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ffmpeg/libavcodec/aac/aacdec.c

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/*
* Common parts of the AAC decoders
* Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org )
* Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com )
* Copyright (c) 2008-2013 Alex Converse <alex.converse@gmail.com>
*
* AAC LATM decoder
* Copyright (c) 2008-2010 Paul Kendall <paul@kcbbs.gen.nz>
* Copyright (c) 2010 Janne Grunau <janne-libav@jannau.net>
*
* AAC decoder fixed-point implementation
* Copyright (c) 2013
* MIPS Technologies, Inc., California.
*
* 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
*/
/* We use several quantization functions here (Q31, Q30),
* for which we need this to be defined for them to work as expected. */
#define USE_FIXED 1
#include "config_components.h"
#include <limits.h>
#include <stddef.h>
#include "aacdec.h"
#include "aacdec_tab.h"
#include "aacdec_usac.h"
#include "libavcodec/aac.h"
#include "libavcodec/aac_defines.h"
#include "libavcodec/aacsbr.h"
#include "libavcodec/aactab.h"
#include "libavcodec/adts_header.h"
#include "libavcodec/avcodec.h"
#include "libavcodec/internal.h"
#include "libavcodec/codec_internal.h"
#include "libavcodec/decode.h"
#include "libavcodec/profiles.h"
#include "libavutil/attributes.h"
#include "libavutil/error.h"
#include "libavutil/log.h"
#include "libavutil/macros.h"
#include "libavutil/mem.h"
#include "libavutil/opt.h"
#include "libavutil/tx.h"
#include "libavutil/version.h"
/*
* supported tools
*
* Support? Name
* N (code in SoC repo) gain control
* Y block switching
* Y window shapes - standard
* N window shapes - Low Delay
* Y filterbank - standard
* N (code in SoC repo) filterbank - Scalable Sample Rate
* Y Temporal Noise Shaping
* Y Long Term Prediction
* Y intensity stereo
* Y channel coupling
* Y frequency domain prediction
* Y Perceptual Noise Substitution
* Y Mid/Side stereo
* N Scalable Inverse AAC Quantization
* N Frequency Selective Switch
* N upsampling filter
* Y quantization & coding - AAC
* N quantization & coding - TwinVQ
* N quantization & coding - BSAC
* N AAC Error Resilience tools
* N Error Resilience payload syntax
* N Error Protection tool
* N CELP
* N Silence Compression
* N HVXC
* N HVXC 4kbits/s VR
* N Structured Audio tools
* N Structured Audio Sample Bank Format
* N MIDI
* N Harmonic and Individual Lines plus Noise
* N Text-To-Speech Interface
* Y Spectral Band Replication
* Y (not in this code) Layer-1
* Y (not in this code) Layer-2
* Y (not in this code) Layer-3
* N SinuSoidal Coding (Transient, Sinusoid, Noise)
* Y Parametric Stereo
* N Direct Stream Transfer
* Y (not in fixed point code) Enhanced AAC Low Delay (ER AAC ELD)
*
* Note: - HE AAC v1 comprises LC AAC with Spectral Band Replication.
* - HE AAC v2 comprises LC AAC with Spectral Band Replication and
Parametric Stereo.
*/
#define overread_err "Input buffer exhausted before END element found\n"
static int count_channels(uint8_t (*layout)[3], int tags)
{
int i, sum = 0;
for (i = 0; i < tags; i++) {
int syn_ele = layout[i][0];
int pos = layout[i][2];
sum += (1 + (syn_ele == TYPE_CPE)) *
(pos != AAC_CHANNEL_OFF && pos != AAC_CHANNEL_CC);
}
return sum;
}
/**
* Check for the channel element in the current channel position configuration.
* If it exists, make sure the appropriate element is allocated and map the
* channel order to match the internal FFmpeg channel layout.
*
* @param che_pos current channel position configuration
* @param type channel element type
* @param id channel element id
* @param channels count of the number of channels in the configuration
*
* @return Returns error status. 0 - OK, !0 - error
*/
static av_cold int che_configure(AACDecContext *ac,
enum ChannelPosition che_pos,
int type, int id, int *channels)
{
if (*channels >= MAX_CHANNELS)
return AVERROR_INVALIDDATA;
if (che_pos) {
if (!ac->che[type][id]) {
int ret = ac->proc.sbr_ctx_alloc_init(ac, &ac->che[type][id], type);
if (ret < 0)
return ret;
}
if (type != TYPE_CCE) {
if (*channels >= MAX_CHANNELS - (type == TYPE_CPE || (type == TYPE_SCE && ac->oc[1].m4ac.ps == 1))) {
av_log(ac->avctx, AV_LOG_ERROR, "Too many channels\n");
return AVERROR_INVALIDDATA;
}
ac->output_element[(*channels)++] = &ac->che[type][id]->ch[0];
if (type == TYPE_CPE ||
(type == TYPE_SCE && ac->oc[1].m4ac.ps == 1)) {
ac->output_element[(*channels)++] = &ac->che[type][id]->ch[1];
}
}
} else {
if (ac->che[type][id]) {
ac->proc.sbr_ctx_close(ac->che[type][id]);
}
av_freep(&ac->che[type][id]);
}
return 0;
}
static int frame_configure_elements(AVCodecContext *avctx)
{
AACDecContext *ac = avctx->priv_data;
int type, id, ch, ret;
/* set channel pointers to internal buffers by default */
for (type = 0; type < 4; type++) {
for (id = 0; id < MAX_ELEM_ID; id++) {
ChannelElement *che = ac->che[type][id];
if (che) {
che->ch[0].output = che->ch[0].ret_buf;
che->ch[1].output = che->ch[1].ret_buf;
}
}
}
/* get output buffer */
av_frame_unref(ac->frame);
if (!avctx->ch_layout.nb_channels)
return 1;
ac->frame->nb_samples = 2048;
if ((ret = ff_get_buffer(avctx, ac->frame, 0)) < 0)
return ret;
/* map output channel pointers to AVFrame data */
for (ch = 0; ch < avctx->ch_layout.nb_channels; ch++) {
if (ac->output_element[ch])
ac->output_element[ch]->output = (void *)ac->frame->extended_data[ch];
}
return 0;
}
struct elem_to_channel {
uint64_t av_position;
uint8_t syn_ele;
uint8_t elem_id;
uint8_t aac_position;
};
static int assign_pair(struct elem_to_channel e2c_vec[MAX_ELEM_ID],
uint8_t (*layout_map)[3], int offset, uint64_t left,
uint64_t right, int pos, uint64_t *layout)
{
if (layout_map[offset][0] == TYPE_CPE) {
e2c_vec[offset] = (struct elem_to_channel) {
.av_position = left | right,
.syn_ele = TYPE_CPE,
.elem_id = layout_map[offset][1],
.aac_position = pos
};
if (e2c_vec[offset].av_position != UINT64_MAX)
*layout |= e2c_vec[offset].av_position;
return 1;
} else {
e2c_vec[offset] = (struct elem_to_channel) {
.av_position = left,
.syn_ele = TYPE_SCE,
.elem_id = layout_map[offset][1],
.aac_position = pos
};
e2c_vec[offset + 1] = (struct elem_to_channel) {
.av_position = right,
.syn_ele = TYPE_SCE,
.elem_id = layout_map[offset + 1][1],
.aac_position = pos
};
if (left != UINT64_MAX)
*layout |= left;
if (right != UINT64_MAX)
*layout |= right;
return 2;
}
}
static int count_paired_channels(uint8_t (*layout_map)[3], int tags, int pos,
int current)
{
int num_pos_channels = 0;
int first_cpe = 0;
int sce_parity = 0;
int i;
for (i = current; i < tags; i++) {
if (layout_map[i][2] != pos)
break;
if (layout_map[i][0] == TYPE_CPE) {
if (sce_parity) {
if (pos == AAC_CHANNEL_FRONT && !first_cpe) {
sce_parity = 0;
} else {
return -1;
}
}
num_pos_channels += 2;
first_cpe = 1;
} else {
num_pos_channels++;
sce_parity ^= (pos != AAC_CHANNEL_LFE);
}
}
if (sce_parity &&
(pos == AAC_CHANNEL_FRONT && first_cpe))
return -1;
return num_pos_channels;
}
static int assign_channels(struct elem_to_channel e2c_vec[MAX_ELEM_ID], uint8_t (*layout_map)[3],
uint64_t *layout, int tags, int layer, int pos, int *current)
{
int i = *current, j = 0;
int nb_channels = count_paired_channels(layout_map, tags, pos, i);
if (nb_channels < 0 || nb_channels > 5)
return 0;
if (pos == AAC_CHANNEL_LFE) {
while (nb_channels) {
if (ff_aac_channel_map[layer][pos - 1][j] == AV_CHAN_NONE)
return -1;
e2c_vec[i] = (struct elem_to_channel) {
.av_position = 1ULL << ff_aac_channel_map[layer][pos - 1][j],
.syn_ele = layout_map[i][0],
.elem_id = layout_map[i][1],
.aac_position = pos
};
*layout |= e2c_vec[i].av_position;
i++;
j++;
nb_channels--;
}
*current = i;
return 0;
}
while (nb_channels & 1) {
if (ff_aac_channel_map[layer][pos - 1][0] == AV_CHAN_NONE)
return -1;
if (ff_aac_channel_map[layer][pos - 1][0] == AV_CHAN_UNUSED)
break;
e2c_vec[i] = (struct elem_to_channel) {
.av_position = 1ULL << ff_aac_channel_map[layer][pos - 1][0],
.syn_ele = layout_map[i][0],
.elem_id = layout_map[i][1],
.aac_position = pos
};
*layout |= e2c_vec[i].av_position;
i++;
nb_channels--;
}
j = (pos != AAC_CHANNEL_SIDE) && nb_channels <= 3 ? 3 : 1;
while (nb_channels >= 2) {
if (ff_aac_channel_map[layer][pos - 1][j] == AV_CHAN_NONE ||
ff_aac_channel_map[layer][pos - 1][j+1] == AV_CHAN_NONE)
return -1;
i += assign_pair(e2c_vec, layout_map, i,
1ULL << ff_aac_channel_map[layer][pos - 1][j],
1ULL << ff_aac_channel_map[layer][pos - 1][j+1],
pos, layout);
j += 2;
nb_channels -= 2;
}
while (nb_channels & 1) {
if (ff_aac_channel_map[layer][pos - 1][5] == AV_CHAN_NONE)
return -1;
e2c_vec[i] = (struct elem_to_channel) {
.av_position = 1ULL << ff_aac_channel_map[layer][pos - 1][5],
.syn_ele = layout_map[i][0],
.elem_id = layout_map[i][1],
.aac_position = pos
};
*layout |= e2c_vec[i].av_position;
i++;
nb_channels--;
}
if (nb_channels)
return -1;
*current = i;
return 0;
}
static uint64_t sniff_channel_order(uint8_t (*layout_map)[3], int tags)
{
int i, n, total_non_cc_elements;
struct elem_to_channel e2c_vec[4 * MAX_ELEM_ID] = { { 0 } };
uint64_t layout = 0;
if (FF_ARRAY_ELEMS(e2c_vec) < tags)
return 0;
for (n = 0, i = 0; n < 3 && i < tags; n++) {
int ret = assign_channels(e2c_vec, layout_map, &layout, tags, n, AAC_CHANNEL_FRONT, &i);
if (ret < 0)
return 0;
ret = assign_channels(e2c_vec, layout_map, &layout, tags, n, AAC_CHANNEL_SIDE, &i);
if (ret < 0)
return 0;
ret = assign_channels(e2c_vec, layout_map, &layout, tags, n, AAC_CHANNEL_BACK, &i);
if (ret < 0)
return 0;
ret = assign_channels(e2c_vec, layout_map, &layout, tags, n, AAC_CHANNEL_LFE, &i);
if (ret < 0)
return 0;
}
total_non_cc_elements = n = i;
if (layout == AV_CH_LAYOUT_22POINT2) {
// For 22.2 reorder the result as needed
FFSWAP(struct elem_to_channel, e2c_vec[2], e2c_vec[0]); // FL & FR first (final), FC third
FFSWAP(struct elem_to_channel, e2c_vec[2], e2c_vec[1]); // FC second (final), FLc & FRc third
FFSWAP(struct elem_to_channel, e2c_vec[6], e2c_vec[2]); // LFE1 third (final), FLc & FRc seventh
FFSWAP(struct elem_to_channel, e2c_vec[4], e2c_vec[3]); // BL & BR fourth (final), SiL & SiR fifth
FFSWAP(struct elem_to_channel, e2c_vec[6], e2c_vec[4]); // FLc & FRc fifth (final), SiL & SiR seventh
FFSWAP(struct elem_to_channel, e2c_vec[7], e2c_vec[6]); // LFE2 seventh (final), SiL & SiR eight (final)
FFSWAP(struct elem_to_channel, e2c_vec[9], e2c_vec[8]); // TpFL & TpFR ninth (final), TFC tenth (final)
FFSWAP(struct elem_to_channel, e2c_vec[11], e2c_vec[10]); // TC eleventh (final), TpSiL & TpSiR twelth
FFSWAP(struct elem_to_channel, e2c_vec[12], e2c_vec[11]); // TpBL & TpBR twelth (final), TpSiL & TpSiR thirteenth (final)
} else {
// For everything else, utilize the AV channel position define as a
// stable sort.
