xA5)
invalid++;
}
if (invalid)
av_log(avctx, AV_LOG_WARNING, "Invalid fill bytes: %i\n",
invalid);
break;
default:
while (len--)
skip_bits(gb, 8);
break;
}
}
}
ac->avctx->profile = AV_PROFILE_AAC_USAC;
ret = ff_aac_usac_reset_state(ac, oc);
if (ret < 0)
return ret;
return 0;
}
static int decode_usac_scale_factors(AACDecContext *ac,
SingleChannelElement *sce,
GetBitContext *gb, uint8_t global_gain)
{
IndividualChannelStream *ics = &sce->ics;
/* Decode all scalefactors. */
int offset_sf = global_gain;
for (int g = 0; g < ics->num_window_groups; g++) {
for (int sfb = 0; sfb < ics->max_sfb; sfb++) {
if (g || sfb)
offset_sf += get_vlc2(gb, ff_vlc_scalefactors, 7, 3) - SCALE_DIFF_ZERO;
if (offset_sf > 255U) {
av_log(ac->avctx, AV_LOG_ERROR,
"Scalefactor (%d) out of range.\n", offset_sf);
return AVERROR_INVALIDDATA;
}
sce->sfo[g*ics->max_sfb + sfb] = offset_sf - 100;
}
}
return 0;
}
/**
* Decode and dequantize arithmetically coded, uniformly quantized value
*
* @param coef array of dequantized, scaled spectral data
* @param sf array of scalefactors or intensity stereo positions
*
* @return Returns error status. 0 - OK, !0 - error
*/
static int decode_spectrum_ac(AACDecContext *s, float coef[1024],
GetBitContext *gb, AACArithState *state,
int reset, uint16_t len, uint16_t N)
{
AACArith ac;
int i, a, b;
uint32_t c;
int gb_count;
GetBitContext gb2;
c = ff_aac_ac_map_process(state, reset, N);
if (!len) {
ff_aac_ac_finish(state, 0, N);
return 0;
}
ff_aac_ac_init(&ac, gb);
/* Backup reader for rolling back by 14 bits at the end */
gb2 = *gb;
gb_count = get_bits_count(&gb2);
for (i = 0; i < len/2; i++) {
/* MSB */
int lvl, esc_nb, m;
c = ff_aac_ac_get_context(state, c, i, N);
for (lvl=esc_nb=0;;) {
uint32_t pki = ff_aac_ac_get_pk(c + (esc_nb << 17));
m = ff_aac_ac_decode(&ac, &gb2, ff_aac_ac_msb_cdfs[pki],
FF_ARRAY_ELEMS(ff_aac_ac_msb_cdfs[pki]));
if (m < FF_AAC_AC_ESCAPE)
break;
lvl++;
/* Cargo-culted value. */
if (lvl > 23)
return AVERROR(EINVAL);
if ((esc_nb = lvl) > 7)
esc_nb = 7;
}
b = m >> 2;
a = m - (b << 2);
/* ARITH_STOP detection */
if (!m) {
if (esc_nb)
break;
a = b = 0;
}
/* LSB */
for (int l = lvl; l > 0; l--) {
int lsbidx = !a ? 1 : (!b ? 0 : 2);
uint8_t r = ff_aac_ac_decode(&ac, &gb2, ff_aac_ac_lsb_cdfs[lsbidx],
FF_ARRAY_ELEMS(ff_aac_ac_lsb_cdfs[lsbidx]));
a = (a << 1) | (r & 1);
b = (b << 1) | ((r >> 1) & 1);
}
/* Dequantize coeffs here */
coef[2*i + 0] = a * cbrt(a);
coef[2*i + 1] = b * cbrt(b);
ff_aac_ac_update_context(state, i, a, b);
}
if (len > 1) {
/* "Rewind" bitstream back by 14 bits */
int gb_count2 = get_bits_count(&gb2);
skip_bits(gb, gb_count2 - gb_count - 14);
} else {
*gb = gb2;
}
ff_aac_ac_finish(state, i, N);
for (; i < N/2; i++) {
coef[2*i + 0] = 0;
coef[2*i + 1] = 0;
}
/* Signs */
for (i = 0; i < len; i++) {
if (coef[i]) {
if (!get_bits1(gb)) /* s */
coef[i] *= -1;
}
}
return 0;
}
static int decode_usac_stereo_cplx(AACDecContext *ac, AACUsacStereo *us,
ChannelElement *cpe, GetBitContext *gb,
int num_window_groups,
int prev_num_window_groups,
int indep_flag)
{
int delta_code_time;
IndividualChannelStream *ics = &cpe->ch[0].ics;
if (!get_bits1(gb)) { /* cplx_pred_all */
for (int g = 0; g < num_window_groups; g++) {
for (int sfb = 0; sfb < cpe->max_sfb_ste; sfb += SFB_PER_PRED_BAND) {
const uint8_t val = get_bits1(gb);
us->pred_used[g*cpe->max_sfb_ste + sfb] = val;
if ((sfb + 1) < cpe->max_sfb_ste)
us->pred_used[g*cpe->max_sfb_ste + sfb + 1] = val;
}
}
} else {
for (int g = 0; g < num_window_groups; g++)
for (int sfb = 0; sfb < cpe->max_sfb_ste; sfb++)
us->pred_used[g*cpe->max_sfb_ste + sfb] = 1;
}
us->pred_dir = get_bits1(gb);
us->complex_coef = get_bits1(gb);
us->use_prev_frame = 0;
if (us->complex_coef && !indep_flag)
us->use_prev_frame = get_bits1(gb);
delta_code_time = 0;
if (!indep_flag)
delta_code_time = get_bits1(gb);
/* TODO: shouldn't be needed */
