[SCM] FFmpeg packaging branch, master.snapshot, updated. debian/0.6-1-18-g081dbd2

siretart at users.alioth.debian.org siretart at users.alioth.debian.org
Tue Jun 29 07:10:44 UTC 2010


The following commit has been merged in the master.snapshot branch:
commit 6af547cc4db57ed13713673148fb58809d4d4d12
Author: Reinhard Tartler <siretart at tauware.de>
Date:   Tue Jun 29 09:04:43 2010 +0200

    Backport-AAC-HE-v2
    
    This patch is currently under consideration for the 0.6.1 release

diff --git a/debian/patches/0003-Backport-AAC-HE-v2.patch b/debian/patches/0003-Backport-AAC-HE-v2.patch
new file mode 100644
index 0000000..babb9f5
--- /dev/null
+++ b/debian/patches/0003-Backport-AAC-HE-v2.patch
@@ -0,0 +1,6774 @@
+From: Reinhard Tartler <siretart at tauware.de>
+Subject: [PATCH] Backport AAC-HE-v2
+
+merge all revision that are related for aac encoder and decoder from trunk
+
+this patch is under consideration for the upcoming 0.6.1 release
+
+--- a/libavcodec/aac.c
++++ /dev/null
+@@ -1,2108 +0,0 @@
+-/*
+- * AAC decoder
+- * Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org )
+- * Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com )
+- *
+- * This file is part of FFmpeg.
+- *
+- * FFmpeg is free software; you can redistribute it and/or
+- * modify it under the terms of the GNU Lesser General Public
+- * License as published by the Free Software Foundation; either
+- * version 2.1 of the License, or (at your option) any later version.
+- *
+- * FFmpeg is distributed in the hope that it will be useful,
+- * but WITHOUT ANY WARRANTY; without even the implied warranty of
+- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+- * Lesser General Public License for more details.
+- *
+- * You should have received a copy of the GNU Lesser General Public
+- * License along with FFmpeg; if not, write to the Free Software
+- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+- */
+-
+-/**
+- * @file
+- * AAC decoder
+- * @author Oded Shimon  ( ods15 ods15 dyndns org )
+- * @author Maxim Gavrilov ( maxim.gavrilov gmail com )
+- */
+-
+-/*
+- * 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
+- * N (code in SoC repo) Long Term Prediction
+- * Y                    intensity stereo
+- * Y                    channel coupling
+- * Y                    frequency domain prediction
+- * Y                    Perceptual Noise Substitution
+- * Y                    Mid/Side stereo
+- * N                    Scalable Inverse AAC Quantization
+- * N                    Frequency Selective Switch
+- * N                    upsampling filter
+- * Y                    quantization & coding - AAC
+- * N                    quantization & coding - TwinVQ
+- * N                    quantization & coding - BSAC
+- * N                    AAC Error Resilience tools
+- * N                    Error Resilience payload syntax
+- * N                    Error Protection tool
+- * N                    CELP
+- * N                    Silence Compression
+- * N                    HVXC
+- * N                    HVXC 4kbits/s VR
+- * N                    Structured Audio tools
+- * N                    Structured Audio Sample Bank Format
+- * N                    MIDI
+- * N                    Harmonic and Individual Lines plus Noise
+- * N                    Text-To-Speech Interface
+- * Y                    Spectral Band Replication
+- * Y (not in this code) Layer-1
+- * Y (not in this code) Layer-2
+- * Y (not in this code) Layer-3
+- * N                    SinuSoidal Coding (Transient, Sinusoid, Noise)
+- * N (planned)          Parametric Stereo
+- * 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.
+- */
+-
+-
+-#include "avcodec.h"
+-#include "internal.h"
+-#include "get_bits.h"
+-#include "dsputil.h"
+-#include "fft.h"
+-#include "lpc.h"
+-
+-#include "aac.h"
+-#include "aactab.h"
+-#include "aacdectab.h"
+-#include "cbrt_tablegen.h"
+-#include "sbr.h"
+-#include "aacsbr.h"
+-#include "mpeg4audio.h"
+-#include "aac_parser.h"
+-
+-#include <assert.h>
+-#include <errno.h>
+-#include <math.h>
+-#include <string.h>
+-
+-#if ARCH_ARM
+-#   include "arm/aac.h"
+-#endif
+-
+-union float754 {
+-    float f;
+-    uint32_t i;
+-};
+-
+-static VLC vlc_scalefactors;
+-static VLC vlc_spectral[11];
+-
+-static const char overread_err[] = "Input buffer exhausted before END element found\n";
+-
+-static ChannelElement *get_che(AACContext *ac, int type, int elem_id)
+-{
+-    if (ac->tag_che_map[type][elem_id]) {
+-        return ac->tag_che_map[type][elem_id];
+-    }
+-    if (ac->tags_mapped >= tags_per_config[ac->m4ac.chan_config]) {
+-        return NULL;
+-    }
+-    switch (ac->m4ac.chan_config) {
+-    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[0] LFE[0]. If we seem to have
+-           encountered such a stream, transfer the LFE[0] element to SCE[1] */
+-        if (ac->tags_mapped == tags_per_config[ac->m4ac.chan_config] - 1 && (type == TYPE_LFE || type == TYPE_SCE)) {
+-            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:
+-        if (ac->tags_mapped == 2 && ac->m4ac.chan_config == 4 && type == TYPE_SCE) {
+-            ac->tags_mapped++;
+-            return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][1];
+-        }
+-    case 3:
+-    case 2:
+-        if (ac->tags_mapped == (ac->m4ac.chan_config != 2) && type == TYPE_CPE) {
+-            ac->tags_mapped++;
+-            return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][0];
+-        } else if (ac->m4ac.chan_config == 2) {
+-            return NULL;
+-        }
+-    case 1:
+-        if (!ac->tags_mapped && type == TYPE_SCE) {
+-            ac->tags_mapped++;
+-            return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][0];
+-        }
+-    default:
+-        return NULL;
+-    }
+-}
+-
+-/**
+- * Check for the channel element in the current channel position configuration.
+- * If it exists, make sure the appropriate element is allocated and map the
+- * channel order to match the internal FFmpeg channel layout.
+- *
+- * @param   che_pos current channel position configuration
+- * @param   type channel element type
+- * @param   id channel element id
+- * @param   channels count of the number of channels in the configuration
+- *
+- * @return  Returns error status. 0 - OK, !0 - error
+- */
+-static av_cold int che_configure(AACContext *ac,
+-                         enum ChannelPosition che_pos[4][MAX_ELEM_ID],
+-                         int type, int id,
+-                         int *channels)
+-{
+-    if (che_pos[type][id]) {
+-        if (!ac->che[type][id] && !(ac->che[type][id] = av_mallocz(sizeof(ChannelElement))))
+-            return AVERROR(ENOMEM);
+-        ff_aac_sbr_ctx_init(&ac->che[type][id]->sbr);
+-        if (type != TYPE_CCE) {
+-            ac->output_data[(*channels)++] = ac->che[type][id]->ch[0].ret;
+-            if (type == TYPE_CPE) {
+-                ac->output_data[(*channels)++] = ac->che[type][id]->ch[1].ret;
+-            }
+-        }
+-    } else {
+-        if (ac->che[type][id])
+-            ff_aac_sbr_ctx_close(&ac->che[type][id]->sbr);
+-        av_freep(&ac->che[type][id]);
+-    }
+-    return 0;
+-}
+-
+-/**
+- * Configure output channel order based on the current program configuration element.
+- *
+- * @param   che_pos current channel position configuration
+- * @param   new_che_pos New channel position configuration - we only do something if it differs from the current one.
+- *
+- * @return  Returns error status. 0 - OK, !0 - error
+- */
+-static av_cold int output_configure(AACContext *ac,
+-                            enum ChannelPosition che_pos[4][MAX_ELEM_ID],
+-                            enum ChannelPosition new_che_pos[4][MAX_ELEM_ID],
+-                            int channel_config, enum OCStatus oc_type)
+-{
+-    AVCodecContext *avctx = ac->avccontext;
+-    int i, type, channels = 0, ret;
+-
+-    memcpy(che_pos, new_che_pos, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]));
+-
+-    if (channel_config) {
+-        for (i = 0; i < tags_per_config[channel_config]; i++) {
+-            if ((ret = che_configure(ac, che_pos,
+-                                     aac_channel_layout_map[channel_config - 1][i][0],
+-                                     aac_channel_layout_map[channel_config - 1][i][1],
+-                                     &channels)))
+-                return ret;
+-        }
+-
+-        memset(ac->tag_che_map, 0,       4 * MAX_ELEM_ID * sizeof(ac->che[0][0]));
+-        ac->tags_mapped = 0;
+-
+-        avctx->channel_layout = aac_channel_layout[channel_config - 1];
+-    } else {
+-        /* Allocate or free elements depending on if they are in the
+-         * current program configuration.
+-         *
+-         * Set up default 1:1 output mapping.
+-         *
+-         * For a 5.1 stream the output order will be:
+-         *    [ Center ] [ Front Left ] [ Front Right ] [ LFE ] [ Surround Left ] [ Surround Right ]
+-         */
+-
+-        for (i = 0; i < MAX_ELEM_ID; i++) {
+-            for (type = 0; type < 4; type++) {
+-                if ((ret = che_configure(ac, che_pos, type, i, &channels)))
+-                    return ret;
+-            }
+-        }
+-
+-        memcpy(ac->tag_che_map, ac->che, 4 * MAX_ELEM_ID * sizeof(ac->che[0][0]));
+-        ac->tags_mapped = 4 * MAX_ELEM_ID;
+-
+-        avctx->channel_layout = 0;
+-    }
+-
+-    avctx->channels = channels;
+-
+-    ac->output_configured = oc_type;
+-
+-    return 0;
+-}
+-
+-/**
+- * Decode an array of 4 bit element IDs, optionally interleaved with a stereo/mono switching bit.
+- *
+- * @param cpe_map Stereo (Channel Pair Element) map, NULL if stereo bit is not present.
+- * @param sce_map mono (Single Channel Element) map
+- * @param type speaker type/position for these channels
+- */
+-static void decode_channel_map(enum ChannelPosition *cpe_map,
+-                               enum ChannelPosition *sce_map,
+-                               enum ChannelPosition type,
+-                               GetBitContext *gb, int n)
+-{
+-    while (n--) {
+-        enum ChannelPosition *map = cpe_map && get_bits1(gb) ? cpe_map : sce_map; // stereo or mono map
+-        map[get_bits(gb, 4)] = type;
+-    }
+-}
+-
+-/**
+- * Decode program configuration element; reference: table 4.2.
+- *
+- * @param   new_che_pos New channel position configuration - we only do something if it differs from the current one.
+- *
+- * @return  Returns error status. 0 - OK, !0 - error
+- */
+-static int decode_pce(AACContext *ac, enum ChannelPosition new_che_pos[4][MAX_ELEM_ID],
+-                      GetBitContext *gb)
+-{
+-    int num_front, num_side, num_back, num_lfe, num_assoc_data, num_cc, sampling_index;
+-    int comment_len;
+-
+-    skip_bits(gb, 2);  // object_type
+-
+-    sampling_index = get_bits(gb, 4);
+-    if (ac->m4ac.sampling_index != sampling_index)
+-        av_log(ac->avccontext, AV_LOG_WARNING, "Sample rate index in program config element does not match the sample rate index configured by the container.\n");
+-
+-    num_front       = get_bits(gb, 4);
+-    num_side        = get_bits(gb, 4);
+-    num_back        = get_bits(gb, 4);
+-    num_lfe         = get_bits(gb, 2);
+-    num_assoc_data  = get_bits(gb, 3);
+-    num_cc          = get_bits(gb, 4);
+-
+-    if (get_bits1(gb))
+-        skip_bits(gb, 4); // mono_mixdown_tag
+-    if (get_bits1(gb))
+-        skip_bits(gb, 4); // stereo_mixdown_tag
+-
+-    if (get_bits1(gb))
+-        skip_bits(gb, 3); // mixdown_coeff_index and pseudo_surround
+-
+-    decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_FRONT, gb, num_front);
+-    decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_SIDE,  gb, num_side );
+-    decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_BACK,  gb, num_back );
+-    decode_channel_map(NULL,                  new_che_pos[TYPE_LFE], AAC_CHANNEL_LFE,   gb, num_lfe  );
+-
+-    skip_bits_long(gb, 4 * num_assoc_data);
+-
+-    decode_channel_map(new_che_pos[TYPE_CCE], new_che_pos[TYPE_CCE], AAC_CHANNEL_CC,    gb, num_cc   );
+-
+-    align_get_bits(gb);
+-
+-    /* comment field, first byte is length */
+-    comment_len = get_bits(gb, 8) * 8;
+-    if (get_bits_left(gb) < comment_len) {
+-        av_log(ac->avccontext, AV_LOG_ERROR, overread_err);
+-        return -1;
+-    }
+-    skip_bits_long(gb, comment_len);
+-    return 0;
+-}
+-
+-/**
+- * Set up channel positions based on a default channel configuration
+- * as specified in table 1.17.
+- *
+- * @param   new_che_pos New channel position configuration - we only do something if it differs from the current one.
+- *
+- * @return  Returns error status. 0 - OK, !0 - error
+- */
+-static av_cold int set_default_channel_config(AACContext *ac,
+-                                      enum ChannelPosition new_che_pos[4][MAX_ELEM_ID],
+-                                      int channel_config)
+-{
+-    if (channel_config < 1 || channel_config > 7) {
+-        av_log(ac->avccontext, AV_LOG_ERROR, "invalid default channel configuration (%d)\n",
+-               channel_config);
+-        return -1;
+-    }
+-
+-    /* default channel configurations:
+-     *
+-     * 1ch : front center (mono)
+-     * 2ch : L + R (stereo)
+-     * 3ch : front center + L + R
+-     * 4ch : front center + L + R + back center
+-     * 5ch : front center + L + R + back stereo
+-     * 6ch : front center + L + R + back stereo + LFE
+-     * 7ch : front center + L + R + outer front left + outer front right + back stereo + LFE
+-     */
+-
+-    if (channel_config != 2)
+-        new_che_pos[TYPE_SCE][0] = AAC_CHANNEL_FRONT; // front center (or mono)
+-    if (channel_config > 1)
+-        new_che_pos[TYPE_CPE][0] = AAC_CHANNEL_FRONT; // L + R (or stereo)
+-    if (channel_config == 4)
+-        new_che_pos[TYPE_SCE][1] = AAC_CHANNEL_BACK;  // back center
+-    if (channel_config > 4)
+-        new_che_pos[TYPE_CPE][(channel_config == 7) + 1]
+-        = AAC_CHANNEL_BACK;  // back stereo
+-    if (channel_config > 5)
+-        new_che_pos[TYPE_LFE][0] = AAC_CHANNEL_LFE;   // LFE
+-    if (channel_config == 7)
+-        new_che_pos[TYPE_CPE][1] = AAC_CHANNEL_FRONT; // outer front left + outer front right
+-
+-    return 0;
+-}
+-
+-/**
+- * Decode GA "General Audio" specific configuration; reference: table 4.1.
+- *
+- * @return  Returns error status. 0 - OK, !0 - error
+- */
+-static int decode_ga_specific_config(AACContext *ac, GetBitContext *gb,
+-                                     int channel_config)
+-{
+-    enum ChannelPosition new_che_pos[4][MAX_ELEM_ID];
+-    int extension_flag, ret;
+-
+-    if (get_bits1(gb)) { // frameLengthFlag
+-        av_log_missing_feature(ac->avccontext, "960/120 MDCT window is", 1);
+-        return -1;
+-    }
+-
+-    if (get_bits1(gb))       // dependsOnCoreCoder
+-        skip_bits(gb, 14);   // coreCoderDelay
+-    extension_flag = get_bits1(gb);
+-
+-    if (ac->m4ac.object_type == AOT_AAC_SCALABLE ||
+-        ac->m4ac.object_type == AOT_ER_AAC_SCALABLE)
+-        skip_bits(gb, 3);     // layerNr
+-
+-    memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]));
+-    if (channel_config == 0) {
+-        skip_bits(gb, 4);  // element_instance_tag
+-        if ((ret = decode_pce(ac, new_che_pos, gb)))
+-            return ret;
+-    } else {
+-        if ((ret = set_default_channel_config(ac, new_che_pos, channel_config)))
+-            return ret;
+-    }
+-    if ((ret = output_configure(ac, ac->che_pos, new_che_pos, channel_config, OC_GLOBAL_HDR)))
+-        return ret;
+-
+-    if (extension_flag) {
+-        switch (ac->m4ac.object_type) {
+-        case AOT_ER_BSAC:
+-            skip_bits(gb, 5);    // numOfSubFrame
+-            skip_bits(gb, 11);   // layer_length
+-            break;
+-        case AOT_ER_AAC_LC:
+-        case AOT_ER_AAC_LTP:
+-        case AOT_ER_AAC_SCALABLE:
+-        case AOT_ER_AAC_LD:
+-            skip_bits(gb, 3);  /* aacSectionDataResilienceFlag
+-                                    * aacScalefactorDataResilienceFlag
+-                                    * aacSpectralDataResilienceFlag
+-                                    */
+-            break;
+-        }
+-        skip_bits1(gb);    // extensionFlag3 (TBD in version 3)
+-    }
+-    return 0;
+-}
+-
+-/**
+- * Decode audio specific configuration; reference: table 1.13.
+- *
+- * @param   data        pointer to AVCodecContext extradata
+- * @param   data_size   size of AVCCodecContext extradata
+- *
+- * @return  Returns error status. 0 - OK, !0 - error
+- */
+-static int decode_audio_specific_config(AACContext *ac, void *data,
+-                                        int data_size)
+-{
+-    GetBitContext gb;
+-    int i;
+-
+-    init_get_bits(&gb, data, data_size * 8);
+-
+-    if ((i = ff_mpeg4audio_get_config(&ac->m4ac, data, data_size)) < 0)
+-        return -1;
+-    if (ac->m4ac.sampling_index > 12) {
+-        av_log(ac->avccontext, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index);
+-        return -1;
+-    }
+-
+-    skip_bits_long(&gb, i);
+-
+-    switch (ac->m4ac.object_type) {
+-    case AOT_AAC_MAIN:
+-    case AOT_AAC_LC:
+-        if (decode_ga_specific_config(ac, &gb, ac->m4ac.chan_config))
+-            return -1;
+-        break;
+-    default:
+-        av_log(ac->avccontext, AV_LOG_ERROR, "Audio object type %s%d is not supported.\n",
+-               ac->m4ac.sbr == 1? "SBR+" : "", ac->m4ac.object_type);
+-        return -1;
+-    }
+-    return 0;
+-}
+-
+-/**
+- * linear congruential pseudorandom number generator
+- *
+- * @param   previous_val    pointer to the current state of the generator
+- *
+- * @return  Returns a 32-bit pseudorandom integer
+- */
+-static av_always_inline int lcg_random(int previous_val)
+-{
+-    return previous_val * 1664525 + 1013904223;
+-}
+-
+-static av_always_inline void reset_predict_state(PredictorState *ps)
+-{
+-    ps->r0   = 0.0f;
+-    ps->r1   = 0.0f;
+-    ps->cor0 = 0.0f;
+-    ps->cor1 = 0.0f;
+-    ps->var0 = 1.0f;
+-    ps->var1 = 1.0f;
+-}
+-
+-static void reset_all_predictors(PredictorState *ps)
+-{
+-    int i;
+-    for (i = 0; i < MAX_PREDICTORS; i++)
+-        reset_predict_state(&ps[i]);
+-}
+-
+-static void reset_predictor_group(PredictorState *ps, int group_num)
+-{
+-    int i;
+-    for (i = group_num - 1; i < MAX_PREDICTORS; i += 30)
+-        reset_predict_state(&ps[i]);
+-}
+-
+-static av_cold int aac_decode_init(AVCodecContext *avccontext)
+-{
+-    AACContext *ac = avccontext->priv_data;
+-    int i;
+-
+-    ac->avccontext = avccontext;
+-    ac->m4ac.sample_rate = avccontext->sample_rate;
+-
+-    if (avccontext->extradata_size > 0) {
+-        if (decode_audio_specific_config(ac, avccontext->extradata, avccontext->extradata_size))
+-            return -1;
+-    }
+-
+-    avccontext->sample_fmt = SAMPLE_FMT_S16;
+-
+-    AAC_INIT_VLC_STATIC( 0, 304);
+-    AAC_INIT_VLC_STATIC( 1, 270);
+-    AAC_INIT_VLC_STATIC( 2, 550);
+-    AAC_INIT_VLC_STATIC( 3, 300);
+-    AAC_INIT_VLC_STATIC( 4, 328);
+-    AAC_INIT_VLC_STATIC( 5, 294);
+-    AAC_INIT_VLC_STATIC( 6, 306);
+-    AAC_INIT_VLC_STATIC( 7, 268);
+-    AAC_INIT_VLC_STATIC( 8, 510);
+-    AAC_INIT_VLC_STATIC( 9, 366);
+-    AAC_INIT_VLC_STATIC(10, 462);
+-
+-    ff_aac_sbr_init();
+-
+-    dsputil_init(&ac->dsp, avccontext);
+-
+-    ac->random_state = 0x1f2e3d4c;
+-
+-    // -1024 - Compensate wrong IMDCT method.
+-    // 32768 - Required to scale values to the correct range for the bias method
+-    //         for float to int16 conversion.
+-
+-    if (ac->dsp.float_to_int16_interleave == ff_float_to_int16_interleave_c) {
+-        ac->add_bias  = 385.0f;
+-        ac->sf_scale  = 1. / (-1024. * 32768.);
+-        ac->sf_offset = 0;
+-    } else {
+-        ac->add_bias  = 0.0f;
+-        ac->sf_scale  = 1. / -1024.;
+-        ac->sf_offset = 60;
+-    }
+-
+-#if !CONFIG_HARDCODED_TABLES
+-    for (i = 0; i < 428; i++)
+-        ff_aac_pow2sf_tab[i] = pow(2, (i - 200) / 4.);
+-#endif /* CONFIG_HARDCODED_TABLES */
+-
+-    INIT_VLC_STATIC(&vlc_scalefactors,7,FF_ARRAY_ELEMS(ff_aac_scalefactor_code),
+-                    ff_aac_scalefactor_bits, sizeof(ff_aac_scalefactor_bits[0]), sizeof(ff_aac_scalefactor_bits[0]),
+-                    ff_aac_scalefactor_code, sizeof(ff_aac_scalefactor_code[0]), sizeof(ff_aac_scalefactor_code[0]),
+-                    352);
+-
+-    ff_mdct_init(&ac->mdct, 11, 1, 1.0);
+-    ff_mdct_init(&ac->mdct_small, 8, 1, 1.0);
+-    // 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);
+-    ff_init_ff_sine_windows(10);
+-    ff_init_ff_sine_windows( 7);
+-
+-    cbrt_tableinit();
+-
+-    return 0;
+-}
+-
+-/**
+- * Skip data_stream_element; reference: table 4.10.
+- */
+-static int skip_data_stream_element(AACContext *ac, GetBitContext *gb)
+-{
+-    int byte_align = get_bits1(gb);
+-    int count = get_bits(gb, 8);
+-    if (count == 255)
+-        count += get_bits(gb, 8);
+-    if (byte_align)
+-        align_get_bits(gb);
+-
+-    if (get_bits_left(gb) < 8 * count) {
+-        av_log(ac->avccontext, AV_LOG_ERROR, overread_err);
+-        return -1;
+-    }
+-    skip_bits_long(gb, 8 * count);
+-    return 0;
+-}
+-
+-static int decode_prediction(AACContext *ac, IndividualChannelStream *ics,
+-                             GetBitContext *gb)
+-{
+-    int sfb;
+-    if (get_bits1(gb)) {
+-        ics->predictor_reset_group = get_bits(gb, 5);
+-        if (ics->predictor_reset_group == 0 || ics->predictor_reset_group > 30) {
+-            av_log(ac->avccontext, AV_LOG_ERROR, "Invalid Predictor Reset Group.\n");
+-            return -1;
+-        }
+-    }
+-    for (sfb = 0; sfb < FFMIN(ics->max_sfb, ff_aac_pred_sfb_max[ac->m4ac.sampling_index]); sfb++) {
+-        ics->prediction_used[sfb] = get_bits1(gb);
+-    }
+-    return 0;
+-}
+-
+-/**
+- * Decode Individual Channel Stream info; reference: table 4.6.
+- *
+- * @param   common_window   Channels have independent [0], or shared [1], Individual Channel Stream information.
+- */
+-static int decode_ics_info(AACContext *ac, IndividualChannelStream *ics,
+-                           GetBitContext *gb, int common_window)
+-{
+-    if (get_bits1(gb)) {
+-        av_log(ac->avccontext, AV_LOG_ERROR, "Reserved bit set.\n");
+-        memset(ics, 0, sizeof(IndividualChannelStream));
+-        return -1;
+-    }
+-    ics->window_sequence[1] = ics->window_sequence[0];
+-    ics->window_sequence[0] = get_bits(gb, 2);
+-    ics->use_kb_window[1]   = ics->use_kb_window[0];
+-    ics->use_kb_window[0]   = get_bits1(gb);
+-    ics->num_window_groups  = 1;
+-    ics->group_len[0]       = 1;
+-    if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
+-        int i;
+-        ics->max_sfb = get_bits(gb, 4);
+-        for (i = 0; i < 7; i++) {
+-            if (get_bits1(gb)) {
+-                ics->group_len[ics->num_window_groups - 1]++;
+-            } else {
+-                ics->num_window_groups++;
+-                ics->group_len[ics->num_window_groups - 1] = 1;
+-            }
+-        }
+-        ics->num_windows       = 8;
+-        ics->swb_offset        =    ff_swb_offset_128[ac->m4ac.sampling_index];
+-        ics->num_swb           =   ff_aac_num_swb_128[ac->m4ac.sampling_index];
+-        ics->tns_max_bands     = ff_tns_max_bands_128[ac->m4ac.sampling_index];
+-        ics->predictor_present = 0;
+-    } else {
+-        ics->max_sfb               = get_bits(gb, 6);
+-        ics->num_windows           = 1;
+-        ics->swb_offset            =    ff_swb_offset_1024[ac->m4ac.sampling_index];
+-        ics->num_swb               =   ff_aac_num_swb_1024[ac->m4ac.sampling_index];
+-        ics->tns_max_bands         = ff_tns_max_bands_1024[ac->m4ac.sampling_index];
+-        ics->predictor_present     = get_bits1(gb);
+-        ics->predictor_reset_group = 0;
+-        if (ics->predictor_present) {
+-            if (ac->m4ac.object_type == AOT_AAC_MAIN) {
+-                if (decode_prediction(ac, ics, gb)) {
+-                    memset(ics, 0, sizeof(IndividualChannelStream));
+-                    return -1;
+-                }
+-            } else if (ac->m4ac.object_type == AOT_AAC_LC) {
+-                av_log(ac->avccontext, AV_LOG_ERROR, "Prediction is not allowed in AAC-LC.\n");
+-                memset(ics, 0, sizeof(IndividualChannelStream));
+-                return -1;
+-            } else {
+-                av_log_missing_feature(ac->avccontext, "Predictor bit set but LTP is", 1);
+-                memset(ics, 0, sizeof(IndividualChannelStream));
+-                return -1;
+-            }
+-        }
+-    }
+-
+-    if (ics->max_sfb > ics->num_swb) {
+-        av_log(ac->avccontext, AV_LOG_ERROR,
+-               "Number of scalefactor bands in group (%d) exceeds limit (%d).\n",
+-               ics->max_sfb, ics->num_swb);
+-        memset(ics, 0, sizeof(IndividualChannelStream));
+-        return -1;
+-    }
+-
+-    return 0;
+-}
+-
+-/**
+- * Decode band types (section_data payload); reference: table 4.46.
+- *
+- * @param   band_type           array of the used band type
+- * @param   band_type_run_end   array of the last scalefactor band of a band type run
+- *
+- * @return  Returns error status. 0 - OK, !0 - error
+- */
+-static int decode_band_types(AACContext *ac, enum BandType band_type[120],
+-                             int band_type_run_end[120], GetBitContext *gb,
+-                             IndividualChannelStream *ics)
+-{
+-    int g, idx = 0;
+-    const int bits = (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) ? 3 : 5;
+-    for (g = 0; g < ics->num_window_groups; g++) {
+-        int k = 0;
+-        while (k < ics->max_sfb) {
+-            uint8_t sect_end = k;
+-            int sect_len_incr;
+-            int sect_band_type = get_bits(gb, 4);
+-            if (sect_band_type == 12) {
+-                av_log(ac->avccontext, AV_LOG_ERROR, "invalid band type\n");
+-                return -1;
+-            }
+-            while ((sect_len_incr = get_bits(gb, bits)) == (1 << bits) - 1)
+-                sect_end += sect_len_incr;
+-            sect_end += sect_len_incr;
+-            if (get_bits_left(gb) < 0) {
+-                av_log(ac->avccontext, AV_LOG_ERROR, overread_err);
+-                return -1;
+-            }
+-            if (sect_end > ics->max_sfb) {
+-                av_log(ac->avccontext, AV_LOG_ERROR,
+-                       "Number of bands (%d) exceeds limit (%d).\n",
+-                       sect_end, ics->max_sfb);
+-                return -1;
+-            }
+-            for (; k < sect_end; k++) {
+-                band_type        [idx]   = sect_band_type;
+-                band_type_run_end[idx++] = sect_end;
+-            }
+-        }
+-    }
+-    return 0;
+-}
+-
+-/**
+- * Decode scalefactors; reference: table 4.47.
+- *
+- * @param   global_gain         first scalefactor value as scalefactors are differentially coded
+- * @param   band_type           array of the used band type
+- * @param   band_type_run_end   array of the last scalefactor band of a band type run
+- * @param   sf                  array of scalefactors or intensity stereo positions
+- *
+- * @return  Returns error status. 0 - OK, !0 - error
+- */
+-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])
+-{
+-    const int sf_offset = ac->sf_offset + (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE ? 12 : 0);
+-    int g, i, idx = 0;
+-    int offset[3] = { global_gain, global_gain - 90, 100 };
+-    int noise_flag = 1;
+-    static const char *sf_str[3] = { "Global gain", "Noise gain", "Intensity stereo position" };
+-    for (g = 0; g < ics->num_window_groups; g++) {
+-        for (i = 0; i < ics->max_sfb;) {
+-            int run_end = band_type_run_end[idx];
+-            if (band_type[idx] == ZERO_BT) {
+-                for (; i < run_end; i++, idx++)
+-                    sf[idx] = 0.;
+-            } else if ((band_type[idx] == INTENSITY_BT) || (band_type[idx] == INTENSITY_BT2)) {
+-                for (; i < run_end; i++, idx++) {
+-                    offset[2] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
+-                    if (offset[2] > 255U) {
+-                        av_log(ac->avccontext, AV_LOG_ERROR,
+-                               "%s (%d) out of range.\n", sf_str[2], offset[2]);
+-                        return -1;
+-                    }
+-                    sf[idx] = ff_aac_pow2sf_tab[-offset[2] + 300];
+-                }
+-            } else if (band_type[idx] == NOISE_BT) {
+-                for (; i < run_end; i++, idx++) {
+-                    if (noise_flag-- > 0)
+-                        offset[1] += get_bits(gb, 9) - 256;
+-                    else
+-                        offset[1] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
+-                    if (offset[1] > 255U) {
+-                        av_log(ac->avccontext, AV_LOG_ERROR,
+-                               "%s (%d) out of range.\n", sf_str[1], offset[1]);
+-                        return -1;
+-                    }
+-                    sf[idx] = -ff_aac_pow2sf_tab[offset[1] + sf_offset + 100];
+-                }
+-            } else {
+-                for (; i < run_end; i++, idx++) {
+-                    offset[0] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
+-                    if (offset[0] > 255U) {
+-                        av_log(ac->avccontext, AV_LOG_ERROR,
+-                               "%s (%d) out of range.\n", sf_str[0], offset[0]);
+-                        return -1;
+-                    }
+-                    sf[idx] = -ff_aac_pow2sf_tab[ offset[0] + sf_offset];
+-                }
+-            }
+-        }
+-    }
+-    return 0;
+-}
+-
+-/**
+- * Decode pulse data; reference: table 4.7.
+- */
+-static int decode_pulses(Pulse *pulse, GetBitContext *gb,
+-                         const uint16_t *swb_offset, int num_swb)
+-{
+-    int i, pulse_swb;
+-    pulse->num_pulse = get_bits(gb, 2) + 1;
+-    pulse_swb        = get_bits(gb, 6);
+-    if (pulse_swb >= num_swb)
+-        return -1;
+-    pulse->pos[0]    = swb_offset[pulse_swb];
+-    pulse->pos[0]   += get_bits(gb, 5);
+-    if (pulse->pos[0] > 1023)
+-        return -1;
+-    pulse->amp[0]    = get_bits(gb, 4);
+-    for (i = 1; i < pulse->num_pulse; i++) {
+-        pulse->pos[i] = get_bits(gb, 5) + pulse->pos[i - 1];
+-        if (pulse->pos[i] > 1023)
+-            return -1;
+-        pulse->amp[i] = get_bits(gb, 4);
+-    }
+-    return 0;
+-}
+-
+-/**
+- * Decode Temporal Noise Shaping data; reference: table 4.48.
+- *
+- * @return  Returns error status. 0 - OK, !0 - error
+- */
+-static int decode_tns(AACContext *ac, TemporalNoiseShaping *tns,
+-                      GetBitContext *gb, const IndividualChannelStream *ics)
+-{
+-    int w, filt, i, coef_len, coef_res, coef_compress;
+-    const int is8 = ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE;
+-    const int tns_max_order = is8 ? 7 : ac->m4ac.object_type == AOT_AAC_MAIN ? 20 : 12;
+-    for (w = 0; w < ics->num_windows; w++) {
+-        if ((tns->n_filt[w] = get_bits(gb, 2 - is8))) {
+-            coef_res = get_bits1(gb);
+-
+-            for (filt = 0; filt < tns->n_filt[w]; filt++) {
+-                int tmp2_idx;
+-                tns->length[w][filt] = get_bits(gb, 6 - 2 * is8);
+-
+-                if ((tns->order[w][filt] = get_bits(gb, 5 - 2 * is8)) > tns_max_order) {
+-                    av_log(ac->avccontext, AV_LOG_ERROR, "TNS filter order %d is greater than maximum %d.\n",
+-                           tns->order[w][filt], tns_max_order);
+-                    tns->order[w][filt] = 0;
+-                    return -1;
+-                }
+-                if (tns->order[w][filt]) {
+-                    tns->direction[w][filt] = get_bits1(gb);
+-                    coef_compress = get_bits1(gb);
+-                    coef_len = coef_res + 3 - coef_compress;
+-                    tmp2_idx = 2 * coef_compress + coef_res;
+-
+-                    for (i = 0; i < tns->order[w][filt]; i++)
+-                        tns->coef[w][filt][i] = tns_tmp2_map[tmp2_idx][get_bits(gb, coef_len)];
+-                }
+-            }
+-        }
+-    }
+-    return 0;
+-}
+-
+-/**
+- * Decode Mid/Side data; reference: table 4.54.
