Coverage Report

Created: 2025-07-01 06:21

/src/aac/libSBRenc/src/resampler.cpp
Line
Count
Source (jump to first uncovered line)
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
0
#define B1 0
118
0
#define B2 1
119
0
#define A1 2
120
0
#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
0
#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
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
0
{
298
0
  UINT i;
299
0
  const struct FILTER_PARAM *currentSet = NULL;
300
301
0
  FDKmemclear(DownSampler->downFilter.states,
302
0
              sizeof(DownSampler->downFilter.states));
303
0
  DownSampler->downFilter.ptr = 0;
304
305
  /*
306
    find applicable parameter set
307
  */
308
0
  currentSet = filter_paramSet[0];
309
0
  for (i = 1; i < sizeof(filter_paramSet) / sizeof(struct FILTER_PARAM *);
310
0
       i++) {
311
0
    if (filter_paramSet[i]->Wc <= Wc) {
312
0
      break;
313
0
    }
314
0
    currentSet = filter_paramSet[i];
315
0
  }
316
317
0
  DownSampler->downFilter.coeffa = currentSet->coeffa;
318
319
0
  DownSampler->downFilter.gain = currentSet->g;
320
0
  FDK_ASSERT(currentSet->noCoeffs <= MAXNR_SECTIONS * 2);
321
322
0
  DownSampler->downFilter.noCoeffs = currentSet->noCoeffs;
323
0
  DownSampler->delay = currentSet->delay;
324
0
  DownSampler->downFilter.Wc = currentSet->Wc;
325
326
0
  DownSampler->ratio = ratio;
327
0
  DownSampler->pending = ratio - 1;
328
0
  return (1);
329
0
}
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
341
static inline INT_PCM AdvanceFilter(
342
    LP_FILTER *downFilter, /*!< pointer to iir filter instance */
343
    INT_PCM *pInput,       /*!< input of filter                */
344
0
    int downRatio) {
345
0
  INT_PCM output;
346
0
  int i, n;
347
348
0
#define BIQUAD_SCALE 12
349
350
0
  FIXP_DBL y = FL2FXCONST_DBL(0.0f);
351
0
  FIXP_DBL input;
352
353
0
  for (n = 0; n < downRatio; n++) {
354
0
    FIXP_BQS(*states)[2] = downFilter->states;
355
0
    const FIXP_SGL *coeff = downFilter->coeffa;
356
0
    int s1, s2;
357
358
0
    s1 = downFilter->ptr;
359
0
    s2 = s1 ^ 1;
360
361
0
#if (SAMPLE_BITS == 16)
362
0
    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
0
    FIXP_BQS state1, state2, state1b, state2b;
370
371
0
    state1 = states[0][s1];
372
0
    state2 = states[0][s2];
373
374
    /* Loop over sections */
375
0
    for (i = 0; i < downFilter->noCoeffs; i++) {
376
0
      FIXP_DBL state0;
377
378
      /* Load merged states (from next section) */
379
0
      state1b = states[i + 1][s1];
380
0
      state2b = states[i + 1][s2];
381
382
0
      state0 = input + fMult(state1, coeff[B1]) + fMult(state2, coeff[B2]);
383
0
      y = state0 - fMult(state1b, coeff[A1]) - fMult(state2b, coeff[A2]);
384
385
      /* Store new feed forward merge state */
386
0
      states[i + 1][s2] = y << 1;
387
      /* Store new feed backward state */
388
0
      states[i][s2] = input << 1;
389
390
      /* Feedback output to next section. */
391
0
      input = y;
392
393
      /* Transfer merged states */
394
0
      state1 = state1b;
395
0
      state2 = state2b;
396
397
      /* Step to next coef set */
398
0
      coeff += BIQUAD_COEFSTEP;
399
0
    }
400
0
    downFilter->ptr ^= 1;
401
0
  }
402
  /* Apply global gain */
403
0
  y = fMult(y, downFilter->gain);
404
405
  /* Apply final gain/scaling to output */
406
0
#if (SAMPLE_BITS == 16)
407
0
  output = (INT_PCM)SATURATE_RIGHT_SHIFT(
408
0
      y + (FIXP_DBL)(1 << (DFRACT_BITS - SAMPLE_BITS - BIQUAD_SCALE - 1)),
409
0
      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
0
  return output;
417
0
}
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
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
0
) {
433
0
  INT i;
434
0
  *numOutSamples = 0;
435
436
0
  for (i = 0; i < numInSamples; i += DownSampler->ratio) {
437
0
    *outSamples = AdvanceFilter(&(DownSampler->downFilter), &inSamples[i],
438
0
                                DownSampler->ratio);
439
0
    outSamples++;
440
0
  }
441
0
  *numOutSamples = numInSamples / DownSampler->ratio;
442
443
0
  return 0;
444
0
}