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 86 \brief Envelope calculation 87 88 The envelope adjustor compares the energies present in the transposed 89 highband to the reference energies conveyed with the bitstream. 90 The highband is amplified (sometimes) or attenuated (mostly) to the 91 desired level. 92 93 The spectral shape of the reference energies can be changed several times per 94 frame if necessary. Each set of energy values corresponding to a certain range 95 in time will be called an <em>envelope</em> here. 96 The bitstream supports several frequency scales and two resolutions. Normally, 97 one or more QMF-subbands are grouped to one SBR-band. An envelope contains 98 reference energies for each SBR-band. 99 In addition to the energy envelopes, noise envelopes are transmitted that 100 define the ratio of energy which is generated by adding noise instead of 101 transposing the lowband. The noise envelopes are given in a coarser time 102 and frequency resolution. 103 If a signal contains strong tonal components, synthetic sines can be 104 generated in individual SBR bands. 105 106 An overlap buffer of 6 QMF-timeslots is used to allow a more 107 flexible alignment of the envelopes in time that is not restricted to the 108 core codec's frame borders. 109 Therefore the envelope adjustor has access to the spectral data of the 110 current frame as well as the last 6 QMF-timeslots of the previous frame. 111 However, in average only the data of 1 frame is being processed as 112 the adjustor is called once per frame. 113 114 Depending on the frequency range set in the bitstream, only QMF-subbands between 115 <em>lowSubband</em> and <em>highSubband</em> are adjusted. 116 117 Scaling of spectral data to maximize SNR (see #QMF_SCALE_FACTOR) as well as a special Mantissa-Exponent format 118 ( see calculateSbrEnvelope() ) are being used. The main entry point for this modules is calculateSbrEnvelope(). 119 120 \sa sbr_scale.h, #QMF_SCALE_FACTOR, calculateSbrEnvelope(), \ref documentationOverview 121 */ 122 123 124 #include "env_calc.h" 125 126 #include "sbrdec_freq_sca.h" 127 #include "env_extr.h" 128 #include "transcendent.h" 129 #include "sbr_ram.h" 130 #include "sbr_rom.h" 131 132 #include "genericStds.h" /* need FDKpow() for debug outputs */ 133 134 #if defined(__arm__) 135 #include "arm/env_calc_arm.cpp" 136 #endif 137 138 typedef struct 139 { 140 FIXP_DBL nrgRef[MAX_FREQ_COEFFS]; 141 FIXP_DBL nrgEst[MAX_FREQ_COEFFS]; 142 FIXP_DBL nrgGain[MAX_FREQ_COEFFS]; 143 FIXP_DBL noiseLevel[MAX_FREQ_COEFFS]; 144 FIXP_DBL nrgSine[MAX_FREQ_COEFFS]; 145 146 SCHAR nrgRef_e[MAX_FREQ_COEFFS]; 147 SCHAR nrgEst_e[MAX_FREQ_COEFFS]; 148 SCHAR nrgGain_e[MAX_FREQ_COEFFS]; 149 SCHAR noiseLevel_e[MAX_FREQ_COEFFS]; 150 SCHAR nrgSine_e[MAX_FREQ_COEFFS]; 151 } 152 ENV_CALC_NRGS; 153 154 /*static*/ void equalizeFiltBufferExp(FIXP_DBL *filtBuffer, 155 SCHAR *filtBuffer_e, 156 FIXP_DBL *NrgGain, 157 SCHAR *NrgGain_e, 158 int subbands); 159 160 /*static*/ void calcNrgPerSubband(FIXP_DBL **analysBufferReal, 161 FIXP_DBL **analysBufferImag, 162 int lowSubband, int highSubband, 163 int start_pos, int next_pos, 164 SCHAR frameExp, 165 FIXP_DBL *nrgEst, 166 SCHAR *nrgEst_e ); 167 168 /*static*/ void calcNrgPerSfb(FIXP_DBL **analysBufferReal, 169 FIXP_DBL **analysBufferImag, 170 int nSfb, 171 UCHAR *freqBandTable, 172 int start_pos, int next_pos, 173 SCHAR input_e, 174 FIXP_DBL *nrg_est, 175 SCHAR *nrg_est_e ); 176 177 /*static*/ void calcSubbandGain(FIXP_DBL nrgRef, SCHAR nrgRef_e, ENV_CALC_NRGS* nrgs, int c, 178 FIXP_DBL tmpNoise, SCHAR tmpNoise_e, 179 UCHAR sinePresentFlag, 180 UCHAR sineMapped, 181 int noNoiseFlag); 182 183 /*static*/ void calcAvgGain(ENV_CALC_NRGS* nrgs, 184 int lowSubband, 185 int highSubband, 186 FIXP_DBL *sumRef_m, 187 SCHAR *sumRef_e, 188 FIXP_DBL *ptrAvgGain_m, 189 SCHAR *ptrAvgGain_e); 190 191 /*static*/ void adjustTimeSlotLC(FIXP_DBL *ptrReal, 192 ENV_CALC_NRGS* nrgs, 193 UCHAR *ptrHarmIndex, 194 int lowSubbands, 195 int noSubbands, 196 int scale_change, 197 int noNoiseFlag, 198 int *ptrPhaseIndex, 199 int fCldfb); 200 /*static*/ void adjustTimeSlotHQ(FIXP_DBL *ptrReal, 201 FIXP_DBL *ptrImag, 202 HANDLE_SBR_CALCULATE_ENVELOPE h_sbr_cal_env, 203 ENV_CALC_NRGS* nrgs, 204 int lowSubbands, 205 int noSubbands, 206 int scale_change, 207 FIXP_SGL smooth_ratio, 208 int noNoiseFlag, 209 int filtBufferNoiseShift); 210 211 212 /*! 213 \brief Map sine flags from bitstream to QMF bands 214 215 The bitstream carries only 1 sine flag per band and frame. 216 This function maps every sine flag from the bitstream to a specific QMF subband 217 and to a specific envelope where the sine shall start. 218 The result is stored in the vector sineMapped which contains one entry per 219 QMF subband. The value of an entry specifies the envelope where a sine 220 shall start. A value of #MAX_ENVELOPES indicates that no sine is present 221 in the subband. 222 The missing harmonics flags from the previous frame (harmFlagsPrev) determine 223 if a sine starts at the beginning of the frame or at the transient position. 224 Additionally, the flags in harmFlagsPrev are being updated by this function 225 for the next frame. 226 */ 227 /*static*/ void mapSineFlags(UCHAR *freqBandTable, /*!< Band borders (there's only 1 flag per band) */ 228 int nSfb, /*!< Number of bands in the table */ 229 UCHAR *addHarmonics, /*!< vector with 1 flag per sfb */ 230 int *harmFlagsPrev, /*!< Packed 'addHarmonics' */ 231 int tranEnv, /*!< Transient position */ 232 SCHAR *sineMapped) /*!< Resulting vector of sine start positions for each QMF band */ 233 234 { 235 int i; 236 int lowSubband2 = freqBandTable[0]<<1; 237 int bitcount = 0; 238 int oldflags = *harmFlagsPrev; 239 int newflags = 0; 240 241 /* 242 Format of harmFlagsPrev: 243 244 first word = flags for highest 16 sfb bands in use 245 second word = flags for next lower 16 sfb bands (if present) 246 third word = flags for lowest 16 sfb bands (if present) 247 248 Up to MAX_FREQ_COEFFS sfb bands can be flagged for a sign. 249 The lowest bit of the first word corresponds to the _highest_ sfb band in use. 250 This is ensures that each flag is mapped to the same QMF band even after a 251 change of the crossover-frequency. 252 */ 253 254 255 /* Reset the output vector first */ 256 FDKmemset(sineMapped, MAX_ENVELOPES,MAX_FREQ_COEFFS); /* MAX_ENVELOPES means 'no sine' */ 257 258 freqBandTable += nSfb; 259 addHarmonics += nSfb-1; 260 261 for (i=nSfb; i!=0; i--) { 262 int ui = *freqBandTable--; /* Upper limit of the current scale factor band. */ 263 int li = *freqBandTable; /* Lower limit of the current scale factor band. */ 264 265 if ( *addHarmonics-- ) { /* There is a sine in this band */ 266 267 unsigned int mask = 1 << bitcount; 268 newflags |= mask; /* Set flag */ 269 270 /* 271 If there was a sine in the last frame, let it continue from the first envelope on 272 else start at the transient position. 273 */ 274 sineMapped[(ui+li-lowSubband2) >> 1] = ( oldflags & mask ) ? 0 : tranEnv; 275 } 276 277 if ((++bitcount == 16) || i==1) { 278 bitcount = 0; 279 *harmFlagsPrev++ = newflags; 280 oldflags = *harmFlagsPrev; /* Fetch 16 of the old flags */ 281 newflags = 0; 282 } 283 } 284 } 285 286 287 /*! 288 \brief Reduce gain-adjustment induced aliasing for real valued filterbank. 289 */ 290 /*static*/ void 291 aliasingReduction(FIXP_DBL* degreeAlias, /*!< estimated aliasing for each QMF channel */ 292 ENV_CALC_NRGS* nrgs, 293 int* useAliasReduction, /*!< synthetic sine engergy for each subband, used as flag */ 294 int noSubbands) /*!< number of QMF channels to process */ 295 { 296 FIXP_DBL* nrgGain = nrgs->nrgGain; /*!< subband gains to be modified */ 297 SCHAR* nrgGain_e = nrgs->nrgGain_e; /*!< subband gains to be modified (exponents) */ 298 FIXP_DBL* nrgEst = nrgs->nrgEst; /*!< subband energy before amplification */ 299 SCHAR* nrgEst_e = nrgs->nrgEst_e; /*!