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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 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[] = {
+ &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 */
+
+{
+ 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;
+}