[SCM] ffmpeg/master.snapshot: remove patches merged upstream

siretart at users.alioth.debian.org siretart at users.alioth.debian.org
Mon Nov 1 11:02:09 UTC 2010


The following commit has been merged in the master.snapshot branch:
commit c7faf5a1c3dcdc29e7b046c49a45cb453d8382a0
Author: Reinhard Tartler <siretart at tauware.de>
Date:   Mon Nov 1 09:15:19 2010 +0100

    remove patches merged upstream

diff --git a/debian/patches/0001-Add-VP80-fourcc.patch b/debian/patches/0001-Add-VP80-fourcc.patch
deleted file mode 100644
index d27c109..0000000
--- a/debian/patches/0001-Add-VP80-fourcc.patch
+++ /dev/null
@@ -1,24 +0,0 @@
-From: Reinhard Tartler <siretart at tauware.de>
-Date: Mon, 28 Jun 2010 23:12:40 +0200
-Subject: [PATCH] Add VP80 fourcc
-
-Patch by Google
-
-backport r23193 by conrad
----
- libavformat/riff.c |    1 +
- 1 files changed, 1 insertions(+), 0 deletions(-)
-
-diff --git a/libavformat/riff.c b/libavformat/riff.c
-index 04b7108..64464ca 100644
---- a/libavformat/riff.c
-+++ b/libavformat/riff.c
-@@ -183,6 +183,7 @@ const AVCodecTag ff_codec_bmp_tags[] = {
-     { CODEC_ID_VP6,          MKTAG('V', 'P', '6', '2') },
-     { CODEC_ID_VP6F,         MKTAG('V', 'P', '6', 'F') },
-     { CODEC_ID_VP6F,         MKTAG('F', 'L', 'V', '4') },
-+    { CODEC_ID_VP8,          MKTAG('V', 'P', '8', '0') },
-     { CODEC_ID_ASV1,         MKTAG('A', 'S', 'V', '1') },
-     { CODEC_ID_ASV2,         MKTAG('A', 'S', 'V', '2') },
-     { CODEC_ID_VCR1,         MKTAG('V', 'C', 'R', '1') },
--- 
diff --git a/debian/patches/0003-Backport-AAC-HE-v2.patch b/debian/patches/0003-Backport-AAC-HE-v2.patch
deleted file mode 100644
index babb9f5..0000000
--- a/debian/patches/0003-Backport-AAC-HE-v2.patch
+++ /dev/null
@@ -1,6774 +0,0 @@
-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 12f1b01..5576601 100644
--- a/debian/patches/series
+++ b/debian/patches/series
@@ -1,4 +1,2 @@
-0001-Add-VP80-fourcc.patch
 0002-Tweak-doxygen-config.patch
-0003-Backport-AAC-HE-v2.patch
 0004-cpp-hack.patch

-- 
FFmpeg packaging



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