1 /* ----------------------------------------------------------------------------- 2 Software License for The Fraunhofer FDK AAC Codec Library for Android 3 4 © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten 5 Forschung e.V. All rights reserved. 6 7 1. INTRODUCTION 8 The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software 9 that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding 10 scheme for digital audio. This FDK AAC Codec software is intended to be used on 11 a wide variety of Android devices. 12 13 AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient 14 general perceptual audio codecs. AAC-ELD is considered the best-performing 15 full-bandwidth communications codec by independent studies and is widely 16 deployed. AAC has been standardized by ISO and IEC as part of the MPEG 17 specifications. 18 19 Patent licenses for necessary patent claims for the FDK AAC Codec (including 20 those of Fraunhofer) may be obtained through Via Licensing 21 (www.vialicensing.com) or through the respective patent owners individually for 22 the purpose of encoding or decoding bit streams in products that are compliant 23 with the ISO/IEC MPEG audio standards. Please note that most manufacturers of 24 Android devices already license these patent claims through Via Licensing or 25 directly from the patent owners, and therefore FDK AAC Codec software may 26 already be covered under those patent licenses when it is used for those 27 licensed purposes only. 28 29 Commercially-licensed AAC software libraries, including floating-point versions 30 with enhanced sound quality, are also available from Fraunhofer. Users are 31 encouraged to check the Fraunhofer website for additional applications 32 information and documentation. 33 34 2. COPYRIGHT LICENSE 35 36 Redistribution and use in source and binary forms, with or without modification, 37 are permitted without payment of copyright license fees provided that you 38 satisfy the following conditions: 39 40 You must retain the complete text of this software license in redistributions of 41 the FDK AAC Codec or your modifications thereto in source code form. 42 43 You must retain the complete text of this software license in the documentation 44 and/or other materials provided with redistributions of the FDK AAC Codec or 45 your modifications thereto in binary form. You must make available free of 46 charge copies of the complete source code of the FDK AAC Codec and your 47 modifications thereto to recipients of copies in binary form. 48 49 The name of Fraunhofer may not be used to endorse or promote products derived 50 from this library without prior written permission. 51 52 You may not charge copyright license fees for anyone to use, copy or distribute 53 the FDK AAC Codec software or your modifications thereto. 54 55 Your modified versions of the FDK AAC Codec must carry prominent notices stating 56 that you changed the software and the date of any change. For modified versions 57 of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" 58 must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK 59 AAC Codec Library for Android." 60 61 3. NO PATENT LICENSE 62 63 NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without 64 limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. 65 Fraunhofer provides no warranty of patent non-infringement with respect to this 66 software. 67 68 You may use this FDK AAC Codec software or modifications thereto only for 69 purposes that are authorized by appropriate patent licenses. 