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diff --git a/fdk-aac/libSBRenc/src/resampler.cpp b/fdk-aac/libSBRenc/src/resampler.cpp new file mode 100644 index 0000000..b1781a7 --- /dev/null +++ b/fdk-aac/libSBRenc/src/resampler.cpp @@ -0,0 +1,444 @@ +/* ----------------------------------------------------------------------------- +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 encoder library ****************************** + + Author(s): + + Description: + +*******************************************************************************/ + +/*! + \file + \brief FDK resampler tool box:$Revision: 91655 $ + \author M. Werner +*/ + +#include "resampler.h" + +#include "genericStds.h" + +/**************************************************************************/ +/* BIQUAD Filter Specifications */ +/**************************************************************************/ + +#define B1 0 +#define B2 1 +#define A1 2 +#define A2 3 + +#define BQC(x) FL2FXCONST_SGL(x / 2) + +struct FILTER_PARAM { + const FIXP_SGL *coeffa; /*! SOS matrix One row/section. Scaled using BQC(). + Order of coefficients: B1,B2,A1,A2. B0=A0=1.0 */ + FIXP_DBL g; /*! overall gain */ + int Wc; /*! normalized passband bandwidth at input samplerate * 1000 */ + int noCoeffs; /*! number of filter coeffs */ + int delay; /*! delay in samples at input samplerate */ +}; + +#define BIQUAD_COEFSTEP 4 + +/** + *\brief Low Pass + Wc = 0,5, order 30, Stop Band -96dB. Wc criteria is "almost 0dB passband", not + the usual -3db gain point. [b,a]=cheby2(30,96,0.505) [sos,g]=tf2sos(b,a) + bandwidth 0.48 + */ +static const FIXP_SGL sos48[] = { + BQC(1.98941075681938), BQC(0.999999996890811), + BQC(0.863264527201963), BQC(0.189553799960663), + BQC(1.90733804822445), BQC(1.00000001736189), + BQC(0.836321575841691), BQC(0.203505809266564), + BQC(1.75616665495325), BQC(0.999999946079721), + BQC(0.784699225121588), BQC(0.230471265506986), + BQC(1.55727745512726), BQC(1.00000011737815), + BQC(0.712515423588351), BQC(0.268752723900498), + BQC(1.33407591943643), BQC(0.999999795953228), + BQC(0.625059117330989), BQC(0.316194685288965), + BQC(1.10689898412458), BQC(1.00000035057114), + BQC(0.52803514366398), BQC(0.370517843224669), + BQC(0.89060371078454), BQC(0.999999343962822), + BQC(0.426920462165257), BQC(0.429608200207746), + BQC(0.694438261209433), BQC(1.0000008629792), + BQC(0.326530699561716), BQC(0.491714450654174), + BQC(0.523237800935322), BQC(1.00000101349782), + BQC(0.230829556274851), BQC(0.555559034843281), + BQC(0.378631165929563), BQC(0.99998986482665), + BQC(0.142906422036095), BQC(0.620338874442411), + BQC(0.260786911308437), BQC(1.00003261460178), + BQC(0.0651008576256505), BQC(0.685759923926262), + BQC(0.168409429188098), BQC(0.999933049695828), + BQC(-0.000790067789975562), BQC(0.751905896602325), + BQC(0.100724533818628), BQC(1.00009472669872), + BQC(-0.0533772830257041), BQC(0.81930744384525), + BQC(0.0561434357867363), BQC(0.999911636304276), + BQC(-0.0913550299236405), BQC(0.88883625875915), + BQC(0.0341680678662057), BQC(1.00003667508676), + BQC(-0.113405185536697), BQC(0.961756638268446)}; + +static