diff --git a/src/audio_core/CMakeLists.txt b/src/audio_core/CMakeLists.txt
index ec71524a3b50f7806fd6f9aeb7e8937588591dd7..82e4850f7425a529207c6372eeaa19914cbd07a0 100644
--- a/src/audio_core/CMakeLists.txt
+++ b/src/audio_core/CMakeLists.txt
@@ -1,4 +1,8 @@
add_library(audio_core STATIC
+ algorithm/filter.cpp
+ algorithm/filter.h
+ algorithm/interpolate.cpp
+ algorithm/interpolate.h
audio_out.cpp
audio_out.h
audio_renderer.cpp
@@ -7,12 +11,12 @@ add_library(audio_core STATIC
codec.cpp
codec.h
null_sink.h
- stream.cpp
- stream.h
sink.h
sink_details.cpp
sink_details.h
sink_stream.h
+ stream.cpp
+ stream.h
$<$<BOOL:${ENABLE_CUBEB}>:cubeb_sink.cpp cubeb_sink.h>
)
diff --git a/src/audio_core/algorithm/filter.cpp b/src/audio_core/algorithm/filter.cpp
new file mode 100644
index 0000000000000000000000000000000000000000..403b8503ffd129563888f80486d9ffd91fcd0d61
--- /dev/null
+++ b/src/audio_core/algorithm/filter.cpp
@@ -0,0 +1,79 @@
+// Copyright 2018 yuzu Emulator Project
+// Licensed under GPLv2 or any later version
+// Refer to the license.txt file included.
+
+#define _USE_MATH_DEFINES
+
+#include <algorithm>
+#include <array>
+#include <cmath>
+#include <vector>
+#include "audio_core/algorithm/filter.h"
+#include "common/common_types.h"
+
+namespace AudioCore {
+
+Filter Filter::LowPass(double cutoff, double Q) {
+ const double w0 = 2.0 * M_PI * cutoff;
+ const double sin_w0 = std::sin(w0);
+ const double cos_w0 = std::cos(w0);
+ const double alpha = sin_w0 / (2 * Q);
+
+ const double a0 = 1 + alpha;
+ const double a1 = -2.0 * cos_w0;
+ const double a2 = 1 - alpha;
+ const double b0 = 0.5 * (1 - cos_w0);
+ const double b1 = 1.0 * (1 - cos_w0);
+ const double b2 = 0.5 * (1 - cos_w0);
+
+ return {a0, a1, a2, b0, b1, b2};
+}
+
+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) {}
+
+void Filter::Process(std::vector<s16>& signal) {
+ const size_t num_frames = signal.size() / 2;
+ for (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++) {
+ 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] -
+ a2 * out[2][ch];
+
+ signal[i * 2 + ch] = std::clamp(out[0][ch], -32768.0, 32767.0);
+ }
+ }
+}
+
+/// 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) {
+ 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) {
+ std::vector<Filter> cascade(cascade_size);
+ for (size_t i = 0; i < cascade_size; i++) {
+ cascade[i] = Filter::LowPass(cutoff, CascadingBiquadQ(cascade_size, i));
+ }
+ return CascadingFilter{std::move(cascade)};
+}
+
+CascadingFilter::CascadingFilter() = default;
+CascadingFilter::CascadingFilter(std::vector<Filter> filters) : filters(std::move(filters)) {}
+
+void CascadingFilter::Process(std::vector<s16>& signal) {
+ for (auto& filter : filters) {
+ filter.Process(signal);
+ }
+}
+
+} // namespace AudioCore
diff --git a/src/audio_core/algorithm/filter.h b/src/audio_core/algorithm/filter.h
new file mode 100644
index 0000000000000000000000000000000000000000..a41beef98bcbb338ab8deb4fa9d09e7735280e61
--- /dev/null
+++ b/src/audio_core/algorithm/filter.h
@@ -0,0 +1,62 @@
+// Copyright 2018 yuzu 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 AudioCore {
+
+/// Digital biquad filter:
+///
+/// b0 + b1 z^-1 + b2 z^-2
+/// H(z) = ------------------------
+/// a0 + a1 z^-1 + b2 z^-2
+class Filter {
+public:
+ /// Creates a low-pass filter.
