1/* ----------------------------------------------------------------------------- 2Software License for The Fraunhofer FDK AAC Codec Library for Android 3 4© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten 5Forschung e.V. All rights reserved. 6 7 1. INTRODUCTION 8The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software 9that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding 10scheme for digital audio. This FDK AAC Codec software is intended to be used on 11a wide variety of Android devices. 12 13AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient 14general perceptual audio codecs. AAC-ELD is considered the best-performing 15full-bandwidth communications codec by independent studies and is widely 16deployed. AAC has been standardized by ISO and IEC as part of the MPEG 17specifications. 18 19Patent licenses for necessary patent claims for the FDK AAC Codec (including 20those of Fraunhofer) may be obtained through Via Licensing 21(www.vialicensing.com) or through the respective patent owners individually for 22the purpose of encoding or decoding bit streams in products that are compliant 23with the ISO/IEC MPEG audio standards. Please note that most manufacturers of 24Android devices already license these patent claims through Via Licensing or 25directly from the patent owners, and therefore FDK AAC Codec software may 26already be covered under those patent licenses when it is used for those 27licensed purposes only. 28 29Commercially-licensed AAC software libraries, including floating-point versions 30with enhanced sound quality, are also available from Fraunhofer. Users are 31encouraged to check the Fraunhofer website for additional applications 32information and documentation. 33 342. COPYRIGHT LICENSE 35 36Redistribution and use in source and binary forms, with or without modification, 37are permitted without payment of copyright license fees provided that you 38satisfy the following conditions: 39 40You must retain the complete text of this software license in redistributions of 41the FDK AAC Codec or your modifications thereto in source code form. 42 43You must retain the complete text of this software license in the documentation 44and/or other materials provided with redistributions of the FDK AAC Codec or 45your modifications thereto in binary form. You must make available free of 46charge copies of the complete source code of the FDK AAC Codec and your 47modifications thereto to recipients of copies in binary form. 48 49The name of Fraunhofer may not be used to endorse or promote products derived 50from this library without prior written permission. 51 52You may not charge copyright license fees for anyone to use, copy or distribute 53the FDK AAC Codec software or your modifications thereto. 54 55Your modified versions of the FDK AAC Codec must carry prominent notices stating 56that you changed the software and the date of any change. For modified versions 57of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" 58must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK 59AAC Codec Library for Android." 60 613. NO PATENT LICENSE 62 63NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without 64limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. 