17 files changed, 1364 insertions, 0 deletions

diff --git a/src/audio_core/CMakeLists.txt b/src/audio_core/CMakeLists.txt
new file mode 100644
index 000000000..ec71524a3
--- /dev/null
+++ b/src/audio_core/CMakeLists.txt
@@ -0,0 +1,27 @@
+add_library(audio_core STATIC
+    audio_out.cpp
+    audio_out.h
+    audio_renderer.cpp
+    audio_renderer.h
+    buffer.h
+    codec.cpp
+    codec.h
+    null_sink.h
+    stream.cpp
+    stream.h
+    sink.h
+    sink_details.cpp
+    sink_details.h
+    sink_stream.h
+
+    $<$<BOOL:${ENABLE_CUBEB}>:cubeb_sink.cpp cubeb_sink.h>
+)
+
+create_target_directory_groups(audio_core)
+
+target_link_libraries(audio_core PUBLIC common core)
+
+if(ENABLE_CUBEB)
+    target_link_libraries(audio_core PRIVATE cubeb)
+    target_compile_definitions(audio_core PRIVATE -DHAVE_CUBEB=1)
+endif()
diff --git a/src/audio_core/audio_out.cpp b/src/audio_core/audio_out.cpp
new file mode 100644
index 000000000..12632a95c
--- /dev/null
+++ b/src/audio_core/audio_out.cpp
@@ -0,0 +1,58 @@
+// Copyright 2018 yuzu Emulator Project
+// Licensed under GPLv2 or any later version
+// Refer to the license.txt file included.
+
+#include "audio_core/audio_out.h"
+#include "audio_core/sink.h"
+#include "audio_core/sink_details.h"
+#include "common/assert.h"
+#include "common/logging/log.h"
+#include "core/settings.h"
+
+namespace AudioCore {
+
+/// Returns the stream format from the specified number of channels
+static Stream::Format ChannelsToStreamFormat(u32 num_channels) {
+    switch (num_channels) {
+    case 1:
+        return Stream::Format::Mono16;
+    case 2:
+        return Stream::Format::Stereo16;
+    case 6:
+        return Stream::Format::Multi51Channel16;
+    }
+
+    LOG_CRITICAL(Audio, "Unimplemented num_channels={}", num_channels);
+    UNREACHABLE();
+    return {};
+}
+
+StreamPtr AudioOut::OpenStream(u32 sample_rate, u32 num_channels, std::string&& name,
+                               Stream::ReleaseCallback&& release_callback) {
+    if (!sink) {
+        const SinkDetails& sink_details = GetSinkDetails(Settings::values.sink_id);
+        sink = sink_details.factory(Settings::values.audio_device_id);
+    }
+
+    return std::make_shared<Stream>(
+        sample_rate, ChannelsToStreamFormat(num_channels), std::move(release_callback),
+        sink->AcquireSinkStream(sample_rate, num_channels, name), std::move(name));
+}
+
+std::vector<Buffer::Tag> AudioOut::GetTagsAndReleaseBuffers(StreamPtr stream, size_t max_count) {
+    return stream->GetTagsAndReleaseBuffers(max_count);
+}
+
+void AudioOut::StartStream(StreamPtr stream) {
+    stream->Play();
+}
+
+void AudioOut::StopStream(StreamPtr stream) {
+    stream->Stop();
+}
+
+bool AudioOut::QueueBuffer(StreamPtr stream, Buffer::Tag tag, std::vector<s16>&& data) {
+    return stream->QueueBuffer(std::make_shared<Buffer>(tag, std::move(data)));
+}
+
+} // namespace AudioCore
diff --git a/src/audio_core/audio_out.h b/src/audio_core/audio_out.h
new file mode 100644
index 000000000..39b7e656b
--- /dev/null
+++ b/src/audio_core/audio_out.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 <memory>
+#include <string>
+#include <vector>
+
+#include "audio_core/buffer.h"
+#include "audio_core/sink.h"
+#include "audio_core/stream.h"
+#include "common/common_types.h"
+
+namespace AudioCore {
+
+/**
+ * Represents an audio playback interface, used to open and play audio streams
+ */
+class AudioOut {
+public:
+    /// Opens a new audio stream
+    StreamPtr OpenStream(u32 sample_rate, u32 num_channels, std::string&& name,
+                         Stream::ReleaseCallback&& release_callback);
+
+    /// Returns a vector of recently released buffers specified by tag for the specified stream
+    std::vector<Buffer::Tag> GetTagsAndReleaseBuffers(StreamPtr stream, size_t max_count);
+
+    /// Starts an audio stream for playback
+    void StartStream(StreamPtr stream);
+
+    /// Stops an audio stream that is currently playing
+    void StopStream(StreamPtr stream);
+
+    /// Queues a buffer into the specified audio stream, returns true on success
+    bool QueueBuffer(StreamPtr stream, Buffer::Tag tag, std::vector<s16>&& data);
+
+private:
+    SinkPtr sink;
+};
+
+} // namespace AudioCore
diff --git a/src/audio_core/audio_renderer.cpp b/src/audio_core/audio_renderer.cpp
new file mode 100644
index 000000000..282f345c5
--- /dev/null
+++ b/src/audio_core/audio_renderer.cpp
@@ -0,0 +1,234 @@
+// Copyright 2018 yuzu Emulator Project
+// Licensed under GPLv2 or any later version
+// Refer to the license.txt file included.
