aboutsummaryrefslogtreecommitdiffstats
path: root/fdk-aac/libSACdec/src/sac_dec.h
diff options
context:
space:
mode:
authorMatthias P. Braendli <matthias.braendli@mpb.li>2019-11-11 11:38:02 +0100
committerMatthias P. Braendli <matthias.braendli@mpb.li>2019-11-11 11:38:02 +0100
commit0e5af65c467b2423a0b857ae3ad98c91acc1e190 (patch)
treed07f69550d8886271e44fe79c4dcfb299cafbd38 /fdk-aac/libSACdec/src/sac_dec.h
parentefe406d9724f959c8bc2a31802559ca6d41fd897 (diff)
downloadODR-AudioEnc-0e5af65c467b2423a0b857ae3ad98c91acc1e190.tar.gz
ODR-AudioEnc-0e5af65c467b2423a0b857ae3ad98c91acc1e190.tar.bz2
ODR-AudioEnc-0e5af65c467b2423a0b857ae3ad98c91acc1e190.zip
Include patched FDK-AAC in the repository
The initial idea was to get the DAB+ patch into upstream, but since that follows the android source releases, there is no place for a custom DAB+ patch there. So instead of having to maintain a patched fdk-aac that has to have the same .so version as the distribution package on which it is installed, we prefer having a separate fdk-aac-dab library to avoid collision. At that point, there's no reason to keep fdk-aac in a separate repository, as odr-audioenc is the only tool that needs DAB+ encoding support. Including it here simplifies installation, and makes it consistent with toolame-dab, also shipped in this repository. DAB+ decoding support (needed by ODR-SourceCompanion, dablin, etisnoop, welle.io and others) can be done using upstream FDK-AAC.
Diffstat (limited to 'fdk-aac/libSACdec/src/sac_dec.h')
-rw-r--r--fdk-aac/libSACdec/src/sac_dec.h539
1 files changed, 539 insertions, 0 deletions
diff --git a/fdk-aac/libSACdec/src/sac_dec.h b/fdk-aac/libSACdec/src/sac_dec.h
new file mode 100644
index 0000000..992acad
--- /dev/null
+++ b/fdk-aac/libSACdec/src/sac_dec.h
@@ -0,0 +1,539 @@
+/* -----------------------------------------------------------------------------
+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
+----------------------------------------------------------------------------- */
+
+/*********************** MPEG surround decoder library *************************
+
+ Author(s):
+
+ Description: SAC Decoder Library structures
+
+*******************************************************************************/
+
+#ifndef SAC_DEC_H
+#define SAC_DEC_H
+
+#include "common_fix.h"
+
+#include "sac_dec_interface.h" /* library interface in ../include */
+
+#include "FDK_qmf_domain.h"
+#include "sac_qmf.h"
+#include "FDK_bitstream.h" /* mp4 bitbuffer */
+#include "sac_calcM1andM2.h"
+#include "FDK_hybrid.h"
+#include "FDK_decorrelate.h"
+#include "sac_reshapeBBEnv.h"
+
+#include "sac_dec_conceal.h"
+
+#include "sac_tsd.h"
+
+#ifndef MAX
+#define MAX(a, b) ((a) > (b) ? (a) : (b))
+#endif
+
+#define ICCdefault 0