+- *
+- * @param   ms_present  Indicates mid/side stereo presence. [0] mask is all 0s;
+- *                      [1] mask is decoded from bitstream; [2] mask is all 1s;
+- *                      [3] reserved for scalable AAC
+- */
+-static void decode_mid_side_stereo(ChannelElement *cpe, GetBitContext *gb,
+-                                   int ms_present)
+-{
+-    int idx;
+-    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) {
+-        memset(cpe->ms_mask, 1, cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb * sizeof(cpe->ms_mask[0]));
+-    }
+-}
+-
+-#ifndef VMUL2
+-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;
+-}
+-#endif
+-
+-#ifndef VMUL4
+-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;
+-}
+-#endif
+-
+-#ifndef VMUL2S
+-static inline float *VMUL2S(float *dst, const float *v, unsigned idx,
+-                            unsigned sign, const float *scale)
+-{
+-    union float754 s0, s1;
+-
+-    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;
+-}
+-#endif
+-
+-#ifndef VMUL4S
+-static inline float *VMUL4S(float *dst, const float *v, unsigned idx,
+-                            unsigned sign, const float *scale)
+-{
+-    unsigned nz = idx >> 12;
+-    union float754 s = { .f = *scale };
+-    union float754 t;
+-
+-    t.i = s.i ^ (sign & 1<<31);
+-    *dst++ = v[idx    & 3] * t.f;
+-
+-    sign <<= nz & 1; nz >>= 1;
+-    t.i = s.i ^ (sign & 1<<31);
+-    *dst++ = v[idx>>2 & 3] * t.f;
+-
+-    sign <<= nz & 1; nz >>= 1;
+-    t.i = s.i ^ (sign & 1<<31);
+-    *dst++ = v[idx>>4 & 3] * t.f;
+-
+-    sign <<= nz & 1; nz >>= 1;
+-    t.i = s.i ^ (sign & 1<<31);
+-    *dst++ = v[idx>>6 & 3] * t.f;
+-
+-    return dst;
+-}
+-#endif
+-
+-/**
+- * 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
+- */
+-static int decode_spectrum_and_dequant(AACContext *ac, float coef[1024],
+-                                       GetBitContext *gb, const float sf[120],
+-                                       int pulse_present, const Pulse *pulse,
+-                                       const IndividualChannelStream *ics,
+-                                       enum BandType band_type[120])
+-{
+-    int i, k, g, idx = 0;
+-    const int c = 1024 / ics->num_windows;
+-    const uint16_t *offsets = ics->swb_offset;
+-    float *coef_base = coef;
+-    int err_idx;
+-
+-    for (g = 0; g < ics->num_windows; g++)
+-        memset(coef + g * 128 + offsets[ics->max_sfb], 0, sizeof(float) * (c - offsets[ics->max_sfb]));
+-
+-    for (g = 0; g < ics->num_window_groups; g++) {
+-        unsigned g_len = ics->group_len[g];
+-
+-        for (i = 0; i < ics->max_sfb; i++, idx++) {
+-            const unsigned cbt_m1 = band_type[idx] - 1;
+-            float *cfo = coef + offsets[i];
+-            int off_len = offsets[i + 1] - offsets[i];
+-            int group;
+-
+-            if (cbt_m1 >= INTENSITY_BT2 - 1) {
+-                for (group = 0; group < g_len; group++, cfo+=128) {
+-                    memset(cfo, 0, off_len * sizeof(float));
+-                }
+-            } else if (cbt_m1 == NOISE_BT - 1) {
+-                for (group = 0; group < g_len; group++, cfo+=128) {
+-                    float scale;
+-                    float band_energy;
+-
+-                    for (k = 0; k < off_len; k++) {
+-                        ac->random_state  = lcg_random(ac->random_state);
+-                        cfo[k] = ac->random_state;
+-                    }
+-
+-                    band_energy = ac->dsp.scalarproduct_float(cfo, cfo, off_len);
+-                    scale = sf[idx] / sqrtf(band_energy);
+-                    ac->dsp.vector_fmul_scalar(cfo, cfo, scale, off_len);
+-                }
+-            } else {
+-                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;
+-                const int cb_size = ff_aac_spectral_sizes[cbt_m1];
+-                OPEN_READER(re, gb);
+-
+-                switch (cbt_m1 >> 1) {
+-                case 0:
+-                    for (group = 0; group < g_len; group++, cfo+=128) {
+-                        float *cf = cfo;
+-                        int len = off_len;
+-
+-                        do {
+-                            int code;
+-                            unsigned cb_idx;
+-
+-                            UPDATE_CACHE(re, gb);
+-                            GET_VLC(code, re, gb, vlc_tab, 8, 2);
+-
+-                            if (code >= cb_size) {
+-                                err_idx = code;
+-                                goto err_cb_overflow;
+-                            }
+-
+-                            cb_idx = cb_vector_idx[code];
+-                            cf = VMUL4(cf, vq, cb_idx, sf + idx);
+-                        } while (len -= 4);
+-                    }
+-                    break;
+-
+-                case 1:
+-                    for (group = 0; group < g_len; group++, cfo+=128) {
+-                        float *cf = cfo;
+-                        int len = off_len;
+-
+-                        do {
+-                            int code;
+-                            unsigned nnz;
+-                            unsigned cb_idx;
+-                            uint32_t bits;
+-
+-                            UPDATE_CACHE(re, gb);
+-                            GET_VLC(code, re, gb, vlc_tab, 8, 2);
+-
+-                            if (code >= cb_size) {
+-                                err_idx = code;
+-                                goto err_cb_overflow;
+-                            }
+-
+-#if MIN_CACHE_BITS < 20
+-                            UPDATE_CACHE(re, gb);
+-#endif
+-                            cb_idx = cb_vector_idx[code];
+-                            nnz = cb_idx >> 8 & 15;
+-                            bits = SHOW_UBITS(re, gb, nnz) << (32-nnz);
+-                            LAST_SKIP_BITS(re, gb, nnz);
+-                            cf = VMUL4S(cf, vq, cb_idx, bits, sf + idx);
+-                        } while (len -= 4);
+-                    }
+-                    break;
+-
+-                case 2:
+-                    for (group = 0; group < g_len; group++, cfo+=128) {
+-                        float *cf = cfo;
+-                        int len = off_len;
+-
+-                        do {
+-                            int code;
+-                            unsigned cb_idx;
+-
+-                            UPDATE_CACHE(re, gb);
+-                            GET_VLC(code, re, gb, vlc_tab, 8, 2);
+-
+-                            if (code >= cb_size) {
+-                                err_idx = code;
+-                                goto err_cb_overflow;
+-                            }
+-
+-                            cb_idx = cb_vector_idx[code];
+-                            cf = VMUL2(cf, vq, cb_idx, sf + idx);
+-                        } while (len -= 2);
+-                    }
+-                    break;
+-
+-                case 3:
+-                case 4:
+-                    for (group = 0; group < g_len; group++, cfo+=128) {
+-                        float *cf = cfo;
+-                        int len = off_len;
+-
+-                        do {
+-                            int code;
+-                            unsigned nnz;
+-                            unsigned cb_idx;
+-                            unsigned sign;
+-
+-                            UPDATE_CACHE(re, gb);
+-                            GET_VLC(code, re, gb, vlc_tab, 8, 2);
+-
+-                            if (code >= cb_size) {
+-                                err_idx = code;
+-                                goto err_cb_overflow;
+-                            }
+-
+-                            cb_idx = cb_vector_idx[code];
+-                            nnz = cb_idx >> 8 & 15;
+-                            sign = SHOW_UBITS(re, gb, nnz) << (cb_idx >> 12);
+-                            LAST_SKIP_BITS(re, gb, nnz);
+-                            cf = VMUL2S(cf, vq, cb_idx, sign, sf + idx);
+-                        } while (len -= 2);
+-                    }
+-                    break;
+-
+-                default:
+-                    for (group = 0; group < g_len; group++, cfo+=128) {
+-                        float *cf = cfo;
+-                        uint32_t *icf = (uint32_t *) cf;
+-                        int len = off_len;
+-
+-                        do {
+-                            int code;
+-                            unsigned nzt, nnz;
+-                            unsigned cb_idx;
+-                            uint32_t bits;
+-                            int j;
+-
+-                            UPDATE_CACHE(re, gb);
+-                            GET_VLC(code, re, gb, vlc_tab, 8, 2);
+-
+-                            if (!code) {
+-                                *icf++ = 0;
+-                                *icf++ = 0;
+-                                continue;
+-                            }
+-
+-                            if (code >= cb_size) {
+-                                err_idx = code;
+-                                goto err_cb_overflow;
+-                            }
+-
+-                            cb_idx = cb_vector_idx[code];
+-                            nnz = cb_idx >> 12;
+-                            nzt = cb_idx >> 8;
+-                            bits = SHOW_UBITS(re, gb, nnz) << (32-nnz);
+-                            LAST_SKIP_BITS(re, gb, nnz);
+-
+-                            for (j = 0; j < 2; j++) {
+-                                if (nzt & 1<<j) {
+-                                    uint32_t b;
+-                                    int n;
+-                                    /* The total length of escape_sequence must be < 22 bits according
+-                                       to the specification (i.e. max is 111111110xxxxxxxxxxxx). */
+-                                    UPDATE_CACHE(re, gb);
+-                                    b = GET_CACHE(re, gb);
+-                                    b = 31 - av_log2(~b);
+-
+-                                    if (b > 8) {
+-                                        av_log(ac->avccontext, AV_LOG_ERROR, "error in spectral data, ESC overflow\n");
+-                                        return -1;
+-                                    }
+-
+-#if MIN_CACHE_BITS < 21
+-                                    LAST_SKIP_BITS(re, gb, b + 1);
+-                                    UPDATE_CACHE(re, gb);
+-#else
+-                                    SKIP_BITS(re, gb, b + 1);
+-#endif
+-                                    b += 4;
+-                                    n = (1 << b) + SHOW_UBITS(re, gb, b);
+-                                    LAST_SKIP_BITS(re, gb, b);
+-                                    *icf++ = cbrt_tab[n] | (bits & 1<<31);
+-                                    bits <<= 1;
+-                                } else {
+-                                    unsigned v = ((const uint32_t*)vq)[cb_idx & 15];
+-                                    *icf++ = (bits & 1<<31) | v;
+-                                    bits <<= !!v;
+-                                }
+-                                cb_idx >>= 4;
+-                            }
+-                        } while (len -= 2);
+-
+-                        ac->dsp.vector_fmul_scalar(cfo, cfo, sf[idx], off_len);
+-                    }
+-                }
+-
+-                CLOSE_READER(re, gb);
+-            }
+-        }
+-        coef += g_len << 7;
+-    }
+-
+-    if (pulse_present) {
+-        idx = 0;
+-        for (i = 0; i < pulse->num_pulse; i++) {
+-            float co = coef_base[ pulse->pos[i] ];
+-            while (offsets[idx + 1] <= pulse->pos[i])
+-                idx++;
+-            if (band_type[idx] != NOISE_BT && sf[idx]) {
+-                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];
+-            }
+-        }
+-    }
+-    return 0;
+-
+-err_cb_overflow:
+-    av_log(ac->avccontext, AV_LOG_ERROR,
+-           "Read beyond end of ff_aac_codebook_vectors[%d][]. index %d >= %d\n",
+-           band_type[idx], err_idx, ff_aac_spectral_sizes[band_type[idx]]);
+-    return -1;
+-}
+-
+-static av_always_inline float flt16_round(float pf)
+-{
+-    union float754 tmp;
+-    tmp.f = pf;
+-    tmp.i = (tmp.i + 0x00008000U) & 0xFFFF0000U;
+-    return tmp.f;
+-}
+-
+-static av_always_inline float flt16_even(float pf)
+-{
+-    union float754 tmp;
+-    tmp.f = pf;
+-    tmp.i = (tmp.i + 0x00007FFFU + (tmp.i & 0x00010000U >> 16)) & 0xFFFF0000U;
+-    return tmp.f;
+-}
+-
+-static av_always_inline float flt16_trunc(float pf)
+-{
+-    union float754 pun;
+-    pun.f = pf;
+-    pun.i &= 0xFFFF0000U;
+-    return pun.f;
+-}
+-
+-static av_always_inline void predict(AACContext *ac, PredictorState *ps, float *coef,
+-                    int output_enable)
+-{
+-    const float a     = 0.953125; // 61.0 / 64
+-    const float alpha = 0.90625;  // 29.0 / 32
+-    float e0, e1;
+-    float pv;
+-    float k1, k2;
+-
+-    k1 = ps->var0 > 1 ? ps->cor0 * flt16_even(a / ps->var0) : 0;
+-    k2 = ps->var1 > 1 ? ps->cor1 * flt16_even(a / ps->var1) : 0;
+-
+-    pv = flt16_round(k1 * ps->r0 + k2 * ps->r1);
+-    if (output_enable)
+-        *coef += pv * ac->sf_scale;
+-
+-    e0 = *coef / ac->sf_scale;
+-    e1 = e0 - k1 * ps->r0;
+-
+-    ps->cor1 = flt16_trunc(alpha * ps->cor1 + ps->r1 * e1);
+-    ps->var1 = flt16_trunc(alpha * ps->var1 + 0.5 * (ps->r1 * ps->r1 + e1 * e1));
+-    ps->cor0 = flt16_trunc(alpha * ps->cor0 + ps->r0 * e0);
+-    ps->var0 = flt16_trunc(alpha * ps->var0 + 0.5 * (ps->r0 * ps->r0 + e0 * e0));
+-
+-    ps->r1 = flt16_trunc(a * (ps->r0 - k1 * e0));
+-    ps->r0 = flt16_trunc(a * e0);
+-}
+-
+-/**
+- * Apply AAC-Main style frequency domain prediction.
+- */
+-static void apply_prediction(AACContext *ac, SingleChannelElement *sce)
+-{
+-    int sfb, k;
+-
+-    if (!sce->ics.predictor_initialized) {
+-        reset_all_predictors(sce->predictor_state);
+-        sce->ics.predictor_initialized = 1;
+-    }
+-
+-    if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
+-        for (sfb = 0; sfb < ff_aac_pred_sfb_max[ac->m4ac.sampling_index]; sfb++) {
+-            for (k = sce->ics.swb_offset[sfb]; k < sce->ics.swb_offset[sfb + 1]; k++) {
+-                predict(ac, &sce->predictor_state[k], &sce->coeffs[k],
+-                        sce->ics.predictor_present && sce->ics.prediction_used[sfb]);
+-            }
+-        }
+-        if (sce->ics.predictor_reset_group)
+-            reset_predictor_group(sce->predictor_state, sce->ics.predictor_reset_group);
+-    } else
+-        reset_all_predictors(sce->predictor_state);
+-}
+-
+-/**
+- * Decode an individual_channel_stream payload; reference: table 4.44.
+- *
+- * @param   common_window   Channels have independent [0], or shared [1], Individual Channel Stream information.
+- * @param   scale_flag      scalable [1] or non-scalable [0] AAC (Unused until scalable AAC is implemented.)
+- *
+- * @return  Returns error status. 0 - OK, !0 - error
+- */
+-static int decode_ics(AACContext *ac, SingleChannelElement *sce,
+-                      GetBitContext *gb, int common_window, int scale_flag)
+-{
+-    Pulse pulse;
+-    TemporalNoiseShaping    *tns = &sce->tns;
+-    IndividualChannelStream *ics = &sce->ics;
+-    float *out = sce->coeffs;
+-    int global_gain, pulse_present = 0;
+-
+-    /* This assignment is to silence a GCC warning about the variable being used
+-     * uninitialized when in fact it always is.
+-     */
+-    pulse.num_pulse = 0;
+-
+-    global_gain = get_bits(gb, 8);
+-
+-    if (!common_window && !scale_flag) {
+-        if (decode_ics_info(ac, ics, gb, 0) < 0)
+-            return -1;
+-    }
+-
+-    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) {
+-                av_log(ac->avccontext, AV_LOG_ERROR, "Pulse tool not allowed in eight short sequence.\n");
+-                return -1;
+-            }
+-            if (decode_pulses(&pulse, gb, ics->swb_offset, ics->num_swb)) {
+-                av_log(ac->avccontext, AV_LOG_ERROR, "Pulse data corrupt or invalid.\n");
+-                return -1;
+-            }
+-        }
+-        if ((tns->present = get_bits1(gb)) && decode_tns(ac, tns, gb, ics))
+-            return -1;
+-        if (get_bits1(gb)) {
+-            av_log_missing_feature(ac->avccontext, "SSR", 1);
+-            return -1;
+-        }
+-    }
+-
+-    if (decode_spectrum_and_dequant(ac, out, gb, sce->sf, pulse_present, &pulse, ics, sce->band_type) < 0)
+-        return -1;
+-
+-    if (ac->m4ac.object_type == AOT_AAC_MAIN && !common_window)
+-        apply_prediction(ac, sce);
+-
+-    return 0;
+-}
+-
+-/**
+- * Mid/Side stereo decoding; reference: 4.6.8.1.3.
+- */
+-static void apply_mid_side_stereo(AACContext *ac, ChannelElement *cpe)
+-{
+-    const IndividualChannelStream *ics = &cpe->ch[0].ics;
+-    float *ch0 = cpe->ch[0].coeffs;
+-    float *ch1 = cpe->ch[1].coeffs;
+-    int g, i, group, idx = 0;
+-    const uint16_t *offsets = ics->swb_offset;
+-    for (g = 0; g < ics->num_window_groups; g++) {
+-        for (i = 0; i < ics->max_sfb; i++, idx++) {
+-            if (cpe->ms_mask[idx] &&
+-                    cpe->ch[0].band_type[idx] < NOISE_BT && cpe->ch[1].band_type[idx] < NOISE_BT) {
+-                for (group = 0; group < ics->group_len[g]; group++) {
+-                    ac->dsp.butterflies_float(ch0 + group * 128 + offsets[i],
+-                                              ch1 + group * 128 + offsets[i],
+-                                              offsets[i+1] - offsets[i]);
+-                }
+-            }
+-        }
+-        ch0 += ics->group_len[g] * 128;
+-        ch1 += ics->group_len[g] * 128;
+-    }
+-}
+-
+-/**
+- * intensity stereo decoding; reference: 4.6.8.2.3
+- *
+- * @param   ms_present  Indicates mid/side stereo presence. [0] mask is all 0s;
+- *                      [1] mask is decoded from bitstream; [2] mask is all 1s;
+- *                      [3] reserved for scalable AAC
+- */
+-static void apply_intensity_stereo(ChannelElement *cpe, int ms_present)
+-{
+-    const IndividualChannelStream *ics = &cpe->ch[1].ics;
+-    SingleChannelElement         *sce1 = &cpe->ch[1];
+-    float *coef0 = cpe->ch[0].coeffs, *coef1 = cpe->ch[1].coeffs;
+-    const uint16_t *offsets = ics->swb_offset;
+-    int g, group, i, k, idx = 0;
+-    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++)
+-                        for (k = offsets[i]; k < offsets[i + 1]; k++)
+-                            coef1[group * 128 + k] = scale * coef0[group * 128 + k];
+-                }
+-            } else {
+-                int bt_run_end = sce1->band_type_run_end[idx];
+-                idx += bt_run_end - i;
+-                i    = bt_run_end;
+-            }
+-        }
+-        coef0 += ics->group_len[g] * 128;
+-        coef1 += ics->group_len[g] * 128;
+-    }
+-}
+-
+-/**
+- * Decode a channel_pair_element; reference: table 4.4.
+- *
+- * @param   elem_id Identifies the instance of a syntax element.
+- *
+- * @return  Returns error status. 0 - OK, !0 - error
+- */
+-static int decode_cpe(AACContext *ac, GetBitContext *gb, ChannelElement *cpe)
+-{
+-    int i, ret, common_window, ms_present = 0;
+-
+-    common_window = get_bits1(gb);
+-    if (common_window) {
+-        if (decode_ics_info(ac, &cpe->ch[0].ics, gb, 1))
+-            return -1;
+-        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;
+-        ms_present = get_bits(gb, 2);
+-        if (ms_present == 3) {
+-            av_log(ac->avccontext, AV_LOG_ERROR, "ms_present = 3 is reserved.\n");
+-            return -1;
+-        } else if (ms_present)
+-            decode_mid_side_stereo(cpe, gb, ms_present);
+-    }
+-    if ((ret = decode_ics(ac, &cpe->ch[0], gb, common_window, 0)))
+-        return ret;
+-    if ((ret = decode_ics(ac, &cpe->ch[1], gb, common_window, 0)))
+-        return ret;
+-
+-    if (common_window) {
+-        if (ms_present)
+-            apply_mid_side_stereo(ac, cpe);
+-        if (ac->m4ac.object_type == AOT_AAC_MAIN) {
+-            apply_prediction(ac, &cpe->ch[0]);
+-            apply_prediction(ac, &cpe->ch[1]);
+-        }
+-    }
+-
+-    apply_intensity_stereo(cpe, ms_present);
+-    return 0;
+-}
+-
+-/**
+- * Decode coupling_channel_element; reference: table 4.8.
+- *
+- * @param   elem_id Identifies the instance of a syntax element.
+- *
+- * @return  Returns error status. 0 - OK, !0 - error
+- */
+-static int decode_cce(AACContext *ac, GetBitContext *gb, ChannelElement *che)
+-{
+-    int num_gain = 0;
+-    int c, g, sfb, ret;
+-    int sign;
+-    float scale;
+-    SingleChannelElement *sce = &che->ch[0];
+-    ChannelCoupling     *coup = &che->coup;
+-
+-    coup->coupling_point = 2 * get_bits1(gb);
+-    coup->num_coupled = get_bits(gb, 3);
+-    for (c = 0; c <= coup->num_coupled; c++) {
+-        num_gain++;
+-        coup->type[c] = get_bits1(gb) ? TYPE_CPE : TYPE_SCE;
+-        coup->id_select[c] = get_bits(gb, 4);
+-        if (coup->type[c] == TYPE_CPE) {
+-            coup->ch_select[c] = get_bits(gb, 2);
+-            if (coup->ch_select[c] == 3)
+-                num_gain++;
+-        } else
+-            coup->ch_select[c] = 2;
+-    }
+-    coup->coupling_point += get_bits1(gb) || (coup->coupling_point >> 1);
+-
+-    sign  = get_bits(gb, 1);
+-    scale = pow(2., pow(2., (int)get_bits(gb, 2) - 3));
+-
+-    if ((ret = decode_ics(ac, sce, gb, 0, 0)))
+-        return ret;
+-
+-    for (c = 0; c < num_gain; c++) {
+-        int idx  = 0;
+-        int cge  = 1;
+-        int gain = 0;
+-        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;
+-            gain_cache = pow(scale, -gain);
+-        }
+-        if (coup->coupling_point == AFTER_IMDCT) {
+-            coup->gain[c][0] = gain_cache;
+-        } else {
+-            for (g = 0; g < sce->ics.num_window_groups; g++) {
+-                for (sfb = 0; sfb < sce->ics.max_sfb; sfb++, idx++) {
+-                    if (sce->band_type[idx] != ZERO_BT) {
+-                        if (!cge) {
+-                            int t = get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
+-                            if (t) {
+-                                int s = 1;
+-                                t = gain += t;
+-                                if (sign) {
+-                                    s  -= 2 * (t & 0x1);
+-                                    t >>= 1;
+-                                }
+-                                gain_cache = pow(scale, -t) * s;
+-                            }
+-                        }
+-                        coup->gain[c][idx] = gain_cache;
+-                    }
+-                }
+-            }
+-        }
+-    }
+-    return 0;
+-}
+-
+-/**
+- * Parse whether channels are to be excluded from Dynamic Range Compression; reference: table 4.53.
+- *
+- * @return  Returns number of bytes consumed.
+- */
+-static int decode_drc_channel_exclusions(DynamicRangeControl *che_drc,
+-                                         GetBitContext *gb)
+-{
+-    int i;
+-    int num_excl_chan = 0;
+-
+-    do {
+-        for (i = 0; i < 7; i++)
+-            che_drc->exclude_mask[num_excl_chan++] = get_bits1(gb);
+-    } while (num_excl_chan < MAX_CHANNELS - 7 && get_bits1(gb));
+-
+-    return num_excl_chan / 7;
+-}
+-
+-/**
+- * Decode dynamic range information; reference: table 4.52.
+- *
+- * @param   cnt length of TYPE_FIL syntactic element in bytes
+- *
+- * @return  Returns number of bytes consumed.
+- */
+-static int decode_dynamic_range(DynamicRangeControl *che_drc,
+-                                GetBitContext *gb, int cnt)
+-{
+-    int n             = 1;
+-    int drc_num_bands = 1;
+-    int i;
+-
+-    /* pce_tag_present? */
+-    if (get_bits1(gb)) {
+-        che_drc->pce_instance_tag  = get_bits(gb, 4);
+-        skip_bits(gb, 4); // tag_reserved_bits
+-        n++;
+-    }
+-
+-    /* excluded_chns_present? */
+-    if (get_bits1(gb)) {
+-        n += decode_drc_channel_exclusions(che_drc, gb);
+-    }
+-
+-    /* drc_bands_present? */
+-    if (get_bits1(gb)) {
+-        che_drc->band_incr            = get_bits(gb, 4);
+-        che_drc->interpolation_scheme = get_bits(gb, 4);
+-        n++;
+-        drc_num_bands += che_drc->band_incr;
+-        for (i = 0; i < drc_num_bands; i++) {
+-            che_drc->band_top[i] = get_bits(gb, 8);
+-            n++;
+-        }
+-    }
+-
+-    /* prog_ref_level_present? */
+-    if (get_bits1(gb)) {
+-        che_drc->prog_ref_level = get_bits(gb, 7);
+-        skip_bits1(gb); // prog_ref_level_reserved_bits
+-        n++;
+-    }
+-
+-    for (i = 0; i < drc_num_bands; i++) {
+-        che_drc->dyn_rng_sgn[i] = get_bits1(gb);
+-        che_drc->dyn_rng_ctl[i] = get_bits(gb, 7);
+-        n++;
+-    }
+-
+-    return n;
+-}
+-
+-/**
+- * Decode extension data (incomplete); reference: table 4.51.
+- *
+- * @param   cnt length of TYPE_FIL syntactic element in bytes
+- *
+- * @return Returns number of bytes consumed
+- */
+-static int decode_extension_payload(AACContext *ac, GetBitContext *gb, int cnt,
+-                                    ChannelElement *che, enum RawDataBlockType elem_type)
+-{
+-    int crc_flag = 0;
+-    int res = cnt;
+-    switch (get_bits(gb, 4)) { // extension type
+-    case EXT_SBR_DATA_CRC:
+-        crc_flag++;
+-    case EXT_SBR_DATA:
+-        if (!che) {
+-            av_log(ac->avccontext, AV_LOG_ERROR, "SBR was found before the first channel element.\n");
+-            return res;
+-        } else if (!ac->m4ac.sbr) {
+-            av_log(ac->avccontext, AV_LOG_ERROR, "SBR signaled to be not-present but was found in the bitstream.\n");
+-            skip_bits_long(gb, 8 * cnt - 4);
+-            return res;
+-        } else if (ac->m4ac.sbr == -1 && ac->output_configured == OC_LOCKED) {
+-            av_log(ac->avccontext, AV_LOG_ERROR, "Implicit SBR was found with a first occurrence after the first frame.\n");
+-            skip_bits_long(gb, 8 * cnt - 4);
+-            return res;
+-        } else {
+-            ac->m4ac.sbr = 1;
+-        }
+-        res = ff_decode_sbr_extension(ac, &che->sbr, gb, crc_flag, cnt, elem_type);
+-        break;
+-    case EXT_DYNAMIC_RANGE:
+-        res = decode_dynamic_range(&ac->che_drc, gb, cnt);
+-        break;
+-    case EXT_FILL:
+-    case EXT_FILL_DATA:
+-    case EXT_DATA_ELEMENT:
+-    default:
+-        skip_bits_long(gb, 8 * cnt - 4);
+-        break;
+-    };
+-    return res;
+-}
+-
+-/**
+- * Decode Temporal Noise Shaping filter coefficients and apply all-pole filters; reference: 4.6.9.3.
+- *
+- * @param   decode  1 if tool is used normally, 0 if tool is used in LTP.
+- * @param   coef    spectral coefficients
+- */
+-static void apply_tns(float coef[1024], TemporalNoiseShaping *tns,
+-                      IndividualChannelStream *ics, int decode)
+-{
+-    const int mmm = FFMIN(ics->tns_max_bands, ics->max_sfb);
+-    int w, filt, m, i;
+-    int bottom, top, order, start, end, size, inc;
+-    float lpc[TNS_MAX_ORDER];
+-
+-    for (w = 0; w < ics->num_windows; w++) {
+-        bottom = ics->num_swb;
+-        for (filt = 0; filt < tns->n_filt[w]; filt++) {
+-            top    = bottom;
+-            bottom = FFMAX(0, top - tns->length[w][filt]);
+-            order  = tns->order[w][filt];
+-            if (order == 0)
+-                continue;
+-
+-            // tns_decode_coef
+-            compute_lpc_coefs(tns->coef[w][filt], order, lpc, 0, 0, 0);
+-
+-            start = ics->swb_offset[FFMIN(bottom, mmm)];
+-            end   = ics->swb_offset[FFMIN(   top, mmm)];
+-            if ((size = end - start) <= 0)
+-                continue;
+-            if (tns->direction[w][filt]) {
+-                inc = -1;
+-                start = end - 1;
+-            } else {
+-                inc = 1;
+-            }
+-            start += w * 128;
+-
+-            // 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];
+-        }
+-    }
+-}
+-
+-/**
+- * Conduct IMDCT and windowing.
+- */
+-static void imdct_and_windowing(AACContext *ac, SingleChannelElement *sce, float bias)
+-{
+-    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;
+-    int i;
+-
+-    // imdct
+-    if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
+-        if (ics->window_sequence[1] == ONLY_LONG_SEQUENCE || ics->window_sequence[1] == LONG_STOP_SEQUENCE)
+-            av_log(ac->avccontext, AV_LOG_WARNING,
+-                   "Transition from an ONLY_LONG or LONG_STOP to an EIGHT_SHORT sequence detected. "
+-                   "If you heard an audible artifact, please submit the sample to the FFmpeg developers.\n");
+-        for (i = 0; i < 1024; i += 128)
+-            ff_imdct_half(&ac->mdct_small, buf + i, in + i);
+-    } else
+-        ff_imdct_half(&ac->mdct, buf, in);
+-
+-    /* window overlapping
+-     * NOTE: To simplify the overlapping code, all 'meaningless' short to long
+-     * and long to short transitions are considered to be short to short
+-     * transitions. This leaves just two cases (long to long and short to short)
+-     * with a little special sauce for EIGHT_SHORT_SEQUENCE.
+-     */
+-    if ((ics->window_sequence[1] == ONLY_LONG_SEQUENCE || ics->window_sequence[1] == LONG_STOP_SEQUENCE) &&
+-            (ics->window_sequence[0] == ONLY_LONG_SEQUENCE || ics->window_sequence[0] == LONG_START_SEQUENCE)) {
+-        ac->dsp.vector_fmul_window(    out,               saved,            buf,         lwindow_prev, bias, 512);
+-    } else {
+-        for (i = 0; i < 448; i++)
+-            out[i] = saved[i] + bias;
+-
+-        if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
+-            ac->dsp.vector_fmul_window(out + 448 + 0*128, saved + 448,      buf + 0*128, swindow_prev, bias, 64);
+-            ac->dsp.vector_fmul_window(out + 448 + 1*128, buf + 0*128 + 64, buf + 1*128, swindow,      bias, 64);
+-            ac->dsp.vector_fmul_window(out + 448 + 2*128, buf + 1*128 + 64, buf + 2*128, swindow,      bias, 64);
+-            ac->dsp.vector_fmul_window(out + 448 + 3*128, buf + 2*128 + 64, buf + 3*128, swindow,      bias, 64);
+-            ac->dsp.vector_fmul_window(temp,              buf + 3*128 + 64, buf + 4*128, swindow,      bias, 64);
+-            memcpy(                    out + 448 + 4*128, temp, 64 * sizeof(float));
+-        } else {
+-            ac->dsp.vector_fmul_window(out + 448,         saved + 448,      buf,         swindow_prev, bias, 64);
+-            for (i = 576; i < 1024; i++)
+-                out[i] = buf[i-512] + bias;
+-        }
+-    }
+-
+-    // buffer update
+-    if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
+-        for (i = 0; i < 64; i++)
+-            saved[i] = temp[64 + i] - bias;
+-        ac->dsp.vector_fmul_window(saved + 64,  buf + 4*128 + 64, buf + 5*128, swindow, 0, 64);
+-        ac->dsp.vector_fmul_window(saved + 192, buf + 5*128 + 64, buf + 6*128, swindow, 0, 64);
+-        ac->dsp.vector_fmul_window(saved + 320, buf + 6*128 + 64, buf + 7*128, swindow, 0, 64);
+-        memcpy(                    saved + 448, buf + 7*128 + 64,  64 * sizeof(float));
+-    } else if (ics->window_sequence[0] == LONG_START_SEQUENCE) {
+-        memcpy(                    saved,       buf + 512,        448 * sizeof(float));
+-        memcpy(                    saved + 448, buf + 7*128 + 64,  64 * sizeof(float));
+-    } else { // LONG_STOP or ONLY_LONG
+-        memcpy(                    saved,       buf + 512,        512 * sizeof(float));
+-    }
+-}
+-
+-/**
+- * Apply dependent channel coupling (applied before IMDCT).
+- *
+- * @param   index   index into coupling gain array
+- */
+-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;
+-    int g, i, group, k, idx = 0;
+-    if (ac->m4ac.object_type == AOT_AAC_LTP) {
+-        av_log(ac->avccontext, AV_LOG_ERROR,
+-               "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++) {
+-            if (cce->ch[0].band_type[idx] != ZERO_BT) {
+-                const float gain = cce->coup.gain[index][idx];
+-                for (group = 0; group < ics->group_len[g]; group++) {
+-                    for (k = offsets[i]; k < offsets[i + 1]; k++) {
+-                        // XXX dsputil-ize
+-                        dest[group * 128 + k] += gain * src[group * 128 + k];
+-                    }
+-                }
+-            }
+-        }
+-        dest += ics->group_len[g] * 128;
+-        src  += ics->group_len[g] * 128;
+-    }
+-}
+-
+-/**
+- * Apply independent channel coupling (applied after IMDCT).
+- *
+- * @param   index   index into coupling gain array
+- */
+-static void apply_independent_coupling(AACContext *ac,
+-                                       SingleChannelElement *target,
+-                                       ChannelElement *cce, int index)
+-{
+-    int i;
+-    const float gain = cce->coup.gain[index][0];
+-    const float bias = ac->add_bias;
+-    const float *src = cce->ch[0].ret;
+-    float *dest = target->ret;
+-    const int len = 1024 << (ac->m4ac.sbr == 1);
+-
+-    for (i = 0; i < len; i++)
+-        dest[i] += gain * (src[i] - bias);
+-}
+-
+-/**
+- * channel coupling transformation interface
+- *
+- * @param   index   index into coupling gain array
+- * @param   apply_coupling_method   pointer to (in)dependent coupling function
+- */
+-static void apply_channel_coupling(AACContext *ac, ChannelElement *cc,
+-                                   enum RawDataBlockType type, int elem_id,
+-                                   enum CouplingPoint coupling_point,
+-                                   void (*apply_coupling_method)(AACContext *ac, SingleChannelElement *target, ChannelElement *cce, int index))
+-{
+-    int i, c;
+-
+-    for (i = 0; i < MAX_ELEM_ID; i++) {
+-        ChannelElement *cce = ac->che[TYPE_CCE][i];
+-        int index = 0;
+-
+-        if (cce && cce->coup.coupling_point == coupling_point) {
+-            ChannelCoupling *coup = &cce->coup;
+-
+-            for (c = 0; c <= coup->num_coupled; c++) {
+-                if (coup->type[c] == type && coup->id_select[c] == elem_id) {
+-                    if (coup->ch_select[c] != 1) {
+-                        apply_coupling_method(ac, &cc->ch[0], cce, index);
+-                        if (coup->ch_select[c] != 0)
+-                            index++;
+-                    }
+-                    if (coup->ch_select[c] != 2)
+-                        apply_coupling_method(ac, &cc->ch[1], cce, index++);
+-                } else
+-                    index += 1 + (coup->ch_select[c] == 3);
+-            }
+-        }
+-    }
+-}
+-
+-/**
+- * Convert spectral data to float samples, applying all supported tools as appropriate.
