Merge branch 'master' into tlds

4 files changed, 16 insertions, 16 deletions

diff --git a/src/audio_core/algorithm/filter.cpp b/src/audio_core/algorithm/filter.cpp
index 9fcd0614d..f65bf64f7 100644
--- a/src/audio_core/algorithm/filter.cpp
+++ b/src/audio_core/algorithm/filter.cpp
@@ -35,12 +35,12 @@ 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 size_t num_frames = signal.size() / 2;
-    for (size_t i = 0; i < num_frames; i++) {
+    const std::size_t num_frames = signal.size() / 2;
+    for (std::size_t i = 0; i < num_frames; i++) {
         std::rotate(in.begin(), in.end() - 1, in.end());
         std::rotate(out.begin(), out.end() - 1, out.end());
 
-        for (size_t ch = 0; ch < channel_count; ch++) {
+        for (std::size_t ch = 0; ch < channel_count; ch++) {
             in[0][ch] = signal[i * channel_count + ch];
 
             out[0][ch] = b0 * in[0][ch] + b1 * in[1][ch] + b2 * in[2][ch] - a1 * out[1][ch] -
@@ -54,14 +54,14 @@ void Filter::Process(std::vector<s16>& signal) {
 /// Calculates the appropriate Q for each biquad in a cascading filter.
 /// @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(size_t total_count, size_t index) {
+static double CascadingBiquadQ(std::size_t total_count, std::size_t index) {
     const double pole = M_PI * (2 * index + 1) / (4.0 * total_count);
     return 1.0 / (2.0 * std::cos(pole));
 }
 
-CascadingFilter CascadingFilter::LowPass(double cutoff, size_t cascade_size) {
+CascadingFilter CascadingFilter::LowPass(double cutoff, std::size_t cascade_size) {
     std::vector<Filter> cascade(cascade_size);
-    for (size_t i = 0; i < cascade_size; i++) {
+    for (std::size_t i = 0; i < cascade_size; i++) {
         cascade[i] = Filter::LowPass(cutoff, CascadingBiquadQ(cascade_size, i));
     }
     return CascadingFilter{std::move(cascade)};
diff --git a/src/audio_core/algorithm/filter.h b/src/audio_core/algorithm/filter.h
index a41beef98..3546d149b 100644
--- a/src/audio_core/algorithm/filter.h
+++ b/src/audio_core/algorithm/filter.h
@@ -30,7 +30,7 @@ public:
     void Process(std::vector<s16>& signal);
 
 private:
-    static constexpr size_t channel_count = 2;
+    static constexpr std::size_t channel_count = 2;
 
     /// Coefficients are in normalized form (a0 = 1.0).
     double a1, a2, b0, b1, b2;
@@ -46,7 +46,7 @@ public:
     /// Creates a cascading low-pass filter.
     /// @param cutoff Determines the cutoff frequency. A value from 0.0 to 1.0.
     /// @param cascade_size Number of biquads in cascade.
-    static CascadingFilter LowPass(double cutoff, size_t cascade_size);
+    static CascadingFilter LowPass(double cutoff, std::size_t cascade_size);
 
     /// Passthrough.
     CascadingFilter();
diff --git a/src/audio_core/algorithm/interpolate.cpp b/src/audio_core/algorithm/interpolate.cpp
index 11459821f..3aea9b0f2 100644
--- a/src/audio_core/algorithm/interpolate.cpp
+++ b/src/audio_core/algorithm/interpolate.cpp
@@ -14,7 +14,7 @@
 namespace AudioCore {
 
 /// The Lanczos kernel
-static double Lanczos(size_t a, double x) {
+static double Lanczos(std::size_t a, double x) {
     if (x == 0.0)
         return 1.0;
     const double px = M_PI * x;
@@ -37,15 +37,15 @@ std::vector<s16> Interpolate(InterpolationState& state, std::vector<s16> input,
     }
     state.nyquist.Process(input);
 
-    constexpr size_t taps = InterpolationState::lanczos_taps;
-    const size_t num_frames = input.size() / 2;
+    constexpr std::size_t taps = InterpolationState::lanczos_taps;
+    const std::size_t num_frames = input.size() / 2;
 
     std::vector<s16> output;
-    output.reserve(static_cast<size_t>(input.size() / ratio + 4));
+    output.reserve(static_cast<std::size_t>(input.size() / ratio + 4));
 
     double& pos = state.position;
     auto& h = state.history;
-    for (size_t i = 0; i < num_frames; ++i) {
+    for (std::size_t i = 0; i < num_frames; ++i) {
         std::rotate(h.begin(), h.end() - 1, h.end());
         h[0][0] = input[i * 2 + 0];
         h[0][1] = input[i * 2 + 1];
@@ -53,7 +53,7 @@ std::vector<s16> Interpolate(InterpolationState& state, std::vector<s16> input,
         while (pos <= 1.0) {
             double l = 0.0;
             double r = 0.0;
-            for (size_t j = 0; j < h.size(); j++) {
+            for (std::size_t j = 0; j < h.size(); j++) {
                 l += Lanczos(taps, pos + j - taps + 1) * h[j][0];
                 r += Lanczos(taps, pos + j - taps + 1) * h[j][1];
             }
diff --git a/src/audio_core/algorithm/interpolate.h b/src/audio_core/algorithm/interpolate.h
index c79c2eef4..edbd6460f 100644
--- a/src/audio_core/algorithm/interpolate.h
+++ b/src/audio_core/algorithm/interpolate.h
@@ -12,8 +12,8 @@
 namespace AudioCore {
 
 struct InterpolationState {
-    static constexpr size_t lanczos_taps = 4;
-    static constexpr size_t history_size = lanczos_taps * 2 - 1;
+    static constexpr std::size_t lanczos_taps = 4;
+    static constexpr std::size_t history_size = lanczos_taps * 2 - 1;
 
     double current_ratio = 0.0;
     CascadingFilter nyquist;