/*
  * AAC decoder
  * Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org )
  * Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com )
  *

  * AAC LATM decoder
  * Copyright (c) 2008-2010 Paul Kendall <paul@kcbbs.gen.nz>

  * Copyright (c) 2010      Janne Grunau <janne-libav@jannau.net>

  *

  * 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

  * Y                    Long Term Prediction

  * Y                    intensity stereo
  * Y                    channel coupling

  * Y                    frequency domain prediction

  * Y                    Perceptual Noise Substitution
  * Y                    Mid/Side stereo
  * N                    Scalable Inverse AAC Quantization
  * N                    Frequency Selective Switch
  * N                    upsampling filter
  * Y                    quantization & coding - AAC
  * N                    quantization & coding - TwinVQ
  * N                    quantization & coding - BSAC
  * N                    AAC Error Resilience tools
  * N                    Error Resilience payload syntax
  * N                    Error Protection tool
  * N                    CELP
  * N                    Silence Compression
  * N                    HVXC
  * N                    HVXC 4kbits/s VR
  * N                    Structured Audio tools
  * N                    Structured Audio Sample Bank Format
  * N                    MIDI
  * N                    Harmonic and Individual Lines plus Noise
  * N                    Text-To-Speech Interface

  * Y                    Spectral Band Replication

  * Y (not in this code) Layer-1
  * Y (not in this code) Layer-2
  * Y (not in this code) Layer-3
  * N                    SinuSoidal Coding (Transient, Sinusoid, Noise)

  * Y                    Parametric Stereo

  * N                    Direct Stream Transfer
  *
  * 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 "libavutil/float_dsp.h"

 #include "libavutil/opt.h"

 #include "avcodec.h"

 #include "internal.h"

 #include "get_bits.h"

 #include "fft.h"

 #include "fmtconvert.h"

 #include "lpc.h"

 #include "kbdwin.h"

 #include "sinewin.h"

 
 #include "aac.h"
 #include "aactab.h"

 #include "aacdectab.h"

 #include "cbrt_tablegen.h"

 #include "sbr.h"
 #include "aacsbr.h"

 #include "mpeg4audio.h"

 #include "aacadtsdec.h"

 #include "libavutil/intfloat.h"

 
 #include <assert.h>
 #include <errno.h>
 #include <math.h>
 #include <string.h>
 

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

 #elif ARCH_MIPS
 #   include "mips/aacdec_mips.h"

 #endif
 

 static VLC vlc_scalefactors;
 static VLC vlc_spectral[11];
 

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

 #define overread_err "Input buffer exhausted before END element found\n"

 static int count_channels(uint8_t (*layout)[3], int tags)

 {

     int i, sum = 0;
     for (i = 0; i < tags; i++) {
         int syn_ele = layout[i][0];
         int pos     = layout[i][2];
         sum += (1 + (syn_ele == TYPE_CPE)) *
                (pos != AAC_CHANNEL_OFF && pos != AAC_CHANNEL_CC);

     }
     return sum;
 }
 

 /**

  * Check for the channel element in the current channel position configuration.
  * If it exists, make sure the appropriate element is allocated and map the
  * channel order to match the internal FFmpeg channel layout.
  *
  * @param   che_pos current channel position configuration
  * @param   type channel element type
  * @param   id channel element id
  * @param   channels count of the number of channels in the configuration
  *
  * @return  Returns error status. 0 - OK, !0 - error
  */

 static av_cold int che_configure(AACContext *ac,

                                  enum ChannelPosition che_pos,

                                  int type, int id, int *channels)

 {

     if (che_pos) {

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

         if (type != TYPE_CCE) {

             if (*channels >= MAX_CHANNELS - (type == TYPE_CPE || (type == TYPE_SCE && ac->oc[1].m4ac.ps == 1))) {

                 av_log(ac->avctx, AV_LOG_ERROR, "Too many channels\n");
                 return AVERROR_INVALIDDATA;
             }

             ac->output_element[(*channels)++] = &ac->che[type][id]->ch[0];

             if (type == TYPE_CPE ||

                 (type == TYPE_SCE && ac->oc[1].m4ac.ps == 1)) {

                 ac->output_element[(*channels)++] = &ac->che[type][id]->ch[1];

             }
         }

     } else {
         if (ac->che[type][id])
             ff_aac_sbr_ctx_close(&ac->che[type][id]->sbr);

         av_freep(&ac->che[type][id]);

     }

     return 0;
 }
 

 static int frame_configure_elements(AVCodecContext *avctx)
 {
     AACContext *ac = avctx->priv_data;
     int type, id, ch, ret;
 
     /* set channel pointers to internal buffers by default */
     for (type = 0; type < 4; type++) {
         for (id = 0; id < MAX_ELEM_ID; id++) {
             ChannelElement *che = ac->che[type][id];
             if (che) {
                 che->ch[0].ret = che->ch[0].ret_buf;
                 che->ch[1].ret = che->ch[1].ret_buf;
             }
         }
     }
 
     /* get output buffer */

     ac->frame->nb_samples = 2048;
     if ((ret = ff_get_buffer(avctx, ac->frame)) < 0) {

         av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
         return ret;
     }
 
     /* map output channel pointers to AVFrame data */
     for (ch = 0; ch < avctx->channels; ch++) {
         if (ac->output_element[ch])

             ac->output_element[ch]->ret = (float *)ac->frame->extended_data[ch];

     }
 
     return 0;
 }
 

 struct elem_to_channel {
     uint64_t av_position;
     uint8_t syn_ele;
     uint8_t elem_id;
     uint8_t aac_position;
 };
 
 static int assign_pair(struct elem_to_channel e2c_vec[MAX_ELEM_ID],

                        uint8_t (*layout_map)[3], int offset, uint64_t left,

     uint64_t right, int pos)
 {
     if (layout_map[offset][0] == TYPE_CPE) {
         e2c_vec[offset] = (struct elem_to_channel) {
             .av_position = left | right, .syn_ele = TYPE_CPE,
             .elem_id = layout_map[offset    ][1], .aac_position = pos };
         return 1;
     } else {
         e2c_vec[offset]   = (struct elem_to_channel) {
             .av_position = left, .syn_ele = TYPE_SCE,
             .elem_id = layout_map[offset    ][1], .aac_position = pos };
         e2c_vec[offset + 1] = (struct elem_to_channel) {
             .av_position = right, .syn_ele = TYPE_SCE,
             .elem_id = layout_map[offset + 1][1], .aac_position = pos };
         return 2;
     }
 }
 
 static int count_paired_channels(uint8_t (*layout_map)[3], int tags, int pos, int *current) {
     int num_pos_channels = 0;
     int first_cpe = 0;
     int sce_parity = 0;
     int i;
     for (i = *current; i < tags; i++) {
         if (layout_map[i][2] != pos)
             break;
         if (layout_map[i][0] == TYPE_CPE) {
             if (sce_parity) {

                 if (pos == AAC_CHANNEL_FRONT && !first_cpe) {

                     sce_parity = 0;
                 } else {
                     return -1;
                 }
             }
             num_pos_channels += 2;
             first_cpe = 1;
         } else {
             num_pos_channels++;
             sce_parity ^= 1;
         }
     }
     if (sce_parity &&
         ((pos == AAC_CHANNEL_FRONT && first_cpe) || pos == AAC_CHANNEL_SIDE))
             return -1;
     *current = i;
     return num_pos_channels;
 }
 
 static uint64_t sniff_channel_order(uint8_t (*layout_map)[3], int tags)
 {
     int i, n, total_non_cc_elements;

     struct elem_to_channel e2c_vec[4*MAX_ELEM_ID] = {{ 0 }};

     int num_front_channels, num_side_channels, num_back_channels;
     uint64_t layout;
 

     if (FF_ARRAY_ELEMS(e2c_vec) < tags)

         return 0;
 

     i = 0;
     num_front_channels =
         count_paired_channels(layout_map, tags, AAC_CHANNEL_FRONT, &i);
     if (num_front_channels < 0)
         return 0;
     num_side_channels =
         count_paired_channels(layout_map, tags, AAC_CHANNEL_SIDE, &i);
     if (num_side_channels < 0)
         return 0;
     num_back_channels =
         count_paired_channels(layout_map, tags, AAC_CHANNEL_BACK, &i);
     if (num_back_channels < 0)
         return 0;
 
     i = 0;
     if (num_front_channels & 1) {
         e2c_vec[i] = (struct elem_to_channel) {
             .av_position = AV_CH_FRONT_CENTER, .syn_ele = TYPE_SCE,
             .elem_id = layout_map[i][1], .aac_position = AAC_CHANNEL_FRONT };
         i++;
         num_front_channels--;
     }
     if (num_front_channels >= 4) {

         i += assign_pair(e2c_vec, layout_map, i,

                          AV_CH_FRONT_LEFT_OF_CENTER,
                          AV_CH_FRONT_RIGHT_OF_CENTER,
                          AAC_CHANNEL_FRONT);
         num_front_channels -= 2;
     }
     if (num_front_channels >= 2) {

         i += assign_pair(e2c_vec, layout_map, i,

                          AV_CH_FRONT_LEFT,
                          AV_CH_FRONT_RIGHT,
                          AAC_CHANNEL_FRONT);
         num_front_channels -= 2;
     }
     while (num_front_channels >= 2) {

         i += assign_pair(e2c_vec, layout_map, i,

                          UINT64_MAX,
                          UINT64_MAX,
                          AAC_CHANNEL_FRONT);
         num_front_channels -= 2;
     }
 
     if (num_side_channels >= 2) {

         i += assign_pair(e2c_vec, layout_map, i,

                          AV_CH_SIDE_LEFT,
                          AV_CH_SIDE_RIGHT,
                          AAC_CHANNEL_FRONT);
         num_side_channels -= 2;
     }
     while (num_side_channels >= 2) {

         i += assign_pair(e2c_vec, layout_map, i,

                          UINT64_MAX,
                          UINT64_MAX,
                          AAC_CHANNEL_SIDE);
         num_side_channels -= 2;
     }
 
     while (num_back_channels >= 4) {

         i += assign_pair(e2c_vec, layout_map, i,

                          UINT64_MAX,
                          UINT64_MAX,
                          AAC_CHANNEL_BACK);
         num_back_channels -= 2;
     }
     if (num_back_channels >= 2) {

         i += assign_pair(e2c_vec, layout_map, i,

                          AV_CH_BACK_LEFT,
                          AV_CH_BACK_RIGHT,
                          AAC_CHANNEL_BACK);
         num_back_channels -= 2;
     }
     if (num_back_channels) {
         e2c_vec[i] = (struct elem_to_channel) {
           .av_position = AV_CH_BACK_CENTER, .syn_ele = TYPE_SCE,
           .elem_id = layout_map[i][1], .aac_position = AAC_CHANNEL_BACK };
         i++;
         num_back_channels--;
     }
 
     if (i < tags && layout_map[i][2] == AAC_CHANNEL_LFE) {
         e2c_vec[i] = (struct elem_to_channel) {
           .av_position = AV_CH_LOW_FREQUENCY, .syn_ele = TYPE_LFE,
           .elem_id = layout_map[i][1], .aac_position = AAC_CHANNEL_LFE };
         i++;
     }
     while (i < tags && layout_map[i][2] == AAC_CHANNEL_LFE) {
         e2c_vec[i] = (struct elem_to_channel) {
           .av_position = UINT64_MAX, .syn_ele = TYPE_LFE,
           .elem_id = layout_map[i][1], .aac_position = AAC_CHANNEL_LFE };
         i++;
     }
 
