mirror of https://github.com/mstorsjo/fdk-aac.git
353 lines
20 KiB
C
353 lines
20 KiB
C
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/* -----------------------------------------------------------------------------------------------------------
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Software License for The Fraunhofer FDK AAC Codec Library for Android
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© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
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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 that implements
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the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
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This FDK AAC Codec software is intended to be used on 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 general perceptual
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audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
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independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
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of the MPEG specifications.
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Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
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may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
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individually for the purpose of encoding or decoding bit streams in products that are compliant with
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the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
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these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
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software may already be covered under those patent licenses when it is used for those licensed purposes only.
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Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
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are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
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applications 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, are permitted without
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payment of copyright license fees provided that you satisfy the following conditions:
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You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
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your modifications thereto in source code form.
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You must retain the complete text of this software license in the documentation and/or other materials
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provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
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You must make available free of 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 from this library without
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prior written permission.
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You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
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software or your modifications thereto.
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Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
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and the date of any change. For modified versions of the FDK AAC Codec, the term
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"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
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"Third-Party Modified Version of the Fraunhofer FDK 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 limitation the patents of Fraunhofer,
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ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
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respect to this software.
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You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
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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 holders and contributors
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"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
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of merchantability and 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, or consequential damages,
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including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
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or business interruption, 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 this software, even if
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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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/*!
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\file
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\brief Sbr decoder
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*/
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#ifndef __PSDEC_H
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#define __PSDEC_H
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#include "sbrdecoder.h"
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/* This PS decoder implements the baseline version. So it always uses the */
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/* hybrid filter structure for 20 stereo bands and does not implemet IPD/OPD */
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/* synthesis. The baseline version has to support the complete PS bitstream */
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/* syntax. But IPD/OPD data is ignored and set to 0. If 34 stereo band config */
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/* is used in the bitstream for IIS/ICC the decoded parameters are mapped to */
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/* 20 stereo bands. */
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#include "FDK_bitstream.h"
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#include "psdec_hybrid.h"
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#define SCAL_HEADROOM ( 2 )