65Fraunhofer provides no warranty of patent non-infringement with respect to this 66software. 67 68You may use this FDK AAC Codec software or modifications thereto only for 69purposes that are authorized by appropriate patent licenses. 70 714. DISCLAIMER 72 73This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright 74holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, 75including but not limited to the implied warranties of merchantability and 76fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR 77CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, 78or consequential damages, including but not limited to procurement of substitute 79goods or services; loss of use, data, or profits, or business interruption, 80however caused and on any theory of liability, whether in contract, strict 81liability, or tort (including negligence), arising in any way out of the use of 82this software, even if advised of the possibility of such damage. 83 845. CONTACT INFORMATION 85 86Fraunhofer Institute for Integrated Circuits IIS 87Attention: Audio and Multimedia Departments - FDK AAC LL 88Am Wolfsmantel 33 8991058 Erlangen, Germany 90 91www.iis.fraunhofer.de/amm 92amm-info@iis.fraunhofer.de 93----------------------------------------------------------------------------- */ 94 95/******************* Library for basic calculation routines ******************** 96 97 Author(s): Markus Lohwasser, Josef Hoepfl, Manuel Jander 98 99 Description: QMF filterbank 100 101*******************************************************************************/ 102 103#ifndef QMF_PCM_H 104#define QMF_PCM_H 105 106/* 107 All Synthesis functions dependent on datatype INT_PCM_QMFOUT 108 Should only be included by qmf.cpp, but not compiled separately, please 109 exclude compilation from project, if done otherwise. Is optional included 110 twice to duplicate all functions with two different pre-definitions, as: 111 #define INT_PCM_QMFOUT LONG 112 and ... 113 #define INT_PCM_QMFOUT SHORT 114 needed to run QMF synthesis in both 16bit and 32bit sample output format. 115*/ 116 117#define QSSCALE (0) 118#define FX_DBL2FX_QSS(x) (x) 119#define FX_QSS2FX_DBL(x) (x) 120 121/*! 122 \brief Perform Synthesis Prototype Filtering on a single slot of input data. 123 124 The filter takes 2 * qmf->no_channels of input data and 125 generates qmf->no_channels time domain output samples. 126*/ 127/* static */ 128#ifndef FUNCTION_qmfSynPrototypeFirSlot 129void qmfSynPrototypeFirSlot( 130#else 131void qmfSynPrototypeFirSlot_fallback( 132#endif 133 HANDLE_QMF_FILTER_BANK qmf, 134 FIXP_DBL *RESTRICT realSlot, /*!< Input: Pointer to real Slot */ 135 FIXP_DBL *RESTRICT imagSlot, /*!< Input: Pointer to imag Slot */ 136 INT_PCM_QMFOUT *RESTRICT timeOut, /*!