| |
| /* ----------------------------------------------------------------------------------------------------------- |
| 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 */ |