70 71 4. DISCLAIMER 72 73 This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright 74 holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, 75 including but not limited to the implied warranties of merchantability and 76 fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR 77 CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, 78 or consequential damages, including but not limited to procurement of substitute 79 goods or services; loss of use, data, or profits, or business interruption, 80 however caused and on any theory of liability, whether in contract, strict 81 liability, or tort (including negligence), arising in any way out of the use of 82 this software, even if advised of the possibility of such damage. 83 84 5. CONTACT INFORMATION 85 86 Fraunhofer Institute for Integrated Circuits IIS 87 Attention: Audio and Multimedia Departments - FDK AAC LL 88 Am Wolfsmantel 33 89 91058 Erlangen, Germany 90 91 www.iis.fraunhofer.de/amm 92 amm-info@iis.fraunhofer.de 93 ----------------------------------------------------------------------------- */ 94 95 /**************************** SBR decoder library ****************************** 96 97 Author(s): 98 99 Description: 100 101 *******************************************************************************/ 102 103 /*! 104 \file 105 \brief Low Power Profile Transposer 106 */ 107 108 #ifndef LPP_TRAN_H 109 #define LPP_TRAN_H 110 111 #include "sbrdecoder.h" 112 #include "hbe.h" 113 #include "qmf.h" 114 115 /* 116 Common 117 */ 118 #define QMF_OUT_SCALE 8 119 120 /* 121 Frequency scales 122 */ 123 124 /* 125 Env-Adjust 126 */ 127 #define MAX_NOISE_ENVELOPES 2 128 #define MAX_NOISE_COEFFS 5 129 #define MAX_NUM_NOISE_VALUES (MAX_NOISE_ENVELOPES * MAX_NOISE_COEFFS) 130 #define MAX_NUM_LIMITERS 12 131 132 /* Set MAX_ENVELOPES to the largest value of all supported BSFORMATs 133 by overriding MAX_ENVELOPES in the correct order: */ 134 #define MAX_ENVELOPES_LEGACY 5 135 #define MAX_ENVELOPES_USAC 8 136 #define MAX_ENVELOPES MAX_ENVELOPES_USAC 137 138 #define MAX_FREQ_COEFFS_DUAL_RATE 48 139 #define MAX_FREQ_COEFFS_QUAD_RATE 56 140 #define MAX_FREQ_COEFFS MAX_FREQ_COEFFS_QUAD_RATE 141 142 #define MAX_FREQ_COEFFS_FS44100 35 143 #define MAX_FREQ_COEFFS_FS48000 32 144 145 #define MAX_NUM_ENVELOPE_VALUES (MAX_ENVELOPES * MAX_FREQ_COEFFS) 146 147 #define MAX_GAIN_EXP 34 148 /* Maximum gain will be sqrt(0.5 * 2^MAX_GAIN_EXP) 149 example: 34=99dB */ 150 #define MAX_GAIN_CONCEAL_EXP 1 151 /* Maximum gain will be sqrt(0.5 * 2^MAX_GAIN_CONCEAL_EXP) in concealment case 152 * (0dB) */ 153 154 /* 155 LPP Transposer 156 */ 157 #define LPC_ORDER 2 158 159 #define MAX_INVF_BANDS MAX_NOISE_COEFFS 160 161 #define MAX_NUM_PATCHES 6 162 #define SHIFT_START_SB 1 /*!< lowest subband of source range */ 163 164 typedef enum { 165 INVF_OFF = 0, 166 INVF_LOW_LEVEL, 167 INVF_MID_LEVEL, 168 INVF_HIGH_LEVEL, 169 INVF_SWITCHED /* not a real choice but used here to control behaviour */ 170 } INVF_MODE; 171 172 /** parameter set for one single patch */ 173 typedef struct { 174 UCHAR sourceStartBand; /*!< first band in lowbands where to take the samples 175 from */ 176 UCHAR 177 sourceStopBand; /*!< first band in lowbands which is not included in the 178 patch anymore */ 179 UCHAR guardStartBand; /*!< first band in highbands to be filled with zeros in 180 order to reduce interferences between patches */ 181 UCHAR 182 targetStartBand; /*!< first band in highbands to be filled with whitened 183 lowband signal */ 184 UCHAR targetBandOffs; /*!< difference between 'startTargetBand' and 185 'startSourceBand' */ 186 UCHAR numBandsInPatch; /*!< number of consecutive bands in this one patch */ 187 } PATCH_PARAM; 188 189 /** whitening factors for different levels of whitening 190 need to be initialized corresponding to crossover frequency */ 191 typedef struct { 192 FIXP_DBL off; /*!