aacdec_template.c 115 KB
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/*
 * AAC decoder
 * 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>
 *
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 * AAC decoder fixed-point implementation
 * Copyright (c) 2013
 *      MIPS Technologies, Inc., California.
 *
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 * 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
 */

/**
 * @file
 * AAC decoder
 * @author Oded Shimon  ( ods15 ods15 dyndns org )
 * @author Maxim Gavrilov ( maxim.gavrilov gmail com )
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 *
 * AAC decoder fixed-point implementation
 * @author Stanislav Ocovaj ( stanislav.ocovaj imgtec com )
 * @author Nedeljko Babic ( nedeljko.babic imgtec com )
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 */

/*
 * 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.
 */

static VLC vlc_scalefactors;
static VLC vlc_spectral[11];

static int output_configure(AACContext *ac,
                            uint8_t layout_map[MAX_ELEM_ID*4][3], int tags,
                            enum OCStatus oc_type, int get_new_frame);

#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(AACContext *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]) {
            if (!(ac->che[type][id] = av_mallocz(sizeof(ChannelElement))))
                return AVERROR(ENOMEM);
            ff_aac_sbr_ctx_init(ac, &ac->che[type][id]->sbr);
        }
        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])
            ff_aac_sbr_ctx_close(&ac->che[type][id]->sbr);
        av_freep(&ac->che[type][id]);
    }
    return 0;
}

static int frame_configure_elements(AVCodecContext *avctx)
{
    AACContext *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].ret = che->ch[0].ret_buf;
                che->ch[1].ret = che->ch[1].ret_buf;
            }
        }
    }

    /* get output buffer */
    av_frame_unref(ac->frame);
    if (!avctx->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->channels; ch++) {
        if (ac->output_element[ch])
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            ac->output_element[ch]->ret = (INTFLOAT *)ac->frame->extended_data[ch];
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    }

    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)
{
    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
        };
        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
        };
        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 ^= 1;
        }
    }
    if (sce_parity &&
        ((pos == AAC_CHANNEL_FRONT && first_cpe) || pos == AAC_CHANNEL_SIDE))
        return -1;
    *current = i;
    return num_pos_channels;
}

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 } };
    int num_front_channels, num_side_channels, num_back_channels;
    uint64_t layout;

    if (FF_ARRAY_ELEMS(e2c_vec) < tags)
        return 0;

    i = 0;
    num_front_channels =
        count_paired_channels(layout_map, tags, AAC_CHANNEL_FRONT, &i);
    if (num_front_channels < 0)
        return 0;
    num_side_channels =
        count_paired_channels(layout_map, tags, AAC_CHANNEL_SIDE, &i);
    if (num_side_channels < 0)
        return 0;
    num_back_channels =
        count_paired_channels(layout_map, tags, AAC_CHANNEL_BACK, &i);
    if (num_back_channels < 0)
        return 0;

    if (num_side_channels == 0 && num_back_channels >= 4) {
        num_side_channels = 2;
        num_back_channels -= 2;
    }

    i = 0;
    if (num_front_channels & 1) {
        e2c_vec[i] = (struct elem_to_channel) {
            .av_position  = AV_CH_FRONT_CENTER,
            .syn_ele      = TYPE_SCE,
            .elem_id      = layout_map[i][1],
            .aac_position = AAC_CHANNEL_FRONT
        };
        i++;
        num_front_channels--;
    }
    if (num_front_channels >= 4) {
        i += assign_pair(e2c_vec, layout_map, i,
                         AV_CH_FRONT_LEFT_OF_CENTER,
                         AV_CH_FRONT_RIGHT_OF_CENTER,
                         AAC_CHANNEL_FRONT);
        num_front_channels -= 2;
    }
    if (num_front_channels >= 2) {
        i += assign_pair(e2c_vec, layout_map, i,
                         AV_CH_FRONT_LEFT,
                         AV_CH_FRONT_RIGHT,
                         AAC_CHANNEL_FRONT);
        num_front_channels -= 2;
    }
    while (num_front_channels >= 2) {
        i += assign_pair(e2c_vec, layout_map, i,
                         UINT64_MAX,
                         UINT64_MAX,
                         AAC_CHANNEL_FRONT);
        num_front_channels -= 2;
    }

    if (num_side_channels >= 2) {
        i += assign_pair(e2c_vec, layout_map, i,
                         AV_CH_SIDE_LEFT,
                         AV_CH_SIDE_RIGHT,
                         AAC_CHANNEL_FRONT);
        num_side_channels -= 2;
    }
    while (num_side_channels >= 2) {
        i += assign_pair(e2c_vec, layout_map, i,
                         UINT64_MAX,
                         UINT64_MAX,
                         AAC_CHANNEL_SIDE);
        num_side_channels -= 2;
    }

    while (num_back_channels >= 4) {
        i += assign_pair(e2c_vec, layout_map, i,
                         UINT64_MAX,
                         UINT64_MAX,
                         AAC_CHANNEL_BACK);
        num_back_channels -= 2;
    }
    if (num_back_channels >= 2) {
        i += assign_pair(e2c_vec, layout_map, i,
                         AV_CH_BACK_LEFT,
                         AV_CH_BACK_RIGHT,
                         AAC_CHANNEL_BACK);
        num_back_channels -= 2;
    }
    if (num_back_channels) {
        e2c_vec[i] = (struct elem_to_channel) {
            .av_position  = AV_CH_BACK_CENTER,
            .syn_ele      = TYPE_SCE,
            .elem_id      = layout_map[i][1],
            .aac_position = AAC_CHANNEL_BACK
        };
        i++;
        num_back_channels--;
    }

    if (i < tags && layout_map[i][2] == AAC_CHANNEL_LFE) {
        e2c_vec[i] = (struct elem_to_channel) {
            .av_position  = AV_CH_LOW_FREQUENCY,
            .syn_ele      = TYPE_LFE,
            .elem_id      = layout_map[i][1],
            .aac_position = AAC_CHANNEL_LFE
        };
        i++;
    }
    while (i < tags && layout_map[i][2] == AAC_CHANNEL_LFE) {
        e2c_vec[i] = (struct elem_to_channel) {
            .av_position  = UINT64_MAX,
            .syn_ele      = TYPE_LFE,
            .elem_id      = layout_map[i][1],
            .aac_position = AAC_CHANNEL_LFE
        };
        i++;
    }

    // Must choose a stable sort
    total_non_cc_elements = n = i;
    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);

    layout = 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;
        if (e2c_vec[i].av_position != UINT64_MAX) {
            layout |= e2c_vec[i].av_position;
        }
    }

    return layout;
}

/**
 * Save current output configuration if and only if it has been locked.
 */
static void push_output_configuration(AACContext *ac) {
    if (ac->oc[1].status == OC_LOCKED || ac->oc[0].status == OC_NONE) {
        ac->oc[0] = ac->oc[1];
    }
    ac->oc[1].status = OC_NONE;
}

/**
 * Restore the previous output configuration if and only if the current
 * configuration is unlocked.
 */
static void pop_output_configuration(AACContext *ac) {
    if (ac->oc[1].status != OC_LOCKED && ac->oc[0].status != OC_NONE) {
        ac->oc[1] = ac->oc[0];
        ac->avctx->channels = ac->oc[1].channels;
        ac->avctx->channel_layout = ac->oc[1].channel_layout;
        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
 */
static int output_configure(AACContext *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]++;
    }
    // Try to sniff a reasonable channel order, otherwise output the
    // channels in the order the PCE declared them.
    if (avctx->request_channel_layout != AV_CH_LAYOUT_NATIVE)
        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;
        }
    }

    if (layout) avctx->channel_layout = layout;
                            ac->oc[1].channel_layout = layout;
    avctx->channels       = ac->oc[1].channels       = channels;
    ac->oc[1].status = oc_type;

    if (get_new_frame) {
        if ((ret = frame_configure_elements(ac->avctx)) < 0)
            return ret;
    }

    return 0;
}

static void flush(AVCodecContext *avctx)
{
    AACContext *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));
                }
            }
        }
    }
}

/**
 * Set up channel positions based on a default channel configuration
 * as specified in table 1.17.
 *
 * @return  Returns error status. 0 - OK, !0 - error
 */
static int set_default_channel_config(AVCodecContext *avctx,
                                      uint8_t (*layout_map)[3],
                                      int *tags,
                                      int channel_config)
{
    if (channel_config < 1 || (channel_config > 7 && channel_config < 11) ||
        channel_config > 12) {
        av_log(avctx, AV_LOG_ERROR,
               "invalid default channel configuration (%d)\n",
               channel_config);
        return AVERROR_INVALIDDATA;
    }
    *tags = tags_per_config[channel_config];
    memcpy(layout_map, 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) {
        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);
        layout_map[2][2] = AAC_CHANNEL_SIDE;
    }

    return 0;
}

static ChannelElement *get_che(AACContext *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 (set_default_channel_config(ac->avctx, layout_map,
                                       &layout_map_tags, 2) < 0)
            return NULL;
        if (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");

        if (set_default_channel_config(ac->avctx, layout_map,
                                       &layout_map_tags, 1) < 0)
            return NULL;
        if (output_configure(ac, layout_map, layout_map_tags,
                             OC_TRIAL_FRAME, 1) < 0)
            return NULL;

        ac->oc[1].m4ac.chan_config = 1;
        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 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 == 2 &&
            ac->oc[1].m4ac.chan_config == 11 &&
            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 == 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 == 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->oc[1].m4ac.chan_config == 2) {
            return NULL;
        }
    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++;
    }
}

/**
 * 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 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) < 4 * (num_front + num_side + num_back + 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;

    align_get_bits(gb);

    /* 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 AACContext, 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(AACContext *ac, AVCodecContext *avctx,
                                     GetBitContext *gb,
                                     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;

    if (get_bits1(gb)) { // frameLengthFlag
        avpriv_request_sample(avctx, "960/120 MDCT window");
        return AVERROR_PATCHWELCOME;
    }
    m4ac->frame_length_short = 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);
        if (tags < 0)
            return tags;
    } else {
        if ((ret = set_default_channel_config(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 = 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(AACContext *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;
877 878 879 880 881 882
#if USE_FIXED
    if (get_bits1(gb)) { // frameLengthFlag
        avpriv_request_sample(avctx, "960/120 MDCT window");
        return AVERROR_PATCHWELCOME;
    }
#else
883
    m4ac->frame_length_short = get_bits1(gb);
884
#endif
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    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 = set_default_channel_config(avctx, layout_map,
                                          &tags, channel_config)))
        return ret;

    if (ac && (ret = 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 AACContext, may be null
 * @param   avctx       pointer to AVCCodecContext, used for logging
 * @param   m4ac        pointer to MPEG4AudioConfig, used for parsing
 * @param   data        pointer to buffer holding an audio specific config
 * @param   bit_size    size of audio specific config or data in bits
 * @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(AACContext *ac,
                                        AVCodecContext *avctx,
                                        MPEG4AudioConfig *m4ac,
                                        const uint8_t *data, int bit_size,
                                        int sync_extension)
{
    GetBitContext gb;
    int i, ret;

    ff_dlog(avctx, "audio specific config size %d\n", 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;

    if ((i = avpriv_mpeg4audio_get_config(m4ac, data, bit_size,
                                          sync_extension)) < 0)
        return AVERROR_INVALIDDATA;
    if (m4ac->sampling_index > 12) {
        av_log(avctx, AV_LOG_ERROR,
               "invalid sampling rate index %d\n",
               m4ac->sampling_index);
        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);
        return AVERROR_INVALIDDATA;
    }

    skip_bits_long(&gb, i);

    switch (m4ac->object_type) {
    case AOT_AAC_MAIN:
    case AOT_AAC_LC:
    case AOT_AAC_LTP:
    case AOT_ER_AAC_LC:
    case AOT_ER_AAC_LD:
        if ((ret = decode_ga_specific_config(ac, avctx, &gb,
                                            m4ac, m4ac->chan_config)) < 0)
            return ret;
        break;
    case AOT_ER_AAC_ELD:
        if ((ret = decode_eld_specific_config(ac, avctx, &gb,
                                              m4ac, m4ac->chan_config)) < 0)
            return ret;
        break;
    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);
}

