summaryrefslogtreecommitdiff
path: root/libavcodec/aac.h
blob: 5b66aefb13d92487f8d17efe150070774cc86e85 (plain)
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
222
223
224
225
226
227
228
229
230
231
232
233
234
235
236
237
238
239
240
241
242
243
244
245
246
247
248
249
250
251
252
253
254
255
256
257
258
259
260
261
262
263
264
265
266
267
268
269
270
271
272
273
274
275
276
277
278
279
280
281
282
283
284
285
286
287
288
289
290
291
292
293
294
/*
 * 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 "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
     * @{
     */
    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 */