do {
int next_n = 0;
for (i = 1; i < n; i++)
if (e2c_vec[i - 1].av_position > e2c_vec[i].av_position) {
FFSWAP(struct elem_to_channel, e2c_vec[i - 1], e2c_vec[i]);
next_n = i;
}
n = next_n;
} while (n > 0);
}
for (i = 0; i < total_non_cc_elements; i++) {
layout_map[i][0] = e2c_vec[i].syn_ele;
layout_map[i][1] = e2c_vec[i].elem_id;
layout_map[i][2] = e2c_vec[i].aac_position;
}
return layout;
}
/**
* Save current output configuration if and only if it has been locked.
*/
static int push_output_configuration(AACDecContext *ac)
{
int pushed = 0;
if (ac->oc[1].status == OC_LOCKED || ac->oc[0].status == OC_NONE) {
ac->oc[0] = ac->oc[1];
pushed = 1;
}
ac->oc[1].status = OC_NONE;
return pushed;
}
/**
* Restore the previous output configuration if and only if the current
* configuration is unlocked.
*/
static void pop_output_configuration(AACDecContext *ac)
{
if (ac->oc[1].status != OC_LOCKED && ac->oc[0].status != OC_NONE) {
ac->oc[1] = ac->oc[0];
ac->avctx->ch_layout = ac->oc[1].ch_layout;
ff_aac_output_configure(ac, ac->oc[1].layout_map, ac->oc[1].layout_map_tags,
ac->oc[1].status, 0);
}
}
/**
* Configure output channel order based on the current program
* configuration element.
*
* @return Returns error status. 0 - OK, !0 - error
*/
int ff_aac_output_configure(AACDecContext *ac,
uint8_t layout_map[MAX_ELEM_ID * 4][3], int tags,
enum OCStatus oc_type, int get_new_frame)
{
AVCodecContext *avctx = ac->avctx;
int i, channels = 0, ret;
uint64_t layout = 0;
uint8_t id_map[TYPE_END][MAX_ELEM_ID] = {{ 0 }};
uint8_t type_counts[TYPE_END] = { 0 };
if (ac->oc[1].layout_map != layout_map) {
memcpy(ac->oc[1].layout_map, layout_map, tags * sizeof(layout_map[0]));
ac->oc[1].layout_map_tags = tags;
}
for (i = 0; i < tags; i++) {
int type = layout_map[i][0];
int id = layout_map[i][1];
id_map[type][id] = type_counts[type]++;
if (id_map[type][id] >= MAX_ELEM_ID) {
avpriv_request_sample(ac->avctx, "Too large remapped id");
return AVERROR_PATCHWELCOME;
}
}
// Try to sniff a reasonable channel order, otherwise output the
// channels in the order the PCE declared them.
if (ac->output_channel_order == CHANNEL_ORDER_DEFAULT)
layout = sniff_channel_order(layout_map, tags);
for (i = 0; i < tags; i++) {
int type = layout_map[i][0];
int id = layout_map[i][1];
int iid = id_map[type][id];
int position = layout_map[i][2];
// Allocate or free elements depending on if they are in the
// current program configuration.
ret = che_configure(ac, position, type, iid, &channels);
if (ret < 0)
return ret;
ac->tag_che_map[type][id] = ac->che[type][iid];
}
if (ac->oc[1].m4ac.ps == 1 && channels == 2) {
if (layout == AV_CH_FRONT_CENTER) {
layout = AV_CH_FRONT_LEFT|AV_CH_FRONT_RIGHT;
} else {
layout = 0;
}
}
av_channel_layout_uninit(&ac->oc[1].ch_layout);
if (layout)
av_channel_layout_from_mask(&ac->oc[1].ch_layout, layout);
else {
ac->oc[1].ch_layout.order = AV_CHANNEL_ORDER_UNSPEC;
ac->oc[1].ch_layout.nb_channels = channels;
}
av_channel_layout_copy(&avctx->ch_layout, &ac->oc[1].ch_layout);
ac->oc[1].status = oc_type;
if (get_new_frame) {
if ((ret = frame_configure_elements(ac->avctx)) < 0)
return ret;
}
return 0;
}
static av_cold void flush(AVCodecContext *avctx)
{
AACDecContext *ac= avctx->priv_data;
int type, i, j;
for (type = 3; type >= 0; type--) {
for (i = 0; i < MAX_ELEM_ID; i++) {
ChannelElement *che = ac->che[type][i];
if (che) {
for (j = 0; j <= 1; j++) {
memset(che->ch[j].saved, 0, sizeof(che->ch[j].saved));
}
}
}
}
ff_aac_usac_reset_state(ac, &ac->oc[1]);
}
/**
* Set up channel positions based on a default channel configuration
* as specified in table 1.17.
*
* @return Returns error status. 0 - OK, !0 - error
*/
int ff_aac_set_default_channel_config(AACDecContext *ac, AVCodecContext *avctx,
uint8_t (*layout_map)[3],
int *tags,
int channel_config)
{
if (channel_config < 1 || (channel_config > 7 && channel_config < 11) ||
channel_config > 14) {
av_log(avctx, AV_LOG_ERROR,
"invalid default channel configuration (%d)\n",
channel_config);
return AVERROR_INVALIDDATA;
}
*tags = ff_tags_per_config[channel_config];
memcpy(layout_map, ff_aac_channel_layout_map[channel_config - 1],
*tags * sizeof(*layout_map));
/*
* AAC specification has 7.1(wide) as a default layout for 8-channel streams.
* However, at least Nero AAC encoder encodes 7.1 streams using the default
* channel config 7, mapping the side channels of the original audio stream
* to the second AAC_CHANNEL_FRONT pair in the AAC stream. Similarly, e.g. FAAD
* decodes the second AAC_CHANNEL_FRONT pair as side channels, therefore decoding
* the incorrect streams as if they were correct (and as the encoder intended).
*
* As actual intended 7.1(wide) streams are very rare, default to assuming a
* 7.1 layout was intended.
*/
if (channel_config == 7 && avctx->strict_std_compliance < FF_COMPLIANCE_STRICT) {
layout_map[2][2] = AAC_CHANNEL_BACK;
if (!ac || !ac->warned_71_wide++) {
av_log(avctx, AV_LOG_INFO, "Assuming an incorrectly encoded 7.1 channel layout"
" instead of a spec-compliant 7.1(wide) layout, use -strict %d to decode"
" according to the specification instead.\n", FF_COMPLIANCE_STRICT);
}
}
return 0;
}
ChannelElement *ff_aac_get_che(AACDecContext *ac, int type, int elem_id)
{
/* For PCE based channel configurations map the channels solely based
* on tags. */
if (!ac->oc[1].m4ac.chan_config) {
return ac->tag_che_map[type][elem_id];
}
// Allow single CPE stereo files to be signalled with mono configuration.
if (!ac->tags_mapped && type == TYPE_CPE &&
ac->oc[1].m4ac.chan_config == 1) {
uint8_t layout_map[MAX_ELEM_ID*4][3];
int layout_map_tags;
push_output_configuration(ac);
av_log(ac->avctx, AV_LOG_DEBUG, "mono with CPE\n");
if (ff_aac_set_default_channel_config(ac, ac->avctx, layout_map,
&layout_map_tags, 2) < 0)
return NULL;
if (ff_aac_output_configure(ac, layout_map, layout_map_tags,
OC_TRIAL_FRAME, 1) < 0)
return NULL;
ac->oc[1].m4ac.chan_config = 2;
ac->oc[1].m4ac.ps = 0;
}
// And vice-versa
if (!ac->tags_mapped && type == TYPE_SCE &&
ac->oc[1].m4ac.chan_config == 2) {
uint8_t layout_map[MAX_ELEM_ID * 4][3];
int layout_map_tags;
push_output_configuration(ac);
av_log(ac->avctx, AV_LOG_DEBUG, "stereo with SCE\n");
layout_map_tags = 2;
layout_map[0][0] = layout_map[1][0] = TYPE_SCE;
layout_map[0][2] = layout_map[1][2] = AAC_CHANNEL_FRONT;
layout_map[0][1] = 0;
layout_map[1][1] = 1;
if (ff_aac_output_configure(ac, layout_map, layout_map_tags,
OC_TRIAL_FRAME, 1) < 0)
return NULL;
if (ac->oc[1].m4ac.sbr)
ac->oc[1].m4ac.ps = -1;
}
/* For indexed channel configurations map the channels solely based
* on position. */
switch (ac->oc[1].m4ac.chan_config) {
case 14:
if (ac->tags_mapped > 2 && ((type == TYPE_CPE && elem_id < 3) ||
(type == TYPE_LFE && elem_id < 1))) {
ac->tags_mapped++;
return ac->tag_che_map[type][elem_id] = ac->che[type][elem_id];
}
case 13:
if (ac->tags_mapped > 3 && ((type == TYPE_CPE && elem_id < 8) ||
(type == TYPE_SCE && elem_id < 6) ||
(type == TYPE_LFE && elem_id < 2))) {
ac->tags_mapped++;
return ac->tag_che_map[type][elem_id] = ac->che[type][elem_id];
}
case 12:
case 7:
if (ac->tags_mapped == 3 && type == TYPE_CPE) {
ac->tags_mapped++;
return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][2];
}
case 11:
if (ac->tags_mapped == 3 && type == TYPE_SCE) {
ac->tags_mapped++;
return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][1];
}
case 6:
/* Some streams incorrectly code 5.1 audio as
* SCE[0] CPE[0] CPE[1] SCE[1]
* instead of
* SCE[0] CPE[0] CPE[1] LFE[0].
