8 files changed, 538 insertions, 12 deletions

diff --git a/src/audio_core/CMakeLists.txt b/src/audio_core/CMakeLists.txt
index 13b5e400e..a72a907ef 100644
--- a/src/audio_core/CMakeLists.txt
+++ b/src/audio_core/CMakeLists.txt
@@ -3,10 +3,12 @@ set(SRCS
             codec.cpp
             hle/dsp.cpp
             hle/filter.cpp
+            hle/mixers.cpp
             hle/pipe.cpp
             hle/source.cpp
             interpolate.cpp
             sink_details.cpp
+            time_stretch.cpp
             )
 
 set(HEADERS
@@ -15,12 +17,14 @@ set(HEADERS
             hle/common.h
             hle/dsp.h
             hle/filter.h
+            hle/mixers.h
             hle/pipe.h
             hle/source.h
             interpolate.h
             null_sink.h
             sink.h
             sink_details.h
+            time_stretch.h
             )
 
 include_directories(../../externals/soundtouch/include)
diff --git a/src/audio_core/hle/dsp.cpp b/src/audio_core/hle/dsp.cpp
index 0cdbdb06a..0640e1eff 100644
--- a/src/audio_core/hle/dsp.cpp
+++ b/src/audio_core/hle/dsp.cpp
@@ -6,13 +6,17 @@
 #include <memory>
 
 #include "audio_core/hle/dsp.h"
+#include "audio_core/hle/mixers.h"
 #include "audio_core/hle/pipe.h"
 #include "audio_core/hle/source.h"
 #include "audio_core/sink.h"
+#include "audio_core/time_stretch.h"
 
 namespace DSP {
 namespace HLE {
 
+// Region management
+
 std::array<SharedMemory, 2> g_regions;
 
 static size_t CurrentRegionIndex() {
@@ -40,43 +44,96 @@ static SharedMemory& WriteRegion() {
     return g_regions[1 - CurrentRegionIndex()];
 }
 
+// Audio processing and mixing
+
 static std::array<Source, num_sources> sources = {
     Source(0), Source(1), Source(2), Source(3), Source(4), Source(5),
     Source(6), Source(7), Source(8), Source(9), Source(10), Source(11),
     Source(12), Source(13), Source(14), Source(15), Source(16), Source(17),
     Source(18), Source(19), Source(20), Source(21), Source(22), Source(23)
 };
+static Mixers mixers;
+
+static StereoFrame16 GenerateCurrentFrame() {
+    SharedMemory& read = ReadRegion();
+    SharedMemory& write = WriteRegion();
+
+    std::array<QuadFrame32, 3> intermediate_mixes = {};
+
+    // Generate intermediate mixes
+    for (size_t i = 0; i < num_sources; i++) {
+        write.source_statuses.status[i] = sources[i].Tick(read.source_configurations.config[i], read.adpcm_coefficients.coeff[i]);
+        for (size_t mix = 0; mix < 3; mix++) {
+            sources[i].MixInto(intermediate_mixes[mix], mix);
+        }
+    }
+
+    // Generate final mix
+    write.dsp_status = mixers.Tick(read.dsp_configuration, read.intermediate_mix_samples, write.intermediate_mix_samples, intermediate_mixes);
+
+    StereoFrame16 output_frame = mixers.GetOutput();
+
+    // Write current output frame to the shared memory region
+    for (size_t samplei = 0; samplei < output_frame.size(); samplei++) {
+        for (size_t channeli = 0; channeli < output_frame[0].size(); channeli++) {
+            write.final_samples.pcm16[samplei][channeli] = s16_le(output_frame[samplei][channeli]);
+        }
+    }
+
+    return output_frame;
+}
+
+// Audio output
 
 static std::unique_ptr<AudioCore::Sink> sink;
+static AudioCore::TimeStretcher time_stretcher;
+
+static void OutputCurrentFrame(const StereoFrame16& frame) {
+    time_stretcher.AddSamples(&frame[0][0], frame.size());
+    sink->EnqueueSamples(time_stretcher.Process(sink->SamplesInQueue()));
+}
+
+// Public Interface
 
 void Init() {
     DSP::HLE::ResetPipes();
+
     for (auto& source : sources) {
         source.Reset();
     }
+
+    mixers.Reset();
+
+    time_stretcher.Reset();
+    if (sink) {
+        time_stretcher.SetOutputSampleRate(sink->GetNativeSampleRate());
+    }
 }
 
 void Shutdown() {
+    time_stretcher.Flush();
+    while (true) {
+        std::vector<s16> residual_audio = time_stretcher.Process(sink->SamplesInQueue());
+        if (residual_audio.empty())
+            break;
+        sink->EnqueueSamples(residual_audio);
+    }
 }
 
 bool Tick() {
-    SharedMemory& read = ReadRegion();
-    SharedMemory& write = WriteRegion();
+    StereoFrame16 current_frame = {};
 
