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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 encoder library ******************************
96 
97    Author(s):
98 
99    Description:
100 
101 *******************************************************************************/
102 
103 /*!
104   \file
105   \brief  FDK resampler tool box:$Revision: 91655 $
106   \author M. Werner
107 */
108 
109 #include "resampler.h"
110 
111 #include "genericStds.h"
112 
113 /**************************************************************************/
114 /*                   BIQUAD Filter Specifications                         */
115 /**************************************************************************/
116 
117 #define B1 0
118 #define B2 1
119 #define A1 2
120 #define A2 3
121 
122 #define BQC(x) FL2FXCONST_SGL(x / 2)
123 
124 struct FILTER_PARAM {
125   const FIXP_SGL *coeffa; /*! SOS matrix One row/section. Scaled using BQC().
126                              Order of coefficients: B1,B2,A1,A2. B0=A0=1.0 */
127   FIXP_DBL g;             /*! overall gain */
128   int Wc;       /*! normalized passband bandwidth at input samplerate * 1000 */
129   int noCoeffs; /*! number of filter coeffs */
130   int delay;    /*! delay in samples at input samplerate */
131 };
132 
133 #define BIQUAD_COEFSTEP 4
134 
135 /**
136  *\brief Low Pass
137  Wc = 0,5, order 30, Stop Band -96dB. Wc criteria is "almost 0dB passband", not
138  the usual -3db gain point. [b,a]=cheby2(30,96,0.505) [sos,g]=tf2sos(b,a)
139  bandwidth 0.48
140  */
141 static const FIXP_SGL sos48[] = {
142     BQC(1.98941075681938),      BQC(0.999999996890811),
143     BQC(0.863264527201963),     BQC(0.189553799960663),
144     BQC(1.90733804822445),      BQC(1.00000001736189),
145     BQC(0.836321575841691),     BQC(0.203505809266564),
146     BQC(1.75616665495325),      BQC(0.999999946079721),
147     BQC(0.784699225121588),     BQC(0.230471265506986),
148     BQC(1.55727745512726),      BQC(1.00000011737815),
149     BQC(0.712515423588351),     BQC(0.268752723900498),
150     BQC(1.33407591943643),      BQC(0.999999795953228),
151     BQC(0.625059117330989),     BQC(0.316194685288965),
152     BQC(1.10689898412458),      BQC(1.00000035057114),
153     BQC(0.52803514366398),      BQC(0.370517843224669),
154     BQC(0.89060371078454),      BQC(0.999999343962822),
155     BQC(0.426920462165257),     BQC(0.429608200207746),
156     BQC(0.694438261209433),     BQC(1.0000008629792),
157     BQC(0.326530699561716),     BQC(0.491714450654174),
158     BQC(0.523237800935322),     BQC(1.00000101349782),
159     BQC(0.230829556274851),     BQC(0.555559034843281),
160     BQC(0.378631165929563),     BQC(0.99998986482665),
161     BQC(0.142906422036095),     BQC(0.620338874442411),
162     BQC(0.260786911308437),     BQC(1.00003261460178),
163     BQC(0.0651008576256505),    BQC(0.685759923926262),
164     BQC(0.168409429188098),     BQC(0.999933049695828),
165     BQC(-0.000790067789975562), BQC(0.751905896602325),
166     BQC(0.100724533818628),     BQC(1.00009472669872),
167     BQC(-0.0533772830257041),   BQC(0.81930744384525),
168     BQC(0.0561434357867363),    BQC(0.999911636304276),
169     BQC(-0.0913550299236405),   BQC(0.88883625875915),
170     BQC(0.0341680678662057),    BQC(1.00003667508676),
171     BQC(-0.113405185536697),    BQC(0.961756638268446)};
