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 /******************* 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 #endif /* QMF_PCM_H */
406