libavcodec/aac.h

   1 /*
   2  * AAC definitions and structures
   3  * Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org )
   4  * Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com )
   5  *
   6  * This file is part of FFmpeg.
   7  *
   8  * FFmpeg is free software; you can redistribute it and/or
   9  * modify it under the terms of the GNU Lesser General Public
  10  * License as published by the Free Software Foundation; either
  11  * version 2.1 of the License, or (at your option) any later version.
  12  *
  13  * FFmpeg is distributed in the hope that it will be useful,
  14  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  15  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  16  * Lesser General Public License for more details.
  17  *
  18  * You should have received a copy of the GNU Lesser General Public
  19  * License along with FFmpeg; if not, write to the Free Software
  20  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  21  */
  22
  23 /**
  24  * @file aac.h
  25  * AAC definitions and structures
  26  * @author Oded Shimon  ( ods15 ods15 dyndns org )
  27  * @author Maxim Gavrilov ( maxim.gavrilov gmail com )
  28  */
  29
  30 #ifndef AVCODEC_AAC_H
  31 #define AVCODEC_AAC_H
  32
  33 #include "avcodec.h"
  34 #include "dsputil.h"
  35 #include "mpeg4audio.h"
  36
  37 #include <stdint.h>
  38
  39 #define AAC_INIT_VLC_STATIC(num, size) \
  40     INIT_VLC_STATIC(&vlc_spectral[num], 6, ff_aac_spectral_sizes[num], \
  41          ff_aac_spectral_bits[num], sizeof( ff_aac_spectral_bits[num][0]), sizeof( ff_aac_spectral_bits[num][0]), \
  42         ff_aac_spectral_codes[num], sizeof(ff_aac_spectral_codes[num][0]), sizeof(ff_aac_spectral_codes[num][0]), \
  43         size);
  44
  45 #define MAX_CHANNELS 64
  46 #define MAX_ELEM_ID 16
  47
  48 #define TNS_MAX_ORDER 20
  49 #define PNS_MEAN_ENERGY 3719550720.0f // sqrt(3.0) * 1<<31
  50
  51 enum AudioObjectType {
  52     AOT_NULL,
  53                                // Support?                Name
  54     AOT_AAC_MAIN,              ///< Y                       Main
  55     AOT_AAC_LC,                ///< Y                       Low Complexity
  56     AOT_AAC_SSR,               ///< N (code in SoC repo)    Scalable Sample Rate
  57     AOT_AAC_LTP,               ///< N (code in SoC repo)    Long Term Prediction
  58     AOT_SBR,                   ///< N (in progress)         Spectral Band Replication
  59     AOT_AAC_SCALABLE,          ///< N                       Scalable
  60     AOT_TWINVQ,                ///< N                       Twin Vector Quantizer
  61     AOT_CELP,                  ///< N                       Code Excited Linear Prediction
  62     AOT_HVXC,                  ///< N                       Harmonic Vector eXcitation Coding
  63     AOT_TTSI             = 12, ///< N                       Text-To-Speech Interface
  64     AOT_MAINSYNTH,             ///< N                       Main Synthesis
  65     AOT_WAVESYNTH,             ///< N                       Wavetable Synthesis
  66     AOT_MIDI,                  ///< N                       General MIDI
  67     AOT_SAFX,                  ///< N                       Algorithmic Synthesis and Audio Effects
  68     AOT_ER_AAC_LC,             ///< N                       Error Resilient Low Complexity
  69     AOT_ER_AAC_LTP       = 19, ///< N                       Error Resilient Long Term Prediction
  70     AOT_ER_AAC_SCALABLE,       ///< N                       Error Resilient Scalable
  71     AOT_ER_TWINVQ,             ///< N                       Error Resilient Twin Vector Quantizer
  72     AOT_ER_BSAC,               ///< N                       Error Resilient Bit-Sliced Arithmetic Coding
