diff --git a/src/audio_core/CMakeLists.txt b/src/audio_core/CMakeLists.txt
index 92322f59bc..82e4850f74 100644
--- a/src/audio_core/CMakeLists.txt
+++ b/src/audio_core/CMakeLists.txt
@@ -1,6 +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
diff --git a/src/audio_core/algorithm/interpolate.cpp b/src/audio_core/algorithm/interpolate.cpp
new file mode 100644
index 0000000000..11459821f9
--- /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 0000000000..c79c2eef41
--- /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 6ebed3fb0d..7bff635b8c 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;
 }
 
diff --git a/src/audio_core/audio_renderer.h b/src/audio_core/audio_renderer.h
index 13c5d0adc9..eba67f28e1 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{};