/* -----------------------------------------------------------------------------
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Software License for The Fraunhofer FDK AAC Codec Library for Android
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© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
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Forschung e.V. All rights reserved.
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1. INTRODUCTION
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The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software
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that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding
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scheme for digital audio. This FDK AAC Codec software is intended to be used on
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a wide variety of Android devices.
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AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient
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general perceptual audio codecs. AAC-ELD is considered the best-performing
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full-bandwidth communications codec by independent studies and is widely
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deployed. AAC has been standardized by ISO and IEC as part of the MPEG
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specifications.
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Patent licenses for necessary patent claims for the FDK AAC Codec (including
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those of Fraunhofer) may be obtained through Via Licensing
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(www.vialicensing.com) or through the respective patent owners individually for
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the purpose of encoding or decoding bit streams in products that are compliant
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with the ISO/IEC MPEG audio standards. Please note that most manufacturers of
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Android devices already license these patent claims through Via Licensing or
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directly from the patent owners, and therefore FDK AAC Codec software may
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already be covered under those patent licenses when it is used for those
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licensed purposes only.
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Commercially-licensed AAC software libraries, including floating-point versions
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with enhanced sound quality, are also available from Fraunhofer. Users are
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encouraged to check the Fraunhofer website for additional applications
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information and documentation.
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2. COPYRIGHT LICENSE
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Redistribution and use in source and binary forms, with or without modification,
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are permitted without payment of copyright license fees provided that you
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satisfy the following conditions:
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You must retain the complete text of this software license in redistributions of
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the FDK AAC Codec or your modifications thereto in source code form.
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You must retain the complete text of this software license in the documentation
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and/or other materials provided with redistributions of the FDK AAC Codec or
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your modifications thereto in binary form. You must make available free of
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charge copies of the complete source code of the FDK AAC Codec and your
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modifications thereto to recipients of copies in binary form.
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The name of Fraunhofer may not be used to endorse or promote products derived
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from this library without prior written permission.
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You may not charge copyright license fees for anyone to use, copy or distribute
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the FDK AAC Codec software or your modifications thereto.
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Your modified versions of the FDK AAC Codec must carry prominent notices stating
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that you changed the software and the date of any change. For modified versions
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of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android"
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must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK
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AAC Codec Library for Android."
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3. NO PATENT LICENSE
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NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without
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limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE.
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Fraunhofer provides no warranty of patent non-infringement with respect to this
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software.
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You may use this FDK AAC Codec software or modifications thereto only for
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purposes that are authorized by appropriate patent licenses.
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4. DISCLAIMER
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This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright
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holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES,
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including but not limited to the implied warranties of merchantability and
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fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
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CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary,
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or consequential damages, including but not limited to procurement of substitute
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goods or services; loss of use, data, or profits, or business interruption,
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however caused and on any theory of liability, whether in contract, strict
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liability, or tort (including negligence), arising in any way out of the use of
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this software, even if advised of the possibility of such damage.
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5. CONTACT INFORMATION
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Fraunhofer Institute for Integrated Circuits IIS
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Attention: Audio and Multimedia Departments - FDK AAC LL
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Am Wolfsmantel 33
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91058 Erlangen, Germany
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www.iis.fraunhofer.de/amm
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amm-info@iis.fraunhofer.de
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----------------------------------------------------------------------------- */
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/**************************** SBR decoder library ******************************
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Author(s):
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Description:
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*******************************************************************************/
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/*!
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\file
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\brief Envelope extraction prototypes
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*/
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#ifndef ENV_EXTR_H
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#define ENV_EXTR_H
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#include "sbrdecoder.h"
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#include "FDK_bitstream.h"
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#include "lpp_tran.h"
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#include "psdec.h"
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#include "pvc_dec.h"
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#define ENV_EXP_FRACT 0
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/*!< Shift raw envelope data to support fractional numbers.
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Can be set to 8 instead of 0 to enhance accuracy during concealment.
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This is not required for conformance and #requantizeEnvelopeData() will
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become more expensive.
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*/
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#define EXP_BITS 6
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/*!< Size of exponent-part of a pseudo float envelope value (should be at least
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6). The remaining bits in each word are used for the mantissa (should be at
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least 10). This format is used in the arrays iEnvelope[] and
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sbrNoiseFloorLevel[] in the FRAME_DATA struct which must fit in a certain part
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of the output buffer (See buffer management in sbr_dec.cpp). Exponents and
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mantissas could also be stored in separate arrays. Accessing the exponent or
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the mantissa would be simplified and the masks #MASK_E resp. #MASK_M would
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no longer be required.
