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diff --git a/Src/external_dependencies/openmpt-trunk/soundlib/Paula.cpp b/Src/external_dependencies/openmpt-trunk/soundlib/Paula.cpp
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+/*
+* Paula.cpp
+* ---------
+* Purpose: Emulating the Amiga's sound chip, Paula, by implementing resampling using band-limited steps (BLEPs)
+* Notes  : The BLEP table generator code is a translation of Antti S. Lankila's original Python code.
+* Authors: OpenMPT Devs
+*          Antti S. Lankila
+* The OpenMPT source code is released under the BSD license. Read LICENSE for more details.
+*/
+
+#include "stdafx.h"
+#include "Paula.h"
+#include "TinyFFT.h"
+#include "Tables.h"
+#include "mpt/base/numbers.hpp"
+
+#include <complex>
+#include <numeric>
+
+OPENMPT_NAMESPACE_BEGIN
+
+namespace Paula
+{
+
+namespace
+{
+
+MPT_NOINLINE std::vector<double> KaiserFIR(int numTaps, double cutoff, double beta)
+{
+	const double izeroBeta = Izero(beta);
+	const double kPi = 4.0 * std::atan(1.0) * cutoff;
+	const double xDiv = 1.0 / ((numTaps / 2) * (numTaps / 2));
+	const int numTapsDiv2 = numTaps / 2;
+	std::vector<double> result(numTaps);
+	for(int i = 0; i < numTaps; i++)
+	{
+		double fsinc;
+		if(i == numTapsDiv2)
+		{
+			fsinc = 1.0;
+		} else
+		{
+			const double x = i - numTapsDiv2;
+			const double xPi = x * kPi;
+			//      -   sinc    -   -                   Kaiser window                   -   -sinc-
+			fsinc = std::sin(xPi) * Izero(beta * std::sqrt(1 - x * x * xDiv)) / (izeroBeta * xPi);
+		}
+
+		result[i] = fsinc * cutoff;
+	}
+	return result;
+}
+
+
+MPT_NOINLINE void FIR_MinPhase(std::vector<double> &table, const TinyFFT &fft)
+{
+	std::vector<std::complex<double>> cepstrum(fft.Size());
+	MPT_ASSERT(cepstrum.size() >= table.size());
+	for(size_t i = 0; i < table.size(); i++)
+		cepstrum[i] = table[i];
+	// Compute the real cepstrum: fft -> abs + ln -> ifft -> real
+	fft.FFT(cepstrum);
+	for(auto &v : cepstrum)
+		v = std::log(std::abs(v));
+	fft.IFFT(cepstrum);
+	fft.Normalize(cepstrum);
+
+	// Window the cepstrum in such a way that anticausal components become rejected
+	for(size_t i = 1; i < cepstrum.size() / 2; i++)
+	{
+		cepstrum[i] *= 2;
+		cepstrum[i + cepstrum.size() / 2] *= 0;
+	}
+
+	// Now cancel the previous steps: fft -> exp -> ifft -> real
+	fft.FFT(cepstrum);
+	for(auto &v : cepstrum)
+		v = std::exp(v);
+	fft.IFFT(cepstrum);
+	fft.Normalize(cepstrum);
+	for(size_t i = 0; i < table.size(); i++)
+		table[i] = cepstrum[i].real();
+}
+
+
+class BiquadFilter
+{
+	double b0, b1, b2, a1, a2, x1 = 0.0, x2 = 0.0, y1 = 0.0, y2 = 0.0;
+
+	double Filter(double x0)
+	{
+		double y0 = b0 * x0 + b1 * x1 + b2 * x2 - a1 * y1 - a2 * y2;
+		x2 = x1;
+		x1 = x0;
+		y2 = y1;
+		y1 = y0;
+		return y0;
+	}
+
+public:
+	BiquadFilter(double b0_, double b1_, double b2_, double a1_, double a2_)
+	    : b0(b0_), b1(b1_), b2(b2_), a1(a1_), a2(a2_)
+	{ }
+
+	std::vector<double> Run(std::vector<double> table)
+	{
+		x1 = 0.0;
+		x2 = 0.0;
+		y1 = 0.0;
+		y2 = 0.0;
+
+		// Initialize filter to stable state
+		for(int i = 0; i < 10000; i++)
+			Filter(table[0]);
+		// Now run the filter
+		for(auto &v : table)
+			v = Filter(v);
+
+		return table;
+	}
+};
+
+