* If we seem to have encountered such a stream, transfer
* the LFE[0] element to the SCE[1]'s mapping */
if (ac->tags_mapped == ff_tags_per_config[ac->oc[1].m4ac.chan_config] - 1 && (type == TYPE_LFE || type == TYPE_SCE)) {
if (!ac->warned_remapping_once && (type != TYPE_LFE || elem_id != 0)) {
av_log(ac->avctx, AV_LOG_WARNING,
"This stream seems to incorrectly report its last channel as %s[%d], mapping to LFE[0]\n",
type == TYPE_SCE ? "SCE" : "LFE", elem_id);
ac->warned_remapping_once++;
}
ac->tags_mapped++;
return ac->tag_che_map[type][elem_id] = ac->che[TYPE_LFE][0];
}
case 5:
if (ac->tags_mapped == 2 && type == TYPE_CPE) {
ac->tags_mapped++;
return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][1];
}
case 4:
/* Some streams incorrectly code 4.0 audio as
* SCE[0] CPE[0] LFE[0]
* instead of
* SCE[0] CPE[0] SCE[1].
* If we seem to have encountered such a stream, transfer
* the SCE[1] element to the LFE[0]'s mapping */
if (ac->tags_mapped == ff_tags_per_config[ac->oc[1].m4ac.chan_config] - 1 && (type == TYPE_LFE || type == TYPE_SCE)) {
if (!ac->warned_remapping_once && (type != TYPE_SCE || elem_id != 1)) {
av_log(ac->avctx, AV_LOG_WARNING,
"This stream seems to incorrectly report its last channel as %s[%d], mapping to SCE[1]\n",
type == TYPE_SCE ? "SCE" : "LFE", elem_id);
ac->warned_remapping_once++;
}
ac->tags_mapped++;
return ac->tag_che_map[type][elem_id] = ac->che[TYPE_SCE][1];
}
if (ac->tags_mapped == 2 &&
ac->oc[1].m4ac.chan_config == 4 &&
type == TYPE_SCE) {
ac->tags_mapped++;
return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][1];
}
case 3:
case 2:
if (ac->tags_mapped == (ac->oc[1].m4ac.chan_config != 2) &&
type == TYPE_CPE) {
ac->tags_mapped++;
return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][0];
} else if (ac->tags_mapped == 1 && ac->oc[1].m4ac.chan_config == 2 &&
type == TYPE_SCE) {
ac->tags_mapped++;
return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][1];
}
case 1:
if (!ac->tags_mapped && type == TYPE_SCE) {
ac->tags_mapped++;
return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][0];
}
default:
return NULL;
}
}
/**
* Decode an array of 4 bit element IDs, optionally interleaved with a
* stereo/mono switching bit.
*
* @param type speaker type/position for these channels
*/
static void decode_channel_map(uint8_t layout_map[][3],
enum ChannelPosition type,
GetBitContext *gb, int n)
{
while (n--) {
enum RawDataBlockType syn_ele;
switch (type) {
case AAC_CHANNEL_FRONT:
case AAC_CHANNEL_BACK:
case AAC_CHANNEL_SIDE:
syn_ele = get_bits1(gb);
break;
case AAC_CHANNEL_CC:
skip_bits1(gb);
syn_ele = TYPE_CCE;
break;
case AAC_CHANNEL_LFE:
syn_ele = TYPE_LFE;
break;
default:
// AAC_CHANNEL_OFF has no channel map
av_assert0(0);
}
layout_map[0][0] = syn_ele;
layout_map[0][1] = get_bits(gb, 4);
layout_map[0][2] = type;
layout_map++;
}
}
static inline void relative_align_get_bits(GetBitContext *gb,
int reference_position) {
int n = (reference_position - get_bits_count(gb) & 7);
if (n)
skip_bits(gb, n);
}
/**
* Decode program configuration element; reference: table 4.2.
*
* @return Returns error status. 0 - OK, !0 - error
*/
static int decode_pce(AVCodecContext *avctx, MPEG4AudioConfig *m4ac,
uint8_t (*layout_map)[3],
GetBitContext *gb, int byte_align_ref)
{
int num_front, num_side, num_back, num_lfe, num_assoc_data, num_cc;
int sampling_index;
int comment_len;
int tags;
skip_bits(gb, 2); // object_type
sampling_index = get_bits(gb, 4);
if (m4ac->sampling_index != sampling_index)
av_log(avctx, AV_LOG_WARNING,
"Sample rate index in program config element does not "
"match the sample rate index configured by the container.\n");
num_front = get_bits(gb, 4);
num_side = get_bits(gb, 4);
num_back = get_bits(gb, 4);
num_lfe = get_bits(gb, 2);
num_assoc_data = get_bits(gb, 3);
num_cc = get_bits(gb, 4);
if (get_bits1(gb))
skip_bits(gb, 4); // mono_mixdown_tag
if (get_bits1(gb))
skip_bits(gb, 4); // stereo_mixdown_tag
if (get_bits1(gb))
skip_bits(gb, 3); // mixdown_coeff_index and pseudo_surround
if (get_bits_left(gb) < 5 * (num_front + num_side + num_back + num_cc) + 4 *(num_lfe + num_assoc_data + num_cc)) {
av_log(avctx, AV_LOG_ERROR, "decode_pce: " overread_err);
return -1;
}
decode_channel_map(layout_map , AAC_CHANNEL_FRONT, gb, num_front);
tags = num_front;
decode_channel_map(layout_map + tags, AAC_CHANNEL_SIDE, gb, num_side);
tags += num_side;
decode_channel_map(layout_map + tags, AAC_CHANNEL_BACK, gb, num_back);
tags += num_back;
decode_channel_map(layout_map + tags, AAC_CHANNEL_LFE, gb, num_lfe);
tags += num_lfe;
skip_bits_long(gb, 4 * num_assoc_data);
decode_channel_map(layout_map + tags, AAC_CHANNEL_CC, gb, num_cc);
tags += num_cc;
relative_align_get_bits(gb, byte_align_ref);
/* comment field, first byte is length */
comment_len = get_bits(gb, 8) * 8;
if (get_bits_left(gb) < comment_len) {
av_log(avctx, AV_LOG_ERROR, "decode_pce: " overread_err);
return AVERROR_INVALIDDATA;
}
skip_bits_long(gb, comment_len);
return tags;
}
/**
* Decode GA "General Audio" specific configuration; reference: table 4.1.
*
* @param ac pointer to AACDecContext, may be null
* @param avctx pointer to AVCCodecContext, used for logging
*
* @return Returns error status. 0 - OK, !0 - error
*/
static int decode_ga_specific_config(AACDecContext *ac, AVCodecContext *avctx,
GetBitContext *gb,
int get_bit_alignment,
MPEG4AudioConfig *m4ac,
int channel_config)
{
int extension_flag, ret, ep_config, res_flags;
uint8_t layout_map[MAX_ELEM_ID*4][3];
int tags = 0;
m4ac->frame_length_short = get_bits1(gb);
if (m4ac->frame_length_short && m4ac->sbr == 1) {
avpriv_report_missing_feature(avctx, "SBR with 960 frame length");
if (ac) ac->warned_960_sbr = 1;
m4ac->sbr = 0;
m4ac->ps = 0;
}
if (get_bits1(gb)) // dependsOnCoreCoder
skip_bits(gb, 14); // coreCoderDelay
extension_flag = get_bits1(gb);
if (m4ac->object_type == AOT_AAC_SCALABLE ||
m4ac->object_type == AOT_ER_AAC_SCALABLE)
skip_bits(gb, 3); // layerNr
if (channel_config == 0) {
skip_bits(gb, 4); // element_instance_tag
tags = decode_pce(avctx, m4ac, layout_map, gb, get_bit_alignment);
if (tags < 0)
return tags;
} else {
if ((ret = ff_aac_set_default_channel_config(ac, avctx, layout_map,
&tags, channel_config)))
return ret;
}
if (count_channels(layout_map, tags) > 1) {
m4ac->ps = 0;
} else if (m4ac->sbr == 1 && m4ac->ps == -1)
m4ac->ps = 1;
if (ac && (ret = ff_aac_output_configure(ac, layout_map, tags, OC_GLOBAL_HDR, 0)))
return ret;
if (extension_flag) {
switch (m4ac->object_type) {
case AOT_ER_BSAC:
skip_bits(gb, 5); // numOfSubFrame
skip_bits(gb, 11); // layer_length
break;
case AOT_ER_AAC_LC:
case AOT_ER_AAC_LTP:
case AOT_ER_AAC_SCALABLE:
case AOT_ER_AAC_LD:
res_flags = get_bits(gb, 3);
if (res_flags) {
avpriv_report_missing_feature(avctx,
"AAC data resilience (flags %x)",
res_flags);
return AVERROR_PATCHWELCOME;
}
break;
}
skip_bits1(gb); // extensionFlag3 (TBD in version 3)
}
switch (m4ac->object_type) {
case AOT_ER_AAC_LC:
case AOT_ER_AAC_LTP:
case AOT_ER_AAC_SCALABLE:
case AOT_ER_AAC_LD:
ep_config = get_bits(gb, 2);
if (ep_config) {
avpriv_report_missing_feature(avctx,
"epConfig %d", ep_config);
return AVERROR_PATCHWELCOME;
}
}
return 0;
}
static int decode_eld_specific_config(AACDecContext *ac, AVCodecContext *avctx,
GetBitContext *gb,
MPEG4AudioConfig *m4ac,
int channel_config)
{
int ret, ep_config, res_flags;
uint8_t layout_map[MAX_ELEM_ID*4][3];
int tags = 0;
const int ELDEXT_TERM = 0;
m4ac->ps = 0;
m4ac->sbr = 0;
m4ac->frame_length_short = get_bits1(gb);
res_flags = get_bits(gb, 3);
if (res_flags) {
avpriv_report_missing_feature(avctx,
"AAC data resilience (flags %x)",
res_flags);
return AVERROR_PATCHWELCOME;
}
if (get_bits1(gb)) { // ldSbrPresentFlag
avpriv_report_missing_feature(avctx,
"Low Delay SBR");
return AVERROR_PATCHWELCOME;
}
while (get_bits(gb, 4) != ELDEXT_TERM) {
int len = get_bits(gb, 4);
if (len == 15)
len += get_bits(gb, 8);
if (len == 15 + 255)
len += get_bits(gb, 16);
if (get_bits_left(gb) < len * 8 + 4) {
av_log(avctx, AV_LOG_ERROR, overread_err);
return AVERROR_INVALIDDATA;
}
skip_bits_long(gb, 8 * len);
}
if ((ret = ff_aac_set_default_channel_config(ac, avctx, layout_map,
&tags, channel_config)))
return ret;
if (ac && (ret = ff_aac_output_configure(ac, layout_map, tags, OC_GLOBAL_HDR, 0)))
return ret;
ep_config = get_bits(gb, 2);
if (ep_config) {
avpriv_report_missing_feature(avctx,
"epConfig %d", ep_config);
return AVERROR_PATCHWELCOME;
}
return 0;
}
/**
* Decode audio specific configuration; reference: table 1.13.
