aacdec.c 104 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 )
 *
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 * AAC LATM decoder
 * Copyright (c) 2008-2010 Paul Kendall <paul@kcbbs.gen.nz>
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 * Copyright (c) 2010      Janne Grunau <janne-libav@jannau.net>
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 *
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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
 */

/**
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 * @file
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 * AAC decoder
 * @author Oded Shimon  ( ods15 ods15 dyndns org )
 * @author Maxim Gavrilov ( maxim.gavrilov gmail com )
 */

/*
 * 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
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 * Y                    Long Term Prediction
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 * Y                    intensity stereo
 * Y                    channel coupling
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 * Y                    frequency domain prediction
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 * 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
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 * Y                    Spectral Band Replication
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 * 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)
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 * Y                    Parametric Stereo
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 * N                    Direct Stream Transfer
 *
 * Note: - HE AAC v1 comprises LC AAC with Spectral Band Replication.
 *       - HE AAC v2 comprises LC AAC with Spectral Band Replication and
           Parametric Stereo.
 */

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#include "libavutil/float_dsp.h"
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#include "avcodec.h"
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#include "internal.h"
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#include "get_bits.h"
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#include "dsputil.h"
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#include "fft.h"
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#include "fmtconvert.h"
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#include "lpc.h"
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#include "kbdwin.h"
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#include "sinewin.h"
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#include "aac.h"
#include "aactab.h"
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#include "aacdectab.h"
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#include "cbrt_tablegen.h"
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#include "sbr.h"
#include "aacsbr.h"
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#include "mpeg4audio.h"
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#include "aacadtsdec.h"
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#include "libavutil/intfloat.h"
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#include <assert.h>
#include <errno.h>
#include <math.h>
#include <string.h>

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#if ARCH_ARM
#   include "arm/aac.h"
#endif

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static VLC vlc_scalefactors;
static VLC vlc_spectral[11];

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#define overread_err "Input buffer exhausted before END element found\n"
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static int count_channels(uint8_t (*layout)[3], int tags)
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{
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    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);
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    }
    return sum;
}

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/**
 * 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
 */
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static av_cold int che_configure(AACContext *ac,
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                                 enum ChannelPosition che_pos,
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                                 int type, int id, int *channels)
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{
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    if (che_pos) {
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        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);
        }
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        if (type != TYPE_CCE) {
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            if (*channels >= MAX_CHANNELS - (type == TYPE_CPE || (type == TYPE_SCE && ac->oc[1].m4ac.ps == 1))) {
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                av_log(ac->avctx, AV_LOG_ERROR, "Too many channels\n");
                return AVERROR_INVALIDDATA;
            }
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            ac->output_element[(*channels)++] = &ac->che[type][id]->ch[0];
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            if (type == TYPE_CPE ||
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                (type == TYPE_SCE && ac->oc[1].m4ac.ps == 1)) {
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                ac->output_element[(*channels)++] = &ac->che[type][id]->ch[1];
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            }
        }
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    } else {
        if (ac->che[type][id])
            ff_aac_sbr_ctx_close(&ac->che[type][id]->sbr);
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        av_freep(&ac->che[type][id]);
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    }
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    return 0;
}

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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 */
    ac->frame.nb_samples = 2048;
    if ((ret = avctx->get_buffer(avctx, &ac->frame)) < 0) {
        av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
        return ret;
    }

    /* map output channel pointers to AVFrame data */
    for (ch = 0; ch < avctx->channels; ch++) {
        if (ac->output_element[ch])
            ac->output_element[ch]->ret = (float *)ac->frame.extended_data[ch];
    }

    return 0;
}

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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],
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                       uint8_t (*layout_map)[3], int offset, uint64_t left,
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    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) {
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                if (pos == AAC_CHANNEL_FRONT && !first_cpe) {
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                    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;
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    struct elem_to_channel e2c_vec[4*MAX_ELEM_ID] = {{ 0 }};
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    int num_front_channels, num_side_channels, num_back_channels;
    uint64_t layout;

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

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    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;

    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) {
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        i += assign_pair(e2c_vec, layout_map, i,
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                         AV_CH_FRONT_LEFT_OF_CENTER,
                         AV_CH_FRONT_RIGHT_OF_CENTER,
                         AAC_CHANNEL_FRONT);
        num_front_channels -= 2;
    }
    if (num_front_channels >= 2) {
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        i += assign_pair(e2c_vec, layout_map, i,
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                         AV_CH_FRONT_LEFT,
                         AV_CH_FRONT_RIGHT,
                         AAC_CHANNEL_FRONT);
        num_front_channels -= 2;
    }
    while (num_front_channels >= 2) {
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        i += assign_pair(e2c_vec, layout_map, i,
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                         UINT64_MAX,
                         UINT64_MAX,
                         AAC_CHANNEL_FRONT);
        num_front_channels -= 2;
    }

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

    while (num_back_channels >= 4) {
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        i += assign_pair(e2c_vec, layout_map, i,
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                         UINT64_MAX,
                         UINT64_MAX,
                         AAC_CHANNEL_BACK);
        num_back_channels -= 2;
    }
    if (num_back_channels >= 2) {
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        i += assign_pair(e2c_vec, layout_map, i,
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                         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;
}

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/**
 * 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] = 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) {
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    if (ac->oc[1].status != OC_LOCKED && ac->oc[0].status != OC_NONE) {
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        ac->oc[1] = ac->oc[0];
        ac->avctx->channels = ac->oc[1].channels;
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        ac->avctx->channel_layout = ac->oc[1].channel_layout;
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    }
}

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/**
 * Configure output channel order based on the current program configuration element.
 *
 * @return  Returns error status. 0 - OK, !0 - error
 */
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static int output_configure(AACContext *ac,
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                            uint8_t layout_map[MAX_ELEM_ID*4][3], int tags,
                            enum OCStatus oc_type, int get_new_frame)
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{
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    AVCodecContext *avctx = ac->avctx;
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    int i, channels = 0, ret;
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    uint64_t layout = 0;
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    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;
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    }
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    // 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 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, id, &channels);
        if (ret < 0)
            return ret;
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    }
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    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;
        }
    }
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    memcpy(ac->tag_che_map, ac->che, 4 * MAX_ELEM_ID * sizeof(ac->che[0][0]));
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    if (layout) avctx->channel_layout = layout;
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    ac->oc[1].channel_layout = layout;
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    avctx->channels = ac->oc[1].channels = channels;
    ac->oc[1].status = oc_type;
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    if (get_new_frame) {
        if ((ret = frame_configure_elements(ac->avctx)) < 0)
            return ret;
    }

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    return 0;
}

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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));
                }
            }
        }
    }
}

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/**
 * Set up channel positions based on a default channel configuration
 * as specified in table 1.17.
 *
 * @return  Returns error status. 0 - OK, !0 - error
 */
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static int set_default_channel_config(AVCodecContext *avctx,
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                                              uint8_t (*layout_map)[3],
                                              int *tags,
                                              int channel_config)
{
    if (channel_config < 1 || channel_config > 7) {
        av_log(avctx, AV_LOG_ERROR, "invalid default channel configuration (%d)\n",
               channel_config);
        return -1;
    }
    *tags = tags_per_config[channel_config];
    memcpy(layout_map, aac_channel_layout_map[channel_config-1], *tags * sizeof(*layout_map));
    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.
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    if (!ac->oc[1].m4ac.chan_config) {
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        return ac->tag_che_map[type][elem_id];
    }
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    // Allow single CPE stereo files to be signalled with mono configuration.
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    if (!ac->tags_mapped && type == TYPE_CPE && ac->oc[1].m4ac.chan_config == 1) {
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        uint8_t layout_map[MAX_ELEM_ID*4][3];
        int layout_map_tags;
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        push_output_configuration(ac);
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        av_log(ac->avctx, AV_LOG_DEBUG, "mono with CPE\n");

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        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,
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                             OC_TRIAL_FRAME, 1) < 0)
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            return NULL;

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        ac->oc[1].m4ac.chan_config = 2;
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        ac->oc[1].m4ac.ps = 0;
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    }
    // And vice-versa
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    if (!ac->tags_mapped && type == TYPE_SCE && ac->oc[1].m4ac.chan_config == 2) {
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        uint8_t layout_map[MAX_ELEM_ID*4][3];
        int layout_map_tags;
        push_output_configuration(ac);

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        av_log(ac->avctx, AV_LOG_DEBUG, "stereo with SCE\n");

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        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,
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                             OC_TRIAL_FRAME, 1) < 0)
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            return NULL;

        ac->oc[1].m4ac.chan_config = 1;
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        if (ac->oc[1].m4ac.sbr)
            ac->oc[1].m4ac.ps = -1;
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    }
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    // For indexed channel configurations map the channels solely based on position.
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    switch (ac->oc[1].m4ac.chan_config) {
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    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 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 */
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        if (ac->tags_mapped == tags_per_config[ac->oc[1].m4ac.chan_config] - 1 && (type == TYPE_LFE || type == TYPE_SCE)) {
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            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:
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        if (ac->tags_mapped == 2 && ac->oc[1].m4ac.chan_config == 4 && type == TYPE_SCE) {
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            ac->tags_mapped++;
            return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][1];
        }
    case 3:
    case 2:
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        if (ac->tags_mapped == (ac->oc[1].m4ac.chan_config != 2) && type == TYPE_CPE) {
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            ac->tags_mapped++;
            return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][0];
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        } else if (ac->oc[1].m4ac.chan_config == 2) {
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            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;
    }
}

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/**
 * Decode an array of 4 bit element IDs, optionally interleaved with a stereo/mono switching bit.
 *
 * @param type speaker type/position for these channels
 */
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static void decode_channel_map(uint8_t layout_map[][3],
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                               enum ChannelPosition type,
                               GetBitContext *gb, int n)
{
    while (n--) {
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        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;
613 614
        default:
            av_assert0(0);
615 616 617 618 619
        }
        layout_map[0][0] = syn_ele;
        layout_map[0][1] = get_bits(gb, 4);
        layout_map[0][2] = type;
        layout_map++;
620 621 622 623 624 625 626 627
    }
}

/**
 * Decode program configuration element; reference: table 4.2.
 *
 * @return  Returns error status. 0 - OK, !0 - error
 */
628
static int decode_pce(AVCodecContext *avctx, MPEG4AudioConfig *m4ac,
629
                      uint8_t (*layout_map)[3],
630 631
                      GetBitContext *gb)
{
632
    int num_front, num_side, num_back, num_lfe, num_assoc_data, num_cc, sampling_index;
633
    int comment_len;
634
    int tags;
635 636 637

    skip_bits(gb, 2);  // object_type

638
    sampling_index = get_bits(gb, 4);
639 640
    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");
641

642 643 644 645 646 647 648
    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);

649 650 651 652
    if (get_bits1(gb))
        skip_bits(gb, 4); // mono_mixdown_tag
    if (get_bits1(gb))
        skip_bits(gb, 4); // stereo_mixdown_tag
653

654 655
    if (get_bits1(gb))
        skip_bits(gb, 3); // mixdown_coeff_index and pseudo_surround
656

657
    if (get_bits_left(gb) < 4 * (num_front + num_side + num_back + num_lfe + num_assoc_data + num_cc)) {
658
        av_log(avctx, AV_LOG_ERROR, "decode_pce: " overread_err);
659 660
        return -1;
    }
661 662 663 664 665 666 667 668
    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;
669 670 671

    skip_bits_long(gb, 4 * num_assoc_data);

672 673
    decode_channel_map(layout_map + tags, AAC_CHANNEL_CC,    gb, num_cc);
    tags += num_cc;
674 675 676 677

    align_get_bits(gb);

    /* comment field, first byte is length */
678 679
    comment_len = get_bits(gb, 8) * 8;
    if (get_bits_left(gb) < comment_len) {
680
        av_log(avctx, AV_LOG_ERROR, "decode_pce: " overread_err);
681 682 683
        return -1;
    }
    skip_bits_long(gb, comment_len);
684
    return tags;
685
}
686

