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1 /* -----------------------------------------------------------------------------
2 Software License for The Fraunhofer FDK AAC Codec Library for Android
3 
4 © Copyright  1995 - 2019 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 /******************* Library for basic calculation routines ********************
96 
97    Author(s):   Markus Lohwasser, Josef Hoepfl, Manuel Jander
98 
99    Description: QMF filterbank
100 
101 *******************************************************************************/
102 
103 #ifndef QMF_PCM_H
104 #define QMF_PCM_H
105 
106 /*
107    All Synthesis functions dependent on datatype INT_PCM_QMFOUT
108    Should only be included by qmf.cpp, but not compiled separately, please
109    exclude compilation from project, if done otherwise. Is optional included
110    twice to duplicate all functions with two different pre-definitions, as:
111         #define INT_PCM_QMFOUT LONG
112     and ...
113         #define INT_PCM_QMFOUT SHORT
114     needed to run QMF synthesis in both 16bit and 32bit sample output format.
115 */
116 
117 #define QSSCALE (0)
118 #define FX_DBL2FX_QSS(x) (x)
119 #define FX_QSS2FX_DBL(x) (x)
120 
121 /*!
122   \brief Perform Synthesis Prototype Filtering on a single slot of input data.
123 
124   The filter takes 2 * qmf->no_channels of input data and
125   generates qmf->no_channels time domain output samples.
126 */
127 /* static */
128 #ifndef FUNCTION_qmfSynPrototypeFirSlot
qmfSynPrototypeFirSlot(HANDLE_QMF_FILTER_BANK qmf,FIXP_DBL * RESTRICT realSlot,FIXP_DBL * RESTRICT imagSlot,INT_PCM_QMFOUT * RESTRICT timeOut,int stride)129 void qmfSynPrototypeFirSlot(
130 #else
131 void qmfSynPrototypeFirSlot_fallback(
132 #endif
133     HANDLE_QMF_FILTER_BANK qmf,
134     FIXP_DBL *RESTRICT realSlot,      /*!< Input: Pointer to real Slot */
135     FIXP_DBL *RESTRICT imagSlot,      /*!< Input: Pointer to imag Slot */
136     INT_PCM_QMFOUT *RESTRICT timeOut, /*!< Time domain data */
137     int stride) {
138   FIXP_QSS *FilterStates = (FIXP_QSS *)qmf->FilterStates;
139   int no_channels = qmf->no_channels;
140   const FIXP_PFT *p_Filter = qmf->p_filter;
141   int p_stride = qmf->p_stride;
142   int j;
143   FIXP_QSS *RESTRICT sta = FilterStates;
144   const FIXP_PFT *RESTRICT p_flt, *RESTRICT p_fltm;
145   int scale = (DFRACT_BITS - SAMPLE_BITS_QMFOUT) - 1 - qmf->outScalefactor -
146               qmf->outGain_e;
147 
148   p_flt =
149       p_Filter + p_stride * QMF_NO_POLY; /*                     5th of 330 */
150   p_fltm = p_Filter + (qmf->FilterSize / 2) -
151            p_stride * QMF_NO_POLY; /* 5 + (320 - 2*5) = 315th of 330 */
152 
153   FIXP_SGL gain = FX_DBL2FX_SGL(qmf->outGain_m);
154 
155   FIXP_DBL rnd_val = 0;
156 
157   if (scale > 0) {
158     if (scale < (DFRACT_BITS - 1))
159       rnd_val = FIXP_DBL(1 << (scale - 1));
160     else
161       scale = (DFRACT_BITS - 1);
162   } else {
163     scale = fMax(scale, -(DFRACT_BITS - 1));
164   }
165 
166   for (j = no_channels - 1; j >= 0; j--) {
167     FIXP_DBL imag = imagSlot[j]; /* no_channels-1 .. 0 */
168     FIXP_DBL real = realSlot[j]; /* no_channels-1 .. 0 */
169     {
170       INT_PCM_QMFOUT tmp;
171       FIXP_DBL Are = fMultAddDiv2(FX_QSS2FX_DBL(sta[0]), p_fltm[0], real);
172 
173       /* This PCM formatting performs:
174          - multiplication with 16-bit gain, if not -1.0f
175          - rounding, if shift right is applied
176          - apply shift left (or right) with saturation to 32 (or 16) bits
