mirror of https://github.com/mstorsjo/fdk-aac.git
276 lines
11 KiB
C
276 lines
11 KiB
C
/* -----------------------------------------------------------------------------
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Software License for The Fraunhofer FDK AAC Codec Library for Android
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© Copyright 1995 - 2023 Fraunhofer-Gesellschaft zur Förderung der angewandten
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Forschung e.V. 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
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that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding
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scheme for digital audio. This FDK AAC Codec software is intended to be used on
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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
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general perceptual audio codecs. AAC-ELD is considered the best-performing
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full-bandwidth communications codec by independent studies and is widely
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deployed. AAC has been standardized by ISO and IEC as part of the MPEG
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specifications.
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Patent licenses for necessary patent claims for the FDK AAC Codec (including
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those of Fraunhofer) may be obtained through Via Licensing
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(www.vialicensing.com) or through the respective patent owners individually for
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the purpose of encoding or decoding bit streams in products that are compliant
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with the ISO/IEC MPEG audio standards. Please note that most manufacturers of
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Android devices already license these patent claims through Via Licensing or
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directly from the patent owners, and therefore FDK AAC Codec software may
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already be covered under those patent licenses when it is used for those
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licensed purposes only.
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Commercially-licensed AAC software libraries, including floating-point versions
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with enhanced sound quality, are also available from Fraunhofer. Users are
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encouraged to check the Fraunhofer website for additional applications
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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,
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are permitted without payment of copyright license fees provided that you
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satisfy the following conditions:
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You must retain the complete text of this software license in redistributions of
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the FDK AAC Codec or your modifications thereto in source code form.
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You must retain the complete text of this software license in the documentation
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and/or other materials provided with redistributions of the FDK AAC Codec or
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your modifications thereto in binary form. You must make available free of
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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
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from this library without prior written permission.
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You may not charge copyright license fees for anyone to use, copy or distribute
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the FDK AAC Codec software or your modifications thereto.
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Your modified versions of the FDK AAC Codec must carry prominent notices stating
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that you changed the software and the date of any change. For modified versions
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of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android"
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must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK
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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
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limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE.
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Fraunhofer provides no warranty of patent non-infringement with respect to this
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software.
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You may use this FDK AAC Codec software or modifications thereto only for
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purposes that are authorized 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
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holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES,
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including but not limited to the implied warranties of merchantability and
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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,
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or consequential damages, including but not limited to procurement of substitute
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goods or services; loss of use, data, or profits, or business interruption,
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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
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this software, even if 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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/**************************** SBR decoder library ******************************
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Author(s):
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Description:
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*******************************************************************************/
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/*!
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\file
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\brief Low Power Profile Transposer
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*/
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#ifndef LPP_TRAN_H
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#define LPP_TRAN_H
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#include "sbrdecoder.h"
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#include "hbe.h"
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#include "qmf.h"
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/*
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Common
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*/
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#define QMF_OUT_SCALE 8
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/*
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Frequency scales
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*/
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/*
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Env-Adjust
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*/
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#define MAX_NOISE_ENVELOPES 2
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#define MAX_NOISE_COEFFS 5
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#define MAX_NUM_NOISE_VALUES (MAX_NOISE_ENVELOPES * MAX_NOISE_COEFFS)
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#define MAX_NUM_LIMITERS 12
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/* Set MAX_ENVELOPES to the largest value of all supported BSFORMATs
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by overriding MAX_ENVELOPES in the correct order: */
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#define MAX_ENVELOPES_LEGACY 5
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#define MAX_ENVELOPES_USAC 8
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#define MAX_ENVELOPES MAX_ENVELOPES_USAC
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#define MAX_FREQ_COEFFS_DUAL_RATE 48
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#define MAX_FREQ_COEFFS_QUAD_RATE 56
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#define MAX_FREQ_COEFFS MAX_FREQ_COEFFS_QUAD_RATE
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#define MAX_FREQ_COEFFS_FS44100 35
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#define MAX_FREQ_COEFFS_FS48000 32
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#define MAX_NUM_ENVELOPE_VALUES (MAX_ENVELOPES * MAX_FREQ_COEFFS)
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#define MAX_GAIN_EXP 34
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/* Maximum gain will be sqrt(0.5 * 2^MAX_GAIN_EXP)
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example: 34=99dB */
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#define MAX_GAIN_CONCEAL_EXP 1
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/* Maximum gain will be sqrt(0.5 * 2^MAX_GAIN_CONCEAL_EXP) in concealment case
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* (0dB) */
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/*
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LPP Transposer
