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 /**************************** SBR decoder library ****************************** 96 97 Author(s): 98 99 Description: 100 101 *******************************************************************************/ 102 103 /*! 104 \file 105 \brief parametric stereo decoder 106 */ 107 108 #include "psdec.h" 109 110 #include "FDK_bitbuffer.h" 111 112 #include "sbr_rom.h" 113 #include "sbr_ram.h" 114 115 #include "FDK_tools_rom.h" 116 117 #include "genericStds.h" 118 119 #include "FDK_trigFcts.h" 120 121 /********************************************************************/ 122 /* MLQUAL DEFINES */ 123 /********************************************************************/ 124 125 #define FRACT_ZERO FRACT_BITS - 1 126 /********************************************************************/ 127 128 SBR_ERROR ResetPsDec(HANDLE_PS_DEC h_ps_d); 129 130 /***** HELPERS *****/ 131 132 /***************************************************************************/ 133 /*! 134 \brief Creates one instance of the PS_DEC struct 135 136 \return Error info 137 138 ****************************************************************************/ 139 int CreatePsDec(HANDLE_PS_DEC *h_PS_DEC, /*!< pointer to the module state */ 140 int aacSamplesPerFrame) { 141 SBR_ERROR errorInfo = SBRDEC_OK; 142 HANDLE_PS_DEC h_ps_d; 143 int i; 144 145 if (*h_PS_DEC == NULL) { 146 /* Get ps dec ram */ 147 h_ps_d = GetRam_ps_dec(); 148 if (h_ps_d == NULL) { 149 goto bail; 150 } 151 } else { 152 /* Reset an open instance */ 153 h_ps_d = *h_PS_DEC; 154 } 155 156 /* 157 * Create Analysis Hybrid filterbank. 158 */ 159 FDKhybridAnalysisOpen(&h_ps_d->specificTo.mpeg.hybridAnalysis, 160 h_ps_d->specificTo.mpeg.pHybridAnaStatesLFdmx, 161 sizeof(h_ps_d->specificTo.mpeg.pHybridAnaStatesLFdmx), 162 NULL, 0); 163 164 /* initialisation */ 165 switch (aacSamplesPerFrame) { 166 case 960: 167 h_ps_d->noSubSamples = 30; /* col */ 168 break; 169 case 1024: 170 h_ps_d->noSubSamples = 32; /* col */ 171 break; 172 default: 173 h_ps_d->noSubSamples = -1; 174 break; 175 } 176 177 if (h_ps_d->noSubSamples > MAX_NUM_COL || h_ps_d->noSubSamples <= 0) { 178 goto bail; 179 } 180 h_ps_d->noChannels = NO_QMF_CHANNELS; /* row */ 181 182 h_ps_d->psDecodedPrv = 0; 183 h_ps_d->procFrameBased = -1; 184 for (i = 0; i < (1) + 1; i++) { 185 h_ps_d->bPsDataAvail[i] = ppt_none; 186 } 187 { 188 int error; 189 error = FDKdecorrelateOpen(&(h_ps_d->specificTo.mpeg.apDecor), 190 h_ps_d->specificTo.mpeg.decorrBufferCplx, 191 (2 * ((825) + (373)))); 192 if (error) goto bail; 193 } 194 195 for (i = 0; i < (1) + 1; i++) { 196 FDKmemclear(&h_ps_d->bsData[i].mpeg, sizeof(MPEG_PS_BS_DATA)); 197 } 198 199 errorInfo = ResetPsDec(h_ps_d); 200 201 if (errorInfo != SBRDEC_OK) goto bail; 202 203 *h_PS_DEC = h_ps_d; 204 205 return 0; 206 207 bail: 208 if (h_ps_d != NULL) { 209 DeletePsDec(&h_ps_d); 210 } 211 212 return -1; 213 } /*END CreatePsDec */ 214 215 /***************************************************************************/ 216 /*! 