1 2 /* ----------------------------------------------------------------------------------------------------------- 3 Software License for The Fraunhofer FDK AAC Codec Library for Android 4 5 Copyright 1995 - 2012 Fraunhofer-Gesellschaft zur Frderung der angewandten Forschung e.V. 6 All rights reserved. 7 8 1. INTRODUCTION 9 The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements 10 the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. 11 This FDK AAC Codec software is intended to be used on a wide variety of Android devices. 12 13 AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual 14 audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by 15 independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part 16 of the MPEG specifications. 17 18 Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) 19 may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners 20 individually for the purpose of encoding or decoding bit streams in products that are compliant with 21 the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license 22 these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec 23 software may already be covered under those patent licenses when it is used for those licensed purposes only. 24 25 Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, 26 are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional 27 applications information and documentation. 28 29 2. COPYRIGHT LICENSE 30 31 Redistribution and use in source and binary forms, with or without modification, are permitted without 32 payment of copyright license fees provided that you satisfy the following conditions: 33 34 You must retain the complete text of this software license in redistributions of the FDK AAC Codec or 35 your modifications thereto in source code form. 36 37 You must retain the complete text of this software license in the documentation and/or other materials 38 provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. 39 You must make available free of charge copies of the complete source code of the FDK AAC Codec and your 40 modifications thereto to recipients of copies in binary form. 41 42 The name of Fraunhofer may not be used to endorse or promote products derived from this library without 43 prior written permission. 44 45 You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec 46 software or your modifications thereto. 47 48 Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software 49 and the date of any change. For modified versions of the FDK AAC Codec, the term 50 "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term 51 "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 52 53 3. NO PATENT LICENSE 54 55 NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, 56 ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with 57 respect to this software. 58 59 You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized 60 by appropriate patent licenses. 61 62 4. DISCLAIMER 63 64 This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors 65 "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties 66 of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR 67 CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, 68 including but not limited to procurement of substitute goods or services; loss of use, data, or profits, 69 or business interruption, however caused and on any theory of liability, whether in contract, strict 70 liability, or tort (including negligence), arising in any way out of the use of this software, even if 71 advised of the possibility of such damage. 72 73 5. CONTACT INFORMATION 74 75 Fraunhofer Institute for Integrated Circuits IIS 76 Attention: Audio and Multimedia Departments - FDK AAC LL 77 Am Wolfsmantel 33 78 91058 Erlangen, Germany 79 80 www.iis.fraunhofer.de/amm 81 amm-info (at) iis.fraunhofer.de 82 ----------------------------------------------------------------------------------------------------------- */ 83 84 /*! 85 \file qmf.h 86 \brief Complex qmf analysis/synthesis 87 \author Markus Werner 88 89 */ 90 #ifndef __QMF_H 91 #define __QMF_H 92 93 94 95 #include "common_fix.h" 96 #include "FDK_tools_rom.h" 97 #include "dct.h" 98 99 /* 100 * Filter coefficient type definition 101 */ 102 #ifdef QMF_DATA_16BIT 103 #define FIXP_QMF FIXP_SGL 104 #define FX_DBL2FX_QMF FX_DBL2FX_SGL 105 #define FX_QMF2FX_DBL FX_SGL2FX_DBL 106 #define QFRACT_BITS FRACT_BITS 107 #else 108 #define FIXP_QMF FIXP_DBL 109 #define FX_DBL2FX_QMF 110 #define FX_QMF2FX_DBL 111 #define QFRACT_BITS DFRACT_BITS 112 #endif 113 114 /* ARM neon optimized QMF analysis filter requires 32 bit input. 