177          - store output with --stride in 32 (or 16) bit format
178       */
179       if (gain != (FIXP_SGL)(-32768)) /* -1.0f */
180       {
181         Are = fMult(Are, gain);
182       }
183       if (scale >= 0) {
184         FDK_ASSERT(
185             Are <=
186             (Are + rnd_val)); /* Round-addition must not overflow, might be
187                                  equal for rnd_val=0 */
188         tmp = (INT_PCM_QMFOUT)(
189             SATURATE_RIGHT_SHIFT(Are + rnd_val, scale, SAMPLE_BITS_QMFOUT));
190       } else {
191         tmp = (INT_PCM_QMFOUT)(
192             SATURATE_LEFT_SHIFT(Are, -scale, SAMPLE_BITS_QMFOUT));
193       }
194 
195       { timeOut[(j)*stride] = tmp; }
196     }
197 
198     sta[0] = FX_DBL2FX_QSS(fMultAddDiv2(FX_QSS2FX_DBL(sta[1]), p_flt[4], imag));
199     sta[1] =
200         FX_DBL2FX_QSS(fMultAddDiv2(FX_QSS2FX_DBL(sta[2]), p_fltm[1], real));
201     sta[2] = FX_DBL2FX_QSS(fMultAddDiv2(FX_QSS2FX_DBL(sta[3]), p_flt[3], imag));
202     sta[3] =
203         FX_DBL2FX_QSS(fMultAddDiv2(FX_QSS2FX_DBL(sta[4]), p_fltm[2], real));
204     sta[4] = FX_DBL2FX_QSS(fMultAddDiv2(FX_QSS2FX_DBL(sta[5]), p_flt[2], imag));
205     sta[5] =
206         FX_DBL2FX_QSS(fMultAddDiv2(FX_QSS2FX_DBL(sta[6]), p_fltm[3], real));
207     sta[6] = FX_DBL2FX_QSS(fMultAddDiv2(FX_QSS2FX_DBL(sta[7]), p_flt[1], imag));
208     sta[7] =
209         FX_DBL2FX_QSS(fMultAddDiv2(FX_QSS2FX_DBL(sta[8]), p_fltm[4], real));
210     sta[8] = FX_DBL2FX_QSS(fMultDiv2(p_flt[0], imag));
211     p_flt += (p_stride * QMF_NO_POLY);
212     p_fltm -= (p_stride * QMF_NO_POLY);
213     sta += 9;  // = (2*QMF_NO_POLY-1);
214   }
215 }
216 
217 #ifndef FUNCTION_qmfSynPrototypeFirSlot_NonSymmetric
218 /*!
219   \brief Perform Synthesis Prototype Filtering on a single slot of input data.
220 
221   The filter takes 2 * qmf->no_channels of input data and
222   generates qmf->no_channels time domain output samples.
223 */
qmfSynPrototypeFirSlot_NonSymmetric(HANDLE_QMF_FILTER_BANK qmf,FIXP_DBL * RESTRICT realSlot,FIXP_DBL * RESTRICT imagSlot,INT_PCM_QMFOUT * RESTRICT timeOut,int stride)224 static void qmfSynPrototypeFirSlot_NonSymmetric(
225     HANDLE_QMF_FILTER_BANK qmf,
226     FIXP_DBL *RESTRICT realSlot,      /*!< Input: Pointer to real Slot */
227     FIXP_DBL *RESTRICT imagSlot,      /*!< Input: Pointer to imag Slot */
228     INT_PCM_QMFOUT *RESTRICT timeOut, /*!< Time domain data */
229     int stride) {
230   FIXP_QSS *FilterStates = (FIXP_QSS *)qmf->FilterStates;
231   int no_channels = qmf->no_channels;
232   const FIXP_PFT *p_Filter = qmf->p_filter;
233   int p_stride = qmf->p_stride;
234   int j;
235   FIXP_QSS *RESTRICT sta = FilterStates;
236   const FIXP_PFT *RESTRICT p_flt, *RESTRICT p_fltm;
237   int scale = (DFRACT_BITS - SAMPLE_BITS_QMFOUT) - 1 - qmf->outScalefactor -
238               qmf->outGain_e;
239 
240   p_flt = p_Filter; /*!< Pointer to first half of filter coefficients */
241   p_fltm =
242       &p_flt[qmf->FilterSize / 2]; /* at index 320, overall 640 coefficients */
243 
244   FIXP_SGL gain = FX_DBL2FX_SGL(qmf->outGain_m);
245 
246   FIXP_DBL rnd_val = (FIXP_DBL)0;
247 
248   if (scale > 0) {
249     if (scale < (DFRACT_BITS - 1))
250       rnd_val = FIXP_DBL(1 << (scale - 1));
251     else
252       scale = (DFRACT_BITS - 1);
253   } else {
254     scale = fMax(scale, -(DFRACT_BITS - 1));
255   }
256 
257   for (j = no_channels - 1; j >= 0; j--) {
258     FIXP_DBL imag = imagSlot[j]; /* no_channels-1 .. 0 */
259     FIXP_DBL real = realSlot[j]; /* no_channels-1 .. 0 */