+- */
+-static void spectral_to_sample(AACContext *ac)
+-{
+-    int i, type;
+-    float imdct_bias = (ac->m4ac.sbr <= 0) ? ac->add_bias : 0.0f;
+-    for (type = 3; type >= 0; type--) {
+-        for (i = 0; i < MAX_ELEM_ID; i++) {
+-            ChannelElement *che = ac->che[type][i];
+-            if (che) {
+-                if (type <= TYPE_CPE)
+-                    apply_channel_coupling(ac, che, type, i, BEFORE_TNS, apply_dependent_coupling);
+-                if (che->ch[0].tns.present)
+-                    apply_tns(che->ch[0].coeffs, &che->ch[0].tns, &che->ch[0].ics, 1);
+-                if (che->ch[1].tns.present)
+-                    apply_tns(che->ch[1].coeffs, &che->ch[1].tns, &che->ch[1].ics, 1);
+-                if (type <= TYPE_CPE)
+-                    apply_channel_coupling(ac, che, type, i, BETWEEN_TNS_AND_IMDCT, apply_dependent_coupling);
+-                if (type != TYPE_CCE || che->coup.coupling_point == AFTER_IMDCT) {
+-                    imdct_and_windowing(ac, &che->ch[0], imdct_bias);
+-                    if (type == TYPE_CPE) {
+-                        imdct_and_windowing(ac, &che->ch[1], imdct_bias);
+-                    }
+-                    if (ac->m4ac.sbr > 0) {
+-                        ff_sbr_apply(ac, &che->sbr, type, che->ch[0].ret, che->ch[1].ret);
+-                    }
+-                }
+-                if (type <= TYPE_CCE)
+-                    apply_channel_coupling(ac, che, type, i, AFTER_IMDCT, apply_independent_coupling);
+-            }
+-        }
+-    }
+-}
+-
+-static int parse_adts_frame_header(AACContext *ac, GetBitContext *gb)
+-{
+-    int size;
+-    AACADTSHeaderInfo hdr_info;
+-
+-    size = ff_aac_parse_header(gb, &hdr_info);
+-    if (size > 0) {
+-        if (ac->output_configured != OC_LOCKED && hdr_info.chan_config) {
+-            enum ChannelPosition new_che_pos[4][MAX_ELEM_ID];
+-            memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]));
+-            ac->m4ac.chan_config = hdr_info.chan_config;
+-            if (set_default_channel_config(ac, new_che_pos, hdr_info.chan_config))
+-                return -7;
+-            if (output_configure(ac, ac->che_pos, new_che_pos, hdr_info.chan_config, OC_TRIAL_FRAME))
+-                return -7;
+-        } else if (ac->output_configured != OC_LOCKED) {
+-            ac->output_configured = OC_NONE;
+-        }
+-        if (ac->output_configured != OC_LOCKED)
+-            ac->m4ac.sbr = -1;
+-        ac->m4ac.sample_rate     = hdr_info.sample_rate;
+-        ac->m4ac.sampling_index  = hdr_info.sampling_index;
+-        ac->m4ac.object_type     = hdr_info.object_type;
+-        if (!ac->avccontext->sample_rate)
+-            ac->avccontext->sample_rate = hdr_info.sample_rate;
+-        if (hdr_info.num_aac_frames == 1) {
+-            if (!hdr_info.crc_absent)
+-                skip_bits(gb, 16);
+-        } else {
+-            av_log_missing_feature(ac->avccontext, "More than one AAC RDB per ADTS frame is", 0);
+-            return -1;
+-        }
+-    }
+-    return size;
+-}
+-
+-static int aac_decode_frame(AVCodecContext *avccontext, void *data,
+-                            int *data_size, AVPacket *avpkt)
+-{
+-    const uint8_t *buf = avpkt->data;
+-    int buf_size = avpkt->size;
+-    AACContext *ac = avccontext->priv_data;
+-    ChannelElement *che = NULL, *che_prev = NULL;
+-    GetBitContext gb;
+-    enum RawDataBlockType elem_type, elem_type_prev = TYPE_END;
+-    int err, elem_id, data_size_tmp;
+-    int buf_consumed;
+-    int samples = 1024, multiplier;
+-    int buf_offset;
+-
+-    init_get_bits(&gb, buf, buf_size * 8);
+-
+-    if (show_bits(&gb, 12) == 0xfff) {
+-        if (parse_adts_frame_header(ac, &gb) < 0) {
+-            av_log(avccontext, AV_LOG_ERROR, "Error decoding AAC frame header.\n");
+-            return -1;
+-        }
+-        if (ac->m4ac.sampling_index > 12) {
+-            av_log(ac->avccontext, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index);
+-            return -1;
+-        }
+-    }
+-
+-    // parse
+-    while ((elem_type = get_bits(&gb, 3)) != TYPE_END) {
+-        elem_id = get_bits(&gb, 4);
+-
+-        if (elem_type < TYPE_DSE && !(che=get_che(ac, elem_type, elem_id))) {
+-            av_log(ac->avccontext, AV_LOG_ERROR, "channel element %d.%d is not allocated\n", elem_type, elem_id);
+-            return -1;
+-        }
+-
+-        switch (elem_type) {
+-
+-        case TYPE_SCE:
+-            err = decode_ics(ac, &che->ch[0], &gb, 0, 0);
+-            break;
+-
+-        case TYPE_CPE:
+-            err = decode_cpe(ac, &gb, che);
+-            break;
+-
+-        case TYPE_CCE:
+-            err = decode_cce(ac, &gb, che);
+-            break;
+-
+-        case TYPE_LFE:
+-            err = decode_ics(ac, &che->ch[0], &gb, 0, 0);
+-            break;
+-
+-        case TYPE_DSE:
+-            err = skip_data_stream_element(ac, &gb);
+-            break;
+-
+-        case TYPE_PCE: {
+-            enum ChannelPosition new_che_pos[4][MAX_ELEM_ID];
+-            memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]));
+-            if ((err = decode_pce(ac, new_che_pos, &gb)))
+-                break;
+-            if (ac->output_configured > OC_TRIAL_PCE)
+-                av_log(avccontext, AV_LOG_ERROR,
+-                       "Not evaluating a further program_config_element as this construct is dubious at best.\n");
+-            else
+-                err = output_configure(ac, ac->che_pos, new_che_pos, 0, OC_TRIAL_PCE);
+-            break;
+-        }
+-
+-        case TYPE_FIL:
+-            if (elem_id == 15)
+-                elem_id += get_bits(&gb, 8) - 1;
+-            if (get_bits_left(&gb) < 8 * elem_id) {
+-                    av_log(avccontext, AV_LOG_ERROR, overread_err);
+-                    return -1;
+-            }
+-            while (elem_id > 0)
+-                elem_id -= decode_extension_payload(ac, &gb, elem_id, che_prev, elem_type_prev);
+-            err = 0; /* FIXME */
+-            break;
+-
+-        default:
+-            err = -1; /* should not happen, but keeps compiler happy */
+-            break;
+-        }
+-
+-        che_prev       = che;
+-        elem_type_prev = elem_type;
+-
+-        if (err)
+-            return err;
+-
+-        if (get_bits_left(&gb) < 3) {
+-            av_log(avccontext, AV_LOG_ERROR, overread_err);
+-            return -1;
+-        }
+-    }
+-
+-    spectral_to_sample(ac);
+-
+-    multiplier = (ac->m4ac.sbr == 1) ? ac->m4ac.ext_sample_rate > ac->m4ac.sample_rate : 0;
+-    samples <<= multiplier;
+-    if (ac->output_configured < OC_LOCKED) {
+-        avccontext->sample_rate = ac->m4ac.sample_rate << multiplier;
+-        avccontext->frame_size = samples;
+-    }
+-
+-    data_size_tmp = samples * avccontext->channels * sizeof(int16_t);
+-    if (*data_size < data_size_tmp) {
+-        av_log(avccontext, AV_LOG_ERROR,
+-               "Output buffer too small (%d) or trying to output too many samples (%d) for this frame.\n",
+-               *data_size, data_size_tmp);
+-        return -1;
+-    }
+-    *data_size = data_size_tmp;
+-
+-    ac->dsp.float_to_int16_interleave(data, (const float **)ac->output_data, samples, avccontext->channels);
+-
+-    if (ac->output_configured)
+-        ac->output_configured = OC_LOCKED;
+-
+-    buf_consumed = (get_bits_count(&gb) + 7) >> 3;
+-    for (buf_offset = buf_consumed; buf_offset < buf_size; buf_offset++)
+-        if (buf[buf_offset])
+-            break;
+-
+-    return buf_size > buf_offset ? buf_consumed : buf_size;
+-}
+-
+-static av_cold int aac_decode_close(AVCodecContext *avccontext)
+-{
+-    AACContext *ac = avccontext->priv_data;
+-    int i, type;
+-
+-    for (i = 0; i < MAX_ELEM_ID; i++) {
+-        for (type = 0; type < 4; type++) {
+-            if (ac->che[type][i])
+-                ff_aac_sbr_ctx_close(&ac->che[type][i]->sbr);
+-            av_freep(&ac->che[type][i]);
+-        }
+-    }
+-
+-    ff_mdct_end(&ac->mdct);
+-    ff_mdct_end(&ac->mdct_small);
+-    return 0;
+-}
+-
+-AVCodec aac_decoder = {
+-    "aac",
+-    AVMEDIA_TYPE_AUDIO,
+-    CODEC_ID_AAC,
+-    sizeof(AACContext),
+-    aac_decode_init,
+-    NULL,
+-    aac_decode_close,
+-    aac_decode_frame,
+-    .long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"),
+-    .sample_fmts = (const enum SampleFormat[]) {
+-        SAMPLE_FMT_S16,SAMPLE_FMT_NONE
+-    },
+-    .channel_layouts = aac_channel_layout,
+-};
+--- a/libavcodec/aacenc.c
++++ b/libavcodec/aacenc.c
+@@ -201,13 +201,11 @@ static av_cold int aac_encode_init(AVCod
+     lengths[1] = ff_aac_num_swb_128[i];
+     ff_psy_init(&s->psy, avctx, 2, sizes, lengths);
+     s->psypp = ff_psy_preprocess_init(avctx);
+-    s->coder = &ff_aac_coders[0];
++    s->coder = &ff_aac_coders[2];
+ 
+     s->lambda = avctx->global_quality ? avctx->global_quality : 120;
+-#if !CONFIG_HARDCODED_TABLES
+-    for (i = 0; i < 428; i++)
+-        ff_aac_pow2sf_tab[i] = pow(2, (i - 200)/4.);
+-#endif /* CONFIG_HARDCODED_TABLES */
++
++    ff_aac_tableinit();
+ 
+     if (avctx->channels > 5)
+         av_log(avctx, AV_LOG_ERROR, "This encoder does not yet enforce the restrictions on LFEs. "
+@@ -234,25 +232,21 @@ static void apply_window_and_mdct(AVCode
+                 s->output[i] = sce->saved[i];
+         }
+         if (sce->ics.window_sequence[0] != LONG_START_SEQUENCE) {
+-            j = channel;
+-            for (i = 0; i < 1024; i++, j += avctx->channels) {
++            for (i = 0, j = channel; i < 1024; i++, j += avctx->channels) {
+                 s->output[i+1024]         = audio[j] * lwindow[1024 - i - 1];
+                 sce->saved[i] = audio[j] * lwindow[i];
+             }
+         } else {
+-            j = channel;
+-            for (i = 0; i < 448; i++, j += avctx->channels)
++            for (i = 0, j = channel; i < 448; i++, j += avctx->channels)
+                 s->output[i+1024]         = audio[j];
+-            for (i = 448; i < 576; i++, j += avctx->channels)
++            for (; i < 576; i++, j += avctx->channels)
+                 s->output[i+1024]         = audio[j] * swindow[576 - i - 1];
+             memset(s->output+1024+576, 0, sizeof(s->output[0]) * 448);
+-            j = channel;
+-            for (i = 0; i < 1024; i++, j += avctx->channels)
++            for (i = 0, j = channel; i < 1024; i++, j += avctx->channels)
+                 sce->saved[i] = audio[j];
+         }
+         ff_mdct_calc(&s->mdct1024, sce->coeffs, s->output);
+     } else {
+-        j = channel;
+         for (k = 0; k < 1024; k += 128) {
+             for (i = 448 + k; i < 448 + k + 256; i++)
+                 s->output[i - 448 - k] = (i < 1024)
+@@ -262,8 +256,7 @@ static void apply_window_and_mdct(AVCode
+             s->dsp.vector_fmul_reverse(s->output+128, s->output+128, swindow, 128);
+             ff_mdct_calc(&s->mdct128, sce->coeffs + k, s->output);
+         }
+-        j = channel;
+-        for (i = 0; i < 1024; i++, j += avctx->channels)
++        for (i = 0, j = channel; i < 1024; i++, j += avctx->channels)
+             sce->saved[i] = audio[j];
+     }
+ }
+@@ -562,6 +555,7 @@ static int aac_encode_frame(AVCodecConte
+             cpe      = &s->cpe[i];
+             for (j = 0; j < chans; j++) {
+                 s->cur_channel = start_ch + j;
++                ff_psy_set_band_info(&s->psy, s->cur_channel, cpe->ch[j].coeffs, &wi[j]);
+                 s->coder->search_for_quantizers(avctx, s, &cpe->ch[j], s->lambda);
+             }
+             cpe->common_window = 0;
+@@ -592,7 +586,6 @@ static int aac_encode_frame(AVCodecConte
+             }
+             for (j = 0; j < chans; j++) {
+                 s->cur_channel = start_ch + j;
+-                ff_psy_set_band_info(&s->psy, s->cur_channel, cpe->ch[j].coeffs, &wi[j]);
+                 encode_individual_channel(avctx, s, &cpe->ch[j], cpe->common_window);
+             }
+             start_ch += chans;
+--- a/libavcodec/aacenc.h
++++ b/libavcodec/aacenc.h
+@@ -64,7 +64,7 @@ typedef struct AACEncContext {
+     int cur_channel;
+     int last_frame;
+     float lambda;
+-    DECLARE_ALIGNED(16, int,   qcoefs)[96][2];   ///< quantized coefficients
++    DECLARE_ALIGNED(16, int,   qcoefs)[96];      ///< quantized coefficients
+     DECLARE_ALIGNED(16, float, scoefs)[1024];    ///< scaled coefficients
+ } AACEncContext;
+ 
+--- /dev/null
++++ b/libavcodec/aacdec.c
+@@ -0,0 +1,2142 @@
++/*
++ * AAC decoder
++ * Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org )
++ * Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com )
++ *
++ * This file is part of FFmpeg.
++ *
++ * FFmpeg is free software; you can redistribute it and/or
++ * modify it under the terms of the GNU Lesser General Public
++ * License as published by the Free Software Foundation; either
++ * version 2.1 of the License, or (at your option) any later version.
++ *
++ * FFmpeg is distributed in the hope that it will be useful,
++ * but WITHOUT ANY WARRANTY; without even the implied warranty of
++ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
++ * Lesser General Public License for more details.
++ *
++ * You should have received a copy of the GNU Lesser General Public
++ * License along with FFmpeg; if not, write to the Free Software
++ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
++ */
++
++/**
++ * @file
++ * AAC decoder
++ * @author Oded Shimon  ( ods15 ods15 dyndns org )
++ * @author Maxim Gavrilov ( maxim.gavrilov gmail com )
++ */
++
++/*
++ * 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
++ * N (code in SoC repo) Long Term Prediction
++ * Y                    intensity stereo
++ * Y                    channel coupling
++ * Y                    frequency domain prediction
++ * Y                    Perceptual Noise Substitution
++ * Y                    Mid/Side stereo
++ * N                    Scalable Inverse AAC Quantization
++ * N                    Frequency Selective Switch
++ * N                    upsampling filter
++ * Y                    quantization & coding - AAC
++ * N                    quantization & coding - TwinVQ
++ * N                    quantization & coding - BSAC
++ * N                    AAC Error Resilience tools
++ * N                    Error Resilience payload syntax
++ * N                    Error Protection tool
++ * N                    CELP
++ * N                    Silence Compression
++ * N                    HVXC
++ * N                    HVXC 4kbits/s VR
++ * N                    Structured Audio tools
++ * N                    Structured Audio Sample Bank Format
++ * N                    MIDI
++ * N                    Harmonic and Individual Lines plus Noise
++ * N                    Text-To-Speech Interface
++ * Y                    Spectral Band Replication
++ * Y (not in this code) Layer-1
++ * Y (not in this code) Layer-2
++ * Y (not in this code) Layer-3
++ * N                    SinuSoidal Coding (Transient, Sinusoid, Noise)
++ * Y                    Parametric Stereo
++ * N                    Direct Stream Transfer
++ *
++ * Note: - HE AAC v1 comprises LC AAC with Spectral Band Replication.
++ *       - HE AAC v2 comprises LC AAC with Spectral Band Replication and
++           Parametric Stereo.
++ */
++
++
++#include "avcodec.h"
++#include "internal.h"
++#include "get_bits.h"
++#include "dsputil.h"
++#include "fft.h"
++#include "lpc.h"
++
++#include "aac.h"
++#include "aactab.h"
++#include "aacdectab.h"
++#include "cbrt_tablegen.h"
++#include "sbr.h"
++#include "aacsbr.h"
++#include "mpeg4audio.h"
++#include "aac_parser.h"
++
++#include <assert.h>
++#include <errno.h>
++#include <math.h>
++#include <string.h>
++
++#if ARCH_ARM
++#   include "arm/aac.h"
++#endif
++
++union float754 {
++    float f;
++    uint32_t i;
++};
++
++static VLC vlc_scalefactors;
++static VLC vlc_spectral[11];
++
++static const char overread_err[] = "Input buffer exhausted before END element found\n";
++
++static ChannelElement *get_che(AACContext *ac, int type, int elem_id)
++{
++    /* Some buggy encoders appear to set all elem_ids to zero and rely on
++    channels always occurring in the same order. This is expressly forbidden
++    by the spec but we will try to work around it.
++    */
++    int err_printed = 0;
++    while (ac->tags_seen_this_frame[type][elem_id] && elem_id < MAX_ELEM_ID) {
++        if (ac->output_configured < OC_LOCKED && !err_printed) {
++            av_log(ac->avctx, AV_LOG_WARNING, "Duplicate channel tag found, attempting to remap.\n");
++            err_printed = 1;
++        }
++        elem_id++;
++    }
++    if (elem_id == MAX_ELEM_ID)
++        return NULL;
++    ac->tags_seen_this_frame[type][elem_id] = 1;
++
++    if (ac->tag_che_map[type][elem_id]) {
++        return ac->tag_che_map[type][elem_id];
++    }
++    if (ac->tags_mapped >= tags_per_config[ac->m4ac.chan_config]) {
++        return NULL;
++    }
++    switch (ac->m4ac.chan_config) {
++    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 */
++        if (ac->tags_mapped == tags_per_config[ac->m4ac.chan_config] - 1 && (type == TYPE_LFE || type == TYPE_SCE)) {
++            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:
++        if (ac->tags_mapped == 2 && ac->m4ac.chan_config == 4 && type == TYPE_SCE) {
++            ac->tags_mapped++;
++            return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][1];
++        }
++    case 3:
++    case 2:
++        if (ac->tags_mapped == (ac->m4ac.chan_config != 2) && type == TYPE_CPE) {
++            ac->tags_mapped++;
++            return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][0];
++        } else if (ac->m4ac.chan_config == 2) {
++            return NULL;
++        }
++    case 1:
++        if (!ac->tags_mapped && type == TYPE_SCE) {
++            ac->tags_mapped++;
++            return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][0];
++        }
++    default:
++        return NULL;
++    }
++}
++
++/**
++ * Check for the channel element in the current channel position configuration.
++ * If it exists, make sure the appropriate element is allocated and map the
++ * channel order to match the internal FFmpeg channel layout.
++ *
++ * @param   che_pos current channel position configuration
++ * @param   type channel element type
++ * @param   id channel element id
++ * @param   channels count of the number of channels in the configuration
++ *
++ * @return  Returns error status. 0 - OK, !0 - error
++ */
++static av_cold int che_configure(AACContext *ac,
++                         enum ChannelPosition che_pos[4][MAX_ELEM_ID],
++                         int type, int id,
++                         int *channels)
++{
++    if (che_pos[type][id]) {
++        if (!ac->che[type][id] && !(ac->che[type][id] = av_mallocz(sizeof(ChannelElement))))
++            return AVERROR(ENOMEM);
++        ff_aac_sbr_ctx_init(&ac->che[type][id]->sbr);
++        if (type != TYPE_CCE) {
++            ac->output_data[(*channels)++] = ac->che[type][id]->ch[0].ret;
++            if (type == TYPE_CPE ||
++                (type == TYPE_SCE && ac->m4ac.ps == 1)) {
++                ac->output_data[(*channels)++] = ac->che[type][id]->ch[1].ret;
++            }
++        }
++    } else {
++        if (ac->che[type][id])
++            ff_aac_sbr_ctx_close(&ac->che[type][id]->sbr);
++        av_freep(&ac->che[type][id]);
++    }
++    return 0;
++}
++
++/**
++ * Configure output channel order based on the current program configuration element.
++ *
++ * @param   che_pos current channel position configuration
++ * @param   new_che_pos New channel position configuration - we only do something if it differs from the current one.
++ *
++ * @return  Returns error status. 0 - OK, !0 - error
++ */
++static av_cold int output_configure(AACContext *ac,
++                            enum ChannelPosition che_pos[4][MAX_ELEM_ID],
++                            enum ChannelPosition new_che_pos[4][MAX_ELEM_ID],
++                            int channel_config, enum OCStatus oc_type)
++{
++    AVCodecContext *avctx = ac->avctx;
++    int i, type, channels = 0, ret;
++
++    if (new_che_pos != che_pos)
++    memcpy(che_pos, new_che_pos, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]));
++
++    if (channel_config) {
++        for (i = 0; i < tags_per_config[channel_config]; i++) {
++            if ((ret = che_configure(ac, che_pos,
++                                     aac_channel_layout_map[channel_config - 1][i][0],
++                                     aac_channel_layout_map[channel_config - 1][i][1],
++                                     &channels)))
++                return ret;
++        }
++
++        memset(ac->tag_che_map, 0,       4 * MAX_ELEM_ID * sizeof(ac->che[0][0]));
++        ac->tags_mapped = 0;
++
++        avctx->channel_layout = aac_channel_layout[channel_config - 1];
++    } else {
++        /* Allocate or free elements depending on if they are in the
++         * current program configuration.
++         *
++         * Set up default 1:1 output mapping.
++         *
++         * For a 5.1 stream the output order will be:
++         *    [ Center ] [ Front Left ] [ Front Right ] [ LFE ] [ Surround Left ] [ Surround Right ]
++         */
++
++        for (i = 0; i < MAX_ELEM_ID; i++) {
++            for (type = 0; type < 4; type++) {
++                if ((ret = che_configure(ac, che_pos, type, i, &channels)))
++                    return ret;
++            }
++        }
++
++        memcpy(ac->tag_che_map, ac->che, 4 * MAX_ELEM_ID * sizeof(ac->che[0][0]));
++        ac->tags_mapped = 4 * MAX_ELEM_ID;
++
++        avctx->channel_layout = 0;
++    }
++
++    avctx->channels = channels;
++
++    ac->output_configured = oc_type;
++
++    return 0;
++}
++
++/**
++ * Decode an array of 4 bit element IDs, optionally interleaved with a stereo/mono switching bit.
++ *
++ * @param cpe_map Stereo (Channel Pair Element) map, NULL if stereo bit is not present.
++ * @param sce_map mono (Single Channel Element) map
++ * @param type speaker type/position for these channels
++ */
++static void decode_channel_map(enum ChannelPosition *cpe_map,
++                               enum ChannelPosition *sce_map,
++                               enum ChannelPosition type,
++                               GetBitContext *gb, int n)
++{
++    while (n--) {
++        enum ChannelPosition *map = cpe_map && get_bits1(gb) ? cpe_map : sce_map; // stereo or mono map
++        map[get_bits(gb, 4)] = type;
++    }
++}
++
++/**
++ * Decode program configuration element; reference: table 4.2.
++ *
++ * @param   new_che_pos New channel position configuration - we only do something if it differs from the current one.
++ *
++ * @return  Returns error status. 0 - OK, !0 - error
++ */
++static int decode_pce(AACContext *ac, enum ChannelPosition new_che_pos[4][MAX_ELEM_ID],
++                      GetBitContext *gb)
++{
++    int num_front, num_side, num_back, num_lfe, num_assoc_data, num_cc, sampling_index;
++    int comment_len;
++
++    skip_bits(gb, 2);  // object_type
++
++    sampling_index = get_bits(gb, 4);
++    if (ac->m4ac.sampling_index != sampling_index)
++        av_log(ac->avctx, AV_LOG_WARNING, "Sample rate index in program config element does not match the sample rate index configured by the container.\n");
++
++    num_front       = get_bits(gb, 4);
++    num_side        = get_bits(gb, 4);
++    num_back        = get_bits(gb, 4);
++    num_lfe         = get_bits(gb, 2);
++    num_assoc_data  = get_bits(gb, 3);
++    num_cc          = get_bits(gb, 4);
++
++    if (get_bits1(gb))
++        skip_bits(gb, 4); // mono_mixdown_tag
++    if (get_bits1(gb))
++        skip_bits(gb, 4); // stereo_mixdown_tag
++
++    if (get_bits1(gb))
++        skip_bits(gb, 3); // mixdown_coeff_index and pseudo_surround
++
++    decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_FRONT, gb, num_front);
++    decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_SIDE,  gb, num_side );
++    decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_BACK,  gb, num_back );
++    decode_channel_map(NULL,                  new_che_pos[TYPE_LFE], AAC_CHANNEL_LFE,   gb, num_lfe  );
++
++    skip_bits_long(gb, 4 * num_assoc_data);
++
++    decode_channel_map(new_che_pos[TYPE_CCE], new_che_pos[TYPE_CCE], AAC_CHANNEL_CC,    gb, num_cc   );
++
++    align_get_bits(gb);
++
++    /* comment field, first byte is length */
++    comment_len = get_bits(gb, 8) * 8;
++    if (get_bits_left(gb) < comment_len) {
++        av_log(ac->avctx, AV_LOG_ERROR, overread_err);
++        return -1;
++    }
++    skip_bits_long(gb, comment_len);
++    return 0;
++}
++
++/**
++ * Set up channel positions based on a default channel configuration
++ * as specified in table 1.17.
++ *
++ * @param   new_che_pos New channel position configuration - we only do something if it differs from the current one.
++ *
++ * @return  Returns error status. 0 - OK, !0 - error
++ */
++static av_cold int set_default_channel_config(AACContext *ac,
++                                      enum ChannelPosition new_che_pos[4][MAX_ELEM_ID],
++                                      int channel_config)
++{
++    if (channel_config < 1 || channel_config > 7) {
++        av_log(ac->avctx, AV_LOG_ERROR, "invalid default channel configuration (%d)\n",
++               channel_config);
++        return -1;
++    }
++
++    /* default channel configurations:
++     *
++     * 1ch : front center (mono)
++     * 2ch : L + R (stereo)
++     * 3ch : front center + L + R
++     * 4ch : front center + L + R + back center
++     * 5ch : front center + L + R + back stereo
++     * 6ch : front center + L + R + back stereo + LFE
++     * 7ch : front center + L + R + outer front left + outer front right + back stereo + LFE
++     */
++
++    if (channel_config != 2)
++        new_che_pos[TYPE_SCE][0] = AAC_CHANNEL_FRONT; // front center (or mono)
++    if (channel_config > 1)
++        new_che_pos[TYPE_CPE][0] = AAC_CHANNEL_FRONT; // L + R (or stereo)
++    if (channel_config == 4)
++        new_che_pos[TYPE_SCE][1] = AAC_CHANNEL_BACK;  // back center
++    if (channel_config > 4)
++        new_che_pos[TYPE_CPE][(channel_config == 7) + 1]
++        = AAC_CHANNEL_BACK;  // back stereo
++    if (channel_config > 5)
++        new_che_pos[TYPE_LFE][0] = AAC_CHANNEL_LFE;   // LFE
++    if (channel_config == 7)
++        new_che_pos[TYPE_CPE][1] = AAC_CHANNEL_FRONT; // outer front left + outer front right
++
++    return 0;
++}
++
++/**
++ * Decode GA "General Audio" specific configuration; reference: table 4.1.
++ *
++ * @return  Returns error status. 0 - OK, !0 - error
++ */
++static int decode_ga_specific_config(AACContext *ac, GetBitContext *gb,
++                                     int channel_config)
++{
++    enum ChannelPosition new_che_pos[4][MAX_ELEM_ID];
++    int extension_flag, ret;
++
++    if (get_bits1(gb)) { // frameLengthFlag
++        av_log_missing_feature(ac->avctx, "960/120 MDCT window is", 1);
++        return -1;
++    }
++
++    if (get_bits1(gb))       // dependsOnCoreCoder
++        skip_bits(gb, 14);   // coreCoderDelay
++    extension_flag = get_bits1(gb);
++
++    if (ac->m4ac.object_type == AOT_AAC_SCALABLE ||
++        ac->m4ac.object_type == AOT_ER_AAC_SCALABLE)
++        skip_bits(gb, 3);     // layerNr
++
++    memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]));
++    if (channel_config == 0) {
++        skip_bits(gb, 4);  // element_instance_tag
++        if ((ret = decode_pce(ac, new_che_pos, gb)))
++            return ret;
++    } else {
++        if ((ret = set_default_channel_config(ac, new_che_pos, channel_config)))
++            return ret;
++    }
++    if ((ret = output_configure(ac, ac->che_pos, new_che_pos, channel_config, OC_GLOBAL_HDR)))
++        return ret;
++
++    if (extension_flag) {
++        switch (ac->m4ac.object_type) {
++        case AOT_ER_BSAC:
++            skip_bits(gb, 5);    // numOfSubFrame
++            skip_bits(gb, 11);   // layer_length
++            break;
++        case AOT_ER_AAC_LC:
++        case AOT_ER_AAC_LTP:
++        case AOT_ER_AAC_SCALABLE:
++        case AOT_ER_AAC_LD:
++            skip_bits(gb, 3);  /* aacSectionDataResilienceFlag
++                                    * aacScalefactorDataResilienceFlag
++                                    * aacSpectralDataResilienceFlag
++                                    */
++            break;
++        }
++        skip_bits1(gb);    // extensionFlag3 (TBD in version 3)
++    }
++    return 0;
++}
++
++/**
++ * Decode audio specific configuration; reference: table 1.13.
++ *
++ * @param   data        pointer to AVCodecContext extradata
++ * @param   data_size   size of AVCCodecContext extradata
++ *
++ * @return  Returns error status. 0 - OK, !0 - error
++ */
++static int decode_audio_specific_config(AACContext *ac, void *data,
++                                        int data_size)
++{
++    GetBitContext gb;
++    int i;
++
++    init_get_bits(&gb, data, data_size * 8);
++
++    if ((i = ff_mpeg4audio_get_config(&ac->m4ac, data, data_size)) < 0)
++        return -1;
++    if (ac->m4ac.sampling_index > 12) {
++        av_log(ac->avctx, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index);
++        return -1;
++    }
++    if (ac->m4ac.sbr == 1 && ac->m4ac.ps == -1)
++        ac->m4ac.ps = 1;
++
++    skip_bits_long(&gb, i);
++
++    switch (ac->m4ac.object_type) {
++    case AOT_AAC_MAIN:
++    case AOT_AAC_LC:
++        if (decode_ga_specific_config(ac, &gb, ac->m4ac.chan_config))
++            return -1;
++        break;
++    default:
++        av_log(ac->avctx, AV_LOG_ERROR, "Audio object type %s%d is not supported.\n",
++               ac->m4ac.sbr == 1? "SBR+" : "", ac->m4ac.object_type);
++        return -1;
++    }
++    return 0;
++}
++
++/**
++ * linear congruential pseudorandom number generator
++ *
++ * @param   previous_val    pointer to the current state of the generator
++ *
++ * @return  Returns a 32-bit pseudorandom integer
++ */
++static av_always_inline int lcg_random(int previous_val)
++{
++    return previous_val * 1664525 + 1013904223;
++}
++
++static av_always_inline void reset_predict_state(PredictorState *ps)
++{
++    ps->r0   = 0.0f;
++    ps->r1   = 0.0f;
++    ps->cor0 = 0.0f;
++    ps->cor1 = 0.0f;
++    ps->var0 = 1.0f;
++    ps->var1 = 1.0f;
++}
++
++static void reset_all_predictors(PredictorState *ps)
++{
++    int i;
++    for (i = 0; i < MAX_PREDICTORS; i++)
++        reset_predict_state(&ps[i]);
++}
++
++static void reset_predictor_group(PredictorState *ps, int group_num)
++{
++    int i;
++    for (i = group_num - 1; i < MAX_PREDICTORS; i += 30)
++        reset_predict_state(&ps[i]);
++}
++
++#define AAC_INIT_VLC_STATIC(num, size) \
++    INIT_VLC_STATIC(&vlc_spectral[num], 8, ff_aac_spectral_sizes[num], \
++         ff_aac_spectral_bits[num], sizeof( ff_aac_spectral_bits[num][0]), sizeof( ff_aac_spectral_bits[num][0]), \
++        ff_aac_spectral_codes[num], sizeof(ff_aac_spectral_codes[num][0]), sizeof(ff_aac_spectral_codes[num][0]), \
++        size);
++
++static av_cold int aac_decode_init(AVCodecContext *avctx)
++{
++    AACContext *ac = avctx->priv_data;
++
++    ac->avctx = avctx;
++    ac->m4ac.sample_rate = avctx->sample_rate;
++
++    if (avctx->extradata_size > 0) {
++        if (decode_audio_specific_config(ac, avctx->extradata, avctx->extradata_size))
++            return -1;
++    }
++
++    avctx->sample_fmt = SAMPLE_FMT_S16;
++
++    AAC_INIT_VLC_STATIC( 0, 304);
++    AAC_INIT_VLC_STATIC( 1, 270);
++    AAC_INIT_VLC_STATIC( 2, 550);
++    AAC_INIT_VLC_STATIC( 3, 300);
++    AAC_INIT_VLC_STATIC( 4, 328);
++    AAC_INIT_VLC_STATIC( 5, 294);
++    AAC_INIT_VLC_STATIC( 6, 306);
++    AAC_INIT_VLC_STATIC( 7, 268);
++    AAC_INIT_VLC_STATIC( 8, 510);
++    AAC_INIT_VLC_STATIC( 9, 366);
++    AAC_INIT_VLC_STATIC(10, 462);
++
++    ff_aac_sbr_init();
++
++    dsputil_init(&ac->dsp, avctx);
++
++    ac->random_state = 0x1f2e3d4c;
++
++    // -1024 - Compensate wrong IMDCT method.
