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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 Envelope extraction prototypes
+*/
+
+#ifndef ENV_EXTR_H
+#define ENV_EXTR_H
+
+#include "sbrdecoder.h"
+
+#include "FDK_bitstream.h"
+#include "lpp_tran.h"
+
+#include "psdec.h"
+#include "pvc_dec.h"
+
+#define ENV_EXP_FRACT 0
+/*!< Shift raw envelope data to support fractional numbers.
+ Can be set to 8 instead of 0 to enhance accuracy during concealment.
+ This is not required for conformance and #requantizeEnvelopeData() will
+ become more expensive.
+*/
+
+#define EXP_BITS 6
+/*!< Size of exponent-part of a pseudo float envelope value (should be at least
+ 6). The remaining bits in each word are used for the mantissa (should be at
+ least 10). This format is used in the arrays iEnvelope[] and
+ sbrNoiseFloorLevel[] in the FRAME_DATA struct which must fit in a certain part
+ of the output buffer (See buffer management in sbr_dec.cpp). Exponents and
+ mantissas could also be stored in separate arrays. Accessing the exponent or
+ the mantissa would be simplified and the masks #MASK_E resp. #MASK_M would
+ no longer be required.
+*/
+
+#define MASK_M \
+ (((1 << (FRACT_BITS - EXP_BITS)) - 1) \
+ << EXP_BITS) /*!< Mask for extracting the mantissa of a pseudo float \
+ envelope value */
+#define MASK_E \
+ ((1 << EXP_BITS) - 1) /*!< Mask for extracting the exponent of a pseudo \
+ float envelope value */
+
+#define SIGN_EXT \
+ (((SCHAR)-1) ^ \
+ MASK_E) /*!< a CHAR-constant with all bits above our sign-bit set */
+#define ROUNDING \
+ ((FIXP_SGL)( \
+ 1 << (EXP_BITS - 1))) /*!< 0.5-offset for rounding the mantissa of a \
+ pseudo-float envelope value */
+#define NRG_EXP_OFFSET \
+ 16 /*!< Will be added to the reference energy's exponent to prevent negative \
+ numbers */
+#define NOISE_EXP_OFFSET \
+ 38 /*!< Will be added to the noise level exponent to prevent negative \
+ numbers */
+
+#define ADD_HARMONICS_FLAGS_SIZE 2 /* ceil(MAX_FREQ_COEFFS/32) */
+
+typedef enum {
+ HEADER_NOT_PRESENT,
+ HEADER_ERROR,
+ HEADER_OK,
+ HEADER_RESET
+} SBR_HEADER_STATUS;
+
+typedef enum {
+ SBR_NOT_INITIALIZED = 0,
+ UPSAMPLING = 1,
+ SBR_HEADER = 2,
+ SBR_ACTIVE = 3
+} SBR_SYNC_STATE;
+
+typedef enum { COUPLING_OFF = 0, COUPLING_LEVEL, COUPLING_BAL } COUPLING_MODE;
+
+typedef struct {
+ UCHAR nSfb[2]; /*!< Number of SBR-bands for low and high freq-resolution */
+ UCHAR nNfb; /*!< Actual number of noise bands to read from the bitstream*/
+ UCHAR numMaster; /*!< Number of SBR-bands in v_k_master */
+ UCHAR lowSubband; /*!< QMF-band where SBR frequency range starts */
+ UCHAR highSubband; /*!< QMF-band where SBR frequency range ends */
+ UCHAR ov_highSubband; /*!< if headerchange applies this value holds the old
+ highband value -> highband value of overlap area;
+ required for overlap in usac when headerchange
+ occurs between XVAR and VARX frame */
+ UCHAR limiterBandTable[MAX_NUM_LIMITERS + 1]; /*!< Limiter band table. */
