mirror of https://github.com/mstorsjo/fdk-aac.git
543 lines
18 KiB
C
543 lines
18 KiB
C
/* -----------------------------------------------------------------------------
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Software License for The Fraunhofer FDK AAC Codec Library for Android
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© Copyright 1995 - 2019 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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/*********************** MPEG surround decoder library *************************
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Author(s):
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Description: SAC Decoder Library structures
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*******************************************************************************/
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#ifndef SAC_DEC_H
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#define SAC_DEC_H
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#include "common_fix.h"
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#include "sac_dec_interface.h" /* library interface in ../include */
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#include "FDK_qmf_domain.h"
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#include "sac_qmf.h"
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#include "FDK_bitstream.h" /* mp4 bitbuffer */
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#include "sac_calcM1andM2.h"
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#include "FDK_hybrid.h"
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#include "FDK_decorrelate.h"
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#include "sac_reshapeBBEnv.h"
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#include "sac_dec_conceal.h"
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#include "sac_tsd.h"
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#ifndef MAX
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#define MAX(a, b) ((a) > (b) ? (a) : (b))
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#endif
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#define ICCdefault 0
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#define IPDdefault 0
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#define arbdmxGainDefault 0
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#define CPCdefault 10
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#define tttCLD1default 15
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#define tttCLD2default 0
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#define IS_HQ_ONLY(aot) \
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((aot) == AOT_ER_AAC_LD || (aot) == AOT_ER_AAC_ELD || (aot) == AOT_USAC || \
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(aot) == AOT_RSVD50)
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#define SCONST(x) FL2FXCONST_DBL(x)
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#define PC_NUM_BANDS (8)
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#define PC_NUM_HYB_BANDS (PC_NUM_BANDS - 3 + 10)
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#define ABS_THR (1e-9f * 32768 * 32768)
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#define MAX_HYBRID_BANDS (MAX_NUM_QMF_BANDS - 3 + 10)
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#define HYBRID_FILTER_DELAY (6)
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#define MAX_RESIDUAL_FRAMES (4)
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#define MAX_RESIDUAL_BISTREAM \
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(836) /* 48000 bps * 3 res / (8 * 44100 / 2048 ) */
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#define MAX_MDCT_COEFFS (1024)
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#define SACDEC_RESIDUAL_BS_BUF_SIZE \
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(1024) /* used to setup and check residual bitstream buffer */
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#define MAX_NUM_PARAMS (MAX_NUM_OTT + 4 * MAX_NUM_TTT + MAX_INPUT_CHANNELS)
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#define MAX_NUM_PARAMETERS (MAX(MAX_NUM_PARAMS, MAX_NUM_OTT))
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#define MAX_PARAMETER_SETS (9)
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#define MAX_M2_INPUT (MAX_OUTPUT_CHANNELS) /* 3 direct + 5 diffuse */
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#define MAX_QMF_BANDS_TO_HYBRID \
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(3) /* 3 bands are filtered again in "40 bands" case */
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#define PROTO_LEN (13)
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#define BUFFER_LEN_LF (PROTO_LEN)
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#define BUFFER_LEN_HF ((PROTO_LEN - 1) / 2)
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#define MAX_NO_DECORR_CHANNELS (MAX_OUTPUT_CHANNELS)
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#define HRTF_AZIMUTHS (5)
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#define MAX_NUM_OTT_AT 0
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/* left out */
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typedef enum {
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UPMIXTYPE_BYPASS = -1, /*just bypass the input channels without processing*/
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UPMIXTYPE_NORMAL = 0 /*multichannel loudspeaker upmix with spatial data*/
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} UPMIXTYPE;
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static inline int isTwoChMode(UPMIXTYPE upmixType) {
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int retval = 0;
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return retval;
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}
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/* left out end */
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#define MPEGS_BYPASSMODE (0x00000001)
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#define MPEGS_CONCEAL (0x00000002)
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typedef struct STP_DEC *HANDLE_STP_DEC;
