[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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