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1 /* -----------------------------------------------------------------------------
2 Software License for The Fraunhofer FDK AAC Codec Library for Android
3 
4 © Copyright  1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
5 Forschung e.V. All rights reserved.
6 
7  1.    INTRODUCTION
8 The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software
9 that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding
10 scheme for digital audio. This FDK AAC Codec software is intended to be used on
11 a wide variety of Android devices.
12 
13 AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient
14 general perceptual audio codecs. AAC-ELD is considered the best-performing
15 full-bandwidth communications codec by independent studies and is widely
16 deployed. AAC has been standardized by ISO and IEC as part of the MPEG
17 specifications.
18 
19 Patent licenses for necessary patent claims for the FDK AAC Codec (including
20 those of Fraunhofer) may be obtained through Via Licensing
21 (www.vialicensing.com) or through the respective patent owners individually for
22 the purpose of encoding or decoding bit streams in products that are compliant
23 with the ISO/IEC MPEG audio standards. Please note that most manufacturers of
24 Android devices already license these patent claims through Via Licensing or
25 directly from the patent owners, and therefore FDK AAC Codec software may
26 already be covered under those patent licenses when it is used for those
27 licensed purposes only.
28 
29 Commercially-licensed AAC software libraries, including floating-point versions
30 with enhanced sound quality, are also available from Fraunhofer. Users are
31 encouraged to check the Fraunhofer website for additional applications
32 information and documentation.
33 
34 2.    COPYRIGHT LICENSE
35 
36 Redistribution and use in source and binary forms, with or without modification,
37 are permitted without payment of copyright license fees provided that you
38 satisfy the following conditions:
39 
40 You must retain the complete text of this software license in redistributions of
41 the FDK AAC Codec or your modifications thereto in source code form.
42 
43 You must retain the complete text of this software license in the documentation
44 and/or other materials provided with redistributions of the FDK AAC Codec or
45 your modifications thereto in binary form. You must make available free of
46 charge copies of the complete source code of the FDK AAC Codec and your
47 modifications thereto to recipients of copies in binary form.
48 
49 The name of Fraunhofer may not be used to endorse or promote products derived
50 from this library without prior written permission.
51 
52 You may not charge copyright license fees for anyone to use, copy or distribute
53 the FDK AAC Codec software or your modifications thereto.
54 
55 Your modified versions of the FDK AAC Codec must carry prominent notices stating
56 that you changed the software and the date of any change. For modified versions
57 of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android"
58 must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK
59 AAC Codec Library for Android."
60 
61 3.    NO PATENT LICENSE
62 
63 NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without
64 limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE.
65 Fraunhofer provides no warranty of patent non-infringement with respect to this
66 software.
67 
68 You may use this FDK AAC Codec software or modifications thereto only for
69 purposes that are authorized by appropriate patent licenses.
70 
71 4.    DISCLAIMER
72 
73 This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright
74 holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES,
75 including but not limited to the implied warranties of merchantability and
76 fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
77 CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary,
78 or consequential damages, including but not limited to procurement of substitute
79 goods or services; loss of use, data, or profits, or business interruption,
80 however caused and on any theory of liability, whether in contract, strict
81 liability, or tort (including negligence), arising in any way out of the use of
82 this software, even if advised of the possibility of such damage.
83 
84 5.    CONTACT INFORMATION
85 
86 Fraunhofer Institute for Integrated Circuits IIS
87 Attention: Audio and Multimedia Departments - FDK AAC LL
88 Am Wolfsmantel 33
89 91058 Erlangen, Germany
90 
91 www.iis.fraunhofer.de/amm
92 amm-info@iis.fraunhofer.de
93 ----------------------------------------------------------------------------- */
94 
95 /**************************** SBR decoder library ******************************
96 
97    Author(s):
98 
99    Description:
100 
101 *******************************************************************************/
102 
103 /*!
