11 files changed, 673 insertions, 10 deletions

diff --git a/src/audio_core/CMakeLists.txt b/src/audio_core/CMakeLists.txt
index 5a2747e78..13b5e400e 100644
--- a/src/audio_core/CMakeLists.txt
+++ b/src/audio_core/CMakeLists.txt
@@ -4,6 +4,7 @@ set(SRCS
             hle/dsp.cpp
             hle/filter.cpp
             hle/pipe.cpp
+            hle/source.cpp
             interpolate.cpp
             sink_details.cpp
             )
@@ -15,6 +16,7 @@ set(HEADERS
             hle/dsp.h
             hle/filter.h
             hle/pipe.h
+            hle/source.h
             interpolate.h
             null_sink.h
             sink.h
@@ -23,7 +25,18 @@ set(HEADERS
 
 include_directories(../../externals/soundtouch/include)
 
+if(SDL2_FOUND)
+    set(SRCS ${SRCS} sdl2_sink.cpp)
+    set(HEADERS ${HEADERS} sdl2_sink.h)
+    include_directories(${SDL2_INCLUDE_DIR})
+endif()
+
 create_directory_groups(${SRCS} ${HEADERS})
 
 add_library(audio_core STATIC ${SRCS} ${HEADERS})
 target_link_libraries(audio_core SoundTouch)
+
+if(SDL2_FOUND)
+    target_link_libraries(audio_core ${SDL2_LIBRARY})
+    set_property(TARGET audio_core APPEND PROPERTY COMPILE_DEFINITIONS HAVE_SDL2)
+endif()
diff --git a/src/audio_core/hle/common.h b/src/audio_core/hle/common.h
index 7910f42ae..596b67eaf 100644
--- a/src/audio_core/hle/common.h
+++ b/src/audio_core/hle/common.h
@@ -27,7 +27,7 @@ using QuadFrame32   = std::array<std::array<s32, 4>, samples_per_frame>;
  */
 template<typename FrameT, typename FilterT>
 void FilterFrame(FrameT& frame, FilterT& filter) {
-    std::transform(frame.begin(), frame.end(), frame.begin(), [&filter](const typename FrameT::value_type& sample) {
+    std::transform(frame.begin(), frame.end(), frame.begin(), [&filter](const auto& sample) {
         return filter.ProcessSample(sample);
     });
 }
diff --git a/src/audio_core/hle/dsp.cpp b/src/audio_core/hle/dsp.cpp
index 4d44bd2d9..0cdbdb06a 100644
--- a/src/audio_core/hle/dsp.cpp
+++ b/src/audio_core/hle/dsp.cpp
@@ -2,10 +2,12 @@
 // Licensed under GPLv2 or any later version
 // Refer to the license.txt file included.
 
+#include <array>
 #include <memory>
 
 #include "audio_core/hle/dsp.h"
 #include "audio_core/hle/pipe.h"
+#include "audio_core/hle/source.h"
 #include "audio_core/sink.h"
 
 namespace DSP {
@@ -38,16 +40,38 @@ static SharedMemory& WriteRegion() {
     return g_regions[1 - CurrentRegionIndex()];
 }
 
+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 std::unique_ptr<AudioCore::Sink> sink;
 
 void Init() {
     DSP::HLE::ResetPipes();
+    for (auto& source : sources) {
+        source.Reset();
+    }
 }
 
 void Shutdown() {
 }
 
 bool Tick() {
+    SharedMemory& read = ReadRegion();
+    SharedMemory& write = WriteRegion();
+
+    std::array<QuadFrame32, 3> 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);
+        }
+    }
+
     return true;
 }
 
diff --git a/src/audio_core/hle/dsp.h b/src/audio_core/hle/dsp.h
index 4f2410c27..f6e53f68f 100644
--- a/src/audio_core/hle/dsp.h
+++ b/src/audio_core/hle/dsp.h
@@ -33,13 +33,9 @@ namespace HLE {
 // double-buffer. The frame counter is located as the very last u16 of each region and is incremented
 // each audio tick.
 
