/*
  * AAC decoder
  * Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org )
  * Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com )
  * Copyright (c) 2008-2013 Alex Converse <alex.converse@gmail.com>
  *
  * AAC LATM decoder
  * Copyright (c) 2008-2010 Paul Kendall <paul@kcbbs.gen.nz>
  * Copyright (c) 2010      Janne Grunau <janne-libav@jannau.net>
  *

  * AAC decoder fixed-point implementation
  * Copyright (c) 2013
  *      MIPS Technologies, Inc., California.
  *

  * 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 )

  *
  * AAC decoder fixed-point implementation
  * @author Stanislav Ocovaj ( stanislav.ocovaj imgtec com )
  * @author Nedeljko Babic ( nedeljko.babic imgtec 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
  * Y  (not in fixed point code) Enhanced AAC Low Delay (ER AAC ELD)
  *
  * 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/thread.h"
 

 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 (*channels >= MAX_CHANNELS)
         return AVERROR_INVALIDDATA;
     if (che_pos) {
         if (!ac->che[type][id]) {
             if (!(ac->che[type][id] = av_mallocz(sizeof(ChannelElement))))
                 return AVERROR(ENOMEM);

             AAC_RENAME(ff_aac_sbr_ctx_init)(ac, &ac->che[type][id]->sbr, type);

         }
         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])

             AAC_RENAME(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 */
     av_frame_unref(ac->frame);
     if (!avctx->channels)
         return 1;
 
     ac->frame->nb_samples = 2048;
     if ((ret = ff_get_buffer(avctx, ac->frame, 0)) < 0)
         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 = (INTFLOAT *)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;
 
     if (num_side_channels == 0 && num_back_channels >= 4) {
         num_side_channels = 2;
         num_back_channels -= 2;
     }
 
     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 int push_output_configuration(AACContext *ac) {
     int pushed = 0;
 

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

         pushed = 1;

     }
     ac->oc[1].status = OC_NONE;

     return pushed;

 }
 
 /**
  * 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;
     uint8_t id_map[TYPE_END][MAX_ELEM_ID] = {{ 0 }};
     uint8_t type_counts[TYPE_END] = { 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;
     }
     for (i = 0; i < tags; i++) {
         int type =         layout_map[i][0];
         int id =           layout_map[i][1];
         id_map[type][id] = type_counts[type]++;

         if (id_map[type][id] >= MAX_ELEM_ID) {

             avpriv_request_sample(ac->avctx, "Too large remapped id");

             return AVERROR_PATCHWELCOME;
         }

     }
     // 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 iid =      id_map[type][id];
         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, iid, &channels);
         if (ret < 0)
             return ret;
         ac->tag_che_map[type][id] = ac->che[type][iid];
     }
     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;
         }
     }
 
     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 && channel_config < 11) ||
         channel_config > 12) {
         av_log(avctx, AV_LOG_ERROR,
                "invalid default channel configuration (%d)\n",
                channel_config);
         return AVERROR_INVALIDDATA;
     }
     *tags = tags_per_config[channel_config];
     memcpy(layout_map, aac_channel_layout_map[channel_config - 1],
            *tags * sizeof(*layout_map));
 
     /*
      * AAC specification has 7.1(wide) as a default layout for 8-channel streams.
      * However, at least Nero AAC encoder encodes 7.1 streams using the default
      * channel config 7, mapping the side channels of the original audio stream
      * to the second AAC_CHANNEL_FRONT pair in the AAC stream. Similarly, e.g. FAAD
      * decodes the second AAC_CHANNEL_FRONT pair as side channels, therefore decoding
      * the incorrect streams as if they were correct (and as the encoder intended).
      *
      * As actual intended 7.1(wide) streams are very rare, default to assuming a
      * 7.1 layout was intended.
      */
     if (channel_config == 7 && avctx->strict_std_compliance < FF_COMPLIANCE_STRICT) {
         av_log(avctx, AV_LOG_INFO, "Assuming an incorrectly encoded 7.1 channel layout"
                " instead of a spec-compliant 7.1(wide) layout, use -strict %d to decode"
                " according to the specification instead.\n", FF_COMPLIANCE_STRICT);
         layout_map[2][2] = AAC_CHANNEL_SIDE;
     }
 
     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 12:
     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 11:
         if (ac->tags_mapped == 2 &&
             ac->oc[1].m4ac.chan_config == 11 &&
             type == TYPE_SCE) {
             ac->tags_mapped++;
             return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][1];
         }
     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)) {
             if (!ac->warned_remapping_once && (type != TYPE_LFE || elem_id != 0)) {
                 av_log(ac->avctx, AV_LOG_WARNING,
                    "This stream seems to incorrectly report its last channel as %s[%d], mapping to LFE[0]\n",
                    type == TYPE_SCE ? "SCE" : "LFE", elem_id);
                 ac->warned_remapping_once++;
             }
             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:
         /* Some streams incorrectly code 4.0 audio as
          * SCE[0] CPE[0] LFE[0]
          * instead of
          * SCE[0] CPE[0] SCE[1].
          * If we seem to have encountered such a stream, transfer
          * the SCE[1] element to the LFE[0]'s mapping */
         if (ac->tags_mapped == tags_per_config[ac->oc[1].m4ac.chan_config] - 1 && (type == TYPE_LFE || type == TYPE_SCE)) {
             if (!ac->warned_remapping_once && (type != TYPE_SCE || elem_id != 1)) {
                 av_log(ac->avctx, AV_LOG_WARNING,
                    "This stream seems to incorrectly report its last channel as %s[%d], mapping to SCE[1]\n",
                    type == TYPE_SCE ? "SCE" : "LFE", elem_id);
                 ac->warned_remapping_once++;
             }
             ac->tags_mapped++;
             return ac->tag_che_map[type][elem_id] = ac->che[TYPE_SCE][1];
         }
         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:
             // AAC_CHANNEL_OFF has no channel map
             av_assert0(0);
         }
         layout_map[0][0] = syn_ele;
         layout_map[0][1] = get_bits(gb, 4);
         layout_map[0][2] = type;
         layout_map++;
     }
 }
 

 static inline void relative_align_get_bits(GetBitContext *gb,
                                            int reference_position) {
     int n = (reference_position - get_bits_count(gb) & 7);
     if (n)
         skip_bits(gb, n);
 }
 

 /**
  * 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 byte_align_ref)

 {
     int num_front, num_side, num_back, num_lfe, num_assoc_data, num_cc;
     int 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) < 5 * (num_front + num_side + num_back + num_cc) + 4 *(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;
 

     relative_align_get_bits(gb, byte_align_ref);

 
     /* 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 AVERROR_INVALIDDATA;
     }
     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,

                                      int get_bit_alignment,

                                      MPEG4AudioConfig *m4ac,
                                      int channel_config)
 {
     int extension_flag, ret, ep_config, res_flags;
     uint8_t layout_map[MAX_ELEM_ID*4][3];
     int tags = 0;
 

 #if USE_FIXED

     if (get_bits1(gb)) { // frameLengthFlag

         avpriv_report_missing_feature(avctx, "Fixed point 960/120 MDCT window");

         return AVERROR_PATCHWELCOME;
     }
     m4ac->frame_length_short = 0;

