text: /*
   * 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>
   *
   * This file is part of FFmpeg.
   *
   * FFmpeg is free software; you can redistribute it and/or
   * modify it under the terms of the GNU Lesser General Public
   * License as published by the Free Software Foundation; either
   * version 2.1 of the License, or (at your option) any later version.
   *
   * FFmpeg is distributed in the hope that it will be useful,
   * but WITHOUT ANY WARRANTY; without even the implied warranty of
   * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
   * Lesser General Public License for more details.
   *
   * You should have received a copy of the GNU Lesser General Public
   * License along with FFmpeg; if not, write to the Free Software
   * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
   */
  
  /**
   * @file
   * AAC decoder
   * @author Oded Shimon  ( ods15 ods15 dyndns org )
   * @author Maxim Gavrilov ( maxim.gavrilov gmail com )
   */
  
  /*
   * supported tools
   *
   * Support?             Name
   * N (code in SoC repo) gain control
   * Y                    block switching
   * Y                    window shapes - standard
   * N                    window shapes - Low Delay
   * Y                    filterbank - standard
   * N (code in SoC repo) filterbank - Scalable Sample Rate
   * Y                    Temporal Noise Shaping
   * Y                    Long Term Prediction
   * Y                    intensity stereo
   * Y                    channel coupling
   * Y                    frequency domain prediction
   * Y                    Perceptual Noise Substitution
   * Y                    Mid/Side stereo
   * N                    Scalable Inverse AAC Quantization
   * N                    Frequency Selective Switch
   * N                    upsampling filter
   * Y                    quantization & coding - AAC
   * N                    quantization & coding - TwinVQ
   * N                    quantization & coding - BSAC
   * N                    AAC Error Resilience tools
   * N                    Error Resilience payload syntax
   * N                    Error Protection tool
   * N                    CELP
   * N                    Silence Compression
   * N                    HVXC
   * N                    HVXC 4kbits/s VR
   * N                    Structured Audio tools
   * N                    Structured Audio Sample Bank Format
   * N                    MIDI
   * N                    Harmonic and Individual Lines plus Noise
   * N                    Text-To-Speech Interface
   * Y                    Spectral Band Replication
   * Y (not in this code) Layer-1
   * Y (not in this code) Layer-2
   * Y (not in this code) Layer-3
   * N                    SinuSoidal Coding (Transient, Sinusoid, Noise)
   * Y                    Parametric Stereo
   * N                    Direct Stream Transfer
   * Y                    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/float_dsp.h"
  #include "libavutil/opt.h"
  #include "avcodec.h"
  #include "internal.h"
  #include "get_bits.h"
  #include "fft.h"
  #include "fmtconvert.h"
  #include "lpc.h"
  #include "kbdwin.h"
  #include "sinewin.h"
  
  #include "aac.h"
  #include "aactab.h"
  #include "aacdectab.h"
  #include "cbrt_tablegen.h"
  #include "sbr.h"
  #include "aacsbr.h"
  #include "mpeg4audio.h"
  #include "aacadtsdec.h"
  #include "libavutil/intfloat.h"
  
  #include <assert.h>
  #include <errno.h>
  #include <math.h>
  #include <stdint.h>
  #include <string.h>
  
  #if ARCH_ARM
  #   include "arm/aac.h"
  #elif ARCH_MIPS
  #   include "mips/aacdec_mips.h"
  #endif
  
  static VLC vlc_scalefactors;
  static VLC vlc_spectral[11];
  
  static int output_configure(AACContext *ac,
                              uint8_t layout_map[MAX_ELEM_ID*4][3], int tags,
                              enum OCStatus oc_type, int get_new_frame);
  
  #define overread_err "Input buffer exhausted before END element found\n"
  
  static int count_channels(uint8_t (*layout)[3], int tags)
  {
      int i, sum = 0;
      for (i = 0; i < tags; i++) {
          int syn_ele = layout[i][0];
          int pos     = layout[i][2];
          sum += (1 + (syn_ele == TYPE_CPE)) *
                 (pos != AAC_CHANNEL_OFF && pos != AAC_CHANNEL_CC);
      }
      return sum;
  }
  
  /**
   * Check for the channel element in the current channel position configuration.
   * If it exists, make sure the appropriate element is allocated and map the
   * channel order to match the internal FFmpeg channel layout.
   *
   * @param   che_pos current channel position configuration
   * @param   type channel element type
   * @param   id channel element id
   * @param   channels count of the number of channels in the configuration
   *
   * @return  Returns error status. 0 - OK, !0 - error
   */
  static av_cold int che_configure(AACContext *ac,
                                   enum ChannelPosition che_pos,
                                   int type, int id, int *channels)
  {
      if (*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);
              ff_aac_sbr_ctx_init(ac, &ac->che[type][id]->sbr);
          }
          if (type != TYPE_CCE) {
              if (*channels >= MAX_CHANNELS - (type == TYPE_CPE || (type == TYPE_SCE && ac->oc[1].m4ac.ps == 1))) {
                  av_log(ac->avctx, AV_LOG_ERROR, "Too many channels\n");
                  return AVERROR_INVALIDDATA;
              }
              ac->output_element[(*channels)++] = &ac->che[type][id]->ch[0];
              if (type == TYPE_CPE ||
                  (type == TYPE_SCE && ac->oc[1].m4ac.ps == 1)) {
                  ac->output_element[(*channels)++] = &ac->che[type][id]->ch[1];
              }
          }
      } else {
          if (ac->che[type][id])
              ff_aac_sbr_ctx_close(&ac->che[type][id]->sbr);
          av_freep(&ac->che[type][id]);
      }
      return 0;
  }
  
  static int frame_configure_elements(AVCodecContext *avctx)
  {
      AACContext *ac = avctx->priv_data;
      int type, id, ch, ret;
  
      /* set channel pointers to internal buffers by default */
      for (type = 0; type < 4; type++) {
          for (id = 0; id < MAX_ELEM_ID; id++) {
              ChannelElement *che = ac->che[type][id];
              if (che) {
                  che->ch[0].ret = che->ch[0].ret_buf;
                  che->ch[1].ret = che->ch[1].ret_buf;
              }
          }
      }
  
      /* get output buffer */
      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 = (float *)ac->frame->extended_data[ch];
      }
  
      return 0;
  }
  
  struct elem_to_channel {
      uint64_t av_position;
      uint8_t syn_ele;
      uint8_t elem_id;
      uint8_t aac_position;
  };
  
  static int assign_pair(struct elem_to_channel e2c_vec[MAX_ELEM_ID],
                         uint8_t (*layout_map)[3], int offset, uint64_t left,
                         uint64_t right, int pos)
  {
      if (layout_map[offset][0] == TYPE_CPE) {
          e2c_vec[offset] = (struct elem_to_channel) {
              .av_position  = left | right,
              .syn_ele      = TYPE_CPE,
              .elem_id      = layout_map[offset][1],
              .aac_position = pos
          };
          return 1;
      } else {
          e2c_vec[offset] = (struct elem_to_channel) {
              .av_position  = left,
              .syn_ele      = TYPE_SCE,
              .elem_id      = layout_map[offset][1],
              .aac_position = pos
          };
          e2c_vec[offset + 1] = (struct elem_to_channel) {
              .av_position  = right,
              .syn_ele      = TYPE_SCE,
              .elem_id      = layout_map[offset + 1][1],
              .aac_position = pos
          };
          return 2;
      }
  }
  
  static int count_paired_channels(uint8_t (*layout_map)[3], int tags, int pos,
                                   int *current)
  {
      int num_pos_channels = 0;
      int first_cpe        = 0;
      int sce_parity       = 0;
      int i;
      for (i = *current; i < tags; i++) {
          if (layout_map[i][2] != pos)
              break;
          if (layout_map[i][0] == TYPE_CPE) {
              if (sce_parity) {
                  if (pos == AAC_CHANNEL_FRONT && !first_cpe) {
                      sce_parity = 0;
                  } else {
                      return -1;
                  }
              }
              num_pos_channels += 2;
              first_cpe         = 1;
          } else {
              num_pos_channels++;
              sce_parity ^= 1;
          }
      }
      if (sce_parity &&
          ((pos == AAC_CHANNEL_FRONT && first_cpe) || pos == AAC_CHANNEL_SIDE))
          return -1;
      *current = i;
      return num_pos_channels;
  }
  
  static uint64_t sniff_channel_order(uint8_t (*layout_map)[3], int tags)
  {
      int i, n, total_non_cc_elements;
      struct elem_to_channel e2c_vec[4 * MAX_ELEM_ID] = { { 0 } };
      int num_front_channels, num_side_channels, num_back_channels;
      uint64_t layout;
  
      if (FF_ARRAY_ELEMS(e2c_vec) < tags)
          return 0;
  
      i = 0;
      num_front_channels =
          count_paired_channels(layout_map, tags, AAC_CHANNEL_FRONT, &i);
      if (num_front_channels < 0)
          return 0;
      num_side_channels =
          count_paired_channels(layout_map, tags, AAC_CHANNEL_SIDE, &i);
      if (num_side_channels < 0)
          return 0;
      num_back_channels =
          count_paired_channels(layout_map, tags, AAC_CHANNEL_BACK, &i);
      if (num_back_channels < 0)
          return 0;
  
      i = 0;
      if (num_front_channels & 1) {
          e2c_vec[i] = (struct elem_to_channel) {
              .av_position  = AV_CH_FRONT_CENTER,
              .syn_ele      = TYPE_SCE,
              .elem_id      = layout_map[i][1],
              .aac_position = AAC_CHANNEL_FRONT
          };
          i++;
          num_front_channels--;
      }
      if (num_front_channels >= 4) {
          i += assign_pair(e2c_vec, layout_map, i,
                           AV_CH_FRONT_LEFT_OF_CENTER,
                           AV_CH_FRONT_RIGHT_OF_CENTER,
                           AAC_CHANNEL_FRONT);
          num_front_channels -= 2;
      }
      if (num_front_channels >= 2) {
          i += assign_pair(e2c_vec, layout_map, i,
                           AV_CH_FRONT_LEFT,
                           AV_CH_FRONT_RIGHT,
                           AAC_CHANNEL_FRONT);
          num_front_channels -= 2;
      }
      while (num_front_channels >= 2) {
          i += assign_pair(e2c_vec, layout_map, i,
                           UINT64_MAX,
                           UINT64_MAX,
                           AAC_CHANNEL_FRONT);
          num_front_channels -= 2;
      }
  
      if (num_side_channels >= 2) {
          i += assign_pair(e2c_vec, layout_map, i,
                           AV_CH_SIDE_LEFT,
                           AV_CH_SIDE_RIGHT,
                           AAC_CHANNEL_FRONT);
          num_side_channels -= 2;
      }
      while (num_side_channels >= 2) {
          i += assign_pair(e2c_vec, layout_map, i,
                           UINT64_MAX,
                           UINT64_MAX,
                           AAC_CHANNEL_SIDE);
          num_side_channels -= 2;
      }
  
      while (num_back_channels >= 4) {
          i += assign_pair(e2c_vec, layout_map, i,
                           UINT64_MAX,
                           UINT64_MAX,
                           AAC_CHANNEL_BACK);
          num_back_channels -= 2;
      }
      if (num_back_channels >= 2) {
          i += assign_pair(e2c_vec, layout_map, i,
                           AV_CH_BACK_LEFT,
                           AV_CH_BACK_RIGHT,
                           AAC_CHANNEL_BACK);
          num_back_channels -= 2;
      }
      if (num_back_channels) {
          e2c_vec[i] = (struct elem_to_channel) {
              .av_position  = AV_CH_BACK_CENTER,
              .syn_ele      = TYPE_SCE,
              .elem_id      = layout_map[i][1],
              .aac_position = AAC_CHANNEL_BACK
          };
          i++;
          num_back_channels--;
      }
  
      if (i < tags && layout_map[i][2] == AAC_CHANNEL_LFE) {
          e2c_vec[i] = (struct elem_to_channel) {
              .av_position  = AV_CH_LOW_FREQUENCY,
              .syn_ele      = TYPE_LFE,
              .elem_id      = layout_map[i][1],
              .aac_position = AAC_CHANNEL_LFE
          };
          i++;
      }
      while (i < tags && layout_map[i][2] == AAC_CHANNEL_LFE) {
          e2c_vec[i] = (struct elem_to_channel) {
              .av_position  = UINT64_MAX,
              .syn_ele      = TYPE_LFE,
              .elem_id      = layout_map[i][1],
              .aac_position = AAC_CHANNEL_LFE
          };
          i++;
      }
  