< Time domain data */ 137 int stride) { 138 FIXP_QSS *FilterStates = (FIXP_QSS *)qmf->FilterStates; 139 int no_channels = qmf->no_channels; 140 const FIXP_PFT *p_Filter = qmf->p_filter; 141 int p_stride = qmf->p_stride; 142 int j; 143 FIXP_QSS *RESTRICT sta = FilterStates; 144 const FIXP_PFT *RESTRICT p_flt, *RESTRICT p_fltm; 145 int scale = (DFRACT_BITS - SAMPLE_BITS_QMFOUT) - 1 - qmf->outScalefactor - 146 qmf->outGain_e; 147 148 p_flt = 149 p_Filter + p_stride * QMF_NO_POLY; /* 5th of 330 */ 150 p_fltm = p_Filter + (qmf->FilterSize / 2) - 151 p_stride * QMF_NO_POLY; /* 5 + (320 - 2*5) = 315th of 330 */ 152 153 FIXP_SGL gain = FX_DBL2FX_SGL(qmf->outGain_m); 154 155 FIXP_DBL rnd_val = 0; 156 157 if (scale > 0) { 158 if (scale < (DFRACT_BITS - 1)) 159 rnd_val = FIXP_DBL(1 << (scale - 1)); 160 else 161 scale = (DFRACT_BITS - 1); 162 } else { 163 scale = fMax(scale, -(DFRACT_BITS - 1)); 164 } 165 166 for (j = no_channels - 1; j >= 0; j--) { 167 FIXP_DBL imag = imagSlot[j]; /* no_channels-1 .. 0 */ 168 FIXP_DBL real = realSlot[j]; /* no_channels-1 .. 0 */ 169 { 170 INT_PCM_QMFOUT tmp; 171 FIXP_DBL Are = fMultAddDiv2(FX_QSS2FX_DBL(sta[0]), p_fltm[0], real); 172 173 /* This PCM formatting performs: 174 - multiplication with 16-bit gain, if not -1.0f 175 - rounding, if shift right is applied 176 - apply shift left (or right) with saturation to 32 (or 16) bits 177 - store output with --stride in 32 (or 16) bit format 178 */ 179 if (gain != (FIXP_SGL)(-32768)) /* -1.0f */ 180 { 181 Are = fMult(Are, gain); 182 } 183 if (scale >= 0) { 184 FDK_ASSERT( 185 Are <= 186 (Are + rnd_val)); /* Round-addition must not overflow, might be 187 equal for rnd_val=0 */ 188 tmp = (INT_PCM_QMFOUT)( 189 SATURATE_RIGHT_SHIFT(Are + rnd_val, scale, SAMPLE_BITS_QMFOUT)); 190 } else { 191 tmp = (INT_PCM_QMFOUT)( 192 SATURATE_LEFT_SHIFT(Are, -scale, SAMPLE_BITS_QMFOUT)); 193 } 194 195 { timeOut[(j)*stride] = tmp; } 196 } 197 198 sta[0] = FX_DBL2FX_QSS(fMultAddDiv2(FX_QSS2FX_DBL(sta[1]), p_flt[4], imag)); 199 sta[1] = 200 FX_DBL2FX_QSS(fMultAddDiv2(FX_QSS2FX_DBL(sta[2]), p_fltm[1], real)); 201 sta[2] = FX_DBL2FX_QSS(fMultAddDiv2(FX_QSS2FX_DBL(sta[3]), p_flt[3], imag)); 202 sta[3] = 203 FX_DBL2FX_QSS(fMultAddDiv2(FX_QSS2FX_DBL(sta[4]), p_fltm[2], real)); 204 sta[4] = FX_DBL2FX_QSS(fMultAddDiv2(FX_QSS2FX_DBL(sta[5]), p_flt[2], imag)); 205 sta[5] = 206 FX_DBL2FX_QSS(fMultAddDiv2(FX_QSS2FX_DBL(sta[6]), p_fltm[3], real)); 207 sta[6] = FX_DBL2FX_QSS(fMultAddDiv2(FX_QSS2FX_DBL(sta[7]), p_flt[1], imag)); 208 sta[7] = 209 FX_DBL2FX_QSS(fMultAddDiv2(FX_QSS2FX_DBL(sta[8]), p_fltm[4], real)); 210 sta[8] = FX_DBL2FX_QSS(fMultDiv2(p_flt[0], imag)); 211 p_flt += (p_stride * QMF_NO_POLY); 212 p_fltm -= (p_stride * QMF_NO_POLY); 213 sta += 9; // = (2*QMF_NO_POLY-1); 214 } 215} 216 217#ifndef FUNCTION_qmfSynPrototypeFirSlot_NonSymmetric 218/*! 219 \brief Perform Synthesis Prototype Filtering on a single slot of input data. 220 221 The filter takes 2 * qmf->no_channels of input data and 222 generates qmf->no_channels time domain output samples. 223*/ 224static void qmfSynPrototypeFirSlot_NonSymmetric( 225 HANDLE_QMF_FILTER_BANK qmf, 226 FIXP_DBL *RESTRICT realSlot, /*!< Input: Pointer to real Slot */ 227 FIXP_DBL *RESTRICT imagSlot, /*!< Input: Pointer to imag Slot */ 228 INT_PCM_QMFOUT *RESTRICT timeOut, /*!< Time domain data */ 229 int stride) { 230 FIXP_QSS *FilterStates = (FIXP_QSS *)qmf->FilterStates; 231 int no_channels = qmf->no_channels; 232 const FIXP_PFT *p_Filter = qmf->p_filter; 233 int p_stride = qmf->p_stride; 234 int j; 235 FIXP_QSS *RESTRICT sta = FilterStates; 236 const FIXP_PFT *RESTRICT p_flt, *RESTRICT p_fltm; 237 int scale = (DFRACT_BITS - SAMPLE_BITS_QMFOUT) - 1 - qmf->outScalefactor - 238 qmf->outGain_e; 239 240 p_flt = p_Filter; /*!