 #else
     m4ac->frame_length_short = get_bits1(gb);
     if (m4ac->frame_length_short && m4ac->sbr == 1) {
       avpriv_report_missing_feature(avctx, "SBR with 960 frame length");
       if (ac) ac->warned_960_sbr = 1;
       m4ac->sbr = 0;
       m4ac->ps = 0;
     }
 #endif

 
     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, get_bit_alignment);

         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:
             res_flags = get_bits(gb, 3);
             if (res_flags) {
                 avpriv_report_missing_feature(avctx,
                                               "AAC data resilience (flags %x)",
                                               res_flags);
                 return AVERROR_PATCHWELCOME;
             }
             break;
         }
         skip_bits1(gb);    // extensionFlag3 (TBD in version 3)
     }
     switch (m4ac->object_type) {
     case AOT_ER_AAC_LC:
     case AOT_ER_AAC_LTP:
     case AOT_ER_AAC_SCALABLE:
     case AOT_ER_AAC_LD:
         ep_config = get_bits(gb, 2);
         if (ep_config) {
             avpriv_report_missing_feature(avctx,
                                           "epConfig %d", ep_config);
             return AVERROR_PATCHWELCOME;
         }
     }
     return 0;
 }
 
 static int decode_eld_specific_config(AACContext *ac, AVCodecContext *avctx,
                                      GetBitContext *gb,
                                      MPEG4AudioConfig *m4ac,
                                      int channel_config)
 {
     int ret, ep_config, res_flags;
     uint8_t layout_map[MAX_ELEM_ID*4][3];
     int tags = 0;
     const int ELDEXT_TERM = 0;
 
     m4ac->ps  = 0;
     m4ac->sbr = 0;

 #if USE_FIXED
     if (get_bits1(gb)) { // frameLengthFlag
         avpriv_request_sample(avctx, "960/120 MDCT window");
         return AVERROR_PATCHWELCOME;
     }
 #else

     m4ac->frame_length_short = get_bits1(gb);

 #endif

     res_flags = get_bits(gb, 3);
     if (res_flags) {
         avpriv_report_missing_feature(avctx,
                                       "AAC data resilience (flags %x)",
                                       res_flags);
         return AVERROR_PATCHWELCOME;
     }
 
     if (get_bits1(gb)) { // ldSbrPresentFlag
         avpriv_report_missing_feature(avctx,
                                       "Low Delay SBR");
         return AVERROR_PATCHWELCOME;
     }
 
     while (get_bits(gb, 4) != ELDEXT_TERM) {
         int len = get_bits(gb, 4);
         if (len == 15)
             len += get_bits(gb, 8);
         if (len == 15 + 255)
             len += get_bits(gb, 16);
         if (get_bits_left(gb) < len * 8 + 4) {
             av_log(avctx, AV_LOG_ERROR, overread_err);
             return AVERROR_INVALIDDATA;
         }
         skip_bits_long(gb, 8 * len);
     }
 
     if ((ret = set_default_channel_config(avctx, layout_map,
                                           &tags, channel_config)))
         return ret;
 
     if (ac && (ret = output_configure(ac, layout_map, tags, OC_GLOBAL_HDR, 0)))
         return ret;
 
     ep_config = get_bits(gb, 2);
     if (ep_config) {
         avpriv_report_missing_feature(avctx,
                                       "epConfig %d", ep_config);
         return AVERROR_PATCHWELCOME;
     }
     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   gb          buffer holding an audio specific config
  * @param   get_bit_alignment relative alignment for byte align operations

  * @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_gb(AACContext *ac,
                                            AVCodecContext *avctx,
                                            MPEG4AudioConfig *m4ac,
                                            GetBitContext *gb,
                                            int get_bit_alignment,
                                            int sync_extension)

 {
     int i, ret;

     GetBitContext gbc = *gb;

     if ((i = ff_mpeg4audio_get_config_gb(m4ac, &gbc, sync_extension)) < 0)

         return AVERROR_INVALIDDATA;
 

     if (m4ac->sampling_index > 12) {
         av_log(avctx, AV_LOG_ERROR,
                "invalid sampling rate index %d\n",
                m4ac->sampling_index);
         return AVERROR_INVALIDDATA;
     }
     if (m4ac->object_type == AOT_ER_AAC_LD &&
         (m4ac->sampling_index < 3 || m4ac->sampling_index > 7)) {
         av_log(avctx, AV_LOG_ERROR,
                "invalid low delay sampling rate index %d\n",
                m4ac->sampling_index);
         return AVERROR_INVALIDDATA;
     }
 

     skip_bits_long(gb, i);

 
     switch (m4ac->object_type) {
     case AOT_AAC_MAIN:
     case AOT_AAC_LC:
     case AOT_AAC_LTP:
     case AOT_ER_AAC_LC:
     case AOT_ER_AAC_LD:

         if ((ret = decode_ga_specific_config(ac, avctx, gb, get_bit_alignment,

                                             m4ac, m4ac->chan_config)) < 0)
             return ret;
         break;
     case AOT_ER_AAC_ELD:

         if ((ret = decode_eld_specific_config(ac, avctx, gb,

                                               m4ac, m4ac->chan_config)) < 0)
             return ret;
         break;
     default:
         avpriv_report_missing_feature(avctx,
                                       "Audio object type %s%d",
                                       m4ac->sbr == 1 ? "SBR+" : "",
                                       m4ac->object_type);
         return AVERROR(ENOSYS);
     }
 
     ff_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);
 }
 
 static int decode_audio_specific_config(AACContext *ac,
                                         AVCodecContext *avctx,
                                         MPEG4AudioConfig *m4ac,
                                         const uint8_t *data, int64_t bit_size,
                                         int sync_extension)
 {
     int i, ret;
     GetBitContext gb;
 
     if (bit_size < 0 || bit_size > INT_MAX) {
         av_log(avctx, AV_LOG_ERROR, "Audio specific config size is invalid\n");
         return AVERROR_INVALIDDATA;
     }
 
     ff_dlog(avctx, "audio specific config size %d\n", (int)bit_size >> 3);
     for (i = 0; i < bit_size >> 3; i++)
         ff_dlog(avctx, "%02x ", data[i]);
     ff_dlog(avctx, "\n");
 
     if ((ret = init_get_bits(&gb, data, bit_size)) < 0)
         return ret;
 
     return decode_audio_specific_config_gb(ac, avctx, m4ac, &gb, 0,
                                            sync_extension);

 }
 
 /**
  * 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 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 void aac_static_table_init(void)

 {
     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);
 
     AAC_RENAME(ff_aac_sbr_init)();
 
     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);
 
     // window initialization
     AAC_RENAME(ff_kbd_window_init)(AAC_RENAME(ff_aac_kbd_long_1024), 4.0, 1024);
     AAC_RENAME(ff_kbd_window_init)(AAC_RENAME(ff_aac_kbd_short_128), 6.0, 128);