+
+#include "audio_core/audio_renderer.h"
+#include "common/assert.h"
+#include "common/logging/log.h"
+#include "core/memory.h"
+
+namespace AudioCore {
+
+constexpr u32 STREAM_SAMPLE_RATE{48000};
+constexpr u32 STREAM_NUM_CHANNELS{2};
+
+AudioRenderer::AudioRenderer(AudioRendererParameter params,
+                             Kernel::SharedPtr<Kernel::Event> buffer_event)
+    : worker_params{params}, buffer_event{buffer_event}, voices(params.voice_count) {
+
+    audio_core = std::make_unique<AudioCore::AudioOut>();
+    stream = audio_core->OpenStream(STREAM_SAMPLE_RATE, STREAM_NUM_CHANNELS, "AudioRenderer",
+                                    [=]() { buffer_event->Signal(); });
+    audio_core->StartStream(stream);
+
+    QueueMixedBuffer(0);
+    QueueMixedBuffer(1);
+    QueueMixedBuffer(2);
+}
+
+std::vector<u8> AudioRenderer::UpdateAudioRenderer(const std::vector<u8>& input_params) {
+    // Copy UpdateDataHeader struct
+    UpdateDataHeader config{};
+    std::memcpy(&config, input_params.data(), sizeof(UpdateDataHeader));
+    u32 memory_pool_count = worker_params.effect_count + (worker_params.voice_count * 4);
+
+    // Copy MemoryPoolInfo structs
+    std::vector<MemoryPoolInfo> mem_pool_info(memory_pool_count);
+    std::memcpy(mem_pool_info.data(),
+                input_params.data() + sizeof(UpdateDataHeader) + config.behavior_size,
+                memory_pool_count * sizeof(MemoryPoolInfo));
+
+    // Copy VoiceInfo structs
+    size_t offset{sizeof(UpdateDataHeader) + config.behavior_size + config.memory_pools_size +
+                  config.voice_resource_size};
+    for (auto& voice : voices) {
+        std::memcpy(&voice.Info(), input_params.data() + offset, sizeof(VoiceInfo));
+        offset += sizeof(VoiceInfo);
+    }
+
+    // Update voices
+    for (auto& voice : voices) {
+        voice.UpdateState();
+        if (!voice.GetInfo().is_in_use) {
+            continue;
+        }
+        if (voice.GetInfo().is_new) {
+            voice.SetWaveIndex(voice.GetInfo().wave_buffer_head);
+        }
+    }
+
+    // Update memory pool state
+    std::vector<MemoryPoolEntry> memory_pool(memory_pool_count);
+    for (size_t index = 0; index < memory_pool.size(); ++index) {
+        if (mem_pool_info[index].pool_state == MemoryPoolStates::RequestAttach) {
+            memory_pool[index].state = MemoryPoolStates::Attached;
+        } else if (mem_pool_info[index].pool_state == MemoryPoolStates::RequestDetach) {
+            memory_pool[index].state = MemoryPoolStates::Detached;
+        }
+    }
+
+    // Release previous buffers and queue next ones for playback
+    ReleaseAndQueueBuffers();
+
+    // Copy output header
+    UpdateDataHeader response_data{worker_params};
+    std::vector<u8> output_params(response_data.total_size);
+    std::memcpy(output_params.data(), &response_data, sizeof(UpdateDataHeader));
+
+    // Copy output memory pool entries
+    std::memcpy(output_params.data() + sizeof(UpdateDataHeader), memory_pool.data(),
+                response_data.memory_pools_size);
+
+    // Copy output voice status
+    size_t voice_out_status_offset{sizeof(UpdateDataHeader) + response_data.memory_pools_size};
+    for (const auto& voice : voices) {
+        std::memcpy(output_params.data() + voice_out_status_offset, &voice.GetOutStatus(),
+                    sizeof(VoiceOutStatus));
+        voice_out_status_offset += sizeof(VoiceOutStatus);
+    }
+
+    return output_params;
+}
+
+void AudioRenderer::VoiceState::SetWaveIndex(size_t index) {
+    wave_index = index & 3;
+    is_refresh_pending = true;
+}
+
+std::vector<s16> AudioRenderer::VoiceState::DequeueSamples(size_t sample_count) {
+    if (!IsPlaying()) {
+        return {};
+    }
+
+    if (is_refresh_pending) {
+        RefreshBuffer();
+    }
+
+    const size_t max_size{samples.size() - offset};
+    const size_t dequeue_offset{offset};
+    size_t size{sample_count * STREAM_NUM_CHANNELS};
+    if (size > max_size) {
+        size = max_size;
+    }
+
+    out_status.played_sample_count += size / STREAM_NUM_CHANNELS;
+    offset += size;
+
+    const auto& wave_buffer{info.wave_buffer[wave_index]};
+    if (offset == samples.size()) {
+        offset = 0;
+
+        if (!wave_buffer.is_looping) {
+            SetWaveIndex(wave_index + 1);
+        }
+
+        out_status.wave_buffer_consumed++;
+
+        if (wave_buffer.end_of_stream) {
+            info.play_state = PlayState::Paused;
+        }
+    }
+
+    return {samples.begin() + dequeue_offset, samples.begin() + dequeue_offset + size};
+}
+
+void AudioRenderer::VoiceState::UpdateState() {
+    if (is_in_use && !info.is_in_use) {
+        // No longer in use, reset state
+        is_refresh_pending = true;
+        wave_index = 0;
+        offset = 0;
+        out_status = {};
+    }
+    is_in_use = info.is_in_use;
+}
+
+void AudioRenderer::VoiceState::RefreshBuffer() {
+    std::vector<s16> new_samples(info.wave_buffer[wave_index].buffer_sz / sizeof(s16));
+    Memory::ReadBlock(info.wave_buffer[wave_index].buffer_addr, new_samples.data(),
+                      info.wave_buffer[wave_index].buffer_sz);
+
+    switch (static_cast<Codec::PcmFormat>(info.sample_format)) {
+    case Codec::PcmFormat::Int16: {
+        // PCM16 is played as-is
+        break;
+    }
+    case Codec::PcmFormat::Adpcm: {
+        // Decode ADPCM to PCM16
+        Codec::ADPCM_Coeff coeffs;
+        Memory::ReadBlock(info.additional_params_addr, coeffs.data(), sizeof(Codec::ADPCM_Coeff));
+        new_samples = Codec::DecodeADPCM(reinterpret_cast<u8*>(new_samples.data()),
+                                         new_samples.size() * sizeof(s16), coeffs, adpcm_state);
+        break;
+    }
+    default:
+        LOG_CRITICAL(Audio, "Unimplemented sample_format={}", info.sample_format);
+        UNREACHABLE();
+        break;
+    }
+
+    switch (info.channel_count) {
+    case 1:
+        // 1 channel is upsampled to 2 channel
+        samples.resize(new_samples.size() * 2);
+        for (size_t index = 0; index < new_samples.size(); ++index) {
+            samples[index * 2] = new_samples[index];
+            samples[index * 2 + 1] = new_samples[index];
+        }
+        break;
+    case 2: {
+        // 2 channel is played as is
+        samples = std::move(new_samples);
+        break;
+    }
+    default:
+        LOG_CRITICAL(Audio, "Unimplemented channel_count={}", info.channel_count);
+        UNREACHABLE();
+        break;
+    }
+
+    is_refresh_pending = false;
+}
+
+static constexpr s16 ClampToS16(s32 value) {
+    return static_cast<s16>(std::clamp(value, -32768, 32767));
+}
+
+void AudioRenderer::QueueMixedBuffer(Buffer::Tag tag) {
+    constexpr size_t BUFFER_SIZE{512};
+    std::vector<s16> buffer(BUFFER_SIZE * stream->GetNumChannels());
+
+    for (auto& voice : voices) {
+        if (!voice.IsPlaying()) {
+            continue;
+        }
+
+        size_t offset{};
+        s64 samples_remaining{BUFFER_SIZE};
+        while (samples_remaining > 0) {
+            const std::vector<s16> samples{voice.DequeueSamples(samples_remaining)};
+
+            if (samples.empty()) {
+                break;
+            }
+
+            samples_remaining -= samples.size();
+
+            for (const auto& sample : samples) {
+                const s32 buffer_sample{buffer[offset]};
+                buffer[offset++] =
+                    ClampToS16(buffer_sample + static_cast<s32>(sample * voice.GetInfo().volume));
+            }
+        }
+    }
+    audio_core->QueueBuffer(stream, tag, std::move(buffer));
+}
+
+void AudioRenderer::ReleaseAndQueueBuffers() {
+    const auto released_buffers{audio_core->GetTagsAndReleaseBuffers(stream, 2)};
+    for (const auto& tag : released_buffers) {
+        QueueMixedBuffer(tag);
+    }
+}
+
+} // namespace AudioCore
diff --git a/src/audio_core/audio_renderer.h b/src/audio_core/audio_renderer.h
new file mode 100644
index 000000000..6950a4681
--- /dev/null
+++ b/src/audio_core/audio_renderer.h
@@ -0,0 +1,206 @@
+// Copyright 2018 yuzu Emulator Project
+// Licensed under GPLv2 or any later version
+// Refer to the license.txt file included.