+ UCHAR noLimiterBands; /*!< Number of limiter bands. */
+ UCHAR nInvfBands; /*!< Number of bands for inverse filtering */
+ UCHAR
+ *freqBandTable[2]; /*!< Pointers to freqBandTableLo and freqBandTableHi */
+ UCHAR freqBandTableLo[MAX_FREQ_COEFFS / 2 + 1];
+ /*!< Mapping of SBR bands to QMF bands for low frequency resolution */
+ UCHAR freqBandTableHi[MAX_FREQ_COEFFS + 1];
+ /*!< Mapping of SBR bands to QMF bands for high frequency resolution */
+ UCHAR freqBandTableNoise[MAX_NOISE_COEFFS + 1];
+ /*!< Mapping of SBR noise bands to QMF bands */
+ UCHAR v_k_master[MAX_FREQ_COEFFS + 1];
+ /*!< Master BandTable which freqBandTable is derived from */
+} FREQ_BAND_DATA;
+
+typedef FREQ_BAND_DATA *HANDLE_FREQ_BAND_DATA;
+
+#define SBRDEC_ELD_GRID 1
+#define SBRDEC_SYNTAX_SCAL 2
+#define SBRDEC_SYNTAX_USAC 4
+#define SBRDEC_SYNTAX_RSVD50 8
+#define SBRDEC_USAC_INDEP \
+ 16 /* Flag indicating that USAC global independency flag is active. */
+#define SBRDEC_LOW_POWER \
+ 32 /* Flag indicating that Low Power QMF mode shall be used. */
+#define SBRDEC_PS_DECODED \
+ 64 /* Flag indicating that PS was decoded and rendered. */
+#define SBRDEC_QUAD_RATE \
+ 128 /* Flag indicating that USAC SBR 4:1 is active. \
+ */
+#define SBRDEC_USAC_HARMONICSBR \
+ 256 /* Flag indicating that USAC HBE tool is active. */
+#define SBRDEC_LD_MPS_QMF \
+ 512 /* Flag indicating that the LD-MPS QMF shall be used. */
+#define SBRDEC_USAC_ITES \
+ 1024 /* Flag indicating that USAC inter TES tool is active. */
+#define SBRDEC_SYNTAX_DRM \
+ 2048 /* Flag indicating that DRM30/DRM+ reverse syntax is being used. */
+#define SBRDEC_ELD_DOWNSCALE \
+ 4096 /* Flag indicating that ELD downscaled mode decoding is used */
+#define SBRDEC_DOWNSAMPLE \
+ 8192 /* Flag indicating that the downsampling mode is used. */
+#define SBRDEC_FLUSH 16384 /* Flag is used to flush all elements in use. */
+#define SBRDEC_FORCE_RESET \
+ 32768 /* Flag is used to force a reset of all elements in use. */
+#define SBRDEC_SKIP_QMF_ANA \
+ (1 << 21) /* Flag indicating that the input data is provided in the QMF \
+ domain. */
+#define SBRDEC_SKIP_QMF_SYN \
+ (1 << 22) /* Flag indicating that the output data is exported in the QMF \
+ domain. */
+
+#define SBRDEC_HDR_STAT_RESET 1
+#define SBRDEC_HDR_STAT_UPDATE 2
+
+typedef struct {
+ UCHAR ampResolution; /*!< Amplitude resolution of envelope values (0: 1.5dB,
+ 1: 3dB) */
+ UCHAR
+ xover_band; /*!< Start index in #v_k_master[] used for dynamic crossover
+ frequency */
+ UCHAR sbr_preprocessing; /*!< SBR prewhitening flag. */
+ UCHAR pvc_mode; /*!< Predictive vector coding mode */
+} SBR_HEADER_DATA_BS_INFO;
+
+typedef struct {
+ /* Changes in these variables causes a reset of the decoder */
+ UCHAR startFreq; /*!< Index for SBR start frequency */
+ UCHAR stopFreq; /*!< Index for SBR highest frequency */
+ UCHAR freqScale; /*!< 0: linear scale, 1-3 logarithmic scales */
+ UCHAR alterScale; /*!< Flag for coarser frequency resolution */