+ /// @param cutoff Determines the cutoff frequency. A value from 0.0 to 1.0.
+ /// @param Q Determines the quality factor of this filter.
+ static Filter LowPass(double cutoff, double Q = 0.7071);
+
+ /// Passthrough filter.
+ Filter();
+
+ Filter(double a0, double a1, double a2, double b0, double b1, double b2);
+
+ void Process(std::vector<s16>& signal);
+
+private:
+ static constexpr size_t channel_count = 2;
+
+ /// Coefficients are in normalized form (a0 = 1.0).
+ double a1, a2, b0, b1, b2;
+ /// Input History
+ std::array<std::array<double, channel_count>, 3> in;
+ /// Output History
+ std::array<std::array<double, channel_count>, 3> out;
+};
+
+/// Cascade filters to build up higher-order filters from lower-order ones.
+class CascadingFilter {
+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);
+
+ /// Passthrough.
+ CascadingFilter();
+
+ explicit CascadingFilter(std::vector<Filter> filters);
+
+ void Process(std::vector<s16>& signal);
+
+private:
+ std::vector<Filter> filters;
+};
+
+} // namespace AudioCore
diff --git a/src/audio_core/algorithm/interpolate.cpp b/src/audio_core/algorithm/interpolate.cpp
new file mode 100644
index 0000000000000000000000000000000000000000..11459821f903fb4b83eed357da274a21b7159738
--- /dev/null
+++ b/src/audio_core/algorithm/interpolate.cpp
@@ -0,0 +1,71 @@
+// Copyright 2018 yuzu Emulator Project
+// Licensed under GPLv2 or any later version
+// Refer to the license.txt file included.
+
+#define _USE_MATH_DEFINES
+
+#include <algorithm>
+#include <cmath>
+#include <vector>
+#include "audio_core/algorithm/interpolate.h"
+#include "common/common_types.h"
+#include "common/logging/log.h"
+
+namespace AudioCore {
+
+/// The Lanczos kernel
+static double Lanczos(size_t a, double x) {
+ if (x == 0.0)
+ return 1.0;
+ const double px = M_PI * x;
+ return a * std::sin(px) * std::sin(px / a) / (px * px);
+}
+
+std::vector<s16> Interpolate(InterpolationState& state, std::vector<s16> input, double ratio) {
+ if (input.size() < 2)
+ return {};
+
+ if (ratio <= 0) {
+ LOG_CRITICAL(Audio, "Nonsensical interpolation ratio {}", ratio);
+ ratio = 1.0;
+ }
+
+ if (ratio != state.current_ratio) {
+ const double cutoff_frequency = std::min(0.5 / ratio, 0.5 * ratio);
+ state.nyquist = CascadingFilter::LowPass(std::clamp(cutoff_frequency, 0.0, 0.4), 3);
+ state.current_ratio = ratio;
+ }
+ state.nyquist.Process(input);
+
+ constexpr size_t taps = InterpolationState::lanczos_taps;
+ const size_t num_frames = input.size() / 2;
+
+ std::vector<s16> output;
+ output.reserve(static_cast<size_t>(input.size() / ratio + 4));
+
+ double& pos = state.position;
+ auto& h = state.history;
+ for (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];
+
+ while (pos <= 1.0) {
+ double l = 0.0;
+ double r = 0.0;
+ for (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];
+ }
+ output.emplace_back(static_cast<s16>(std::clamp(l, -32768.0, 32767.0)));
+ output.emplace_back(static_cast<s16>(std::clamp(r, -32768.0, 32767.0)));
+
+ pos += ratio;
+ }
+ pos -= 1.0;
+ }
+
+ return output;
+}
+
+} // namespace AudioCore
diff --git a/src/audio_core/algorithm/interpolate.h b/src/audio_core/algorithm/interpolate.h
new file mode 100644
index 0000000000000000000000000000000000000000..c79c2eef41ea2e265956b41e810b42a810f51112
--- /dev/null
+++ b/src/audio_core/algorithm/interpolate.h
@@ -0,0 +1,43 @@
+// Copyright 2018 yuzu Emulator Project
+// Licensed under GPLv2 or any later version
+// Refer to the license.txt file included.