< subband energy before amplification (exponents) */ 300 int grouping = 0, index = 0, noGroups, k; 301 int groupVector[MAX_FREQ_COEFFS]; 302 303 /* Calculate grouping*/ 304 for (k = 0; k < noSubbands-1; k++ ){ 305 if ( (degreeAlias[k + 1] != FL2FXCONST_DBL(0.0f)) && useAliasReduction[k] ) { 306 if(grouping==0){ 307 groupVector[index++] = k; 308 grouping = 1; 309 } 310 else{ 311 if(groupVector[index-1] + 3 == k){ 312 groupVector[index++] = k + 1; 313 grouping = 0; 314 } 315 } 316 } 317 else{ 318 if(grouping){ 319 if(useAliasReduction[k]) 320 groupVector[index++] = k + 1; 321 else 322 groupVector[index++] = k; 323 grouping = 0; 324 } 325 } 326 } 327 328 if(grouping){ 329 groupVector[index++] = noSubbands; 330 } 331 noGroups = index >> 1; 332 333 334 /*Calculate new gain*/ 335 for (int group = 0; group < noGroups; group ++) { 336 FIXP_DBL nrgOrig = FL2FXCONST_DBL(0.0f); /* Original signal energy in current group of bands */ 337 SCHAR nrgOrig_e = 0; 338 FIXP_DBL nrgAmp = FL2FXCONST_DBL(0.0f); /* Amplified signal energy in group (using current gains) */ 339 SCHAR nrgAmp_e = 0; 340 FIXP_DBL nrgMod = FL2FXCONST_DBL(0.0f); /* Signal energy in group when applying modified gains */ 341 SCHAR nrgMod_e = 0; 342 FIXP_DBL groupGain; /* Total energy gain in group */ 343 SCHAR groupGain_e; 344 FIXP_DBL compensation; /* Compensation factor for the energy change when applying modified gains */ 345 SCHAR compensation_e; 346 347 int startGroup = groupVector[2*group]; 348 int stopGroup = groupVector[2*group+1]; 349 350 /* Calculate total energy in group before and after amplification with current gains: */ 351 for(k = startGroup; k < stopGroup; k++){ 352 /* Get original band energy */ 353 FIXP_DBL tmp = nrgEst[k]; 354 SCHAR tmp_e = nrgEst_e[k]; 355 356 FDK_add_MantExp(tmp, tmp_e, nrgOrig, nrgOrig_e, &nrgOrig, &nrgOrig_e); 357 358 /* Multiply band energy with current gain */ 359 tmp = fMult(tmp,nrgGain[k]); 360 tmp_e = tmp_e + nrgGain_e[k]; 361 362 FDK_add_MantExp(tmp, tmp_e, nrgAmp, nrgAmp_e, &nrgAmp, &nrgAmp_e); 363 } 364 365 /* Calculate total energy gain in group */ 366 FDK_divide_MantExp(nrgAmp, nrgAmp_e, 367 nrgOrig, nrgOrig_e, 368 &groupGain, &groupGain_e); 369 370 for(k = startGroup; k < stopGroup; k++){ 371 FIXP_DBL tmp; 372 SCHAR tmp_e; 373 374 FIXP_DBL alpha = degreeAlias[k]; 375 if (k < noSubbands - 1) { 376 if (degreeAlias[k + 1] > alpha) 377 alpha = degreeAlias[k + 1]; 378 } 379 380 /* Modify gain depending on the degree of aliasing */ 381 FDK_add_MantExp( fMult(alpha,groupGain), groupGain_e, 382 fMult(/*FL2FXCONST_DBL(1.0f)*/ (FIXP_DBL)MAXVAL_DBL - alpha,nrgGain[k]), nrgGain_e[k], 383 &nrgGain[k], &nrgGain_e[k] ); 384 385 /* Apply modified gain to original energy */ 386 tmp = fMult(nrgGain[k],nrgEst[k]); 387 tmp_e = nrgGain_e[k] + nrgEst_e[k]; 388 389 /* Accumulate energy with modified gains applied */ 390 FDK_add_MantExp( tmp, tmp_e, 391 nrgMod, nrgMod_e, 392 &nrgMod, &nrgMod_e ); 393 } 394 395 /* Calculate compensation factor to retain the energy of the amplified signal */ 396 FDK_divide_MantExp(nrgAmp, nrgAmp_e, 397 nrgMod, nrgMod_e, 398 &compensation, &compensation_e); 399 400 /* Apply compensation factor to all gains of the group */ 401 for(k = startGroup; k < stopGroup; k++){ 402 nrgGain[k] = fMult(nrgGain[k],compensation); 403 nrgGain_e[k] = nrgGain_e[k] + compensation_e; 404 } 405 } 406 } 407 408 409 /* Convert headroom bits to exponent */ 410 #define SCALE2EXP(s) (15-(s)) 411 #define EXP2SCALE(e) (15-(e)) 412 413 /*! 414 \brief Apply spectral envelope to subband samples 415 416 This function is called from sbr_dec.cpp in each frame. 417 418 To enhance accuracy and due to the usage of tables for squareroots and 419 inverse, some calculations are performed with the operands being split 420 into mantissa and exponent. The variable names in the source code carry 421 the suffixes <em>_m</em> and <em>_e</em> respectively. The control data 422 in #hFrameData containts envelope data which is represented by this format but 423 stored in single words. (See requantizeEnvelopeData() for details). This data 424 is unpacked within calculateSbrEnvelope() to follow the described suffix convention. 425 426 The actual value (comparable to the corresponding float-variable in the 427 research-implementation) of a mantissa/exponent-pair can be calculated as 428 429 \f$ value = value\_m * 2^{value\_e} \f$ 430 431 All energies and noise levels decoded from the bitstream suit for an 432 original signal magnitude of \f$\pm 32768 \f$ rather than \f$ \pm 1\f$. Therefore, 433 the scale factor <em>hb_scale</em> passed into this function will be converted 434 to an 'input exponent' (#input_e), which fits the internal representation. 435 436 Before the actual processing, an exponent #adj_e for resulting adjusted 437 samples is derived from the maximum reference energy. 438 439 Then, for each envelope, the following steps are performed: 440 441 \li Calculate energy in the signal to be adjusted. Depending on the the value of 442 #interpolFreq (interpolation mode), this is either done seperately 443 for each QMF-subband or for each SBR-band. 444 The resulting energies are stored in #nrgEst_m[#MAX_FREQ_COEFFS] (mantissas) 445 and #nrgEst_e[#MAX_FREQ_COEFFS] (exponents). 446 \li Calculate gain and noise level for each subband:<br> 447 \f$ gain = \sqrt{ \frac{nrgRef}{nrgEst} \cdot (1 - noiseRatio) } 448 \hspace{2cm} 449 noise = \sqrt{ nrgRef \cdot noiseRatio } 450 \f$<br> 451 where <em>noiseRatio</em> and <em>nrgRef</em> are extracted from the 452 bitstream and <em>nrgEst</em> is the subband energy before adjustment. 453 The resulting gains are stored in #nrgGain_m[#MAX_FREQ_COEFFS] 454 (mantissas) and #nrgGain_e[#MAX_FREQ_COEFFS] (exponents), the noise levels 455 are stored in #noiseLevel_m[#MAX_FREQ_COEFFS] and #noiseLevel_e[#MAX_FREQ_COEFFS] 456 (exponents). 457 The sine levels are stored in #nrgSine_m[#MAX_FREQ_COEFFS] 458 and #nrgSine_e[#MAX_FREQ_COEFFS]. 459 \li Noise limiting: The gain for each subband is limited both absolutely 460 and relatively compared to the total gain over all subbands. 461 \li Boost gain: Calculate and apply boost factor for each limiter band 462 in order to compensate for the energy loss imposed by the limiting. 463 \li Apply gains and add noise: The gains and noise levels are applied 464 to all timeslots of the current envelope. A short FIR-filter (length 4 465 QMF-timeslots) can be used to smooth the sudden change at the envelope borders. 466 Each complex subband sample of the current timeslot is multiplied by the 467 smoothed gain, then random noise with the calculated level is added. 468 469 \note 470 To reduce the stack size, some of the local arrays could be located within 471 the time output buffer. Of the 512 samples temporarily available there, 472 about half the size is already used by #SBR_FRAME_DATA. A pointer to the 473 remaining free memory could be supplied by an additional argument to calculateSbrEnvelope() 474 in sbr_dec: 475 476 \par 477 \code 478 calculateSbrEnvelope (&hSbrDec->sbrScaleFactor, 479 &hSbrDec->SbrCalculateEnvelope, 480 hHeaderData, 481 hFrameData, 482 QmfBufferReal, 483 QmfBufferImag, 484 timeOutPtr + sizeof(SBR_FRAME_DATA)/sizeof(Float) + 1); 485 \endcode 486 487 \par 488 Within calculateSbrEnvelope(), some pointers could be defined instead of the arrays 489 #nrgRef_m, #nrgRef_e, #nrgEst_m, #nrgEst_e, #noiseLevel_m: 490 491 \par 492 \code 493 fract* nrgRef_m = timeOutPtr; 494 SCHAR* nrgRef_e = nrgRef_m + MAX_FREQ_COEFFS; 495 fract* nrgEst_m = nrgRef_e + MAX_FREQ_COEFFS; 496 SCHAR* nrgEst_e = nrgEst_m + MAX_FREQ_COEFFS; 497 fract* noiseLevel_m = nrgEst_e + MAX_FREQ_COEFFS; 498 \endcode 499 500 <br> 501 */ 502 void 503 calculateSbrEnvelope (QMF_SCALE_FACTOR *sbrScaleFactor, /*!< Scaling factors */ 504 HANDLE_SBR_CALCULATE_ENVELOPE h_sbr_cal_env, /*!< Handle to struct filled by the create-function */ 505 HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Static control data */ 506 HANDLE_SBR_FRAME_DATA hFrameData, /*!< Control data of current frame */ 507 FIXP_DBL **analysBufferReal, /*!< Real part of subband samples to be processed */ 508 FIXP_DBL **analysBufferImag, /*!< Imag part of subband samples to be processed */ 509 const int useLP, 510 FIXP_DBL *degreeAlias, /*!< Estimated aliasing for each QMF channel */ 511 const UINT flags, 512 const int frameErrorFlag 513 ) 514 { 515 int c, i, j, envNoise = 0; 516 UCHAR* borders = hFrameData->frameInfo.borders; 517 518 FIXP_SGL *noiseLevels = hFrameData->sbrNoiseFloorLevel; 519 HANDLE_FREQ_BAND_DATA hFreq = &hHeaderData->freqBandData; 520 521 int lowSubband = hFreq->lowSubband; 522 int highSubband = hFreq->highSubband; 523 int noSubbands = highSubband - lowSubband; 524 525 int noNoiseBands = hFreq->nNfb; 526 int no_cols = hHeaderData->numberTimeSlots * hHeaderData->timeStep; 527 UCHAR first_start = borders[0] * hHeaderData->timeStep; 528 529 SCHAR sineMapped[MAX_FREQ_COEFFS]; 530 SCHAR ov_adj_e = SCALE2EXP(sbrScaleFactor->ov_hb_scale); 531 SCHAR adj_e = 0; 532 SCHAR output_e; 533 SCHAR final_e = 0; 534 535 SCHAR maxGainLimit_e = (frameErrorFlag) ? MAX_GAIN_CONCEAL_EXP : MAX_GAIN_EXP; 536 537 int useAliasReduction[64]; 538 UCHAR smooth_length = 0; 539 540 FIXP_SGL * pIenv = hFrameData->iEnvelope; 541 542 /* 543 Extract sine flags for all QMF bands 544 */ 545 mapSineFlags(hFreq->freqBandTable[1], 546 hFreq->nSfb[1], 547 hFrameData->addHarmonics, 548 h_sbr_cal_env->harmFlagsPrev, 549 hFrameData->frameInfo.tranEnv, 550 sineMapped); 551 552 553 /* 554 Scan for maximum in bufferd noise levels. 555 This is needed in case that we had strong noise in the previous frame 556 which is smoothed into the current frame. 557 The resulting exponent is used as start value for the maximum search 558 in reference energies 559 */ 560 if (!useLP) 561 adj_e = h_sbr_cal_env->filtBufferNoise_e - getScalefactor(h_sbr_cal_env->filtBufferNoise, noSubbands); 562 563 /* 564 Scan for maximum reference energy to be able 565 to select appropriate values for adj_e and final_e. 566 */ 567 568 for (i = 0; i < hFrameData->frameInfo.nEnvelopes; i++) { 569 INT maxSfbNrg_e = -FRACT_BITS+NRG_EXP_OFFSET; /* start value for maximum search */ 570 571 /* Fetch frequency resolution for current envelope: */ 572 for (j=hFreq->nSfb[hFrameData->frameInfo.freqRes[i]]; j!