+
+#pragma once
+
+#include <array>
+#include <memory>
+#include <vector>
+
+#include "audio_core/audio_out.h"
+#include "audio_core/codec.h"
+#include "audio_core/stream.h"
+#include "common/common_types.h"
+#include "common/swap.h"
+#include "core/hle/kernel/event.h"
+
+namespace AudioCore {
+
+enum class PlayState : u8 {
+    Started = 0,
+    Stopped = 1,
+    Paused = 2,
+};
+
+struct AudioRendererParameter {
+    u32_le sample_rate;
+    u32_le sample_count;
+    u32_le unknown_8;
+    u32_le unknown_c;
+    u32_le voice_count;
+    u32_le sink_count;
+    u32_le effect_count;
+    u32_le unknown_1c;
+    u8 unknown_20;
+    INSERT_PADDING_BYTES(3);
+    u32_le splitter_count;
+    u32_le unknown_2c;
+    INSERT_PADDING_WORDS(1);
+    u32_le revision;
+};
+static_assert(sizeof(AudioRendererParameter) == 52, "AudioRendererParameter is an invalid size");
+
+enum class MemoryPoolStates : u32 { // Should be LE
+    Invalid = 0x0,
+    Unknown = 0x1,
+    RequestDetach = 0x2,
+    Detached = 0x3,
+    RequestAttach = 0x4,
+    Attached = 0x5,
+    Released = 0x6,
+};
+
+struct MemoryPoolEntry {
+    MemoryPoolStates state;
+    u32_le unknown_4;
+    u32_le unknown_8;
+    u32_le unknown_c;
+};
+static_assert(sizeof(MemoryPoolEntry) == 0x10, "MemoryPoolEntry has wrong size");
+
+struct MemoryPoolInfo {
+    u64_le pool_address;
+    u64_le pool_size;
+    MemoryPoolStates pool_state;
+    INSERT_PADDING_WORDS(3); // Unknown
+};
+static_assert(sizeof(MemoryPoolInfo) == 0x20, "MemoryPoolInfo has wrong size");
+struct BiquadFilter {
+    u8 enable;
+    INSERT_PADDING_BYTES(1);
+    std::array<s16_le, 3> numerator;
+    std::array<s16_le, 2> denominator;
+};
+static_assert(sizeof(BiquadFilter) == 0xc, "BiquadFilter has wrong size");
+
+struct WaveBuffer {
+    u64_le buffer_addr;
+    u64_le buffer_sz;
+    s32_le start_sample_offset;
+    s32_le end_sample_offset;
+    u8 is_looping;
+    u8 end_of_stream;
+    u8 sent_to_server;
+    INSERT_PADDING_BYTES(5);
+    u64 context_addr;
+    u64 context_sz;
+    INSERT_PADDING_BYTES(8);
+};
+static_assert(sizeof(WaveBuffer) == 0x38, "WaveBuffer has wrong size");
+
+struct VoiceInfo {
+    u32_le id;
+    u32_le node_id;
+    u8 is_new;
+    u8 is_in_use;
+    PlayState play_state;
+    u8 sample_format;
+    u32_le sample_rate;
+    u32_le priority;
+    u32_le sorting_order;
+    u32_le channel_count;
+    float_le pitch;
+    float_le volume;
+    std::array<BiquadFilter, 2> biquad_filter;
+    u32_le wave_buffer_count;
+    u32_le wave_buffer_head;
+    INSERT_PADDING_WORDS(1);
+    u64_le additional_params_addr;
+    u64_le additional_params_sz;
+    u32_le mix_id;
+    u32_le splitter_info_id;
+    std::array<WaveBuffer, 4> wave_buffer;
+    std::array<u32_le, 6> voice_channel_resource_ids;
+    INSERT_PADDING_BYTES(24);
+};
+static_assert(sizeof(VoiceInfo) == 0x170, "VoiceInfo is wrong size");
+
+struct VoiceOutStatus {
+    u64_le played_sample_count;
+    u32_le wave_buffer_consumed;
+    u32_le voice_drops_count;
+};
+static_assert(sizeof(VoiceOutStatus) == 0x10, "VoiceOutStatus has wrong size");
+
+struct UpdateDataHeader {
+    UpdateDataHeader() {}
+
+    explicit UpdateDataHeader(const AudioRendererParameter& config) {
+        revision = Common::MakeMagic('R', 'E', 'V', '4'); // 5.1.0 Revision
+        behavior_size = 0xb0;
+        memory_pools_size = (config.effect_count + (config.voice_count * 4)) * 0x10;
+        voices_size = config.voice_count * 0x10;
+        voice_resource_size = 0x0;
+        effects_size = config.effect_count * 0x10;
+        mixes_size = 0x0;
+        sinks_size = config.sink_count * 0x20;
+        performance_manager_size = 0x10;
+        total_size = sizeof(UpdateDataHeader) + behavior_size + memory_pools_size + voices_size +
+                     effects_size + sinks_size + performance_manager_size;
+    }
+
+    u32_le revision;
+    u32_le behavior_size;
+    u32_le memory_pools_size;
+    u32_le voices_size;
+    u32_le voice_resource_size;
+    u32_le effects_size;
+    u32_le mixes_size;
+    u32_le sinks_size;
+    u32_le performance_manager_size;
+    INSERT_PADDING_WORDS(6);
+    u32_le total_size;
+};
+static_assert(sizeof(UpdateDataHeader) == 0x40, "UpdateDataHeader has wrong size");
+
+class AudioRenderer {
+public:
+    AudioRenderer(AudioRendererParameter params, Kernel::SharedPtr<Kernel::Event> buffer_event);
+    std::vector<u8> UpdateAudioRenderer(const std::vector<u8>& input_params);
+    void QueueMixedBuffer(Buffer::Tag tag);
+    void ReleaseAndQueueBuffers();
+
+private:
+    class VoiceState {
+    public:
