text: /*
   * AAC definitions and structures
   * Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org )
   * Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com )
   *
   * This file is part of FFmpeg.
   *
   * FFmpeg is free software; you can redistribute it and/or
   * modify it under the terms of the GNU Lesser General Public
   * License as published by the Free Software Foundation; either
   * version 2.1 of the License, or (at your option) any later version.
   *
   * FFmpeg is distributed in the hope that it will be useful,
   * but WITHOUT ANY WARRANTY; without even the implied warranty of
   * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
   * Lesser General Public License for more details.
   *
   * You should have received a copy of the GNU Lesser General Public
   * License along with FFmpeg; if not, write to the Free Software
   * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
   */
  
  /**
   * @file libavcodec/aac.h
   * AAC definitions and structures
   * @author Oded Shimon  ( ods15 ods15 dyndns org )
   * @author Maxim Gavrilov ( maxim.gavrilov gmail com )
   */
  
  #ifndef AVCODEC_AAC_H
  #define AVCODEC_AAC_H
  
  #include "libavutil/internal.h"
  #include "avcodec.h"
  #include "dsputil.h"
  #include "mpeg4audio.h"
  
  #include <stdint.h>
  
  #define AAC_INIT_VLC_STATIC(num, size) \
      INIT_VLC_STATIC(&vlc_spectral[num], 6, ff_aac_spectral_sizes[num], \
           ff_aac_spectral_bits[num], sizeof( ff_aac_spectral_bits[num][0]), sizeof( ff_aac_spectral_bits[num][0]), \
          ff_aac_spectral_codes[num], sizeof(ff_aac_spectral_codes[num][0]), sizeof(ff_aac_spectral_codes[num][0]), \
          size);
  
  #define MAX_CHANNELS 64
  #define MAX_ELEM_ID 16
  
  #define TNS_MAX_ORDER 20
  
  enum AudioObjectType {
      AOT_NULL,
                                 // Support?                Name
      AOT_AAC_MAIN,              ///< Y                       Main
      AOT_AAC_LC,                ///< Y                       Low Complexity
      AOT_AAC_SSR,               ///< N (code in SoC repo)    Scalable Sample Rate
      AOT_AAC_LTP,               ///< N (code in SoC repo)    Long Term Prediction
      AOT_SBR,                   ///< N (in progress)         Spectral Band Replication
      AOT_AAC_SCALABLE,          ///< N                       Scalable
      AOT_TWINVQ,                ///< N                       Twin Vector Quantizer
      AOT_CELP,                  ///< N                       Code Excited Linear Prediction
      AOT_HVXC,                  ///< N                       Harmonic Vector eXcitation Coding
      AOT_TTSI             = 12, ///< N                       Text-To-Speech Interface
      AOT_MAINSYNTH,             ///< N                       Main Synthesis
      AOT_WAVESYNTH,             ///< N                       Wavetable Synthesis
      AOT_MIDI,                  ///< N                       General MIDI
      AOT_SAFX,                  ///< N                       Algorithmic Synthesis and Audio Effects
      AOT_ER_AAC_LC,             ///< N                       Error Resilient Low Complexity
      AOT_ER_AAC_LTP       = 19, ///< N                       Error Resilient Long Term Prediction
      AOT_ER_AAC_SCALABLE,       ///< N                       Error Resilient Scalable
      AOT_ER_TWINVQ,             ///< N                       Error Resilient Twin Vector Quantizer
      AOT_ER_BSAC,               ///< N                       Error Resilient Bit-Sliced Arithmetic Coding
      AOT_ER_AAC_LD,             ///< N                       Error Resilient Low Delay
      AOT_ER_CELP,               ///< N                       Error Resilient Code Excited Linear Prediction
      AOT_ER_HVXC,               ///< N                       Error Resilient Harmonic Vector eXcitation Coding
      AOT_ER_HILN,               ///< N                       Error Resilient Harmonic and Individual Lines plus Noise
      AOT_ER_PARAM,              ///< N                       Error Resilient Parametric
      AOT_SSC,                   ///< N                       SinuSoidal Coding
  };
  