104   \file
105   \brief  parametric stereo decoder
106 */
107 
108 #include "psdec.h"
109 
110 #include "FDK_bitbuffer.h"
111 
112 #include "sbr_rom.h"
113 #include "sbr_ram.h"
114 
115 #include "FDK_tools_rom.h"
116 
117 #include "genericStds.h"
118 
119 #include "FDK_trigFcts.h"
120 
121 /********************************************************************/
122 /*                       MLQUAL DEFINES                             */
123 /********************************************************************/
124 
125 #define FRACT_ZERO FRACT_BITS - 1
126 /********************************************************************/
127 
128 SBR_ERROR ResetPsDec(HANDLE_PS_DEC h_ps_d);
129 
130 /***** HELPERS *****/
131 
132 /***************************************************************************/
133 /*!
134   \brief  Creates one instance of the PS_DEC struct
135 
136   \return Error info
137 
138 ****************************************************************************/
CreatePsDec(HANDLE_PS_DEC * h_PS_DEC,int aacSamplesPerFrame)139 int CreatePsDec(HANDLE_PS_DEC *h_PS_DEC, /*!< pointer to the module state */
140                 int aacSamplesPerFrame) {
141   SBR_ERROR errorInfo = SBRDEC_OK;
142   HANDLE_PS_DEC h_ps_d;
143   int i;
144 
145   if (*h_PS_DEC == NULL) {
146     /* Get ps dec ram */
147     h_ps_d = GetRam_ps_dec();
148     if (h_ps_d == NULL) {
149       goto bail;
150     }
151   } else {
152     /* Reset an open instance */
153     h_ps_d = *h_PS_DEC;
154   }
155 
156   /*
157    * Create Analysis Hybrid filterbank.
158    */
159   FDKhybridAnalysisOpen(&h_ps_d->specificTo.mpeg.hybridAnalysis,
160                         h_ps_d->specificTo.mpeg.pHybridAnaStatesLFdmx,
161                         sizeof(h_ps_d->specificTo.mpeg.pHybridAnaStatesLFdmx),
162                         NULL, 0);
163 
164   /* initialisation */
165   switch (aacSamplesPerFrame) {
166     case 960:
167       h_ps_d->noSubSamples = 30; /* col */
168       break;
169     case 1024:
170       h_ps_d->noSubSamples = 32; /* col */
171       break;
172     default:
173       h_ps_d->noSubSamples = -1;
174       break;
175   }
176 
177   if (h_ps_d->noSubSamples > MAX_NUM_COL || h_ps_d->noSubSamples <= 0) {
178     goto bail;
179   }
180   h_ps_d->noChannels = NO_QMF_CHANNELS; /* row */
181 
182   h_ps_d->psDecodedPrv = 0;
183   h_ps_d->procFrameBased = -1;
184   for (i = 0; i < (1) + 1; i++) {
185     h_ps_d->bPsDataAvail[i] = ppt_none;
186   }
187   {
188     int error;
189     error = FDKdecorrelateOpen(&(h_ps_d->specificTo.mpeg.apDecor),
190                                h_ps_d->specificTo.mpeg.decorrBufferCplx,
191                                (2 * ((825) + (373))));
192     if (error) goto bail;
193   }
194 
195   for (i = 0; i < (1) + 1; i++) {
196     FDKmemclear(&h_ps_d->bsData[i].mpeg, sizeof(MPEG_PS_BS_DATA));
197   }
198 
199   errorInfo = ResetPsDec(h_ps_d);
200 
201   if (errorInfo != SBRDEC_OK) goto bail;
202 
203   *h_PS_DEC = h_ps_d;
204 
205   return 0;
206 
207 bail:
208   if (h_ps_d != NULL) {
209     DeletePsDec(&h_ps_d);
210   }
211 
212   return -1;
213 } /*END CreatePsDec */
214 
215 /***************************************************************************/
216 /*!
217   \brief  Delete one instance of the PS_DEC struct
218 
219   \return Error info
220 
221 ****************************************************************************/
DeletePsDec(HANDLE_PS_DEC * h_PS_DEC)222 int DeletePsDec(HANDLE_PS_DEC *h_PS_DEC) /*!< pointer to the module state */
223 {
224   if (*h_PS_DEC == NULL) {
225     return -1;
226   }
227 
228   {
229     HANDLE_PS_DEC h_ps_d = *h_PS_DEC;
230     FDKdecorrelateClose(&(h_ps_d->specificTo.mpeg.apDecor));
231   }
232 
233   FreeRam_ps_dec(h_PS_DEC);
234 
235   return 0;
236 } /*END DeletePsDec */
237 
238 /***************************************************************************/
239 /*!