*
* @param ac pointer to AACDecContext, may be null
* @param avctx pointer to AVCCodecContext, used for logging
* @param m4ac pointer to MPEG4AudioConfig, used for parsing
* @param gb buffer holding an audio specific config
* @param get_bit_alignment relative alignment for byte align operations
* @param sync_extension look for an appended sync extension
*
* @return Returns error status or number of consumed bits. <0 - error
*/
static int decode_audio_specific_config_gb(AACDecContext *ac,
AVCodecContext *avctx,
OutputConfiguration *oc,
GetBitContext *gb,
int get_bit_alignment,
int sync_extension)
{
int i, ret;
GetBitContext gbc = *gb;
MPEG4AudioConfig *m4ac = &oc->m4ac;
MPEG4AudioConfig m4ac_bak = *m4ac;
if ((i = ff_mpeg4audio_get_config_gb(m4ac, &gbc, sync_extension, avctx)) < 0) {
*m4ac = m4ac_bak;
return AVERROR_INVALIDDATA;
}
if (m4ac->sampling_index > 12) {
av_log(avctx, AV_LOG_ERROR,
"invalid sampling rate index %d\n",
m4ac->sampling_index);
*m4ac = m4ac_bak;
return AVERROR_INVALIDDATA;
}
if (m4ac->object_type == AOT_ER_AAC_LD &&
(m4ac->sampling_index < 3 || m4ac->sampling_index > 7)) {
av_log(avctx, AV_LOG_ERROR,
"invalid low delay sampling rate index %d\n",
m4ac->sampling_index);
*m4ac = m4ac_bak;
return AVERROR_INVALIDDATA;
}
skip_bits_long(gb, i);
switch (m4ac->object_type) {
case AOT_AAC_MAIN:
case AOT_AAC_LC:
case AOT_AAC_SSR:
case AOT_AAC_LTP:
case AOT_ER_AAC_LC:
case AOT_ER_AAC_LD:
if ((ret = decode_ga_specific_config(ac, avctx, gb, get_bit_alignment,
&oc->m4ac, m4ac->chan_config)) < 0)
return ret;
break;
case AOT_ER_AAC_ELD:
if ((ret = decode_eld_specific_config(ac, avctx, gb,
&oc->m4ac, m4ac->chan_config)) < 0)
return ret;
break;
#if CONFIG_AAC_DECODER
case AOT_USAC:
if ((ret = ff_aac_usac_config_decode(ac, avctx, gb,
oc, m4ac->chan_config)) < 0)
return ret;
break;
#endif
default:
avpriv_report_missing_feature(avctx,
"Audio object type %s%d",
m4ac->sbr == 1 ? "SBR+" : "",
m4ac->object_type);
return AVERROR(ENOSYS);
}
ff_dlog(avctx,
"AOT %d chan config %d sampling index %d (%d) SBR %d PS %d\n",
m4ac->object_type, m4ac->chan_config, m4ac->sampling_index,
m4ac->sample_rate, m4ac->sbr,
m4ac->ps);
return get_bits_count(gb);
}
static int decode_audio_specific_config(AACDecContext *ac,
AVCodecContext *avctx,
OutputConfiguration *oc,
const uint8_t *data, int64_t bit_size,
int sync_extension)
{
int i, ret;
GetBitContext gb;
if (bit_size < 0 || bit_size > INT_MAX) {
av_log(avctx, AV_LOG_ERROR, "Audio specific config size is invalid\n");
return AVERROR_INVALIDDATA;
}
ff_dlog(avctx, "audio specific config size %d\n", (int)bit_size >> 3);
for (i = 0; i < bit_size >> 3; i++)
ff_dlog(avctx, "%02x ", data[i]);
ff_dlog(avctx, "\n");
if ((ret = init_get_bits(&gb, data, bit_size)) < 0)
return ret;
return decode_audio_specific_config_gb(ac, avctx, oc, &gb, 0,
sync_extension);
}
static int sample_rate_idx (int rate)
{
if (92017 <= rate) return 0;
else if (75132 <= rate) return 1;
else if (55426 <= rate) return 2;
else if (46009 <= rate) return 3;
else if (37566 <= rate) return 4;
else if (27713 <= rate) return 5;
else if (23004 <= rate) return 6;
else if (18783 <= rate) return 7;
else if (13856 <= rate) return 8;
else if (11502 <= rate) return 9;
else if (9391 <= rate) return 10;
else return 11;
}
static av_cold int decode_close(AVCodecContext *avctx)
{
AACDecContext *ac = avctx->priv_data;
for (int i = 0; i < 2; i++) {
OutputConfiguration *oc = &ac->oc[i];
AACUSACConfig *usac = &oc->usac;
for (int j = 0; j < usac->nb_elems; j++) {
AACUsacElemConfig *ec = &usac->elems[i];
av_freep(&ec->ext.pl_data);
}
}
for (int type = 0; type < FF_ARRAY_ELEMS(ac->che); type++) {
for (int i = 0; i < MAX_ELEM_ID; i++) {
if (ac->che[type][i]) {
ac->proc.sbr_ctx_close(ac->che[type][i]);
av_freep(&ac->che[type][i]);
}
}
}
av_tx_uninit(&ac->mdct96);
av_tx_uninit(&ac->mdct120);
av_tx_uninit(&ac->mdct128);
av_tx_uninit(&ac->mdct480);
av_tx_uninit(&ac->mdct512);
av_tx_uninit(&ac->mdct768);
av_tx_uninit(&ac->mdct960);
av_tx_uninit(&ac->mdct1024);
av_tx_uninit(&ac->mdct_ltp);
// Compiler will optimize this branch away.
if (ac->is_fixed)
av_freep(&ac->RENAME_FIXED(fdsp));
else
av_freep(&ac->fdsp);
return 0;
}
static av_cold int init_dsp(AVCodecContext *avctx)
{
AACDecContext *ac = avctx->priv_data;
int is_fixed = ac->is_fixed, ret;
float scale_fixed, scale_float;
const float *const scalep = is_fixed ? &scale_fixed : &scale_float;
enum AVTXType tx_type = is_fixed ? AV_TX_INT32_MDCT : AV_TX_FLOAT_MDCT;
#define MDCT_INIT(s, fn, len, sval) \
scale_fixed = (sval) * 128.0f; \
scale_float = (sval) / 32768.0f; \
ret = av_tx_init(&s, &fn, tx_type, 1, len, scalep, 0); \
if (ret < 0) \
return ret
MDCT_INIT(ac->mdct96, ac->mdct96_fn, 96, 1.0/96);
MDCT_INIT(ac->mdct120, ac->mdct120_fn, 120, 1.0/120);
MDCT_INIT(ac->mdct128, ac->mdct128_fn, 128, 1.0/128);
MDCT_INIT(ac->mdct480, ac->mdct480_fn, 480, 1.0/480);
MDCT_INIT(ac->mdct512, ac->mdct512_fn, 512, 1.0/512);
MDCT_INIT(ac->mdct768, ac->mdct768_fn, 768, 1.0/768);
MDCT_INIT(ac->mdct960, ac->mdct960_fn, 960, 1.0/960);
MDCT_INIT(ac->mdct1024, ac->mdct1024_fn, 1024, 1.0/1024);
#undef MDCT_INIT
/* LTP forward MDCT */
scale_fixed = -1.0;
scale_float = -32786.0*2 + 36;
ret = av_tx_init(&ac->mdct_ltp, &ac->mdct_ltp_fn, tx_type, 0, 1024, scalep, 0);
if (ret < 0)
return ret;
return 0;
}
av_cold int ff_aac_decode_init(AVCodecContext *avctx)
{
AACDecContext *ac = avctx->priv_data;
int ret;
if (avctx->sample_rate > 96000)
return AVERROR_INVALIDDATA;
ff_aacdec_common_init_once();
ac->avctx = avctx;
ac->oc[1].m4ac.sample_rate = avctx->sample_rate;
if (avctx->extradata_size > 0) {
if ((ret = decode_audio_specific_config(ac, ac->avctx, &ac->oc[1],
avctx->extradata,
avctx->extradata_size * 8LL,
1)) < 0)
return ret;
} else {
int sr, i;
uint8_t layout_map[MAX_ELEM_ID*4][3];
int layout_map_tags;
sr = sample_rate_idx(avctx->sample_rate);
ac->oc[1].m4ac.sampling_index = sr;
ac->oc[1].m4ac.channels = avctx->ch_layout.nb_channels;
ac->oc[1].m4ac.sbr = -1;
ac->oc[1].m4ac.ps = -1;
for (i = 0; i < FF_ARRAY_ELEMS(ff_mpeg4audio_channels); i++)
if (ff_mpeg4audio_channels[i] == avctx->ch_layout.nb_channels)
break;
if (i == FF_ARRAY_ELEMS(ff_mpeg4audio_channels)) {
i = 0;
}
ac->oc[1].m4ac.chan_config = i;
if (ac->oc[1].m4ac.chan_config) {
int ret = ff_aac_set_default_channel_config(ac, avctx, layout_map,
&layout_map_tags,
ac->oc[1].m4ac.chan_config);
if (!ret)
ff_aac_output_configure(ac, layout_map, layout_map_tags,
OC_GLOBAL_HDR, 0);
else if (avctx->err_recognition & AV_EF_EXPLODE)
return AVERROR_INVALIDDATA;
}
}
if (avctx->ch_layout.nb_channels > MAX_CHANNELS) {
av_log(avctx, AV_LOG_ERROR, "Too many channels\n");
return AVERROR_INVALIDDATA;
}
ac->random_state = 0x1f2e3d4c;
return init_dsp(avctx);
}
/**
* Skip data_stream_element; reference: table 4.10.
*/
static int skip_data_stream_element(AACDecContext *ac, GetBitContext *gb)
{
int byte_align = get_bits1(gb);
int count = get_bits(gb, 8);
if (count == 255)
count += get_bits(gb, 8);
if (byte_align)
align_get_bits(gb);
if (get_bits_left(gb) < 8 * count) {
av_log(ac->avctx, AV_LOG_ERROR, "skip_data_stream_element: "overread_err);
return AVERROR_INVALIDDATA;
}
skip_bits_long(gb, 8 * count);
return 0;
}
static int decode_prediction(AACDecContext *ac, IndividualChannelStream *ics,
GetBitContext *gb)
{
int sfb;
if (get_bits1(gb)) {
ics->predictor_reset_group = get_bits(gb, 5);
if (ics->predictor_reset_group == 0 ||
ics->predictor_reset_group > 30) {
av_log(ac->avctx, AV_LOG_ERROR,
"Invalid Predictor Reset Group.\n");
return AVERROR_INVALIDDATA;
}
}
for (sfb = 0; sfb < FFMIN(ics->max_sfb, ff_aac_pred_sfb_max[ac->oc[1].m4ac.sampling_index]); sfb++) {
ics->prediction_used[sfb] = get_bits1(gb);
}
return 0;
}
/**
* Decode Long Term Prediction data; reference: table 4.xx.
*/
static void decode_ltp(AACDecContext *ac, LongTermPrediction *ltp,
GetBitContext *gb, uint8_t max_sfb)
{
int sfb;
ltp->lag = get_bits(gb, 11);
if (CONFIG_AAC_FIXED_DECODER && ac->is_fixed)
ltp->coef_fixed = Q30(ff_ltp_coef[get_bits(gb, 3)]);
else if (CONFIG_AAC_DECODER)
ltp->coef = ff_ltp_coef[get_bits(gb, 3)];
for (sfb = 0; sfb < FFMIN(max_sfb, MAX_LTP_LONG_SFB); sfb++)
ltp->used[sfb] = get_bits1(gb);
}
/**
* Decode Individual Channel Stream info; reference: table 4.6.