++    // 32768 - Required to scale values to the correct range for the bias method
++    //         for float to int16 conversion.
++
++    if (ac->dsp.float_to_int16_interleave == ff_float_to_int16_interleave_c) {
++        ac->add_bias  = 385.0f;
++        ac->sf_scale  = 1. / (-1024. * 32768.);
++        ac->sf_offset = 0;
++    } else {
++        ac->add_bias  = 0.0f;
++        ac->sf_scale  = 1. / -1024.;
++        ac->sf_offset = 60;
++    }
++
++    ff_aac_tableinit();
++
++    INIT_VLC_STATIC(&vlc_scalefactors,7,FF_ARRAY_ELEMS(ff_aac_scalefactor_code),
++                    ff_aac_scalefactor_bits, sizeof(ff_aac_scalefactor_bits[0]), sizeof(ff_aac_scalefactor_bits[0]),
++                    ff_aac_scalefactor_code, sizeof(ff_aac_scalefactor_code[0]), sizeof(ff_aac_scalefactor_code[0]),
++                    352);
++
++    ff_mdct_init(&ac->mdct, 11, 1, 1.0);
++    ff_mdct_init(&ac->mdct_small, 8, 1, 1.0);
++    // 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);
++    ff_init_ff_sine_windows(10);
++    ff_init_ff_sine_windows( 7);
++
++    cbrt_tableinit();
++
++    return 0;
++}
++
++/**
++ * Skip data_stream_element; reference: table 4.10.
++ */
++static int skip_data_stream_element(AACContext *ac, GetBitContext *gb)
++{
++    int byte_align = get_bits1(gb);
++    int count = get_bits(gb, 8);
++    if (count == 255)
++        count += get_bits(gb, 8);
++    if (byte_align)
++        align_get_bits(gb);
++
++    if (get_bits_left(gb) < 8 * count) {
++        av_log(ac->avctx, AV_LOG_ERROR, overread_err);
++        return -1;
++    }
++    skip_bits_long(gb, 8 * count);
++    return 0;
++}
++
++static int decode_prediction(AACContext *ac, IndividualChannelStream *ics,
++                             GetBitContext *gb)
++{
++    int sfb;
++    if (get_bits1(gb)) {
++        ics->predictor_reset_group = get_bits(gb, 5);
++        if (ics->predictor_reset_group == 0 || ics->predictor_reset_group > 30) {
++            av_log(ac->avctx, AV_LOG_ERROR, "Invalid Predictor Reset Group.\n");
++            return -1;
++        }
++    }
++    for (sfb = 0; sfb < FFMIN(ics->max_sfb, ff_aac_pred_sfb_max[ac->m4ac.sampling_index]); sfb++) {
++        ics->prediction_used[sfb] = get_bits1(gb);
++    }
++    return 0;
++}
++
++/**
++ * Decode Individual Channel Stream info; reference: table 4.6.
++ *
++ * @param   common_window   Channels have independent [0], or shared [1], Individual Channel Stream information.
++ */
++static int decode_ics_info(AACContext *ac, IndividualChannelStream *ics,
++                           GetBitContext *gb, int common_window)
++{
++    if (get_bits1(gb)) {
++        av_log(ac->avctx, AV_LOG_ERROR, "Reserved bit set.\n");
++        memset(ics, 0, sizeof(IndividualChannelStream));
++        return -1;
++    }
++    ics->window_sequence[1] = ics->window_sequence[0];
++    ics->window_sequence[0] = get_bits(gb, 2);
++    ics->use_kb_window[1]   = ics->use_kb_window[0];
++    ics->use_kb_window[0]   = get_bits1(gb);
++    ics->num_window_groups  = 1;
++    ics->group_len[0]       = 1;
++    if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
++        int i;
++        ics->max_sfb = get_bits(gb, 4);
++        for (i = 0; i < 7; i++) {
++            if (get_bits1(gb)) {
++                ics->group_len[ics->num_window_groups - 1]++;
++            } else {
++                ics->num_window_groups++;
++                ics->group_len[ics->num_window_groups - 1] = 1;
++            }
++        }
++        ics->num_windows       = 8;
++        ics->swb_offset        =    ff_swb_offset_128[ac->m4ac.sampling_index];
++        ics->num_swb           =   ff_aac_num_swb_128[ac->m4ac.sampling_index];
++        ics->tns_max_bands     = ff_tns_max_bands_128[ac->m4ac.sampling_index];
++        ics->predictor_present = 0;
++    } else {
++        ics->max_sfb               = get_bits(gb, 6);
++        ics->num_windows           = 1;
++        ics->swb_offset            =    ff_swb_offset_1024[ac->m4ac.sampling_index];
++        ics->num_swb               =   ff_aac_num_swb_1024[ac->m4ac.sampling_index];
++        ics->tns_max_bands         = ff_tns_max_bands_1024[ac->m4ac.sampling_index];
++        ics->predictor_present     = get_bits1(gb);
++        ics->predictor_reset_group = 0;
++        if (ics->predictor_present) {
++            if (ac->m4ac.object_type == AOT_AAC_MAIN) {
++                if (decode_prediction(ac, ics, gb)) {
++                    memset(ics, 0, sizeof(IndividualChannelStream));
++                    return -1;
++                }
++            } else if (ac->m4ac.object_type == AOT_AAC_LC) {
++                av_log(ac->avctx, AV_LOG_ERROR, "Prediction is not allowed in AAC-LC.\n");
++                memset(ics, 0, sizeof(IndividualChannelStream));
++                return -1;
++            } else {
++                av_log_missing_feature(ac->avctx, "Predictor bit set but LTP is", 1);
++                memset(ics, 0, sizeof(IndividualChannelStream));
++                return -1;
++            }
++        }
++    }
++
++    if (ics->max_sfb > ics->num_swb) {
++        av_log(ac->avctx, AV_LOG_ERROR,
++               "Number of scalefactor bands in group (%d) exceeds limit (%d).\n",
++               ics->max_sfb, ics->num_swb);
++        memset(ics, 0, sizeof(IndividualChannelStream));
++        return -1;
++    }
++
++    return 0;
++}
++
++/**
++ * Decode band types (section_data payload); reference: table 4.46.
++ *
++ * @param   band_type           array of the used band type
++ * @param   band_type_run_end   array of the last scalefactor band of a band type run
++ *
++ * @return  Returns error status. 0 - OK, !0 - error
++ */
++static int decode_band_types(AACContext *ac, enum BandType band_type[120],
++                             int band_type_run_end[120], GetBitContext *gb,
++                             IndividualChannelStream *ics)
++{
++    int g, idx = 0;
++    const int bits = (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) ? 3 : 5;
++    for (g = 0; g < ics->num_window_groups; g++) {
++        int k = 0;
++        while (k < ics->max_sfb) {
++            uint8_t sect_end = k;
++            int sect_len_incr;
++            int sect_band_type = get_bits(gb, 4);
++            if (sect_band_type == 12) {
++                av_log(ac->avctx, AV_LOG_ERROR, "invalid band type\n");
++                return -1;
++            }
++            while ((sect_len_incr = get_bits(gb, bits)) == (1 << bits) - 1)
++                sect_end += sect_len_incr;
++            sect_end += sect_len_incr;
++            if (get_bits_left(gb) < 0) {
++                av_log(ac->avctx, AV_LOG_ERROR, overread_err);
++                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;
++            }
++            for (; k < sect_end; k++) {
++                band_type        [idx]   = sect_band_type;
++                band_type_run_end[idx++] = sect_end;
++            }
++        }
++    }
++    return 0;
++}
++
++/**
++ * Decode scalefactors; reference: table 4.47.
++ *
++ * @param   global_gain         first scalefactor value as scalefactors are differentially coded
++ * @param   band_type           array of the used band type
++ * @param   band_type_run_end   array of the last scalefactor band of a band type run
++ * @param   sf                  array of scalefactors or intensity stereo positions
++ *
++ * @return  Returns error status. 0 - OK, !0 - error
++ */
++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])
++{
++    const int sf_offset = ac->sf_offset + (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE ? 12 : 0);
++    int g, i, idx = 0;
++    int offset[3] = { global_gain, global_gain - 90, 100 };
++    int noise_flag = 1;
++    static const char *sf_str[3] = { "Global gain", "Noise gain", "Intensity stereo position" };
++    for (g = 0; g < ics->num_window_groups; g++) {
++        for (i = 0; i < ics->max_sfb;) {
++            int run_end = band_type_run_end[idx];
++            if (band_type[idx] == ZERO_BT) {
++                for (; i < run_end; i++, idx++)
++                    sf[idx] = 0.;
++            } else if ((band_type[idx] == INTENSITY_BT) || (band_type[idx] == INTENSITY_BT2)) {
++                for (; i < run_end; i++, idx++) {
++                    offset[2] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
++                    if (offset[2] > 255U) {
++                        av_log(ac->avctx, AV_LOG_ERROR,
++                               "%s (%d) out of range.\n", sf_str[2], offset[2]);
++                        return -1;
++                    }
++                    sf[idx] = ff_aac_pow2sf_tab[-offset[2] + 300];
++                }
++            } else if (band_type[idx] == NOISE_BT) {
++                for (; i < run_end; i++, idx++) {
++                    if (noise_flag-- > 0)
++                        offset[1] += get_bits(gb, 9) - 256;
++                    else
++                        offset[1] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
++                    if (offset[1] > 255U) {
++                        av_log(ac->avctx, AV_LOG_ERROR,
++                               "%s (%d) out of range.\n", sf_str[1], offset[1]);
++                        return -1;
++                    }
++                    sf[idx] = -ff_aac_pow2sf_tab[offset[1] + sf_offset + 100];
++                }
++            } else {
++                for (; i < run_end; i++, idx++) {
++                    offset[0] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
++                    if (offset[0] > 255U) {
++                        av_log(ac->avctx, AV_LOG_ERROR,
++                               "%s (%d) out of range.\n", sf_str[0], offset[0]);
++                        return -1;
++                    }
++                    sf[idx] = -ff_aac_pow2sf_tab[ offset[0] + sf_offset];
++                }
++            }
++        }
++    }
++    return 0;
++}
++
++/**
++ * Decode pulse data; reference: table 4.7.
++ */
++static int decode_pulses(Pulse *pulse, GetBitContext *gb,
++                         const uint16_t *swb_offset, int num_swb)
++{
++    int i, pulse_swb;
++    pulse->num_pulse = get_bits(gb, 2) + 1;
++    pulse_swb        = get_bits(gb, 6);
++    if (pulse_swb >= num_swb)
++        return -1;
++    pulse->pos[0]    = swb_offset[pulse_swb];
++    pulse->pos[0]   += get_bits(gb, 5);
++    if (pulse->pos[0] > 1023)
++        return -1;
++    pulse->amp[0]    = get_bits(gb, 4);
++    for (i = 1; i < pulse->num_pulse; i++) {
++        pulse->pos[i] = get_bits(gb, 5) + pulse->pos[i - 1];
++        if (pulse->pos[i] > 1023)
++            return -1;
++        pulse->amp[i] = get_bits(gb, 4);
++    }
++    return 0;
++}
++
++/**
++ * Decode Temporal Noise Shaping data; reference: table 4.48.
++ *
++ * @return  Returns error status. 0 - OK, !0 - error
++ */
++static int decode_tns(AACContext *ac, TemporalNoiseShaping *tns,
++                      GetBitContext *gb, const IndividualChannelStream *ics)
++{
++    int w, filt, i, coef_len, coef_res, coef_compress;
++    const int is8 = ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE;
++    const int tns_max_order = is8 ? 7 : ac->m4ac.object_type == AOT_AAC_MAIN ? 20 : 12;
++    for (w = 0; w < ics->num_windows; w++) {
++        if ((tns->n_filt[w] = get_bits(gb, 2 - is8))) {
++            coef_res = get_bits1(gb);
++
++            for (filt = 0; filt < tns->n_filt[w]; filt++) {
++                int tmp2_idx;
++                tns->length[w][filt] = get_bits(gb, 6 - 2 * is8);
++
++                if ((tns->order[w][filt] = get_bits(gb, 5 - 2 * is8)) > tns_max_order) {
++                    av_log(ac->avctx, AV_LOG_ERROR, "TNS filter order %d is greater than maximum %d.\n",
++                           tns->order[w][filt], tns_max_order);
++                    tns->order[w][filt] = 0;
++                    return -1;
++                }
++                if (tns->order[w][filt]) {
++                    tns->direction[w][filt] = get_bits1(gb);
++                    coef_compress = get_bits1(gb);
++                    coef_len = coef_res + 3 - coef_compress;
++                    tmp2_idx = 2 * coef_compress + coef_res;
++
++                    for (i = 0; i < tns->order[w][filt]; i++)
++                        tns->coef[w][filt][i] = tns_tmp2_map[tmp2_idx][get_bits(gb, coef_len)];
++                }
++            }
++        }
++    }
++    return 0;
++}
++
++/**
++ * Decode Mid/Side data; reference: table 4.54.
++ *
++ * @param   ms_present  Indicates mid/side stereo presence. [0] mask is all 0s;
++ *                      [1] mask is decoded from bitstream; [2] mask is all 1s;
++ *                      [3] reserved for scalable AAC
++ */
++static void decode_mid_side_stereo(ChannelElement *cpe, GetBitContext *gb,
++                                   int ms_present)
++{
++    int idx;
++    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) {
++        memset(cpe->ms_mask, 1, cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb * sizeof(cpe->ms_mask[0]));
++    }
++}
++
++#ifndef VMUL2
++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;
++}
++#endif
++
++#ifndef VMUL4
++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;
++}
++#endif
++
++#ifndef VMUL2S
++static inline float *VMUL2S(float *dst, const float *v, unsigned idx,
++                            unsigned sign, const float *scale)
++{
++    union float754 s0, s1;
++
++    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;
++}
++#endif
++
++#ifndef VMUL4S
++static inline float *VMUL4S(float *dst, const float *v, unsigned idx,
++                            unsigned sign, const float *scale)
++{
++    unsigned nz = idx >> 12;
++    union float754 s = { .f = *scale };
++    union float754 t;
++
++    t.i = s.i ^ (sign & 1<<31);
++    *dst++ = v[idx    & 3] * t.f;
++
++    sign <<= nz & 1; nz >>= 1;
++    t.i = s.i ^ (sign & 1<<31);
++    *dst++ = v[idx>>2 & 3] * t.f;
++
++    sign <<= nz & 1; nz >>= 1;
++    t.i = s.i ^ (sign & 1<<31);
++    *dst++ = v[idx>>4 & 3] * t.f;
++
++    sign <<= nz & 1; nz >>= 1;
++    t.i = s.i ^ (sign & 1<<31);
++    *dst++ = v[idx>>6 & 3] * t.f;
++
++    return dst;
++}
++#endif
++
++/**
++ * 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
++ */
++static int decode_spectrum_and_dequant(AACContext *ac, float coef[1024],
++                                       GetBitContext *gb, const float sf[120],
++                                       int pulse_present, const Pulse *pulse,
++                                       const IndividualChannelStream *ics,
++                                       enum BandType band_type[120])
++{
++    int i, k, g, idx = 0;
++    const int c = 1024 / ics->num_windows;
++    const uint16_t *offsets = ics->swb_offset;
++    float *coef_base = coef;
++    int err_idx;
++
++    for (g = 0; g < ics->num_windows; g++)
++        memset(coef + g * 128 + offsets[ics->max_sfb], 0, sizeof(float) * (c - offsets[ics->max_sfb]));
++
++    for (g = 0; g < ics->num_window_groups; g++) {
++        unsigned g_len = ics->group_len[g];
++
++        for (i = 0; i < ics->max_sfb; i++, idx++) {
++            const unsigned cbt_m1 = band_type[idx] - 1;
++            float *cfo = coef + offsets[i];
++            int off_len = offsets[i + 1] - offsets[i];
++            int group;
++
++            if (cbt_m1 >= INTENSITY_BT2 - 1) {
++                for (group = 0; group < g_len; group++, cfo+=128) {
++                    memset(cfo, 0, off_len * sizeof(float));
++                }
++            } else if (cbt_m1 == NOISE_BT - 1) {
++                for (group = 0; group < g_len; group++, cfo+=128) {
++                    float scale;
++                    float band_energy;
++
++                    for (k = 0; k < off_len; k++) {
++                        ac->random_state  = lcg_random(ac->random_state);
++                        cfo[k] = ac->random_state;
++                    }
++
++                    band_energy = ac->dsp.scalarproduct_float(cfo, cfo, off_len);
++                    scale = sf[idx] / sqrtf(band_energy);
++                    ac->dsp.vector_fmul_scalar(cfo, cfo, scale, off_len);
++                }
++            } else {
++                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;
++                const int cb_size = ff_aac_spectral_sizes[cbt_m1];
++                OPEN_READER(re, gb);
++
++                switch (cbt_m1 >> 1) {
++                case 0:
++                    for (group = 0; group < g_len; group++, cfo+=128) {
++                        float *cf = cfo;
++                        int len = off_len;
++
++                        do {
++                            int code;
++                            unsigned cb_idx;
++
++                            UPDATE_CACHE(re, gb);
++                            GET_VLC(code, re, gb, vlc_tab, 8, 2);
++
++                            if (code >= cb_size) {
++                                err_idx = code;
++                                goto err_cb_overflow;
++                            }
++
++                            cb_idx = cb_vector_idx[code];
++                            cf = VMUL4(cf, vq, cb_idx, sf + idx);
++                        } while (len -= 4);
++                    }
++                    break;
++
++                case 1:
++                    for (group = 0; group < g_len; group++, cfo+=128) {
++                        float *cf = cfo;
++                        int len = off_len;
++
++                        do {
++                            int code;
++                            unsigned nnz;
++                            unsigned cb_idx;
++                            uint32_t bits;
++
++                            UPDATE_CACHE(re, gb);
++                            GET_VLC(code, re, gb, vlc_tab, 8, 2);
++
++                            if (code >= cb_size) {
++                                err_idx = code;
++                                goto err_cb_overflow;
++                            }
++
++#if MIN_CACHE_BITS < 20
++                            UPDATE_CACHE(re, gb);
++#endif
++                            cb_idx = cb_vector_idx[code];
++                            nnz = cb_idx >> 8 & 15;
++                            bits = SHOW_UBITS(re, gb, nnz) << (32-nnz);
++                            LAST_SKIP_BITS(re, gb, nnz);
++                            cf = VMUL4S(cf, vq, cb_idx, bits, sf + idx);
++                        } while (len -= 4);
++                    }
++                    break;
++
++                case 2:
++                    for (group = 0; group < g_len; group++, cfo+=128) {
++                        float *cf = cfo;
++                        int len = off_len;
++
++                        do {
++                            int code;
++                            unsigned cb_idx;
++
++                            UPDATE_CACHE(re, gb);
++                            GET_VLC(code, re, gb, vlc_tab, 8, 2);
++
++                            if (code >= cb_size) {
++                                err_idx = code;
++                                goto err_cb_overflow;
++                            }
++
++                            cb_idx = cb_vector_idx[code];
++                            cf = VMUL2(cf, vq, cb_idx, sf + idx);
++                        } while (len -= 2);
++                    }
++                    break;
++
++                case 3:
++                case 4:
++                    for (group = 0; group < g_len; group++, cfo+=128) {
++                        float *cf = cfo;
++                        int len = off_len;
++
++                        do {
++                            int code;
++                            unsigned nnz;
++                            unsigned cb_idx;
++                            unsigned sign;
++
++                            UPDATE_CACHE(re, gb);
++                            GET_VLC(code, re, gb, vlc_tab, 8, 2);
++
++                            if (code >= cb_size) {
++                                err_idx = code;
++                                goto err_cb_overflow;
++                            }
++
++                            cb_idx = cb_vector_idx[code];
++                            nnz = cb_idx >> 8 & 15;
++                            sign = SHOW_UBITS(re, gb, nnz) << (cb_idx >> 12);
++                            LAST_SKIP_BITS(re, gb, nnz);
++                            cf = VMUL2S(cf, vq, cb_idx, sign, sf + idx);
++                        } while (len -= 2);
++                    }
++                    break;
++
++                default:
++                    for (group = 0; group < g_len; group++, cfo+=128) {
++                        float *cf = cfo;
++                        uint32_t *icf = (uint32_t *) cf;
++                        int len = off_len;
++
++                        do {
++                            int code;
++                            unsigned nzt, nnz;
++                            unsigned cb_idx;
++                            uint32_t bits;
++                            int j;
++
++                            UPDATE_CACHE(re, gb);
++                            GET_VLC(code, re, gb, vlc_tab, 8, 2);
++
++                            if (!code) {
++                                *icf++ = 0;
++                                *icf++ = 0;
++                                continue;
++                            }
++
++                            if (code >= cb_size) {
++                                err_idx = code;
++                                goto err_cb_overflow;
++                            }
++
++                            cb_idx = cb_vector_idx[code];
++                            nnz = cb_idx >> 12;
++                            nzt = cb_idx >> 8;
++                            bits = SHOW_UBITS(re, gb, nnz) << (32-nnz);
++                            LAST_SKIP_BITS(re, gb, nnz);
++
++                            for (j = 0; j < 2; j++) {
++                                if (nzt & 1<<j) {
++                                    uint32_t b;
++                                    int n;
++                                    /* The total length of escape_sequence must be < 22 bits according
++                                       to the specification (i.e. max is 111111110xxxxxxxxxxxx). */
++                                    UPDATE_CACHE(re, gb);
++                                    b = GET_CACHE(re, gb);
++                                    b = 31 - av_log2(~b);
++
++                                    if (b > 8) {
++                                        av_log(ac->avctx, AV_LOG_ERROR, "error in spectral data, ESC overflow\n");
++                                        return -1;
++                                    }
++
++#if MIN_CACHE_BITS < 21
++                                    LAST_SKIP_BITS(re, gb, b + 1);
++                                    UPDATE_CACHE(re, gb);
++#else
++                                    SKIP_BITS(re, gb, b + 1);
++#endif
++                                    b += 4;
++                                    n = (1 << b) + SHOW_UBITS(re, gb, b);
++                                    LAST_SKIP_BITS(re, gb, b);
++                                    *icf++ = cbrt_tab[n] | (bits & 1<<31);
++                                    bits <<= 1;
++                                } else {
++                                    unsigned v = ((const uint32_t*)vq)[cb_idx & 15];
++                                    *icf++ = (bits & 1<<31) | v;
++                                    bits <<= !!v;
++                                }
++                                cb_idx >>= 4;
++                            }
++                        } while (len -= 2);
++
++                        ac->dsp.vector_fmul_scalar(cfo, cfo, sf[idx], off_len);
++                    }
++                }
++
++                CLOSE_READER(re, gb);
++            }
++        }
++        coef += g_len << 7;
++    }
++
++    if (pulse_present) {
++        idx = 0;
++        for (i = 0; i < pulse->num_pulse; i++) {
++            float co = coef_base[ pulse->pos[i] ];
++            while (offsets[idx + 1] <= pulse->pos[i])
++                idx++;
++            if (band_type[idx] != NOISE_BT && sf[idx]) {
++                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];
++            }
++        }
++    }
++    return 0;
++
++err_cb_overflow:
++    av_log(ac->avctx, AV_LOG_ERROR,
++           "Read beyond end of ff_aac_codebook_vectors[%d][]. index %d >= %d\n",
++           band_type[idx], err_idx, ff_aac_spectral_sizes[band_type[idx]]);
++    return -1;
++}
++
++static av_always_inline float flt16_round(float pf)
++{
++    union float754 tmp;
++    tmp.f = pf;
++    tmp.i = (tmp.i + 0x00008000U) & 0xFFFF0000U;
++    return tmp.f;
++}
++
++static av_always_inline float flt16_even(float pf)
++{
++    union float754 tmp;
++    tmp.f = pf;
++    tmp.i = (tmp.i + 0x00007FFFU + (tmp.i & 0x00010000U >> 16)) & 0xFFFF0000U;
++    return tmp.f;
++}
++
++static av_always_inline float flt16_trunc(float pf)
++{
++    union float754 pun;
++    pun.f = pf;
++    pun.i &= 0xFFFF0000U;
++    return pun.f;
++}
++
++static av_always_inline void predict(AACContext *ac, PredictorState *ps, float *coef,
++                    int output_enable)
++{
++    const float a     = 0.953125; // 61.0 / 64
++    const float alpha = 0.90625;  // 29.0 / 32
++    float e0, e1;
++    float pv;
++    float k1, k2;
++
++    k1 = ps->var0 > 1 ? ps->cor0 * flt16_even(a / ps->var0) : 0;
++    k2 = ps->var1 > 1 ? ps->cor1 * flt16_even(a / ps->var1) : 0;
++
++    pv = flt16_round(k1 * ps->r0 + k2 * ps->r1);
++    if (output_enable)
++        *coef += pv * ac->sf_scale;
++
++    e0 = *coef / ac->sf_scale;
++    e1 = e0 - k1 * ps->r0;
++
++    ps->cor1 = flt16_trunc(alpha * ps->cor1 + ps->r1 * e1);
++    ps->var1 = flt16_trunc(alpha * ps->var1 + 0.5 * (ps->r1 * ps->r1 + e1 * e1));
++    ps->cor0 = flt16_trunc(alpha * ps->cor0 + ps->r0 * e0);
++    ps->var0 = flt16_trunc(alpha * ps->var0 + 0.5 * (ps->r0 * ps->r0 + e0 * e0));
++
++    ps->r1 = flt16_trunc(a * (ps->r0 - k1 * e0));
++    ps->r0 = flt16_trunc(a * e0);
++}
++
++/**
++ * Apply AAC-Main style frequency domain prediction.
++ */
++static void apply_prediction(AACContext *ac, SingleChannelElement *sce)
++{
++    int sfb, k;
++
++    if (!sce->ics.predictor_initialized) {
++        reset_all_predictors(sce->predictor_state);
++        sce->ics.predictor_initialized = 1;
++    }
++
++    if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
++        for (sfb = 0; sfb < ff_aac_pred_sfb_max[ac->m4ac.sampling_index]; sfb++) {
++            for (k = sce->ics.swb_offset[sfb]; k < sce->ics.swb_offset[sfb + 1]; k++) {
++                predict(ac, &sce->predictor_state[k], &sce->coeffs[k],
++                        sce->ics.predictor_present && sce->ics.prediction_used[sfb]);
++            }
++        }
++        if (sce->ics.predictor_reset_group)
++            reset_predictor_group(sce->predictor_state, sce->ics.predictor_reset_group);
++    } else
++        reset_all_predictors(sce->predictor_state);
++}
++
++/**
++ * Decode an individual_channel_stream payload; reference: table 4.44.
++ *
++ * @param   common_window   Channels have independent [0], or shared [1], Individual Channel Stream information.
++ * @param   scale_flag      scalable [1] or non-scalable [0] AAC (Unused until scalable AAC is implemented.)
++ *
++ * @return  Returns error status. 0 - OK, !0 - error
++ */
++static int decode_ics(AACContext *ac, SingleChannelElement *sce,
++                      GetBitContext *gb, int common_window, int scale_flag)
++{
++    Pulse pulse;
++    TemporalNoiseShaping    *tns = &sce->tns;
++    IndividualChannelStream *ics = &sce->ics;
++    float *out = sce->coeffs;
++    int global_gain, pulse_present = 0;
++
++    /* This assignment is to silence a GCC warning about the variable being used
++     * uninitialized when in fact it always is.
++     */
++    pulse.num_pulse = 0;
++
++    global_gain = get_bits(gb, 8);
++
++    if (!common_window && !scale_flag) {
++        if (decode_ics_info(ac, ics, gb, 0) < 0)
++            return -1;
++    }
++
++    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) {
++                av_log(ac->avctx, AV_LOG_ERROR, "Pulse tool not allowed in eight short sequence.\n");
++                return -1;
++            }
++            if (decode_pulses(&pulse, gb, ics->swb_offset, ics->num_swb)) {
++                av_log(ac->avctx, AV_LOG_ERROR, "Pulse data corrupt or invalid.\n");
++                return -1;
++            }
++        }
++        if ((tns->present = get_bits1(gb)) && decode_tns(ac, tns, gb, ics))
++            return -1;
++        if (get_bits1(gb)) {
++            av_log_missing_feature(ac->avctx, "SSR", 1);
++            return -1;
++        }
++    }
++
++    if (decode_spectrum_and_dequant(ac, out, gb, sce->sf, pulse_present, &pulse, ics, sce->band_type) < 0)
++        return -1;
++
++    if (ac->m4ac.object_type == AOT_AAC_MAIN && !common_window)
++        apply_prediction(ac, sce);
++
++    return 0;
++}
++
++/**
++ * Mid/Side stereo decoding; reference: 4.6.8.1.3.
++ */
++static void apply_mid_side_stereo(AACContext *ac, ChannelElement *cpe)
++{
++    const IndividualChannelStream *ics = &cpe->ch[0].ics;
++    float *ch0 = cpe->ch[0].coeffs;
++    float *ch1 = cpe->ch[1].coeffs;
++    int g, i, group, idx = 0;
++    const uint16_t *offsets = ics->swb_offset;
++    for (g = 0; g < ics->num_window_groups; g++) {
++        for (i = 0; i < ics->max_sfb; i++, idx++) {
++            if (cpe->ms_mask[idx] &&
++                    cpe->ch[0].band_type[idx] < NOISE_BT && cpe->ch[1].band_type[idx] < NOISE_BT) {
++                for (group = 0; group < ics->group_len[g]; group++) {
++                    ac->dsp.butterflies_float(ch0 + group * 128 + offsets[i],
++                                              ch1 + group * 128 + offsets[i],
++                                              offsets[i+1] - offsets[i]);
++                }
++            }
++        }
++        ch0 += ics->group_len[g] * 128;
++        ch1 += ics->group_len[g] * 128;
++    }
++}
++
++/**
++ * intensity stereo decoding; reference: 4.6.8.2.3
++ *
++ * @param   ms_present  Indicates mid/side stereo presence. [0] mask is all 0s;
++ *                      [1] mask is decoded from bitstream; [2] mask is all 1s;
++ *                      [3] reserved for scalable AAC
++ */
++static void apply_intensity_stereo(ChannelElement *cpe, int ms_present)
++{
++    const IndividualChannelStream *ics = &cpe->ch[1].ics;
++    SingleChannelElement         *sce1 = &cpe->ch[1];
++    float *coef0 = cpe->ch[0].coeffs, *coef1 = cpe->ch[1].coeffs;
++    const uint16_t *offsets = ics->swb_offset;
++    int g, group, i, k, idx = 0;
++    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++)
++                        for (k = offsets[i]; k < offsets[i + 1]; k++)
++                            coef1[group * 128 + k] = scale * coef0[group * 128 + k];
++                }
++            } else {
++                int bt_run_end = sce1->band_type_run_end[idx];
++                idx += bt_run_end - i;
++                i    = bt_run_end;
++            }
++        }
++        coef0 += ics->group_len[g] * 128;
++        coef1 += ics->group_len[g] * 128;
++    }
++}
++
++/**
++ * Decode a channel_pair_element; reference: table 4.4.
++ *
++ * @param   elem_id Identifies the instance of a syntax element.
++ *
++ * @return  Returns error status. 0 - OK, !0 - error
++ */
++static int decode_cpe(AACContext *ac, GetBitContext *gb, ChannelElement *cpe)
++{
++    int i, ret, common_window, ms_present = 0;
++
++    common_window = get_bits1(gb);
++    if (common_window) {
++        if (decode_ics_info(ac, &cpe->ch[0].ics, gb, 1))
++            return -1;
++        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;
++        ms_present = get_bits(gb, 2);
++        if (ms_present == 3) {
++            av_log(ac->avctx, AV_LOG_ERROR, "ms_present = 3 is reserved.\n");
++            return -1;
++        } else if (ms_present)
++            decode_mid_side_stereo(cpe, gb, ms_present);
++    }
++    if ((ret = decode_ics(ac, &cpe->ch[0], gb, common_window, 0)))
++        return ret;
++    if ((ret = decode_ics(ac, &cpe->ch[1], gb, common_window, 0)))
++        return ret;
++
++    if (common_window) {
++        if (ms_present)
++            apply_mid_side_stereo(ac, cpe);
++        if (ac->m4ac.object_type == AOT_AAC_MAIN) {
++            apply_prediction(ac, &cpe->ch[0]);
++            apply_prediction(ac, &cpe->ch[1]);
++        }
++    }
++
++    apply_intensity_stereo(cpe, ms_present);
++    return 0;
++}
++
++/**
++ * Decode coupling_channel_element; reference: table 4.8.
++ *
++ * @param   elem_id Identifies the instance of a syntax element.
++ *
++ * @return  Returns error status. 0 - OK, !0 - error
++ */
++static int decode_cce(AACContext *ac, GetBitContext *gb, ChannelElement *che)
++{
++    int num_gain = 0;
++    int c, g, sfb, ret;
++    int sign;
++    float scale;
++    SingleChannelElement *sce = &che->ch[0];
++    ChannelCoupling     *coup = &che->coup;
++
++    coup->coupling_point = 2 * get_bits1(gb);
++    coup->num_coupled = get_bits(gb, 3);
++    for (c = 0; c <= coup->num_coupled; c++) {
++        num_gain++;
++        coup->type[c] = get_bits1(gb) ? TYPE_CPE : TYPE_SCE;
++        coup->id_select[c] = get_bits(gb, 4);
++        if (coup->type[c] == TYPE_CPE) {
++            coup->ch_select[c] = get_bits(gb, 2);
++            if (coup->ch_select[c] == 3)
++                num_gain++;
++        } else
++            coup->ch_select[c] = 2;
++    }
++    coup->coupling_point += get_bits1(gb) || (coup->coupling_point >> 1);
++
++    sign  = get_bits(gb, 1);
++    scale = pow(2., pow(2., (int)get_bits(gb, 2) - 3));
++
++    if ((ret = decode_ics(ac, sce, gb, 0, 0)))
++        return ret;
++
++    for (c = 0; c < num_gain; c++) {
++        int idx  = 0;
++        int cge  = 1;
++        int gain = 0;
++        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;
++            gain_cache = pow(scale, -gain);
++        }
++        if (coup->coupling_point == AFTER_IMDCT) {
++            coup->gain[c][0] = gain_cache;
++        } else {
++            for (g = 0; g < sce->ics.num_window_groups; g++) {
++                for (sfb = 0; sfb < sce->ics.max_sfb; sfb++, idx++) {
++                    if (sce->band_type[idx] != ZERO_BT) {
++                        if (!cge) {
++                            int t = get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
++                            if (t) {
++                                int s = 1;
++                                t = gain += t;
++                                if (sign) {
++                                    s  -= 2 * (t & 0x1);
++                                    t >>= 1;
++                                }
++                                gain_cache = pow(scale, -t) * s;
++                            }
++                        }
++                        coup->gain[c][idx] = gain_cache;
++                    }
++                }
++            }
++        }
++    }
++    return 0;
++}
++
++/**
++ * Parse whether channels are to be excluded from Dynamic Range Compression; reference: table 4.53.
++ *
++ * @return  Returns number of bytes consumed.
++ */
++static int decode_drc_channel_exclusions(DynamicRangeControl *che_drc,
++                                         GetBitContext *gb)
++{
++    int i;
++    int num_excl_chan = 0;
++
++    do {
++        for (i = 0; i < 7; i++)
++            che_drc->exclude_mask[num_excl_chan++] = get_bits1(gb);
++    } while (num_excl_chan < MAX_CHANNELS - 7 && get_bits1(gb));
++
++    return num_excl_chan / 7;
++}
++
++/**
++ * Decode dynamic range information; reference: table 4.52.