687 688 689
/**
 * Decode GA "General Audio" specific configuration; reference: table 4.1.
 *
690 691 692
 * @param   ac          pointer to AACContext, may be null
 * @param   avctx       pointer to AVCCodecContext, used for logging
 *
693 694
 * @return  Returns error status. 0 - OK, !0 - error
 */
695 696
static int decode_ga_specific_config(AACContext *ac, AVCodecContext *avctx,
                                     GetBitContext *gb,
697
                                     MPEG4AudioConfig *m4ac,
698 699
                                     int channel_config)
{
700
    int extension_flag, ret;
701 702
    uint8_t layout_map[MAX_ELEM_ID*4][3];
    int tags = 0;
703

704
    if (get_bits1(gb)) { // frameLengthFlag
705
        av_log_missing_feature(avctx, "960/120 MDCT window", 1);
706
        return AVERROR_PATCHWELCOME;
707 708 709 710 711 712
    }

    if (get_bits1(gb))       // dependsOnCoreCoder
        skip_bits(gb, 14);   // coreCoderDelay
    extension_flag = get_bits1(gb);

713 714
    if (m4ac->object_type == AOT_AAC_SCALABLE ||
        m4ac->object_type == AOT_ER_AAC_SCALABLE)
715 716 717 718
        skip_bits(gb, 3);     // layerNr

    if (channel_config == 0) {
        skip_bits(gb, 4);  // element_instance_tag
719 720 721
        tags = decode_pce(avctx, m4ac, layout_map, gb);
        if (tags < 0)
            return tags;
722
    } else {
723
        if ((ret = set_default_channel_config(avctx, layout_map, &tags, channel_config)))
724 725
            return ret;
    }
726

727
    if (count_channels(layout_map, tags) > 1) {
728 729 730 731
        m4ac->ps = 0;
    } else if (m4ac->sbr == 1 && m4ac->ps == -1)
        m4ac->ps = 1;

732
    if (ac && (ret = output_configure(ac, layout_map, tags, OC_GLOBAL_HDR, 0)))
733 734 735
        return ret;

    if (extension_flag) {
736
        switch (m4ac->object_type) {
737 738 739 740 741 742 743 744 745
        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:
            skip_bits(gb, 3);  /* aacSectionDataResilienceFlag
746 747 748
                                    * aacScalefactorDataResilienceFlag
                                    * aacSpectralDataResilienceFlag
                                    */
749
            break;
750 751 752 753 754 755 756 757 758
        }
        skip_bits1(gb);    // extensionFlag3 (TBD in version 3)
    }
    return 0;
}

/**
 * Decode audio specific configuration; reference: table 1.13.
 *
759 760 761
 * @param   ac          pointer to AACContext, may be null
 * @param   avctx       pointer to AVCCodecContext, used for logging
 * @param   m4ac        pointer to MPEG4AudioConfig, used for parsing
762 763 764
 * @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
765
 *
766
 * @return  Returns error status or number of consumed bits. <0 - error
767
 */
768
static int decode_audio_specific_config(AACContext *ac,
769 770
                                        AVCodecContext *avctx,
                                        MPEG4AudioConfig *m4ac,
771 772
                                        const uint8_t *data, int bit_size,
                                        int sync_extension)
773
{
774 775 776
    GetBitContext gb;
    int i;

777 778 779
    av_dlog(avctx, "audio specific config size %d\n", bit_size >> 3);
    for (i = 0; i < bit_size >> 3; i++)
         av_dlog(avctx, "%02x ", data[i]);
780 781
    av_dlog(avctx, "\n");

782
    init_get_bits(&gb, data, bit_size);
783

784
    if ((i = avpriv_mpeg4audio_get_config(m4ac, data, bit_size, sync_extension)) < 0)
785
        return -1;
786
    if (m4ac->sampling_index > 12) {
787
        av_log(avctx, AV_LOG_ERROR, "invalid sampling rate index %d\n", m4ac->sampling_index);
788 789 790 791 792
        return -1;
    }

    skip_bits_long(&gb, i);

793
    switch (m4ac->object_type) {
794
    case AOT_AAC_MAIN:
795
    case AOT_AAC_LC:
796
    case AOT_AAC_LTP:
797
        if (decode_ga_specific_config(ac, avctx, &gb, m4ac, m4ac->chan_config))
798 799 800
            return -1;
        break;
    default:
801
        av_log(avctx, AV_LOG_ERROR, "Audio object type %s%d is not supported.\n",
802
               m4ac->sbr == 1? "SBR+" : "", m4ac->object_type);
803 804
        return -1;
    }
805

806 807 808 809
    av_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);

810
    return get_bits_count(&gb);
811 812
}

813 814 815 816 817 818 819
/**
 * linear congruential pseudorandom number generator
 *
 * @param   previous_val    pointer to the current state of the generator
 *
 * @return  Returns a 32-bit pseudorandom integer
 */
820
static av_always_inline int lcg_random(unsigned previous_val)
821
{
822 823
    union { unsigned u; int s; } v = { previous_val * 1664525u + 1013904223 };
    return v.s;
824 825
}

826
static av_always_inline void reset_predict_state(PredictorState *ps)
827 828 829
{
    ps->r0   = 0.0f;
    ps->r1   = 0.0f;
830 831 832 833 834 835
    ps->cor0 = 0.0f;
    ps->cor1 = 0.0f;
    ps->var0 = 1.0f;
    ps->var1 = 1.0f;
}

836 837
static void reset_all_predictors(PredictorState *ps)
{
838 839 840 841 842
    int i;
    for (i = 0; i < MAX_PREDICTORS; i++)
        reset_predict_state(&ps[i]);
}

843 844 845 846 847 848 849 850 851 852 853 854 855 856 857 858
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;
}

859 860
static void reset_predictor_group(PredictorState *ps, int group_num)
{
861
    int i;
862
    for (i = group_num - 1; i < MAX_PREDICTORS; i += 30)
863 864 865
        reset_predict_state(&ps[i]);
}

866 867 868 869 870 871
#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);

872
static av_cold int aac_decode_init(AVCodecContext *avctx)
873
{
874
    AACContext *ac = avctx->priv_data;
875

876
    ac->avctx = avctx;
877
    ac->oc[1].m4ac.sample_rate = avctx->sample_rate;
878

879
    avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
880

881
    if (avctx->extradata_size > 0) {
882
        if (decode_audio_specific_config(ac, ac->avctx, &ac->oc[1].m4ac,
883
                                         avctx->extradata,
884
                                         avctx->extradata_size*8, 1) < 0)
885
            return -1;
886 887
    } else {
        int sr, i;
888 889
        uint8_t layout_map[MAX_ELEM_ID*4][3];
        int layout_map_tags;
890 891

        sr = sample_rate_idx(avctx->sample_rate);
892 893 894 895
        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;
896 897 898 899 900 901 902

        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;
        }
903
        ac->oc[1].m4ac.chan_config = i;
904

905
        if (ac->oc[1].m4ac.chan_config) {
906
            int ret = set_default_channel_config(avctx, layout_map,
907
                &layout_map_tags, ac->oc[1].m4ac.chan_config);
908
            if (!ret)
909
                output_configure(ac, layout_map, layout_map_tags,
910
                                 OC_GLOBAL_HDR, 0);
911
            else if (avctx->err_recognition & AV_EF_EXPLODE)
912
                return AVERROR_INVALIDDATA;
913
        }
914
    }
915

916 917 918 919 920 921 922 923 924 925 926
    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);
927

928 929
    ff_aac_sbr_init();

930
    ff_dsputil_init(&ac->dsp, avctx);
931
    ff_fmt_convert_init(&ac->fmt_conv, avctx);
932
    avpriv_float_dsp_init(&ac->fdsp, avctx->flags & CODEC_FLAG_BITEXACT);
933

934 935
    ac->random_state = 0x1f2e3d4c;

936
    ff_aac_tableinit();
937

938
    INIT_VLC_STATIC(&vlc_scalefactors,7,FF_ARRAY_ELEMS(ff_aac_scalefactor_code),
939 940 941
                    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);
942

943 944 945
    ff_mdct_init(&ac->mdct,       11, 1, 1.0 / (32768.0 * 1024.0));
    ff_mdct_init(&ac->mdct_small,  8, 1, 1.0 / (32768.0 * 128.0));
    ff_mdct_init(&ac->mdct_ltp,   11, 0, -2.0 * 32768.0);
946 947 948
    // window initialization
    ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024);
    ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128);
949 950
    ff_init_ff_sine_windows(10);
    ff_init_ff_sine_windows( 7);
951

952
    cbrt_tableinit();
953

954 955 956
    avcodec_get_frame_defaults(&ac->frame);
    avctx->coded_frame = &ac->frame;

957 958 959
    return 0;
}

960 961 962
/**
 * Skip data_stream_element; reference: table 4.10.
 */
963
static int skip_data_stream_element(AACContext *ac, GetBitContext *gb)
964
{
965 966 967 968 969 970
    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);
971 972

    if (get_bits_left(gb) < 8 * count) {
973
        av_log(ac->avctx, AV_LOG_ERROR, "skip_data_stream_element: "overread_err);
974 975
        return -1;
    }
976
    skip_bits_long(gb, 8 * count);
977
    return 0;
978 979
}

980 981 982
static int decode_prediction(AACContext *ac, IndividualChannelStream *ics,
                             GetBitContext *gb)
{
983 984 985 986
    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) {
987
            av_log(ac->avctx, AV_LOG_ERROR, "Invalid Predictor Reset Group.\n");
988 989 990
            return -1;
        }
    }
991
    for (sfb = 0; sfb < FFMIN(ics->max_sfb, ff_aac_pred_sfb_max[ac->oc[1].m4ac.sampling_index]); sfb++) {
992 993 994 995 996
        ics->prediction_used[sfb] = get_bits1(gb);
    }
    return 0;
}

997 998 999
/**
 * Decode Long Term Prediction data; reference: table 4.xx.
 */
1000
static void decode_ltp(LongTermPrediction *ltp,
1001 1002 1003 1004 1005
                       GetBitContext *gb, uint8_t max_sfb)
{
    int sfb;

    ltp->lag  = get_bits(gb, 11);
1006
    ltp->coef = ltp_coef[get_bits(gb, 3)];
1007 1008 1009 1010
    for (sfb = 0; sfb < FFMIN(max_sfb, MAX_LTP_LONG_SFB); sfb++)
        ltp->used[sfb] = get_bits1(gb);
}

1011 1012 1013
/**
 * Decode Individual Channel Stream info; reference: table 4.6.
 */
1014
static int decode_ics_info(AACContext *ac, IndividualChannelStream *ics,
1015
                           GetBitContext *gb)
1016
{
1017
    if (get_bits1(gb)) {
1018
        av_log(ac->avctx, AV_LOG_ERROR, "Reserved bit set.\n");
1019
        return AVERROR_INVALIDDATA;
1020 1021 1022
    }
    ics->window_sequence[1] = ics->window_sequence[0];
    ics->window_sequence[0] = get_bits(gb, 2);
1023 1024 1025 1026
    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;
1027 1028 1029 1030 1031
    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)) {
1032
                ics->group_len[ics->num_window_groups - 1]++;
1033 1034
            } else {
                ics->num_window_groups++;
1035
                ics->group_len[ics->num_window_groups - 1] = 1;
1036 1037
            }
        }
1038
        ics->num_windows       = 8;
1039 1040 1041
        ics->swb_offset        =    ff_swb_offset_128[ac->oc[1].m4ac.sampling_index];
        ics->num_swb           =   ff_aac_num_swb_128[ac->oc[1].m4ac.sampling_index];
        ics->tns_max_bands     = ff_tns_max_bands_128[ac->oc[1].m4ac.sampling_index];
1042
        ics->predictor_present = 0;
1043
    } else {
1044 1045
        ics->max_sfb               = get_bits(gb, 6);
        ics->num_windows           = 1;
1046 1047 1048
        ics->swb_offset            =    ff_swb_offset_1024[ac->oc[1].m4ac.sampling_index];
        ics->num_swb               =   ff_aac_num_swb_1024[ac->oc[1].m4ac.sampling_index];
        ics->tns_max_bands         = ff_tns_max_bands_1024[ac->oc[1].m4ac.sampling_index];
1049
        ics->predictor_present     = get_bits1(gb);
1050 1051
        ics->predictor_reset_group = 0;
        if (ics->predictor_present) {
1052
            if (ac->oc[1].m4ac.object_type == AOT_AAC_MAIN) {
1053
                if (decode_prediction(ac, ics, gb)) {
1054
                    goto fail;
1055
                }
1056
            } else if (ac->oc[1].m4ac.object_type == AOT_AAC_LC) {
1057
                av_log(ac->avctx, AV_LOG_ERROR, "Prediction is not allowed in AAC-LC.\n");
1058
                goto fail;
1059
            } else {
1060
                if ((ics->ltp.present = get_bits(gb, 1)))
1061
                    decode_ltp(&ics->ltp, gb, ics->max_sfb);
1062
            }
1063 1064 1065
        }
    }

