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+/* -----------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
+Forschung e.V. All rights reserved.
+
+ 1. INTRODUCTION
+The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software
+that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding
+scheme for digital audio. This FDK AAC Codec software is intended to be used on
+a wide variety of Android devices.
+
+AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient
+general perceptual audio codecs. AAC-ELD is considered the best-performing
+full-bandwidth communications codec by independent studies and is widely
+deployed. AAC has been standardized by ISO and IEC as part of the MPEG
+specifications.
+
+Patent licenses for necessary patent claims for the FDK AAC Codec (including
+those of Fraunhofer) may be obtained through Via Licensing
+(www.vialicensing.com) or through the respective patent owners individually for
+the purpose of encoding or decoding bit streams in products that are compliant
+with the ISO/IEC MPEG audio standards. Please note that most manufacturers of
+Android devices already license these patent claims through Via Licensing or
+directly from the patent owners, and therefore FDK AAC Codec software may
+already be covered under those patent licenses when it is used for those
+licensed purposes only.
+
+Commercially-licensed AAC software libraries, including floating-point versions
+with enhanced sound quality, are also available from Fraunhofer. Users are
+encouraged to check the Fraunhofer website for additional applications
+information and documentation.
+
+2. COPYRIGHT LICENSE
+
+Redistribution and use in source and binary forms, with or without modification,
+are permitted without payment of copyright license fees provided that you
+satisfy the following conditions:
+
+You must retain the complete text of this software license in redistributions of
+the FDK AAC Codec or your modifications thereto in source code form.
+
+You must retain the complete text of this software license in the documentation
+and/or other materials provided with redistributions of the FDK AAC Codec or
+your modifications thereto in binary form. You must make available free of
+charge copies of the complete source code of the FDK AAC Codec and your
+modifications thereto to recipients of copies in binary form.
+
+The name of Fraunhofer may not be used to endorse or promote products derived
+from this library without prior written permission.
+
+You may not charge copyright license fees for anyone to use, copy or distribute
+the FDK AAC Codec software or your modifications thereto.
+
+Your modified versions of the FDK AAC Codec must carry prominent notices stating
+that you changed the software and the date of any change. For modified versions
+of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android"
+must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK
+AAC Codec Library for Android."
+
+3. NO PATENT LICENSE
+
+NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without
+limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE.
+Fraunhofer provides no warranty of patent non-infringement with respect to this
+software.
+
+You may use this FDK AAC Codec software or modifications thereto only for
+purposes that are authorized by appropriate patent licenses.
+
+4. DISCLAIMER
+
+This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright
+holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES,
+including but not limited to the implied warranties of merchantability and
+fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
+CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary,
+or consequential damages, including but not limited to procurement of substitute
+goods or services; loss of use, data, or profits, or business interruption,
+however caused and on any theory of liability, whether in contract, strict
+liability, or tort (including negligence), arising in any way out of the use of
+this software, even if advised of the possibility of such damage.
+
+5. CONTACT INFORMATION
+
+Fraunhofer Institute for Integrated Circuits IIS
+Attention: Audio and Multimedia Departments - FDK AAC LL
+Am Wolfsmantel 33
+91058 Erlangen, Germany
+
+www.iis.fraunhofer.de/amm
+amm-info@iis.fraunhofer.de
+----------------------------------------------------------------------------- */
+