*/
static int decode_ics_info(AACDecContext *ac, IndividualChannelStream *ics,
GetBitContext *gb)
{
const MPEG4AudioConfig *const m4ac = &ac->oc[1].m4ac;
const int aot = m4ac->object_type;
const int sampling_index = m4ac->sampling_index;
int ret_fail = AVERROR_INVALIDDATA;
if (aot != AOT_ER_AAC_ELD) {
if (get_bits1(gb)) {
av_log(ac->avctx, AV_LOG_ERROR, "Reserved bit set.\n");
if (ac->avctx->err_recognition & AV_EF_BITSTREAM)
return AVERROR_INVALIDDATA;
}
ics->window_sequence[1] = ics->window_sequence[0];
ics->window_sequence[0] = get_bits(gb, 2);
if (aot == AOT_ER_AAC_LD &&
ics->window_sequence[0] != ONLY_LONG_SEQUENCE) {
av_log(ac->avctx, AV_LOG_ERROR,
"AAC LD is only defined for ONLY_LONG_SEQUENCE but "
"window sequence %d found.\n", ics->window_sequence[0]);
ics->window_sequence[0] = ONLY_LONG_SEQUENCE;
return AVERROR_INVALIDDATA;
}
ics->use_kb_window[1] = ics->use_kb_window[0];
ics->use_kb_window[0] = get_bits1(gb);
}
ics->prev_num_window_groups = FFMAX(ics->num_window_groups, 1);
ics->num_window_groups = 1;
ics->group_len[0] = 1;
if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
int i;
ics->max_sfb = get_bits(gb, 4);
for (i = 0; i < 7; i++) {
if (get_bits1(gb)) {
ics->group_len[ics->num_window_groups - 1]++;
} else {
ics->num_window_groups++;
ics->group_len[ics->num_window_groups - 1] = 1;
}
}
ics->num_windows = 8;
if (m4ac->frame_length_short) {
ics->swb_offset = ff_swb_offset_120[sampling_index];
ics->num_swb = ff_aac_num_swb_120[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_128[sampling_index];
ics->predictor_present = 0;
} else {
ics->max_sfb = get_bits(gb, 6);
ics->num_windows = 1;
if (aot == AOT_ER_AAC_LD || aot == AOT_ER_AAC_ELD) {
if (m4ac->frame_length_short) {
ics->swb_offset = ff_swb_offset_480[sampling_index];
ics->num_swb = ff_aac_num_swb_480[sampling_index];
ics->tns_max_bands = ff_tns_max_bands_480[sampling_index];
} else {
ics->swb_offset = ff_swb_offset_512[sampling_index];
ics->num_swb = ff_aac_num_swb_512[sampling_index];
ics->tns_max_bands = ff_tns_max_bands_512[sampling_index];
}
if (!ics->num_swb || !ics->swb_offset) {
ret_fail = AVERROR_BUG;
goto fail;
}
} else {
if (m4ac->frame_length_short) {
ics->num_swb = ff_aac_num_swb_960[sampling_index];
ics->swb_offset = ff_swb_offset_960[sampling_index];
} else {
ics->num_swb = ff_aac_num_swb_1024[sampling_index];
ics->swb_offset = ff_swb_offset_1024[sampling_index];
}
ics->tns_max_bands = ff_tns_max_bands_1024[sampling_index];
}
if (aot != AOT_ER_AAC_ELD) {
ics->predictor_present = get_bits1(gb);
ics->predictor_reset_group = 0;
}
if (ics->predictor_present) {
if (aot == AOT_AAC_MAIN) {
if (decode_prediction(ac, ics, gb)) {
goto fail;
}
} else if (aot == AOT_AAC_LC ||
aot == AOT_ER_AAC_LC) {
av_log(ac->avctx, AV_LOG_ERROR,
"Prediction is not allowed in AAC-LC.\n");
goto fail;
} else {
if (aot == AOT_ER_AAC_LD) {
av_log(ac->avctx, AV_LOG_ERROR,
"LTP in ER AAC LD not yet implemented.\n");
ret_fail = AVERROR_PATCHWELCOME;
goto fail;
}
if ((ics->ltp.present = get_bits(gb, 1)))
decode_ltp(ac, &ics->ltp, gb, ics->max_sfb);
}
}
}
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);
goto fail;
}
return 0;
fail:
ics->max_sfb = 0;
return ret_fail;
}
/**
* Decode band types (section_data payload); reference: table 4.46.
*
* @param band_type array of the used band type
* @param band_type_run_end array of the last scalefactor band of a band type run
*
* @return Returns error status. 0 - OK, !0 - error
*/
static int decode_band_types(AACDecContext *ac, SingleChannelElement *sce,
GetBitContext *gb)
{
IndividualChannelStream *ics = &sce->ics;
const int bits = (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) ? 3 : 5;
for (int g = 0; g < ics->num_window_groups; g++) {
int k = 0;
while (k < ics->max_sfb) {
uint8_t sect_end = k;
int sect_len_incr;
int sect_band_type = get_bits(gb, 4);
if (sect_band_type == 12) {
av_log(ac->avctx, AV_LOG_ERROR, "invalid band type\n");
return AVERROR_INVALIDDATA;
}
do {
sect_len_incr = get_bits(gb, bits);
sect_end += sect_len_incr;
if (get_bits_left(gb) < 0) {
av_log(ac->avctx, AV_LOG_ERROR, "decode_band_types: "overread_err);
return AVERROR_INVALIDDATA;
}
if (sect_end > ics->max_sfb) {
av_log(ac->avctx, AV_LOG_ERROR,
"Number of bands (%d) exceeds limit (%d).\n",
sect_end, ics->max_sfb);
return AVERROR_INVALIDDATA;
}
} while (sect_len_incr == (1 << bits) - 1);
for (; k < sect_end; k++)
sce->band_type[g*ics->max_sfb + k] = sect_band_type;
}
}
return 0;
}
/**
* Decode scalefactors; reference: table 4.47.
*
* @param global_gain first scalefactor value as scalefactors are differentially coded
* @param band_type array of the used band type
* @param band_type_run_end array of the last scalefactor band of a band type run
* @param sf array of scalefactors or intensity stereo positions
*
* @return Returns error status. 0 - OK, !0 - error
*/
static int decode_scalefactors(AACDecContext *ac, SingleChannelElement *sce,
GetBitContext *gb, unsigned int global_gain)
{
IndividualChannelStream *ics = &sce->ics;
int offset[3] = { global_gain, global_gain - NOISE_OFFSET, 0 };
int clipped_offset;
int noise_flag = 1;
for (int g = 0; g < ics->num_window_groups; g++) {
for (int sfb = 0; sfb < ics->max_sfb; sfb++) {
switch (sce->band_type[g*ics->max_sfb + sfb]) {
case ZERO_BT:
sce->sfo[g*ics->max_sfb + sfb] = 0;
break;
case INTENSITY_BT: /* fallthrough */
case INTENSITY_BT2:
offset[2] += get_vlc2(gb, ff_vlc_scalefactors, 7, 3) - SCALE_DIFF_ZERO;
clipped_offset = av_clip(offset[2], -155, 100);
if (offset[2] != clipped_offset) {
avpriv_request_sample(ac->avctx,
"If you heard an audible artifact, there may be a bug in the decoder. "
"Clipped intensity stereo position (%d -> %d)",
offset[2], clipped_offset);
}
sce->sfo[g*ics->max_sfb + sfb] = clipped_offset - 100;
break;
case NOISE_BT:
if (noise_flag-- > 0)
offset[1] += get_bits(gb, NOISE_PRE_BITS) - NOISE_PRE;
else
offset[1] += get_vlc2(gb, ff_vlc_scalefactors, 7, 3) - SCALE_DIFF_ZERO;
clipped_offset = av_clip(offset[1], -100, 155);
if (offset[1] != clipped_offset) {
avpriv_request_sample(ac->avctx,
"If you heard an audible artifact, there may be a bug in the decoder. "
"Clipped noise gain (%d -> %d)",
offset[1], clipped_offset);
}
sce->sfo[g*ics->max_sfb + sfb] = clipped_offset;
break;
default:
offset[0] += get_vlc2(gb, ff_vlc_scalefactors, 7, 3) - SCALE_DIFF_ZERO;
if (offset[0] > 255U) {
av_log(ac->avctx, AV_LOG_ERROR,
"Scalefactor (%d) out of range.\n", offset[0]);
return AVERROR_INVALIDDATA;
}
sce->sfo[g*ics->max_sfb + sfb] = offset[0] - 100;
break;
}
}
}
return 0;
}
/**
* Decode pulse data; reference: table 4.7.
*/
static int decode_pulses(Pulse *pulse, GetBitContext *gb,
const uint16_t *swb_offset, int num_swb)
{
int i, pulse_swb;
pulse->num_pulse = get_bits(gb, 2) + 1;
pulse_swb = get_bits(gb, 6);
if (pulse_swb >= num_swb)
return -1;
pulse->pos[0] = swb_offset[pulse_swb];
pulse->pos[0] += get_bits(gb, 5);
if (pulse->pos[0] >= swb_offset[num_swb])
return -1;
pulse->amp[0] = get_bits(gb, 4);
for (i = 1; i < pulse->num_pulse; i++) {
pulse->pos[i] = get_bits(gb, 5) + pulse->pos[i - 1];
if (pulse->pos[i] >= swb_offset[num_swb])
return -1;
pulse->amp[i] = get_bits(gb, 4);
}
return 0;
}
/**
* Decode Temporal Noise Shaping data; reference: table 4.48.
*
* @return Returns error status. 0 - OK, !0 - error
*/
int ff_aac_decode_tns(AACDecContext *ac, TemporalNoiseShaping *tns,
GetBitContext *gb, const IndividualChannelStream *ics)
{
int tns_max_order = INT32_MAX;
const int is_usac = ac->oc[1].m4ac.object_type == AOT_USAC;
int w, filt, i, coef_len, coef_res, coef_compress;
const int is8 = ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE;
/* USAC doesn't seem to have a limit */
if (!is_usac)
tns_max_order = is8 ? 7 : ac->oc[1].m4ac.object_type == AOT_AAC_MAIN ? 20 : 12;
for (w = 0; w < ics->num_windows; w++) {
if ((tns->n_filt[w] = get_bits(gb, 2 - is8))) {
coef_res = get_bits1(gb);
for (filt = 0; filt < tns->n_filt[w]; filt++) {
int tmp2_idx;
tns->length[w][filt] = get_bits(gb, 6 - 2 * is8);
if (is_usac)
tns->order[w][filt] = get_bits(gb, 4 - is8);
else
tns->order[w][filt] = get_bits(gb, 5 - (2 * is8));
if (tns->order[w][filt] > tns_max_order) {
av_log(ac->avctx, AV_LOG_ERROR,
"TNS filter order %d is greater than maximum %d.\n",
tns->order[w][filt], tns_max_order);
tns->order[w][filt] = 0;
return AVERROR_INVALIDDATA;
}
if (tns->order[w][filt]) {
tns->direction[w][filt] = get_bits1(gb);
coef_compress = get_bits1(gb);
coef_len = coef_res + 3 - coef_compress;
tmp2_idx = 2 * coef_compress + coef_res;
for (i = 0; i < tns->order[w][filt]; i++) {
if (CONFIG_AAC_FIXED_DECODER && ac->is_fixed)
tns->coef_fixed[w][filt][i] = Q31(ff_tns_tmp2_map[tmp2_idx][get_bits(gb, coef_len)]);
else if (CONFIG_AAC_DECODER)
tns->coef[w][filt][i] = ff_tns_tmp2_map[tmp2_idx][get_bits(gb, coef_len)];
}
}
}
}
}
return 0;
}
/**
* Decode Mid/Side data; reference: table 4.54.