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#define PS_EXTENSION_SIZE_BITS ( 4 )
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#define PS_EXTENSION_ESC_COUNT_BITS ( 8 )
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#define NO_QMF_CHANNELS ( 64 )
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#define MAX_NUM_COL ( 32 )
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#define NO_QMF_BANDS_HYBRID20 ( 3 )
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#define NO_SUB_QMF_CHANNELS ( 12 )
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#define NRG_INT_COEFF ( 0.75f )
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#define INT_FILTER_COEFF (FL2FXCONST_DBL( 1.0f - NRG_INT_COEFF ))
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#define PEAK_DECAY_FACTOR (FL2FXCONST_DBL( 0.765928338364649f ))
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#define TRANSIENT_IMPACT_FACTOR (FL2FXCONST_DBL( 2.0 / 3.0 ))
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#define NO_SERIAL_ALLPASS_LINKS ( 3 )
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#define MAX_NO_PS_ENV ( 4 + 1 ) /* +1 needed for VAR_BORDER */
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#define MAX_DELAY_BUFFER_SIZE ( 14 )
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#define NO_DELAY_BUFFER_BANDS ( 35 )
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#define NO_HI_RES_BINS ( 34 )
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#define NO_MID_RES_BINS ( 20 )
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#define NO_LOW_RES_BINS ( 10 )
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#define FIRST_DELAY_SB ( 23 )
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#define NO_SAMPLE_DELAY_ALLPASS ( 2 )
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#define NO_DELAY_LENGTH_VECTORS ( 12 ) /* d(m): d(0)=3 + d(1)=4 + d(2)=5 */
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#define NO_HI_RES_IID_BINS ( NO_HI_RES_BINS )
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#define NO_HI_RES_ICC_BINS ( NO_HI_RES_BINS )
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#define NO_MID_RES_IID_BINS ( NO_MID_RES_BINS )
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#define NO_MID_RES_ICC_BINS ( NO_MID_RES_BINS )
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#define NO_LOW_RES_IID_BINS ( NO_LOW_RES_BINS )
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#define NO_LOW_RES_ICC_BINS ( NO_LOW_RES_BINS )
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#define SUBQMF_GROUPS ( 10 )
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#define QMF_GROUPS ( 12 )
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#define SUBQMF_GROUPS_HI_RES ( 32 )
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#define QMF_GROUPS_HI_RES ( 18 )
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#define NO_IID_GROUPS ( SUBQMF_GROUPS + QMF_GROUPS )
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#define NO_IID_GROUPS_HI_RES ( SUBQMF_GROUPS_HI_RES + QMF_GROUPS_HI_RES )
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#define NO_IID_STEPS ( 7 ) /* 1 .. + 7 */
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#define NO_IID_STEPS_FINE ( 15 ) /* 1 .. +15 */
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#define NO_ICC_STEPS ( 8 ) /* 0 .. + 7 */
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#define NO_IID_LEVELS ( 2 * NO_IID_STEPS + 1 ) /* - 7 .. + 7 */
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#define NO_IID_LEVELS_FINE ( 2 * NO_IID_STEPS_FINE + 1 ) /* -15 .. +15 */
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#define NO_ICC_LEVELS ( NO_ICC_STEPS ) /* 0 .. + 7 */
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#define FIXP_SQRT05 ((FIXP_DBL)0x5a827980) /* 1/SQRT2 */
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struct PS_DEC_COEFFICIENTS {
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FIXP_DBL H11r[NO_IID_GROUPS]; /*!< coefficients of the sub-subband groups */
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FIXP_DBL H12r[NO_IID_GROUPS]; /*!< coefficients of the sub-subband groups */
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FIXP_DBL H21r[NO_IID_GROUPS]; /*!< coefficients of the sub-subband groups */
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FIXP_DBL H22r[NO_IID_GROUPS]; /*!< coefficients of the sub-subband groups */
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FIXP_DBL DeltaH11r[NO_IID_GROUPS]; /*!< coefficients of the sub-subband groups */
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FIXP_DBL DeltaH12r[NO_IID_GROUPS]; /*!< coefficients of the sub-subband groups */
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FIXP_DBL DeltaH21r[NO_IID_GROUPS]; /*!< coefficients of the sub-subband groups */
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FIXP_DBL DeltaH22r[NO_IID_GROUPS]; /*!< coefficients of the sub-subband groups */
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SCHAR aaIidIndexMapped[MAX_NO_PS_ENV][NO_HI_RES_IID_BINS]; /*!< The mapped IID index for all envelopes and all IID bins */
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SCHAR aaIccIndexMapped[MAX_NO_PS_ENV][NO_HI_RES_ICC_BINS]; /*!< The mapped ICC index for all envelopes and all ICC bins */
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};
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typedef enum {
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ppt_none = 0,
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ppt_mpeg = 1,
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ppt_drm = 2
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} PS_PAYLOAD_TYPE;
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typedef struct {
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UCHAR bPsHeaderValid; /*!< set if new header is available from bitstream */
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UCHAR bEnableIid; /*!< One bit denoting the presence of IID parameters */
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UCHAR bEnableIcc; /*!< One bit denoting the presence of ICC parameters */
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UCHAR bEnableExt; /*!< The PS extension layer is enabled using the enable_ext bit.
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If it is set to %1 the IPD and OPD parameters are sent.