=0; j--) { 573 maxSfbNrg_e = fixMax(maxSfbNrg_e,(INT)((LONG)(*pIenv++) & MASK_E)); 574 } 575 maxSfbNrg_e -= NRG_EXP_OFFSET; 576 577 /* Energy -> magnitude (sqrt halfens exponent) */ 578 maxSfbNrg_e = (maxSfbNrg_e+1) >> 1; /* +1 to go safe (round to next higher int) */ 579 580 /* Some safety margin is needed for 2 reasons: 581 - The signal energy is not equally spread over all subband samples in 582 a specific sfb of an envelope (Nrg could be too high by a factor of 583 envWidth * sfbWidth) 584 - Smoothing can smear high gains of the previous envelope into the current 585 */ 586 maxSfbNrg_e += 6; 587 588 if (borders[i] < hHeaderData->numberTimeSlots) 589 /* This envelope affects timeslots that belong to the output frame */ 590 adj_e = (maxSfbNrg_e > adj_e) ? maxSfbNrg_e : adj_e; 591 592 if (borders[i+1] > hHeaderData->numberTimeSlots) 593 /* This envelope affects timeslots after the output frame */ 594 final_e = (maxSfbNrg_e > final_e) ? maxSfbNrg_e : final_e; 595 596 } 597 598 /* 599 Calculate adjustment factors and apply them for every envelope. 600 */ 601 pIenv = hFrameData->iEnvelope; 602 603 for (i = 0; i < hFrameData->frameInfo.nEnvelopes; i++) { 604 605 int k, noNoiseFlag; 606 SCHAR noise_e, input_e = SCALE2EXP(sbrScaleFactor->hb_scale); 607 C_ALLOC_SCRATCH_START(pNrgs, ENV_CALC_NRGS, 1); 608 609 /* 610 Helper variables. 611 */ 612 UCHAR start_pos = hHeaderData->timeStep * borders[i]; /* Start-position in time (subband sample) for current envelope. */ 613 UCHAR stop_pos = hHeaderData->timeStep * borders[i+1]; /* Stop-position in time (subband sample) for current envelope. */ 614 UCHAR freq_res = hFrameData->frameInfo.freqRes[i]; /* Frequency resolution for current envelope. */ 615 616 617 /* Always do fully initialize the temporary energy table. This prevents negative energies and extreme gain factors in 618 cases where the number of limiter bands exceeds the number of subbands. The latter can be caused by undetected bit 619 errors and is tested by some streams from the certification set. */ 620 FDKmemclear(pNrgs, sizeof(ENV_CALC_NRGS)); 621 622 /* If the start-pos of the current envelope equals the stop pos of the current 623 noise envelope, increase the pointer (i.e. choose the next noise-floor).*/ 624 if (borders[i] == hFrameData->frameInfo.bordersNoise[envNoise+1]){ 625 noiseLevels += noNoiseBands; /* The noise floor data is stored in a row [noiseFloor1 noiseFloor2...].*/ 626 envNoise++; 627 } 628 629 if(i==hFrameData->frameInfo.tranEnv || i==h_sbr_cal_env->prevTranEnv) /* attack */ 630 { 631 noNoiseFlag = 1; 632 if (!useLP) 633 smooth_length = 0; /* No smoothing on attacks! */ 634 } 635 else { 636 noNoiseFlag = 0; 637 if (!useLP) 638 smooth_length = (1 - hHeaderData->bs_data.smoothingLength) << 2; /* can become either 0 or 4 */ 639 } 640 641 642 /* 643 Energy estimation in transposed highband. 644 */ 645 if (hHeaderData->bs_data.interpolFreq) 646 calcNrgPerSubband(analysBufferReal, 647 (useLP) ? NULL : analysBufferImag, 648 lowSubband, highSubband, 649 start_pos, stop_pos, 650 input_e, 651 pNrgs->nrgEst, 652 pNrgs->nrgEst_e); 653 else 654 calcNrgPerSfb(analysBufferReal, 655 (useLP) ? NULL : analysBufferImag, 656 hFreq->nSfb[freq_res], 657 hFreq->freqBandTable[freq_res], 658 start_pos, stop_pos, 659 input_e, 660 pNrgs->nrgEst, 661 pNrgs->nrgEst_e); 662 663 /* 664 Calculate subband gains 665 */ 666 { 667 UCHAR * table = hFreq->freqBandTable[freq_res]; 668 UCHAR * pUiNoise = &hFreq->freqBandTableNoise[1]; /*! Upper limit of the current noise floor band. */ 669 670 FIXP_SGL * pNoiseLevels = noiseLevels; 671 672 FIXP_DBL tmpNoise = FX_SGL2FX_DBL((FIXP_SGL)((LONG)(*pNoiseLevels) & MASK_M)); 673 SCHAR tmpNoise_e = (UCHAR)((LONG)(*pNoiseLevels++) & MASK_E) - NOISE_EXP_OFFSET; 674 675 int cc = 0; 676 c = 0; 677 for (j = 0; j < hFreq->nSfb[freq_res]; j++) { 678 679 FIXP_DBL refNrg = FX_SGL2FX_DBL((FIXP_SGL)((LONG)(*pIenv) & MASK_M)); 680 SCHAR refNrg_e = (SCHAR)((LONG)(*pIenv) & MASK_E) - NRG_EXP_OFFSET; 681 682 UCHAR sinePresentFlag = 0; 683 int li = table[j]; 684 int ui = table[j+1]; 685 686 for (k=li; k<ui; k++) { 687 sinePresentFlag |= (i >= sineMapped[cc]); 688 cc++; 689 } 690 691 for (k=li; k<ui; k++) { 692 if (k >= *pUiNoise) { 693 tmpNoise = FX_SGL2FX_DBL((FIXP_SGL)((LONG)(*pNoiseLevels) & MASK_M)); 694 tmpNoise_e = (SCHAR)((LONG)(*pNoiseLevels++) & MASK_E) - NOISE_EXP_OFFSET; 695 696 pUiNoise++; 697 } 698 699 FDK_ASSERT(k >= lowSubband); 700 701 if (useLP) 702 useAliasReduction[k-lowSubband] = !sinePresentFlag; 703 704 pNrgs->nrgSine[c] = FL2FXCONST_DBL(0.0f); 705 pNrgs->nrgSine_e[c] = 0; 706 707 calcSubbandGain(refNrg, refNrg_e, pNrgs, c, 708 tmpNoise, tmpNoise_e, 709 sinePresentFlag, i >= sineMapped[c], 710 noNoiseFlag); 711 712 pNrgs->nrgRef[c] = refNrg; 713 pNrgs->nrgRef_e[c] = refNrg_e; 714 715 c++; 716 } 717 pIenv++; 718 } 719 } 720 721 /* 722 Noise limiting 723 */ 724 725 for (c = 0; c < hFreq->noLimiterBands; c++) { 726 727 FIXP_DBL sumRef, boostGain, maxGain; 728 FIXP_DBL accu = FL2FXCONST_DBL(0.0f); 729 SCHAR sumRef_e, boostGain_e, maxGain_e, accu_e = 0; 730 731 calcAvgGain(pNrgs, 732 hFreq->limiterBandTable[c], hFreq->limiterBandTable[c+1], 733 &sumRef, &sumRef_e, 734 &maxGain, &maxGain_e); 735 736 /* Multiply maxGain with limiterGain: */ 737 maxGain = fMult(maxGain, FDK_sbrDecoder_sbr_limGains_m[hHeaderData->bs_data.limiterGains]); 738 maxGain_e += FDK_sbrDecoder_sbr_limGains_e[hHeaderData->bs_data.limiterGains]; 739 740 /* Scale mantissa of MaxGain into range between 0.5 and 1: */ 741 if (maxGain == FL2FXCONST_DBL(0.0f)) 742 maxGain_e = -FRACT_BITS; 743 else { 744 SCHAR charTemp = CountLeadingBits(maxGain); 745 maxGain_e -= charTemp; 746 maxGain <<= (int)charTemp; 747 } 748 749 if (maxGain_e >= maxGainLimit_e) { /* upper limit (e.g. 96 dB) */ 750 maxGain = FL2FXCONST_DBL(0.5f); 751 maxGain_e = maxGainLimit_e; 752 } 753 754 755 /* Every subband gain is compared to the scaled "average gain" 756 and limited if necessary: */ 757 for (k = hFreq->limiterBandTable[c]; k < hFreq->limiterBandTable[c+1]; k++) { 758 if ( (pNrgs->nrgGain_e[k] > maxGain_e) || (pNrgs->nrgGain_e[k] == maxGain_e && pNrgs->nrgGain[k]>maxGain) ) { 759 760 FIXP_DBL noiseAmp; 761 SCHAR noiseAmp_e; 762 763 FDK_divide_MantExp(maxGain, maxGain_e, pNrgs->nrgGain[k], pNrgs->nrgGain_e[k], &noiseAmp, &noiseAmp_e); 764 pNrgs->noiseLevel[k] = fMult(pNrgs->noiseLevel[k],noiseAmp); 765 pNrgs->noiseLevel_e[k] += noiseAmp_e; 766 pNrgs->nrgGain[k] = maxGain; 767 pNrgs->nrgGain_e[k] = maxGain_e; 768 } 769 } 770 771 /* -- Boost gain 772 Calculate and apply boost factor for each limiter band: 773 1. Check how much energy would be present when using the limited gain 774 2. Calculate boost factor by comparison with reference energy 775 3. Apply boost factor to compensate for the energy loss due to limiting 776 */ 777 for (k = hFreq->limiterBandTable[c]; k < hFreq->limiterBandTable[c + 1]; k++) { 778 779 /* 1.a Add energy of adjusted signal (using preliminary gain) */ 780 FIXP_DBL tmp = fMult(pNrgs->nrgGain[k],pNrgs->nrgEst[k]); 781 SCHAR tmp_e = pNrgs->nrgGain_e[k] + pNrgs->nrgEst_e[k]; 782 FDK_add_MantExp(tmp, tmp_e, accu, accu_e, &accu, &accu_e); 783 784 /* 1.b Add sine energy (if present) */ 785 if(pNrgs->nrgSine[k] != FL2FXCONST_DBL(0.0f)) { 786 FDK_add_MantExp(pNrgs->nrgSine[k], pNrgs->nrgSine_e[k], accu, accu_e, &accu, &accu_e); 787 } 788 else { 789 /* 1.c Add noise energy (if present) */ 790 if(noNoiseFlag == 0) { 791 FDK_add_MantExp(pNrgs->noiseLevel[k], pNrgs->noiseLevel_e[k], accu, accu_e, &accu, &accu_e); 792 } 793 } 794 } 795 796 /* 2.a Calculate ratio of wanted energy and accumulated energy */ 797 if (accu == (FIXP_DBL)0) { /* If divisor is 0, limit quotient to +4 dB */ 798 boostGain = FL2FXCONST_DBL(0.6279716f); 799 boostGain_e = 2; 800 } else { 801 INT div_e; 802 boostGain = fDivNorm(sumRef, accu, &div_e); 803 boostGain_e = sumRef_e - accu_e + div_e; 804 } 805 806 807 /* 2.b Result too high? --> Limit the boost factor to +4 dB */ 808 if((boostGain_e > 3) || 809 (boostGain_e == 2 && boostGain > FL2FXCONST_DBL(0.6279716f)) || 810 (boostGain_e == 3 && boostGain > FL2FXCONST_DBL(0.3139858f)) ) 811 { 812 boostGain = FL2FXCONST_DBL(0.6279716f); 813 boostGain_e = 2; 814 } 815 /* 3. Multiply all signal components with the boost factor */ 816 for (k = hFreq->limiterBandTable[c]; k < hFreq->limiterBandTable[c + 1]; k++) { 817 pNrgs->nrgGain[k] = fMultDiv2(pNrgs->nrgGain[k],boostGain); 818 pNrgs->nrgGain_e[k] = pNrgs->nrgGain_e[k] + boostGain_e + 1; 819 820 pNrgs->nrgSine[k] = fMultDiv2(pNrgs->nrgSine[k],boostGain); 821 pNrgs->nrgSine_e[k] = pNrgs->nrgSine_e[k] + boostGain_e + 1; 822 823 pNrgs->noiseLevel[k] = fMultDiv2(pNrgs->noiseLevel[k],boostGain); 824 pNrgs->noiseLevel_e[k] = pNrgs->noiseLevel_e[k] + boostGain_e + 1; 825 } 826 } 827 /* End of noise limiting */ 828 829 if (useLP) 830 aliasingReduction(degreeAlias+lowSubband, 831 pNrgs, 832 useAliasReduction, 833 noSubbands); 834 835 /* For the timeslots within the range for the output frame, 836 use the same scale for the noise levels. 