     // Must choose a stable sort
     total_non_cc_elements = n = i;
     do {
         int next_n = 0;
         for (i = 1; i < n; i++) {
             if (e2c_vec[i-1].av_position > e2c_vec[i].av_position) {
                 FFSWAP(struct elem_to_channel, e2c_vec[i-1], e2c_vec[i]);
                 next_n = i;
             }
         }
         n = next_n;
     } while (n > 0);
 
     layout = 0;
     for (i = 0; i < total_non_cc_elements; i++) {
         layout_map[i][0] = e2c_vec[i].syn_ele;
         layout_map[i][1] = e2c_vec[i].elem_id;
         layout_map[i][2] = e2c_vec[i].aac_position;
         if (e2c_vec[i].av_position != UINT64_MAX) {
             layout |= e2c_vec[i].av_position;
         }
     }
 
     return layout;
 }
 

 /**

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

     if (ac->oc[1].status != OC_LOCKED && ac->oc[0].status != OC_NONE) {

         ac->oc[1] = ac->oc[0];
         ac->avctx->channels = ac->oc[1].channels;

         ac->avctx->channel_layout = ac->oc[1].channel_layout;

         output_configure(ac, ac->oc[1].layout_map, ac->oc[1].layout_map_tags,

                          ac->oc[1].status, 0);

     }
 }
 
 /**

  * Configure output channel order based on the current program configuration element.
  *
  * @return  Returns error status. 0 - OK, !0 - error
  */

 static int output_configure(AACContext *ac,

                             uint8_t layout_map[MAX_ELEM_ID*4][3], int tags,
                             enum OCStatus oc_type, int get_new_frame)

 {

     AVCodecContext *avctx = ac->avctx;

     int i, channels = 0, ret;

     uint64_t layout = 0;

     if (ac->oc[1].layout_map != layout_map) {
         memcpy(ac->oc[1].layout_map, layout_map, tags * sizeof(layout_map[0]));
         ac->oc[1].layout_map_tags = tags;

     }

     // Try to sniff a reasonable channel order, otherwise output the
     // channels in the order the PCE declared them.
     if (avctx->request_channel_layout != AV_CH_LAYOUT_NATIVE)
         layout = sniff_channel_order(layout_map, tags);
     for (i = 0; i < tags; i++) {
         int type =     layout_map[i][0];
         int id =       layout_map[i][1];
         int position = layout_map[i][2];
         // Allocate or free elements depending on if they are in the
         // current program configuration.
         ret = che_configure(ac, position, type, id, &channels);
         if (ret < 0)
             return ret;

     }

     if (ac->oc[1].m4ac.ps == 1 && channels == 2) {
         if (layout == AV_CH_FRONT_CENTER) {
             layout = AV_CH_FRONT_LEFT|AV_CH_FRONT_RIGHT;
         } else {
             layout = 0;
         }
     }

     memcpy(ac->tag_che_map, ac->che, 4 * MAX_ELEM_ID * sizeof(ac->che[0][0]));

     if (layout) avctx->channel_layout = layout;

     ac->oc[1].channel_layout = layout;

     avctx->channels = ac->oc[1].channels = channels;
     ac->oc[1].status = oc_type;

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

     return 0;
 }
 

 static void flush(AVCodecContext *avctx)
 {
     AACContext *ac= avctx->priv_data;
     int type, i, j;
 
     for (type = 3; type >= 0; type--) {
         for (i = 0; i < MAX_ELEM_ID; i++) {
             ChannelElement *che = ac->che[type][i];
             if (che) {
                 for (j = 0; j <= 1; j++) {
                     memset(che->ch[j].saved, 0, sizeof(che->ch[j].saved));
                 }
             }
         }
     }
 }
 

 /**

  * Set up channel positions based on a default channel configuration
  * as specified in table 1.17.
  *
  * @return  Returns error status. 0 - OK, !0 - error
  */

 static int set_default_channel_config(AVCodecContext *avctx,

                                               uint8_t (*layout_map)[3],
                                               int *tags,
                                               int channel_config)
 {
     if (channel_config < 1 || channel_config > 7) {
         av_log(avctx, AV_LOG_ERROR, "invalid default channel configuration (%d)\n",
                channel_config);
         return -1;
     }
     *tags = tags_per_config[channel_config];
     memcpy(layout_map, aac_channel_layout_map[channel_config-1], *tags * sizeof(*layout_map));
     return 0;
 }
 
 static ChannelElement *get_che(AACContext *ac, int type, int elem_id)
 {
     // For PCE based channel configurations map the channels solely based on tags.

     if (!ac->oc[1].m4ac.chan_config) {

         return ac->tag_che_map[type][elem_id];
     }

     // Allow single CPE stereo files to be signalled with mono configuration.

     if (!ac->tags_mapped && type == TYPE_CPE && ac->oc[1].m4ac.chan_config == 1) {

         uint8_t layout_map[MAX_ELEM_ID*4][3];
         int layout_map_tags;

         push_output_configuration(ac);

         av_log(ac->avctx, AV_LOG_DEBUG, "mono with CPE\n");
 

         if (set_default_channel_config(ac->avctx, layout_map, &layout_map_tags,
                                        2) < 0)
             return NULL;

         if (output_configure(ac, layout_map, layout_map_tags,

                              OC_TRIAL_FRAME, 1) < 0)

             return NULL;
 

         ac->oc[1].m4ac.chan_config = 2;

         ac->oc[1].m4ac.ps = 0;

     }
     // And vice-versa

     if (!ac->tags_mapped && type == TYPE_SCE && ac->oc[1].m4ac.chan_config == 2) {

         uint8_t layout_map[MAX_ELEM_ID*4][3];
         int layout_map_tags;
         push_output_configuration(ac);
 

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

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

         if (output_configure(ac, layout_map, layout_map_tags,

                              OC_TRIAL_FRAME, 1) < 0)

             return NULL;
 
         ac->oc[1].m4ac.chan_config = 1;

         if (ac->oc[1].m4ac.sbr)
             ac->oc[1].m4ac.ps = -1;

     }

     // For indexed channel configurations map the channels solely based on position.

     switch (ac->oc[1].m4ac.chan_config) {

     case 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->oc[1].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->oc[1].m4ac.chan_config == 4 && type == TYPE_SCE) {

             ac->tags_mapped++;
             return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][1];
         }
     case 3:
     case 2:

         if (ac->tags_mapped == (ac->oc[1].m4ac.chan_config != 2) && type == TYPE_CPE) {

             ac->tags_mapped++;
             return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][0];

         } else if (ac->oc[1].m4ac.chan_config == 2) {

             return NULL;
         }
     case 1:
         if (!ac->tags_mapped && type == TYPE_SCE) {
             ac->tags_mapped++;
             return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][0];
         }
     default:
         return NULL;
     }
 }
 
 /**

  * Decode an array of 4 bit element IDs, optionally interleaved with a stereo/mono switching bit.
  *
  * @param type speaker type/position for these channels
  */

 static void decode_channel_map(uint8_t layout_map[][3],

                                enum ChannelPosition type,
                                GetBitContext *gb, int n)
 {
     while (n--) {

         enum RawDataBlockType syn_ele;
         switch (type) {
         case AAC_CHANNEL_FRONT:
         case AAC_CHANNEL_BACK:
         case AAC_CHANNEL_SIDE:
             syn_ele = get_bits1(gb);
             break;
         case AAC_CHANNEL_CC:
             skip_bits1(gb);
             syn_ele = TYPE_CCE;
             break;
         case AAC_CHANNEL_LFE:
             syn_ele = TYPE_LFE;
             break;

         default:
             av_assert0(0);

         }
         layout_map[0][0] = syn_ele;
         layout_map[0][1] = get_bits(gb, 4);
         layout_map[0][2] = type;
         layout_map++;

     }
 }
 
 /**
  * Decode program configuration element; reference: table 4.2.
  *
  * @return  Returns error status. 0 - OK, !0 - error
  */

 static int decode_pce(AVCodecContext *avctx, MPEG4AudioConfig *m4ac,

                       uint8_t (*layout_map)[3],

                       GetBitContext *gb)
 {

     int num_front, num_side, num_back, num_lfe, num_assoc_data, num_cc, sampling_index;

     int comment_len;

     int tags;

 
     skip_bits(gb, 2);  // object_type
 

     sampling_index = get_bits(gb, 4);

     if (m4ac->sampling_index != sampling_index)
         av_log(avctx, AV_LOG_WARNING, "Sample rate index in program config element does not match the sample rate index configured by the container.\n");

     num_front       = get_bits(gb, 4);
     num_side        = get_bits(gb, 4);
     num_back        = get_bits(gb, 4);
     num_lfe         = get_bits(gb, 2);
     num_assoc_data  = get_bits(gb, 3);
     num_cc          = get_bits(gb, 4);
 

     if (get_bits1(gb))
         skip_bits(gb, 4); // mono_mixdown_tag
     if (get_bits1(gb))
         skip_bits(gb, 4); // stereo_mixdown_tag