240   \brief resets some values of the PS handle to default states
241 
242   \return
243 
244 ****************************************************************************/
ResetPsDec(HANDLE_PS_DEC h_ps_d)245 SBR_ERROR ResetPsDec(HANDLE_PS_DEC h_ps_d) /*!< pointer to the module state */
246 {
247   SBR_ERROR errorInfo = SBRDEC_OK;
248   INT i;
249 
250   /* explicitly init state variables to safe values (until first ps header
251    * arrives) */
252 
253   h_ps_d->specificTo.mpeg.lastUsb = 0;
254 
255   /*
256    * Initialize Analysis Hybrid filterbank.
257    */
258   FDKhybridAnalysisInit(&h_ps_d->specificTo.mpeg.hybridAnalysis, THREE_TO_TEN,
259                         NO_QMF_BANDS_HYBRID20, NO_QMF_BANDS_HYBRID20, 1);
260 
261   /*
262    * Initialize Synthesis Hybrid filterbank.
263    */
264   for (i = 0; i < 2; i++) {
265     FDKhybridSynthesisInit(&h_ps_d->specificTo.mpeg.hybridSynthesis[i],
266                            THREE_TO_TEN, NO_QMF_CHANNELS, NO_QMF_CHANNELS);
267   }
268   {
269     INT error;
270     error = FDKdecorrelateInit(&h_ps_d->specificTo.mpeg.apDecor, 71, DECORR_PS,
271                                DUCKER_AUTOMATIC, 0, 0, 0, 0, 1, /* isLegacyPS */
272                                1);
273     if (error) return SBRDEC_NOT_INITIALIZED;
274   }
275 
276   for (i = 0; i < NO_IID_GROUPS; i++) {
277     h_ps_d->specificTo.mpeg.h11rPrev[i] = FL2FXCONST_DBL(0.5f);
278     h_ps_d->specificTo.mpeg.h12rPrev[i] = FL2FXCONST_DBL(0.5f);
279   }
280 
281   FDKmemclear(h_ps_d->specificTo.mpeg.h21rPrev,
282               sizeof(h_ps_d->specificTo.mpeg.h21rPrev));
283   FDKmemclear(h_ps_d->specificTo.mpeg.h22rPrev,
284               sizeof(h_ps_d->specificTo.mpeg.h22rPrev));
285 
286   return errorInfo;
287 }
288 
289 /***************************************************************************/
290 /*!
291   \brief  Feed delaylines when parametric stereo is switched on.
292   \return
293 ****************************************************************************/
PreparePsProcessing(HANDLE_PS_DEC h_ps_d,const FIXP_DBL * const * const rIntBufferLeft,const FIXP_DBL * const * const iIntBufferLeft,const int scaleFactorLowBand)294 void PreparePsProcessing(HANDLE_PS_DEC h_ps_d,
295                          const FIXP_DBL *const *const rIntBufferLeft,
296                          const FIXP_DBL *const *const iIntBufferLeft,
297                          const int scaleFactorLowBand) {
298   if (h_ps_d->procFrameBased ==
299       1) /* If we have switched from frame to slot based processing  */
300   {      /* fill hybrid delay buffer.                                */
301     int i, j;
302 
303     for (i = 0; i < HYBRID_FILTER_DELAY; i++) {
304       FIXP_DBL qmfInputData[2][NO_QMF_BANDS_HYBRID20];
305       FIXP_DBL hybridOutputData[2][NO_SUB_QMF_CHANNELS];
306 
307       for (j = 0; j < NO_QMF_BANDS_HYBRID20; j++) {
308         qmfInputData[0][j] =
309             scaleValue(rIntBufferLeft[i][j], scaleFactorLowBand);
310         qmfInputData[1][j] =
311             scaleValue(iIntBufferLeft[i][j], scaleFactorLowBand);
312       }
313 
314       FDKhybridAnalysisApply(&h_ps_d->specificTo.mpeg.hybridAnalysis,
315                              qmfInputData[0], qmfInputData[1],
316                              hybridOutputData[0], hybridOutputData[1]);
317     }
318     h_ps_d->procFrameBased = 0; /* switch to slot based processing. */
319 
320   } /* procFrameBased==1 */
321 }
322 
initSlotBasedRotation(HANDLE_PS_DEC h_ps_d,int env,int usb)323 void initSlotBasedRotation(
324     HANDLE_PS_DEC h_ps_d, /*!< pointer to the module state */