837 Drawback: If the envelope exceeds the frame border, the noise levels 838 will have to be rescaled later to fit final_e of 839 the gain-values. 840 */ 841 noise_e = (start_pos < no_cols) ? adj_e : final_e; 842 843 /* 844 Convert energies to amplitude levels 845 */ 846 for (k=0; k<noSubbands; k++) { 847 FDK_sqrt_MantExp(&pNrgs->nrgSine[k], &pNrgs->nrgSine_e[k], &noise_e); 848 FDK_sqrt_MantExp(&pNrgs->nrgGain[k], &pNrgs->nrgGain_e[k], &pNrgs->nrgGain_e[k]); 849 FDK_sqrt_MantExp(&pNrgs->noiseLevel[k], &pNrgs->noiseLevel_e[k], &noise_e); 850 } 851 852 853 854 /* 855 Apply calculated gains and adaptive noise 856 */ 857 858 /* assembleHfSignals() */ 859 { 860 int scale_change, sc_change; 861 FIXP_SGL smooth_ratio; 862 int filtBufferNoiseShift=0; 863 864 /* Initialize smoothing buffers with the first valid values */ 865 if (h_sbr_cal_env->startUp) 866 { 867 if (!useLP) { 868 h_sbr_cal_env->filtBufferNoise_e = noise_e; 869 870 FDKmemcpy(h_sbr_cal_env->filtBuffer_e, pNrgs->nrgGain_e, noSubbands*sizeof(SCHAR)); 871 FDKmemcpy(h_sbr_cal_env->filtBufferNoise, pNrgs->noiseLevel, noSubbands*sizeof(FIXP_DBL)); 872 FDKmemcpy(h_sbr_cal_env->filtBuffer, pNrgs->nrgGain, noSubbands*sizeof(FIXP_DBL)); 873 874 } 875 h_sbr_cal_env->startUp = 0; 876 } 877 878 if (!useLP) { 879 880 equalizeFiltBufferExp(h_sbr_cal_env->filtBuffer, /* buffered */ 881 h_sbr_cal_env->filtBuffer_e, /* buffered */ 882 pNrgs->nrgGain, /* current */ 883 pNrgs->nrgGain_e, /* current */ 884 noSubbands); 885 886 /* Adapt exponent of buffered noise levels to the current exponent 887 so they can easily be smoothed */ 888 if((h_sbr_cal_env->filtBufferNoise_e - noise_e)>=0) { 889 int shift = fixMin(DFRACT_BITS-1,(int)(h_sbr_cal_env->filtBufferNoise_e - noise_e)); 890 for (k=0; k<noSubbands; k++) 891 h_sbr_cal_env->filtBufferNoise[k] <<= shift; 892 } 893 else { 894 int shift = fixMin(DFRACT_BITS-1,-(int)(h_sbr_cal_env->filtBufferNoise_e - noise_e)); 895 for (k=0; k<noSubbands; k++) 896 h_sbr_cal_env->filtBufferNoise[k] >>= shift; 897 } 898 899 h_sbr_cal_env->filtBufferNoise_e = noise_e; 900 } 901 902 /* find best scaling! */ 903 scale_change = -(DFRACT_BITS-1); 904 for(k=0;k<noSubbands;k++) { 905 scale_change = fixMax(scale_change,(int)pNrgs->nrgGain_e[k]); 906 } 907 sc_change = (start_pos<no_cols)? adj_e - input_e : final_e - input_e; 908 909 if ((scale_change-sc_change+1)<0) 910 scale_change-=(scale_change-sc_change+1); 911 912 scale_change = (scale_change-sc_change)+1; 913 914 for(k=0;k<noSubbands;k++) { 915 int sc = scale_change-pNrgs->nrgGain_e[k] + (sc_change-1); 916 pNrgs->nrgGain[k] >>= sc; 917 pNrgs->nrgGain_e[k] += sc; 918 } 919 920 if (!useLP) { 921 for(k=0;k<noSubbands;k++) { 922 int sc = scale_change-h_sbr_cal_env->filtBuffer_e[k] + (sc_change-1); 923 h_sbr_cal_env->filtBuffer[k] >>= sc; 924 } 925 } 926 927 for (j = start_pos; j < stop_pos; j++) 928 { 929 /* This timeslot is located within the first part of the processing buffer 930 and will be fed into the QMF-synthesis for the current frame. 931 adj_e - input_e 932 This timeslot will not yet be fed into the QMF so we do not care 933 about the adj_e. 934 sc_change = final_e - input_e 935 */ 936 if ( (j==no_cols) && (start_pos<no_cols) ) 937 { 938 int shift = (int) (noise_e - final_e); 939 if (!useLP) 940 filtBufferNoiseShift = shift; /* shifting of h_sbr_cal_env->filtBufferNoise[k] will be applied in function adjustTimeSlotHQ() */ 941 if (shift>=0) { 942 shift = fixMin(DFRACT_BITS-1,shift); 943 for (k=0; k<noSubbands; k++) { 944 pNrgs->nrgSine[k] <<= shift; 945 pNrgs->noiseLevel[k] <<= shift; 946 /* 947 if (!useLP) 948 h_sbr_cal_env->filtBufferNoise[k] <<= shift; 949 */ 950 } 951 } 952 else { 953 shift = fixMin(DFRACT_BITS-1,-shift); 954 for (k=0; k<noSubbands; k++) { 955 pNrgs->nrgSine[k] >>= shift; 956 pNrgs->noiseLevel[k] >>= shift; 957 /* 958 if (!useLP) 959 h_sbr_cal_env->filtBufferNoise[k] >>= shift; 960 */ 961 } 962 } 963 964 /* update noise scaling */ 965 noise_e = final_e; 966 if (!useLP) 967 h_sbr_cal_env->filtBufferNoise_e = noise_e; /* scaling value unused! */ 968 969 /* update gain buffer*/ 970 sc_change -= (final_e - input_e); 971 972 if (sc_change<0) { 973 for(k=0;k<noSubbands;k++) { 974 pNrgs->nrgGain[k] >>= -sc_change; 975 pNrgs->nrgGain_e[k] += -sc_change; 976 } 977 if (!useLP) { 978 for(k=0;k<noSubbands;k++) { 979 h_sbr_cal_env->filtBuffer[k] >>= -sc_change; 980 } 981 } 982 } else { 983 scale_change+=sc_change; 984 } 985 986 } // if 987 988 if (!useLP) { 989 990 /* Prevent the smoothing filter from running on constant levels */ 991 if (j-start_pos < smooth_length) 992 smooth_ratio = FDK_sbrDecoder_sbr_smoothFilter[j-start_pos]; 993 994 else 995 smooth_ratio = FL2FXCONST_SGL(0.0f); 996 997 adjustTimeSlotHQ(&analysBufferReal[j][lowSubband], 998 &analysBufferImag[j][lowSubband], 999 h_sbr_cal_env, 1000 pNrgs, 1001 lowSubband, 1002 noSubbands, 1003 scale_change, 1004 smooth_ratio, 1005 noNoiseFlag, 1006 filtBufferNoiseShift); 1007 } 1008 else 1009 { 1010 adjustTimeSlotLC(&analysBufferReal[j][lowSubband], 1011 pNrgs, 1012 &h_sbr_cal_env->harmIndex, 1013 lowSubband, 1014 noSubbands, 1015 scale_change, 1016 noNoiseFlag, 1017 &h_sbr_cal_env->phaseIndex, 1018 (flags & SBRDEC_ELD_GRID)); 1019 } 1020 } // for 1021 1022 if (!useLP) { 1023 /* Update time-smoothing-buffers for gains and noise levels 1024 The gains and the noise values of the current envelope are copied into the buffer. 1025 This has to be done at the end of each envelope as the values are required for 1026 a smooth transition to the next envelope. */ 1027 FDKmemcpy(h_sbr_cal_env->filtBuffer, pNrgs->nrgGain, noSubbands*sizeof(FIXP_DBL)); 1028 FDKmemcpy(h_sbr_cal_env->filtBuffer_e, pNrgs->nrgGain_e, noSubbands*sizeof(SCHAR)); 1029 FDKmemcpy(h_sbr_cal_env->filtBufferNoise, pNrgs->noiseLevel, noSubbands*sizeof(FIXP_DBL)); 1030 } 1031 1032 } 1033 C_ALLOC_SCRATCH_END(pNrgs, ENV_CALC_NRGS, 1); 1034 } 1035 1036 /* Rescale output samples */ 1037 { 1038 FIXP_DBL maxVal; 1039 int ov_reserve, reserve; 1040 1041 /* Determine headroom in old adjusted samples */ 1042 maxVal = maxSubbandSample( analysBufferReal, 1043 (useLP) ? NULL : analysBufferImag, 1044 lowSubband, 1045 highSubband, 1046 0, 1047 first_start); 1048 1049 ov_reserve = fNorm(maxVal); 1050 1051 /* Determine headroom in new adjusted samples */ 1052 maxVal = maxSubbandSample( analysBufferReal, 1053 (useLP) ? NULL : analysBufferImag, 1054 lowSubband, 1055 highSubband, 1056 first_start, 1057 no_cols); 1058 1059 reserve = fNorm(maxVal); 1060 1061 /* Determine common output exponent */ 1062 if (ov_adj_e - ov_reserve > adj_e - reserve ) /* set output_e to the maximum */ 1063 output_e = ov_adj_e - ov_reserve; 1064 else 1065 output_e = adj_e - reserve; 1066 1067 /* Rescale old samples */ 1068 rescaleSubbandSamples( analysBufferReal, 1069 (useLP) ? NULL : analysBufferImag, 1070 lowSubband, highSubband, 1071 0, first_start, 1072 ov_adj_e - output_e); 1073 1074 /* Rescale new samples */ 1075 rescaleSubbandSamples( analysBufferReal, 1076 (useLP) ? NULL : analysBufferImag, 1077 lowSubband, highSubband, 1078 first_start, no_cols, 1079 adj_e - output_e); 1080 } 1081 1082 /* Update hb_scale */ 1083 sbrScaleFactor->hb_scale = EXP2SCALE(output_e); 1084 1085 /* Save the current final exponent for the next frame: */ 1086 sbrScaleFactor->ov_hb_scale = EXP2SCALE(final_e); 1087 1088 1089 /* We need to remeber to the next frame that the transient 1090 will occur in the first envelope (if tranEnv == nEnvelopes). */ 1091 if(hFrameData->frameInfo.tranEnv == hFrameData->frameInfo.nEnvelopes) 1092 h_sbr_cal_env->prevTranEnv = 0; 1093 else 1094 h_sbr_cal_env->prevTranEnv = -1; 1095 1096 } 1097 1098 1099 /*! 1100 \brief Create envelope instance 1101 1102 Must be called once for each channel before calculateSbrEnvelope() can be used. 1103 1104 \return errorCode, 0 if successful 1105 */ 1106 SBR_ERROR 1107 createSbrEnvelopeCalc (HANDLE_SBR_CALCULATE_ENVELOPE hs, /*!< pointer to envelope instance */ 1108 HANDLE_SBR_HEADER_DATA hHeaderData, /*!< static SBR control data, initialized with defaults */ 1109 const int chan, /*!< Channel for which to assign buffers */ 1110 const UINT flags) 1111 { 1112 SBR_ERROR err = SBRDEC_OK; 1113 int i; 1114 1115 /* Clear previous missing harmonics flags */ 1116 for (i=0; i<(MAX_FREQ_COEFFS+15)>>4; i++) { 1117 hs->harmFlagsPrev[i] = 0; 1118 } 1119 hs->harmIndex = 0; 1120 1121 /* 1122 Setup pointers for time smoothing. 1123 The buffer itself will be initialized later triggered by the startUp-flag. 1124 */ 1125 hs->prevTranEnv = -1; 1126 1127 1128 /* initialization */ 1129 resetSbrEnvelopeCalc(hs); 1130 1131 if (chan==0) { /* do this only once */ 1132 err = resetFreqBandTables(hHeaderData, flags); 1133 } 1134 1135 return err; 1136 } 1137 1138 /*! 