++ *
++ * @param   cnt length of TYPE_FIL syntactic element in bytes
++ *
++ * @return  Returns number of bytes consumed.
++ */
++static int decode_dynamic_range(DynamicRangeControl *che_drc,
++                                GetBitContext *gb, int cnt)
++{
++    int n             = 1;
++    int drc_num_bands = 1;
++    int i;
++
++    /* pce_tag_present? */
++    if (get_bits1(gb)) {
++        che_drc->pce_instance_tag  = get_bits(gb, 4);
++        skip_bits(gb, 4); // tag_reserved_bits
++        n++;
++    }
++
++    /* excluded_chns_present? */
++    if (get_bits1(gb)) {
++        n += decode_drc_channel_exclusions(che_drc, gb);
++    }
++
++    /* drc_bands_present? */
++    if (get_bits1(gb)) {
++        che_drc->band_incr            = get_bits(gb, 4);
++        che_drc->interpolation_scheme = get_bits(gb, 4);
++        n++;
++        drc_num_bands += che_drc->band_incr;
++        for (i = 0; i < drc_num_bands; i++) {
++            che_drc->band_top[i] = get_bits(gb, 8);
++            n++;
++        }
++    }
++
++    /* prog_ref_level_present? */
++    if (get_bits1(gb)) {
++        che_drc->prog_ref_level = get_bits(gb, 7);
++        skip_bits1(gb); // prog_ref_level_reserved_bits
++        n++;
++    }
++
++    for (i = 0; i < drc_num_bands; i++) {
++        che_drc->dyn_rng_sgn[i] = get_bits1(gb);
++        che_drc->dyn_rng_ctl[i] = get_bits(gb, 7);
++        n++;
++    }
++
++    return n;
++}
++
++/**
++ * Decode extension data (incomplete); reference: table 4.51.
++ *
++ * @param   cnt length of TYPE_FIL syntactic element in bytes
++ *
++ * @return Returns number of bytes consumed
++ */
++static int decode_extension_payload(AACContext *ac, GetBitContext *gb, int cnt,
++                                    ChannelElement *che, enum RawDataBlockType elem_type)
++{
++    int crc_flag = 0;
++    int res = cnt;
++    switch (get_bits(gb, 4)) { // extension type
++    case EXT_SBR_DATA_CRC:
++        crc_flag++;
++    case EXT_SBR_DATA:
++        if (!che) {
++            av_log(ac->avctx, AV_LOG_ERROR, "SBR was found before the first channel element.\n");
++            return res;
++        } else if (!ac->m4ac.sbr) {
++            av_log(ac->avctx, AV_LOG_ERROR, "SBR signaled to be not-present but was found in the bitstream.\n");
++            skip_bits_long(gb, 8 * cnt - 4);
++            return res;
++        } else if (ac->m4ac.sbr == -1 && ac->output_configured == OC_LOCKED) {
++            av_log(ac->avctx, AV_LOG_ERROR, "Implicit SBR was found with a first occurrence after the first frame.\n");
++            skip_bits_long(gb, 8 * cnt - 4);
++            return res;
++        } else if (ac->m4ac.ps == -1 && ac->output_configured < OC_LOCKED && ac->avctx->channels == 1) {
++            ac->m4ac.sbr = 1;
++            ac->m4ac.ps = 1;
++            output_configure(ac, ac->che_pos, ac->che_pos, ac->m4ac.chan_config, ac->output_configured);
++        } else {
++            ac->m4ac.sbr = 1;
++        }
++        res = ff_decode_sbr_extension(ac, &che->sbr, gb, crc_flag, cnt, elem_type);
++        break;
++    case EXT_DYNAMIC_RANGE:
++        res = decode_dynamic_range(&ac->che_drc, gb, cnt);
++        break;
++    case EXT_FILL:
++    case EXT_FILL_DATA:
++    case EXT_DATA_ELEMENT:
++    default:
++        skip_bits_long(gb, 8 * cnt - 4);
++        break;
++    };
++    return res;
++}
++
++/**
++ * Decode Temporal Noise Shaping filter coefficients and apply all-pole filters; reference: 4.6.9.3.
++ *
++ * @param   decode  1 if tool is used normally, 0 if tool is used in LTP.
++ * @param   coef    spectral coefficients
++ */
++static void apply_tns(float coef[1024], TemporalNoiseShaping *tns,
++                      IndividualChannelStream *ics, int decode)
++{
++    const int mmm = FFMIN(ics->tns_max_bands, ics->max_sfb);
++    int w, filt, m, i;
++    int bottom, top, order, start, end, size, inc;
++    float lpc[TNS_MAX_ORDER];
++
++    for (w = 0; w < ics->num_windows; w++) {
++        bottom = ics->num_swb;
++        for (filt = 0; filt < tns->n_filt[w]; filt++) {
++            top    = bottom;
++            bottom = FFMAX(0, top - tns->length[w][filt]);
++            order  = tns->order[w][filt];
++            if (order == 0)
++                continue;
++
++            // tns_decode_coef
++            compute_lpc_coefs(tns->coef[w][filt], order, lpc, 0, 0, 0);
++
++            start = ics->swb_offset[FFMIN(bottom, mmm)];
++            end   = ics->swb_offset[FFMIN(   top, mmm)];
++            if ((size = end - start) <= 0)
++                continue;
++            if (tns->direction[w][filt]) {
++                inc = -1;
++                start = end - 1;
++            } else {
++                inc = 1;
++            }
++            start += w * 128;
++
++            // 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];
++        }
++    }
++}
++
++/**
++ * Conduct IMDCT and windowing.
++ */
++static void imdct_and_windowing(AACContext *ac, SingleChannelElement *sce, float bias)
++{
++    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;
++    int i;
++
++    // imdct
++    if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
++        if (ics->window_sequence[1] == ONLY_LONG_SEQUENCE || ics->window_sequence[1] == LONG_STOP_SEQUENCE)
++            av_log(ac->avctx, AV_LOG_WARNING,
++                   "Transition from an ONLY_LONG or LONG_STOP to an EIGHT_SHORT sequence detected. "
++                   "If you heard an audible artifact, please submit the sample to the FFmpeg developers.\n");
++        for (i = 0; i < 1024; i += 128)
++            ff_imdct_half(&ac->mdct_small, buf + i, in + i);
++    } else
++        ff_imdct_half(&ac->mdct, buf, in);
++
++    /* window overlapping
++     * NOTE: To simplify the overlapping code, all 'meaningless' short to long
++     * and long to short transitions are considered to be short to short
++     * transitions. This leaves just two cases (long to long and short to short)
++     * with a little special sauce for EIGHT_SHORT_SEQUENCE.
++     */
++    if ((ics->window_sequence[1] == ONLY_LONG_SEQUENCE || ics->window_sequence[1] == LONG_STOP_SEQUENCE) &&
++            (ics->window_sequence[0] == ONLY_LONG_SEQUENCE || ics->window_sequence[0] == LONG_START_SEQUENCE)) {
++        ac->dsp.vector_fmul_window(    out,               saved,            buf,         lwindow_prev, bias, 512);
++    } else {
++        for (i = 0; i < 448; i++)
++            out[i] = saved[i] + bias;
++
++        if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
++            ac->dsp.vector_fmul_window(out + 448 + 0*128, saved + 448,      buf + 0*128, swindow_prev, bias, 64);
++            ac->dsp.vector_fmul_window(out + 448 + 1*128, buf + 0*128 + 64, buf + 1*128, swindow,      bias, 64);
++            ac->dsp.vector_fmul_window(out + 448 + 2*128, buf + 1*128 + 64, buf + 2*128, swindow,      bias, 64);
++            ac->dsp.vector_fmul_window(out + 448 + 3*128, buf + 2*128 + 64, buf + 3*128, swindow,      bias, 64);
++            ac->dsp.vector_fmul_window(temp,              buf + 3*128 + 64, buf + 4*128, swindow,      bias, 64);
++            memcpy(                    out + 448 + 4*128, temp, 64 * sizeof(float));
++        } else {
++            ac->dsp.vector_fmul_window(out + 448,         saved + 448,      buf,         swindow_prev, bias, 64);
++            for (i = 576; i < 1024; i++)
++                out[i] = buf[i-512] + bias;
++        }
++    }
++
++    // buffer update
++    if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
++        for (i = 0; i < 64; i++)
++            saved[i] = temp[64 + i] - bias;
++        ac->dsp.vector_fmul_window(saved + 64,  buf + 4*128 + 64, buf + 5*128, swindow, 0, 64);
++        ac->dsp.vector_fmul_window(saved + 192, buf + 5*128 + 64, buf + 6*128, swindow, 0, 64);
++        ac->dsp.vector_fmul_window(saved + 320, buf + 6*128 + 64, buf + 7*128, swindow, 0, 64);
++        memcpy(                    saved + 448, buf + 7*128 + 64,  64 * sizeof(float));
++    } else if (ics->window_sequence[0] == LONG_START_SEQUENCE) {
++        memcpy(                    saved,       buf + 512,        448 * sizeof(float));
++        memcpy(                    saved + 448, buf + 7*128 + 64,  64 * sizeof(float));
++    } else { // LONG_STOP or ONLY_LONG
++        memcpy(                    saved,       buf + 512,        512 * sizeof(float));
++    }
++}
++
++/**
++ * Apply dependent channel coupling (applied before IMDCT).
++ *
++ * @param   index   index into coupling gain array
++ */
++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;
++    int g, i, group, k, idx = 0;
++    if (ac->m4ac.object_type == AOT_AAC_LTP) {
++        av_log(ac->avctx, AV_LOG_ERROR,
++               "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++) {
++            if (cce->ch[0].band_type[idx] != ZERO_BT) {
++                const float gain = cce->coup.gain[index][idx];
++                for (group = 0; group < ics->group_len[g]; group++) {
++                    for (k = offsets[i]; k < offsets[i + 1]; k++) {
++                        // XXX dsputil-ize
++                        dest[group * 128 + k] += gain * src[group * 128 + k];
++                    }
++                }
++            }
++        }
++        dest += ics->group_len[g] * 128;
++        src  += ics->group_len[g] * 128;
++    }
++}
++
++/**
++ * Apply independent channel coupling (applied after IMDCT).
++ *
++ * @param   index   index into coupling gain array
++ */
++static void apply_independent_coupling(AACContext *ac,
++                                       SingleChannelElement *target,
++                                       ChannelElement *cce, int index)
++{
++    int i;
++    const float gain = cce->coup.gain[index][0];
++    const float bias = ac->add_bias;
++    const float *src = cce->ch[0].ret;
++    float *dest = target->ret;
++    const int len = 1024 << (ac->m4ac.sbr == 1);
++
++    for (i = 0; i < len; i++)
++        dest[i] += gain * (src[i] - bias);
++}
++
++/**
++ * channel coupling transformation interface
++ *
++ * @param   index   index into coupling gain array
++ * @param   apply_coupling_method   pointer to (in)dependent coupling function
++ */
++static void apply_channel_coupling(AACContext *ac, ChannelElement *cc,
++                                   enum RawDataBlockType type, int elem_id,
++                                   enum CouplingPoint coupling_point,
++                                   void (*apply_coupling_method)(AACContext *ac, SingleChannelElement *target, ChannelElement *cce, int index))
++{
++    int i, c;
++
++    for (i = 0; i < MAX_ELEM_ID; i++) {
++        ChannelElement *cce = ac->che[TYPE_CCE][i];
++        int index = 0;
++
++        if (cce && cce->coup.coupling_point == coupling_point) {
++            ChannelCoupling *coup = &cce->coup;
++
++            for (c = 0; c <= coup->num_coupled; c++) {
++                if (coup->type[c] == type && coup->id_select[c] == elem_id) {
++                    if (coup->ch_select[c] != 1) {
++                        apply_coupling_method(ac, &cc->ch[0], cce, index);
++                        if (coup->ch_select[c] != 0)
++                            index++;
++                    }
++                    if (coup->ch_select[c] != 2)
++                        apply_coupling_method(ac, &cc->ch[1], cce, index++);
++                } else
++                    index += 1 + (coup->ch_select[c] == 3);
++            }
++        }
++    }
++}
++
++/**
++ * Convert spectral data to float samples, applying all supported tools as appropriate.
++ */
++static void spectral_to_sample(AACContext *ac)
++{
++    int i, type;
++    float imdct_bias = (ac->m4ac.sbr <= 0) ? ac->add_bias : 0.0f;
++    for (type = 3; type >= 0; type--) {
++        for (i = 0; i < MAX_ELEM_ID; i++) {
++            ChannelElement *che = ac->che[type][i];
++            if (che) {
++                if (type <= TYPE_CPE)
++                    apply_channel_coupling(ac, che, type, i, BEFORE_TNS, apply_dependent_coupling);
++                if (che->ch[0].tns.present)
++                    apply_tns(che->ch[0].coeffs, &che->ch[0].tns, &che->ch[0].ics, 1);
++                if (che->ch[1].tns.present)
++                    apply_tns(che->ch[1].coeffs, &che->ch[1].tns, &che->ch[1].ics, 1);
++                if (type <= TYPE_CPE)
++                    apply_channel_coupling(ac, che, type, i, BETWEEN_TNS_AND_IMDCT, apply_dependent_coupling);
++                if (type != TYPE_CCE || che->coup.coupling_point == AFTER_IMDCT) {
++                    imdct_and_windowing(ac, &che->ch[0], imdct_bias);
++                    if (type == TYPE_CPE) {
++                        imdct_and_windowing(ac, &che->ch[1], imdct_bias);
++                    }
++                    if (ac->m4ac.sbr > 0) {
++                        ff_sbr_apply(ac, &che->sbr, type, che->ch[0].ret, che->ch[1].ret);
++                    }
++                }
++                if (type <= TYPE_CCE)
++                    apply_channel_coupling(ac, che, type, i, AFTER_IMDCT, apply_independent_coupling);
++            }
++        }
++    }
++}
++
++static int parse_adts_frame_header(AACContext *ac, GetBitContext *gb)
++{
++    int size;
++    AACADTSHeaderInfo hdr_info;
++
++    size = ff_aac_parse_header(gb, &hdr_info);
++    if (size > 0) {
++        if (ac->output_configured != OC_LOCKED && hdr_info.chan_config) {
++            enum ChannelPosition new_che_pos[4][MAX_ELEM_ID];
++            memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]));
++            ac->m4ac.chan_config = hdr_info.chan_config;
++            if (set_default_channel_config(ac, new_che_pos, hdr_info.chan_config))
++                return -7;
++            if (output_configure(ac, ac->che_pos, new_che_pos, hdr_info.chan_config, OC_TRIAL_FRAME))
++                return -7;
++        } else if (ac->output_configured != OC_LOCKED) {
++            ac->output_configured = OC_NONE;
++        }
++        if (ac->output_configured != OC_LOCKED) {
++            ac->m4ac.sbr = -1;
++            ac->m4ac.ps  = -1;
++        }
++        ac->m4ac.sample_rate     = hdr_info.sample_rate;
++        ac->m4ac.sampling_index  = hdr_info.sampling_index;
++        ac->m4ac.object_type     = hdr_info.object_type;
++        if (!ac->avctx->sample_rate)
++            ac->avctx->sample_rate = hdr_info.sample_rate;
++        if (hdr_info.num_aac_frames == 1) {
++            if (!hdr_info.crc_absent)
++                skip_bits(gb, 16);
++        } else {
++            av_log_missing_feature(ac->avctx, "More than one AAC RDB per ADTS frame is", 0);
++            return -1;
++        }
++    }
++    return size;
++}
++
++static int aac_decode_frame(AVCodecContext *avctx, void *data,
++                            int *data_size, AVPacket *avpkt)
++{
++    const uint8_t *buf = avpkt->data;
++    int buf_size = avpkt->size;
++    AACContext *ac = avctx->priv_data;
++    ChannelElement *che = NULL, *che_prev = NULL;
++    GetBitContext gb;
++    enum RawDataBlockType elem_type, elem_type_prev = TYPE_END;
++    int err, elem_id, data_size_tmp;
++    int buf_consumed;
++    int samples = 0, multiplier;
++    int buf_offset;
++
++    init_get_bits(&gb, buf, buf_size * 8);
++
++    if (show_bits(&gb, 12) == 0xfff) {
++        if (parse_adts_frame_header(ac, &gb) < 0) {
++            av_log(avctx, AV_LOG_ERROR, "Error decoding AAC frame header.\n");
++            return -1;
++        }
++        if (ac->m4ac.sampling_index > 12) {
++            av_log(ac->avctx, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index);
++            return -1;
++        }
++    }
++
++    memset(ac->tags_seen_this_frame, 0, sizeof(ac->tags_seen_this_frame));
++    // parse
++    while ((elem_type = get_bits(&gb, 3)) != TYPE_END) {
++        elem_id = get_bits(&gb, 4);
++
++        if (elem_type < TYPE_DSE) {
++            if (!(che=get_che(ac, elem_type, elem_id))) {
++                av_log(ac->avctx, AV_LOG_ERROR, "channel element %d.%d is not allocated\n",
++                       elem_type, elem_id);
++                return -1;
++            }
++            samples = 1024;
++        }
++
++        switch (elem_type) {
++
++        case TYPE_SCE:
++            err = decode_ics(ac, &che->ch[0], &gb, 0, 0);
++            break;
++
++        case TYPE_CPE:
++            err = decode_cpe(ac, &gb, che);
++            break;
++
++        case TYPE_CCE:
++            err = decode_cce(ac, &gb, che);
++            break;
++
++        case TYPE_LFE:
++            err = decode_ics(ac, &che->ch[0], &gb, 0, 0);
++            break;
++
++        case TYPE_DSE:
++            err = skip_data_stream_element(ac, &gb);
++            break;
++
++        case TYPE_PCE: {
++            enum ChannelPosition new_che_pos[4][MAX_ELEM_ID];
++            memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]));
++            if ((err = decode_pce(ac, new_che_pos, &gb)))
++                break;
++            if (ac->output_configured > OC_TRIAL_PCE)
++                av_log(avctx, AV_LOG_ERROR,
++                       "Not evaluating a further program_config_element as this construct is dubious at best.\n");
++            else
++                err = output_configure(ac, ac->che_pos, new_che_pos, 0, OC_TRIAL_PCE);
++            break;
++        }
++
++        case TYPE_FIL:
++            if (elem_id == 15)
++                elem_id += get_bits(&gb, 8) - 1;
++            if (get_bits_left(&gb) < 8 * elem_id) {
++                    av_log(avctx, AV_LOG_ERROR, overread_err);
++                    return -1;
++            }
++            while (elem_id > 0)
++                elem_id -= decode_extension_payload(ac, &gb, elem_id, che_prev, elem_type_prev);
++            err = 0; /* FIXME */
++            break;
++
++        default:
++            err = -1; /* should not happen, but keeps compiler happy */
++            break;
++        }
++
++        che_prev       = che;
++        elem_type_prev = elem_type;
++
++        if (err)
++            return err;
++
++        if (get_bits_left(&gb) < 3) {
++            av_log(avctx, AV_LOG_ERROR, overread_err);
++            return -1;
++        }
++    }
++
++    spectral_to_sample(ac);
++
++    multiplier = (ac->m4ac.sbr == 1) ? ac->m4ac.ext_sample_rate > ac->m4ac.sample_rate : 0;
++    samples <<= multiplier;
++    if (ac->output_configured < OC_LOCKED) {
++        avctx->sample_rate = ac->m4ac.sample_rate << multiplier;
++        avctx->frame_size = samples;
++    }
++
++    data_size_tmp = samples * avctx->channels * sizeof(int16_t);
++    if (*data_size < data_size_tmp) {
++        av_log(avctx, AV_LOG_ERROR,
++               "Output buffer too small (%d) or trying to output too many samples (%d) for this frame.\n",
++               *data_size, data_size_tmp);
++        return -1;
++    }
++    *data_size = data_size_tmp;
++
++    if (samples)
++        ac->dsp.float_to_int16_interleave(data, (const float **)ac->output_data, samples, avctx->channels);
++
++    if (ac->output_configured)
++        ac->output_configured = OC_LOCKED;
++
++    buf_consumed = (get_bits_count(&gb) + 7) >> 3;
++    for (buf_offset = buf_consumed; buf_offset < buf_size; buf_offset++)
++        if (buf[buf_offset])
++            break;
++
++    return buf_size > buf_offset ? buf_consumed : buf_size;
++}
++
++static av_cold int aac_decode_close(AVCodecContext *avctx)
++{
++    AACContext *ac = avctx->priv_data;
++    int i, type;
++
++    for (i = 0; i < MAX_ELEM_ID; i++) {
++        for (type = 0; type < 4; type++) {
++            if (ac->che[type][i])
++                ff_aac_sbr_ctx_close(&ac->che[type][i]->sbr);
++            av_freep(&ac->che[type][i]);
++        }
++    }
++
++    ff_mdct_end(&ac->mdct);
++    ff_mdct_end(&ac->mdct_small);
++    return 0;
++}
++
++AVCodec aac_decoder = {
++    "aac",
++    AVMEDIA_TYPE_AUDIO,
++    CODEC_ID_AAC,
++    sizeof(AACContext),
++    aac_decode_init,
++    NULL,
++    aac_decode_close,
++    aac_decode_frame,
++    .long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"),
++    .sample_fmts = (const enum SampleFormat[]) {
++        SAMPLE_FMT_S16,SAMPLE_FMT_NONE
++    },
++    .channel_layouts = aac_channel_layout,
++};
+--- a/libavcodec/aac.h
++++ b/libavcodec/aac.h
+@@ -38,12 +38,6 @@
+ 
+ #include <stdint.h>
+ 
+-#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);
+-
+ #define MAX_CHANNELS 64
+ #define MAX_ELEM_ID 16
+ 
+@@ -241,7 +235,7 @@ typedef struct {
+  * main AAC context
+  */
+ typedef struct {
+-    AVCodecContext * avccontext;
++    AVCodecContext *avctx;
+ 
+     MPEG4AudioConfig m4ac;
+ 
+@@ -255,8 +249,9 @@ typedef struct {
+     enum ChannelPosition che_pos[4][MAX_ELEM_ID]; /**< channel element channel mapping with the
+                                                    *   first index as the first 4 raw data block types
+                                                    */
+-    ChannelElement * che[4][MAX_ELEM_ID];
+-    ChannelElement * tag_che_map[4][MAX_ELEM_ID];
++    ChannelElement          *che[4][MAX_ELEM_ID];
++    ChannelElement  *tag_che_map[4][MAX_ELEM_ID];
++    uint8_t tags_seen_this_frame[4][MAX_ELEM_ID];
+     int tags_mapped;
+     /** @} */
+ 
+--- /dev/null
++++ b/libavcodec/aac_tablegen_decl.h
+@@ -0,0 +1,34 @@
++/*
++ * Header file for hardcoded AAC tables
++ *
++ * Copyright (c) 2010 Alex Converse <alex.converse at gmail.com>
++ *
++ * 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
++ */
++
++#ifndef AAC_TABLEGEN_INIT_H
++#define AAC_TABLEGEN_INIT_H
++
++#if CONFIG_HARDCODED_TABLES
++#define ff_aac_tableinit()
++extern const float ff_aac_pow2sf_tab[428];
++#else
++void ff_aac_tableinit(void);
++extern       float ff_aac_pow2sf_tab[428];
++#endif /* CONFIG_HARDCODED_TABLES */
++
++#endif /* AAC_TABLEGEN_INIT_H */
+--- /dev/null
++++ b/libavcodec/aacps.c
+@@ -0,0 +1,1037 @@
++/*
++ * MPEG-4 Parametric Stereo decoding functions
++ * Copyright (c) 2010 Alex Converse <alex.converse at gmail.com>
++ *
++ * This file is part of FFmpeg.
++ *
++ * FFmpeg is free software; you can redistribute it and/or
++ * modify it under the terms of the GNU Lesser General Public
++ * License as published by the Free Software Foundation; either
++ * version 2.1 of the License, or (at your option) any later version.
++ *
++ * FFmpeg is distributed in the hope that it will be useful,
++ * but WITHOUT ANY WARRANTY; without even the implied warranty of
++ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
++ * Lesser General Public License for more details.
++ *
++ * You should have received a copy of the GNU Lesser General Public
++ * License along with FFmpeg; if not, write to the Free Software
++ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
++ */
++
++#include <stdint.h>
++#include "libavutil/common.h"
++#include "libavutil/mathematics.h"
++#include "avcodec.h"
++#include "get_bits.h"
++#include "aacps.h"
++#include "aacps_tablegen.h"
++#include "aacpsdata.c"
++
++#define PS_BASELINE 0  //< Operate in Baseline PS mode
++                       //< Baseline implies 10 or 20 stereo bands,
++                       //< mixing mode A, and no ipd/opd
++
++#define numQMFSlots 32 //numTimeSlots * RATE
++
++static const int8_t num_env_tab[2][4] = {
++    { 0, 1, 2, 4, },
++    { 1, 2, 3, 4, },
++};
++
++static const int8_t nr_iidicc_par_tab[] = {
++    10, 20, 34, 10, 20, 34,
++};
++
++static const int8_t nr_iidopd_par_tab[] = {
++     5, 11, 17,  5, 11, 17,
++};
++
++enum {
++    huff_iid_df1,
++    huff_iid_dt1,
++    huff_iid_df0,
++    huff_iid_dt0,
++    huff_icc_df,
++    huff_icc_dt,
++    huff_ipd_df,
++    huff_ipd_dt,
++    huff_opd_df,
++    huff_opd_dt,
++};
++
++static const int huff_iid[] = {
++    huff_iid_df0,
++    huff_iid_df1,
++    huff_iid_dt0,
++    huff_iid_dt1,
++};
++
++static VLC vlc_ps[10];
++
++/**
++ * Read Inter-channel Intensity Difference/Inter-Channel Coherence/
++ * Inter-channel Phase Difference/Overall Phase Difference parameters from the
++ * bitstream.
++ *
++ * @param avctx contains the current codec context
++ * @param gb    pointer to the input bitstream
++ * @param ps    pointer to the Parametric Stereo context
++ * @param par   pointer to the parameter to be read
++ * @param e     envelope to decode
++ * @param dt    1: time delta-coded, 0: frequency delta-coded
++ */
++#define READ_PAR_DATA(PAR, OFFSET, MASK, ERR_CONDITION) \
++static int read_ ## PAR ## _data(AVCodecContext *avctx, GetBitContext *gb, PSContext *ps, \
++                        int8_t (*PAR)[PS_MAX_NR_IIDICC], int table_idx, int e, int dt) \
++{ \
++    int b, num = ps->nr_ ## PAR ## _par; \
++    VLC_TYPE (*vlc_table)[2] = vlc_ps[table_idx].table; \
++    if (dt) { \
++        int e_prev = e ? e - 1 : ps->num_env_old - 1; \
++        e_prev = FFMAX(e_prev, 0); \
++        for (b = 0; b < num; b++) { \
++            int val = PAR[e_prev][b] + get_vlc2(gb, vlc_table, 9, 3) - OFFSET; \
++            if (MASK) val &= MASK; \
++            PAR[e][b] = val; \
++            if (ERR_CONDITION) \
++                goto err; \
++        } \
++    } else { \
++        int val = 0; \
++        for (b = 0; b < num; b++) { \
++            val += get_vlc2(gb, vlc_table, 9, 3) - OFFSET; \
++            if (MASK) val &= MASK; \
++            PAR[e][b] = val; \
++            if (ERR_CONDITION) \
++                goto err; \
++        } \
++    } \
++    return 0; \
++err: \
++    av_log(avctx, AV_LOG_ERROR, "illegal "#PAR"\n"); \
++    return -1; \
++}
++
++READ_PAR_DATA(iid,    huff_offset[table_idx],    0, FFABS(ps->iid_par[e][b]) > 7 + 8 * ps->iid_quant)
++READ_PAR_DATA(icc,    huff_offset[table_idx],    0, ps->icc_par[e][b] > 7U)
++READ_PAR_DATA(ipdopd,                      0, 0x07, 0)
++
++static int ps_read_extension_data(GetBitContext *gb, PSContext *ps, int ps_extension_id)
++{
++    int e;
++    int count = get_bits_count(gb);
++
++    if (ps_extension_id)
++        return 0;
++
++    ps->enable_ipdopd = get_bits1(gb);
++    if (ps->enable_ipdopd) {
++        for (e = 0; e < ps->num_env; e++) {
++            int dt = get_bits1(gb);
++            read_ipdopd_data(NULL, gb, ps, ps->ipd_par, dt ? huff_ipd_dt : huff_ipd_df, e, dt);
++            dt = get_bits1(gb);
++            read_ipdopd_data(NULL, gb, ps, ps->opd_par, dt ? huff_opd_dt : huff_opd_df, e, dt);
++        }
++    }
++    skip_bits1(gb);      //reserved_ps
++    return get_bits_count(gb) - count;
++}
++
++static void ipdopd_reset(int8_t *opd_hist, int8_t *ipd_hist)
++{
++    int i;
++    for (i = 0; i < PS_MAX_NR_IPDOPD; i++) {
++        opd_hist[i] = 0;
++        ipd_hist[i] = 0;
++    }
++}
++
++int ff_ps_read_data(AVCodecContext *avctx, GetBitContext *gb_host, PSContext *ps, int bits_left)
++{
++    int e;
++    int bit_count_start = get_bits_count(gb_host);
++    int header;
++    int bits_consumed;
++    GetBitContext gbc = *gb_host, *gb = &gbc;
++
++    header = get_bits1(gb);
++    if (header) {     //enable_ps_header
++        ps->enable_iid = get_bits1(gb);
++        if (ps->enable_iid) {
++            int iid_mode = get_bits(gb, 3);
++            if (iid_mode > 5) {
++                av_log(avctx, AV_LOG_ERROR, "iid_mode %d is reserved.\n",
++                       iid_mode);
++                goto err;
++            }
++            ps->nr_iid_par    = nr_iidicc_par_tab[iid_mode];
++            ps->iid_quant     = iid_mode > 2;
++            ps->nr_ipdopd_par = nr_iidopd_par_tab[iid_mode];
++        }
++        ps->enable_icc = get_bits1(gb);
++        if (ps->enable_icc) {
++            ps->icc_mode = get_bits(gb, 3);
++            if (ps->icc_mode > 5) {
++                av_log(avctx, AV_LOG_ERROR, "icc_mode %d is reserved.\n",
++                       ps->icc_mode);
++                goto err;
++            }
++            ps->nr_icc_par = nr_iidicc_par_tab[ps->icc_mode];
++        }
++        ps->enable_ext = get_bits1(gb);
++    }
++
++    ps->frame_class = get_bits1(gb);
++    ps->num_env_old = ps->num_env;
++    ps->num_env     = num_env_tab[ps->frame_class][get_bits(gb, 2)];
++
++    ps->border_position[0] = -1;
++    if (ps->frame_class) {
++        for (e = 1; e <= ps->num_env; e++)
++            ps->border_position[e] = get_bits(gb, 5);
++    } else
++        for (e = 1; e <= ps->num_env; e++)
++            ps->border_position[e] = (e * numQMFSlots >> ff_log2_tab[ps->num_env]) - 1;
++
++    if (ps->enable_iid) {
++        for (e = 0; e < ps->num_env; e++) {
++            int dt = get_bits1(gb);
++            if (read_iid_data(avctx, gb, ps, ps->iid_par, huff_iid[2*dt+ps->iid_quant], e, dt))
++                goto err;
++        }
++    } else
++        memset(ps->iid_par, 0, sizeof(ps->iid_par));
++
++    if (ps->enable_icc)
++        for (e = 0; e < ps->num_env; e++) {
++            int dt = get_bits1(gb);
++            if (read_icc_data(avctx, gb, ps, ps->icc_par, dt ? huff_icc_dt : huff_icc_df, e, dt))
++                goto err;
++        }
++    else
++        memset(ps->icc_par, 0, sizeof(ps->icc_par));
++
++    if (ps->enable_ext) {
++        int cnt = get_bits(gb, 4);
++        if (cnt == 15) {
++            cnt += get_bits(gb, 8);
++        }
++        cnt *= 8;
++        while (cnt > 7) {
++            int ps_extension_id = get_bits(gb, 2);
++            cnt -= 2 + ps_read_extension_data(gb, ps, ps_extension_id);
++        }
++        if (cnt < 0) {
++            av_log(avctx, AV_LOG_ERROR, "ps extension overflow %d", cnt);
++            goto err;
++        }
++        skip_bits(gb, cnt);
++    }
++
++    ps->enable_ipdopd &= !PS_BASELINE;
++
++    //Fix up envelopes
++    if (!ps->num_env || ps->border_position[ps->num_env] < numQMFSlots - 1) {
++        //Create a fake envelope
++        int source = ps->num_env ? ps->num_env - 1 : ps->num_env_old - 1;
++        if (source >= 0 && source != ps->num_env) {
++            if (ps->enable_iid) {
++                memcpy(ps->iid_par+ps->num_env, ps->iid_par+source, sizeof(ps->iid_par[0]));
++            }
++            if (ps->enable_icc) {
++                memcpy(ps->icc_par+ps->num_env, ps->icc_par+source, sizeof(ps->icc_par[0]));
++            }
++            if (ps->enable_ipdopd) {
++                memcpy(ps->ipd_par+ps->num_env, ps->ipd_par+source, sizeof(ps->ipd_par[0]));
++                memcpy(ps->opd_par+ps->num_env, ps->opd_par+source, sizeof(ps->opd_par[0]));
++            }
++        }
++        ps->num_env++;
++        ps->border_position[ps->num_env] = numQMFSlots - 1;
++    }
++
++
++    ps->is34bands_old = ps->is34bands;
++    if (!PS_BASELINE && (ps->enable_iid || ps->enable_icc))
++        ps->is34bands = (ps->enable_iid && ps->nr_iid_par == 34) ||
++                        (ps->enable_icc && ps->nr_icc_par == 34);
++
++    //Baseline
++    if (!ps->enable_ipdopd) {
++        memset(ps->ipd_par, 0, sizeof(ps->ipd_par));
++        memset(ps->opd_par, 0, sizeof(ps->opd_par));
++    }
++
++    if (header)
++        ps->start = 1;
++
++    bits_consumed = get_bits_count(gb) - bit_count_start;
++    if (bits_consumed <= bits_left) {
++        skip_bits_long(gb_host, bits_consumed);
++        return bits_consumed;
++    }
++    av_log(avctx, AV_LOG_ERROR, "Expected to read %d PS bits actually read %d.\n", bits_left, bits_consumed);
++err:
++    ps->start = 0;
++    skip_bits_long(gb_host, bits_left);
++    return bits_left;
++}
++
++/** Split one subband into 2 subsubbands with a symmetric real filter.
