1 2 /* ----------------------------------------------------------------------------------------------------------- 3 Software License for The Fraunhofer FDK AAC Codec Library for Android 4 5 Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Frderung der angewandten Forschung e.V. 6 All rights reserved. 7 8 1. INTRODUCTION 9 The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements 10 the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. 11 This FDK AAC Codec software is intended to be used on a wide variety of Android devices. 12 13 AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual 14 audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by 15 independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part 16 of the MPEG specifications. 17 18 Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) 19 may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners 20 individually for the purpose of encoding or decoding bit streams in products that are compliant with 21 the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license 22 these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec 23 software may already be covered under those patent licenses when it is used for those licensed purposes only. 24 25 Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, 26 are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional 27 applications information and documentation. 28 29 2. COPYRIGHT LICENSE 30 31 Redistribution and use in source and binary forms, with or without modification, are permitted without 32 payment of copyright license fees provided that you satisfy the following conditions: 33 34 You must retain the complete text of this software license in redistributions of the FDK AAC Codec or 35 your modifications thereto in source code form. 36 37 You must retain the complete text of this software license in the documentation and/or other materials 38 provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. 39 You must make available free of charge copies of the complete source code of the FDK AAC Codec and your 40 modifications thereto to recipients of copies in binary form. 41 42 The name of Fraunhofer may not be used to endorse or promote products derived from this library without 43 prior written permission. 44 45 You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec 46 software or your modifications thereto. 47 48 Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software 49 and the date of any change. For modified versions of the FDK AAC Codec, the term 50 "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term 51 "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 52 53 3. NO PATENT LICENSE 54 55 NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, 56 ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with 57 respect to this software. 58 59 You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized 60 by appropriate patent licenses. 61 62 4. DISCLAIMER 63 64 This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors 65 "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties 66 of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR 67 CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, 68 including but not limited to procurement of substitute goods or services; loss of use, data, or profits, 69 or business interruption, however caused and on any theory of liability, whether in contract, strict 70 liability, or tort (including negligence), arising in any way out of the use of this software, even if 71 advised of the possibility of such damage. 