for (int g = 0; g < num_window_groups; g++) {
for (int sfb = 0; sfb < cpe->max_sfb_ste; sfb += SFB_PER_PRED_BAND) {
float last_alpha_q_re = 0;
float last_alpha_q_im = 0;
if (delta_code_time) {
if (g) {
/* Transient, after the first group - use the current frame,
* previous window, alpha values. */
last_alpha_q_re = us->alpha_q_re[(g - 1)*cpe->max_sfb_ste + sfb];
last_alpha_q_im = us->alpha_q_im[(g - 1)*cpe->max_sfb_ste + sfb];
} else if (!g &&
(ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) &&
(ics->window_sequence[1] == EIGHT_SHORT_SEQUENCE)) {
/* The spec doesn't explicitly mention this, but it doesn't make
* any other sense otherwise! */
const int wg = prev_num_window_groups - 1;
last_alpha_q_re = us->prev_alpha_q_re[wg*cpe->max_sfb_ste + sfb];
last_alpha_q_im = us->prev_alpha_q_im[wg*cpe->max_sfb_ste + sfb];
} else {
last_alpha_q_re = us->prev_alpha_q_re[g*cpe->max_sfb_ste + sfb];
last_alpha_q_im = us->prev_alpha_q_im[g*cpe->max_sfb_ste + sfb];
}
} else {
if (sfb) {
last_alpha_q_re = us->alpha_q_re[g*cpe->max_sfb_ste + sfb - 1];
last_alpha_q_im = us->alpha_q_im[g*cpe->max_sfb_ste + sfb - 1];
}
}
if (us->pred_used[g*cpe->max_sfb_ste + sfb]) {
int val = -get_vlc2(gb, ff_vlc_scalefactors, 7, 3) + 60;
last_alpha_q_re += val * 0.1f;
if (us->complex_coef) {
val = -get_vlc2(gb, ff_vlc_scalefactors, 7, 3) + 60;
last_alpha_q_im += val * 0.1f;
}
us->alpha_q_re[g*cpe->max_sfb_ste + sfb] = last_alpha_q_re;
us->alpha_q_im[g*cpe->max_sfb_ste + sfb] = last_alpha_q_im;
} else {
us->alpha_q_re[g*cpe->max_sfb_ste + sfb] = 0;
us->alpha_q_im[g*cpe->max_sfb_ste + sfb] = 0;
}
if ((sfb + 1) < cpe->max_sfb_ste) {
us->alpha_q_re[g*cpe->max_sfb_ste + sfb + 1] =
us->alpha_q_re[g*cpe->max_sfb_ste + sfb];
us->alpha_q_im[g*cpe->max_sfb_ste + sfb + 1] =
us->alpha_q_im[g*cpe->max_sfb_ste + sfb];
}
}
}
return 0;
}
static int setup_sce(AACDecContext *ac, SingleChannelElement *sce,
AACUSACConfig *usac)
{
AACUsacElemData *ue = &sce->ue;
IndividualChannelStream *ics = &sce->ics;
const int sampling_index = ac->oc[1].m4ac.sampling_index;
/* Setup window parameters */
ics->prev_num_window_groups = FFMAX(ics->num_window_groups, 1);
if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
if (usac->core_frame_len == 768) {
ics->swb_offset = ff_swb_offset_96[sampling_index];
ics->num_swb = ff_aac_num_swb_96[sampling_index];
} else {
ics->swb_offset = ff_swb_offset_128[sampling_index];
ics->num_swb = ff_aac_num_swb_128[sampling_index];
}
ics->tns_max_bands = ff_tns_max_bands_usac_128[sampling_index];
/* Setup scalefactor grouping. 7 bit mask. */
ics->num_window_groups = 0;
for (int j = 0; j < 7; j++) {
ics->group_len[j] = 1;
if (ue->scale_factor_grouping & (1 << (6 - j)))
ics->group_len[ics->num_window_groups] += 1;
else
ics->num_window_groups++;
}
ics->group_len[7] = 1;
ics->num_window_groups++;
ics->num_windows = 8;
} else {
if (usac->core_frame_len == 768) {
ics->swb_offset = ff_swb_offset_768[sampling_index];
ics->num_swb = ff_aac_num_swb_768[sampling_index];
} else {
ics->swb_offset = ff_swb_offset_1024[sampling_index];
ics->num_swb = ff_aac_num_swb_1024[sampling_index];
}
ics->tns_max_bands = ff_tns_max_bands_usac_1024[sampling_index];
ics->group_len[0] = 1;
ics->num_window_groups = 1;
ics->num_windows = 1;
}
if (ics->max_sfb > ics->num_swb) {
av_log(ac->avctx, AV_LOG_ERROR,
"Number of scalefactor bands in group (%d) "
"exceeds limit (%d).\n",
ics->max_sfb, ics->num_swb);
ics->max_sfb = 0;
return AVERROR(EINVAL);
}
/* Just some defaults for the band types */
for (int i = 0; i < FF_ARRAY_ELEMS(sce->band_type); i++)
sce->band_type[i] = ESC_BT;
return 0;
}
static int decode_usac_stereo_info(AACDecContext *ac, AACUSACConfig *usac,
AACUsacElemConfig *ec, ChannelElement *cpe,
GetBitContext *gb, int indep_flag)
{
int ret, tns_active;
AACUsacStereo *us = &cpe->us;
SingleChannelElement *sce1 = &cpe->ch[0];