1139 \brief Create envelope instance 1140 1141 Must be called once for each channel before calculateSbrEnvelope() can be used. 1142 1143 \return errorCode, 0 if successful 1144 */ 1145 int 1146 deleteSbrEnvelopeCalc (HANDLE_SBR_CALCULATE_ENVELOPE hs) 1147 { 1148 return 0; 1149 } 1150 1151 1152 /*! 1153 \brief Reset envelope instance 1154 1155 This function must be called for each channel on a change of configuration. 1156 Note that resetFreqBandTables should also be called in this case. 1157 1158 \return errorCode, 0 if successful 1159 */ 1160 void 1161 resetSbrEnvelopeCalc (HANDLE_SBR_CALCULATE_ENVELOPE hCalEnv) /*!< pointer to envelope instance */ 1162 { 1163 hCalEnv->phaseIndex = 0; 1164 1165 /* Noise exponent needs to be reset because the output exponent for the next frame depends on it */ 1166 hCalEnv->filtBufferNoise_e = 0; 1167 1168 hCalEnv->startUp = 1; 1169 } 1170 1171 1172 /*! 1173 \brief Equalize exponents of the buffered gain values and the new ones 1174 1175 After equalization of exponents, the FIR-filter addition for smoothing 1176 can be performed. 1177 This function is called once for each envelope before adjusting. 1178 */ 1179 /*static*/ void equalizeFiltBufferExp(FIXP_DBL *filtBuffer, /*!< bufferd gains */ 1180 SCHAR *filtBuffer_e, /*!< exponents of bufferd gains */ 1181 FIXP_DBL *nrgGain, /*!< gains for current envelope */ 1182 SCHAR *nrgGain_e, /*!< exponents of gains for current envelope */ 1183 int subbands) /*!< Number of QMF subbands */ 1184 { 1185 int band; 1186 int diff; 1187 1188 for (band=0; band<subbands; band++){ 1189 diff = (int) (nrgGain_e[band] - filtBuffer_e[band]); 1190 if (diff>0) { 1191 filtBuffer[band] >>= diff; /* Compensate for the scale change by shifting the mantissa. */ 1192 filtBuffer_e[band] += diff; /* New gain is bigger, use its exponent */ 1193 } 1194 else if (diff<0) { 1195 /* The buffered gains seem to be larger, but maybe there 1196 are some unused bits left in the mantissa */ 1197 1198 int reserve = CntLeadingZeros(fixp_abs(filtBuffer[band]))-1; 1199 1200 if ((-diff) <= reserve) { 1201 /* There is enough space in the buffered mantissa so 1202 that we can take the new exponent as common. 1203 */ 1204 filtBuffer[band] <<= (-diff); 1205 filtBuffer_e[band] += diff; /* becomes equal to *ptrNewExp */ 1206 } 1207 else { 1208 filtBuffer[band] <<= reserve; /* Shift the mantissa as far as possible: */ 1209 filtBuffer_e[band] -= reserve; /* Compensate in the exponent: */ 1210 1211 /* For the remaining difference, change the new gain value */ 1212 diff = fixMin(-(reserve + diff),DFRACT_BITS-1); 1213 nrgGain[band] >>= diff; 1214 nrgGain_e[band] += diff; 1215 } 1216 } 1217 } 1218 } 1219 1220 /*! 1221 \brief Shift left the mantissas of all subband samples 1222 in the giventime and frequency range by the specified number of bits. 1223 1224 This function is used to rescale the audio data in the overlap buffer 1225 which has already been envelope adjusted with the last frame. 1226 */ 1227 void rescaleSubbandSamples(FIXP_DBL ** re, /*!< Real part of input and output subband samples */ 1228 FIXP_DBL ** im, /*!< Imaginary part of input and output subband samples */ 1229 int lowSubband, /*!< Begin of frequency range to process */ 1230 int highSubband, /*!< End of frequency range to process */ 1231 int start_pos, /*!< Begin of time rage (QMF-timeslot) */ 1232 int next_pos, /*!< End of time rage (QMF-timeslot) */ 1233 int shift) /*!< number of bits to shift */ 1234 { 1235 int width = highSubband-lowSubband; 1236 1237 if ( (width > 0) && (shift!=0) ) { 1238 if (im!=NULL) { 1239 for (int l=start_pos; l<next_pos; l++) { 1240 scaleValues(&re[l][lowSubband], width, shift); 1241 scaleValues(&im[l][lowSubband], width, shift); 1242 } 1243 } else 1244 { 1245 for (int l=start_pos; l<next_pos; l++) { 1246 scaleValues(&re[l][lowSubband], width, shift); 1247 } 1248 } 1249 } 1250 } 1251 1252 1253 /*! 1254 \brief Determine headroom for shifting 1255 1256 Determine by how much the spectrum can be shifted left 1257 for better accuracy in later processing. 1258 1259 \return Number of free bits in the biggest spectral value 1260 */ 1261 1262 FIXP_DBL maxSubbandSample( FIXP_DBL ** re, /*!< Real part of input and output subband samples */ 1263 FIXP_DBL ** im, /*!< Real part of input and output subband samples */ 1264 int lowSubband, /*!< Begin of frequency range to process */ 1265 int highSubband, /*!< Number of QMF bands to process */ 1266 int start_pos, /*!< Begin of time rage (QMF-timeslot) */ 1267 int next_pos /*!< End of time rage (QMF-timeslot) */ 1268 ) 1269 { 1270 FIXP_DBL maxVal = FL2FX_DBL(0.0f); 1271 unsigned int width = highSubband - lowSubband; 1272 1273 FDK_ASSERT(width <= (64)); 1274 1275 if ( width > 0 ) { 1276 if (im!=NULL) 1277 { 1278 for (int l=start_pos; l<next_pos; l++) 1279 { 1280 #ifdef FUNCTION_FDK_get_maxval 1281 maxVal = FDK_get_maxval(maxVal, &re[l][lowSubband], &im[l][lowSubband], width); 1282 #else 1283 int k=width; 1284 FIXP_DBL *reTmp = &re[l][lowSubband]; 1285 FIXP_DBL *imTmp = &im[l][lowSubband]; 1286 do{ 1287 FIXP_DBL tmp1 = *(reTmp++); 1288 FIXP_DBL tmp2 = *(imTmp++); 1289 maxVal |= (FIXP_DBL)((LONG)(tmp1)^((LONG)tmp1>>(DFRACT_BITS-1))); 1290 maxVal |= (FIXP_DBL)((LONG)(tmp2)^((LONG)tmp2>>(DFRACT_BITS-1))); 1291 } while(--k!=0); 1292 #endif 1293 } 1294 } else 1295 { 1296 for (int l=start_pos; l<next_pos; l++) { 1297 int k=width; 1298 FIXP_DBL *reTmp = &re[l][lowSubband]; 1299 do{ 1300 FIXP_DBL tmp = *(reTmp++); 1301 maxVal |= (FIXP_DBL)((LONG)(tmp)^((LONG)tmp>>(DFRACT_BITS-1))); 1302 }while(--k!=0); 1303 } 1304 } 1305 } 1306 1307 return(maxVal); 1308 } 1309 1310 #define SHIFT_BEFORE_SQUARE (3) /* (7/2) */ 1311 /*!< 1312 If the accumulator does not provide enough overflow bits or 1313 does not provide a high dynamic range, the below energy calculation 1314 requires an additional shift operation for each sample. 1315 On the other hand, doing the shift allows using a single-precision 1316 multiplication for the square (at least 16bit x 16bit). 1317 For even values of OVRFLW_BITS (0, 2, 4, 6), saturated arithmetic 1318 is required for the energy accumulation. 1319 Theoretically, the sample-squares can sum up to a value of 76, 1320 requiring 7 overflow bits. However since such situations are *very* 1321 rare, accu can be limited to 64. 1322 In case native saturated arithmetic is not available, overflows 1323 can be prevented by replacing the above #define by 1324 #define SHIFT_BEFORE_SQUARE ((8 - OVRFLW_BITS) / 2) 1325 which will result in slightly reduced accuracy. 1326 */ 1327 1328 /*! 1329 \brief Estimates the mean energy of each filter-bank channel for the 1330 duration of the current envelope 1331 1332 This function is used when interpolFreq is true. 1333 */ 1334 /*static*/ void calcNrgPerSubband(FIXP_DBL **analysBufferReal, /*!< Real part of subband samples */ 1335 FIXP_DBL **analysBufferImag, /*!< Imaginary part of subband samples */ 1336 int lowSubband, /*!< Begin of the SBR frequency range */ 1337 int highSubband, /*!< High end of the SBR frequency range */ 1338 int start_pos, /*!< First QMF-slot of current envelope */ 1339 int next_pos, /*!< Last QMF-slot of current envelope + 1 */ 1340 SCHAR frameExp, /*!< Common exponent for all input samples */ 1341 FIXP_DBL *nrgEst, /*!< resulting Energy (0..1) */ 1342 SCHAR *nrgEst_e ) /*!< Exponent of resulting Energy */ 1343 { 1344 FIXP_SGL invWidth; 1345 SCHAR preShift; 1346 SCHAR shift; 1347 FIXP_DBL sum; 1348 int k,l; 1349 1350 /* Divide by width of envelope later: */ 1351 invWidth = FX_DBL2FX_SGL(GetInvInt(next_pos - start_pos)); 1352 /* The common exponent needs to be doubled because all mantissas are squared: */ 1353 frameExp = frameExp << 1; 1354 1355 for (k=lowSubband; k<highSubband; k++) { 1356 FIXP_DBL bufferReal[(((1024)/(32))+(6))]; 1357 FIXP_DBL bufferImag[(((1024)/(32))+(6))]; 1358 FIXP_DBL maxVal = FL2FX_DBL(0.0f); 1359 1360 if (analysBufferImag!=NULL) 1361 { 1362 for (l=start_pos;l<next_pos;l++) 1363 { 1364 bufferImag[l] = analysBufferImag[l][k]; 1365 maxVal |= (FIXP_DBL)((LONG)(bufferImag[l])^((LONG)bufferImag[l]>>(DFRACT_BITS-1))); 1366 bufferReal[l] = analysBufferReal[l][k]; 1367 maxVal |= (FIXP_DBL)((LONG)(bufferReal[l])^((LONG)bufferReal[l]>>(DFRACT_BITS-1))); 1368 } 1369 } 1370 else 1371 { 1372 for (l=start_pos;l<next_pos;l++) 1373 { 1374 bufferReal[l] = analysBufferReal[l][k]; 1375 maxVal |= (FIXP_DBL)((LONG)(bufferReal[l])^((LONG)bufferReal[l]>>(DFRACT_BITS-1))); 1376 } 1377 } 1378 1379 if (maxVal!=FL2FXCONST_DBL(0.f)) { 1380 1381 1382 /* If the accu does not provide enough overflow bits, we cannot 1383 shift the samples up to the limit. 1384 Instead, keep up to 3 free bits in each sample, i.e. up to 1385 6 bits after calculation of square. 1386 Please note the comment on saturated arithmetic above! 1387 */ 1388 FIXP_DBL accu = FL2FXCONST_DBL(0.0f); 1389 preShift = CntLeadingZeros(maxVal)-1; 1390 preShift -= SHIFT_BEFORE_SQUARE; 1391 1392 if (preShift>=0) { 1393 if (analysBufferImag!