 #if !USE_FIXED
     AAC_RENAME(ff_kbd_window_init)(AAC_RENAME(ff_aac_kbd_long_960), 4.0, 960);
     AAC_RENAME(ff_kbd_window_init)(AAC_RENAME(ff_aac_kbd_short_120), 6.0, 120);
     AAC_RENAME(ff_sine_window_init)(AAC_RENAME(ff_sine_960), 960);
     AAC_RENAME(ff_sine_window_init)(AAC_RENAME(ff_sine_120), 120);
 #endif

     AAC_RENAME(ff_init_ff_sine_windows)(10);
     AAC_RENAME(ff_init_ff_sine_windows)( 9);
     AAC_RENAME(ff_init_ff_sine_windows)( 7);
 

     AAC_RENAME(ff_cbrt_tableinit)();

 }
 

 static AVOnce aac_table_init = AV_ONCE_INIT;
 

 static av_cold int aac_decode_init(AVCodecContext *avctx)
 {
     AACContext *ac = avctx->priv_data;
     int ret;
 

     ret = ff_thread_once(&aac_table_init, &aac_static_table_init);

     if (ret != 0)
         return AVERROR_UNKNOWN;
 

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

 #if USE_FIXED
     avctx->sample_fmt = AV_SAMPLE_FMT_S32P;
 #else

     avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;

 #endif /* USE_FIXED */

 
     if (avctx->extradata_size > 0) {
         if ((ret = decode_audio_specific_config(ac, ac->avctx, &ac->oc[1].m4ac,
                                                 avctx->extradata,

                                                 avctx->extradata_size * 8LL,

                                                 1)) < 0)
             return ret;
     } 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;
     }
 

 #if USE_FIXED

     ac->fdsp = avpriv_alloc_fixed_dsp(avctx->flags & AV_CODEC_FLAG_BITEXACT);

 #else

     ac->fdsp = avpriv_float_dsp_alloc(avctx->flags & AV_CODEC_FLAG_BITEXACT);

 #endif /* USE_FIXED */

     if (!ac->fdsp) {
         return AVERROR(ENOMEM);
     }
 
     ac->random_state = 0x1f2e3d4c;
 

     AAC_RENAME_32(ff_mdct_init)(&ac->mdct,       11, 1, 1.0 / RANGE15(1024.0));
     AAC_RENAME_32(ff_mdct_init)(&ac->mdct_ld,    10, 1, 1.0 / RANGE15(512.0));
     AAC_RENAME_32(ff_mdct_init)(&ac->mdct_small,  8, 1, 1.0 / RANGE15(128.0));
     AAC_RENAME_32(ff_mdct_init)(&ac->mdct_ltp,   11, 0, RANGE15(-2.0));
 #if !USE_FIXED

     ret = ff_mdct15_init(&ac->mdct120, 1, 3, 1.0f/(16*1024*120*2));
     if (ret < 0)
         return ret;

     ret = ff_mdct15_init(&ac->mdct480, 1, 5, 1.0f/(16*1024*960));

     if (ret < 0)
         return ret;

     ret = ff_mdct15_init(&ac->mdct960, 1, 6, 1.0f/(16*1024*960*2));
     if (ret < 0)
         return ret;

 #endif

 
     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 AVERROR_INVALIDDATA;
     }
     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 AVERROR_INVALIDDATA;
         }
     }
     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)
 {
     const MPEG4AudioConfig *const m4ac = &ac->oc[1].m4ac;
     const int aot = m4ac->object_type;
     const int sampling_index = m4ac->sampling_index;

     int ret_fail = AVERROR_INVALIDDATA;
 

     if (aot != AOT_ER_AAC_ELD) {
         if (get_bits1(gb)) {
             av_log(ac->avctx, AV_LOG_ERROR, "Reserved bit set.\n");
             if (ac->avctx->err_recognition & AV_EF_BITSTREAM)
                 return AVERROR_INVALIDDATA;
         }
         ics->window_sequence[1] = ics->window_sequence[0];
         ics->window_sequence[0] = get_bits(gb, 2);
         if (aot == AOT_ER_AAC_LD &&
             ics->window_sequence[0] != ONLY_LONG_SEQUENCE) {
             av_log(ac->avctx, AV_LOG_ERROR,
                    "AAC LD is only defined for ONLY_LONG_SEQUENCE but "
                    "window sequence %d found.\n", ics->window_sequence[0]);
             ics->window_sequence[0] = ONLY_LONG_SEQUENCE;
             return AVERROR_INVALIDDATA;
         }
         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;

         if (m4ac->frame_length_short) {
             ics->swb_offset    =  ff_swb_offset_120[sampling_index];
             ics->num_swb       = ff_aac_num_swb_120[sampling_index];
         } else {
             ics->swb_offset    =  ff_swb_offset_128[sampling_index];
             ics->num_swb       = ff_aac_num_swb_128[sampling_index];
         }

         ics->tns_max_bands     = ff_tns_max_bands_128[sampling_index];
         ics->predictor_present = 0;
     } else {
         ics->max_sfb           = get_bits(gb, 6);
         ics->num_windows       = 1;
         if (aot == AOT_ER_AAC_LD || aot == AOT_ER_AAC_ELD) {
             if (m4ac->frame_length_short) {
                 ics->swb_offset    =     ff_swb_offset_480[sampling_index];
                 ics->num_swb       =    ff_aac_num_swb_480[sampling_index];
                 ics->tns_max_bands =  ff_tns_max_bands_480[sampling_index];
             } else {
                 ics->swb_offset    =     ff_swb_offset_512[sampling_index];
                 ics->num_swb       =    ff_aac_num_swb_512[sampling_index];
                 ics->tns_max_bands =  ff_tns_max_bands_512[sampling_index];
             }

             if (!ics->num_swb || !ics->swb_offset) {
                 ret_fail = AVERROR_BUG;
                 goto fail;
             }

         } else {

             if (m4ac->frame_length_short) {
                 ics->num_swb    = ff_aac_num_swb_960[sampling_index];
                 ics->swb_offset = ff_swb_offset_960[sampling_index];
             } else {
                 ics->num_swb    = ff_aac_num_swb_1024[sampling_index];
                 ics->swb_offset = ff_swb_offset_1024[sampling_index];
             }

             ics->tns_max_bands = ff_tns_max_bands_1024[sampling_index];
         }
         if (aot != AOT_ER_AAC_ELD) {
             ics->predictor_present     = get_bits1(gb);
             ics->predictor_reset_group = 0;
         }
         if (ics->predictor_present) {
             if (aot == AOT_AAC_MAIN) {
                 if (decode_prediction(ac, ics, gb)) {
                     goto fail;
                 }
             } else if (aot == AOT_AAC_LC ||
                        aot == AOT_ER_AAC_LC) {
                 av_log(ac->avctx, AV_LOG_ERROR,
                        "Prediction is not allowed in AAC-LC.\n");
                 goto fail;
             } else {
                 if (aot == AOT_ER_AAC_LD) {
                     av_log(ac->avctx, AV_LOG_ERROR,
                            "LTP in ER AAC LD not yet implemented.\n");

                     ret_fail = AVERROR_PATCHWELCOME;
                     goto fail;

                 }
                 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 ret_fail;