+        bool IsPlaying() const {
+            return is_in_use && info.play_state == PlayState::Started;
+        }
+
+        const VoiceOutStatus& GetOutStatus() const {
+            return out_status;
+        }
+
+        const VoiceInfo& GetInfo() const {
+            return info;
+        }
+
+        VoiceInfo& Info() {
+            return info;
+        }
+
+        void SetWaveIndex(size_t index);
+        std::vector<s16> DequeueSamples(size_t sample_count);
+        void UpdateState();
+        void RefreshBuffer();
+
+    private:
+        bool is_in_use{};
+        bool is_refresh_pending{};
+        size_t wave_index{};
+        size_t offset{};
+        Codec::ADPCMState adpcm_state{};
+        std::vector<s16> samples;
+        VoiceOutStatus out_status{};
+        VoiceInfo info{};
+    };
+
+    AudioRendererParameter worker_params;
+    Kernel::SharedPtr<Kernel::Event> buffer_event;
+    std::vector<VoiceState> voices;
+    std::unique_ptr<AudioCore::AudioOut> audio_core;
+    AudioCore::StreamPtr stream;
+};
+
+} // namespace AudioCore
diff --git a/src/audio_core/buffer.h b/src/audio_core/buffer.h
new file mode 100644
index 000000000..a323b23ec
--- /dev/null
+++ b/src/audio_core/buffer.h
@@ -0,0 +1,45 @@
+// Copyright 2018 yuzu Emulator Project
+// Licensed under GPLv2 or any later version
+// Refer to the license.txt file included.
+
+#pragma once
+
+#include <memory>
+#include <vector>
+
+#include "common/common_types.h"
+
+namespace AudioCore {
+
+/**
+ * Represents a buffer of audio samples to be played in an audio stream
+ */
+class Buffer {
+public:
+    using Tag = u64;
+
+    Buffer(Tag tag, std::vector<s16>&& samples) : tag{tag}, samples{std::move(samples)} {}
+
+    /// Returns the raw audio data for the buffer
+    std::vector<s16>& Samples() {
+        return samples;
+    }
+
+    /// Returns the raw audio data for the buffer
+    const std::vector<s16>& GetSamples() const {
+        return samples;
+    }
+
+    /// Returns the buffer tag, this is provided by the game to the audout service
+    Tag GetTag() const {
+        return tag;
+    }
+
+private:
+    Tag tag;
+    std::vector<s16> samples;
+};
+
+using BufferPtr = std::shared_ptr<Buffer>;
+
+} // namespace AudioCore
diff --git a/src/audio_core/codec.cpp b/src/audio_core/codec.cpp
new file mode 100644
index 000000000..c3021403f
--- /dev/null
+++ b/src/audio_core/codec.cpp
@@ -0,0 +1,77 @@
+// Copyright 2018 yuzu Emulator Project
+// Licensed under GPLv2 or any later version
+// Refer to the license.txt file included.
+
+#include <algorithm>
+
+#include "audio_core/codec.h"
+
+namespace AudioCore::Codec {
+
+std::vector<s16> DecodeADPCM(const u8* const data, size_t size, const ADPCM_Coeff& coeff,
+                             ADPCMState& state) {
+    // GC-ADPCM with scale factor and variable coefficients.
+    // Frames are 8 bytes long containing 14 samples each.
+    // Samples are 4 bits (one nibble) long.
+
+    constexpr size_t FRAME_LEN = 8;
+    constexpr size_t SAMPLES_PER_FRAME = 14;
+    constexpr std::array<int, 16> SIGNED_NIBBLES = {
+        {0, 1, 2, 3, 4, 5, 6, 7, -8, -7, -6, -5, -4, -3, -2, -1}};
+
+    const size_t sample_count = (size / FRAME_LEN) * SAMPLES_PER_FRAME;
+    const size_t ret_size =
+        sample_count % 2 == 0 ? sample_count : sample_count + 1; // Ensure multiple of two.
+    std::vector<s16> ret(ret_size);
+
+    int yn1 = state.yn1, yn2 = state.yn2;
+
+    const size_t NUM_FRAMES =
+        (sample_count + (SAMPLES_PER_FRAME - 1)) / SAMPLES_PER_FRAME; // Round up.
+    for (size_t framei = 0; framei < NUM_FRAMES; framei++) {
+        const int frame_header = data[framei * FRAME_LEN];
+        const int scale = 1 << (frame_header & 0xF);
+        const int idx = (frame_header >> 4) & 0x7;
+
+        // Coefficients are fixed point with 11 bits fractional part.
+        const int coef1 = coeff[idx * 2 + 0];
+        const int coef2 = coeff[idx * 2 + 1];
+
+        // Decodes an audio sample. One nibble produces one sample.
+        const auto decode_sample = [&](const int nibble) -> s16 {
+            const int xn = nibble * scale;
+            // We first transform everything into 11 bit fixed point, perform the second order
+            // digital filter, then transform back.
+            // 0x400 == 0.5 in 11 bit fixed point.
+            // Filter: y[n] = x[n] + 0.5 + c1 * y[n-1] + c2 * y[n-2]
+            int val = ((xn << 11) + 0x400 + coef1 * yn1 + coef2 * yn2) >> 11;
+            // Clamp to output range.
+            val = std::clamp<s32>(val, -32768, 32767);
+            // Advance output feedback.