115 Implemented for RVCT only, currently disabled. See src/arm/qmf_arm.cpp:45 */ 116 #define FIXP_QAS FIXP_PCM 117 #define QAS_BITS SAMPLE_BITS 118 119 #ifdef QMFSYN_STATES_16BIT 120 #define FIXP_QSS FIXP_SGL 121 #define QSS_BITS FRACT_BITS 122 #else 123 #define FIXP_QSS FIXP_DBL 124 #define QSS_BITS DFRACT_BITS 125 #endif 126 127 /* Flags for QMF intialization */ 128 /* Low Power mode flag */ 129 #define QMF_FLAG_LP 1 130 /* Filter is not symetric. This flag is set internally in the QMF initialization as required. */ 131 #define QMF_FLAG_NONSYMMETRIC 2 132 /* Complex Low Delay Filter Bank (or std symmetric filter bank) */ 133 #define QMF_FLAG_CLDFB 4 134 /* Flag indicating that the states should be kept. */ 135 #define QMF_FLAG_KEEP_STATES 8 136 /* Complex Low Delay Filter Bank used in MPEG Surround Encoder */ 137 #define QMF_FLAG_MPSLDFB 16 138 /* Complex Low Delay Filter Bank used in MPEG Surround Encoder allows a optimized calculation of the modulation in qmfForwardModulationHQ() */ 139 #define QMF_FLAG_MPSLDFB_OPTIMIZE_MODULATION 32 140 141 142 typedef struct 143 { 144 int lb_scale; /*!< Scale of low band area */ 145 int ov_lb_scale; /*!< Scale of adjusted overlap low band area */ 146 int hb_scale; /*!< Scale of high band area */ 147 int ov_hb_scale; /*!< Scale of adjusted overlap high band area */ 148 } QMF_SCALE_FACTOR; 149 150 struct QMF_FILTER_BANK 151 { 152 const FIXP_PFT *p_filter; /*!< Pointer to filter coefficients */ 153 154 void *FilterStates; /*!< Pointer to buffer of filter states 155 FIXP_PCM in analyse and 156 FIXP_DBL in synthesis filter */ 157 int FilterSize; /*!< Size of prototype filter. */ 158 const FIXP_QTW *t_cos; /*!< Modulation tables. */ 159 const FIXP_QTW *t_sin; 160 int filterScale; /*!< filter scale */ 161 162 int no_channels; /*!< Total number of channels (subbands) */ 163 int no_col; /*!< Number of time slots */ 164 int lsb; /*!< Top of low subbands */ 165 int usb; /*!< Top of high subbands */ 166 167 int outScalefactor; /*!< Scale factor of output data (syn only) */ 168 FIXP_DBL outGain; /*!< Gain output data (syn only) (init with 0x80000000 to ignore) */ 169 170 UINT flags; /*!< flags */ 171 UCHAR p_stride; /*!< Stride Factor of polyphase filters */ 172 173 }; 174 175 typedef struct QMF_FILTER_BANK *HANDLE_QMF_FILTER_BANK; 176 177 void 178 qmfAnalysisFiltering( HANDLE_QMF_FILTER_BANK anaQmf, /*!< Handle of Qmf Analysis Bank */ 179 FIXP_QMF **qmfReal, /*!< Pointer to real subband slots */ 180 FIXP_QMF **qmfImag, /*!< Pointer to imag subband slots */ 181 QMF_SCALE_FACTOR *scaleFactor, /*!< Scale factors of QMF data */ 182 const INT_PCM *timeIn, /*!< Time signal */ 183 const int stride, /*!< Stride factor of audio data */ 184 FIXP_QMF *pWorkBuffer /*!< pointer to temporal working buffer */ 185 ); 186 187 void 188 qmfSynthesisFiltering( HANDLE_QMF_FILTER_BANK synQmf, /*!< Handle of Qmf Synthesis Bank */ 189 FIXP_QMF **QmfBufferReal, /*!< Pointer to real subband slots */ 190 FIXP_QMF **QmfBufferImag, /*!< Pointer to imag subband slots */ 191 const QMF_SCALE_FACTOR *scaleFactor, /*!< Scale factors of QMF data */ 192 const int ov_len, /*!< Length of band overlap */ 193 INT_PCM *timeOut, /*!< Time signal */ 194 const int stride, /*!< Stride factor of audio data */ 195 FIXP_QMF *pWorkBuffer /*!< pointer to temporal working buffer */ 196 ); 197 198 int 199 qmfInitAnalysisFilterBank( HANDLE_QMF_FILTER_BANK h_Qmf, /*!< QMF Handle */ 200 FIXP_QAS *pFilterStates, /*!< Pointer to filter state buffer */ 201 int noCols, /*!< Number of time slots */ 202 int lsb, /*!< Number of lower bands */ 203 int usb, /*!< Number of upper bands */ 204 int no_channels, /*!< Number of critically sampled bands */ 205 int flags); /*!< Flags */ 206 207 void 208 qmfAnalysisFilteringSlot( HANDLE_QMF_FILTER_BANK anaQmf, /*!< Handle of Qmf Synthesis Bank */ 209 FIXP_QMF *qmfReal, /*!< Low and High band, real */ 210 FIXP_QMF *qmfImag, /*!< Low and High band, imag */ 211 const INT_PCM *timeIn, /*!< Pointer to input */ 212 const int stride, /*!< stride factor of input */ 213 FIXP_QMF *pWorkBuffer /*!< pointer to temporal working buffer */ 214 ); 215 216 int 217 qmfInitSynthesisFilterBank( HANDLE_QMF_FILTER_BANK h_Qmf, /*!< QMF Handle */ 218 FIXP_QSS *pFilterStates, /*!< Pointer to filter state buffer */ 219 int noCols, /*!< Number of time slots */ 220 int lsb, /*!< Number of lower bands */ 221 int usb, /*!< Number of upper bands */ 222 int no_channels, /*!< Number of critically sampled bands */ 223 int flags); /*!< Flags */ 224 225 void qmfSynthesisFilteringSlot( HANDLE_QMF_FILTER_BANK synQmf, 226 const FIXP_QMF *realSlot, 227 const FIXP_QMF *imagSlot, 228 const int scaleFactorLowBand, 229 const int scaleFactorHighBand, 230 INT_PCM *timeOut, 231 const int stride, 232 FIXP_QMF *pWorkBuffer); 233 234 void 235 qmfChangeOutScalefactor (HANDLE_QMF_FILTER_BANK synQmf, /*!< Handle of Qmf Synthesis Bank */ 236 int outScalefactor /*!< New scaling factor for output data */ 237 ); 238 239 void 240 qmfChangeOutGain (HANDLE_QMF_FILTER_BANK synQmf, /*!< Handle of Qmf Synthesis Bank */ 241 FIXP_DBL outputGain /*!< New gain for output data */ 242 ); 243 244 245 246 #endif /* __QMF_H */ 247