172 
173 static const FIXP_DBL g48 =
174     FL2FXCONST_DBL(0.002712866530047) - (FIXP_DBL)0x8000;
175 
176 static const struct FILTER_PARAM param_set48 = {
177     sos48, g48, 480, 15, 4 /* LF 2 */
178 };
179 
180 /**
181  *\brief Low Pass
182  Wc = 0,5, order 24, Stop Band -96dB. Wc criteria is "almost 0dB passband", not
183  the usual -3db gain point. [b,a]=cheby2(24,96,0.5) [sos,g]=tf2sos(b,a)
184  bandwidth 0.45
185  */
186 static const FIXP_SGL sos45[] = {
187     BQC(1.982962601444),     BQC(1.00000000007504),    BQC(0.646113303737836),
188     BQC(0.10851149979981),   BQC(1.85334094281111),    BQC(0.999999999677192),
189     BQC(0.612073220102006),  BQC(0.130022141698044),   BQC(1.62541051415425),
190     BQC(1.00000000080398),   BQC(0.547879702855959),   BQC(0.171165825133192),
191     BQC(1.34554656923247),   BQC(0.9999999980169),     BQC(0.460373914508491),
192     BQC(0.228677463376354),  BQC(1.05656568503116),    BQC(1.00000000569363),
193     BQC(0.357891894038287),  BQC(0.298676843912185),   BQC(0.787967587877312),
194     BQC(0.999999984415017),  BQC(0.248826893211877),   BQC(0.377441803512978),
195     BQC(0.555480971120497),  BQC(1.00000003583307),    BQC(0.140614263345315),
196     BQC(0.461979302213679),  BQC(0.364986207070964),   BQC(0.999999932084303),
197     BQC(0.0392669446074516), BQC(0.55033451180825),    BQC(0.216827267631558),
198     BQC(1.00000010534682),   BQC(-0.0506232228865103), BQC(0.641691581560946),
199     BQC(0.108951672277119),  BQC(0.999999871167516),   BQC(-0.125584840183225),
200     BQC(0.736367748771803),  BQC(0.0387988607229035),  BQC(1.00000011205574),
201     BQC(-0.182814849097974), BQC(0.835802108714964),   BQC(0.0042866175809225),
202     BQC(0.999999954830813),  BQC(-0.21965740617151),   BQC(0.942623047782363)};
203 
204 static const FIXP_DBL g45 =
205     FL2FXCONST_DBL(0.00242743980909524) - (FIXP_DBL)0x8000;
206 
207 static const struct FILTER_PARAM param_set45 = {
208     sos45, g45, 450, 12, 4 /* LF 2 */
209 };
210 
211 /*
212  Created by Octave 2.1.73, Mon Oct 13 17:31:32 2008 CEST
213  Wc = 0,5, order 16, Stop Band -96dB damping.
214  [b,a]=cheby2(16,96,0.5)
215  [sos,g]=tf2sos(b,a)
216  bandwidth = 0.41
217  */
218 
219 static const FIXP_SGL sos41[] = {
220     BQC(1.96193625292),      BQC(0.999999999999964),   BQC(0.169266178786789),
221     BQC(0.0128823300475907), BQC(1.68913437662092),    BQC(1.00000000000053),
222     BQC(0.124751503206552),  BQC(0.0537472273950989),  BQC(1.27274692366017),
223     BQC(0.999999999995674),  BQC(0.0433108625178357),  BQC(0.131015753236317),
224     BQC(0.85214175088395),   BQC(1.00000000001813),    BQC(-0.0625658152550408),
225     BQC(0.237763778993806),  BQC(0.503841579939009),   BQC(0.999999999953223),
226     BQC(-0.179176128722865), BQC(0.367475236424474),   BQC(0.249990711986162),
227     BQC(1.00000000007952),   BQC(-0.294425165824676),  BQC(0.516594857170212),
228     BQC(0.087971668680286),  BQC(0.999999999915528),   BQC(-0.398956566777928),
229     BQC(0.686417767801123),  BQC(0.00965373325350294), BQC(1.00000000003744),
230     BQC(-0.48579173764817),  BQC(0.884931534239068)};
231 
232 static const FIXP_DBL g41 = FL2FXCONST_DBL(0.00155956951169248);
233 
234 static const struct FILTER_PARAM param_set41 = {
235     sos41, g41, 410, 8, 5 /* LF 3 */
236 };