+// Observe: a and b are reversed here. To be absolutely clear:
+// a is the nominator and b is the denominator. :-/
+BiquadFilter ZTransform(double a0, double a1, double a2, double b0, double b1, double b2, double fc, double fs)
+{
+	// Prewarp s - domain coefficients
+	const double wp = 2.0 * fs * std::tan(mpt::numbers::pi * fc / fs);
+	a2 /= wp * wp;
+	a1 /= wp;
+	b2 /= wp * wp;
+	b1 /= wp;
+
+	// Compute bilinear transform and return it
+	const double bd = 4 * b2 * fs * fs + 2 * b1 * fs + b0;
+	return BiquadFilter(
+	    (4 * a2 * fs * fs + 2 * a1 * fs + a0) / bd,
+	    (2 * a0 - 8 * a2 * fs * fs) / bd,
+	    (4 * a2 * fs * fs - 2 * a1 * fs + a0) / bd,
+	    (2 * b0 - 8 * b2 * fs * fs) / bd,
+	    (4 * b2 * fs * fs - 2 * b1 * fs + b0) / bd);
+}
+
+
+BiquadFilter MakeRCLowpass(double sampleRate, double freq)
+{
+	const double omega = (2.0 * mpt::numbers::pi) * freq / sampleRate;
+	const double term = 1 + 1 / omega;
+	return BiquadFilter(1 / term, 0.0, 0.0, -1.0 + 1.0 / term, 0.0);
+}
+
+
+BiquadFilter MakeButterworth(double fs, double fc, double res_dB = 0)
+{
+	// 2nd-order Butterworth s-domain coefficients are:
+	//
+	// b0 = 1.0  b1 = 0        b2 = 0
+	// a0 = 1    a1 = sqrt(2)  a2 = 1
+	//
+	// by tweaking the a1 parameter, some resonance can be produced.
+
+	const double res = std::pow(10.0, (-res_dB / 10.0 / 2.0));
+	return ZTransform(1, 0, 0, 1, std::sqrt(2) * res, 1, fc, fs);
+}
+
+
+MPT_NOINLINE void Integrate(std::vector<double> &table)
+{
+	const double total = std::accumulate(table.begin(), table.end(), 0.0);
+	double startVal = -total;
+
+	for(auto &v : table)
+	{
+		startVal += v;
+		v = startVal;
+	}
+}
+
+
+MPT_NOINLINE void Quantize(const std::vector<double> &in, Paula::BlepArray &quantized)
+{
+	MPT_ASSERT(in.size() == Paula::BLEP_SIZE);
+	constexpr int fact = 1 << Paula::BLEP_SCALE;
+	const double cv = fact / (in.back() - in.front());
+
+	for(int i = 0; i < Paula::BLEP_SIZE; i++)
+	{
+		double val = in[i] * cv;
+#ifdef MPT_INTMIXER
+		val = mpt::round(val);
+#endif
+		quantized[i] = static_cast<mixsample_t>(-val);
+	}
+}
+
+}  // namespace
+
+void BlepTables::InitTables()
+{
+	constexpr double sampleRate = Paula::PAULA_HZ;
+
+	// Because Amiga only has 84 dB SNR, the noise floor is low enough with -90 dB.
+	// A500 model uses slightly lower-quality kaiser window to obtain slightly
+	// steeper stopband attenuation. The fixed filters attenuates the sidelobes by
+	// 12 dB, compensating for the worse performance of the kaiser window.
+
+	// 21 kHz stopband is not fully attenuated by 22 kHz. If the sampling frequency
+	// is 44.1 kHz, all frequencies above 22 kHz will alias over 20 kHz, thus inaudible.
+	// The output should be aliasingless for 48 kHz sampling frequency.
+	auto unfilteredA500 = KaiserFIR(Paula::BLEP_SIZE, 21000.0 / sampleRate * 2.0, 8.0);
+	auto unfilteredA1200 = KaiserFIR(Paula::BLEP_SIZE, 21000.0 / sampleRate * 2.0, 9.0);
+	// Move filtering effects to start to allow IIRs more time to settle
+	constexpr size_t padSize = 8;
+	constexpr int fftSize = static_cast<int>(mpt::bit_width(size_t(Paula::BLEP_SIZE)) + mpt::bit_width(padSize) - 2);
+	const TinyFFT fft(fftSize);
+	FIR_MinPhase(unfilteredA500, fft);
+	FIR_MinPhase(unfilteredA1200, fft);
+
+	// Make digital models for the filters on Amiga 500 and 1200.