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typedef struct {
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SCHAR bsQuantCoarseXXXprev;
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SCHAR bsQuantCoarseXXXprevParse;
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} LOSSLESSSTATE;
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typedef struct {
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SCHAR bsXXXDataMode[MAX_PARAMETER_SETS];
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SCHAR bsQuantCoarseXXX[MAX_PARAMETER_SETS];
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SCHAR bsFreqResStrideXXX[MAX_PARAMETER_SETS];
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SCHAR nocmpQuantCoarseXXX[MAX_PARAMETER_SETS];
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LOSSLESSSTATE *state; /* Link to persistent state information */
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} LOSSLESSDATA;
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struct SPATIAL_BS_FRAME_struct {
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UCHAR bsIndependencyFlag;
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UCHAR newBsData;
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UCHAR numParameterSets;
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/*
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If bsFramingType == 0, then the paramSlot[ps] for 0 <= ps < numParamSets is
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calculated as follows: paramSlot[ps] = ceil(numSlots*(ps+1)/numParamSets) - 1
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Otherwise, it is
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paramSlot[ps] = bsParamSlot[ps]
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*/
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INT paramSlot[MAX_PARAMETER_SETS];
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/* These arrays contain the compact indices, only one value per pbstride, only
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* paramsets actually containing data. */
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/* These values are written from the parser in ecDataDec() and read during
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* decode in mapIndexData() */
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SCHAR cmpOttCLDidx[MAX_NUM_OTT + MAX_NUM_OTT_AT][MAX_PARAMETER_SETS]
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[MAX_PARAMETER_BANDS];
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SCHAR cmpOttICCidx[MAX_NUM_OTT][MAX_PARAMETER_SETS][MAX_PARAMETER_BANDS];
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/* Smoothing */
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UCHAR bsSmoothMode[MAX_PARAMETER_SETS];
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UCHAR bsSmoothTime[MAX_PARAMETER_SETS];
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UCHAR bsFreqResStrideSmg[MAX_PARAMETER_SETS];
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UCHAR bsSmgData[MAX_PARAMETER_SETS]
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[MAX_PARAMETER_BANDS]; /* smoothing flags, one if band is
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smoothed, otherwise zero */
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/* Arbitrary Downmix */
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SCHAR (*cmpArbdmxGainIdx)[MAX_PARAMETER_SETS][MAX_PARAMETER_BANDS];
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/* Lossless control */
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LOSSLESSDATA *CLDLosslessData;
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LOSSLESSDATA *ICCLosslessData;
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/* LOSSLESSDATA *ADGLosslessData; -> is stored in CLDLosslessData[offset] */
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LOSSLESSDATA *IPDLosslessData;
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SCHAR (*cmpOttIPDidx)[MAX_PARAMETER_SETS][MAX_PARAMETER_BANDS];
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int phaseMode;
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int OpdSmoothingMode;
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UCHAR tempShapeEnableChannelGES[MAX_OUTPUT_CHANNELS]; /*!< GES side info. */
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UCHAR bsEnvShapeData[MAX_OUTPUT_CHANNELS]
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[MAX_TIME_SLOTS]; /*!< GES side info (quantized). */
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UCHAR tempShapeEnableChannelSTP[MAX_OUTPUT_CHANNELS]; /*!< STP side info. */
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TSD_DATA TsdData[1]; /*!< TSD data structure. */
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};
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typedef struct {
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/* Lossless state */
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LOSSLESSSTATE CLDLosslessState[MAX_NUM_PARAMETERS];
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LOSSLESSSTATE ICCLosslessState[MAX_NUM_PARAMETERS];
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LOSSLESSSTATE IPDLosslessState[MAX_NUM_PARAMETERS];
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} BS_LL_STATE;
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typedef struct {
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int prevParamSlot;
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int prevSmgTime;
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UCHAR prevSmgData[MAX_PARAMETER_BANDS];
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FIXP_DBL opdLeftState__FDK[MAX_PARAMETER_BANDS];
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FIXP_DBL opdRightState__FDK[MAX_PARAMETER_BANDS];
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} SMOOTHING_STATE;
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typedef struct {
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FIXP_DBL alpha__FDK;
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FIXP_DBL beta__FDK;
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FIXP_DBL partNrgPrev__FDK[2 * MAX_OUTPUT_CHANNELS + MAX_INPUT_CHANNELS]