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If it is disabled, i.e. %0, the extension layer is skipped. */
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UCHAR modeIid; /*!< The configuration of IID parameters (number of bands and
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quantisation grid, iid_quant) is determined by iid_mode. */
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UCHAR modeIcc; /*!< The configuration of Inter-channel Coherence parameters
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(number of bands and quantisation grid) is determined by
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icc_mode. */
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UCHAR freqResIid; /*!< 0=low, 1=mid or 2=high frequency resolution for iid */
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UCHAR freqResIcc; /*!< 0=low, 1=mid or 2=high frequency resolution for icc */
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UCHAR bFineIidQ; /*!< Use fine Iid quantisation. */
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UCHAR bFrameClass; /*!< The frame_class bit determines whether the parameter
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positions of the current frame are uniformly spaced
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accross the frame or they are defined using the positions
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described by border_position. */
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UCHAR noEnv; /*!< The number of envelopes per frame */
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UCHAR aEnvStartStop[MAX_NO_PS_ENV+1]; /*!< In case of variable parameter spacing the parameter
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positions are determined by border_position */
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SCHAR abIidDtFlag[MAX_NO_PS_ENV]; /*!< Deltacoding time/freq flag for IID, 0 => freq */
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SCHAR abIccDtFlag[MAX_NO_PS_ENV]; /*!< Deltacoding time/freq flag for ICC, 0 => freq */
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SCHAR aaIidIndex[MAX_NO_PS_ENV][NO_HI_RES_IID_BINS]; /*!< The IID index for all envelopes and all IID bins */
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SCHAR aaIccIndex[MAX_NO_PS_ENV][NO_HI_RES_ICC_BINS]; /*!< The ICC index for all envelopes and all ICC bins */
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} MPEG_PS_BS_DATA;
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struct PS_DEC {
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SCHAR noSubSamples;
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SCHAR noChannels;
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SCHAR procFrameBased; /*!< Helper to detected switching from frame based to slot based
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processing */
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PS_PAYLOAD_TYPE bPsDataAvail[(1)+1]; /*!< set if new data available from bitstream */
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UCHAR psDecodedPrv; /*!< set if PS has been processed in the last frame */
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/* helpers for frame delay line */
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UCHAR bsLastSlot; /*!< Index of last read slot. */
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UCHAR bsReadSlot; /*!< Index of current read slot for additional delay. */
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UCHAR processSlot; /*!< Index of current slot for processing (need for add. delay). */
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INT rescal;
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INT sf_IntBuffer;
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union { /* Bitstream data */
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MPEG_PS_BS_DATA mpeg; /*!< Struct containing all MPEG specific PS data from bitstream. */
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} bsData[(1)+1];
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shouldBeUnion { /* Static data */
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struct {
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SCHAR aIidPrevFrameIndex[NO_HI_RES_IID_BINS]; /*!< The IID index for previous frame */
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SCHAR aIccPrevFrameIndex[NO_HI_RES_ICC_BINS]; /*!< The ICC index for previous frame */
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UCHAR delayBufIndex; /*!< Pointer to where the latest sample is in buffer */
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UCHAR noSampleDelay; /*!< How many QMF samples delay is used. */
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UCHAR lastUsb; /*!< uppermost WMF delay band of last frame */
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UCHAR aDelayRBufIndexSer[NO_SERIAL_ALLPASS_LINKS]; /*!< Delay buffer for reverb filter */
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UCHAR aDelayBufIndexDelayQmf[NO_QMF_CHANNELS-FIRST_DELAY_SB]; /*!< Delay buffer for ICC group 20 & 21 */
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SCHAR scaleFactorPsDelayBuffer; /*!< Scale factor for ps delay buffer */
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/* hybrid filter bank delay lines */
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FIXP_DBL aaQmfDelayBufReal[(NO_QMF_CHANNELS-FIRST_DELAY_SB) + (MAX_DELAY_BUFFER_SIZE-1)*(NO_DELAY_BUFFER_BANDS-FIRST_DELAY_SB)];
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FIXP_DBL aaQmfDelayBufImag[(NO_QMF_CHANNELS-FIRST_DELAY_SB) + (MAX_DELAY_BUFFER_SIZE-1)*(NO_DELAY_BUFFER_BANDS-FIRST_DELAY_SB)];
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FIXP_DBL *pAaRealDelayBufferQmf[MAX_DELAY_BUFFER_SIZE]; /*!< Real part delay buffer */