 }
 
 /**
  * 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 AVERROR_INVALIDDATA;
             }
             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 AVERROR_INVALIDDATA;
                 }
                 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 AVERROR_INVALIDDATA;
                 }
             } 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, INTFLOAT 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 - NOISE_OFFSET, 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] = FIXR(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) - SCALE_DIFF_ZERO;
                     clipped_offset = av_clip(offset[2], -155, 100);
                     if (offset[2] != clipped_offset) {
                         avpriv_request_sample(ac->avctx,
                                               "If you heard an audible artifact, there may be a bug in the decoder. "
                                               "Clipped intensity stereo position (%d -> %d)",
                                               offset[2], clipped_offset);
                     }

 #if USE_FIXED
                     sf[idx] = 100 - clipped_offset;
 #else

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

 #endif /* USE_FIXED */

                 }
             } else if (band_type[idx] == NOISE_BT) {
                 for (; i < run_end; i++, idx++) {
                     if (noise_flag-- > 0)
                         offset[1] += get_bits(gb, NOISE_PRE_BITS) - NOISE_PRE;
                     else
                         offset[1] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - SCALE_DIFF_ZERO;
                     clipped_offset = av_clip(offset[1], -100, 155);
                     if (offset[1] != clipped_offset) {
                         avpriv_request_sample(ac->avctx,
                                               "If you heard an audible artifact, there may be a bug in the decoder. "
                                               "Clipped noise gain (%d -> %d)",
                                               offset[1], clipped_offset);
                     }

 #if USE_FIXED
                     sf[idx] = -(100 + clipped_offset);
 #else

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

 #endif /* USE_FIXED */

                 }
             } else {
                 for (; i < run_end; i++, idx++) {
                     offset[0] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - SCALE_DIFF_ZERO;
                     if (offset[0] > 255U) {
                         av_log(ac->avctx, AV_LOG_ERROR,
                                "Scalefactor (%d) out of range.\n", offset[0]);
                         return AVERROR_INVALIDDATA;
                     }

 #if USE_FIXED
                     sf[idx] = -offset[0];
 #else

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

 #endif /* USE_FIXED */

                 }
             }
         }
     }
     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] >= swb_offset[num_swb])
         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] >= swb_offset[num_swb])
             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 AVERROR_INVALIDDATA;
                 }
                 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;
     int max_idx = cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb;
     if (ms_present == 1) {
         for (idx = 0; idx < max_idx; idx++)
             cpe->ms_mask[idx] = get_bits1(gb);
     } else if (ms_present == 2) {
         memset(cpe->ms_mask, 1, max_idx * sizeof(cpe->ms_mask[0]));
     }
 }
 
 /**
  * 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, INTFLOAT coef[1024],
                                        GetBitContext *gb, const INTFLOAT 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;

     INTFLOAT *coef_base = coef;

 
     for (g = 0; g < ics->num_windows; g++)
         memset(coef + g * 128 + offsets[ics->max_sfb], 0,

                sizeof(INTFLOAT) * (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;

             INTFLOAT *cfo = coef + offsets[i];

             int off_len = offsets[i + 1] - offsets[i];
             int group;
 
             if (cbt_m1 >= INTENSITY_BT2 - 1) {

                 for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
                     memset(cfo, 0, off_len * sizeof(*cfo));

                 }
             } else if (cbt_m1 == NOISE_BT - 1) {

                 for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
 #if !USE_FIXED

                     float scale;

 #endif /* !USE_FIXED */
                     INTFLOAT band_energy;

 
                     for (k = 0; k < off_len; k++) {
                         ac->random_state  = lcg_random(ac->random_state);

 #if USE_FIXED
                         cfo[k] = ac->random_state >> 3;
 #else

                         cfo[k] = ac->random_state;

 #endif /* USE_FIXED */

                     }
 

 #if USE_FIXED
                     band_energy = ac->fdsp->scalarproduct_fixed(cfo, cfo, off_len);
                     band_energy = fixed_sqrt(band_energy, 31);
                     noise_scale(cfo, sf[idx], band_energy, off_len);
 #else

                     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);

 #endif /* USE_FIXED */

                 }
             } else {

 #if !USE_FIXED

                 const float *vq = ff_aac_codebook_vector_vals[cbt_m1];

 #endif /* !USE_FIXED */

                 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 < (AAC_SIGNE)g_len; group++, cfo+=128) {
                         INTFLOAT *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];

 #if USE_FIXED
                             cf = DEC_SQUAD(cf, cb_idx);
 #else

                             cf = VMUL4(cf, vq, cb_idx, sf + idx);

 #endif /* USE_FIXED */

                         } while (len -= 4);
                     }
                     break;
 
                 case 1:

                     for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
                         INTFLOAT *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);

 #if USE_FIXED
                             cf = DEC_UQUAD(cf, cb_idx, bits);
 #else

                             cf = VMUL4S(cf, vq, cb_idx, bits, sf + idx);

 #endif /* USE_FIXED */

                         } while (len -= 4);
                     }
                     break;
 
                 case 2:

                     for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
                         INTFLOAT *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];

 #if USE_FIXED
                             cf = DEC_SPAIR(cf, cb_idx);
 #else

                             cf = VMUL2(cf, vq, cb_idx, sf + idx);

 #endif /* USE_FIXED */

                         } while (len -= 2);
                     }
                     break;
 
                 case 3:
                 case 4:

                     for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
                         INTFLOAT *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);

 #if USE_FIXED
                             cf = DEC_UPAIR(cf, cb_idx, sign);
 #else

                             cf = VMUL2S(cf, vq, cb_idx, sign, sf + idx);

 #endif /* USE_FIXED */

                         } while (len -= 2);
                     }
                     break;
 
                 default:

                     for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
 #if USE_FIXED
                         int *icf = cfo;
                         int v;
 #else

                         float *cf = cfo;
                         uint32_t *icf = (uint32_t *) cf;

 #endif /* USE_FIXED */

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

 #if USE_FIXED
                                     v = n;
                                     if (bits & 1U<<31)
                                         v = -v;
                                     *icf++ = v;
 #else

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

 #endif /* USE_FIXED */

                                     bits <<= 1;
                                 } else {

 #if USE_FIXED
                                     v = cb_idx & 15;
                                     if (bits & 1U<<31)
                                         v = -v;
                                     *icf++ = v;
 #else

                                     unsigned v = ((const uint32_t*)vq)[cb_idx & 15];
                                     *icf++ = (bits & 1U<<31) | v;

 #endif /* USE_FIXED */

                                     bits <<= !!v;
                                 }
                                 cb_idx >>= 4;
                             }
                         } while (len -= 2);

 #if !USE_FIXED

                         ac->fdsp->vector_fmul_scalar(cfo, cfo, sf[idx], off_len);

 #endif /* !USE_FIXED */

                     }
                 }
 
                 CLOSE_READER(re, gb);
             }
         }
         coef += g_len << 7;
     }
 
     if (pulse_present) {
         idx = 0;
         for (i = 0; i < pulse->num_pulse; i++) {

             INTFLOAT co = coef_base[ pulse->pos[i] ];

             while (offsets[idx + 1] <= pulse->pos[i])
                 idx++;
             if (band_type[idx] != NOISE_BT && sf[idx]) {