< Pointer to first half of filter coefficients */ 241 p_fltm = 242 &p_flt[qmf->FilterSize / 2]; /* at index 320, overall 640 coefficients */ 243 244 FIXP_SGL gain = FX_DBL2FX_SGL(qmf->outGain_m); 245 246 FIXP_DBL rnd_val = (FIXP_DBL)0; 247 248 if (scale > 0) { 249 if (scale < (DFRACT_BITS - 1)) 250 rnd_val = FIXP_DBL(1 << (scale - 1)); 251 else 252 scale = (DFRACT_BITS - 1); 253 } else { 254 scale = fMax(scale, -(DFRACT_BITS - 1)); 255 } 256 257 for (j = no_channels - 1; j >= 0; j--) { 258 FIXP_DBL imag = imagSlot[j]; /* no_channels-1 .. 0 */ 259 FIXP_DBL real = realSlot[j]; /* no_channels-1 .. 0 */ 260 { 261 INT_PCM_QMFOUT tmp; 262 FIXP_DBL Are = sta[0] + FX_DBL2FX_QSS(fMultDiv2(p_fltm[4], real)); 263 264 /* This PCM formatting performs: 265 - multiplication with 16-bit gain, if not -1.0f 266 - rounding, if shift right is applied 267 - apply shift left (or right) with saturation to 32 (or 16) bits 268 - store output with --stride in 32 (or 16) bit format 269 */ 270 if (gain != (FIXP_SGL)(-32768)) /* -1.0f */ 271 { 272 Are = fMult(Are, gain); 273 } 274 if (scale > 0) { 275 FDK_ASSERT(Are < 276 (Are + rnd_val)); /* Round-addition must not overflow */ 277 tmp = (INT_PCM_QMFOUT)( 278 SATURATE_RIGHT_SHIFT(Are + rnd_val, scale, SAMPLE_BITS_QMFOUT)); 279 } else { 280 tmp = (INT_PCM_QMFOUT)( 281 SATURATE_LEFT_SHIFT(Are, -scale, SAMPLE_BITS_QMFOUT)); 282 } 283 timeOut[j * stride] = tmp; 284 } 285 286 sta[0] = sta[1] + FX_DBL2FX_QSS(fMultDiv2(p_flt[4], imag)); 287 sta[1] = sta[2] + FX_DBL2FX_QSS(fMultDiv2(p_fltm[3], real)); 288 sta[2] = sta[3] + FX_DBL2FX_QSS(fMultDiv2(p_flt[3], imag)); 289 290 sta[3] = sta[4] + FX_DBL2FX_QSS(fMultDiv2(p_fltm[2], real)); 291 sta[4] = sta[5] + FX_DBL2FX_QSS(fMultDiv2(p_flt[2], imag)); 292 sta[5] = sta[6] + FX_DBL2FX_QSS(fMultDiv2(p_fltm[1], real)); 293 sta[6] = sta[7] + FX_DBL2FX_QSS(fMultDiv2(p_flt[1], imag)); 294 295 sta[7] = sta[8] + FX_DBL2FX_QSS(fMultDiv2(p_fltm[0], real)); 296 sta[8] = FX_DBL2FX_QSS(fMultDiv2(p_flt[0], imag)); 297 298 p_flt += (p_stride * QMF_NO_POLY); 299 p_fltm += (p_stride * QMF_NO_POLY); 300 sta += 9; // = (2*QMF_NO_POLY-1); 301 } 302} 303#endif /* FUNCTION_qmfSynPrototypeFirSlot_NonSymmetric */ 304 305void qmfSynthesisFilteringSlot(HANDLE_QMF_FILTER_BANK synQmf, 306 const FIXP_DBL *realSlot, 307 const FIXP_DBL *imagSlot, 308 const int scaleFactorLowBand, 309 const int scaleFactorHighBand, 310 INT_PCM_QMFOUT *timeOut, const int stride, 311 FIXP_DBL *pWorkBuffer) { 312 if (!(synQmf->flags & QMF_FLAG_LP)) 313 qmfInverseModulationHQ(synQmf, realSlot, imagSlot, scaleFactorLowBand, 314 scaleFactorHighBand, pWorkBuffer); 315 else { 316 if (synQmf->flags & QMF_FLAG_CLDFB) { 317 qmfInverseModulationLP_odd(synQmf, realSlot, scaleFactorLowBand, 318 scaleFactorHighBand, pWorkBuffer); 319 } else { 320 qmfInverseModulationLP_even(synQmf, realSlot, scaleFactorLowBand, 321 scaleFactorHighBand, pWorkBuffer); 322 } 323 } 324 325 if (synQmf->flags & QMF_FLAG_NONSYMMETRIC) { 326 qmfSynPrototypeFirSlot_NonSymmetric(synQmf, pWorkBuffer, 327 pWorkBuffer + synQmf->no_channels, 328 timeOut, stride); 329 } else { 330 qmfSynPrototypeFirSlot(synQmf, pWorkBuffer, 331 pWorkBuffer + synQmf->no_channels, timeOut, stride); 332 } 333} 334 335/*! 