237 
238 /*
239  # Created by Octave 2.1.73, Mon Oct 13 17:55:33 2008 CEST
240  Wc = 0,5, order 12, Stop Band -96dB damping.
241  [b,a]=cheby2(12,96,0.5);
242  [sos,g]=tf2sos(b,a)
243 */
244 static const FIXP_SGL sos35[] = {
245     BQC(1.93299325235762),   BQC(0.999999999999985),  BQC(-0.140733187246596),
246     BQC(0.0124139497836062), BQC(1.4890416764109),    BQC(1.00000000000011),
247     BQC(-0.198215402588504), BQC(0.0746730616584138), BQC(0.918450161309795),
248     BQC(0.999999999999619),  BQC(-0.30133912791941),  BQC(0.192276468839529),
249     BQC(0.454877024246818),  BQC(1.00000000000086),   BQC(-0.432337328809815),
250     BQC(0.356852933642815),  BQC(0.158017147118507),  BQC(0.999999999998876),
251     BQC(-0.574817494249777), BQC(0.566380436970833),  BQC(0.0171834649478749),
252     BQC(1.00000000000055),   BQC(-0.718581178041165), BQC(0.83367484487889)};
253 
254 static const FIXP_DBL g35 = FL2FXCONST_DBL(0.00162580994125131);
255 
256 static const struct FILTER_PARAM param_set35 = {sos35, g35, 350, 6, 4};
257 
258 /*
259  # Created by Octave 2.1.73, Mon Oct 13 18:15:38 2008 CEST
260  Wc = 0,5, order 8, Stop Band -96dB damping.
261  [b,a]=cheby2(8,96,0.5);
262  [sos,g]=tf2sos(b,a)
263 */
264 static const FIXP_SGL sos25[] = {
265     BQC(1.85334094301225),   BQC(1.0),
266     BQC(-0.702127214212663), BQC(0.132452403998767),
267     BQC(1.056565682167),     BQC(0.999999999999997),
268     BQC(-0.789503667880785), BQC(0.236328693569128),
269     BQC(0.364986307455489),  BQC(0.999999999999996),
270     BQC(-0.955191189843375), BQC(0.442966457936379),
271     BQC(0.0387985751642125), BQC(1.0),
272     BQC(-1.19817786088084),  BQC(0.770493895456328)};
273 
274 static const FIXP_DBL g25 = FL2FXCONST_DBL(0.000945182835294559);
275 
276 static const struct FILTER_PARAM param_set25 = {sos25, g25, 250, 4, 5};
277 
278 /* Must be sorted in descending order */
279 static const struct FILTER_PARAM *const filter_paramSet[] = {
280     &param_set48, &param_set45, &param_set41, &param_set35, &param_set25};
281 
282 /**************************************************************************/
283 /*                         Resampler Functions                            */
284 /**************************************************************************/
285 
286 /*!
287   \brief   Reset downsampler instance and clear delay lines
288 
289   \return  success of operation
290 */
291 
FDKaacEnc_InitDownsampler(DOWNSAMPLER * DownSampler,int Wc,int ratio)292 INT FDKaacEnc_InitDownsampler(
293     DOWNSAMPLER *DownSampler, /*!< pointer to downsampler instance */
294     int Wc,                   /*!< normalized cutoff freq * 1000*  */
295     int ratio)                /*!< downsampler ratio */
296 
297 {
298   UINT i;
299   const struct FILTER_PARAM *currentSet = NULL;
300 
301   FDKmemclear(DownSampler->downFilter.states,
302               sizeof(DownSampler->downFilter.states));
303   DownSampler->downFilter.ptr = 0;
304 
305   /*
306     find applicable parameter set
307   */
308   currentSet = filter_paramSet[0];
309   for (i = 1; i < sizeof(filter_paramSet) / sizeof(struct FILTER_PARAM *);
310        i++) {
311     if (filter_paramSet[i]->Wc <= Wc) {
312       break;
313     }
314     currentSet = filter_paramSet[i];
315   }
316 
317   DownSampler->downFilter.coeffa = currentSet->coeffa;
318 
319   DownSampler->downFilter.gain = currentSet->g;
320   FDK_ASSERT(currentSet->noCoeffs <= MAXNR_SECTIONS * 2);
321 
322   DownSampler->downFilter.noCoeffs = currentSet->noCoeffs;
323   DownSampler->delay = currentSet->delay;
324   DownSampler->downFilter.Wc = currentSet->Wc;
325 
326   DownSampler->ratio = ratio;
327   DownSampler->pending = ratio - 1;
328   return (1);
329 }
330 
331 /*!