-    std::array<QuadFrame32, 3> intermediate_mixes = {};
+    // TODO: Check dsp::DSP semaphore (which indicates emulated application has finished writing to shared memory region)
+    current_frame = GenerateCurrentFrame();
 
-    for (size_t i = 0; i < num_sources; i++) {
-        write.source_statuses.status[i] = sources[i].Tick(read.source_configurations.config[i], read.adpcm_coefficients.coeff[i]);
-        for (size_t mix = 0; mix < 3; mix++) {
-            sources[i].MixInto(intermediate_mixes[mix], mix);
-        }
-    }
+    OutputCurrentFrame(current_frame);
 
     return true;
 }
 
 void SetSink(std::unique_ptr<AudioCore::Sink> sink_) {
     sink = std::move(sink_);
+    time_stretcher.SetOutputSampleRate(sink->GetNativeSampleRate());
 }
 
 } // namespace HLE
diff --git a/src/audio_core/hle/dsp.h b/src/audio_core/hle/dsp.h
index f6e53f68f..9275cd7de 100644
--- a/src/audio_core/hle/dsp.h
+++ b/src/audio_core/hle/dsp.h
@@ -428,7 +428,7 @@ ASSERT_DSP_STRUCT(DspStatus, 32);
 /// Final mixed output in PCM16 stereo format, what you hear out of the speakers.
 /// When the application writes to this region it has no effect.
 struct FinalMixSamples {
-    s16_le pcm16[2 * samples_per_frame];
+    s16_le pcm16[samples_per_frame][2];
 };
 ASSERT_DSP_STRUCT(FinalMixSamples, 640);
 
