Mercurial > libavcodec.hg
annotate aac.h @ 8990:cc8f95accbff libavcodec
Fixing a value returning issue
author | romansh |
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date | Fri, 20 Feb 2009 02:00:44 +0000 |
parents | 9396eb8808ca |
children | 955835e2bc99 |
rev | line source |
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1 /* |
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2 * AAC definitions and structures |
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3 * Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org ) |
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4 * Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com ) |
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5 * |
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6 * This file is part of FFmpeg. |
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7 * |
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8 * FFmpeg is free software; you can redistribute it and/or |
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9 * modify it under the terms of the GNU Lesser General Public |
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10 * License as published by the Free Software Foundation; either |
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11 * version 2.1 of the License, or (at your option) any later version. |
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12 * |
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13 * FFmpeg is distributed in the hope that it will be useful, |
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14 * but WITHOUT ANY WARRANTY; without even the implied warranty of |
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15 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
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16 * Lesser General Public License for more details. |
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17 * |
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18 * You should have received a copy of the GNU Lesser General Public |
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19 * License along with FFmpeg; if not, write to the Free Software |
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20 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
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21 */ |
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22 |
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23 /** |
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24 * @file libavcodec/aac.h |
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25 * AAC definitions and structures |
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26 * @author Oded Shimon ( ods15 ods15 dyndns org ) |
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27 * @author Maxim Gavrilov ( maxim.gavrilov gmail com ) |
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28 */ |
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29 |
7760 | 30 #ifndef AVCODEC_AAC_H |
31 #define AVCODEC_AAC_H | |
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32 |
8957 | 33 #include "libavutil/internal.h" |
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34 #include "avcodec.h" |
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35 #include "dsputil.h" |
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36 #include "mpeg4audio.h" |
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37 |
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38 #include <stdint.h> |
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39 |
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40 #define AAC_INIT_VLC_STATIC(num, size) \ |
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41 INIT_VLC_STATIC(&vlc_spectral[num], 6, ff_aac_spectral_sizes[num], \ |
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42 ff_aac_spectral_bits[num], sizeof( ff_aac_spectral_bits[num][0]), sizeof( ff_aac_spectral_bits[num][0]), \ |
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43 ff_aac_spectral_codes[num], sizeof(ff_aac_spectral_codes[num][0]), sizeof(ff_aac_spectral_codes[num][0]), \ |
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44 size); |
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45 |
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46 #define MAX_CHANNELS 64 |
7539 | 47 #define MAX_ELEM_ID 16 |
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48 |
7581 | 49 #define TNS_MAX_ORDER 20 |
50 | |
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51 enum AudioObjectType { |
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52 AOT_NULL, |
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53 // Support? Name |
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54 AOT_AAC_MAIN, ///< Y Main |
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55 AOT_AAC_LC, ///< Y Low Complexity |
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56 AOT_AAC_SSR, ///< N (code in SoC repo) Scalable Sample Rate |
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57 AOT_AAC_LTP, ///< N (code in SoC repo) Long Term Prediction |
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58 AOT_SBR, ///< N (in progress) Spectral Band Replication |
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59 AOT_AAC_SCALABLE, ///< N Scalable |
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60 AOT_TWINVQ, ///< N Twin Vector Quantizer |
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61 AOT_CELP, ///< N Code Excited Linear Prediction |
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62 AOT_HVXC, ///< N Harmonic Vector eXcitation Coding |
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63 AOT_TTSI = 12, ///< N Text-To-Speech Interface |
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64 AOT_MAINSYNTH, ///< N Main Synthesis |
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65 AOT_WAVESYNTH, ///< N Wavetable Synthesis |
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66 AOT_MIDI, ///< N General MIDI |
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67 AOT_SAFX, ///< N Algorithmic Synthesis and Audio Effects |
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68 AOT_ER_AAC_LC, ///< N Error Resilient Low Complexity |
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69 AOT_ER_AAC_LTP = 19, ///< N Error Resilient Long Term Prediction |
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70 AOT_ER_AAC_SCALABLE, ///< N Error Resilient Scalable |
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71 AOT_ER_TWINVQ, ///< N Error Resilient Twin Vector Quantizer |
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72 AOT_ER_BSAC, ///< N Error Resilient Bit-Sliced Arithmetic Coding |
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73 AOT_ER_AAC_LD, ///< N Error Resilient Low Delay |
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74 AOT_ER_CELP, ///< N Error Resilient Code Excited Linear Prediction |
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75 AOT_ER_HVXC, ///< N Error Resilient Harmonic Vector eXcitation Coding |
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76 AOT_ER_HILN, ///< N Error Resilient Harmonic and Individual Lines plus Noise |
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77 AOT_ER_PARAM, ///< N Error Resilient Parametric |
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78 AOT_SSC, ///< N SinuSoidal Coding |
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79 }; |
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80 |
7539 | 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 | |
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92 enum ExtensionPayloadID { |
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93 EXT_FILL, |
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94 EXT_FILL_DATA, |
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95 EXT_DATA_ELEMENT, |
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96 EXT_DYNAMIC_RANGE = 0xb, |
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97 EXT_SBR_DATA = 0xd, |
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98 EXT_SBR_DATA_CRC = 0xe, |
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99 }; |
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100 |
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101 enum WindowSequence { |
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102 ONLY_LONG_SEQUENCE, |
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103 LONG_START_SEQUENCE, |
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104 EIGHT_SHORT_SEQUENCE, |
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105 LONG_STOP_SEQUENCE, |
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106 }; |
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107 |
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108 enum BandType { |
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109 ZERO_BT = 0, ///< Scalefactors and spectral data are all zero. |
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110 FIRST_PAIR_BT = 5, ///< This and later band types encode two values (rather than four) with one code word. |