-struct SharedMemory;
-
 constexpr VAddr region0_base = 0x1FF50000;
 constexpr VAddr region1_base = 0x1FF70000;
 
-extern std::array<SharedMemory, 2> g_regions;
-
 /**
  * The DSP is native 16-bit. The DSP also appears to be big-endian. When reading 32-bit numbers from
  * its memory regions, the higher and lower 16-bit halves are swapped compared to the little-endian
@@ -169,9 +165,9 @@ struct SourceConfiguration {
         float_le rate_multiplier;
 
         enum class InterpolationMode : u8 {
-            None = 0,
+            Polyphase = 0,
             Linear = 1,
-            Polyphase = 2
+            None = 2
         };
 
         InterpolationMode interpolation_mode;
@@ -318,10 +314,10 @@ ASSERT_DSP_STRUCT(SourceConfiguration::Configuration::Buffer, 20);
 struct SourceStatus {
     struct Status {
         u8 is_enabled;               ///< Is this channel enabled? (Doesn't have to be playing anything.)
-        u8 previous_buffer_id_dirty; ///< Non-zero when previous_buffer_id changes
+        u8 current_buffer_id_dirty;  ///< Non-zero when current_buffer_id changes
         u16_le sync;                 ///< Is set by the DSP to the value of SourceConfiguration::sync
         u32_dsp buffer_position;     ///< Number of samples into the current buffer
-        u16_le previous_buffer_id;   ///< Updated when a buffer finishes playing
+        u16_le current_buffer_id;    ///< Updated when a buffer finishes playing
         INSERT_PADDING_DSPWORDS(1);
     };
 
@@ -507,6 +503,8 @@ struct SharedMemory {
 };
 ASSERT_DSP_STRUCT(SharedMemory, 0x8000);
 
+extern std::array<SharedMemory, 2> g_regions;
+
 // Structures must have an offset that is a multiple of two.
 static_assert(offsetof(SharedMemory, frame_counter) % 2 == 0, "Structures in DSP::HLE::SharedMemory must be 2-byte aligned");
 static_assert(offsetof(SharedMemory, source_configurations) % 2 == 0, "Structures in DSP::HLE::SharedMemory must be 2-byte aligned");
diff --git a/src/audio_core/hle/filter.h b/src/audio_core/hle/filter.h
index 75738f600..43d2035cd 100644
--- a/src/audio_core/hle/filter.h
+++ b/src/audio_core/hle/filter.h
@@ -16,6 +16,7 @@ namespace HLE {
 
 /// Preprocessing filters. There is an independent set of filters for each Source.
 class SourceFilters final {
+public:
     SourceFilters() { Reset(); }
 