  enum RawDataBlockType {
      TYPE_SCE,
      TYPE_CPE,
      TYPE_CCE,
      TYPE_LFE,
      TYPE_DSE,
      TYPE_PCE,
      TYPE_FIL,
      TYPE_END,
  };
  
  enum ExtensionPayloadID {
      EXT_FILL,
      EXT_FILL_DATA,
      EXT_DATA_ELEMENT,
      EXT_DYNAMIC_RANGE = 0xb,
      EXT_SBR_DATA      = 0xd,
      EXT_SBR_DATA_CRC  = 0xe,
  };
  
  enum WindowSequence {
      ONLY_LONG_SEQUENCE,
      LONG_START_SEQUENCE,
      EIGHT_SHORT_SEQUENCE,
      LONG_STOP_SEQUENCE,
  };
  
  enum BandType {
      ZERO_BT        = 0,     ///< Scalefactors and spectral data are all zero.
      FIRST_PAIR_BT  = 5,     ///< This and later band types encode two values (rather than four) with one code word.
      ESC_BT         = 11,    ///< Spectral data are coded with an escape sequence.
      NOISE_BT       = 13,    ///< Spectral data are scaled white noise not coded in the bitstream.
      INTENSITY_BT2  = 14,    ///< Scalefactor data are intensity stereo positions.
      INTENSITY_BT   = 15,    ///< Scalefactor data are intensity stereo positions.
  };
  
  #define IS_CODEBOOK_UNSIGNED(x) ((x - 1) & 10)
  
  enum ChannelPosition {
      AAC_CHANNEL_FRONT = 1,
      AAC_CHANNEL_SIDE  = 2,
      AAC_CHANNEL_BACK  = 3,
      AAC_CHANNEL_LFE   = 4,
      AAC_CHANNEL_CC    = 5,
  };
  
  /**
   * The point during decoding at which channel coupling is applied.
   */
  enum CouplingPoint {
      BEFORE_TNS,
      BETWEEN_TNS_AND_IMDCT,
      AFTER_IMDCT = 3,
  };
  
  /**
   * Predictor State
   */
  typedef struct {
      float cor0;
      float cor1;
      float var0;
      float var1;
      float r0;
      float r1;
  } PredictorState;
  
  #define MAX_PREDICTORS 672
  
  /**
   * Individual Channel Stream
   */
  typedef struct {
      uint8_t max_sfb;            ///< number of scalefactor bands per group
      enum WindowSequence window_sequence[2];
      uint8_t use_kb_window[2];   ///< If set, use Kaiser-Bessel window, otherwise use a sinus window.
      int num_window_groups;
      uint8_t group_len[8];
      const uint16_t *swb_offset; ///< table of offsets to the lowest spectral coefficient of a scalefactor band, sfb, for a particular window
      int num_swb;                ///< number of scalefactor window bands
      int num_windows;
      int tns_max_bands;
      int predictor_present;
      int predictor_initialized;
      int predictor_reset_group;
      uint8_t prediction_used[41];
  } IndividualChannelStream;
  
  /**
   * Temporal Noise Shaping
   */
  typedef struct {
      int present;
      int n_filt[8];
      int length[8][4];
      int direction[8][4];
      int order[8][4];
      float coef[8][4][TNS_MAX_ORDER];
  } TemporalNoiseShaping;
  