SingleChannelElement *sce2 = &cpe->ch[1];
IndividualChannelStream *ics1 = &sce1->ics;
IndividualChannelStream *ics2 = &sce2->ics;
AACUsacElemData *ue1 = &sce1->ue;
AACUsacElemData *ue2 = &sce2->ue;
us->common_window = 0;
us->common_tw = 0;
/* Alpha values must always be zeroed out for the current frame,
* as they are propagated to the next frame and may be used. */
memset(us->alpha_q_re, 0, sizeof(us->alpha_q_re));
memset(us->alpha_q_im, 0, sizeof(us->alpha_q_im));
if (!(!ue1->core_mode && !ue2->core_mode))
return 0;
tns_active = get_bits1(gb);
us->common_window = get_bits1(gb);
if (!us->common_window || indep_flag) {
memset(us->prev_alpha_q_re, 0, sizeof(us->prev_alpha_q_re));
memset(us->prev_alpha_q_im, 0, sizeof(us->prev_alpha_q_im));
}
if (us->common_window) {
/* ics_info() */
ics1->window_sequence[1] = ics1->window_sequence[0];
ics2->window_sequence[1] = ics2->window_sequence[0];
ics1->window_sequence[0] = ics2->window_sequence[0] = get_bits(gb, 2);
ics1->use_kb_window[1] = ics1->use_kb_window[0];
ics2->use_kb_window[1] = ics2->use_kb_window[0];
ics1->use_kb_window[0] = ics2->use_kb_window[0] = get_bits1(gb);
/* If there's a change in the transform sequence, zero out last frame's
* stereo prediction coefficients */
if ((ics1->window_sequence[0] == EIGHT_SHORT_SEQUENCE &&
ics1->window_sequence[1] != EIGHT_SHORT_SEQUENCE) ||
(ics1->window_sequence[1] == EIGHT_SHORT_SEQUENCE &&
ics1->window_sequence[0] != EIGHT_SHORT_SEQUENCE) ||
(ics2->window_sequence[0] == EIGHT_SHORT_SEQUENCE &&
ics2->window_sequence[1] != EIGHT_SHORT_SEQUENCE) ||
(ics2->window_sequence[1] == EIGHT_SHORT_SEQUENCE &&
ics2->window_sequence[0] != EIGHT_SHORT_SEQUENCE)) {
memset(us->prev_alpha_q_re, 0, sizeof(us->prev_alpha_q_re));
memset(us->prev_alpha_q_im, 0, sizeof(us->prev_alpha_q_im));
}
if (ics1->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
ics1->max_sfb = ics2->max_sfb = get_bits(gb, 4);
ue1->scale_factor_grouping = ue2->scale_factor_grouping = get_bits(gb, 7);
} else {
ics1->max_sfb = ics2->max_sfb = get_bits(gb, 6);
}
if (!get_bits1(gb)) { /* common_max_sfb */
if (ics2->window_sequence[0] == EIGHT_SHORT_SEQUENCE)
ics2->max_sfb = get_bits(gb, 4);
else
ics2->max_sfb = get_bits(gb, 6);
}
ret = setup_sce(ac, sce1, usac);
if (ret < 0) {
ics2->max_sfb = 0;
return ret;
}
ret = setup_sce(ac, sce2, usac);
if (ret < 0)
return ret;
cpe->max_sfb_ste = FFMAX(ics1->max_sfb, ics2->max_sfb);
us->ms_mask_mode = get_bits(gb, 2); /* ms_mask_present */
memset(cpe->ms_mask, 0, sizeof(cpe->ms_mask));
if (us->ms_mask_mode == 1) {
for (int g = 0; g < ics1->num_window_groups; g++)
for (int sfb = 0; sfb < cpe->max_sfb_ste; sfb++)
cpe->ms_mask[g*cpe->max_sfb_ste + sfb] = get_bits1(gb);
} else if (us->ms_mask_mode == 2) {
memset(cpe->ms_mask, 0xFF, sizeof(cpe->ms_mask));
} else if ((us->ms_mask_mode == 3) && !ec->stereo_config_index) {
ret = decode_usac_stereo_cplx(ac, us, cpe, gb,
ics1->num_window_groups,
ics1->prev_num_window_groups,
indep_flag);
if (ret < 0)
return ret;
}
}
if (ec->tw_mdct) {
us->common_tw = get_bits1(gb);
avpriv_report_missing_feature(ac->avctx,
"AAC USAC timewarping");
return AVERROR_PATCHWELCOME;
}
us->tns_on_lr = 0;
ue1->tns_data_present = ue2->tns_data_present = 0;
if (tns_active) {
int common_tns = 0;
if (us->common_window)
common_tns = get_bits1(gb);
us->tns_on_lr = get_bits1(gb);
if (common_tns) {
ret = ff_aac_decode_tns(ac, &sce1->tns, gb, ics1);
if (ret < 0)
return ret;
memcpy(&sce2->tns, &sce1->tns, sizeof(sce1->tns));
sce2->tns.present = 1;
sce1->tns.present = 1;
ue1->tns_data_present = 0;
ue2->tns_data_present = 0;
} else {
if (get_bits1(gb)) {
ue1->tns_data_present = 1;
ue2->tns_data_present = 1;
} else {
ue2->tns_data_present = get_bits1(gb);
ue1->tns_data_present = !ue2->tns_data_present;
}
}
}
return 0;
}
/* 7.2.4 Generation of random signs for spectral noise filling