1066
    if (ics->max_sfb > ics->num_swb) {
1067
        av_log(ac->avctx, AV_LOG_ERROR,
1068 1069
               "Number of scalefactor bands in group (%d) exceeds limit (%d).\n",
               ics->max_sfb, ics->num_swb);
1070
        goto fail;
1071 1072
    }

1073
    return 0;
1074 1075 1076
fail:
    ics->max_sfb = 0;
    return AVERROR_INVALIDDATA;
1077 1078 1079 1080 1081 1082 1083 1084 1085 1086
}

/**
 * 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
 */
1087 1088 1089 1090
static int decode_band_types(AACContext *ac, enum BandType band_type[120],
                             int band_type_run_end[120], GetBitContext *gb,
                             IndividualChannelStream *ics)
{
1091 1092 1093 1094 1095
    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) {
1096
            uint8_t sect_end = k;
1097 1098 1099
            int sect_len_incr;
            int sect_band_type = get_bits(gb, 4);
            if (sect_band_type == 12) {
1100
                av_log(ac->avctx, AV_LOG_ERROR, "invalid band type\n");
1101 1102
                return -1;
            }
1103 1104
            do {
                sect_len_incr = get_bits(gb, bits);
1105
                sect_end += sect_len_incr;
1106
                if (get_bits_left(gb) < 0) {
1107
                    av_log(ac->avctx, AV_LOG_ERROR, "decode_band_types: "overread_err);
1108 1109 1110 1111 1112 1113 1114 1115 1116
                    return -1;
                }
                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 -1;
                }
            } while (sect_len_incr == (1 << bits) - 1);
1117
            for (; k < sect_end; k++) {
1118
                band_type        [idx]   = sect_band_type;
1119
                band_type_run_end[idx++] = sect_end;
1120
            }
1121 1122 1123 1124
        }
    }
    return 0;
}
1125

1126 1127
/**
 * Decode scalefactors; reference: table 4.47.
1128 1129 1130 1131 1132 1133 1134 1135
 *
 * @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
 */
1136 1137 1138 1139 1140 1141
static int decode_scalefactors(AACContext *ac, float sf[120], GetBitContext *gb,
                               unsigned int global_gain,
                               IndividualChannelStream *ics,
                               enum BandType band_type[120],
                               int band_type_run_end[120])
{
1142
    int g, i, idx = 0;
1143 1144
    int offset[3] = { global_gain, global_gain - 90, 0 };
    int clipped_offset;
1145 1146 1147 1148 1149
    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) {
1150
                for (; i < run_end; i++, idx++)
1151
                    sf[idx] = 0.;
1152 1153
            } else if ((band_type[idx] == INTENSITY_BT) || (band_type[idx] == INTENSITY_BT2)) {
                for (; i < run_end; i++, idx++) {
1154
                    offset[2] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
1155 1156 1157 1158 1159 1160
                    clipped_offset = av_clip(offset[2], -155, 100);
                    if (offset[2] != clipped_offset) {
                        av_log_ask_for_sample(ac->avctx, "Intensity stereo "
                                "position clipped (%d -> %d).\nIf you heard an "
                                "audible artifact, there may be a bug in the "
                                "decoder. ", offset[2], clipped_offset);
1161
                    }
1162
                    sf[idx] = ff_aac_pow2sf_tab[-clipped_offset + POW_SF2_ZERO];
1163
                }
1164 1165 1166
            } else if (band_type[idx] == NOISE_BT) {
                for (; i < run_end; i++, idx++) {
                    if (noise_flag-- > 0)
1167 1168 1169
                        offset[1] += get_bits(gb, 9) - 256;
                    else
                        offset[1] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
1170
                    clipped_offset = av_clip(offset[1], -100, 155);
1171
                    if (offset[1] != clipped_offset) {
1172 1173 1174 1175
                        av_log_ask_for_sample(ac->avctx, "Noise gain clipped "
                                "(%d -> %d).\nIf you heard an audible "
                                "artifact, there may be a bug in the decoder. ",
                                offset[1], clipped_offset);
1176
                    }
1177
                    sf[idx] = -ff_aac_pow2sf_tab[clipped_offset + POW_SF2_ZERO];
1178
                }
1179 1180
            } else {
                for (; i < run_end; i++, idx++) {
1181
                    offset[0] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
1182
                    if (offset[0] > 255U) {
1183
                        av_log(ac->avctx, AV_LOG_ERROR,
1184
                               "Scalefactor (%d) out of range.\n", offset[0]);
1185 1186
                        return -1;
                    }
1187
                    sf[idx] = -ff_aac_pow2sf_tab[offset[0] - 100 + POW_SF2_ZERO];
1188 1189 1190 1191 1192 1193 1194 1195 1196 1197
                }
            }
        }
    }
    return 0;
}

/**
 * Decode pulse data; reference: table 4.7.
 */
1198 1199 1200
static int decode_pulses(Pulse *pulse, GetBitContext *gb,
                         const uint16_t *swb_offset, int num_swb)
{
1201
    int i, pulse_swb;
1202
    pulse->num_pulse = get_bits(gb, 2) + 1;
1203 1204 1205 1206
    pulse_swb        = get_bits(gb, 6);
    if (pulse_swb >= num_swb)
        return -1;
    pulse->pos[0]    = swb_offset[pulse_swb];
1207
    pulse->pos[0]   += get_bits(gb, 5);
1208 1209
    if (pulse->pos[0] > 1023)
        return -1;
1210 1211
    pulse->amp[0]    = get_bits(gb, 4);
    for (i = 1; i < pulse->num_pulse; i++) {
1212
        pulse->pos[i] = get_bits(gb, 5) + pulse->pos[i - 1];
1213 1214
        if (pulse->pos[i] > 1023)
            return -1;
1215
        pulse->amp[i] = get_bits(gb, 4);
1216
    }
1217
    return 0;
1218 1219
}

1220 1221 1222 1223 1224
/**
 * Decode Temporal Noise Shaping data; reference: table 4.48.
 *
 * @return  Returns error status. 0 - OK, !0 - error
 */
1225 1226 1227
static int decode_tns(AACContext *ac, TemporalNoiseShaping *tns,
                      GetBitContext *gb, const IndividualChannelStream *ics)
{
1228 1229
    int w, filt, i, coef_len, coef_res, coef_compress;
    const int is8 = ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE;
1230
    const int tns_max_order = is8 ? 7 : ac->oc[1].m4ac.object_type == AOT_AAC_MAIN ? 20 : 12;
1231
    for (w = 0; w < ics->num_windows; w++) {
1232
        if ((tns->n_filt[w] = get_bits(gb, 2 - is8))) {
1233 1234
            coef_res = get_bits1(gb);

1235 1236
            for (filt = 0; filt < tns->n_filt[w]; filt++) {
                int tmp2_idx;
1237
                tns->length[w][filt] = get_bits(gb, 6 - 2 * is8);
1238

1239
                if ((tns->order[w][filt] = get_bits(gb, 5 - 2 * is8)) > tns_max_order) {
1240
                    av_log(ac->avctx, AV_LOG_ERROR, "TNS filter order %d is greater than maximum %d.\n",
1241 1242 1243 1244
                           tns->order[w][filt], tns_max_order);
                    tns->order[w][filt] = 0;
                    return -1;
                }
1245
                if (tns->order[w][filt]) {
1246 1247 1248
                    tns->direction[w][filt] = get_bits1(gb);
                    coef_compress = get_bits1(gb);
                    coef_len = coef_res + 3 - coef_compress;
1249
                    tmp2_idx = 2 * coef_compress + coef_res;
1250

1251 1252
                    for (i = 0; i < tns->order[w][filt]; i++)
                        tns->coef[w][filt][i] = tns_tmp2_map[tmp2_idx][get_bits(gb, coef_len)];
1253
                }
1254
            }
1255
        }
1256 1257 1258 1259
    }
    return 0;
}

1260 1261 1262 1263 1264 1265 1266
/**
 * 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
 */
1267 1268 1269
static void decode_mid_side_stereo(ChannelElement *cpe, GetBitContext *gb,
                                   int ms_present)
{
1270 1271 1272 1273 1274
    int idx;
    if (ms_present == 1) {
        for (idx = 0; idx < cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb; idx++)
            cpe->ms_mask[idx] = get_bits1(gb);
    } else if (ms_present == 2) {
1275
        memset(cpe->ms_mask, 1,  sizeof(cpe->ms_mask[0]) * cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb);
1276 1277
    }
}
1278

1279
#ifndef VMUL2
1280 1281 1282 1283 1284 1285 1286 1287
static inline float *VMUL2(float *dst, const float *v, unsigned idx,
                           const float *scale)
{
    float s = *scale;
    *dst++ = v[idx    & 15] * s;
    *dst++ = v[idx>>4 & 15] * s;
    return dst;
}
1288
#endif
1289

1290
#ifndef VMUL4
1291 1292 1293 1294 1295 1296 1297 1298 1299 1300
static inline float *VMUL4(float *dst, const float *v, unsigned idx,
                           const float *scale)
{
    float s = *scale;
    *dst++ = v[idx    & 3] * s;
    *dst++ = v[idx>>2 & 3] * s;
    *dst++ = v[idx>>4 & 3] * s;
    *dst++ = v[idx>>6 & 3] * s;
    return dst;
}
1301
#endif
1302

1303
#ifndef VMUL2S
1304 1305 1306
static inline float *VMUL2S(float *dst, const float *v, unsigned idx,
                            unsigned sign, const float *scale)
{
1307
    union av_intfloat32 s0, s1;
1308 1309 1310 1311 1312 1313 1314 1315 1316 1317

    s0.f = s1.f = *scale;
    s0.i ^= sign >> 1 << 31;
    s1.i ^= sign      << 31;

    *dst++ = v[idx    & 15] * s0.f;
    *dst++ = v[idx>>4 & 15] * s1.f;

    return dst;
}
1318
#endif
1319

1320
#ifndef VMUL4S
1321 1322 1323 1324
static inline float *VMUL4S(float *dst, const float *v, unsigned idx,
                            unsigned sign, const float *scale)
{
    unsigned nz = idx >> 12;
1325 1326
    union av_intfloat32 s = { .f = *scale };
    union av_intfloat32 t;
1327

1328
    t.i = s.i ^ (sign & 1U<<31);
1329 1330 1331
    *dst++ = v[idx    & 3] * t.f;

    sign <<= nz & 1; nz >>= 1;
1332
    t.i = s.i ^ (sign & 1U<<31);
1333 1334 1335
    *dst++ = v[idx>>2 & 3] * t.f;

    sign <<= nz & 1; nz >>= 1;
1336
    t.i = s.i ^ (sign & 1U<<31);
1337 1338
    *dst++ = v[idx>>4 & 3] * t.f;

1339
    sign <<= nz & 1;
1340
    t.i = s.i ^ (sign & 1U<<31);
1341 1342 1343 1344
    *dst++ = v[idx>>6 & 3] * t.f;

    return dst;
}
1345
#endif
1346

1347 1348 1349 1350 1351 1352 1353 1354 1355 1356 1357 1358
/**
 * 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
 */
1359
static int decode_spectrum_and_dequant(AACContext *ac, float coef[1024],
1360
                                       GetBitContext *gb, const float sf[120],
1361 1362 1363 1364
                                       int pulse_present, const Pulse *pulse,
                                       const IndividualChannelStream *ics,
                                       enum BandType band_type[120])
{
1365
    int i, k, g, idx = 0;
1366 1367
    const int c = 1024 / ics->num_windows;
    const uint16_t *offsets = ics->swb_offset;
1368 1369 1370
    float *coef_base = coef;

    for (g = 0; g < ics->num_windows; g++)
1371
        memset(coef + g * 128 + offsets[ics->max_sfb], 0, sizeof(float) * (c - offsets[ics->max_sfb]));
1372 1373

    for (g = 0; g < ics->num_window_groups; g++) {
1374 1375
        unsigned g_len = ics->group_len[g];