*
* @param ms_present Indicates mid/side stereo presence. [0] mask is all 0s;
* [1] mask is decoded from bitstream; [2] mask is all 1s;
* [3] reserved for scalable AAC
*/
static void decode_mid_side_stereo(ChannelElement *cpe, GetBitContext *gb,
int ms_present)
{
int idx;
int max_idx = cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb;
cpe->max_sfb_ste = cpe->ch[0].ics.max_sfb;
if (ms_present == 1) {
for (idx = 0; idx < max_idx; idx++)
cpe->ms_mask[idx] = get_bits1(gb);
} else if (ms_present == 2) {
memset(cpe->ms_mask, 1, max_idx * sizeof(cpe->ms_mask[0]));
}
}
static void decode_gain_control(SingleChannelElement * sce, GetBitContext * gb)
{
// wd_num, wd_test, aloc_size
static const uint8_t gain_mode[4][3] = {
{1, 0, 5}, // ONLY_LONG_SEQUENCE = 0,
{2, 1, 2}, // LONG_START_SEQUENCE,
{8, 0, 2}, // EIGHT_SHORT_SEQUENCE,
{2, 1, 5}, // LONG_STOP_SEQUENCE
};
const int mode = sce->ics.window_sequence[0];
uint8_t bd, wd, ad;
// FIXME: Store the gain control data on |sce| and do something with it.
uint8_t max_band = get_bits(gb, 2);
for (bd = 0; bd < max_band; bd++) {
for (wd = 0; wd < gain_mode[mode][0]; wd++) {
uint8_t adjust_num = get_bits(gb, 3);
for (ad = 0; ad < adjust_num; ad++) {
skip_bits(gb, 4 + ((wd == 0 && gain_mode[mode][1])
? 4
: gain_mode[mode][2]));
}
}
}
}
/**
* Decode an individual_channel_stream payload; reference: table 4.44.
*
* @param common_window Channels have independent [0], or shared [1], Individual Channel Stream information.
* @param scale_flag scalable [1] or non-scalable [0] AAC (Unused until scalable AAC is implemented.)
*
* @return Returns error status. 0 - OK, !0 - error
*/
int ff_aac_decode_ics(AACDecContext *ac, SingleChannelElement *sce,
GetBitContext *gb, int common_window, int scale_flag)
{
Pulse pulse;
TemporalNoiseShaping *tns = &sce->tns;
IndividualChannelStream *ics = &sce->ics;
int global_gain, eld_syntax, er_syntax, pulse_present = 0;
int ret;
eld_syntax = ac->oc[1].m4ac.object_type == AOT_ER_AAC_ELD;
er_syntax = ac->oc[1].m4ac.object_type == AOT_ER_AAC_LC ||
ac->oc[1].m4ac.object_type == AOT_ER_AAC_LTP ||
ac->oc[1].m4ac.object_type == AOT_ER_AAC_LD ||
ac->oc[1].m4ac.object_type == AOT_ER_AAC_ELD;
/* This assignment is to silence a GCC warning about the variable being used
* uninitialized when in fact it always is.
*/
pulse.num_pulse = 0;
global_gain = get_bits(gb, 8);
if (!common_window && !scale_flag) {
ret = decode_ics_info(ac, ics, gb);
if (ret < 0)
goto fail;
}
if ((ret = decode_band_types(ac, sce, gb)) < 0)
goto fail;
if ((ret = decode_scalefactors(ac, sce, gb, global_gain)) < 0)
goto fail;
ac->dsp.dequant_scalefactors(sce);
pulse_present = 0;
if (!scale_flag) {
if (!eld_syntax && (pulse_present = get_bits1(gb))) {
if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
av_log(ac->avctx, AV_LOG_ERROR,
"Pulse tool not allowed in eight short sequence.\n");
ret = AVERROR_INVALIDDATA;
goto fail;
}
if (decode_pulses(&pulse, gb, ics->swb_offset, ics->num_swb)) {
av_log(ac->avctx, AV_LOG_ERROR,
"Pulse data corrupt or invalid.\n");
ret = AVERROR_INVALIDDATA;
goto fail;
}
}
tns->present = get_bits1(gb);
if (tns->present && !er_syntax) {
ret = ff_aac_decode_tns(ac, tns, gb, ics);
if (ret < 0)
goto fail;
}
if (!eld_syntax && get_bits1(gb)) {
decode_gain_control(sce, gb);
if (!ac->warned_gain_control) {
avpriv_report_missing_feature(ac->avctx, "Gain control");
ac->warned_gain_control = 1;
}
}
// I see no textual basis in the spec for this occurring after SSR gain
// control, but this is what both reference and real implmentations do
if (tns->present && er_syntax) {
ret = ff_aac_decode_tns(ac, tns, gb, ics);
if (ret < 0)
goto fail;
}
}
ret = ac->proc.decode_spectrum_and_dequant(ac, gb,
pulse_present ? &pulse : NULL,
sce);
if (ret < 0)
goto fail;
if (ac->oc[1].m4ac.object_type == AOT_AAC_MAIN && !common_window)
ac->dsp.apply_prediction(ac, sce);
return 0;
fail:
tns->present = 0;
return ret;
}
/**
* Decode a channel_pair_element; reference: table 4.4.
*
* @return Returns error status. 0 - OK, !0 - error
*/
static int decode_cpe(AACDecContext *ac, GetBitContext *gb, ChannelElement *cpe)
{
int i, ret, common_window, ms_present = 0;
int eld_syntax = ac->oc[1].m4ac.object_type == AOT_ER_AAC_ELD;
common_window = eld_syntax || get_bits1(gb);
if (common_window) {
if (decode_ics_info(ac, &cpe->ch[0].ics, gb))
return AVERROR_INVALIDDATA;
i = cpe->ch[1].ics.use_kb_window[0];
cpe->ch[1].ics = cpe->ch[0].ics;
cpe->ch[1].ics.use_kb_window[1] = i;
if (cpe->ch[1].ics.predictor_present &&
(ac->oc[1].m4ac.object_type != AOT_AAC_MAIN))
if ((cpe->ch[1].ics.ltp.present = get_bits(gb, 1)))
decode_ltp(ac, &cpe->ch[1].ics.ltp, gb, cpe->ch[1].ics.max_sfb);
ms_present = get_bits(gb, 2);
if (ms_present == 3) {
av_log(ac->avctx, AV_LOG_ERROR, "ms_present = 3 is reserved.\n");
return AVERROR_INVALIDDATA;
} else if (ms_present)
decode_mid_side_stereo(cpe, gb, ms_present);
}
if ((ret = ff_aac_decode_ics(ac, &cpe->ch[0], gb, common_window, 0)))
return ret;
if ((ret = ff_aac_decode_ics(ac, &cpe->ch[1], gb, common_window, 0)))
return ret;
if (common_window) {
if (ms_present)
ac->dsp.apply_mid_side_stereo(ac, cpe);
if (ac->oc[1].m4ac.object_type == AOT_AAC_MAIN) {
ac->dsp.apply_prediction(ac, &cpe->ch[0]);
ac->dsp.apply_prediction(ac, &cpe->ch[1]);
}
}
ac->dsp.apply_intensity_stereo(ac, cpe, ms_present);
return 0;
}
/**
* Parse whether channels are to be excluded from Dynamic Range Compression; reference: table 4.53.
*
* @return Returns number of bytes consumed.
*/
static int decode_drc_channel_exclusions(DynamicRangeControl *che_drc,
GetBitContext *gb)
{
int i;
int num_excl_chan = 0;
do {
for (i = 0; i < 7; i++)
che_drc->exclude_mask[num_excl_chan++] = get_bits1(gb);
} while (num_excl_chan < MAX_CHANNELS - 7 && get_bits1(gb));
return num_excl_chan / 7;
}
/**
* Decode dynamic range information; reference: table 4.52.
*
* @return Returns number of bytes consumed.
*/
static int decode_dynamic_range(DynamicRangeControl *che_drc,
GetBitContext *gb)
{
int n = 1;
int drc_num_bands = 1;
int i;
/* pce_tag_present? */
if (get_bits1(gb)) {
che_drc->pce_instance_tag = get_bits(gb, 4);
skip_bits(gb, 4); // tag_reserved_bits
n++;
}
/* excluded_chns_present? */
if (get_bits1(gb)) {
n += decode_drc_channel_exclusions(che_drc, gb);
}
/* drc_bands_present? */
if (get_bits1(gb)) {
che_drc->band_incr = get_bits(gb, 4);
che_drc->interpolation_scheme = get_bits(gb, 4);
n++;
drc_num_bands += che_drc->band_incr;
for (i = 0; i < drc_num_bands; i++) {
che_drc->band_top[i] = get_bits(gb, 8);
n++;
}
}
/* prog_ref_level_present? */
if (get_bits1(gb)) {
che_drc->prog_ref_level = get_bits(gb, 7);
skip_bits1(gb); // prog_ref_level_reserved_bits
n++;
}
for (i = 0; i < drc_num_bands; i++) {
che_drc->dyn_rng_sgn[i] = get_bits1(gb);
che_drc->dyn_rng_ctl[i] = get_bits(gb, 7);
n++;
}
return n;
}
static int decode_fill(AACDecContext *ac, GetBitContext *gb, int len) {
uint8_t buf[256];
int i, major, minor;
if (len < 13+7*8)
goto unknown;
get_bits(gb, 13); len -= 13;
for(i=0; i+1<sizeof(buf) && len>=8; i++, len-=8)
buf[i] = get_bits(gb, 8);
buf[i] = 0;
if (ac->avctx->debug & FF_DEBUG_PICT_INFO)
av_log(ac->avctx, AV_LOG_DEBUG, "FILL:%s\n", buf);
if (sscanf(buf, "libfaac %d.%d", &major, &minor) == 2){
ac->avctx->internal->skip_samples = 1024;
}
unknown:
skip_bits_long(gb, len);
return 0;
}
/**
* Decode extension data (incomplete); reference: table 4.51.