/**
 * linear congruential pseudorandom number generator
 *
 * @param   previous_val    pointer to the current state of the generator
 *
 * @return  Returns a 32-bit pseudorandom integer
 */
static av_always_inline int lcg_random(unsigned previous_val)
{
    union { unsigned u; int s; } v = { previous_val * 1664525u + 1013904223 };
    return v.s;
}

static void reset_all_predictors(PredictorState *ps)
{
    int i;
    for (i = 0; i < MAX_PREDICTORS; i++)
        reset_predict_state(&ps[i]);
}

static int sample_rate_idx (int rate)
{
         if (92017 <= rate) return 0;
    else if (75132 <= rate) return 1;
    else if (55426 <= rate) return 2;
    else if (46009 <= rate) return 3;
    else if (37566 <= rate) return 4;
    else if (27713 <= rate) return 5;
    else if (23004 <= rate) return 6;
    else if (18783 <= rate) return 7;
    else if (13856 <= rate) return 8;
    else if (11502 <= rate) return 9;
    else if (9391  <= rate) return 10;
    else                    return 11;
}

static void reset_predictor_group(PredictorState *ps, int group_num)
{
    int i;
    for (i = group_num - 1; i < MAX_PREDICTORS; i += 30)
        reset_predict_state(&ps[i]);
}

#define AAC_INIT_VLC_STATIC(num, size)                                     \
    INIT_VLC_STATIC(&vlc_spectral[num], 8, ff_aac_spectral_sizes[num],     \
         ff_aac_spectral_bits[num], sizeof(ff_aac_spectral_bits[num][0]),  \
                                    sizeof(ff_aac_spectral_bits[num][0]),  \
        ff_aac_spectral_codes[num], sizeof(ff_aac_spectral_codes[num][0]), \
                                    sizeof(ff_aac_spectral_codes[num][0]), \
        size);

static void aacdec_init(AACContext *ac);

static av_cold int aac_decode_init(AVCodecContext *avctx)
{
    AACContext *ac = avctx->priv_data;
    int ret;

    ac->avctx = avctx;
    ac->oc[1].m4ac.sample_rate = avctx->sample_rate;

    aacdec_init(ac);
1070 1071 1072
#if USE_FIXED
    avctx->sample_fmt = AV_SAMPLE_FMT_S32P;
#else
1073
    avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
1074
#endif /* USE_FIXED */
1075 1076 1077 1078 1079 1080 1081 1082 1083 1084 1085 1086 1087 1088 1089 1090 1091 1092 1093 1094 1095 1096 1097 1098 1099 1100 1101 1102 1103 1104 1105 1106 1107 1108 1109 1110 1111 1112 1113 1114 1115 1116 1117 1118 1119 1120 1121 1122 1123 1124 1125 1126 1127 1128 1129 1130

    if (avctx->extradata_size > 0) {
        if ((ret = decode_audio_specific_config(ac, ac->avctx, &ac->oc[1].m4ac,
                                                avctx->extradata,
                                                avctx->extradata_size * 8,
                                                1)) < 0)
            return ret;
    } else {
        int sr, i;
        uint8_t layout_map[MAX_ELEM_ID*4][3];
        int layout_map_tags;

        sr = sample_rate_idx(avctx->sample_rate);
        ac->oc[1].m4ac.sampling_index = sr;
        ac->oc[1].m4ac.channels = avctx->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->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 = set_default_channel_config(avctx, layout_map,
                &layout_map_tags, ac->oc[1].m4ac.chan_config);
            if (!ret)
                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->channels > MAX_CHANNELS) {
        av_log(avctx, AV_LOG_ERROR, "Too many channels\n");
        return AVERROR_INVALIDDATA;
    }

    AAC_INIT_VLC_STATIC( 0, 304);
    AAC_INIT_VLC_STATIC( 1, 270);
    AAC_INIT_VLC_STATIC( 2, 550);
    AAC_INIT_VLC_STATIC( 3, 300);
    AAC_INIT_VLC_STATIC( 4, 328);
    AAC_INIT_VLC_STATIC( 5, 294);
    AAC_INIT_VLC_STATIC( 6, 306);
    AAC_INIT_VLC_STATIC( 7, 268);
    AAC_INIT_VLC_STATIC( 8, 510);
    AAC_INIT_VLC_STATIC( 9, 366);
    AAC_INIT_VLC_STATIC(10, 462);

    ff_aac_sbr_init();

1131 1132 1133
#if USE_FIXED
    ac->fdsp = avpriv_alloc_fixed_dsp(avctx->flags & CODEC_FLAG_BITEXACT);
#else
1134
    ac->fdsp = avpriv_float_dsp_alloc(avctx->flags & CODEC_FLAG_BITEXACT);
1135
#endif /* USE_FIXED */
1136 1137 1138 1139 1140 1141 1142 1143 1144 1145 1146 1147 1148 1149 1150 1151 1152 1153
    if (!ac->fdsp) {
        return AVERROR(ENOMEM);
    }

    ac->random_state = 0x1f2e3d4c;

    ff_aac_tableinit();

    INIT_VLC_STATIC(&vlc_scalefactors, 7,
                    FF_ARRAY_ELEMS(ff_aac_scalefactor_code),
                    ff_aac_scalefactor_bits,
                    sizeof(ff_aac_scalefactor_bits[0]),
                    sizeof(ff_aac_scalefactor_bits[0]),
                    ff_aac_scalefactor_code,
                    sizeof(ff_aac_scalefactor_code[0]),
                    sizeof(ff_aac_scalefactor_code[0]),
                    352);

1154 1155 1156 1157 1158
    AAC_RENAME_32(ff_mdct_init)(&ac->mdct,       11, 1, 1.0 / RANGE15(1024.0));
    AAC_RENAME_32(ff_mdct_init)(&ac->mdct_ld,    10, 1, 1.0 / RANGE15(512.0));
    AAC_RENAME_32(ff_mdct_init)(&ac->mdct_small,  8, 1, 1.0 / RANGE15(128.0));
    AAC_RENAME_32(ff_mdct_init)(&ac->mdct_ltp,   11, 0, RANGE15(-2.0));
#if !USE_FIXED
1159 1160 1161
    ret = ff_imdct15_init(&ac->mdct480, 5);
    if (ret < 0)
        return ret;
1162
#endif
1163
    // window initialization
1164 1165 1166 1167 1168
    AAC_RENAME(ff_kbd_window_init)(AAC_RENAME(ff_aac_kbd_long_1024), 4.0, 1024);
    AAC_RENAME(ff_kbd_window_init)(AAC_RENAME(ff_aac_kbd_short_128), 6.0, 128);
    AAC_RENAME(ff_init_ff_sine_windows)(10);
    AAC_RENAME(ff_init_ff_sine_windows)( 9);
    AAC_RENAME(ff_init_ff_sine_windows)( 7);
1169

1170
    AAC_RENAME(cbrt_tableinit)();
1171 1172 1173 1174 1175 1176 1177 1178 1179 1180 1181 1182 1183 1184 1185 1186 1187 1188 1189 1190 1191 1192 1193 1194 1195 1196 1197 1198 1199 1200 1201 1202 1203 1204 1205 1206 1207 1208 1209 1210 1211 1212 1213 1214 1215 1216 1217 1218 1219 1220 1221 1222 1223 1224 1225 1226 1227 1228 1229 1230 1231 1232 1233 1234 1235 1236 1237 1238 1239 1240 1241 1242 1243 1244 1245 1246 1247 1248 1249 1250 1251 1252 1253 1254 1255 1256 1257 1258 1259 1260 1261 1262 1263 1264 1265 1266 1267 1268 1269 1270 1271 1272 1273 1274 1275 1276 1277 1278 1279 1280 1281 1282 1283 1284 1285 1286 1287 1288 1289 1290 1291 1292 1293 1294 1295 1296 1297 1298 1299 1300 1301 1302 1303 1304 1305 1306 1307 1308 1309 1310 1311 1312 1313 1314 1315 1316 1317 1318 1319 1320 1321 1322 1323 1324 1325 1326 1327 1328 1329 1330 1331 1332 1333 1334 1335 1336 1337 1338 1339 1340 1341 1342 1343 1344 1345 1346 1347 1348 1349 1350 1351 1352 1353 1354 1355 1356 1357 1358 1359 1360 1361 1362 1363 1364 1365 1366 1367 1368 1369 1370 1371 1372 1373 1374 1375 1376 1377 1378 1379 1380 1381 1382 1383 1384 1385 1386 1387 1388 1389 1390

    return 0;
}

/**
 * Skip data_stream_element; reference: table 4.10.
 */
static int skip_data_stream_element(AACContext *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(AACContext *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(LongTermPrediction *ltp,
                       GetBitContext *gb, uint8_t max_sfb)
{
    int sfb;

    ltp->lag  = get_bits(gb, 11);
    ltp->coef = 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(AACContext *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;
    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->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;
        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)
                return AVERROR_BUG;
        } else {
            ics->swb_offset    =    ff_swb_offset_1024[sampling_index];
            ics->num_swb       =   ff_aac_num_swb_1024[sampling_index];
            ics->tns_max_bands = ff_tns_max_bands_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");
                    return AVERROR_PATCHWELCOME;
                }
                if ((ics->ltp.present = get_bits(gb, 1)))
                    decode_ltp(&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 AVERROR_INVALIDDATA;
}