+            yn2 = yn1;
+            yn1 = val;
+            return static_cast<s16>(val);
+        };
+
+        size_t outputi = framei * SAMPLES_PER_FRAME;
+        size_t datai = framei * FRAME_LEN + 1;
+        for (size_t i = 0; i < SAMPLES_PER_FRAME && outputi < sample_count; i += 2) {
+            const s16 sample1 = decode_sample(SIGNED_NIBBLES[data[datai] >> 4]);
+            ret[outputi] = sample1;
+            outputi++;
+
+            const s16 sample2 = decode_sample(SIGNED_NIBBLES[data[datai] & 0xF]);
+            ret[outputi] = sample2;
+            outputi++;
+
+            datai++;
+        }
+    }
+
+    state.yn1 = yn1;
+    state.yn2 = yn2;
+
+    return ret;
+}
+
+} // namespace AudioCore::Codec
diff --git a/src/audio_core/codec.h b/src/audio_core/codec.h
new file mode 100644
index 000000000..3f845c42c
--- /dev/null
+++ b/src/audio_core/codec.h
@@ -0,0 +1,44 @@
+// 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::Codec {
+
+enum class PcmFormat : u32 {
+    Invalid = 0,
+    Int8 = 1,
+    Int16 = 2,
+    Int24 = 3,
+    Int32 = 4,
+    PcmFloat = 5,
+    Adpcm = 6,
+};
+
+/// See: Codec::DecodeADPCM
+struct ADPCMState {
+    // Two historical samples from previous processed buffer,
+    // required for ADPCM decoding
+    s16 yn1; ///< y[n-1]
+    s16 yn2; ///< y[n-2]
+};
+
+using ADPCM_Coeff = std::array<s16, 16>;
+
+/**
+ * @param data Pointer to buffer that contains ADPCM data to decode
+ * @param size Size of buffer in bytes
+ * @param coeff ADPCM coefficients
+ * @param state ADPCM state, this is updated with new state
+ * @return Decoded stereo signed PCM16 data, sample_count in length
+ */
+std::vector<s16> DecodeADPCM(const u8* const data, size_t size, const ADPCM_Coeff& coeff,
+                             ADPCMState& state);
+
+}; // namespace AudioCore::Codec
diff --git a/src/audio_core/cubeb_sink.cpp b/src/audio_core/cubeb_sink.cpp
new file mode 100644
index 000000000..1501ef1f4
--- /dev/null
+++ b/src/audio_core/cubeb_sink.cpp
@@ -0,0 +1,200 @@
+// Copyright 2018 yuzu Emulator Project
+// Licensed under GPLv2 or any later version
+// Refer to the license.txt file included.
+
+#include <algorithm>
+#include <cstring>
+
+#include "audio_core/cubeb_sink.h"
+#include "audio_core/stream.h"
+#include "common/logging/log.h"
+
+namespace AudioCore {
+
+class SinkStreamImpl final : public SinkStream {
+public:
+    SinkStreamImpl(cubeb* ctx, u32 sample_rate, u32 num_channels_, cubeb_devid output_device,
+                   const std::string& name)
+        : ctx{ctx}, num_channels{num_channels_} {
+
+        if (num_channels == 6) {
+            // 6-channel audio does not seem to work with cubeb + SDL, so we downsample this to 2
+            // channel for now
+            is_6_channel = true;
+            num_channels = 2;
+        }
+
+        cubeb_stream_params params{};
+        params.rate = sample_rate;
+        params.channels = num_channels;
+        params.format = CUBEB_SAMPLE_S16NE;
+        params.layout = num_channels == 1 ? CUBEB_LAYOUT_MONO : CUBEB_LAYOUT_STEREO;
+
+        u32 minimum_latency{};
+        if (cubeb_get_min_latency(ctx, &params, &minimum_latency) != CUBEB_OK) {
+            LOG_CRITICAL(Audio_Sink, "Error getting minimum latency");
+        }
+
+        if (cubeb_stream_init(ctx, &stream_backend, name.c_str(), nullptr, nullptr, output_device,
+                              &params, std::max(512u, minimum_latency),
+                              &SinkStreamImpl::DataCallback, &SinkStreamImpl::StateCallback,
+                              this) != CUBEB_OK) {
+            LOG_CRITICAL(Audio_Sink, "Error initializing cubeb stream");
+            return;
+        }
+
+        if (cubeb_stream_start(stream_backend) != CUBEB_OK) {
+            LOG_CRITICAL(Audio_Sink, "Error starting cubeb stream");
+            return;
+        }
+    }
+
+    ~SinkStreamImpl() {
+        if (!ctx) {
+            return;
+        }
+
+        if (cubeb_stream_stop(stream_backend) != CUBEB_OK) {
+            LOG_CRITICAL(Audio_Sink, "Error stopping cubeb stream");
+        }
+
+        cubeb_stream_destroy(stream_backend);
+    }
+
+    void EnqueueSamples(u32 num_channels, const std::vector<s16>& samples) override {
+        if (!ctx) {
+            return;
+        }
+
+        queue.reserve(queue.size() + samples.size() * GetNumChannels());
+
+        if (is_6_channel) {
+            // Downsample 6 channels to 2
+            const size_t sample_count_copy_size = samples.size() * 2;
+            queue.reserve(sample_count_copy_size);
+            for (size_t i = 0; i < samples.size(); i += num_channels) {
+                queue.push_back(samples[i]);
+                queue.push_back(samples[i + 1]);
+            }
+        } else {
+            // Copy as-is
+            std::copy(samples.begin(), samples.end(), std::back_inserter(queue));
+        }
+    }
+
+    u32 GetNumChannels() const {
+        return num_channels;
+    }
+
+private:
+    std::vector<std::string> device_list;
+
+    cubeb* ctx{};
+    cubeb_stream* stream_backend{};
+    u32 num_channels{};
+    bool is_6_channel{};
+
+    std::vector<s16> queue;
+
+    static long DataCallback(cubeb_stream* stream, void* user_data, const void* input_buffer,
+                             void* output_buffer, long num_frames);
+    static void StateCallback(cubeb_stream* stream, void* user_data, cubeb_state state);
+};
+
+CubebSink::CubebSink(std::string target_device_name) {
+    if (cubeb_init(&ctx, "yuzu", nullptr) != CUBEB_OK) {
+        LOG_CRITICAL(Audio_Sink, "cubeb_init failed");
+        return;
+    }
+
+    if (target_device_name != auto_device_name && !target_device_name.empty()) {
+        cubeb_device_collection collection;
+        if (cubeb_enumerate_devices(ctx, CUBEB_DEVICE_TYPE_OUTPUT, &collection) != CUBEB_OK) {