260     {
261       INT_PCM_QMFOUT tmp;
262       FIXP_DBL Are = sta[0] + FX_DBL2FX_QSS(fMultDiv2(p_fltm[4], real));
263 
264       /* This PCM formatting performs:
265          - multiplication with 16-bit gain, if not -1.0f
266          - rounding, if shift right is applied
267          - apply shift left (or right) with saturation to 32 (or 16) bits
268          - store output with --stride in 32 (or 16) bit format
269       */
270       if (gain != (FIXP_SGL)(-32768)) /* -1.0f */
271       {
272         Are = fMult(Are, gain);
273       }
274       if (scale > 0) {
275         FDK_ASSERT(Are <
276                    (Are + rnd_val)); /* Round-addition must not overflow */
277         tmp = (INT_PCM_QMFOUT)(
278             SATURATE_RIGHT_SHIFT(Are + rnd_val, scale, SAMPLE_BITS_QMFOUT));
279       } else {
280         tmp = (INT_PCM_QMFOUT)(
281             SATURATE_LEFT_SHIFT(Are, -scale, SAMPLE_BITS_QMFOUT));
282       }
283       timeOut[j * stride] = tmp;
284     }
285 
286     sta[0] = sta[1] + FX_DBL2FX_QSS(fMultDiv2(p_flt[4], imag));
287     sta[1] = sta[2] + FX_DBL2FX_QSS(fMultDiv2(p_fltm[3], real));
288     sta[2] = sta[3] + FX_DBL2FX_QSS(fMultDiv2(p_flt[3], imag));
289 
290     sta[3] = sta[4] + FX_DBL2FX_QSS(fMultDiv2(p_fltm[2], real));
291     sta[4] = sta[5] + FX_DBL2FX_QSS(fMultDiv2(p_flt[2], imag));
292     sta[5] = sta[6] + FX_DBL2FX_QSS(fMultDiv2(p_fltm[1], real));
293     sta[6] = sta[7] + FX_DBL2FX_QSS(fMultDiv2(p_flt[1], imag));
294 
295     sta[7] = sta[8] + FX_DBL2FX_QSS(fMultDiv2(p_fltm[0], real));
296     sta[8] = FX_DBL2FX_QSS(fMultDiv2(p_flt[0], imag));
297 
298     p_flt += (p_stride * QMF_NO_POLY);
299     p_fltm += (p_stride * QMF_NO_POLY);
300     sta += 9;  // = (2*QMF_NO_POLY-1);
301   }
302 }
303 #endif /* FUNCTION_qmfSynPrototypeFirSlot_NonSymmetric */
304 
qmfSynthesisFilteringSlot(HANDLE_QMF_FILTER_BANK synQmf,const FIXP_DBL * realSlot,const FIXP_DBL * imagSlot,const int scaleFactorLowBand,const int scaleFactorHighBand,INT_PCM_QMFOUT * timeOut,const int stride,FIXP_DBL * pWorkBuffer)305 void qmfSynthesisFilteringSlot(HANDLE_QMF_FILTER_BANK synQmf,
306                                const FIXP_DBL *realSlot,
307                                const FIXP_DBL *imagSlot,
308                                const int scaleFactorLowBand,
309                                const int scaleFactorHighBand,
310                                INT_PCM_QMFOUT *timeOut, const int stride,
311                                FIXP_DBL *pWorkBuffer) {
312   if (!(synQmf->flags & QMF_FLAG_LP))
313     qmfInverseModulationHQ(synQmf, realSlot, imagSlot, scaleFactorLowBand,
314                            scaleFactorHighBand, pWorkBuffer);
315   else {
316     if (synQmf->flags & QMF_FLAG_CLDFB) {
317       qmfInverseModulationLP_odd(synQmf, realSlot, scaleFactorLowBand,
318                                  scaleFactorHighBand, pWorkBuffer);
319     } else {
320       qmfInverseModulationLP_even(synQmf, realSlot, scaleFactorLowBand,
321                                   scaleFactorHighBand, pWorkBuffer);
322     }
323   }
324 
325   if (synQmf->flags & QMF_FLAG_NONSYMMETRIC) {
326     qmfSynPrototypeFirSlot_NonSymmetric(synQmf, pWorkBuffer,
327                                         pWorkBuffer + synQmf->no_channels,
328                                         timeOut, stride);
329   } else {
330     qmfSynPrototypeFirSlot(synQmf, pWorkBuffer,
331                            pWorkBuffer + synQmf->no_channels, timeOut, stride);
332   }
333 }
334 
335 /*!