/**
 * 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(AACContext *ac, enum BandType band_type[120],
                             int band_type_run_end[120], GetBitContext *gb,
                             IndividualChannelStream *ics)
{
    int g, idx = 0;
    const int bits = (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) ? 3 : 5;
    for (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++) {
                band_type        [idx]   = sect_band_type;
                band_type_run_end[idx++] = sect_end;
            }
        }
    }
    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
 */
1391
static int decode_scalefactors(AACContext *ac, INTFLOAT sf[120], GetBitContext *gb,
1392 1393 1394 1395 1396 1397 1398 1399 1400 1401 1402 1403 1404 1405
                               unsigned int global_gain,
                               IndividualChannelStream *ics,
                               enum BandType band_type[120],
                               int band_type_run_end[120])
{
    int g, i, idx = 0;
    int offset[3] = { global_gain, global_gain - NOISE_OFFSET, 0 };
    int clipped_offset;
    int noise_flag = 1;
    for (g = 0; g < ics->num_window_groups; g++) {
        for (i = 0; i < ics->max_sfb;) {
            int run_end = band_type_run_end[idx];
            if (band_type[idx] == ZERO_BT) {
                for (; i < run_end; i++, idx++)
1406
                    sf[idx] = FIXR(0.);
1407 1408 1409 1410 1411 1412 1413 1414 1415 1416 1417
            } else if ((band_type[idx] == INTENSITY_BT) ||
                       (band_type[idx] == INTENSITY_BT2)) {
                for (; i < run_end; i++, idx++) {
                    offset[2] += get_vlc2(gb, vlc_scalefactors.table, 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);
                    }
1418 1419 1420
#if USE_FIXED
                    sf[idx] = 100 - clipped_offset;
#else
1421
                    sf[idx] = ff_aac_pow2sf_tab[-clipped_offset + POW_SF2_ZERO];
1422
#endif /* USE_FIXED */
1423 1424 1425 1426 1427 1428 1429 1430 1431 1432 1433 1434 1435 1436
                }
            } else if (band_type[idx] == NOISE_BT) {
                for (; i < run_end; i++, idx++) {
                    if (noise_flag-- > 0)
                        offset[1] += get_bits(gb, NOISE_PRE_BITS) - NOISE_PRE;
                    else
                        offset[1] += get_vlc2(gb, vlc_scalefactors.table, 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);
                    }
1437 1438 1439
#if USE_FIXED
                    sf[idx] = -(100 + clipped_offset);
#else
1440
                    sf[idx] = -ff_aac_pow2sf_tab[clipped_offset + POW_SF2_ZERO];
1441
#endif /* USE_FIXED */
1442 1443 1444 1445 1446 1447 1448 1449 1450
                }
            } else {
                for (; i < run_end; i++, idx++) {
                    offset[0] += get_vlc2(gb, vlc_scalefactors.table, 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;
                    }
1451 1452 1453
#if USE_FIXED
                    sf[idx] = -offset[0];
#else
1454
                    sf[idx] = -ff_aac_pow2sf_tab[offset[0] - 100 + POW_SF2_ZERO];
1455
#endif /* USE_FIXED */
1456 1457 1458 1459 1460 1461 1462 1463 1464 1465 1466 1467 1468 1469 1470 1471 1472 1473 1474 1475 1476 1477 1478 1479 1480 1481 1482 1483 1484 1485 1486 1487 1488 1489 1490 1491 1492 1493 1494 1495 1496 1497 1498 1499 1500 1501 1502 1503 1504 1505 1506 1507 1508 1509 1510 1511 1512 1513 1514 1515 1516 1517 1518 1519 1520 1521 1522 1523 1524 1525 1526 1527 1528 1529 1530 1531 1532 1533 1534 1535 1536 1537 1538 1539 1540 1541 1542 1543 1544 1545 1546 1547 1548 1549 1550 1551 1552 1553 1554 1555 1556 1557 1558 1559 1560
                }
            }
        }
    }
    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
 */
static int decode_tns(AACContext *ac, TemporalNoiseShaping *tns,
                      GetBitContext *gb, const IndividualChannelStream *ics)
{
    int w, filt, i, coef_len, coef_res, coef_compress;
    const int is8 = ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE;
    const int 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 ((tns->order[w][filt] = get_bits(gb, 5 - 2 * is8)) > 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++)
                        tns->coef[w][filt][i] = 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;
    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]));
    }
}

/**
 * Decode spectral data; reference: table 4.50.
 * Dequantize and scale spectral data; reference: 4.6.3.3.
 *
 * @param   coef            array of dequantized, scaled spectral data
 * @param   sf              array of scalefactors or intensity stereo positions
 * @param   pulse_present   set if pulses are present
 * @param   pulse           pointer to pulse data struct
 * @param   band_type       array of the used band type
 *
 * @return  Returns error status. 0 - OK, !0 - error
 */
1561 1562
static int decode_spectrum_and_dequant(AACContext *ac, INTFLOAT coef[1024],
                                       GetBitContext *gb, const INTFLOAT sf[120],
1563 1564 1565 1566 1567 1568 1569
                                       int pulse_present, const Pulse *pulse,
                                       const IndividualChannelStream *ics,
                                       enum BandType band_type[120])
{
    int i, k, g, idx = 0;
    const int c = 1024 / ics->num_windows;
    const uint16_t *offsets = ics->swb_offset;
1570
    INTFLOAT *coef_base = coef;
1571 1572 1573

    for (g = 0; g < ics->num_windows; g++)
        memset(coef + g * 128 + offsets[ics->max_sfb], 0,
1574
               sizeof(INTFLOAT) * (c - offsets[ics->max_sfb]));
1575 1576 1577 1578 1579 1580

    for (g = 0; g < ics->num_window_groups; g++) {
        unsigned g_len = ics->group_len[g];

        for (i = 0; i < ics->max_sfb; i++, idx++) {
            const unsigned cbt_m1 = band_type[idx] - 1;
1581
            INTFLOAT *cfo = coef + offsets[i];
1582 1583 1584 1585
            int off_len = offsets[i + 1] - offsets[i];
            int group;

            if (cbt_m1 >= INTENSITY_BT2 - 1) {
1586 1587
                for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
                    memset(cfo, 0, off_len * sizeof(*cfo));
1588 1589
                }
            } else if (cbt_m1 == NOISE_BT - 1) {
1590 1591
                for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
#if !USE_FIXED
1592
                    float scale;
1593 1594
#endif /* !USE_FIXED */
                    INTFLOAT band_energy;
1595 1596 1597

                    for (k = 0; k < off_len; k++) {
                        ac->random_state  = lcg_random(ac->random_state);
1598 1599 1600
#if USE_FIXED
                        cfo[k] = ac->random_state >> 3;
#else
1601
                        cfo[k] = ac->random_state;
1602
#endif /* USE_FIXED */
1603 1604
                    }

1605 1606 1607 1608 1609
#if USE_FIXED
                    band_energy = ac->fdsp->scalarproduct_fixed(cfo, cfo, off_len);
                    band_energy = fixed_sqrt(band_energy, 31);
                    noise_scale(cfo, sf[idx], band_energy, off_len);
#else
1610 1611 1612
                    band_energy = ac->fdsp->scalarproduct_float(cfo, cfo, off_len);
                    scale = sf[idx] / sqrtf(band_energy);
                    ac->fdsp->vector_fmul_scalar(cfo, cfo, scale, off_len);
1613
#endif /* USE_FIXED */
1614 1615
                }
            } else {
1616
#if !USE_FIXED
1617
                const float *vq = ff_aac_codebook_vector_vals[cbt_m1];
1618
#endif /* !USE_FIXED */
1619 1620 1621 1622 1623 1624
                const uint16_t *cb_vector_idx = ff_aac_codebook_vector_idx[cbt_m1];
                VLC_TYPE (*vlc_tab)[2] = vlc_spectral[cbt_m1].table;
                OPEN_READER(re, gb);

                switch (cbt_m1 >> 1) {
                case 0:
1625 1626
                    for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
                        INTFLOAT *cf = cfo;
1627 1628 1629 1630 1631 1632 1633 1634 1635
                        int len = off_len;

                        do {
                            int code;
                            unsigned cb_idx;

                            UPDATE_CACHE(re, gb);
                            GET_VLC(code, re, gb, vlc_tab, 8, 2);
                            cb_idx = cb_vector_idx[code];
1636 1637 1638
#if USE_FIXED
                            cf = DEC_SQUAD(cf, cb_idx);
#else
1639
                            cf = VMUL4(cf, vq, cb_idx, sf + idx);
1640
#endif /* USE_FIXED */
1641 1642 1643 1644 1645
                        } while (len -= 4);
                    }
                    break;

                case 1:
1646 1647
                    for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
                        INTFLOAT *cf = cfo;
1648 1649 1650 1651 1652 1653 1654 1655 1656 1657 1658 1659 1660 1661
                        int len = off_len;

                        do {
                            int code;
                            unsigned nnz;
                            unsigned cb_idx;
                            uint32_t bits;

                            UPDATE_CACHE(re, gb);
                            GET_VLC(code, re, gb, vlc_tab, 8, 2);
                            cb_idx = cb_vector_idx[code];
                            nnz = cb_idx >> 8 & 15;
                            bits = nnz ? GET_CACHE(re, gb) : 0;
                            LAST_SKIP_BITS(re, gb, nnz);
1662 1663 1664
#if USE_FIXED
                            cf = DEC_UQUAD(cf, cb_idx, bits);
#else
1665
                            cf = VMUL4S(cf, vq, cb_idx, bits, sf + idx);
1666
#endif /* USE_FIXED */
1667 1668 1669 1670 1671
                        } while (len -= 4);
                    }
                    break;

                case 2:
1672 1673
                    for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
                        INTFLOAT *cf = cfo;
1674 1675 1676 1677 1678 1679 1680 1681 1682
                        int len = off_len;

                        do {
                            int code;
                            unsigned cb_idx;

                            UPDATE_CACHE(re, gb);
                            GET_VLC(code, re, gb, vlc_tab, 8, 2);
                            cb_idx = cb_vector_idx[code];
1683 1684 1685
#if USE_FIXED
                            cf = DEC_SPAIR(cf, cb_idx);
#else
1686
                            cf = VMUL2(cf, vq, cb_idx, sf + idx);
1687
#endif /* USE_FIXED */
1688 1689 1690 1691 1692 1693
                        } while (len -= 2);
                    }
                    break;

                case 3:
                case 4:
1694 1695
                    for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
                        INTFLOAT *cf = cfo;
1696 1697 1698 1699 1700 1701 1702 1703 1704 1705 1706 1707 1708 1709
                        int len = off_len;

                        do {
                            int code;
                            unsigned nnz;
                            unsigned cb_idx;
                            unsigned sign;

