Imported Debian version 0.1.3.1
[deb_fdk-aac.git] / libFDK / include / qmf.h
1
2 /* -----------------------------------------------------------------------------------------------------------
3 Software License for The Fraunhofer FDK AAC Codec Library for Android
4
5 © Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
6 All rights reserved.
7
8 1. INTRODUCTION
9 The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
10 the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
11 This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
12
13 AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
14 audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
15 independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
16 of the MPEG specifications.
17
18 Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
19 may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
20 individually for the purpose of encoding or decoding bit streams in products that are compliant with
21 the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
22 these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
23 software may already be covered under those patent licenses when it is used for those licensed purposes only.
24
25 Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
26 are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
27 applications information and documentation.
28
29 2. COPYRIGHT LICENSE
30
31 Redistribution and use in source and binary forms, with or without modification, are permitted without
32 payment of copyright license fees provided that you satisfy the following conditions:
33
34 You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
35 your modifications thereto in source code form.
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37 You must retain the complete text of this software license in the documentation and/or other materials
38 provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
39 You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
40 modifications thereto to recipients of copies in binary form.
41
42 The name of Fraunhofer may not be used to endorse or promote products derived from this library without
43 prior written permission.
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45 You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
46 software or your modifications thereto.
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48 Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
49 and the date of any change. For modified versions of the FDK AAC Codec, the term
50 "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
51 "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
52
53 3. NO PATENT LICENSE
54
55 NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
56 ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
57 respect to this software.
58
59 You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
60 by appropriate patent licenses.
61
62 4. DISCLAIMER
63
64 This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
65 "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
66 of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
67 CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
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69 or business interruption, however caused and on any theory of liability, whether in contract, strict
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71 advised of the possibility of such damage.
72
73 5. CONTACT INFORMATION
74
75 Fraunhofer Institute for Integrated Circuits IIS
76 Attention: Audio and Multimedia Departments - FDK AAC LL
77 Am Wolfsmantel 33
78 91058 Erlangen, Germany
79
80 www.iis.fraunhofer.de/amm
81 amm-info@iis.fraunhofer.de
82 ----------------------------------------------------------------------------------------------------------- */
83
84 /*!
85 \file qmf.h
86 \brief Complex qmf analysis/synthesis
87 \author Markus Werner
88
89 */
90 #ifndef __QMF_H
91 #define __QMF_H
92
93
94
95 #include "common_fix.h"
96 #include "FDK_tools_rom.h"
97 #include "dct.h"
98
99 /*
100 * Filter coefficient type definition
101 */
102 #ifdef QMF_DATA_16BIT
103 #define FIXP_QMF FIXP_SGL
104 #define FX_DBL2FX_QMF FX_DBL2FX_SGL
105 #define FX_QMF2FX_DBL FX_SGL2FX_DBL
106 #define QFRACT_BITS FRACT_BITS
107 #else
108 #define FIXP_QMF FIXP_DBL
109 #define FX_DBL2FX_QMF
110 #define FX_QMF2FX_DBL
111 #define QFRACT_BITS DFRACT_BITS
112 #endif
113
114 /* ARM neon optimized QMF analysis filter requires 32 bit input.
115 Implemented for RVCT only, currently disabled. See src/arm/qmf_arm.cpp:45 */
116 #define FIXP_QAS FIXP_PCM
117 #define QAS_BITS SAMPLE_BITS
118
119 #ifdef QMFSYN_STATES_16BIT
120 #define FIXP_QSS FIXP_SGL
121 #define QSS_BITS FRACT_BITS
122 #else
123 #define FIXP_QSS FIXP_DBL
124 #define QSS_BITS DFRACT_BITS
125 #endif
126
127 /* Flags for QMF intialization */
128 /* Low Power mode flag */
