Mercurial > libavcodec.hg
annotate aac.h @ 8161:bc4ddc399192 libavcodec
Revert previous commit: the libavcodec/i386/fft_*.o objects depend on
libavcodec/i386/fft_mmx.o, which depends on yasm.
author | diego |
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date | Mon, 17 Nov 2008 21:56:34 +0000 |
parents | 7baa25290231 |
children | 8a8d40fe77d9 |
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 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 |
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33 #include "avcodec.h" |
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34 #include "dsputil.h" |
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35 #include "mpeg4audio.h" |
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36 |
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37 #include <stdint.h> |
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38 |
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39 #define AAC_INIT_VLC_STATIC(num, size) \ |
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40 INIT_VLC_STATIC(&vlc_spectral[num], 6, ff_aac_spectral_sizes[num], \ |
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41 ff_aac_spectral_bits[num], sizeof( ff_aac_spectral_bits[num][0]), sizeof( ff_aac_spectral_bits[num][0]), \ |
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42 ff_aac_spectral_codes[num], sizeof(ff_aac_spectral_codes[num][0]), sizeof(ff_aac_spectral_codes[num][0]), \ |
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43 size); |
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44 |
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45 #define MAX_CHANNELS 64 |
7539 | 46 #define MAX_ELEM_ID 16 |
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47 |
7581 | 48 #define TNS_MAX_ORDER 20 |
49 | |
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50 enum AudioObjectType { |
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51 AOT_NULL, |
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52 // Support? Name |
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53 AOT_AAC_MAIN, ///< Y Main |
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54 AOT_AAC_LC, ///< Y Low Complexity |
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55 AOT_AAC_SSR, ///< N (code in SoC repo) Scalable Sample Rate |
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56 AOT_AAC_LTP, ///< N (code in SoC repo) Long Term Prediction |
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57 AOT_SBR, ///< N (in progress) Spectral Band Replication |
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58 AOT_AAC_SCALABLE, ///< N Scalable |
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59 AOT_TWINVQ, ///< N Twin Vector Quantizer |
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60 AOT_CELP, ///< N Code Excited Linear Prediction |
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61 AOT_HVXC, ///< N Harmonic Vector eXcitation Coding |
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62 AOT_TTSI = 12, ///< N Text-To-Speech Interface |
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63 AOT_MAINSYNTH, ///< N Main Synthesis |
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64 AOT_WAVESYNTH, ///< N Wavetable Synthesis |
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65 AOT_MIDI, ///< N General MIDI |
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66 AOT_SAFX, ///< N Algorithmic Synthesis and Audio Effects |
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67 AOT_ER_AAC_LC, ///< N Error Resilient Low Complexity |
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68 AOT_ER_AAC_LTP = 19, ///< N Error Resilient Long Term Prediction |
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69 AOT_ER_AAC_SCALABLE, ///< N Error Resilient Scalable |
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70 AOT_ER_TWINVQ, ///< N Error Resilient Twin Vector Quantizer |
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71 AOT_ER_BSAC, ///< N Error Resilient Bit-Sliced Arithmetic Coding |
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72 AOT_ER_AAC_LD, ///< N Error Resilient Low Delay |
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73 AOT_ER_CELP, ///< N Error Resilient Code Excited Linear Prediction |
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74 AOT_ER_HVXC, ///< N Error Resilient Harmonic Vector eXcitation Coding |
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75 AOT_ER_HILN, ///< N Error Resilient Harmonic and Individual Lines plus Noise |
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76 AOT_ER_PARAM, ///< N Error Resilient Parametric |
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77 AOT_SSC, ///< N SinuSoidal Coding |
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78 }; |
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79 |
7539 | 80 enum 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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91 enum ExtensionPayloadID { |
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92 EXT_FILL, |
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93 EXT_FILL_DATA, |
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94 EXT_DATA_ELEMENT, |
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95 EXT_DYNAMIC_RANGE = 0xb, |
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96 EXT_SBR_DATA = 0xd, |
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97 EXT_SBR_DATA_CRC = 0xe, |
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98 }; |
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99 |
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100 enum WindowSequence { |
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101 ONLY_LONG_SEQUENCE, |
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102 LONG_START_SEQUENCE, |
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103 EIGHT_SHORT_SEQUENCE, |
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104 LONG_STOP_SEQUENCE, |
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105 }; |
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106 |
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107 enum BandType { |
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108 ZERO_BT = 0, ///< Scalefactors and spectral data are all zero. |
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109 FIRST_PAIR_BT = 5, ///< This and later band types encode two values (rather than four) with one code word. |
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110 ESC_BT = 11, ///< Spectral data are coded with an escape sequence. |