1376
        for (i = 0; i < ics->max_sfb; i++, idx++) {
1377 1378 1379
            const unsigned cbt_m1 = band_type[idx] - 1;
            float *cfo = coef + offsets[i];
            int off_len = offsets[i + 1] - offsets[i];
1380
            int group;
1381 1382 1383 1384

            if (cbt_m1 >= INTENSITY_BT2 - 1) {
                for (group = 0; group < g_len; group++, cfo+=128) {
                    memset(cfo, 0, off_len * sizeof(float));
1385
                }
1386 1387
            } else if (cbt_m1 == NOISE_BT - 1) {
                for (group = 0; group < g_len; group++, cfo+=128) {
1388
                    float scale;
1389
                    float band_energy;
1390

1391
                    for (k = 0; k < off_len; k++) {
1392
                        ac->random_state  = lcg_random(ac->random_state);
1393
                        cfo[k] = ac->random_state;
1394
                    }
1395

1396
                    band_energy = ac->dsp.scalarproduct_float(cfo, cfo, off_len);
1397
                    scale = sf[idx] / sqrtf(band_energy);
1398
                    ac->fdsp.vector_fmul_scalar(cfo, cfo, scale, off_len);
1399
                }
1400
            } else {
1401 1402 1403
                const float *vq = ff_aac_codebook_vector_vals[cbt_m1];
                const uint16_t *cb_vector_idx = ff_aac_codebook_vector_idx[cbt_m1];
                VLC_TYPE (*vlc_tab)[2] = vlc_spectral[cbt_m1].table;
1404
                OPEN_READER(re, gb);
1405

1406 1407 1408 1409 1410
                switch (cbt_m1 >> 1) {
                case 0:
                    for (group = 0; group < g_len; group++, cfo+=128) {
                        float *cf = cfo;
                        int len = off_len;
1411

1412
                        do {
1413
                            int code;
1414 1415
                            unsigned cb_idx;

1416 1417 1418
                            UPDATE_CACHE(re, gb);
                            GET_VLC(code, re, gb, vlc_tab, 8, 2);
                            cb_idx = cb_vector_idx[code];
1419 1420
                            cf = VMUL4(cf, vq, cb_idx, sf + idx);
                        } while (len -= 4);
1421 1422 1423 1424 1425 1426 1427 1428
                    }
                    break;

                case 1:
                    for (group = 0; group < g_len; group++, cfo+=128) {
                        float *cf = cfo;
                        int len = off_len;

1429
                        do {
1430
                            int code;
1431 1432 1433 1434
                            unsigned nnz;
                            unsigned cb_idx;
                            uint32_t bits;

1435 1436 1437
                            UPDATE_CACHE(re, gb);
                            GET_VLC(code, re, gb, vlc_tab, 8, 2);
                            cb_idx = cb_vector_idx[code];
1438
                            nnz = cb_idx >> 8 & 15;
1439
                            bits = nnz ? GET_CACHE(re, gb) : 0;
1440
                            LAST_SKIP_BITS(re, gb, nnz);
1441 1442
                            cf = VMUL4S(cf, vq, cb_idx, bits, sf + idx);
                        } while (len -= 4);
1443 1444 1445 1446 1447 1448 1449 1450
                    }
                    break;

                case 2:
                    for (group = 0; group < g_len; group++, cfo+=128) {
                        float *cf = cfo;
                        int len = off_len;

1451
                        do {
1452
                            int code;
1453 1454
                            unsigned cb_idx;

1455 1456 1457
                            UPDATE_CACHE(re, gb);
                            GET_VLC(code, re, gb, vlc_tab, 8, 2);
                            cb_idx = cb_vector_idx[code];
1458 1459
                            cf = VMUL2(cf, vq, cb_idx, sf + idx);
                        } while (len -= 2);
1460 1461 1462 1463 1464 1465 1466 1467 1468
                    }
                    break;

                case 3:
                case 4:
                    for (group = 0; group < g_len; group++, cfo+=128) {
                        float *cf = cfo;
                        int len = off_len;

1469
                        do {
1470
                            int code;
1471 1472 1473 1474
                            unsigned nnz;
                            unsigned cb_idx;
                            unsigned sign;

1475 1476 1477
                            UPDATE_CACHE(re, gb);
                            GET_VLC(code, re, gb, vlc_tab, 8, 2);
                            cb_idx = cb_vector_idx[code];
1478
                            nnz = cb_idx >> 8 & 15;
1479
                            sign = nnz ? SHOW_UBITS(re, gb, nnz) << (cb_idx >> 12) : 0;
1480
                            LAST_SKIP_BITS(re, gb, nnz);
1481 1482
                            cf = VMUL2S(cf, vq, cb_idx, sign, sf + idx);
                        } while (len -= 2);
1483 1484 1485 1486 1487 1488 1489 1490 1491
                    }
                    break;

                default:
                    for (group = 0; group < g_len; group++, cfo+=128) {
                        float *cf = cfo;
                        uint32_t *icf = (uint32_t *) cf;
                        int len = off_len;

1492
                        do {
1493
                            int code;
1494 1495 1496 1497 1498
                            unsigned nzt, nnz;
                            unsigned cb_idx;
                            uint32_t bits;
                            int j;

1499 1500 1501 1502
                            UPDATE_CACHE(re, gb);
                            GET_VLC(code, re, gb, vlc_tab, 8, 2);

                            if (!code) {
1503 1504
                                *icf++ = 0;
                                *icf++ = 0;
1505 1506 1507
                                continue;
                            }

1508
                            cb_idx = cb_vector_idx[code];
1509 1510
                            nnz = cb_idx >> 12;
                            nzt = cb_idx >> 8;
1511 1512
                            bits = SHOW_UBITS(re, gb, nnz) << (32-nnz);
                            LAST_SKIP_BITS(re, gb, nnz);
1513 1514 1515

                            for (j = 0; j < 2; j++) {
                                if (nzt & 1<<j) {
1516 1517
                                    uint32_t b;
                                    int n;
1518 1519
                                    /* The total length of escape_sequence must be < 22 bits according
                                       to the specification (i.e. max is 111111110xxxxxxxxxxxx). */
1520 1521 1522 1523 1524
                                    UPDATE_CACHE(re, gb);
                                    b = GET_CACHE(re, gb);
                                    b = 31 - av_log2(~b);

                                    if (b > 8) {
1525
                                        av_log(ac->avctx, AV_LOG_ERROR, "error in spectral data, ESC overflow\n");
1526 1527
                                        return -1;
                                    }
1528 1529 1530 1531 1532

                                    SKIP_BITS(re, gb, b + 1);
                                    b += 4;
                                    n = (1 << b) + SHOW_UBITS(re, gb, b);
                                    LAST_SKIP_BITS(re, gb, b);
1533
                                    *icf++ = cbrt_tab[n] | (bits & 1U<<31);
1534 1535 1536
                                    bits <<= 1;
                                } else {
                                    unsigned v = ((const uint32_t*)vq)[cb_idx & 15];
1537
                                    *icf++ = (bits & 1U<<31) | v;
1538
                                    bits <<= !!v;
1539
                                }
1540
                                cb_idx >>= 4;
1541
                            }
1542
                        } while (len -= 2);
1543

1544
                        ac->fdsp.vector_fmul_scalar(cfo, cfo, sf[idx], off_len);
1545
                    }
1546
                }
1547 1548

                CLOSE_READER(re, gb);
1549 1550
            }
        }
1551
        coef += g_len << 7;
1552 1553 1554
    }

    if (pulse_present) {
1555
        idx = 0;
1556 1557 1558
        for (i = 0; i < pulse->num_pulse; i++) {
            float co = coef_base[ pulse->pos[i] ];
            while (offsets[idx + 1] <= pulse->pos[i])
1559 1560
                idx++;
            if (band_type[idx] != NOISE_BT && sf[idx]) {
1561 1562 1563 1564 1565 1566
                float ico = -pulse->amp[i];
                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];
1567
            }
1568 1569 1570 1571 1572
        }
    }
    return 0;
}

1573 1574
static av_always_inline float flt16_round(float pf)
{
1575
    union av_intfloat32 tmp;
1576 1577 1578
    tmp.f = pf;
    tmp.i = (tmp.i + 0x00008000U) & 0xFFFF0000U;
    return tmp.f;
1579 1580
}

1581 1582
static av_always_inline float flt16_even(float pf)
{
1583
    union av_intfloat32 tmp;
1584
    tmp.f = pf;
1585
    tmp.i = (tmp.i + 0x00007FFFU + (tmp.i & 0x00010000U >> 16)) & 0xFFFF0000U;
1586
    return tmp.f;
1587 1588
}

1589 1590
static av_always_inline float flt16_trunc(float pf)
{
1591
    union av_intfloat32 pun;
1592 1593 1594
    pun.f = pf;
    pun.i &= 0xFFFF0000U;
    return pun.f;
1595 1596
}

1597
static av_always_inline void predict(PredictorState *ps, float *coef,
1598
                                     int output_enable)
1599 1600 1601
{
    const float a     = 0.953125; // 61.0 / 64
    const float alpha = 0.90625;  // 29.0 / 32
1602 1603 1604
    float e0, e1;
    float pv;
    float k1, k2;
1605 1606 1607
    float   r0 = ps->r0,     r1 = ps->r1;
    float cor0 = ps->cor0, cor1 = ps->cor1;
    float var0 = ps->var0, var1 = ps->var1;
1608

1609 1610
    k1 = var0 > 1 ? cor0 * flt16_even(a / var0) : 0;
    k2 = var1 > 1 ? cor1 * flt16_even(a / var1) : 0;
1611

1612
    pv = flt16_round(k1 * r0 + k2 * r1);
1613
    if (output_enable)
1614
        *coef += pv;
1615

1616
    e0 = *coef;
1617
    e1 = e0 - k1 * r0;
1618

1619 1620 1621 1622
    ps->cor1 = flt16_trunc(alpha * cor1 + r1 * e1);
    ps->var1 = flt16_trunc(alpha * var1 + 0.5f * (r1 * r1 + e1 * e1));
    ps->cor0 = flt16_trunc(alpha * cor0 + r0 * e0);
    ps->var0 = flt16_trunc(alpha * var0 + 0.5f * (r0 * r0 + e0 * e0));
1623

1624
    ps->r1 = flt16_trunc(a * (r0 - k1 * e0));
1625 1626 1627 1628 1629 1630
    ps->r0 = flt16_trunc(a * e0);
}

/**
 * Apply AAC-Main style frequency domain prediction.
 */
1631 1632
static void apply_prediction(AACContext *ac, SingleChannelElement *sce)
{
1633 1634 1635
    int sfb, k;

    if (!sce->ics.predictor_initialized) {
1636
        reset_all_predictors(sce->predictor_state);
1637 1638 1639 1640
        sce->ics.predictor_initialized = 1;
    }

    if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
1641
        for (sfb = 0; sfb < ff_aac_pred_sfb_max[ac->oc[1].m4ac.sampling_index]; sfb++) {
1642
            for (k = sce->ics.swb_offset[sfb]; k < sce->ics.swb_offset[sfb + 1]; k++) {
1643
                predict(&sce->predictor_state[k], &sce->coeffs[k],
1644
                        sce->ics.predictor_present && sce->ics.prediction_used[sfb]);
1645 1646 1647
            }
        }
        if (sce->ics.predictor_reset_group)
1648
            reset_predictor_group(sce->predictor_state, sce->ics.predictor_reset_group);
1649
    } else
1650
        reset_all_predictors(sce->predictor_state);
1651 1652
}

