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 /************************* MPEG-D DRC decoder library ************************** 96 97 Author(s): 98 99 Description: 100 101 *******************************************************************************/ 102 103 #ifndef DRCDEC_GAINDECODER_H 104 #define DRCDEC_GAINDECODER_H 105 106 #include "drcDecoder.h" 107 108 /* Definitions common to gainDecoder submodule */ 109 110 #define NUM_LNB_FRAMES \ 111 5 /* previous frame + this frame + one frame for DM_REGULAR_DELAY + (maximum \ 112 delaySamples)/frameSize */ 113 114 /* QMF64 */ 115 #define SUBBAND_NUM_BANDS_QMF64 64 116 #define SUBBAND_DOWNSAMPLING_FACTOR_QMF64 64 117 #define SUBBAND_ANALYSIS_DELAY_QMF64 320 118 119 /* QMF71 (according to ISO/IEC 23003-1:2007) */ 120 #define SUBBAND_NUM_BANDS_QMF71 71 121 #define SUBBAND_DOWNSAMPLING_FACTOR_QMF71 64 122 #define SUBBAND_ANALYSIS_DELAY_QMF71 320 + 384 123 124 /* STFT256 (according to ISO/IEC 23008-3:2015/AMD3) */ 125 #define SUBBAND_NUM_BANDS_STFT256 256 126 #define SUBBAND_DOWNSAMPLING_FACTOR_STFT256 256 127 #define SUBBAND_ANALYSIS_DELAY_STFT256 256 128 129 typedef enum { 130 GAIN_DEC_DRC1, 131 GAIN_DEC_DRC1_DRC2, 132 GAIN_DEC_DRC2, 133 GAIN_DEC_DRC3, 134 GAIN_DEC_DRC2_DRC3 135 } GAIN_DEC_LOCATION; 136 137 typedef struct { 138 FIXP_DBL gainLin; /* e = 7 */ 139 SHORT time; 140 } NODE_LIN; 141 142 typedef struct { 143 GAIN_INTERPOLATION_TYPE gainInterpolationType; 144 int nNodes[NUM_LNB_FRAMES]; /* number of nodes, saturated to 16 */ 145 NODE_LIN linearNode[NUM_LNB_FRAMES][16]; 146 } LINEAR_NODE_BUFFER; 147 148 typedef struct { 149 int lnbPointer; 150 LINEAR_NODE_BUFFER linearNodeBuffer[12]; 151 LINEAR_NODE_BUFFER dummyLnb; 152 FIXP_DBL channelGain[8][NUM_LNB_FRAMES]; /* e = 8 */ 153 } DRC_GAIN_BUFFERS; 154 155 typedef struct { 156 int activeDrcOffset; 157 DRC_INSTRUCTIONS_UNI_DRC* pInst; 158 DRC_COEFFICIENTS_UNI_DRC* pCoef; 159 160 DUCKING_MODIFICATION duckingModificationForChannelGroup[8]; 161 SCHAR channelGroupForChannel[8]; 162 163 UCHAR bandCountForChannelGroup[8]; 164 UCHAR gainElementForGroup[8]; 165 UCHAR channelGroupIsParametricDrc[8]; 166 UCHAR gainElementCount; /* number of different DRC gains inluding all DRC 167 bands */ 168 int lnbIndexForChannel[8][NUM_LNB_FRAMES]; 169 int subbandGainsReady; 170 } ACTIVE_DRC; 171 172 typedef struct { 173 int deltaTminDefault; 174 INT frameSize; 175 FIXP_DBL loudnessNormalisationGainDb; 176 DELAY_MODE delayMode; 177 178 int nActiveDrcs; 179 ACTIVE_DRC activeDrc[MAX_ACTIVE_DRCS]; 180 int multiBandActiveDrcIndex; 181 int channelGainActiveDrcIndex; 182 FIXP_DBL channelGain[8]; /* e = 8 */ 183 184 DRC_GAIN_BUFFERS drcGainBuffers; 185 FIXP_DBL subbandGains[12][4 * 1024 / 256]; 186 FIXP_DBL dummySubbandGains[4 * 1024 / 256]; 187 