                 INTFLOAT ico = -pulse->amp[i];
 #if USE_FIXED
                 if (co) {
                     ico = co + (co > 0 ? -ico : ico);
                 }
                 coef_base[ pulse->pos[i] ] = ico;
 #else

                 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];

 #endif /* USE_FIXED */
             }
         }
     }
 #if USE_FIXED
     coef = coef_base;
     idx = 0;
     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;
             int *cfo = coef + offsets[i];
             int off_len = offsets[i + 1] - offsets[i];
             int group;
 
             if (cbt_m1 < NOISE_BT - 1) {
                 for (group = 0; group < (int)g_len; group++, cfo+=128) {
                     ac->vector_pow43(cfo, off_len);
                     ac->subband_scale(cfo, cfo, sf[idx], 34, off_len);
                 }

             }
         }

         coef += g_len << 7;

     }

 #endif /* USE_FIXED */

     return 0;
 }
 
 /**
  * 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;

     INTFLOAT *out = sce->coeffs;

     int global_gain, eld_syntax, er_syntax, pulse_present = 0;
     int ret;
 
     eld_syntax = ac->oc[1].m4ac.object_type == AOT_ER_AAC_ELD;
     er_syntax  = ac->oc[1].m4ac.object_type == AOT_ER_AAC_LC ||
                  ac->oc[1].m4ac.object_type == AOT_ER_AAC_LTP ||
                  ac->oc[1].m4ac.object_type == AOT_ER_AAC_LD ||
                  ac->oc[1].m4ac.object_type == AOT_ER_AAC_ELD;
 
     /* 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) {

         ret = decode_ics_info(ac, ics, gb);
         if (ret < 0)
             goto fail;

     }
 
     if ((ret = decode_band_types(ac, sce->band_type,
                                  sce->band_type_run_end, gb, ics)) < 0)

         goto fail;

     if ((ret = decode_scalefactors(ac, sce->sf, gb, global_gain, ics,
                                   sce->band_type, sce->band_type_run_end)) < 0)

         goto fail;

 
     pulse_present = 0;
     if (!scale_flag) {
         if (!eld_syntax && (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");

                 ret = AVERROR_INVALIDDATA;
                 goto fail;

             }
             if (decode_pulses(&pulse, gb, ics->swb_offset, ics->num_swb)) {
                 av_log(ac->avctx, AV_LOG_ERROR,
                        "Pulse data corrupt or invalid.\n");

                 ret = AVERROR_INVALIDDATA;
                 goto fail;

             }
         }
         tns->present = get_bits1(gb);

         if (tns->present && !er_syntax) {
             ret = decode_tns(ac, tns, gb, ics);
             if (ret < 0)
                 goto fail;
         }

         if (!eld_syntax && get_bits1(gb)) {
             avpriv_request_sample(ac->avctx, "SSR");

             ret = AVERROR_PATCHWELCOME;
             goto fail;

         }
         // I see no textual basis in the spec for this occurring after SSR gain
         // control, but this is what both reference and real implmentations do

         if (tns->present && er_syntax) {
             ret = decode_tns(ac, tns, gb, ics);
             if (ret < 0)
                 goto fail;
         }

     }
 

     ret = decode_spectrum_and_dequant(ac, out, gb, sce->sf, pulse_present,
                                     &pulse, ics, sce->band_type);
     if (ret < 0)
         goto fail;

 
     if (ac->oc[1].m4ac.object_type == AOT_AAC_MAIN && !common_window)
         apply_prediction(ac, sce);
 
     return 0;

 fail:
     tns->present = 0;
     return ret;

 }
 
 /**
  * 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;

     INTFLOAT *ch0 = cpe->ch[0].coeffs;
     INTFLOAT *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) {

 #if USE_FIXED
                 for (group = 0; group < ics->group_len[g]; group++) {
                     ac->fdsp->butterflies_fixed(ch0 + group * 128 + offsets[i],
                                                 ch1 + group * 128 + offsets[i],
                                                 offsets[i+1] - offsets[i]);
 #else

                 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]);

 #endif /* USE_FIXED */

                 }
             }
         }
         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];

     INTFLOAT *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;

     INTFLOAT 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++)

 #if USE_FIXED
                         ac->subband_scale(coef1 + group * 128 + offsets[i],
                                       coef0 + group * 128 + offsets[i],
                                       scale,
                                       23,
                                       offsets[i + 1] - offsets[i]);
 #else

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

 #endif /* USE_FIXED */

                 }
             } 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;
     int eld_syntax = ac->oc[1].m4ac.object_type == AOT_ER_AAC_ELD;
 
     common_window = eld_syntax || 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 AVERROR_INVALIDDATA;
         } 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;

     INTFLOAT 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);

 #if USE_FIXED
     scale = get_bits(gb, 2);
 #else
     scale = cce_scale[get_bits(gb, 2)];
 #endif

 
     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;

         INTFLOAT gain_cache = FIXR10(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 = GET_GAIN(scale, gain);

 #if USE_FIXED
             if ((abs(gain_cache)-1024) >> 3 > 30)
                 return AVERROR(ERANGE);
 #endif

         }
         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 = GET_GAIN(scale, t) * s;

 #if USE_FIXED
                                 if ((abs(gain_cache)-1024) >> 3 > 30)
                                     return AVERROR(ERANGE);
 #endif

                             }
                         }
                         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;
     int type = get_bits(gb, 4);
 
     if (ac->avctx->debug & FF_DEBUG_STARTCODE)
         av_log(ac->avctx, AV_LOG_DEBUG, "extension type: %d len:%d\n", type, cnt);
 
     switch (type) { // 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.frame_length_short) {
             if (!ac->warned_960_sbr)
               avpriv_report_missing_feature(ac->avctx,
                                             "SBR with 960 frame length");
             ac->warned_960_sbr = 1;
             skip_bits_long(gb, 8 * cnt - 4);
             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;
             ac->avctx->profile = FF_PROFILE_AAC_HE_V2;
             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;
             ac->avctx->profile = FF_PROFILE_AAC_HE;
         }

         res = AAC_RENAME(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(INTFLOAT coef_param[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;

     INTFLOAT lpc[TNS_MAX_ORDER];
     INTFLOAT tmp[TNS_MAX_ORDER+1];

     UINTFLOAT *coef = coef_param;

 
     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

             AAC_RENAME(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] -= AAC_MUL26((INTFLOAT)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] += AAC_MUL26(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, INTFLOAT *out,
                                    INTFLOAT *in, IndividualChannelStream *ics)

 {

     const INTFLOAT *lwindow      = ics->use_kb_window[0] ? AAC_RENAME(ff_aac_kbd_long_1024) : AAC_RENAME(ff_sine_1024);
     const INTFLOAT *swindow      = ics->use_kb_window[0] ? AAC_RENAME(ff_aac_kbd_short_128) : AAC_RENAME(ff_sine_128);
     const INTFLOAT *lwindow_prev = ics->use_kb_window[1] ? AAC_RENAME(ff_aac_kbd_long_1024) : AAC_RENAME(ff_sine_1024);
     const INTFLOAT *swindow_prev = ics->use_kb_window[1] ? AAC_RENAME(ff_aac_kbd_short_128) : AAC_RENAME(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(*in));