325     int env, int usb) {
326   INT group = 0;
327   INT bin = 0;
328   INT noIidSteps;
329 
330   FIXP_SGL invL;
331   FIXP_DBL ScaleL, ScaleR;
332   FIXP_DBL Alpha, Beta;
333   FIXP_DBL h11r, h12r, h21r, h22r;
334 
335   const FIXP_DBL *PScaleFactors;
336 
337   if (h_ps_d->bsData[h_ps_d->processSlot].mpeg.bFineIidQ) {
338     PScaleFactors = ScaleFactorsFine; /* values are shiftet right by one */
339     noIidSteps = NO_IID_STEPS_FINE;
340   } else {
341     PScaleFactors = ScaleFactors; /* values are shiftet right by one */
342     noIidSteps = NO_IID_STEPS;
343   }
344 
345   /* dequantize and decode */
346   for (group = 0; group < NO_IID_GROUPS; group++) {
347     bin = bins2groupMap20[group];
348 
349     /*!
350     <h3> type 'A' rotation </h3>
351     mixing procedure R_a, used in baseline version<br>
352 
353      Scale-factor vectors c1 and c2 are precalculated in initPsTables () and
354     stored in scaleFactors[] and scaleFactorsFine[] = pScaleFactors []. From the
355     linearized IID parameters (intensity differences), two scale factors are
356      calculated. They are used to obtain the coefficients h11... h22.
357     */
358 
359     /* ScaleR and ScaleL are scaled by 1 shift right */
360 
361     ScaleR = PScaleFactors[noIidSteps + h_ps_d->specificTo.mpeg.pCoef
362                                             ->aaIidIndexMapped[env][bin]];
363     ScaleL = PScaleFactors[noIidSteps - h_ps_d->specificTo.mpeg.pCoef
364                                             ->aaIidIndexMapped[env][bin]];
365 
366     Beta = fMult(
367         fMult(Alphas[h_ps_d->specificTo.mpeg.pCoef->aaIccIndexMapped[env][bin]],
368               (ScaleR - ScaleL)),
369         FIXP_SQRT05);
370     Alpha =
371         Alphas[h_ps_d->specificTo.mpeg.pCoef->aaIccIndexMapped[env][bin]] >> 1;
372 
373     /* Alpha and Beta are now both scaled by 2 shifts right */
374 
375     /* calculate the coefficients h11... h22 from scale-factors and ICC
376      * parameters */
377 
378     /* h values are scaled by 1 shift right */
379     {
380       FIXP_DBL trigData[4];
381 
382       inline_fixp_cos_sin(Beta + Alpha, Beta - Alpha, 2, trigData);
383       h11r = fMult(ScaleL, trigData[0]);
384       h12r = fMult(ScaleR, trigData[2]);
385       h21r = fMult(ScaleL, trigData[1]);
386       h22r = fMult(ScaleR, trigData[3]);
387     }
388     /*****************************************************************************************/
389     /* Interpolation of the matrices H11... H22: */
390     /*                                                                                       */
391     /* H11(k,n) = H11(k,n[e]) + (n-n[e]) * (H11(k,n[e+1] - H11(k,n[e])) /
392      * (n[e+1] - n[e])    */
393     /* ... */
394     /*****************************************************************************************/
395 
396     /* invL = 1/(length of envelope) */
397     invL = FX_DBL2FX_SGL(GetInvInt(
398         h_ps_d->bsData[h_ps_d->processSlot].mpeg.aEnvStartStop[env + 1] -
399         h_ps_d->bsData[h_ps_d->processSlot].mpeg.aEnvStartStop[env]));
400 
401     h_ps_d->specificTo.mpeg.pCoef->H11r[group] =
402         h_ps_d->specificTo.mpeg.h11rPrev[group];
403     h_ps_d->specificTo.mpeg.pCoef->H12r[group] =
404         h_ps_d->specificTo.mpeg.h12rPrev[group];
405     h_ps_d->specificTo.mpeg.pCoef->H21r[group] =
406         h_ps_d->specificTo.mpeg.h21rPrev[group];
407     h_ps_d->specificTo.mpeg.pCoef->H22r[group] =
408         h_ps_d->specificTo.mpeg.h22rPrev[group];
409 
410     h_ps_d->specificTo.mpeg.pCoef->DeltaH11r[group] =