  73     AOT_ER_AAC_LD,             ///< N                       Error Resilient Low Delay
  74     AOT_ER_CELP,               ///< N                       Error Resilient Code Excited Linear Prediction
  75     AOT_ER_HVXC,               ///< N                       Error Resilient Harmonic Vector eXcitation Coding
  76     AOT_ER_HILN,               ///< N                       Error Resilient Harmonic and Individual Lines plus Noise
  77     AOT_ER_PARAM,              ///< N                       Error Resilient Parametric
  78     AOT_SSC,                   ///< N                       SinuSoidal Coding
  79 };
  80
  81 enum RawDataBlockType {
  82     TYPE_SCE,
  83     TYPE_CPE,
  84     TYPE_CCE,
  85     TYPE_LFE,
  86     TYPE_DSE,
  87     TYPE_PCE,
  88     TYPE_FIL,
  89     TYPE_END,
  90 };
  91
  92 enum ExtensionPayloadID {
  93     EXT_FILL,
  94     EXT_FILL_DATA,
  95     EXT_DATA_ELEMENT,
  96     EXT_DYNAMIC_RANGE = 0xb,
  97     EXT_SBR_DATA      = 0xd,
  98     EXT_SBR_DATA_CRC  = 0xe,
  99 };
 100
 101 enum WindowSequence {
 102     ONLY_LONG_SEQUENCE,
 103     LONG_START_SEQUENCE,
 104     EIGHT_SHORT_SEQUENCE,
 105     LONG_STOP_SEQUENCE,
 106 };
 107
 108 enum BandType {
 109     ZERO_BT        = 0,     ///< Scalefactors and spectral data are all zero.
 110     FIRST_PAIR_BT  = 5,     ///< This and later band types encode two values (rather than four) with one code word.
 111     ESC_BT         = 11,    ///< Spectral data are coded with an escape sequence.
 112     NOISE_BT       = 13,    ///< Spectral data are scaled white noise not coded in the bitstream.
 113     INTENSITY_BT2  = 14,    ///< Scalefactor data are intensity stereo positions.
 114     INTENSITY_BT   = 15,    ///< Scalefactor data are intensity stereo positions.
 115 };
 116
 117 #define IS_CODEBOOK_UNSIGNED(x) ((x - 1) & 10)
 118
 119 enum ChannelPosition {
 120     AAC_CHANNEL_FRONT = 1,
 121     AAC_CHANNEL_SIDE  = 2,
 122     AAC_CHANNEL_BACK  = 3,
 123     AAC_CHANNEL_LFE   = 4,
 124     AAC_CHANNEL_CC    = 5,
 125 };
 126
 127 /**
 128  * The point during decoding at which channel coupling is applied.
 129  */
 130 enum CouplingPoint {
 131     BEFORE_TNS,
 132     BETWEEN_TNS_AND_IMDCT,
 133     AFTER_IMDCT = 3,
 134 };
 135
 136 /**
 137  * Individual Channel Stream
 138  */
 139 typedef struct {
 140     uint8_t max_sfb;            ///< number of scalefactor bands per group
 141     enum WindowSequence window_sequence[2];
 142     uint8_t use_kb_window[2];   ///< If set, use Kaiser-Bessel window, otherwise use a sinus window.
 143     int num_window_groups;
 144     uint8_t group_len[8];
 145     const uint16_t *swb_offset; ///< table of offsets to the lowest spectral coefficient of a scalefactor band, sfb, for a particular window
 146     int num_swb;                ///< number of scalefactor window bands
 147     int num_windows;
 148     int tns_max_bands;
 149 } IndividualChannelStream;
 150
 151 /**
 152  * Temporal Noise Shaping
 153  */
 154 typedef struct {
 155     int present;
 156     int n_filt[8];
 157     int length[8][4];
 158     int direction[8][4];
 159     int order[8][4];
 160     float coef[8][4][TNS_MAX_ORDER];
 161 } TemporalNoiseShaping;
 162
 163 /**
 164  * Dynamic Range Control - decoded from the bitstream but not processed further.
 165  */
 166 typedef struct {
 167     int pce_instance_tag;                           ///< Indicates with which program the DRC info is associated.
 168     int dyn_rng_sgn[17];                            ///< DRC sign information; 0 - positive, 1 - negative
 169     int dyn_rng_ctl[17];                            ///< DRC magnitude information
 170     int exclude_mask[MAX_CHANNELS];                 ///< Channels to be excluded from DRC processing.
