4 files changed, 45 insertions, 8 deletions
diff --git a/src/audio_core/algorithm/filter.cpp b/src/audio_core/algorithm/filter.cpp
index f65bf64f7..01b8dff6b 100644
--- a/src/audio_core/algorithm/filter.cpp
+++ b/src/audio_core/algorithm/filter.cpp
@@ -31,8 +31,8 @@ Filter Filter::LowPass(double cutoff, double Q) {
 
 Filter::Filter() : Filter(1.0, 0.0, 0.0, 1.0, 0.0, 0.0) {}
 
-Filter::Filter(double a0, double a1, double a2, double b0, double b1, double b2)
-    : a1(a1 / a0), a2(a2 / a0), b0(b0 / a0), b1(b1 / a0), b2(b2 / a0) {}
+Filter::Filter(double a0_, double a1_, double a2_, double b0_, double b1_, double b2_)
+    : a1(a1_ / a0_), a2(a2_ / a0_), b0(b0_ / a0_), b1(b1_ / a0_), b2(b2_ / a0_) {}
 
 void Filter::Process(std::vector<s16>& signal) {
     const std::size_t num_frames = signal.size() / 2;
@@ -55,7 +55,8 @@ void Filter::Process(std::vector<s16>& signal) {
 /// @param total_count The total number of biquads to be cascaded.
 /// @param index 0-index of the biquad to calculate the Q value for.
 static double CascadingBiquadQ(std::size_t total_count, std::size_t index) {
-    const double pole = M_PI * (2 * index + 1) / (4.0 * total_count);
+    const auto pole =
+        M_PI * static_cast<double>(2 * index + 1) / (4.0 * static_cast<double>(total_count));
     return 1.0 / (2.0 * std::cos(pole));
 }
 
@@ -68,7 +69,7 @@ CascadingFilter CascadingFilter::LowPass(double cutoff, std::size_t cascade_size
 }
 
 CascadingFilter::CascadingFilter() = default;
-CascadingFilter::CascadingFilter(std::vector<Filter> filters) : filters(std::move(filters)) {}
+CascadingFilter::CascadingFilter(std::vector<Filter> filters_) : filters(std::move(filters_)) {}
 
 void CascadingFilter::Process(std::vector<s16>& signal) {
     for (auto& filter : filters) {
diff --git a/src/audio_core/algorithm/filter.h b/src/audio_core/algorithm/filter.h
index 3546d149b..a291fe79b 100644
--- a/src/audio_core/algorithm/filter.h
+++ b/src/audio_core/algorithm/filter.h
@@ -25,7 +25,7 @@ public:
     /// Passthrough filter.
     Filter();
 
-    Filter(double a0, double a1, double a2, double b0, double b1, double b2);
+    Filter(double a0_, double a1_, double a2_, double b0_, double b1_, double b2_);
 
     void Process(std::vector<s16>& signal);
 
@@ -51,7 +51,7 @@ public:
     /// Passthrough.
     CascadingFilter();
 
-    explicit CascadingFilter(std::vector<Filter> filters);
+    explicit CascadingFilter(std::vector<Filter> filters_);
 
     void Process(std::vector<s16>& signal);
 
diff --git a/src/audio_core/algorithm/interpolate.cpp b/src/audio_core/algorithm/interpolate.cpp
index 49ab9d3e1..3b4144e21 100644
--- a/src/audio_core/algorithm/interpolate.cpp
+++ b/src/audio_core/algorithm/interpolate.cpp
@@ -146,7 +146,7 @@ std::vector<s16> Interpolate(InterpolationState& state, std::vector<s16> input,
         return {};
 
     if (ratio <= 0) {
-        LOG_CRITICAL(Audio, "Nonsensical interpolation ratio {}", ratio);
+        LOG_ERROR(Audio, "Nonsensical interpolation ratio {}", ratio);
         return input;
     }
 
@@ -164,7 +164,8 @@ std::vector<s16> Interpolate(InterpolationState& state, std::vector<s16> input,
     const std::size_t num_frames{input.size() / 2};
 
     std::vector<s16> output;
-    output.reserve(static_cast<std::size_t>(input.size() / ratio + InterpolationState::taps));
+    output.reserve(static_cast<std::size_t>(static_cast<double>(input.size()) / ratio +
+                                            InterpolationState::taps));
 
     for (std::size_t frame{}; frame < num_frames; ++frame) {
         const std::size_t lut_index{(state.fraction >> 8) * InterpolationState::taps};
@@ -197,4 +198,36 @@ std::vector<s16> Interpolate(InterpolationState& state, std::vector<s16> input,
     return output;
 }
 
+void Resample(s32* output, const s32* input, s32 pitch, s32& fraction, std::size_t sample_count) {
+    const std::array<s16, 512>& lut = [pitch] {
+        if (pitch > 0xaaaa) {
+            return curve_lut0;
+        }
+        if (pitch <= 0x8000) {
+            return curve_lut1;
+        }
+        return curve_lut2;
+    }();
+
+    std::size_t index{};
+
+    for (std::size_t i = 0; i < sample_count; i++) {
+        const std::size_t lut_index{(static_cast<std::size_t>(fraction) >> 8) * 4};
+        const auto l0 = lut[lut_index + 0];
+        const auto l1 = lut[lut_index + 1];
+        const auto l2 = lut[lut_index + 2];
+        const auto l3 = lut[lut_index + 3];
+
+        const auto s0 = static_cast<s32>(input[index + 0]);
+        const auto s1 = static_cast<s32>(input[index + 1]);
+        const auto s2 = static_cast<s32>(input[index + 2]);
+        const auto s3 = static_cast<s32>(input[index + 3]);
+
+        output[i] = (l0 * s0 + l1 * s1 + l2 * s2 + l3 * s3) >> 15;
+        fraction += pitch;
+        index += (fraction >> 15);
+        fraction &= 0x7fff;
+    }
+}
+
 } // namespace AudioCore
diff --git a/src/audio_core/algorithm/interpolate.h b/src/audio_core/algorithm/interpolate.h
index ab1a31754..d534077af 100644
--- a/src/audio_core/algorithm/interpolate.h
+++ b/src/audio_core/algorithm/interpolate.h
@@ -38,4 +38,7 @@ inline std::vector<s16> Interpolate(InterpolationState& state, std::vector<s16>
     return Interpolate(state, std::move(input), ratio);
 }
 
+/// Nintendo Switchs DSP resampling algorithm. Based on a single channel
+void Resample(s32* output, const s32* input, s32 pitch, s32& fraction, std::size_t sample_count);
+
 } // namespace AudioCore