72 73 5. CONTACT INFORMATION 74 75 Fraunhofer Institute for Integrated Circuits IIS 76 Attention: Audio and Multimedia Departments - FDK AAC LL 77 Am Wolfsmantel 33 78 91058 Erlangen, Germany 79 80 www.iis.fraunhofer.de/amm 81 amm-info (at) iis.fraunhofer.de 82 ----------------------------------------------------------------------------------------------------------- */ 83 84 #include "nf_est.h" 85 86 #include "sbr_misc.h" 87 88 #include "genericStds.h" 89 90 /* smoothFilter[4] = {0.05857864376269f, 0.2f, 0.34142135623731f, 0.4f}; */ 91 static const FIXP_DBL smoothFilter[4] = { 0x077f813d, 0x19999995, 0x2bb3b1f5, 0x33333335 }; 92 93 /* static const INT smoothFilterLength = 4; */ 94 95 static const FIXP_DBL QuantOffset = (INT)0xfc000000; /* ld64(0.25) */ 96 97 #ifndef min 98 #define min(a,b) ( a < b ? a:b) 99 #endif 100 101 #ifndef max 102 #define max(a,b) ( a > b ? a:b) 103 #endif 104 105 #define NOISE_FLOOR_OFFSET_SCALING (4) 106 107 108 109 /**************************************************************************/ 110 /*! 111 \brief The function applies smoothing to the noise levels. 112 113 114 115 \return none 116 117 */ 118 /**************************************************************************/ 119 static void 120 smoothingOfNoiseLevels(FIXP_DBL *NoiseLevels, /*!< pointer to noise-floor levels.*/ 121 INT nEnvelopes, /*!< Number of noise floor envelopes.*/ 122 INT noNoiseBands, /*!< Number of noise bands for every noise floor envelope. */ 123 FIXP_DBL prevNoiseLevels[NF_SMOOTHING_LENGTH][MAX_NUM_NOISE_VALUES],/*!< Previous noise floor envelopes. */ 124 const FIXP_DBL *smoothFilter, /*!< filter used for smoothing the noise floor levels. */ 125 INT transientFlag) /*!< flag indicating if a transient is present*/ 126 127 { 128 INT i,band,env; 129 FIXP_DBL accu; 130 131 for(env = 0; env < nEnvelopes; env++){ 132 if(transientFlag){ 133 for (i = 0; i < NF_SMOOTHING_LENGTH; i++){ 134 FDKmemcpy(prevNoiseLevels[i],NoiseLevels+env*noNoiseBands,noNoiseBands*sizeof(FIXP_DBL)); 135 } 136 } 137 else { 138 for (i = 1; i < NF_SMOOTHING_LENGTH; i++){ 139 FDKmemcpy(prevNoiseLevels[i - 1],prevNoiseLevels[i],noNoiseBands*sizeof(FIXP_DBL)); 140 } 141 FDKmemcpy(prevNoiseLevels[NF_SMOOTHING_LENGTH - 1],NoiseLevels+env*noNoiseBands,noNoiseBands*sizeof(FIXP_DBL)); 142 } 143 144 for (band = 0; band < noNoiseBands; band++){ 145 accu = FL2FXCONST_DBL(0.0f); 146 for (i = 0; i < NF_SMOOTHING_LENGTH; i++){ 147 accu += fMultDiv2(smoothFilter[i], prevNoiseLevels[i][band]); 148 } 149 FDK_ASSERT( (band + env*noNoiseBands) < MAX_NUM_NOISE_VALUES); 150 NoiseLevels[band+ env*noNoiseBands] = accu<<1; 151 } 152 } 153 } 154 155 /**************************************************************************/ 156 /*! 157 \brief Does the noise floor level estiamtion. 158 159 The noiseLevel samples are scaled by the factor 0.25 160 161 \return none 162 163 */ 164 /**************************************************************************/ 165 static void 166 qmfBasedNoiseFloorDetection(FIXP_DBL *noiseLevel, /*!< Pointer to vector to store the noise levels in.*/ 167 FIXP_DBL ** quotaMatrixOrig, /*!< Matrix holding the quota values of the original. */ 168 SCHAR *indexVector, /*!< Index vector to obtain the patched data. */ 169 INT startIndex, /*!< Start index. */ 170 INT stopIndex, /*!