      // Must choose a stable sort
      total_non_cc_elements = n = i;
      do {
          int next_n = 0;
          for (i = 1; i < n; i++)
              if (e2c_vec[i - 1].av_position > e2c_vec[i].av_position) {
                  FFSWAP(struct elem_to_channel, e2c_vec[i - 1], e2c_vec[i]);
                  next_n = i;
              }
          n = next_n;
      } while (n > 0);
  
      layout = 0;
      for (i = 0; i < total_non_cc_elements; i++) {
          layout_map[i][0] = e2c_vec[i].syn_ele;
          layout_map[i][1] = e2c_vec[i].elem_id;
          layout_map[i][2] = e2c_vec[i].aac_position;
          if (e2c_vec[i].av_position != UINT64_MAX) {
              layout |= e2c_vec[i].av_position;
          }
      }
  
      return layout;
  }
  
  /**
   * Save current output configuration if and only if it has been locked.
   */
  static void push_output_configuration(AACContext *ac) {
      if (ac->oc[1].status == OC_LOCKED) {
          ac->oc[0] = ac->oc[1];
      }
      ac->oc[1].status = OC_NONE;
  }
  
  /**
   * Restore the previous output configuration if and only if the current
   * configuration is unlocked.
   */
  static void pop_output_configuration(AACContext *ac) {
      if (ac->oc[1].status != OC_LOCKED && ac->oc[0].status != OC_NONE) {
          ac->oc[1] = ac->oc[0];
          ac->avctx->channels = ac->oc[1].channels;
          ac->avctx->channel_layout = ac->oc[1].channel_layout;
          output_configure(ac, ac->oc[1].layout_map, ac->oc[1].layout_map_tags,
                           ac->oc[1].status, 0);
      }
  }
  
  /**
   * Configure output channel order based on the current program
   * configuration element.
   *
   * @return  Returns error status. 0 - OK, !0 - error
   */
  static int output_configure(AACContext *ac,
                              uint8_t layout_map[MAX_ELEM_ID * 4][3], int tags,
                              enum OCStatus oc_type, int get_new_frame)
  {
      AVCodecContext *avctx = ac->avctx;
      int i, channels = 0, ret;
      uint64_t layout = 0;
  
      if (ac->oc[1].layout_map != layout_map) {
          memcpy(ac->oc[1].layout_map, layout_map, tags * sizeof(layout_map[0]));
          ac->oc[1].layout_map_tags = tags;
      }
  
      // Try to sniff a reasonable channel order, otherwise output the
      // channels in the order the PCE declared them.
      if (avctx->request_channel_layout != AV_CH_LAYOUT_NATIVE)
          layout = sniff_channel_order(layout_map, tags);
      for (i = 0; i < tags; i++) {
          int type =     layout_map[i][0];
          int id =       layout_map[i][1];
          int position = layout_map[i][2];
          // Allocate or free elements depending on if they are in the
          // current program configuration.
          ret = che_configure(ac, position, type, id, &channels);
          if (ret < 0)
              return ret;
      }
      if (ac->oc[1].m4ac.ps == 1 && channels == 2) {
          if (layout == AV_CH_FRONT_CENTER) {
              layout = AV_CH_FRONT_LEFT|AV_CH_FRONT_RIGHT;
          } else {
              layout = 0;
          }
      }
  
      memcpy(ac->tag_che_map, ac->che, 4 * MAX_ELEM_ID * sizeof(ac->che[0][0]));
      if (layout) avctx->channel_layout = layout;
                              ac->oc[1].channel_layout = layout;
      avctx->channels       = ac->oc[1].channels       = channels;
      ac->oc[1].status = oc_type;
  
      if (get_new_frame) {
          if ((ret = frame_configure_elements(ac->avctx)) < 0)
              return ret;
      }
  
      return 0;
  }
  
  static void flush(AVCodecContext *avctx)
  {
      AACContext *ac= avctx->priv_data;
      int type, i, j;
  
      for (type = 3; type >= 0; type--) {
          for (i = 0; i < MAX_ELEM_ID; i++) {
              ChannelElement *che = ac->che[type][i];
              if (che) {
                  for (j = 0; j <= 1; j++) {
                      memset(che->ch[j].saved, 0, sizeof(che->ch[j].saved));
                  }
              }
          }
      }
  }
  
  /**
   * Set up channel positions based on a default channel configuration
   * as specified in table 1.17.
   *
   * @return  Returns error status. 0 - OK, !0 - error
   */
  static int set_default_channel_config(AVCodecContext *avctx,
                                        uint8_t (*layout_map)[3],
                                        int *tags,
                                        int channel_config)
  {
      if (channel_config < 1 || channel_config > 7) {
          av_log(avctx, AV_LOG_ERROR,
                 "invalid default channel configuration (%d)\n",
                 channel_config);
          return 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 7:
          if (ac->tags_mapped == 3 && type == TYPE_CPE) {
              ac->tags_mapped++;
              return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][2];
          }
      case 6:
          /* Some streams incorrectly code 5.1 audio as
           * SCE[0] CPE[0] CPE[1] SCE[1]
           * instead of
           * SCE[0] CPE[0] CPE[1] LFE[0].
           * If we seem to have encountered such a stream, transfer
           * the LFE[0] element to the SCE[1]'s mapping */
          if (ac->tags_mapped == tags_per_config[ac->oc[1].m4ac.chan_config] - 1 && (type == TYPE_LFE || type == TYPE_SCE)) {
              ac->tags_mapped++;
              return ac->tag_che_map[type][elem_id] = ac->che[TYPE_LFE][0];
          }
      case 5:
          if (ac->tags_mapped == 2 && type == TYPE_CPE) {
              ac->tags_mapped++;
              return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][1];
          }
      case 4:
          if (ac->tags_mapped == 2 &&
              ac->oc[1].m4ac.chan_config == 4 &&
              type == TYPE_SCE) {
              ac->tags_mapped++;
              return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][1];
          }
      case 3:
      case 2:
          if (ac->tags_mapped == (ac->oc[1].m4ac.chan_config != 2) &&
              type == TYPE_CPE) {
              ac->tags_mapped++;
              return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][0];
          } else if (ac->oc[1].m4ac.chan_config == 2) {
              return NULL;
          }
      case 1:
          if (!ac->tags_mapped && type == TYPE_SCE) {
              ac->tags_mapped++;
              return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][0];
          }
      default:
          return NULL;
      }
  }
  
  /**
   * Decode an array of 4 bit element IDs, optionally interleaved with a
   * stereo/mono switching bit.
   *
   * @param type speaker type/position for these channels
   */
  static void decode_channel_map(uint8_t layout_map[][3],
                                 enum ChannelPosition type,
                                 GetBitContext *gb, int n)
  {
      while (n--) {
          enum RawDataBlockType syn_ele;
          switch (type) {
          case AAC_CHANNEL_FRONT:
          case AAC_CHANNEL_BACK:
          case AAC_CHANNEL_SIDE:
              syn_ele = get_bits1(gb);
              break;
          case AAC_CHANNEL_CC:
              skip_bits1(gb);
              syn_ele = TYPE_CCE;
              break;
          case AAC_CHANNEL_LFE:
              syn_ele = TYPE_LFE;
              break;
          default:
              av_assert0(0);
          }
          layout_map[0][0] = syn_ele;
          layout_map[0][1] = get_bits(gb, 4);
          layout_map[0][2] = type;
          layout_map++;
      }
  }
  
  /**
   * Decode program configuration element; reference: table 4.2.
   *
   * @return  Returns error status. 0 - OK, !0 - error
   */
  static int decode_pce(AVCodecContext *avctx, MPEG4AudioConfig *m4ac,
                        uint8_t (*layout_map)[3],
                        GetBitContext *gb)
  {
      int num_front, num_side, num_back, num_lfe, num_assoc_data, num_cc;
      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) < 4 * (num_front + num_side + num_back + num_lfe + num_assoc_data + num_cc)) {
          av_log(avctx, AV_LOG_ERROR, "decode_pce: " overread_err);
          return -1;
      }
      decode_channel_map(layout_map       , AAC_CHANNEL_FRONT, gb, num_front);
      tags = num_front;
      decode_channel_map(layout_map + tags, AAC_CHANNEL_SIDE,  gb, num_side);
      tags += num_side;
      decode_channel_map(layout_map + tags, AAC_CHANNEL_BACK,  gb, num_back);
      tags += num_back;
      decode_channel_map(layout_map + tags, AAC_CHANNEL_LFE,   gb, num_lfe);
      tags += num_lfe;
  
      skip_bits_long(gb, 4 * num_assoc_data);
  
      decode_channel_map(layout_map + tags, AAC_CHANNEL_CC,    gb, num_cc);
      tags += num_cc;
  
      align_get_bits(gb);
  
      /* comment field, first byte is length */
      comment_len = get_bits(gb, 8) * 8;
      if (get_bits_left(gb) < comment_len) {
          av_log(avctx, AV_LOG_ERROR, "decode_pce: " overread_err);
          return 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,
                                       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 (get_bits1(gb)) { // frameLengthFlag
          avpriv_request_sample(avctx, "960/120 MDCT window");
          return AVERROR_PATCHWELCOME;
      }
  
      if (get_bits1(gb))       // dependsOnCoreCoder
          skip_bits(gb, 14);   // coreCoderDelay
      extension_flag = get_bits1(gb);
  
      if (m4ac->object_type == AOT_AAC_SCALABLE ||
          m4ac->object_type == AOT_ER_AAC_SCALABLE)
          skip_bits(gb, 3);     // layerNr
  
      if (channel_config == 0) {
          skip_bits(gb, 4);  // element_instance_tag
          tags = decode_pce(avctx, m4ac, layout_map, gb);
          if (tags < 0)
              return tags;
      } else {
          if ((ret = set_default_channel_config(avctx, layout_map,
                                                &tags, channel_config)))
              return ret;
      }
  
      if (count_channels(layout_map, tags) > 1) {
          m4ac->ps = 0;
      } else if (m4ac->sbr == 1 && m4ac->ps == -1)
          m4ac->ps = 1;
  
      if (ac && (ret = output_configure(ac, layout_map, tags, OC_GLOBAL_HDR, 0)))
          return ret;
  
      if (extension_flag) {
          switch (m4ac->object_type) {
          case AOT_ER_BSAC:
              skip_bits(gb, 5);    // numOfSubFrame
              skip_bits(gb, 11);   // layer_length
              break;
          case AOT_ER_AAC_LC:
          case AOT_ER_AAC_LTP:
          case AOT_ER_AAC_SCALABLE:
          case AOT_ER_AAC_LD:
              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 (get_bits1(gb)) { // frameLengthFlag
          avpriv_request_sample(avctx, "960/120 MDCT window");
          return AVERROR_PATCHWELCOME;
      }
  