+#define IPDdefault 0
+#define arbdmxGainDefault 0
+#define CPCdefault 10
+#define tttCLD1default 15
+#define tttCLD2default 0
+
+#define IS_HQ_ONLY(aot) \
+ ((aot) == AOT_ER_AAC_LD || (aot) == AOT_ER_AAC_ELD || (aot) == AOT_USAC || \
+ (aot) == AOT_RSVD50)
+
+#define SCONST(x) FL2FXCONST_DBL(x)
+
+#define PC_NUM_BANDS (8)
+#define PC_NUM_HYB_BANDS (PC_NUM_BANDS - 3 + 10)
+#define ABS_THR (1e-9f * 32768 * 32768)
+
+#define MAX_HYBRID_BANDS (MAX_NUM_QMF_BANDS - 3 + 10)
+#define HYBRID_FILTER_DELAY (6)
+
+#define MAX_RESIDUAL_FRAMES (4)
+#define MAX_RESIDUAL_BISTREAM \
+ (836) /* 48000 bps * 3 res / (8 * 44100 / 2048 ) */
+#define MAX_MDCT_COEFFS (1024)
+#define SACDEC_RESIDUAL_BS_BUF_SIZE \
+ (1024) /* used to setup and check residual bitstream buffer */
+
+#define MAX_NUM_PARAMS (MAX_NUM_OTT + 4 * MAX_NUM_TTT + MAX_INPUT_CHANNELS)
+#define MAX_NUM_PARAMETERS (MAX(MAX_NUM_PARAMS, MAX_NUM_OTT))
+
+#define MAX_PARAMETER_SETS (9)
+
+#define MAX_M2_INPUT (MAX_OUTPUT_CHANNELS) /* 3 direct + 5 diffuse */
+
+#define MAX_QMF_BANDS_TO_HYBRID \
+ (3) /* 3 bands are filtered again in "40 bands" case */
+#define PROTO_LEN (13)
+#define BUFFER_LEN_LF (PROTO_LEN)
+#define BUFFER_LEN_HF ((PROTO_LEN - 1) / 2)
+
+#define MAX_NO_DECORR_CHANNELS (MAX_OUTPUT_CHANNELS)
+#define HRTF_AZIMUTHS (5)
+
+#define MAX_NUM_OTT_AT 0
+
+/* left out */
+
+typedef enum {
+ UPMIXTYPE_BYPASS = -1, /*just bypass the input channels without processing*/
+ UPMIXTYPE_NORMAL = 0 /*multichannel loudspeaker upmix with spatial data*/
+} UPMIXTYPE;
+
+static inline int isTwoChMode(UPMIXTYPE upmixType) {
+ int retval = 0;
+ return retval;
+}
+
+ /* left out end */
+
+#define MPEGS_BYPASSMODE (0x00000001)
+#define MPEGS_CONCEAL (0x00000002)
+
+typedef struct STP_DEC *HANDLE_STP_DEC;
+
+typedef struct {
+ SCHAR bsQuantCoarseXXXprev;
+ SCHAR bsQuantCoarseXXXprevParse;
+} LOSSLESSSTATE;
+
+typedef struct {
+ SCHAR bsXXXDataMode[MAX_PARAMETER_SETS];
+ SCHAR bsQuantCoarseXXX[MAX_PARAMETER_SETS];
+ SCHAR bsFreqResStrideXXX[MAX_PARAMETER_SETS];
+ SCHAR nocmpQuantCoarseXXX[MAX_PARAMETER_SETS];
+ LOSSLESSSTATE *state; /* Link to persistent state information */
+} LOSSLESSDATA;
+
+struct SPATIAL_BS_FRAME_struct {
+ UCHAR bsIndependencyFlag;
+ UCHAR newBsData;
+ UCHAR numParameterSets;
+
+ /*
+ If bsFramingType == 0, then the paramSlot[ps] for 0 <= ps < numParamSets is
+ calculated as follows: paramSlot[ps] = ceil(numSlots*(ps+1)/numParamSets) - 1
+ Otherwise, it is
+ paramSlot[ps] = bsParamSlot[ps]
+ */
+ INT paramSlot[MAX_PARAMETER_SETS];
+
+ /* These arrays contain the compact indices, only one value per pbstride, only
+ * paramsets actually containing data. */
+ /* These values are written from the parser in ecDataDec() and read during
+ * decode in mapIndexData() */