1653
/**
1654 1655 1656 1657 1658 1659 1660
 * 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
 */
1661 1662 1663
static int decode_ics(AACContext *ac, SingleChannelElement *sce,
                      GetBitContext *gb, int common_window, int scale_flag)
{
1664
    Pulse pulse;
1665 1666 1667
    TemporalNoiseShaping    *tns = &sce->tns;
    IndividualChannelStream *ics = &sce->ics;
    float *out = sce->coeffs;
1668 1669
    int global_gain, pulse_present = 0;

1670 1671
    /* This assignment is to silence a GCC warning about the variable being used
     * uninitialized when in fact it always is.
1672 1673 1674 1675 1676 1677
     */
    pulse.num_pulse = 0;

    global_gain = get_bits(gb, 8);

    if (!common_window && !scale_flag) {
1678 1679
        if (decode_ics_info(ac, ics, gb) < 0)
            return AVERROR_INVALIDDATA;
1680 1681 1682 1683 1684 1685 1686 1687 1688 1689 1690
    }

    if (decode_band_types(ac, sce->band_type, sce->band_type_run_end, gb, ics) < 0)
        return -1;
    if (decode_scalefactors(ac, sce->sf, gb, global_gain, ics, sce->band_type, sce->band_type_run_end) < 0)
        return -1;

    pulse_present = 0;
    if (!scale_flag) {
        if ((pulse_present = get_bits1(gb))) {
            if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
1691
                av_log(ac->avctx, AV_LOG_ERROR, "Pulse tool not allowed in eight short sequence.\n");
1692 1693
                return -1;
            }
1694
            if (decode_pulses(&pulse, gb, ics->swb_offset, ics->num_swb)) {
1695
                av_log(ac->avctx, AV_LOG_ERROR, "Pulse data corrupt or invalid.\n");
1696 1697
                return -1;
            }
1698 1699 1700 1701
        }
        if ((tns->present = get_bits1(gb)) && decode_tns(ac, tns, gb, ics))
            return -1;
        if (get_bits1(gb)) {
1702
            av_log_missing_feature(ac->avctx, "SSR", 1);
1703
            return AVERROR_PATCHWELCOME;
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        }
    }

1707
    if (decode_spectrum_and_dequant(ac, out, gb, sce->sf, pulse_present, &pulse, ics, sce->band_type) < 0)
1708
        return -1;
1709

1710
    if (ac->oc[1].m4ac.object_type == AOT_AAC_MAIN && !common_window)
1711 1712
        apply_prediction(ac, sce);

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    return 0;
}

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/**
 * Mid/Side stereo decoding; reference: 4.6.8.1.3.
 */
1719
static void apply_mid_side_stereo(AACContext *ac, ChannelElement *cpe)
1720 1721
{
    const IndividualChannelStream *ics = &cpe->ch[0].ics;
1722 1723
    float *ch0 = cpe->ch[0].coeffs;
    float *ch1 = cpe->ch[1].coeffs;
1724
    int g, i, group, idx = 0;
1725
    const uint16_t *offsets = ics->swb_offset;
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    for (g = 0; g < ics->num_window_groups; g++) {
        for (i = 0; i < ics->max_sfb; i++, idx++) {
            if (cpe->ms_mask[idx] &&
1729
                    cpe->ch[0].band_type[idx] < NOISE_BT && cpe->ch[1].band_type[idx] < NOISE_BT) {
1730
                for (group = 0; group < ics->group_len[g]; group++) {
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                    ac->dsp.butterflies_float(ch0 + group * 128 + offsets[i],
                                              ch1 + group * 128 + offsets[i],
                                              offsets[i+1] - offsets[i]);
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                }
            }
        }
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        ch0 += ics->group_len[g] * 128;
        ch1 += ics->group_len[g] * 128;
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    }
}

/**
 * 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
 */
1749
static void apply_intensity_stereo(AACContext *ac, ChannelElement *cpe, int ms_present)
1750 1751 1752
{
    const IndividualChannelStream *ics = &cpe->ch[1].ics;
    SingleChannelElement         *sce1 = &cpe->ch[1];
1753
    float *coef0 = cpe->ch[0].coeffs, *coef1 = cpe->ch[1].coeffs;
1754
    const uint16_t *offsets = ics->swb_offset;
1755
    int g, group, i, idx = 0;
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    int c;
    float scale;
    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++)
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                        ac->fdsp.vector_fmul_scalar(coef1 + group * 128 + offsets[i],
                                                    coef0 + group * 128 + offsets[i],
                                                    scale,
                                                    offsets[i + 1] - offsets[i]);
1772 1773 1774 1775 1776 1777 1778
                }
            } else {
                int bt_run_end = sce1->band_type_run_end[idx];
                idx += bt_run_end - i;
                i    = bt_run_end;
            }
        }
1779 1780
        coef0 += ics->group_len[g] * 128;
        coef1 += ics->group_len[g] * 128;
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    }
}

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/**
 * Decode a channel_pair_element; reference: table 4.4.
 *
 * @return  Returns error status. 0 - OK, !0 - error
 */
1789 1790
static int decode_cpe(AACContext *ac, GetBitContext *gb, ChannelElement *cpe)
{
1791 1792 1793 1794
    int i, ret, common_window, ms_present = 0;

    common_window = get_bits1(gb);
    if (common_window) {
1795 1796
        if (decode_ics_info(ac, &cpe->ch[0].ics, gb))
            return AVERROR_INVALIDDATA;
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        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;
1800
        if (cpe->ch[1].ics.predictor_present && (ac->oc[1].m4ac.object_type != AOT_AAC_MAIN))
1801
            if ((cpe->ch[1].ics.ltp.present = get_bits(gb, 1)))
1802
                decode_ltp(&cpe->ch[1].ics.ltp, gb, cpe->ch[1].ics.max_sfb);
1803
        ms_present = get_bits(gb, 2);
1804
        if (ms_present == 3) {
1805
            av_log(ac->avctx, AV_LOG_ERROR, "ms_present = 3 is reserved.\n");
1806
            return -1;
1807
        } else if (ms_present)
1808 1809 1810 1811 1812 1813 1814
            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;

1815 1816
    if (common_window) {
        if (ms_present)
1817
            apply_mid_side_stereo(ac, cpe);
1818
        if (ac->oc[1].m4ac.object_type == AOT_AAC_MAIN) {
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            apply_prediction(ac, &cpe->ch[0]);
            apply_prediction(ac, &cpe->ch[1]);
        }
    }
1823

1824
    apply_intensity_stereo(ac, cpe, ms_present);
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    return 0;
}

1828 1829 1830 1831 1832 1833 1834
static const float cce_scale[] = {
    1.09050773266525765921, //2^(1/8)
    1.18920711500272106672, //2^(1/4)
    M_SQRT2,
    2,
};

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/**
 * Decode coupling_channel_element; reference: table 4.8.
 *
 * @return  Returns error status. 0 - OK, !0 - error
 */
1840 1841
static int decode_cce(AACContext *ac, GetBitContext *gb, ChannelElement *che)
{
1842
    int num_gain = 0;
1843
    int c, g, sfb, ret;
1844 1845
    int sign;
    float scale;
1846 1847
    SingleChannelElement *sce = &che->ch[0];
    ChannelCoupling     *coup = &che->coup;
1848

1849
    coup->coupling_point = 2 * get_bits1(gb);
1850 1851 1852 1853 1854 1855 1856 1857 1858 1859
    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
1860
            coup->ch_select[c] = 2;
1861
    }
1862
    coup->coupling_point += get_bits1(gb) || (coup->coupling_point >> 1);
1863

1864
    sign  = get_bits(gb, 1);
1865
    scale = cce_scale[get_bits(gb, 2)];
1866 1867 1868 1869 1870

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

    for (c = 0; c < num_gain; c++) {
1871 1872
        int idx  = 0;
        int cge  = 1;
1873 1874 1875 1876 1877
        int gain = 0;
        float gain_cache = 1.;
        if (c) {
            cge = coup->coupling_point == AFTER_IMDCT ? 1 : get_bits1(gb);
            gain = cge ? get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60: 0;
1878
            gain_cache = powf(scale, -gain);
1879
        }
1880 1881 1882
        if (coup->coupling_point == AFTER_IMDCT) {
            coup->gain[c][0] = gain_cache;
        } else {
1883 1884 1885 1886 1887
            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;
1888
                            if (t) {
1889 1890 1891 1892 1893 1894
                                int s = 1;
                                t = gain += t;
                                if (sign) {
                                    s  -= 2 * (t & 0x1);
                                    t >>= 1;
                                }
1895
                                gain_cache = powf(scale, -t) * s;
1896 1897
                            }
                        }
1898
                        coup->gain[c][idx] = gain_cache;
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                    }
                }
1901 1902
            }
        }
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    }
    return 0;
}

/**
 * Parse whether channels are to be excluded from Dynamic Range Compression; reference: table 4.53.
 *
 * @return  Returns number of bytes consumed.
 */
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static int decode_drc_channel_exclusions(DynamicRangeControl *che_drc,
                                         GetBitContext *gb)
{
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    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;
}

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/**
 * Decode dynamic range information; reference: table 4.52.
 *
 * @return  Returns number of bytes consumed.
 */
1931
static int decode_dynamic_range(DynamicRangeControl *che_drc,
1932
                                GetBitContext *gb)
1933 1934
{
    int n             = 1;
1935 1936 1937 1938
    int drc_num_bands = 1;
    int i;

    /* pce_tag_present? */
1939
    if (get_bits1(gb)) {
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        che_drc->pce_instance_tag  = get_bits(gb, 4);
        skip_bits(gb, 4); // tag_reserved_bits
        n++;
    }

    /* excluded_chns_present? */
1946
    if (get_bits1(gb)) {
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 1974 1975 1976 1977
        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;
}

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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;
}

2004 2005 2006 2007 2008 2009 2010
/**
 * Decode extension data (incomplete); reference: table 4.51.
 *
 * @param   cnt length of TYPE_FIL syntactic element in bytes
 *
 * @return Returns number of bytes consumed
 */
2011 2012
static int decode_extension_payload(AACContext *ac, GetBitContext *gb, int cnt,
                                    ChannelElement *che, enum RawDataBlockType elem_type)
2013
{
2014 2015 2016
    int crc_flag = 0;
    int res = cnt;
    switch (get_bits(gb, 4)) { // extension type
2017 2018 2019
    case EXT_SBR_DATA_CRC:
        crc_flag++;
    case EXT_SBR_DATA:
2020
        if (!che) {
2021
            av_log(ac->avctx, AV_LOG_ERROR, "SBR was found before the first channel element.\n");
2022
            return res;
2023
        } else if (!ac->oc[1].m4ac.sbr) {
2024
            av_log(ac->avctx, AV_LOG_ERROR, "SBR signaled to be not-present but was found in the bitstream.\n");
2025 2026
            skip_bits_long(gb, 8 * cnt - 4);
            return res;
2027
        } else if (ac->oc[1].m4ac.sbr == -1 && ac->oc[1].status == OC_LOCKED) {
2028
            av_log(ac->avctx, AV_LOG_ERROR, "Implicit SBR was found with a first occurrence after the first frame.\n");
2029 2030
            skip_bits_long(gb, 8 * cnt - 4);
            return res;
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        } 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;
            output_configure(ac, ac->oc[1].layout_map, ac->oc[1].layout_map_tags,
2035
                             ac->oc[1].status, 1);
2036
        } else {
2037
            ac->oc[1].m4ac.sbr = 1;
2038 2039
        }
        res = ff_decode_sbr_extension(ac, &che->sbr, gb, crc_flag, cnt, elem_type);
2040 2041
        break;
    case EXT_DYNAMIC_RANGE:
2042
        res = decode_dynamic_range(&ac->che_drc, gb);
2043 2044
        break;
    case EXT_FILL:
2045 2046
        decode_fill(ac, gb, 8 * cnt - 4);
        break;
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    case EXT_FILL_DATA:
    case EXT_DATA_ELEMENT:
    default:
        skip_bits_long(gb, 8 * cnt - 4);
        break;
2052 2053 2054 2055
    };
    return res;
}