=NULL) { 1394 for (l=start_pos; l<next_pos; l++) { 1395 FIXP_DBL temp1 = bufferReal[l] << (int)preShift; 1396 FIXP_DBL temp2 = bufferImag[l] << (int)preShift; 1397 accu = fPow2AddDiv2(accu, temp1); 1398 accu = fPow2AddDiv2(accu, temp2); 1399 } 1400 } else 1401 { 1402 for (l=start_pos; l<next_pos; l++) { 1403 FIXP_DBL temp = bufferReal[l] << (int)preShift; 1404 accu = fPow2AddDiv2(accu, temp); 1405 } 1406 } 1407 } 1408 else { /* if negative shift value */ 1409 int negpreShift = -preShift; 1410 if (analysBufferImag!=NULL) { 1411 for (l=start_pos; l<next_pos; l++) { 1412 FIXP_DBL temp1 = bufferReal[l] >> (int)negpreShift; 1413 FIXP_DBL temp2 = bufferImag[l] >> (int)negpreShift; 1414 accu = fPow2AddDiv2(accu, temp1); 1415 accu = fPow2AddDiv2(accu, temp2); 1416 } 1417 } else 1418 { 1419 for (l=start_pos; l<next_pos; l++) { 1420 FIXP_DBL temp = bufferReal[l] >> (int)negpreShift; 1421 accu = fPow2AddDiv2(accu, temp); 1422 } 1423 } 1424 } 1425 accu <<= 1; 1426 1427 /* Convert double precision to Mantissa/Exponent: */ 1428 shift = fNorm(accu); 1429 sum = accu << (int)shift; 1430 1431 /* Divide by width of envelope and apply frame scale: */ 1432 *nrgEst++ = fMult(sum, invWidth); 1433 shift += 2 * preShift; 1434 if (analysBufferImag!=NULL) 1435 *nrgEst_e++ = frameExp - shift; 1436 else 1437 *nrgEst_e++ = frameExp - shift + 1; /* +1 due to missing imag. part */ 1438 } /* maxVal!=0 */ 1439 else { 1440 1441 /* Prevent a zero-mantissa-number from being misinterpreted 1442 due to its exponent. */ 1443 *nrgEst++ = FL2FXCONST_DBL(0.0f); 1444 *nrgEst_e++ = 0; 1445 } 1446 } 1447 } 1448 1449 /*! 1450 \brief Estimates the mean energy of each Scale factor band for the 1451 duration of the current envelope. 1452 1453 This function is used when interpolFreq is false. 1454 */ 1455 /*static*/ void calcNrgPerSfb(FIXP_DBL **analysBufferReal, /*!< Real part of subband samples */ 1456 FIXP_DBL **analysBufferImag, /*!< Imaginary part of subband samples */ 1457 int nSfb, /*!< Number of scale factor bands */ 1458 UCHAR *freqBandTable, /*!< First Subband for each Sfb */ 1459 int start_pos, /*!< First QMF-slot of current envelope */ 1460 int next_pos, /*!< Last QMF-slot of current envelope + 1 */ 1461 SCHAR input_e, /*!< Common exponent for all input samples */ 1462 FIXP_DBL *nrgEst, /*!< resulting Energy (0..1) */ 1463 SCHAR *nrgEst_e ) /*!< Exponent of resulting Energy */ 1464 { 1465 FIXP_SGL invWidth; 1466 FIXP_DBL temp; 1467 SCHAR preShift; 1468 SCHAR shift, sum_e; 1469 FIXP_DBL sum; 1470 1471 int j,k,l,li,ui; 1472 FIXP_DBL sumAll, sumLine; /* Single precision would be sufficient, 1473 but overflow bits are required for accumulation */ 1474 1475 /* Divide by width of envelope later: */ 1476 invWidth = FX_DBL2FX_SGL(GetInvInt(next_pos - start_pos)); 1477 /* The common exponent needs to be doubled because all mantissas are squared: */ 1478 input_e = input_e << 1; 1479 1480 for(j=0; j<nSfb; j++) { 1481 li = freqBandTable[j]; 1482 ui = freqBandTable[j+1]; 1483 1484 FIXP_DBL maxVal = maxSubbandSample( analysBufferReal, 1485 analysBufferImag, 1486 li, 1487 ui, 1488 start_pos, 1489 next_pos ); 1490 1491 if (maxVal!=FL2FXCONST_DBL(0.f)) { 1492 1493 preShift = CntLeadingZeros(maxVal)-1; 1494 1495 /* If the accu does not provide enough overflow bits, we cannot 1496 shift the samples up to the limit. 1497 Instead, keep up to 3 free bits in each sample, i.e. up to 1498 6 bits after calculation of square. 1499 Please note the comment on saturated arithmetic above! 1500 */ 1501 preShift -= SHIFT_BEFORE_SQUARE; 1502 1503 sumAll = FL2FXCONST_DBL(0.0f); 1504 1505 1506 for (k=li; k<ui; k++) { 1507 1508 sumLine = FL2FXCONST_DBL(0.0f); 1509 1510 if (analysBufferImag!=NULL) { 1511 if (preShift>=0) { 1512 for (l=start_pos; l<next_pos; l++) { 1513 temp = analysBufferReal[l][k] << (int)preShift; 1514 sumLine += fPow2Div2(temp); 1515 temp = analysBufferImag[l][k] << (int)preShift; 1516 sumLine += fPow2Div2(temp); 1517 1518 } 1519 } else { 1520 for (l=start_pos; l<next_pos; l++) { 1521 temp = analysBufferReal[l][k] >> -(int)preShift; 1522 sumLine += fPow2Div2(temp); 1523 temp = analysBufferImag[l][k] >> -(int)preShift; 1524 sumLine += fPow2Div2(temp); 1525 } 1526 } 1527 } else 1528 { 1529 if (preShift>=0) { 1530 for (l=start_pos; l<next_pos; l++) { 1531 temp = analysBufferReal[l][k] << (int)preShift; 1532 sumLine += fPow2Div2(temp); 1533 } 1534 } else { 1535 for (l=start_pos; l<next_pos; l++) { 1536 temp = analysBufferReal[l][k] >> -(int)preShift; 1537 sumLine += fPow2Div2(temp); 1538 } 1539 } 1540 } 1541 1542 /* The number of QMF-channels per SBR bands may be up to 15. 1543 Shift right to avoid overflows in sum over all channels. */ 1544 sumLine = sumLine >> (4-1); 1545 sumAll += sumLine; 1546 } 1547 1548 /* Convert double precision to Mantissa/Exponent: */ 1549 shift = fNorm(sumAll); 1550 sum = sumAll << (int)shift; 1551 1552 /* Divide by width of envelope: */ 1553 sum = fMult(sum,invWidth); 1554 1555 /* Divide by width of Sfb: */ 1556 sum = fMult(sum, FX_DBL2FX_SGL(GetInvInt(ui-li))); 1557 1558 /* Set all Subband energies in the Sfb to the average energy: */ 1559 if (analysBufferImag!=NULL) 1560 sum_e = input_e + 4 - shift; /* -4 to compensate right-shift */ 1561 else 1562 sum_e = input_e + 4 + 1 - shift; /* -4 to compensate right-shift; +1 due to missing imag. part */ 1563 1564 sum_e -= 2 * preShift; 1565 } /* maxVal!=0 */ 1566 else { 1567 1568 /* Prevent a zero-mantissa-number from being misinterpreted 1569 due to its exponent. */ 1570 sum = FL2FXCONST_DBL(0.0f); 1571 sum_e = 0; 1572 } 1573 1574 for (k=li; k<ui; k++) 1575 { 1576 *nrgEst++ = sum; 1577 *nrgEst_e++ = sum_e; 1578 } 1579 } 1580 } 1581 1582 1583 /*! 1584 \brief Calculate gain, noise, and additional sine level for one subband. 1585 1586 The resulting energy gain is given by mantissa and exponent. 1587 */ 1588 /*static*/ void calcSubbandGain(FIXP_DBL nrgRef, /*!< Reference Energy according to envelope data */ 1589 SCHAR nrgRef_e, /*!< Reference Energy according to envelope data (exponent) */ 1590 ENV_CALC_NRGS* nrgs, 1591 int i, 1592 FIXP_DBL tmpNoise, /*!< Relative noise level */ 1593 SCHAR tmpNoise_e, /*!< Relative noise level (exponent) */ 1594 UCHAR sinePresentFlag, /*!< Indicates if sine is present on band */ 1595 UCHAR sineMapped, /*!< Indicates if sine must be added */ 1596 int noNoiseFlag) /*!< Flag to suppress noise addition */ 1597 { 1598 FIXP_DBL nrgEst = nrgs->nrgEst[i]; /*!< Energy in transposed signal */ 1599 SCHAR nrgEst_e = nrgs->nrgEst_e[i]; /*!< Energy in transposed signal (exponent) */ 1600 FIXP_DBL *ptrNrgGain = &nrgs->nrgGain[i]; /*!< Resulting energy gain */ 1601 SCHAR *ptrNrgGain_e = &nrgs->nrgGain_e[i]; /*!< Resulting energy gain (exponent) */ 1602 FIXP_DBL *ptrNoiseLevel = &nrgs->noiseLevel[i]; /*!< Resulting absolute noise energy */ 1603 SCHAR *ptrNoiseLevel_e = &nrgs->noiseLevel_e[i]; /*!< Resulting absolute noise energy (exponent) */ 1604 FIXP_DBL *ptrNrgSine = &nrgs->nrgSine[i]; /*!< Additional sine energy */ 1605 SCHAR *ptrNrgSine_e = &nrgs->nrgSine_e[i]; /*!< Additional sine energy (exponent) */ 1606 1607 FIXP_DBL a, b, c; 1608 SCHAR a_e, b_e, c_e; 1609 1610 /* 1611 This addition of 1 prevents divisions by zero in the reference code. 1612 For very small energies in nrgEst, it prevents the gains from becoming 1613 very high which could cause some trouble due to the smoothing. 1614 */ 1615 b_e = (int)(nrgEst_e - 1); 1616 if (b_e>=0) { 1617 nrgEst = (FL2FXCONST_DBL(0.5f) >> (INT)fixMin(b_e+1,DFRACT_BITS-1)) + (nrgEst >> 1); 1618 nrgEst_e += 1; /* shift by 1 bit to avoid overflow */ 1619 1620 } else { 1621 nrgEst = (nrgEst >> (INT)(fixMin(-b_e+1,DFRACT_BITS-1))) + (FL2FXCONST_DBL(0.5f) >> 1); 1622 nrgEst_e = 2; /* shift by 1 bit to avoid overflow */ 1623 } 1624 1625 /* A = NrgRef * TmpNoise */ 1626 a = fMult(nrgRef,tmpNoise); 1627 a_e = nrgRef_e + tmpNoise_e; 1628 1629 /* B = 1 + TmpNoise */ 1630 b_e = (int)(tmpNoise_e - 1); 1631 if (b_e>=0) { 1632 b = (FL2FXCONST_DBL(0.5f) >> (INT)fixMin(b_e+1,DFRACT_BITS-1)) + (tmpNoise >> 1); 1633 b_e = tmpNoise_e + 1; /* shift by 1 bit to avoid overflow */ 1634 } else { 1635 b = (tmpNoise >> (INT)(fixMin(-b_e+1,DFRACT_BITS-1))) + (FL2FXCONST_DBL(0.5f) >> 1); 1636 b_e = 2; /* shift by 1 bit to avoid overflow */ 1637 } 1638 1639 /* noiseLevel = A / B = (NrgRef * TmpNoise) / (1 + TmpNoise) */ 1640 FDK_divide_MantExp( a, a_e, 1641 b, b_e, 1642 ptrNoiseLevel, ptrNoiseLevel_e); 1643 1644 if (sinePresentFlag) { 1645 1646 /* C = (1 + TmpNoise) * NrgEst */ 1647 c = fMult(b,nrgEst); 1648 c_e = b_e + nrgEst_e; 1649 1650 /* gain = A / C = (NrgRef * TmpNoise) / (1 + TmpNoise) * NrgEst */ 1651 FDK_divide_MantExp( a, a_e, 1652 c, c_e, 1653 ptrNrgGain, ptrNrgGain_e); 1654 1655 if (sineMapped) { 1656 1657 /* sineLevel = nrgRef/ (1 + TmpNoise) */ 1658 FDK_divide_MantExp( nrgRef, nrgRef_e, 1659 b, b_e, 1660 ptrNrgSine, ptrNrgSine_e); 1661 } 1662 } 1663 else { 1664 if (noNoiseFlag) { 1665 /* B = NrgEst */ 1666 b = nrgEst; 1667 b_e = nrgEst_e; 1668 } 1669 else { 1670 /* B = NrgEst * (1 + TmpNoise) */ 1671 b = fMult(b,nrgEst); 1672 b_e = b_e + nrgEst_e; 1673 } 1674 1675 1676 /* gain = nrgRef / B */ 1677 FDK_divide_MantExp( nrgRef, nrgRef_e, 1678 b, b_e, 1679 ptrNrgGain, ptrNrgGain_e); 1680 } 1681 } 1682 1683 1684 /*! 