129 #define QMF_FLAG_LP 1
130 /* Filter is not symetric. This flag is set internally in the QMF initialization as required. */
131 #define QMF_FLAG_NONSYMMETRIC 2
132 /* Complex Low Delay Filter Bank (or std symmetric filter bank) */
133 #define QMF_FLAG_CLDFB 4
134 /* Flag indicating that the states should be kept. */
135 #define QMF_FLAG_KEEP_STATES 8
136 /* Complex Low Delay Filter Bank used in MPEG Surround Encoder */
137 #define QMF_FLAG_MPSLDFB 16
138 /* Complex Low Delay Filter Bank used in MPEG Surround Encoder allows a optimized calculation of the modulation in qmfForwardModulationHQ() */
139 #define QMF_FLAG_MPSLDFB_OPTIMIZE_MODULATION 32
140 /* Flag to indicate HE-AAC down-sampled SBR mode (decoder) -> adapt analysis post twiddling */
141 #define QMF_FLAG_DOWNSAMPLED 64
142
143
144 typedef struct
145 {
146 int lb_scale; /*!< Scale of low band area */
147 int ov_lb_scale; /*!< Scale of adjusted overlap low band area */
148 int hb_scale; /*!< Scale of high band area */
149 int ov_hb_scale; /*!< Scale of adjusted overlap high band area */
150 } QMF_SCALE_FACTOR;
151
152 struct QMF_FILTER_BANK
153 {
154 const FIXP_PFT *p_filter; /*!< Pointer to filter coefficients */
155
156 void *FilterStates; /*!< Pointer to buffer of filter states
157 FIXP_PCM in analyse and
158 FIXP_DBL in synthesis filter */
159 int FilterSize; /*!< Size of prototype filter. */
160 const FIXP_QTW *t_cos; /*!< Modulation tables. */
161 const FIXP_QTW *t_sin;
162 int filterScale; /*!< filter scale */
163
164 int no_channels; /*!< Total number of channels (subbands) */
165 int no_col; /*!< Number of time slots */
166 int lsb; /*!< Top of low subbands */
167 int usb; /*!< Top of high subbands */
168
169 int outScalefactor; /*!< Scale factor of output data (syn only) */
170 FIXP_DBL outGain; /*!< Gain output data (syn only) (init with 0x80000000 to ignore) */
171
172 UINT flags; /*!< flags */
173 UCHAR p_stride; /*!< Stride Factor of polyphase filters */
174
175 };
176
177 typedef struct QMF_FILTER_BANK *HANDLE_QMF_FILTER_BANK;
178
179 void
180 qmfAnalysisFiltering( HANDLE_QMF_FILTER_BANK anaQmf, /*!< Handle of Qmf Analysis Bank */
181 FIXP_QMF **qmfReal, /*!< Pointer to real subband slots */
182 FIXP_QMF **qmfImag, /*!< Pointer to imag subband slots */
183 QMF_SCALE_FACTOR *scaleFactor, /*!< Scale factors of QMF data */
184 const INT_PCM *timeIn, /*!< Time signal */
185 const int stride, /*!< Stride factor of audio data */
186 FIXP_QMF *pWorkBuffer /*!< pointer to temporal working buffer */
187 );
188
189 void
190 qmfSynthesisFiltering( HANDLE_QMF_FILTER_BANK synQmf, /*!< Handle of Qmf Synthesis Bank */
191 FIXP_QMF **QmfBufferReal, /*!< Pointer to real subband slots */
192 FIXP_QMF **QmfBufferImag, /*!< Pointer to imag subband slots */
193 const QMF_SCALE_FACTOR *scaleFactor, /*!< Scale factors of QMF data */
194 const int ov_len, /*!< Length of band overlap */
195 INT_PCM *timeOut, /*!< Time signal */
196 const int stride, /*!< Stride factor of audio data */
197 FIXP_QMF *pWorkBuffer /*!< pointer to temporal working buffer */
198 );
199
200 int
201 qmfInitAnalysisFilterBank( HANDLE_QMF_FILTER_BANK h_Qmf, /*!< QMF Handle */
202 FIXP_QAS *pFilterStates, /*!< Pointer to filter state buffer */
203 int noCols, /*!< Number of time slots */
204 int lsb, /*!< Number of lower bands */
205 int usb, /*!< Number of upper bands */
206 int no_channels, /*!< Number of critically sampled bands */
207 int flags); /*!< Flags */
208
209 void
210 qmfAnalysisFilteringSlot( HANDLE_QMF_FILTER_BANK anaQmf, /*!< Handle of Qmf Synthesis Bank */
211 FIXP_QMF *qmfReal, /*!< Low and High band, real */
212 FIXP_QMF *qmfImag, /*!< Low and High band, imag */
213 const INT_PCM *timeIn, /*!< Pointer to input */
214 const int stride, /*!< stride factor of input */
215 FIXP_QMF *pWorkBuffer /*!< pointer to temporal working buffer */
216 );
217
218 int
219 qmfInitSynthesisFilterBank( HANDLE_QMF_FILTER_BANK h_Qmf, /*!< QMF Handle */
220 FIXP_QSS *pFilterStates, /*!< Pointer to filter state buffer */
221 int noCols, /*!< Number of time slots */
222 int lsb, /*!< Number of lower bands */
223 int usb, /*!< Number of upper bands */
224 int no_channels, /*!< Number of critically sampled bands */
225 int flags); /*!< Flags */
226
227 void qmfSynthesisFilteringSlot( HANDLE_QMF_FILTER_BANK synQmf,
228 const FIXP_QMF *realSlot,
229 const FIXP_QMF *imagSlot,
230 const int scaleFactorLowBand,
231 const int scaleFactorHighBand,
232 INT_PCM *timeOut,
233 const int stride,
234 FIXP_QMF *pWorkBuffer);
235
236 void
237 qmfChangeOutScalefactor (HANDLE_QMF_FILTER_BANK synQmf, /*!< Handle of Qmf Synthesis Bank */
238 int outScalefactor /*!< New scaling factor for output data */
239 );
240
241 void
242 qmfChangeOutGain (HANDLE_QMF_FILTER_BANK synQmf, /*!< Handle of Qmf Synthesis Bank */
243 FIXP_DBL outputGain /*!< New gain for output data */
244 );
245
246
247
248 #endif /* __QMF_H */