diff --git a/src/audio_core/hle/mixers.cpp b/src/audio_core/hle/mixers.cpp
new file mode 100644
index 000000000..18335f7f0
--- /dev/null
+++ b/src/audio_core/hle/mixers.cpp
@@ -0,0 +1,201 @@
+// Copyright 2016 Citra Emulator Project
+// Licensed under GPLv2 or any later version
+// Refer to the license.txt file included.
+
+#include <cstddef>
+
+#include "audio_core/hle/common.h"
+#include "audio_core/hle/dsp.h"
+#include "audio_core/hle/mixers.h"
+
+#include "common/assert.h"
+#include "common/logging/log.h"
+#include "common/math_util.h"
+
+namespace DSP {
+namespace HLE {
+
+void Mixers::Reset() {
+    current_frame.fill({});
+    state = {};
+}
+
+DspStatus Mixers::Tick(DspConfiguration& config,
+        const IntermediateMixSamples& read_samples,
+        IntermediateMixSamples& write_samples,
+        const std::array<QuadFrame32, 3>& input)
+{
+    ParseConfig(config);
+
+    AuxReturn(read_samples);
+    AuxSend(write_samples, input);
+
+    MixCurrentFrame();
+
+    return GetCurrentStatus();
+}
+
+void Mixers::ParseConfig(DspConfiguration& config) {
+    if (!config.dirty_raw) {
+        return;
+    }
+
+    if (config.mixer1_enabled_dirty) {
+        config.mixer1_enabled_dirty.Assign(0);
+        state.mixer1_enabled = config.mixer1_enabled != 0;
+        LOG_TRACE(Audio_DSP, "mixers mixer1_enabled = %hu", config.mixer1_enabled);
+    }
+
+    if (config.mixer2_enabled_dirty) {
+        config.mixer2_enabled_dirty.Assign(0);
+        state.mixer2_enabled = config.mixer2_enabled != 0;
+        LOG_TRACE(Audio_DSP, "mixers mixer2_enabled = %hu", config.mixer2_enabled);
+    }
+
+    if (config.volume_0_dirty) {
+        config.volume_0_dirty.Assign(0);
+        state.intermediate_mixer_volume[0] = config.volume[0];
+        LOG_TRACE(Audio_DSP, "mixers volume[0] = %f", config.volume[0]);
+    }
+
+    if (config.volume_1_dirty) {
+        config.volume_1_dirty.Assign(0);
+        state.intermediate_mixer_volume[1] = config.volume[1];
+        LOG_TRACE(Audio_DSP, "mixers volume[1] = %f", config.volume[1]);
+    }
+
+    if (config.volume_2_dirty) {
+        config.volume_2_dirty.Assign(0);
+        state.intermediate_mixer_volume[2] = config.volume[2];
+        LOG_TRACE(Audio_DSP, "mixers volume[2] = %f", config.volume[2]);
+    }
+
+    if (config.output_format_dirty) {
+        config.output_format_dirty.Assign(0);
+        state.output_format = config.output_format;
+        LOG_TRACE(Audio_DSP, "mixers output_format = %zu", static_cast<size_t>(config.output_format));
+    }
+
+    if (config.headphones_connected_dirty) {
+        config.headphones_connected_dirty.Assign(0);
+        // Do nothing.
+        // (Note: Whether headphones are connected does affect coefficients used for surround sound.)
+        LOG_TRACE(Audio_DSP, "mixers headphones_connected=%hu", config.headphones_connected);
+    }
+
+    if (config.dirty_raw) {
+        LOG_DEBUG(Audio_DSP, "mixers remaining_dirty=%x", config.dirty_raw);
+    }
+
+    config.dirty_raw = 0;
+}
+
+static s16 ClampToS16(s32 value) {
+    return static_cast<s16>(MathUtil::Clamp(value, -32768, 32767));
+}
+
+static std::array<s16, 2> AddAndClampToS16(const std::array<s16, 2>& a, const std::array<s16, 2>& b) {
+    return {
+        ClampToS16(static_cast<s32>(a[0]) + static_cast<s32>(b[0])),
+        ClampToS16(static_cast<s32>(a[1]) + static_cast<s32>(b[1]))
+    };
+}
+
+void Mixers::DownmixAndMixIntoCurrentFrame(float gain, const QuadFrame32& samples) {
+    // TODO(merry): Limiter. (Currently we're performing final mixing assuming a disabled limiter.)
+
+    switch (state.output_format) {
+    case OutputFormat::Mono:
+        std::transform(current_frame.begin(), current_frame.end(), samples.begin(), current_frame.begin(),
+            [gain](const std::array<s16, 2>& accumulator, const std::array<s32, 4>& sample) -> std::array<s16, 2> {
+                // Downmix to mono
+                s16 mono = ClampToS16(static_cast<s32>((gain * sample[0] + gain * sample[1] + gain * sample[2] + gain * sample[3]) / 2));
+                // Mix into current frame
+                return AddAndClampToS16(accumulator, { mono, mono });
+            });
+        return;
+
+    case OutputFormat::Surround:
+        // TODO(merry): Implement surround sound.
+        // fallthrough
+
+    case OutputFormat::Stereo:
+        std::transform(current_frame.begin(), current_frame.end(), samples.begin(), current_frame.begin(),
+            [gain](const std::array<s16, 2>& accumulator, const std::array<s32, 4>& sample) -> std::array<s16, 2> {
+                // Downmix to stereo
+                s16 left = ClampToS16(static_cast<s32>(gain * sample[0] + gain * sample[2]));
+                s16 right = ClampToS16(static_cast<s32>(gain * sample[1] + gain * sample[3]));
+                // Mix into current frame
+                return AddAndClampToS16(accumulator, { left, right });
+            });
+        return;
+    }
+
+    UNREACHABLE_MSG("Invalid output_format %zu", static_cast<size_t>(state.output_format));