1685 \brief Calculate "average gain" for the specified subband range. 1686 1687 This is rather a gain of the average magnitude than the average 1688 of gains! 1689 The result is used as a relative limit for all gains within the 1690 current "limiter band" (a certain frequency range). 1691 */ 1692 /*static*/ void calcAvgGain(ENV_CALC_NRGS* nrgs, 1693 int lowSubband, /*!< Begin of the limiter band */ 1694 int highSubband, /*!< High end of the limiter band */ 1695 FIXP_DBL *ptrSumRef, 1696 SCHAR *ptrSumRef_e, 1697 FIXP_DBL *ptrAvgGain, /*!< Resulting overall gain (mantissa) */ 1698 SCHAR *ptrAvgGain_e) /*!< Resulting overall gain (exponent) */ 1699 { 1700 FIXP_DBL *nrgRef = nrgs->nrgRef; /*!< Reference Energy according to envelope data */ 1701 SCHAR *nrgRef_e = nrgs->nrgRef_e; /*!< Reference Energy according to envelope data (exponent) */ 1702 FIXP_DBL *nrgEst = nrgs->nrgEst; /*!< Energy in transposed signal */ 1703 SCHAR *nrgEst_e = nrgs->nrgEst_e; /*!< Energy in transposed signal (exponent) */ 1704 1705 FIXP_DBL sumRef = 1; 1706 FIXP_DBL sumEst = 1; 1707 SCHAR sumRef_e = -FRACT_BITS; 1708 SCHAR sumEst_e = -FRACT_BITS; 1709 int k; 1710 1711 for (k=lowSubband; k<highSubband; k++){ 1712 /* Add nrgRef[k] to sumRef: */ 1713 FDK_add_MantExp( sumRef, sumRef_e, 1714 nrgRef[k], nrgRef_e[k], 1715 &sumRef, &sumRef_e ); 1716 1717 /* Add nrgEst[k] to sumEst: */ 1718 FDK_add_MantExp( sumEst, sumEst_e, 1719 nrgEst[k], nrgEst_e[k], 1720 &sumEst, &sumEst_e ); 1721 } 1722 1723 FDK_divide_MantExp(sumRef, sumRef_e, 1724 sumEst, sumEst_e, 1725 ptrAvgGain, ptrAvgGain_e); 1726 1727 *ptrSumRef = sumRef; 1728 *ptrSumRef_e = sumRef_e; 1729 } 1730 1731 1732 /*! 1733 \brief Amplify one timeslot of the signal with the calculated gains 1734 and add the noisefloor. 1735 */ 1736 1737 /*static*/ void adjustTimeSlotLC(FIXP_DBL *ptrReal, /*!< Subband samples to be adjusted, real part */ 1738 ENV_CALC_NRGS* nrgs, 1739 UCHAR *ptrHarmIndex, /*!< Harmonic index */ 1740 int lowSubband, /*!< Lowest QMF-channel in the currently used SBR range. */ 1741 int noSubbands, /*!< Number of QMF subbands */ 1742 int scale_change, /*!< Number of bits to shift adjusted samples */ 1743 int noNoiseFlag, /*!< Flag to suppress noise addition */ 1744 int *ptrPhaseIndex, /*!< Start index to random number array */ 1745 int fCldfb) /*!< CLDFB 80 flag */ 1746 { 1747 FIXP_DBL *pGain = nrgs->nrgGain; /*!< Gains of current envelope */ 1748 FIXP_DBL *pNoiseLevel = nrgs->noiseLevel; /*!< Noise levels of current envelope */ 1749 FIXP_DBL *pSineLevel = nrgs->nrgSine; /*!< Sine levels */ 1750 1751 int k; 1752 int index = *ptrPhaseIndex; 1753 UCHAR harmIndex = *ptrHarmIndex; 1754 UCHAR freqInvFlag = (lowSubband & 1); 1755 FIXP_DBL signalReal, sineLevel, sineLevelNext, sineLevelPrev; 1756 int tone_count = 0; 1757 int sineSign = 1; 1758 1759 #define C1 ((FIXP_SGL)FL2FXCONST_SGL(2.f*0.00815f)) 1760 #define C1_CLDFB ((FIXP_SGL)FL2FXCONST_SGL(2.f*0.16773f)) 1761 1762 /* 1763 First pass for k=0 pulled out of the loop: 1764 */ 1765 1766 index = (index + 1) & (SBR_NF_NO_RANDOM_VAL - 1); 1767 1768 /* 1769 The next multiplication constitutes the actual envelope adjustment 1770 of the signal and should be carried out with full accuracy 1771 (supplying #FRACT_BITS valid bits). 1772 */ 1773 signalReal = fMultDiv2(*ptrReal,*pGain++) << ((int)scale_change); 1774 sineLevel = *pSineLevel++; 1775 sineLevelNext = (noSubbands > 1) ? pSineLevel[0] : FL2FXCONST_DBL(0.0f); 1776 1777 if (sineLevel!=FL2FXCONST_DBL(0.0f)) tone_count++; 1778 1779 else if (!noNoiseFlag) 1780 /* Add noisefloor to the amplified signal */ 1781 signalReal += (fMultDiv2(FDK_sbrDecoder_sbr_randomPhase[index][0], pNoiseLevel[0])<<4); 1782 1783 if (fCldfb) { 1784 1785 if (!(harmIndex&0x1)) { 1786 /* harmIndex 0,2 */ 1787 signalReal += (harmIndex&0x2) ? -sineLevel : sineLevel; 1788 *ptrReal++ = signalReal; 1789 } 1790 else { 1791 /* harmIndex 1,3 in combination with freqInvFlag */ 1792 int shift = (int) (scale_change+1); 1793 shift = (shift>=0) ? fixMin(DFRACT_BITS-1,shift) : fixMax(-(DFRACT_BITS-1),shift); 1794 1795 FIXP_DBL tmp1 = scaleValue( fMultDiv2(C1_CLDFB, sineLevel), -shift ); 1796 1797 FIXP_DBL tmp2 = fMultDiv2(C1_CLDFB, sineLevelNext); 1798 1799 1800 /* save switch and compare operations and reduce to XOR statement */ 1801 if ( ((harmIndex>>1)&0x1)^freqInvFlag) { 1802 *(ptrReal-1) += tmp1; 1803 signalReal -= tmp2; 1804 } else { 1805 *(ptrReal-1) -= tmp1; 1806 signalReal += tmp2; 1807 } 1808 *ptrReal++ = signalReal; 1809 freqInvFlag = !freqInvFlag; 1810 } 1811 1812 } else 1813 { 1814 if (!(harmIndex&0x1)) { 1815 /* harmIndex 0,2 */ 1816 signalReal += (harmIndex&0x2) ? -sineLevel : sineLevel; 1817 *ptrReal++ = signalReal; 1818 } 1819 else { 1820 /* harmIndex 1,3 in combination with freqInvFlag */ 1821 int shift = (int) (scale_change+1); 1822 shift = (shift>=0) ? fixMin(DFRACT_BITS-1,shift) : fixMax(-(DFRACT_BITS-1),shift); 1823 1824 FIXP_DBL tmp1 = (shift>=0) ? ( fMultDiv2(C1, sineLevel) >> shift ) 1825 : ( fMultDiv2(C1, sineLevel) << (-shift) ); 1826 FIXP_DBL tmp2 = fMultDiv2(C1, sineLevelNext); 1827 1828 1829 /* save switch and compare operations and reduce to XOR statement */ 1830 if ( ((harmIndex>>1)&0x1)^freqInvFlag) { 1831 *(ptrReal-1) += tmp1; 1832 signalReal -= tmp2; 1833 } else { 1834 *(ptrReal-1) -= tmp1; 1835 signalReal += tmp2; 1836 } 1837 *ptrReal++ = signalReal; 1838 freqInvFlag = !freqInvFlag; 1839 } 1840 } 1841 1842 pNoiseLevel++; 1843 1844 if ( noSubbands > 2 ) { 1845 if (!(harmIndex&0x1)) { 1846 /* harmIndex 0,2 */ 1847 if(!harmIndex) 1848 { 1849 sineSign = 0; 1850 } 1851 1852 for (k=noSubbands-2; k!=0; k--) { 1853 FIXP_DBL sinelevel = *pSineLevel++; 1854 index++; 1855 if (((signalReal = (sineSign ? -sinelevel : sinelevel)) == FL2FXCONST_DBL(0.0f)) && !noNoiseFlag) 1856 { 1857 /* Add noisefloor to the amplified signal */ 1858 index &= (SBR_NF_NO_RANDOM_VAL - 1); 1859 signalReal += (fMultDiv2(FDK_sbrDecoder_sbr_randomPhase[index][0], pNoiseLevel[0])<<4); 1860 } 1861 1862 /* The next multiplication constitutes the actual envelope adjustment of the signal. */ 1863 signalReal += fMultDiv2(*ptrReal,*pGain++) << ((int)scale_change); 1864 1865 pNoiseLevel++; 1866 *ptrReal++ = signalReal; 1867 } /* for ... */ 1868 } 1869 else { 1870 /* harmIndex 1,3 in combination with freqInvFlag */ 1871 if (harmIndex==1) freqInvFlag = !freqInvFlag; 1872 1873 for (k=noSubbands-2; k!=0; k--) { 1874 index++; 1875 /* The next multiplication constitutes the actual envelope adjustment of the signal. */ 1876 signalReal = fMultDiv2(*ptrReal,*pGain++) << ((int)scale_change); 1877 1878 if (*pSineLevel++!=FL2FXCONST_DBL(0.0f)) tone_count++; 1879 else if (!noNoiseFlag) { 1880 /* Add noisefloor to the amplified signal */ 1881 index &= (SBR_NF_NO_RANDOM_VAL - 1); 1882 signalReal += (fMultDiv2(FDK_sbrDecoder_sbr_randomPhase[index][0], pNoiseLevel[0])<<4); 1883 } 1884 1885 pNoiseLevel++; 1886 1887 if (tone_count <= 16) { 1888 FIXP_DBL addSine = fMultDiv2((pSineLevel[-2] - pSineLevel[0]), C1); 1889 signalReal += (freqInvFlag) ? (-addSine) : (addSine); 1890 } 1891 1892 *ptrReal++ = signalReal; 1893 freqInvFlag = !freqInvFlag; 1894 } /* for ... */ 1895 } 1896 } 1897 1898 if (noSubbands > -1) { 1899 index++; 1900 /* The next multiplication constitutes the actual envelope adjustment of the signal. */ 1901 signalReal = fMultDiv2(*ptrReal,*pGain) << ((int)scale_change); 1902 sineLevelPrev = fMultDiv2(pSineLevel[-1],FL2FX_SGL(0.0163f)); 1903 sineLevel = pSineLevel[0]; 1904 1905 if (pSineLevel[0]!=FL2FXCONST_DBL(0.0f)) tone_count++; 1906 else if (!noNoiseFlag) { 1907 /* Add noisefloor to the amplified signal */ 1908 index &= (SBR_NF_NO_RANDOM_VAL - 1); 1909 signalReal = signalReal + (fMultDiv2(FDK_sbrDecoder_sbr_randomPhase[index][0], pNoiseLevel[0])<<4); 1910 } 1911 1912 if (!(harmIndex&0x1)) { 1913 /* harmIndex 0,2 */ 1914 *ptrReal = signalReal + ( (sineSign) ? -sineLevel : sineLevel); 1915 } 1916 else { 1917 /* harmIndex 1,3 in combination with freqInvFlag */ 1918 if(tone_count <= 16){ 1919 if (freqInvFlag) { 1920 *ptrReal++ = signalReal - sineLevelPrev; 1921 if (noSubbands + lowSubband < 63) 1922 *ptrReal = *ptrReal + fMultDiv2(C1, sineLevel); 1923 } 1924 else { 1925 *ptrReal++ = signalReal + sineLevelPrev; 1926 if (noSubbands + lowSubband < 63) 1927 *ptrReal = *ptrReal - fMultDiv2(C1, sineLevel); 1928 } 1929 } 1930 else *ptrReal = signalReal; 1931 } 1932 } 1933 *ptrHarmIndex = (harmIndex + 1) & 3; 1934 *ptrPhaseIndex = index & (SBR_NF_NO_RANDOM_VAL - 1); 1935 } 1936 void adjustTimeSlotHQ(FIXP_DBL *RESTRICT ptrReal, /*!< Subband samples to be adjusted, real part */ 1937 FIXP_DBL *RESTRICT ptrImag, /*!< Subband samples to be adjusted, imag part */ 1938 HANDLE_SBR_CALCULATE_ENVELOPE h_sbr_cal_env, 1939 ENV_CALC_NRGS* nrgs, 1940 int lowSubband, /*!< Lowest QMF-channel in the currently used SBR range. */ 1941 int noSubbands, /*!< Number of QMF subbands */ 1942 int scale_change, /*!< Number of bits to shift adjusted samples */ 1943 FIXP_SGL smooth_ratio, /*!