217 \brief Delete one instance of the PS_DEC struct 218 219 \return Error info 220 221 ****************************************************************************/ 222 int DeletePsDec(HANDLE_PS_DEC *h_PS_DEC) /*!< pointer to the module state */ 223 { 224 if (*h_PS_DEC == NULL) { 225 return -1; 226 } 227 228 { 229 HANDLE_PS_DEC h_ps_d = *h_PS_DEC; 230 FDKdecorrelateClose(&(h_ps_d->specificTo.mpeg.apDecor)); 231 } 232 233 FreeRam_ps_dec(h_PS_DEC); 234 235 return 0; 236 } /*END DeletePsDec */ 237 238 /***************************************************************************/ 239 /*! 240 \brief resets some values of the PS handle to default states 241 242 \return 243 244 ****************************************************************************/ 245 SBR_ERROR ResetPsDec(HANDLE_PS_DEC h_ps_d) /*!< pointer to the module state */ 246 { 247 SBR_ERROR errorInfo = SBRDEC_OK; 248 INT i; 249 250 /* explicitly init state variables to safe values (until first ps header 251 * arrives) */ 252 253 h_ps_d->specificTo.mpeg.lastUsb = 0; 254 255 /* 256 * Initialize Analysis Hybrid filterbank. 257 */ 258 FDKhybridAnalysisInit(&h_ps_d->specificTo.mpeg.hybridAnalysis, THREE_TO_TEN, 259 NO_QMF_BANDS_HYBRID20, NO_QMF_BANDS_HYBRID20, 1); 260 261 /* 262 * Initialize Synthesis Hybrid filterbank. 263 */ 264 for (i = 0; i < 2; i++) { 265 FDKhybridSynthesisInit(&h_ps_d->specificTo.mpeg.hybridSynthesis[i], 266 THREE_TO_TEN, NO_QMF_CHANNELS, NO_QMF_CHANNELS); 267 } 268 { 269 INT error; 270 error = FDKdecorrelateInit(&h_ps_d->specificTo.mpeg.apDecor, 71, DECORR_PS, 271 DUCKER_AUTOMATIC, 0, 0, 0, 0, 1, /* isLegacyPS */ 272 1); 273 if (error) return SBRDEC_NOT_INITIALIZED; 274 } 275 276 for (i = 0; i < NO_IID_GROUPS; i++) { 277 h_ps_d->specificTo.mpeg.h11rPrev[i] = FL2FXCONST_DBL(0.5f); 278 h_ps_d->specificTo.mpeg.h12rPrev[i] = FL2FXCONST_DBL(0.5f); 279 } 280 281 FDKmemclear(h_ps_d->specificTo.mpeg.h21rPrev, 282 sizeof(h_ps_d->specificTo.mpeg.h21rPrev)); 283 FDKmemclear(h_ps_d->specificTo.mpeg.h22rPrev, 284 sizeof(h_ps_d->specificTo.mpeg.h22rPrev)); 285 286 return errorInfo; 287 } 288 289 /***************************************************************************/ 290 /*! 291 \brief Feed delaylines when parametric stereo is switched on. 292 \return 293 ****************************************************************************/ 294 void PreparePsProcessing(HANDLE_PS_DEC h_ps_d, 295 const FIXP_DBL *const *const rIntBufferLeft, 296 const FIXP_DBL *const *const iIntBufferLeft, 297 const int scaleFactorLowBand) { 298 if (h_ps_d->procFrameBased == 299 1) /* If we have switched from frame to slot based processing */ 300 { /* fill hybrid delay buffer. */ 301 int i, j; 302 303 for (i = 0; i < HYBRID_FILTER_DELAY; i++) { 304 FIXP_DBL qmfInputData[2][NO_QMF_BANDS_HYBRID20]; 305 FIXP_DBL hybridOutputData[2][NO_SUB_QMF_CHANNELS]; 306 307 for (j = 0; j < NO_QMF_BANDS_HYBRID20; j++) { 308 qmfInputData[0][j] = 309 scaleValue(rIntBufferLeft[i][j], scaleFactorLowBand); 310 qmfInputData[1][j] = 311 scaleValue(iIntBufferLeft[i][j], scaleFactorLowBand); 312 } 313 314 FDKhybridAnalysisApply(&h_ps_d->specificTo.mpeg.hybridAnalysis, 315 qmfInputData[0], qmfInputData[1], 316 hybridOutputData[0], hybridOutputData[1]); 317 } 318 h_ps_d->procFrameBased = 0; /* switch to slot based processing. */ 319 320 } /* procFrameBased==1 */ 321 } 322 323 void initSlotBasedRotation( 324 HANDLE_PS_DEC h_ps_d, /*!