                            UPDATE_CACHE(re, gb);
                            GET_VLC(code, re, gb, vlc_tab, 8, 2);
                            cb_idx = cb_vector_idx[code];
                            nnz = cb_idx >> 8 & 15;
                            sign = nnz ? SHOW_UBITS(re, gb, nnz) << (cb_idx >> 12) : 0;
                            LAST_SKIP_BITS(re, gb, nnz);
1710 1711 1712
#if USE_FIXED
                            cf = DEC_UPAIR(cf, cb_idx, sign);
#else
1713
                            cf = VMUL2S(cf, vq, cb_idx, sign, sf + idx);
1714
#endif /* USE_FIXED */
1715 1716 1717 1718 1719
                        } while (len -= 2);
                    }
                    break;

                default:
1720 1721 1722 1723 1724
                    for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
#if USE_FIXED
                        int *icf = cfo;
                        int v;
#else
1725 1726
                        float *cf = cfo;
                        uint32_t *icf = (uint32_t *) cf;
1727
#endif /* USE_FIXED */
1728 1729 1730 1731 1732 1733 1734 1735 1736 1737 1738 1739 1740 1741 1742 1743 1744 1745 1746 1747 1748 1749 1750 1751 1752 1753 1754 1755 1756 1757 1758 1759 1760 1761 1762 1763 1764 1765 1766 1767 1768 1769 1770
                        int len = off_len;

                        do {
                            int code;
                            unsigned nzt, nnz;
                            unsigned cb_idx;
                            uint32_t bits;
                            int j;

                            UPDATE_CACHE(re, gb);
                            GET_VLC(code, re, gb, vlc_tab, 8, 2);

                            if (!code) {
                                *icf++ = 0;
                                *icf++ = 0;
                                continue;
                            }

                            cb_idx = cb_vector_idx[code];
                            nnz = cb_idx >> 12;
                            nzt = cb_idx >> 8;
                            bits = SHOW_UBITS(re, gb, nnz) << (32-nnz);
                            LAST_SKIP_BITS(re, gb, nnz);

                            for (j = 0; j < 2; j++) {
                                if (nzt & 1<<j) {
                                    uint32_t b;
                                    int n;
                                    /* The total length of escape_sequence must be < 22 bits according
                                       to the specification (i.e. max is 111111110xxxxxxxxxxxx). */
                                    UPDATE_CACHE(re, gb);
                                    b = GET_CACHE(re, gb);
                                    b = 31 - av_log2(~b);

                                    if (b > 8) {
                                        av_log(ac->avctx, AV_LOG_ERROR, "error in spectral data, ESC overflow\n");
                                        return AVERROR_INVALIDDATA;
                                    }

                                    SKIP_BITS(re, gb, b + 1);
                                    b += 4;
                                    n = (1 << b) + SHOW_UBITS(re, gb, b);
                                    LAST_SKIP_BITS(re, gb, b);
1771 1772 1773 1774 1775 1776
#if USE_FIXED
                                    v = n;
                                    if (bits & 1U<<31)
                                        v = -v;
                                    *icf++ = v;
#else
1777
                                    *icf++ = cbrt_tab[n] | (bits & 1U<<31);
1778
#endif /* USE_FIXED */
1779 1780
                                    bits <<= 1;
                                } else {
1781 1782 1783 1784 1785 1786
#if USE_FIXED
                                    v = cb_idx & 15;
                                    if (bits & 1U<<31)
                                        v = -v;
                                    *icf++ = v;
#else
1787 1788
                                    unsigned v = ((const uint32_t*)vq)[cb_idx & 15];
                                    *icf++ = (bits & 1U<<31) | v;
1789
#endif /* USE_FIXED */
1790 1791 1792 1793 1794
                                    bits <<= !!v;
                                }
                                cb_idx >>= 4;
                            }
                        } while (len -= 2);
1795
#if !USE_FIXED
1796
                        ac->fdsp->vector_fmul_scalar(cfo, cfo, sf[idx], off_len);
1797
#endif /* !USE_FIXED */
1798 1799 1800 1801 1802 1803 1804 1805 1806 1807 1808 1809
                    }
                }

                CLOSE_READER(re, gb);
            }
        }
        coef += g_len << 7;
    }

    if (pulse_present) {
        idx = 0;
        for (i = 0; i < pulse->num_pulse; i++) {
1810
            INTFLOAT co = coef_base[ pulse->pos[i] ];
1811 1812 1813
            while (offsets[idx + 1] <= pulse->pos[i])
                idx++;
            if (band_type[idx] != NOISE_BT && sf[idx]) {
1814 1815 1816 1817 1818 1819 1820
                INTFLOAT ico = -pulse->amp[i];
#if USE_FIXED
                if (co) {
                    ico = co + (co > 0 ? -ico : ico);
                }
                coef_base[ pulse->pos[i] ] = ico;
#else
1821 1822 1823 1824 1825
                if (co) {
                    co /= sf[idx];
                    ico = co / sqrtf(sqrtf(fabsf(co))) + (co > 0 ? -ico : ico);
                }
                coef_base[ pulse->pos[i] ] = cbrtf(fabsf(ico)) * ico * sf[idx];
1826 1827 1828 1829 1830 1831 1832 1833 1834 1835 1836 1837 1838 1839 1840 1841 1842 1843 1844 1845 1846
#endif /* USE_FIXED */
            }
        }
    }
#if USE_FIXED
    coef = coef_base;
    idx = 0;
    for (g = 0; g < ics->num_window_groups; g++) {
        unsigned g_len = ics->group_len[g];

        for (i = 0; i < ics->max_sfb; i++, idx++) {
            const unsigned cbt_m1 = band_type[idx] - 1;
            int *cfo = coef + offsets[i];
            int off_len = offsets[i + 1] - offsets[i];
            int group;

            if (cbt_m1 < NOISE_BT - 1) {
                for (group = 0; group < (int)g_len; group++, cfo+=128) {
                    ac->vector_pow43(cfo, off_len);
                    ac->subband_scale(cfo, cfo, sf[idx], 34, off_len);
                }
1847 1848
            }
        }
1849
        coef += g_len << 7;
1850
    }
1851
#endif /* USE_FIXED */
1852 1853 1854 1855 1856 1857 1858 1859 1860 1861 1862 1863 1864 1865 1866 1867 1868 1869 1870 1871 1872 1873 1874 1875 1876 1877 1878 1879 1880 1881 1882 1883 1884 1885 1886 1887 1888 1889 1890 1891 1892 1893 1894 1895 1896 1897 1898 1899
    return 0;
}

/**
 * Apply AAC-Main style frequency domain prediction.
 */
static void apply_prediction(AACContext *ac, SingleChannelElement *sce)
{
    int sfb, k;

    if (!sce->ics.predictor_initialized) {
        reset_all_predictors(sce->predictor_state);
        sce->ics.predictor_initialized = 1;
    }

    if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
        for (sfb = 0;
             sfb < ff_aac_pred_sfb_max[ac->oc[1].m4ac.sampling_index];
             sfb++) {
            for (k = sce->ics.swb_offset[sfb];
                 k < sce->ics.swb_offset[sfb + 1];
                 k++) {
                predict(&sce->predictor_state[k], &sce->coeffs[k],
                        sce->ics.predictor_present &&
                        sce->ics.prediction_used[sfb]);
            }
        }
        if (sce->ics.predictor_reset_group)
            reset_predictor_group(sce->predictor_state,
                                  sce->ics.predictor_reset_group);
    } else
        reset_all_predictors(sce->predictor_state);
}

/**
 * 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
 */
static int decode_ics(AACContext *ac, SingleChannelElement *sce,
                      GetBitContext *gb, int common_window, int scale_flag)
{
    Pulse pulse;
    TemporalNoiseShaping    *tns = &sce->tns;
    IndividualChannelStream *ics = &sce->ics;
1900
    INTFLOAT *out = sce->coeffs;
1901 1902 1903 1904 1905 1906 1907 1908 1909 1910 1911 1912 1913 1914 1915 1916 1917 1918 1919 1920 1921 1922 1923 1924 1925 1926 1927 1928 1929 1930 1931 1932 1933 1934 1935 1936 1937 1938 1939 1940 1941 1942 1943 1944 1945 1946 1947 1948 1949 1950 1951 1952 1953 1954 1955 1956 1957 1958 1959 1960 1961 1962 1963 1964 1965 1966 1967 1968 1969 1970 1971 1972 1973
    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) {
        if (decode_ics_info(ac, ics, gb) < 0)
            return AVERROR_INVALIDDATA;
    }

    if ((ret = decode_band_types(ac, sce->band_type,
                                 sce->band_type_run_end, gb, ics)) < 0)
        return ret;
    if ((ret = decode_scalefactors(ac, sce->sf, gb, global_gain, ics,
                                  sce->band_type, sce->band_type_run_end)) < 0)
        return ret;

    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");
                return AVERROR_INVALIDDATA;
            }
            if (decode_pulses(&pulse, gb, ics->swb_offset, ics->num_swb)) {
                av_log(ac->avctx, AV_LOG_ERROR,
                       "Pulse data corrupt or invalid.\n");
                return AVERROR_INVALIDDATA;
            }
        }
        tns->present = get_bits1(gb);
        if (tns->present && !er_syntax)
            if (decode_tns(ac, tns, gb, ics) < 0)
                return AVERROR_INVALIDDATA;
        if (!eld_syntax && get_bits1(gb)) {
            avpriv_request_sample(ac->avctx, "SSR");
            return AVERROR_PATCHWELCOME;
        }
        // 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)
            if (decode_tns(ac, tns, gb, ics) < 0)
                return AVERROR_INVALIDDATA;
    }

    if (decode_spectrum_and_dequant(ac, out, gb, sce->sf, pulse_present,
                                    &pulse, ics, sce->band_type) < 0)
        return AVERROR_INVALIDDATA;

    if (ac->oc[1].m4ac.object_type == AOT_AAC_MAIN && !common_window)
        apply_prediction(ac, sce);

    return 0;
}

/**
 * Mid/Side stereo decoding; reference: 4.6.8.1.3.
 */
static void apply_mid_side_stereo(AACContext *ac, ChannelElement *cpe)
{
    const IndividualChannelStream *ics = &cpe->ch[0].ics;
1974 1975
    INTFLOAT *ch0 = cpe->ch[0].coeffs;
    INTFLOAT *ch1 = cpe->ch[1].coeffs;
1976 1977 1978 1979 1980 1981 1982
    int g, i, group, idx = 0;
    const uint16_t *offsets = ics->swb_offset;
    for (g = 0; g < ics->num_window_groups; g++) {
        for (i = 0; i < ics->max_sfb; i++, idx++) {
            if (cpe->ms_mask[idx] &&
                cpe->ch[0].band_type[idx] < NOISE_BT &&
                cpe->ch[1].band_type[idx] < NOISE_BT) {
1983 1984 1985 1986 1987 1988
#if USE_FIXED
                for (group = 0; group < ics->group_len[g]; group++) {
                    ac->fdsp->butterflies_fixed(ch0 + group * 128 + offsets[i],
                                                ch1 + group * 128 + offsets[i],
                                                offsets[i+1] - offsets[i]);
#else
1989 1990 1991 1992
                for (group = 0; group < ics->group_len[g]; group++) {
                    ac->fdsp->butterflies_float(ch0 + group * 128 + offsets[i],
                                               ch1 + group * 128 + offsets[i],
                                               offsets[i+1] - offsets[i]);
1993
#endif /* USE_FIXED */
1994 1995 1996 1997 1998 1999 2000 2001 2002 2003 2004 2005 2006 2007 2008 2009 2010 2011 2012 2013
                }
            }
        }
        ch0 += ics->group_len[g] * 128;
        ch1 += ics->group_len[g] * 128;
    }
}