++ * The filter must have its non-center even coefficients equal to zero. */
++static void hybrid2_re(float (*in)[2], float (*out)[32][2], const float filter[7], int len, int reverse)
++{
++    int i, j;
++    for (i = 0; i < len; i++, in++) {
++        float re_in = filter[6] * in[6][0];          //real inphase
++        float re_op = 0.0f;                          //real out of phase
++        float im_in = filter[6] * in[6][1];          //imag inphase
++        float im_op = 0.0f;                          //imag out of phase
++        for (j = 0; j < 6; j += 2) {
++            re_op += filter[j+1] * (in[j+1][0] + in[12-j-1][0]);
++            im_op += filter[j+1] * (in[j+1][1] + in[12-j-1][1]);
++        }
++        out[ reverse][i][0] = re_in + re_op;
++        out[ reverse][i][1] = im_in + im_op;
++        out[!reverse][i][0] = re_in - re_op;
++        out[!reverse][i][1] = im_in - im_op;
++    }
++}
++
++/** Split one subband into 6 subsubbands with a complex filter */
++static void hybrid6_cx(float (*in)[2], float (*out)[32][2], const float (*filter)[7][2], int len)
++{
++    int i, j, ssb;
++    int N = 8;
++    float temp[8][2];
++
++    for (i = 0; i < len; i++, in++) {
++        for (ssb = 0; ssb < N; ssb++) {
++            float sum_re = filter[ssb][6][0] * in[6][0], sum_im = filter[ssb][6][0] * in[6][1];
++            for (j = 0; j < 6; j++) {
++                float in0_re = in[j][0];
++                float in0_im = in[j][1];
++                float in1_re = in[12-j][0];
++                float in1_im = in[12-j][1];
++                sum_re += filter[ssb][j][0] * (in0_re + in1_re) - filter[ssb][j][1] * (in0_im - in1_im);
++                sum_im += filter[ssb][j][0] * (in0_im + in1_im) + filter[ssb][j][1] * (in0_re - in1_re);
++            }
++            temp[ssb][0] = sum_re;
++            temp[ssb][1] = sum_im;
++        }
++        out[0][i][0] = temp[6][0];
++        out[0][i][1] = temp[6][1];
++        out[1][i][0] = temp[7][0];
++        out[1][i][1] = temp[7][1];
++        out[2][i][0] = temp[0][0];
++        out[2][i][1] = temp[0][1];
++        out[3][i][0] = temp[1][0];
++        out[3][i][1] = temp[1][1];
++        out[4][i][0] = temp[2][0] + temp[5][0];
++        out[4][i][1] = temp[2][1] + temp[5][1];
++        out[5][i][0] = temp[3][0] + temp[4][0];
++        out[5][i][1] = temp[3][1] + temp[4][1];
++    }
++}
++
++static void hybrid4_8_12_cx(float (*in)[2], float (*out)[32][2], const float (*filter)[7][2], int N, int len)
++{
++    int i, j, ssb;
++
++    for (i = 0; i < len; i++, in++) {
++        for (ssb = 0; ssb < N; ssb++) {
++            float sum_re = filter[ssb][6][0] * in[6][0], sum_im = filter[ssb][6][0] * in[6][1];
++            for (j = 0; j < 6; j++) {
++                float in0_re = in[j][0];
++                float in0_im = in[j][1];
++                float in1_re = in[12-j][0];
++                float in1_im = in[12-j][1];
++                sum_re += filter[ssb][j][0] * (in0_re + in1_re) - filter[ssb][j][1] * (in0_im - in1_im);
++                sum_im += filter[ssb][j][0] * (in0_im + in1_im) + filter[ssb][j][1] * (in0_re - in1_re);
++            }
++            out[ssb][i][0] = sum_re;
++            out[ssb][i][1] = sum_im;
++        }
++    }
++}
++
++static void hybrid_analysis(float out[91][32][2], float in[5][44][2], float L[2][38][64], int is34, int len)
++{
++    int i, j;
++    for (i = 0; i < 5; i++) {
++        for (j = 0; j < 38; j++) {
++            in[i][j+6][0] = L[0][j][i];
++            in[i][j+6][1] = L[1][j][i];
++        }
++    }
++    if (is34) {
++        hybrid4_8_12_cx(in[0], out,    f34_0_12, 12, len);
++        hybrid4_8_12_cx(in[1], out+12, f34_1_8,   8, len);
++        hybrid4_8_12_cx(in[2], out+20, f34_2_4,   4, len);
++        hybrid4_8_12_cx(in[3], out+24, f34_2_4,   4, len);
++        hybrid4_8_12_cx(in[4], out+28, f34_2_4,   4, len);
++        for (i = 0; i < 59; i++) {
++            for (j = 0; j < len; j++) {
++                out[i+32][j][0] = L[0][j][i+5];
++                out[i+32][j][1] = L[1][j][i+5];
++            }
++        }
++    } else {
++        hybrid6_cx(in[0], out, f20_0_8, len);
++        hybrid2_re(in[1], out+6, g1_Q2, len, 1);
++        hybrid2_re(in[2], out+8, g1_Q2, len, 0);
++        for (i = 0; i < 61; i++) {
++            for (j = 0; j < len; j++) {
++                out[i+10][j][0] = L[0][j][i+3];
++                out[i+10][j][1] = L[1][j][i+3];
++            }
++        }
++    }
++    //update in_buf
++    for (i = 0; i < 5; i++) {
++        memcpy(in[i], in[i]+32, 6 * sizeof(in[i][0]));
++    }
++}
++
++static void hybrid_synthesis(float out[2][38][64], float in[91][32][2], int is34, int len)
++{
++    int i, n;
++    if (is34) {
++        for (n = 0; n < len; n++) {
++            memset(out[0][n], 0, 5*sizeof(out[0][n][0]));
++            memset(out[1][n], 0, 5*sizeof(out[1][n][0]));
++            for (i = 0; i < 12; i++) {
++                out[0][n][0] += in[   i][n][0];
++                out[1][n][0] += in[   i][n][1];
++            }
++            for (i = 0; i < 8; i++) {
++                out[0][n][1] += in[12+i][n][0];
++                out[1][n][1] += in[12+i][n][1];
++            }
++            for (i = 0; i < 4; i++) {
++                out[0][n][2] += in[20+i][n][0];
++                out[1][n][2] += in[20+i][n][1];
++                out[0][n][3] += in[24+i][n][0];
++                out[1][n][3] += in[24+i][n][1];
++                out[0][n][4] += in[28+i][n][0];
++                out[1][n][4] += in[28+i][n][1];
++            }
++        }
++        for (i = 0; i < 59; i++) {
++            for (n = 0; n < len; n++) {
++                out[0][n][i+5] = in[i+32][n][0];
++                out[1][n][i+5] = in[i+32][n][1];
++            }
++        }
++    } else {
++        for (n = 0; n < len; n++) {
++            out[0][n][0] = in[0][n][0] + in[1][n][0] + in[2][n][0] +
++                           in[3][n][0] + in[4][n][0] + in[5][n][0];
++            out[1][n][0] = in[0][n][1] + in[1][n][1] + in[2][n][1] +
++                           in[3][n][1] + in[4][n][1] + in[5][n][1];
++            out[0][n][1] = in[6][n][0] + in[7][n][0];
++            out[1][n][1] = in[6][n][1] + in[7][n][1];
++            out[0][n][2] = in[8][n][0] + in[9][n][0];
++            out[1][n][2] = in[8][n][1] + in[9][n][1];
++        }
++        for (i = 0; i < 61; i++) {
++            for (n = 0; n < len; n++) {
++                out[0][n][i+3] = in[i+10][n][0];
++                out[1][n][i+3] = in[i+10][n][1];
++            }
++        }
++    }
++}
++
++/// All-pass filter decay slope
++#define DECAY_SLOPE      0.05f
++/// Number of frequency bands that can be addressed by the parameter index, b(k)
++static const int   NR_PAR_BANDS[]      = { 20, 34 };
++/// Number of frequency bands that can be addressed by the sub subband index, k
++static const int   NR_BANDS[]          = { 71, 91 };
++/// Start frequency band for the all-pass filter decay slope
++static const int   DECAY_CUTOFF[]      = { 10, 32 };
++/// Number of all-pass filer bands
++static const int   NR_ALLPASS_BANDS[]  = { 30, 50 };
++/// First stereo band using the short one sample delay
++static const int   SHORT_DELAY_BAND[]  = { 42, 62 };
++
++/** Table 8.46 */
++static void map_idx_10_to_20(int8_t *par_mapped, const int8_t *par, int full)
++{
++    int b;
++    if (full)
++        b = 9;
++    else {
++        b = 4;
++        par_mapped[10] = 0;
++    }
++    for (; b >= 0; b--) {
++        par_mapped[2*b+1] = par_mapped[2*b] = par[b];
++    }
++}
++
++static void map_idx_34_to_20(int8_t *par_mapped, const int8_t *par, int full)
++{
++    par_mapped[ 0] = (2*par[ 0] +   par[ 1]) / 3;
++    par_mapped[ 1] = (  par[ 1] + 2*par[ 2]) / 3;
++    par_mapped[ 2] = (2*par[ 3] +   par[ 4]) / 3;
++    par_mapped[ 3] = (  par[ 4] + 2*par[ 5]) / 3;
++    par_mapped[ 4] = (  par[ 6] +   par[ 7]) / 2;
++    par_mapped[ 5] = (  par[ 8] +   par[ 9]) / 2;
++    par_mapped[ 6] =    par[10];
++    par_mapped[ 7] =    par[11];
++    par_mapped[ 8] = (  par[12] +   par[13]) / 2;
++    par_mapped[ 9] = (  par[14] +   par[15]) / 2;
++    par_mapped[10] =    par[16];
++    if (full) {
++        par_mapped[11] =    par[17];
++        par_mapped[12] =    par[18];
++        par_mapped[13] =    par[19];
++        par_mapped[14] = (  par[20] +   par[21]) / 2;
++        par_mapped[15] = (  par[22] +   par[23]) / 2;
++        par_mapped[16] = (  par[24] +   par[25]) / 2;
++        par_mapped[17] = (  par[26] +   par[27]) / 2;
++        par_mapped[18] = (  par[28] +   par[29] +   par[30] +   par[31]) / 4;
++        par_mapped[19] = (  par[32] +   par[33]) / 2;
++    }
++}
++
++static void map_val_34_to_20(float par[PS_MAX_NR_IIDICC])
++{
++    par[ 0] = (2*par[ 0] +   par[ 1]) * 0.33333333f;
++    par[ 1] = (  par[ 1] + 2*par[ 2]) * 0.33333333f;
++    par[ 2] = (2*par[ 3] +   par[ 4]) * 0.33333333f;
++    par[ 3] = (  par[ 4] + 2*par[ 5]) * 0.33333333f;
++    par[ 4] = (  par[ 6] +   par[ 7]) * 0.5f;
++    par[ 5] = (  par[ 8] +   par[ 9]) * 0.5f;
++    par[ 6] =    par[10];
++    par[ 7] =    par[11];
++    par[ 8] = (  par[12] +   par[13]) * 0.5f;
++    par[ 9] = (  par[14] +   par[15]) * 0.5f;
++    par[10] =    par[16];
++    par[11] =    par[17];
++    par[12] =    par[18];
++    par[13] =    par[19];
++    par[14] = (  par[20] +   par[21]) * 0.5f;
++    par[15] = (  par[22] +   par[23]) * 0.5f;
++    par[16] = (  par[24] +   par[25]) * 0.5f;
++    par[17] = (  par[26] +   par[27]) * 0.5f;
++    par[18] = (  par[28] +   par[29] +   par[30] +   par[31]) * 0.25f;
++    par[19] = (  par[32] +   par[33]) * 0.5f;
++}
++
++static void map_idx_10_to_34(int8_t *par_mapped, const int8_t *par, int full)
++{
++    if (full) {
++        par_mapped[33] = par[9];
++        par_mapped[32] = par[9];
++        par_mapped[31] = par[9];
++        par_mapped[30] = par[9];
++        par_mapped[29] = par[9];
++        par_mapped[28] = par[9];
++        par_mapped[27] = par[8];
++        par_mapped[26] = par[8];
++        par_mapped[25] = par[8];
++        par_mapped[24] = par[8];
++        par_mapped[23] = par[7];
++        par_mapped[22] = par[7];
++        par_mapped[21] = par[7];
++        par_mapped[20] = par[7];
++        par_mapped[19] = par[6];
++        par_mapped[18] = par[6];
++        par_mapped[17] = par[5];
++        par_mapped[16] = par[5];
++    } else {
++        par_mapped[16] =      0;
++    }
++    par_mapped[15] = par[4];
++    par_mapped[14] = par[4];
++    par_mapped[13] = par[4];
++    par_mapped[12] = par[4];
++    par_mapped[11] = par[3];
++    par_mapped[10] = par[3];
++    par_mapped[ 9] = par[2];
++    par_mapped[ 8] = par[2];
++    par_mapped[ 7] = par[2];
++    par_mapped[ 6] = par[2];
++    par_mapped[ 5] = par[1];
++    par_mapped[ 4] = par[1];
++    par_mapped[ 3] = par[1];
++    par_mapped[ 2] = par[0];
++    par_mapped[ 1] = par[0];
++    par_mapped[ 0] = par[0];
++}
++
++static void map_idx_20_to_34(int8_t *par_mapped, const int8_t *par, int full)
++{
++    if (full) {
++        par_mapped[33] =  par[19];
++        par_mapped[32] =  par[19];
++        par_mapped[31] =  par[18];
++        par_mapped[30] =  par[18];
++        par_mapped[29] =  par[18];
++        par_mapped[28] =  par[18];
++        par_mapped[27] =  par[17];
++        par_mapped[26] =  par[17];
++        par_mapped[25] =  par[16];
++        par_mapped[24] =  par[16];
++        par_mapped[23] =  par[15];
++        par_mapped[22] =  par[15];
++        par_mapped[21] =  par[14];
++        par_mapped[20] =  par[14];
++        par_mapped[19] =  par[13];
++        par_mapped[18] =  par[12];
++        par_mapped[17] =  par[11];
++    }
++    par_mapped[16] =  par[10];
++    par_mapped[15] =  par[ 9];
++    par_mapped[14] =  par[ 9];
++    par_mapped[13] =  par[ 8];
++    par_mapped[12] =  par[ 8];
++    par_mapped[11] =  par[ 7];
++    par_mapped[10] =  par[ 6];
++    par_mapped[ 9] =  par[ 5];
++    par_mapped[ 8] =  par[ 5];
++    par_mapped[ 7] =  par[ 4];
++    par_mapped[ 6] =  par[ 4];
++    par_mapped[ 5] =  par[ 3];
++    par_mapped[ 4] = (par[ 2] + par[ 3]) / 2;
++    par_mapped[ 3] =  par[ 2];
++    par_mapped[ 2] =  par[ 1];
++    par_mapped[ 1] = (par[ 0] + par[ 1]) / 2;
++    par_mapped[ 0] =  par[ 0];
++}
++
++static void map_val_20_to_34(float par[PS_MAX_NR_IIDICC])
++{
++    par[33] =  par[19];
++    par[32] =  par[19];
++    par[31] =  par[18];
++    par[30] =  par[18];
++    par[29] =  par[18];
++    par[28] =  par[18];
++    par[27] =  par[17];
++    par[26] =  par[17];
++    par[25] =  par[16];
++    par[24] =  par[16];
++    par[23] =  par[15];
++    par[22] =  par[15];
++    par[21] =  par[14];
++    par[20] =  par[14];
++    par[19] =  par[13];
++    par[18] =  par[12];
++    par[17] =  par[11];
++    par[16] =  par[10];
++    par[15] =  par[ 9];
++    par[14] =  par[ 9];
++    par[13] =  par[ 8];
++    par[12] =  par[ 8];
++    par[11] =  par[ 7];
++    par[10] =  par[ 6];
++    par[ 9] =  par[ 5];
++    par[ 8] =  par[ 5];
++    par[ 7] =  par[ 4];
++    par[ 6] =  par[ 4];
++    par[ 5] =  par[ 3];
++    par[ 4] = (par[ 2] + par[ 3]) * 0.5f;
++    par[ 3] =  par[ 2];
++    par[ 2] =  par[ 1];
++    par[ 1] = (par[ 0] + par[ 1]) * 0.5f;
++    par[ 0] =  par[ 0];
++}
++
++static void decorrelation(PSContext *ps, float (*out)[32][2], const float (*s)[32][2], int is34)
++{
++    float power[34][PS_QMF_TIME_SLOTS] = {{0}};
++    float transient_gain[34][PS_QMF_TIME_SLOTS];
++    float *peak_decay_nrg = ps->peak_decay_nrg;
++    float *power_smooth = ps->power_smooth;
++    float *peak_decay_diff_smooth = ps->peak_decay_diff_smooth;
++    float (*delay)[PS_QMF_TIME_SLOTS + PS_MAX_DELAY][2] = ps->delay;
++    float (*ap_delay)[PS_AP_LINKS][PS_QMF_TIME_SLOTS + PS_MAX_AP_DELAY][2] = ps->ap_delay;
++    const int8_t *k_to_i = is34 ? k_to_i_34 : k_to_i_20;
++    const float peak_decay_factor = 0.76592833836465f;
++    const float transient_impact  = 1.5f;
++    const float a_smooth          = 0.25f; //< Smoothing coefficient
++    int i, k, m, n;
++    int n0 = 0, nL = 32;
++    static const int link_delay[] = { 3, 4, 5 };
++    static const float a[] = { 0.65143905753106f,
++                               0.56471812200776f,
++                               0.48954165955695f };
++
++    if (is34 != ps->is34bands_old) {
++        memset(ps->peak_decay_nrg,         0, sizeof(ps->peak_decay_nrg));
++        memset(ps->power_smooth,           0, sizeof(ps->power_smooth));
++        memset(ps->peak_decay_diff_smooth, 0, sizeof(ps->peak_decay_diff_smooth));
++        memset(ps->delay,                  0, sizeof(ps->delay));
++        memset(ps->ap_delay,               0, sizeof(ps->ap_delay));
++    }
++
++    for (n = n0; n < nL; n++) {
++        for (k = 0; k < NR_BANDS[is34]; k++) {
++            int i = k_to_i[k];
++            power[i][n] += s[k][n][0] * s[k][n][0] + s[k][n][1] * s[k][n][1];
++        }
++    }
++
++    //Transient detection
++    for (i = 0; i < NR_PAR_BANDS[is34]; i++) {
++        for (n = n0; n < nL; n++) {
++            float decayed_peak = peak_decay_factor * peak_decay_nrg[i];
++            float denom;
++            peak_decay_nrg[i] = FFMAX(decayed_peak, power[i][n]);
++            power_smooth[i] += a_smooth * (power[i][n] - power_smooth[i]);
++            peak_decay_diff_smooth[i] += a_smooth * (peak_decay_nrg[i] - power[i][n] - peak_decay_diff_smooth[i]);
++            denom = transient_impact * peak_decay_diff_smooth[i];
++            transient_gain[i][n]   = (denom > power_smooth[i]) ?
++                                         power_smooth[i] / denom : 1.0f;
++        }
++    }
++
++    //Decorrelation and transient reduction
++    //                         PS_AP_LINKS - 1
++    //                               -----
++    //                                | |  Q_fract_allpass[k][m]*z^-link_delay[m] - a[m]*g_decay_slope[k]
++    //H[k][z] = z^-2 * phi_fract[k] * | | ----------------------------------------------------------------
++    //                                | | 1 - a[m]*g_decay_slope[k]*Q_fract_allpass[k][m]*z^-link_delay[m]
++    //                               m = 0
++    //d[k][z] (out) = transient_gain_mapped[k][z] * H[k][z] * s[k][z]
++    for (k = 0; k < NR_ALLPASS_BANDS[is34]; k++) {
++        int b = k_to_i[k];
++        float g_decay_slope = 1.f - DECAY_SLOPE * (k - DECAY_CUTOFF[is34]);
++        float ag[PS_AP_LINKS];
++        g_decay_slope = av_clipf(g_decay_slope, 0.f, 1.f);
++        memcpy(delay[k], delay[k]+nL, PS_MAX_DELAY*sizeof(delay[k][0]));
++        memcpy(delay[k]+PS_MAX_DELAY, s[k], numQMFSlots*sizeof(delay[k][0]));
++        for (m = 0; m < PS_AP_LINKS; m++) {
++            memcpy(ap_delay[k][m],   ap_delay[k][m]+numQMFSlots,           5*sizeof(ap_delay[k][m][0]));
++            ag[m] = a[m] * g_decay_slope;
++        }
++        for (n = n0; n < nL; n++) {
++            float in_re = delay[k][n+PS_MAX_DELAY-2][0] * phi_fract[is34][k][0] -
++                          delay[k][n+PS_MAX_DELAY-2][1] * phi_fract[is34][k][1];
++            float in_im = delay[k][n+PS_MAX_DELAY-2][0] * phi_fract[is34][k][1] +
++                          delay[k][n+PS_MAX_DELAY-2][1] * phi_fract[is34][k][0];
++            for (m = 0; m < PS_AP_LINKS; m++) {
++                float a_re                = ag[m] * in_re;
++                float a_im                = ag[m] * in_im;
++                float link_delay_re       = ap_delay[k][m][n+5-link_delay[m]][0];
++                float link_delay_im       = ap_delay[k][m][n+5-link_delay[m]][1];
++                float fractional_delay_re = Q_fract_allpass[is34][k][m][0];
++                float fractional_delay_im = Q_fract_allpass[is34][k][m][1];
++                ap_delay[k][m][n+5][0] = in_re;
++                ap_delay[k][m][n+5][1] = in_im;
++                in_re = link_delay_re * fractional_delay_re - link_delay_im * fractional_delay_im - a_re;
++                in_im = link_delay_re * fractional_delay_im + link_delay_im * fractional_delay_re - a_im;
++                ap_delay[k][m][n+5][0] += ag[m] * in_re;
++                ap_delay[k][m][n+5][1] += ag[m] * in_im;
++            }
++            out[k][n][0] = transient_gain[b][n] * in_re;
++            out[k][n][1] = transient_gain[b][n] * in_im;
++        }
++    }
++    for (; k < SHORT_DELAY_BAND[is34]; k++) {
++        memcpy(delay[k], delay[k]+nL, PS_MAX_DELAY*sizeof(delay[k][0]));
++        memcpy(delay[k]+PS_MAX_DELAY, s[k], numQMFSlots*sizeof(delay[k][0]));
++        for (n = n0; n < nL; n++) {
++            //H = delay 14
++            out[k][n][0] = transient_gain[k_to_i[k]][n] * delay[k][n+PS_MAX_DELAY-14][0];
++            out[k][n][1] = transient_gain[k_to_i[k]][n] * delay[k][n+PS_MAX_DELAY-14][1];
++        }
++    }
++    for (; k < NR_BANDS[is34]; k++) {
++        memcpy(delay[k], delay[k]+nL, PS_MAX_DELAY*sizeof(delay[k][0]));
++        memcpy(delay[k]+PS_MAX_DELAY, s[k], numQMFSlots*sizeof(delay[k][0]));
++        for (n = n0; n < nL; n++) {
++            //H = delay 1
++            out[k][n][0] = transient_gain[k_to_i[k]][n] * delay[k][n+PS_MAX_DELAY-1][0];
++            out[k][n][1] = transient_gain[k_to_i[k]][n] * delay[k][n+PS_MAX_DELAY-1][1];
++        }
++    }
++}
++
++static void remap34(int8_t (**p_par_mapped)[PS_MAX_NR_IIDICC],
++                    int8_t           (*par)[PS_MAX_NR_IIDICC],
++                    int num_par, int num_env, int full)
++{
++    int8_t (*par_mapped)[PS_MAX_NR_IIDICC] = *p_par_mapped;
++    int e;
++    if (num_par == 20 || num_par == 11) {
++        for (e = 0; e < num_env; e++) {
++            map_idx_20_to_34(par_mapped[e], par[e], full);
++        }
++    } else if (num_par == 10 || num_par == 5) {
++        for (e = 0; e < num_env; e++) {
++            map_idx_10_to_34(par_mapped[e], par[e], full);
++        }
++    } else {
++        *p_par_mapped = par;
++    }
++}
++
++static void remap20(int8_t (**p_par_mapped)[PS_MAX_NR_IIDICC],
++                    int8_t           (*par)[PS_MAX_NR_IIDICC],
++                    int num_par, int num_env, int full)
++{
++    int8_t (*par_mapped)[PS_MAX_NR_IIDICC] = *p_par_mapped;
++    int e;
++    if (num_par == 34 || num_par == 17) {
++        for (e = 0; e < num_env; e++) {
++            map_idx_34_to_20(par_mapped[e], par[e], full);
++        }
++    } else if (num_par == 10 || num_par == 5) {
++        for (e = 0; e < num_env; e++) {
++            map_idx_10_to_20(par_mapped[e], par[e], full);
++        }
++    } else {
++        *p_par_mapped = par;
++    }
++}
++
++static void stereo_processing(PSContext *ps, float (*l)[32][2], float (*r)[32][2], int is34)
++{
++    int e, b, k, n;
++
++    float (*H11)[PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC] = ps->H11;
++    float (*H12)[PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC] = ps->H12;
++    float (*H21)[PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC] = ps->H21;
++    float (*H22)[PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC] = ps->H22;
++    int8_t *opd_hist = ps->opd_hist;
++    int8_t *ipd_hist = ps->ipd_hist;
++    int8_t iid_mapped_buf[PS_MAX_NUM_ENV][PS_MAX_NR_IIDICC];
++    int8_t icc_mapped_buf[PS_MAX_NUM_ENV][PS_MAX_NR_IIDICC];
++    int8_t ipd_mapped_buf[PS_MAX_NUM_ENV][PS_MAX_NR_IIDICC];
++    int8_t opd_mapped_buf[PS_MAX_NUM_ENV][PS_MAX_NR_IIDICC];
++    int8_t (*iid_mapped)[PS_MAX_NR_IIDICC] = iid_mapped_buf;
++    int8_t (*icc_mapped)[PS_MAX_NR_IIDICC] = icc_mapped_buf;
++    int8_t (*ipd_mapped)[PS_MAX_NR_IIDICC] = ipd_mapped_buf;
++    int8_t (*opd_mapped)[PS_MAX_NR_IIDICC] = opd_mapped_buf;
++    const int8_t *k_to_i = is34 ? k_to_i_34 : k_to_i_20;
++    const float (*H_LUT)[8][4] = (PS_BASELINE || ps->icc_mode < 3) ? HA : HB;
++
++    //Remapping
++    memcpy(H11[0][0], H11[0][ps->num_env_old], PS_MAX_NR_IIDICC*sizeof(H11[0][0][0]));
++    memcpy(H11[1][0], H11[1][ps->num_env_old], PS_MAX_NR_IIDICC*sizeof(H11[1][0][0]));
++    memcpy(H12[0][0], H12[0][ps->num_env_old], PS_MAX_NR_IIDICC*sizeof(H12[0][0][0]));
++    memcpy(H12[1][0], H12[1][ps->num_env_old], PS_MAX_NR_IIDICC*sizeof(H12[1][0][0]));
++    memcpy(H21[0][0], H21[0][ps->num_env_old], PS_MAX_NR_IIDICC*sizeof(H21[0][0][0]));
++    memcpy(H21[1][0], H21[1][ps->num_env_old], PS_MAX_NR_IIDICC*sizeof(H21[1][0][0]));
++    memcpy(H22[0][0], H22[0][ps->num_env_old], PS_MAX_NR_IIDICC*sizeof(H22[0][0][0]));
++    memcpy(H22[1][0], H22[1][ps->num_env_old], PS_MAX_NR_IIDICC*sizeof(H22[1][0][0]));
++    if (is34) {
++        remap34(&iid_mapped, ps->iid_par, ps->nr_iid_par, ps->num_env, 1);
++        remap34(&icc_mapped, ps->icc_par, ps->nr_icc_par, ps->num_env, 1);
++        if (ps->enable_ipdopd) {
++            remap34(&ipd_mapped, ps->ipd_par, ps->nr_ipdopd_par, ps->num_env, 0);
++            remap34(&opd_mapped, ps->opd_par, ps->nr_ipdopd_par, ps->num_env, 0);
++        }
++        if (!ps->is34bands_old) {
++            map_val_20_to_34(H11[0][0]);
++            map_val_20_to_34(H11[1][0]);
++            map_val_20_to_34(H12[0][0]);
++            map_val_20_to_34(H12[1][0]);
++            map_val_20_to_34(H21[0][0]);
++            map_val_20_to_34(H21[1][0]);
++            map_val_20_to_34(H22[0][0]);
++            map_val_20_to_34(H22[1][0]);
++            ipdopd_reset(ipd_hist, opd_hist);
++        }
++    } else {
++        remap20(&iid_mapped, ps->iid_par, ps->nr_iid_par, ps->num_env, 1);
++        remap20(&icc_mapped, ps->icc_par, ps->nr_icc_par, ps->num_env, 1);
++        if (ps->enable_ipdopd) {
++            remap20(&ipd_mapped, ps->ipd_par, ps->nr_ipdopd_par, ps->num_env, 0);
++            remap20(&opd_mapped, ps->opd_par, ps->nr_ipdopd_par, ps->num_env, 0);
++        }
++        if (ps->is34bands_old) {
++            map_val_34_to_20(H11[0][0]);
++            map_val_34_to_20(H11[1][0]);
++            map_val_34_to_20(H12[0][0]);
++            map_val_34_to_20(H12[1][0]);
++            map_val_34_to_20(H21[0][0]);
++            map_val_34_to_20(H21[1][0]);
++            map_val_34_to_20(H22[0][0]);
++            map_val_34_to_20(H22[1][0]);
++            ipdopd_reset(ipd_hist, opd_hist);
++        }
++    }
++
++    //Mixing
++    for (e = 0; e < ps->num_env; e++) {
++        for (b = 0; b < NR_PAR_BANDS[is34]; b++) {
++            float h11, h12, h21, h22;
++            h11 = H_LUT[iid_mapped[e][b] + 7 + 23 * ps->iid_quant][icc_mapped[e][b]][0];
++            h12 = H_LUT[iid_mapped[e][b] + 7 + 23 * ps->iid_quant][icc_mapped[e][b]][1];
++            h21 = H_LUT[iid_mapped[e][b] + 7 + 23 * ps->iid_quant][icc_mapped[e][b]][2];
++            h22 = H_LUT[iid_mapped[e][b] + 7 + 23 * ps->iid_quant][icc_mapped[e][b]][3];
++            if (!PS_BASELINE && ps->enable_ipdopd && b < ps->nr_ipdopd_par) {
++                //The spec say says to only run this smoother when enable_ipdopd
++                //is set but the reference decoder appears to run it constantly
++                float h11i, h12i, h21i, h22i;
++                float ipd_adj_re, ipd_adj_im;
++                int opd_idx = opd_hist[b] * 8 + opd_mapped[e][b];
++                int ipd_idx = ipd_hist[b] * 8 + ipd_mapped[e][b];
++                float opd_re = pd_re_smooth[opd_idx];
++                float opd_im = pd_im_smooth[opd_idx];
++                float ipd_re = pd_re_smooth[ipd_idx];
++                float ipd_im = pd_im_smooth[ipd_idx];
++                opd_hist[b] = opd_idx & 0x3F;
++                ipd_hist[b] = ipd_idx & 0x3F;
++
++                ipd_adj_re = opd_re*ipd_re + opd_im*ipd_im;
++                ipd_adj_im = opd_im*ipd_re - opd_re*ipd_im;
++                h11i = h11 * opd_im;
++                h11  = h11 * opd_re;
++                h12i = h12 * ipd_adj_im;
++                h12  = h12 * ipd_adj_re;
++                h21i = h21 * opd_im;
++                h21  = h21 * opd_re;
++                h22i = h22 * ipd_adj_im;
++                h22  = h22 * ipd_adj_re;
++                H11[1][e+1][b] = h11i;
++                H12[1][e+1][b] = h12i;
++                H21[1][e+1][b] = h21i;
++                H22[1][e+1][b] = h22i;
++            }
++            H11[0][e+1][b] = h11;
++            H12[0][e+1][b] = h12;
++            H21[0][e+1][b] = h21;
++            H22[0][e+1][b] = h22;
++        }
++        for (k = 0; k < NR_BANDS[is34]; k++) {
++            float h11r, h12r, h21r, h22r;
++            float h11i, h12i, h21i, h22i;
++            float h11r_step, h12r_step, h21r_step, h22r_step;
++            float h11i_step, h12i_step, h21i_step, h22i_step;
++            int start = ps->border_position[e];
++            int stop  = ps->border_position[e+1];
++            float width = 1.f / (stop - start);
++            b = k_to_i[k];
++            h11r = H11[0][e][b];
++            h12r = H12[0][e][b];
++            h21r = H21[0][e][b];
++            h22r = H22[0][e][b];
++            if (!PS_BASELINE && ps->enable_ipdopd) {
++            //Is this necessary? ps_04_new seems unchanged
++            if ((is34 && k <= 13 && k >= 9) || (!is34 && k <= 1)) {
++                h11i = -H11[1][e][b];
++                h12i = -H12[1][e][b];
++                h21i = -H21[1][e][b];
++                h22i = -H22[1][e][b];
++            } else {
++                h11i = H11[1][e][b];
++                h12i = H12[1][e][b];
++                h21i = H21[1][e][b];
++                h22i = H22[1][e][b];
++            }
++            }
++            //Interpolation
++            h11r_step = (H11[0][e+1][b] - h11r) * width;
++            h12r_step = (H12[0][e+1][b] - h12r) * width;
++            h21r_step = (H21[0][e+1][b] - h21r) * width;
++            h22r_step = (H22[0][e+1][b] - h22r) * width;
++            if (!PS_BASELINE && ps->enable_ipdopd) {
++                h11i_step = (H11[1][e+1][b] - h11i) * width;
++                h12i_step = (H12[1][e+1][b] - h12i) * width;
++                h21i_step = (H21[1][e+1][b] - h21i) * width;
++                h22i_step = (H22[1][e+1][b] - h22i) * width;
++            }