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[BB_ENV_SIZE];
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FIXP_DBL normNrgPrev__FDK[2 * MAX_OUTPUT_CHANNELS + MAX_INPUT_CHANNELS];
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FIXP_DBL frameNrgPrev__FDK[2 * MAX_OUTPUT_CHANNELS + MAX_INPUT_CHANNELS];
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INT partNrgPrevSF[2 * MAX_OUTPUT_CHANNELS + MAX_INPUT_CHANNELS];
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INT partNrgPrev2SF[2 * MAX_OUTPUT_CHANNELS + MAX_INPUT_CHANNELS];
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INT normNrgPrevSF[2 * MAX_OUTPUT_CHANNELS + MAX_INPUT_CHANNELS];
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INT frameNrgPrevSF[2 * MAX_OUTPUT_CHANNELS + MAX_INPUT_CHANNELS];
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} RESHAPE_BBENV_STATE;
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typedef struct {
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int maxNumInputChannels;
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int maxNumOutputChannels;
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int maxNumQmfBands;
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int maxNumHybridBands;
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int maxNumXChannels;
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int maxNumVChannels;
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int maxNumDecorChannels;
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int maxNumCmplxQmfBands;
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int maxNumCmplxHybBands;
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int maxNumResChannels;
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int bProcResidual; /* process residual */
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int maxNumResidualChannels;
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int maxNumOttBoxes;
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int maxNumParams;
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} SACDEC_CREATION_PARAMS;
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struct spatialDec_struct {
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SACDEC_ERROR
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errInt; /* Field to store internal errors.
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Will be clear at the very beginning of each process call. */
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int staticDecScale; /* static scale of decoder */
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/* GENERAL */
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int samplingFreq; /* [Hz] */
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CFG_LEVEL decoderLevel; /* 0..5 */
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CFG_EXTENT decoderMode;
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CFG_BINAURAL binauralMode;
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SACDEC_CREATION_PARAMS createParams;
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int numComplexProcessingBands;
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int treeConfig; /* TREE_5151 = 5151, TREE_5152 = 5152, TREE_525 = 525, defined
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in sac_bitdec.h */
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int numInputChannels; /* 1 (M) or 2 (L,R) */
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int numOutputChannels; /* 6 for 3/2.1 (FL,FR,FC,LF,BL,BR) */
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int numOttBoxes; /* number of ott boxes */
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int numM2rows;
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int numOutputChannelsAT; /* Number of output channels after arbitrary tree
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processing */
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int quantMode; /* QUANT_FINE, QUANT_EBQ1, QUANT_EBQ2, defined in sac_bitdec.h
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*/
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int arbitraryDownmix; /* (arbitraryDownmix != 0) 1 arbitrary downmix data
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present, 2 arbitrary downmix residual data present*/
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int residualCoding; /* (residualCoding != 0) => residual coding data present
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*/
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UCHAR nrResidualFrame;
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UCHAR nrArbDownmixResidualFrame;
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FDK_BITSTREAM **hResidualBitstreams;
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int tempShapeConfig; /* */
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int decorrType; /* Indicates to use PS or none PS decorrelator. */
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int decorrConfig; /* chosen decorrelator */
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int envQuantMode; /* quantization mode of envelope reshaping data */
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FIXP_DBL clipProtectGain__FDK; /* global gain for upmix */
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char clipProtectGainSF__FDK; /* global gain for upmix */
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/* Currently ignoring center decorr
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numVChannels = numDirektSignals + numDecorSignals */
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int numDirektSignals; /* needed for W, Number of direkt signals 515 -> 1 525
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-> 3 */
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int wStartResidualIdx; /* Where to start read residuals for W, = 0 for 515, =
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1 for 525 since one residual is used in V */
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int numDecorSignals; /* needed for W, Number of residual and decorrelated
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signals, = 2, 3 for center deccorelation*/
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int numVChannels; /* direct signals + decorelator signals */
