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authorMatthias P. Braendli <matthias.braendli@mpb.li>2016-09-10 20:15:44 +0200
committerMatthias P. Braendli <matthias.braendli@mpb.li>2016-09-10 20:15:44 +0200
commit14c7b800eaa23e9da7c92c7c4df397d0c191f097 (patch)
treed840b6ec41ff74d1184ca1dcd7731d08f1e9ebbb /libSBRenc/src/resampler.cpp
parent78a801e4d716c6f2403cc56cf6c5b6f138f24b2f (diff)
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Remove FDK-AAC
Diffstat (limited to 'libSBRenc/src/resampler.cpp')
-rw-r--r--libSBRenc/src/resampler.cpp507
1 files changed, 0 insertions, 507 deletions
diff --git a/libSBRenc/src/resampler.cpp b/libSBRenc/src/resampler.cpp
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-
-/* -----------------------------------------------------------------------------------------------------------
-Software License for The Fraunhofer FDK AAC Codec Library for Android
-
-© Copyright 1995 - 2013 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 FDK resampler tool box:
- \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)
-};
-
-#ifdef RS_BIQUAD_SCATTERGAIN
-static const FIXP_DBL g48 = FL2FXCONST_DBL(0.67436532061161992682404480717671 - 0.001);
-#else
-static const FIXP_DBL g48 = FL2FXCONST_DBL(0.002712866530047) - (FIXP_DBL)0x8000;
-#endif
-
-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)
-};
-
-#ifdef RS_BIQUAD_SCATTERGAIN
-static const FIXP_DBL g45 = FL2FXCONST_DBL(0.60547428891341319051142629706723 - 0.001);
-#else
-static const FIXP_DBL g45 = FL2FXCONST_DBL(0.00242743980909524) - (FIXP_DBL)0x8000;
-#endif
-
-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)
-};
-
-#ifdef RS_BIQUAD_SCATTERGAIN
-static const FIXP_DBL g41 = FL2FXCONST_DBL(0.44578514476476679750811222123569);
-#else
-static const FIXP_DBL g41 = FL2FXCONST_DBL(0.00155956951169248);
-#endif
-
-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)
-};
-
-#ifdef RS_BIQUAD_SCATTERGAIN
-static const FIXP_DBL g35 = FL2FXCONST_DBL(0.34290853574973898694521267606792);
-#else
-static const FIXP_DBL g35 = FL2FXCONST_DBL(0.00162580994125131);
-#endif
-
-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)
-};
-
-#ifdef RS_BIQUAD_SCATTERGAIN
-static const FIXP_DBL g25 = FL2FXCONST_DBL(0.17533917408936346960080259950471);
-#else
-static const FIXP_DBL g25 = FL2FXCONST_DBL(0.000945182835294559);
-#endif
-
-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[] = {
- &param_set48,
- &param_set45,
- &param_set41,
- &param_set35,
- &param_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 (only 2 supported at the momment) */
-
-{
- UINT i;
- const struct FILTER_PARAM *currentSet=NULL;
-
- FDK_ASSERT(ratio == 2);
- 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 inStride)
-{
- INT_PCM output;
- int i, n;
-
-
-#ifdef RS_BIQUAD_SCATTERGAIN
-#define BIQUAD_SCALE 3
-#else
-#define BIQUAD_SCALE 12
-#endif
-
- 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*inStride]) << (DFRACT_BITS-SAMPLE_BITS-BIQUAD_SCALE);
-#elif (SAMPLE_BITS == 32)
- input = pInput[n*inStride] >> BIQUAD_SCALE;
-#else
-#error NOT IMPLEMENTED
-#endif
-
-#ifndef RS_BIQUAD_SCATTERGAIN /* Merged Direct form I */
-
- 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);
-
-#else /* Direct form II */
-
- /* Loop over sections */
- for (i=0; i<downFilter->noCoeffs; i++)
- {
- FIXP_BQS state1, state2;
- FIXP_DBL state0;
-
- /* Load states */
- state1 = states[i][s1];
- state2 = states[i][s2];
-
- state0 = input - fMult(state1, coeff[A1]) - fMult(state2, coeff[A2]);
- y = state0 + fMult(state1, coeff[B1]) + fMult(state2, coeff[B2]);
- /* Apply scattered gain */
- y = fMult(y, downFilter->gain);
-
- /* Store new state in normalized form */
-#ifdef RS_BIQUAD_STATES16
- /* Do not saturate any state value ! The result would be unacceptable. Rounding makes SNR around 10dB better. */
- states[i][s2] = (FIXP_BQS)(LONG)((state0 + (FIXP_DBL)(1<<(DFRACT_BITS-FRACT_BITS-2))) >> (DFRACT_BITS-FRACT_BITS-1));
-#else
- states[i][s2] = state0<<1;
-#endif
-
- /* Feedback output to next section. */
- input=y;
-
- /* Step to next coef set */
- coeff += BIQUAD_COEFSTEP;
- }
- downFilter->ptr ^= 1;
- }
-
-#endif
-
- /* 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 inStride, /*!< increment of input samples */
- INT_PCM *outSamples, /*!< pointer to output samples */
- INT *numOutSamples, /*!< pointer tp number of output samples */
- INT outStride /*!< increment of output samples */
- )
-{
- INT i;
- *numOutSamples=0;
-
- for(i=0; i<numInSamples; i+=DownSampler->ratio)
- {
- *outSamples = AdvanceFilter(&(DownSampler->downFilter), &inSamples[i*inStride], DownSampler->ratio, inStride);
- outSamples += outStride;
- }
- *numOutSamples = numInSamples/DownSampler->ratio;
-
- return 0;
-}
-