336  *
337  * \brief Perform complex-valued subband synthesis of the
338  *        low band and the high band and store the
339  *        time domain data in timeOut
340  *
341  * First step: Calculate the proper scaling factor of current
342  * spectral data in qmfReal/qmfImag, old spectral data in the overlap
343  * range and filter states.
344  *
345  * Second step: Perform Frequency-to-Time mapping with inverse
346  * Modulation slot-wise.
347  *
348  * Third step: Perform FIR-filter slot-wise. To save space for filter
349  * states, the MAC operations are executed directly on the filter states
350  * instead of accumulating several products in the accumulator. The
351  * buffer shift at the end of the function should be replaced by a
352  * modulo operation, which is available on some DSPs.
353  *
354  * Last step: Copy the upper part of the spectral data to the overlap buffer.
355  *
356  * The qmf coefficient table is symmetric. The symmetry is exploited by
357  * shrinking the coefficient table to half the size. The addressing mode
358  * takes care of the symmetries.  If the #define #QMFTABLE_FULL is set,
359  * coefficient addressing works on the full table size. The code will be
360  * slightly faster and slightly more compact.
361  *
362  * Workbuffer requirement: 2 x sizeof(**QmfBufferReal) * synQmf->no_channels
363  * The workbuffer must be aligned
364  */
qmfSynthesisFiltering(HANDLE_QMF_FILTER_BANK synQmf,FIXP_DBL ** QmfBufferReal,FIXP_DBL ** QmfBufferImag,const QMF_SCALE_FACTOR * scaleFactor,const INT ov_len,INT_PCM_QMFOUT * timeOut,const INT stride,FIXP_DBL * pWorkBuffer)365 void qmfSynthesisFiltering(
366     HANDLE_QMF_FILTER_BANK synQmf, /*!< Handle of Qmf Synthesis Bank  */
367     FIXP_DBL **QmfBufferReal,      /*!< Low and High band, real */
368     FIXP_DBL **QmfBufferImag,      /*!< Low and High band, imag */
369     const QMF_SCALE_FACTOR *scaleFactor,
370     const INT ov_len,        /*!< split Slot of overlap and actual slots */
371     INT_PCM_QMFOUT *timeOut, /*!< Pointer to output */
372     const INT stride,        /*!< stride factor of output */
373     FIXP_DBL *pWorkBuffer    /*!< pointer to temporal working buffer */
374 ) {
375   int i;
376   int L = synQmf->no_channels;
377   int scaleFactorHighBand;
378   int scaleFactorLowBand_ov, scaleFactorLowBand_no_ov;
379 
380   FDK_ASSERT(synQmf->no_channels >= synQmf->lsb);
381   FDK_ASSERT(synQmf->no_channels >= synQmf->usb);
382 
383   /* adapt scaling */
384   scaleFactorHighBand = -ALGORITHMIC_SCALING_IN_ANALYSIS_FILTERBANK -
385                         scaleFactor->hb_scale - synQmf->filterScale;
386   scaleFactorLowBand_ov = -ALGORITHMIC_SCALING_IN_ANALYSIS_FILTERBANK -
387                           scaleFactor->ov_lb_scale - synQmf->filterScale;
388   scaleFactorLowBand_no_ov = -ALGORITHMIC_SCALING_IN_ANALYSIS_FILTERBANK -
389                              scaleFactor->lb_scale - synQmf->filterScale;
390 
391   for (i = 0; i < synQmf->no_col; i++) /* ----- no_col loop ----- */
392   {
393     const FIXP_DBL *QmfBufferImagSlot = NULL;
394 
395     int scaleFactorLowBand =
396         (i < ov_len) ? scaleFactorLowBand_ov : scaleFactorLowBand_no_ov;
397 
398     if (!(synQmf->flags & QMF_FLAG_LP)) QmfBufferImagSlot = QmfBufferImag[i];
399 
400     qmfSynthesisFilteringSlot(synQmf, QmfBufferReal[i], QmfBufferImagSlot,
401                               scaleFactorLowBand, scaleFactorHighBand,
402                               timeOut + (i * L * stride), stride, pWorkBuffer);
403   } /* no_col loop  i  */
404 }
405 
406 /*!