< Stop index. */ 171 INT startChannel, /*!< Start channel of the current noise floor band.*/ 172 INT stopChannel, /*!< Stop channel of the current noise floor band. */ 173 FIXP_DBL ana_max_level, /*!< Maximum level of the adaptive noise.*/ 174 FIXP_DBL noiseFloorOffset, /*!< Noise floor offset. */ 175 INT missingHarmonicFlag, /*!< Flag indicating if a strong tonal component is missing.*/ 176 FIXP_DBL weightFac, /*!< Weightening factor for the difference between orig and sbr. */ 177 INVF_MODE diffThres, /*!< Threshold value to control the inverse filtering decision.*/ 178 INVF_MODE inverseFilteringLevel) /*!< Inverse filtering level of the current band.*/ 179 { 180 INT scale, l, k; 181 FIXP_DBL meanOrig=FL2FXCONST_DBL(0.0f), meanSbr=FL2FXCONST_DBL(0.0f), diff; 182 FIXP_DBL invIndex = GetInvInt(stopIndex-startIndex); 183 FIXP_DBL invChannel = GetInvInt(stopChannel-startChannel); 184 FIXP_DBL accu; 185 186 /* 187 Calculate the mean value, over the current time segment, for the original, the HFR 188 and the difference, over all channels in the current frequency range. 189 */ 190 191 if(missingHarmonicFlag == 1){ 192 for(l = startChannel; l < stopChannel;l++){ 193 /* tonalityOrig */ 194 accu = FL2FXCONST_DBL(0.0f); 195 for(k = startIndex ; k < stopIndex; k++){ 196 accu += fMultDiv2(quotaMatrixOrig[k][l], invIndex); 197 } 198 meanOrig = fixMax(meanOrig,(accu<<1)); 199 200 /* tonalitySbr */ 201 accu = FL2FXCONST_DBL(0.0f); 202 for(k = startIndex ; k < stopIndex; k++){ 203 accu += fMultDiv2(quotaMatrixOrig[k][indexVector[l]], invIndex); 204 } 205 meanSbr = fixMax(meanSbr,(accu<<1)); 206 207 } 208 } 209 else{ 210 for(l = startChannel; l < stopChannel;l++){ 211 /* tonalityOrig */ 212 accu = FL2FXCONST_DBL(0.0f); 213 for(k = startIndex ; k < stopIndex; k++){ 214 accu += fMultDiv2(quotaMatrixOrig[k][l], invIndex); 215 } 216 meanOrig += fMult((accu<<1), invChannel); 217 218 /* tonalitySbr */ 219 accu = FL2FXCONST_DBL(0.0f); 220 for(k = startIndex ; k < stopIndex; k++){ 221 accu += fMultDiv2(quotaMatrixOrig[k][indexVector[l]], invIndex); 222 } 223 meanSbr += fMult((accu<<1), invChannel); 224 } 225 } 226 227 /* Small fix to avoid noise during silent passages.*/ 228 if( meanOrig <= FL2FXCONST_DBL(0.000976562f*RELAXATION_FLOAT) && 229 meanSbr <= FL2FXCONST_DBL(0.000976562f*RELAXATION_FLOAT) ) 230 { 231 meanOrig = FL2FXCONST_DBL(101.5936673f*RELAXATION_FLOAT); 232 meanSbr = FL2FXCONST_DBL(101.5936673f*RELAXATION_FLOAT); 233 } 234 235 meanOrig = fixMax(meanOrig,RELAXATION); 236 meanSbr = fixMax(meanSbr,RELAXATION); 237 238 if (missingHarmonicFlag == 1 || 239 inverseFilteringLevel == INVF_MID_LEVEL || 240 inverseFilteringLevel == INVF_LOW_LEVEL || 241 inverseFilteringLevel == INVF_OFF || 242 inverseFilteringLevel <= diffThres) 243 { 244 diff = RELAXATION; 245 } 246 else { 247 accu = fDivNorm(meanSbr, meanOrig, &scale); 248 249 diff = fixMax( RELAXATION, 250 fMult(RELAXATION_FRACT,fMult(weightFac,accu)) >>( RELAXATION_SHIFT-scale ) ) ; 251 } 252 253 /* 254 * noise Level is now a positive value, i.e. 255 * the more harmonic the signal is the higher noise level, 256 * this makes no sense so we change the sign. 257 *********************************************************/ 258 accu = fDivNorm(diff, meanOrig, &scale); 259 scale -= 2; 260 261 if ( (scale>0) && (accu > ((FIXP_DBL)MAXVAL_DBL)>>scale) ) { 262 *noiseLevel = (FIXP_DBL)MAXVAL_DBL; 263 } 264 else { 265 *noiseLevel = scaleValue(accu, scale); 