Strong filtering function for future RV40 loop filter
[libav.git] / libavcodec / aac.h
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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
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30#ifndef AVCODEC_AAC_H
31#define AVCODEC_AAC_H
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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
cc0591da 45#define MAX_CHANNELS 64
9cc04edf 46#define MAX_ELEM_ID 16
cc0591da 47
9ffd5c1c 48#define TNS_MAX_ORDER 20
9ffd5c1c 49
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50enum AudioObjectType {
51 AOT_NULL,
52 // Support? Name
53 AOT_AAC_MAIN, ///< Y Main
54 AOT_AAC_LC, ///< Y Low Complexity
55 AOT_AAC_SSR, ///< N (code in SoC repo) Scalable Sample Rate
56 AOT_AAC_LTP, ///< N (code in SoC repo) Long Term Prediction
57 AOT_SBR, ///< N (in progress) Spectral Band Replication
58 AOT_AAC_SCALABLE, ///< N Scalable
59 AOT_TWINVQ, ///< N Twin Vector Quantizer
60 AOT_CELP, ///< N Code Excited Linear Prediction
61 AOT_HVXC, ///< N Harmonic Vector eXcitation Coding
62 AOT_TTSI = 12, ///< N Text-To-Speech Interface
63 AOT_MAINSYNTH, ///< N Main Synthesis
64 AOT_WAVESYNTH, ///< N Wavetable Synthesis
65 AOT_MIDI, ///< N General MIDI
66 AOT_SAFX, ///< N Algorithmic Synthesis and Audio Effects
67 AOT_ER_AAC_LC, ///< N Error Resilient Low Complexity
68 AOT_ER_AAC_LTP = 19, ///< N Error Resilient Long Term Prediction
69 AOT_ER_AAC_SCALABLE, ///< N Error Resilient Scalable
70 AOT_ER_TWINVQ, ///< N Error Resilient Twin Vector Quantizer
71 AOT_ER_BSAC, ///< N Error Resilient Bit-Sliced Arithmetic Coding
72 AOT_ER_AAC_LD, ///< N Error Resilient Low Delay
73 AOT_ER_CELP, ///< N Error Resilient Code Excited Linear Prediction
74 AOT_ER_HVXC, ///< N Error Resilient Harmonic Vector eXcitation Coding
75 AOT_ER_HILN, ///< N Error Resilient Harmonic and Individual Lines plus Noise
76 AOT_ER_PARAM, ///< N Error Resilient Parametric
77 AOT_SSC, ///< N SinuSoidal Coding
78};
79
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80enum RawDataBlockType {
81 TYPE_SCE,
82 TYPE_CPE,
83 TYPE_CCE,
84 TYPE_LFE,
85 TYPE_DSE,
86 TYPE_PCE,
87 TYPE_FIL,
88 TYPE_END,
89};
90
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91enum ExtensionPayloadID {
92 EXT_FILL,
93 EXT_FILL_DATA,
94 EXT_DATA_ELEMENT,
95 EXT_DYNAMIC_RANGE = 0xb,
96 EXT_SBR_DATA = 0xd,
97 EXT_SBR_DATA_CRC = 0xe,
98};
99
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100enum WindowSequence {
101 ONLY_LONG_SEQUENCE,
102 LONG_START_SEQUENCE,
103 EIGHT_SHORT_SEQUENCE,
104 LONG_STOP_SEQUENCE,
105};
106
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107enum BandType {
108 ZERO_BT = 0, ///< Scalefactors and spectral data are all zero.
109 FIRST_PAIR_BT = 5, ///< This and later band types encode two values (rather than four) with one code word.
110 ESC_BT = 11, ///< Spectral data are coded with an escape sequence.
111 NOISE_BT = 13, ///< Spectral data are scaled white noise not coded in the bitstream.
112 INTENSITY_BT2 = 14, ///< Scalefactor data are intensity stereo positions.
113 INTENSITY_BT = 15, ///< Scalefactor data are intensity stereo positions.
114};
115
116#define IS_CODEBOOK_UNSIGNED(x) ((x - 1) & 10)
117
118enum ChannelPosition {
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119 AAC_CHANNEL_FRONT = 1,
120 AAC_CHANNEL_SIDE = 2,
121 AAC_CHANNEL_BACK = 3,
122 AAC_CHANNEL_LFE = 4,
123 AAC_CHANNEL_CC = 5,
124};
125
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126/**
127 * The point during decoding at which channel coupling is applied.
128 */
129enum CouplingPoint {
130 BEFORE_TNS,
131 BETWEEN_TNS_AND_IMDCT,
132 AFTER_IMDCT = 3,
133};
134
135/**
136 * Individual Channel Stream
137 */
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138typedef struct {
139 uint8_t max_sfb; ///< number of scalefactor bands per group
140 enum WindowSequence window_sequence[2];
141 uint8_t use_kb_window[2]; ///< If set, use Kaiser-Bessel window, otherwise use a sinus window.
