| /* ----------------------------------------------------------------------------- |
| Software License for The Fraunhofer FDK AAC Codec Library for Android |
| |
| © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten |
| Forschung e.V. All rights reserved. |
| |
| 1. INTRODUCTION |
| The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software |
| that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding |
| scheme for digital audio. This FDK AAC Codec software is intended to be used on |
| a wide variety of Android devices. |
| |
| AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient |
| general perceptual audio codecs. AAC-ELD is considered the best-performing |
| full-bandwidth communications codec by independent studies and is widely |
| deployed. AAC has been standardized by ISO and IEC as part of the MPEG |
| specifications. |
| |
| Patent licenses for necessary patent claims for the FDK AAC Codec (including |
| those of Fraunhofer) may be obtained through Via Licensing |
| (www.vialicensing.com) or through the respective patent owners individually for |
| the purpose of encoding or decoding bit streams in products that are compliant |
| with the ISO/IEC MPEG audio standards. Please note that most manufacturers of |
| Android devices already license these patent claims through Via Licensing or |
| directly from the patent owners, and therefore FDK AAC Codec software may |
| already be covered under those patent licenses when it is used for those |
| licensed purposes only. |
| |
| Commercially-licensed AAC software libraries, including floating-point versions |
| with enhanced sound quality, are also available from Fraunhofer. Users are |
| encouraged to check the Fraunhofer website for additional applications |
| information and documentation. |
| |
| 2. COPYRIGHT LICENSE |
| |
| Redistribution and use in source and binary forms, with or without modification, |
| are permitted without payment of copyright license fees provided that you |
| satisfy the following conditions: |
| |
| You must retain the complete text of this software license in redistributions of |
| the FDK AAC Codec or your modifications thereto in source code form. |
| |
| You must retain the complete text of this software license in the documentation |
| and/or other materials provided with redistributions of the FDK AAC Codec or |
| your modifications thereto in binary form. You must make available free of |
| charge copies of the complete source code of the FDK AAC Codec and your |
| modifications thereto to recipients of copies in binary form. |
| |
| The name of Fraunhofer may not be used to endorse or promote products derived |
| from this library without prior written permission. |
| |
| You may not charge copyright license fees for anyone to use, copy or distribute |
| the FDK AAC Codec software or your modifications thereto. |
| |
| Your modified versions of the FDK AAC Codec must carry prominent notices stating |
| that you changed the software and the date of any change. For modified versions |
| of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" |
| must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK |
| AAC Codec Library for Android." |
| |
| 3. NO PATENT LICENSE |
| |
| NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without |
| limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. |
| Fraunhofer provides no warranty of patent non-infringement with respect to this |
| software. |
| |
| You may use this FDK AAC Codec software or modifications thereto only for |
| purposes that are authorized by appropriate patent licenses. |
| |
| 4. DISCLAIMER |
| |
| This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright |
| holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, |
| including but not limited to the implied warranties of merchantability and |
| fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR |
| CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, |
| or consequential damages, including but not limited to procurement of substitute |
| goods or services; loss of use, data, or profits, or business interruption, |
| however caused and on any theory of liability, whether in contract, strict |
| liability, or tort (including negligence), arising in any way out of the use of |
| this software, even if advised of the possibility of such damage. |
| |
| 5. CONTACT INFORMATION |
| |
| Fraunhofer Institute for Integrated Circuits IIS |
| Attention: Audio and Multimedia Departments - FDK AAC LL |
| Am Wolfsmantel 33 |
| 91058 Erlangen, Germany |
| |
| www.iis.fraunhofer.de/amm |
| amm-info@iis.fraunhofer.de |
| ----------------------------------------------------------------------------- */ |
| |
| /**************************** SBR decoder library ****************************** |
| |
| Author(s): |
| |
| Description: |
| |
| *******************************************************************************/ |
| |
| /*! |
| \file |
| \brief Low Power Profile Transposer |
| */ |
| |
| #ifndef LPP_TRAN_H |
| #define LPP_TRAN_H |
| |
| #include "sbrdecoder.h" |
| #include "hbe.h" |
| #include "qmf.h" |
| |
| /* |
| Common |
| */ |
| #define QMF_OUT_SCALE 8 |
| |
| /* |
| Frequency scales |
| */ |
| |
| /* |
| Env-Adjust |
| */ |
| #define MAX_NOISE_ENVELOPES 2 |
| #define MAX_NOISE_COEFFS 5 |
| #define MAX_NUM_NOISE_VALUES (MAX_NOISE_ENVELOPES * MAX_NOISE_COEFFS) |
| #define MAX_NUM_LIMITERS 12 |
| |
| /* Set MAX_ENVELOPES to the largest value of all supported BSFORMATs |
| by overriding MAX_ENVELOPES in the correct order: */ |
| #define MAX_ENVELOPES_LEGACY 5 |
| #define MAX_ENVELOPES_USAC 8 |
| #define MAX_ENVELOPES MAX_ENVELOPES_USAC |
| |
| #define MAX_FREQ_COEFFS_DUAL_RATE 48 |
| #define MAX_FREQ_COEFFS_QUAD_RATE 56 |
| #define MAX_FREQ_COEFFS MAX_FREQ_COEFFS_QUAD_RATE |
| |
| #define MAX_FREQ_COEFFS_FS44100 35 |
| #define MAX_FREQ_COEFFS_FS48000 32 |
| |
| #define MAX_NUM_ENVELOPE_VALUES (MAX_ENVELOPES * MAX_FREQ_COEFFS) |
| |
| #define MAX_GAIN_EXP 34 |
| /* Maximum gain will be sqrt(0.5 * 2^MAX_GAIN_EXP) |
| example: 34=99dB */ |
| #define MAX_GAIN_CONCEAL_EXP 1 |
| /* Maximum gain will be sqrt(0.5 * 2^MAX_GAIN_CONCEAL_EXP) in concealment case |
| * (0dB) */ |
| |
| /* |
| LPP Transposer |
| */ |
| #define LPC_ORDER 2 |
| |
| #define MAX_INVF_BANDS MAX_NOISE_COEFFS |
| |
| #define MAX_NUM_PATCHES 6 |
| #define SHIFT_START_SB 1 /*!< lowest subband of source range */ |
| |
| typedef enum { |
| INVF_OFF = 0, |
| INVF_LOW_LEVEL, |
| INVF_MID_LEVEL, |
| INVF_HIGH_LEVEL, |
| INVF_SWITCHED /* not a real choice but used here to control behaviour */ |
| } INVF_MODE; |
| |
| /** parameter set for one single patch */ |
| typedef struct { |
| UCHAR sourceStartBand; /*!< first band in lowbands where to take the samples |
| from */ |
| UCHAR |
| sourceStopBand; /*!< first band in lowbands which is not included in the |
| patch anymore */ |
| UCHAR guardStartBand; /*!< first band in highbands to be filled with zeros in |
| order to reduce interferences between patches */ |
| UCHAR |
| targetStartBand; /*!< first band in highbands to be filled with whitened |
| lowband signal */ |
| UCHAR targetBandOffs; /*!< difference between 'startTargetBand' and |
| 'startSourceBand' */ |
| UCHAR numBandsInPatch; /*!< number of consecutive bands in this one patch */ |
| } PATCH_PARAM; |
| |
| /** whitening factors for different levels of whitening |
| need to be initialized corresponding to crossover frequency */ |