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

     if (get_bits_left(gb) < 4 * (num_front + num_side + num_back + num_lfe + num_assoc_data + num_cc)) {

         av_log(avctx, AV_LOG_ERROR, "decode_pce: " overread_err);

         return -1;
     }

     decode_channel_map(layout_map       , AAC_CHANNEL_FRONT, gb, num_front);
     tags = num_front;
     decode_channel_map(layout_map + tags, AAC_CHANNEL_SIDE,  gb, num_side);
     tags += num_side;
     decode_channel_map(layout_map + tags, AAC_CHANNEL_BACK,  gb, num_back);
     tags += num_back;
     decode_channel_map(layout_map + tags, AAC_CHANNEL_LFE,   gb, num_lfe);
     tags += num_lfe;

 
     skip_bits_long(gb, 4 * num_assoc_data);
 

     decode_channel_map(layout_map + tags, AAC_CHANNEL_CC,    gb, num_cc);
     tags += num_cc;

 
     align_get_bits(gb);
 
     /* comment field, first byte is length */

     comment_len = get_bits(gb, 8) * 8;
     if (get_bits_left(gb) < comment_len) {

         av_log(avctx, AV_LOG_ERROR, "decode_pce: " overread_err);

         return -1;
     }
     skip_bits_long(gb, comment_len);

     return tags;

 }

 /**

  * Decode GA "General Audio" specific configuration; reference: table 4.1.
  *

  * @param   ac          pointer to AACContext, may be null
  * @param   avctx       pointer to AVCCodecContext, used for logging
  *

  * @return  Returns error status. 0 - OK, !0 - error
  */

 static int decode_ga_specific_config(AACContext *ac, AVCodecContext *avctx,
                                      GetBitContext *gb,

                                      MPEG4AudioConfig *m4ac,

                                      int channel_config)
 {

     int extension_flag, ret;

     uint8_t layout_map[MAX_ELEM_ID*4][3];
     int tags = 0;

     if (get_bits1(gb)) { // frameLengthFlag

         av_log_missing_feature(avctx, "960/120 MDCT window", 1);

         return AVERROR_PATCHWELCOME;

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

     if (m4ac->object_type == AOT_AAC_SCALABLE ||
         m4ac->object_type == AOT_ER_AAC_SCALABLE)

         skip_bits(gb, 3);     // layerNr
 
     if (channel_config == 0) {
         skip_bits(gb, 4);  // element_instance_tag

         tags = decode_pce(avctx, m4ac, layout_map, gb);
         if (tags < 0)
             return tags;

     } else {

         if ((ret = set_default_channel_config(avctx, layout_map, &tags, channel_config)))

             return ret;
     }

     if (count_channels(layout_map, tags) > 1) {

         m4ac->ps = 0;
     } else if (m4ac->sbr == 1 && m4ac->ps == -1)
         m4ac->ps = 1;
 

     if (ac && (ret = output_configure(ac, layout_map, tags, OC_GLOBAL_HDR, 0)))

         return ret;
 
     if (extension_flag) {

         switch (m4ac->object_type) {

         case AOT_ER_BSAC:
             skip_bits(gb, 5);    // numOfSubFrame
             skip_bits(gb, 11);   // layer_length
             break;
         case AOT_ER_AAC_LC:
         case AOT_ER_AAC_LTP:
         case AOT_ER_AAC_SCALABLE:
         case AOT_ER_AAC_LD:
             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   ac          pointer to AACContext, may be null
  * @param   avctx       pointer to AVCCodecContext, used for logging
  * @param   m4ac        pointer to MPEG4AudioConfig, used for parsing

  * @param   data        pointer to buffer holding an audio specific config
  * @param   bit_size    size of audio specific config or data in bits
  * @param   sync_extension look for an appended sync extension

  *

  * @return  Returns error status or number of consumed bits. <0 - error

  */

 static int decode_audio_specific_config(AACContext *ac,

                                         AVCodecContext *avctx,
                                         MPEG4AudioConfig *m4ac,

                                         const uint8_t *data, int bit_size,
                                         int sync_extension)

 {

     GetBitContext gb;
     int i;

     int ret;

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

     av_dlog(avctx, "\n");
 

     if ((ret = init_get_bits(&gb, data, bit_size)) < 0)
         return ret;

     if ((i = avpriv_mpeg4audio_get_config(m4ac, data, bit_size, sync_extension)) < 0)

         return -1;

     if (m4ac->sampling_index > 12) {

         av_log(avctx, AV_LOG_ERROR, "invalid sampling rate index %d\n", m4ac->sampling_index);

         return -1;
     }
 
     skip_bits_long(&gb, i);
 

     switch (m4ac->object_type) {

     case AOT_AAC_MAIN:

     case AOT_AAC_LC:

     case AOT_AAC_LTP:

         if (decode_ga_specific_config(ac, avctx, &gb, m4ac, m4ac->chan_config))

             return -1;
         break;
     default:

         av_log(avctx, AV_LOG_ERROR, "Audio object type %s%d is not supported.\n",

                m4ac->sbr == 1? "SBR+" : "", m4ac->object_type);

         return -1;
     }

     av_dlog(avctx, "AOT %d chan config %d sampling index %d (%d) SBR %d PS %d\n",
             m4ac->object_type, m4ac->chan_config, m4ac->sampling_index,
             m4ac->sample_rate, m4ac->sbr, m4ac->ps);
 

     return get_bits_count(&gb);

 }
 

 /**
  * linear congruential pseudorandom number generator
  *
  * @param   previous_val    pointer to the current state of the generator
  *
  * @return  Returns a 32-bit pseudorandom integer
  */

 static av_always_inline int lcg_random(unsigned previous_val)

 {

     union { unsigned u; int s; } v = { previous_val * 1664525u + 1013904223 };
     return v.s;

 }
 

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

 static void reset_predictor_group(PredictorState *ps, int group_num)
 {

     int i;

     for (i = group_num - 1; i < MAX_PREDICTORS; i += 30)

         reset_predict_state(&ps[i]);
 }
 

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

 static void aacdec_init(AACContext *ac);
 

 static av_cold int aac_decode_init(AVCodecContext *avctx)

 {

     AACContext *ac = avctx->priv_data;

     ac->avctx = avctx;

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

     aacdec_init(ac);
 

     avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;

     if (avctx->extradata_size > 0) {

         if (decode_audio_specific_config(ac, ac->avctx, &ac->oc[1].m4ac,

                                          avctx->extradata,

                                          avctx->extradata_size*8, 1) < 0)

             return -1;

     } else {
         int sr, i;

         uint8_t layout_map[MAX_ELEM_ID*4][3];
         int layout_map_tags;

 
         sr = sample_rate_idx(avctx->sample_rate);

         ac->oc[1].m4ac.sampling_index = sr;
         ac->oc[1].m4ac.channels = avctx->channels;
         ac->oc[1].m4ac.sbr = -1;
         ac->oc[1].m4ac.ps = -1;

 
         for (i = 0; i < FF_ARRAY_ELEMS(ff_mpeg4audio_channels); i++)
             if (ff_mpeg4audio_channels[i] == avctx->channels)
                 break;
         if (i == FF_ARRAY_ELEMS(ff_mpeg4audio_channels)) {
             i = 0;
         }

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

         if (ac->oc[1].m4ac.chan_config) {

             int ret = set_default_channel_config(avctx, layout_map,

                 &layout_map_tags, ac->oc[1].m4ac.chan_config);

             if (!ret)

                 output_configure(ac, layout_map, layout_map_tags,

                                  OC_GLOBAL_HDR, 0);

             else if (avctx->err_recognition & AV_EF_EXPLODE)

                 return AVERROR_INVALIDDATA;

         }

     }

     if (avctx->channels > MAX_CHANNELS) {
         av_log(avctx, AV_LOG_ERROR, "Too many channels\n");
         return AVERROR_INVALIDDATA;
     }
 

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

     ff_aac_sbr_init();
 

     ff_fmt_convert_init(&ac->fmt_conv, avctx);

     avpriv_float_dsp_init(&ac->fdsp, avctx->flags & CODEC_FLAG_BITEXACT);

     ac->random_state = 0x1f2e3d4c;
 

     ff_aac_tableinit();

     INIT_VLC_STATIC(&vlc_scalefactors,7,FF_ARRAY_ELEMS(ff_aac_scalefactor_code),

                     ff_aac_scalefactor_bits, sizeof(ff_aac_scalefactor_bits[0]), sizeof(ff_aac_scalefactor_bits[0]),
                     ff_aac_scalefactor_code, sizeof(ff_aac_scalefactor_code[0]), sizeof(ff_aac_scalefactor_code[0]),
                     352);

     ff_mdct_init(&ac->mdct,       11, 1, 1.0 / (32768.0 * 1024.0));
     ff_mdct_init(&ac->mdct_small,  8, 1, 1.0 / (32768.0 * 128.0));
     ff_mdct_init(&ac->mdct_ltp,   11, 0, -2.0 * 32768.0);

     // 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, "skip_data_stream_element: "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->oc[1].m4ac.sampling_index]); sfb++) {

         ics->prediction_used[sfb] = get_bits1(gb);
     }
     return 0;
 }
 

 /**

  * Decode Long Term Prediction data; reference: table 4.xx.
  */

 static void decode_ltp(LongTermPrediction *ltp,

                        GetBitContext *gb, uint8_t max_sfb)
 {
     int sfb;
 
     ltp->lag  = get_bits(gb, 11);

     ltp->coef = ltp_coef[get_bits(gb, 3)];

     for (sfb = 0; sfb < FFMIN(max_sfb, MAX_LTP_LONG_SFB); sfb++)
         ltp->used[sfb] = get_bits1(gb);
 }
 