+/**************************** SBR decoder library ******************************
+
+ Author(s):
+
+ Description:
+
+*******************************************************************************/
+
+/*!
+ \file
+ \brief Sbr decoder
+*/
+#ifndef PSDEC_H
+#define PSDEC_H
+
+#include "sbrdecoder.h"
+#include "FDK_hybrid.h"
+
+#include "FDK_decorrelate.h"
+
+/* This PS decoder implements the baseline version. So it always uses the */
+/* hybrid filter structure for 20 stereo bands and does not implemet IPD/OPD */
+/* synthesis. The baseline version has to support the complete PS bitstream */
+/* syntax. But IPD/OPD data is ignored and set to 0. If 34 stereo band config */
+/* is used in the bitstream for IIS/ICC the decoded parameters are mapped to */
+/* 20 stereo bands. */
+
+#include "FDK_bitstream.h"
+
+#define SCAL_HEADROOM (2)
+
+#define PS_EXTENSION_SIZE_BITS (4)
+#define PS_EXTENSION_ESC_COUNT_BITS (8)
+
+#define NO_QMF_CHANNELS (64)
+#define MAX_NUM_COL (32)
+
+#define NO_QMF_BANDS_HYBRID20 (3)
+#define NO_SUB_QMF_CHANNELS (12)
+#define HYBRID_FILTER_DELAY (6)
+
+#define MAX_NO_PS_ENV (4 + 1) /* +1 needed for VAR_BORDER */
+
+#define NO_HI_RES_BINS (34)
+#define NO_MID_RES_BINS (20)
+#define NO_LOW_RES_BINS (10)
+
+#define NO_HI_RES_IID_BINS (NO_HI_RES_BINS)
+#define NO_HI_RES_ICC_BINS (NO_HI_RES_BINS)
+
+#define NO_MID_RES_IID_BINS (NO_MID_RES_BINS)
+#define NO_MID_RES_ICC_BINS (NO_MID_RES_BINS)
+
+#define NO_LOW_RES_IID_BINS (NO_LOW_RES_BINS)
+#define NO_LOW_RES_ICC_BINS (NO_LOW_RES_BINS)
+
+#define SUBQMF_GROUPS (10)
+#define QMF_GROUPS (12)
+
+//#define SUBQMF_GROUPS_HI_RES ( 32 )
+//#define QMF_GROUPS_HI_RES ( 18 )
+
+#define NO_IID_GROUPS (SUBQMF_GROUPS + QMF_GROUPS)
+//#define NO_IID_GROUPS_HI_RES ( SUBQMF_GROUPS_HI_RES +
+// QMF_GROUPS_HI_RES )
+
+#define NO_IID_STEPS (7) /* 1 .. + 7 */
+#define NO_IID_STEPS_FINE (15) /* 1 .. +15 */
+#define NO_ICC_STEPS (8) /* 0 .. + 7 */
+
+#define NO_IID_LEVELS (2 * NO_IID_STEPS + 1) /* - 7 .. + 7 */
+#define NO_IID_LEVELS_FINE (2 * NO_IID_STEPS_FINE + 1) /* -15 .. +15 */
+#define NO_ICC_LEVELS (NO_ICC_STEPS) /* 0 .. + 7 */
+
+#define FIXP_SQRT05 ((FIXP_DBL)0x5a827980) /* 1/SQRT2 */
+
+struct PS_DEC_COEFFICIENTS {
+ FIXP_DBL H11r[NO_IID_GROUPS]; /*!< coefficients of the sub-subband groups */
+ FIXP_DBL H12r[NO_IID_GROUPS]; /*!< coefficients of the sub-subband groups */
+ FIXP_DBL H21r[NO_IID_GROUPS]; /*!< coefficients of the sub-subband groups */
+ FIXP_DBL H22r[NO_IID_GROUPS]; /*!< coefficients of the sub-subband groups */
+
+ FIXP_DBL
+ DeltaH11r[NO_IID_GROUPS]; /*!< coefficients of the sub-subband groups */
+ FIXP_DBL
+ DeltaH12r[NO_IID_GROUPS]; /*!< coefficients of the sub-subband groups */
+ FIXP_DBL
+ DeltaH21r[NO_IID_GROUPS]; /*!< coefficients of the sub-subband groups */
+ FIXP_DBL
+ DeltaH22r[NO_IID_GROUPS]; /*!< coefficients of the sub-subband groups */
+
+ SCHAR
+ aaIidIndexMapped[MAX_NO_PS_ENV]
+ [NO_HI_RES_IID_BINS]; /*!< The mapped IID index for all
+ envelopes and all IID bins */
+ SCHAR
+ aaIccIndexMapped[MAX_NO_PS_ENV]
+ [NO_HI_RES_ICC_BINS]; /*!< The mapped ICC index for all
+ envelopes and all ICC bins */
+};
+
+typedef enum { ppt_none = 0, ppt_mpeg = 1, ppt_drm = 2 } PS_PAYLOAD_TYPE;
+
+typedef struct {
+ UCHAR bPsHeaderValid; /*!< set if new header is available from bitstream */
+
+ UCHAR bEnableIid; /*!< One bit denoting the presence of IID parameters */
+ UCHAR bEnableIcc; /*!< One bit denoting the presence of ICC parameters */
+ UCHAR bEnableExt; /*!< The PS extension layer is enabled using the enable_ext
+ bit. If it is set to %1 the IPD and OPD parameters are
+ sent. If it is disabled, i.e. %0, the extension layer is
+ skipped. */
+
+ UCHAR
+ modeIid; /*!< The configuration of IID parameters (number of bands and
+ quantisation grid, iid_quant) is determined by iid_mode. */
+ UCHAR modeIcc; /*!< The configuration of Inter-channel Coherence parameters
+ (number of bands and quantisation grid) is determined by
+ icc_mode. */
+
+ UCHAR freqResIid; /*!< 0=low, 1=mid or 2=high frequency resolution for iid */
+ UCHAR freqResIcc; /*!< 0=low, 1=mid or 2=high frequency resolution for icc */
+
+ UCHAR bFineIidQ; /*!< Use fine Iid quantisation. */
+
+ UCHAR bFrameClass; /*!< The frame_class bit determines whether the parameter
+ positions of the current frame are uniformly spaced
+ accross the frame or they are defined using the
+ positions described by border_position.