411         fMult(h11r - h_ps_d->specificTo.mpeg.pCoef->H11r[group], invL);
412     h_ps_d->specificTo.mpeg.pCoef->DeltaH12r[group] =
413         fMult(h12r - h_ps_d->specificTo.mpeg.pCoef->H12r[group], invL);
414     h_ps_d->specificTo.mpeg.pCoef->DeltaH21r[group] =
415         fMult(h21r - h_ps_d->specificTo.mpeg.pCoef->H21r[group], invL);
416     h_ps_d->specificTo.mpeg.pCoef->DeltaH22r[group] =
417         fMult(h22r - h_ps_d->specificTo.mpeg.pCoef->H22r[group], invL);
418 
419     /* update prev coefficients for interpolation in next envelope */
420 
421     h_ps_d->specificTo.mpeg.h11rPrev[group] = h11r;
422     h_ps_d->specificTo.mpeg.h12rPrev[group] = h12r;
423     h_ps_d->specificTo.mpeg.h21rPrev[group] = h21r;
424     h_ps_d->specificTo.mpeg.h22rPrev[group] = h22r;
425 
426   } /* group loop */
427 }
428 
429 static const UCHAR groupTable[NO_IID_GROUPS + 1] = {
430     0,  1,  2,  3,  4,  5,  6,  7,  8,  9,  10, 11,
431     12, 13, 14, 15, 16, 18, 21, 25, 30, 42, 71};
432 
applySlotBasedRotation(HANDLE_PS_DEC h_ps_d,FIXP_DBL * mHybridRealLeft,FIXP_DBL * mHybridImagLeft,FIXP_DBL * mHybridRealRight,FIXP_DBL * mHybridImagRight)433 static void applySlotBasedRotation(
434     HANDLE_PS_DEC h_ps_d, /*!< pointer to the module state */
435 
436     FIXP_DBL *mHybridRealLeft, /*!< hybrid values real left  */
437     FIXP_DBL *mHybridImagLeft, /*!< hybrid values imag left  */
438 
439     FIXP_DBL *mHybridRealRight, /*!< hybrid values real right  */
440     FIXP_DBL *mHybridImagRight  /*!< hybrid values imag right  */
441 ) {
442   INT group;
443   INT subband;
444 
445   /**********************************************************************************************/
446   /*!
447   <h2> Mapping </h2>
448 
449   The number of stereo bands that is actually used depends on the number of
450   availble parameters for IID and ICC: <pre> nr. of IID para.| nr. of ICC para.
451   | nr. of Stereo bands
452    ----------------|------------------|-------------------
453      10,20         |     10,20        |        20
454      10,20         |     34           |        34
455      34            |     10,20        |        34
456      34            |     34           |        34
457   </pre>
458   In the case the number of parameters for IIS and ICC differs from the number
459   of stereo bands, a mapping from the lower number to the higher number of
460   parameters is applied. Index mapping of IID and ICC parameters is already done
461   in psbitdec.cpp. Further mapping is not needed here in baseline version.
462   **********************************************************************************************/
463 
464   /************************************************************************************************/
465   /*!
466   <h2> Mixing </h2>
467 
468   To generate the QMF subband signals for the subband samples n = n[e]+1 ,,,
469   n_[e+1] the parameters at position n[e] and n[e+1] are required as well as the
470   subband domain signals s_k(n) and d_k(n) for n = n[e]+1... n_[e+1]. n[e]
471   represents the start position for envelope e. The border positions n[e] are
472   handled in DecodePS().
473 
474   The stereo sub subband signals are constructed as:
475   <pre>
476   l_k(n) = H11(k,n) s_k(n) + H21(k,n) d_k(n)
477   r_k(n) = H21(k,n) s_k(n) + H22(k,n) d_k(n)
478   </pre>
479   In order to obtain the matrices H11(k,n)... H22 (k,n), the vectors h11(b)...
480   h22(b) need to be calculated first (b: parameter index). Depending on ICC mode
481   either mixing procedure R_a or R_b is used for that. For both procedures, the
482   parameters for parameter position n[e+1] is used.