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FIXP_DBL *pAaImagDelayBufferQmf[MAX_DELAY_BUFFER_SIZE]; /*!< Imaginary part delay buffer */
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FIXP_DBL aaRealDelayBufferQmf[NO_SAMPLE_DELAY_ALLPASS][FIRST_DELAY_SB]; /*!< Real part delay buffer */
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FIXP_DBL aaImagDelayBufferQmf[NO_SAMPLE_DELAY_ALLPASS][FIRST_DELAY_SB]; /*!< Imaginary part delay buffer*/
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FIXP_DBL aaRealDelayBufferSubQmf[NO_SAMPLE_DELAY_ALLPASS][NO_SUB_QMF_CHANNELS]; /*!< Real part delay buffer */
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FIXP_DBL aaImagDelayBufferSubQmf[NO_SAMPLE_DELAY_ALLPASS][NO_SUB_QMF_CHANNELS]; /*!< Imaginary part delay buffer */
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FIXP_DBL aaaRealDelayRBufferSerQmf[FIRST_DELAY_SB][NO_DELAY_LENGTH_VECTORS]; /*!< Real part delay buffer */
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FIXP_DBL aaaImagDelayRBufferSerQmf[FIRST_DELAY_SB][NO_DELAY_LENGTH_VECTORS]; /*!< Imaginary part delay buffer */
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FIXP_DBL aaaRealDelayRBufferSerSubQmf[NO_SUB_QMF_CHANNELS][NO_DELAY_LENGTH_VECTORS]; /*!< Real part delay buffer */
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FIXP_DBL aaaImagDelayRBufferSerSubQmf[NO_SUB_QMF_CHANNELS][NO_DELAY_LENGTH_VECTORS]; /*!< Imaginary part delay buffer */
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HYBRID hybrid; /*!< hybrid filter bank struct 1 or 2. */
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FIXP_DBL aPrevNrgBin[NO_MID_RES_BINS]; /*!< energy of previous frame */
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FIXP_DBL aPrevPeakDiffBin[NO_MID_RES_BINS]; /*!< peak difference of previous frame */
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FIXP_DBL aPeakDecayFastBin[NO_MID_RES_BINS]; /*!< Saved max. peak decay value per bin */
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SCHAR aPowerPrevScal[NO_MID_RES_BINS]; /*!< Last power value (each bin) of previous frame */
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FIXP_DBL h11rPrev[NO_IID_GROUPS]; /*!< previous calculated h(xy) coefficients */
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FIXP_DBL h12rPrev[NO_IID_GROUPS]; /*!< previous calculated h(xy) coefficients */
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FIXP_DBL h21rPrev[NO_IID_GROUPS]; /*!< previous calculated h(xy) coefficients */
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FIXP_DBL h22rPrev[NO_IID_GROUPS]; /*!< previous calculated h(xy) coefficients */
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PS_DEC_COEFFICIENTS coef; /*!< temporal coefficients (reusable scratch memory) */
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} mpeg;
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} specificTo;
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};
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typedef struct PS_DEC *HANDLE_PS_DEC;
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int CreatePsDec(HANDLE_PS_DEC *h_PS_DEC, int aacSamplesPerFrame);
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int DeletePsDec(HANDLE_PS_DEC *h_PS_DEC);
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void
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scalFilterBankValues( HANDLE_PS_DEC h_ps_d, /* parametric stereo decoder handle */
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FIXP_DBL **fixpQmfReal, /* qmf filterbank values */
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FIXP_DBL **fixpQmfImag, /* qmf filterbank values */
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int lsb, /* sbr start subband */
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int scaleFactorLowBandSplitLow,
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int scaleFactorLowBandSplitHigh,
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SCHAR *scaleFactorLowBand_lb,
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SCHAR *scaleFactorLowBand_hb,
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int scaleFactorHighBands,
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INT *scaleFactorHighBand,
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INT noCols);
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void
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rescalFilterBankValues( HANDLE_PS_DEC h_ps_d, /* parametric stereo decoder handle */
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FIXP_DBL **QmfBufferReal, /* qmf filterbank values */
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FIXP_DBL **QmfBufferImag, /* qmf filterbank values */
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int lsb, /* sbr start subband */
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INT noCols);
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void
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initSlotBasedRotation( HANDLE_PS_DEC h_ps_d,
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int env,
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int usb);
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void
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ApplyPsSlot( HANDLE_PS_DEC h_ps_d, /* parametric stereo decoder handle */
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FIXP_DBL **rIntBufferLeft, /* real values of left qmf timeslot */
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FIXP_DBL **iIntBufferLeft, /* imag values of left qmf timeslot */
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FIXP_DBL *rIntBufferRight, /* real values of right qmf timeslot */
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FIXP_DBL *iIntBufferRight); /* imag values of right qmf timeslot */
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#endif /* __PSDEC_H */
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