+ SCHAR cmpOttCLDidx[MAX_NUM_OTT + MAX_NUM_OTT_AT][MAX_PARAMETER_SETS]
+ [MAX_PARAMETER_BANDS];
+ SCHAR cmpOttICCidx[MAX_NUM_OTT][MAX_PARAMETER_SETS][MAX_PARAMETER_BANDS];
+
+ /* Smoothing */
+ UCHAR bsSmoothMode[MAX_PARAMETER_SETS];
+ UCHAR bsSmoothTime[MAX_PARAMETER_SETS];
+ UCHAR bsFreqResStrideSmg[MAX_PARAMETER_SETS];
+ UCHAR bsSmgData[MAX_PARAMETER_SETS]
+ [MAX_PARAMETER_BANDS]; /* smoothing flags, one if band is
+ smoothed, otherwise zero */
+
+ /* Arbitrary Downmix */
+ SCHAR (*cmpArbdmxGainIdx)[MAX_PARAMETER_SETS][MAX_PARAMETER_BANDS];
+
+ /* Lossless control */
+ LOSSLESSDATA *CLDLosslessData;
+ LOSSLESSDATA *ICCLosslessData;
+ /* LOSSLESSDATA *ADGLosslessData; -> is stored in CLDLosslessData[offset] */
+
+ LOSSLESSDATA *IPDLosslessData;
+ SCHAR (*cmpOttIPDidx)[MAX_PARAMETER_SETS][MAX_PARAMETER_BANDS];
+ int phaseMode;
+ int OpdSmoothingMode;
+
+ UCHAR tempShapeEnableChannelGES[MAX_OUTPUT_CHANNELS]; /*!< GES side info. */
+ UCHAR bsEnvShapeData[MAX_OUTPUT_CHANNELS]
+ [MAX_TIME_SLOTS]; /*!< GES side info (quantized). */
+
+ UCHAR tempShapeEnableChannelSTP[MAX_OUTPUT_CHANNELS]; /*!< STP side info. */
+
+ TSD_DATA TsdData[1]; /*!< TSD data structure. */
+};
+
+typedef struct {
+ /* Lossless state */
+ LOSSLESSSTATE CLDLosslessState[MAX_NUM_PARAMETERS];
+ LOSSLESSSTATE ICCLosslessState[MAX_NUM_PARAMETERS];
+ LOSSLESSSTATE IPDLosslessState[MAX_NUM_PARAMETERS];
+} BS_LL_STATE;
+
+typedef struct {
+ int prevParamSlot;
+ int prevSmgTime;
+ UCHAR prevSmgData[MAX_PARAMETER_BANDS];
+
+ FIXP_DBL opdLeftState__FDK[MAX_PARAMETER_BANDS];
+ FIXP_DBL opdRightState__FDK[MAX_PARAMETER_BANDS];
+
+} SMOOTHING_STATE;
+
+typedef struct {
+ FIXP_DBL alpha__FDK;
+ FIXP_DBL beta__FDK;
+ FIXP_DBL partNrgPrev__FDK[2 * MAX_OUTPUT_CHANNELS + MAX_INPUT_CHANNELS]
+ [BB_ENV_SIZE];
+ FIXP_DBL normNrgPrev__FDK[2 * MAX_OUTPUT_CHANNELS + MAX_INPUT_CHANNELS];
+ FIXP_DBL frameNrgPrev__FDK[2 * MAX_OUTPUT_CHANNELS + MAX_INPUT_CHANNELS];
+ INT partNrgPrevSF[2 * MAX_OUTPUT_CHANNELS + MAX_INPUT_CHANNELS];
+ INT partNrgPrev2SF[2 * MAX_OUTPUT_CHANNELS + MAX_INPUT_CHANNELS];
+ INT normNrgPrevSF[2 * MAX_OUTPUT_CHANNELS + MAX_INPUT_CHANNELS];
+ INT frameNrgPrevSF[2 * MAX_OUTPUT_CHANNELS + MAX_INPUT_CHANNELS];
+} RESHAPE_BBENV_STATE;
+
+typedef struct {
+ int maxNumInputChannels;
+ int maxNumOutputChannels;
+ int maxNumQmfBands;
+ int maxNumHybridBands;
+ int maxNumXChannels;
+ int maxNumVChannels;
+ int maxNumDecorChannels;
+ int maxNumCmplxQmfBands;
+ int maxNumCmplxHybBands;
+ int maxNumResChannels;
+ int bProcResidual; /* process residual */
+ int maxNumResidualChannels;
+ int maxNumOttBoxes;
+ int maxNumParams;
+
+} SACDEC_CREATION_PARAMS;
+
+struct spatialDec_struct {
+ SACDEC_ERROR
+ errInt; /* Field to store internal errors.