407  *
408  * \brief Create QMF filter bank instance
409  *
410  *
411  * \return 0 if successful
412  *
413  */
qmfInitAnalysisFilterBank(HANDLE_QMF_FILTER_BANK h_Qmf,FIXP_QAS * pFilterStates,int noCols,int lsb,int usb,int no_channels,int flags)414 int qmfInitAnalysisFilterBank(
415     HANDLE_QMF_FILTER_BANK h_Qmf, /*!< Returns handle */
416     FIXP_QAS *pFilterStates,      /*!< Handle to filter states */
417     int noCols,                   /*!< Number of timeslots per frame */
418     int lsb,                      /*!< lower end of QMF */
419     int usb,                      /*!< upper end of QMF */
420     int no_channels,              /*!< Number of channels (bands) */
421     int flags)                    /*!< Low Power flag */
422 {
423   int err = qmfInitFilterBank(h_Qmf, pFilterStates, noCols, lsb, usb,
424                               no_channels, flags, 0);
425   if (!(flags & QMF_FLAG_KEEP_STATES) && (h_Qmf->FilterStates != NULL)) {
426     FDKmemclear(h_Qmf->FilterStates,
427                 (2 * QMF_NO_POLY - 1) * h_Qmf->no_channels * sizeof(FIXP_QAS));
428   }
429 
430   FDK_ASSERT(h_Qmf->no_channels >= h_Qmf->lsb);
431 
432   return err;
433 }
434 
435 #ifndef FUNCTION_qmfAnaPrototypeFirSlot
436 /*!
437   \brief Perform Analysis Prototype Filtering on a single slot of input data.
438 */
qmfAnaPrototypeFirSlot(FIXP_DBL * analysisBuffer,INT no_channels,const FIXP_PFT * p_filter,INT p_stride,FIXP_QAS * RESTRICT pFilterStates)439 static void qmfAnaPrototypeFirSlot(
440     FIXP_DBL *analysisBuffer,
441     INT no_channels, /*!< Number channels of analysis filter */
442     const FIXP_PFT *p_filter, INT p_stride, /*!< Stride of analysis filter    */
443     FIXP_QAS *RESTRICT pFilterStates) {
444   INT k;
445 
446   FIXP_DBL accu;
447   const FIXP_PFT *RESTRICT p_flt = p_filter;
448   FIXP_DBL *RESTRICT pData_0 = analysisBuffer + 2 * no_channels - 1;
449   FIXP_DBL *RESTRICT pData_1 = analysisBuffer;
450 
451   FIXP_QAS *RESTRICT sta_0 = (FIXP_QAS *)pFilterStates;
452   FIXP_QAS *RESTRICT sta_1 =
453       (FIXP_QAS *)pFilterStates + (2 * QMF_NO_POLY * no_channels) - 1;
454   INT pfltStep = QMF_NO_POLY * (p_stride);
455   INT staStep1 = no_channels << 1;
456   INT staStep2 = (no_channels << 3) - 1; /* Rewind one less */
457 
458   /* FIR filters 127..64 0..63 */
459   for (k = 0; k < no_channels; k++) {
460     accu = fMultDiv2(p_flt[0], *sta_1);
461     sta_1 -= staStep1;
462     accu += fMultDiv2(p_flt[1], *sta_1);
463     sta_1 -= staStep1;
464     accu += fMultDiv2(p_flt[2], *sta_1);
465     sta_1 -= staStep1;
466     accu += fMultDiv2(p_flt[3], *sta_1);
467     sta_1 -= staStep1;
468     accu += fMultDiv2(p_flt[4], *sta_1);
469     *pData_1++ = (accu << 1);
470     sta_1 += staStep2;
471 
472     p_flt += pfltStep;
473     accu = fMultDiv2(p_flt[0], *sta_0);
474     sta_0 += staStep1;
475     accu += fMultDiv2(p_flt[1], *sta_0);
476     sta_0 += staStep1;
477     accu += fMultDiv2(p_flt[2], *sta_0);
478     sta_0 += staStep1;
479     accu += fMultDiv2(p_flt[3], *sta_0);
480     sta_0 += staStep1;
481     accu += fMultDiv2(p_flt[4], *sta_0);
482     *pData_0-- = (accu << 1);
483     sta_0 -= staStep2;
484   }
485 }
486 #endif /* !defined(FUNCTION_qmfAnaPrototypeFirSlot) */
487 
488 #ifndef FUNCTION_qmfAnaPrototypeFirSlot_NonSymmetric
489 /*!