* This function is exactly defined, though we've helped the definition
* along with being slightly faster. */
static inline float noise_random_sign(unsigned int *seed)
{
unsigned int new_seed = *seed = ((*seed) * 69069) + 5;
if (((new_seed) & 0x10000) > 0)
return -1.f;
return +1.f;
}
static void apply_noise_fill(AACDecContext *ac, SingleChannelElement *sce,
AACUsacElemData *ue)
{
float *coef;
IndividualChannelStream *ics = &sce->ics;
float noise_val = powf(2, ((float)ue->noise.level - 14.0f)/3.0f);
int noise_offset = ue->noise.offset - 16;
int band_off;
band_off = ff_usac_noise_fill_start_offset[ac->oc[1].m4ac.frame_length_short]
[ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE];
coef = sce->coeffs;
for (int g = 0; g < ics->num_window_groups; g++) {
unsigned g_len = ics->group_len[g];
for (int sfb = 0; sfb < ics->max_sfb; sfb++) {
float *cb = coef + ics->swb_offset[sfb];
int cb_len = ics->swb_offset[sfb + 1] - ics->swb_offset[sfb];
int band_quantized_to_zero = 1;
if (ics->swb_offset[sfb] < band_off)
continue;
for (int group = 0; group < (unsigned)g_len; group++, cb += 128) {
for (int z = 0; z < cb_len; z++) {
if (cb[z] == 0)
cb[z] = noise_random_sign(&sce->ue.noise.seed) * noise_val;
else
band_quantized_to_zero = 0;
}
}
if (band_quantized_to_zero) {
sce->sfo[g*ics->max_sfb + sfb] = FFMAX(sce->sfo[g*ics->max_sfb + sfb] + noise_offset, -200);
}
}
coef += g_len << 7;
}
}
static void spectrum_scale(AACDecContext *ac, SingleChannelElement *sce,
AACUsacElemData *ue)
{
IndividualChannelStream *ics = &sce->ics;
float *coef;
/* Synthesise noise */
if (ue->noise.level)
apply_noise_fill(ac, sce, ue);
/* Noise filling may apply an offset to the scalefactor offset */
ac->dsp.dequant_scalefactors(sce);
/* Apply scalefactors */
coef = sce->coeffs;
for (int g = 0; g < ics->num_window_groups; g++) {
unsigned g_len = ics->group_len[g];
for (int sfb = 0; sfb < ics->max_sfb; sfb++) {
float *cb = coef + ics->swb_offset[sfb];
int cb_len = ics->swb_offset[sfb + 1] - ics->swb_offset[sfb];
float sf = sce->sf[g*ics->max_sfb + sfb];
for (int group = 0; group < (unsigned)g_len; group++, cb += 128)
ac->fdsp->vector_fmul_scalar(cb, cb, sf, cb_len);
}
coef += g_len << 7;
}
}
static void complex_stereo_downmix_prev(AACDecContext *ac, ChannelElement *cpe,
float *dmix_re)
{
IndividualChannelStream *ics = &cpe->ch[0].ics;
int sign = !cpe->us.pred_dir ? +1 : -1;
float *coef1 = cpe->ch[0].coeffs;
float *coef2 = cpe->ch[1].coeffs;
for (int g = 0; g < ics->num_window_groups; g++) {
unsigned g_len = ics->group_len[g];
for (int sfb = 0; sfb < cpe->max_sfb_ste; sfb++) {
int off = ics->swb_offset[sfb];
int cb_len = ics->swb_offset[sfb + 1] - off;
float *c1 = coef1 + off;
float *c2 = coef2 + off;
float *dm = dmix_re + off;
for (int group = 0; group < (unsigned)g_len;
group++, c1 += 128, c2 += 128, dm += 128) {
for (int z = 0; z < cb_len; z++)
dm[z] = 0.5*(c1[z] + sign*c2[z]);
}
}
coef1 += g_len << 7;
coef2 += g_len << 7;
dmix_re += g_len << 7;
}
}
static void complex_stereo_downmix_cur(AACDecContext *ac, ChannelElement *cpe,
float *dmix_re)
{
AACUsacStereo *us = &cpe->us;
IndividualChannelStream *ics = &cpe->ch[0].ics;
int sign = !cpe->us.pred_dir ? +1 : -1;
float *coef1 = cpe->ch[0].coeffs;
float *coef2 = cpe->ch[1].coeffs;
for (int g = 0; g < ics->num_window_groups; g++) {
unsigned g_len = ics->group_len[g];
for (int sfb = 0; sfb < cpe->max_sfb_ste; sfb++) {
int off = ics->swb_offset[sfb];
int cb_len = ics->swb_offset[sfb + 1] - off;
float *c1 = coef1 + off;
float *c2 = coef2 + off;
float *dm = dmix_re + off;
if (us->pred_used[g*cpe->max_sfb_ste + sfb]) {
for (int group = 0; group < (unsigned)g_len;
group++, c1 += 128, c2 += 128, dm += 128) {
for (int z = 0; z < cb_len; z++)
dm[z] = 0.5*(c1[z] + sign*c2[z]);
}
} else {
for (int group = 0; group < (unsigned)g_len;
group++, c1 += 128, c2 += 128, dm += 128) {
for (int z = 0; z < cb_len; z++)
