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authorThe Android Open Source Project <initial-contribution@android.com>2012-07-11 10:15:24 -0700
committerThe Android Open Source Project <initial-contribution@android.com>2012-07-11 10:15:24 -0700
commit2228e360595641dd906bf1773307f43d304f5b2e (patch)
tree57f3d390ebb0782cc0de0fb984c8ea7e45b4f386 /libSBRenc/src/resampler.cpp
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Change-Id: If584e579464f28b97d50e51fc76ba654a5536c54
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+
+/* -----------------------------------------------------------------------------------------------------------
+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 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;
+}
+