+            LOG_WARNING(Audio_Sink, "Audio output device enumeration not supported");
+        } else {
+            const auto collection_end{collection.device + collection.count};
+            const auto device{std::find_if(collection.device, collection_end,
+                                           [&](const cubeb_device_info& device) {
+                                               return target_device_name == device.friendly_name;
+                                           })};
+            if (device != collection_end) {
+                output_device = device->devid;
+            }
+            cubeb_device_collection_destroy(ctx, &collection);
+        }
+    }
+}
+
+CubebSink::~CubebSink() {
+    if (!ctx) {
+        return;
+    }
+
+    for (auto& sink_stream : sink_streams) {
+        sink_stream.reset();
+    }
+
+    cubeb_destroy(ctx);
+}
+
+SinkStream& CubebSink::AcquireSinkStream(u32 sample_rate, u32 num_channels,
+                                         const std::string& name) {
+    sink_streams.push_back(
+        std::make_unique<SinkStreamImpl>(ctx, sample_rate, num_channels, output_device, name));
+    return *sink_streams.back();
+}
+
+long SinkStreamImpl::DataCallback(cubeb_stream* stream, void* user_data, const void* input_buffer,
+                                  void* output_buffer, long num_frames) {
+    SinkStreamImpl* impl = static_cast<SinkStreamImpl*>(user_data);
+    u8* buffer = reinterpret_cast<u8*>(output_buffer);
+
+    if (!impl) {
+        return {};
+    }
+
+    const size_t frames_to_write{
+        std::min(impl->queue.size() / impl->GetNumChannels(), static_cast<size_t>(num_frames))};
+
+    memcpy(buffer, impl->queue.data(), frames_to_write * sizeof(s16) * impl->GetNumChannels());
+    impl->queue.erase(impl->queue.begin(),
+                      impl->queue.begin() + frames_to_write * impl->GetNumChannels());
+
+    if (frames_to_write < num_frames) {
+        // Fill the rest of the frames with silence
+        memset(buffer + frames_to_write * sizeof(s16) * impl->GetNumChannels(), 0,
+               (num_frames - frames_to_write) * sizeof(s16) * impl->GetNumChannels());
+    }
+
+    return num_frames;
+}
+
+void SinkStreamImpl::StateCallback(cubeb_stream* stream, void* user_data, cubeb_state state) {}
+
+std::vector<std::string> ListCubebSinkDevices() {
+    std::vector<std::string> device_list;
+    cubeb* ctx;
+
+    if (cubeb_init(&ctx, "Citra Device Enumerator", nullptr) != CUBEB_OK) {
+        LOG_CRITICAL(Audio_Sink, "cubeb_init failed");
+        return {};
+    }
+
+    cubeb_device_collection collection;
+    if (cubeb_enumerate_devices(ctx, CUBEB_DEVICE_TYPE_OUTPUT, &collection) != CUBEB_OK) {
+        LOG_WARNING(Audio_Sink, "Audio output device enumeration not supported");
+    } else {
+        for (size_t i = 0; i < collection.count; i++) {
+            const cubeb_device_info& device = collection.device[i];
+            if (device.friendly_name) {
+                device_list.emplace_back(device.friendly_name);
+            }
+        }
+        cubeb_device_collection_destroy(ctx, &collection);
+    }
+
+    cubeb_destroy(ctx);
+    return device_list;
+}
+
+} // namespace AudioCore
diff --git a/src/audio_core/cubeb_sink.h b/src/audio_core/cubeb_sink.h
new file mode 100644
index 000000000..59cbf05e9
--- /dev/null
+++ b/src/audio_core/cubeb_sink.h
@@ -0,0 +1,32 @@
+// Copyright 2018 yuzu Emulator Project
+// Licensed under GPLv2 or any later version
+// Refer to the license.txt file included.
+
+#pragma once
+
+#include <string>
+#include <vector>
+
+#include <cubeb/cubeb.h>
+
+#include "audio_core/sink.h"
+
+namespace AudioCore {
+
+class CubebSink final : public Sink {
+public:
+    explicit CubebSink(std::string device_id);
+    ~CubebSink() override;
+
+    SinkStream& AcquireSinkStream(u32 sample_rate, u32 num_channels,
+                                  const std::string& name) override;
+
+private:
+    cubeb* ctx{};
+    cubeb_devid output_device{};
+    std::vector<SinkStreamPtr> sink_streams;
+};
+
+std::vector<std::string> ListCubebSinkDevices();
+
+} // namespace AudioCore
diff --git a/src/audio_core/null_sink.h b/src/audio_core/null_sink.h
new file mode 100644
index 000000000..f235d93e5
--- /dev/null
+++ b/src/audio_core/null_sink.h
@@ -0,0 +1,27 @@
+// Copyright 2018 yuzu Emulator Project
+// Licensed under GPLv2 or any later version
+// Refer to the license.txt file included.
+
+#pragma once
+
+#include "audio_core/sink.h"
+
+namespace AudioCore {
+
+class NullSink final : public Sink {
+public:
+    explicit NullSink(std::string){};
+    ~NullSink() override = default;
+
+    SinkStream& AcquireSinkStream(u32 /*sample_rate*/, u32 /*num_channels*/,
+                                  const std::string& /*name*/) override {
+        return null_sink_stream;
+    }
+
+private:
+    struct NullSinkStreamImpl final : SinkStream {
+        void EnqueueSamples(u32 /*num_channels*/, const std::vector<s16>& /*samples*/) override {}
+    } null_sink_stream;
+};
+
+} // namespace AudioCore
diff --git a/src/audio_core/sink.h b/src/audio_core/sink.h
new file mode 100644
index 000000000..95c7b2b6e
--- /dev/null
+++ b/src/audio_core/sink.h
@@ -0,0 +1,31 @@
+// Copyright 2018 yuzu Emulator Project
+// Licensed under GPLv2 or any later version
+// Refer to the license.txt file included.
+
+#pragma once
+
+#include <memory>
+#include <string>
+
+#include "audio_core/sink_stream.h"
+#include "common/common_types.h"
+
+namespace AudioCore {
+
+constexpr char auto_device_name[] = "auto";
+
+/**
+ * This class is an interface for an audio sink. An audio sink accepts samples in stereo signed
+ * PCM16 format to be output. Sinks *do not* handle resampling and expect the correct sample rate.