*
* @param cnt length of TYPE_FIL syntactic element in bytes
*
* @return Returns number of bytes consumed
*/
static int decode_extension_payload(AACDecContext *ac, GetBitContext *gb, int cnt,
ChannelElement *che, enum RawDataBlockType elem_type)
{
int crc_flag = 0;
int res = cnt;
int type = get_bits(gb, 4);
if (ac->avctx->debug & FF_DEBUG_STARTCODE)
av_log(ac->avctx, AV_LOG_DEBUG, "extension type: %d len:%d\n", type, cnt);
switch (type) { // extension type
case EXT_SBR_DATA_CRC:
crc_flag++;
case EXT_SBR_DATA:
if (!che) {
av_log(ac->avctx, AV_LOG_ERROR, "SBR was found before the first channel element.\n");
return res;
} else if (ac->oc[1].m4ac.frame_length_short) {
if (!ac->warned_960_sbr)
avpriv_report_missing_feature(ac->avctx,
"SBR with 960 frame length");
ac->warned_960_sbr = 1;
skip_bits_long(gb, 8 * cnt - 4);
return res;
} else if (!ac->oc[1].m4ac.sbr) {
av_log(ac->avctx, AV_LOG_ERROR, "SBR signaled to be not-present but was found in the bitstream.\n");
skip_bits_long(gb, 8 * cnt - 4);
return res;
} else if (ac->oc[1].m4ac.sbr == -1 && ac->oc[1].status == OC_LOCKED) {
av_log(ac->avctx, AV_LOG_ERROR, "Implicit SBR was found with a first occurrence after the first frame.\n");
skip_bits_long(gb, 8 * cnt - 4);
return res;
} else if (ac->oc[1].m4ac.ps == -1 && ac->oc[1].status < OC_LOCKED &&
ac->avctx->ch_layout.nb_channels == 1) {
ac->oc[1].m4ac.sbr = 1;
ac->oc[1].m4ac.ps = 1;
ac->avctx->profile = AV_PROFILE_AAC_HE_V2;
ff_aac_output_configure(ac, ac->oc[1].layout_map, ac->oc[1].layout_map_tags,
ac->oc[1].status, 1);
} else {
ac->oc[1].m4ac.sbr = 1;
ac->avctx->profile = AV_PROFILE_AAC_HE;
}
ac->proc.sbr_decode_extension(ac, che, gb, crc_flag, cnt, elem_type);
if (ac->oc[1].m4ac.ps == 1 && !ac->warned_he_aac_mono) {
av_log(ac->avctx, AV_LOG_VERBOSE, "Treating HE-AAC mono as stereo.\n");
ac->warned_he_aac_mono = 1;
}
break;
case EXT_DYNAMIC_RANGE:
res = decode_dynamic_range(&ac->che_drc, gb);
break;
case EXT_FILL:
decode_fill(ac, gb, 8 * cnt - 4);
break;
case EXT_FILL_DATA:
case EXT_DATA_ELEMENT:
default:
skip_bits_long(gb, 8 * cnt - 4);
break;
};
return res;
}
/**
* channel coupling transformation interface
*
* @param apply_coupling_method pointer to (in)dependent coupling function
*/
static void apply_channel_coupling(AACDecContext *ac, ChannelElement *cc,
enum RawDataBlockType type, int elem_id,
enum CouplingPoint coupling_point,
void (*apply_coupling_method)(AACDecContext *ac, SingleChannelElement *target, ChannelElement *cce, int index))
{
int i, c;
for (i = 0; i < MAX_ELEM_ID; i++) {
ChannelElement *cce = ac->che[TYPE_CCE][i];
int index = 0;
if (cce && cce->coup.coupling_point == coupling_point) {
ChannelCoupling *coup = &cce->coup;
for (c = 0; c <= coup->num_coupled; c++) {
if (coup->type[c] == type && coup->id_select[c] == elem_id) {
if (coup->ch_select[c] != 1) {
apply_coupling_method(ac, &cc->ch[0], cce, index);
if (coup->ch_select[c] != 0)
index++;
}
if (coup->ch_select[c] != 2)
apply_coupling_method(ac, &cc->ch[1], cce, index++);
} else
index += 1 + (coup->ch_select[c] == 3);
}
}
}
}
/**
* Convert spectral data to samples, applying all supported tools as appropriate.
*/
static void spectral_to_sample(AACDecContext *ac, int samples)
{
int i, type;
void (*imdct_and_window)(AACDecContext *ac, SingleChannelElement *sce);
switch (ac->oc[1].m4ac.object_type) {
case AOT_ER_AAC_LD:
imdct_and_window = ac->dsp.imdct_and_windowing_ld;
break;
case AOT_ER_AAC_ELD:
imdct_and_window = ac->dsp.imdct_and_windowing_eld;
break;
default:
if (ac->oc[1].m4ac.frame_length_short)
imdct_and_window = ac->dsp.imdct_and_windowing_960;
else
imdct_and_window = ac->dsp.imdct_and_windowing;
}
for (type = 3; type >= 0; type--) {
for (i = 0; i < MAX_ELEM_ID; i++) {
ChannelElement *che = ac->che[type][i];
if (che && che->present) {
if (type <= TYPE_CPE)
apply_channel_coupling(ac, che, type, i, BEFORE_TNS, ac->dsp.apply_dependent_coupling);
if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP) {
if (che->ch[0].ics.predictor_present) {
if (che->ch[0].ics.ltp.present)
ac->dsp.apply_ltp(ac, &che->ch[0]);
if (che->ch[1].ics.ltp.present && type == TYPE_CPE)
ac->dsp.apply_ltp(ac, &che->ch[1]);
}
}
if (che->ch[0].tns.present)
ac->dsp.apply_tns(che->ch[0].coeffs,
&che->ch[0].tns, &che->ch[0].ics, 1);
if (che->ch[1].tns.present)
ac->dsp.apply_tns(che->ch[1].coeffs,
&che->ch[1].tns, &che->ch[1].ics, 1);
if (type <= TYPE_CPE)
apply_channel_coupling(ac, che, type, i, BETWEEN_TNS_AND_IMDCT, ac->dsp.apply_dependent_coupling);
if (type != TYPE_CCE || che->coup.coupling_point == AFTER_IMDCT) {
imdct_and_window(ac, &che->ch[0]);
if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP)
ac->dsp.update_ltp(ac, &che->ch[0]);
if (type == TYPE_CPE) {
imdct_and_window(ac, &che->ch[1]);
if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP)
ac->dsp.update_ltp(ac, &che->ch[1]);
}
if (ac->oc[1].m4ac.sbr > 0) {
ac->proc.sbr_apply(ac, che, type,
che->ch[0].output,
che->ch[1].output);
}
}
if (type <= TYPE_CCE)
apply_channel_coupling(ac, che, type, i, AFTER_IMDCT, ac->dsp.apply_independent_coupling);
ac->dsp.clip_output(ac, che, type, samples);
che->present = 0;
} else if (che) {
av_log(ac->avctx, AV_LOG_VERBOSE, "ChannelElement %d.%d missing \n", type, i);
}
}
}
}
static int parse_adts_frame_header(AACDecContext *ac, GetBitContext *gb)
{
int size;
AACADTSHeaderInfo hdr_info;
uint8_t layout_map[MAX_ELEM_ID*4][3];
int layout_map_tags, ret;
size = ff_adts_header_parse(gb, &hdr_info);
if (size > 0) {
if (!ac->warned_num_aac_frames && hdr_info.num_aac_frames != 1) {
// This is 2 for "VLB " audio in NSV files.
// See samples/nsv/vlb_audio.
avpriv_report_missing_feature(ac->avctx,
"More than one AAC RDB per ADTS frame");
ac->warned_num_aac_frames = 1;
}
push_output_configuration(ac);
if (hdr_info.chan_config) {
ac->oc[1].m4ac.chan_config = hdr_info.chan_config;
if ((ret = ff_aac_set_default_channel_config(ac, ac->avctx,
layout_map,
&layout_map_tags,
hdr_info.chan_config)) < 0)
return ret;
if ((ret = ff_aac_output_configure(ac, layout_map, layout_map_tags,
FFMAX(ac->oc[1].status,
OC_TRIAL_FRAME), 0)) < 0)
return ret;
} else {
ac->oc[1].m4ac.chan_config = 0;
/**
* dual mono frames in Japanese DTV can have chan_config 0
* WITHOUT specifying PCE.
* thus, set dual mono as default.
*/
if (ac->dmono_mode && ac->oc[0].status == OC_NONE) {
layout_map_tags = 2;
layout_map[0][0] = layout_map[1][0] = TYPE_SCE;
layout_map[0][2] = layout_map[1][2] = AAC_CHANNEL_FRONT;
layout_map[0][1] = 0;
layout_map[1][1] = 1;
if (ff_aac_output_configure(ac, layout_map, layout_map_tags,
OC_TRIAL_FRAME, 0))
return -7;
}
}
ac->oc[1].m4ac.sample_rate = hdr_info.sample_rate;
ac->oc[1].m4ac.sampling_index = hdr_info.sampling_index;
ac->oc[1].m4ac.object_type = hdr_info.object_type;
ac->oc[1].m4ac.frame_length_short = 0;
if (ac->oc[0].status != OC_LOCKED ||
ac->oc[0].m4ac.chan_config != hdr_info.chan_config ||
ac->oc[0].m4ac.sample_rate != hdr_info.sample_rate) {
ac->oc[1].m4ac.sbr = -1;
ac->oc[1].m4ac.ps = -1;
}
if (!hdr_info.crc_absent)
skip_bits(gb, 16);
}
return size;
}
static int aac_decode_er_frame(AVCodecContext *avctx, AVFrame *frame,
int *got_frame_ptr, GetBitContext *gb)
{
AACDecContext *ac = avctx->priv_data;
const MPEG4AudioConfig *const m4ac = &ac->oc[1].m4ac;
ChannelElement *che;
int err, i;
int samples = m4ac->frame_length_short ? 960 : 1024;
int chan_config = m4ac->chan_config;
int aot = m4ac->object_type;
if (aot == AOT_ER_AAC_LD || aot == AOT_ER_AAC_ELD)
samples >>= 1;
ac->frame = frame;
if ((err = frame_configure_elements(avctx)) < 0)
return err;
// The AV_PROFILE_AAC_* defines are all object_type - 1
// This may lead to an undefined profile being signaled
ac->avctx->profile = aot - 1;
ac->tags_mapped = 0;
if (chan_config < 0 || (chan_config >= 8 && chan_config < 11) || chan_config >= 13) {
avpriv_request_sample(avctx, "Unknown ER channel configuration %d",
chan_config);
return AVERROR_INVALIDDATA;
}
for (i = 0; i < ff_tags_per_config[chan_config]; i++) {
const int elem_type = ff_aac_channel_layout_map[chan_config-1][i][0];
const int elem_id = ff_aac_channel_layout_map[chan_config-1][i][1];
if (!(che=ff_aac_get_che(ac, elem_type, elem_id))) {
av_log(ac->avctx, AV_LOG_ERROR,
"channel element %d.%d is not allocated\n",
elem_type, elem_id);
return AVERROR_INVALIDDATA;
}
che->present = 1;
if (aot != AOT_ER_AAC_ELD)
skip_bits(gb, 4);
switch (elem_type) {
case TYPE_SCE:
err = ff_aac_decode_ics(ac, &che->ch[0], gb, 0, 0);
break;
case TYPE_CPE:
err = decode_cpe(ac, gb, che);
break;
case TYPE_LFE:
err = ff_aac_decode_ics(ac, &che->ch[0], gb, 0, 0);
break;
}
if (err < 0)
return err;
}
spectral_to_sample(ac, samples);
if (!ac->frame->data[0] && samples) {
av_log(avctx, AV_LOG_ERROR, "no frame data found\n");
return AVERROR_INVALIDDATA;
}
ac->frame->nb_samples = samples;
ac->frame->sample_rate = avctx->sample_rate;
*got_frame_ptr = 1;
skip_bits_long(gb, get_bits_left(gb));
return 0;
}
static int decode_frame_ga(AVCodecContext *avctx, AACDecContext *ac,
GetBitContext *gb, int *got_frame_ptr)
{
int err;
int is_dmono;
int elem_id;
enum RawDataBlockType elem_type, che_prev_type = TYPE_END;
uint8_t che_presence[4][MAX_ELEM_ID] = {{0}};
ChannelElement *che = NULL, *che_prev = NULL;
int samples = 0, multiplier, audio_found = 0, pce_found = 0, sce_count = 0;