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*/
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#define MASK_M \
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(((1 << (FRACT_BITS - EXP_BITS)) - 1) \
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<< EXP_BITS) /*!< Mask for extracting the mantissa of a pseudo float \
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envelope value */
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#define MASK_E \
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((1 << EXP_BITS) - 1) /*!< Mask for extracting the exponent of a pseudo \
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float envelope value */
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#define SIGN_EXT \
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(((SCHAR)-1) ^ \
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MASK_E) /*!< a CHAR-constant with all bits above our sign-bit set */
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#define ROUNDING \
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((FIXP_SGL)( \
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1 << (EXP_BITS - 1))) /*!< 0.5-offset for rounding the mantissa of a \
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pseudo-float envelope value */
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#define NRG_EXP_OFFSET \
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16 /*!< Will be added to the reference energy's exponent to prevent negative \
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numbers */
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#define NOISE_EXP_OFFSET \
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38 /*!< Will be added to the noise level exponent to prevent negative \
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numbers */
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#define ADD_HARMONICS_FLAGS_SIZE 2 /* ceil(MAX_FREQ_COEFFS/32) */
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typedef enum {
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HEADER_NOT_PRESENT,
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HEADER_ERROR,
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HEADER_OK,
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HEADER_RESET
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} SBR_HEADER_STATUS;
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typedef enum {
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SBR_NOT_INITIALIZED = 0,
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UPSAMPLING = 1,
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SBR_HEADER = 2,
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SBR_ACTIVE = 3
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} SBR_SYNC_STATE;
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typedef enum { COUPLING_OFF = 0, COUPLING_LEVEL, COUPLING_BAL } COUPLING_MODE;
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typedef struct {
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UCHAR nSfb[2]; /*!< Number of SBR-bands for low and high freq-resolution */
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UCHAR nNfb; /*!< Actual number of noise bands to read from the bitstream*/
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UCHAR numMaster; /*!< Number of SBR-bands in v_k_master */
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UCHAR lowSubband; /*!< QMF-band where SBR frequency range starts */
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UCHAR highSubband; /*!< QMF-band where SBR frequency range ends */
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UCHAR ov_highSubband; /*!< if headerchange applies this value holds the old
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highband value -> highband value of overlap area;
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required for overlap in usac when headerchange
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occurs between XVAR and VARX frame */
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UCHAR limiterBandTable[MAX_NUM_LIMITERS + 1]; /*!< Limiter band table. */
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UCHAR noLimiterBands; /*!< Number of limiter bands. */
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UCHAR nInvfBands; /*!< Number of bands for inverse filtering */
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UCHAR
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*freqBandTable[2]; /*!< Pointers to freqBandTableLo and freqBandTableHi */
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UCHAR freqBandTableLo[MAX_FREQ_COEFFS / 2 + 1];
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/*!< Mapping of SBR bands to QMF bands for low frequency resolution */
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UCHAR freqBandTableHi[MAX_FREQ_COEFFS + 1];
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/*!< Mapping of SBR bands to QMF bands for high frequency resolution */
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UCHAR freqBandTableNoise[MAX_NOISE_COEFFS + 1];
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/*!< Mapping of SBR noise bands to QMF bands */
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UCHAR v_k_master[MAX_FREQ_COEFFS + 1];
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/*!< Master BandTable which freqBandTable is derived from */
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} FREQ_BAND_DATA;
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typedef FREQ_BAND_DATA *HANDLE_FREQ_BAND_DATA;
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#define SBRDEC_ELD_GRID 1
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#define SBRDEC_SYNTAX_SCAL 2
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#define SBRDEC_SYNTAX_USAC 4
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#define SBRDEC_SYNTAX_RSVD50 8
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#define SBRDEC_USAC_INDEP \
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16 /* Flag indicating that USAC global independency flag is active. */
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#define SBRDEC_LOW_POWER \
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32 /* Flag indicating that Low Power QMF mode shall be used. */
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#define SBRDEC_PS_DECODED \
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64 /* Flag indicating that PS was decoded and rendered. */
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#define SBRDEC_QUAD_RATE \
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128 /* Flag indicating that USAC SBR 4:1 is active. \
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*/
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#define SBRDEC_USAC_HARMONICSBR \
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256 /* Flag indicating that USAC HBE tool is active. */
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#define SBRDEC_LD_MPS_QMF \
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512 /* Flag indicating that the LD-MPS QMF shall be used. */
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#define SBRDEC_USAC_ITES \
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1024 /* Flag indicating that USAC inter TES tool is active. */
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#define SBRDEC_SYNTAX_DRM \
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2048 /* Flag indicating that DRM30/DRM+ reverse syntax is being used. */
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#define SBRDEC_ELD_DOWNSCALE \