| typedef struct { |
| FIXP_DBL off; /*!< bw factor for signal OFF */ |
| FIXP_DBL transitionLevel; |
| FIXP_DBL lowLevel; /*!< bw factor for signal LOW_LEVEL */ |
| FIXP_DBL midLevel; /*!< bw factor for signal MID_LEVEL */ |
| FIXP_DBL highLevel; /*!< bw factor for signal HIGH_LEVEL */ |
| } WHITENING_FACTORS; |
| |
| /*! The transposer settings are calculated on a header reset and are shared by |
| * both channels. */ |
| typedef struct { |
| UCHAR nCols; /*!< number subsamples of a codec frame */ |
| UCHAR noOfPatches; /*!< number of patches */ |
| UCHAR lbStartPatching; /*!< first band of lowbands that will be patched */ |
| UCHAR lbStopPatching; /*!< first band that won't be patched anymore*/ |
| UCHAR bwBorders[MAX_NUM_NOISE_VALUES]; /*!< spectral bands with different |
| inverse filtering levels */ |
| |
| PATCH_PARAM |
| patchParam[MAX_NUM_PATCHES]; /*!< new parameter set for patching */ |
| WHITENING_FACTORS |
| whFactors; /*!< the pole moving factors for certain |
| whitening levels as indicated in the bitstream |
| depending on the crossover frequency */ |
| UCHAR overlap; /*!< Overlap size */ |
| } TRANSPOSER_SETTINGS; |
| |
| typedef struct { |
| TRANSPOSER_SETTINGS *pSettings; /*!< Common settings for both channels */ |
| FIXP_DBL |
| bwVectorOld[MAX_NUM_PATCHES]; /*!< pole moving factors of past frame */ |
| FIXP_DBL lpcFilterStatesRealLegSBR[LPC_ORDER + (3 * (4))][( |
| 32)]; /*!< pointer array to save filter states */ |
| |
| FIXP_DBL lpcFilterStatesImagLegSBR[LPC_ORDER + (3 * (4))][( |
| 32)]; /*!< pointer array to save filter states */ |
| |
| FIXP_DBL lpcFilterStatesRealHBE[LPC_ORDER + (3 * (4))][( |
| 64)]; /*!< pointer array to save filter states */ |
| FIXP_DBL lpcFilterStatesImagHBE[LPC_ORDER + (3 * (4))][( |
| 64)]; /*!< pointer array to save filter states */ |
| } SBR_LPP_TRANS; |
| |
| typedef SBR_LPP_TRANS *HANDLE_SBR_LPP_TRANS; |
| |
| void lppTransposer(HANDLE_SBR_LPP_TRANS hLppTrans, |
| QMF_SCALE_FACTOR *sbrScaleFactor, FIXP_DBL **qmfBufferReal, |
| |
| FIXP_DBL *degreeAlias, FIXP_DBL **qmfBufferImag, |
| const int useLP, const int fPreWhitening, |
| const int v_k_master0, const int timeStep, |
| const int firstSlotOffset, const int lastSlotOffset, |
| const int nInvfBands, INVF_MODE *sbr_invf_mode, |
| INVF_MODE *sbr_invf_mode_prev); |
| |
| void lppTransposerHBE( |
| HANDLE_SBR_LPP_TRANS hLppTrans, /*!< Handle of lpp transposer */ |
| HANDLE_HBE_TRANSPOSER hQmfTransposer, |
| QMF_SCALE_FACTOR *sbrScaleFactor, /*!< Scaling factors */ |
| FIXP_DBL **qmfBufferReal, /*!< Pointer to pointer to real part of subband |
| samples (source) */ |
| FIXP_DBL **qmfBufferImag, /*!< Pointer to pointer to imaginary part of |
| subband samples (source) */ |
| const int timeStep, /*!< Time step of envelope */ |
| const int firstSlotOffs, /*!< Start position in time */ |
| const int lastSlotOffs, /*!< Number of overlap-slots into next frame */ |
| const int nInvfBands, /*!< Number of bands for inverse filtering */ |
| INVF_MODE *sbr_invf_mode, /*!< Current inverse filtering modes */ |
| INVF_MODE *sbr_invf_mode_prev /*!< Previous inverse filtering modes */ |
| ); |
| |
| SBR_ERROR |
| createLppTransposer(HANDLE_SBR_LPP_TRANS hLppTrans, |
| TRANSPOSER_SETTINGS *pSettings, const int highBandStartSb, |
| UCHAR *v_k_master, const int numMaster, const int usb, |
| const int timeSlots, const int nCols, UCHAR *noiseBandTable, |
| const int noNoiseBands, UINT fs, const int chan, |
| const int overlap); |
| |
| SBR_ERROR |
| resetLppTransposer(HANDLE_SBR_LPP_TRANS hLppTrans, UCHAR highBandStartSb, |
| UCHAR *v_k_master, UCHAR numMaster, UCHAR *noiseBandTable, |
| UCHAR noNoiseBands, UCHAR usb, UINT fs); |
| |
| #endif /* LPP_TRAN_H */ |