      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(ac->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   data        pointer to buffer holding an audio specific config
   * @param   bit_size    size of audio specific config or data in bits
   * @param   sync_extension look for an appended sync extension
   *
   * @return  Returns error status or number of consumed bits. <0 - error
   */
  static int decode_audio_specific_config(AACContext *ac,
                                          AVCodecContext *avctx,
                                          MPEG4AudioConfig *m4ac,
                                          const uint8_t *data, int bit_size,
                                          int sync_extension)
  {
      GetBitContext gb;
      int i, ret;
  
      av_dlog(avctx, "audio specific config size %d\n", bit_size >> 3);
      for (i = 0; i < bit_size >> 3; i++)
          av_dlog(avctx, "%02x ", data[i]);
      av_dlog(avctx, "\n");
  
      if ((ret = init_get_bits(&gb, data, bit_size)) < 0)
          return ret;
  
      if ((i = avpriv_mpeg4audio_get_config(m4ac, data, bit_size,
                                            sync_extension)) < 0)
          return 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,
                                              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);
      }
  
      av_dlog(avctx,
              "AOT %d chan config %d sampling index %d (%d) SBR %d PS %d\n",
              m4ac->object_type, m4ac->chan_config, m4ac->sampling_index,
              m4ac->sample_rate, m4ac->sbr,
              m4ac->ps);
  
      return get_bits_count(&gb);
  }
  
  /**
   * linear congruential pseudorandom number generator
   *
   * @param   previous_val    pointer to the current state of the generator
   *
   * @return  Returns a 32-bit pseudorandom integer
   */
  static av_always_inline int lcg_random(unsigned previous_val)
  {
      union { unsigned u; int s; } v = { previous_val * 1664525u + 1013904223 };
      return v.s;
  }
  
  static av_always_inline void reset_predict_state(PredictorState *ps)
  {
      ps->r0   = 0.0f;
      ps->r1   = 0.0f;
      ps->cor0 = 0.0f;
      ps->cor1 = 0.0f;
      ps->var0 = 1.0f;
      ps->var1 = 1.0f;
  }
  
  static void reset_all_predictors(PredictorState *ps)
  {
      int i;
      for (i = 0; i < MAX_PREDICTORS; i++)
          reset_predict_state(&ps[i]);
  }
  
  static int sample_rate_idx (int rate)
  {
           if (92017 <= rate) return 0;
      else if (75132 <= rate) return 1;
      else if (55426 <= rate) return 2;
      else if (46009 <= rate) return 3;
      else if (37566 <= rate) return 4;
      else if (27713 <= rate) return 5;
      else if (23004 <= rate) return 6;
      else if (18783 <= rate) return 7;
      else if (13856 <= rate) return 8;
      else if (11502 <= rate) return 9;
      else if (9391  <= rate) return 10;
      else                    return 11;
  }
  
  static void reset_predictor_group(PredictorState *ps, int group_num)
  {
      int i;
      for (i = group_num - 1; i < MAX_PREDICTORS; i += 30)
          reset_predict_state(&ps[i]);
  }
  
  #define AAC_INIT_VLC_STATIC(num, size)                                     \
      INIT_VLC_STATIC(&vlc_spectral[num], 8, ff_aac_spectral_sizes[num],     \
           ff_aac_spectral_bits[num], sizeof(ff_aac_spectral_bits[num][0]),  \
                                      sizeof(ff_aac_spectral_bits[num][0]),  \
          ff_aac_spectral_codes[num], sizeof(ff_aac_spectral_codes[num][0]), \
                                      sizeof(ff_aac_spectral_codes[num][0]), \
          size);
  
  static void aacdec_init(AACContext *ac);
  
  static av_cold int aac_decode_init(AVCodecContext *avctx)
  {
      AACContext *ac = avctx->priv_data;
      int ret;
  
      ac->avctx = avctx;
      ac->oc[1].m4ac.sample_rate = avctx->sample_rate;
  
      aacdec_init(ac);
  
      avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
  
      if (avctx->extradata_size > 0) {
          if ((ret = decode_audio_specific_config(ac, ac->avctx, &ac->oc[1].m4ac,
                                                  avctx->extradata,
                                                  avctx->extradata_size * 8,
                                                  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;
      }
  
      AAC_INIT_VLC_STATIC( 0, 304);
      AAC_INIT_VLC_STATIC( 1, 270);
      AAC_INIT_VLC_STATIC( 2, 550);
      AAC_INIT_VLC_STATIC( 3, 300);
      AAC_INIT_VLC_STATIC( 4, 328);
      AAC_INIT_VLC_STATIC( 5, 294);
      AAC_INIT_VLC_STATIC( 6, 306);
      AAC_INIT_VLC_STATIC( 7, 268);
      AAC_INIT_VLC_STATIC( 8, 510);
      AAC_INIT_VLC_STATIC( 9, 366);
      AAC_INIT_VLC_STATIC(10, 462);
  
      ff_aac_sbr_init();
  
      ff_fmt_convert_init(&ac->fmt_conv, avctx);
      avpriv_float_dsp_init(&ac->fdsp, avctx->flags & CODEC_FLAG_BITEXACT);
  
      ac->random_state = 0x1f2e3d4c;
  
      ff_aac_tableinit();
  
      INIT_VLC_STATIC(&vlc_scalefactors, 7,
                      FF_ARRAY_ELEMS(ff_aac_scalefactor_code),
                      ff_aac_scalefactor_bits,
                      sizeof(ff_aac_scalefactor_bits[0]),
                      sizeof(ff_aac_scalefactor_bits[0]),
                      ff_aac_scalefactor_code,
                      sizeof(ff_aac_scalefactor_code[0]),
                      sizeof(ff_aac_scalefactor_code[0]),
                      352);
  
      ff_mdct_init(&ac->mdct,       11, 1, 1.0 / (32768.0 * 1024.0));
      ff_mdct_init(&ac->mdct_ld,    10, 1, 1.0 / (32768.0 * 512.0));
      ff_mdct_init(&ac->mdct_small,  8, 1, 1.0 / (32768.0 * 128.0));
      ff_mdct_init(&ac->mdct_ltp,   11, 0, -2.0 * 32768.0);
      // window initialization
      ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024);
      ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128);
      ff_init_ff_sine_windows(10);
      ff_init_ff_sine_windows( 9);
      ff_init_ff_sine_windows( 7);
  
      cbrt_tableinit();
  
      return 0;
  }
  
  /**
   * Skip data_stream_element; reference: table 4.10.
   */
  static int skip_data_stream_element(AACContext *ac, GetBitContext *gb)
  {
      int byte_align = get_bits1(gb);
      int count = get_bits(gb, 8);
      if (count == 255)
          count += get_bits(gb, 8);
      if (byte_align)
          align_get_bits(gb);
  
      if (get_bits_left(gb) < 8 * count) {
          av_log(ac->avctx, AV_LOG_ERROR, "skip_data_stream_element: "overread_err);
          return 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)
  {
      int aot = ac->oc[1].m4ac.object_type;
      if (aot != AOT_ER_AAC_ELD) {
          if (get_bits1(gb)) {
              av_log(ac->avctx, AV_LOG_ERROR, "Reserved bit set.\n");
              return AVERROR_INVALIDDATA;
          }
          ics->window_sequence[1] = ics->window_sequence[0];
          ics->window_sequence[0] = get_bits(gb, 2);
          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;
          ics->swb_offset        =    ff_swb_offset_128[ac->oc[1].m4ac.sampling_index];
          ics->num_swb           =   ff_aac_num_swb_128[ac->oc[1].m4ac.sampling_index];
          ics->tns_max_bands     = ff_tns_max_bands_128[ac->oc[1].m4ac.sampling_index];
          ics->predictor_present = 0;
      } else {
          ics->max_sfb               = get_bits(gb, 6);
          ics->num_windows           = 1;
          if (aot == AOT_ER_AAC_LD || aot == AOT_ER_AAC_ELD) {
              ics->swb_offset        =     ff_swb_offset_512[ac->oc[1].m4ac.sampling_index];
              ics->num_swb           =    ff_aac_num_swb_512[ac->oc[1].m4ac.sampling_index];
              ics->tns_max_bands     =  ff_tns_max_bands_512[ac->oc[1].m4ac.sampling_index];
              if (!ics->num_swb || !ics->swb_offset)
                  return AVERROR_BUG;
          } else {
              ics->swb_offset        =    ff_swb_offset_1024[ac->oc[1].m4ac.sampling_index];
              ics->num_swb           =   ff_aac_num_swb_1024[ac->oc[1].m4ac.sampling_index];
              ics->tns_max_bands     = ff_tns_max_bands_1024[ac->oc[1].m4ac.sampling_index];
          }
          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");
                      return AVERROR_PATCHWELCOME;
                  }
                  if ((ics->ltp.present = get_bits(gb, 1)))
                      decode_ltp(&ics->ltp, gb, ics->max_sfb);
              }
          }
      }
  
      if (ics->max_sfb > ics->num_swb) {
          av_log(ac->avctx, AV_LOG_ERROR,
                 "Number of scalefactor bands in group (%d) "
                 "exceeds limit (%d).\n",
                 ics->max_sfb, ics->num_swb);
          goto fail;
      }
  
      return 0;
  fail:
      ics->max_sfb = 0;
      return AVERROR_INVALIDDATA;
  }
  
  /**
   * Decode band types (section_data payload); reference: table 4.46.
   *
   * @param   band_type           array of the used band type
   * @param   band_type_run_end   array of the last scalefactor band of a band type run
   *
   * @return  Returns error status. 0 - OK, !0 - error
   */
  static int decode_band_types(AACContext *ac, enum BandType band_type[120],
                               int band_type_run_end[120], GetBitContext *gb,
                               IndividualChannelStream *ics)
  {
      int g, idx = 0;
      const int bits = (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) ? 3 : 5;
      for (g = 0; g < ics->num_window_groups; g++) {
          int k = 0;
          while (k < ics->max_sfb) {
              uint8_t sect_end = k;
              int sect_len_incr;
              int sect_band_type = get_bits(gb, 4);
              if (sect_band_type == 12) {
                  av_log(ac->avctx, AV_LOG_ERROR, "invalid band type\n");
                  return 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, float sf[120], GetBitContext *gb,
                                 unsigned int global_gain,
                                 IndividualChannelStream *ics,
                                 enum BandType band_type[120],
                                 int band_type_run_end[120])
  {
      int g, i, idx = 0;
      int offset[3] = { global_gain, global_gain - 90, 0 };
      int clipped_offset;
      int noise_flag = 1;
      for (g = 0; g < ics->num_window_groups; g++) {
          for (i = 0; i < ics->max_sfb;) {
              int run_end = band_type_run_end[idx];
              if (band_type[idx] == ZERO_BT) {
                  for (; i < run_end; i++, idx++)
                      sf[idx] = 0.0;
              } else if ((band_type[idx] == INTENSITY_BT) ||
                         (band_type[idx] == INTENSITY_BT2)) {
                  for (; i < run_end; i++, idx++) {
                      offset[2] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
                      clipped_offset = av_clip(offset[2], -155, 100);
                      if (offset[2] != clipped_offset) {
                          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);
                      }
                      sf[idx] = ff_aac_pow2sf_tab[-clipped_offset + POW_SF2_ZERO];
                  }
              } else if (band_type[idx] == NOISE_BT) {
                  for (; i < run_end; i++, idx++) {
                      if (noise_flag-- > 0)
                          offset[1] += get_bits(gb, 9) - 256;
                      else
                          offset[1] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
                      clipped_offset = av_clip(offset[1], -100, 155);
                      if (offset[1] != clipped_offset) {
                          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);
                      }
                      sf[idx] = -ff_aac_pow2sf_tab[clipped_offset + POW_SF2_ZERO];
                  }
              } else {
                  for (; i < run_end; i++, idx++) {
                      offset[0] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
                      if (offset[0] > 255U) {
                          av_log(ac->avctx, AV_LOG_ERROR,
                                 "Scalefactor (%d) out of range.\n", offset[0]);
                          return AVERROR_INVALIDDATA;
                      }
                      sf[idx] = -ff_aac_pow2sf_tab[offset[0] - 100 + POW_SF2_ZERO];
                  }
              }
          }
      }
      return 0;
  }
  
  /**
   * Decode pulse data; reference: table 4.7.
   */
  static int decode_pulses(Pulse *pulse, GetBitContext *gb,
                           const uint16_t *swb_offset, int num_swb)
  {
      int i, pulse_swb;
      pulse->num_pulse = get_bits(gb, 2) + 1;
      pulse_swb        = get_bits(gb, 6);
      if (pulse_swb >= num_swb)
          return -1;
      pulse->pos[0]    = swb_offset[pulse_swb];
      pulse->pos[0]   += get_bits(gb, 5);
      if (pulse->pos[0] >= 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;
      if (ms_present == 1) {
          for (idx = 0;
               idx < cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb;
               idx++)
              cpe->ms_mask[idx] = get_bits1(gb);
      } else if (ms_present == 2) {
          memset(cpe->ms_mask, 1,  sizeof(cpe->ms_mask[0]) * cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb);
      }
  }
  
  #ifndef VMUL2
  static inline float *VMUL2(float *dst, const float *v, unsigned idx,
                             const float *scale)
  {
      float s = *scale;
      *dst++ = v[idx    & 15] * s;
      *dst++ = v[idx>>4 & 15] * s;
      return dst;
  }
  #endif
  
  #ifndef VMUL4
  static inline float *VMUL4(float *dst, const float *v, unsigned idx,
                             const float *scale)
  {
      float s = *scale;
      *dst++ = v[idx    & 3] * s;
      *dst++ = v[idx>>2 & 3] * s;
      *dst++ = v[idx>>4 & 3] * s;
      *dst++ = v[idx>>6 & 3] * s;
      return dst;
  }
  #endif
  