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4096 /* Flag indicating that ELD downscaled mode decoding is used */
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#define SBRDEC_DOWNSAMPLE \
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8192 /* Flag indicating that the downsampling mode is used. */
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#define SBRDEC_FLUSH 16384 /* Flag is used to flush all elements in use. */
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#define SBRDEC_FORCE_RESET \
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32768 /* Flag is used to force a reset of all elements in use. */
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#define SBRDEC_SKIP_QMF_ANA \
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(1 << 21) /* Flag indicating that the input data is provided in the QMF \
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domain. */
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#define SBRDEC_SKIP_QMF_SYN \
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(1 << 22) /* Flag indicating that the output data is exported in the QMF \
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domain. */
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#define SBRDEC_HDR_STAT_RESET 1
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#define SBRDEC_HDR_STAT_UPDATE 2
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typedef struct {
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UCHAR ampResolution; /*!< Amplitude resolution of envelope values (0: 1.5dB,
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1: 3dB) */
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UCHAR
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xover_band; /*!< Start index in #v_k_master[] used for dynamic crossover
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frequency */
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UCHAR sbr_preprocessing; /*!< SBR prewhitening flag. */
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UCHAR pvc_mode; /*!< Predictive vector coding mode */
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} SBR_HEADER_DATA_BS_INFO;
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typedef struct {
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/* Changes in these variables causes a reset of the decoder */
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UCHAR startFreq; /*!< Index for SBR start frequency */
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UCHAR stopFreq; /*!< Index for SBR highest frequency */
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UCHAR freqScale; /*!< 0: linear scale, 1-3 logarithmic scales */
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UCHAR alterScale; /*!< Flag for coarser frequency resolution */
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UCHAR noise_bands; /*!< Noise bands per octave, read from bitstream*/
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/* don't require reset */
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UCHAR limiterBands; /*!< Index for number of limiter bands per octave */
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UCHAR limiterGains; /*!< Index to select gain limit */
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UCHAR interpolFreq; /*!< Select gain calculation method (1: per QMF channel,
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0: per SBR band) */
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UCHAR smoothingLength; /*!< Smoothing of gains over time (0: on 1: off) */
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} SBR_HEADER_DATA_BS;
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typedef struct {
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SBR_SYNC_STATE
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syncState; /*!< The current initialization status of the header */
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UCHAR status; /*!< Flags field used for signaling a reset right before the
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processing starts and an update from config (e.g. ASC). */
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UCHAR
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frameErrorFlag; /*!< Frame data valid flag. CAUTION: This variable will be
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overwritten by the flag stored in the element
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structure. This is necessary because of the frame
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delay. There it might happen that different slots use
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the same header. */
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UCHAR numberTimeSlots; /*!< AAC: 16,15 */
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UCHAR numberOfAnalysisBands; /*!< Number of QMF analysis bands */
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UCHAR timeStep; /*!< Time resolution of SBR in QMF-slots */
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UINT
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sbrProcSmplRate; /*!< SBR processing sampling frequency (!=
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OutputSamplingRate) (always: CoreSamplingRate *
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UpSamplingFactor; even in single rate mode) */
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SBR_HEADER_DATA_BS bs_data; /*!< current SBR header. */
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SBR_HEADER_DATA_BS bs_dflt; /*!< Default sbr header. */
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SBR_HEADER_DATA_BS_INFO bs_info; /*!< SBR info. */
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FREQ_BAND_DATA freqBandData; /*!< Pointer to struct #FREQ_BAND_DATA */
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UCHAR pvcIDprev;
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} SBR_HEADER_DATA;
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typedef SBR_HEADER_DATA *HANDLE_SBR_HEADER_DATA;
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typedef struct {
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UCHAR frameClass; /*!< Select grid type */
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UCHAR nEnvelopes; /*!< Number of envelopes */
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UCHAR borders[MAX_ENVELOPES + 1]; /*!< Envelope borders (in SBR-timeslots,
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e.g. mp3PRO: 0..11) */
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UCHAR freqRes[MAX_ENVELOPES]; /*!< Frequency resolution for each envelope
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(0=low, 1=high) */
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SCHAR tranEnv; /*!< Transient envelope, -1 if none */
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UCHAR nNoiseEnvelopes; /*!< Number of noise envelopes */
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UCHAR
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bordersNoise[MAX_NOISE_ENVELOPES + 1]; /*!< borders of noise envelopes */
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UCHAR pvcBorders[MAX_PVC_ENVELOPES + 1];
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UCHAR noisePosition;
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UCHAR varLength;
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} FRAME_INFO;
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typedef struct {
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FIXP_SGL sfb_nrg_prev[MAX_FREQ_COEFFS]; /*!< Previous envelope (required for
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differential-coded values) */