142 int num_window_groups;
143 uint8_t group_len[8];
144 const uint16_t *swb_offset; ///< table of offsets to the lowest spectral coefficient of a scalefactor band, sfb, for a particular window
145 int num_swb; ///< number of scalefactor window bands
146 int num_windows;
147 int tns_max_bands;
148} IndividualChannelStream;
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149
150/**
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151 * Temporal Noise Shaping
152 */
153typedef struct {
154 int present;
155 int n_filt[8];
156 int length[8][4];
157 int direction[8][4];
158 int order[8][4];
159 float coef[8][4][TNS_MAX_ORDER];
160} TemporalNoiseShaping;
161
162/**
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163 * Dynamic Range Control - decoded from the bitstream but not processed further.
164 */
165typedef struct {
166 int pce_instance_tag; ///< Indicates with which program the DRC info is associated.
167 int dyn_rng_sgn[17]; ///< DRC sign information; 0 - positive, 1 - negative
168 int dyn_rng_ctl[17]; ///< DRC magnitude information
169 int exclude_mask[MAX_CHANNELS]; ///< Channels to be excluded from DRC processing.
170 int band_incr; ///< Number of DRC bands greater than 1 having DRC info.
171 int interpolation_scheme; ///< Indicates the interpolation scheme used in the SBR QMF domain.
172 int band_top[17]; ///< Indicates the top of the i-th DRC band in units of 4 spectral lines.
173 int prog_ref_level; /**< A reference level for the long-term program audio level for all
174 * channels combined.
175 */
176} DynamicRangeControl;
177
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178typedef struct {
179 int num_pulse;
848a5815 180 int pos[4];
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181 int amp[4];
182} Pulse;
183
184/**
185 * coupling parameters
186 */
187typedef struct {
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188 enum CouplingPoint coupling_point; ///< The point during decoding at which coupling is applied.
189 int num_coupled; ///< number of target elements
190 enum RawDataBlockType type[8]; ///< Type of channel element to be coupled - SCE or CPE.
191 int id_select[8]; ///< element id
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192 int ch_select[8]; /**< [0] shared list of gains; [1] list of gains for right channel;
193 * [2] list of gains for left channel; [3] lists of gains for both channels
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194 */
195 float gain[16][120];
196} ChannelCoupling;
197
198/**
199 * Single Channel Element - used for both SCE and LFE elements.
200 */
201typedef struct {
202 IndividualChannelStream ics;
203 TemporalNoiseShaping tns;
204 enum BandType band_type[120]; ///< band types
205 int band_type_run_end[120]; ///< band type run end points
206 float sf[120]; ///< scalefactors
207 DECLARE_ALIGNED_16(float, coeffs[1024]); ///< coefficients for IMDCT
b0f5852a 208 DECLARE_ALIGNED_16(float, saved[512]); ///< overlap
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209 DECLARE_ALIGNED_16(float, ret[1024]); ///< PCM output
210} SingleChannelElement;
211
212/**
213 * channel element - generic struct for SCE/CPE/CCE/LFE
214 */
215typedef struct {
216 // CPE specific
217 uint8_t ms_mask[120]; ///< Set if mid/side stereo is used for each scalefactor window band
218 // shared
219 SingleChannelElement ch[2];
220 // CCE specific
221 ChannelCoupling coup;
222} ChannelElement;
cc0591da 223
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224/**
225 * main AAC context
226 */
227typedef struct {
228 AVCodecContext * avccontext;
229
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230 MPEG4AudioConfig m4ac;
231
232 int is_saved; ///< Set if elements have stored overlap from previous frame.
233 DynamicRangeControl che_drc;
234
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235 /**
236 * @defgroup elements
237 * @{
238 */
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239 enum ChannelPosition che_pos[4][MAX_ELEM_ID]; /**< channel element channel mapping with the
240 * first index as the first 4 raw data block types
241 */
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242 ChannelElement * che[4][MAX_ELEM_ID];
243 /** @} */
cc0591da 244
589ce6e6 245 /**
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246 * @defgroup temporary aligned temporary buffers (We do not want to have these on the stack.)
247 * @{
248 */
b0f5852a 249 DECLARE_ALIGNED_16(float, buf_mdct[1024]);
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250 /** @} */
251
252 /**
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253 * @defgroup tables Computed / set up during initialization.
254 * @{
255 */
256 MDCTContext mdct;
257 MDCTContext mdct_small;
258 DSPContext dsp;
9cc04edf 259 int random_state;
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260 /** @} */
261
262 /**
cc0591da 263 * @defgroup output Members used for output interleaving.
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264 * @{
265 */
cc0591da 266 float *output_data[MAX_CHANNELS]; ///< Points to each element's 'ret' buffer (PCM output).
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267 float add_bias; ///< offset for dsp.float_to_int16
268 float sf_scale; ///< Pre-scale for correct IMDCT and dsp.float_to_int16.
269 int sf_offset; ///< offset into pow2sf_tab as appropriate for dsp.float_to_int16
270 /** @} */
271
272} AACContext;
273
98790382 274#endif /* AVCODEC_AAC_H */