< bw factor for signal OFF */ 193 FIXP_DBL transitionLevel; 194 FIXP_DBL lowLevel; /*!< bw factor for signal LOW_LEVEL */ 195 FIXP_DBL midLevel; /*!< bw factor for signal MID_LEVEL */ 196 FIXP_DBL highLevel; /*!< bw factor for signal HIGH_LEVEL */ 197 } WHITENING_FACTORS; 198 199 /*! The transposer settings are calculated on a header reset and are shared by 200 * both channels. */ 201 typedef struct { 202 UCHAR nCols; /*!< number subsamples of a codec frame */ 203 UCHAR noOfPatches; /*!< number of patches */ 204 UCHAR lbStartPatching; /*!< first band of lowbands that will be patched */ 205 UCHAR lbStopPatching; /*!< first band that won't be patched anymore*/ 206 UCHAR bwBorders[MAX_NUM_NOISE_VALUES]; /*!< spectral bands with different 207 inverse filtering levels */ 208 209 PATCH_PARAM 210 patchParam[MAX_NUM_PATCHES]; /*!< new parameter set for patching */ 211 WHITENING_FACTORS 212 whFactors; /*!< the pole moving factors for certain 213 whitening levels as indicated in the bitstream 214 depending on the crossover frequency */ 215 UCHAR overlap; /*!< Overlap size */ 216 } TRANSPOSER_SETTINGS; 217 218 typedef struct { 219 TRANSPOSER_SETTINGS *pSettings; /*!< Common settings for both channels */ 220 FIXP_DBL 221 bwVectorOld[MAX_NUM_PATCHES]; /*!< pole moving factors of past frame */ 222 FIXP_DBL lpcFilterStatesRealLegSBR[LPC_ORDER + (3 * (4))][( 223 32)]; /*!< pointer array to save filter states */ 224 225 FIXP_DBL lpcFilterStatesImagLegSBR[LPC_ORDER + (3 * (4))][( 226 32)]; /*!< pointer array to save filter states */ 227 228 FIXP_DBL lpcFilterStatesRealHBE[LPC_ORDER + (3 * (4))][( 229 64)]; /*!< pointer array to save filter states */ 230 FIXP_DBL lpcFilterStatesImagHBE[LPC_ORDER + (3 * (4))][( 231 64)]; /*!< pointer array to save filter states */ 232 } SBR_LPP_TRANS; 233 234 typedef SBR_LPP_TRANS *HANDLE_SBR_LPP_TRANS; 235 236 void lppTransposer(HANDLE_SBR_LPP_TRANS hLppTrans, 237 QMF_SCALE_FACTOR *sbrScaleFactor, FIXP_DBL **qmfBufferReal, 238 239 FIXP_DBL *degreeAlias, FIXP_DBL **qmfBufferImag, 240 const int useLP, const int fPreWhitening, 241 const int v_k_master0, const int timeStep, 242 const int firstSlotOffset, const int lastSlotOffset, 243 const int nInvfBands, INVF_MODE *sbr_invf_mode, 244 INVF_MODE *sbr_invf_mode_prev); 245 246 void lppTransposerHBE( 247 HANDLE_SBR_LPP_TRANS hLppTrans, /*!< Handle of lpp transposer */ 248 HANDLE_HBE_TRANSPOSER hQmfTransposer, 249 QMF_SCALE_FACTOR *sbrScaleFactor, /*!< Scaling factors */ 250 FIXP_DBL **qmfBufferReal, /*!< Pointer to pointer to real part of subband 251 samples (source) */ 252 FIXP_DBL **qmfBufferImag, /*!< Pointer to pointer to imaginary part of 253 subband samples (source) */ 254 const int timeStep, /*!< Time step of envelope */ 255 const int firstSlotOffs, /*!< Start position in time */ 256 const int lastSlotOffs, /*!< Number of overlap-slots into next frame */ 257 const int nInvfBands, /*!< Number of bands for inverse filtering */ 258 INVF_MODE *sbr_invf_mode, /*!< Current inverse filtering modes */ 259 INVF_MODE *sbr_invf_mode_prev /*!< Previous inverse filtering modes */ 260 ); 261 262 SBR_ERROR 263 createLppTransposer(HANDLE_SBR_LPP_TRANS hLppTrans, 264 TRANSPOSER_SETTINGS *pSettings, const int highBandStartSb, 265 UCHAR *v_k_master, const int numMaster, const int usb, 266 const int timeSlots, const int nCols, UCHAR *noiseBandTable, 267 const int noNoiseBands, UINT fs, const int chan, 268 const int overlap); 269 270 SBR_ERROR 271 resetLppTransposer(HANDLE_SBR_LPP_TRANS hLppTrans, UCHAR highBandStartSb, 272 UCHAR *v_k_master, UCHAR numMaster, UCHAR *noiseBandTable, 273 UCHAR noNoiseBands, UCHAR usb, UINT fs); 274 275 #endif /* LPP_TRAN_H */ 276