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*/
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#define LPC_ORDER 2
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#define MAX_INVF_BANDS MAX_NOISE_COEFFS
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#define MAX_NUM_PATCHES 6
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#define SHIFT_START_SB 1 /*!< lowest subband of source range */
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typedef enum {
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INVF_OFF = 0,
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INVF_LOW_LEVEL,
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INVF_MID_LEVEL,
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INVF_HIGH_LEVEL,
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INVF_SWITCHED /* not a real choice but used here to control behaviour */
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} INVF_MODE;
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/** parameter set for one single patch */
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typedef struct {
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UCHAR sourceStartBand; /*!< first band in lowbands where to take the samples
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from */
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UCHAR
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sourceStopBand; /*!< first band in lowbands which is not included in the
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patch anymore */
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UCHAR guardStartBand; /*!< first band in highbands to be filled with zeros in
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order to reduce interferences between patches */
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UCHAR
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targetStartBand; /*!< first band in highbands to be filled with whitened
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lowband signal */
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UCHAR targetBandOffs; /*!< difference between 'startTargetBand' and
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'startSourceBand' */
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UCHAR numBandsInPatch; /*!< number of consecutive bands in this one patch */
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} PATCH_PARAM;
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/** whitening factors for different levels of whitening
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need to be initialized corresponding to crossover frequency */
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typedef struct {
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FIXP_DBL off; /*!< bw factor for signal OFF */
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FIXP_DBL transitionLevel;
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FIXP_DBL lowLevel; /*!< bw factor for signal LOW_LEVEL */
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FIXP_DBL midLevel; /*!< bw factor for signal MID_LEVEL */
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FIXP_DBL highLevel; /*!< bw factor for signal HIGH_LEVEL */
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} WHITENING_FACTORS;
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/*! The transposer settings are calculated on a header reset and are shared by
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* both channels. */
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typedef struct {
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UCHAR nCols; /*!< number subsamples of a codec frame */
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UCHAR noOfPatches; /*!< number of patches */
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UCHAR lbStartPatching; /*!< first band of lowbands that will be patched */
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UCHAR lbStopPatching; /*!< first band that won't be patched anymore*/
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UCHAR bwBorders[MAX_NUM_NOISE_VALUES]; /*!< spectral bands with different
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inverse filtering levels */
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PATCH_PARAM
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patchParam[MAX_NUM_PATCHES + 1]; /*!< new parameter set for patching */
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WHITENING_FACTORS
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whFactors; /*!< the pole moving factors for certain
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whitening levels as indicated in the bitstream
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depending on the crossover frequency */
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UCHAR overlap; /*!< Overlap size */
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} TRANSPOSER_SETTINGS;
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typedef struct {
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TRANSPOSER_SETTINGS *pSettings; /*!< Common settings for both channels */
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FIXP_DBL
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bwVectorOld[MAX_NUM_PATCHES]; /*!< pole moving factors of past frame */
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FIXP_DBL lpcFilterStatesRealLegSBR[LPC_ORDER + (3 * (4))][(
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32)]; /*!< pointer array to save filter states */
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FIXP_DBL lpcFilterStatesImagLegSBR[LPC_ORDER + (3 * (4))][(
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32)]; /*!< pointer array to save filter states */
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FIXP_DBL lpcFilterStatesRealHBE[LPC_ORDER + (3 * (4))][(
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64)]; /*!< pointer array to save filter states */
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FIXP_DBL lpcFilterStatesImagHBE[LPC_ORDER + (3 * (4))][(
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64)]; /*!< pointer array to save filter states */
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} SBR_LPP_TRANS;
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typedef SBR_LPP_TRANS *HANDLE_SBR_LPP_TRANS;
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void lppTransposer(HANDLE_SBR_LPP_TRANS hLppTrans,
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QMF_SCALE_FACTOR *sbrScaleFactor, FIXP_DBL **qmfBufferReal,
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FIXP_DBL *degreeAlias, FIXP_DBL **qmfBufferImag,
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const int useLP, const int fPreWhitening,
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const int v_k_master0, const int timeStep,
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const int firstSlotOffset, const int lastSlotOffset,
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const int nInvfBands, INVF_MODE *sbr_invf_mode,
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INVF_MODE *sbr_invf_mode_prev);
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void lppTransposerHBE(
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HANDLE_SBR_LPP_TRANS hLppTrans, /*!< Handle of lpp transposer */
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HANDLE_HBE_TRANSPOSER hQmfTransposer,
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QMF_SCALE_FACTOR *sbrScaleFactor, /*!< Scaling factors */
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FIXP_DBL **qmfBufferReal, /*!< Pointer to pointer to real part of subband
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samples (source) */
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FIXP_DBL **qmfBufferImag, /*!< Pointer to pointer to imaginary part of
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subband samples (source) */
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const int timeStep, /*!< Time step of envelope */
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const int firstSlotOffs, /*!< Start position in time */
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const int lastSlotOffs, /*!< Number of overlap-slots into next frame */
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const int nInvfBands, /*!< Number of bands for inverse filtering */
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INVF_MODE *sbr_invf_mode, /*!< Current inverse filtering modes */
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INVF_MODE *sbr_invf_mode_prev /*!< Previous inverse filtering modes */
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);
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SBR_ERROR
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createLppTransposer(HANDLE_SBR_LPP_TRANS hLppTrans,
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TRANSPOSER_SETTINGS *pSettings, const int highBandStartSb,
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UCHAR *v_k_master, const int numMaster, const int usb,
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const int timeSlots, const int nCols, UCHAR *noiseBandTable,
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const int noNoiseBands, UINT fs, const int chan,
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const int overlap);
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SBR_ERROR
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resetLppTransposer(HANDLE_SBR_LPP_TRANS hLppTrans, UCHAR highBandStartSb,
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UCHAR *v_k_master, UCHAR numMaster, UCHAR *noiseBandTable,
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UCHAR noNoiseBands, UCHAR usb, UINT fs);
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#endif /* LPP_TRAN_H */
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