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int numXChannels; /* direct input signals + TTT-residuals */
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int timeSlots; /* length of spatial frame in QMF samples */
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int curTimeSlot; /* pointer to the current time slot used for hyperframing */
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int prevTimeSlot; /* */
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int curPs;
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int frameLength; /* number of output waveform samples/channel/frame */
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UPMIXTYPE upmixType;
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int partiallyComplex;
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int useFDreverb;
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int bShareDelayWithSBR;
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int tp_hybBandBorder; /* Hybrid band indicating the HP filter cut-off. */
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/* FREQUENCY MAPPING */
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int qmfBands;
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int hybridBands;
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const SCHAR *kernels; /* Mapping hybrid band to parameter band. */
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int TsdTs; /**< TSD QMF slot counter 0<= ts < numSlots */
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int *param2hyb; /* Mapping parameter bands to hybrid bands */
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int kernels_width[MAX_PARAMETER_BANDS]; /* Mapping parmeter band to hybrid
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band offsets. */
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/* Residual coding */
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int residualSamplingFreq;
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UCHAR residualPresent[MAX_NUM_OTT + MAX_NUM_TTT];
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UCHAR residualBands[MAX_NUM_OTT + MAX_NUM_TTT]; /* 0, if no residual data
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present for this box */
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UCHAR residualQMFBands[MAX_NUM_OTT + MAX_NUM_TTT]; /* needed for optimized
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mdct2qmf calculation */
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SPATIAL_SPECIFIC_CONFIG *pConfigCurrent;
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int arbdmxFramesPerSpatialFrame;
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int arbdmxUpdQMF;
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int numParameterBands; /* Number of parameter bands 40, 28, 20, 14, 10, ...
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.*/
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int bitstreamParameterBands;
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int *numOttBands; /* number of bands for each ott, is != numParameterBands for
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LFEs */
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/* 1 MAPPING */
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UCHAR extendFrame;
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UCHAR numParameterSetsPrev;
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int *smgTime;
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UCHAR **smgData;
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/* PARAMETER DATA decoded and dequantized */
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/* Last parameters from prev frame required during decode in mapIndexData()
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* and not touched during parse */
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SCHAR **ottCLDidxPrev;
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SCHAR **ottICCidxPrev;
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SCHAR **arbdmxGainIdxPrev;
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SCHAR **ottIPDidxPrev;
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SCHAR ***outIdxData; /* is this really persistent memory ? */
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/* State mem required during parse in SpatialDecParseFrameData() */
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SCHAR **cmpOttCLDidxPrev;
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SCHAR **cmpOttICCidxPrev;
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SCHAR ***ottICCdiffidx;
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SCHAR **cmpOttIPDidxPrev;
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/* State mem required in parseArbitraryDownmixData */
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SCHAR **cmpArbdmxGainIdxPrev;
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SCHAR ***ottCLD__FDK;
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SCHAR ***ottICC__FDK;
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SCHAR ***arbdmxGain__FDK; /* Holds the artistic downmix correction index.*/
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FIXP_DBL *arbdmxAlpha__FDK;
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FIXP_DBL *arbdmxAlphaPrev__FDK;
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UCHAR stereoConfigIndex;
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int highRateMode;
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int phaseCoding;
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SCHAR ***ottIPD__FDK;
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FIXP_DBL PhaseLeft__FDK[MAX_PARAMETER_BANDS];
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FIXP_DBL PhaseRight__FDK[MAX_PARAMETER_BANDS];
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FIXP_DBL PhasePrevLeft__FDK[MAX_PARAMETER_BANDS];
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FIXP_DBL PhasePrevRight__FDK[MAX_PARAMETER_BANDS];
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int numOttBandsIPD;
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/* GAIN MATRICIES FOR CURRENT and PREVIOUS PARMATER SET(s)*/