dm[z] = c1[z];
}
}
}
coef1 += g_len << 7;
coef2 += g_len << 7;
dmix_re += g_len << 7;
}
}
static void complex_stereo_interpolate_imag(float *im, float *re, const float f[7],
int len, int factor_even, int factor_odd)
{
int i = 0;
float s;
s = f[6]*re[2] + f[5]*re[1] + f[4]*re[0] +
f[3]*re[0] +
f[2]*re[1] + f[1]*re[2] + f[0]*re[3];
im[i] += s*factor_even;
i = 1;
s = f[6]*re[1] + f[5]*re[0] + f[4]*re[0] +
f[3]*re[1] +
f[2]*re[2] + f[1]*re[3] + f[0]*re[4];
im[i] += s*factor_odd;
i = 2;
s = f[6]*re[0] + f[5]*re[0] + f[4]*re[1] +
f[3]*re[2] +
f[2]*re[3] + f[1]*re[4] + f[0]*re[5];
im[i] += s*factor_even;
for (i = 3; i < len - 4; i += 2) {
s = f[6]*re[i-3] + f[5]*re[i-2] + f[4]*re[i-1] +
f[3]*re[i] +
f[2]*re[i+1] + f[1]*re[i+2] + f[0]*re[i+3];
im[i+0] += s*factor_odd;
s = f[6]*re[i-2] + f[5]*re[i-1] + f[4]*re[i] +
f[3]*re[i+1] +
f[2]*re[i+2] + f[1]*re[i+3] + f[0]*re[i+4];
im[i+1] += s*factor_even;
}
i = len - 3;
s = f[6]*re[i-3] + f[5]*re[i-2] + f[4]*re[i-1] +
f[3]*re[i] +
f[2]*re[i+1] + f[1]*re[i+2] + f[0]*re[i+2];
im[i] += s*factor_odd;
i = len - 2;
s = f[6]*re[i-3] + f[5]*re[i-2] + f[4]*re[i-1] +
f[3]*re[i] +
f[2]*re[i+1] + f[1]*re[i+1] + f[0]*re[i];
im[i] += s*factor_even;
i = len - 1;
s = f[6]*re[i-3] + f[5]*re[i-2] + f[4]*re[i-1] +
f[3]*re[i] +
f[2]*re[i] + f[1]*re[i-1] + f[0]*re[i-2];
im[i] += s*factor_odd;
}
static void apply_complex_stereo(AACDecContext *ac, ChannelElement *cpe)
{
AACUsacStereo *us = &cpe->us;
IndividualChannelStream *ics = &cpe->ch[0].ics;
float *coef1 = cpe->ch[0].coeffs;
float *coef2 = cpe->ch[1].coeffs;
float *dmix_im = us->dmix_im;
for (int g = 0; g < ics->num_window_groups; g++) {
unsigned g_len = ics->group_len[g];
for (int sfb = 0; sfb < cpe->max_sfb_ste; sfb++) {
int off = ics->swb_offset[sfb];
int cb_len = ics->swb_offset[sfb + 1] - off;
float *c1 = coef1 + off;
float *c2 = coef2 + off;
float *dm_im = dmix_im + off;
float alpha_re = us->alpha_q_re[g*cpe->max_sfb_ste + sfb];
float alpha_im = us->alpha_q_im[g*cpe->max_sfb_ste + sfb];
if (!us->pred_used[g*cpe->max_sfb_ste + sfb])
continue;
if (!cpe->us.pred_dir) {
for (int group = 0; group < (unsigned)g_len;
group++, c1 += 128, c2 += 128, dm_im += 128) {
for (int z = 0; z < cb_len; z++) {
float side;
side = c2[z] - alpha_re*c1[z] - alpha_im*dm_im[z];
c2[z] = c1[z] - side;
c1[z] = c1[z] + side;
}
}
} else {
for (int group = 0; group < (unsigned)g_len;
group++, c1 += 128, c2 += 128, dm_im += 128) {
for (int z = 0; z < cb_len; z++) {
float mid;
mid = c2[z] - alpha_re*c1[z] - alpha_im*dm_im[z];
c2[z] = mid - c1[z];
c1[z] = mid + c1[z];
}
}
}
}
coef1 += g_len << 7;
coef2 += g_len << 7;
dmix_im += g_len << 7;
}
}
static const float *complex_stereo_get_filter(ChannelElement *cpe, int is_prev)
{
int win, shape;
if (!is_prev) {
switch (cpe->ch[0].ics.window_sequence[0]) {
default:
case ONLY_LONG_SEQUENCE:
case EIGHT_SHORT_SEQUENCE:
win = 0;
break;
case LONG_START_SEQUENCE:
win = 1;
break;
case LONG_STOP_SEQUENCE:
win = 2;
break;
}
if (cpe->ch[0].ics.use_kb_window[0] == 0 &&
cpe->ch[0].ics.use_kb_window[1] == 0)
shape = 0;
else if (cpe->ch[0].ics.use_kb_window[0] == 1 &&
cpe->ch[0].ics.use_kb_window[1] == 1)
shape = 1;
else if (cpe->ch[0].ics.use_kb_window[0] == 0 &&
cpe->ch[0].ics.use_kb_window[1] == 1)
shape = 2;
else if (cpe->ch[0].ics.use_kb_window[0] == 1 &&
cpe->ch[0].ics.use_kb_window[1] == 0)
shape = 3;
else
shape = 3;
} else {
win = cpe->ch[0].ics.window_sequence[0] == LONG_STOP_SEQUENCE;
shape = cpe->ch[0].ics.use_kb_window[1];
}
return ff_aac_usac_mdst_filt_cur[win][shape];
}
static void spectrum_decode(AACDecContext *ac, AACUSACConfig *usac,
ChannelElement *cpe, int nb_channels)
{
AACUsacStereo *us = &cpe->us;
for (int ch = 0; ch < nb_channels; ch++) {
SingleChannelElement *sce = &cpe->ch[ch];
AACUsacElemData *ue = &sce->ue;
spectrum_scale(ac, sce, ue);
}
if (nb_channels > 1 && us->common_window) {
