1 /* ----------------------------------------------------------------------------- 2 Software License for The Fraunhofer FDK AAC Codec Library for Android 3 4 Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Frderung 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 (at) iis.fraunhofer.de 93 ----------------------------------------------------------------------------- */ 94 95 /******************* Library for basic calculation routines ******************** 96 97 Author(s): 98 99 Description: 100 101 *******************************************************************************/ 102 103 /*! 104 \file qmf.h 105 \brief Complex qmf analysis/synthesis 106 \author Markus Werner 107 108 */ 109 110 #ifndef QMF_H 111 #define QMF_H 112 113 #include "common_fix.h" 114 #include "FDK_tools_rom.h" 115 #include "dct.h" 116 117 #define FIXP_QAS FIXP_PCM 118 #define QAS_BITS SAMPLE_BITS 119 120 #define FIXP_QSS FIXP_DBL 121 #define QSS_BITS DFRACT_BITS 122 123 /* Flags for QMF intialization */ 124 /* Low Power mode flag */ 125 #define QMF_FLAG_LP 1 126 /* Filter is not symmetric. This flag is set internally in the QMF 127 * initialization as required. */ 128 /* DO NOT PASS THIS FLAG TO qmfInitAnalysisFilterBank or 129 * qmfInitSynthesisFilterBank */ 130 #define QMF_FLAG_NONSYMMETRIC 2 131 /* Complex Low Delay Filter Bank (or std symmetric filter bank) */ 132 #define QMF_FLAG_CLDFB 4 133 /* Flag indicating that the states should be kept. */ 134 #define QMF_FLAG_KEEP_STATES 8 135 /* Complex Low Delay Filter Bank used in MPEG Surround Encoder */ 136 #define QMF_FLAG_MPSLDFB 16 137 /* Complex Low Delay Filter Bank used in MPEG Surround Encoder allows a 138 * optimized calculation of the modulation in qmfForwardModulationHQ() */ 139 #define QMF_FLAG_MPSLDFB_OPTIMIZE_MODULATION 32 140 /* Flag to indicate HE-AAC down-sampled SBR mode (decoder) -> adapt analysis 141 * post twiddling */ 142 #define QMF_FLAG_DOWNSAMPLED 64 143 144 #define QMF_MAX_SYNTHESIS_BANDS (64) 145 146 /*! 147 * \brief Algorithmic scaling in sbrForwardModulation() 148 * 149 * The scaling in sbrForwardModulation() is caused by: 150 * 151 * \li 1 R_SHIFT in sbrForwardModulation() 152 * \li 5/6 R_SHIFT in dct3() if using 32/64 Bands 153 * \li 1 omitted gain of 2.0 in qmfForwardModulation() 154 */ 155 #define ALGORITHMIC_SCALING_IN_ANALYSIS_FILTERBANK 7 156 157 /*! 158 * \brief Algorithmic scaling in cplxSynthesisQmfFiltering() 159 * 160 * The scaling in cplxSynthesisQmfFiltering() is caused by: 161 * 162 * \li 5/6 R_SHIFT in dct2() if using 32/64 Bands 163 * \li 1 omitted gain of 2.0 in qmfInverseModulation() 164 * \li -6 division by 64 in synthesis filterbank 165 * \li x bits external influence 166 */ 167 #define ALGORITHMIC_SCALING_IN_SYNTHESIS_FILTERBANK 1 168 169 typedef struct { 170 int lb_scale; /*!< Scale of low band area */ 171 int ov_lb_scale; /*!< Scale of adjusted overlap low band area */ 172 int hb_scale; /*!< Scale of high band area */ 173 int ov_hb_scale; /*!< Scale of adjusted overlap high band area */ 174 } QMF_SCALE_FACTOR; 175 176 struct QMF_FILTER_BANK { 177 const FIXP_PFT *p_filter; /*!< Pointer to filter coefficients */ 178 179 void *FilterStates; /*!< Pointer to buffer of filter states 180 FIXP_PCM in analyse and 181 FIXP_DBL in synthesis filter */ 182 int FilterSize; /*!< Size of prototype filter. */ 183 const FIXP_QTW *t_cos; /*!< Modulation tables. */ 184 const FIXP_QTW *t_sin; 185 int filterScale; /*!< filter scale */ 186 187 int no_channels; /*!< Total number of channels (subbands) */ 188 int no_col; /*!< Number of time slots */ 189 int lsb; /*!< Top of low subbands */ 190 int usb; /*!< Top of high subbands */ 191 192 int synScalefactor; /*!< Scale factor of synthesis qmf (syn only) */ 193 int outScalefactor; /*!< Scale factor of output data (syn only) */ 194 FIXP_DBL outGain_m; /*!< Mantissa of gain output data (syn only) (init with 195 0x80000000 to ignore) */ 196 int outGain_e; /*!< Exponent of gain output data (syn only) */ 197 198 UINT flags; /*!< flags */ 199 UCHAR p_stride; /*!< Stride Factor of polyphase filters */ 200 }; 201 202 typedef struct QMF_FILTER_BANK *HANDLE_QMF_FILTER_BANK; 203 204 void qmfAnalysisFiltering( 205 HANDLE_QMF_FILTER_BANK anaQmf, /*!