+
+#pragma once
+
+#include <array>
+#include <vector>
+#include "audio_core/algorithm/filter.h"
+#include "common/common_types.h"
+
+namespace AudioCore {
+
+struct InterpolationState {
+ static constexpr size_t lanczos_taps = 4;
+ static constexpr size_t history_size = lanczos_taps * 2 - 1;
+
+ double current_ratio = 0.0;
+ CascadingFilter nyquist;
+ std::array<std::array<s16, 2>, history_size> history = {};
+ double position = 0;
+};
+
+/// Interpolates input signal to produce output signal.
+/// @param input The signal to interpolate.
+/// @param ratio Interpolation ratio.
+/// ratio > 1.0 results in fewer output samples.
+/// ratio < 1.0 results in more output samples.
+/// @returns Output signal.
+std::vector<s16> Interpolate(InterpolationState& state, std::vector<s16> input, double ratio);
+
+/// Interpolates input signal to produce output signal.
+/// @param input The signal to interpolate.
+/// @param input_rate The sample rate of input.
+/// @param output_rate The desired sample rate of the output.
+/// @returns Output signal.
+inline std::vector<s16> Interpolate(InterpolationState& state, std::vector<s16> input,
+ u32 input_rate, u32 output_rate) {
+ const double ratio = static_cast<double>(input_rate) / static_cast<double>(output_rate);
+ return Interpolate(state, std::move(input), ratio);
+}
+
+} // namespace AudioCore
diff --git a/src/audio_core/audio_renderer.cpp b/src/audio_core/audio_renderer.cpp
index 6ebed3fb0d2ec34a4d40ef5da24c3cd62c596167..397b107f582b35a75f2401ef6278d617a107f44a 100644
--- a/src/audio_core/audio_renderer.cpp
+++ b/src/audio_core/audio_renderer.cpp
@@ -2,6 +2,7 @@
// Licensed under GPLv2 or any later version
// Refer to the license.txt file included.
+#include "audio_core/algorithm/interpolate.h"
#include "audio_core/audio_renderer.h"
#include "common/assert.h"
#include "common/logging/log.h"
@@ -199,6 +200,8 @@ void AudioRenderer::VoiceState::RefreshBuffer() {
break;
}
+ samples = Interpolate(interp_state, std::move(samples), Info().sample_rate, STREAM_SAMPLE_RATE);
+
is_refresh_pending = false;
}
@@ -224,7 +227,7 @@ void AudioRenderer::QueueMixedBuffer(Buffer::Tag tag) {
break;
}
- samples_remaining -= samples.size();
+ samples_remaining -= samples.size() / stream->GetNumChannels();
for (const auto& sample : samples) {
const s32 buffer_sample{buffer[offset]};
diff --git a/src/audio_core/audio_renderer.h b/src/audio_core/audio_renderer.h
index 13c5d0adc9d1b2f0fba95c211588593ed0243179..eba67f28e1a42306bbf1dac3d2e050555203cbb1 100644
--- a/src/audio_core/audio_renderer.h
+++ b/src/audio_core/audio_renderer.h
@@ -8,6 +8,7 @@
#include <memory>
#include <vector>
+#include "audio_core/algorithm/interpolate.h"
#include "audio_core/audio_out.h"
#include "audio_core/codec.h"
#include "audio_core/stream.h"
@@ -194,6 +195,7 @@ private:
size_t wave_index{};
size_t offset{};
Codec::ADPCMState adpcm_state{};
+ InterpolationState interp_state{};
std::vector<s16> samples;
VoiceOutStatus out_status{};
VoiceInfo info{};