Add audio interpolation helpers

src/core/audio/audio_interpolation.cpp

@@ -0,0 +1,73 @@
+// Copyright 2016 Citra Emulator Project
+// Licensed under GPLv2 or any later version
+// Refer to the license.txt file included.
+
+#include "audio/audio_interpolation.hpp"
+
+#include <algorithm>
+
+#include "helpers.hpp"
+
+namespace Audio::Interpolation {
+	// Calculations are done in fixed point with 24 fractional bits.
+	// (This is not verified. This was chosen for minimal error.)
+	static constexpr u64 scaleFactor = 1 << 24;
+	static constexpr u64 scaleMask = scaleFactor - 1;
+
+	/// Here we step over the input in steps of rate, until we consume all of the input.
+	/// Three adjacent samples are passed to fn each step.
+	template <typename Function>
+	static void stepOverSamples(State& state, StereoBuffer16& input, float rate, StereoFrame16& output, usize& outputi, Function fn) {
+		if (input.empty()) {
+			return;
+		}
+
+		input.insert(input.begin(), {state.xn2, state.xn1});
+
+		const u64 step_size = static_cast<u64>(rate * scaleFactor);
+		u64 fposition = state.fposition;
+		usize inputi = 0;
+
+		while (outputi < output.size()) {
+			inputi = static_cast<usize>(fposition / scaleFactor);
+
+			if (inputi + 2 >= input.size()) {
+				inputi = input.size() - 2;
+				break;
+			}
+
+			u64 fraction = fposition & scaleMask;
+			output[outputi++] = fn(fraction, input[inputi], input[inputi + 1], input[inputi + 2]);
+
+			fposition += step_size;
+		}
+
+		state.xn2 = input[inputi];
+		state.xn1 = input[inputi + 1];
+		state.fposition = fposition - inputi * scaleFactor;
+
+		input.erase(input.begin(), std::next(input.begin(), inputi + 2));
+	}
+
+	void none(State& state, StereoBuffer16& input, float rate, StereoFrame16& output, usize& outputi) {
+		stepOverSamples(state, input, rate, output, outputi, [](u64 fraction, const auto& x0, const auto& x1, const auto& x2) { return x0; });
+	}
+
+	void linear(State& state, StereoBuffer16& input, float rate, StereoFrame16& output, usize& outputi) {
+		// Note on accuracy: Some values that this produces are +/- 1 from the actual firmware.
+		stepOverSamples(state, input, rate, output, outputi, [](u64 fraction, const auto& x0, const auto& x1, const auto& x2) {
+			// This is a saturated subtraction. (Verified by black-box fuzzing.)
+			s64 delta0 = std::clamp<s64>(x1[0] - x0[0], -32768, 32767);
+			s64 delta1 = std::clamp<s64>(x1[1] - x0[1], -32768, 32767);
+
+			return std::array<s16, 2>{
+				static_cast<s16>(x0[0] + fraction * delta0 / scaleFactor),
+				static_cast<s16>(x0[1] + fraction * delta1 / scaleFactor),
+			};
+		});
+	}
+
+	void polyphase(State& state, StereoBuffer16& input, float rate, StereoFrame16& output, usize& outputi) {
+		linear(state, input, rate, output, outputi);
+	}
+}  // namespace Audio::Interpolation

src/core/audio/hle_core.cpp

@@ -732,6 +732,7 @@ namespace Audio {
 		rateMultiplier = 1.f;
 
 		buffers = {};
+        interpolationState = {};
 		currentSamples.clear();
 
 		gains.fill({});