/**
 * intensity stereo decoding; reference: 4.6.8.2.3
 *
 * @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 apply_intensity_stereo(AACContext *ac,
                                   ChannelElement *cpe, int ms_present)
{
    const IndividualChannelStream *ics = &cpe->ch[1].ics;
    SingleChannelElement         *sce1 = &cpe->ch[1];
2014
    INTFLOAT *coef0 = cpe->ch[0].coeffs, *coef1 = cpe->ch[1].coeffs;
2015 2016 2017
    const uint16_t *offsets = ics->swb_offset;
    int g, group, i, idx = 0;
    int c;
2018
    INTFLOAT scale;
2019 2020 2021 2022 2023 2024 2025 2026 2027 2028 2029
    for (g = 0; g < ics->num_window_groups; g++) {
        for (i = 0; i < ics->max_sfb;) {
            if (sce1->band_type[idx] == INTENSITY_BT ||
                sce1->band_type[idx] == INTENSITY_BT2) {
                const int bt_run_end = sce1->band_type_run_end[idx];
                for (; i < bt_run_end; i++, idx++) {
                    c = -1 + 2 * (sce1->band_type[idx] - 14);
                    if (ms_present)
                        c *= 1 - 2 * cpe->ms_mask[idx];
                    scale = c * sce1->sf[idx];
                    for (group = 0; group < ics->group_len[g]; group++)
2030 2031 2032 2033 2034 2035 2036
#if USE_FIXED
                        ac->subband_scale(coef1 + group * 128 + offsets[i],
                                      coef0 + group * 128 + offsets[i],
                                      scale,
                                      23,
                                      offsets[i + 1] - offsets[i]);
#else
2037 2038 2039 2040
                        ac->fdsp->vector_fmul_scalar(coef1 + group * 128 + offsets[i],
                                                    coef0 + group * 128 + offsets[i],
                                                    scale,
                                                    offsets[i + 1] - offsets[i]);
2041
#endif /* USE_FIXED */
2042 2043 2044 2045 2046 2047 2048 2049 2050 2051 2052 2053 2054 2055 2056 2057 2058 2059 2060 2061 2062 2063 2064 2065 2066 2067 2068 2069 2070 2071 2072 2073 2074 2075 2076 2077 2078 2079 2080 2081 2082 2083 2084 2085 2086 2087 2088 2089 2090 2091 2092 2093 2094 2095 2096 2097 2098 2099 2100 2101 2102 2103 2104 2105 2106 2107 2108 2109 2110 2111 2112 2113 2114 2115 2116
                }
            } else {
                int bt_run_end = sce1->band_type_run_end[idx];
                idx += bt_run_end - i;
                i    = bt_run_end;
            }
        }
        coef0 += ics->group_len[g] * 128;
        coef1 += ics->group_len[g] * 128;
    }
}

/**
 * Decode a channel_pair_element; reference: table 4.4.
 *
 * @return  Returns error status. 0 - OK, !0 - error
 */
static int decode_cpe(AACContext *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(&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 = decode_ics(ac, &cpe->ch[0], gb, common_window, 0)))
        return ret;
    if ((ret = decode_ics(ac, &cpe->ch[1], gb, common_window, 0)))
        return ret;

    if (common_window) {
        if (ms_present)
            apply_mid_side_stereo(ac, cpe);
        if (ac->oc[1].m4ac.object_type == AOT_AAC_MAIN) {
            apply_prediction(ac, &cpe->ch[0]);
            apply_prediction(ac, &cpe->ch[1]);
        }
    }

    apply_intensity_stereo(ac, cpe, ms_present);
    return 0;
}

static const float cce_scale[] = {
    1.09050773266525765921, //2^(1/8)
    1.18920711500272106672, //2^(1/4)
    M_SQRT2,
    2,
};

/**
 * Decode coupling_channel_element; reference: table 4.8.
 *
 * @return  Returns error status. 0 - OK, !0 - error
 */
static int decode_cce(AACContext *ac, GetBitContext *gb, ChannelElement *che)
{
    int num_gain = 0;
    int c, g, sfb, ret;
    int sign;
2117
    INTFLOAT scale;
2118 2119 2120 2121 2122 2123 2124 2125 2126 2127 2128 2129 2130 2131 2132 2133 2134 2135 2136
    SingleChannelElement *sce = &che->ch[0];
    ChannelCoupling     *coup = &che->coup;

    coup->coupling_point = 2 * get_bits1(gb);
    coup->num_coupled = get_bits(gb, 3);
    for (c = 0; c <= coup->num_coupled; c++) {
        num_gain++;
        coup->type[c] = get_bits1(gb) ? TYPE_CPE : TYPE_SCE;
        coup->id_select[c] = get_bits(gb, 4);
        if (coup->type[c] == TYPE_CPE) {
            coup->ch_select[c] = get_bits(gb, 2);
            if (coup->ch_select[c] == 3)
                num_gain++;
        } else
            coup->ch_select[c] = 2;
    }
    coup->coupling_point += get_bits1(gb) || (coup->coupling_point >> 1);

    sign  = get_bits(gb, 1);
2137
    scale = AAC_RENAME(cce_scale)[get_bits(gb, 2)];
2138 2139 2140 2141 2142 2143 2144 2145

    if ((ret = decode_ics(ac, sce, gb, 0, 0)))
        return ret;

    for (c = 0; c < num_gain; c++) {
        int idx  = 0;
        int cge  = 1;
        int gain = 0;
2146
        INTFLOAT gain_cache = FIXR10(1.);
2147 2148 2149
        if (c) {
            cge = coup->coupling_point == AFTER_IMDCT ? 1 : get_bits1(gb);
            gain = cge ? get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60: 0;
2150
            gain_cache = GET_GAIN(scale, gain);
2151 2152 2153 2154 2155 2156 2157 2158 2159 2160 2161 2162 2163 2164 2165 2166
        }
        if (coup->coupling_point == AFTER_IMDCT) {
            coup->gain[c][0] = gain_cache;
        } else {
            for (g = 0; g < sce->ics.num_window_groups; g++) {
                for (sfb = 0; sfb < sce->ics.max_sfb; sfb++, idx++) {
                    if (sce->band_type[idx] != ZERO_BT) {
                        if (!cge) {
                            int t = get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
                            if (t) {
                                int s = 1;
                                t = gain += t;
                                if (sign) {
                                    s  -= 2 * (t & 0x1);
                                    t >>= 1;
                                }
2167
                                gain_cache = GET_GAIN(scale, t) * s;
2168 2169 2170 2171 2172 2173 2174 2175 2176 2177 2178 2179 2180 2181 2182 2183 2184 2185 2186 2187 2188 2189 2190 2191 2192 2193 2194 2195 2196 2197 2198 2199 2200 2201 2202 2203 2204 2205 2206 2207 2208 2209 2210 2211 2212 2213 2214 2215 2216 2217 2218 2219 2220 2221 2222 2223 2224 2225 2226 2227 2228 2229 2230 2231 2232 2233 2234 2235 2236 2237 2238 2239 2240 2241 2242 2243 2244 2245 2246 2247 2248 2249 2250 2251 2252 2253 2254 2255 2256 2257 2258 2259 2260 2261 2262 2263 2264 2265 2266 2267 2268 2269 2270 2271 2272 2273 2274 2275 2276 2277 2278 2279 2280 2281 2282 2283 2284 2285 2286 2287 2288 2289 2290 2291 2292 2293 2294 2295 2296 2297 2298 2299 2300 2301 2302 2303 2304 2305 2306 2307 2308 2309 2310 2311 2312 2313 2314 2315 2316 2317 2318 2319 2320 2321 2322 2323 2324 2325 2326 2327 2328 2329 2330 2331 2332 2333 2334 2335 2336 2337 2338 2339 2340
                            }
                        }
                        coup->gain[c][idx] = gain_cache;
                    }
                }
            }
        }
    }
    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(AACContext *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(AACContext *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.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->channels == 1) {
            ac->oc[1].m4ac.sbr = 1;
            ac->oc[1].m4ac.ps = 1;
            ac->avctx->profile = FF_PROFILE_AAC_HE_V2;
            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 = FF_PROFILE_AAC_HE;
        }
        res = ff_decode_sbr_extension(ac, &che->sbr, gb, crc_flag, cnt, elem_type);
        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;
}

/**
 * Decode Temporal Noise Shaping filter coefficients and apply all-pole filters; reference: 4.6.9.3.
 *
 * @param   decode  1 if tool is used normally, 0 if tool is used in LTP.
 * @param   coef    spectral coefficients
 */
2341
static void apply_tns(INTFLOAT coef[1024], TemporalNoiseShaping *tns,
2342 2343 2344 2345 2346
                      IndividualChannelStream *ics, int decode)
{
    const int mmm = FFMIN(ics->tns_max_bands, ics->max_sfb);
    int w, filt, m, i;
    int bottom, top, order, start, end, size, inc;
2347 2348
    INTFLOAT lpc[TNS_MAX_ORDER];
    INTFLOAT tmp[TNS_MAX_ORDER+1];
2349 2350 2351 2352 2353 2354 2355 2356 2357 2358 2359 2360 2361 2362 2363 2364 2365 2366 2367 2368 2369 2370 2371 2372 2373 2374 2375 2376 2377

    for (w = 0; w < ics->num_windows; w++) {
        bottom = ics->num_swb;
        for (filt = 0; filt < tns->n_filt[w]; filt++) {
            top    = bottom;
            bottom = FFMAX(0, top - tns->length[w][filt]);
            order  = tns->order[w][filt];
            if (order == 0)
                continue;

            // tns_decode_coef
            compute_lpc_coefs(tns->coef[w][filt], order, lpc, 0, 0, 0);

            start = ics->swb_offset[FFMIN(bottom, mmm)];
            end   = ics->swb_offset[FFMIN(   top, mmm)];
            if ((size = end - start) <= 0)
                continue;
            if (tns->direction[w][filt]) {
                inc = -1;
                start = end - 1;
            } else {
                inc = 1;
            }
            start += w * 128;

            if (decode) {
                // ar filter
                for (m = 0; m < size; m++, start += inc)
                    for (i = 1; i <= FFMIN(m, order); i++)
2378
                        coef[start] -= AAC_MUL26(coef[start - i * inc], lpc[i - 1]);
2379 2380 2381 2382 2383
            } else {
                // ma filter
                for (m = 0; m < size; m++, start += inc) {
                    tmp[0] = coef[start];
                    for (i = 1; i <= FFMIN(m, order); i++)
2384
                        coef[start] += AAC_MUL26(tmp[i], lpc[i - 1]);
2385 2386 2387 2388 2389 2390 2391 2392 2393 2394 2395 2396
                    for (i = order; i > 0; i--)
                        tmp[i] = tmp[i - 1];
                }
            }
        }
    }
}