for (int ch = 0; ch < nb_channels; ch++) {
SingleChannelElement *sce = &cpe->ch[ch];
/* Apply TNS, if the tns_on_lr bit is not set. */
if (sce->tns.present && !us->tns_on_lr)
ac->dsp.apply_tns(sce->coeffs, &sce->tns, &sce->ics, 1);
}
if (us->ms_mask_mode == 3) {
const float *filt;
complex_stereo_downmix_cur(ac, cpe, us->dmix_re);
complex_stereo_downmix_prev(ac, cpe, us->prev_dmix_re);
filt = complex_stereo_get_filter(cpe, 0);
complex_stereo_interpolate_imag(us->dmix_im, us->dmix_re, filt,
usac->core_frame_len, 1, 1);
if (us->use_prev_frame) {
filt = complex_stereo_get_filter(cpe, 1);
complex_stereo_interpolate_imag(us->dmix_im, us->prev_dmix_re, filt,
usac->core_frame_len, -1, 1);
}
apply_complex_stereo(ac, cpe);
} else if (us->ms_mask_mode > 0) {
ac->dsp.apply_mid_side_stereo(ac, cpe);
}
}
/* Save coefficients and alpha values for prediction reasons */
if (nb_channels > 1) {
AACUsacStereo *us = &cpe->us;
for (int ch = 0; ch < nb_channels; ch++) {
SingleChannelElement *sce = &cpe->ch[ch];
memcpy(sce->prev_coeffs, sce->coeffs, sizeof(sce->coeffs));
}
memcpy(us->prev_alpha_q_re, us->alpha_q_re, sizeof(us->alpha_q_re));
memcpy(us->prev_alpha_q_im, us->alpha_q_im, sizeof(us->alpha_q_im));
}
for (int ch = 0; ch < nb_channels; ch++) {
SingleChannelElement *sce = &cpe->ch[ch];
/* Apply TNS, if it hasn't been applied yet. */
if (sce->tns.present && ((nb_channels == 1) || (us->tns_on_lr)))
ac->dsp.apply_tns(sce->coeffs, &sce->tns, &sce->ics, 1);
ac->oc[1].m4ac.frame_length_short ? ac->dsp.imdct_and_windowing_768(ac, sce) :
ac->dsp.imdct_and_windowing(ac, sce);
}
}
static int decode_usac_core_coder(AACDecContext *ac, AACUSACConfig *usac,
AACUsacElemConfig *ec, ChannelElement *che,
GetBitContext *gb, int indep_flag, int nb_channels)
{
int ret;
int arith_reset_flag;
AACUsacStereo *us = &che->us;
int core_nb_channels = nb_channels;
/* Local symbols */
uint8_t global_gain;
us->common_window = 0;
for (int ch = 0; ch < core_nb_channels; ch++) {
SingleChannelElement *sce = &che->ch[ch];
AACUsacElemData *ue = &sce->ue;
sce->tns.present = 0;
ue->tns_data_present = 0;
ue->core_mode = get_bits1(gb);
}
if (nb_channels > 1 && ec->stereo_config_index == 1)
core_nb_channels = 1;
if (core_nb_channels == 2) {
ret = decode_usac_stereo_info(ac, usac, ec, che, gb, indep_flag);
if (ret)
return ret;
}
for (int ch = 0; ch < core_nb_channels; ch++) {
SingleChannelElement *sce = &che->ch[ch];
IndividualChannelStream *ics = &sce->ics;
AACUsacElemData *ue = &sce->ue;
if (ue->core_mode) { /* lpd_channel_stream */
ret = ff_aac_ldp_parse_channel_stream(ac, usac, ue, gb);
if (ret < 0)
return ret;
continue;
}
if ((core_nb_channels == 1) ||
(che->ch[0].ue.core_mode != che->ch[1].ue.core_mode))
ue->tns_data_present = get_bits1(gb);
/* fd_channel_stream */
global_gain = get_bits(gb, 8);
ue->noise.level = 0;
if (ec->noise_fill) {
ue->noise.level = get_bits(gb, 3);
ue->noise.offset = get_bits(gb, 5);
}
if (!us->common_window) {
/* ics_info() */
ics->window_sequence[1] = ics->window_sequence[0];
ics->window_sequence[0] = get_bits(gb, 2);
ics->use_kb_window[1] = ics->use_kb_window[0];
ics->use_kb_window[0] = get_bits1(gb);
if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
ics->max_sfb = get_bits(gb, 4);
ue->scale_factor_grouping = get_bits(gb, 7);
} else {
ics->max_sfb = get_bits(gb, 6);
}
ret = setup_sce(ac, sce, usac);
if (ret < 0)
return ret;
}
if (ec->tw_mdct && !us->common_tw) {
/* tw_data() */
if (get_bits1(gb)) { /* tw_data_present */
/* Time warping is not supported in baseline profile streams. */
avpriv_report_missing_feature(ac->avctx,
"AAC USAC timewarping");
return AVERROR_PATCHWELCOME;
}
}
ret = decode_usac_scale_factors(ac, sce, gb, global_gain);
if (ret < 0)
return ret;
if (ue->tns_data_present) {
sce->tns.present = 1;
ret = ff_aac_decode_tns(ac, &sce->tns, gb, ics);
if (ret < 0)
return ret;