490   \brief Perform Analysis Prototype Filtering on a single slot of input data.
491 */
qmfAnaPrototypeFirSlot_NonSymmetric(FIXP_DBL * analysisBuffer,int no_channels,const FIXP_PFT * p_filter,int p_stride,FIXP_QAS * RESTRICT pFilterStates)492 static void qmfAnaPrototypeFirSlot_NonSymmetric(
493     FIXP_DBL *analysisBuffer,
494     int no_channels, /*!< Number channels of analysis filter */
495     const FIXP_PFT *p_filter, int p_stride, /*!< Stride of analysis filter    */
496     FIXP_QAS *RESTRICT pFilterStates) {
497   const FIXP_PFT *RESTRICT p_flt = p_filter;
498   int p, k;
499 
500   for (k = 0; k < 2 * no_channels; k++) {
501     FIXP_DBL accu = (FIXP_DBL)0;
502 
503     p_flt += QMF_NO_POLY * (p_stride - 1);
504 
505     /*
506       Perform FIR-Filter
507     */
508     for (p = 0; p < QMF_NO_POLY; p++) {
509       accu += fMultDiv2(*p_flt++, pFilterStates[2 * no_channels * p]);
510     }
511     analysisBuffer[2 * no_channels - 1 - k] = (accu << 1);
512     pFilterStates++;
513   }
514 }
515 #endif /* FUNCTION_qmfAnaPrototypeFirSlot_NonSymmetric */
516 
517 /*
518  * \brief Perform one QMF slot analysis of the time domain data of timeIn
519  *        with specified stride and stores the real part of the subband
520  *        samples in rSubband, and the imaginary part in iSubband
521  *
522  *        Note: anaQmf->lsb can be greater than anaQmf->no_channels in case
523  *        of implicit resampling (USAC with reduced 3/4 core frame length).