< pointer to the module state */ 325 int env, int usb) { 326 INT group = 0; 327 INT bin = 0; 328 INT noIidSteps; 329 330 FIXP_SGL invL; 331 FIXP_DBL ScaleL, ScaleR; 332 FIXP_DBL Alpha, Beta; 333 FIXP_DBL h11r, h12r, h21r, h22r; 334 335 const FIXP_DBL *PScaleFactors; 336 337 if (h_ps_d->bsData[h_ps_d->processSlot].mpeg.bFineIidQ) { 338 PScaleFactors = ScaleFactorsFine; /* values are shiftet right by one */ 339 noIidSteps = NO_IID_STEPS_FINE; 340 } else { 341 PScaleFactors = ScaleFactors; /* values are shiftet right by one */ 342 noIidSteps = NO_IID_STEPS; 343 } 344 345 /* dequantize and decode */ 346 for (group = 0; group < NO_IID_GROUPS; group++) { 347 bin = bins2groupMap20[group]; 348 349 /*! 350 <h3> type 'A' rotation </h3> 351 mixing procedure R_a, used in baseline version<br> 352 353 Scale-factor vectors c1 and c2 are precalculated in initPsTables () and 354 stored in scaleFactors[] and scaleFactorsFine[] = pScaleFactors []. From the 355 linearized IID parameters (intensity differences), two scale factors are 356 calculated. They are used to obtain the coefficients h11... h22. 357 */ 358 359 /* ScaleR and ScaleL are scaled by 1 shift right */ 360 361 ScaleR = PScaleFactors[noIidSteps + h_ps_d->specificTo.mpeg.pCoef 362 ->aaIidIndexMapped[env][bin]]; 363 ScaleL = PScaleFactors[noIidSteps - h_ps_d->specificTo.mpeg.pCoef 364 ->aaIidIndexMapped[env][bin]]; 365 366 Beta = fMult( 367 fMult(Alphas[h_ps_d->specificTo.mpeg.pCoef->aaIccIndexMapped[env][bin]], 368 (ScaleR - ScaleL)), 369 FIXP_SQRT05); 370 Alpha = 371 Alphas[h_ps_d->specificTo.mpeg.pCoef->aaIccIndexMapped[env][bin]] >> 1; 372 373 /* Alpha and Beta are now both scaled by 2 shifts right */ 374 375 /* calculate the coefficients h11... h22 from scale-factors and ICC 376 * parameters */ 377 378 /* h values are scaled by 1 shift right */ 379 { 380 FIXP_DBL trigData[4]; 381 382 inline_fixp_cos_sin(Beta + Alpha, Beta - Alpha, 2, trigData); 383 h11r = fMult(ScaleL, trigData[0]); 384 h12r = fMult(ScaleR, trigData[2]); 385 h21r = fMult(ScaleL, trigData[1]); 386 h22r = fMult(ScaleR, trigData[3]); 387 } 388 /*****************************************************************************************/ 389 /* Interpolation of the matrices H11... H22: */ 390 /* */ 391 /* H11(k,n) = H11(k,n[e]) + (n-n[e]) * (H11(k,n[e+1] - H11(k,n[e])) / 392 * (n[e+1] - n[e]) */ 393 /* ... */ 394 /*****************************************************************************************/ 395 396 /* invL = 1/(length of envelope) */ 397 invL = FX_DBL2FX_SGL(GetInvInt( 398 h_ps_d->bsData[h_ps_d->processSlot].mpeg.aEnvStartStop[env + 1] - 399 h_ps_d->bsData[h_ps_d->processSlot].mpeg.aEnvStartStop[env])); 400 401 h_ps_d->specificTo.mpeg.pCoef->H11r[group] = 402 h_ps_d->specificTo.mpeg.h11rPrev[group]; 403 h_ps_d->specificTo.mpeg.pCoef->H12r[group] = 404 h_ps_d->specificTo.mpeg.h12rPrev[group]; 405 h_ps_d->specificTo.mpeg.pCoef->H21r[group] = 