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FIXP_DBL ***M2Real__FDK;
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FIXP_DBL ***M2Imag__FDK;
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FIXP_DBL ***M2RealPrev__FDK;
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FIXP_DBL ***M2ImagPrev__FDK;
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/* INPUT SIGNALS */
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FIXP_DBL ***qmfInputRealDelayBuffer__FDK;
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FIXP_DBL ***qmfInputImagDelayBuffer__FDK;
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int pc_filterdelay; /* additional delay to align HQ with LP before hybird
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analysis */
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int qmfInputDelayBufPos;
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FIXP_DBL **qmfInputReal__FDK;
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FIXP_DBL **qmfInputImag__FDK;
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FIXP_DBL **hybInputReal__FDK;
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FIXP_DBL **hybInputImag__FDK;
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FIXP_DBL **binInputReverb;
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FIXP_DBL binGain, reverbGain;
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FIXP_DBL binCenterGain, reverbCenterGain;
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/* RESIDUAL SIGNALS */
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FIXP_DBL ***qmfResidualReal__FDK;
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FIXP_DBL ***qmfResidualImag__FDK;
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FIXP_DBL **hybResidualReal__FDK;
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FIXP_DBL **hybResidualImag__FDK;
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int qmfOutputRealDryDelayBufPos;
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FIXP_DBL ***qmfOutputRealDryDelayBuffer__FDK;
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FIXP_DBL ***qmfOutputImagDryFilterBuffer__FDK;
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FIXP_DBL *qmfOutputImagDryFilterBufferBase__FDK;
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/* TEMPORARY SIGNALS */
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FIXP_DBL **wReal__FDK;
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FIXP_DBL **wImag__FDK;
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/* OUTPUT SIGNALS */
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FIXP_DBL **hybOutputRealDry__FDK;
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FIXP_DBL **hybOutputImagDry__FDK;
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FIXP_DBL **hybOutputRealWet__FDK;
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FIXP_DBL **hybOutputImagWet__FDK;
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|
PCM_MPS *timeOut__FDK;
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HANDLE_FDK_QMF_DOMAIN pQmfDomain;
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FDK_ANA_HYB_FILTER
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*hybridAnalysis; /*!< pointer Analysis hybrid filterbank array. */
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FDK_SYN_HYB_FILTER
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*hybridSynthesis; /*!< pointer Synthesis hybrid filterbank array. */
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|
FIXP_DBL **
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|
pHybridAnaStatesLFdmx; /*!< pointer to analysis hybrid filter states LF */
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|
FIXP_DBL **
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|
pHybridAnaStatesHFdmx; /*!< pointer to analysis hybrid filter states HF */
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|
FIXP_DBL **
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|
pHybridAnaStatesLFres; /*!< pointer to analysis hybrid filter states LF */
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|
FIXP_DBL **
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|
pHybridAnaStatesHFres; /*!< pointer to analysis hybrid filter states HF */
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|
|
|
DECORR_DEC *apDecor; /*!< pointer decorrelator array. */
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|
FIXP_DBL **pDecorBufferCplx;
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|
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|
SMOOTHING_STATE *smoothState; /*!< Pointer to smoothing states. */
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|
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|
RESHAPE_BBENV_STATE *reshapeBBEnvState; /*!< GES handle. */
|
|
SCHAR row2channelDmxGES[MAX_OUTPUT_CHANNELS];
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|
|
|
HANDLE_STP_DEC hStpDec; /*!< STP handle. */
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|
|
|
const UCHAR *pActivM2ParamBands;
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|
|
|
int bOverwriteM1M2prev; /* Overwrite previous M2/M2 params with first set of
|
|
new frame after SSC change (aka
|
|
decodeAfterConfigHasChangedFlag). */
|
|
SpatialDecConcealmentInfo concealInfo;
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|
|
|
INT sacInDataHeadroom; /* Headroom of the SAC input time signal to prevent
|
|
clipping */
|
|
};
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|
|
|
#define SACDEC_SYNTAX_MPS 1
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|
#define SACDEC_SYNTAX_USAC 2
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|
#define SACDEC_SYNTAX_RSVD50 4
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|
#define SACDEC_SYNTAX_L2 8
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|
#define SACDEC_SYNTAX_L3 16
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|
#define SACDEC_SYNTAX_LD 32
|
|
|
|
static inline int GetProcBand(spatialDec_struct *self, int qs) {
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|
return self->kernels[qs];
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|
}
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|
|
|
#endif /* SAC_DEC_H */
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