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111 ESC_BT = 11, ///< Spectral data are coded with an escape sequence. |
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112 NOISE_BT = 13, ///< Spectral data are scaled white noise not coded in the bitstream. |
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113 INTENSITY_BT2 = 14, ///< Scalefactor data are intensity stereo positions. |
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114 INTENSITY_BT = 15, ///< Scalefactor data are intensity stereo positions. |
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115 }; |
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116 |
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117 #define IS_CODEBOOK_UNSIGNED(x) ((x - 1) & 10) |
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118 |
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119 enum ChannelPosition { |
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120 AAC_CHANNEL_FRONT = 1, |
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121 AAC_CHANNEL_SIDE = 2, |
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122 AAC_CHANNEL_BACK = 3, |
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123 AAC_CHANNEL_LFE = 4, |
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124 AAC_CHANNEL_CC = 5, |
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125 }; |
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126 |
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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 /** | |
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137 * Predictor State |
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138 */ |
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139 typedef struct { |
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140 float cor0; |
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141 float cor1; |
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142 float var0; |
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143 float var1; |
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144 float r0; |
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145 float r1; |
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146 } PredictorState; |
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147 |
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148 #define MAX_PREDICTORS 672 |
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149 |
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150 /** |
7539 | 151 * Individual Channel Stream |
152 */ | |
7578 | 153 typedef struct { |
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 | |
161 int num_windows; | |
162 int tns_max_bands; | |
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163 int predictor_present; |
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164 int predictor_initialized; |
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165 int predictor_reset_group; |
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166 uint8_t prediction_used[41]; |
7578 | 167 } IndividualChannelStream; |
7539 | 168 |
169 /** | |
7609 | 170 * Temporal Noise Shaping |
171 */ | |
172 typedef struct { | |
173 int present; | |
174 int n_filt[8]; | |
175 int length[8][4]; | |
176 int direction[8][4]; | |
177 int order[8][4]; | |
178 float coef[8][4][TNS_MAX_ORDER]; | |
179 } TemporalNoiseShaping; | |
180 | |
181 /** | |
7539 | 182 * Dynamic Range Control - decoded from the bitstream but not processed further. |
183 */ | |
184 typedef struct { | |
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 | |
193 * channels combined. | |
194 */ | |
195 } DynamicRangeControl; | |
196 | |
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197 typedef struct { |
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198 int num_pulse; |
7580 | 199 int pos[4]; |
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200 int amp[4]; |
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201 } Pulse; |
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202 |
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203 /** |
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204 * coupling parameters |
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205 */ |
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206 typedef struct { |
7578 | 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 | |
8005 | 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 | |
7578 | 213 */ |
214 float gain[16][120]; | |
215 } ChannelCoupling; | |
216 | |
217 /** | |
218 * Single Channel Element - used for both SCE and LFE elements. | |
219 */ | |
220 typedef struct { | |
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 | |
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227 DECLARE_ALIGNED_16(float, saved[512]); ///< overlap |
7578 | 228 DECLARE_ALIGNED_16(float, ret[1024]); ///< PCM output |
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229 PredictorState predictor_state[MAX_PREDICTORS]; |
7578 | 230 } SingleChannelElement; |
231 | |
232 /** | |
233 * channel element - generic struct for SCE/CPE/CCE/LFE | |
234 */ | |
235 typedef struct { | |
236 // CPE specific | |
237 uint8_t ms_mask[120]; ///< Set if mid/side stereo is used for each scalefactor window band | |
238 // shared | |
239 SingleChannelElement ch[2]; | |
240 // CCE specific | |
241 ChannelCoupling coup; | |
242 } ChannelElement; | |
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243 |
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244 /** |
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245 * main AAC context |
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246 */ |
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247 typedef struct { |
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248 AVCodecContext * avccontext; |
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249 |
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250 MPEG4AudioConfig m4ac; |
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251 |
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252 int is_saved; ///< Set if elements have stored overlap from previous frame. |
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253 DynamicRangeControl che_drc; |
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254 |
7539 | 255 /** |
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256 * @defgroup elements Channel element related data. |
7539 | 257 * @{ |
258 */ | |
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259 enum ChannelPosition che_pos[4][MAX_ELEM_ID]; /**< channel element channel mapping with the |
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260 * first index as the first 4 raw data block types |
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261 */ |
7539 | 262 ChannelElement * che[4][MAX_ELEM_ID]; |
263 /** @} */ | |
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264 |
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265 /** |
7608 | 266 * @defgroup temporary aligned temporary buffers (We do not want to have these on the stack.) |
267 * @{ | |
268 */ | |
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269 DECLARE_ALIGNED_16(float, buf_mdct[1024]); |
7608 | 270 /** @} */ |
271 | |
272 /** | |
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273 * @defgroup tables Computed / set up during initialization. |
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274 * @{ |
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275 */ |
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276 MDCTContext mdct; |
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277 MDCTContext mdct_small; |
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278 DSPContext dsp; |
7539 | 279 int random_state; |
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280 /** @} */ |
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281 |
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282 /** |
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283 * @defgroup output Members used for output interleaving. |
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284 * @{ |
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285 */ |
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286 float *output_data[MAX_CHANNELS]; ///< Points to each element's 'ret' buffer (PCM output). |
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287 float add_bias; ///< offset for dsp.float_to_int16 |
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288 float sf_scale; ///< Pre-scale for correct IMDCT and dsp.float_to_int16. |
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289 int sf_offset; ///< offset into pow2sf_tab as appropriate for dsp.float_to_int16 |
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290 /** @} */ |
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291 |
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292 DECLARE_ALIGNED(16, float, temp[128]); |
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293 } AACContext; |
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294 |
7760 | 295 #endif /* AVCODEC_AAC_H */ |