  #ifndef VMUL2S
  static inline float *VMUL2S(float *dst, const float *v, unsigned idx,
                              unsigned sign, const float *scale)
  {
      union av_intfloat32 s0, s1;
  
      s0.f = s1.f = *scale;
      s0.i ^= sign >> 1 << 31;
      s1.i ^= sign      << 31;
  
      *dst++ = v[idx    & 15] * s0.f;
      *dst++ = v[idx>>4 & 15] * s1.f;
  
      return dst;
  }
  #endif
  
  #ifndef VMUL4S
  static inline float *VMUL4S(float *dst, const float *v, unsigned idx,
                              unsigned sign, const float *scale)
  {
      unsigned nz = idx >> 12;
      union av_intfloat32 s = { .f = *scale };
      union av_intfloat32 t;
  
      t.i = s.i ^ (sign & 1U<<31);
      *dst++ = v[idx    & 3] * t.f;
  
      sign <<= nz & 1; nz >>= 1;
      t.i = s.i ^ (sign & 1U<<31);
      *dst++ = v[idx>>2 & 3] * t.f;
  
      sign <<= nz & 1; nz >>= 1;
      t.i = s.i ^ (sign & 1U<<31);
      *dst++ = v[idx>>4 & 3] * t.f;
  
      sign <<= nz & 1;
      t.i = s.i ^ (sign & 1U<<31);
      *dst++ = v[idx>>6 & 3] * t.f;
  
      return dst;
  }
  #endif
  
  /**
   * Decode spectral data; reference: table 4.50.
   * Dequantize and scale spectral data; reference: 4.6.3.3.
   *
   * @param   coef            array of dequantized, scaled spectral data
   * @param   sf              array of scalefactors or intensity stereo positions
   * @param   pulse_present   set if pulses are present
   * @param   pulse           pointer to pulse data struct
   * @param   band_type       array of the used band type
   *
   * @return  Returns error status. 0 - OK, !0 - error
   */
  static int decode_spectrum_and_dequant(AACContext *ac, float coef[1024],
                                         GetBitContext *gb, const float sf[120],
                                         int pulse_present, const Pulse *pulse,
                                         const IndividualChannelStream *ics,
                                         enum BandType band_type[120])
  {
      int i, k, g, idx = 0;
      const int c = 1024 / ics->num_windows;
      const uint16_t *offsets = ics->swb_offset;
      float *coef_base = coef;
  
      for (g = 0; g < ics->num_windows; g++)
          memset(coef + g * 128 + offsets[ics->max_sfb], 0,
                 sizeof(float) * (c - offsets[ics->max_sfb]));
  
      for (g = 0; g < ics->num_window_groups; g++) {
          unsigned g_len = ics->group_len[g];
  
          for (i = 0; i < ics->max_sfb; i++, idx++) {
              const unsigned cbt_m1 = band_type[idx] - 1;
              float *cfo = coef + offsets[i];
              int off_len = offsets[i + 1] - offsets[i];
              int group;
  
              if (cbt_m1 >= INTENSITY_BT2 - 1) {
                  for (group = 0; group < g_len; group++, cfo+=128) {
                      memset(cfo, 0, off_len * sizeof(float));
                  }
              } else if (cbt_m1 == NOISE_BT - 1) {
                  for (group = 0; group < g_len; group++, cfo+=128) {
                      float scale;
                      float band_energy;
  
                      for (k = 0; k < off_len; k++) {
                          ac->random_state  = lcg_random(ac->random_state);
                          cfo[k] = ac->random_state;
                      }
  
                      band_energy = ac->fdsp.scalarproduct_float(cfo, cfo, off_len);
                      scale = sf[idx] / sqrtf(band_energy);
                      ac->fdsp.vector_fmul_scalar(cfo, cfo, scale, off_len);
                  }
              } else {
                  const float *vq = ff_aac_codebook_vector_vals[cbt_m1];
                  const uint16_t *cb_vector_idx = ff_aac_codebook_vector_idx[cbt_m1];
                  VLC_TYPE (*vlc_tab)[2] = vlc_spectral[cbt_m1].table;
                  OPEN_READER(re, gb);
  
                  switch (cbt_m1 >> 1) {
                  case 0:
                      for (group = 0; group < g_len; group++, cfo+=128) {
                          float *cf = cfo;
                          int len = off_len;
  
                          do {
                              int code;
                              unsigned cb_idx;
  
                              UPDATE_CACHE(re, gb);
                              GET_VLC(code, re, gb, vlc_tab, 8, 2);
                              cb_idx = cb_vector_idx[code];
                              cf = VMUL4(cf, vq, cb_idx, sf + idx);
                          } while (len -= 4);
                      }
                      break;
  
                  case 1:
                      for (group = 0; group < g_len; group++, cfo+=128) {
                          float *cf = cfo;
                          int len = off_len;
  
                          do {
                              int code;
                              unsigned nnz;
                              unsigned cb_idx;
                              uint32_t bits;
  
                              UPDATE_CACHE(re, gb);
                              GET_VLC(code, re, gb, vlc_tab, 8, 2);
                              cb_idx = cb_vector_idx[code];
                              nnz = cb_idx >> 8 & 15;
                              bits = nnz ? GET_CACHE(re, gb) : 0;
                              LAST_SKIP_BITS(re, gb, nnz);
                              cf = VMUL4S(cf, vq, cb_idx, bits, sf + idx);
                          } while (len -= 4);
                      }
                      break;
  
                  case 2:
                      for (group = 0; group < g_len; group++, cfo+=128) {
                          float *cf = cfo;
                          int len = off_len;
  
                          do {
                              int code;
                              unsigned cb_idx;
  
                              UPDATE_CACHE(re, gb);
                              GET_VLC(code, re, gb, vlc_tab, 8, 2);
                              cb_idx = cb_vector_idx[code];
                              cf = VMUL2(cf, vq, cb_idx, sf + idx);
                          } while (len -= 2);
                      }
                      break;
  
                  case 3:
                  case 4:
                      for (group = 0; group < g_len; group++, cfo+=128) {
                          float *cf = cfo;
                          int len = off_len;
  
                          do {
                              int code;
                              unsigned nnz;
                              unsigned cb_idx;
                              unsigned sign;
  
                              UPDATE_CACHE(re, gb);
                              GET_VLC(code, re, gb, vlc_tab, 8, 2);
                              cb_idx = cb_vector_idx[code];
                              nnz = cb_idx >> 8 & 15;
                              sign = nnz ? SHOW_UBITS(re, gb, nnz) << (cb_idx >> 12) : 0;
                              LAST_SKIP_BITS(re, gb, nnz);
                              cf = VMUL2S(cf, vq, cb_idx, sign, sf + idx);
                          } while (len -= 2);
                      }
                      break;
  
                  default:
                      for (group = 0; group < g_len; group++, cfo+=128) {
                          float *cf = cfo;
                          uint32_t *icf = (uint32_t *) cf;
                          int len = off_len;
  
                          do {
                              int code;
                              unsigned nzt, nnz;
                              unsigned cb_idx;
                              uint32_t bits;
                              int j;
  
                              UPDATE_CACHE(re, gb);
                              GET_VLC(code, re, gb, vlc_tab, 8, 2);
  
                              if (!code) {
                                  *icf++ = 0;
                                  *icf++ = 0;
                                  continue;
                              }
  
                              cb_idx = cb_vector_idx[code];
                              nnz = cb_idx >> 12;
                              nzt = cb_idx >> 8;
                              bits = SHOW_UBITS(re, gb, nnz) << (32-nnz);
                              LAST_SKIP_BITS(re, gb, nnz);
  
                              for (j = 0; j < 2; j++) {
                                  if (nzt & 1<<j) {
                                      uint32_t b;
                                      int n;
                                      /* The total length of escape_sequence must be < 22 bits according
                                         to the specification (i.e. max is 111111110xxxxxxxxxxxx). */
                                      UPDATE_CACHE(re, gb);
                                      b = GET_CACHE(re, gb);
                                      b = 31 - av_log2(~b);
  
                                      if (b > 8) {
                                          av_log(ac->avctx, AV_LOG_ERROR, "error in spectral data, ESC overflow\n");
                                          return AVERROR_INVALIDDATA;
                                      }
  
                                      SKIP_BITS(re, gb, b + 1);
                                      b += 4;
                                      n = (1 << b) + SHOW_UBITS(re, gb, b);
                                      LAST_SKIP_BITS(re, gb, b);
                                      *icf++ = cbrt_tab[n] | (bits & 1U<<31);
                                      bits <<= 1;
                                  } else {
                                      unsigned v = ((const uint32_t*)vq)[cb_idx & 15];
                                      *icf++ = (bits & 1U<<31) | v;
                                      bits <<= !!v;
                                  }
                                  cb_idx >>= 4;
                              }
                          } while (len -= 2);
  
                          ac->fdsp.vector_fmul_scalar(cfo, cfo, sf[idx], off_len);
                      }
                  }
  
                  CLOSE_READER(re, gb);
              }
          }
          coef += g_len << 7;
      }
  
      if (pulse_present) {
          idx = 0;
          for (i = 0; i < pulse->num_pulse; i++) {
              float co = coef_base[ pulse->pos[i] ];
              while (offsets[idx + 1] <= pulse->pos[i])
                  idx++;
              if (band_type[idx] != NOISE_BT && sf[idx]) {
                  float ico = -pulse->amp[i];
                  if (co) {
                      co /= sf[idx];
                      ico = co / sqrtf(sqrtf(fabsf(co))) + (co > 0 ? -ico : ico);
                  }
                  coef_base[ pulse->pos[i] ] = cbrtf(fabsf(ico)) * ico * sf[idx];
              }
          }
      }
      return 0;
  }
  
  static av_always_inline float flt16_round(float pf)
  {
      union av_intfloat32 tmp;
      tmp.f = pf;
      tmp.i = (tmp.i + 0x00008000U) & 0xFFFF0000U;
      return tmp.f;
  }
  
  static av_always_inline float flt16_even(float pf)
  {
      union av_intfloat32 tmp;
      tmp.f = pf;
      tmp.i = (tmp.i + 0x00007FFFU + (tmp.i & 0x00010000U >> 16)) & 0xFFFF0000U;
      return tmp.f;
  }
  
  static av_always_inline float flt16_trunc(float pf)
  {
      union av_intfloat32 pun;
      pun.f = pf;
      pun.i &= 0xFFFF0000U;
      return pun.f;
  }
  
  static av_always_inline void predict(PredictorState *ps, float *coef,
                                       int output_enable)
  {
      const float a     = 0.953125; // 61.0 / 64
      const float alpha = 0.90625;  // 29.0 / 32
      float e0, e1;
      float pv;
      float k1, k2;
      float   r0 = ps->r0,     r1 = ps->r1;
      float cor0 = ps->cor0, cor1 = ps->cor1;
      float var0 = ps->var0, var1 = ps->var1;
  
      k1 = var0 > 1 ? cor0 * flt16_even(a / var0) : 0;
      k2 = var1 > 1 ? cor1 * flt16_even(a / var1) : 0;
  
      pv = flt16_round(k1 * r0 + k2 * r1);
      if (output_enable)
          *coef += pv;
  
      e0 = *coef;
      e1 = e0 - k1 * r0;
  
      ps->cor1 = flt16_trunc(alpha * cor1 + r1 * e1);
      ps->var1 = flt16_trunc(alpha * var1 + 0.5f * (r1 * r1 + e1 * e1));
      ps->cor0 = flt16_trunc(alpha * cor0 + r0 * e0);
      ps->var0 = flt16_trunc(alpha * var0 + 0.5f * (r0 * r0 + e0 * e0));
  
      ps->r1 = flt16_trunc(a * (r0 - k1 * e0));
      ps->r0 = flt16_trunc(a * e0);
  }
  
  /**
   * Apply AAC-Main style frequency domain prediction.
   */
  static void apply_prediction(AACContext *ac, SingleChannelElement *sce)
  {
      int sfb, k;
  
      if (!sce->ics.predictor_initialized) {
          reset_all_predictors(sce->predictor_state);
          sce->ics.predictor_initialized = 1;
      }
  
      if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
          for (sfb = 0;
               sfb < ff_aac_pred_sfb_max[ac->oc[1].m4ac.sampling_index];
               sfb++) {
              for (k = sce->ics.swb_offset[sfb];
                   k < sce->ics.swb_offset[sfb + 1];
                   k++) {
                  predict(&sce->predictor_state[k], &sce->coeffs[k],
                          sce->ics.predictor_present &&
                          sce->ics.prediction_used[sfb]);
              }
          }
          if (sce->ics.predictor_reset_group)
              reset_predictor_group(sce->predictor_state,
                                    sce->ics.predictor_reset_group);
      } else
          reset_all_predictors(sce->predictor_state);
  }
  