         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(*in));

     }
     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) {

         INTFLOAT *predTime = sce->ret;
         INTFLOAT *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] = AAC_MUL30(sce->ltp_state[i + 2048 - ltp->lag], ltp->coef);
         memset(&predTime[i], 0, (2048 - i) * sizeof(*predTime));

 
         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;

     INTFLOAT *saved     = sce->saved;
     INTFLOAT *saved_ltp = sce->coeffs;
     const INTFLOAT *lwindow = ics->use_kb_window[0] ? AAC_RENAME(ff_aac_kbd_long_1024) : AAC_RENAME(ff_sine_1024);
     const INTFLOAT *swindow = ics->use_kb_window[0] ? AAC_RENAME(ff_aac_kbd_short_128) : AAC_RENAME(ff_sine_128);

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

         memcpy(saved_ltp,       saved, 512 * sizeof(*saved_ltp));
         memset(saved_ltp + 576, 0,     448 * sizeof(*saved_ltp));

         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] = AAC_MUL31(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(*saved_ltp));
         memset(saved_ltp + 576, 0,                  448 * sizeof(*saved_ltp));

         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] = AAC_MUL31(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] = AAC_MUL31(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;

     INTFLOAT *in    = sce->coeffs;
     INTFLOAT *out   = sce->ret;
     INTFLOAT *saved = sce->saved;
     const INTFLOAT *swindow      = ics->use_kb_window[0] ? AAC_RENAME(ff_aac_kbd_short_128) : AAC_RENAME(ff_sine_128);
     const INTFLOAT *lwindow_prev = ics->use_kb_window[1] ? AAC_RENAME(ff_aac_kbd_long_1024) : AAC_RENAME(ff_sine_1024);
     const INTFLOAT *swindow_prev = ics->use_kb_window[1] ? AAC_RENAME(ff_aac_kbd_short_128) : AAC_RENAME(ff_sine_128);
     INTFLOAT *buf  = ac->buf_mdct;
     INTFLOAT *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);

 #if USE_FIXED
         for (i=0; i<1024; i++)
           buf[i] = (buf[i] + 4) >> 3;
 #endif /* USE_FIXED */
     }

 
     /* 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(*out));

 
         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(*out));

         } else {
             ac->fdsp->vector_fmul_window(out + 448,         saved + 448,      buf,         swindow_prev, 64);

             memcpy(                     out + 576,         buf + 64,         448 * sizeof(*out));

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

         memcpy(                     saved,       temp + 64,         64 * sizeof(*saved));

         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(*saved));

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

         memcpy(                     saved,       buf + 512,        448 * sizeof(*saved));
         memcpy(                     saved + 448, buf + 7*128 + 64,  64 * sizeof(*saved));

     } else { // LONG_STOP or ONLY_LONG

         memcpy(                     saved,       buf + 512,        512 * sizeof(*saved));

     }
 }
 

 /**
  * Conduct IMDCT and windowing.
  */
 static void imdct_and_windowing_960(AACContext *ac, SingleChannelElement *sce)
 {
 #if !USE_FIXED
     IndividualChannelStream *ics = &sce->ics;
     INTFLOAT *in    = sce->coeffs;
     INTFLOAT *out   = sce->ret;
     INTFLOAT *saved = sce->saved;
     const INTFLOAT *swindow      = ics->use_kb_window[0] ? AAC_RENAME(ff_aac_kbd_short_120) : AAC_RENAME(ff_sine_120);
     const INTFLOAT *lwindow_prev = ics->use_kb_window[1] ? AAC_RENAME(ff_aac_kbd_long_960) : AAC_RENAME(ff_sine_960);
     const INTFLOAT *swindow_prev = ics->use_kb_window[1] ? AAC_RENAME(ff_aac_kbd_short_120) : AAC_RENAME(ff_sine_120);
     INTFLOAT *buf  = ac->buf_mdct;
     INTFLOAT *temp = ac->temp;
     int i;
 
     // imdct
     if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
         for (i = 0; i < 8; i++)
             ac->mdct120->imdct_half(ac->mdct120, buf + i * 120, in + i * 128, 1);
     } else {
         ac->mdct960->imdct_half(ac->mdct960, buf, in, 1);
     }
 
     /* 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, 480);
     } else {
         memcpy(                          out,               saved,            420 * sizeof(*out));
 
         if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
             ac->fdsp->vector_fmul_window(out + 420 + 0*120, saved + 420,      buf + 0*120, swindow_prev, 60);
             ac->fdsp->vector_fmul_window(out + 420 + 1*120, buf + 0*120 + 60, buf + 1*120, swindow,      60);
             ac->fdsp->vector_fmul_window(out + 420 + 2*120, buf + 1*120 + 60, buf + 2*120, swindow,      60);
             ac->fdsp->vector_fmul_window(out + 420 + 3*120, buf + 2*120 + 60, buf + 3*120, swindow,      60);
             ac->fdsp->vector_fmul_window(temp,              buf + 3*120 + 60, buf + 4*120, swindow,      60);
             memcpy(                      out + 420 + 4*120, temp, 60 * sizeof(*out));
         } else {
             ac->fdsp->vector_fmul_window(out + 420,         saved + 420,      buf,         swindow_prev, 60);
             memcpy(                      out + 540,         buf + 60,         420 * sizeof(*out));
         }
     }
 
     // buffer update
     if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
         memcpy(                      saved,       temp + 60,         60 * sizeof(*saved));
         ac->fdsp->vector_fmul_window(saved + 60,  buf + 4*120 + 60, buf + 5*120, swindow, 60);
         ac->fdsp->vector_fmul_window(saved + 180, buf + 5*120 + 60, buf + 6*120, swindow, 60);
         ac->fdsp->vector_fmul_window(saved + 300, buf + 6*120 + 60, buf + 7*120, swindow, 60);
         memcpy(                      saved + 420, buf + 7*120 + 60,  60 * sizeof(*saved));
     } else if (ics->window_sequence[0] == LONG_START_SEQUENCE) {
         memcpy(                      saved,       buf + 480,        420 * sizeof(*saved));
         memcpy(                      saved + 420, buf + 7*120 + 60,  60 * sizeof(*saved));
     } else { // LONG_STOP or ONLY_LONG
         memcpy(                      saved,       buf + 480,        480 * sizeof(*saved));
     }
 #endif
 }

 static void imdct_and_windowing_ld(AACContext *ac, SingleChannelElement *sce)
 {
     IndividualChannelStream *ics = &sce->ics;

     INTFLOAT *in    = sce->coeffs;
     INTFLOAT *out   = sce->ret;
     INTFLOAT *saved = sce->saved;
     INTFLOAT *buf  = ac->buf_mdct;
 #if USE_FIXED
     int i;
 #endif /* USE_FIXED */

 
     // imdct
     ac->mdct.imdct_half(&ac->mdct_ld, buf, in);
 