+ Will be clear at the very beginning of each process call. */
+ int staticDecScale; /* static scale of decoder */
+
+ /* GENERAL */
+ int samplingFreq; /* [Hz] */
+ CFG_LEVEL decoderLevel; /* 0..5 */
+ CFG_EXTENT decoderMode;
+ CFG_BINAURAL binauralMode;
+
+ SACDEC_CREATION_PARAMS createParams;
+
+ int numComplexProcessingBands;
+
+ int treeConfig; /* TREE_5151 = 5151, TREE_5152 = 5152, TREE_525 = 525, defined
+ in sac_bitdec.h */
+
+ int numInputChannels; /* 1 (M) or 2 (L,R) */
+ int numOutputChannels; /* 6 for 3/2.1 (FL,FR,FC,LF,BL,BR) */
+ int numOttBoxes; /* number of ott boxes */
+ int numM2rows;
+
+ int numOutputChannelsAT; /* Number of output channels after arbitrary tree
+ processing */
+
+ int quantMode; /* QUANT_FINE, QUANT_EBQ1, QUANT_EBQ2, defined in sac_bitdec.h
+ */
+ int arbitraryDownmix; /* (arbitraryDownmix != 0) 1 arbitrary downmix data
+ present, 2 arbitrary downmix residual data present*/
+ int residualCoding; /* (residualCoding != 0) => residual coding data present
+ */
+ UCHAR nrResidualFrame;
+ UCHAR nrArbDownmixResidualFrame;
+ FDK_BITSTREAM **hResidualBitstreams;
+ int tempShapeConfig; /* */
+ int decorrType; /* Indicates to use PS or none PS decorrelator. */
+ int decorrConfig; /* chosen decorrelator */
+ int envQuantMode; /* quantization mode of envelope reshaping data */
+
+ FIXP_DBL clipProtectGain__FDK; /* global gain for upmix */
+ char clipProtectGainSF__FDK; /* global gain for upmix */
+
+ /* Currently ignoring center decorr
+ numVChannels = numDirektSignals + numDecorSignals */
+ int numDirektSignals; /* needed for W, Number of direkt signals 515 -> 1 525
+ -> 3 */
+ int wStartResidualIdx; /* Where to start read residuals for W, = 0 for 515, =
+ 1 for 525 since one residual is used in V */
+ int numDecorSignals; /* needed for W, Number of residual and decorrelated
+ signals, = 2, 3 for center deccorelation*/
+ int numVChannels; /* direct signals + decorelator signals */
+ int numXChannels; /* direct input signals + TTT-residuals */
+
+ int timeSlots; /* length of spatial frame in QMF samples */
+ int curTimeSlot; /* pointer to the current time slot used for hyperframing */
+ int prevTimeSlot; /* */
+ int curPs;
+ int frameLength; /* number of output waveform samples/channel/frame */
+ UPMIXTYPE upmixType;
+ int partiallyComplex;
+ int useFDreverb;
+
+ int bShareDelayWithSBR;
+
+ int tp_hybBandBorder; /* Hybrid band indicating the HP filter cut-off. */
+
+ /* FREQUENCY MAPPING */
+ int qmfBands;
+ int hybridBands;
+ const SCHAR *kernels; /* Mapping hybrid band to parameter band. */
+
+ int TsdTs; /**< TSD QMF slot counter 0<= ts < numSlots */
+
+ int *param2hyb; /* Mapping parameter bands to hybrid bands */
+ int kernels_width[MAX_PARAMETER_BANDS]; /* Mapping parmeter band to hybrid
+ band offsets. */
+
+ /* Residual coding */
+ int residualSamplingFreq;
+ UCHAR residualPresent[MAX_NUM_OTT + MAX_NUM_TTT];
+ UCHAR residualBands[MAX_NUM_OTT + MAX_NUM_TTT]; /* 0, if no residual data
+ present for this box */
+ UCHAR residualQMFBands[MAX_NUM_OTT + MAX_NUM_TTT]; /* needed for optimized
+ mdct2qmf calculation */
+ SPATIAL_SPECIFIC_CONFIG *pConfigCurrent;
+
+ int arbdmxFramesPerSpatialFrame;
+ int arbdmxUpdQMF;
+
+ int numParameterBands; /* Number of parameter bands 40, 28, 20, 14, 10, ...