  /**
   * Decode an individual_channel_stream payload; reference: table 4.44.
   *
   * @param   common_window   Channels have independent [0], or shared [1], Individual Channel Stream information.
   * @param   scale_flag      scalable [1] or non-scalable [0] AAC (Unused until scalable AAC is implemented.)
   *
   * @return  Returns error status. 0 - OK, !0 - error
   */
  static int decode_ics(AACContext *ac, SingleChannelElement *sce,
                        GetBitContext *gb, int common_window, int scale_flag)
  {
      Pulse pulse;
      TemporalNoiseShaping    *tns = &sce->tns;
      IndividualChannelStream *ics = &sce->ics;
      float *out = sce->coeffs;
      int global_gain, 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) {
          if (decode_ics_info(ac, ics, gb) < 0)
              return AVERROR_INVALIDDATA;
      }
  
      if ((ret = decode_band_types(ac, sce->band_type,
                                   sce->band_type_run_end, gb, ics)) < 0)
          return ret;
      if ((ret = decode_scalefactors(ac, sce->sf, gb, global_gain, ics,
                                    sce->band_type, sce->band_type_run_end)) < 0)
          return ret;
  
      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");
                  return AVERROR_INVALIDDATA;
              }
              if (decode_pulses(&pulse, gb, ics->swb_offset, ics->num_swb)) {
                  av_log(ac->avctx, AV_LOG_ERROR,
                         "Pulse data corrupt or invalid.\n");
                  return AVERROR_INVALIDDATA;
              }
          }
          tns->present = get_bits1(gb);
          if (tns->present && !er_syntax)
              if (decode_tns(ac, tns, gb, ics) < 0)
                  return AVERROR_INVALIDDATA;
          if (!eld_syntax && get_bits1(gb)) {
              avpriv_request_sample(ac->avctx, "SSR");
              return AVERROR_PATCHWELCOME;
          }
          // I see no textual basis in the spec for this occuring after SSR gain
          // control, but this is what both reference and real implmentations do
          if (tns->present && er_syntax)
              if (decode_tns(ac, tns, gb, ics) < 0)
                  return AVERROR_INVALIDDATA;
      }
  
      if (decode_spectrum_and_dequant(ac, out, gb, sce->sf, pulse_present,
                                      &pulse, ics, sce->band_type) < 0)
          return AVERROR_INVALIDDATA;
  
      if (ac->oc[1].m4ac.object_type == AOT_AAC_MAIN && !common_window)
          apply_prediction(ac, sce);
  
      return 0;
  }
  
  /**
   * Mid/Side stereo decoding; reference: 4.6.8.1.3.
   */
  static void apply_mid_side_stereo(AACContext *ac, ChannelElement *cpe)
  {
      const IndividualChannelStream *ics = &cpe->ch[0].ics;
      float *ch0 = cpe->ch[0].coeffs;
      float *ch1 = cpe->ch[1].coeffs;
      int g, i, group, idx = 0;
      const uint16_t *offsets = ics->swb_offset;
      for (g = 0; g < ics->num_window_groups; g++) {
          for (i = 0; i < ics->max_sfb; i++, idx++) {
              if (cpe->ms_mask[idx] &&
                  cpe->ch[0].band_type[idx] < NOISE_BT &&
                  cpe->ch[1].band_type[idx] < NOISE_BT) {
                  for (group = 0; group < ics->group_len[g]; group++) {
                      ac->fdsp.butterflies_float(ch0 + group * 128 + offsets[i],
                                                 ch1 + group * 128 + offsets[i],
                                                 offsets[i+1] - offsets[i]);
                  }
              }
          }
          ch0 += ics->group_len[g] * 128;
          ch1 += ics->group_len[g] * 128;
      }
  }
  
  /**
   * intensity stereo decoding; reference: 4.6.8.2.3
   *
   * @param   ms_present  Indicates mid/side stereo presence. [0] mask is all 0s;
   *                      [1] mask is decoded from bitstream; [2] mask is all 1s;
   *                      [3] reserved for scalable AAC
   */
  static void apply_intensity_stereo(AACContext *ac,
                                     ChannelElement *cpe, int ms_present)
  {
      const IndividualChannelStream *ics = &cpe->ch[1].ics;
      SingleChannelElement         *sce1 = &cpe->ch[1];
      float *coef0 = cpe->ch[0].coeffs, *coef1 = cpe->ch[1].coeffs;
      const uint16_t *offsets = ics->swb_offset;
      int g, group, i, idx = 0;
      int c;
      float scale;
      for (g = 0; g < ics->num_window_groups; g++) {
          for (i = 0; i < ics->max_sfb;) {
              if (sce1->band_type[idx] == INTENSITY_BT ||
                  sce1->band_type[idx] == INTENSITY_BT2) {
                  const int bt_run_end = sce1->band_type_run_end[idx];
                  for (; i < bt_run_end; i++, idx++) {
                      c = -1 + 2 * (sce1->band_type[idx] - 14);
                      if (ms_present)
                          c *= 1 - 2 * cpe->ms_mask[idx];
                      scale = c * sce1->sf[idx];
                      for (group = 0; group < ics->group_len[g]; group++)
                          ac->fdsp.vector_fmul_scalar(coef1 + group * 128 + offsets[i],
                                                      coef0 + group * 128 + offsets[i],
                                                      scale,
                                                      offsets[i + 1] - offsets[i]);
                  }
              } else {
                  int bt_run_end = sce1->band_type_run_end[idx];
                  idx += bt_run_end - i;
                  i    = bt_run_end;
              }
          }
          coef0 += ics->group_len[g] * 128;
          coef1 += ics->group_len[g] * 128;
      }
  }
  
  /**
   * Decode a channel_pair_element; reference: table 4.4.
   *
   * @return  Returns error status. 0 - OK, !0 - error
   */
  static int decode_cpe(AACContext *ac, GetBitContext *gb, ChannelElement *cpe)
  {
      int i, ret, common_window, ms_present = 0;
      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;
      float scale;
      SingleChannelElement *sce = &che->ch[0];
      ChannelCoupling     *coup = &che->coup;
  
      coup->coupling_point = 2 * get_bits1(gb);
      coup->num_coupled = get_bits(gb, 3);
      for (c = 0; c <= coup->num_coupled; c++) {
          num_gain++;
          coup->type[c] = get_bits1(gb) ? TYPE_CPE : TYPE_SCE;
          coup->id_select[c] = get_bits(gb, 4);
          if (coup->type[c] == TYPE_CPE) {
              coup->ch_select[c] = get_bits(gb, 2);
              if (coup->ch_select[c] == 3)
                  num_gain++;
          } else
              coup->ch_select[c] = 2;
      }
      coup->coupling_point += get_bits1(gb) || (coup->coupling_point >> 1);
  
      sign  = get_bits(gb, 1);
      scale = cce_scale[get_bits(gb, 2)];
  
      if ((ret = decode_ics(ac, sce, gb, 0, 0)))
          return ret;
  
      for (c = 0; c < num_gain; c++) {
          int idx  = 0;
          int cge  = 1;
          int gain = 0;
          float gain_cache = 1.0;
          if (c) {
              cge = coup->coupling_point == AFTER_IMDCT ? 1 : get_bits1(gb);
              gain = cge ? get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60: 0;
              gain_cache = powf(scale, -gain);
          }
          if (coup->coupling_point == AFTER_IMDCT) {
              coup->gain[c][0] = gain_cache;
          } else {
              for (g = 0; g < sce->ics.num_window_groups; g++) {
                  for (sfb = 0; sfb < sce->ics.max_sfb; sfb++, idx++) {
                      if (sce->band_type[idx] != ZERO_BT) {
                          if (!cge) {
                              int t = get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
                              if (t) {
                                  int s = 1;
                                  t = gain += t;
                                  if (sign) {
                                      s  -= 2 * (t & 0x1);
                                      t >>= 1;
                                  }
                                  gain_cache = powf(scale, -t) * s;
                              }
                          }
                          coup->gain[c][idx] = gain_cache;
                      }
                  }
              }
          }
      }
      return 0;
  }
  
  /**
   * Parse whether channels are to be excluded from Dynamic Range Compression; reference: table 4.53.
   *
   * @return  Returns number of bytes consumed.
   */
  static int decode_drc_channel_exclusions(DynamicRangeControl *che_drc,
                                           GetBitContext *gb)
  {
      int i;
      int num_excl_chan = 0;
  
      do {
          for (i = 0; i < 7; i++)
              che_drc->exclude_mask[num_excl_chan++] = get_bits1(gb);
      } while (num_excl_chan < MAX_CHANNELS - 7 && get_bits1(gb));
  
      return num_excl_chan / 7;
  }
  
  /**
   * Decode dynamic range information; reference: table 4.52.
   *
   * @return  Returns number of bytes consumed.
   */
  static int decode_dynamic_range(DynamicRangeControl *che_drc,
                                  GetBitContext *gb)
  {
      int n             = 1;
      int drc_num_bands = 1;
      int i;
  
      /* pce_tag_present? */
      if (get_bits1(gb)) {
          che_drc->pce_instance_tag  = get_bits(gb, 4);
          skip_bits(gb, 4); // tag_reserved_bits
          n++;
      }
  
      /* excluded_chns_present? */
      if (get_bits1(gb)) {
          n += decode_drc_channel_exclusions(che_drc, gb);
      }
  
      /* drc_bands_present? */
      if (get_bits1(gb)) {
          che_drc->band_incr            = get_bits(gb, 4);
          che_drc->interpolation_scheme = get_bits(gb, 4);
          n++;
          drc_num_bands += che_drc->band_incr;
          for (i = 0; i < drc_num_bands; i++) {
              che_drc->band_top[i] = get_bits(gb, 8);
              n++;
          }
      }
  
      /* prog_ref_level_present? */
      if (get_bits1(gb)) {
          che_drc->prog_ref_level = get_bits(gb, 7);
          skip_bits1(gb); // prog_ref_level_reserved_bits
          n++;
      }
  
      for (i = 0; i < drc_num_bands; i++) {
          che_drc->dyn_rng_sgn[i] = get_bits1(gb);
          che_drc->dyn_rng_ctl[i] = get_bits(gb, 7);
          n++;
      }
  
      return n;
  }
  
  static int decode_fill(AACContext *ac, GetBitContext *gb, int len) {
      uint8_t buf[256];
      int i, major, minor;
  
      if (len < 13+7*8)
          goto unknown;
  
      get_bits(gb, 13); len -= 13;
  
      for(i=0; i+1<sizeof(buf) && len>=8; i++, len-=8)
          buf[i] = get_bits(gb, 8);
  
      buf[i] = 0;
      if (ac->avctx->debug & FF_DEBUG_PICT_INFO)
          av_log(ac->avctx, AV_LOG_DEBUG, "FILL:%s\n", buf);
  
      if (sscanf(buf, "libfaac %d.%d", &major, &minor) == 2){
          ac->avctx->internal->skip_samples = 1024;
      }
  
  unknown:
      skip_bits_long(gb, len);
  
      return 0;
  }
  
  /**
   * Decode extension data (incomplete); reference: table 4.51.
   *
   * @param   cnt length of TYPE_FIL syntactic element in bytes
   *
   * @return Returns number of bytes consumed
   */
  static int decode_extension_payload(AACContext *ac, GetBitContext *gb, int cnt,
                                      ChannelElement *che, enum RawDataBlockType elem_type)
  {
      int crc_flag = 0;
      int res = cnt;
      switch (get_bits(gb, 4)) { // extension type
      case EXT_SBR_DATA_CRC:
          crc_flag++;
      case EXT_SBR_DATA:
          if (!che) {
              av_log(ac->avctx, AV_LOG_ERROR, "SBR was found before the first channel element.\n");
              return res;
          } else if (!ac->oc[1].m4ac.sbr) {
              av_log(ac->avctx, AV_LOG_ERROR, "SBR signaled to be not-present but was found in the bitstream.\n");
              skip_bits_long(gb, 8 * cnt - 4);
              return res;
          } else if (ac->oc[1].m4ac.sbr == -1 && ac->oc[1].status == OC_LOCKED) {
              av_log(ac->avctx, AV_LOG_ERROR, "Implicit SBR was found with a first occurrence after the first frame.\n");
              skip_bits_long(gb, 8 * cnt - 4);
              return res;
          } else if (ac->oc[1].m4ac.ps == -1 && ac->oc[1].status < OC_LOCKED && ac->avctx->channels == 1) {
              ac->oc[1].m4ac.sbr = 1;
              ac->oc[1].m4ac.ps = 1;
              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 = ff_decode_sbr_extension(ac, &che->sbr, gb, crc_flag, cnt, elem_type);
          break;
      case EXT_DYNAMIC_RANGE:
          res = decode_dynamic_range(&ac->che_drc, gb);
          break;
      case EXT_FILL:
          decode_fill(ac, gb, 8 * cnt - 4);
          break;
      case EXT_FILL_DATA:
      case EXT_DATA_ELEMENT:
      default:
          skip_bits_long(gb, 8 * cnt - 4);
          break;
      };
      return res;
  }
  