483   ************************************************************************************************/
484 
485   /************************************************************************************************/
486   /*!
487   <h2>Phase parameters </h2>
488   With disabled phase parameters (which is the case in baseline version), the
489   H-matrices are just calculated by:
490 
491   <pre>
492   H11(k,n[e+1] = h11(b(k))
493   (...)
494   b(k): parameter index according to mapping table
495   </pre>
496 
497   <h2>Processing of the samples in the sub subbands </h2>
498   this loop includes the interpolation of the coefficients Hxx
499   ************************************************************************************************/
500 
501   /******************************************************/
502   /* construct stereo sub subband signals according to: */
503   /*                                                    */
504   /* l_k(n) = H11(k,n) s_k(n) + H21(k,n) d_k(n)         */
505   /* r_k(n) = H12(k,n) s_k(n) + H22(k,n) d_k(n)         */
506   /******************************************************/
507   PS_DEC_COEFFICIENTS *pCoef = h_ps_d->specificTo.mpeg.pCoef;
508 
509   for (group = 0; group < NO_IID_GROUPS; group++) {
510     pCoef->H11r[group] += pCoef->DeltaH11r[group];
511     pCoef->H12r[group] += pCoef->DeltaH12r[group];
512     pCoef->H21r[group] += pCoef->DeltaH21r[group];
513     pCoef->H22r[group] += pCoef->DeltaH22r[group];
514 
515     const int start = groupTable[group];
516     const int stop = groupTable[group + 1];
517     for (subband = start; subband < stop; subband++) {
518       FIXP_DBL tmpLeft =
519           fMultAdd(fMultDiv2(pCoef->H11r[group], mHybridRealLeft[subband]),
520                    pCoef->H21r[group], mHybridRealRight[subband]);
521       FIXP_DBL tmpRight =
522           fMultAdd(fMultDiv2(pCoef->H12r[group], mHybridRealLeft[subband]),
523                    pCoef->H22r[group], mHybridRealRight[subband]);
524       mHybridRealLeft[subband] = tmpLeft;
525       mHybridRealRight[subband] = tmpRight;
526 
527       tmpLeft =
528           fMultAdd(fMultDiv2(pCoef->H11r[group], mHybridImagLeft[subband]),
529                    pCoef->H21r[group], mHybridImagRight[subband]);
530       tmpRight =
531           fMultAdd(fMultDiv2(pCoef->H12r[group], mHybridImagLeft[subband]),
532                    pCoef->H22r[group], mHybridImagRight[subband]);
533       mHybridImagLeft[subband] = tmpLeft;
534       mHybridImagRight[subband] = tmpRight;
535     } /* subband */
536   }
537 }
538 
539 /***************************************************************************/
540 /*!
541   \brief  Applies IID, ICC, IPD and OPD parameters to the current frame.
542 
543   \return none
544 
545 ****************************************************************************/
ApplyPsSlot(HANDLE_PS_DEC h_ps_d,FIXP_DBL ** rIntBufferLeft,FIXP_DBL ** iIntBufferLeft,FIXP_DBL * rIntBufferRight,FIXP_DBL * iIntBufferRight,const int scaleFactorLowBand_no_ov,const int scaleFactorLowBand,const int scaleFactorHighBand,const int lsb,const int usb)546 void ApplyPsSlot(
547     HANDLE_PS_DEC h_ps_d,      /*!< handle PS_DEC*/
548     FIXP_DBL **rIntBufferLeft, /*!< real bands left qmf channel (38x64)  */
549     FIXP_DBL **iIntBufferLeft, /*!< imag bands left qmf channel (38x64)  */
550     FIXP_DBL *rIntBufferRight, /*!< real bands right qmf channel (38x64) */
551     FIXP_DBL *iIntBufferRight, /*!< imag bands right qmf channel (38x64) */
552     const int scaleFactorLowBand_no_ov, const int scaleFactorLowBand,
553     const int scaleFactorHighBand, const int lsb, const int usb) {
554 /*!