++            for (n = start + 1; n <= stop; n++) {
++                //l is s, r is d
++                float l_re = l[k][n][0];
++                float l_im = l[k][n][1];
++                float r_re = r[k][n][0];
++                float r_im = r[k][n][1];
++                h11r += h11r_step;
++                h12r += h12r_step;
++                h21r += h21r_step;
++                h22r += h22r_step;
++                if (!PS_BASELINE && ps->enable_ipdopd) {
++                    h11i += h11i_step;
++                    h12i += h12i_step;
++                    h21i += h21i_step;
++                    h22i += h22i_step;
++
++                    l[k][n][0] = h11r*l_re + h21r*r_re - h11i*l_im - h21i*r_im;
++                    l[k][n][1] = h11r*l_im + h21r*r_im + h11i*l_re + h21i*r_re;
++                    r[k][n][0] = h12r*l_re + h22r*r_re - h12i*l_im - h22i*r_im;
++                    r[k][n][1] = h12r*l_im + h22r*r_im + h12i*l_re + h22i*r_re;
++                } else {
++                    l[k][n][0] = h11r*l_re + h21r*r_re;
++                    l[k][n][1] = h11r*l_im + h21r*r_im;
++                    r[k][n][0] = h12r*l_re + h22r*r_re;
++                    r[k][n][1] = h12r*l_im + h22r*r_im;
++                }
++            }
++        }
++    }
++}
++
++int ff_ps_apply(AVCodecContext *avctx, PSContext *ps, float L[2][38][64], float R[2][38][64], int top)
++{
++    float Lbuf[91][32][2];
++    float Rbuf[91][32][2];
++    const int len = 32;
++    int is34 = ps->is34bands;
++
++    top += NR_BANDS[is34] - 64;
++    memset(ps->delay+top, 0, (NR_BANDS[is34] - top)*sizeof(ps->delay[0]));
++    if (top < NR_ALLPASS_BANDS[is34])
++        memset(ps->ap_delay + top, 0, (NR_ALLPASS_BANDS[is34] - top)*sizeof(ps->ap_delay[0]));
++
++    hybrid_analysis(Lbuf, ps->in_buf, L, is34, len);
++    decorrelation(ps, Rbuf, Lbuf, is34);
++    stereo_processing(ps, Lbuf, Rbuf, is34);
++    hybrid_synthesis(L, Lbuf, is34, len);
++    hybrid_synthesis(R, Rbuf, is34, len);
++
++    return 0;
++}
++
++#define PS_INIT_VLC_STATIC(num, size) \
++    INIT_VLC_STATIC(&vlc_ps[num], 9, ps_tmp[num].table_size / ps_tmp[num].elem_size,    \
++                    ps_tmp[num].ps_bits, 1, 1,                                          \
++                    ps_tmp[num].ps_codes, ps_tmp[num].elem_size, ps_tmp[num].elem_size, \
++                    size);
++
++#define PS_VLC_ROW(name) \
++    { name ## _codes, name ## _bits, sizeof(name ## _codes), sizeof(name ## _codes[0]) }
++
++av_cold void ff_ps_init(void) {
++    // Syntax initialization
++    static const struct {
++        const void *ps_codes, *ps_bits;
++        const unsigned int table_size, elem_size;
++    } ps_tmp[] = {
++        PS_VLC_ROW(huff_iid_df1),
++        PS_VLC_ROW(huff_iid_dt1),
++        PS_VLC_ROW(huff_iid_df0),
++        PS_VLC_ROW(huff_iid_dt0),
++        PS_VLC_ROW(huff_icc_df),
++        PS_VLC_ROW(huff_icc_dt),
++        PS_VLC_ROW(huff_ipd_df),
++        PS_VLC_ROW(huff_ipd_dt),
++        PS_VLC_ROW(huff_opd_df),
++        PS_VLC_ROW(huff_opd_dt),
++    };
++
++    PS_INIT_VLC_STATIC(0, 1544);
++    PS_INIT_VLC_STATIC(1,  832);
++    PS_INIT_VLC_STATIC(2, 1024);
++    PS_INIT_VLC_STATIC(3, 1036);
++    PS_INIT_VLC_STATIC(4,  544);
++    PS_INIT_VLC_STATIC(5,  544);
++    PS_INIT_VLC_STATIC(6,  512);
++    PS_INIT_VLC_STATIC(7,  512);
++    PS_INIT_VLC_STATIC(8,  512);
++    PS_INIT_VLC_STATIC(9,  512);
++
++    ps_tableinit();
++}
++
++av_cold void ff_ps_ctx_init(PSContext *ps)
++{
++}
+--- /dev/null
++++ b/libavcodec/aacps.h
+@@ -0,0 +1,82 @@
++/*
++ * MPEG-4 Parametric Stereo definitions and declarations
++ * Copyright (c) 2010 Alex Converse <alex.converse at gmail.com>
++ *
++ * 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
++ */
++
++#ifndef AVCODEC_PS_H
++#define AVCODEC_PS_H
++
++#include <stdint.h>
++
++#include "avcodec.h"
++#include "get_bits.h"
++
++#define PS_MAX_NUM_ENV 5
++#define PS_MAX_NR_IIDICC 34
++#define PS_MAX_NR_IPDOPD 17
++#define PS_MAX_SSB 91
++#define PS_MAX_AP_BANDS 50
++#define PS_QMF_TIME_SLOTS 32
++#define PS_MAX_DELAY 14
++#define PS_AP_LINKS 3
++#define PS_MAX_AP_DELAY 5
++
++typedef struct {
++    int    start;
++    int    enable_iid;
++    int    iid_quant;
++    int    nr_iid_par;
++    int    nr_ipdopd_par;
++    int    enable_icc;
++    int    icc_mode;
++    int    nr_icc_par;
++    int    enable_ext;
++    int    frame_class;
++    int    num_env_old;
++    int    num_env;
++    int    enable_ipdopd;
++    int    border_position[PS_MAX_NUM_ENV+1];
++    int8_t iid_par[PS_MAX_NUM_ENV][PS_MAX_NR_IIDICC]; //<Inter-channel Intensity Difference Parameters
++    int8_t icc_par[PS_MAX_NUM_ENV][PS_MAX_NR_IIDICC]; //<Inter-Channel Coherence Parameters
++    /* ipd/opd is iid/icc sized so that the same functions can handle both */
++    int8_t ipd_par[PS_MAX_NUM_ENV][PS_MAX_NR_IIDICC]; //<Inter-channel Phase Difference Parameters
++    int8_t opd_par[PS_MAX_NUM_ENV][PS_MAX_NR_IIDICC]; //<Overall Phase Difference Parameters
++    int    is34bands;
++    int    is34bands_old;
++
++    float  in_buf[5][44][2];
++    float  delay[PS_MAX_SSB][PS_QMF_TIME_SLOTS + PS_MAX_DELAY][2];
++    float  ap_delay[PS_MAX_AP_BANDS][PS_AP_LINKS][PS_QMF_TIME_SLOTS + PS_MAX_AP_DELAY][2];
++    float  peak_decay_nrg[34];
++    float  power_smooth[34];
++    float  peak_decay_diff_smooth[34];
++    float  H11[2][PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC];
++    float  H12[2][PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC];
++    float  H21[2][PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC];
++    float  H22[2][PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC];
++    int8_t opd_hist[PS_MAX_NR_IIDICC];
++    int8_t ipd_hist[PS_MAX_NR_IIDICC];
++} PSContext;
++
++void ff_ps_init(void);
++void ff_ps_ctx_init(PSContext *ps);
++int ff_ps_read_data(AVCodecContext *avctx, GetBitContext *gb, PSContext *ps, int bits_left);
++int ff_ps_apply(AVCodecContext *avctx, PSContext *ps, float L[2][38][64], float R[2][38][64], int top);
++
++#endif /* AVCODEC_PS_H */
+--- /dev/null
++++ b/libavcodec/aacpsdata.c
+@@ -0,0 +1,163 @@
++/*
++ * MPEG-4 Parametric Stereo data tables
++ * Copyright (c) 2010 Alex Converse <alex.converse at gmail.com>
++ *
++ * 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
++ */
++
++static const uint8_t huff_iid_df1_bits[] = {
++    18, 18, 18, 18, 18, 18, 18, 18, 18, 17, 18, 17, 17, 16, 16, 15, 14, 14,
++    13, 12, 12, 11, 10, 10,  8,  7,  6,  5,  4,  3,  1,  3,  4,  5,  6,  7,
++     8,  9, 10, 11, 11, 12, 13, 14, 14, 15, 16, 16, 17, 17, 18, 17, 18, 18,
++    18, 18, 18, 18, 18, 18, 18,
++};
++
++static const uint32_t huff_iid_df1_codes[] = {
++    0x01FEB4, 0x01FEB5, 0x01FD76, 0x01FD77, 0x01FD74, 0x01FD75, 0x01FE8A,
++    0x01FE8B, 0x01FE88, 0x00FE80, 0x01FEB6, 0x00FE82, 0x00FEB8, 0x007F42,
++    0x007FAE, 0x003FAF, 0x001FD1, 0x001FE9, 0x000FE9, 0x0007EA, 0x0007FB,
++    0x0003FB, 0x0001FB, 0x0001FF, 0x00007C, 0x00003C, 0x00001C, 0x00000C,
++    0x000000, 0x000001, 0x000001, 0x000002, 0x000001, 0x00000D, 0x00001D,
++    0x00003D, 0x00007D, 0x0000FC, 0x0001FC, 0x0003FC, 0x0003F4, 0x0007EB,
++    0x000FEA, 0x001FEA, 0x001FD6, 0x003FD0, 0x007FAF, 0x007F43, 0x00FEB9,
++    0x00FE83, 0x01FEB7, 0x00FE81, 0x01FE89, 0x01FE8E, 0x01FE8F, 0x01FE8C,
++    0x01FE8D, 0x01FEB2, 0x01FEB3, 0x01FEB0, 0x01FEB1,
++};
++
++static const uint8_t huff_iid_dt1_bits[] = {
++    16, 16, 16, 16, 16, 16, 16, 16, 16, 15, 15, 15, 15, 15, 15, 14, 14, 13,
++    13, 13, 12, 12, 11, 10,  9,  9,  7,  6,  5,  3,  1,  2,  5,  6,  7,  8,
++     9, 10, 11, 11, 12, 12, 13, 13, 14, 14, 15, 15, 15, 15, 16, 16, 16, 16,
++    16, 16, 16, 16, 16, 16, 16,
++};
++
++static const uint16_t huff_iid_dt1_codes[] = {
++    0x004ED4, 0x004ED5, 0x004ECE, 0x004ECF, 0x004ECC, 0x004ED6, 0x004ED8,
++    0x004F46, 0x004F60, 0x002718, 0x002719, 0x002764, 0x002765, 0x00276D,
++    0x0027B1, 0x0013B7, 0x0013D6, 0x0009C7, 0x0009E9, 0x0009ED, 0x0004EE,
++    0x0004F7, 0x000278, 0x000139, 0x00009A, 0x00009F, 0x000020, 0x000011,
++    0x00000A, 0x000003, 0x000001, 0x000000, 0x00000B, 0x000012, 0x000021,
++    0x00004C, 0x00009B, 0x00013A, 0x000279, 0x000270, 0x0004EF, 0x0004E2,
++    0x0009EA, 0x0009D8, 0x0013D7, 0x0013D0, 0x0027B2, 0x0027A2, 0x00271A,
++    0x00271B, 0x004F66, 0x004F67, 0x004F61, 0x004F47, 0x004ED9, 0x004ED7,
++    0x004ECD, 0x004ED2, 0x004ED3, 0x004ED0, 0x004ED1,
++};
++
++static const uint8_t huff_iid_df0_bits[] = {
++    17, 17, 17, 17, 16, 15, 13, 10,  9,  7,  6,  5,  4,  3,  1,  3,  4,  5,
++     6,  6,  8, 11, 13, 14, 14, 15, 17, 18, 18,
++};
++
++static const uint32_t huff_iid_df0_codes[] = {
++    0x01FFFB, 0x01FFFC, 0x01FFFD, 0x01FFFA, 0x00FFFC, 0x007FFC, 0x001FFD,
++    0x0003FE, 0x0001FE, 0x00007E, 0x00003C, 0x00001D, 0x00000D, 0x000005,
++    0x000000, 0x000004, 0x00000C, 0x00001C, 0x00003D, 0x00003E, 0x0000FE,
++    0x0007FE, 0x001FFC, 0x003FFC, 0x003FFD, 0x007FFD, 0x01FFFE, 0x03FFFE,
++    0x03FFFF,
++};
++
++static const uint8_t huff_iid_dt0_bits[] = {
++    19, 19, 19, 20, 20, 20, 17, 15, 12, 10,  8,  6,  4,  2,  1,  3,  5,  7,
++     9, 11, 13, 14, 17, 19, 20, 20, 20, 20, 20,
++};
++
++static const uint32_t huff_iid_dt0_codes[] = {
++    0x07FFF9, 0x07FFFA, 0x07FFFB, 0x0FFFF8, 0x0FFFF9, 0x0FFFFA, 0x01FFFD,
++    0x007FFE, 0x000FFE, 0x0003FE, 0x0000FE, 0x00003E, 0x00000E, 0x000002,
++    0x000000, 0x000006, 0x00001E, 0x00007E, 0x0001FE, 0x0007FE, 0x001FFE,
++    0x003FFE, 0x01FFFC, 0x07FFF8, 0x0FFFFB, 0x0FFFFC, 0x0FFFFD, 0x0FFFFE,
++    0x0FFFFF,
++};
++
++static const uint8_t huff_icc_df_bits[] = {
++    14, 14, 12, 10, 7, 5, 3, 1, 2, 4, 6, 8, 9, 11, 13,
++};
++
++static const uint16_t huff_icc_df_codes[] = {
++    0x3FFF, 0x3FFE, 0x0FFE, 0x03FE, 0x007E, 0x001E, 0x0006, 0x0000,
++    0x0002, 0x000E, 0x003E, 0x00FE, 0x01FE, 0x07FE, 0x1FFE,
++};
++
++static const uint8_t huff_icc_dt_bits[] = {
++    14, 13, 11, 9, 7, 5, 3, 1, 2, 4, 6, 8, 10, 12, 14,
++};
++
++static const uint16_t huff_icc_dt_codes[] = {
++    0x3FFE, 0x1FFE, 0x07FE, 0x01FE, 0x007E, 0x001E, 0x0006, 0x0000,
++    0x0002, 0x000E, 0x003E, 0x00FE, 0x03FE, 0x0FFE, 0x3FFF,
++};
++
++static const uint8_t huff_ipd_df_bits[] = {
++    1, 3, 4, 4, 4, 4, 4, 4,
++};
++
++static const uint8_t huff_ipd_df_codes[] = {
++    0x01, 0x00, 0x06, 0x04, 0x02, 0x03, 0x05, 0x07,
++};
++
++static const uint8_t huff_ipd_dt_bits[] = {
++    1, 3, 4, 5, 5, 4, 4, 3,
++};
++
++static const uint8_t huff_ipd_dt_codes[] = {
++    0x01, 0x02, 0x02, 0x03, 0x02, 0x00, 0x03, 0x03,
++};
++
++static const uint8_t huff_opd_df_bits[] = {
++    1, 3, 4, 4, 5, 5, 4, 3,
++};
++
++static const uint8_t huff_opd_df_codes[] = {
++    0x01, 0x01, 0x06, 0x04, 0x0F, 0x0E, 0x05, 0x00,
++};
++
++static const uint8_t huff_opd_dt_bits[] = {
++    1, 3, 4, 5, 5, 4, 4, 3,
++};
++
++static const uint8_t huff_opd_dt_codes[] = {
++    0x01, 0x02, 0x01, 0x07, 0x06, 0x00, 0x02, 0x03,
++};
++
++static const int8_t huff_offset[] = {
++    30, 30,
++    14, 14,
++    7, 7,
++    0, 0,
++    0, 0,
++};
++
++///Table 8.48
++static const int8_t k_to_i_20[] = {
++     1,  0,  0,  1,  2,  3,  4,  5,  6,  7,  8,  9, 10, 11, 12, 13, 14, 14, 15,
++    15, 15, 16, 16, 16, 16, 17, 17, 17, 17, 17, 18, 18, 18, 18, 18, 18, 18, 18,
++    18, 18, 18, 18, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19,
++    19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19
++};
++///Table 8.49
++static const int8_t k_to_i_34[] = {
++     0,  1,  2,  3,  4,  5,  6,  6,  7,  2,  1,  0, 10, 10,  4,  5,  6,  7,  8,
++     9, 10, 11, 12,  9, 14, 11, 12, 13, 14, 15, 16, 13, 16, 17, 18, 19, 20, 21,
++    22, 22, 23, 23, 24, 24, 25, 25, 26, 26, 27, 27, 27, 28, 28, 28, 29, 29, 29,
++    30, 30, 30, 31, 31, 31, 31, 32, 32, 32, 32, 33, 33, 33, 33, 33, 33, 33, 33,
++    33, 33, 33, 33, 33, 33, 33, 33, 33, 33, 33, 33, 33, 33, 33
++};
++
++static const float g1_Q2[] = {
++    0.0f,  0.01899487526049f, 0.0f, -0.07293139167538f,
++    0.0f,  0.30596630545168f, 0.5f
++};
+--- a/libavcodec/sbr.h
++++ b/libavcodec/sbr.h
+@@ -31,6 +31,7 @@
+ 
+ #include <stdint.h>
+ #include "fft.h"
++#include "aacps.h"
+ 
+ /**
+  * Spectral Band Replication header - spectrum parameters that invoke a reset if they differ from the previous header.
+@@ -133,6 +134,7 @@ typedef struct {
+     ///The number of frequency bands in f_master
+     unsigned           n_master;
+     SBRData            data[2];
++    PSContext          ps;
+     ///N_Low and N_High respectively, the number of frequency bands for low and high resolution
+     unsigned           n[2];
+     ///Number of noise floor bands
+@@ -157,7 +159,7 @@ typedef struct {
+     ///QMF output of the HF generator
+     float              X_high[64][40][2];
+     ///QMF values of the reconstructed signal
+-    DECLARE_ALIGNED(16, float, X)[2][2][32][64];
++    DECLARE_ALIGNED(16, float, X)[2][2][38][64];
+     ///Zeroth coefficient used to filter the subband signals
+     float              alpha0[64][2];
+     ///First coefficient used to filter the subband signals
+@@ -176,7 +178,7 @@ typedef struct {
+     float              s_m[7][48];
+     float              gain[7][48];
+     DECLARE_ALIGNED(16, float, qmf_filter_scratch)[5][64];
+-    RDFTContext        rdft;
++    FFTContext         mdct_ana;
+     FFTContext         mdct;
+ } SpectralBandReplication;
+ 
+--- a/libavcodec/Makefile
++++ b/libavcodec/Makefile
+@@ -43,7 +43,7 @@ OBJS-$(CONFIG_VAAPI)                   +
+ OBJS-$(CONFIG_VDPAU)                   += vdpau.o
+ 
+ # decoders/encoders/hardware accelerators
+-OBJS-$(CONFIG_AAC_DECODER)             += aac.o aactab.o aacsbr.o
++OBJS-$(CONFIG_AAC_DECODER)             += aacdec.o aactab.o aacsbr.o aacps.o
+ OBJS-$(CONFIG_AAC_ENCODER)             += aacenc.o aaccoder.o    \
+                                           aacpsy.o aactab.o      \
+                                           psymodel.o iirfilter.o \
+--- a/libavcodec/aacsbr.c
++++ b/libavcodec/aacsbr.c
+@@ -31,6 +31,7 @@
+ #include "aacsbr.h"
+ #include "aacsbrdata.h"
+ #include "fft.h"
++#include "aacps.h"
+ 
+ #include <stdint.h>
+ #include <float.h>
+@@ -71,9 +72,6 @@ enum {
+ static VLC vlc_sbr[10];
+ static const int8_t vlc_sbr_lav[10] =
+     { 60, 60, 24, 24, 31, 31, 12, 12, 31, 12 };
+-static DECLARE_ALIGNED(16, float, analysis_cos_pre)[64];
+-static DECLARE_ALIGNED(16, float, analysis_sin_pre)[64];
+-static DECLARE_ALIGNED(16, float, analysis_cossin_post)[32][2];
+ static const DECLARE_ALIGNED(16, float, zero64)[64];
+ 
+ #define SBR_INIT_VLC_STATIC(num, size) \
+@@ -87,7 +85,7 @@ static const DECLARE_ALIGNED(16, float, 
+ 
+ av_cold void ff_aac_sbr_init(void)
+ {
+-    int n, k;
++    int n;
+     static const struct {
+         const void *sbr_codes, *sbr_bits;
+         const unsigned int table_size, elem_size;
+@@ -116,16 +114,6 @@ av_cold void ff_aac_sbr_init(void)
+     SBR_INIT_VLC_STATIC(8, 592);
+     SBR_INIT_VLC_STATIC(9, 512);
+ 
+-    for (n = 0; n < 64; n++) {
+-        float pre = M_PI * n / 64;
+-        analysis_cos_pre[n] = cosf(pre);
+-        analysis_sin_pre[n] = sinf(pre);
+-    }
+-    for (k = 0; k < 32; k++) {
+-        float post = M_PI * (k + 0.5) / 128;
+-        analysis_cossin_post[k][0] =  4.0 * cosf(post);
+-        analysis_cossin_post[k][1] = -4.0 * sinf(post);
+-    }
+     for (n = 1; n < 320; n++)
+         sbr_qmf_window_us[320 + n] = sbr_qmf_window_us[320 - n];
+     sbr_qmf_window_us[384] = -sbr_qmf_window_us[384];
+@@ -133,6 +121,8 @@ av_cold void ff_aac_sbr_init(void)
+ 
+     for (n = 0; n < 320; n++)
+         sbr_qmf_window_ds[n] = sbr_qmf_window_us[2*n];
++
++    ff_ps_init();
+ }
+ 
+ av_cold void ff_aac_sbr_ctx_init(SpectralBandReplication *sbr)
+@@ -142,13 +132,14 @@ av_cold void ff_aac_sbr_ctx_init(Spectra
+     sbr->data[0].synthesis_filterbank_samples_offset = SBR_SYNTHESIS_BUF_SIZE - (1280 - 128);
+     sbr->data[1].synthesis_filterbank_samples_offset = SBR_SYNTHESIS_BUF_SIZE - (1280 - 128);
+     ff_mdct_init(&sbr->mdct, 7, 1, 1.0/64);
+-    ff_rdft_init(&sbr->rdft, 6, IDFT_R2C);
++    ff_mdct_init(&sbr->mdct_ana, 7, 1, -2.0);
++    ff_ps_ctx_init(&sbr->ps);
+ }
+ 
+ av_cold void ff_aac_sbr_ctx_close(SpectralBandReplication *sbr)
+ {
+     ff_mdct_end(&sbr->mdct);
+-    ff_rdft_end(&sbr->rdft);
++    ff_mdct_end(&sbr->mdct_ana);
+ }
+ 
+ static int qsort_comparison_function_int16(const void *a, const void *b)
+@@ -293,15 +284,15 @@ static void make_bands(int16_t* bands, i
+     bands[num_bands-1] = stop - previous;
+ }
+ 
+-static int check_n_master(AVCodecContext *avccontext, int n_master, int bs_xover_band)
++static int check_n_master(AVCodecContext *avctx, int n_master, int bs_xover_band)
+ {
+     // Requirements (14496-3 sp04 p205)
+     if (n_master <= 0) {
+-        av_log(avccontext, AV_LOG_ERROR, "Invalid n_master: %d\n", n_master);
++        av_log(avctx, AV_LOG_ERROR, "Invalid n_master: %d\n", n_master);
+         return -1;
+     }
+     if (bs_xover_band >= n_master) {
+-        av_log(avccontext, AV_LOG_ERROR,
++        av_log(avctx, AV_LOG_ERROR,
+                "Invalid bitstream, crossover band index beyond array bounds: %d\n",
+                bs_xover_band);
+         return -1;
+@@ -349,7 +340,7 @@ static int sbr_make_f_master(AACContext 
+         sbr_offset_ptr = sbr_offset[5];
+         break;
+     default:
+-        av_log(ac->avccontext, AV_LOG_ERROR,
++        av_log(ac->avctx, AV_LOG_ERROR,
+                "Unsupported sample rate for SBR: %d\n", sbr->sample_rate);
+         return -1;
+     }
+@@ -367,7 +358,7 @@ static int sbr_make_f_master(AACContext 
+     } else if (spectrum->bs_stop_freq == 15) {
+         sbr->k[2] = 3*sbr->k[0];
+     } else {
+-        av_log(ac->avccontext, AV_LOG_ERROR,
++        av_log(ac->avctx, AV_LOG_ERROR,
+                "Invalid bs_stop_freq: %d\n", spectrum->bs_stop_freq);
+         return -1;
+     }
+@@ -382,18 +373,17 @@ static int sbr_make_f_master(AACContext 
+         max_qmf_subbands = 32;
+ 
+     if (sbr->k[2] - sbr->k[0] > max_qmf_subbands) {
+-        av_log(ac->avccontext, AV_LOG_ERROR,
++        av_log(ac->avctx, AV_LOG_ERROR,
+                "Invalid bitstream, too many QMF subbands: %d\n", sbr->k[2] - sbr->k[0]);
+         return -1;
+     }
+ 
+     if (!spectrum->bs_freq_scale) {
+-        unsigned int dk;
+-        int k2diff;
++        int dk, k2diff;
+ 
+         dk = spectrum->bs_alter_scale + 1;
+         sbr->n_master = ((sbr->k[2] - sbr->k[0] + (dk&2)) >> dk) << 1;
+-        if (check_n_master(ac->avccontext, sbr->n_master, sbr->spectrum_params.bs_xover_band))
++        if (check_n_master(ac->avctx, sbr->n_master, sbr->spectrum_params.bs_xover_band))
+             return -1;
+ 
+         for (k = 1; k <= sbr->n_master; k++)
+@@ -428,7 +418,7 @@ static int sbr_make_f_master(AACContext 
+         num_bands_0 = lrintf(half_bands * log2f(sbr->k[1] / (float)sbr->k[0])) * 2;
+ 
+         if (num_bands_0 <= 0) { // Requirements (14496-3 sp04 p205)
+-            av_log(ac->avccontext, AV_LOG_ERROR, "Invalid num_bands_0: %d\n", num_bands_0);
++            av_log(ac->avctx, AV_LOG_ERROR, "Invalid num_bands_0: %d\n", num_bands_0);
+             return -1;
+         }
+ 
+@@ -442,7 +432,7 @@ static int sbr_make_f_master(AACContext 
+         vk0[0] = sbr->k[0];
+         for (k = 1; k <= num_bands_0; k++) {
+             if (vk0[k] <= 0) { // Requirements (14496-3 sp04 p205)
+-                av_log(ac->avccontext, AV_LOG_ERROR, "Invalid vDk0[%d]: %d\n", k, vk0[k]);
++                av_log(ac->avctx, AV_LOG_ERROR, "Invalid vDk0[%d]: %d\n", k, vk0[k]);
+                 return -1;
+             }
+             vk0[k] += vk0[k-1];
+@@ -472,14 +462,14 @@ static int sbr_make_f_master(AACContext 
+             vk1[0] = sbr->k[1];
+             for (k = 1; k <= num_bands_1; k++) {
+                 if (vk1[k] <= 0) { // Requirements (14496-3 sp04 p205)
+-                    av_log(ac->avccontext, AV_LOG_ERROR, "Invalid vDk1[%d]: %d\n", k, vk1[k]);
++                    av_log(ac->avctx, AV_LOG_ERROR, "Invalid vDk1[%d]: %d\n", k, vk1[k]);
+                     return -1;
+                 }
+                 vk1[k] += vk1[k-1];
+             }
+ 
+             sbr->n_master = num_bands_0 + num_bands_1;
+-            if (check_n_master(ac->avccontext, sbr->n_master, sbr->spectrum_params.bs_xover_band))
++            if (check_n_master(ac->avctx, sbr->n_master, sbr->spectrum_params.bs_xover_band))
+                 return -1;
+             memcpy(&sbr->f_master[0],               vk0,
+                    (num_bands_0 + 1) * sizeof(sbr->f_master[0]));
+@@ -488,7 +478,7 @@ static int sbr_make_f_master(AACContext 
+ 
+         } else {
+             sbr->n_master = num_bands_0;
+-            if (check_n_master(ac->avccontext, sbr->n_master, sbr->spectrum_params.bs_xover_band))
++            if (check_n_master(ac->avctx, sbr->n_master, sbr->spectrum_params.bs_xover_band))
+                 return -1;
+             memcpy(sbr->f_master, vk0, (num_bands_0 + 1) * sizeof(sbr->f_master[0]));
+         }
+@@ -524,7 +514,7 @@ static int sbr_hf_calc_npatches(AACConte
+         // illegal however the Coding Technologies decoder check stream has a final
+         // count of 6 patches
+         if (sbr->num_patches > 5) {
+-            av_log(ac->avccontext, AV_LOG_ERROR, "Too many patches: %d\n", sbr->num_patches);
++            av_log(ac->avctx, AV_LOG_ERROR, "Too many patches: %d\n", sbr->num_patches);
+             return -1;
+         }
+ 
+@@ -563,12 +553,12 @@ static int sbr_make_f_derived(AACContext
+ 
+     // Requirements (14496-3 sp04 p205)
+     if (sbr->kx[1] + sbr->m[1] > 64) {
+-        av_log(ac->avccontext, AV_LOG_ERROR,
++        av_log(ac->avctx, AV_LOG_ERROR,
+                "Stop frequency border too high: %d\n", sbr->kx[1] + sbr->m[1]);
+         return -1;
+     }
+     if (sbr->kx[1] > 32) {
+-        av_log(ac->avccontext, AV_LOG_ERROR, "Start frequency border too high: %d\n", sbr->kx[1]);
++        av_log(ac->avctx, AV_LOG_ERROR, "Start frequency border too high: %d\n", sbr->kx[1]);
+         return -1;
+     }
+ 
+@@ -580,7 +570,7 @@ static int sbr_make_f_derived(AACContext
+     sbr->n_q = FFMAX(1, lrintf(sbr->spectrum_params.bs_noise_bands *
+                                log2f(sbr->k[2] / (float)sbr->kx[1]))); // 0 <= bs_noise_bands <= 3
+     if (sbr->n_q > 5) {
+-        av_log(ac->avccontext, AV_LOG_ERROR, "Too many noise floor scale factors: %d\n", sbr->n_q);
++        av_log(ac->avctx, AV_LOG_ERROR, "Too many noise floor scale factors: %d\n", sbr->n_q);
+         return -1;
+     }
+ 
+@@ -638,7 +628,7 @@ static int read_sbr_grid(AACContext *ac,
+             ch_data->bs_amp_res = 0;
+ 
+         if (ch_data->bs_num_env > 4) {
+-            av_log(ac->avccontext, AV_LOG_ERROR,
++            av_log(ac->avctx, AV_LOG_ERROR,
+                    "Invalid bitstream, too many SBR envelopes in FIXFIX type SBR frame: %d\n",
+                    ch_data->bs_num_env);
+             return -1;
+@@ -693,7 +683,7 @@ static int read_sbr_grid(AACContext *ac,
+         ch_data->bs_num_env                 = num_rel_lead + num_rel_trail + 1;
+ 
+         if (ch_data->bs_num_env > 5) {
+-            av_log(ac->avccontext, AV_LOG_ERROR,
++            av_log(ac->avctx, AV_LOG_ERROR,
+                    "Invalid bitstream, too many SBR envelopes in VARVAR type SBR frame: %d\n",
+                    ch_data->bs_num_env);
+             return -1;
+@@ -714,7 +704,7 @@ static int read_sbr_grid(AACContext *ac,
+     }
+ 
+     if (bs_pointer > ch_data->bs_num_env + 1) {
+-        av_log(ac->avccontext, AV_LOG_ERROR,
++        av_log(ac->avctx, AV_LOG_ERROR,
+                "Invalid bitstream, bs_pointer points to a middle noise border outside the time borders table: %d\n",
+                bs_pointer);
+         return -1;
+@@ -722,7 +712,7 @@ static int read_sbr_grid(AACContext *ac,
+ 
+     for (i = 1; i <= ch_data->bs_num_env; i++) {
+         if (ch_data->t_env[i-1] > ch_data->t_env[i]) {
+-            av_log(ac->avccontext, AV_LOG_ERROR, "Non monotone time borders\n");
++            av_log(ac->avctx, AV_LOG_ERROR, "Non monotone time borders\n");
+             return -1;
+         }
+     }
+@@ -903,25 +893,24 @@ static void read_sbr_extension(AACContex
+                                GetBitContext *gb,
+                                int bs_extension_id, int *num_bits_left)
+ {
+-//TODO - implement ps_data for parametric stereo parsing
+     switch (bs_extension_id) {
+     case EXTENSION_ID_PS:
+         if (!ac->m4ac.ps) {
+-            av_log(ac->avccontext, AV_LOG_ERROR, "Parametric Stereo signaled to be not-present but was found in the bitstream.\n");
++            av_log(ac->avctx, AV_LOG_ERROR, "Parametric Stereo signaled to be not-present but was found in the bitstream.\n");
+             skip_bits_long(gb, *num_bits_left); // bs_fill_bits
+             *num_bits_left = 0;
+         } else {
+-#if 0
+-            *num_bits_left -= ff_ps_data(gb, ps);
++#if 1
++            *num_bits_left -= ff_ps_read_data(ac->avctx, gb, &sbr->ps, *num_bits_left);
+ #else
+-            av_log_missing_feature(ac->avccontext, "Parametric Stereo is", 0);
++            av_log_missing_feature(ac->avctx, "Parametric Stereo is", 0);
+             skip_bits_long(gb, *num_bits_left); // bs_fill_bits
+             *num_bits_left = 0;
+ #endif
+         }
+         break;
+     default:
+-        av_log_missing_feature(ac->avccontext, "Reserved SBR extensions are", 1);
++        av_log_missing_feature(ac->avctx, "Reserved SBR extensions are", 1);
+         skip_bits_long(gb, *num_bits_left); // bs_fill_bits
+         *num_bits_left = 0;
+         break;
+@@ -1006,7 +995,7 @@ static unsigned int read_sbr_data(AACCon
+             return get_bits_count(gb) - cnt;
+         }
+     } else {
+-        av_log(ac->avccontext, AV_LOG_ERROR,
++        av_log(ac->avctx, AV_LOG_ERROR,
+             "Invalid bitstream - cannot apply SBR to element type %d\n", id_aac);
+         sbr->start = 0;
+         return get_bits_count(gb) - cnt;
+@@ -1021,6 +1010,11 @@ static unsigned int read_sbr_data(AACCon
+             num_bits_left -= 2;
+             read_sbr_extension(ac, sbr, gb, get_bits(gb, 2), &num_bits_left); // bs_extension_id
+         }
++        if (num_bits_left < 0) {
++            av_log(ac->avctx, AV_LOG_ERROR, "SBR Extension over read.\n");
++        }
++        if (num_bits_left > 0)
++            skip_bits(gb, num_bits_left);
+     }
+ 
+     return get_bits_count(gb) - cnt;
+@@ -1033,7 +1027,7 @@ static void sbr_reset(AACContext *ac, Sp
+     if (err >= 0)
+         err = sbr_make_f_derived(ac, sbr);
+     if (err < 0) {
+-        av_log(ac->avccontext, AV_LOG_ERROR,
++        av_log(ac->avctx, AV_LOG_ERROR,
+                "SBR reset failed. Switching SBR to pure upsampling mode.\n");
+         sbr->start = 0;
+     }
+@@ -1085,7 +1079,7 @@ int ff_decode_sbr_extension(AACContext *
+     bytes_read = ((num_sbr_bits + num_align_bits + 4) >> 3);
+ 
+     if (bytes_read > cnt) {
+-        av_log(ac->avccontext, AV_LOG_ERROR,
++        av_log(ac->avctx, AV_LOG_ERROR,
+                "Expected to read %d SBR bytes actually read %d.\n", cnt, bytes_read);
+     }
+     return cnt;
+@@ -1139,7 +1133,7 @@ static void sbr_dequant(SpectralBandRepl
+  * @param   x       pointer to the beginning of the first sample window
+  * @param   W       array of complex-valued samples split into subbands
+  */
+-static void sbr_qmf_analysis(DSPContext *dsp, RDFTContext *rdft, const float *in, float *x,
++static void sbr_qmf_analysis(DSPContext *dsp, FFTContext *mdct, const float *in, float *x,
+                              float z[320], float W[2][32][32][2],
+                              float scale)
+ {
+@@ -1152,23 +1146,23 @@ static void sbr_qmf_analysis(DSPContext 
+         memcpy(x+288, in, 1024*sizeof(*x));
+     for (i = 0; i < 32; i++) { // numTimeSlots*RATE = 16*2 as 960 sample frames
+                                // are not supported
+-        float re, im;