+ */
+
+ UCHAR noEnv; /*!< The number of envelopes per frame */
+ UCHAR aEnvStartStop[MAX_NO_PS_ENV + 1]; /*!< In case of variable parameter
+ spacing the parameter positions are
+ determined by border_position */
+
+ SCHAR abIidDtFlag[MAX_NO_PS_ENV]; /*!< Deltacoding time/freq flag for IID, 0
+ => freq */
+ SCHAR abIccDtFlag[MAX_NO_PS_ENV]; /*!< Deltacoding time/freq flag for ICC, 0
+ => freq */
+
+ SCHAR
+ aaIidIndex[MAX_NO_PS_ENV]
+ [NO_HI_RES_IID_BINS]; /*!< The IID index for all envelopes and
+ all IID bins */
+ SCHAR
+ aaIccIndex[MAX_NO_PS_ENV]
+ [NO_HI_RES_ICC_BINS]; /*!< The ICC index for all envelopes and
+ all ICC bins */
+
+} MPEG_PS_BS_DATA;
+
+struct PS_DEC {
+ SCHAR noSubSamples;
+ SCHAR noChannels;
+
+ SCHAR procFrameBased; /*!< Helper to detected switching from frame based to
+ slot based processing
+ */
+
+ PS_PAYLOAD_TYPE
+ bPsDataAvail[(1) + 1]; /*!< set if new data available from bitstream */
+ UCHAR psDecodedPrv; /*!< set if PS has been processed in the last frame */
+
+ /* helpers for frame delay line */
+ UCHAR bsLastSlot; /*!< Index of last read slot. */
+ UCHAR bsReadSlot; /*!< Index of current read slot for additional delay. */
+ UCHAR processSlot; /*!< Index of current slot for processing (need for add.
+ delay). */
+
+ union { /* Bitstream data */
+ MPEG_PS_BS_DATA
+ mpeg; /*!< Struct containing all MPEG specific PS data from bitstream.
+ */
+ } bsData[(1) + 1];
+
+ shouldBeUnion { /* Static data */
+ struct {
+ SCHAR aIidPrevFrameIndex[NO_HI_RES_IID_BINS]; /*!< The IID index for
+ previous frame */
+ SCHAR aIccPrevFrameIndex[NO_HI_RES_ICC_BINS]; /*!< The ICC index for
+ previous frame */
+ UCHAR
+ bPrevFrameFineIidQ; /*!< The IID quantization of the previous frame */
+ UCHAR prevFreqResIid; /*!< Frequency resolution for IID of the previous
+ frame */
+ UCHAR prevFreqResIcc; /*!< Frequency resolution for ICC of the previous
+ frame */
+ UCHAR lastUsb; /*!< uppermost WMF delay band of last frame */
+
+ FIXP_DBL pHybridAnaStatesLFdmx
+ [2 * 13 * NO_QMF_BANDS_HYBRID20]; /*!< Memory used in hybrid analysis
+ for filter states. */
+ FDK_ANA_HYB_FILTER hybridAnalysis;
+ FDK_SYN_HYB_FILTER hybridSynthesis[2];
+
+ DECORR_DEC apDecor; /*!< Decorrelator instance. */
+ FIXP_DBL decorrBufferCplx[(2 * ((825) + (373)))];
+
+ FIXP_DBL h11rPrev[NO_IID_GROUPS]; /*!< previous calculated h(xy)
+ coefficients */
+ FIXP_DBL h12rPrev[NO_IID_GROUPS]; /*!< previous calculated h(xy)
+ coefficients */
+ FIXP_DBL h21rPrev[NO_IID_GROUPS]; /*!< previous calculated h(xy)
+ coefficients */
+ FIXP_DBL h22rPrev[NO_IID_GROUPS]; /*!< previous calculated h(xy)
+ coefficients */
+
+ PS_DEC_COEFFICIENTS
+ *pCoef; /*!< temporal coefficients are on reusable scratch memory */
+
+ } mpeg;
+ }
+ specificTo;
+};
+
+typedef struct PS_DEC *HANDLE_PS_DEC;
+
+int CreatePsDec(HANDLE_PS_DEC *h_PS_DEC, int aacSamplesPerFrame);
+
+int DeletePsDec(HANDLE_PS_DEC *h_PS_DEC);
+
+void PreparePsProcessing(HANDLE_PS_DEC h_ps_d,
+ const FIXP_DBL *const *const rIntBufferLeft,
+ const FIXP_DBL *const *const iIntBufferLeft,
+ const int scaleFactorLowBand);
+
+void initSlotBasedRotation(HANDLE_PS_DEC h_ps_d, int env, int usb);
+
+void ApplyPsSlot(
+ HANDLE_PS_DEC h_ps_d, /* parametric stereo decoder handle */
+ FIXP_DBL **rIntBufferLeft, /* real values of left qmf timeslot */
+ FIXP_DBL **iIntBufferLeft, /* imag values of left qmf timeslot */
+ FIXP_DBL *rIntBufferRight, /* real values of right qmf timeslot */
+ FIXP_DBL *iIntBufferRight, /* imag values of right qmf timeslot */
+ const int scaleFactorLowBand_no_ov, const int scaleFactorLowBand,
+ const int scaleFactorHighBand, const int lsb, const int usb);
+
+#endif /* PSDEC_H */