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FIXP_SGL
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prevNoiseLevel[MAX_NOISE_COEFFS]; /*!< Previous noise envelope (required
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for differential-coded values) */
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COUPLING_MODE coupling; /*!< Stereo-mode of previous frame */
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INVF_MODE sbr_invf_mode[MAX_INVF_BANDS]; /*!< Previous strength of filtering
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in transposer */
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UCHAR ampRes; /*!< Previous amplitude resolution (0: 1.5dB, 1: 3dB) */
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UCHAR stopPos; /*!< Position in time where last envelope ended */
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UCHAR frameErrorFlag; /*!< Previous frame status */
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UCHAR prevSbrPitchInBins; /*!< Previous frame pitchInBins */
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FRAME_INFO prevFrameInfo;
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} SBR_PREV_FRAME_DATA;
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typedef SBR_PREV_FRAME_DATA *HANDLE_SBR_PREV_FRAME_DATA;
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typedef struct {
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int nScaleFactors; /*!< total number of scalefactors in frame */
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FRAME_INFO frameInfo; /*!< time grid for current frame */
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UCHAR domain_vec[MAX_ENVELOPES]; /*!< Bitfield containing direction of
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delta-coding for each envelope
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(0:frequency, 1:time) */
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UCHAR domain_vec_noise
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[MAX_NOISE_ENVELOPES]; /*!< Same as above, but for noise envelopes */
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INVF_MODE
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sbr_invf_mode[MAX_INVF_BANDS]; /*!< Strength of filtering in transposer */
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COUPLING_MODE coupling; /*!< Stereo-mode */
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int ampResolutionCurrentFrame; /*!< Amplitude resolution of envelope values
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(0: 1.5dB, 1: 3dB) */
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ULONG addHarmonics[ADD_HARMONICS_FLAGS_SIZE]; /*!< Flags for synthetic sine
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addition (aligned to MSB) */
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FIXP_SGL iEnvelope[MAX_NUM_ENVELOPE_VALUES]; /*!< Envelope data */
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FIXP_SGL sbrNoiseFloorLevel[MAX_NUM_NOISE_VALUES]; /*!< Noise envelope data */
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UCHAR iTESactive; /*!< One flag for each envelope to enable USAC inter-TES */
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UCHAR
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interTempShapeMode[MAX_ENVELOPES]; /*!< USAC inter-TES:
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bs_inter_temp_shape_mode[ch][env]
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value */
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UCHAR pvcID[PVC_NTIMESLOT]; /*!< One PVC ID value for each time slot */
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UCHAR ns;
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UCHAR sinusoidal_position;
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UCHAR sbrPatchingMode;
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UCHAR sbrOversamplingFlag;
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UCHAR sbrPitchInBins;
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} SBR_FRAME_DATA;
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typedef SBR_FRAME_DATA *HANDLE_SBR_FRAME_DATA;
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/*!
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\brief Maps sampling frequencies to frequencies for which setup tables are
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available
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Maps arbitary sampling frequency to nearest neighbors for which setup tables
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are available (e.g. 25600 -> 24000).
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Used for startFreq calculation.
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The mapping is defined in 14496-3 (4.6.18.2.6), fs(SBR), and table 4.82
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\return mapped sampling frequency
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*/
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UINT sbrdec_mapToStdSampleRate(UINT fs,
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UINT isUsac); /*!< Output sampling frequency */
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void initSbrPrevFrameData(HANDLE_SBR_PREV_FRAME_DATA h_prev_data,
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int timeSlots);
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int sbrGetChannelElement(HANDLE_SBR_HEADER_DATA hHeaderData,
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HANDLE_SBR_FRAME_DATA hFrameDataLeft,
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HANDLE_SBR_FRAME_DATA hFrameDataRight,
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HANDLE_SBR_PREV_FRAME_DATA hFrameDataLeftPrev,
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UCHAR pvc_mode_last, HANDLE_FDK_BITSTREAM hBitBuf,
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HANDLE_PS_DEC hParametricStereoDec, const UINT flags,
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const int overlap);
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SBR_HEADER_STATUS
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sbrGetHeaderData(HANDLE_SBR_HEADER_DATA headerData,
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HANDLE_FDK_BITSTREAM hBitBuf, const UINT flags,
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const int fIsSbrData, const UCHAR configMode);
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/*!
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\brief Initialize SBR header data
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Copy default values to the header data struct and patch some entries
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depending on the core codec.
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*/
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SBR_ERROR
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initHeaderData(HANDLE_SBR_HEADER_DATA hHeaderData, const int sampleRateIn,
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const int sampleRateOut, const INT downscaleFactor,
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const int samplesPerFrame, const UINT flags,
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const int setDefaultHdr);
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#endif
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/* Convert headroom bits to exponent */
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#define SCALE2EXP(s) (15 - (s))
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#define EXP2SCALE(e) (15 - (e))
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