406 h_ps_d->specificTo.mpeg.h21rPrev[group]; 407 h_ps_d->specificTo.mpeg.pCoef->H22r[group] = 408 h_ps_d->specificTo.mpeg.h22rPrev[group]; 409 410 h_ps_d->specificTo.mpeg.pCoef->DeltaH11r[group] = 411 fMult(h11r - h_ps_d->specificTo.mpeg.pCoef->H11r[group], invL); 412 h_ps_d->specificTo.mpeg.pCoef->DeltaH12r[group] = 413 fMult(h12r - h_ps_d->specificTo.mpeg.pCoef->H12r[group], invL); 414 h_ps_d->specificTo.mpeg.pCoef->DeltaH21r[group] = 415 fMult(h21r - h_ps_d->specificTo.mpeg.pCoef->H21r[group], invL); 416 h_ps_d->specificTo.mpeg.pCoef->DeltaH22r[group] = 417 fMult(h22r - h_ps_d->specificTo.mpeg.pCoef->H22r[group], invL); 418 419 /* update prev coefficients for interpolation in next envelope */ 420 421 h_ps_d->specificTo.mpeg.h11rPrev[group] = h11r; 422 h_ps_d->specificTo.mpeg.h12rPrev[group] = h12r; 423 h_ps_d->specificTo.mpeg.h21rPrev[group] = h21r; 424 h_ps_d->specificTo.mpeg.h22rPrev[group] = h22r; 425 426 } /* group loop */ 427 } 428 429 static const UCHAR groupTable[NO_IID_GROUPS + 1] = { 430 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 431 12, 13, 14, 15, 16, 18, 21, 25, 30, 42, 71}; 432 433 static void applySlotBasedRotation( 434 HANDLE_PS_DEC h_ps_d, /*!< pointer to the module state */ 435 436 FIXP_DBL *mHybridRealLeft, /*!< hybrid values real left */ 437 FIXP_DBL *mHybridImagLeft, /*!< hybrid values imag left */ 438 439 FIXP_DBL *mHybridRealRight, /*!< hybrid values real right */ 440 FIXP_DBL *mHybridImagRight /*!< hybrid values imag right */ 441 ) { 442 INT group; 443 INT subband; 444 445 /**********************************************************************************************/ 446 /*! 447 <h2> Mapping </h2> 448 449 The number of stereo bands that is actually used depends on the number of 450 availble parameters for IID and ICC: <pre> nr. of IID para.| nr. of ICC para. 451 | nr. of Stereo bands 452 ----------------|------------------|------------------- 453 10,20 | 10,20 | 20 454 10,20 | 34 | 34 455 34 | 10,20 | 34 456 34 | 34 | 34 457 </pre> 458 In the case the number of parameters for IIS and ICC differs from the number 459 of stereo bands, a mapping from the lower number to the higher number of 460 parameters is applied. Index mapping of IID and ICC parameters is already done 461 in psbitdec.cpp. Further mapping is not needed here in baseline version. 462 **********************************************************************************************/ 463 464 /************************************************************************************************/ 465 /*! 466 <h2> Mixing </h2> 467 468 To generate the QMF subband signals for the subband samples n = n[e]+1 ,,, 469 n_[e+1] the parameters at position n[e] and n[e+1] are required as well as the 470 subband domain signals s_k(n) and d_k(n) for n = n[e]+1... n_[e+1]. n[e] 471 represents the start position for envelope e. The border positions n[e] are 472 handled in DecodePS(). 473 474 The stereo sub subband signals are constructed as: 475 <pre> 476 l_k(n) = H11(k,n) s_k(n) + H21(k,n) d_k(n) 477 r_k(n) = H21(k,n) s_k(n) + H22(k,n) d_k(n) 478 </pre> 479 In order to obtain the matrices H11(k,n)... H22 (k,n), the vectors h11(b)... 