+ .*/
+ int bitstreamParameterBands;
+ int *numOttBands; /* number of bands for each ott, is != numParameterBands for
+ LFEs */
+
+ /* 1 MAPPING */
+ UCHAR extendFrame;
+ UCHAR numParameterSetsPrev;
+
+ int *smgTime;
+ UCHAR **smgData;
+
+ /* PARAMETER DATA decoded and dequantized */
+
+ /* Last parameters from prev frame required during decode in mapIndexData()
+ * and not touched during parse */
+ SCHAR **ottCLDidxPrev;
+ SCHAR **ottICCidxPrev;
+ SCHAR **arbdmxGainIdxPrev;
+ SCHAR **ottIPDidxPrev;
+ SCHAR ***outIdxData; /* is this really persistent memory ? */
+
+ /* State mem required during parse in SpatialDecParseFrameData() */
+ SCHAR **cmpOttCLDidxPrev;
+ SCHAR **cmpOttICCidxPrev;
+ SCHAR ***ottICCdiffidx;
+ SCHAR **cmpOttIPDidxPrev;
+
+ /* State mem required in parseArbitraryDownmixData */
+ SCHAR **cmpArbdmxGainIdxPrev;
+
+ SCHAR ***ottCLD__FDK;
+ SCHAR ***ottICC__FDK;
+
+ SCHAR ***arbdmxGain__FDK; /* Holds the artistic downmix correction index.*/
+
+ FIXP_DBL *arbdmxAlpha__FDK;
+ FIXP_DBL *arbdmxAlphaPrev__FDK;
+
+ UCHAR stereoConfigIndex;
+ int highRateMode;
+
+ int phaseCoding;
+
+ SCHAR ***ottIPD__FDK;
+
+ FIXP_DBL PhaseLeft__FDK[MAX_PARAMETER_BANDS];
+ FIXP_DBL PhaseRight__FDK[MAX_PARAMETER_BANDS];
+ FIXP_DBL PhasePrevLeft__FDK[MAX_PARAMETER_BANDS];
+ FIXP_DBL PhasePrevRight__FDK[MAX_PARAMETER_BANDS];
+ int numOttBandsIPD;
+
+ /* GAIN MATRICIES FOR CURRENT and PREVIOUS PARMATER SET(s)*/
+ FIXP_DBL ***M2Real__FDK;
+ FIXP_DBL ***M2Imag__FDK;
+ FIXP_DBL ***M2RealPrev__FDK;
+ FIXP_DBL ***M2ImagPrev__FDK;
+
+ /* INPUT SIGNALS */
+ FIXP_DBL ***qmfInputRealDelayBuffer__FDK;
+ FIXP_DBL ***qmfInputImagDelayBuffer__FDK;
+
+ int pc_filterdelay; /* additional delay to align HQ with LP before hybird
+ analysis */
+ int qmfInputDelayBufPos;
+ FIXP_DBL **qmfInputReal__FDK;
+ FIXP_DBL **qmfInputImag__FDK;
+
+ FIXP_DBL **hybInputReal__FDK;
+ FIXP_DBL **hybInputImag__FDK;
+
+ FIXP_DBL **binInputReverb;
+
+ FIXP_DBL binGain, reverbGain;
+ FIXP_DBL binCenterGain, reverbCenterGain;
+
+ /* RESIDUAL SIGNALS */
+