+         dsp->vector_fmul_reverse(z, sbr_qmf_window_ds, x, 320);
+         for (k = 0; k < 64; k++) {
+             float f = z[k] + z[k + 64] + z[k + 128] + z[k + 192] + z[k + 256];
+-            z[k] = f * analysis_cos_pre[k];
+-            z[k+64] = f;
++            z[k] = f;
+         }
+-        ff_rdft_calc(rdft, z);
+-        re = z[0] * 0.5f;
+-        im = 0.5f * dsp->scalarproduct_float(z+64, analysis_sin_pre, 64);
+-        W[1][i][0][0] = re * analysis_cossin_post[0][0] - im * analysis_cossin_post[0][1];
+-        W[1][i][0][1] = re * analysis_cossin_post[0][1] + im * analysis_cossin_post[0][0];
++        //Shuffle to IMDCT
++        z[64] = z[0];
+         for (k = 1; k < 32; k++) {
+-            re = z[2*k  ] - re;
+-            im = z[2*k+1] - im;
+-            W[1][i][k][0] = re * analysis_cossin_post[k][0] - im * analysis_cossin_post[k][1];
+-            W[1][i][k][1] = re * analysis_cossin_post[k][1] + im * analysis_cossin_post[k][0];
++            z[64+2*k-1] =  z[   k];
++            z[64+2*k  ] = -z[64-k];
++        }
++        z[64+63] = z[32];
++
++        ff_imdct_half(mdct, z, z+64);
++        for (k = 0; k < 32; k++) {
++            W[1][i][k][0] = -z[63-k];
++            W[1][i][k][1] = z[k];
+         }
+         x += 32;
+     }
+@@ -1179,7 +1173,7 @@ static void sbr_qmf_analysis(DSPContext 
+  * (14496-3 sp04 p206)
+  */
+ static void sbr_qmf_synthesis(DSPContext *dsp, FFTContext *mdct,
+-                              float *out, float X[2][32][64],
++                              float *out, float X[2][38][64],
+                               float mdct_buf[2][64],
+                               float *v0, int *v_off, const unsigned int div,
+                               float bias, float scale)
+@@ -1197,21 +1191,22 @@ static void sbr_qmf_synthesis(DSPContext
+             *v_off -= 128 >> div;
+         }
+         v = v0 + *v_off;
+-        for (n = 1; n < 64 >> div; n+=2) {
+-            X[1][i][n] = -X[1][i][n];
+-        }
+-        if (div) {
+-            memset(X[0][i]+32, 0, 32*sizeof(float));
+-            memset(X[1][i]+32, 0, 32*sizeof(float));
+-        }
+-        ff_imdct_half(mdct, mdct_buf[0], X[0][i]);
+-        ff_imdct_half(mdct, mdct_buf[1], X[1][i]);
+         if (div) {
+             for (n = 0; n < 32; n++) {
+-                v[      n] = -mdct_buf[0][63 - 2*n] + mdct_buf[1][2*n    ];
+-                v[ 63 - n] =  mdct_buf[0][62 - 2*n] + mdct_buf[1][2*n + 1];
++                X[0][i][   n] = -X[0][i][n];
++                X[0][i][32+n] =  X[1][i][31-n];
++            }
++            ff_imdct_half(mdct, mdct_buf[0], X[0][i]);
++            for (n = 0; n < 32; n++) {
++                v[     n] =  mdct_buf[0][63 - 2*n];
++                v[63 - n] = -mdct_buf[0][62 - 2*n];
+             }
+         } else {
++            for (n = 1; n < 64; n+=2) {
++                X[1][i][n] = -X[1][i][n];
++            }
++            ff_imdct_half(mdct, mdct_buf[0], X[0][i]);
++            ff_imdct_half(mdct, mdct_buf[1], X[1][i]);
+             for (n = 0; n < 64; n++) {
+                 v[      n] = -mdct_buf[0][63 -   n] + mdct_buf[1][  n    ];
+                 v[127 - n] =  mdct_buf[0][63 -   n] + mdct_buf[1][  n    ];
+@@ -1380,7 +1375,7 @@ static int sbr_hf_gen(AACContext *ac, Sp
+             g--;
+ 
+             if (g < 0) {
+-                av_log(ac->avccontext, AV_LOG_ERROR,
++                av_log(ac->avctx, AV_LOG_ERROR,
+                        "ERROR : no subband found for frequency %d\n", k);
+                 return -1;
+             }
+@@ -1414,7 +1409,7 @@ static int sbr_hf_gen(AACContext *ac, Sp
+ }
+ 
+ /// Generate the subband filtered lowband
+-static int sbr_x_gen(SpectralBandReplication *sbr, float X[2][32][64],
++static int sbr_x_gen(SpectralBandReplication *sbr, float X[2][38][64],
+                      const float X_low[32][40][2], const float Y[2][38][64][2],
+                      int ch)
+ {
+@@ -1436,7 +1431,7 @@ static int sbr_x_gen(SpectralBandReplica
+     }
+ 
+     for (k = 0; k < sbr->kx[1]; k++) {
+-        for (i = i_Temp; i < i_f; i++) {
++        for (i = i_Temp; i < 38; i++) {
+             X[0][i][k] = X_low[k][i + ENVELOPE_ADJUSTMENT_OFFSET][0];
+             X[1][i][k] = X_low[k][i + ENVELOPE_ADJUSTMENT_OFFSET][1];
+         }
+@@ -1730,7 +1725,7 @@ void ff_sbr_apply(AACContext *ac, Spectr
+     }
+     for (ch = 0; ch < nch; ch++) {
+         /* decode channel */
+-        sbr_qmf_analysis(&ac->dsp, &sbr->rdft, ch ? R : L, sbr->data[ch].analysis_filterbank_samples,
++        sbr_qmf_analysis(&ac->dsp, &sbr->mdct_ana, ch ? R : L, sbr->data[ch].analysis_filterbank_samples,
+                          (float*)sbr->qmf_filter_scratch,
+                          sbr->data[ch].W, 1/(-1024 * ac->sf_scale));
+         sbr_lf_gen(ac, sbr, sbr->X_low, sbr->data[ch].W);
+@@ -1752,6 +1747,16 @@ void ff_sbr_apply(AACContext *ac, Spectr
+         /* synthesis */
+         sbr_x_gen(sbr, sbr->X[ch], sbr->X_low, sbr->data[ch].Y, ch);
+     }
++
++    if (ac->m4ac.ps == 1) {
++        if (sbr->ps.start) {
++            ff_ps_apply(ac->avctx, &sbr->ps, sbr->X[0], sbr->X[1], sbr->kx[1] + sbr->m[1]);
++        } else {
++            memcpy(sbr->X[1], sbr->X[0], sizeof(sbr->X[0]));
++        }
++        nch = 2;
++    }
++
+     sbr_qmf_synthesis(&ac->dsp, &sbr->mdct, L, sbr->X[0], sbr->qmf_filter_scratch,
+                       sbr->data[0].synthesis_filterbank_samples,
+                       &sbr->data[0].synthesis_filterbank_samples_offset,
+--- a/libavcodec/aactab.c
++++ b/libavcodec/aactab.c
+@@ -29,6 +29,7 @@
+ 
+ #include "libavutil/mem.h"
+ #include "aac.h"
++#include "aac_tablegen.h"
+ 
+ #include <stdint.h>
+ 
+@@ -1204,129 +1205,3 @@ const uint8_t ff_tns_max_bands_128[] = {
+     9, 9, 10, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14
+ };
+ // @}
+-
+-
+-#if CONFIG_HARDCODED_TABLES
+-
+-/**
+- * Table of pow(2, (i - 200)/4.) used for different purposes depending on the
+- * range of indices to the table:
+- * [ 0, 255] scale factor decoding when using C dsp.float_to_int16
+- * [60, 315] scale factor decoding when using SIMD dsp.float_to_int16
+- * [45, 300] intensity stereo position decoding mapped in reverse order i.e. 0->300, 1->299, ..., 254->46, 255->45
+- */
+-const float ff_aac_pow2sf_tab[428] = {
+-    8.88178420e-16, 1.05622810e-15, 1.25607397e-15, 1.49373210e-15,
+-    1.77635684e-15, 2.11245619e-15, 2.51214793e-15, 2.98746420e-15,
+-    3.55271368e-15, 4.22491238e-15, 5.02429587e-15, 5.97492839e-15,
+-    7.10542736e-15, 8.44982477e-15, 1.00485917e-14, 1.19498568e-14,
+-    1.42108547e-14, 1.68996495e-14, 2.00971835e-14, 2.38997136e-14,
+-    2.84217094e-14, 3.37992991e-14, 4.01943669e-14, 4.77994272e-14,
+-    5.68434189e-14, 6.75985982e-14, 8.03887339e-14, 9.55988543e-14,
+-    1.13686838e-13, 1.35197196e-13, 1.60777468e-13, 1.91197709e-13,
+-    2.27373675e-13, 2.70394393e-13, 3.21554936e-13, 3.82395417e-13,
+-    4.54747351e-13, 5.40788785e-13, 6.43109871e-13, 7.64790834e-13,
+-    9.09494702e-13, 1.08157757e-12, 1.28621974e-12, 1.52958167e-12,
+-    1.81898940e-12, 2.16315514e-12, 2.57243948e-12, 3.05916334e-12,
+-    3.63797881e-12, 4.32631028e-12, 5.14487897e-12, 6.11832668e-12,
+-    7.27595761e-12, 8.65262056e-12, 1.02897579e-11, 1.22366534e-11,
+-    1.45519152e-11, 1.73052411e-11, 2.05795159e-11, 2.44733067e-11,
+-    2.91038305e-11, 3.46104823e-11, 4.11590317e-11, 4.89466134e-11,
+-    5.82076609e-11, 6.92209645e-11, 8.23180635e-11, 9.78932268e-11,
+-    1.16415322e-10, 1.38441929e-10, 1.64636127e-10, 1.95786454e-10,
+-    2.32830644e-10, 2.76883858e-10, 3.29272254e-10, 3.91572907e-10,
+-    4.65661287e-10, 5.53767716e-10, 6.58544508e-10, 7.83145814e-10,
+-    9.31322575e-10, 1.10753543e-09, 1.31708902e-09, 1.56629163e-09,
+-    1.86264515e-09, 2.21507086e-09, 2.63417803e-09, 3.13258326e-09,
+-    3.72529030e-09, 4.43014173e-09, 5.26835606e-09, 6.26516652e-09,
+-    7.45058060e-09, 8.86028346e-09, 1.05367121e-08, 1.25303330e-08,
+-    1.49011612e-08, 1.77205669e-08, 2.10734243e-08, 2.50606661e-08,
+-    2.98023224e-08, 3.54411338e-08, 4.21468485e-08, 5.01213321e-08,
+-    5.96046448e-08, 7.08822677e-08, 8.42936970e-08, 1.00242664e-07,
+-    1.19209290e-07, 1.41764535e-07, 1.68587394e-07, 2.00485328e-07,
+-    2.38418579e-07, 2.83529071e-07, 3.37174788e-07, 4.00970657e-07,
+-    4.76837158e-07, 5.67058141e-07, 6.74349576e-07, 8.01941314e-07,
+-    9.53674316e-07, 1.13411628e-06, 1.34869915e-06, 1.60388263e-06,
+-    1.90734863e-06, 2.26823256e-06, 2.69739830e-06, 3.20776526e-06,
+-    3.81469727e-06, 4.53646513e-06, 5.39479661e-06, 6.41553051e-06,
+-    7.62939453e-06, 9.07293026e-06, 1.07895932e-05, 1.28310610e-05,
+-    1.52587891e-05, 1.81458605e-05, 2.15791864e-05, 2.56621220e-05,
+-    3.05175781e-05, 3.62917210e-05, 4.31583729e-05, 5.13242441e-05,
+-    6.10351562e-05, 7.25834421e-05, 8.63167458e-05, 1.02648488e-04,
+-    1.22070312e-04, 1.45166884e-04, 1.72633492e-04, 2.05296976e-04,
+-    2.44140625e-04, 2.90333768e-04, 3.45266983e-04, 4.10593953e-04,
+-    4.88281250e-04, 5.80667537e-04, 6.90533966e-04, 8.21187906e-04,
+-    9.76562500e-04, 1.16133507e-03, 1.38106793e-03, 1.64237581e-03,
+-    1.95312500e-03, 2.32267015e-03, 2.76213586e-03, 3.28475162e-03,
+-    3.90625000e-03, 4.64534029e-03, 5.52427173e-03, 6.56950324e-03,
+-    7.81250000e-03, 9.29068059e-03, 1.10485435e-02, 1.31390065e-02,
+-    1.56250000e-02, 1.85813612e-02, 2.20970869e-02, 2.62780130e-02,
+-    3.12500000e-02, 3.71627223e-02, 4.41941738e-02, 5.25560260e-02,
+-    6.25000000e-02, 7.43254447e-02, 8.83883476e-02, 1.05112052e-01,
+-    1.25000000e-01, 1.48650889e-01, 1.76776695e-01, 2.10224104e-01,
+-    2.50000000e-01, 2.97301779e-01, 3.53553391e-01, 4.20448208e-01,
+-    5.00000000e-01, 5.94603558e-01, 7.07106781e-01, 8.40896415e-01,
+-    1.00000000e+00, 1.18920712e+00, 1.41421356e+00, 1.68179283e+00,
+-    2.00000000e+00, 2.37841423e+00, 2.82842712e+00, 3.36358566e+00,
+-    4.00000000e+00, 4.75682846e+00, 5.65685425e+00, 6.72717132e+00,
+-    8.00000000e+00, 9.51365692e+00, 1.13137085e+01, 1.34543426e+01,
+-    1.60000000e+01, 1.90273138e+01, 2.26274170e+01, 2.69086853e+01,
+-    3.20000000e+01, 3.80546277e+01, 4.52548340e+01, 5.38173706e+01,
+-    6.40000000e+01, 7.61092554e+01, 9.05096680e+01, 1.07634741e+02,
+-    1.28000000e+02, 1.52218511e+02, 1.81019336e+02, 2.15269482e+02,
+-    2.56000000e+02, 3.04437021e+02, 3.62038672e+02, 4.30538965e+02,
+-    5.12000000e+02, 6.08874043e+02, 7.24077344e+02, 8.61077929e+02,
+-    1.02400000e+03, 1.21774809e+03, 1.44815469e+03, 1.72215586e+03,
+-    2.04800000e+03, 2.43549617e+03, 2.89630938e+03, 3.44431172e+03,
+-    4.09600000e+03, 4.87099234e+03, 5.79261875e+03, 6.88862343e+03,
+-    8.19200000e+03, 9.74198469e+03, 1.15852375e+04, 1.37772469e+04,
+-    1.63840000e+04, 1.94839694e+04, 2.31704750e+04, 2.75544937e+04,
+-    3.27680000e+04, 3.89679387e+04, 4.63409500e+04, 5.51089875e+04,
+-    6.55360000e+04, 7.79358775e+04, 9.26819000e+04, 1.10217975e+05,
+-    1.31072000e+05, 1.55871755e+05, 1.85363800e+05, 2.20435950e+05,
+-    2.62144000e+05, 3.11743510e+05, 3.70727600e+05, 4.40871900e+05,
+-    5.24288000e+05, 6.23487020e+05, 7.41455200e+05, 8.81743800e+05,
+-    1.04857600e+06, 1.24697404e+06, 1.48291040e+06, 1.76348760e+06,
+-    2.09715200e+06, 2.49394808e+06, 2.96582080e+06, 3.52697520e+06,
+-    4.19430400e+06, 4.98789616e+06, 5.93164160e+06, 7.05395040e+06,
+-    8.38860800e+06, 9.97579232e+06, 1.18632832e+07, 1.41079008e+07,
+-    1.67772160e+07, 1.99515846e+07, 2.37265664e+07, 2.82158016e+07,
+-    3.35544320e+07, 3.99031693e+07, 4.74531328e+07, 5.64316032e+07,
+-    6.71088640e+07, 7.98063385e+07, 9.49062656e+07, 1.12863206e+08,
+-    1.34217728e+08, 1.59612677e+08, 1.89812531e+08, 2.25726413e+08,
+-    2.68435456e+08, 3.19225354e+08, 3.79625062e+08, 4.51452825e+08,
+-    5.36870912e+08, 6.38450708e+08, 7.59250125e+08, 9.02905651e+08,
+-    1.07374182e+09, 1.27690142e+09, 1.51850025e+09, 1.80581130e+09,
+-    2.14748365e+09, 2.55380283e+09, 3.03700050e+09, 3.61162260e+09,
+-    4.29496730e+09, 5.10760567e+09, 6.07400100e+09, 7.22324521e+09,
+-    8.58993459e+09, 1.02152113e+10, 1.21480020e+10, 1.44464904e+10,
+-    1.71798692e+10, 2.04304227e+10, 2.42960040e+10, 2.88929808e+10,
+-    3.43597384e+10, 4.08608453e+10, 4.85920080e+10, 5.77859616e+10,
+-    6.87194767e+10, 8.17216907e+10, 9.71840160e+10, 1.15571923e+11,
+-    1.37438953e+11, 1.63443381e+11, 1.94368032e+11, 2.31143847e+11,
+-    2.74877907e+11, 3.26886763e+11, 3.88736064e+11, 4.62287693e+11,
+-    5.49755814e+11, 6.53773525e+11, 7.77472128e+11, 9.24575386e+11,
+-    1.09951163e+12, 1.30754705e+12, 1.55494426e+12, 1.84915077e+12,
+-    2.19902326e+12, 2.61509410e+12, 3.10988851e+12, 3.69830155e+12,
+-    4.39804651e+12, 5.23018820e+12, 6.21977702e+12, 7.39660309e+12,
+-    8.79609302e+12, 1.04603764e+13, 1.24395540e+13, 1.47932062e+13,
+-    1.75921860e+13, 2.09207528e+13, 2.48791081e+13, 2.95864124e+13,
+-    3.51843721e+13, 4.18415056e+13, 4.97582162e+13, 5.91728247e+13,
+-    7.03687442e+13, 8.36830112e+13, 9.95164324e+13, 1.18345649e+14,
+-    1.40737488e+14, 1.67366022e+14, 1.99032865e+14, 2.36691299e+14,
+-    2.81474977e+14, 3.34732045e+14, 3.98065730e+14, 4.73382598e+14,
+-    5.62949953e+14, 6.69464090e+14, 7.96131459e+14, 9.46765196e+14,
+-    1.12589991e+15, 1.33892818e+15, 1.59226292e+15, 1.89353039e+15,
+-    2.25179981e+15, 2.67785636e+15, 3.18452584e+15, 3.78706078e+15,
+-    4.50359963e+15, 5.35571272e+15, 6.36905167e+15, 7.57412156e+15,
+-    9.00719925e+15, 1.07114254e+16, 1.27381033e+16, 1.51482431e+16,
+-    1.80143985e+16, 2.14228509e+16, 2.54762067e+16, 3.02964863e+16,
+-    3.60287970e+16, 4.28457018e+16, 5.09524134e+16, 6.05929725e+16,
+-    7.20575940e+16, 8.56914035e+16, 1.01904827e+17, 1.21185945e+17,
+-};
+-
+-#else
+-
+-float ff_aac_pow2sf_tab[428];
+-
+-#endif /* CONFIG_HARDCODED_TABLES */
+--- a/libavcodec/aactab.h
++++ b/libavcodec/aactab.h
+@@ -32,6 +32,7 @@
+ 
+ #include "libavutil/mem.h"
+ #include "aac.h"
++#include "aac_tablegen_decl.h"
+ 
+ #include <stdint.h>
+ 
+@@ -73,10 +74,4 @@ extern const uint16_t * const ff_swb_off
+ extern const uint8_t ff_tns_max_bands_1024[13];
+ extern const uint8_t ff_tns_max_bands_128 [13];
+ 
+-#if CONFIG_HARDCODED_TABLES
+-extern const float ff_aac_pow2sf_tab[428];
+-#else
+-extern       float ff_aac_pow2sf_tab[428];
+-#endif /* CONFIG_HARDCODED_TABLES */
+-
+ #endif /* AVCODEC_AACTAB_H */
+--- /dev/null
++++ b/libavcodec/aac_tablegen.c
+@@ -0,0 +1,39 @@
++/*
++ * Generate a header file for hardcoded AAC tables
++ *
++ * Copyright (c) 2010 Alex Converse <alex.converse at gmail.com>
++ *
++ * This file is part of FFmpeg.
++ *
++ * FFmpeg is free software; you can redistribute it and/or
++ * modify it under the terms of the GNU Lesser General Public
++ * License as published by the Free Software Foundation; either
++ * version 2.1 of the License, or (at your option) any later version.
++ *
++ * FFmpeg is distributed in the hope that it will be useful,
++ * but WITHOUT ANY WARRANTY; without even the implied warranty of
++ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
++ * Lesser General Public License for more details.
++ *
++ * You should have received a copy of the GNU Lesser General Public
++ * License along with FFmpeg; if not, write to the Free Software
++ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
++ */
++
++#include <stdlib.h>
++#define CONFIG_HARDCODED_TABLES 0
++#include "aac_tablegen.h"
++#include "tableprint.h"
++
++int main(void)
++{
++    ff_aac_tableinit();
++
++    write_fileheader();
++
++    printf("const float ff_aac_pow2sf_tab[428] = {\n");
++    write_float_array(ff_aac_pow2sf_tab, 428);
++    printf("};\n");
++
++    return 0;
++}
+--- /dev/null
++++ b/libavcodec/aac_tablegen.h
+@@ -0,0 +1,42 @@
++/*
++ * Header file for hardcoded AAC tables
++ *
++ * Copyright (c) 2010 Alex Converse <alex.converse at gmail.com>
++ *
++ * 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
++ */
++
++#ifndef AAC_TABLEGEN_H
++#define AAC_TABLEGEN_H
++
++#include "aac_tablegen_decl.h"
++
++#if CONFIG_HARDCODED_TABLES
++#include "libavcodec/aac_tables.h"
++#else
++#include "../libavutil/mathematics.h"
++float ff_aac_pow2sf_tab[428];
++
++void ff_aac_tableinit(void)
++{
++    int i;
++    for (i = 0; i < 428; i++)
++        ff_aac_pow2sf_tab[i] = pow(2, (i - 200) / 4.);
++}
++#endif /* CONFIG_HARDCODED_TABLES */
++
++#endif /* AAC_TABLEGEN_H */
+--- /dev/null
++++ b/libavcodec/aacps_tablegen.c
+@@ -0,0 +1,93 @@
++/*
++ * Generate a header file for hardcoded Parametric Stereo tables
++ *
++ * Copyright (c) 2010 Alex Converse <alex.converse at gmail.com>
++ *
++ * This file is part of FFmpeg.
++ *
++ * FFmpeg is free software; you can redistribute it and/or
++ * modify it under the terms of the GNU Lesser General Public
++ * License as published by the Free Software Foundation; either
++ * version 2.1 of the License, or (at your option) any later version.
++ *
++ * FFmpeg is distributed in the hope that it will be useful,
++ * but WITHOUT ANY WARRANTY; without even the implied warranty of
++ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
++ * Lesser General Public License for more details.
++ *
++ * You should have received a copy of the GNU Lesser General Public
++ * License along with FFmpeg; if not, write to the Free Software
++ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
++ */
++
++#include <stdlib.h>
++#define CONFIG_HARDCODED_TABLES 0
++#include "aacps_tablegen.h"
++#include "tableprint.h"
++
++void write_float_3d_array (const void *p, int b, int c, int d)
++{
++    int i;
++    const float *f = p;
++    for (i = 0; i < b; i++) {
++        printf("{\n");
++        write_float_2d_array(f, c, d);
++        printf("},\n");
++        f += c * d;
++    }
++}
++
++void write_float_4d_array (const void *p, int a, int b, int c, int d)
++{
++    int i;
++    const float *f = p;
++    for (i = 0; i < a; i++) {
++        printf("{\n");
++        write_float_3d_array(f, b, c, d);
++        printf("},\n");
++        f += b * c * d;
++    }
++}
++
++int main(void)
++{
++    ps_tableinit();
++
++    write_fileheader();
++
++    printf("static const float pd_re_smooth[8*8*8] = {\n");
++    write_float_array(pd_re_smooth, 8*8*8);
++    printf("};\n");
++    printf("static const float pd_im_smooth[8*8*8] = {\n");
++    write_float_array(pd_im_smooth, 8*8*8);
++    printf("};\n");
++
++    printf("static const float HA[46][8][4] = {\n");
++    write_float_3d_array(HA, 46, 8, 4);
++    printf("};\n");
++    printf("static const float HB[46][8][4] = {\n");
++    write_float_3d_array(HB, 46, 8, 4);
++    printf("};\n");
++
++    printf("static const float f20_0_8[8][7][2] = {\n");
++    write_float_3d_array(f20_0_8, 8, 7, 2);
++    printf("};\n");
++    printf("static const float f34_0_12[12][7][2] = {\n");
++    write_float_3d_array(f34_0_12, 12, 7, 2);
++    printf("};\n");
++    printf("static const float f34_1_8[8][7][2] = {\n");
++    write_float_3d_array(f34_1_8, 8, 7, 2);
++    printf("};\n");
++    printf("static const float f34_2_4[4][7][2] = {\n");
++    write_float_3d_array(f34_2_4, 4, 7, 2);
++    printf("};\n");
++
++    printf("static const float Q_fract_allpass[2][50][3][2] = {\n");
++    write_float_4d_array(Q_fract_allpass, 2, 50, 3, 2);
++    printf("};\n");
++    printf("static const float phi_fract[2][50][2] = {\n");
++    write_float_3d_array(phi_fract, 2, 50, 2);
++    printf("};\n");
++
++    return 0;
++}
+--- /dev/null
++++ b/libavcodec/aacps_tablegen.h
+@@ -0,0 +1,212 @@
++/*
++ * Header file for hardcoded Parametric Stereo tables
++ *
++ * Copyright (c) 2010 Alex Converse <alex.converse at gmail.com>
++ *
++ * 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
++ */
++
++#ifndef AACPS_TABLEGEN_H
++#define AACPS_TABLEGEN_H
++
++#include <stdint.h>
++
++#if CONFIG_HARDCODED_TABLES
++#define ps_tableinit()
++#include "libavcodec/aacps_tables.h"
++#else
++#include "../libavutil/common.h"
++#include "../libavutil/mathematics.h"
++#define NR_ALLPASS_BANDS20 30
++#define NR_ALLPASS_BANDS34 50
++#define PS_AP_LINKS 3
++static float pd_re_smooth[8*8*8];
++static float pd_im_smooth[8*8*8];
++static float HA[46][8][4];
++static float HB[46][8][4];
++static float f20_0_8 [ 8][7][2];
++static float f34_0_12[12][7][2];
++static float f34_1_8 [ 8][7][2];
++static float f34_2_4 [ 4][7][2];
++static float Q_fract_allpass[2][50][3][2];
++static float phi_fract[2][50][2];
++
++static const float g0_Q8[] = {
++    0.00746082949812f, 0.02270420949825f, 0.04546865930473f, 0.07266113929591f,
++    0.09885108575264f, 0.11793710567217f, 0.125f
++};
++
++static const float g0_Q12[] = {
++    0.04081179924692f, 0.03812810994926f, 0.05144908135699f, 0.06399831151592f,
++    0.07428313801106f, 0.08100347892914f, 0.08333333333333f
++};
++
++static const float g1_Q8[] = {
++    0.01565675600122f, 0.03752716391991f, 0.05417891378782f, 0.08417044116767f,
++    0.10307344158036f, 0.12222452249753f, 0.125f
++};
++
++static const float g2_Q4[] = {
++    -0.05908211155639f, -0.04871498374946f, 0.0f,   0.07778723915851f,
++     0.16486303567403f,  0.23279856662996f, 0.25f
++};
++
++static void make_filters_from_proto(float (*filter)[7][2], const float *proto, int bands)
++{
++    int q, n;
++    for (q = 0; q < bands; q++) {
++        for (n = 0; n < 7; n++) {
++            double theta = 2 * M_PI * (q + 0.5) * (n - 6) / bands;
++            filter[q][n][0] = proto[n] *  cos(theta);
++            filter[q][n][1] = proto[n] * -sin(theta);
++        }
++    }
++}
++
++static void ps_tableinit(void)
++{
++    static const float ipdopd_sin[] = { 0, M_SQRT1_2, 1,  M_SQRT1_2,  0, -M_SQRT1_2, -1, -M_SQRT1_2 };
++    static const float ipdopd_cos[] = { 1, M_SQRT1_2, 0, -M_SQRT1_2, -1, -M_SQRT1_2,  0,  M_SQRT1_2 };
++    int pd0, pd1, pd2;
++
++    static const float iid_par_dequant[] = {
++        //iid_par_dequant_default
++        0.05623413251903, 0.12589254117942, 0.19952623149689, 0.31622776601684,
++        0.44668359215096, 0.63095734448019, 0.79432823472428, 1,
++        1.25892541179417, 1.58489319246111, 2.23872113856834, 3.16227766016838,
++        5.01187233627272, 7.94328234724282, 17.7827941003892,
++        //iid_par_dequant_fine
++        0.00316227766017, 0.00562341325190, 0.01,             0.01778279410039,
++        0.03162277660168, 0.05623413251903, 0.07943282347243, 0.11220184543020,
++        0.15848931924611, 0.22387211385683, 0.31622776601684, 0.39810717055350,
++        0.50118723362727, 0.63095734448019, 0.79432823472428, 1,
++        1.25892541179417, 1.58489319246111, 1.99526231496888, 2.51188643150958,
++        3.16227766016838, 4.46683592150963, 6.30957344480193, 8.91250938133745,
++        12.5892541179417, 17.7827941003892, 31.6227766016838, 56.2341325190349,
++        100,              177.827941003892, 316.227766016837,
++    };
++    static const float icc_invq[] = {
++        1, 0.937,      0.84118,    0.60092,    0.36764,   0,      -0.589,    -1
++    };
++    static const float acos_icc_invq[] = {
++        0, 0.35685527, 0.57133466, 0.92614472, 1.1943263, M_PI/2, 2.2006171, M_PI
++    };
++    int iid, icc;
++
++    int k, m;
++    static const int8_t f_center_20[] = {
++        -3, -1, 1, 3, 5, 7, 10, 14, 18, 22,
++    };
++    static const int8_t f_center_34[] = {
++         2,  6, 10, 14, 18, 22, 26, 30,
++        34,-10, -6, -2, 51, 57, 15, 21,
++        27, 33, 39, 45, 54, 66, 78, 42,
++       102, 66, 78, 90,102,114,126, 90,
++    };
++    static const float fractional_delay_links[] = { 0.43f, 0.75f, 0.347f };
++    const float fractional_delay_gain = 0.39f;
++
++    for (pd0 = 0; pd0 < 8; pd0++) {
++        float pd0_re = ipdopd_cos[pd0];
++        float pd0_im = ipdopd_sin[pd0];
++        for (pd1 = 0; pd1 < 8; pd1++) {
++            float pd1_re = ipdopd_cos[pd1];
++            float pd1_im = ipdopd_sin[pd1];
++            for (pd2 = 0; pd2 < 8; pd2++) {
++                float pd2_re = ipdopd_cos[pd2];
++                float pd2_im = ipdopd_sin[pd2];
++                float re_smooth = 0.25f * pd0_re + 0.5f * pd1_re + pd2_re;
++                float im_smooth = 0.25f * pd0_im + 0.5f * pd1_im + pd2_im;
++                float pd_mag = 1 / sqrt(im_smooth * im_smooth + re_smooth * re_smooth);
++                pd_re_smooth[pd0*64+pd1*8+pd2] = re_smooth * pd_mag;
++                pd_im_smooth[pd0*64+pd1*8+pd2] = im_smooth * pd_mag;
++            }
++        }
++    }
++
++    for (iid = 0; iid < 46; iid++) {
++        float c = iid_par_dequant[iid]; //<Linear Inter-channel Intensity Difference
++        float c1 = (float)M_SQRT2 / sqrtf(1.0f + c*c);
++        float c2 = c * c1;
++        for (icc = 0; icc < 8; icc++) {
++            /*if (PS_BASELINE || ps->icc_mode < 3)*/ {
++                float alpha = 0.5f * acos_icc_invq[icc];
++                float beta  = alpha * (c1 - c2) * (float)M_SQRT1_2;
++                HA[iid][icc][0] = c2 * cosf(beta + alpha);
++                HA[iid][icc][1] = c1 * cosf(beta - alpha);
++                HA[iid][icc][2] = c2 * sinf(beta + alpha);
++                HA[iid][icc][3] = c1 * sinf(beta - alpha);
++            } /* else */ {
++                float alpha, gamma, mu, rho;
++                float alpha_c, alpha_s, gamma_c, gamma_s;
++                rho = FFMAX(icc_invq[icc], 0.05f);
++                alpha = 0.5f * atan2f(2.0f * c * rho, c*c - 1.0f);
++                mu = c + 1.0f / c;
++                mu = sqrtf(1 + (4 * rho * rho - 4)/(mu * mu));
++                gamma = atanf(sqrtf((1.0f - mu)/(1.0f + mu)));
++                if (alpha < 0) alpha += M_PI/2;
++                alpha_c = cosf(alpha);
++                alpha_s = sinf(alpha);
++                gamma_c = cosf(gamma);
++                gamma_s = sinf(gamma);
++                HB[iid][icc][0] =  M_SQRT2 * alpha_c * gamma_c;
++                HB[iid][icc][1] =  M_SQRT2 * alpha_s * gamma_c;
++                HB[iid][icc][2] = -M_SQRT2 * alpha_s * gamma_s;
++                HB[iid][icc][3] =  M_SQRT2 * alpha_c * gamma_s;
++            }
++        }
++    }
++
++    for (k = 0; k < NR_ALLPASS_BANDS20; k++) {
++        double f_center, theta;
++        if (k < FF_ARRAY_ELEMS(f_center_20))
++            f_center = f_center_20[k] * 0.125;
++        else
++            f_center = k - 6.5f;
++        for (m = 0; m < PS_AP_LINKS; m++) {
++            theta = -M_PI * fractional_delay_links[m] * f_center;
++            Q_fract_allpass[0][k][m][0] = cos(theta);
++            Q_fract_allpass[0][k][m][1] = sin(theta);
++        }
++        theta = -M_PI*fractional_delay_gain*f_center;
++        phi_fract[0][k][0] = cos(theta);
++        phi_fract[0][k][1] = sin(theta);
++    }
++    for (k = 0; k < NR_ALLPASS_BANDS34; k++) {
++        double f_center, theta;
++        if (k < FF_ARRAY_ELEMS(f_center_34))
++            f_center = f_center_34[k] / 24.;
++        else
++            f_center = k - 26.5f;
++        for (m = 0; m < PS_AP_LINKS; m++) {
++            theta = -M_PI * fractional_delay_links[m] * f_center;
++            Q_fract_allpass[1][k][m][0] = cos(theta);
++            Q_fract_allpass[1][k][m][1] = sin(theta);
++        }
++        theta = -M_PI*fractional_delay_gain*f_center;
++        phi_fract[1][k][0] = cos(theta);
++        phi_fract[1][k][1] = sin(theta);
++    }
++
++    make_filters_from_proto(f20_0_8,  g0_Q8,   8);
++    make_filters_from_proto(f34_0_12, g0_Q12, 12);
++    make_filters_from_proto(f34_1_8,  g1_Q8,   8);
++    make_filters_from_proto(f34_2_4,  g2_Q4,   4);
++}
++#endif /* CONFIG_HARDCODED_TABLES */
++
++#endif /* AACPS_TABLEGEN_H */
diff --git a/debian/patches/series b/debian/patches/series
index 8105776..4db8bde 100644
--- a/debian/patches/series
+++ b/debian/patches/series
@@ -1,2 +1,3 @@
 0001-Add-VP80-fourcc.patch
 0002-Tweak-doxygen-config.patch
+0003-Backport-AAC-HE-v2.patch

-- 
FFmpeg packaging



More information about the pkg-multimedia-commits mailing list