}
/* ac_spectral_data */
arith_reset_flag = indep_flag;
if (!arith_reset_flag)
arith_reset_flag = get_bits1(gb);
/* Decode coeffs */
memset(&sce->coeffs[0], 0, 1024*sizeof(float));
for (int win = 0; win < ics->num_windows; win++) {
int lg = ics->swb_offset[ics->max_sfb];
int N;
if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE)
N = usac->core_frame_len / 8;
else
N = usac->core_frame_len;
ret = decode_spectrum_ac(ac, sce->coeffs + win*128, gb, &ue->ac,
arith_reset_flag && (win == 0), lg, N);
if (ret < 0)
return ret;
}
if (get_bits1(gb)) { /* fac_data_present */
const uint16_t len_8 = usac->core_frame_len / 8;
const uint16_t len_16 = usac->core_frame_len / 16;
const uint16_t fac_len = ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE ? len_16 : len_8;
ret = ff_aac_parse_fac_data(ue, gb, 1, fac_len);
if (ret < 0)
return ret;
}
}
if (ec->sbr.ratio) {
int sbr_ch = nb_channels;
if (nb_channels == 2 &&
!(ec->stereo_config_index == 0 || ec->stereo_config_index == 3))
sbr_ch = 1;
ret = ff_aac_sbr_decode_usac_data(ac, che, ec, gb, sbr_ch, indep_flag);
if (ret < 0)
return ret;
}
if (ec->stereo_config_index) {
avpriv_report_missing_feature(ac->avctx, "AAC USAC Mps212");
return AVERROR_PATCHWELCOME;
}
spectrum_decode(ac, usac, che, core_nb_channels);
if (ac->oc[1].m4ac.sbr > 0) {
ac->proc.sbr_apply(ac, che, nb_channels == 2 ? TYPE_CPE : TYPE_SCE,
che->ch[0].output,
che->ch[1].output);
}
return 0;
}
static int parse_audio_preroll(AACDecContext *ac, GetBitContext *gb)
{
int ret = 0;
GetBitContext gbc;
OutputConfiguration *oc = &ac->oc[1];
MPEG4AudioConfig *m4ac = &oc->m4ac;
MPEG4AudioConfig m4ac_bak = oc->m4ac;
uint8_t temp_data[512];
uint8_t *tmp_buf = temp_data;
size_t tmp_buf_size = sizeof(temp_data);
av_unused int crossfade;
int num_preroll_frames;
int config_len = get_escaped_value(gb, 4, 4, 8);
/* Implementations are free to pad the config to any length, so use a
* different reader for this. */
gbc = *gb;
ret = ff_aac_usac_config_decode(ac, ac->avctx, &gbc, oc, m4ac->chan_config);
if (ret < 0) {
*m4ac = m4ac_bak;
return ret;
} else {
ac->oc[1].m4ac.chan_config = 0;
}
/* 7.18.3.3 Bitrate adaption
* If configuration didn't change after applying preroll, continue
* without decoding it. */
if (!memcmp(m4ac, &m4ac_bak, sizeof(m4ac_bak)))
return 0;
skip_bits_long(gb, config_len*8);
crossfade = get_bits1(gb); /* applyCrossfade */
skip_bits1(gb); /* reserved */
num_preroll_frames = get_escaped_value(gb, 2, 4, 0); /* numPreRollFrames */
for (int i = 0; i < num_preroll_frames; i++) {
int got_frame_ptr = 0;
int au_len = get_escaped_value(gb, 16, 16, 0);
if (au_len*8 > tmp_buf_size) {
uint8_t *tmp2;
tmp_buf = tmp_buf == temp_data ? NULL : tmp_buf;
tmp2 = av_realloc_array(tmp_buf, au_len, 8);
if (!tmp2) {
if (tmp_buf != temp_data)
av_free(tmp_buf);
return AVERROR(ENOMEM);
}
tmp_buf = tmp2;
}
/* Byte alignment is not guaranteed. */
for (int i = 0; i < au_len; i++)
tmp_buf[i] = get_bits(gb, 8);
ret = init_get_bits8(&gbc, tmp_buf, au_len);
if (ret < 0)
break;
ret = ff_aac_usac_decode_frame(ac->avctx, ac, &gbc, &got_frame_ptr);
if (ret < 0)
break;
}
if (tmp_buf != temp_data)
av_free(tmp_buf);
return 0;
}
static int parse_ext_ele(AACDecContext *ac, AACUsacElemConfig *e,
GetBitContext *gb)
{
uint8_t pl_frag_start = 1;
uint8_t pl_frag_end = 1;
uint32_t len;
if (!get_bits1(gb)) /* usacExtElementPresent */
return 0;
if (get_bits1(gb)) { /* usacExtElementUseDefaultLength */
len = e->ext.default_len;
} else {
len = get_bits(gb, 8); /* usacExtElementPayloadLength */
if (len == 255)
len += get_bits(gb, 16) - 2;
}
if (!len)
return 0;
if (e->ext.payload_frag) {
pl_frag_start = get_bits1(gb); /* usacExtElementStart */
pl_frag_end = get_bits1(gb); /* usacExtElementStop */
}
if (pl_frag_start)
e->ext.pl_data_offset = 0;
/* If an extension starts and ends this packet, we can directly use it below.