 /**

  * Decode Individual Channel Stream info; reference: table 4.6.
  */

 static int decode_ics_info(AACContext *ac, IndividualChannelStream *ics,

                            GetBitContext *gb)

 {

     if (get_bits1(gb)) {

         av_log(ac->avctx, AV_LOG_ERROR, "Reserved bit set.\n");

         return AVERROR_INVALIDDATA;

     }
     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->oc[1].m4ac.sampling_index];
         ics->num_swb           =   ff_aac_num_swb_128[ac->oc[1].m4ac.sampling_index];
         ics->tns_max_bands     = ff_tns_max_bands_128[ac->oc[1].m4ac.sampling_index];

         ics->predictor_present = 0;

     } else {

         ics->max_sfb               = get_bits(gb, 6);
         ics->num_windows           = 1;

         ics->swb_offset            =    ff_swb_offset_1024[ac->oc[1].m4ac.sampling_index];
         ics->num_swb               =   ff_aac_num_swb_1024[ac->oc[1].m4ac.sampling_index];
         ics->tns_max_bands         = ff_tns_max_bands_1024[ac->oc[1].m4ac.sampling_index];

         ics->predictor_present     = get_bits1(gb);

         ics->predictor_reset_group = 0;
         if (ics->predictor_present) {

             if (ac->oc[1].m4ac.object_type == AOT_AAC_MAIN) {

                 if (decode_prediction(ac, ics, gb)) {

                     goto fail;

                 }

             } else if (ac->oc[1].m4ac.object_type == AOT_AAC_LC) {

                 av_log(ac->avctx, AV_LOG_ERROR, "Prediction is not allowed in AAC-LC.\n");

                 goto fail;

             } else {

                 if ((ics->ltp.present = get_bits(gb, 1)))

                     decode_ltp(&ics->ltp, gb, ics->max_sfb);

             }

         }
     }
 

     if (ics->max_sfb > ics->num_swb) {

         av_log(ac->avctx, AV_LOG_ERROR,

                "Number of scalefactor bands in group (%d) exceeds limit (%d).\n",
                ics->max_sfb, ics->num_swb);

         goto fail;

     }
 

     return 0;

 fail:
     ics->max_sfb = 0;
     return AVERROR_INVALIDDATA;

 }
 
 /**
  * Decode band types (section_data payload); reference: table 4.46.
  *
  * @param   band_type           array of the used band type
  * @param   band_type_run_end   array of the last scalefactor band of a band type run
  *
  * @return  Returns error status. 0 - OK, !0 - error
  */

 static int decode_band_types(AACContext *ac, enum BandType band_type[120],
                              int band_type_run_end[120], GetBitContext *gb,
                              IndividualChannelStream *ics)
 {

     int g, idx = 0;
     const int bits = (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) ? 3 : 5;
     for (g = 0; g < ics->num_window_groups; g++) {
         int k = 0;
         while (k < ics->max_sfb) {

             uint8_t sect_end = k;

             int sect_len_incr;
             int sect_band_type = get_bits(gb, 4);
             if (sect_band_type == 12) {

                 av_log(ac->avctx, AV_LOG_ERROR, "invalid band type\n");

                 return -1;
             }

             do {
                 sect_len_incr = get_bits(gb, bits);

                 sect_end += sect_len_incr;

                 if (get_bits_left(gb) < 0) {

                     av_log(ac->avctx, AV_LOG_ERROR, "decode_band_types: "overread_err);

                     return -1;
                 }
                 if (sect_end > ics->max_sfb) {
                     av_log(ac->avctx, AV_LOG_ERROR,
                            "Number of bands (%d) exceeds limit (%d).\n",
                            sect_end, ics->max_sfb);
                     return -1;
                 }
             } while (sect_len_incr == (1 << bits) - 1);

             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])
 {

     int g, i, idx = 0;

     int offset[3] = { global_gain, global_gain - 90, 0 };
     int clipped_offset;

     int noise_flag = 1;
     for (g = 0; g < ics->num_window_groups; g++) {
         for (i = 0; i < ics->max_sfb;) {
             int run_end = band_type_run_end[idx];
             if (band_type[idx] == ZERO_BT) {

                 for (; i < run_end; i++, idx++)

                     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;

                     clipped_offset = av_clip(offset[2], -155, 100);
                     if (offset[2] != clipped_offset) {
                         av_log_ask_for_sample(ac->avctx, "Intensity stereo "
                                 "position clipped (%d -> %d).\nIf you heard an "
                                 "audible artifact, there may be a bug in the "
                                 "decoder. ", offset[2], clipped_offset);

                     }

                     sf[idx] = ff_aac_pow2sf_tab[-clipped_offset + POW_SF2_ZERO];

                 }

             } 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;

                     clipped_offset = av_clip(offset[1], -100, 155);

                     if (offset[1] != clipped_offset) {

                         av_log_ask_for_sample(ac->avctx, "Noise gain clipped "
                                 "(%d -> %d).\nIf you heard an audible "
                                 "artifact, there may be a bug in the decoder. ",
                                 offset[1], clipped_offset);

                     }

                     sf[idx] = -ff_aac_pow2sf_tab[clipped_offset + POW_SF2_ZERO];

                 }

             } 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,

                                "Scalefactor (%d) out of range.\n", offset[0]);

                         return -1;
                     }

                     sf[idx] = -ff_aac_pow2sf_tab[offset[0] - 100 + POW_SF2_ZERO];

                 }
             }
         }
     }
     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->oc[1].m4ac.object_type == AOT_AAC_MAIN ? 20 : 12;

     for (w = 0; w < ics->num_windows; w++) {

         if ((tns->n_filt[w] = get_bits(gb, 2 - is8))) {

             coef_res = get_bits1(gb);
 

             for (filt = 0; filt < tns->n_filt[w]; filt++) {
                 int tmp2_idx;

                 tns->length[w][filt] = get_bits(gb, 6 - 2 * is8);

                 if ((tns->order[w][filt] = get_bits(gb, 5 - 2 * is8)) > tns_max_order) {

                     av_log(ac->avctx, AV_LOG_ERROR, "TNS filter order %d is greater than maximum %d.\n",

                            tns->order[w][filt], tns_max_order);
                     tns->order[w][filt] = 0;
                     return -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,  sizeof(cpe->ms_mask[0]) * cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb);

     }
 }

 #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 av_intfloat32 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 av_intfloat32 s = { .f = *scale };
     union av_intfloat32 t;

     t.i = s.i ^ (sign & 1U<<31);

     *dst++ = v[idx    & 3] * t.f;
 
     sign <<= nz & 1; nz >>= 1;

     t.i = s.i ^ (sign & 1U<<31);

     *dst++ = v[idx>>2 & 3] * t.f;
 
     sign <<= nz & 1; nz >>= 1;

     t.i = s.i ^ (sign & 1U<<31);

     *dst++ = v[idx>>4 & 3] * t.f;
 

     sign <<= nz & 1;

     t.i = s.i ^ (sign & 1U<<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;
 
     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->fdsp.scalarproduct_float(cfo, cfo, off_len);

                     scale = sf[idx] / sqrtf(band_energy);

                     ac->fdsp.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;

                 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);
                             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);
                             cb_idx = cb_vector_idx[code];

                             nnz = cb_idx >> 8 & 15;

                             bits = nnz ? GET_CACHE(re, gb) : 0;

                             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);
                             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);
                             cb_idx = cb_vector_idx[code];

                             nnz = cb_idx >> 8 & 15;

                             sign = nnz ? SHOW_UBITS(re, gb, nnz) << (cb_idx >> 12) : 0;

                             LAST_SKIP_BITS(re, gb, nnz);

                             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;
                             }
 

                             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;
                                     }

 
                                     SKIP_BITS(re, gb, b + 1);
                                     b += 4;
                                     n = (1 << b) + SHOW_UBITS(re, gb, b);
                                     LAST_SKIP_BITS(re, gb, b);

                                     *icf++ = cbrt_tab[n] | (bits & 1U<<31);

                                     bits <<= 1;
                                 } else {
                                     unsigned v = ((const uint32_t*)vq)[cb_idx & 15];

                                     *icf++ = (bits & 1U<<31) | v;

                                     bits <<= !!v;

                                 }

                                 cb_idx >>= 4;

                             }

                         } while (len -= 2);

                         ac->fdsp.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;
 }
 

 static av_always_inline float flt16_round(float pf)
 {

     union av_intfloat32 tmp;

     tmp.f = pf;
     tmp.i = (tmp.i + 0x00008000U) & 0xFFFF0000U;
     return tmp.f;

 }
 

 static av_always_inline float flt16_even(float pf)
 {

     union av_intfloat32 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 av_intfloat32 pun;

     pun.f = pf;
     pun.i &= 0xFFFF0000U;
     return pun.f;

 }
 

 static av_always_inline void predict(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;

     float   r0 = ps->r0,     r1 = ps->r1;
     float cor0 = ps->cor0, cor1 = ps->cor1;
     float var0 = ps->var0, var1 = ps->var1;

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

     pv = flt16_round(k1 * r0 + k2 * r1);

     if (output_enable)

         *coef += pv;

     e0 = *coef;

     e1 = e0 - k1 * r0;

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

     ps->r1 = flt16_trunc(a * (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->oc[1].m4ac.sampling_index]; sfb++) {

             for (k = sce->ics.swb_offset[sfb]; k < sce->ics.swb_offset[sfb + 1]; k++) {

                 predict(&sce->predictor_state[k], &sce->coeffs[k],

                         sce->ics.predictor_present && sce->ics.prediction_used[sfb]);

             }
         }
         if (sce->ics.predictor_reset_group)

             reset_predictor_group(sce->predictor_state, sce->ics.predictor_reset_group);

     } else

         reset_all_predictors(sce->predictor_state);

 }
 

 /**

  * Decode an individual_channel_stream payload; reference: table 4.44.
  *
  * @param   common_window   Channels have independent [0], or shared [1], Individual Channel Stream information.
  * @param   scale_flag      scalable [1] or non-scalable [0] AAC (Unused until scalable AAC is implemented.)
  *
  * @return  Returns error status. 0 - OK, !0 - error
  */

 static int decode_ics(AACContext *ac, SingleChannelElement *sce,
                       GetBitContext *gb, int common_window, int scale_flag)
 {