332   \brief   faster simple folding operation
333            Filter:
334            H(z) = A(z)/B(z)
335            with
336            A(z) = a[0]*z^0 + a[1]*z^1 + a[2]*z^2 ... a[n]*z^n
337 
338   \return  filtered value
339 */
340 
AdvanceFilter(LP_FILTER * downFilter,INT_PCM * pInput,int downRatio)341 static inline INT_PCM AdvanceFilter(
342     LP_FILTER *downFilter, /*!< pointer to iir filter instance */
343     INT_PCM *pInput,       /*!< input of filter                */
344     int downRatio) {
345   INT_PCM output;
346   int i, n;
347 
348 #define BIQUAD_SCALE 12
349 
350   FIXP_DBL y = FL2FXCONST_DBL(0.0f);
351   FIXP_DBL input;
352 
353   for (n = 0; n < downRatio; n++) {
354     FIXP_BQS(*states)[2] = downFilter->states;
355     const FIXP_SGL *coeff = downFilter->coeffa;
356     int s1, s2;
357 
358     s1 = downFilter->ptr;
359     s2 = s1 ^ 1;
360 
361 #if (SAMPLE_BITS == 16)
362     input = ((FIXP_DBL)pInput[n]) << (DFRACT_BITS - SAMPLE_BITS - BIQUAD_SCALE);
363 #elif (SAMPLE_BITS == 32)
364     input = pInput[n] >> BIQUAD_SCALE;
365 #else
366 #error NOT IMPLEMENTED
367 #endif
368 
369     FIXP_BQS state1, state2, state1b, state2b;
370 
371     state1 = states[0][s1];
372     state2 = states[0][s2];
373 
374     /* Loop over sections */
375     for (i = 0; i < downFilter->noCoeffs; i++) {
376       FIXP_DBL state0;
377 
378       /* Load merged states (from next section) */
379       state1b = states[i + 1][s1];
380       state2b = states[i + 1][s2];
381 
382       state0 = input + fMult(state1, coeff[B1]) + fMult(state2, coeff[B2]);
383       y = state0 - fMult(state1b, coeff[A1]) - fMult(state2b, coeff[A2]);
384 
385       /* Store new feed forward merge state */
386       states[i + 1][s2] = y << 1;
387       /* Store new feed backward state */
388       states[i][s2] = input << 1;
389 
390       /* Feedback output to next section. */
391       input = y;
392 
393       /* Transfer merged states */
394       state1 = state1b;
395       state2 = state2b;
396 
397       /* Step to next coef set */
398       coeff += BIQUAD_COEFSTEP;
399     }
400     downFilter->ptr ^= 1;
401   }
402   /* Apply global gain */
403   y = fMult(y, downFilter->gain);
404 
405   /* Apply final gain/scaling to output */
406 #if (SAMPLE_BITS == 16)
407   output = (INT_PCM)SATURATE_RIGHT_SHIFT(
408       y + (FIXP_DBL)(1 << (DFRACT_BITS - SAMPLE_BITS - BIQUAD_SCALE - 1)),
409       DFRACT_BITS - SAMPLE_BITS - BIQUAD_SCALE, SAMPLE_BITS);
410   // output = (INT_PCM) SATURATE_RIGHT_SHIFT(y,
411   // DFRACT_BITS-SAMPLE_BITS-BIQUAD_SCALE, SAMPLE_BITS);
412 #else
413   output = SATURATE_LEFT_SHIFT(y, BIQUAD_SCALE, SAMPLE_BITS);
414 #endif
415 
416   return output;
417 }
418 
419 /*!
420   \brief   FDKaacEnc_Downsample numInSamples of type INT_PCM
421            Returns number of output samples in numOutSamples
422 
423   \return  success of operation
424 */
425 
FDKaacEnc_Downsample(DOWNSAMPLER * DownSampler,INT_PCM * inSamples,INT numInSamples,INT_PCM * outSamples,INT * numOutSamples)426 INT FDKaacEnc_Downsample(
427     DOWNSAMPLER *DownSampler, /*!< pointer to downsampler instance */
428     INT_PCM *inSamples,       /*!< pointer to input samples */
429     INT numInSamples,         /*!< number  of input samples  */
430     INT_PCM *outSamples,      /*!< pointer to output samples */
431     INT *numOutSamples        /*!< pointer tp number of output samples */
432 ) {
433   INT i;
434   *numOutSamples = 0;
435 
436   for (i = 0; i < numInSamples; i += DownSampler->ratio) {
437     *outSamples = AdvanceFilter(&(DownSampler->downFilter), &inSamples[i],
438                                 DownSampler->ratio);
439     outSamples++;
440   }
441   *numOutSamples = numInSamples / DownSampler->ratio;
442 
443   return 0;
444 }
445