555 The 64-band QMF representation of the monaural signal generated by the SBR tool
556 is used as input of the PS tool. After the PS processing, the outputs of the
557 left and right hybrid synthesis filterbanks are used to generate the stereo
558 output signal.
559 
560 <pre>
561 
562            -------------            ----------            -------------
563           | Hybrid      | M_n[k,m] |          | L_n[k,m] | Hybrid      | l[n]
564  m[n] --->| analysis    |--------->|          |--------->| synthesis   |----->
565            -------------           | Stereo   |           -------------
566                  |                 | recon-   |
567                  |                 | stuction |
568                 \|/                |          |
569            -------------           |          |
570           | De-         | D_n[k,m] |          |
571           | correlation |--------->|          |
572            -------------           |          |           -------------
573                                    |          | R_n[k,m] | Hybrid      | r[n]
574                                    |          |--------->| synthesis   |----->
575  IID, ICC ------------------------>|          |          | filter bank |
576 (IPD, OPD)                          ----------            -------------
577 
578 m[n]:      QMF represantation of the mono input
579 M_n[k,m]:  (sub-)sub-band domain signals of the mono input
580 D_n[k,m]:  decorrelated (sub-)sub-band domain signals
581 L_n[k,m]:  (sub-)sub-band domain signals of the left output
582 R_n[k,m]:  (sub-)sub-band domain signals of the right output
583 l[n],r[n]: left/right output signals
584 
585 </pre>
586 */
587 #define NO_HYBRID_DATA_BANDS (71)
588 
589   int i;
590   FIXP_DBL qmfInputData[2][NO_QMF_BANDS_HYBRID20];
591   FIXP_DBL *hybridData[2][2];
592   C_ALLOC_SCRATCH_START(pHybridData, FIXP_DBL, 4 * NO_HYBRID_DATA_BANDS);
593 
594   hybridData[0][0] =
595       pHybridData + 0 * NO_HYBRID_DATA_BANDS; /* left real hybrid data */
596   hybridData[0][1] =
597       pHybridData + 1 * NO_HYBRID_DATA_BANDS; /* left imag hybrid data */
598   hybridData[1][0] =
599       pHybridData + 2 * NO_HYBRID_DATA_BANDS; /* right real hybrid data */
600   hybridData[1][1] =
601       pHybridData + 3 * NO_HYBRID_DATA_BANDS; /* right imag hybrid data */
602 
603   /*!
604   Hybrid analysis filterbank:
605   The lower 3 (5) of the 64 QMF subbands are further split to provide better
606   frequency resolution. for PS processing. For the 10 and 20 stereo bands
607   configuration, the QMF band H_0(w) is split up into 8 (sub-) sub-bands and the
608   QMF bands H_1(w) and H_2(w) are spit into 2 (sub-) 4th. (See figures 8.20
609   and 8.22 of ISO/IEC 14496-3:2001/FDAM 2:2004(E) )
610   */
611 
612   /*
613    * Hybrid analysis.