+ * They are dumb outputs.
+ */
+class Sink {
+public:
+    virtual ~Sink() = default;
+    virtual SinkStream& AcquireSinkStream(u32 sample_rate, u32 num_channels,
+                                          const std::string& name) = 0;
+};
+
+using SinkPtr = std::unique_ptr<Sink>;
+
+} // namespace AudioCore
diff --git a/src/audio_core/sink_details.cpp b/src/audio_core/sink_details.cpp
new file mode 100644
index 000000000..955ba20fb
--- /dev/null
+++ b/src/audio_core/sink_details.cpp
@@ -0,0 +1,44 @@
+// Copyright 2018 yuzu Emulator Project
+// Licensed under GPLv2 or any later version
+// Refer to the license.txt file included.
+
+#include <algorithm>
+#include <memory>
+#include <string>
+#include <vector>
+#include "audio_core/null_sink.h"
+#include "audio_core/sink_details.h"
+#ifdef HAVE_CUBEB
+#include "audio_core/cubeb_sink.h"
+#endif
+#include "common/logging/log.h"
+
+namespace AudioCore {
+
+// g_sink_details is ordered in terms of desirability, with the best choice at the top.
+const std::vector<SinkDetails> g_sink_details = {
+#ifdef HAVE_CUBEB
+    SinkDetails{"cubeb", &std::make_unique<CubebSink, std::string>, &ListCubebSinkDevices},
+#endif
+    SinkDetails{"null", &std::make_unique<NullSink, std::string>,
+                [] { return std::vector<std::string>{"null"}; }},
+};
+
+const SinkDetails& GetSinkDetails(std::string sink_id) {
+    auto iter =
+        std::find_if(g_sink_details.begin(), g_sink_details.end(),
+                     [sink_id](const auto& sink_detail) { return sink_detail.id == sink_id; });
+
+    if (sink_id == "auto" || iter == g_sink_details.end()) {
+        if (sink_id != "auto") {
+            LOG_ERROR(Audio, "AudioCore::SelectSink given invalid sink_id {}", sink_id);
+        }
+        // Auto-select.
+        // g_sink_details is ordered in terms of desirability, with the best choice at the front.
+        iter = g_sink_details.begin();
+    }
+
+    return *iter;
+}
+
+} // namespace AudioCore
diff --git a/src/audio_core/sink_details.h b/src/audio_core/sink_details.h
new file mode 100644
index 000000000..ea666c554
--- /dev/null
+++ b/src/audio_core/sink_details.h
@@ -0,0 +1,35 @@
+// Copyright 2018 yuzu Emulator Project
+// Licensed under GPLv2 or any later version
+// Refer to the license.txt file included.
+
+#pragma once
+
+#include <functional>
+#include <memory>
+#include <utility>
+#include <vector>
+
+namespace AudioCore {
+
+class Sink;
+
+struct SinkDetails {
+    using FactoryFn = std::function<std::unique_ptr<Sink>(std::string)>;
+    using ListDevicesFn = std::function<std::vector<std::string>()>;
+
+    SinkDetails(const char* id_, FactoryFn factory_, ListDevicesFn list_devices_)
+        : id(id_), factory(std::move(factory_)), list_devices(std::move(list_devices_)) {}
+
+    /// Name for this sink.
+    const char* id;
+    /// A method to call to construct an instance of this type of sink.
+    FactoryFn factory;
+    /// A method to call to list available devices.
+    ListDevicesFn list_devices;
+};
+
+extern const std::vector<SinkDetails> g_sink_details;
+
+const SinkDetails& GetSinkDetails(std::string sink_id);
+
+} // namespace AudioCore
diff --git a/src/audio_core/sink_stream.h b/src/audio_core/sink_stream.h
new file mode 100644
index 000000000..41b6736d8
--- /dev/null
+++ b/src/audio_core/sink_stream.h
@@ -0,0 +1,32 @@
+// Copyright 2018 yuzu Emulator Project
+// Licensed under GPLv2 or any later version
+// Refer to the license.txt file included.
+
+#pragma once
+
+#include <memory>
+#include <vector>
+
+#include "common/common_types.h"
+
+namespace AudioCore {
+
+/**
+ * Accepts samples in stereo signed PCM16 format to be output. Sinks *do not* handle resampling and
+ * expect the correct sample rate. They are dumb outputs.
+ */
+class SinkStream {
+public:
+    virtual ~SinkStream() = default;
+
+    /**
+     * Feed stereo samples to sink.
+     * @param num_channels Number of channels used.
+     * @param samples Samples in interleaved stereo PCM16 format.
+     */
+    virtual void EnqueueSamples(u32 num_channels, const std::vector<s16>& samples) = 0;
+};
+
+using SinkStreamPtr = std::unique_ptr<SinkStream>;
+
+} // namespace AudioCore
diff --git a/src/audio_core/stream.cpp b/src/audio_core/stream.cpp
new file mode 100644
index 000000000..ad9e2915c
--- /dev/null
+++ b/src/audio_core/stream.cpp
@@ -0,0 +1,127 @@
+// Copyright 2018 yuzu Emulator Project
+// Licensed under GPLv2 or any later version
+// Refer to the license.txt file included.
