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authorThe Android Open Source Project <initial-contribution@android.com>2012-07-11 10:15:24 -0700
committerThe Android Open Source Project <initial-contribution@android.com>2012-07-11 10:15:24 -0700
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+
+/* -----------------------------------------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright 1995 - 2012 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 */