+}
+
+void Mixers::AuxReturn(const IntermediateMixSamples& read_samples) {
+    // NOTE: read_samples.mix{1,2}.pcm32 annoyingly have their dimensions in reverse order to QuadFrame32.
+
+    if (state.mixer1_enabled) {
+        for (size_t sample = 0; sample < samples_per_frame; sample++) {
+            for (size_t channel = 0; channel < 4; channel++) {
+                state.intermediate_mix_buffer[1][sample][channel] = read_samples.mix1.pcm32[channel][sample];
+            }
+        }
+    }
+
+    if (state.mixer2_enabled) {
+        for (size_t sample = 0; sample < samples_per_frame; sample++) {
+            for (size_t channel = 0; channel < 4; channel++) {
+                state.intermediate_mix_buffer[2][sample][channel] = read_samples.mix2.pcm32[channel][sample];
+            }
+        }
+    }
+}
+
+void Mixers::AuxSend(IntermediateMixSamples& write_samples, const std::array<QuadFrame32, 3>& input) {
+    // NOTE: read_samples.mix{1,2}.pcm32 annoyingly have their dimensions in reverse order to QuadFrame32.
+
+    state.intermediate_mix_buffer[0] = input[0];
+
+    if (state.mixer1_enabled) {
+        for (size_t sample = 0; sample < samples_per_frame; sample++) {
+            for (size_t channel = 0; channel < 4; channel++) {
+                write_samples.mix1.pcm32[channel][sample] = input[1][sample][channel];
+            }
+        }
+    } else {
+        state.intermediate_mix_buffer[1] = input[1];
+    }
+
+    if (state.mixer2_enabled) {
+        for (size_t sample = 0; sample < samples_per_frame; sample++) {
+            for (size_t channel = 0; channel < 4; channel++) {
+                write_samples.mix2.pcm32[channel][sample] = input[2][sample][channel];
+            }
+        }
+    } else {
+        state.intermediate_mix_buffer[2] = input[2];
+    }
+}
+
+void Mixers::MixCurrentFrame() {
+    current_frame.fill({});
+
+    for (size_t mix = 0; mix < 3; mix++) {
+        DownmixAndMixIntoCurrentFrame(state.intermediate_mixer_volume[mix], state.intermediate_mix_buffer[mix]);
+    }
+
+    // TODO(merry): Compressor. (We currently assume a disabled compressor.)
+}
+
+DspStatus Mixers::GetCurrentStatus() const {
+    DspStatus status;
+    status.unknown = 0;
+    status.dropped_frames = 0;
+    return status;
+}
+
+} // namespace HLE
+} // namespace DSP
diff --git a/src/audio_core/hle/mixers.h b/src/audio_core/hle/mixers.h
new file mode 100644
index 000000000..b52952eb5
--- /dev/null
+++ b/src/audio_core/hle/mixers.h
@@ -0,0 +1,63 @@
+// Copyright 2016 Citra Emulator Project
+// Licensed under GPLv2 or any later version
+// Refer to the license.txt file included.
+
+#pragma once
+
+#include <array>
+
+#include "audio_core/hle/common.h"
+#include "audio_core/hle/dsp.h"
+
+namespace DSP {
+namespace HLE {
+
+class Mixers final {
+public:
+    Mixers() {
+        Reset();
+    }
+
+    void Reset();
+
+    DspStatus Tick(DspConfiguration& config,
+                   const IntermediateMixSamples& read_samples,
+                   IntermediateMixSamples& write_samples,
+                   const std::array<QuadFrame32, 3>& input);
+
+    StereoFrame16 GetOutput() const {
+        return current_frame;
+    }
+
+private:
+    StereoFrame16 current_frame = {};
+
+    using OutputFormat = DspConfiguration::OutputFormat;
+
+    struct {
+        std::array<float, 3> intermediate_mixer_volume = {};
+
+        bool mixer1_enabled = false;
+        bool mixer2_enabled = false;
+        std::array<QuadFrame32, 3> intermediate_mix_buffer = {};
+
+        OutputFormat output_format = OutputFormat::Stereo;
+
+    } state;
+
+    /// INTERNAL: Update our internal state based on the current config.
+    void ParseConfig(DspConfiguration& config);
+    /// INTERNAL: Read samples from shared memory that have been modified by the ARM11.
+    void AuxReturn(const IntermediateMixSamples& read_samples);
+    /// INTERNAL: Write samples to shared memory for the ARM11 to modify.
+    void AuxSend(IntermediateMixSamples& write_samples, const std::array<QuadFrame32, 3>& input);
+    /// INTERNAL: Mix current_frame.
+    void MixCurrentFrame();
+    /// INTERNAL: Downmix from quadraphonic to stereo based on status.output_format and accumulate into current_frame.
+    void DownmixAndMixIntoCurrentFrame(float gain, const QuadFrame32& samples);
+    /// INTERNAL: Generate DspStatus based on internal state.
+    DspStatus GetCurrentStatus() const;
+};
+
+} // namespace HLE
+} // namespace DSP
diff --git a/src/audio_core/hle/source.cpp b/src/audio_core/hle/source.cpp
index daaf6e3f3..30552fe26 100644
--- a/src/audio_core/hle/source.cpp
+++ b/src/audio_core/hle/source.cpp
@@ -126,13 +126,13 @@ void Source::ParseConfig(SourceConfiguration::Configuration& config, const s16_l
     if (config.simple_filter_dirty) {
         config.simple_filter_dirty.Assign(0);
         state.filters.Configure(config.simple_filter);
-        LOG_TRACE(Audio_DSP, "source_id=%zu simple filter update");
+        LOG_TRACE(Audio_DSP, "source_id=%zu simple filter update", source_id);
     }
 