188 int status; 189 int timeDomainSupported; 190 SUBBAND_DOMAIN_MODE subbandDomainSupported; 191 } DRC_GAIN_DECODER, *HANDLE_DRC_GAIN_DECODER; 192 193 /* init functions */ 194 DRC_ERROR 195 drcDec_GainDecoder_Open(HANDLE_DRC_GAIN_DECODER* phGainDec); 196 197 DRC_ERROR 198 drcDec_GainDecoder_Init(HANDLE_DRC_GAIN_DECODER hGainDec, const int frameSize, 199 const int sampleRate); 200 201 DRC_ERROR 202 drcDec_GainDecoder_SetCodecDependentParameters( 203 HANDLE_DRC_GAIN_DECODER hGainDec, const DELAY_MODE delayMode, 204 const int timeDomainSupported, 205 const SUBBAND_DOMAIN_MODE subbandDomainSupported); 206 207 DRC_ERROR 208 drcDec_GainDecoder_Config(HANDLE_DRC_GAIN_DECODER hGainDec, 209 HANDLE_UNI_DRC_CONFIG hUniDrcConfig, 210 const UCHAR numSelectedDrcSets, 211 const SCHAR* selectedDrcSetIds, 212 const UCHAR* selectedDownmixIds); 213 214 /* close functions */ 215 DRC_ERROR 216 drcDec_GainDecoder_Close(HANDLE_DRC_GAIN_DECODER* phGainDec); 217 218 /* process functions */ 219 220 /* call drcDec_GainDecoder_Preprocess first */ 221 DRC_ERROR 222 drcDec_GainDecoder_Preprocess(HANDLE_DRC_GAIN_DECODER hGainDec, 223 HANDLE_UNI_DRC_GAIN hUniDrcGain, 224 const FIXP_DBL loudnessNormalizationGainDb, 225 const FIXP_SGL boost, const FIXP_SGL compress); 226 227 /* Then call one of drcDec_GainDecoder_ProcessTimeDomain or 228 * drcDec_GainDecoder_ProcessSubbandDomain */ 229 DRC_ERROR 230 drcDec_GainDecoder_ProcessTimeDomain( 231 HANDLE_DRC_GAIN_DECODER hGainDec, const int delaySamples, 232 const GAIN_DEC_LOCATION drcLocation, const int channelOffset, 233 const int drcChannelOffset, const int numChannelsProcessed, 234 const int timeDataChannelOffset, FIXP_DBL* audioIOBuffer); 235 236 DRC_ERROR 237 drcDec_GainDecoder_ProcessSubbandDomain( 238 HANDLE_DRC_GAIN_DECODER hGainDec, const int delaySamples, 239 GAIN_DEC_LOCATION drcLocation, const int channelOffset, 240 const int drcChannelOffset, const int numChannelsProcessed, 241 const int processSingleTimeslot, FIXP_DBL* audioIOBufferReal[], 242 FIXP_DBL* audioIOBufferImag[]); 243 244 DRC_ERROR 245 drcDec_GainDecoder_Conceal(HANDLE_DRC_GAIN_DECODER hGainDec, 246 HANDLE_UNI_DRC_CONFIG hUniDrcConfig, 247 HANDLE_UNI_DRC_GAIN hUniDrcGain); 248 249 DRC_ERROR 250 drcDec_GainDecoder_SetLoudnessNormalizationGainDb( 251 HANDLE_DRC_GAIN_DECODER hGainDec, FIXP_DBL loudnessNormalizationGainDb); 252 253 int drcDec_GainDecoder_GetFrameSize(HANDLE_DRC_GAIN_DECODER hGainDec); 254 255 int drcDec_GainDecoder_GetDeltaTminDefault(HANDLE_DRC_GAIN_DECODER hGainDec); 256 257 void drcDec_GainDecoder_SetChannelGains(HANDLE_DRC_GAIN_DECODER hGainDec, 258 const int numChannels, 259 const int frameSize, 260 const FIXP_DBL* channelGainDb, 261 const int audioBufferChannelOffset, 262 FIXP_DBL* audioBuffer); 263 264 #endif 265