 171     int band_incr;                                  ///< Number of DRC bands greater than 1 having DRC info.
 172     int interpolation_scheme;                       ///< Indicates the interpolation scheme used in the SBR QMF domain.
 173     int band_top[17];                               ///< Indicates the top of the i-th DRC band in units of 4 spectral lines.
 174     int prog_ref_level;                             /**< A reference level for the long-term program audio level for all
 175                                                      *   channels combined.
 176                                                      */
 177 } DynamicRangeControl;
 178
 179 typedef struct {
 180     int num_pulse;
 181     int pos[4];
 182     int amp[4];
 183 } Pulse;
 184
 185 /**
 186  * coupling parameters
 187  */
 188 typedef struct {
 189     enum CouplingPoint coupling_point;  ///< The point during decoding at which coupling is applied.
 190     int num_coupled;       ///< number of target elements
 191     enum RawDataBlockType type[8];   ///< Type of channel element to be coupled - SCE or CPE.
 192     int id_select[8];      ///< element id
 193     int ch_select[8];      /**< [0] shared list of gains; [1] list of gains for left channel;
 194                             *   [2] list of gains for right channel; [3] lists of gains for both channels
 195                             */
 196     float gain[16][120];
 197 } ChannelCoupling;
 198
 199 /**
 200  * Single Channel Element - used for both SCE and LFE elements.
 201  */
 202 typedef struct {
 203     IndividualChannelStream ics;
 204     TemporalNoiseShaping tns;
 205     enum BandType band_type[120];             ///< band types
 206     int band_type_run_end[120];               ///< band type run end points
 207     float sf[120];                            ///< scalefactors
 208     DECLARE_ALIGNED_16(float, coeffs[1024]);  ///< coefficients for IMDCT
 209     DECLARE_ALIGNED_16(float, saved[512]);    ///< overlap
 210     DECLARE_ALIGNED_16(float, ret[1024]);     ///< PCM output
 211 } SingleChannelElement;
 212
 213 /**
 214  * channel element - generic struct for SCE/CPE/CCE/LFE
 215  */
 216 typedef struct {
 217     // CPE specific
 218     uint8_t ms_mask[120];     ///< Set if mid/side stereo is used for each scalefactor window band
 219     // shared
 220     SingleChannelElement ch[2];
 221     // CCE specific
 222     ChannelCoupling coup;
 223 } ChannelElement;
 224
 225 /**
 226  * main AAC context
 227  */
 228 typedef struct {
 229     AVCodecContext * avccontext;
 230
 231     MPEG4AudioConfig m4ac;
 232
 233     int is_saved;                 ///< Set if elements have stored overlap from previous frame.
 234     DynamicRangeControl che_drc;
 235
 236     /**
 237      * @defgroup elements
 238      * @{
 239      */
 240     enum ChannelPosition che_pos[4][MAX_ELEM_ID]; /**< channel element channel mapping with the
 241                                                    *   first index as the first 4 raw data block types
 242                                                    */
 243     ChannelElement * che[4][MAX_ELEM_ID];
 244     /** @} */
 245
 246     /**
 247      * @defgroup temporary aligned temporary buffers (We do not want to have these on the stack.)
 248      * @{
 249      */
 250     DECLARE_ALIGNED_16(float, buf_mdct[1024]);
 251     /** @} */
 252
 253     /**
 254      * @defgroup tables   Computed / set up during initialization.
 255      * @{
 256      */
 257     MDCTContext mdct;
 258     MDCTContext mdct_small;
 259     DSPContext dsp;
 260     int random_state;
 261     /** @} */
 262
 263     /**
 264      * @defgroup output   Members used for output interleaving.
 265      * @{
 266      */
 267     float *output_data[MAX_CHANNELS];                 ///< Points to each element's 'ret' buffer (PCM output).
 268     float add_bias;                                   ///< offset for dsp.float_to_int16
 269     float sf_scale;                                   ///< Pre-scale for correct IMDCT and dsp.float_to_int16.
 270     int sf_offset;                                    ///< offset into pow2sf_tab as appropriate for dsp.float_to_int16
 271     /** @} */
 272
 273 } AACContext;
 274
 275 #endif /* AVCODEC_AAC_H */