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111 NOISE_BT = 13, ///< Spectral data are scaled white noise not coded in the bitstream. |
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112 INTENSITY_BT2 = 14, ///< Scalefactor data are intensity stereo positions. |
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113 INTENSITY_BT = 15, ///< Scalefactor data are intensity stereo positions. |
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114 }; |
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115 |
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116 #define IS_CODEBOOK_UNSIGNED(x) ((x - 1) & 10) |
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117 |
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118 enum ChannelPosition { |
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119 AAC_CHANNEL_FRONT = 1, |
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120 AAC_CHANNEL_SIDE = 2, |
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121 AAC_CHANNEL_BACK = 3, |
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122 AAC_CHANNEL_LFE = 4, |
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123 AAC_CHANNEL_CC = 5, |
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124 }; |
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125 |
7539 | 126 /** |
127 * The point during decoding at which channel coupling is applied. | |
128 */ | |
129 enum CouplingPoint { | |
130 BEFORE_TNS, | |
131 BETWEEN_TNS_AND_IMDCT, | |
132 AFTER_IMDCT = 3, | |
133 }; | |
134 | |
135 /** | |
136 * Individual Channel Stream | |
137 */ | |
7578 | 138 typedef 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; | |
7539 | 149 |
150 /** | |
7609 | 151 * Temporal Noise Shaping |
152 */ | |
153 typedef 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 /** | |
7539 | 163 * Dynamic Range Control - decoded from the bitstream but not processed further. |
164 */ | |
165 typedef 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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178 typedef struct { |
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179 int num_pulse; |
7580 | 180 int pos[4]; |
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181 int amp[4]; |
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182 } Pulse; |
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183 |
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184 /** |
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185 * coupling parameters |
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186 */ |
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187 typedef struct { |
7578 | 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 | |
8005 | 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 | |
7578 | 194 */ |
195 float gain[16][120]; | |
196 } ChannelCoupling; | |
197 | |
198 /** | |
199 * Single Channel Element - used for both SCE and LFE elements. | |
200 */ | |
201 typedef 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 | |
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208 DECLARE_ALIGNED_16(float, saved[512]); ///< overlap |
7578 | 209 DECLARE_ALIGNED_16(float, ret[1024]); ///< PCM output |
210 } SingleChannelElement; | |
211 | |
212 /** | |
213 * channel element - generic struct for SCE/CPE/CCE/LFE | |
214 */ | |
215 typedef 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; | |
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223 |
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224 /** |
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225 * main AAC context |
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226 */ |
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227 typedef struct { |
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228 AVCodecContext * avccontext; |
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229 |
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230 MPEG4AudioConfig m4ac; |
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231 |
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232 int is_saved; ///< Set if elements have stored overlap from previous frame. |
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233 DynamicRangeControl che_drc; |
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234 |
7539 | 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 |
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240 * first index as the first 4 raw data block types |
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241 */ |
7539 | 242 ChannelElement * che[4][MAX_ELEM_ID]; |
243 /** @} */ | |
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244 |
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245 /** |
7608 | 246 * @defgroup temporary aligned temporary buffers (We do not want to have these on the stack.) |
247 * @{ | |
248 */ | |
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249 DECLARE_ALIGNED_16(float, buf_mdct[1024]); |
7608 | 250 /** @} */ |
251 | |
252 /** | |
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253 * @defgroup tables Computed / set up during initialization. |
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254 * @{ |
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255 */ |
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256 MDCTContext mdct; |
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257 MDCTContext mdct_small; |
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258 DSPContext dsp; |
7539 | 259 int random_state; |
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260 /** @} */ |
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261 |
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262 /** |
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263 * @defgroup output Members used for output interleaving. |
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264 * @{ |
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265 */ |
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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 |
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268 float sf_scale; ///< Pre-scale for correct IMDCT and dsp.float_to_int16. |
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269 int sf_offset; ///< offset into pow2sf_tab as appropriate for dsp.float_to_int16 |
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270 /** @} */ |
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271 |
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272 } AACContext; |
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273 |
7760 | 274 #endif /* AVCODEC_AAC_H */ |