266 } 267 268 /* 269 * Add a noise floor offset to compensate for bias in the detector 270 *****************************************************************/ 271 if(!missingHarmonicFlag) 272 *noiseLevel = fMult(*noiseLevel, noiseFloorOffset)<<(NOISE_FLOOR_OFFSET_SCALING); 273 274 /* 275 * check to see that we don't exceed the maximum allowed level 276 **************************************************************/ 277 *noiseLevel = fixMin(*noiseLevel, ana_max_level); /* ana_max_level is scaled with factor 0.25 */ 278 } 279 280 /**************************************************************************/ 281 /*! 282 \brief Does the noise floor level estiamtion. 283 The function calls the Noisefloor estimation function 284 for the time segments decided based upon the transient 285 information. The block is always divided into one or two segments. 286 287 288 \return none 289 290 */ 291 /**************************************************************************/ 292 void 293 FDKsbrEnc_sbrNoiseFloorEstimateQmf(HANDLE_SBR_NOISE_FLOOR_ESTIMATE h_sbrNoiseFloorEstimate, /*!< Handle to SBR_NOISE_FLOOR_ESTIMATE struct */ 294 const SBR_FRAME_INFO *frame_info, /*!< Time frequency grid of the current frame. */ 295 FIXP_DBL *noiseLevels, /*!< Pointer to vector to store the noise levels in.*/ 296 FIXP_DBL **quotaMatrixOrig, /*!< Matrix holding the quota values of the original. */ 297 SCHAR *indexVector, /*!< Index vector to obtain the patched data. */ 298 INT missingHarmonicsFlag, /*!< Flag indicating if a strong tonal component will be missing. */ 299 INT startIndex, /*!< Start index. */ 300 int numberOfEstimatesPerFrame, /*!< The number of tonality estimates per frame. */ 301 int transientFrame, /*!< A flag indicating if a transient is present. */ 302 INVF_MODE* pInvFiltLevels, /*!< Pointer to the vector holding the inverse filtering levels. */ 303 UINT sbrSyntaxFlags 304 ) 305 306 { 307 308 INT nNoiseEnvelopes, startPos[2], stopPos[2], env, band; 309 310 INT noNoiseBands = h_sbrNoiseFloorEstimate->noNoiseBands; 311 INT *freqBandTable = h_sbrNoiseFloorEstimate->freqBandTableQmf; 312 313 nNoiseEnvelopes = frame_info->nNoiseEnvelopes; 314 315 if (sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY) { 316 nNoiseEnvelopes = 1; 317 startPos[0] = startIndex; 318 stopPos[0] = startIndex + min(numberOfEstimatesPerFrame,2); 319 } else 320 if(nNoiseEnvelopes == 1){ 321 startPos[0] = startIndex; 322 stopPos[0] = startIndex + 2; 323 } 324 else{ 325 startPos[0] = startIndex; 326 stopPos[0] = startIndex + 1; 327 startPos[1] = startIndex + 1; 328 stopPos[1] = startIndex + 2; 329 } 330 331 /* 332 * Estimate the noise floor. 333 **************************************/ 334 for(env = 0; env < nNoiseEnvelopes; env++){ 335 for(band = 0; band < noNoiseBands; band++){ 336 FDK_ASSERT( (band + env*noNoiseBands) < MAX_NUM_NOISE_VALUES); 337 qmfBasedNoiseFloorDetection(&noiseLevels[band + env*noNoiseBands], 338 quotaMatrixOrig, 339 indexVector, 340 startPos[env], 341 stopPos[env], 342 freqBandTable[band], 343 freqBandTable[band+1], 344 h_sbrNoiseFloorEstimate->ana_max_level, 345 h_sbrNoiseFloorEstimate->noiseFloorOffset[band], 346 missingHarmonicsFlag, 347 h_sbrNoiseFloorEstimate->weightFac, 348 h_sbrNoiseFloorEstimate->diffThres, 349 pInvFiltLevels[band]); 350 } 351 } 352 353 354 /* 355 * Smoothing of the values. 