+	auto filterFixed5kHz = MakeRCLowpass(sampleRate, 4900.0);
+	// The leakage filter seems to reduce treble in both models a bit
+	// The A500 filter seems to be well modelled only with a 4.9 kHz
+	// filter although the component values would suggest 5 kHz filter.
+	auto filterLeakage = MakeRCLowpass(sampleRate, 32000.0);
+	auto filterLED = MakeButterworth(sampleRate, 3275.0, -0.70);
+
+	// Apply fixed filter to A500
+	auto amiga500Off = filterFixed5kHz.Run(unfilteredA500);
+	// Produce the filtered outputs
+	auto amiga1200Off = filterLeakage.Run(unfilteredA1200);
+
+	// Produce LED filters
+	auto amiga500On = filterLED.Run(amiga500Off);
+	auto amiga1200On = filterLED.Run(amiga1200Off);
+
+	// Integrate to produce blep
+	Integrate(amiga500Off);
+	Integrate(amiga500On);
+	Integrate(amiga1200Off);
+	Integrate(amiga1200On);
+	Integrate(unfilteredA1200);
+
+	// Quantize and scale
+	Quantize(amiga500Off, WinSincIntegral[A500Off]);
+	Quantize(amiga500On, WinSincIntegral[A500On]);
+	Quantize(amiga1200Off, WinSincIntegral[A1200Off]);
+	Quantize(amiga1200On, WinSincIntegral[A1200On]);
+	Quantize(unfilteredA1200, WinSincIntegral[Unfiltered]);
+}
+
+
+const Paula::BlepArray &BlepTables::GetAmigaTable(Resampling::AmigaFilter amigaType, bool enableFilter) const
+{
+	if(amigaType == Resampling::AmigaFilter::A500)
+		return enableFilter ? WinSincIntegral[A500On] : WinSincIntegral[A500Off];
+	if(amigaType == Resampling::AmigaFilter::A1200)
+		return enableFilter ? WinSincIntegral[A1200On] : WinSincIntegral[A1200Off];
+	return WinSincIntegral[Unfiltered];
+}
+
+
+// we do not initialize blepState here
+// cppcheck-suppress uninitMemberVar
+State::State(uint32 sampleRate)
+{
+	double amigaClocksPerSample = static_cast<double>(PAULA_HZ) / sampleRate;
+	numSteps = static_cast<int>(amigaClocksPerSample / MINIMUM_INTERVAL);
+	stepRemainder = SamplePosition::FromDouble(amigaClocksPerSample - numSteps * MINIMUM_INTERVAL);
+	remainder = SamplePosition(0);
+}
+
+
+void State::Reset()
+{
+	remainder = SamplePosition(0);
+	activeBleps = 0;
+	firstBlep = MAX_BLEPS / 2u;
+	globalOutputLevel = 0;
+}
+
+
+void State::InputSample(int16 sample)
+{
+	if(sample != globalOutputLevel)
+	{
+		// Start a new blep: level is the difference, age (or phase) is 0 clocks.
+		firstBlep = (firstBlep - 1u) % MAX_BLEPS;
+		if(activeBleps < std::size(blepState))
+			activeBleps++;
+		blepState[firstBlep].age = 0;
+		blepState[firstBlep].level = sample - globalOutputLevel;
+		globalOutputLevel = sample;
+	}
+}
+
+
+// Return output simulated as series of bleps
+int State::OutputSample(const BlepArray &WinSincIntegral)
+{
+	int output = globalOutputLevel * (1 << Paula::BLEP_SCALE);
+	uint32 lastBlep = firstBlep + activeBleps;
+	for(uint32 i = firstBlep; i != lastBlep; i++)
+	{
+		const auto &blep = blepState[i % MAX_BLEPS];
+		output -= WinSincIntegral[blep.age] * blep.level;
+	}
+	output /= (1 << (Paula::BLEP_SCALE - 2));	// - 2 to compensate for the fact that we reduced the input sample bit depth
+
+	return output;
+}
+
+
+// Advance the simulation by given number of clock ticks
+void State::Clock(int cycles)
+{
+	uint32 lastBlep = firstBlep + activeBleps;
+	for(uint32 i = firstBlep; i != lastBlep; i++)
+	{
+		auto &blep = blepState[i % MAX_BLEPS];
+		blep.age += static_cast<uint16>(cycles);
+		if(blep.age >= Paula::BLEP_SIZE)
+		{
+			activeBleps = static_cast<uint16>(i - firstBlep);
+			return;
+		}
+	}
+}
+
+}
+
+OPENMPT_NAMESPACE_END