  /**
   * Decode Temporal Noise Shaping filter coefficients and apply all-pole filters; reference: 4.6.9.3.
   *
   * @param   decode  1 if tool is used normally, 0 if tool is used in LTP.
   * @param   coef    spectral coefficients
   */
  static void apply_tns(float coef[1024], TemporalNoiseShaping *tns,
                        IndividualChannelStream *ics, int decode)
  {
      const int mmm = FFMIN(ics->tns_max_bands, ics->max_sfb);
      int w, filt, m, i;
      int bottom, top, order, start, end, size, inc;
      float lpc[TNS_MAX_ORDER];
      float tmp[TNS_MAX_ORDER+1];
  
      for (w = 0; w < ics->num_windows; w++) {
          bottom = ics->num_swb;
          for (filt = 0; filt < tns->n_filt[w]; filt++) {
              top    = bottom;
              bottom = FFMAX(0, top - tns->length[w][filt]);
              order  = tns->order[w][filt];
              if (order == 0)
                  continue;
  
              // tns_decode_coef
              compute_lpc_coefs(tns->coef[w][filt], order, lpc, 0, 0, 0);
  
              start = ics->swb_offset[FFMIN(bottom, mmm)];
              end   = ics->swb_offset[FFMIN(   top, mmm)];
              if ((size = end - start) <= 0)
                  continue;
              if (tns->direction[w][filt]) {
                  inc = -1;
                  start = end - 1;
              } else {
                  inc = 1;
              }
              start += w * 128;
  
              if (decode) {
                  // ar filter
                  for (m = 0; m < size; m++, start += inc)
                      for (i = 1; i <= FFMIN(m, order); i++)
                          coef[start] -= coef[start - i * inc] * lpc[i - 1];
              } else {
                  // ma filter
                  for (m = 0; m < size; m++, start += inc) {
                      tmp[0] = coef[start];
                      for (i = 1; i <= FFMIN(m, order); i++)
                          coef[start] += tmp[i] * lpc[i - 1];
                      for (i = order; i > 0; i--)
                          tmp[i] = tmp[i - 1];
                  }
              }
          }
      }
  }
  
  /**
   *  Apply windowing and MDCT to obtain the spectral
   *  coefficient from the predicted sample by LTP.
   */
  static void windowing_and_mdct_ltp(AACContext *ac, float *out,
                                     float *in, IndividualChannelStream *ics)
  {
      const float *lwindow      = ics->use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
      const float *swindow      = ics->use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
      const float *lwindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;
      const float *swindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
  
      if (ics->window_sequence[0] != LONG_STOP_SEQUENCE) {
          ac->fdsp.vector_fmul(in, in, lwindow_prev, 1024);
      } else {
          memset(in, 0, 448 * sizeof(float));
          ac->fdsp.vector_fmul(in + 448, in + 448, swindow_prev, 128);
      }
      if (ics->window_sequence[0] != LONG_START_SEQUENCE) {
          ac->fdsp.vector_fmul_reverse(in + 1024, in + 1024, lwindow, 1024);
      } else {
          ac->fdsp.vector_fmul_reverse(in + 1024 + 448, in + 1024 + 448, swindow, 128);
          memset(in + 1024 + 576, 0, 448 * sizeof(float));
      }
      ac->mdct_ltp.mdct_calc(&ac->mdct_ltp, out, in);
  }
  
  /**
   * Apply the long term prediction
   */
  static void apply_ltp(AACContext *ac, SingleChannelElement *sce)
  {
      const LongTermPrediction *ltp = &sce->ics.ltp;
      const uint16_t *offsets = sce->ics.swb_offset;
      int i, sfb;
  
      if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
          float *predTime = sce->ret;
          float *predFreq = ac->buf_mdct;
          int16_t num_samples = 2048;
  
          if (ltp->lag < 1024)
              num_samples = ltp->lag + 1024;
          for (i = 0; i < num_samples; i++)
              predTime[i] = sce->ltp_state[i + 2048 - ltp->lag] * ltp->coef;
          memset(&predTime[i], 0, (2048 - i) * sizeof(float));
  
          ac->windowing_and_mdct_ltp(ac, predFreq, predTime, &sce->ics);
  
          if (sce->tns.present)
              ac->apply_tns(predFreq, &sce->tns, &sce->ics, 0);
  
          for (sfb = 0; sfb < FFMIN(sce->ics.max_sfb, MAX_LTP_LONG_SFB); sfb++)
              if (ltp->used[sfb])
                  for (i = offsets[sfb]; i < offsets[sfb + 1]; i++)
                      sce->coeffs[i] += predFreq[i];
      }
  }
  
  /**
   * Update the LTP buffer for next frame
   */
  static void update_ltp(AACContext *ac, SingleChannelElement *sce)
  {
      IndividualChannelStream *ics = &sce->ics;
      float *saved     = sce->saved;
      float *saved_ltp = sce->coeffs;
      const float *lwindow = ics->use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
      const float *swindow = ics->use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
      int i;
  
      if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
          memcpy(saved_ltp,       saved, 512 * sizeof(float));
          memset(saved_ltp + 576, 0,     448 * sizeof(float));
          ac->fdsp.vector_fmul_reverse(saved_ltp + 448, ac->buf_mdct + 960,     &swindow[64],      64);
          for (i = 0; i < 64; i++)
              saved_ltp[i + 512] = ac->buf_mdct[1023 - i] * swindow[63 - i];
      } else if (ics->window_sequence[0] == LONG_START_SEQUENCE) {
          memcpy(saved_ltp,       ac->buf_mdct + 512, 448 * sizeof(float));
          memset(saved_ltp + 576, 0,                  448 * sizeof(float));
          ac->fdsp.vector_fmul_reverse(saved_ltp + 448, ac->buf_mdct + 960,     &swindow[64],      64);
          for (i = 0; i < 64; i++)
              saved_ltp[i + 512] = ac->buf_mdct[1023 - i] * swindow[63 - i];
      } else { // LONG_STOP or ONLY_LONG
          ac->fdsp.vector_fmul_reverse(saved_ltp,       ac->buf_mdct + 512,     &lwindow[512],     512);
          for (i = 0; i < 512; i++)
              saved_ltp[i + 512] = ac->buf_mdct[1023 - i] * lwindow[511 - i];
      }
  
      memcpy(sce->ltp_state,      sce->ltp_state+1024, 1024 * sizeof(*sce->ltp_state));
      memcpy(sce->ltp_state+1024, sce->ret,            1024 * sizeof(*sce->ltp_state));
      memcpy(sce->ltp_state+2048, saved_ltp,           1024 * sizeof(*sce->ltp_state));
  }
  
  /**
   * Conduct IMDCT and windowing.
   */
  static void imdct_and_windowing(AACContext *ac, SingleChannelElement *sce)
  {
      IndividualChannelStream *ics = &sce->ics;
      float *in    = sce->coeffs;
      float *out   = sce->ret;
      float *saved = sce->saved;
      const float *swindow      = ics->use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
      const float *lwindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;
      const float *swindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
      float *buf  = ac->buf_mdct;
      float *temp = ac->temp;
      int i;
  
      // imdct
      if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
          for (i = 0; i < 1024; i += 128)
              ac->mdct_small.imdct_half(&ac->mdct_small, buf + i, in + i);
      } else
          ac->mdct.imdct_half(&ac->mdct, buf, in);
  
      /* window overlapping
       * NOTE: To simplify the overlapping code, all 'meaningless' short to long
       * and long to short transitions are considered to be short to short
       * transitions. This leaves just two cases (long to long and short to short)
       * with a little special sauce for EIGHT_SHORT_SEQUENCE.
       */
      if ((ics->window_sequence[1] == ONLY_LONG_SEQUENCE || ics->window_sequence[1] == LONG_STOP_SEQUENCE) &&
              (ics->window_sequence[0] == ONLY_LONG_SEQUENCE || ics->window_sequence[0] == LONG_START_SEQUENCE)) {
          ac->fdsp.vector_fmul_window(    out,               saved,            buf,         lwindow_prev, 512);
      } else {
          memcpy(                         out,               saved,            448 * sizeof(float));
  
          if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
              ac->fdsp.vector_fmul_window(out + 448 + 0*128, saved + 448,      buf + 0*128, swindow_prev, 64);
              ac->fdsp.vector_fmul_window(out + 448 + 1*128, buf + 0*128 + 64, buf + 1*128, swindow,      64);
              ac->fdsp.vector_fmul_window(out + 448 + 2*128, buf + 1*128 + 64, buf + 2*128, swindow,      64);
              ac->fdsp.vector_fmul_window(out + 448 + 3*128, buf + 2*128 + 64, buf + 3*128, swindow,      64);
              ac->fdsp.vector_fmul_window(temp,              buf + 3*128 + 64, buf + 4*128, swindow,      64);
              memcpy(                     out + 448 + 4*128, temp, 64 * sizeof(float));
          } else {
              ac->fdsp.vector_fmul_window(out + 448,         saved + 448,      buf,         swindow_prev, 64);
              memcpy(                     out + 576,         buf + 64,         448 * sizeof(float));
          }
      }
  
      // buffer update
      if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
          memcpy(                     saved,       temp + 64,         64 * sizeof(float));
          ac->fdsp.vector_fmul_window(saved + 64,  buf + 4*128 + 64, buf + 5*128, swindow, 64);
          ac->fdsp.vector_fmul_window(saved + 192, buf + 5*128 + 64, buf + 6*128, swindow, 64);
          ac->fdsp.vector_fmul_window(saved + 320, buf + 6*128 + 64, buf + 7*128, swindow, 64);
          memcpy(                     saved + 448, buf + 7*128 + 64,  64 * sizeof(float));
      } else if (ics->window_sequence[0] == LONG_START_SEQUENCE) {
          memcpy(                     saved,       buf + 512,        448 * sizeof(float));
          memcpy(                     saved + 448, buf + 7*128 + 64,  64 * sizeof(float));
      } else { // LONG_STOP or ONLY_LONG
          memcpy(                     saved,       buf + 512,        512 * sizeof(float));
      }
  }
  
  static void imdct_and_windowing_ld(AACContext *ac, SingleChannelElement *sce)
  {
      IndividualChannelStream *ics = &sce->ics;
      float *in    = sce->coeffs;
      float *out   = sce->ret;
      float *saved = sce->saved;
      float *buf  = ac->buf_mdct;
  
      // imdct
      ac->mdct.imdct_half(&ac->mdct_ld, buf, in);
  
      // 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(float));
          ac->fdsp.vector_fmul_window(out + 192, saved + 192, buf, ff_sine_128, 64);
          memcpy(                     out + 320, buf + 64, 192 * sizeof(float));
      } else {
          ac->fdsp.vector_fmul_window(out, saved, buf, ff_sine_512, 256);
      }
  
      // buffer update
      memcpy(saved, buf + 256, 256 * sizeof(float));
  }
  
  static void imdct_and_windowing_eld(AACContext *ac, SingleChannelElement *sce)
  {
      float *in    = sce->coeffs;
      float *out   = sce->ret;
      float *saved = sce->saved;
      const float *const window = ff_aac_eld_window;
      float *buf  = ac->buf_mdct;
      int i;
      const int n  = 512;
      const int n2 = n >> 1;
      const int n4 = n >> 2;
  