356 **************************/ 357 smoothingOfNoiseLevels(noiseLevels, 358 nNoiseEnvelopes, 359 h_sbrNoiseFloorEstimate->noNoiseBands, 360 h_sbrNoiseFloorEstimate->prevNoiseLevels, 361 h_sbrNoiseFloorEstimate->smoothFilter, 362 transientFrame); 363 364 365 /* quantisation*/ 366 for(env = 0; env < nNoiseEnvelopes; env++){ 367 for(band = 0; band < noNoiseBands; band++){ 368 FDK_ASSERT( (band + env*noNoiseBands) < MAX_NUM_NOISE_VALUES); 369 noiseLevels[band + env*noNoiseBands] = 370 (FIXP_DBL)NOISE_FLOOR_OFFSET_64 - (FIXP_DBL)CalcLdData(noiseLevels[band + env*noNoiseBands]+(FIXP_DBL)1) + QuantOffset; 371 } 372 } 373 } 374 375 /**************************************************************************/ 376 /*! 377 \brief 378 379 380 \return errorCode, noError if successful 381 382 */ 383 /**************************************************************************/ 384 static INT 385 downSampleLoRes(INT *v_result, /*!< */ 386 INT num_result, /*!< */ 387 const UCHAR *freqBandTableRef,/*!< */ 388 INT num_Ref) /*!< */ 389 { 390 INT step; 391 INT i,j; 392 INT org_length,result_length; 393 INT v_index[MAX_FREQ_COEFFS/2]; 394 395 /* init */ 396 org_length=num_Ref; 397 result_length=num_result; 398 399 v_index[0]=0; /* Always use left border */ 400 i=0; 401 while(org_length > 0) /* Create downsample vector */ 402 { 403 i++; 404 step=org_length/result_length; /* floor; */ 405 org_length=org_length - step; 406 result_length--; 407 v_index[i]=v_index[i-1]+step; 408 } 409 410 if(i != num_result ) /* Should never happen */ 411 return (1);/* error downsampling */ 412 413 for(j=0;j<=i;j++) /* Use downsample vector to index LoResolution vector. */ 414 { 415 v_result[j]=freqBandTableRef[v_index[j]]; 416 } 417 418 return (0); 419 } 420 421 /**************************************************************************/ 422 /*! 423 \brief Initialize an instance of the noise floor level estimation module. 424 425 426 \return errorCode, noError if successful 427 428 */ 429 /**************************************************************************/ 430 INT 431 FDKsbrEnc_InitSbrNoiseFloorEstimate (HANDLE_SBR_NOISE_FLOOR_ESTIMATE h_sbrNoiseFloorEstimate, /*!< Handle to SBR_NOISE_FLOOR_ESTIMATE struct */ 432 INT ana_max_level, /*!< Maximum level of the adaptive noise. */ 433 const UCHAR *freqBandTable, /*!< Frequany band table. */ 434 INT nSfb, /*!< Number of frequency bands. */ 435 INT noiseBands, /*!< Number of noise bands per octave. */ 436 INT noiseFloorOffset, /*!< Noise floor offset. */ 437 INT timeSlots, /*!< Number of time slots in a frame. */ 438 UINT useSpeechConfig /*!< Flag: adapt tuning parameters according to speech */ 439 ) 440 { 441 INT i, qexp, qtmp; 442 FIXP_DBL tmp, exp; 443 444 FDKmemclear(h_sbrNoiseFloorEstimate,sizeof(SBR_NOISE_FLOOR_ESTIMATE)); 445 446 h_sbrNoiseFloorEstimate->smoothFilter = smoothFilter; 447 if (useSpeechConfig) { 448 h_sbrNoiseFloorEstimate->weightFac = (FIXP_DBL)MAXVAL_DBL; 449 h_sbrNoiseFloorEstimate->diffThres = INVF_LOW_LEVEL; 450 } 451 else { 452 h_sbrNoiseFloorEstimate->weightFac = FL2FXCONST_DBL(0.25f); 453 h_sbrNoiseFloorEstimate->diffThres = INVF_MID_LEVEL; 454 } 455 456 h_sbrNoiseFloorEstimate->timeSlots = timeSlots; 457 h_sbrNoiseFloorEstimate->noiseBands = noiseBands; 458 459 /* h_sbrNoiseFloorEstimate->ana_max_level is scaled by 0.25 */ 460 switch(ana_max_level) 461 { 462 case 6: 463 h_sbrNoiseFloorEstimate->ana_max_level = (FIXP_DBL)MAXVAL_DBL; 464 break; 465 case 3: 466 h_sbrNoiseFloorEstimate->ana_max_level = FL2FXCONST_DBL(0.5); 467 break; 