     Pulse pulse;

     TemporalNoiseShaping    *tns = &sce->tns;
     IndividualChannelStream *ics = &sce->ics;
     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)
             return AVERROR_INVALIDDATA;

     }
 
     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 AVERROR_PATCHWELCOME;

         }
     }
 

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

         return -1;

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

         apply_prediction(ac, sce);
 

     return 0;
 }
 
 /**

  * Mid/Side stereo decoding; reference: 4.6.8.1.3.
  */

 static void apply_mid_side_stereo(AACContext *ac, ChannelElement *cpe)

 {
     const IndividualChannelStream *ics = &cpe->ch[0].ics;

     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->fdsp.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(AACContext *ac, 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, 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++)

                         ac->fdsp.vector_fmul_scalar(coef1 + group * 128 + offsets[i],
                                                     coef0 + group * 128 + offsets[i],
                                                     scale,
                                                     offsets[i + 1] - offsets[i]);

                 }
             } else {
                 int bt_run_end = sce1->band_type_run_end[idx];
                 idx += bt_run_end - i;
                 i    = bt_run_end;
             }
         }

         coef0 += ics->group_len[g] * 128;
         coef1 += ics->group_len[g] * 128;

     }
 }
 
 /**

  * Decode a channel_pair_element; reference: table 4.4.
  *
  * @return  Returns error status. 0 - OK, !0 - error
  */

 static int decode_cpe(AACContext *ac, GetBitContext *gb, ChannelElement *cpe)
 {

     int i, ret, common_window, ms_present = 0;
 
     common_window = get_bits1(gb);
     if (common_window) {

         if (decode_ics_info(ac, &cpe->ch[0].ics, gb))
             return AVERROR_INVALIDDATA;

         i = cpe->ch[1].ics.use_kb_window[0];
         cpe->ch[1].ics = cpe->ch[0].ics;
         cpe->ch[1].ics.use_kb_window[1] = i;

         if (cpe->ch[1].ics.predictor_present && (ac->oc[1].m4ac.object_type != AOT_AAC_MAIN))

             if ((cpe->ch[1].ics.ltp.present = get_bits(gb, 1)))

                 decode_ltp(&cpe->ch[1].ics.ltp, gb, cpe->ch[1].ics.max_sfb);

         ms_present = get_bits(gb, 2);

         if (ms_present == 3) {

             av_log(ac->avctx, AV_LOG_ERROR, "ms_present = 3 is reserved.\n");

             return -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->oc[1].m4ac.object_type == AOT_AAC_MAIN) {

             apply_prediction(ac, &cpe->ch[0]);
             apply_prediction(ac, &cpe->ch[1]);
         }
     }

     apply_intensity_stereo(ac, cpe, ms_present);

     return 0;
 }
 

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

 /**
  * Decode coupling_channel_element; reference: table 4.8.
  *
  * @return  Returns error status. 0 - OK, !0 - error
  */

 static int decode_cce(AACContext *ac, GetBitContext *gb, ChannelElement *che)
 {

     int num_gain = 0;

     int c, g, sfb, ret;

     int sign;
     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 = cce_scale[get_bits(gb, 2)];

 
     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 = powf(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 = powf(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.
  *
  * @return  Returns number of bytes consumed.
  */

 static int decode_dynamic_range(DynamicRangeControl *che_drc,

                                 GetBitContext *gb)

 {
     int n             = 1;

     int drc_num_bands = 1;
     int i;
 
     /* pce_tag_present? */

     if (get_bits1(gb)) {

         che_drc->pce_instance_tag  = get_bits(gb, 4);
         skip_bits(gb, 4); // tag_reserved_bits
         n++;
     }
 
     /* excluded_chns_present? */

     if (get_bits1(gb)) {

         n += decode_drc_channel_exclusions(che_drc, gb);
     }
 
     /* drc_bands_present? */
     if (get_bits1(gb)) {
         che_drc->band_incr            = get_bits(gb, 4);
         che_drc->interpolation_scheme = get_bits(gb, 4);
         n++;
         drc_num_bands += che_drc->band_incr;
         for (i = 0; i < drc_num_bands; i++) {
             che_drc->band_top[i] = get_bits(gb, 8);
             n++;
         }
     }
 
     /* prog_ref_level_present? */
     if (get_bits1(gb)) {
         che_drc->prog_ref_level = get_bits(gb, 7);
         skip_bits1(gb); // prog_ref_level_reserved_bits
         n++;
     }
 
     for (i = 0; i < drc_num_bands; i++) {
         che_drc->dyn_rng_sgn[i] = get_bits1(gb);
         che_drc->dyn_rng_ctl[i] = get_bits(gb, 7);
         n++;
     }
 
     return n;
 }
 

 static int decode_fill(AACContext *ac, GetBitContext *gb, int len) {
     uint8_t buf[256];
     int i, major, minor;
 
     if (len < 13+7*8)
         goto unknown;
 
     get_bits(gb, 13); len -= 13;
 
     for(i=0; i+1<sizeof(buf) && len>=8; i++, len-=8)
         buf[i] = get_bits(gb, 8);
 
     buf[i] = 0;
     if (ac->avctx->debug & FF_DEBUG_PICT_INFO)
         av_log(ac->avctx, AV_LOG_DEBUG, "FILL:%s\n", buf);
 
     if (sscanf(buf, "libfaac %d.%d", &major, &minor) == 2){
         ac->avctx->internal->skip_samples = 1024;
     }
 
 unknown:
     skip_bits_long(gb, len);
 
     return 0;
 }
 

 /**
  * Decode extension data (incomplete); reference: table 4.51.
  *
  * @param   cnt length of TYPE_FIL syntactic element in bytes
  *
  * @return Returns number of bytes consumed
  */

 static int decode_extension_payload(AACContext *ac, GetBitContext *gb, int cnt,
                                     ChannelElement *che, enum RawDataBlockType elem_type)

 {

     int crc_flag = 0;
     int res = cnt;
     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->oc[1].m4ac.sbr) {
             av_log(ac->avctx, AV_LOG_ERROR, "SBR signaled to be not-present but was found in the bitstream.\n");
             skip_bits_long(gb, 8 * cnt - 4);
             return res;
         } else if (ac->oc[1].m4ac.sbr == -1 && ac->oc[1].status == OC_LOCKED) {
             av_log(ac->avctx, AV_LOG_ERROR, "Implicit SBR was found with a first occurrence after the first frame.\n");
             skip_bits_long(gb, 8 * cnt - 4);
             return res;
         } else if (ac->oc[1].m4ac.ps == -1 && ac->oc[1].status < OC_LOCKED && ac->avctx->channels == 1) {

             ac->oc[1].m4ac.sbr = 1;

             ac->oc[1].m4ac.ps = 1;
             output_configure(ac, ac->oc[1].layout_map, ac->oc[1].layout_map_tags,

                              ac->oc[1].status, 1);

         } else {
             ac->oc[1].m4ac.sbr = 1;

         }
         res = ff_decode_sbr_extension(ac, &che->sbr, gb, crc_flag, cnt, elem_type);

         break;
     case EXT_DYNAMIC_RANGE:

         res = decode_dynamic_range(&ac->che_drc, gb);

         break;
     case EXT_FILL:

         decode_fill(ac, gb, 8 * cnt - 4);
         break;

     case EXT_FILL_DATA:
     case EXT_DATA_ELEMENT:
     default:
         skip_bits_long(gb, 8 * cnt - 4);
         break;

     };
     return res;
 }
 

 /**
  * Decode Temporal Noise Shaping filter coefficients and apply all-pole filters; reference: 4.6.9.3.
  *
  * @param   decode  1 if tool is used normally, 0 if tool is used in LTP.
  * @param   coef    spectral coefficients
  */

 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];

     float tmp[TNS_MAX_ORDER+1];

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

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

             start = ics->swb_offset[FFMIN(bottom, mmm)];
             end   = ics->swb_offset[FFMIN(   top, mmm)];
             if ((size = end - start) <= 0)
                 continue;
             if (tns->direction[w][filt]) {

                 inc = -1;
                 start = end - 1;

             } else {
                 inc = 1;
             }
             start += w * 128;
 

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

         }
     }
 }
 

 /**

  *  Apply windowing and MDCT to obtain the spectral
  *  coefficient from the predicted sample by LTP.
  */
 static void windowing_and_mdct_ltp(AACContext *ac, float *out,
                                    float *in, IndividualChannelStream *ics)
 {
     const float *lwindow      = ics->use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
     const float *swindow      = ics->use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
     const float *lwindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;
     const float *swindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
 
     if (ics->window_sequence[0] != LONG_STOP_SEQUENCE) {

         ac->fdsp.vector_fmul(in, in, lwindow_prev, 1024);

     } else {
         memset(in, 0, 448 * sizeof(float));

         ac->fdsp.vector_fmul(in + 448, in + 448, swindow_prev, 128);

     }
     if (ics->window_sequence[0] != LONG_START_SEQUENCE) {

         ac->fdsp.vector_fmul_reverse(in + 1024, in + 1024, lwindow, 1024);

     } else {

         ac->fdsp.vector_fmul_reverse(in + 1024 + 448, in + 1024 + 448, swindow, 128);

         memset(in + 1024 + 576, 0, 448 * sizeof(float));
     }

     ac->mdct_ltp.mdct_calc(&ac->mdct_ltp, out, in);

 }
 
 /**
  * Apply the long term prediction
  */
 static void apply_ltp(AACContext *ac, SingleChannelElement *sce)
 {
     const LongTermPrediction *ltp = &sce->ics.ltp;
     const uint16_t *offsets = sce->ics.swb_offset;
     int i, sfb;
 
     if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {

         float *predTime = sce->ret;
         float *predFreq = ac->buf_mdct;

         int16_t num_samples = 2048;
 
         if (ltp->lag < 1024)
             num_samples = ltp->lag + 1024;
         for (i = 0; i < num_samples; i++)
             predTime[i] = sce->ltp_state[i + 2048 - ltp->lag] * ltp->coef;
         memset(&predTime[i], 0, (2048 - i) * sizeof(float));
 