      // 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) {
          float 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;
      }
      ac->mdct.imdct_half(&ac->mdct_ld, buf, in);
      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] =    buf[n2 - 1 - i]       * window[i       - n4] +
                         saved[      i + n2]     * window[i +   n - n4] +
                        -saved[  n + n2 - 1 - i] * window[i + 2*n - n4] +
                        -saved[2*n + n2 + i]     * window[i + 3*n - n4];
      }
      for (i = 0; i < n2; i ++) {
          out[n4 + i] =    buf[i]               * window[i + n2       - n4] +
                        -saved[      n - 1 - i] * window[i + n2 +   n - n4] +
                        -saved[  n + i]         * window[i + n2 + 2*n - n4] +
                         saved[2*n + n - 1 - i] * window[i + n2 + 3*n - n4];
      }
      for (i = 0; i < n4; i ++) {
          out[n2 + n4 + i] =    buf[      i + n2]     * window[i +   n - n4] +
                             -saved[      n2 - 1 - i] * window[i + 2*n - n4] +
                             -saved[  n + n2 + i]     * window[i + 3*n - n4];
      }
  
      // buffer update
      memmove(saved + n, saved, 2 * n * sizeof(float));
      memcpy( saved,       buf,     n * sizeof(float));
  }
  
  /**
   * Apply dependent channel coupling (applied before IMDCT).
   *
   * @param   index   index into coupling gain array
   */
  static void apply_dependent_coupling(AACContext *ac,
                                       SingleChannelElement *target,
                                       ChannelElement *cce, int index)
  {
      IndividualChannelStream *ics = &cce->ch[0].ics;
      const uint16_t *offsets = ics->swb_offset;
      float *dest = target->coeffs;
      const float *src = cce->ch[0].coeffs;
      int g, i, group, k, idx = 0;
      if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP) {
          av_log(ac->avctx, AV_LOG_ERROR,
                 "Dependent coupling is not supported together with LTP\n");
          return;
      }
      for (g = 0; g < ics->num_window_groups; g++) {
          for (i = 0; i < ics->max_sfb; i++, idx++) {
              if (cce->ch[0].band_type[idx] != ZERO_BT) {
                  const float gain = cce->coup.gain[index][idx];
                  for (group = 0; group < ics->group_len[g]; group++) {
                      for (k = offsets[i]; k < offsets[i + 1]; k++) {
                          // XXX dsputil-ize
                          dest[group * 128 + k] += gain * src[group * 128 + k];
                      }
                  }
              }
          }
          dest += ics->group_len[g] * 128;
          src  += ics->group_len[g] * 128;
      }
  }
  
  /**
   * Apply independent channel coupling (applied after IMDCT).
   *
   * @param   index   index into coupling gain array
   */
  static void apply_independent_coupling(AACContext *ac,
                                         SingleChannelElement *target,
                                         ChannelElement *cce, int index)
  {
      int i;
      const float gain = cce->coup.gain[index][0];
      const float *src = cce->ch[0].ret;
      float *dest = target->ret;
      const int len = 1024 << (ac->oc[1].m4ac.sbr == 1);
  
      for (i = 0; i < len; i++)
          dest[i] += gain * src[i];
  }
  
  /**
   * channel coupling transformation interface
   *
   * @param   apply_coupling_method   pointer to (in)dependent coupling function
   */
  static void apply_channel_coupling(AACContext *ac, ChannelElement *cc,
                                     enum RawDataBlockType type, int elem_id,
                                     enum CouplingPoint coupling_point,
                                     void (*apply_coupling_method)(AACContext *ac, SingleChannelElement *target, ChannelElement *cce, int index))
  {
      int i, c;
  
      for (i = 0; i < MAX_ELEM_ID; i++) {
          ChannelElement *cce = ac->che[TYPE_CCE][i];
          int index = 0;
  
          if (cce && cce->coup.coupling_point == coupling_point) {
              ChannelCoupling *coup = &cce->coup;
  
              for (c = 0; c <= coup->num_coupled; c++) {
                  if (coup->type[c] == type && coup->id_select[c] == elem_id) {
                      if (coup->ch_select[c] != 1) {
                          apply_coupling_method(ac, &cc->ch[0], cce, index);
                          if (coup->ch_select[c] != 0)
                              index++;
                      }
                      if (coup->ch_select[c] != 2)
                          apply_coupling_method(ac, &cc->ch[1], cce, index++);
                  } else
                      index += 1 + (coup->ch_select[c] == 3);
              }
          }
      }
  }
  
  /**
   * Convert spectral data to float samples, applying all supported tools as appropriate.
   */
  static void spectral_to_sample(AACContext *ac)
  {
      int i, type;
      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:
          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) {
                  if (type <= TYPE_CPE)
                      apply_channel_coupling(ac, che, type, i, BEFORE_TNS, apply_dependent_coupling);
                  if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP) {
                      if (che->ch[0].ics.predictor_present) {
                          if (che->ch[0].ics.ltp.present)
                              ac->apply_ltp(ac, &che->ch[0]);
                          if (che->ch[1].ics.ltp.present && type == TYPE_CPE)
                              ac->apply_ltp(ac, &che->ch[1]);
                      }
                  }
                  if (che->ch[0].tns.present)
                      ac->apply_tns(che->ch[0].coeffs, &che->ch[0].tns, &che->ch[0].ics, 1);
                  if (che->ch[1].tns.present)
                      ac->apply_tns(che->ch[1].coeffs, &che->ch[1].tns, &che->ch[1].ics, 1);
                  if (type <= TYPE_CPE)
                      apply_channel_coupling(ac, che, type, i, BETWEEN_TNS_AND_IMDCT, apply_dependent_coupling);
                  if (type != TYPE_CCE || che->coup.coupling_point == AFTER_IMDCT) {
                      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) {
                          ff_sbr_apply(ac, &che->sbr, type, che->ch[0].ret, che->ch[1].ret);
                      }
                  }
                  if (type <= TYPE_CCE)
                      apply_channel_coupling(ac, che, type, i, AFTER_IMDCT, apply_independent_coupling);
              }
          }
      }
  }
  
  static int parse_adts_frame_header(AACContext *ac, GetBitContext *gb)
  {
      int size;
      AACADTSHeaderInfo hdr_info;
      uint8_t layout_map[MAX_ELEM_ID*4][3];
      int layout_map_tags, ret;
  
      size = avpriv_aac_parse_header(gb, &hdr_info);
      if (size > 0) {
          if (!ac->warned_num_aac_frames && hdr_info.num_aac_frames != 1) {
              // This is 2 for "VLB " audio in NSV files.
              // See samples/nsv/vlb_audio.
              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;
          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;
      ChannelElement *che;
      int err, i;
      int samples = 1024;
      int chan_config = ac->oc[1].m4ac.chan_config;
      int aot = ac->oc[1].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 = ac->oc[1].m4ac.object_type - 1;
  
      ac->tags_mapped = 0;
  
      if (chan_config < 0 || chan_config >= 8) {
          avpriv_request_sample(avctx, "Unknown ER channel configuration %d",
                                ac->oc[1].m4ac.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;
          }
          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);
  
      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, elem_type_prev = TYPE_END;
      int err, elem_id;
      int samples = 0, multiplier, audio_found = 0, pce_found = 0;
      int is_dmono, sce_count = 0;
  
      ac->frame = data;
  
      if (show_bits(gb, 12) == 0xfff) {
          if ((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;
  
      ac->tags_mapped = 0;
      // parse
      while ((elem_type = get_bits(gb, 3)) != TYPE_END) {
          elem_id = get_bits(gb, 4);
  
          if (elem_type < TYPE_DSE) {
              if (!(che=get_che(ac, elem_type, elem_id))) {
                  av_log(ac->avctx, AV_LOG_ERROR, "channel element %d.%d is not allocated\n",
                         elem_type, elem_id);
                  err = AVERROR_INVALIDDATA;
                  goto fail;
              }
              samples = 1024;
          }
  
          switch (elem_type) {
  
          case TYPE_SCE:
              err = decode_ics(ac, &che->ch[0], gb, 0, 0);
              audio_found = 1;
              sce_count++;
              break;
  
          case TYPE_CPE:
              err = decode_cpe(ac, gb, che);
              audio_found = 1;
              break;
  
          case TYPE_CCE:
              err = decode_cce(ac, gb, che);
              break;
  
          case TYPE_LFE:
              err = decode_ics(ac, &che->ch[0], gb, 0, 0);
              audio_found = 1;
              break;
  
          case TYPE_DSE:
              err = skip_data_stream_element(ac, gb);
              break;
  
          case TYPE_PCE: {
              uint8_t layout_map[MAX_ELEM_ID*4][3];
              int tags;
              push_output_configuration(ac);
              tags = decode_pce(avctx, &ac->oc[1].m4ac, layout_map, gb);
              if (tags < 0) {
                  err = tags;
                  break;
              }
              if (pce_found) {
                  av_log(avctx, AV_LOG_ERROR,
                         "Not evaluating a further program_config_element as this construct is dubious at best.\n");
              } else {
                  err = output_configure(ac, layout_map, tags, OC_TRIAL_PCE, 1);
                  if (!err)
                      ac->oc[1].m4ac.chan_config = 0;
                  pce_found = 1;
              }
              break;
          }
  
          case TYPE_FIL:
              if (elem_id == 15)
                  elem_id += get_bits(gb, 8) - 1;
              if (get_bits_left(gb) < 8 * elem_id) {
                      av_log(avctx, AV_LOG_ERROR, "TYPE_FIL: "overread_err);
                      err = AVERROR_INVALIDDATA;
                      goto fail;
              }
              while (elem_id > 0)
                  elem_id -= decode_extension_payload(ac, gb, elem_id, che_prev, elem_type_prev);
              err = 0; /* FIXME */
              break;
  
          default:
              err = AVERROR_BUG; /* should not happen, but keeps compiler happy */
              break;
          }
  
          che_prev       = che;
          elem_type_prev = elem_type;
  
          if (err)
              goto fail;
  
          if (get_bits_left(gb) < 3) {
              av_log(avctx, AV_LOG_ERROR, overread_err);
              err = AVERROR_INVALIDDATA;
              goto fail;
          }
      }
  
      spectral_to_sample(ac);
  
      multiplier = (ac->oc[1].m4ac.sbr == 1) ? ac->oc[1].m4ac.ext_sample_rate > ac->oc[1].m4ac.sample_rate : 0;
      samples <<= multiplier;
  
      if (ac->oc[1].status && audio_found) {
          avctx->sample_rate = ac->oc[1].m4ac.sample_rate << multiplier;
          avctx->frame_size = samples;
          ac->oc[1].status = OC_LOCKED;
      }
  
      if (multiplier) {
          int side_size;
          const uint8_t *side = av_packet_get_side_data(avpkt, AV_PKT_DATA_SKIP_SAMPLES, &side_size);
          if (side && side_size>=4)
              AV_WL32(side, 2*AV_RL32(side));
      }
  
      *got_frame_ptr = !!samples;
      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 +
                                        FF_INPUT_BUFFER_PADDING_SIZE);
          if (!avctx->extradata)
              return AVERROR(ENOMEM);
          avctx->extradata_size = new_extradata_size;
          memcpy(avctx->extradata, new_extradata, new_extradata_size);
          push_output_configuration(ac);
          if (decode_audio_specific_config(ac, ac->avctx, &ac->oc[1].m4ac,
                                           avctx->extradata,
                                           avctx->extradata_size*8, 1) < 0) {
              pop_output_configuration(ac);
              return AVERROR_INVALIDDATA;
          }
      }
  
      ac->dmono_mode = 0;
      if (jp_dualmono && jp_dualmono_size > 0)
          ac->dmono_mode =  1 + *jp_dualmono;
      if (ac->force_dmono_mode >= 0)
          ac->dmono_mode = ac->force_dmono_mode;
  
      if (INT_MAX / 8 <= buf_size)
          return AVERROR_INVALIDDATA;
  
      if ((err = init_get_bits(&gb, buf, buf_size * 8)) < 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])
                  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);
      return 0;
  }
  
  
  #define LOAS_SYNC_WORD   0x2b7       ///< 11 bits LOAS sync word
  
  struct LATMContext {
      AACContext aac_ctx;     ///< containing AACContext
      int initialized;        ///< initialized after a valid extradata was seen
  
