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 )
6 * This file is part of FFmpeg.
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.
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.
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
24 * @file libavcodec/aac.h
25 * AAC definitions and structures
26 * @author Oded Shimon ( ods15 ods15 dyndns org )
27 * @author Maxim Gavrilov ( maxim.gavrilov gmail com )
33 #include "libavutil/internal.h"
36 #include "mpeg4audio.h"
40 #define AAC_INIT_VLC_STATIC(num, size) \
41 INIT_VLC_STATIC(&vlc_spectral[num], 6, ff_aac_spectral_sizes[num], \
42 ff_aac_spectral_bits[num], sizeof( ff_aac_spectral_bits[num][0]), sizeof( ff_aac_spectral_bits[num][0]), \
43 ff_aac_spectral_codes[num], sizeof(ff_aac_spectral_codes[num][0]), sizeof(ff_aac_spectral_codes[num][0]), \
46 #define MAX_CHANNELS 64
47 #define MAX_ELEM_ID 16
49 #define TNS_MAX_ORDER 20
51 enum AudioObjectType {
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
81 enum RawDataBlockType {
92 enum ExtensionPayloadID {
96 EXT_DYNAMIC_RANGE = 0xb,
98 EXT_SBR_DATA_CRC = 0xe,
101 enum WindowSequence {
104 EIGHT_SHORT_SEQUENCE,
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.
117 #define IS_CODEBOOK_UNSIGNED(x) ((x - 1) & 10)
119 enum ChannelPosition {
120 AAC_CHANNEL_FRONT = 1,
121 AAC_CHANNEL_SIDE = 2,
122 AAC_CHANNEL_BACK = 3,
128 * The point during decoding at which channel coupling is applied.
132 BETWEEN_TNS_AND_IMDCT,
148 #define MAX_PREDICTORS 672
151 * Individual Channel Stream
154 uint8_t max_sfb; ///< number of scalefactor bands per group
155 enum WindowSequence window_sequence[2];
156 uint8_t use_kb_window[2]; ///< If set, use Kaiser-Bessel window, otherwise use a sinus window.
157 int num_window_groups;
158 uint8_t group_len[8];
159 const uint16_t *swb_offset; ///< table of offsets to the lowest spectral coefficient of a scalefactor band, sfb, for a particular window
160 int num_swb; ///< number of scalefactor window bands
163 int predictor_present;
164 int predictor_initialized;
165 int predictor_reset_group;
166 uint8_t prediction_used[41];
167 } IndividualChannelStream;
170 * Temporal Noise Shaping
178 float coef[8][4][TNS_MAX_ORDER];
179 } TemporalNoiseShaping;
182 * Dynamic Range Control - decoded from the bitstream but not processed further.
185 int pce_instance_tag; ///< Indicates with which program the DRC info is associated.
186 int dyn_rng_sgn[17]; ///< DRC sign information; 0 - positive, 1 - negative
187 int dyn_rng_ctl[17]; ///< DRC magnitude information
188 int exclude_mask[MAX_CHANNELS]; ///< Channels to be excluded from DRC processing.
189 int band_incr; ///< Number of DRC bands greater than 1 having DRC info.
190 int interpolation_scheme; ///< Indicates the interpolation scheme used in the SBR QMF domain.
191 int band_top[17]; ///< Indicates the top of the i-th DRC band in units of 4 spectral lines.
192 int prog_ref_level; /**< A reference level for the long-term program audio level for all
195 } DynamicRangeControl;
204 * coupling parameters
207 enum CouplingPoint coupling_point; ///< The point during decoding at which coupling is applied.
208 int num_coupled; ///< number of target elements
209 enum RawDataBlockType type[8]; ///< Type of channel element to be coupled - SCE or CPE.
210 int id_select[8]; ///< element id
211 int ch_select[8]; /**< [0] shared list of gains; [1] list of gains for right channel;
212 * [2] list of gains for left channel; [3] lists of gains for both channels
218 * Single Channel Element - used for both SCE and LFE elements.
221 IndividualChannelStream ics;
222 TemporalNoiseShaping tns;
223 enum BandType band_type[120]; ///< band types
224 int band_type_run_end[120]; ///< band type run end points
225 float sf[120]; ///< scalefactors
226 DECLARE_ALIGNED_16(float, coeffs[1024]); ///< coefficients for IMDCT
227 DECLARE_ALIGNED_16(float, saved[512]); ///< overlap
228 DECLARE_ALIGNED_16(float, ret[1024]); ///< PCM output
229 PredictorState predictor_state[MAX_PREDICTORS];
230 } SingleChannelElement;
233 * channel element - generic struct for SCE/CPE/CCE/LFE
237 uint8_t ms_mask[120]; ///< Set if mid/side stereo is used for each scalefactor window band
239 SingleChannelElement ch[2];
241 ChannelCoupling coup;
248 AVCodecContext * avccontext;
250 MPEG4AudioConfig m4ac;
252 int is_saved; ///< Set if elements have stored overlap from previous frame.
253 DynamicRangeControl che_drc;
256 * @defgroup elements Channel element related data.
259 enum ChannelPosition che_pos[4][MAX_ELEM_ID]; /**< channel element channel mapping with the
260 * first index as the first 4 raw data block types
262 ChannelElement * che[4][MAX_ELEM_ID];
263 ChannelElement * tag_che_map[4][MAX_ELEM_ID];
268 * @defgroup temporary aligned temporary buffers (We do not want to have these on the stack.)
271 DECLARE_ALIGNED_16(float, buf_mdct[1024]);
275 * @defgroup tables Computed / set up during initialization.
279 MDCTContext mdct_small;
285 * @defgroup output Members used for output interleaving.
288 float *output_data[MAX_CHANNELS]; ///< Points to each element's 'ret' buffer (PCM output).
289 float add_bias; ///< offset for dsp.float_to_int16
290 float sf_scale; ///< Pre-scale for correct IMDCT and dsp.float_to_int16.
291 int sf_offset; ///< offset into pow2sf_tab as appropriate for dsp.float_to_int16
294 DECLARE_ALIGNED(16, float, temp[128]);
297 #endif /* AVCODEC_AAC_H */