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

 
         if (sce->tns.present)

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

 
         for (sfb = 0; sfb < FFMIN(sce->ics.max_sfb, MAX_LTP_LONG_SFB); sfb++)
             if (ltp->used[sfb])
                 for (i = offsets[sfb]; i < offsets[sfb + 1]; i++)
                     sce->coeffs[i] += predFreq[i];
     }
 }
 
 /**
  * Update the LTP buffer for next frame
  */
 static void update_ltp(AACContext *ac, SingleChannelElement *sce)
 {
     IndividualChannelStream *ics = &sce->ics;
     float *saved     = sce->saved;
     float *saved_ltp = sce->coeffs;
     const float *lwindow = ics->use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
     const float *swindow = ics->use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
     int i;
 
     if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
         memcpy(saved_ltp,       saved, 512 * sizeof(float));
         memset(saved_ltp + 576, 0,     448 * sizeof(float));

         ac->fdsp.vector_fmul_reverse(saved_ltp + 448, ac->buf_mdct + 960,     &swindow[64],      64);

         for (i = 0; i < 64; i++)
             saved_ltp[i + 512] = ac->buf_mdct[1023 - i] * swindow[63 - i];

     } else if (ics->window_sequence[0] == LONG_START_SEQUENCE) {
         memcpy(saved_ltp,       ac->buf_mdct + 512, 448 * sizeof(float));
         memset(saved_ltp + 576, 0,                  448 * sizeof(float));

         ac->fdsp.vector_fmul_reverse(saved_ltp + 448, ac->buf_mdct + 960,     &swindow[64],      64);

         for (i = 0; i < 64; i++)
             saved_ltp[i + 512] = ac->buf_mdct[1023 - i] * swindow[63 - i];

     } else { // LONG_STOP or ONLY_LONG

         ac->fdsp.vector_fmul_reverse(saved_ltp,       ac->buf_mdct + 512,     &lwindow[512],     512);

         for (i = 0; i < 512; i++)
             saved_ltp[i + 512] = ac->buf_mdct[1023 - i] * lwindow[511 - i];

     }
 

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

 }
 
 /**

  * Conduct IMDCT and windowing.
  */

 static void imdct_and_windowing(AACContext *ac, SingleChannelElement *sce)

 {
     IndividualChannelStream *ics = &sce->ics;
     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) {

         for (i = 0; i < 1024; i += 128)

             ac->mdct_small.imdct_half(&ac->mdct_small, buf + i, in + i);

     } else

         ac->mdct.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->fdsp.vector_fmul_window(    out,               saved,            buf,         lwindow_prev, 512);

     } else {

         memcpy(                         out,               saved,            448 * sizeof(float));

         if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {

             ac->fdsp.vector_fmul_window(out + 448 + 0*128, saved + 448,      buf + 0*128, swindow_prev, 64);
             ac->fdsp.vector_fmul_window(out + 448 + 1*128, buf + 0*128 + 64, buf + 1*128, swindow,      64);
             ac->fdsp.vector_fmul_window(out + 448 + 2*128, buf + 1*128 + 64, buf + 2*128, swindow,      64);
             ac->fdsp.vector_fmul_window(out + 448 + 3*128, buf + 2*128 + 64, buf + 3*128, swindow,      64);
             ac->fdsp.vector_fmul_window(temp,              buf + 3*128 + 64, buf + 4*128, swindow,      64);
             memcpy(                     out + 448 + 4*128, temp, 64 * sizeof(float));

         } else {

             ac->fdsp.vector_fmul_window(out + 448,         saved + 448,      buf,         swindow_prev, 64);
             memcpy(                     out + 576,         buf + 64,         448 * sizeof(float));

         }
     }

     // buffer update
     if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {

         memcpy(                     saved,       temp + 64,         64 * sizeof(float));
         ac->fdsp.vector_fmul_window(saved + 64,  buf + 4*128 + 64, buf + 5*128, swindow, 64);
         ac->fdsp.vector_fmul_window(saved + 192, buf + 5*128 + 64, buf + 6*128, swindow, 64);
         ac->fdsp.vector_fmul_window(saved + 320, buf + 6*128 + 64, buf + 7*128, swindow, 64);
         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->oc[1].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 *src = cce->ch[0].ret;
     float *dest = target->ret;

     const int len = 1024 << (ac->oc[1].m4ac.sbr == 1);

     for (i = 0; i < len; i++)

         dest[i] += gain * src[i];

 }
 

 /**
  * channel coupling transformation interface
  *
  * @param   apply_coupling_method   pointer to (in)dependent coupling function
  */

 static void apply_channel_coupling(AACContext *ac, ChannelElement *cc,
                                    enum RawDataBlockType type, int elem_id,
                                    enum CouplingPoint coupling_point,
                                    void (*apply_coupling_method)(AACContext *ac, SingleChannelElement *target, ChannelElement *cce, int index))

 {

     int i, c;
 
     for (i = 0; i < MAX_ELEM_ID; i++) {
         ChannelElement *cce = ac->che[TYPE_CCE][i];
         int index = 0;
 
         if (cce && cce->coup.coupling_point == coupling_point) {

             ChannelCoupling *coup = &cce->coup;

 
             for (c = 0; c <= coup->num_coupled; c++) {
                 if (coup->type[c] == type && coup->id_select[c] == elem_id) {
                     if (coup->ch_select[c] != 1) {
                         apply_coupling_method(ac, &cc->ch[0], cce, index);
                         if (coup->ch_select[c] != 0)
                             index++;
                     }
                     if (coup->ch_select[c] != 2)
                         apply_coupling_method(ac, &cc->ch[1], cce, index++);
                 } else
                     index += 1 + (coup->ch_select[c] == 3);

             }
         }
     }
 }
 
 /**
  * Convert spectral data to float samples, applying all supported tools as appropriate.
  */

 static void spectral_to_sample(AACContext *ac)
 {

     int i, type;
     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 (ac->oc[1].m4ac.object_type == AOT_AAC_LTP) {

                     if (che->ch[0].ics.predictor_present) {
                         if (che->ch[0].ics.ltp.present)

                             ac->apply_ltp(ac, &che->ch[0]);

                         if (che->ch[1].ics.ltp.present && type == TYPE_CPE)

                             ac->apply_ltp(ac, &che->ch[1]);

                     }
                 }

                 if (che->ch[0].tns.present)

                     ac->apply_tns(che->ch[0].coeffs, &che->ch[0].tns, &che->ch[0].ics, 1);

                 if (che->ch[1].tns.present)

                     ac->apply_tns(che->ch[1].coeffs, &che->ch[1].tns, &che->ch[1].ics, 1);

                 if (type <= TYPE_CPE)

                     apply_channel_coupling(ac, che, type, i, BETWEEN_TNS_AND_IMDCT, apply_dependent_coupling);

                 if (type != TYPE_CCE || che->coup.coupling_point == AFTER_IMDCT) {

                     ac->imdct_and_windowing(ac, &che->ch[0]);

                     if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP)

                         ac->update_ltp(ac, &che->ch[0]);

                     if (type == TYPE_CPE) {

                         ac->imdct_and_windowing(ac, &che->ch[1]);

                         if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP)

                             ac->update_ltp(ac, &che->ch[1]);

                     }

                     if (ac->oc[1].m4ac.sbr > 0) {

                         ff_sbr_apply(ac, &che->sbr, type, che->ch[0].ret, che->ch[1].ret);
                     }

                 }

                 if (type <= TYPE_CCE)

                     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;

     uint8_t layout_map[MAX_ELEM_ID*4][3];
     int layout_map_tags;

     size = avpriv_aac_parse_header(gb, &hdr_info);

     if (size > 0) {

         if (!ac->warned_num_aac_frames && hdr_info.num_aac_frames != 1) {
             // This is 2 for "VLB " audio in NSV files.
             // See samples/nsv/vlb_audio.

             av_log_missing_feature(ac->avctx, "More than one AAC RDB per ADTS frame", 0);

             ac->warned_num_aac_frames = 1;

         }
         push_output_configuration(ac);

         if (hdr_info.chan_config) {

             ac->oc[1].m4ac.chan_config = hdr_info.chan_config;

             if (set_default_channel_config(ac->avctx, layout_map,
                     &layout_map_tags, hdr_info.chan_config))

                 return -7;

             if (output_configure(ac, layout_map, layout_map_tags,

                                  FFMAX(ac->oc[1].status, OC_TRIAL_FRAME), 0))

                 return -7;

         } else {

             ac->oc[1].m4ac.chan_config = 0;

             /**
              * dual mono frames in Japanese DTV can have chan_config 0
              * WITHOUT specifying PCE.
              *  thus, set dual mono as default.
              */

             if (ac->dmono_mode && ac->oc[0].status == OC_NONE) {

                 layout_map_tags = 2;
                 layout_map[0][0] = layout_map[1][0] = TYPE_SCE;
                 layout_map[0][2] = layout_map[1][2] = AAC_CHANNEL_FRONT;
                 layout_map[0][1] = 0;
                 layout_map[1][1] = 1;

                 if (output_configure(ac, layout_map, layout_map_tags,

                                      OC_TRIAL_FRAME, 0))

                     return -7;
             }

         }

         ac->oc[1].m4ac.sample_rate     = hdr_info.sample_rate;
         ac->oc[1].m4ac.sampling_index  = hdr_info.sampling_index;
         ac->oc[1].m4ac.object_type     = hdr_info.object_type;
         if (ac->oc[0].status != OC_LOCKED ||
             ac->oc[0].m4ac.chan_config != hdr_info.chan_config ||
             ac->oc[0].m4ac.sample_rate != hdr_info.sample_rate) {
             ac->oc[1].m4ac.sbr = -1;
             ac->oc[1].m4ac.ps  = -1;

         }

         if (!hdr_info.crc_absent)
             skip_bits(gb, 16);