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/**
 * 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
 */
2062 2063 2064 2065
static void apply_tns(float coef[1024], TemporalNoiseShaping *tns,
                      IndividualChannelStream *ics, int decode)
{
    const int mmm = FFMIN(ics->tns_max_bands, ics->max_sfb);
Robert Swain's avatar
Robert Swain committed
2066
    int w, filt, m, i;
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    int bottom, top, order, start, end, size, inc;
    float lpc[TNS_MAX_ORDER];
2069
    float tmp[TNS_MAX_ORDER+1];
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    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;

2080 2081
            // tns_decode_coef
            compute_lpc_coefs(tns->coef[w][filt], order, lpc, 0, 0, 0);
2082

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            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]) {
2088 2089
                inc = -1;
                start = end - 1;
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            } else {
                inc = 1;
            }
            start += w * 128;

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            if (decode) {
                // ar filter
                for (m = 0; m < size; m++, start += inc)
                    for (i = 1; i <= FFMIN(m, order); i++)
                        coef[start] -= coef[start - i * inc] * lpc[i - 1];
            } else {
                // ma filter
                for (m = 0; m < size; m++, start += inc) {
                    tmp[0] = coef[start];
                    for (i = 1; i <= FFMIN(m, order); i++)
                        coef[start] += tmp[i] * lpc[i - 1];
                    for (i = order; i > 0; i--)
                        tmp[i] = tmp[i - 1];
                }
            }
2110 2111 2112 2113
        }
    }
}

2114 2115 2116 2117 2118 2119 2120 2121 2122 2123 2124 2125 2126
/**
 *  Apply windowing and MDCT to obtain the spectral
 *  coefficient from the predicted sample by LTP.
 */
static void windowing_and_mdct_ltp(AACContext *ac, float *out,
                                   float *in, IndividualChannelStream *ics)
{
    const float *lwindow      = ics->use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
    const float *swindow      = ics->use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
    const float *lwindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;
    const float *swindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;

    if (ics->window_sequence[0] != LONG_STOP_SEQUENCE) {
2127
        ac->fdsp.vector_fmul(in, in, lwindow_prev, 1024);
2128 2129
    } else {
        memset(in, 0, 448 * sizeof(float));
2130
        ac->fdsp.vector_fmul(in + 448, in + 448, swindow_prev, 128);
2131 2132 2133 2134 2135 2136 2137
    }
    if (ics->window_sequence[0] != LONG_START_SEQUENCE) {
        ac->dsp.vector_fmul_reverse(in + 1024, in + 1024, lwindow, 1024);
    } else {
        ac->dsp.vector_fmul_reverse(in + 1024 + 448, in + 1024 + 448, swindow, 128);
        memset(in + 1024 + 576, 0, 448 * sizeof(float));
    }
2138
    ac->mdct_ltp.mdct_calc(&ac->mdct_ltp, out, in);
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}

/**
 * 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) {
2151 2152
        float *predTime = sce->ret;
        float *predFreq = ac->buf_mdct;
2153 2154 2155 2156 2157 2158 2159 2160 2161 2162 2163 2164 2165 2166 2167 2168 2169 2170 2171 2172 2173 2174 2175 2176 2177 2178 2179 2180 2181 2182 2183 2184 2185 2186 2187
        int16_t num_samples = 2048;

        if (ltp->lag < 1024)
            num_samples = ltp->lag + 1024;
        for (i = 0; i < num_samples; i++)
            predTime[i] = sce->ltp_state[i + 2048 - ltp->lag] * ltp->coef;
        memset(&predTime[i], 0, (2048 - i) * sizeof(float));

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

        if (sce->tns.present)
            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;
    float *saved     = sce->saved;
    float *saved_ltp = sce->coeffs;
    const float *lwindow = ics->use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
    const float *swindow = ics->use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
    int i;

    if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
        memcpy(saved_ltp,       saved, 512 * sizeof(float));
        memset(saved_ltp + 576, 0,     448 * sizeof(float));
2188 2189 2190
        ac->dsp.vector_fmul_reverse(saved_ltp + 448, ac->buf_mdct + 960,     &swindow[64],      64);
        for (i = 0; i < 64; i++)
            saved_ltp[i + 512] = ac->buf_mdct[1023 - i] * swindow[63 - i];
2191 2192 2193
    } else if (ics->window_sequence[0] == LONG_START_SEQUENCE) {
        memcpy(saved_ltp,       ac->buf_mdct + 512, 448 * sizeof(float));
        memset(saved_ltp + 576, 0,                  448 * sizeof(float));
2194 2195 2196
        ac->dsp.vector_fmul_reverse(saved_ltp + 448, ac->buf_mdct + 960,     &swindow[64],      64);
        for (i = 0; i < 64; i++)
            saved_ltp[i + 512] = ac->buf_mdct[1023 - i] * swindow[63 - i];
2197
    } else { // LONG_STOP or ONLY_LONG
2198 2199 2200
        ac->dsp.vector_fmul_reverse(saved_ltp,       ac->buf_mdct + 512,     &lwindow[512],     512);
        for (i = 0; i < 512; i++)
            saved_ltp[i + 512] = ac->buf_mdct[1023 - i] * lwindow[511 - i];
2201 2202
    }

2203 2204 2205
    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));
2206 2207
}

2208 2209 2210
/**
 * Conduct IMDCT and windowing.
 */
2211
static void imdct_and_windowing(AACContext *ac, SingleChannelElement *sce)
2212 2213 2214 2215 2216 2217 2218 2219 2220 2221
{
    IndividualChannelStream *ics = &sce->ics;
    float *in    = sce->coeffs;
    float *out   = sce->ret;
    float *saved = sce->saved;
    const float *swindow      = ics->use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
    const float *lwindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;
    const float *swindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
    float *buf  = ac->buf_mdct;
    float *temp = ac->temp;
2222 2223
    int i;

2224
    // imdct
2225
    if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
2226
        for (i = 0; i < 1024; i += 128)
2227
            ac->mdct_small.imdct_half(&ac->mdct_small, buf + i, in + i);
2228
    } else
2229
        ac->mdct.imdct_half(&ac->mdct, buf, in);
2230 2231 2232 2233 2234 2235 2236 2237

    /* 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) &&
2238
            (ics->window_sequence[0] == ONLY_LONG_SEQUENCE || ics->window_sequence[0] == LONG_START_SEQUENCE)) {
2239
        ac->dsp.vector_fmul_window(    out,               saved,            buf,         lwindow_prev, 512);
2240
    } else {
2241
        memcpy(                        out,               saved,            448 * sizeof(float));
2242

2243
        if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
2244 2245 2246 2247 2248
            ac->dsp.vector_fmul_window(out + 448 + 0*128, saved + 448,      buf + 0*128, swindow_prev, 64);
            ac->dsp.vector_fmul_window(out + 448 + 1*128, buf + 0*128 + 64, buf + 1*128, swindow,      64);
            ac->dsp.vector_fmul_window(out + 448 + 2*128, buf + 1*128 + 64, buf + 2*128, swindow,      64);
            ac->dsp.vector_fmul_window(out + 448 + 3*128, buf + 2*128 + 64, buf + 3*128, swindow,      64);
            ac->dsp.vector_fmul_window(temp,              buf + 3*128 + 64, buf + 4*128, swindow,      64);
2249
            memcpy(                    out + 448 + 4*128, temp, 64 * sizeof(float));
2250
        } else {
2251
            ac->dsp.vector_fmul_window(out + 448,         saved + 448,      buf,         swindow_prev, 64);
2252
            memcpy(                    out + 576,         buf + 64,         448 * sizeof(float));
2253 2254
        }
    }
2255

2256 2257
    // buffer update
    if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
2258
        memcpy(                    saved,       temp + 64,         64 * sizeof(float));
2259 2260 2261
        ac->dsp.vector_fmul_window(saved + 64,  buf + 4*128 + 64, buf + 5*128, swindow, 64);
        ac->dsp.vector_fmul_window(saved + 192, buf + 5*128 + 64, buf + 6*128, swindow, 64);
        ac->dsp.vector_fmul_window(saved + 320, buf + 6*128 + 64, buf + 7*128, swindow, 64);
2262
        memcpy(                    saved + 448, buf + 7*128 + 64,  64 * sizeof(float));
2263
    } else if (ics->window_sequence[0] == LONG_START_SEQUENCE) {
2264 2265
        memcpy(                    saved,       buf + 512,        448 * sizeof(float));
        memcpy(                    saved + 448, buf + 7*128 + 64,  64 * sizeof(float));
2266
    } else { // LONG_STOP or ONLY_LONG
2267
        memcpy(                    saved,       buf + 512,        512 * sizeof(float));
2268 2269 2270
    }
}

2271 2272 2273 2274 2275
/**
 * Apply dependent channel coupling (applied before IMDCT).
 *
 * @param   index   index into coupling gain array
 */
2276 2277 2278 2279 2280 2281 2282 2283
static void apply_dependent_coupling(AACContext *ac,
                                     SingleChannelElement *target,
                                     ChannelElement *cce, int index)
{
    IndividualChannelStream *ics = &cce->ch[0].ics;
    const uint16_t *offsets = ics->swb_offset;
    float *dest = target->coeffs;
    const float *src = cce->ch[0].coeffs;
2284
    int g, i, group, k, idx = 0;
2285
    if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP) {
2286
        av_log(ac->avctx, AV_LOG_ERROR,
2287 2288 2289 2290 2291
               "Dependent coupling is not supported together with LTP\n");
        return;
    }
    for (g = 0; g < ics->num_window_groups; g++) {
        for (i = 0; i < ics->max_sfb; i++, idx++) {
2292
            if (cce->ch[0].band_type[idx] != ZERO_BT) {
2293
                const float gain = cce->coup.gain[index][idx];
2294
                for (group = 0; group < ics->group_len[g]; group++) {
2295
                    for (k = offsets[i]; k < offsets[i + 1]; k++) {
2296
                        // XXX dsputil-ize
2297
                        dest[group * 128 + k] += gain * src[group * 128 + k];
2298 2299 2300 2301
                    }
                }
            }
        }
2302 2303
        dest += ics->group_len[g] * 128;
        src  += ics->group_len[g] * 128;
2304 2305 2306 2307 2308 2309 2310 2311
    }
}

/**
 * Apply independent channel coupling (applied after IMDCT).
 *
 * @param   index   index into coupling gain array
 */
2312 2313 2314 2315
static void apply_independent_coupling(AACContext *ac,
                                       SingleChannelElement *target,
                                       ChannelElement *cce, int index)
{
2316
    int i;
2317
    const float gain = cce->coup.gain[index][0];
2318 2319
    const float *src = cce->ch[0].ret;
    float *dest = target->ret;
2320
    const int len = 1024 << (ac->oc[1].m4ac.sbr == 1);
2321

2322
    for (i = 0; i < len; i++)
2323
        dest[i] += gain * src[i];
2324 2325
}

2326 2327 2328 2329 2330
/**
 * channel coupling transformation interface
 *
 * @param   apply_coupling_method   pointer to (in)dependent coupling function
 */
2331 2332 2333 2334
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))
2335
{
2336 2337 2338 2339 2340 2341 2342
    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) {
2343
            ChannelCoupling *coup = &cce->coup;
2344 2345 2346 2347 2348 2349 2350 2351 2352 2353 2354 2355

            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);
2356 2357 2358 2359 2360 2361 2362 2363
            }
        }
    }
}

