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-rw-r--r--src/audio_core/CMakeLists.txt2
-rw-r--r--src/audio_core/interpolate.cpp85
-rw-r--r--src/audio_core/interpolate.h41
3 files changed, 128 insertions, 0 deletions
diff --git a/src/audio_core/CMakeLists.txt b/src/audio_core/CMakeLists.txt
index 869da5e83..b1c86462f 100644
--- a/src/audio_core/CMakeLists.txt
+++ b/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
)
diff --git a/src/audio_core/interpolate.cpp b/src/audio_core/interpolate.cpp
new file mode 100644
index 000000000..fcd3aa066
--- /dev/null
+++ b/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
diff --git a/src/audio_core/interpolate.h b/src/audio_core/interpolate.h
new file mode 100644
index 000000000..a4c0a453d
--- /dev/null
+++ b/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