Method process

src/EvenVCO.cpp:128–276

	int oversamplingIndex = 2; 	// default is 2^oversamplingIndex == x4 oversampling

	void process(const ProcessArgs& args) override {

		// pitch inputs determine number of polyphony engines
		const int channels = std::max({1, inputs[PITCH1_INPUT].getChannels(), inputs[PITCH2_INPUT].getChannels()});

		const float pitchKnobs = 1.f + std::round(params[OCTAVE_PARAM].getValue()) + params[TUNE_PARAM].getValue() / 12.f;
		const int oversamplingRatio = oversampler[0][0].getOversamplingRatio();

		for (int c = 0; c < channels; c += 4) {
			float_4 pw = simd::clamp(params[PWM_PARAM].getValue() + inputs[PWM_INPUT].getPolyVoltageSimd<float_4>(c) / 5.f, -1.f, 1.f);
			if (limitPW) {
				pw = simd::rescale(pw, -1, +1, 0.05f, 0.95f);
			}
			else {
				pw = simd::rescale(pw, -1.f, +1.f, 0.f, 1.f);
			}

			const float_4 pitch = inputs[PITCH1_INPUT].getPolyVoltageSimd<float_4>(c) + inputs[PITCH2_INPUT].getPolyVoltageSimd<float_4>(c);

			// pulsewave waveform doesn't have DC even for non 50% duty cycles, but Befaco team would like the option
			// for it to be added back in for hardware compatibility reasons
			const float_4 pulseDCOffset = (!removePulseDC) * 2.f * (0.5f - pw);

			// input oversampling buffers
			float_4* osBufferSync = oversamplerInputs[SYNC_INPUT_UP][c / 4].getOSBuffer();
			float_4* osBufferFM = oversamplerInputs[FM_INPUT_UP][c / 4].getOSBuffer();

			// upsample hard sync input (if connected)
			if (inputs[SYNC_INPUT].isConnected()) {
				oversamplerInputs[SYNC_INPUT_UP][c].upsample(inputs[SYNC_INPUT].getPolyVoltageSimd<float_4>(c));
			}
			else {
				std::fill(osBufferSync, &osBufferSync[oversamplingRatio], float_4::zero());
			}
			// upsample FM input (if connected)
			if (inputs[FM_INPUT].isConnected()) {
				oversamplerInputs[FM_INPUT_UP][c].upsample(inputs[FM_INPUT].getPolyVoltageSimd<float_4>(c));
			}
			else {
				std::fill(osBufferFM, &osBufferFM[oversamplingRatio], float_4::zero());
			}

			float_4* osBufferTri = oversampler[TRI_OUTPUT][c / 4].getOSBuffer();
			float_4* osBufferSaw = oversampler[SAW_OUTPUT][c / 4].getOSBuffer();
			float_4* osBufferSin = oversampler[SINE_OUTPUT][c / 4].getOSBuffer();
			float_4* osBufferSquare = oversampler[SQUARE_OUTPUT][c / 4].getOSBuffer();
			float_4* osBufferEven = oversampler[EVEN_OUTPUT][c / 4].getOSBuffer();
			for (int i = 0; i < oversamplingRatio; ++i) {
				// use upsampled FM input
				const float_4 fmVoltage = osBufferFM[i] * 0.25f;
				const float_4 freq = dsp::FREQ_C4 * simd::pow(2.f, pitchKnobs + pitch + fmVoltage);
				const float_4 deltaBasePhase = simd::clamp(freq * args.sampleTime / oversamplingRatio, 1e-6, 0.5f);
				// floating point arithmetic doesn't work well at low frequencies, specifically because the finite difference denominator
				// becomes tiny - we check for that scenario and use naive / 1st order waveforms in that frequency regime (as aliasing isn't
				// a problem there). With no oversampling, at 44100Hz, the threshold frequency is 44.1Hz.
				const float_4 lowFreqRegime = simd::abs(deltaBasePhase) < 1e-3;
				// 1 / denominator for the second-order FD
				const float_4 denominatorInv = 0.25 / (deltaBasePhase * deltaBasePhase);