/**
 *  Apply windowing and MDCT to obtain the spectral
 *  coefficient from the predicted sample by LTP.
 */
2397 2398
static void windowing_and_mdct_ltp(AACContext *ac, INTFLOAT *out,
                                   INTFLOAT *in, IndividualChannelStream *ics)
2399
{
2400 2401 2402 2403
    const INTFLOAT *lwindow      = ics->use_kb_window[0] ? AAC_RENAME(ff_aac_kbd_long_1024) : AAC_RENAME(ff_sine_1024);
    const INTFLOAT *swindow      = ics->use_kb_window[0] ? AAC_RENAME(ff_aac_kbd_short_128) : AAC_RENAME(ff_sine_128);
    const INTFLOAT *lwindow_prev = ics->use_kb_window[1] ? AAC_RENAME(ff_aac_kbd_long_1024) : AAC_RENAME(ff_sine_1024);
    const INTFLOAT *swindow_prev = ics->use_kb_window[1] ? AAC_RENAME(ff_aac_kbd_short_128) : AAC_RENAME(ff_sine_128);
2404 2405 2406 2407

    if (ics->window_sequence[0] != LONG_STOP_SEQUENCE) {
        ac->fdsp->vector_fmul(in, in, lwindow_prev, 1024);
    } else {
2408
        memset(in, 0, 448 * sizeof(*in));
2409 2410 2411 2412 2413 2414
        ac->fdsp->vector_fmul(in + 448, in + 448, swindow_prev, 128);
    }
    if (ics->window_sequence[0] != LONG_START_SEQUENCE) {
        ac->fdsp->vector_fmul_reverse(in + 1024, in + 1024, lwindow, 1024);
    } else {
        ac->fdsp->vector_fmul_reverse(in + 1024 + 448, in + 1024 + 448, swindow, 128);
2415
        memset(in + 1024 + 576, 0, 448 * sizeof(*in));
2416 2417 2418 2419 2420 2421 2422 2423 2424 2425 2426 2427 2428 2429
    }
    ac->mdct_ltp.mdct_calc(&ac->mdct_ltp, out, in);
}

/**
 * Apply the long term prediction
 */
static void apply_ltp(AACContext *ac, SingleChannelElement *sce)
{
    const LongTermPrediction *ltp = &sce->ics.ltp;
    const uint16_t *offsets = sce->ics.swb_offset;
    int i, sfb;

    if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
2430 2431
        INTFLOAT *predTime = sce->ret;
        INTFLOAT *predFreq = ac->buf_mdct;
2432 2433 2434 2435 2436
        int16_t num_samples = 2048;

        if (ltp->lag < 1024)
            num_samples = ltp->lag + 1024;
        for (i = 0; i < num_samples; i++)
2437 2438
            predTime[i] = AAC_MUL30(sce->ltp_state[i + 2048 - ltp->lag], ltp->coef);
        memset(&predTime[i], 0, (2048 - i) * sizeof(*predTime));
2439 2440 2441 2442 2443 2444 2445 2446 2447 2448 2449 2450 2451 2452 2453 2454 2455 2456 2457

        ac->windowing_and_mdct_ltp(ac, predFreq, predTime, &sce->ics);

        if (sce->tns.present)
            ac->apply_tns(predFreq, &sce->tns, &sce->ics, 0);

        for (sfb = 0; sfb < FFMIN(sce->ics.max_sfb, MAX_LTP_LONG_SFB); sfb++)
            if (ltp->used[sfb])
                for (i = offsets[sfb]; i < offsets[sfb + 1]; i++)
                    sce->coeffs[i] += predFreq[i];
    }
}

/**
 * Update the LTP buffer for next frame
 */
static void update_ltp(AACContext *ac, SingleChannelElement *sce)
{
    IndividualChannelStream *ics = &sce->ics;
2458 2459 2460 2461
    INTFLOAT *saved     = sce->saved;
    INTFLOAT *saved_ltp = sce->coeffs;
    const INTFLOAT *lwindow = ics->use_kb_window[0] ? AAC_RENAME(ff_aac_kbd_long_1024) : AAC_RENAME(ff_sine_1024);
    const INTFLOAT *swindow = ics->use_kb_window[0] ? AAC_RENAME(ff_aac_kbd_short_128) : AAC_RENAME(ff_sine_128);
2462 2463 2464
    int i;

    if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
2465 2466
        memcpy(saved_ltp,       saved, 512 * sizeof(*saved_ltp));
        memset(saved_ltp + 576, 0,     448 * sizeof(*saved_ltp));
2467
        ac->fdsp->vector_fmul_reverse(saved_ltp + 448, ac->buf_mdct + 960,     &swindow[64],      64);
2468

2469
        for (i = 0; i < 64; i++)
2470
            saved_ltp[i + 512] = AAC_MUL31(ac->buf_mdct[1023 - i], swindow[63 - i]);
2471
    } else if (ics->window_sequence[0] == LONG_START_SEQUENCE) {
2472 2473
        memcpy(saved_ltp,       ac->buf_mdct + 512, 448 * sizeof(*saved_ltp));
        memset(saved_ltp + 576, 0,                  448 * sizeof(*saved_ltp));
2474
        ac->fdsp->vector_fmul_reverse(saved_ltp + 448, ac->buf_mdct + 960,     &swindow[64],      64);
2475

2476
        for (i = 0; i < 64; i++)
2477
            saved_ltp[i + 512] = AAC_MUL31(ac->buf_mdct[1023 - i], swindow[63 - i]);
2478 2479
    } else { // LONG_STOP or ONLY_LONG
        ac->fdsp->vector_fmul_reverse(saved_ltp,       ac->buf_mdct + 512,     &lwindow[512],     512);
2480

2481
        for (i = 0; i < 512; i++)
2482
            saved_ltp[i + 512] = AAC_MUL31(ac->buf_mdct[1023 - i], lwindow[511 - i]);
2483 2484 2485 2486 2487 2488 2489 2490 2491 2492 2493 2494 2495
    }

    memcpy(sce->ltp_state,      sce->ltp_state+1024, 1024 * sizeof(*sce->ltp_state));
    memcpy(sce->ltp_state+1024, sce->ret,            1024 * sizeof(*sce->ltp_state));
    memcpy(sce->ltp_state+2048, saved_ltp,           1024 * sizeof(*sce->ltp_state));
}

/**
 * Conduct IMDCT and windowing.
 */
static void imdct_and_windowing(AACContext *ac, SingleChannelElement *sce)
{
    IndividualChannelStream *ics = &sce->ics;
2496 2497 2498 2499 2500 2501 2502 2503
    INTFLOAT *in    = sce->coeffs;
    INTFLOAT *out   = sce->ret;
    INTFLOAT *saved = sce->saved;
    const INTFLOAT *swindow      = ics->use_kb_window[0] ? AAC_RENAME(ff_aac_kbd_short_128) : AAC_RENAME(ff_sine_128);
    const INTFLOAT *lwindow_prev = ics->use_kb_window[1] ? AAC_RENAME(ff_aac_kbd_long_1024) : AAC_RENAME(ff_sine_1024);
    const INTFLOAT *swindow_prev = ics->use_kb_window[1] ? AAC_RENAME(ff_aac_kbd_short_128) : AAC_RENAME(ff_sine_128);
    INTFLOAT *buf  = ac->buf_mdct;
    INTFLOAT *temp = ac->temp;
2504 2505 2506 2507 2508 2509
    int i;

    // imdct
    if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
        for (i = 0; i < 1024; i += 128)
            ac->mdct_small.imdct_half(&ac->mdct_small, buf + i, in + i);
2510
    } else {
2511
        ac->mdct.imdct_half(&ac->mdct, buf, in);
2512 2513 2514 2515 2516
#if USE_FIXED
        for (i=0; i<1024; i++)
          buf[i] = (buf[i] + 4) >> 3;
#endif /* USE_FIXED */
    }
2517 2518 2519 2520 2521 2522 2523 2524 2525 2526 2527

    /* window overlapping
     * NOTE: To simplify the overlapping code, all 'meaningless' short to long
     * and long to short transitions are considered to be short to short
     * transitions. This leaves just two cases (long to long and short to short)
     * with a little special sauce for EIGHT_SHORT_SEQUENCE.
     */
    if ((ics->window_sequence[1] == ONLY_LONG_SEQUENCE || ics->window_sequence[1] == LONG_STOP_SEQUENCE) &&
            (ics->window_sequence[0] == ONLY_LONG_SEQUENCE || ics->window_sequence[0] == LONG_START_SEQUENCE)) {
        ac->fdsp->vector_fmul_window(    out,               saved,            buf,         lwindow_prev, 512);
    } else {
2528
        memcpy(                         out,               saved,            448 * sizeof(*out));
2529 2530 2531 2532 2533 2534 2535

        if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
            ac->fdsp->vector_fmul_window(out + 448 + 0*128, saved + 448,      buf + 0*128, swindow_prev, 64);
            ac->fdsp->vector_fmul_window(out + 448 + 1*128, buf + 0*128 + 64, buf + 1*128, swindow,      64);
            ac->fdsp->vector_fmul_window(out + 448 + 2*128, buf + 1*128 + 64, buf + 2*128, swindow,      64);
            ac->fdsp->vector_fmul_window(out + 448 + 3*128, buf + 2*128 + 64, buf + 3*128, swindow,      64);
            ac->fdsp->vector_fmul_window(temp,              buf + 3*128 + 64, buf + 4*128, swindow,      64);
2536
            memcpy(                     out + 448 + 4*128, temp, 64 * sizeof(*out));
2537 2538
        } else {
            ac->fdsp->vector_fmul_window(out + 448,         saved + 448,      buf,         swindow_prev, 64);
2539
            memcpy(                     out + 576,         buf + 64,         448 * sizeof(*out));
2540 2541 2542 2543 2544
        }
    }

    // buffer update
    if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
2545
        memcpy(                     saved,       temp + 64,         64 * sizeof(*saved));
2546 2547 2548
        ac->fdsp->vector_fmul_window(saved + 64,  buf + 4*128 + 64, buf + 5*128, swindow, 64);
        ac->fdsp->vector_fmul_window(saved + 192, buf + 5*128 + 64, buf + 6*128, swindow, 64);
        ac->fdsp->vector_fmul_window(saved + 320, buf + 6*128 + 64, buf + 7*128, swindow, 64);
2549
        memcpy(                     saved + 448, buf + 7*128 + 64,  64 * sizeof(*saved));
2550
    } else if (ics->window_sequence[0] == LONG_START_SEQUENCE) {
2551 2552
        memcpy(                     saved,       buf + 512,        448 * sizeof(*saved));
        memcpy(                     saved + 448, buf + 7*128 + 64,  64 * sizeof(*saved));
2553
    } else { // LONG_STOP or ONLY_LONG
2554
        memcpy(                     saved,       buf + 512,        512 * sizeof(*saved));
2555 2556 2557 2558 2559 2560
    }
}

static void imdct_and_windowing_ld(AACContext *ac, SingleChannelElement *sce)
{
    IndividualChannelStream *ics = &sce->ics;
2561 2562 2563 2564 2565 2566 2567
    INTFLOAT *in    = sce->coeffs;
    INTFLOAT *out   = sce->ret;
    INTFLOAT *saved = sce->saved;
    INTFLOAT *buf  = ac->buf_mdct;
#if USE_FIXED
    int i;
#endif /* USE_FIXED */
2568 2569 2570 2571