+ UCHAR noise_bands; /*!< Noise bands per octave, read from bitstream*/
+
+ /* don't require reset */
+ UCHAR limiterBands; /*!< Index for number of limiter bands per octave */
+ UCHAR limiterGains; /*!< Index to select gain limit */
+ UCHAR interpolFreq; /*!< Select gain calculation method (1: per QMF channel,
+ 0: per SBR band) */
+ UCHAR smoothingLength; /*!< Smoothing of gains over time (0: on 1: off) */
+
+} SBR_HEADER_DATA_BS;
+
+typedef struct {
+ SBR_SYNC_STATE
+ syncState; /*!< The current initialization status of the header */
+
+ UCHAR status; /*!< Flags field used for signaling a reset right before the
+ processing starts and an update from config (e.g. ASC). */
+ UCHAR
+ frameErrorFlag; /*!< Frame data valid flag. CAUTION: This variable will be
+ overwritten by the flag stored in the element
+ structure. This is necessary because of the frame
+ delay. There it might happen that different slots use
+ the same header. */
+ UCHAR numberTimeSlots; /*!< AAC: 16,15 */
+ UCHAR numberOfAnalysisBands; /*!< Number of QMF analysis bands */
+ UCHAR timeStep; /*!< Time resolution of SBR in QMF-slots */
+ UINT
+ sbrProcSmplRate; /*!< SBR processing sampling frequency (!=
+ OutputSamplingRate) (always: CoreSamplingRate *
+ UpSamplingFactor; even in single rate mode) */
+
+ SBR_HEADER_DATA_BS bs_data; /*!< current SBR header. */
+ SBR_HEADER_DATA_BS bs_dflt; /*!< Default sbr header. */
+ SBR_HEADER_DATA_BS_INFO bs_info; /*!< SBR info. */
+
+ FREQ_BAND_DATA freqBandData; /*!< Pointer to struct #FREQ_BAND_DATA */
+ UCHAR pvcIDprev;
+} SBR_HEADER_DATA;
+
+typedef SBR_HEADER_DATA *HANDLE_SBR_HEADER_DATA;
+
+typedef struct {
+ UCHAR frameClass; /*!< Select grid type */
+ UCHAR nEnvelopes; /*!< Number of envelopes */
+ UCHAR borders[MAX_ENVELOPES + 1]; /*!< Envelope borders (in SBR-timeslots,
+ e.g. mp3PRO: 0..11) */
+ UCHAR freqRes[MAX_ENVELOPES]; /*!< Frequency resolution for each envelope
+ (0=low, 1=high) */
+ SCHAR tranEnv; /*!< Transient envelope, -1 if none */
+ UCHAR nNoiseEnvelopes; /*!< Number of noise envelopes */
+ UCHAR
+ bordersNoise[MAX_NOISE_ENVELOPES + 1]; /*!< borders of noise envelopes */
+ UCHAR pvcBorders[MAX_PVC_ENVELOPES + 1];
+ UCHAR noisePosition;
+ UCHAR varLength;
+} FRAME_INFO;
+
+typedef struct {
+ FIXP_SGL sfb_nrg_prev[MAX_FREQ_COEFFS]; /*!< Previous envelope (required for
+ differential-coded values) */
+ FIXP_SGL
+ prevNoiseLevel[MAX_NOISE_COEFFS]; /*!< Previous noise envelope (required
+ for differential-coded values) */
+ COUPLING_MODE coupling; /*!< Stereo-mode of previous frame */
+ INVF_MODE sbr_invf_mode[MAX_INVF_BANDS]; /*!< Previous strength of filtering
+ in transposer */
+ UCHAR ampRes; /*!< Previous amplitude resolution (0: 1.5dB, 1: 3dB) */
+ UCHAR stopPos; /*!< Position in time where last envelope ended */
+ UCHAR frameErrorFlag; /*!< Previous frame status */
+ UCHAR prevSbrPitchInBins; /*!< Previous frame pitchInBins */
+ FRAME_INFO prevFrameInfo;