     if (config.biquad_filter_dirty) {
         config.biquad_filter_dirty.Assign(0);
         state.filters.Configure(config.biquad_filter);
-        LOG_TRACE(Audio_DSP, "source_id=%zu biquad filter update");
+        LOG_TRACE(Audio_DSP, "source_id=%zu biquad filter update", source_id);
     }
 
     if (config.interpolation_dirty) {
diff --git a/src/audio_core/time_stretch.cpp b/src/audio_core/time_stretch.cpp
new file mode 100644
index 000000000..ea38f40d0
--- /dev/null
+++ b/src/audio_core/time_stretch.cpp
@@ -0,0 +1,144 @@
+// Copyright 2016 Citra Emulator Project
+// Licensed under GPLv2 or any later version
+// Refer to the license.txt file included.
+
+#include <chrono>
+#include <cmath>
+#include <vector>
+
+#include <SoundTouch.h>
+
+#include "audio_core/audio_core.h"
+#include "audio_core/time_stretch.h"
+
+#include "common/common_types.h"
+#include "common/logging/log.h"
+#include "common/math_util.h"
+
+using steady_clock = std::chrono::steady_clock;
+
+namespace AudioCore {
+
+constexpr double MIN_RATIO = 0.1;
+constexpr double MAX_RATIO = 100.0;
+
+static double ClampRatio(double ratio) {
+    return MathUtil::Clamp(ratio, MIN_RATIO, MAX_RATIO);
+}
+
+constexpr double MIN_DELAY_TIME = 0.05; // Units: seconds
+constexpr double MAX_DELAY_TIME = 0.25; // Units: seconds
+constexpr size_t DROP_FRAMES_SAMPLE_DELAY = 16000; // Units: samples
+
+constexpr double SMOOTHING_FACTOR = 0.007;
+
+struct TimeStretcher::Impl {
+    soundtouch::SoundTouch soundtouch;
+
+    steady_clock::time_point frame_timer = steady_clock::now();
+    size_t samples_queued = 0;
+
+    double smoothed_ratio = 1.0;
+
+    double sample_rate = static_cast<double>(native_sample_rate);
+};
+
+std::vector<s16> TimeStretcher::Process(size_t samples_in_queue) {
+    // This is a very simple algorithm without any fancy control theory. It works and is stable.
+
+    double ratio = CalculateCurrentRatio();
+    ratio = CorrectForUnderAndOverflow(ratio, samples_in_queue);
+    impl->smoothed_ratio = (1.0 - SMOOTHING_FACTOR) * impl->smoothed_ratio + SMOOTHING_FACTOR * ratio;
+    impl->smoothed_ratio = ClampRatio(impl->smoothed_ratio);
+
+    // SoundTouch's tempo definition the inverse of our ratio definition.
+    impl->soundtouch.setTempo(1.0 / impl->smoothed_ratio);
+
+    std::vector<s16> samples = GetSamples();
+    if (samples_in_queue >= DROP_FRAMES_SAMPLE_DELAY) {
+        samples.clear();
+        LOG_DEBUG(Audio, "Dropping frames!");
+    }
+    return samples;
+}
+
+TimeStretcher::TimeStretcher() : impl(std::make_unique<Impl>()) {
+    impl->soundtouch.setPitch(1.0);
+    impl->soundtouch.setChannels(2);
+    impl->soundtouch.setSampleRate(native_sample_rate);
+    Reset();
+}
+
+TimeStretcher::~TimeStretcher() {
+    impl->soundtouch.clear();
+}
+
+void TimeStretcher::SetOutputSampleRate(unsigned int sample_rate) {
+    impl->sample_rate = static_cast<double>(sample_rate);
+    impl->soundtouch.setRate(static_cast<double>(native_sample_rate) / impl->sample_rate);
+}
+
+void TimeStretcher::AddSamples(const s16* buffer, size_t num_samples) {
+    impl->soundtouch.putSamples(buffer, static_cast<uint>(num_samples));
+    impl->samples_queued += num_samples;
+}
+
+void TimeStretcher::Flush() {
+    impl->soundtouch.flush();
+}
+
+void TimeStretcher::Reset() {
+    impl->soundtouch.setTempo(1.0);
+    impl->soundtouch.clear();
+    impl->smoothed_ratio = 1.0;
+    impl->frame_timer = steady_clock::now();
+    impl->samples_queued = 0;
+    SetOutputSampleRate(native_sample_rate);
+}
+