* Otherwise, we have to copy it to a buffer and accumulate it. */
if (!(pl_frag_start && pl_frag_end)) {
/* Reallocate the data */
uint8_t *tmp_buf = av_refstruct_alloc_ext(e->ext.pl_data_offset + len,
AV_REFSTRUCT_FLAG_NO_ZEROING,
NULL, NULL);
if (!tmp_buf)
return AVERROR(ENOMEM);
/* Copy the data over only if we had saved data to begin with */
if (e->ext.pl_buf)
memcpy(tmp_buf, e->ext.pl_buf, e->ext.pl_data_offset);
av_refstruct_unref(&e->ext.pl_buf);
e->ext.pl_buf = tmp_buf;
/* Readout data to a buffer */
for (int i = 0; i < len; i++)
e->ext.pl_buf[e->ext.pl_data_offset + i] = get_bits(gb, 8);
}
e->ext.pl_data_offset += len;
if (pl_frag_end) {
int ret = 0;
int start_bits = get_bits_count(gb);
const int pl_len = e->ext.pl_data_offset;
GetBitContext *gb2 = gb;
GetBitContext gbc;
if (!(pl_frag_start && pl_frag_end)) {
ret = init_get_bits8(&gbc, e->ext.pl_buf, pl_len);
if (ret < 0)
return ret;
gb2 = &gbc;
}
switch (e->ext.type) {
case ID_EXT_ELE_FILL:
/* Filler elements have no usable payload */
break;
case ID_EXT_ELE_AUDIOPREROLL:
ret = parse_audio_preroll(ac, gb2);
break;
default:
/* This should never happen */
av_assert0(0);
}
av_refstruct_unref(&e->ext.pl_buf);
if (ret < 0)
return ret;
skip_bits_long(gb, pl_len*8 - (get_bits_count(gb) - start_bits));
}
return 0;
}
int ff_aac_usac_decode_frame(AVCodecContext *avctx, AACDecContext *ac,
GetBitContext *gb, int *got_frame_ptr)
{
int ret, is_dmono = 0;
int indep_flag, samples = 0;
int audio_found = 0;
int elem_id[3 /* SCE, CPE, LFE */] = { 0, 0, 0 };
AVFrame *frame = ac->frame;
int ratio_mult, ratio_dec;
AACUSACConfig *usac = &ac->oc[1].usac;
int sbr_ratio = usac->core_sbr_frame_len_idx == 2 ? 2 :
usac->core_sbr_frame_len_idx == 3 ? 3 :
usac->core_sbr_frame_len_idx == 4 ? 1 :
0;
if (sbr_ratio == 2) {
ratio_mult = 8;
ratio_dec = 3;
} else if (sbr_ratio == 3) {
ratio_mult = 2;
ratio_dec = 1;
} else if (sbr_ratio == 4) {
ratio_mult = 4;
ratio_dec = 1;
} else {
ratio_mult = 1;
ratio_dec = 1;
}
ff_aac_output_configure(ac, ac->oc[1].layout_map, ac->oc[1].layout_map_tags,
ac->oc[1].status, 0);
ac->avctx->profile = AV_PROFILE_AAC_USAC;
indep_flag = get_bits1(gb);
for (int i = 0; i < ac->oc[1].usac.nb_elems; i++) {
int layout_id;
int layout_type;
AACUsacElemConfig *e = &ac->oc[1].usac.elems[i];
ChannelElement *che;
if (e->type == ID_USAC_SCE) {
layout_id = elem_id[0]++;
layout_type = TYPE_SCE;
che = ff_aac_get_che(ac, TYPE_SCE, layout_id);
} else if (e->type == ID_USAC_CPE) {
layout_id = elem_id[1]++;
layout_type = TYPE_CPE;
che = ff_aac_get_che(ac, TYPE_CPE, layout_id);
} else if (e->type == ID_USAC_LFE) {
layout_id = elem_id[2]++;
layout_type = TYPE_LFE;
che = ff_aac_get_che(ac, TYPE_LFE, layout_id);
}
if (e->type != ID_USAC_EXT && !che) {
av_log(ac->avctx, AV_LOG_ERROR,
"channel element %d.%d is not allocated\n",
layout_type, layout_id);
return AVERROR_INVALIDDATA;
}
switch (e->type) {
case ID_USAC_LFE:
/* Fallthrough */
case ID_USAC_SCE:
ret = decode_usac_core_coder(ac, &ac->oc[1].usac, e, che, gb,
indep_flag, 1);
if (ret < 0)
return ret;
audio_found = 1;
che->present = 1;
break;
case ID_USAC_CPE:
ret = decode_usac_core_coder(ac, &ac->oc[1].usac, e, che, gb,
indep_flag, 2);
if (ret < 0)
return ret;
audio_found = 1;
che->present = 1;
break;
case ID_USAC_EXT:
ret = parse_ext_ele(ac, e, gb);
if (ret < 0)
return ret;
break;
}
}
if (audio_found)
samples = ac->oc[1].m4ac.frame_length_short ? 768 : 1024;
samples = (samples * ratio_mult) / ratio_dec;
if (ac->oc[1].status && audio_found) {
avctx->sample_rate = ac->oc[1].m4ac.ext_sample_rate;
avctx->frame_size = samples;
ac->oc[1].status = OC_LOCKED;
}
if (!frame->data[0] && samples) {
av_log(avctx, AV_LOG_ERROR, "no frame data found\n");
return AVERROR_INVALIDDATA;
}
if (samples) {
frame->nb_samples = samples;
frame->sample_rate = avctx->sample_rate;
frame->flags = indep_flag ? AV_FRAME_FLAG_KEY : 0x0;
*got_frame_ptr = 1;
} else {
av_frame_unref(ac->frame);
frame->flags = indep_flag ? AV_FRAME_FLAG_KEY : 0x0;
*got_frame_ptr = 0;
}
/* for dual-mono audio (SCE + SCE) */
is_dmono = ac->dmono_mode && elem_id[0] == 2 &&
!av_channel_layout_compare(&ac->oc[1].ch_layout,
&(AVChannelLayout)AV_CHANNEL_LAYOUT_STEREO);
if (is_dmono) {
if (ac->dmono_mode == 1)
frame->data[1] = frame->data[0];
else if (ac->dmono_mode == 2)
frame->data[0] = frame->data[1];
}
return 0;
}
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