+} SBR_PREV_FRAME_DATA;
+
+typedef SBR_PREV_FRAME_DATA *HANDLE_SBR_PREV_FRAME_DATA;
+
+typedef struct {
+ int nScaleFactors; /*!< total number of scalefactors in frame */
+
+ FRAME_INFO frameInfo; /*!< time grid for current frame */
+ UCHAR domain_vec[MAX_ENVELOPES]; /*!< Bitfield containing direction of
+ delta-coding for each envelope
+ (0:frequency, 1:time) */
+ UCHAR domain_vec_noise
+ [MAX_NOISE_ENVELOPES]; /*!< Same as above, but for noise envelopes */
+
+ INVF_MODE
+ sbr_invf_mode[MAX_INVF_BANDS]; /*!< Strength of filtering in transposer */
+ COUPLING_MODE coupling; /*!< Stereo-mode */
+ int ampResolutionCurrentFrame; /*!< Amplitude resolution of envelope values
+ (0: 1.5dB, 1: 3dB) */
+
+ ULONG addHarmonics[ADD_HARMONICS_FLAGS_SIZE]; /*!< Flags for synthetic sine
+ addition (aligned to MSB) */
+
+ FIXP_SGL iEnvelope[MAX_NUM_ENVELOPE_VALUES]; /*!< Envelope data */
+ FIXP_SGL sbrNoiseFloorLevel[MAX_NUM_NOISE_VALUES]; /*!< Noise envelope data */
+ UCHAR iTESactive; /*!< One flag for each envelope to enable USAC inter-TES */
+ UCHAR
+ interTempShapeMode[MAX_ENVELOPES]; /*!< USAC inter-TES:
+ bs_inter_temp_shape_mode[ch][env]
+ value */
+ UCHAR pvcID[PVC_NTIMESLOT]; /*!< One PVC ID value for each time slot */
+ UCHAR ns;
+ UCHAR sinusoidal_position;
+
+ UCHAR sbrPatchingMode;
+ UCHAR sbrOversamplingFlag;
+ UCHAR sbrPitchInBins;
+} SBR_FRAME_DATA;
+
+typedef SBR_FRAME_DATA *HANDLE_SBR_FRAME_DATA;
+
+/*!
+\brief Maps sampling frequencies to frequencies for which setup tables are
+available
+
+Maps arbitary sampling frequency to nearest neighbors for which setup tables
+are available (e.g. 25600 -> 24000).
+Used for startFreq calculation.
+The mapping is defined in 14496-3 (4.6.18.2.6), fs(SBR), and table 4.82
+
+\return mapped sampling frequency
+*/
+UINT sbrdec_mapToStdSampleRate(UINT fs,
+ UINT isUsac); /*!< Output sampling frequency */
+
+void initSbrPrevFrameData(HANDLE_SBR_PREV_FRAME_DATA h_prev_data,
+ int timeSlots);
+
+int sbrGetChannelElement(HANDLE_SBR_HEADER_DATA hHeaderData,
+ HANDLE_SBR_FRAME_DATA hFrameDataLeft,
+ HANDLE_SBR_FRAME_DATA hFrameDataRight,
+ HANDLE_SBR_PREV_FRAME_DATA hFrameDataLeftPrev,
+ UCHAR pvc_mode_last, HANDLE_FDK_BITSTREAM hBitBuf,
+ HANDLE_PS_DEC hParametricStereoDec, const UINT flags,
+ const int overlap);
+
+SBR_HEADER_STATUS
+sbrGetHeaderData(HANDLE_SBR_HEADER_DATA headerData,
+ HANDLE_FDK_BITSTREAM hBitBuf, const UINT flags,
+ const int fIsSbrData, const UCHAR configMode);
+
+/*!
+ \brief Initialize SBR header data
+
+ Copy default values to the header data struct and patch some entries
+ depending on the core codec.
+*/
+SBR_ERROR
+initHeaderData(HANDLE_SBR_HEADER_DATA hHeaderData, const int sampleRateIn,
+ const int sampleRateOut, const INT downscaleFactor,
+ const int samplesPerFrame, const UINT flags,
+ const int setDefaultHdr);
+#endif
+
+/* Convert headroom bits to exponent */
+#define SCALE2EXP(s) (15 - (s))
+#define EXP2SCALE(e) (15 - (e))