+double TimeStretcher::CalculateCurrentRatio() {
+    const steady_clock::time_point now = steady_clock::now();
+    const std::chrono::duration<double> duration = now - impl->frame_timer;
+
+    const double expected_time = static_cast<double>(impl->samples_queued) / static_cast<double>(native_sample_rate);
+    const double actual_time = duration.count();
+
+    double ratio;
+    if (expected_time != 0) {
+        ratio = ClampRatio(actual_time / expected_time);
+    } else {
+        ratio = impl->smoothed_ratio;
+    }
+
+    impl->frame_timer = now;
+    impl->samples_queued = 0;
+
+    return ratio;
+}
+
+double TimeStretcher::CorrectForUnderAndOverflow(double ratio, size_t sample_delay) const {
+    const size_t min_sample_delay = static_cast<size_t>(MIN_DELAY_TIME * impl->sample_rate);
+    const size_t max_sample_delay = static_cast<size_t>(MAX_DELAY_TIME * impl->sample_rate);
+
+    if (sample_delay < min_sample_delay) {
+        // Make the ratio bigger.
+        ratio = ratio > 1.0 ? ratio * ratio : sqrt(ratio);
+    } else if (sample_delay > max_sample_delay) {
+        // Make the ratio smaller.
+        ratio = ratio > 1.0 ? sqrt(ratio) : ratio * ratio;
+    }
+
+    return ClampRatio(ratio);
+}
+
+std::vector<s16> TimeStretcher::GetSamples() {
+    uint available = impl->soundtouch.numSamples();
+
+    std::vector<s16> output(static_cast<size_t>(available) * 2);
+
+    impl->soundtouch.receiveSamples(output.data(), available);
+
+    return output;
+}
+
+} // namespace AudioCore
diff --git a/src/audio_core/time_stretch.h b/src/audio_core/time_stretch.h
new file mode 100644
index 000000000..1fde3f72a
--- /dev/null
+++ b/src/audio_core/time_stretch.h
@@ -0,0 +1,57 @@
+// Copyright 2016 Citra Emulator Project
+// Licensed under GPLv2 or any later version
+// Refer to the license.txt file included.
+
+#include <cstddef>
+#include <memory>
+#include <vector>
+
+#include "common/common_types.h"
+
+namespace AudioCore {
+
+class TimeStretcher final {
+public:
+    TimeStretcher();
+    ~TimeStretcher();
+
+    /**
+     * Set sample rate for the samples that Process returns.
+     * @param sample_rate The sample rate.
+     */
+    void SetOutputSampleRate(unsigned int sample_rate);
+
+    /**
+     * Add samples to be processed.
+     * @param sample_buffer Buffer of samples in interleaved stereo PCM16 format.
+     * @param num_sample Number of samples.
+     */
+    void AddSamples(const s16* sample_buffer, size_t num_samples);
+
+    /// Flush audio remaining in internal buffers.
+    void Flush();
+
+    /// Resets internal state and clears buffers.
+    void Reset();
+
+    /**
+     * Does audio stretching and produces the time-stretched samples.
+     * Timer calculations use sample_delay to determine how much of a margin we have.
+     * @param sample_delay How many samples are buffered downstream of this module and haven't been played yet.
+     * @return Samples to play in interleaved stereo PCM16 format.
+     */
+    std::vector<s16> Process(size_t sample_delay);
+
+private:
+    struct Impl;
+    std::unique_ptr<Impl> impl;
+
+    /// INTERNAL: ratio = wallclock time / emulated time
+    double CalculateCurrentRatio();
+    /// INTERNAL: If we have too many or too few samples downstream, nudge ratio in the appropriate direction.
+    double CorrectForUnderAndOverflow(double ratio, size_t sample_delay) const;
+    /// INTERNAL: Gets the time-stretched samples from SoundTouch.
+    std::vector<s16> GetSamples();
+};
+
+} // namespace AudioCore