     }

     return size;
 }
 

 static int aac_decode_frame_int(AVCodecContext *avctx, void *data,

                                 int *got_frame_ptr, GetBitContext *gb, AVPacket *avpkt)

 {

     AACContext *ac = avctx->priv_data;

     ChannelElement *che = NULL, *che_prev = NULL;
     enum RawDataBlockType elem_type, elem_type_prev = TYPE_END;

     int err, elem_id;

     int samples = 0, multiplier, audio_found = 0, pce_found = 0;

     int is_dmono, sce_count = 0;

     ac->frame = data;
 

     if (show_bits(gb, 12) == 0xfff) {
         if (parse_adts_frame_header(ac, gb) < 0) {

             av_log(avctx, AV_LOG_ERROR, "Error decoding AAC frame header.\n");

             err = -1;
             goto fail;

         }

         if (ac->oc[1].m4ac.sampling_index > 12) {
             av_log(ac->avctx, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->oc[1].m4ac.sampling_index);
             err = -1;
             goto fail;

         }

     }
 

     if (frame_configure_elements(avctx) < 0) {
         err = -1;
         goto fail;
     }
 

     ac->tags_mapped = 0;

     // 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);

                 err = -1;
                 goto fail;

             }

             samples = 1024;

         }

         switch (elem_type) {
 
         case TYPE_SCE:

             err = decode_ics(ac, &che->ch[0], gb, 0, 0);

             audio_found = 1;

             sce_count++;

             break;
 
         case TYPE_CPE:

             err = decode_cpe(ac, gb, che);

             audio_found = 1;

             break;
 
         case TYPE_CCE:

             err = decode_cce(ac, gb, che);

             break;
 
         case TYPE_LFE:

             err = decode_ics(ac, &che->ch[0], gb, 0, 0);

             audio_found = 1;

             break;
 
         case TYPE_DSE:

             err = skip_data_stream_element(ac, gb);

             break;
 

         case TYPE_PCE: {

             uint8_t layout_map[MAX_ELEM_ID*4][3];
             int tags;

             push_output_configuration(ac);
             tags = decode_pce(avctx, &ac->oc[1].m4ac, layout_map, gb);

             if (tags < 0) {
                 err = tags;

                 break;

             }

             if (pce_found) {

                 av_log(avctx, AV_LOG_ERROR,
                        "Not evaluating a further program_config_element as this construct is dubious at best.\n");
             } else {

                 err = output_configure(ac, layout_map, tags, OC_TRIAL_PCE, 1);

                 if (!err)
                     ac->oc[1].m4ac.chan_config = 0;
                 pce_found = 1;

             }

             break;
         }
 
         case TYPE_FIL:
             if (elem_id == 15)

                 elem_id += get_bits(gb, 8) - 1;
             if (get_bits_left(gb) < 8 * elem_id) {

                     av_log(avctx, AV_LOG_ERROR, "TYPE_FIL: "overread_err);

                     err = -1;
                     goto fail;

             }

             while (elem_id > 0)

                 elem_id -= decode_extension_payload(ac, gb, elem_id, che_prev, elem_type_prev);

             err = 0; /* FIXME */
             break;
 
         default:
             err = -1; /* should not happen, but keeps compiler happy */
             break;
         }
 

         che_prev       = che;
         elem_type_prev = elem_type;
 

         if (err)

             goto fail;

         if (get_bits_left(gb) < 3) {

             av_log(avctx, AV_LOG_ERROR, overread_err);

             err = -1;
             goto fail;

         }

     }
 
     spectral_to_sample(ac);
 

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

     samples <<= multiplier;

     /* for dual-mono audio (SCE + SCE) */

     is_dmono = ac->dmono_mode && sce_count == 2 &&

                ac->oc[1].channel_layout == (AV_CH_FRONT_LEFT | AV_CH_FRONT_RIGHT);
 

     if (samples)
         ac->frame->nb_samples = samples;

     *got_frame_ptr = !!samples;

     if (is_dmono) {

         if (ac->dmono_mode == 1)

             ((AVFrame *)data)->data[1] =((AVFrame *)data)->data[0];

         else if (ac->dmono_mode == 2)

             ((AVFrame *)data)->data[0] =((AVFrame *)data)->data[1];

     }

     if (ac->oc[1].status && audio_found) {
         avctx->sample_rate = ac->oc[1].m4ac.sample_rate << multiplier;
         avctx->frame_size = samples;
         ac->oc[1].status = OC_LOCKED;
     }

     if (multiplier) {
         int side_size;

         const uint8_t *side = av_packet_get_side_data(avpkt, AV_PKT_DATA_SKIP_SAMPLES, &side_size);

         if (side && side_size>=4)
             AV_WL32(side, 2*AV_RL32(side));
     }

     return 0;

 fail:
     pop_output_configuration(ac);
     return err;

 }
 
 static int aac_decode_frame(AVCodecContext *avctx, void *data,

                             int *got_frame_ptr, AVPacket *avpkt)

 {

     AACContext *ac = avctx->priv_data;

     const uint8_t *buf = avpkt->data;
     int buf_size = avpkt->size;
     GetBitContext gb;
     int buf_consumed;
     int buf_offset;
     int err;

     int new_extradata_size;
     const uint8_t *new_extradata = av_packet_get_side_data(avpkt,
                                        AV_PKT_DATA_NEW_EXTRADATA,
                                        &new_extradata_size);

     int jp_dualmono_size;
     const uint8_t *jp_dualmono   = av_packet_get_side_data(avpkt,
                                        AV_PKT_DATA_JP_DUALMONO,
                                        &jp_dualmono_size);

     if (new_extradata && 0) {

         av_free(avctx->extradata);
         avctx->extradata = av_mallocz(new_extradata_size +
                                       FF_INPUT_BUFFER_PADDING_SIZE);
         if (!avctx->extradata)
             return AVERROR(ENOMEM);
         avctx->extradata_size = new_extradata_size;
         memcpy(avctx->extradata, new_extradata, new_extradata_size);

         push_output_configuration(ac);
         if (decode_audio_specific_config(ac, ac->avctx, &ac->oc[1].m4ac,

                                          avctx->extradata,

                                          avctx->extradata_size*8, 1) < 0) {
             pop_output_configuration(ac);

             return AVERROR_INVALIDDATA;

         }

     }

     ac->dmono_mode = 0;
     if (jp_dualmono && jp_dualmono_size > 0)

         ac->dmono_mode =  1 + *jp_dualmono;

     if (ac->force_dmono_mode >= 0)
         ac->dmono_mode = ac->force_dmono_mode;

     if (INT_MAX / 8 <= buf_size)
         return AVERROR_INVALIDDATA;
 

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

     if ((err = aac_decode_frame_int(avctx, data, got_frame_ptr, &gb, avpkt)) < 0)

         return err;
 

     buf_consumed = (get_bits_count(&gb) + 7) >> 3;

     for (buf_offset = buf_consumed; buf_offset < buf_size; buf_offset++)
         if (buf[buf_offset])
             break;
 
     return buf_size > buf_offset ? buf_consumed : buf_size;

 }
 

 static av_cold int aac_decode_close(AVCodecContext *avctx)

 {

     AACContext *ac = avctx->priv_data;

     int i, type;

     for (i = 0; i < MAX_ELEM_ID; i++) {

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

             av_freep(&ac->che[type][i]);

         }

     }
 
     ff_mdct_end(&ac->mdct);
     ff_mdct_end(&ac->mdct_small);

     ff_mdct_end(&ac->mdct_ltp);

     return 0;

 }
 

 
 #define LOAS_SYNC_WORD   0x2b7       ///< 11 bits LOAS sync word
 
 struct LATMContext {
     AACContext      aac_ctx;             ///< containing AACContext

     int             initialized;         ///< initialized after a valid extradata was seen

 
     // parser data
     int             audio_mux_version_A; ///< LATM syntax version
     int             frame_length_type;   ///< 0/1 variable/fixed frame length
     int             frame_length;        ///< frame length for fixed frame length
 };
 
 static inline uint32_t latm_get_value(GetBitContext *b)
 {
     int length = get_bits(b, 2);
 
     return get_bits_long(b, (length+1)*8);
 }
 
 static int latm_decode_audio_specific_config(struct LATMContext *latmctx,

                                              GetBitContext *gb, int asclen)

 {

     AACContext *ac        = &latmctx->aac_ctx;
     AVCodecContext *avctx = ac->avctx;

     MPEG4AudioConfig m4ac = { 0 };

     int config_start_bit  = get_bits_count(gb);
     int sync_extension    = 0;
     int bits_consumed, esize;
 
     if (asclen) {
         sync_extension = 1;
         asclen         = FFMIN(asclen, get_bits_left(gb));
     } else
         asclen         = get_bits_left(gb);

 
     if (config_start_bit % 8) {

         av_log_missing_feature(latmctx->aac_ctx.avctx,
                                "Non-byte-aligned audio-specific config", 1);

         return AVERROR_PATCHWELCOME;

     }

     if (asclen <= 0)
         return AVERROR_INVALIDDATA;

     bits_consumed = decode_audio_specific_config(NULL, avctx, &m4ac,

                                          gb->buffer + (config_start_bit / 8),

                                          asclen, sync_extension);

     if (bits_consumed < 0)
         return AVERROR_INVALIDDATA;
 

     if (!latmctx->initialized ||
         ac->oc[1].m4ac.sample_rate != m4ac.sample_rate ||

         ac->oc[1].m4ac.chan_config != m4ac.chan_config) {

         if(latmctx->initialized) {
             av_log(avctx, AV_LOG_INFO, "audio config changed\n");
         } else {

             av_log(avctx, AV_LOG_DEBUG, "initializing latmctx\n");