 #if USE_FIXED
     for (i = 0; i < 1024; i++)
         buf[i] = (buf[i] + 2) >> 2;
 #endif /* USE_FIXED */
 

     // window overlapping
     if (ics->use_kb_window[1]) {
         // AAC LD uses a low overlap sine window instead of a KBD window

         memcpy(out, saved, 192 * sizeof(*out));
         ac->fdsp->vector_fmul_window(out + 192, saved + 192, buf, AAC_RENAME(ff_sine_128), 64);
         memcpy(                     out + 320, buf + 64, 192 * sizeof(*out));

     } else {

         ac->fdsp->vector_fmul_window(out, saved, buf, AAC_RENAME(ff_sine_512), 256);

     }
 
     // buffer update

     memcpy(saved, buf + 256, 256 * sizeof(*saved));

 }
 
 static void imdct_and_windowing_eld(AACContext *ac, SingleChannelElement *sce)
 {

     INTFLOAT *in    = sce->coeffs;
     INTFLOAT *out   = sce->ret;
     INTFLOAT *saved = sce->saved;
     INTFLOAT *buf  = ac->buf_mdct;

     int i;
     const int n  = ac->oc[1].m4ac.frame_length_short ? 480 : 512;
     const int n2 = n >> 1;
     const int n4 = n >> 2;

     const INTFLOAT *const window = n == 480 ? AAC_RENAME(ff_aac_eld_window_480) :

                                            AAC_RENAME(ff_aac_eld_window_512);

 
     // Inverse transform, mapped to the conventional IMDCT by
     // Chivukula, R.K.; Reznik, Y.A.; Devarajan, V.,
     // "Efficient algorithms for MPEG-4 AAC-ELD, AAC-LD and AAC-LC filterbanks,"
     // International Conference on Audio, Language and Image Processing, ICALIP 2008.
     // URL: http://ieeexplore.ieee.org/stamp/stamp.jsp?tp=&arnumber=4590245&isnumber=4589950
     for (i = 0; i < n2; i+=2) {

         INTFLOAT temp;

         temp =  in[i    ]; in[i    ] = -in[n - 1 - i]; in[n - 1 - i] = temp;
         temp = -in[i + 1]; in[i + 1] =  in[n - 2 - i]; in[n - 2 - i] = temp;
     }

 #if !USE_FIXED

     if (n == 480)

         ac->mdct480->imdct_half(ac->mdct480, buf, in, 1);

     else

 #endif

         ac->mdct.imdct_half(&ac->mdct_ld, buf, in);

 
 #if USE_FIXED
     for (i = 0; i < 1024; i++)
       buf[i] = (buf[i] + 1) >> 1;
 #endif /* USE_FIXED */
 

     for (i = 0; i < n; i+=2) {
         buf[i] = -buf[i];
     }
     // Like with the regular IMDCT at this point we still have the middle half
     // of a transform but with even symmetry on the left and odd symmetry on
     // the right
 
     // window overlapping
     // The spec says to use samples [0..511] but the reference decoder uses
     // samples [128..639].
     for (i = n4; i < n2; i ++) {

         out[i - n4] = AAC_MUL31(   buf[    n2 - 1 - i] , window[i       - n4]) +
                       AAC_MUL31( saved[        i + n2] , window[i +   n - n4]) +
                       AAC_MUL31(-saved[n + n2 - 1 - i] , window[i + 2*n - n4]) +
                       AAC_MUL31(-saved[  2*n + n2 + i] , window[i + 3*n - n4]);

     }
     for (i = 0; i < n2; i ++) {

         out[n4 + i] = AAC_MUL31(   buf[              i] , window[i + n2       - n4]) +
                       AAC_MUL31(-saved[      n - 1 - i] , window[i + n2 +   n - n4]) +
                       AAC_MUL31(-saved[          n + i] , window[i + n2 + 2*n - n4]) +
                       AAC_MUL31( saved[2*n + n - 1 - i] , window[i + n2 + 3*n - n4]);

     }
     for (i = 0; i < n4; i ++) {

         out[n2 + n4 + i] = AAC_MUL31(   buf[    i + n2] , window[i +   n - n4]) +
                            AAC_MUL31(-saved[n2 - 1 - i] , window[i + 2*n - n4]) +
                            AAC_MUL31(-saved[n + n2 + i] , window[i + 3*n - n4]);

     }
 
     // buffer update

     memmove(saved + n, saved, 2 * n * sizeof(*saved));
     memcpy( saved,       buf,     n * sizeof(*saved));

 }
 
 /**
  * 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 samples, applying all supported tools as appropriate.

  */

 static void spectral_to_sample(AACContext *ac, int samples)

 {
     int i, type;
     void (*imdct_and_window)(AACContext *ac, SingleChannelElement *sce);
     switch (ac->oc[1].m4ac.object_type) {
     case AOT_ER_AAC_LD:
         imdct_and_window = imdct_and_windowing_ld;
         break;
     case AOT_ER_AAC_ELD:
         imdct_and_window = imdct_and_windowing_eld;
         break;
     default:

         if (ac->oc[1].m4ac.frame_length_short)
             imdct_and_window = imdct_and_windowing_960;
         else
             imdct_and_window = ac->imdct_and_windowing;

     }
     for (type = 3; type >= 0; type--) {
         for (i = 0; i < MAX_ELEM_ID; i++) {
             ChannelElement *che = ac->che[type][i];
             if (che && che->present) {
                 if (type <= TYPE_CPE)

                     apply_channel_coupling(ac, che, type, i, BEFORE_TNS, AAC_RENAME(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, AAC_RENAME(apply_dependent_coupling));

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

                         AAC_RENAME(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, AAC_RENAME(apply_independent_coupling));
 
 #if USE_FIXED
                 {
                     int j;
                     /* preparation for resampler */

                     for(j = 0; j<samples; j++){

                         che->ch[0].ret[j] = (int32_t)av_clip64((int64_t)che->ch[0].ret[j]*128, INT32_MIN, INT32_MAX-0x8000)+0x8000;

                         if(type == TYPE_CPE)

                             che->ch[1].ret[j] = (int32_t)av_clip64((int64_t)che->ch[1].ret[j]*128, INT32_MIN, INT32_MAX-0x8000)+0x8000;

                     }
                 }
 #endif /* USE_FIXED */

                 che->present = 0;
             } else if (che) {
                 av_log(ac->avctx, AV_LOG_VERBOSE, "ChannelElement %d.%d missing \n", type, i);
             }
         }
     }
 }
 
 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, ret;
 

     size = ff_adts_header_parse(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.
             avpriv_report_missing_feature(ac->avctx,
                                           "More than one AAC RDB per ADTS frame");
             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 ((ret = set_default_channel_config(ac->avctx,
                                                   layout_map,
                                                   &layout_map_tags,
                                                   hdr_info.chan_config)) < 0)
                 return ret;
             if ((ret = output_configure(ac, layout_map, layout_map_tags,
                                         FFMAX(ac->oc[1].status,
                                               OC_TRIAL_FRAME), 0)) < 0)
                 return ret;
         } 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;
         ac->oc[1].m4ac.frame_length_short = 0;
         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_er_frame(AVCodecContext *avctx, void *data,
                                int *got_frame_ptr, GetBitContext *gb)
 {
     AACContext *ac = avctx->priv_data;
     const MPEG4AudioConfig *const m4ac = &ac->oc[1].m4ac;
     ChannelElement *che;
     int err, i;
     int samples = m4ac->frame_length_short ? 960 : 1024;
     int chan_config = m4ac->chan_config;
     int aot = m4ac->object_type;
 
     if (aot == AOT_ER_AAC_LD || aot == AOT_ER_AAC_ELD)
         samples >>= 1;
 
     ac->frame = data;
 
     if ((err = frame_configure_elements(avctx)) < 0)
         return err;
 