+
+#include <algorithm>
+#include <cmath>
+
+#include "audio_core/sink.h"
+#include "audio_core/sink_details.h"
+#include "audio_core/stream.h"
+#include "common/assert.h"
+#include "common/logging/log.h"
+#include "core/core_timing.h"
+#include "core/core_timing_util.h"
+#include "core/settings.h"
+
+namespace AudioCore {
+
+constexpr size_t MaxAudioBufferCount{32};
+
+u32 Stream::GetNumChannels() const {
+    switch (format) {
+    case Format::Mono16:
+        return 1;
+    case Format::Stereo16:
+        return 2;
+    case Format::Multi51Channel16:
+        return 6;
+    }
+    LOG_CRITICAL(Audio, "Unimplemented format={}", static_cast<u32>(format));
+    UNREACHABLE();
+    return {};
+}
+
+Stream::Stream(u32 sample_rate, Format format, ReleaseCallback&& release_callback,
+               SinkStream& sink_stream, std::string&& name_)
+    : sample_rate{sample_rate}, format{format}, release_callback{std::move(release_callback)},
+      sink_stream{sink_stream}, name{std::move(name_)} {
+
+    release_event = CoreTiming::RegisterEvent(
+        name, [this](u64 userdata, int cycles_late) { ReleaseActiveBuffer(); });
+}
+
+void Stream::Play() {
+    state = State::Playing;
+    PlayNextBuffer();
+}
+
+void Stream::Stop() {
+    ASSERT_MSG(false, "Unimplemented");
+}
+
+s64 Stream::GetBufferReleaseCycles(const Buffer& buffer) const {
+    const size_t num_samples{buffer.GetSamples().size() / GetNumChannels()};
+    return CoreTiming::usToCycles((static_cast<u64>(num_samples) * 1000000) / sample_rate);
+}
+
+static void VolumeAdjustSamples(std::vector<s16>& samples) {
+    const float volume{std::clamp(Settings::values.volume, 0.0f, 1.0f)};
+
+    if (volume == 1.0f) {
+        return;
+    }
+
+    // Implementation of a volume slider with a dynamic range of 60 dB
+    const float volume_scale_factor{std::exp(6.90775f * volume) * 0.001f};
+    for (auto& sample : samples) {
+        sample = static_cast<s16>(sample * volume_scale_factor);
+    }
+}
+
+void Stream::PlayNextBuffer() {
+    if (!IsPlaying()) {
+        // Ensure we are in playing state before playing the next buffer
+        return;
+    }
+
+    if (active_buffer) {
+        // Do not queue a new buffer if we are already playing a buffer
+        return;
+    }
+
+    if (queued_buffers.empty()) {
+        // No queued buffers - we are effectively paused
+        return;
+    }
+
+    active_buffer = queued_buffers.front();
+    queued_buffers.pop();
+
+    VolumeAdjustSamples(active_buffer->Samples());
+    sink_stream.EnqueueSamples(GetNumChannels(), active_buffer->GetSamples());
+
+    CoreTiming::ScheduleEventThreadsafe(GetBufferReleaseCycles(*active_buffer), release_event, {});
+}
+
+void Stream::ReleaseActiveBuffer() {
+    ASSERT(active_buffer);
+    released_buffers.push(std::move(active_buffer));
+    release_callback();
+    PlayNextBuffer();
+}
+
+bool Stream::QueueBuffer(BufferPtr&& buffer) {
+    if (queued_buffers.size() < MaxAudioBufferCount) {
+        queued_buffers.push(std::move(buffer));
+        PlayNextBuffer();
+        return true;
+    }
+    return false;
+}
+
+bool Stream::ContainsBuffer(Buffer::Tag tag) const {
+    ASSERT_MSG(false, "Unimplemented");
+    return {};
+}
+
+std::vector<Buffer::Tag> Stream::GetTagsAndReleaseBuffers(size_t max_count) {
+    std::vector<Buffer::Tag> tags;
+    for (size_t count = 0; count < max_count && !released_buffers.empty(); ++count) {
+        tags.push_back(released_buffers.front()->GetTag());
+        released_buffers.pop();
+    }
+    return tags;
+}
+
+} // namespace AudioCore
diff --git a/src/audio_core/stream.h b/src/audio_core/stream.h
new file mode 100644
index 000000000..049b92ca9
--- /dev/null
+++ b/src/audio_core/stream.h
@@ -0,0 +1,102 @@
+// Copyright 2018 yuzu Emulator Project
+// Licensed under GPLv2 or any later version
+// Refer to the license.txt file included.
+
+#pragma once
+
+#include <functional>
+#include <memory>
+#include <string>
+#include <vector>
+#include <queue>
+
+#include "audio_core/buffer.h"
+#include "audio_core/sink_stream.h"
+#include "common/assert.h"
+#include "common/common_types.h"
+#include "core/core_timing.h"
+
+namespace AudioCore {
+
+/**
+ * Represents an audio stream, which is a sequence of queued buffers, to be outputed by AudioOut
+ */
+class Stream {
+public:
+    /// Audio format of the stream
+    enum class Format {
+        Mono16,
+        Stereo16,
+        Multi51Channel16,
+    };
+
+    /// Callback function type, used to change guest state on a buffer being released
+    using ReleaseCallback = std::function<void()>;
+
+    Stream(u32 sample_rate, Format format, ReleaseCallback&& release_callback,
+           SinkStream& sink_stream, std::string&& name_);
+
+    /// Plays the audio stream
+    void Play();
+
+    /// Stops the audio stream
+    void Stop();
+
+    /// Queues a buffer into the audio stream, returns true on success
+    bool QueueBuffer(BufferPtr&& buffer);
+
+    /// Returns true if the audio stream contains a buffer with the specified tag
+    bool ContainsBuffer(Buffer::Tag tag) const;
+
+    /// Returns a vector of recently released buffers specified by tag
+    std::vector<Buffer::Tag> GetTagsAndReleaseBuffers(size_t max_count);
+
+    /// Returns true if the stream is currently playing
+    bool IsPlaying() const {
+        return state == State::Playing;
+    }
+
+    /// Returns the number of queued buffers
+    size_t GetQueueSize() const {
+        return queued_buffers.size();
+    }
+
+    /// Gets the sample rate
+    u32 GetSampleRate() const {
+        return sample_rate;
+    }
+
+    /// Gets the number of channels
+    u32 GetNumChannels() const;
+
+private:
+    /// Current state of the stream
+    enum class State {
+        Stopped,
+        Playing,
+    };
+
+    /// Plays the next queued buffer in the audio stream, starting playback if necessary
+    void PlayNextBuffer();
+
+    /// Releases the actively playing buffer, signalling that it has been completed
+    void ReleaseActiveBuffer();
+
+    /// Gets the number of core cycles when the specified buffer will be released
+    s64 GetBufferReleaseCycles(const Buffer& buffer) const;
+
+    u32 sample_rate;                        ///< Sample rate of the stream
+    Format format;                          ///< Format of the stream
+    ReleaseCallback release_callback;       ///< Buffer release callback for the stream
+    State state{State::Stopped};            ///< Playback state of the stream
+    CoreTiming::EventType* release_event{}; ///< Core timing release event for the stream
+    BufferPtr active_buffer;                ///< Actively playing buffer in the stream
+    std::queue<BufferPtr> queued_buffers;   ///< Buffers queued to be played in the stream
+    std::queue<BufferPtr> released_buffers; ///< Buffers recently released from the stream
+    SinkStream& sink_stream;                ///< Output sink for the stream
+    std::string name;                       ///< Name of the stream, must be unique
+};
+
+using StreamPtr = std::shared_ptr<Stream>;
+
+} // namespace AudioCore