         }

         latmctx->initialized = 0;

 
         esize = (bits_consumed+7) / 8;
 

         if (avctx->extradata_size < esize) {

             av_free(avctx->extradata);
             avctx->extradata = av_malloc(esize + FF_INPUT_BUFFER_PADDING_SIZE);
             if (!avctx->extradata)
                 return AVERROR(ENOMEM);
         }
 
         avctx->extradata_size = esize;
         memcpy(avctx->extradata, gb->buffer + (config_start_bit/8), esize);
         memset(avctx->extradata+esize, 0, FF_INPUT_BUFFER_PADDING_SIZE);
     }

     skip_bits_long(gb, bits_consumed);

 
     return bits_consumed;
 }
 
 static int read_stream_mux_config(struct LATMContext *latmctx,
                                   GetBitContext *gb)
 {
     int ret, audio_mux_version = get_bits(gb, 1);
 
     latmctx->audio_mux_version_A = 0;
     if (audio_mux_version)
         latmctx->audio_mux_version_A = get_bits(gb, 1);
 
     if (!latmctx->audio_mux_version_A) {
 
         if (audio_mux_version)
             latm_get_value(gb);                 // taraFullness
 
         skip_bits(gb, 1);                       // allStreamSameTimeFraming
         skip_bits(gb, 6);                       // numSubFrames
         // numPrograms
         if (get_bits(gb, 4)) {                  // numPrograms
             av_log_missing_feature(latmctx->aac_ctx.avctx,

                                    "Multiple programs", 1);

             return AVERROR_PATCHWELCOME;
         }
 

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

         // for each layer (which there is only one in DVB)

         if (get_bits(gb, 3)) {                   // numLayer
             av_log_missing_feature(latmctx->aac_ctx.avctx,

                                    "Multiple layers", 1);

             return AVERROR_PATCHWELCOME;
         }
 
         // for all but first stream: use_same_config = get_bits(gb, 1);
         if (!audio_mux_version) {

             if ((ret = latm_decode_audio_specific_config(latmctx, gb, 0)) < 0)

                 return ret;
         } else {
             int ascLen = latm_get_value(gb);

             if ((ret = latm_decode_audio_specific_config(latmctx, gb, ascLen)) < 0)

                 return ret;
             ascLen -= ret;
             skip_bits_long(gb, ascLen);
         }
 
         latmctx->frame_length_type = get_bits(gb, 3);
         switch (latmctx->frame_length_type) {
         case 0:
             skip_bits(gb, 8);       // latmBufferFullness
             break;
         case 1:
             latmctx->frame_length = get_bits(gb, 9);
             break;
         case 3:
         case 4:
         case 5:
             skip_bits(gb, 6);       // CELP frame length table index
             break;
         case 6:
         case 7:
             skip_bits(gb, 1);       // HVXC frame length table index
             break;
         }
 
         if (get_bits(gb, 1)) {                  // other data
             if (audio_mux_version) {
                 latm_get_value(gb);             // other_data_bits
             } else {
                 int esc;
                 do {
                     esc = get_bits(gb, 1);
                     skip_bits(gb, 8);
                 } while (esc);
             }
         }
 
         if (get_bits(gb, 1))                     // crc present
             skip_bits(gb, 8);                    // config_crc
     }
 
     return 0;
 }
 
 static int read_payload_length_info(struct LATMContext *ctx, GetBitContext *gb)
 {
     uint8_t tmp;
 
     if (ctx->frame_length_type == 0) {
         int mux_slot_length = 0;
         do {
             tmp = get_bits(gb, 8);
             mux_slot_length += tmp;
         } while (tmp == 255);
         return mux_slot_length;
     } else if (ctx->frame_length_type == 1) {
         return ctx->frame_length;
     } else if (ctx->frame_length_type == 3 ||
                ctx->frame_length_type == 5 ||
                ctx->frame_length_type == 7) {
         skip_bits(gb, 2);          // mux_slot_length_coded
     }
     return 0;
 }
 
 static int read_audio_mux_element(struct LATMContext *latmctx,
                                   GetBitContext *gb)
 {
     int err;
     uint8_t use_same_mux = get_bits(gb, 1);
     if (!use_same_mux) {
         if ((err = read_stream_mux_config(latmctx, gb)) < 0)
             return err;
     } else if (!latmctx->aac_ctx.avctx->extradata) {
         av_log(latmctx->aac_ctx.avctx, AV_LOG_DEBUG,
                "no decoder config found\n");
         return AVERROR(EAGAIN);
     }
     if (latmctx->audio_mux_version_A == 0) {
         int mux_slot_length_bytes = read_payload_length_info(latmctx, gb);
         if (mux_slot_length_bytes * 8 > get_bits_left(gb)) {
             av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR, "incomplete frame\n");
             return AVERROR_INVALIDDATA;
         } else if (mux_slot_length_bytes * 8 + 256 < get_bits_left(gb)) {
             av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR,
                    "frame length mismatch %d << %d\n",
                    mux_slot_length_bytes * 8, get_bits_left(gb));
             return AVERROR_INVALIDDATA;
         }
     }
     return 0;
 }
 
 

 static int latm_decode_frame(AVCodecContext *avctx, void *out,
                              int *got_frame_ptr, AVPacket *avpkt)

 {
     struct LATMContext *latmctx = avctx->priv_data;
     int                 muxlength, err;
     GetBitContext       gb;
 

     if ((err = init_get_bits8(&gb, avpkt->data, avpkt->size)) < 0)
         return err;

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

     muxlength = get_bits(&gb, 13) + 3;

     // not enough data, the parser should have sorted this out

     if (muxlength > avpkt->size)

         return AVERROR_INVALIDDATA;
 
     if ((err = read_audio_mux_element(latmctx, &gb)) < 0)
         return err;
 
     if (!latmctx->initialized) {
         if (!avctx->extradata) {

             *got_frame_ptr = 0;

             return avpkt->size;
         } else {

             push_output_configuration(&latmctx->aac_ctx);

             if ((err = decode_audio_specific_config(

                     &latmctx->aac_ctx, avctx, &latmctx->aac_ctx.oc[1].m4ac,
                     avctx->extradata, avctx->extradata_size*8, 1)) < 0) {
                 pop_output_configuration(&latmctx->aac_ctx);

                 return err;

             }

             latmctx->initialized = 1;
         }
     }
 
     if (show_bits(&gb, 12) == 0xfff) {
         av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR,
                "ADTS header detected, probably as result of configuration "
                "misparsing\n");
         return AVERROR_INVALIDDATA;
     }
 

     if ((err = aac_decode_frame_int(avctx, out, got_frame_ptr, &gb, avpkt)) < 0)

         return err;
 
     return muxlength;
 }
 

 static av_cold int latm_decode_init(AVCodecContext *avctx)

 {
     struct LATMContext *latmctx = avctx->priv_data;

     int ret = aac_decode_init(avctx);

     if (avctx->extradata_size > 0)

         latmctx->initialized = !ret;
 
     return ret;
 }
 

 static void aacdec_init(AACContext *c)
 {
     c->imdct_and_windowing                      = imdct_and_windowing;
     c->apply_ltp                                = apply_ltp;
     c->apply_tns                                = apply_tns;
     c->windowing_and_mdct_ltp                   = windowing_and_mdct_ltp;
     c->update_ltp                               = update_ltp;
 
     if(ARCH_MIPS)
         ff_aacdec_init_mips(c);
 }

 /**
  * AVOptions for Japanese DTV specific extensions (ADTS only)
  */
 #define AACDEC_FLAGS AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM
 static const AVOption options[] = {
     {"dual_mono_mode", "Select the channel to decode for dual mono",
      offsetof(AACContext, force_dmono_mode), AV_OPT_TYPE_INT, {.i64=-1}, -1, 2,
      AACDEC_FLAGS, "dual_mono_mode"},
 
     {"auto", "autoselection",            0, AV_OPT_TYPE_CONST, {.i64=-1}, INT_MIN, INT_MAX, AACDEC_FLAGS, "dual_mono_mode"},
     {"main", "Select Main/Left channel", 0, AV_OPT_TYPE_CONST, {.i64= 1}, INT_MIN, INT_MAX, AACDEC_FLAGS, "dual_mono_mode"},
     {"sub" , "Select Sub/Right channel", 0, AV_OPT_TYPE_CONST, {.i64= 2}, INT_MIN, INT_MAX, AACDEC_FLAGS, "dual_mono_mode"},
     {"both", "Select both channels",     0, AV_OPT_TYPE_CONST, {.i64= 0}, INT_MIN, INT_MAX, AACDEC_FLAGS, "dual_mono_mode"},
 
     {NULL},
 };
 
 static const AVClass aac_decoder_class = {
     .class_name = "AAC decoder",
     .item_name  = av_default_item_name,
     .option     = options,
     .version    = LIBAVUTIL_VERSION_INT,
 };

 AVCodec ff_aac_decoder = {

     .name            = "aac",
     .type            = AVMEDIA_TYPE_AUDIO,

     .id              = AV_CODEC_ID_AAC,

     .priv_data_size  = sizeof(AACContext),
     .init            = aac_decode_init,
     .close           = aac_decode_close,
     .decode          = aac_decode_frame,

     .long_name       = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"),

     .sample_fmts     = (const enum AVSampleFormat[]) {

         AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE

     },

     .capabilities    = CODEC_CAP_CHANNEL_CONF | CODEC_CAP_DR1,

     .channel_layouts = aac_channel_layout,

     .flush = flush,

     .priv_class      = &aac_decoder_class,

 };

 
 /*
     Note: This decoder filter is intended to decode LATM streams transferred
     in MPEG transport streams which only contain one program.
     To do a more complex LATM demuxing a separate LATM demuxer should be used.
 */

 AVCodec ff_aac_latm_decoder = {

     .name            = "aac_latm",
     .type            = AVMEDIA_TYPE_AUDIO,

     .id              = AV_CODEC_ID_AAC_LATM,

     .priv_data_size  = sizeof(struct LATMContext),
     .init            = latm_decode_init,
     .close           = aac_decode_close,
     .decode          = latm_decode_frame,

     .long_name       = NULL_IF_CONFIG_SMALL("AAC LATM (Advanced Audio Coding LATM syntax)"),

     .sample_fmts     = (const enum AVSampleFormat[]) {

         AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE

     },

     .capabilities    = CODEC_CAP_CHANNEL_CONF | CODEC_CAP_DR1,

     .channel_layouts = aac_channel_layout,

     .flush = flush,

 };