468 case -3: 469 h_sbrNoiseFloorEstimate->ana_max_level = FL2FXCONST_DBL(0.125); 470 break; 471 default: 472 /* Should not enter here */ 473 h_sbrNoiseFloorEstimate->ana_max_level = (FIXP_DBL)MAXVAL_DBL; 474 break; 475 } 476 477 /* 478 calculate number of noise bands and allocate 479 */ 480 if(FDKsbrEnc_resetSbrNoiseFloorEstimate(h_sbrNoiseFloorEstimate,freqBandTable,nSfb)) 481 return(1); 482 483 if(noiseFloorOffset == 0) { 484 tmp = ((FIXP_DBL)MAXVAL_DBL)>>NOISE_FLOOR_OFFSET_SCALING; 485 } 486 else { 487 /* noiseFloorOffset has to be smaller than 12, because 488 the result of the calculation below must be smaller than 1: 489 (2^(noiseFloorOffset/3))*2^4<1 */ 490 FDK_ASSERT(noiseFloorOffset<12); 491 492 /* Assumes the noise floor offset in tuning table are in q31 */ 493 /* Change the qformat here when non-zero values would be filled */ 494 exp = fDivNorm((FIXP_DBL)noiseFloorOffset, 3, &qexp); 495 tmp = fPow(2, DFRACT_BITS-1, exp, qexp, &qtmp); 496 tmp = scaleValue(tmp, qtmp-NOISE_FLOOR_OFFSET_SCALING); 497 } 498 499 for(i=0;i<h_sbrNoiseFloorEstimate->noNoiseBands;i++) { 500 h_sbrNoiseFloorEstimate->noiseFloorOffset[i] = tmp; 501 } 502 503 return (0); 504 } 505 506 /**************************************************************************/ 507 /*! 508 \brief Resets the current instance of the noise floor estiamtion 509 module. 510 511 512 \return errorCode, noError if successful 513 514 */ 515 /**************************************************************************/ 516 INT 517 FDKsbrEnc_resetSbrNoiseFloorEstimate (HANDLE_SBR_NOISE_FLOOR_ESTIMATE h_sbrNoiseFloorEstimate, /*!< Handle to SBR_NOISE_FLOOR_ESTIMATE struct */ 518 const UCHAR *freqBandTable, /*!< Frequany band table. */ 519 INT nSfb) /*!< Number of bands in the frequency band table. */ 520 { 521 INT k2,kx; 522 523 /* 524 * Calculate number of noise bands 525 ***********************************/ 526 k2=freqBandTable[nSfb]; 527 kx=freqBandTable[0]; 528 if(h_sbrNoiseFloorEstimate->noiseBands == 0){ 529 h_sbrNoiseFloorEstimate->noNoiseBands = 1; 530 } 531 else{ 532 /* 533 * Calculate number of noise bands 1,2 or 3 bands/octave 534 ********************************************************/ 535 FIXP_DBL tmp, ratio, lg2; 536 INT ratio_e, qlg2, nNoiseBands; 537 538 ratio = fDivNorm(k2, kx, &ratio_e); 539 lg2 = fLog2(ratio, ratio_e, &qlg2); 540 tmp = fMult((FIXP_DBL)(h_sbrNoiseFloorEstimate->noiseBands<<24), lg2); 541 tmp = scaleValue(tmp, qlg2-23); 542 543 nNoiseBands = (INT)((tmp + (FIXP_DBL)1) >> 1); 544 545 546 if (nNoiseBands > MAX_NUM_NOISE_COEFFS ) { 547 nNoiseBands = MAX_NUM_NOISE_COEFFS; 548 } 549 550 if( nNoiseBands == 0 ) { 551 nNoiseBands = 1; 552 } 553 554 h_sbrNoiseFloorEstimate->noNoiseBands = nNoiseBands; 555 556 } 557 558 559 return(downSampleLoRes(h_sbrNoiseFloorEstimate->freqBandTableQmf, 560 h_sbrNoiseFloorEstimate->noNoiseBands, 561 freqBandTable,nSfb)); 562 } 563 564 /**************************************************************************/ 565 /*! 566 \brief Deletes the current instancce of the noise floor level 567 estimation module. 568 569 570 \return none 571 572 */ 573 /**************************************************************************/ 574 void 575 FDKsbrEnc_deleteSbrNoiseFloorEstimate (HANDLE_SBR_NOISE_FLOOR_ESTIMATE h_sbrNoiseFloorEstimate) /*!< Handle to SBR_NOISE_FLOOR_ESTIMATE struct */ 576 { 577 578 if (h_sbrNoiseFloorEstimate) { 579 /* 580 nothing to do 581 */ 582 } 583 } 584
