/*
 * 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]);
        memset(ac->output_element, 0, sizeof(ac->output_element));
    }
    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));
                }
            }
        }
    }

#if CONFIG_AAC_DECODER
    ff_aac_usac_reset_state(ac, &ac->oc[1]);
#endif
}

/**
 * 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 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[j];
            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 = ff_aac_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;
    ac->frame->flags |= AV_FRAME_FLAG_KEY;
    *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;
        ac->frame->flags |= AV_FRAME_FLAG_KEY;
        *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