+ FIXP_DBL ***qmfResidualReal__FDK;
+ FIXP_DBL ***qmfResidualImag__FDK;
+
+ FIXP_DBL **hybResidualReal__FDK;
+ FIXP_DBL **hybResidualImag__FDK;
+
+ int qmfOutputRealDryDelayBufPos;
+ FIXP_DBL ***qmfOutputRealDryDelayBuffer__FDK;
+ FIXP_DBL ***qmfOutputImagDryFilterBuffer__FDK;
+ FIXP_DBL *qmfOutputImagDryFilterBufferBase__FDK;
+
+ /* TEMPORARY SIGNALS */
+
+ FIXP_DBL **wReal__FDK;
+ FIXP_DBL **wImag__FDK;
+
+ /* OUTPUT SIGNALS */
+ FIXP_DBL **hybOutputRealDry__FDK;
+ FIXP_DBL **hybOutputImagDry__FDK;
+ FIXP_DBL **hybOutputRealWet__FDK;
+ FIXP_DBL **hybOutputImagWet__FDK;
+ PCM_MPS *timeOut__FDK;
+
+ HANDLE_FDK_QMF_DOMAIN pQmfDomain;
+
+ FDK_ANA_HYB_FILTER
+ *hybridAnalysis; /*!< pointer Analysis hybrid filterbank array. */
+ FDK_SYN_HYB_FILTER
+ *hybridSynthesis; /*!< pointer Synthesis hybrid filterbank array. */
+ FIXP_DBL **
+ pHybridAnaStatesLFdmx; /*!< pointer to analysis hybrid filter states LF */
+ FIXP_DBL **
+ pHybridAnaStatesHFdmx; /*!< pointer to analysis hybrid filter states HF */
+ FIXP_DBL **
+ pHybridAnaStatesLFres; /*!< pointer to analysis hybrid filter states LF */
+ FIXP_DBL **
+ pHybridAnaStatesHFres; /*!< pointer to analysis hybrid filter states HF */
+
+ DECORR_DEC *apDecor; /*!< pointer decorrelator array. */
+ FIXP_DBL **pDecorBufferCplx;
+
+ SMOOTHING_STATE *smoothState; /*!< Pointer to smoothing states. */
+
+ RESHAPE_BBENV_STATE *reshapeBBEnvState; /*!< GES handle. */
+ SCHAR row2channelDmxGES[MAX_OUTPUT_CHANNELS];
+
+ HANDLE_STP_DEC hStpDec; /*!< STP handle. */
+
+ const UCHAR *pActivM2ParamBands;
+
+ int bOverwriteM1M2prev; /* Overwrite previous M2/M2 params with first set of
+ new frame after SSC change (aka
+ decodeAfterConfigHasChangedFlag). */
+ SpatialDecConcealmentInfo concealInfo;
+};
+
+#define SACDEC_SYNTAX_MPS 1
+#define SACDEC_SYNTAX_USAC 2
+#define SACDEC_SYNTAX_RSVD50 4
+#define SACDEC_SYNTAX_L2 8
+#define SACDEC_SYNTAX_L3 16
+#define SACDEC_SYNTAX_LD 32
+
+static inline int GetProcBand(spatialDec_struct *self, int qs) {
+ return self->kernels[qs];
+}
+
+#endif /* SAC_DEC_H */