const FIXP_DBL g48 = + FL2FXCONST_DBL(0.002712866530047) - (FIXP_DBL)0x8000; + +static const struct FILTER_PARAM param_set48 = { + sos48, g48, 480, 15, 4 /* LF 2 */ +}; + +/** + *\brief Low Pass + Wc = 0,5, order 24, Stop Band -96dB. Wc criteria is "almost 0dB passband", not + the usual -3db gain point. [b,a]=cheby2(24,96,0.5) [sos,g]=tf2sos(b,a) + bandwidth 0.45 + */ +static const FIXP_SGL sos45[] = { + BQC(1.982962601444), BQC(1.00000000007504), BQC(0.646113303737836), + BQC(0.10851149979981), BQC(1.85334094281111), BQC(0.999999999677192), + BQC(0.612073220102006), BQC(0.130022141698044), BQC(1.62541051415425), + BQC(1.00000000080398), BQC(0.547879702855959), BQC(0.171165825133192), + BQC(1.34554656923247), BQC(0.9999999980169), BQC(0.460373914508491), + BQC(0.228677463376354), BQC(1.05656568503116), BQC(1.00000000569363), + BQC(0.357891894038287), BQC(0.298676843912185), BQC(0.787967587877312), + BQC(0.999999984415017), BQC(0.248826893211877), BQC(0.377441803512978), + BQC(0.555480971120497), BQC(1.00000003583307), BQC(0.140614263345315), + BQC(0.461979302213679), BQC(0.364986207070964), BQC(0.999999932084303), + BQC(0.0392669446074516), BQC(0.55033451180825), BQC(0.216827267631558), + BQC(1.00000010534682), BQC(-0.0506232228865103), BQC(0.641691581560946), + BQC(0.108951672277119), BQC(0.999999871167516), BQC(-0.125584840183225), + BQC(0.736367748771803), BQC(0.0387988607229035), BQC(1.00000011205574), + BQC(-0.182814849097974), BQC(0.835802108714964), BQC(0.0042866175809225), + BQC(0.999999954830813), BQC(-0.21965740617151), BQC(0.942623047782363)}; + +static const FIXP_DBL g45 = + FL2FXCONST_DBL(0.00242743980909524) - (FIXP_DBL)0x8000; + +static const struct FILTER_PARAM param_set45 = { + sos45, g45, 450, 12, 4 /* LF 2 */ +}; + +/* + Created by Octave 2.1.73, Mon Oct 13 17:31:32 2008 CEST + Wc = 0,5, order 16, Stop Band -96dB damping. + [b,a]=cheby2(16,96,0.5) + [sos,g]=tf2sos(b,a) + bandwidth = 0.41 + */ + +static const FIXP_SGL sos41[] = { + BQC(1.96193625292), BQC(0.999999999999964), BQC(0.169266178786789), + BQC(0.0128823300475907), BQC(1.68913437662092), BQC(1.00000000000053), + BQC(0.124751503206552), BQC(0.0537472273950989), BQC(1.27274692366017), + BQC(0.999999999995674), BQC(0.0433108625178357), BQC(0.131015753236317), + BQC(0.85214175088395), BQC(1.00000000001813), BQC(-0.0625658152550408), + BQC(0.237763778993806), BQC(0.503841579939009), BQC(0.999999999953223), + BQC(-0.179176128722865), BQC(0.367475236424474), BQC(0.249990711986162), + BQC(1.00000000007952), BQC(-0.294425165824676), BQC(0.516594857170212), + BQC(0.087971668680286), BQC(0.999999999915528), BQC(-0.398956566777928), + BQC(0.686417767801123), BQC(0.00965373325350294), BQC(1.00000000003744), + BQC(-0.48579173764817), BQC(0.884931534239068)}; + +static const FIXP_DBL g41 = FL2FXCONST_DBL(0.00155956951169248); + +static const struct FILTER_PARAM param_set41 = { + sos41, g41, 410, 8, 5 /* LF 3 */ +}; + +/* + # Created by Octave 2.1.73, Mon Oct 13 17:55:33 2008 CEST + Wc = 0,5, order 12, Stop Band -96dB damping. + [b,a]=cheby2(12,96,0.5); + [sos,g]=tf2sos(b,a) +*/ +static const FIXP_SGL sos35[] = { + BQC(1.93299325235762), BQC(0.999999999999985), BQC(-0.140733187246596), + BQC(0.0124139497836062), BQC(1.4890416764109), BQC(1.00000000000011), + BQC(-0.198215402588504), BQC(0.0746730616584138), BQC(0.918450161309795), + BQC(0.999999999999619), BQC(-0.30133912791941), BQC(0.192276468839529), + BQC(0.454877024246818), BQC(1.00000000000086), BQC(-0.432337328809815), + BQC(0.356852933642815), BQC(0.158017147118507), BQC(0.999999999998876), + BQC(-0.574817494249777), BQC(0.566380436970833), BQC(0.0171834649478749), + BQC(1.00000000000055), BQC(-0.718581178041165), BQC(0.83367484487889)}; + +static const FIXP_DBL g35 = FL2FXCONST_DBL(0.00162580994125131); + +static const struct FILTER_PARAM param_set35 = {sos35, g35, 350, 6, 4}; + +/* + # Created by Octave 2.1.73, Mon Oct 13 18:15:38 2008 CEST + Wc = 0,5, order 8, Stop Band -96dB damping. + [b,a]=cheby2(8,96,0.5); + [sos,g]=tf2sos(b,a) +*/ +static const FIXP_SGL sos25[] = { + BQC(1.85334094301225), BQC(1.0), + BQC(-0.702127214212663), BQC(0.132452403998767), + BQC(1.056565682167), BQC(0.999999999999997), + BQC(-0.789503667880785), BQC(0.236328693569128), + BQC(0.364986307455489), BQC(0.999999999999996), + BQC(-0.955191189843375), BQC(0.442966457936379), + BQC(0.0387985751642125), BQC(1.0), + BQC(-1.19817786088084), BQC(0.770493895456328)}; + +static const FIXP_DBL g25 = FL2FXCONST_DBL(0.000945182835294559); + +static const struct FILTER_PARAM param_set25 = {sos25, g25, 250, 4, 5}; + +/* Must be sorted in descending order */ +static const struct FILTER_PARAM *const filter_paramSet[] = { + ¶m_set48, ¶m_set45, ¶m_set41, ¶m_set35, ¶m_set25}; + +/**************************************************************************/ +/* Resampler Functions */ +/**************************************************************************/ + +/*! + \brief Reset downsampler instance and clear delay lines + + \return success of operation +*/ + +INT FDKaacEnc_InitDownsampler( + DOWNSAMPLER *DownSampler, /*!< pointer to downsampler instance */ + int Wc, /*!< normalized cutoff freq * 1000* */ + int ratio) /*!< downsampler ratio */ + +{ + UINT i; + const struct FILTER_PARAM *currentSet = NULL; + + FDKmemclear(DownSampler->downFilter.states, + sizeof(DownSampler->downFilter.states)); + DownSampler->downFilter.ptr = 0; + + /* + find applicable parameter set + */ + currentSet = filter_paramSet[0]; + for (i = 1; i < sizeof(filter_paramSet) / sizeof(struct FILTER_PARAM *); + i++) { + if (filter_paramSet[i]->Wc <= Wc) { + break; + } + currentSet = filter_paramSet[i]; + } + + DownSampler->downFilter.coeffa = currentSet->coeffa; + + DownSampler->downFilter.gain = currentSet->g; + FDK_ASSERT(currentSet->noCoeffs <= MAXNR_SECTIONS * 2); + + DownSampler->downFilter.noCoeffs = currentSet->noCoeffs; + DownSampler->delay = currentSet->delay; + DownSampler->downFilter.Wc = currentSet->Wc; + + DownSampler->ratio = ratio; + DownSampler->pending = ratio - 1; + return (1); +} + +/*! + \brief faster simple folding operation + Filter: + H(z) = A(z)/B(z) + with + A(z) = a[0]*z^0 + a[1]*z^1 + a[2]*z^2 ... a[n]*z^n + + \return filtered value +*/ + +static inline INT_PCM AdvanceFilter( + LP_FILTER *downFilter, /*!