< Handle of Qmf Analysis Bank */ 206 FIXP_DBL **qmfReal, /*!< Pointer to real subband slots */ 207 FIXP_DBL **qmfImag, /*!< Pointer to imag subband slots */ 208 QMF_SCALE_FACTOR *scaleFactor, /*!< Scale factors of QMF data */ 209 const LONG *timeIn, /*!< Time signal */ 210 const int timeIn_e, /*!< Exponent of audio data */ 211 const int stride, /*!< Stride factor of audio data */ 212 FIXP_DBL *pWorkBuffer /*!< pointer to temporary working buffer */ 213 ); 214 215 void qmfAnalysisFiltering( 216 HANDLE_QMF_FILTER_BANK anaQmf, /*!< Handle of Qmf Analysis Bank */ 217 FIXP_DBL **qmfReal, /*!< Pointer to real subband slots */ 218 FIXP_DBL **qmfImag, /*!< Pointer to imag subband slots */ 219 QMF_SCALE_FACTOR *scaleFactor, /*!< Scale factors of QMF data */ 220 const INT_PCM *timeIn, /*!< Time signal */ 221 const int timeIn_e, /*!< Exponent of audio data */ 222 const int stride, /*!< Stride factor of audio data */ 223 FIXP_DBL *pWorkBuffer /*!< pointer to temporal working buffer */ 224 ); 225 226 void qmfSynthesisFiltering( 227 HANDLE_QMF_FILTER_BANK synQmf, /*!< Handle of Qmf Synthesis Bank */ 228 FIXP_DBL **QmfBufferReal, /*!< Pointer to real subband slots */ 229 FIXP_DBL **QmfBufferImag, /*!< Pointer to imag subband slots */ 230 const QMF_SCALE_FACTOR *scaleFactor, /*!< Scale factors of QMF data */ 231 const int ov_len, /*!< Length of band overlap */ 232 INT_PCM *timeOut, /*!< Time signal */ 233 const INT stride, /*!< Stride factor of audio data */ 234 FIXP_DBL *pWorkBuffer /*!< pointer to temporary working buffer. It must be 235 aligned */ 236 ); 237 238 int qmfInitAnalysisFilterBank( 239 HANDLE_QMF_FILTER_BANK h_Qmf, /*!< QMF Handle */ 240 FIXP_QAS *pFilterStates, /*!< Pointer to filter state buffer */ 241 int noCols, /*!< Number of time slots */ 242 int lsb, /*!< Number of lower bands */ 243 int usb, /*!< Number of upper bands */ 244 int no_channels, /*!< Number of critically sampled bands */ 245 int flags); /*!< Flags */ 246 247 void qmfAnalysisFilteringSlot( 248 HANDLE_QMF_FILTER_BANK anaQmf, /*!< Handle of Qmf Synthesis Bank */ 249 FIXP_DBL *qmfReal, /*!< Low and High band, real */ 250 FIXP_DBL *qmfImag, /*!< Low and High band, imag */ 251 const LONG *timeIn, /*!< Pointer to input */ 252 const int stride, /*!< stride factor of input */ 253 FIXP_DBL *pWorkBuffer /*!< pointer to temporary working buffer */ 254 ); 255 256 void qmfAnalysisFilteringSlot( 257 HANDLE_QMF_FILTER_BANK anaQmf, /*!< Handle of Qmf Synthesis Bank */ 258 FIXP_DBL *qmfReal, /*!< Low and High band, real */ 259 FIXP_DBL *qmfImag, /*!< Low and High band, imag */ 260 const INT_PCM *timeIn, /*!< Pointer to input */ 261 const int stride, /*!< stride factor of input */ 262 FIXP_DBL *pWorkBuffer /*!< pointer to temporal working buffer */ 263 ); 264 int qmfInitSynthesisFilterBank( 265 HANDLE_QMF_FILTER_BANK h_Qmf, /*!< QMF Handle */ 266 FIXP_QSS *pFilterStates, /*!< Pointer to filter state buffer */ 267 int noCols, /*!< Number of time slots */ 268 int lsb, /*!< Number of lower bands */ 269 int usb, /*!< Number of upper bands */ 270 int no_channels, /*!< Number of critically sampled bands */ 271 int flags); /*!< Flags */ 272 273 void qmfSynthesisFilteringSlot(HANDLE_QMF_FILTER_BANK synQmf, 274 const FIXP_DBL *realSlot, 275 const FIXP_DBL *imagSlot, 276 const int scaleFactorLowBand, 277 const int scaleFactorHighBand, INT_PCM *timeOut, 278 const int timeOut_e, FIXP_DBL *pWorkBuffer); 279 280 void qmfChangeOutScalefactor( 281 HANDLE_QMF_FILTER_BANK synQmf, /*!< Handle of Qmf Synthesis Bank */ 282 int outScalefactor /*!< New scaling factor for output data */ 283 ); 284 285 int qmfGetOutScalefactor( 286 HANDLE_QMF_FILTER_BANK synQmf /*!< Handle of Qmf Synthesis Bank */ 287 ); 288 289 void qmfChangeOutGain( 290 HANDLE_QMF_FILTER_BANK synQmf, /*!< Handle of Qmf Synthesis Bank */ 291 FIXP_DBL outputGain, /*!< New gain for output data (mantissa) */ 292 int outputGainScale /*!< New gain for output data (exponent) */ 293 ); 294 void qmfSynPrototypeFirSlot( 295 HANDLE_QMF_FILTER_BANK qmf, 296 FIXP_DBL *RESTRICT realSlot, /*!< Input: Pointer to real Slot */ 297 FIXP_DBL *RESTRICT imagSlot, /*!< Input: Pointer to imag Slot */ 298 INT_PCM *RESTRICT timeOut, /*!< Time domain data */ 299 const int timeOut_e); 300 301 #endif /*ifndef QMF_H */ 302