      // parser data
      int audio_mux_version_A; ///< LATM syntax version
      int frame_length_type;   ///< 0/1 variable/fixed frame length
      int frame_length;        ///< frame length for fixed frame length
  };
  
  static inline uint32_t latm_get_value(GetBitContext *b)
  {
      int length = get_bits(b, 2);
  
      return get_bits_long(b, (length+1)*8);
  }
  
  static int latm_decode_audio_specific_config(struct LATMContext *latmctx,
                                               GetBitContext *gb, int asclen)
  {
      AACContext *ac        = &latmctx->aac_ctx;
      AVCodecContext *avctx = ac->avctx;
      MPEG4AudioConfig m4ac = { 0 };
      int config_start_bit  = get_bits_count(gb);
      int sync_extension    = 0;
      int bits_consumed, esize;
  
      if (asclen) {
          sync_extension = 1;
          asclen         = FFMIN(asclen, get_bits_left(gb));
      } else
          asclen         = get_bits_left(gb);
  
      if (config_start_bit % 8) {
          avpriv_request_sample(latmctx->aac_ctx.avctx,
                                "Non-byte-aligned audio-specific config");
          return AVERROR_PATCHWELCOME;
      }
      if (asclen <= 0)
          return AVERROR_INVALIDDATA;
      bits_consumed = decode_audio_specific_config(NULL, avctx, &m4ac,
                                           gb->buffer + (config_start_bit / 8),
                                           asclen, sync_extension);
  
      if (bits_consumed < 0)
          return AVERROR_INVALIDDATA;
  
      if (!latmctx->initialized ||
          ac->oc[1].m4ac.sample_rate != m4ac.sample_rate ||
          ac->oc[1].m4ac.chan_config != m4ac.chan_config) {
  
          if(latmctx->initialized) {
              av_log(avctx, AV_LOG_INFO, "audio config changed\n");
          } else {
              av_log(avctx, AV_LOG_DEBUG, "initializing latmctx\n");
          }
          latmctx->initialized = 0;
  
          esize = (bits_consumed+7) / 8;
  
          if (avctx->extradata_size < esize) {
              av_free(avctx->extradata);
              avctx->extradata = av_malloc(esize + FF_INPUT_BUFFER_PADDING_SIZE);
              if (!avctx->extradata)
                  return AVERROR(ENOMEM);
          }
  
          avctx->extradata_size = esize;
          memcpy(avctx->extradata, gb->buffer + (config_start_bit/8), esize);
          memset(avctx->extradata+esize, 0, FF_INPUT_BUFFER_PADDING_SIZE);
      }
      skip_bits_long(gb, bits_consumed);
  
      return bits_consumed;
  }
  
  static int read_stream_mux_config(struct LATMContext *latmctx,
                                    GetBitContext *gb)
  {
      int ret, audio_mux_version = get_bits(gb, 1);
  
      latmctx->audio_mux_version_A = 0;
      if (audio_mux_version)
          latmctx->audio_mux_version_A = get_bits(gb, 1);
  
      if (!latmctx->audio_mux_version_A) {
  
          if (audio_mux_version)
              latm_get_value(gb);                 // taraFullness
  
          skip_bits(gb, 1);                       // allStreamSameTimeFraming
          skip_bits(gb, 6);                       // numSubFrames
          // numPrograms
          if (get_bits(gb, 4)) {                  // numPrograms
              avpriv_request_sample(latmctx->aac_ctx.avctx, "Multiple programs");
              return AVERROR_PATCHWELCOME;
          }
  
          // for each program (which there is only one in DVB)
  
          // for each layer (which there is only one in DVB)
          if (get_bits(gb, 3)) {                   // numLayer
              avpriv_request_sample(latmctx->aac_ctx.avctx, "Multiple layers");
              return AVERROR_PATCHWELCOME;
          }
  
          // for all but first stream: use_same_config = get_bits(gb, 1);
          if (!audio_mux_version) {
              if ((ret = latm_decode_audio_specific_config(latmctx, gb, 0)) < 0)
                  return ret;
          } else {
              int ascLen = latm_get_value(gb);
              if ((ret = latm_decode_audio_specific_config(latmctx, gb, ascLen)) < 0)
                  return ret;
              ascLen -= ret;
              skip_bits_long(gb, ascLen);
          }
  
          latmctx->frame_length_type = get_bits(gb, 3);
          switch (latmctx->frame_length_type) {
          case 0:
              skip_bits(gb, 8);       // latmBufferFullness
              break;
          case 1:
              latmctx->frame_length = get_bits(gb, 9);
              break;
          case 3:
          case 4:
          case 5:
              skip_bits(gb, 6);       // CELP frame length table index
              break;
          case 6:
          case 7:
              skip_bits(gb, 1);       // HVXC frame length table index
              break;
          }
  
          if (get_bits(gb, 1)) {                  // other data
              if (audio_mux_version) {
                  latm_get_value(gb);             // other_data_bits
              } else {
                  int esc;
                  do {
                      esc = get_bits(gb, 1);
                      skip_bits(gb, 8);
                  } while (esc);
              }
          }
  
          if (get_bits(gb, 1))                     // crc present
              skip_bits(gb, 8);                    // config_crc
      }
  
      return 0;
  }
  
  static int read_payload_length_info(struct LATMContext *ctx, GetBitContext *gb)
  {
      uint8_t tmp;
  
      if (ctx->frame_length_type == 0) {
          int mux_slot_length = 0;
          do {
              tmp = get_bits(gb, 8);
              mux_slot_length += tmp;
          } while (tmp == 255);
          return mux_slot_length;
      } else if (ctx->frame_length_type == 1) {
          return ctx->frame_length;
      } else if (ctx->frame_length_type == 3 ||
                 ctx->frame_length_type == 5 ||
                 ctx->frame_length_type == 7) {
          skip_bits(gb, 2);          // mux_slot_length_coded
      }
      return 0;
  }
  
  static int read_audio_mux_element(struct LATMContext *latmctx,
                                    GetBitContext *gb)
  {
      int err;
      uint8_t use_same_mux = get_bits(gb, 1);
      if (!use_same_mux) {
          if ((err = read_stream_mux_config(latmctx, gb)) < 0)
              return err;
      } else if (!latmctx->aac_ctx.avctx->extradata) {
          av_log(latmctx->aac_ctx.avctx, AV_LOG_DEBUG,
                 "no decoder config found\n");
          return AVERROR(EAGAIN);
      }
      if (latmctx->audio_mux_version_A == 0) {
          int mux_slot_length_bytes = read_payload_length_info(latmctx, gb);
          if (mux_slot_length_bytes * 8 > get_bits_left(gb)) {
              av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR, "incomplete frame\n");
              return AVERROR_INVALIDDATA;
          } else if (mux_slot_length_bytes * 8 + 256 < get_bits_left(gb)) {
              av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR,
                     "frame length mismatch %d << %d\n",
                     mux_slot_length_bytes * 8, get_bits_left(gb));
              return AVERROR_INVALIDDATA;
          }
      }
      return 0;
  }
  
  
  static int latm_decode_frame(AVCodecContext *avctx, void *out,
                               int *got_frame_ptr, AVPacket *avpkt)
  {
      struct LATMContext *latmctx = avctx->priv_data;
      int                 muxlength, err;
      GetBitContext       gb;
  
      if ((err = init_get_bits8(&gb, avpkt->data, avpkt->size)) < 0)
          return err;
  
      // check for LOAS sync word
      if (get_bits(&gb, 11) != LOAS_SYNC_WORD)
          return AVERROR_INVALIDDATA;
  
      muxlength = get_bits(&gb, 13) + 3;
      // not enough data, the parser should have sorted this out
      if (muxlength > avpkt->size)
          return AVERROR_INVALIDDATA;
  
      if ((err = read_audio_mux_element(latmctx, &gb)) < 0)
          return err;
  
      if (!latmctx->initialized) {
          if (!avctx->extradata) {
              *got_frame_ptr = 0;
              return avpkt->size;
          } else {
              push_output_configuration(&latmctx->aac_ctx);
              if ((err = decode_audio_specific_config(
                      &latmctx->aac_ctx, avctx, &latmctx->aac_ctx.oc[1].m4ac,
                      avctx->extradata, avctx->extradata_size*8, 1)) < 0) {
                  pop_output_configuration(&latmctx->aac_ctx);
                  return err;
              }
              latmctx->initialized = 1;
          }
      }
  
      if (show_bits(&gb, 12) == 0xfff) {
          av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR,
                 "ADTS header detected, probably as result of configuration "
                 "misparsing\n");
          return AVERROR_INVALIDDATA;
      }
  
      if ((err = aac_decode_frame_int(avctx, out, got_frame_ptr, &gb, avpkt)) < 0)
          return err;
  
      return muxlength;
  }
  
  static av_cold int latm_decode_init(AVCodecContext *avctx)
  {
      struct LATMContext *latmctx = avctx->priv_data;
      int ret = aac_decode_init(avctx);
  
      if (avctx->extradata_size > 0)
          latmctx->initialized = !ret;
  
      return ret;
  }
  
  static void aacdec_init(AACContext *c)
  {
      c->imdct_and_windowing                      = imdct_and_windowing;
      c->apply_ltp                                = apply_ltp;
      c->apply_tns                                = apply_tns;
      c->windowing_and_mdct_ltp                   = windowing_and_mdct_ltp;
      c->update_ltp                               = update_ltp;
  
      if(ARCH_MIPS)
          ff_aacdec_init_mips(c);
  }
  /**
   * AVOptions for Japanese DTV specific extensions (ADTS only)
   */
  #define AACDEC_FLAGS AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM
  static const AVOption options[] = {
      {"dual_mono_mode", "Select the channel to decode for dual mono",
       offsetof(AACContext, force_dmono_mode), AV_OPT_TYPE_INT, {.i64=-1}, -1, 2,
       AACDEC_FLAGS, "dual_mono_mode"},
  
      {"auto", "autoselection",            0, AV_OPT_TYPE_CONST, {.i64=-1}, INT_MIN, INT_MAX, AACDEC_FLAGS, "dual_mono_mode"},
      {"main", "Select Main/Left channel", 0, AV_OPT_TYPE_CONST, {.i64= 1}, INT_MIN, INT_MAX, AACDEC_FLAGS, "dual_mono_mode"},
      {"sub" , "Select Sub/Right channel", 0, AV_OPT_TYPE_CONST, {.i64= 2}, INT_MIN, INT_MAX, AACDEC_FLAGS, "dual_mono_mode"},
      {"both", "Select both channels",     0, AV_OPT_TYPE_CONST, {.i64= 0}, INT_MIN, INT_MAX, AACDEC_FLAGS, "dual_mono_mode"},
  
      {NULL},
  };
  
  static const AVClass aac_decoder_class = {
      .class_name = "AAC decoder",
      .item_name  = av_default_item_name,
      .option     = options,
      .version    = LIBAVUTIL_VERSION_INT,
  };
  
  AVCodec ff_aac_decoder = {
      .name            = "aac",
      .long_name       = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"),
      .type            = AVMEDIA_TYPE_AUDIO,
      .id              = AV_CODEC_ID_AAC,
      .priv_data_size  = sizeof(AACContext),
      .init            = aac_decode_init,
      .close           = aac_decode_close,
      .decode          = aac_decode_frame,
      .sample_fmts     = (const enum AVSampleFormat[]) {
          AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE
      },
      .capabilities    = CODEC_CAP_CHANNEL_CONF | CODEC_CAP_DR1,
      .channel_layouts = aac_channel_layout,
      .flush = flush,
      .priv_class      = &aac_decoder_class,
  };
  
  /*
      Note: This decoder filter is intended to decode LATM streams transferred
      in MPEG transport streams which only contain one program.
      To do a more complex LATM demuxing a separate LATM demuxer should be used.
  */
  AVCodec ff_aac_latm_decoder = {
      .name            = "aac_latm",
      .long_name       = NULL_IF_CONFIG_SMALL("AAC LATM (Advanced Audio Coding LATM syntax)"),
      .type            = AVMEDIA_TYPE_AUDIO,
      .id              = AV_CODEC_ID_AAC_LATM,
      .priv_data_size  = sizeof(struct LATMContext),
      .init            = latm_decode_init,
      .close           = aac_decode_close,
      .decode          = latm_decode_frame,
      .sample_fmts     = (const enum AVSampleFormat[]) {
          AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE
      },
      .capabilities    = CODEC_CAP_CHANNEL_CONF | CODEC_CAP_DR1,
      .channel_layouts = aac_channel_layout,
      .flush = flush,
  };
  