    // imdct
    ac->mdct.imdct_half(&ac->mdct_ld, buf, in);

2572 2573 2574 2575 2576
#if USE_FIXED
    for (i = 0; i < 1024; i++)
        buf[i] = (buf[i] + 2) >> 2;
#endif /* USE_FIXED */

2577 2578 2579
    // window overlapping
    if (ics->use_kb_window[1]) {
        // AAC LD uses a low overlap sine window instead of a KBD window
2580 2581 2582
        memcpy(out, saved, 192 * sizeof(*out));
        ac->fdsp->vector_fmul_window(out + 192, saved + 192, buf, AAC_RENAME(ff_sine_128), 64);
        memcpy(                     out + 320, buf + 64, 192 * sizeof(*out));
2583
    } else {
2584
        ac->fdsp->vector_fmul_window(out, saved, buf, AAC_RENAME(ff_sine_512), 256);
2585 2586 2587
    }

    // buffer update
2588
    memcpy(saved, buf + 256, 256 * sizeof(*saved));
2589 2590 2591 2592
}

static void imdct_and_windowing_eld(AACContext *ac, SingleChannelElement *sce)
{
2593 2594 2595 2596
    INTFLOAT *in    = sce->coeffs;
    INTFLOAT *out   = sce->ret;
    INTFLOAT *saved = sce->saved;
    INTFLOAT *buf  = ac->buf_mdct;
2597 2598 2599 2600
    int i;
    const int n  = ac->oc[1].m4ac.frame_length_short ? 480 : 512;
    const int n2 = n >> 1;
    const int n4 = n >> 2;
2601 2602
    const INTFLOAT *const window = n == 480 ? ff_aac_eld_window_480 :
                                           AAC_RENAME(ff_aac_eld_window_512);
2603 2604 2605 2606 2607 2608 2609

    // Inverse transform, mapped to the conventional IMDCT by
    // Chivukula, R.K.; Reznik, Y.A.; Devarajan, V.,
    // "Efficient algorithms for MPEG-4 AAC-ELD, AAC-LD and AAC-LC filterbanks,"
    // International Conference on Audio, Language and Image Processing, ICALIP 2008.
    // URL: http://ieeexplore.ieee.org/stamp/stamp.jsp?tp=&arnumber=4590245&isnumber=4589950
    for (i = 0; i < n2; i+=2) {
2610
        INTFLOAT temp;
2611 2612 2613
        temp =  in[i    ]; in[i    ] = -in[n - 1 - i]; in[n - 1 - i] = temp;
        temp = -in[i + 1]; in[i + 1] =  in[n - 2 - i]; in[n - 2 - i] = temp;
    }
2614
#if !USE_FIXED
2615 2616 2617
    if (n == 480)
        ac->mdct480->imdct_half(ac->mdct480, buf, in, 1, -1.f/(16*1024*960));
    else
2618
#endif
2619
        ac->mdct.imdct_half(&ac->mdct_ld, buf, in);
2620 2621 2622 2623 2624 2625

#if USE_FIXED
    for (i = 0; i < 1024; i++)
      buf[i] = (buf[i] + 1) >> 1;
#endif /* USE_FIXED */

2626 2627 2628 2629 2630 2631 2632 2633 2634 2635 2636
    for (i = 0; i < n; i+=2) {
        buf[i] = -buf[i];
    }
    // Like with the regular IMDCT at this point we still have the middle half
    // of a transform but with even symmetry on the left and odd symmetry on
    // the right

    // window overlapping
    // The spec says to use samples [0..511] but the reference decoder uses
    // samples [128..639].
    for (i = n4; i < n2; i ++) {
2637 2638 2639 2640
        out[i - n4] = AAC_MUL31(   buf[    n2 - 1 - i] , window[i       - n4]) +
                      AAC_MUL31( saved[        i + n2] , window[i +   n - n4]) +
                      AAC_MUL31(-saved[n + n2 - 1 - i] , window[i + 2*n - n4]) +
                      AAC_MUL31(-saved[  2*n + n2 + i] , window[i + 3*n - n4]);
2641 2642
    }
    for (i = 0; i < n2; i ++) {
2643 2644 2645 2646
        out[n4 + i] = AAC_MUL31(   buf[              i] , window[i + n2       - n4]) +
                      AAC_MUL31(-saved[      n - 1 - i] , window[i + n2 +   n - n4]) +
                      AAC_MUL31(-saved[          n + i] , window[i + n2 + 2*n - n4]) +
                      AAC_MUL31( saved[2*n + n - 1 - i] , window[i + n2 + 3*n - n4]);
2647 2648
    }
    for (i = 0; i < n4; i ++) {
2649 2650 2651
        out[n2 + n4 + i] = AAC_MUL31(   buf[    i + n2] , window[i +   n - n4]) +
                           AAC_MUL31(-saved[n2 - 1 - i] , window[i + 2*n - n4]) +
                           AAC_MUL31(-saved[n + n2 + i] , window[i + 3*n - n4]);
2652 2653 2654
    }

    // buffer update
2655 2656
    memmove(saved + n, saved, 2 * n * sizeof(*saved));
    memcpy( saved,       buf,     n * sizeof(*saved));
2657 2658 2659 2660 2661 2662 2663 2664 2665 2666 2667 2668 2669 2670 2671 2672 2673 2674 2675 2676 2677 2678 2679 2680 2681 2682 2683 2684 2685 2686 2687 2688 2689 2690 2691 2692 2693 2694
}

/**
 * channel coupling transformation interface
 *
 * @param   apply_coupling_method   pointer to (in)dependent coupling function
 */
static void apply_channel_coupling(AACContext *ac, ChannelElement *cc,
                                   enum RawDataBlockType type, int elem_id,
                                   enum CouplingPoint coupling_point,
                                   void (*apply_coupling_method)(AACContext *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);
            }
        }
    }
}

/**
2695
 * Convert spectral data to samples, applying all supported tools as appropriate.
2696 2697 2698 2699 2700 2701 2702 2703 2704 2705 2706 2707 2708 2709 2710 2711 2712 2713 2714 2715
 */
static void spectral_to_sample(AACContext *ac)
{
    int i, type;
    void (*imdct_and_window)(AACContext *ac, SingleChannelElement *sce);
    switch (ac->oc[1].m4ac.object_type) {
    case AOT_ER_AAC_LD:
        imdct_and_window = imdct_and_windowing_ld;
        break;
    case AOT_ER_AAC_ELD:
        imdct_and_window = imdct_and_windowing_eld;
        break;
    default:
        imdct_and_window = ac->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)
2716
                    apply_channel_coupling(ac, che, type, i, BEFORE_TNS, AAC_RENAME(apply_dependent_coupling));
2717 2718 2719 2720 2721 2722 2723 2724 2725 2726 2727 2728 2729
                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->apply_ltp(ac, &che->ch[0]);
                        if (che->ch[1].ics.ltp.present && type == TYPE_CPE)
                            ac->apply_ltp(ac, &che->ch[1]);
                    }
                }
                if (che->ch[0].tns.present)
                    ac->apply_tns(che->ch[0].coeffs, &che->ch[0].tns, &che->ch[0].ics, 1);
                if (che->ch[1].tns.present)
                    ac->apply_tns(che->ch[1].coeffs, &che->ch[1].tns, &che->ch[1].ics, 1);
                if (type <= TYPE_CPE)
2730
                    apply_channel_coupling(ac, che, type, i, BETWEEN_TNS_AND_IMDCT, AAC_RENAME(apply_dependent_coupling));
2731 2732 2733 2734 2735 2736 2737 2738 2739 2740 2741 2742 2743 2744
                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->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->update_ltp(ac, &che->ch[1]);
                    }
                    if (ac->oc[1].m4ac.sbr > 0) {
                        ff_sbr_apply(ac, &che->sbr, type, che->ch[0].ret, che->ch[1].ret);
                    }
                }
                if (type <= TYPE_CCE)
2745 2746 2747 2748 2749 2750 2751 2752 2753 2754 2755 2756
                    apply_channel_coupling(ac, che, type, i, AFTER_IMDCT, AAC_RENAME(apply_independent_coupling));

#if USE_FIXED
                {
                    int j;
                    /* preparation for resampler */
                    for(j = 0; j<2048; j++){
                        che->ch[0].ret[j] = (int32_t)av_clipl_int32((int64_t)che->ch[0].ret[j]<<7)+0x8000;
                        che->ch[1].ret[j] = (int32_t)av_clipl_int32((int64_t)che->ch[1].ret[j]<<7)+0x8000;
                    }
                }
#endif /* USE_FIXED */
2757 2758 2759 2760 2761 2762 2763 2764 2765 2766 2767 2768 2769 2770 2771 2772 2773 2774 2775 2776 2777 2778 2779 2780 2781 2782 2783 2784 2785 2786 2787 2788 2789 2790 2791 2792 2793 2794 2795 2796 2797 2798 2799 2800 2801 2802 2803 2804 2805 2806 2807 2808 2809 2810 2811 2812 2813 2814 2815 2816 2817 2818 2819 2820 2821 2822 2823 2824 2825 2826 2827 2828 2829 2830 2831 2832 2833 2834 2835 2836 2837 2838 2839 2840 2841 2842 2843 2844 2845 2846 2847 2848 2849 2850 2851 2852 2853 2854 2855 2856 2857 2858 2859 2860 2861 2862 2863 2864 2865 2866 2867 2868 2869 2870 2871 2872 2873 2874 2875 2876 2877 2878 2879 2880 2881 2882 2883 2884 2885 2886 2887 2888 2889 2890 2891 2892 2893 2894 2895 2896 2897 2898 2899 2900 2901 2902 2903 2904 2905 2906 2907 2908 2909 2910 2911 2912 2913 2914 2915 2916 2917 2918 2919 2920 2921 2922 2923 2924 2925 2926 2927 2928 2929 2930 2931 2932 2933 2934 2935 2936 2937 2938 2939 2940 2941 2942 2943 2944 2945 2946 2947 2948 2949 2950 2951 2952 2953 2954 2955 2956 2957 2958 2959 2960 2961 2962 2963 2964 2965 2966 2967 2968 2969 2970 2971 2972 2973 2974 2975 2976 2977 2978 2979 2980 2981 2982 2983 2984 2985 2986 2987 2988 2989 2990 2991 2992 2993 2994 2995 2996 2997 2998 2999 3000 3001 3002 3003 3004 3005 3006 3007 3008 3009 3010 3011 3012 3013 3014 3015 3016 3017 3018 3019 3020 3021 3022 3023 3024 3025 3026 3027 3028 3029 3030 3031 3032 3033 3034 3035 3036 3037 3038 3039 3040 3041 3042 3043 3044 3045 3046 3047 3048 3049 3050 3051 3052 3053 3054 3055 3056 3057 3058 3059 3060 3061 3062 3063 3064 3065 3066 3067 3068 3069 3070 3071 3072 3073 3074 3075 3076 3077 3078 3079 3080 3081 3082 3083 3084 3085 3086 3087 3088 3089 3090 3091 3092 3093 3094 3095 3096 3097 3098 3099 3100 3101 3102 3103 3104 3105 3106 3107 3108 3109 3110 3111 3112 3113 3114 3115 3116 3117 3118 3119 3120 3121 3122 3123 3124 3125 3126 3127 3128 3129 3130 3131 3132 3133 3134 3135 3136 3137 3138 3139 3140 3141 3142 3143 3144 3145 3146 3147 3148 3149 3150 3151 3152 3153 3154 3155 3156 3157 3158 3159 3160 3161 3162 3163 3164
                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(AACContext *ac, GetBitContext *gb)
{
    int size;
    AACADTSHeaderInfo hdr_info;
    uint8_t layout_map[MAX_ELEM_ID*4][3];
    int layout_map_tags, ret;

    size = avpriv_aac_parse_header(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 = set_default_channel_config(ac->avctx,
                                                  layout_map,
                                                  &layout_map_tags,
                                                  hdr_info.chan_config)) < 0)
                return ret;
            if ((ret = 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 (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, void *data,
                               int *got_frame_ptr, GetBitContext *gb)
{
    AACContext *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 = data;

    if ((err = frame_configure_elements(avctx)) < 0)
        return err;