     // The FF_PROFILE_AAC_* defines are all object_type - 1
     // This may lead to an undefined profile being signaled
     ac->avctx->profile = aot - 1;
 
     ac->tags_mapped = 0;
 
     if (chan_config < 0 || (chan_config >= 8 && chan_config < 11) || chan_config >= 13) {
         avpriv_request_sample(avctx, "Unknown ER channel configuration %d",
                               chan_config);
         return AVERROR_INVALIDDATA;
     }
     for (i = 0; i < tags_per_config[chan_config]; i++) {
         const int elem_type = aac_channel_layout_map[chan_config-1][i][0];
         const int elem_id   = aac_channel_layout_map[chan_config-1][i][1];
         if (!(che=get_che(ac, elem_type, elem_id))) {
             av_log(ac->avctx, AV_LOG_ERROR,
                    "channel element %d.%d is not allocated\n",
                    elem_type, elem_id);
             return AVERROR_INVALIDDATA;
         }
         che->present = 1;
         if (aot != AOT_ER_AAC_ELD)
             skip_bits(gb, 4);
         switch (elem_type) {
         case TYPE_SCE:
             err = decode_ics(ac, &che->ch[0], gb, 0, 0);
             break;
         case TYPE_CPE:
             err = decode_cpe(ac, gb, che);
             break;
         case TYPE_LFE:
             err = decode_ics(ac, &che->ch[0], gb, 0, 0);
             break;
         }
         if (err < 0)
             return err;
     }
 

     spectral_to_sample(ac, samples);

     if (!ac->frame->data[0] && samples) {
         av_log(avctx, AV_LOG_ERROR, "no frame data found\n");
         return AVERROR_INVALIDDATA;
     }
 

     ac->frame->nb_samples = samples;
     ac->frame->sample_rate = avctx->sample_rate;
     *got_frame_ptr = 1;
 
     skip_bits_long(gb, get_bits_left(gb));
     return 0;
 }
 
 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, che_prev_type = TYPE_END;

     int err, elem_id;
     int samples = 0, multiplier, audio_found = 0, pce_found = 0;
     int is_dmono, sce_count = 0;

     int payload_alignment;

 
     ac->frame = data;
 
     if (show_bits(gb, 12) == 0xfff) {
         if ((err = parse_adts_frame_header(ac, gb)) < 0) {
             av_log(avctx, AV_LOG_ERROR, "Error decoding AAC frame header.\n");
             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 = AVERROR_INVALIDDATA;
             goto fail;
         }
     }
 
     if ((err = frame_configure_elements(avctx)) < 0)
         goto fail;
 
     // The FF_PROFILE_AAC_* defines are all object_type - 1
     // This may lead to an undefined profile being signaled
     ac->avctx->profile = ac->oc[1].m4ac.object_type - 1;
 

     payload_alignment = get_bits_count(gb);

     ac->tags_mapped = 0;
     // parse
     while ((elem_type = get_bits(gb, 3)) != TYPE_END) {
         elem_id = get_bits(gb, 4);
 
         if (avctx->debug & FF_DEBUG_STARTCODE)
             av_log(avctx, AV_LOG_DEBUG, "Elem type:%x id:%x\n", elem_type, elem_id);
 
         if (!avctx->channels && elem_type != TYPE_PCE) {
             err = AVERROR_INVALIDDATA;
             goto fail;
         }
 
         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 = AVERROR_INVALIDDATA;
                 goto fail;
             }

             samples = ac->oc[1].m4ac.frame_length_short ? 960 : 1024;

             che->present = 1;
         }
 
         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;

 
             int pushed = push_output_configuration(ac);
             if (pce_found && !pushed) {
                 err = AVERROR_INVALIDDATA;
                 goto fail;
             }
 

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

             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");

                 pop_output_configuration(ac);

             } 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 = AVERROR_INVALIDDATA;
                     goto fail;
             }
             while (elem_id > 0)

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

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

         if (elem_type < TYPE_DSE) {

             che_prev      = che;
             che_prev_type = elem_type;

         }

 
         if (err)
             goto fail;
 
         if (get_bits_left(gb) < 3) {
             av_log(avctx, AV_LOG_ERROR, overread_err);
             err = AVERROR_INVALIDDATA;
             goto fail;
         }
     }
 
     if (!avctx->channels) {
         *got_frame_ptr = 0;
         return 0;
     }
 
     multiplier = (ac->oc[1].m4ac.sbr == 1) ? ac->oc[1].m4ac.ext_sample_rate > ac->oc[1].m4ac.sample_rate : 0;
     samples <<= multiplier;
 

     spectral_to_sample(ac, samples);
 

     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)
         avctx->internal->skip_samples_multiplier = 2;

 
     if (!ac->frame->data[0] && samples) {
         av_log(avctx, AV_LOG_ERROR, "no frame data found\n");
         err = AVERROR_INVALIDDATA;
         goto fail;
     }
 
     if (samples) {
         ac->frame->nb_samples = samples;
         ac->frame->sample_rate = avctx->sample_rate;
     } else
         av_frame_unref(ac->frame);
     *got_frame_ptr = !!samples;
 
     /* 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 (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];
     }
 
     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 +

                                       AV_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*8LL, 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;
 

     if ((err = init_get_bits8(&gb, buf, buf_size)) < 0)

         return err;
 
     switch (ac->oc[1].m4ac.object_type) {
     case AOT_ER_AAC_LC:
     case AOT_ER_AAC_LTP:
     case AOT_ER_AAC_LD:
     case AOT_ER_AAC_ELD:
         err = aac_decode_er_frame(avctx, data, got_frame_ptr, &gb);
         break;
     default:
         err = aac_decode_frame_int(avctx, data, got_frame_ptr, &gb, avpkt);
     }
     if (err < 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])

                 AAC_RENAME(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_ld);
     ff_mdct_end(&ac->mdct_ltp);

 #if !USE_FIXED

     ff_mdct15_uninit(&ac->mdct120);

     ff_mdct15_uninit(&ac->mdct480);

     ff_mdct15_uninit(&ac->mdct960);

 #endif

     av_freep(&ac->fdsp);
     return 0;
 }
 
 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 USE_FIXED
     c->vector_pow43                             = vector_pow43;
     c->subband_scale                            = subband_scale;
 #endif

 #if !USE_FIXED

     if(ARCH_MIPS)
         ff_aacdec_init_mips(c);

 #endif /* !USE_FIXED */

 }
 /**
  * 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,
 };