480 h22(b) need to be calculated first (b: parameter index). Depending on ICC mode 481 either mixing procedure R_a or R_b is used for that. For both procedures, the 482 parameters for parameter position n[e+1] is used. 483 ************************************************************************************************/ 484 485 /************************************************************************************************/ 486 /*! 487 <h2>Phase parameters </h2> 488 With disabled phase parameters (which is the case in baseline version), the 489 H-matrices are just calculated by: 490 491 <pre> 492 H11(k,n[e+1] = h11(b(k)) 493 (...) 494 b(k): parameter index according to mapping table 495 </pre> 496 497 <h2>Processing of the samples in the sub subbands </h2> 498 this loop includes the interpolation of the coefficients Hxx 499 ************************************************************************************************/ 500 501 /******************************************************/ 502 /* construct stereo sub subband signals according to: */ 503 /* */ 504 /* l_k(n) = H11(k,n) s_k(n) + H21(k,n) d_k(n) */ 505 /* r_k(n) = H12(k,n) s_k(n) + H22(k,n) d_k(n) */ 506 /******************************************************/ 507 PS_DEC_COEFFICIENTS *pCoef = h_ps_d->specificTo.mpeg.pCoef; 508 509 for (group = 0; group < NO_IID_GROUPS; group++) { 510 pCoef->H11r[group] += pCoef->DeltaH11r[group]; 511 pCoef->H12r[group] += pCoef->DeltaH12r[group]; 512 pCoef->H21r[group] += pCoef->DeltaH21r[group]; 513 pCoef->H22r[group] += pCoef->DeltaH22r[group]; 514 515 const int start = groupTable[group]; 516 const int stop = groupTable[group + 1]; 517 for (subband = start; subband < stop; subband++) { 518 FIXP_DBL tmpLeft = 519 fMultAdd(fMultDiv2(pCoef->H11r[group], mHybridRealLeft[subband]), 520 pCoef->H21r[group], mHybridRealRight[subband]); 521 FIXP_DBL tmpRight = 522 fMultAdd(fMultDiv2(pCoef->H12r[group], mHybridRealLeft[subband]), 523 pCoef->H22r[group], mHybridRealRight[subband]); 524 mHybridRealLeft[subband] = tmpLeft; 525 mHybridRealRight[subband] = tmpRight; 526 527 tmpLeft = 528 fMultAdd(fMultDiv2(pCoef->H11r[group], mHybridImagLeft[subband]), 529 pCoef->H21r[group], mHybridImagRight[subband]); 530 tmpRight = 531 fMultAdd(fMultDiv2(pCoef->H12r[group], mHybridImagLeft[subband]), 532 pCoef->H22r[group], mHybridImagRight[subband]); 533 mHybridImagLeft[subband] = tmpLeft; 534 mHybridImagRight[subband] = tmpRight; 535 } /* subband */ 536 } 537 } 538 539 /***************************************************************************/ 540 /*! 541 \brief Applies IID, ICC, IPD and OPD parameters to the current frame. 542 543 \return none 544 545 ****************************************************************************/ 546 void ApplyPsSlot( 547 HANDLE_PS_DEC h_ps_d, /*!< handle PS_DEC*/ 548 FIXP_DBL **rIntBufferLeft, /*!< real bands left qmf channel (38x64) */ 549 FIXP_DBL **iIntBufferLeft, /*!< imag bands left qmf channel (38x64) */ 550 FIXP_DBL *rIntBufferRight, /*!< real bands right qmf channel (38x64) */ 551 FIXP_DBL *iIntBufferRight, /*!