    // The FF_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 < tags_per_config[chan_config]; i++) {
        const int elem_type = aac_channel_layout_map[chan_config-1][i][0];
        const int elem_id   = aac_channel_layout_map[chan_config-1][i][1];
        if (!(che=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 = decode_ics(ac, &che->ch[0], gb, 0, 0);
            break;
        case TYPE_CPE:
            err = decode_cpe(ac, gb, che);
            break;
        case TYPE_LFE:
            err = decode_ics(ac, &che->ch[0], gb, 0, 0);
            break;
        }
        if (err < 0)
            return err;
    }

    spectral_to_sample(ac);

    ac->frame->nb_samples = samples;
    ac->frame->sample_rate = avctx->sample_rate;
    *got_frame_ptr = 1;

    skip_bits_long(gb, get_bits_left(gb));
    return 0;
}

static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
                                int *got_frame_ptr, GetBitContext *gb, AVPacket *avpkt)
{
    AACContext *ac = avctx->priv_data;
    ChannelElement *che = NULL, *che_prev = NULL;
    enum RawDataBlockType elem_type, elem_type_prev = TYPE_END;
    int err, elem_id;
    int samples = 0, multiplier, audio_found = 0, pce_found = 0;
    int is_dmono, sce_count = 0;

    ac->frame = data;

    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 FF_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;
    // 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->channels && elem_type != TYPE_PCE) {
            err = AVERROR_INVALIDDATA;
            goto fail;
        }

        if (elem_type < TYPE_DSE) {
            if (!(che=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);
                err = AVERROR_INVALIDDATA;
                goto fail;
            }
            samples = 1024;
            che->present = 1;
        }

        switch (elem_type) {

        case TYPE_SCE:
            err = 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 = decode_cce(ac, gb, che);
            break;

        case TYPE_LFE:
            err = 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];
            int tags;
            push_output_configuration(ac);
            tags = decode_pce(avctx, &ac->oc[1].m4ac, layout_map, gb);
            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");
            } else {
                err = 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);
                    err = AVERROR_INVALIDDATA;
                    goto fail;
            }
            while (elem_id > 0)
                elem_id -= decode_extension_payload(ac, gb, elem_id, che_prev, elem_type_prev);
            err = 0; /* FIXME */
            break;

        default:
            err = AVERROR_BUG; /* should not happen, but keeps compiler happy */
            break;
        }

        che_prev       = che;
        elem_type_prev = elem_type;

        if (err)
            goto fail;

        if (get_bits_left(gb) < 3) {
            av_log(avctx, AV_LOG_ERROR, overread_err);
            err = AVERROR_INVALIDDATA;
            goto fail;
        }
    }

    if (!avctx->channels) {
        *got_frame_ptr = 0;
        return 0;
    }

    spectral_to_sample(ac);

    multiplier = (ac->oc[1].m4ac.sbr == 1) ? ac->oc[1].m4ac.ext_sample_rate > ac->oc[1].m4ac.sample_rate : 0;
    samples <<= multiplier;

    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 (multiplier) {
        int side_size;
        const uint8_t *side = av_packet_get_side_data(avpkt, AV_PKT_DATA_SKIP_SAMPLES, &side_size);
        if (side && side_size>=4)
            AV_WL32(side, 2*AV_RL32(side));
    }

    if (!ac->frame->data[0] && samples) {
        av_log(avctx, AV_LOG_ERROR, "no frame data found\n");
        err = AVERROR_INVALIDDATA;
        goto fail;
    }

    if (samples) {
        ac->frame->nb_samples = samples;
        ac->frame->sample_rate = avctx->sample_rate;
    } else
        av_frame_unref(ac->frame);
    *got_frame_ptr = !!samples;

    /* for dual-mono audio (SCE + SCE) */
    is_dmono = ac->dmono_mode && sce_count == 2 &&
               ac->oc[1].channel_layout == (AV_CH_FRONT_LEFT | AV_CH_FRONT_RIGHT);
    if (is_dmono) {
        if (ac->dmono_mode == 1)
            ((AVFrame *)data)->data[1] =((AVFrame *)data)->data[0];
        else if (ac->dmono_mode == 2)
            ((AVFrame *)data)->data[0] =((AVFrame *)data)->data[1];
    }

    return 0;
fail:
    pop_output_configuration(ac);
    return err;
}

static int aac_decode_frame(AVCodecContext *avctx, void *data,
                            int *got_frame_ptr, AVPacket *avpkt)
{
    AACContext *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;
    int new_extradata_size;
    const uint8_t *new_extradata = av_packet_get_side_data(avpkt,
                                       AV_PKT_DATA_NEW_EXTRADATA,
                                       &new_extradata_size);
    int 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 && 0) {
        av_free(avctx->extradata);
        avctx->extradata = av_mallocz(new_extradata_size +
                                      FF_INPUT_BUFFER_PADDING_SIZE);
        if (!avctx->extradata)
            return AVERROR(ENOMEM);
        avctx->extradata_size = new_extradata_size;
        memcpy(avctx->extradata, new_extradata, new_extradata_size);
        push_output_configuration(ac);
        if (decode_audio_specific_config(ac, ac->avctx, &ac->oc[1].m4ac,
                                         avctx->extradata,
                                         avctx->extradata_size*8, 1) < 0) {
            pop_output_configuration(ac);
            return AVERROR_INVALIDDATA;
        }
    }

    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_bits(&gb, buf, buf_size * 8)) < 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, data, got_frame_ptr, &gb);
        break;
    default:
        err = aac_decode_frame_int(avctx, data, 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;
}

static av_cold int aac_decode_close(AVCodecContext *avctx)
{
    AACContext *ac = avctx->priv_data;
    int i, type;

    for (i = 0; i < MAX_ELEM_ID; i++) {
        for (type = 0; type < 4; type++) {
            if (ac->che[type][i])
                ff_aac_sbr_ctx_close(&ac->che[type][i]->sbr);
            av_freep(&ac->che[type][i]);
        }
    }

    ff_mdct_end(&ac->mdct);
    ff_mdct_end(&ac->mdct_small);
    ff_mdct_end(&ac->mdct_ld);
    ff_mdct_end(&ac->mdct_ltp);
3165
#if !USE_FIXED
3166
    ff_imdct15_uninit(&ac->mdct480);
3167
#endif
3168 3169 3170 3171 3172 3173 3174 3175 3176 3177 3178
    av_freep(&ac->fdsp);
    return 0;
}

static void aacdec_init(AACContext *c)
{
    c->imdct_and_windowing                      = imdct_and_windowing;
    c->apply_ltp                                = apply_ltp;
    c->apply_tns                                = apply_tns;
    c->windowing_and_mdct_ltp                   = windowing_and_mdct_ltp;
    c->update_ltp                               = update_ltp;
3179 3180 3181 3182
#if USE_FIXED
    c->vector_pow43                             = vector_pow43;
    c->subband_scale                            = subband_scale;
#endif
3183

3184
#if !USE_FIXED
3185 3186
    if(ARCH_MIPS)
        ff_aacdec_init_mips(c);
3187
#endif /* !USE_FIXED */
3188 3189 3190 3191 3192 3193 3194 3195 3196 3197 3198 3199 3200 3201 3202 3203 3204 3205 3206 3207 3208 3209 3210 3211 3212 3213 3214 3215 3216 3217 3218 3219 3220 3221 3222 3223
}
/**
 * AVOptions for Japanese DTV specific extensions (ADTS only)
 */
#define AACDEC_FLAGS AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM
static const AVOption options[] = {
    {"dual_mono_mode", "Select the channel to decode for dual mono",
     offsetof(AACContext, force_dmono_mode), AV_OPT_TYPE_INT, {.i64=-1}, -1, 2,
     AACDEC_FLAGS, "dual_mono_mode"},

    {"auto", "autoselection",            0, AV_OPT_TYPE_CONST, {.i64=-1}, INT_MIN, INT_MAX, AACDEC_FLAGS, "dual_mono_mode"},
    {"main", "Select Main/Left channel", 0, AV_OPT_TYPE_CONST, {.i64= 1}, INT_MIN, INT_MAX, AACDEC_FLAGS, "dual_mono_mode"},
    {"sub" , "Select Sub/Right channel", 0, AV_OPT_TYPE_CONST, {.i64= 2}, INT_MIN, INT_MAX, AACDEC_FLAGS, "dual_mono_mode"},
    {"both", "Select both channels",     0, AV_OPT_TYPE_CONST, {.i64= 0}, INT_MIN, INT_MAX, AACDEC_FLAGS, "dual_mono_mode"},

    {NULL},
};

static const AVClass aac_decoder_class = {
    .class_name = "AAC decoder",
    .item_name  = av_default_item_name,
    .option     = options,
    .version    = LIBAVUTIL_VERSION_INT,
};

static const AVProfile profiles[] = {
    { FF_PROFILE_AAC_MAIN,  "Main"     },
    { FF_PROFILE_AAC_LOW,   "LC"       },
    { FF_PROFILE_AAC_SSR,   "SSR"      },
    { FF_PROFILE_AAC_LTP,   "LTP"      },
    { FF_PROFILE_AAC_HE,    "HE-AAC"   },
    { FF_PROFILE_AAC_HE_V2, "HE-AACv2" },
    { FF_PROFILE_AAC_LD,    "LD"       },
    { FF_PROFILE_AAC_ELD,   "ELD"      },
    { FF_PROFILE_UNKNOWN },
};