  /**
   * Dynamic Range Control - decoded from the bitstream but not processed further.
   */
  typedef struct {
      int pce_instance_tag;                           ///< Indicates with which program the DRC info is associated.
      int dyn_rng_sgn[17];                            ///< DRC sign information; 0 - positive, 1 - negative
      int dyn_rng_ctl[17];                            ///< DRC magnitude information
      int exclude_mask[MAX_CHANNELS];                 ///< Channels to be excluded from DRC processing.
      int band_incr;                                  ///< Number of DRC bands greater than 1 having DRC info.
      int interpolation_scheme;                       ///< Indicates the interpolation scheme used in the SBR QMF domain.
      int band_top[17];                               ///< Indicates the top of the i-th DRC band in units of 4 spectral lines.
      int prog_ref_level;                             /**< A reference level for the long-term program audio level for all
                                                       *   channels combined.
                                                       */
  } DynamicRangeControl;
  
  typedef struct {
      int num_pulse;
      int pos[4];
      int amp[4];
  } Pulse;
  
  /**
   * coupling parameters
   */
  typedef struct {
      enum CouplingPoint coupling_point;  ///< The point during decoding at which coupling is applied.
      int num_coupled;       ///< number of target elements
      enum RawDataBlockType type[8];   ///< Type of channel element to be coupled - SCE or CPE.
      int id_select[8];      ///< element id
      int ch_select[8];      /**< [0] shared list of gains; [1] list of gains for right channel;
                              *   [2] list of gains for left channel; [3] lists of gains for both channels
                              */
      float gain[16][120];
  } ChannelCoupling;
  
  /**
   * Single Channel Element - used for both SCE and LFE elements.
   */
  typedef struct {
      IndividualChannelStream ics;
      TemporalNoiseShaping tns;
      enum BandType band_type[120];             ///< band types
      int band_type_run_end[120];               ///< band type run end points
      float sf[120];                            ///< scalefactors
      DECLARE_ALIGNED_16(float, coeffs[1024]);  ///< coefficients for IMDCT
      DECLARE_ALIGNED_16(float, saved[512]);    ///< overlap
      DECLARE_ALIGNED_16(float, ret[1024]);     ///< PCM output
      PredictorState predictor_state[MAX_PREDICTORS];
  } SingleChannelElement;
  
  /**
   * channel element - generic struct for SCE/CPE/CCE/LFE
   */
  typedef struct {
      // CPE specific
      uint8_t ms_mask[120];     ///< Set if mid/side stereo is used for each scalefactor window band
      // shared
      SingleChannelElement ch[2];
      // CCE specific
      ChannelCoupling coup;
  } ChannelElement;
  
  /**
   * main AAC context
   */
  typedef struct {
      AVCodecContext * avccontext;
  
      MPEG4AudioConfig m4ac;
  
      int is_saved;                 ///< Set if elements have stored overlap from previous frame.
      DynamicRangeControl che_drc;
  
      /**
       * @defgroup elements Channel element related data.
       * @{
       */
      enum ChannelPosition che_pos[4][MAX_ELEM_ID]; /**< channel element channel mapping with the
                                                     *   first index as the first 4 raw data block types
                                                     */
      ChannelElement * che[4][MAX_ELEM_ID];
      /** @} */
  
      /**
       * @defgroup temporary aligned temporary buffers (We do not want to have these on the stack.)
       * @{
       */
      DECLARE_ALIGNED_16(float, buf_mdct[1024]);
      /** @} */
  
      /**
       * @defgroup tables   Computed / set up during initialization.
       * @{
       */
      MDCTContext mdct;
      MDCTContext mdct_small;
      DSPContext dsp;
      int random_state;
      /** @} */
  
      /**
       * @defgroup output   Members used for output interleaving.
       * @{
       */
      float *output_data[MAX_CHANNELS];                 ///< Points to each element's 'ret' buffer (PCM output).
      float add_bias;                                   ///< offset for dsp.float_to_int16
      float sf_scale;                                   ///< Pre-scale for correct IMDCT and dsp.float_to_int16.
      int sf_offset;                                    ///< offset into pow2sf_tab as appropriate for dsp.float_to_int16
      /** @} */
  
      DECLARE_ALIGNED(16, float, temp[128]);
  } AACContext;
  
  #endif /* AVCODEC_AAC_H */
  