AudioCore: Implement interpolation

src/audio_core/CMakeLists.txt

@@ -4,6 +4,7 @@ set(SRCS
             hle/dsp.cpp
             hle/filter.cpp
             hle/pipe.cpp
+            interpolate.cpp
             )
 
 set(HEADERS
@@ -13,6 +14,7 @@ set(HEADERS
             hle/dsp.h
             hle/filter.h
             hle/pipe.h
+            interpolate.h
             sink.h
             )
 

src/audio_core/interpolate.cpp

@@ -0,0 +1,85 @@
+// Copyright 2016 Citra Emulator Project
+// Licensed under GPLv2 or any later version
+// Refer to the license.txt file included.
+
+#include "audio_core/interpolate.h"
+
+#include "common/assert.h"
+#include "common/math_util.h"
+
+namespace AudioInterp {
+
+// Calculations are done in fixed point with 24 fractional bits.
+// (This is not verified. This was chosen for minimal error.)
+constexpr u64 scale_factor = 1 << 24;
+constexpr u64 scale_mask = scale_factor - 1;
+
+/// Here we step over the input in steps of rate_multiplier, until we consume all of the input.
+/// Three adjacent samples are passed to fn each step.
+template <typename Function>
+static StereoBuffer16 StepOverSamples(State& state, const StereoBuffer16& input, float rate_multiplier, Function fn) {
+    ASSERT(rate_multiplier > 0);
+
+    if (input.size() < 2)
+        return {};
+
+    StereoBuffer16 output;
+    output.reserve(static_cast<size_t>(input.size() / rate_multiplier));
+
+    u64 step_size = static_cast<u64>(rate_multiplier * scale_factor);
+
+    u64 fposition = 0;
+    const u64 max_fposition = input.size() * scale_factor;
+
+    while (fposition < 1 * scale_factor) {
+        u64 fraction = fposition & scale_mask;
+
+        output.push_back(fn(fraction, state.xn2, state.xn1, input[0]));
+
+        fposition += step_size;
+    }
+
+    while (fposition < 2 * scale_factor) {
+        u64 fraction = fposition & scale_mask;
+
+        output.push_back(fn(fraction, state.xn1, input[0], input[1]));
+
+        fposition += step_size;
+    }
+
+    while (fposition < max_fposition) {
+        u64 fraction = fposition & scale_mask;
+
+        size_t index = static_cast<size_t>(fposition / scale_factor);
+        output.push_back(fn(fraction, input[index - 2], input[index - 1], input[index]));
+
+        fposition += step_size;
+    }
+
+    state.xn2 = input[input.size() - 2];
+    state.xn1 = input[input.size() - 1];
+
+    return output;
+}
+
+StereoBuffer16 None(State& state, const StereoBuffer16& input, float rate_multiplier) {
+    return StepOverSamples(state, input, rate_multiplier, [](u64 fraction, const auto& x0, const auto& x1, const auto& x2) {
+        return x0;
+    });
+}
+
+StereoBuffer16 Linear(State& state, const StereoBuffer16& input, float rate_multiplier) {
+    // Note on accuracy: Some values that this produces are +/- 1 from the actual firmware.
+    return StepOverSamples(state, input, rate_multiplier, [](u64 fraction, const auto& x0, const auto& x1, const auto& x2) {
+        // This is a saturated subtraction. (Verified by black-box fuzzing.)
+        s64 delta0 = MathUtil::Clamp<s64>(x1[0] - x0[0], -32768, 32767);
+        s64 delta1 = MathUtil::Clamp<s64>(x1[1] - x0[1], -32768, 32767);
+
+        return std::array<s16, 2> {
+            static_cast<s16>(x0[0] + fraction * delta0 / scale_factor),
+            static_cast<s16>(x0[1] + fraction * delta1 / scale_factor)
+        };
+    });
+}
+
+} // namespace AudioInterp

src/audio_core/interpolate.h

@@ -0,0 +1,41 @@
+// Copyright 2016 Citra Emulator Project
+// Licensed under GPLv2 or any later version
+// Refer to the license.txt file included.
+
+#pragma once
+
+#include <array>
+#include <vector>
+
+#include "common/common_types.h"
+
+namespace AudioInterp {
+
+/// A variable length buffer of signed PCM16 stereo samples.
+using StereoBuffer16 = std::vector<std::array<s16, 2>>;
+
+struct State {
+    // Two historical samples.
+    std::array<s16, 2> xn1 = {}; ///< x[n-1]
+    std::array<s16, 2> xn2 = {}; ///< x[n-2]
+};
+
+/**
+ * No interpolation. This is equivalent to a zero-order hold. There is a two-sample predelay.
+ * @param input Input buffer.
+ * @param rate_multiplier Stretch factor. Must be a positive non-zero value.
+ *                        rate_multiplier > 1.0 performs decimation and rate_multipler < 1.0 performs upsampling.
+ * @return The resampled audio buffer.
+ */
+StereoBuffer16 None(State& state, const StereoBuffer16& input, float rate_multiplier);
+
+/**
+ * Linear interpolation. This is equivalent to a first-order hold. There is a two-sample predelay.
+ * @param input Input buffer.
+ * @param rate_multiplier Stretch factor. Must be a positive non-zero value.
+ *                        rate_multiplier > 1.0 performs decimation and rate_multipler < 1.0 performs upsampling.
+ * @return The resampled audio buffer.
+ */
+StereoBuffer16 Linear(State& state, const StereoBuffer16& input, float rate_multiplier);
+
+} // namespace AudioInterp