614    */
615 
616   /* Get qmf input data and apply descaling */
617   for (i = 0; i < NO_QMF_BANDS_HYBRID20; i++) {
618     qmfInputData[0][i] = scaleValue(rIntBufferLeft[HYBRID_FILTER_DELAY][i],
619                                     scaleFactorLowBand_no_ov);
620     qmfInputData[1][i] = scaleValue(iIntBufferLeft[HYBRID_FILTER_DELAY][i],
621                                     scaleFactorLowBand_no_ov);
622   }
623 
624   /* LF - part */
625   FDKhybridAnalysisApply(&h_ps_d->specificTo.mpeg.hybridAnalysis,
626                          qmfInputData[0], qmfInputData[1], hybridData[0][0],
627                          hybridData[0][1]);
628 
629   /* HF - part */
630   /* bands up to lsb */
631   scaleValues(&hybridData[0][0][NO_SUB_QMF_CHANNELS - 2],
632               &rIntBufferLeft[0][NO_QMF_BANDS_HYBRID20],
633               lsb - NO_QMF_BANDS_HYBRID20, scaleFactorLowBand);
634   scaleValues(&hybridData[0][1][NO_SUB_QMF_CHANNELS - 2],
635               &iIntBufferLeft[0][NO_QMF_BANDS_HYBRID20],
636               lsb - NO_QMF_BANDS_HYBRID20, scaleFactorLowBand);
637 
638   /* bands from lsb to usb */
639   scaleValues(&hybridData[0][0][lsb + (NO_SUB_QMF_CHANNELS - 2 -
640                                        NO_QMF_BANDS_HYBRID20)],
641               &rIntBufferLeft[0][lsb], usb - lsb, scaleFactorHighBand);
642   scaleValues(&hybridData[0][1][lsb + (NO_SUB_QMF_CHANNELS - 2 -
643                                        NO_QMF_BANDS_HYBRID20)],
644               &iIntBufferLeft[0][lsb], usb - lsb, scaleFactorHighBand);
645 
646   /* bands from usb to NO_SUB_QMF_CHANNELS which should be zero for non-overlap
647      slots but can be non-zero for overlap slots */
648   FDKmemcpy(
649       &hybridData[0][0]
650                  [usb + (NO_SUB_QMF_CHANNELS - 2 - NO_QMF_BANDS_HYBRID20)],
651       &rIntBufferLeft[0][usb], sizeof(FIXP_DBL) * (NO_QMF_CHANNELS - usb));
652   FDKmemcpy(
653       &hybridData[0][1]
654                  [usb + (NO_SUB_QMF_CHANNELS - 2 - NO_QMF_BANDS_HYBRID20)],
655       &iIntBufferLeft[0][usb], sizeof(FIXP_DBL) * (NO_QMF_CHANNELS - usb));
656 
657   /*!
658   Decorrelation:
659   By means of all-pass filtering and delaying, the (sub-)sub-band samples s_k(n)
660   are converted into de-correlated (sub-)sub-band samples d_k(n).
661   - k: frequency in hybrid spectrum
662   - n: time index
663   */
664 
665   FDKdecorrelateApply(&h_ps_d->specificTo.mpeg.apDecor,
666                       &hybridData[0][0][0], /* left real hybrid data */
667                       &hybridData[0][1][0], /* left imag hybrid data */
668                       &hybridData[1][0][0], /* right real hybrid data */
669                       &hybridData[1][1][0], /* right imag hybrid data */
670                       0                     /* startHybBand */
671   );
672 
673   /*!
674   Stereo Processing:
675   The sets of (sub-)sub-band samples s_k(n) and d_k(n) are processed according
676   to the stereo cues which are defined per stereo band.
677   */
678 
679   applySlotBasedRotation(h_ps_d,
680                          &hybridData[0][0][0], /* left real hybrid data */
681                          &hybridData[0][1][0], /* left imag hybrid data */
682                          &hybridData[1][0][0], /* right real hybrid data */
683                          &hybridData[1][1][0]  /* right imag hybrid data */
684   );
685 
686   /*!
687   Hybrid synthesis filterbank:
688   The stereo processed hybrid subband signals l_k(n) and r_k(n) are fed into the
689   hybrid synthesis filterbanks which are identical to the 64 complex synthesis
690   filterbank of the SBR tool. The input to the filterbank are slots of 64 QMF
691   samples. For each slot the filterbank outputs one block of 64 samples of one
692   reconstructed stereo channel. The hybrid synthesis filterbank is computed
693   seperatly for the left and right channel.
694   */
695 
696   /*
697    * Hybrid synthesis.
698    */
699   for (i = 0; i < 2; i++) {
700     FDKhybridSynthesisApply(
701         &h_ps_d->specificTo.mpeg.hybridSynthesis[i],
702         hybridData[i][0], /* real hybrid data */
703         hybridData[i][1], /* imag hybrid data */
704         (i == 0) ? rIntBufferLeft[0]
705                  : rIntBufferRight, /* output real qmf buffer */
706         (i == 0) ? iIntBufferLeft[0]
707                  : iIntBufferRight /* output imag qmf buffer */
708     );
709   }
710 
711   /* free temporary hybrid qmf values of one timeslot */
712   C_ALLOC_SCRATCH_END(pHybridData, FIXP_DBL, 4 * NO_HYBRID_DATA_BANDS);
713 
714 } /* END ApplyPsSlot */
715