< imag bands right qmf channel (38x64) */ 552 const int scaleFactorLowBand_no_ov, const int scaleFactorLowBand, 553 const int scaleFactorHighBand, const int lsb, const int usb) { 554 /*! 555 The 64-band QMF representation of the monaural signal generated by the SBR tool 556 is used as input of the PS tool. After the PS processing, the outputs of the 557 left and right hybrid synthesis filterbanks are used to generate the stereo 558 output signal. 559 560 <pre> 561 562 ------------- ---------- ------------- 563 | Hybrid | M_n[k,m] | | L_n[k,m] | Hybrid | l[n] 564 m[n] --->| analysis |--------->| |--------->| synthesis |-----> 565 ------------- | Stereo | ------------- 566 | | recon- | 567 | | stuction | 568 \|/ | | 569 ------------- | | 570 | De- | D_n[k,m] | | 571 | correlation |--------->| | 572 ------------- | | ------------- 573 | | R_n[k,m] | Hybrid | r[n] 574 | |--------->| synthesis |-----> 575 IID, ICC ------------------------>| | | filter bank | 576 (IPD, OPD) ---------- ------------- 577 578 m[n]: QMF represantation of the mono input 579 M_n[k,m]: (sub-)sub-band domain signals of the mono input 580 D_n[k,m]: decorrelated (sub-)sub-band domain signals 581 L_n[k,m]: (sub-)sub-band domain signals of the left output 582 R_n[k,m]: (sub-)sub-band domain signals of the right output 583 l[n],r[n]: left/right output signals 584 585 </pre> 586 */ 587 #define NO_HYBRID_DATA_BANDS (71) 588 589 int i; 590 FIXP_DBL qmfInputData[2][NO_QMF_BANDS_HYBRID20]; 591 FIXP_DBL *hybridData[2][2]; 592 C_ALLOC_SCRATCH_START(pHybridData, FIXP_DBL, 4 * NO_HYBRID_DATA_BANDS); 593 594 hybridData[0][0] = 595 pHybridData + 0 * NO_HYBRID_DATA_BANDS; /* left real hybrid data */ 596 hybridData[0][1] = 597 pHybridData + 1 * NO_HYBRID_DATA_BANDS; /* left imag hybrid data */ 598 hybridData[1][0] = 599 pHybridData + 2 * NO_HYBRID_DATA_BANDS; /* right real hybrid data */ 600 hybridData[1][1] = 601 pHybridData + 3 * NO_HYBRID_DATA_BANDS; /* right imag hybrid data */ 602 603 /*! 604 Hybrid analysis filterbank: 605 The lower 3 (5) of the 64 QMF subbands are further split to provide better 606 frequency resolution. for PS processing. For the 10 and 20 stereo bands 607 configuration, the QMF band H_0(w) is split up into 8 (sub-) sub-bands and the 608 QMF bands H_1(w) and H_2(w) are spit into 2 (sub-) 4th. (See figures 8.20 609 and 8.22 of ISO/IEC 14496-3:2001/FDAM 2:2004(E) ) 610 */ 611 612 /* 613 * Hybrid analysis. 614 */ 615 616 /* Get qmf input data and apply descaling */ 617 for (i = 0; i < NO_QMF_BANDS_HYBRID20; i++) { 618 qmfInputData[0][i] = scaleValue(rIntBufferLeft[HYBRID_FILTER_DELAY][i], 619 scaleFactorLowBand_no_ov); 620 qmfInputData[1][i] = scaleValue(iIntBufferLeft[HYBRID_FILTER_DELAY][i], 621 scaleFactorLowBand_no_ov); 622 } 623 624 /* LF - part */ 625 FDKhybridAnalysisApply(&h_ps_d->specificTo.mpeg.hybridAnalysis, 626 qmfInputData[0], qmfInputData[1], hybridData[0][0], 627 hybridData[0][1]); 628 629 /* HF - part */ 630 /* bands up to lsb */ 631 scaleValues(&hybridData[0][0][NO_SUB_QMF_CHANNELS - 2], 632 &rIntBufferLeft[0][NO_QMF_BANDS_HYBRID20], 633 lsb - NO_QMF_BANDS_HYBRID20, scaleFactorLowBand); 