524  */
qmfAnalysisFilteringSlot(HANDLE_QMF_FILTER_BANK anaQmf,FIXP_DBL * qmfReal,FIXP_DBL * qmfImag,const INT_PCM_QMFIN * RESTRICT timeIn,const int stride,FIXP_DBL * pWorkBuffer)525 void qmfAnalysisFilteringSlot(
526     HANDLE_QMF_FILTER_BANK anaQmf,        /*!< Handle of Qmf Synthesis Bank  */
527     FIXP_DBL *qmfReal,                    /*!< Low and High band, real */
528     FIXP_DBL *qmfImag,                    /*!< Low and High band, imag */
529     const INT_PCM_QMFIN *RESTRICT timeIn, /*!< Pointer to input */
530     const int stride,                     /*!< stride factor of input */
531     FIXP_DBL *pWorkBuffer /*!< pointer to temporal working buffer */
532 ) {
533   int offset = anaQmf->no_channels * (QMF_NO_POLY * 2 - 1);
534   /*
535     Feed time signal into oldest anaQmf->no_channels states
536   */
537   {
538     FIXP_QAS *FilterStatesAnaTmp = ((FIXP_QAS *)anaQmf->FilterStates) + offset;
539 
540     /* Feed and scale actual time in slot */
541     for (int i = anaQmf->no_channels >> 1; i != 0; i--) {
542       /* Place INT_PCM value left aligned in scaledTimeIn */
543       *FilterStatesAnaTmp++ = (FIXP_QAS)*timeIn;
544       timeIn += stride;
545       *FilterStatesAnaTmp++ = (FIXP_QAS)*timeIn;
546       timeIn += stride;
547     }
548   }
549 
550   if (anaQmf->flags & QMF_FLAG_NONSYMMETRIC) {
551     qmfAnaPrototypeFirSlot_NonSymmetric(pWorkBuffer, anaQmf->no_channels,
552                                         anaQmf->p_filter, anaQmf->p_stride,
553                                         (FIXP_QAS *)anaQmf->FilterStates);
554   } else {
555     qmfAnaPrototypeFirSlot(pWorkBuffer, anaQmf->no_channels, anaQmf->p_filter,
556                            anaQmf->p_stride, (FIXP_QAS *)anaQmf->FilterStates);
557   }
558 
559   if (anaQmf->flags & QMF_FLAG_LP) {
560     if (anaQmf->flags & QMF_FLAG_CLDFB)
561       qmfForwardModulationLP_odd(anaQmf, pWorkBuffer, qmfReal);
562     else
563       qmfForwardModulationLP_even(anaQmf, pWorkBuffer, qmfReal);
564 
565   } else {
566     qmfForwardModulationHQ(anaQmf, pWorkBuffer, qmfReal, qmfImag);
567   }
568   /*
569     Shift filter states
570 
571     Should be realized with modulo addressing on a DSP instead of a true buffer
572     shift
573   */
574   FDKmemmove(anaQmf->FilterStates,
575              (FIXP_QAS *)anaQmf->FilterStates + anaQmf->no_channels,
576              offset * sizeof(FIXP_QAS));
577 }
578 
579 /*!
580  *
581  * \brief Perform complex-valued subband filtering of the time domain
582  *        data of timeIn and stores the real part of the subband
583  *        samples in rAnalysis, and the imaginary part in iAnalysis
584  * The qmf coefficient table is symmetric. The symmetry is expoited by
585  * shrinking the coefficient table to half the size. The addressing mode
586  * takes care of the symmetries.
587  *
588  *
589  * \sa PolyphaseFiltering
590  */
qmfAnalysisFiltering(HANDLE_QMF_FILTER_BANK anaQmf,FIXP_DBL ** qmfReal,FIXP_DBL ** qmfImag,QMF_SCALE_FACTOR * scaleFactor,const INT_PCM_QMFIN * timeIn,const int timeIn_e,const int stride,FIXP_DBL * pWorkBuffer)591 void qmfAnalysisFiltering(
592     HANDLE_QMF_FILTER_BANK anaQmf, /*!< Handle of Qmf Analysis Bank */
593     FIXP_DBL **qmfReal,            /*!< Pointer to real subband slots */
594     FIXP_DBL **qmfImag,            /*!< Pointer to imag subband slots */
595     QMF_SCALE_FACTOR *scaleFactor,
596     const INT_PCM_QMFIN *timeIn, /*!< Time signal */
597     const int timeIn_e, const int stride,
598     FIXP_DBL *pWorkBuffer /*!< pointer to temporal working buffer */
599 ) {
600   int i;
601   int no_channels = anaQmf->no_channels;
602 
603   scaleFactor->lb_scale =
604       -ALGORITHMIC_SCALING_IN_ANALYSIS_FILTERBANK - timeIn_e;
605   scaleFactor->lb_scale -= anaQmf->filterScale;
606 
607   for (i = 0; i < anaQmf->no_col; i++) {
608     FIXP_DBL *qmfImagSlot = NULL;
609 
610     if (!(anaQmf->flags & QMF_FLAG_LP)) {
611       qmfImagSlot = qmfImag[i];
612     }
613 
614     qmfAnalysisFilteringSlot(anaQmf, qmfReal[i], qmfImagSlot, timeIn, stride,
615                              pWorkBuffer);
616 
617     timeIn += no_channels * stride;
618 
619   } /* no_col loop  i  */
620 }
621 #endif /* QMF_PCM_H */
622