< pointer to iir filter instance */ + INT_PCM *pInput, /*!< input of filter */ + int downRatio) { + INT_PCM output; + int i, n; + +#define BIQUAD_SCALE 12 + + FIXP_DBL y = FL2FXCONST_DBL(0.0f); + FIXP_DBL input; + + for (n = 0; n < downRatio; n++) { + FIXP_BQS(*states)[2] = downFilter->states; + const FIXP_SGL *coeff = downFilter->coeffa; + int s1, s2; + + s1 = downFilter->ptr; + s2 = s1 ^ 1; + +#if (SAMPLE_BITS == 16) + input = ((FIXP_DBL)pInput[n]) << (DFRACT_BITS - SAMPLE_BITS - BIQUAD_SCALE); +#elif (SAMPLE_BITS == 32) + input = pInput[n] >> BIQUAD_SCALE; +#else +#error NOT IMPLEMENTED +#endif + + FIXP_BQS state1, state2, state1b, state2b; + + state1 = states[0][s1]; + state2 = states[0][s2]; + + /* Loop over sections */ + for (i = 0; i < downFilter->noCoeffs; i++) { + FIXP_DBL state0; + + /* Load merged states (from next section) */ + state1b = states[i + 1][s1]; + state2b = states[i + 1][s2]; + + state0 = input + fMult(state1, coeff[B1]) + fMult(state2, coeff[B2]); + y = state0 - fMult(state1b, coeff[A1]) - fMult(state2b, coeff[A2]); + + /* Store new feed forward merge state */ + states[i + 1][s2] = y << 1; + /* Store new feed backward state */ + states[i][s2] = input << 1; + + /* Feedback output to next section. */ + input = y; + + /* Transfer merged states */ + state1 = state1b; + state2 = state2b; + + /* Step to next coef set */ + coeff += BIQUAD_COEFSTEP; + } + downFilter->ptr ^= 1; + } + /* Apply global gain */ + y = fMult(y, downFilter->gain); + + /* Apply final gain/scaling to output */ +#if (SAMPLE_BITS == 16) + output = (INT_PCM)SATURATE_RIGHT_SHIFT( + y + (FIXP_DBL)(1 << (DFRACT_BITS - SAMPLE_BITS - BIQUAD_SCALE - 1)), + DFRACT_BITS - SAMPLE_BITS - BIQUAD_SCALE, SAMPLE_BITS); + // output = (INT_PCM) SATURATE_RIGHT_SHIFT(y, + // DFRACT_BITS-SAMPLE_BITS-BIQUAD_SCALE, SAMPLE_BITS); +#else + output = SATURATE_LEFT_SHIFT(y, BIQUAD_SCALE, SAMPLE_BITS); +#endif + + return output; +} + +/*! + \brief FDKaacEnc_Downsample numInSamples of type INT_PCM + Returns number of output samples in numOutSamples + + \return success of operation +*/ + +INT FDKaacEnc_Downsample( + DOWNSAMPLER *DownSampler, /*!< pointer to downsampler instance */ + INT_PCM *inSamples, /*!< pointer to input samples */ + INT numInSamples, /*!< number of input samples */ + INT_PCM *outSamples, /*!< pointer to output samples */ + INT *numOutSamples /*!< pointer tp number of output samples */ +) { + INT i; + *numOutSamples = 0; + + for (i = 0; i < numInSamples; i += DownSampler->ratio) { + *outSamples = AdvanceFilter(&(DownSampler->downFilter), &inSamples[i], + DownSampler->ratio); + outSamples++; + } + *numOutSamples = numInSamples / DownSampler->ratio; + + return 0; +} |