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/* -----------------------------------------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android
� Copyright 1995 - 2012 Fraunhofer-Gesellschaft zur F�rderung 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
----------------------------------------------------------------------------------------------------------- */
/*!
\file
\brief Low Power Profile Transposer,
*/
#ifndef _LPP_TRANS_H
#define _LPP_TRANS_H
#include "sbrdecoder.h"
#include "qmf.h"
/*
Common
*/
#define QMF_OUT_SCALE 8
/*
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_HEAAC 5
#define MAX_ENVELOPES MAX_ENVELOPES_HEAAC
#define MAX_FREQ_COEFFS 48
#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 lpcFilterStatesReal[LPC_ORDER][(32)]; /*!< pointer array to save filter states */
FIXP_DBL lpcFilterStatesImag[LPC_ORDER][(32)]; /*!< 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 timeStep,
const int firstSlotOffset,
const int lastSlotOffset,
const int nInvfBands,
INVF_MODE *sbr_invf_mode,
INVF_MODE *sbr_invf_mode_prev
);
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_TRANS_H */
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