AVFrame *frame = ac->frame;
int payload_alignment = get_bits_count(gb);
// parse
while ((elem_type = get_bits(gb, 3)) != TYPE_END) {
elem_id = get_bits(gb, 4);
if (avctx->debug & FF_DEBUG_STARTCODE)
av_log(avctx, AV_LOG_DEBUG, "Elem type:%x id:%x\n", elem_type, elem_id);
if (!avctx->ch_layout.nb_channels && elem_type != TYPE_PCE)
return AVERROR_INVALIDDATA;
if (elem_type < TYPE_DSE) {
if (che_presence[elem_type][elem_id]) {
int error = che_presence[elem_type][elem_id] > 1;
av_log(ac->avctx, error ? AV_LOG_ERROR : AV_LOG_DEBUG, "channel element %d.%d duplicate\n",
elem_type, elem_id);
if (error)
return AVERROR_INVALIDDATA;
}
che_presence[elem_type][elem_id]++;
if (!(che=ff_aac_get_che(ac, elem_type, elem_id))) {
av_log(ac->avctx, AV_LOG_ERROR, "channel element %d.%d is not allocated\n",
elem_type, elem_id);
return AVERROR_INVALIDDATA;
}
samples = ac->oc[1].m4ac.frame_length_short ? 960 : 1024;
che->present = 1;
}
switch (elem_type) {
case TYPE_SCE:
err = ff_aac_decode_ics(ac, &che->ch[0], gb, 0, 0);
audio_found = 1;
sce_count++;
break;
case TYPE_CPE:
err = decode_cpe(ac, gb, che);
audio_found = 1;
break;
case TYPE_CCE:
err = ac->proc.decode_cce(ac, gb, che);
break;
case TYPE_LFE:
err = ff_aac_decode_ics(ac, &che->ch[0], gb, 0, 0);
audio_found = 1;
break;
case TYPE_DSE:
err = skip_data_stream_element(ac, gb);
break;
case TYPE_PCE: {
uint8_t layout_map[MAX_ELEM_ID*4][3] = {{0}};
int tags;
int pushed = push_output_configuration(ac);
if (pce_found && !pushed)
return AVERROR_INVALIDDATA;
tags = decode_pce(avctx, &ac->oc[1].m4ac, layout_map, gb,
payload_alignment);
if (tags < 0) {
err = tags;
break;
}
if (pce_found) {
av_log(avctx, AV_LOG_ERROR,
"Not evaluating a further program_config_element as this construct is dubious at best.\n");
pop_output_configuration(ac);
} else {
err = ff_aac_output_configure(ac, layout_map, tags, OC_TRIAL_PCE, 1);
if (!err)
ac->oc[1].m4ac.chan_config = 0;
pce_found = 1;
}
break;
}
case TYPE_FIL:
if (elem_id == 15)
elem_id += get_bits(gb, 8) - 1;
if (get_bits_left(gb) < 8 * elem_id) {
av_log(avctx, AV_LOG_ERROR, "TYPE_FIL: "overread_err);
return AVERROR_INVALIDDATA;
}
err = 0;
while (elem_id > 0) {
int ret = decode_extension_payload(ac, gb, elem_id, che_prev, che_prev_type);
if (ret < 0) {
err = ret;
break;
}
elem_id -= ret;
}
break;
default:
err = AVERROR_BUG; /* should not happen, but keeps compiler happy */
break;
}
if (elem_type < TYPE_DSE) {
che_prev = che;
che_prev_type = elem_type;
}
if (err)
return err;
if (get_bits_left(gb) < 3) {
av_log(avctx, AV_LOG_ERROR, overread_err);
return AVERROR_INVALIDDATA;
}
}
if (!avctx->ch_layout.nb_channels)
return 0;
multiplier = (ac->oc[1].m4ac.sbr == 1) ? ac->oc[1].m4ac.ext_sample_rate > ac->oc[1].m4ac.sample_rate : 0;
samples <<= multiplier;
spectral_to_sample(ac, samples);
if (ac->oc[1].status && audio_found) {
avctx->sample_rate = ac->oc[1].m4ac.sample_rate << multiplier;
avctx->frame_size = samples;
ac->oc[1].status = OC_LOCKED;
}
if (!ac->frame->data[0] && samples) {
av_log(avctx, AV_LOG_ERROR, "no frame data found\n");
return AVERROR_INVALIDDATA;
}
if (samples) {
ac->frame->nb_samples = samples;
ac->frame->sample_rate = avctx->sample_rate;
*got_frame_ptr = 1;
} else {
av_frame_unref(ac->frame);
*got_frame_ptr = 0;
}
/* for dual-mono audio (SCE + SCE) */
is_dmono = ac->dmono_mode && sce_count == 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;
}
static int aac_decode_frame_int(AVCodecContext *avctx, AVFrame *frame,
int *got_frame_ptr, GetBitContext *gb,
const AVPacket *avpkt)
{
int err;
AACDecContext *ac = avctx->priv_data;
ac->frame = frame;
*got_frame_ptr = 0;
if (show_bits(gb, 12) == 0xfff) {
if ((err = parse_adts_frame_header(ac, gb)) < 0) {
av_log(avctx, AV_LOG_ERROR, "Error decoding AAC frame header.\n");
goto fail;
}
if (ac->oc[1].m4ac.sampling_index > 12) {
av_log(ac->avctx, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->oc[1].m4ac.sampling_index);
err = AVERROR_INVALIDDATA;
goto fail;
}
}
if ((err = frame_configure_elements(avctx)) < 0)
goto fail;
// The AV_PROFILE_AAC_* defines are all object_type - 1
// This may lead to an undefined profile being signaled
ac->avctx->profile = ac->oc[1].m4ac.object_type - 1;
ac->tags_mapped = 0;
if (ac->oc[1].m4ac.object_type == AOT_USAC) {
if (ac->is_fixed) {
avpriv_report_missing_feature(ac->avctx,
"AAC USAC fixed-point decoding");
return AVERROR_PATCHWELCOME;
}
#if CONFIG_AAC_DECODER
err = ff_aac_usac_decode_frame(avctx, ac, gb, got_frame_ptr);
if (err < 0)
goto fail;
#endif
} else {
err = decode_frame_ga(avctx, ac, gb, got_frame_ptr);
if (err < 0)
goto fail;
}
return err;
fail:
pop_output_configuration(ac);
return err;
}
static int aac_decode_frame(AVCodecContext *avctx, AVFrame *frame,
int *got_frame_ptr, AVPacket *avpkt)
{
AACDecContext *ac = avctx->priv_data;
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
GetBitContext gb;
int buf_consumed;
int buf_offset;
int err;
size_t new_extradata_size;
const uint8_t *new_extradata = av_packet_get_side_data(avpkt,
AV_PKT_DATA_NEW_EXTRADATA,
&new_extradata_size);
size_t jp_dualmono_size;
const uint8_t *jp_dualmono = av_packet_get_side_data(avpkt,
AV_PKT_DATA_JP_DUALMONO,
&jp_dualmono_size);
if (new_extradata) {
/* discard previous configuration */
ac->oc[1].status = OC_NONE;
err = decode_audio_specific_config(ac, ac->avctx, &ac->oc[1],
new_extradata,
new_extradata_size * 8LL, 1);
if (err < 0) {
return err;
}
}
ac->dmono_mode = 0;
if (jp_dualmono && jp_dualmono_size > 0)
ac->dmono_mode = 1 + *jp_dualmono;
if (ac->force_dmono_mode >= 0)
ac->dmono_mode = ac->force_dmono_mode;
if (INT_MAX / 8 <= buf_size)
return AVERROR_INVALIDDATA;
if ((err = init_get_bits8(&gb, buf, buf_size)) < 0)
return err;
switch (ac->oc[1].m4ac.object_type) {
case AOT_ER_AAC_LC:
case AOT_ER_AAC_LTP:
case AOT_ER_AAC_LD:
case AOT_ER_AAC_ELD:
err = aac_decode_er_frame(avctx, frame, got_frame_ptr, &gb);
break;
default:
err = aac_decode_frame_int(avctx, frame, got_frame_ptr, &gb, avpkt);
}
if (err < 0)
return err;
buf_consumed = (get_bits_count(&gb) + 7) >> 3;
for (buf_offset = buf_consumed; buf_offset < buf_size; buf_offset++)
if (buf[buf_offset])
break;
return buf_size > buf_offset ? buf_consumed : buf_size;
}
#if CONFIG_AAC_LATM_DECODER
#include "aacdec_latm.h"
#endif
#define AACDEC_FLAGS AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM
#define OFF(field) offsetof(AACDecContext, field)
static const AVOption options[] = {
/**
* AVOptions for Japanese DTV specific extensions (ADTS only)
*/
{"dual_mono_mode", "Select the channel to decode for dual mono",
OFF(force_dmono_mode), AV_OPT_TYPE_INT, {.i64=-1}, -1, 2,
AACDEC_FLAGS, .unit = "dual_mono_mode"},
{"auto", "autoselection", 0, AV_OPT_TYPE_CONST, {.i64=-1}, INT_MIN, INT_MAX, AACDEC_FLAGS, .unit = "dual_mono_mode"},
{"main", "Select Main/Left channel", 0, AV_OPT_TYPE_CONST, {.i64= 1}, INT_MIN, INT_MAX, AACDEC_FLAGS, .unit = "dual_mono_mode"},
{"sub" , "Select Sub/Right channel", 0, AV_OPT_TYPE_CONST, {.i64= 2}, INT_MIN, INT_MAX, AACDEC_FLAGS, .unit = "dual_mono_mode"},
{"both", "Select both channels", 0, AV_OPT_TYPE_CONST, {.i64= 0}, INT_MIN, INT_MAX, AACDEC_FLAGS, .unit = "dual_mono_mode"},
{ "channel_order", "Order in which the channels are to be exported",
OFF(output_channel_order), AV_OPT_TYPE_INT,
{ .i64 = CHANNEL_ORDER_DEFAULT }, 0, 1, AACDEC_FLAGS, .unit = "channel_order" },
{ "default", "normal libavcodec channel order", 0, AV_OPT_TYPE_CONST,
{ .i64 = CHANNEL_ORDER_DEFAULT }, .flags = AACDEC_FLAGS, .unit = "channel_order" },
{ "coded", "order in which the channels are coded in the bitstream",
0, AV_OPT_TYPE_CONST, { .i64 = CHANNEL_ORDER_CODED }, .flags = AACDEC_FLAGS, .unit = "channel_order" },
{NULL},
};
static const AVClass decoder_class = {
.class_name = "AAC decoder",
.item_name = av_default_item_name,
.option = options,
.version = LIBAVUTIL_VERSION_INT,
};
#if CONFIG_AAC_DECODER
const FFCodec ff_aac_decoder = {
.p.name = "aac",
CODEC_LONG_NAME("AAC (Advanced Audio Coding)"),
.p.type = AVMEDIA_TYPE_AUDIO,
.p.id = AV_CODEC_ID_AAC,
.p.priv_class = &decoder_class,
.priv_data_size = sizeof(AACDecContext),
.init = ff_aac_decode_init_float,
.close = decode_close,
FF_CODEC_DECODE_CB(aac_decode_frame),
.p.sample_fmts = (const enum AVSampleFormat[]) {
AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE
},
.p.capabilities = AV_CODEC_CAP_CHANNEL_CONF | AV_CODEC_CAP_DR1,
.caps_internal = FF_CODEC_CAP_INIT_CLEANUP,
.p.ch_layouts = ff_aac_ch_layout,
.flush = flush,
.p.profiles = NULL_IF_CONFIG_SMALL(ff_aac_profiles),
};
#endif
#if CONFIG_AAC_FIXED_DECODER
const FFCodec ff_aac_fixed_decoder = {
.p.name = "aac_fixed",
CODEC_LONG_NAME("AAC (Advanced Audio Coding)"),
.p.type = AVMEDIA_TYPE_AUDIO,
.p.id = AV_CODEC_ID_AAC,
.p.priv_class = &decoder_class,
.priv_data_size = sizeof(AACDecContext),
.init = ff_aac_decode_init_fixed,
.close = decode_close,
FF_CODEC_DECODE_CB(aac_decode_frame),
.p.sample_fmts = (const enum AVSampleFormat[]) {
AV_SAMPLE_FMT_S32P, AV_SAMPLE_FMT_NONE
},
.p.capabilities = AV_CODEC_CAP_CHANNEL_CONF | AV_CODEC_CAP_DR1,
.caps_internal = FF_CODEC_CAP_INIT_CLEANUP,
.p.ch_layouts = ff_aac_ch_layout,
.p.profiles = NULL_IF_CONFIG_SMALL(ff_aac_profiles),
.flush = flush,
};
#endif
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