< Impact of last envelope */ 1944 int noNoiseFlag, /*!< Start index to random number array */ 1945 int filtBufferNoiseShift) /*!< Shift factor of filtBufferNoise */ 1946 { 1947 1948 FIXP_DBL *RESTRICT gain = nrgs->nrgGain; /*!< Gains of current envelope */ 1949 FIXP_DBL *RESTRICT noiseLevel = nrgs->noiseLevel; /*!< Noise levels of current envelope */ 1950 FIXP_DBL *RESTRICT pSineLevel = nrgs->nrgSine; /*!< Sine levels */ 1951 1952 FIXP_DBL *RESTRICT filtBuffer = h_sbr_cal_env->filtBuffer; /*!< Gains of last envelope */ 1953 FIXP_DBL *RESTRICT filtBufferNoise = h_sbr_cal_env->filtBufferNoise; /*!< Noise levels of last envelope */ 1954 UCHAR *RESTRICT ptrHarmIndex =&h_sbr_cal_env->harmIndex; /*!< Harmonic index */ 1955 int *RESTRICT ptrPhaseIndex =&h_sbr_cal_env->phaseIndex; /*!< Start index to random number array */ 1956 1957 int k; 1958 FIXP_DBL signalReal, signalImag; 1959 FIXP_DBL noiseReal, noiseImag; 1960 FIXP_DBL smoothedGain, smoothedNoise; 1961 FIXP_SGL direct_ratio = /*FL2FXCONST_SGL(1.0f) */ (FIXP_SGL)MAXVAL_SGL - smooth_ratio; 1962 int index = *ptrPhaseIndex; 1963 UCHAR harmIndex = *ptrHarmIndex; 1964 register int freqInvFlag = (lowSubband & 1); 1965 FIXP_DBL sineLevel; 1966 int shift; 1967 1968 *ptrPhaseIndex = (index+noSubbands) & (SBR_NF_NO_RANDOM_VAL - 1); 1969 *ptrHarmIndex = (harmIndex + 1) & 3; 1970 1971 /* 1972 Possible optimization: 1973 smooth_ratio and harmIndex stay constant during the loop. 1974 It might be faster to include a separate loop in each path. 1975 1976 the check for smooth_ratio is now outside the loop and the workload 1977 of the whole function decreased by about 20 % 1978 */ 1979 1980 filtBufferNoiseShift += 1; /* due to later use of fMultDiv2 instead of fMult */ 1981 if (filtBufferNoiseShift<0) 1982 shift = fixMin(DFRACT_BITS-1,-filtBufferNoiseShift); 1983 else 1984 shift = fixMin(DFRACT_BITS-1, filtBufferNoiseShift); 1985 1986 if (smooth_ratio > FL2FXCONST_SGL(0.0f)) { 1987 1988 for (k=0; k<noSubbands; k++) { 1989 /* 1990 Smoothing: The old envelope has been bufferd and a certain ratio 1991 of the old gains and noise levels is used. 1992 */ 1993 1994 smoothedGain = fMult(smooth_ratio,filtBuffer[k]) + 1995 fMult(direct_ratio,gain[k]); 1996 1997 if (filtBufferNoiseShift<0) { 1998 smoothedNoise = (fMultDiv2(smooth_ratio,filtBufferNoise[k])>>shift) + 1999 fMult(direct_ratio,noiseLevel[k]); 2000 } 2001 else { 2002 smoothedNoise = (fMultDiv2(smooth_ratio,filtBufferNoise[k])<<shift) + 2003 fMult(direct_ratio,noiseLevel[k]); 2004 } 2005 2006 /* 2007 The next 2 multiplications constitute the actual envelope adjustment 2008 of the signal and should be carried out with full accuracy 2009 (supplying #DFRACT_BITS valid bits). 2010 */ 2011 signalReal = fMultDiv2(*ptrReal,smoothedGain)<<((int)scale_change); 2012 signalImag = fMultDiv2(*ptrImag,smoothedGain)<<((int)scale_change); 2013 2014 index++; 2015 2016 if (pSineLevel[k] != FL2FXCONST_DBL(0.0f)) { 2017 sineLevel = pSineLevel[k]; 2018 2019 switch(harmIndex) { 2020 case 0: 2021 *ptrReal++ = (signalReal + sineLevel); 2022 *ptrImag++ = (signalImag); 2023 break; 2024 case 2: 2025 *ptrReal++ = (signalReal - sineLevel); 2026 *ptrImag++ = (signalImag); 2027 break; 2028 case 1: 2029 *ptrReal++ = (signalReal); 2030 if (freqInvFlag) 2031 *ptrImag++ = (signalImag - sineLevel); 2032 else 2033 *ptrImag++ = (signalImag + sineLevel); 2034 break; 2035 case 3: 2036 *ptrReal++ = signalReal; 2037 if (freqInvFlag) 2038 *ptrImag++ = (signalImag + sineLevel); 2039 else 2040 *ptrImag++ = (signalImag - sineLevel); 2041 break; 2042 } 2043 } 2044 else { 2045 if (noNoiseFlag) { 2046 /* Just the amplified signal is saved */ 2047 *ptrReal++ = (signalReal); 2048 *ptrImag++ = (signalImag); 2049 } 2050 else { 2051 /* Add noisefloor to the amplified signal */ 2052 index &= (SBR_NF_NO_RANDOM_VAL - 1); 2053 noiseReal = fMultDiv2(FDK_sbrDecoder_sbr_randomPhase[index][0], smoothedNoise)<<4; 2054 noiseImag = fMultDiv2(FDK_sbrDecoder_sbr_randomPhase[index][1], smoothedNoise)<<4; 2055 *ptrReal++ = (signalReal + noiseReal); 2056 *ptrImag++ = (signalImag + noiseImag); 2057 } 2058 } 2059 freqInvFlag ^= 1; 2060 } 2061 2062 } 2063 else 2064 { 2065 for (k=0; k<noSubbands; k++) 2066 { 2067 smoothedGain = gain[k]; 2068 signalReal = fMultDiv2(*ptrReal, smoothedGain) << scale_change; 2069 signalImag = fMultDiv2(*ptrImag, smoothedGain) << scale_change; 2070 2071 index++; 2072 2073 if ((sineLevel = pSineLevel[k]) != FL2FXCONST_DBL(0.0f)) 2074 { 2075 switch (harmIndex) 2076 { 2077 case 0: 2078 signalReal += sineLevel; 2079 break; 2080 case 1: 2081 if (freqInvFlag) 2082 signalImag -= sineLevel; 2083 else 2084 signalImag += sineLevel; 2085 break; 2086 case 2: 2087 signalReal -= sineLevel; 2088 break; 2089 case 3: 2090 if (freqInvFlag) 2091 signalImag += sineLevel; 2092 else 2093 signalImag -= sineLevel; 2094 break; 2095 } 2096 } 2097 else 2098 { 2099 if (noNoiseFlag == 0) 2100 { 2101 /* Add noisefloor to the amplified signal */ 2102 smoothedNoise = noiseLevel[k]; 2103 index &= (SBR_NF_NO_RANDOM_VAL - 1); 2104 noiseReal = fMultDiv2(FDK_sbrDecoder_sbr_randomPhase[index][0], smoothedNoise); 2105 noiseImag = fMultDiv2(FDK_sbrDecoder_sbr_randomPhase[index][1], smoothedNoise); 2106 signalReal += noiseReal<<4; 2107 signalImag += noiseImag<<4; 2108 } 2109 } 2110 *ptrReal++ = signalReal; 2111 *ptrImag++ = signalImag; 2112 2113 freqInvFlag ^= 1; 2114 } 2115 } 2116 } 2117 2118 2119 /*! 2120 \brief Reset limiter bands. 2121 2122 Build frequency band table for the gain limiter dependent on 2123 the previously generated transposer patch areas. 2124 2125 \return SBRDEC_OK if ok, SBRDEC_UNSUPPORTED_CONFIG on error 2126 */ 2127 SBR_ERROR 2128 ResetLimiterBands ( UCHAR *limiterBandTable, /*!< Resulting band borders in QMF channels */ 2129 UCHAR *noLimiterBands, /*!< Resulting number of limiter band */ 2130 UCHAR *freqBandTable, /*!< Table with possible band borders */ 2131 int noFreqBands, /*!< Number of bands in freqBandTable */ 2132 const PATCH_PARAM *patchParam, /*!< Transposer patch parameters */ 2133 int noPatches, /*!< Number of transposer patches */ 2134 int limiterBands) /*!< Selected 'band density' from bitstream */ 2135 { 2136 int i, k, isPatchBorder[2], loLimIndex, hiLimIndex, tempNoLim, nBands; 2137 UCHAR workLimiterBandTable[MAX_FREQ_COEFFS / 2 + MAX_NUM_PATCHES + 1]; 2138 int patchBorders[MAX_NUM_PATCHES + 1]; 2139 int kx, k2; 2140 FIXP_DBL temp; 2141 2142 int lowSubband = freqBandTable[0]; 2143 int highSubband = freqBandTable[noFreqBands]; 2144 2145 /* 1 limiter band. */ 2146 if(limiterBands == 0) { 2147 limiterBandTable[0] = 0; 2148 limiterBandTable[1] = highSubband - lowSubband; 2149 nBands = 1; 2150 } else { 2151 for (i = 0; i < noPatches; i++) { 2152 patchBorders[i] = patchParam[i].guardStartBand - lowSubband; 2153 } 2154 patchBorders[i] = highSubband - lowSubband; 2155 2156 /* 1.2, 2, or 3 limiter bands/octave plus bandborders at patchborders. */ 2157 for (k = 0; k <= noFreqBands; k++) { 2158 workLimiterBandTable[k] = freqBandTable[k] - lowSubband; 2159 } 2160 for (k = 1; k < noPatches; k++) { 2161 workLimiterBandTable[noFreqBands + k] = patchBorders[k]; 2162 } 2163 2164 tempNoLim = nBands = noFreqBands + noPatches - 1; 2165 shellsort(workLimiterBandTable, tempNoLim + 1); 2166 2167 loLimIndex = 0; 2168 hiLimIndex = 1; 2169 2170 2171 while (hiLimIndex <= tempNoLim) { 2172 k2 = workLimiterBandTable[hiLimIndex] + lowSubband; 2173 kx = workLimiterBandTable[loLimIndex] + lowSubband; 2174 2175 temp = FX_SGL2FX_DBL(FDK_getNumOctavesDiv8(kx,k2)); /* Number of octaves */ 2176 temp = fMult(temp, FDK_sbrDecoder_sbr_limiterBandsPerOctaveDiv4[limiterBands]); 2177 2178 if (temp < FL2FXCONST_DBL (0.49f)>>5) { 2179 if (workLimiterBandTable[hiLimIndex] == workLimiterBandTable[loLimIndex]) { 2180 workLimiterBandTable[hiLimIndex] = highSubband; 2181 nBands--; 2182 hiLimIndex++; 2183 continue; 2184 } 2185 isPatchBorder[0] = isPatchBorder[1] = 0; 2186 for (k = 0; k <= noPatches; k++) { 2187 if (workLimiterBandTable[hiLimIndex] == patchBorders[k]) { 2188 isPatchBorder[1] = 1; 2189 break; 2190 } 2191 } 2192 if (!isPatchBorder[1]) { 2193 workLimiterBandTable[hiLimIndex] = highSubband; 2194 nBands--; 2195 hiLimIndex++; 2196 continue; 2197 } 2198 for (k = 0; k <= noPatches; k++) { 2199 if (workLimiterBandTable[loLimIndex] == patchBorders[k]) { 2200 isPatchBorder[0] = 1; 2201 break; 2202 } 2203 } 2204 if (!isPatchBorder[0]) { 2205 workLimiterBandTable[loLimIndex] = highSubband; 2206 nBands--; 2207 } 2208 } 2209 loLimIndex = hiLimIndex; 2210 hiLimIndex++; 2211 2212 } 2213 shellsort(workLimiterBandTable, tempNoLim + 1); 2214 2215 /* Test if algorithm exceeded maximum allowed limiterbands */ 2216 if( nBands > MAX_NUM_LIMITERS || nBands <= 0) { 2217 return SBRDEC_UNSUPPORTED_CONFIG; 2218 } 2219 2220 /* Copy limiterbands from working buffer into final destination */ 2221 for (k = 0; k <= nBands; k++) { 2222 limiterBandTable[k] = workLimiterBandTable[k]; 2223 } 2224 } 2225 *noLimiterBands = nBands; 2226 2227 return SBRDEC_OK; 2228 } 2229 2230
