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-
-/* -----------------------------------------------------------------------------------------------------------
-Software License for The Fraunhofer FDK AAC Codec Library for Android
-
-© Copyright 1995 - 2013 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
------------------------------------------------------------------------------------------------------------ */
-
-/*!
- \file
- \brief Sbr decoder
-*/
-#ifndef __PSDEC_H
-#define __PSDEC_H
-
-#include "sbrdecoder.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"
-
-#include "psdec_hybrid.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 NRG_INT_COEFF ( 0.75f )
- #define INT_FILTER_COEFF (FL2FXCONST_DBL( 1.0f - NRG_INT_COEFF ))
- #define PEAK_DECAY_FACTOR (FL2FXCONST_DBL( 0.765928338364649f ))
- #define TRANSIENT_IMPACT_FACTOR (FL2FXCONST_DBL( 2.0 / 3.0 ))
-
- #define NO_SERIAL_ALLPASS_LINKS ( 3 )
- #define MAX_NO_PS_ENV ( 4 + 1 ) /* +1 needed for VAR_BORDER */
-
- #define MAX_DELAY_BUFFER_SIZE ( 14 )
- #define NO_DELAY_BUFFER_BANDS ( 35 )
-
- #define NO_HI_RES_BINS ( 34 )
- #define NO_MID_RES_BINS ( 20 )
- #define NO_LOW_RES_BINS ( 10 )
-
- #define FIRST_DELAY_SB ( 23 )
- #define NO_SAMPLE_DELAY_ALLPASS ( 2 )
- #define NO_DELAY_LENGTH_VECTORS ( 12 ) /* d(m): d(0)=3 + d(1)=4 + d(2)=5 */
-
- #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). */
-
-
- INT rescal;
- INT sf_IntBuffer;
-
- 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 delayBufIndex; /*!< Pointer to where the latest sample is in buffer */
- UCHAR noSampleDelay; /*!< How many QMF samples delay is used. */
- UCHAR lastUsb; /*!< uppermost WMF delay band of last frame */
-
- UCHAR aDelayRBufIndexSer[NO_SERIAL_ALLPASS_LINKS]; /*!< Delay buffer for reverb filter */
- UCHAR aDelayBufIndexDelayQmf[NO_QMF_CHANNELS-FIRST_DELAY_SB]; /*!< Delay buffer for ICC group 20 & 21 */
-
- SCHAR scaleFactorPsDelayBuffer; /*!< Scale factor for ps delay buffer */
-
- /* hybrid filter bank delay lines */
- FIXP_DBL aaQmfDelayBufReal[(NO_QMF_CHANNELS-FIRST_DELAY_SB) + (MAX_DELAY_BUFFER_SIZE-1)*(NO_DELAY_BUFFER_BANDS-FIRST_DELAY_SB)];
- FIXP_DBL aaQmfDelayBufImag[(NO_QMF_CHANNELS-FIRST_DELAY_SB) + (MAX_DELAY_BUFFER_SIZE-1)*(NO_DELAY_BUFFER_BANDS-FIRST_DELAY_SB)];
-
- FIXP_DBL *pAaRealDelayBufferQmf[MAX_DELAY_BUFFER_SIZE]; /*!< Real part delay buffer */
- FIXP_DBL *pAaImagDelayBufferQmf[MAX_DELAY_BUFFER_SIZE]; /*!< Imaginary part delay buffer */
-
- FIXP_DBL aaRealDelayBufferQmf[NO_SAMPLE_DELAY_ALLPASS][FIRST_DELAY_SB]; /*!< Real part delay buffer */
- FIXP_DBL aaImagDelayBufferQmf[NO_SAMPLE_DELAY_ALLPASS][FIRST_DELAY_SB]; /*!< Imaginary part delay buffer*/
-
- FIXP_DBL aaRealDelayBufferSubQmf[NO_SAMPLE_DELAY_ALLPASS][NO_SUB_QMF_CHANNELS]; /*!< Real part delay buffer */
- FIXP_DBL aaImagDelayBufferSubQmf[NO_SAMPLE_DELAY_ALLPASS][NO_SUB_QMF_CHANNELS]; /*!< Imaginary part delay buffer */
-
- FIXP_DBL aaaRealDelayRBufferSerQmf[FIRST_DELAY_SB][NO_DELAY_LENGTH_VECTORS]; /*!< Real part delay buffer */
- FIXP_DBL aaaImagDelayRBufferSerQmf[FIRST_DELAY_SB][NO_DELAY_LENGTH_VECTORS]; /*!< Imaginary part delay buffer */
-
- FIXP_DBL aaaRealDelayRBufferSerSubQmf[NO_SUB_QMF_CHANNELS][NO_DELAY_LENGTH_VECTORS]; /*!< Real part delay buffer */
- FIXP_DBL aaaImagDelayRBufferSerSubQmf[NO_SUB_QMF_CHANNELS][NO_DELAY_LENGTH_VECTORS]; /*!< Imaginary part delay buffer */
-
- HYBRID hybrid; /*!< hybrid filter bank struct 1 or 2. */
-
- FIXP_DBL aPrevNrgBin[NO_MID_RES_BINS]; /*!< energy of previous frame */
- FIXP_DBL aPrevPeakDiffBin[NO_MID_RES_BINS]; /*!< peak difference of previous frame */
- FIXP_DBL aPeakDecayFastBin[NO_MID_RES_BINS]; /*!< Saved max. peak decay value per bin */
- SCHAR aPowerPrevScal[NO_MID_RES_BINS]; /*!< Last power value (each bin) of previous frame */
-
- 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 coef; /*!< temporal coefficients (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
-scalFilterBankValues( HANDLE_PS_DEC h_ps_d, /* parametric stereo decoder handle */
- FIXP_DBL **fixpQmfReal, /* qmf filterbank values */
- FIXP_DBL **fixpQmfImag, /* qmf filterbank values */
- int lsb, /* sbr start subband */
- int scaleFactorLowBandSplitLow,
- int scaleFactorLowBandSplitHigh,
- SCHAR *scaleFactorLowBand_lb,
- SCHAR *scaleFactorLowBand_hb,
- int scaleFactorHighBands,
- INT *scaleFactorHighBand,
- INT noCols);
-
-void
-rescalFilterBankValues( HANDLE_PS_DEC h_ps_d, /* parametric stereo decoder handle */
- FIXP_DBL **QmfBufferReal, /* qmf filterbank values */
- FIXP_DBL **QmfBufferImag, /* qmf filterbank values */
- int lsb, /* sbr start subband */
- INT noCols);
-
-
-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 */
-
-
-
-#endif /* __PSDEC_H */