1 2 /* ----------------------------------------------------------------------------------------------------------- 3 Software License for The Fraunhofer FDK AAC Codec Library for Android 4 5 Copyright 1995 - 2013 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 /* Flag to indicate HE-AAC down-sampled SBR mode (decoder) -> adapt analysis post twiddling */ 141 #define QMF_FLAG_DOWNSAMPLED 64 142 143 144 typedef struct 145 { 146 int lb_scale; /*!< Scale of low band area */ 147 int ov_lb_scale; /*!< Scale of adjusted overlap low band area */ 148 int hb_scale; /*!< Scale of high band area */ 149 int ov_hb_scale; /*!< Scale of adjusted overlap high band area */ 150 } QMF_SCALE_FACTOR; 151 152 struct QMF_FILTER_BANK 153 { 154 const FIXP_PFT *p_filter; /*!< Pointer to filter coefficients */ 155 156 void *FilterStates; /*!< Pointer to buffer of filter states 157 FIXP_PCM in analyse and 158 FIXP_DBL in synthesis filter */ 159 int FilterSize; /*!< Size of prototype filter. */ 160 const FIXP_QTW *t_cos; /*!< Modulation tables. */ 161 const FIXP_QTW *t_sin; 162 int filterScale; /*!< filter scale */ 163 164 int no_channels; /*!< Total number of channels (subbands) */ 165 int no_col; /*!< Number of time slots */ 166 int lsb; /*!< Top of low subbands */ 167 int usb; /*!< Top of high subbands */ 168 169 int outScalefactor; /*!< Scale factor of output data (syn only) */ 170 FIXP_DBL outGain; /*!< Gain output data (syn only) (init with 0x80000000 to ignore) */ 171 172 UINT flags; /*!< flags */ 173 UCHAR p_stride; /*!< Stride Factor of polyphase filters */ 174 175 }; 176 177 typedef struct QMF_FILTER_BANK *HANDLE_QMF_FILTER_BANK; 178 179 void 180 qmfAnalysisFiltering( HANDLE_QMF_FILTER_BANK anaQmf, /*!< Handle of Qmf Analysis Bank */ 181 FIXP_QMF **qmfReal, /*!< Pointer to real subband slots */ 182 FIXP_QMF **qmfImag, /*!< Pointer to imag subband slots */ 183 QMF_SCALE_FACTOR *scaleFactor, /*!< Scale factors of QMF data */ 184 const INT_PCM *timeIn, /*!< Time signal */ 185 const int stride, /*!< Stride factor of audio data */ 186 FIXP_QMF *pWorkBuffer /*!< pointer to temporal working buffer */ 187 ); 188 189 void 190 qmfSynthesisFiltering( HANDLE_QMF_FILTER_BANK synQmf, /*!< Handle of Qmf Synthesis Bank */ 191 FIXP_QMF **QmfBufferReal, /*!< Pointer to real subband slots */ 192 FIXP_QMF **QmfBufferImag, /*!< Pointer to imag subband slots */ 193 const QMF_SCALE_FACTOR *scaleFactor, /*!< Scale factors of QMF data */ 194 const int ov_len, /*!< Length of band overlap */ 195 INT_PCM *timeOut, /*!< Time signal */ 196 const int stride, /*!< Stride factor of audio data */ 197 FIXP_QMF *pWorkBuffer /*!< pointer to temporal working buffer */ 198 ); 199 200 int 201 qmfInitAnalysisFilterBank( HANDLE_QMF_FILTER_BANK h_Qmf, /*!< QMF Handle */ 202 FIXP_QAS *pFilterStates, /*!< Pointer to filter state buffer */ 203 int noCols, /*!< Number of time slots */ 204 int lsb, /*!< Number of lower bands */ 205 int usb, /*!< Number of upper bands */ 206 int no_channels, /*!< Number of critically sampled bands */ 207 int flags); /*!< Flags */ 208 209 void 210 qmfAnalysisFilteringSlot( HANDLE_QMF_FILTER_BANK anaQmf, /*!< Handle of Qmf Synthesis Bank */ 211 FIXP_QMF *qmfReal, /*!< Low and High band, real */ 212 FIXP_QMF *qmfImag, /*!< Low and High band, imag */ 213 const INT_PCM *timeIn, /*!< Pointer to input */ 214 const int stride, /*!< stride factor of input */ 215 FIXP_QMF *pWorkBuffer /*!< pointer to temporal working buffer */ 216 ); 217 218 int 219 qmfInitSynthesisFilterBank( HANDLE_QMF_FILTER_BANK h_Qmf, /*!< QMF Handle */ 220 FIXP_QSS *pFilterStates, /*!< Pointer to filter state buffer */ 221 int noCols, /*!< Number of time slots */ 222 int lsb, /*!< Number of lower bands */ 223 int usb, /*!< Number of upper bands */ 224 int no_channels, /*!< Number of critically sampled bands */ 225 int flags); /*!< Flags */ 226 227 void qmfSynthesisFilteringSlot( HANDLE_QMF_FILTER_BANK synQmf, 228 const FIXP_QMF *realSlot, 229 const FIXP_QMF *imagSlot, 230 const int scaleFactorLowBand, 231 const int scaleFactorHighBand, 232 INT_PCM *timeOut, 233 const int stride, 234 FIXP_QMF *pWorkBuffer); 235 236 void 237 qmfChangeOutScalefactor (HANDLE_QMF_FILTER_BANK synQmf, /*!< Handle of Qmf Synthesis Bank */ 238 int outScalefactor /*!< New scaling factor for output data */ 239 ); 240 241 void 242 qmfChangeOutGain (HANDLE_QMF_FILTER_BANK synQmf, /*!< Handle of Qmf Synthesis Bank */ 243 FIXP_DBL outputGain /*!< New gain for output data */ 244 ); 245 246 247 248 #endif /* __QMF_H */ 249