336 * 337 * \brief Perform complex-valued subband synthesis of the 338 * low band and the high band and store the 339 * time domain data in timeOut 340 * 341 * First step: Calculate the proper scaling factor of current 342 * spectral data in qmfReal/qmfImag, old spectral data in the overlap 343 * range and filter states. 344 * 345 * Second step: Perform Frequency-to-Time mapping with inverse 346 * Modulation slot-wise. 347 * 348 * Third step: Perform FIR-filter slot-wise. To save space for filter 349 * states, the MAC operations are executed directly on the filter states 350 * instead of accumulating several products in the accumulator. The 351 * buffer shift at the end of the function should be replaced by a 352 * modulo operation, which is available on some DSPs. 353 * 354 * Last step: Copy the upper part of the spectral data to the overlap buffer. 355 * 356 * The qmf coefficient table is symmetric. The symmetry is exploited by 357 * shrinking the coefficient table to half the size. The addressing mode 358 * takes care of the symmetries. If the #define #QMFTABLE_FULL is set, 359 * coefficient addressing works on the full table size. The code will be 360 * slightly faster and slightly more compact. 361 * 362 * Workbuffer requirement: 2 x sizeof(**QmfBufferReal) * synQmf->no_channels 363 * The workbuffer must be aligned 364 */ 365void qmfSynthesisFiltering( 366 HANDLE_QMF_FILTER_BANK synQmf, /*!< Handle of Qmf Synthesis Bank */ 367 FIXP_DBL **QmfBufferReal, /*!< Low and High band, real */ 368 FIXP_DBL **QmfBufferImag, /*!< Low and High band, imag */ 369 const QMF_SCALE_FACTOR *scaleFactor, 370 const INT ov_len, /*!< split Slot of overlap and actual slots */ 371 INT_PCM_QMFOUT *timeOut, /*!< Pointer to output */ 372 const INT stride, /*!< stride factor of output */ 373 FIXP_DBL *pWorkBuffer /*!< pointer to temporal working buffer */ 374) { 375 int i; 376 int L = synQmf->no_channels; 377 int scaleFactorHighBand; 378 int scaleFactorLowBand_ov, scaleFactorLowBand_no_ov; 379 380 FDK_ASSERT(synQmf->no_channels >= synQmf->lsb); 381 FDK_ASSERT(synQmf->no_channels >= synQmf->usb); 382 383 /* adapt scaling */ 384 scaleFactorHighBand = -ALGORITHMIC_SCALING_IN_ANALYSIS_FILTERBANK - 385 scaleFactor->hb_scale - synQmf->filterScale; 386 scaleFactorLowBand_ov = -ALGORITHMIC_SCALING_IN_ANALYSIS_FILTERBANK - 387 scaleFactor->ov_lb_scale - synQmf->filterScale; 388 scaleFactorLowBand_no_ov = -ALGORITHMIC_SCALING_IN_ANALYSIS_FILTERBANK - 389 scaleFactor->lb_scale - synQmf->filterScale; 390 391 for (i = 0; i < synQmf->no_col; i++) /* ----- no_col loop ----- */ 392 { 393 const FIXP_DBL *QmfBufferImagSlot = NULL; 394 395 int scaleFactorLowBand = 396 (i < ov_len) ? scaleFactorLowBand_ov : scaleFactorLowBand_no_ov; 397 398 if (!(synQmf->flags & QMF_FLAG_LP)) QmfBufferImagSlot = QmfBufferImag[i]; 399 400 qmfSynthesisFilteringSlot(synQmf, QmfBufferReal[i], QmfBufferImagSlot, 401 scaleFactorLowBand, scaleFactorHighBand, 402 timeOut + (i * L * stride), stride, pWorkBuffer); 403 } /* no_col loop i */ 404} 405#endif /* QMF_PCM_H */ 406