/**
 * Convert spectral data to float samples, applying all supported tools as appropriate.
 */
2364 2365
static void spectral_to_sample(AACContext *ac)
{
2366 2367
    int i, type;
    for (type = 3; type >= 0; type--) {
2368
        for (i = 0; i < MAX_ELEM_ID; i++) {
2369
            ChannelElement *che = ac->che[type][i];
2370 2371
            if (che) {
                if (type <= TYPE_CPE)
2372
                    apply_channel_coupling(ac, che, type, i, BEFORE_TNS, apply_dependent_coupling);
2373
                if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP) {
2374 2375 2376 2377 2378 2379 2380
                    if (che->ch[0].ics.predictor_present) {
                        if (che->ch[0].ics.ltp.present)
                            apply_ltp(ac, &che->ch[0]);
                        if (che->ch[1].ics.ltp.present && type == TYPE_CPE)
                            apply_ltp(ac, &che->ch[1]);
                    }
                }
2381
                if (che->ch[0].tns.present)
2382
                    apply_tns(che->ch[0].coeffs, &che->ch[0].tns, &che->ch[0].ics, 1);
2383
                if (che->ch[1].tns.present)
2384
                    apply_tns(che->ch[1].coeffs, &che->ch[1].tns, &che->ch[1].ics, 1);
2385
                if (type <= TYPE_CPE)
2386
                    apply_channel_coupling(ac, che, type, i, BETWEEN_TNS_AND_IMDCT, apply_dependent_coupling);
2387
                if (type != TYPE_CCE || che->coup.coupling_point == AFTER_IMDCT) {
2388
                    imdct_and_windowing(ac, &che->ch[0]);
2389
                    if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP)
2390
                        update_ltp(ac, &che->ch[0]);
Alex Converse's avatar
Alex Converse committed
2391
                    if (type == TYPE_CPE) {
2392
                        imdct_and_windowing(ac, &che->ch[1]);
2393
                        if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP)
2394
                            update_ltp(ac, &che->ch[1]);
Alex Converse's avatar
Alex Converse committed
2395
                    }
2396
                    if (ac->oc[1].m4ac.sbr > 0) {
2397 2398
                        ff_sbr_apply(ac, &che->sbr, type, che->ch[0].ret, che->ch[1].ret);
                    }
2399
                }
2400
                if (type <= TYPE_CCE)
2401
                    apply_channel_coupling(ac, che, type, i, AFTER_IMDCT, apply_independent_coupling);
2402 2403 2404 2405 2406
            }
        }
    }
}

2407 2408
static int parse_adts_frame_header(AACContext *ac, GetBitContext *gb)
{
2409 2410
    int size;
    AACADTSHeaderInfo hdr_info;
2411 2412
    uint8_t layout_map[MAX_ELEM_ID*4][3];
    int layout_map_tags;
2413

2414
    size = avpriv_aac_parse_header(gb, &hdr_info);
2415
    if (size > 0) {
2416 2417 2418
        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.
2419
            av_log_missing_feature(ac->avctx, "More than one AAC RDB per ADTS frame", 0);
2420
            ac->warned_num_aac_frames = 1;
2421 2422
        }
        push_output_configuration(ac);
2423
        if (hdr_info.chan_config) {
2424
            ac->oc[1].m4ac.chan_config = hdr_info.chan_config;
2425 2426
            if (set_default_channel_config(ac->avctx, layout_map,
                    &layout_map_tags, hdr_info.chan_config))
2427
                return -7;
2428
            if (output_configure(ac, layout_map, layout_map_tags,
2429
                                 FFMAX(ac->oc[1].status, OC_TRIAL_FRAME), 0))
2430
                return -7;
2431
        } else {
2432
            ac->oc[1].m4ac.chan_config = 0;
2433 2434 2435 2436 2437
            /**
             * dual mono frames in Japanese DTV can have chan_config 0
             * WITHOUT specifying PCE.
             *  thus, set dual mono as default.
             */
2438
#if 0
2439 2440 2441 2442 2443 2444 2445
            if (ac->enable_jp_dmono && 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,
2446
                                     OC_TRIAL_FRAME))
2447 2448
                    return -7;
            }
2449
#endif
2450
        }
2451 2452 2453 2454 2455 2456 2457 2458
        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;
        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;
2459
        }
2460 2461
        if (!hdr_info.crc_absent)
            skip_bits(gb, 16);
2462
    }
2463 2464 2465
    return size;
}

2466
static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
2467
                                int *got_frame_ptr, GetBitContext *gb, AVPacket *avpkt)
2468
{
2469
    AACContext *ac = avctx->priv_data;
2470 2471
    ChannelElement *che = NULL, *che_prev = NULL;
    enum RawDataBlockType elem_type, elem_type_prev = TYPE_END;
2472
    int err, elem_id;
2473
    int samples = 0, multiplier, audio_found = 0, pce_found = 0;
2474 2475
    int is_dmono, sce_count = 0;
    float *tmp = NULL;
2476

2477 2478
    if (show_bits(gb, 12) == 0xfff) {
        if (parse_adts_frame_header(ac, gb) < 0) {
2479
            av_log(avctx, AV_LOG_ERROR, "Error decoding AAC frame header.\n");
2480 2481
            err = -1;
            goto fail;
2482
        }
2483 2484 2485 2486
        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 = -1;
            goto fail;
2487
        }
2488 2489
    }

2490 2491 2492 2493 2494
    if (frame_configure_elements(avctx) < 0) {
        err = -1;
        goto fail;
    }

2495
    ac->tags_mapped = 0;
2496
    // parse
2497 2498
    while ((elem_type = get_bits(gb, 3)) != TYPE_END) {
        elem_id = get_bits(gb, 4);
2499

2500
        if (elem_type < TYPE_DSE) {
2501 2502 2503
            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);
2504 2505
                err = -1;
                goto fail;
2506
            }
2507
            samples = 1024;
2508
        }
2509

2510 2511 2512
        switch (elem_type) {

        case TYPE_SCE:
2513
            err = decode_ics(ac, &che->ch[0], gb, 0, 0);
2514
            audio_found = 1;
2515
            sce_count++;
2516 2517 2518
            break;

        case TYPE_CPE:
2519
            err = decode_cpe(ac, gb, che);
2520
            audio_found = 1;
2521 2522 2523
            break;

        case TYPE_CCE:
2524
            err = decode_cce(ac, gb, che);
2525 2526 2527
            break;

        case TYPE_LFE:
2528
            err = decode_ics(ac, &che->ch[0], gb, 0, 0);
2529
            audio_found = 1;
2530 2531 2532
            break;

        case TYPE_DSE:
2533
            err = skip_data_stream_element(ac, gb);
2534 2535
            break;

2536
        case TYPE_PCE: {
2537 2538
            uint8_t layout_map[MAX_ELEM_ID*4][3];
            int tags;
2539 2540
            push_output_configuration(ac);
            tags = decode_pce(avctx, &ac->oc[1].m4ac, layout_map, gb);
2541 2542
            if (tags < 0) {
                err = tags;
2543
                break;
2544
            }
2545
            if (pce_found) {
2546 2547 2548 2549
                av_log(avctx, AV_LOG_ERROR,
                       "Not evaluating a further program_config_element as this construct is dubious at best.\n");
                pop_output_configuration(ac);
            } else {
2550
                err = output_configure(ac, layout_map, tags, OC_TRIAL_PCE, 1);
2551 2552 2553
                if (!err)
                    ac->oc[1].m4ac.chan_config = 0;
                pce_found = 1;
2554
            }
2555 2556 2557 2558 2559
            break;
        }

        case TYPE_FIL:
            if (elem_id == 15)
2560 2561
                elem_id += get_bits(gb, 8) - 1;
            if (get_bits_left(gb) < 8 * elem_id) {
2562
                    av_log(avctx, AV_LOG_ERROR, "TYPE_FIL: "overread_err);
2563 2564
                    err = -1;
                    goto fail;
2565
            }
2566
            while (elem_id > 0)
2567
                elem_id -= decode_extension_payload(ac, gb, elem_id, che_prev, elem_type_prev);
2568 2569 2570 2571 2572 2573 2574 2575
            err = 0; /* FIXME */
            break;

        default:
            err = -1; /* should not happen, but keeps compiler happy */
            break;
        }

2576 2577 2578
        che_prev       = che;
        elem_type_prev = elem_type;

2579
        if (err)
2580
            goto fail;
2581

2582
        if (get_bits_left(gb) < 3) {
2583
            av_log(avctx, AV_LOG_ERROR, overread_err);
2584 2585
            err = -1;
            goto fail;
2586
        }
2587 2588 2589 2590
    }

    spectral_to_sample(ac);

2591
    multiplier = (ac->oc[1].m4ac.sbr == 1) ? ac->oc[1].m4ac.ext_sample_rate > ac->oc[1].m4ac.sample_rate : 0;
2592
    samples <<= multiplier;
2593
#if 0
2594 2595 2596 2597 2598 2599 2600 2601 2602 2603 2604 2605 2606
    /* for dual-mono audio (SCE + SCE) */
    is_dmono = ac->enable_jp_dmono && sce_count == 2 &&
               ac->oc[1].channel_layout == (AV_CH_FRONT_LEFT | AV_CH_FRONT_RIGHT);

    if (is_dmono) {
        if (ac->dmono_mode == 0) {
            tmp = ac->output_data[1];
            ac->output_data[1] = ac->output_data[0];
        } else if (ac->dmono_mode == 1) {
            tmp = ac->output_data[0];
            ac->output_data[0] = ac->output_data[1];
        }
    }
2607
#endif
2608
    if (samples) {
2609 2610
        ac->frame.nb_samples = samples;
        *(AVFrame *)data = ac->frame;
2611
    }
2612
    *got_frame_ptr = !!samples;
2613
#if 0
2614 2615 2616 2617 2618 2619
    if (is_dmono) {
        if (ac->dmono_mode == 0)
            ac->output_data[1] = tmp;
        else if (ac->dmono_mode == 1)
            ac->output_data[0] = tmp;
    }
2620
#endif
2621 2622 2623 2624 2625
    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;
    }
2626

2627 2628 2629 2630 2631 2632
    if (multiplier) {
        int side_size;
        uint32_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));
    }
2633
    return 0;
2634 2635 2636
fail:
    pop_output_configuration(ac);
    return err;
2637 2638 2639
}

static int aac_decode_frame(AVCodecContext *avctx, void *data,
2640
                            int *got_frame_ptr, AVPacket *avpkt)
2641
{
2642
    AACContext *ac = avctx->priv_data;
2643 2644 2645 2646 2647 2648
    const uint8_t *buf = avpkt->data;
    int buf_size = avpkt->size;
    GetBitContext gb;
    int buf_consumed;
    int buf_offset;
    int err;
2649 2650 2651 2652
    int new_extradata_size;
    const uint8_t *new_extradata = av_packet_get_side_data(avpkt,
                                       AV_PKT_DATA_NEW_EXTRADATA,
                                       &new_extradata_size);
2653 2654 2655 2656
    int jp_dualmono_size;
    const uint8_t *jp_dualmono   = av_packet_get_side_data(avpkt,
                                       AV_PKT_DATA_JP_DUALMONO,
                                       &jp_dualmono_size);
2657

2658
    if (new_extradata && 0) {
2659 2660 2661 2662 2663 2664 2665
        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);
2666 2667
        push_output_configuration(ac);
        if (decode_audio_specific_config(ac, ac->avctx, &ac->oc[1].m4ac,
2668
                                         avctx->extradata,
2669 2670
                                         avctx->extradata_size*8, 1) < 0) {
            pop_output_configuration(ac);
2671
            return AVERROR_INVALIDDATA;
2672
        }
2673
    }
2674

2675 2676 2677 2678 2679
    ac->enable_jp_dmono = !!jp_dualmono;
    ac->dmono_mode = 0;
    if (jp_dualmono && jp_dualmono_size > 0)
        ac->dmono_mode = *jp_dualmono;

2680 2681
    init_get_bits(&gb, buf, buf_size * 8);

2682
    if ((err = aac_decode_frame_int(avctx, data, got_frame_ptr, &gb, avpkt)) < 0)
2683 2684
        return err;

2685
    buf_consumed = (get_bits_count(&gb) + 7) >> 3;
2686 2687 2688 2689 2690
    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;
2691 2692
}

2693
static av_cold int aac_decode_close(AVCodecContext *avctx)
2694
{
2695
    AACContext *ac = avctx->priv_data;
2696
    int i, type;
2697

2698
    for (i = 0; i < MAX_ELEM_ID; i++) {
2699 2700 2701
        for (type = 0; type < 4; type++) {
            if (ac->che[type][i])
                ff_aac_sbr_ctx_close(&ac->che[type][i]->sbr);
2702
            av_freep(&ac->che[type][i]);
2703
        }
2704 2705 2706 2707
    }

    ff_mdct_end(&ac->mdct);
    ff_mdct_end(&ac->mdct_small);
2708
    ff_mdct_end(&ac->mdct_ltp);
2709
    return 0;
2710 2711
}