634 scaleValues(&hybridData[0][1][NO_SUB_QMF_CHANNELS - 2], 635 &iIntBufferLeft[0][NO_QMF_BANDS_HYBRID20], 636 lsb - NO_QMF_BANDS_HYBRID20, scaleFactorLowBand); 637 638 /* bands from lsb to usb */ 639 scaleValues(&hybridData[0][0][lsb + (NO_SUB_QMF_CHANNELS - 2 - 640 NO_QMF_BANDS_HYBRID20)], 641 &rIntBufferLeft[0][lsb], usb - lsb, scaleFactorHighBand); 642 scaleValues(&hybridData[0][1][lsb + (NO_SUB_QMF_CHANNELS - 2 - 643 NO_QMF_BANDS_HYBRID20)], 644 &iIntBufferLeft[0][lsb], usb - lsb, scaleFactorHighBand); 645 646 /* bands from usb to NO_SUB_QMF_CHANNELS which should be zero for non-overlap 647 slots but can be non-zero for overlap slots */ 648 FDKmemcpy( 649 &hybridData[0][0] 650 [usb + (NO_SUB_QMF_CHANNELS - 2 - NO_QMF_BANDS_HYBRID20)], 651 &rIntBufferLeft[0][usb], sizeof(FIXP_DBL) * (NO_QMF_CHANNELS - usb)); 652 FDKmemcpy( 653 &hybridData[0][1] 654 [usb + (NO_SUB_QMF_CHANNELS - 2 - NO_QMF_BANDS_HYBRID20)], 655 &iIntBufferLeft[0][usb], sizeof(FIXP_DBL) * (NO_QMF_CHANNELS - usb)); 656 657 /*! 658 Decorrelation: 659 By means of all-pass filtering and delaying, the (sub-)sub-band samples s_k(n) 660 are converted into de-correlated (sub-)sub-band samples d_k(n). 661 - k: frequency in hybrid spectrum 662 - n: time index 663 */ 664 665 FDKdecorrelateApply(&h_ps_d->specificTo.mpeg.apDecor, 666 &hybridData[0][0][0], /* left real hybrid data */ 667 &hybridData[0][1][0], /* left imag hybrid data */ 668 &hybridData[1][0][0], /* right real hybrid data */ 669 &hybridData[1][1][0], /* right imag hybrid data */ 670 0 /* startHybBand */ 671 ); 672 673 /*! 674 Stereo Processing: 675 The sets of (sub-)sub-band samples s_k(n) and d_k(n) are processed according 676 to the stereo cues which are defined per stereo band. 677 */ 678 679 applySlotBasedRotation(h_ps_d, 680 &hybridData[0][0][0], /* left real hybrid data */ 681 &hybridData[0][1][0], /* left imag hybrid data */ 682 &hybridData[1][0][0], /* right real hybrid data */ 683 &hybridData[1][1][0] /* right imag hybrid data */ 684 ); 685 686 /*! 687 Hybrid synthesis filterbank: 688 The stereo processed hybrid subband signals l_k(n) and r_k(n) are fed into the 689 hybrid synthesis filterbanks which are identical to the 64 complex synthesis 690 filterbank of the SBR tool. The input to the filterbank are slots of 64 QMF 691 samples. For each slot the filterbank outputs one block of 64 samples of one 692 reconstructed stereo channel. The hybrid synthesis filterbank is computed 693 seperatly for the left and right channel. 694 */ 695 696 /* 697 * Hybrid synthesis. 698 */ 699 for (i = 0; i < 2; i++) { 700 FDKhybridSynthesisApply( 701 &h_ps_d->specificTo.mpeg.hybridSynthesis[i], 702 hybridData[i][0], /* real hybrid data */ 703 hybridData[i][1], /* imag hybrid data */ 704 (i == 0) ? rIntBufferLeft[0] 705 : rIntBufferRight, /* output real qmf buffer */ 706 (i == 0) ? iIntBufferLeft[0] 707 : iIntBufferRight /* output imag qmf buffer */ 708 ); 709 } 710 711 /* free temporary hybrid qmf values of one timeslot */ 712 C_ALLOC_SCRATCH_END(pHybridData, FIXP_DBL, 4 * NO_HYBRID_DATA_BANDS); 713 714 } /* END ApplyPsSlot */ 715