2712 2713 2714 2715 2716

#define LOAS_SYNC_WORD   0x2b7       ///< 11 bits LOAS sync word

struct LATMContext {
    AACContext      aac_ctx;             ///< containing AACContext
2717
    int             initialized;         ///< initialized after a valid extradata was seen
2718 2719 2720 2721 2722 2723 2724 2725 2726 2727 2728 2729 2730 2731 2732

    // parser data
    int             audio_mux_version_A; ///< LATM syntax version
    int             frame_length_type;   ///< 0/1 variable/fixed frame length
    int             frame_length;        ///< frame length for fixed frame length
};

static inline uint32_t latm_get_value(GetBitContext *b)
{
    int length = get_bits(b, 2);

    return get_bits_long(b, (length+1)*8);
}

static int latm_decode_audio_specific_config(struct LATMContext *latmctx,
2733
                                             GetBitContext *gb, int asclen)
2734
{
2735 2736
    AACContext *ac        = &latmctx->aac_ctx;
    AVCodecContext *avctx = ac->avctx;
2737
    MPEG4AudioConfig m4ac = { 0 };
2738 2739 2740 2741 2742 2743 2744 2745 2746
    int config_start_bit  = get_bits_count(gb);
    int sync_extension    = 0;
    int bits_consumed, esize;

    if (asclen) {
        sync_extension = 1;
        asclen         = FFMIN(asclen, get_bits_left(gb));
    } else
        asclen         = get_bits_left(gb);
2747 2748

    if (config_start_bit % 8) {
2749 2750
        av_log_missing_feature(latmctx->aac_ctx.avctx,
                               "Non-byte-aligned audio-specific config", 1);
2751
        return AVERROR_PATCHWELCOME;
2752
    }
2753 2754
    if (asclen <= 0)
        return AVERROR_INVALIDDATA;
2755
    bits_consumed = decode_audio_specific_config(NULL, avctx, &m4ac,
2756
                                         gb->buffer + (config_start_bit / 8),
2757
                                         asclen, sync_extension);
2758

2759 2760 2761
    if (bits_consumed < 0)
        return AVERROR_INVALIDDATA;

2762 2763
    if (!latmctx->initialized ||
        ac->oc[1].m4ac.sample_rate != m4ac.sample_rate ||
2764
        ac->oc[1].m4ac.chan_config != m4ac.chan_config) {
2765

2766 2767 2768 2769 2770
        if(latmctx->initialized) {
            av_log(avctx, AV_LOG_INFO, "audio config changed\n");
        } else {
            av_log(avctx, AV_LOG_INFO, "initializing latmctx\n");
        }
2771
        latmctx->initialized = 0;
2772 2773 2774

        esize = (bits_consumed+7) / 8;

2775
        if (avctx->extradata_size < esize) {
2776 2777 2778 2779 2780 2781 2782 2783 2784 2785
            av_free(avctx->extradata);
            avctx->extradata = av_malloc(esize + FF_INPUT_BUFFER_PADDING_SIZE);
            if (!avctx->extradata)
                return AVERROR(ENOMEM);
        }

        avctx->extradata_size = esize;
        memcpy(avctx->extradata, gb->buffer + (config_start_bit/8), esize);
        memset(avctx->extradata+esize, 0, FF_INPUT_BUFFER_PADDING_SIZE);
    }
2786
    skip_bits_long(gb, bits_consumed);
2787 2788 2789 2790 2791 2792 2793 2794 2795 2796 2797 2798 2799 2800 2801 2802 2803 2804 2805 2806 2807 2808 2809

    return bits_consumed;
}

static int read_stream_mux_config(struct LATMContext *latmctx,
                                  GetBitContext *gb)
{
    int ret, audio_mux_version = get_bits(gb, 1);

    latmctx->audio_mux_version_A = 0;
    if (audio_mux_version)
        latmctx->audio_mux_version_A = get_bits(gb, 1);

    if (!latmctx->audio_mux_version_A) {

        if (audio_mux_version)
            latm_get_value(gb);                 // taraFullness

        skip_bits(gb, 1);                       // allStreamSameTimeFraming
        skip_bits(gb, 6);                       // numSubFrames
        // numPrograms
        if (get_bits(gb, 4)) {                  // numPrograms
            av_log_missing_feature(latmctx->aac_ctx.avctx,
2810
                                   "Multiple programs", 1);
2811 2812 2813
            return AVERROR_PATCHWELCOME;
        }

2814
        // for each program (which there is only one in DVB)
2815

2816
        // for each layer (which there is only one in DVB)
2817 2818
        if (get_bits(gb, 3)) {                   // numLayer
            av_log_missing_feature(latmctx->aac_ctx.avctx,
2819
                                   "Multiple layers", 1);
2820 2821 2822 2823 2824
            return AVERROR_PATCHWELCOME;
        }

        // for all but first stream: use_same_config = get_bits(gb, 1);
        if (!audio_mux_version) {
2825
            if ((ret = latm_decode_audio_specific_config(latmctx, gb, 0)) < 0)
2826 2827 2828
                return ret;
        } else {
            int ascLen = latm_get_value(gb);
2829
            if ((ret = latm_decode_audio_specific_config(latmctx, gb, ascLen)) < 0)
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
                return ret;
            ascLen -= ret;
            skip_bits_long(gb, ascLen);
        }

        latmctx->frame_length_type = get_bits(gb, 3);
        switch (latmctx->frame_length_type) {
        case 0:
            skip_bits(gb, 8);       // latmBufferFullness
            break;
        case 1:
            latmctx->frame_length = get_bits(gb, 9);
            break;
        case 3:
        case 4:
        case 5:
            skip_bits(gb, 6);       // CELP frame length table index
            break;
        case 6:
        case 7:
            skip_bits(gb, 1);       // HVXC frame length table index
            break;
        }

        if (get_bits(gb, 1)) {                  // other data
            if (audio_mux_version) {
                latm_get_value(gb);             // other_data_bits
            } else {
                int esc;
                do {
                    esc = get_bits(gb, 1);
                    skip_bits(gb, 8);
                } while (esc);
            }
        }

        if (get_bits(gb, 1))                     // crc present
            skip_bits(gb, 8);                    // config_crc
    }

    return 0;
}

static int read_payload_length_info(struct LATMContext *ctx, GetBitContext *gb)
{
    uint8_t tmp;

    if (ctx->frame_length_type == 0) {
        int mux_slot_length = 0;
        do {
            tmp = get_bits(gb, 8);
            mux_slot_length += tmp;
        } while (tmp == 255);
        return mux_slot_length;
    } else if (ctx->frame_length_type == 1) {
        return ctx->frame_length;
    } else if (ctx->frame_length_type == 3 ||
               ctx->frame_length_type == 5 ||
               ctx->frame_length_type == 7) {
        skip_bits(gb, 2);          // mux_slot_length_coded
    }
    return 0;
}

static int read_audio_mux_element(struct LATMContext *latmctx,
                                  GetBitContext *gb)
{
    int err;
    uint8_t use_same_mux = get_bits(gb, 1);
    if (!use_same_mux) {
        if ((err = read_stream_mux_config(latmctx, gb)) < 0)
            return err;
    } else if (!latmctx->aac_ctx.avctx->extradata) {
        av_log(latmctx->aac_ctx.avctx, AV_LOG_DEBUG,
               "no decoder config found\n");
        return AVERROR(EAGAIN);
    }
    if (latmctx->audio_mux_version_A == 0) {
        int mux_slot_length_bytes = read_payload_length_info(latmctx, gb);
        if (mux_slot_length_bytes * 8 > get_bits_left(gb)) {
            av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR, "incomplete frame\n");
            return AVERROR_INVALIDDATA;
        } else if (mux_slot_length_bytes * 8 + 256 < get_bits_left(gb)) {
            av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR,
                   "frame length mismatch %d << %d\n",
                   mux_slot_length_bytes * 8, get_bits_left(gb));
            return AVERROR_INVALIDDATA;
        }
    }
    return 0;
}


2923 2924
static int latm_decode_frame(AVCodecContext *avctx, void *out,
                             int *got_frame_ptr, AVPacket *avpkt)
2925 2926 2927 2928 2929 2930 2931 2932 2933 2934 2935
{
    struct LATMContext *latmctx = avctx->priv_data;
    int                 muxlength, err;
    GetBitContext       gb;

    init_get_bits(&gb, avpkt->data, avpkt->size * 8);

    // check for LOAS sync word
    if (get_bits(&gb, 11) != LOAS_SYNC_WORD)
        return AVERROR_INVALIDDATA;

2936
    muxlength = get_bits(&gb, 13) + 3;
2937
    // not enough data, the parser should have sorted this out
2938
    if (muxlength > avpkt->size)
2939 2940 2941 2942 2943 2944 2945
        return AVERROR_INVALIDDATA;

    if ((err = read_audio_mux_element(latmctx, &gb)) < 0)
        return err;

    if (!latmctx->initialized) {
        if (!avctx->extradata) {
2946
            *got_frame_ptr = 0;
2947 2948
            return avpkt->size;
        } else {
2949
            push_output_configuration(&latmctx->aac_ctx);
2950
            if ((err = decode_audio_specific_config(
2951 2952 2953
                    &latmctx->aac_ctx, avctx, &latmctx->aac_ctx.oc[1].m4ac,
                    avctx->extradata, avctx->extradata_size*8, 1)) < 0) {
                pop_output_configuration(&latmctx->aac_ctx);
2954
                return err;
2955
            }
2956 2957 2958 2959 2960 2961 2962 2963 2964 2965 2966
            latmctx->initialized = 1;
        }
    }

    if (show_bits(&gb, 12) == 0xfff) {
        av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR,
               "ADTS header detected, probably as result of configuration "
               "misparsing\n");
        return AVERROR_INVALIDDATA;
    }

2967
    if ((err = aac_decode_frame_int(avctx, out, got_frame_ptr, &gb, avpkt)) < 0)
2968 2969 2970 2971 2972
        return err;

    return muxlength;
}

2973
static av_cold int latm_decode_init(AVCodecContext *avctx)
2974 2975
{
    struct LATMContext *latmctx = avctx->priv_data;
2976
    int ret = aac_decode_init(avctx);
2977

2978
    if (avctx->extradata_size > 0)
2979 2980 2981 2982 2983 2984
        latmctx->initialized = !ret;

    return ret;
}


2985
AVCodec ff_aac_decoder = {
2986 2987
    .name            = "aac",
    .type            = AVMEDIA_TYPE_AUDIO,
2988
    .id              = AV_CODEC_ID_AAC,
2989 2990 2991 2992
    .priv_data_size  = sizeof(AACContext),
    .init            = aac_decode_init,
    .close           = aac_decode_close,
    .decode          = aac_decode_frame,
2993
    .long_name       = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"),
2994
    .sample_fmts     = (const enum AVSampleFormat[]) {
2995
        AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE
2996
    },
2997
    .capabilities    = CODEC_CAP_CHANNEL_CONF | CODEC_CAP_DR1,
2998
    .channel_layouts = aac_channel_layout,
2999
    .flush = flush,
3000
};
3001 3002 3003 3004 3005 3006

/*
    Note: This decoder filter is intended to decode LATM streams transferred
    in MPEG transport streams which only contain one program.
    To do a more complex LATM demuxing a separate LATM demuxer should be used.
*/
3007
AVCodec ff_aac_latm_decoder = {
3008 3009
    .name            = "aac_latm",
    .type            = AVMEDIA_TYPE_AUDIO,
3010
    .id              = AV_CODEC_ID_AAC_LATM,
3011 3012 3013 3014
    .priv_data_size  = sizeof(struct LATMContext),
    .init            = latm_decode_init,
    .close           = aac_decode_close,
    .decode          = latm_decode_frame,
3015
    .long_name       = NULL_IF_CONFIG_SMALL("AAC LATM (Advanced Audio Coding LATM syntax)"),
3016
    .sample_fmts     = (const enum AVSampleFormat[]) {
3017
        AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE
3018
    },
3019
    .capabilities    = CODEC_CAP_CHANNEL_CONF | CODEC_CAP_DR1,
3020
    .channel_layouts = aac_channel_layout,
3021
    .flush = flush,
3022
};