     /// Reset internal state.
diff --git a/src/audio_core/hle/source.cpp b/src/audio_core/hle/source.cpp
new file mode 100644
index 000000000..daaf6e3f3
--- /dev/null
+++ b/src/audio_core/hle/source.cpp
@@ -0,0 +1,320 @@
+// Copyright 2016 Citra Emulator Project
+// Licensed under GPLv2 or any later version
+// Refer to the license.txt file included.
+
+#include <algorithm>
+#include <array>
+
+#include "audio_core/codec.h"
+#include "audio_core/hle/common.h"
+#include "audio_core/hle/source.h"
+#include "audio_core/interpolate.h"
+
+#include "common/assert.h"
+#include "common/logging/log.h"
+
+#include "core/memory.h"
+
+namespace DSP {
+namespace HLE {
+
+SourceStatus::Status Source::Tick(SourceConfiguration::Configuration& config, const s16_le (&adpcm_coeffs)[16]) {
+    ParseConfig(config, adpcm_coeffs);
+
+    if (state.enabled) {
+        GenerateFrame();
+    }
+
+    return GetCurrentStatus();
+}
+
+void Source::MixInto(QuadFrame32& dest, size_t intermediate_mix_id) const {
+    if (!state.enabled)
+        return;
+
+    const std::array<float, 4>& gains = state.gain.at(intermediate_mix_id);
+    for (size_t samplei = 0; samplei < samples_per_frame; samplei++) {
+        // Conversion from stereo (current_frame) to quadraphonic (dest) occurs here.
+        dest[samplei][0] += static_cast<s32>(gains[0] * current_frame[samplei][0]);
+        dest[samplei][1] += static_cast<s32>(gains[1] * current_frame[samplei][1]);
+        dest[samplei][2] += static_cast<s32>(gains[2] * current_frame[samplei][0]);
+        dest[samplei][3] += static_cast<s32>(gains[3] * current_frame[samplei][1]);
+    }
+}
+
+void Source::Reset() {
+    current_frame.fill({});
+    state = {};
+}
+
+void Source::ParseConfig(SourceConfiguration::Configuration& config, const s16_le (&adpcm_coeffs)[16]) {
+    if (!config.dirty_raw) {
+        return;
+    }
+
+    if (config.reset_flag) {
+        config.reset_flag.Assign(0);
+        Reset();
+        LOG_TRACE(Audio_DSP, "source_id=%zu reset", source_id);
+    }
+
+    if (config.partial_reset_flag) {
+        config.partial_reset_flag.Assign(0);
+        state.input_queue = std::priority_queue<Buffer, std::vector<Buffer>, BufferOrder>{};
+        LOG_TRACE(Audio_DSP, "source_id=%zu partial_reset", source_id);
+    }
+
+    if (config.enable_dirty) {
+        config.enable_dirty.Assign(0);
+        state.enabled = config.enable != 0;
+        LOG_TRACE(Audio_DSP, "source_id=%zu enable=%d", source_id, state.enabled);
+    }
+
+    if (config.sync_dirty) {
+        config.sync_dirty.Assign(0);
+        state.sync = config.sync;
+        LOG_TRACE(Audio_DSP, "source_id=%zu sync=%u", source_id, state.sync);
+    }
+
+    if (config.rate_multiplier_dirty) {
+        config.rate_multiplier_dirty.Assign(0);
+        state.rate_multiplier = config.rate_multiplier;
+        LOG_TRACE(Audio_DSP, "source_id=%zu rate=%f", source_id, state.rate_multiplier);
+
+        if (state.rate_multiplier <= 0) {
+            LOG_ERROR(Audio_DSP, "Was given an invalid rate multiplier: source_id=%zu rate=%f", source_id, state.rate_multiplier);
+            state.rate_multiplier = 1.0f;
+            // Note: Actual firmware starts producing garbage if this occurs.
+        }
+    }
+
+    if (config.adpcm_coefficients_dirty) {
+        config.adpcm_coefficients_dirty.Assign(0);
+        std::transform(adpcm_coeffs, adpcm_coeffs + state.adpcm_coeffs.size(), state.adpcm_coeffs.begin(),
+            [](const auto& coeff) { return static_cast<s16>(coeff); });
+        LOG_TRACE(Audio_DSP, "source_id=%zu adpcm update", source_id);
+    }
+
+    if (config.gain_0_dirty) {
+        config.gain_0_dirty.Assign(0);
+        std::transform(config.gain[0], config.gain[0] + state.gain[0].size(), state.gain[0].begin(),
+            [](const auto& coeff) { return static_cast<float>(coeff); });
+        LOG_TRACE(Audio_DSP, "source_id=%zu gain 0 update", source_id);
+    }
+
+    if (config.gain_1_dirty) {
+        config.gain_1_dirty.Assign(0);
+        std::transform(config.gain[1], config.gain[1] + state.gain[1].size(), state.gain[1].begin(),
+            [](const auto& coeff) { return static_cast<float>(coeff); });
+        LOG_TRACE(Audio_DSP, "source_id=%zu gain 1 update", source_id);
+    }
+
+    if (config.gain_2_dirty) {
+        config.gain_2_dirty.Assign(0);
+        std::transform(config.gain[2], config.gain[2] + state.gain[2].size(), state.gain[2].begin(),
+            [](const auto& coeff) { return static_cast<float>(coeff); });
+        LOG_TRACE(Audio_DSP, "source_id=%zu gain 2 update", source_id);
+    }
+
+    if (config.filters_enabled_dirty) {
+        config.filters_enabled_dirty.Assign(0);
+        state.filters.Enable(config.simple_filter_enabled.ToBool(), config.biquad_filter_enabled.ToBool());
+        LOG_TRACE(Audio_DSP, "source_id=%zu enable_simple=%hu enable_biquad=%hu",
+                  source_id, config.simple_filter_enabled.Value(), config.biquad_filter_enabled.Value());
+    }
+
+    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");
+    }
+
+    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");
+    }
+
+    if (config.interpolation_dirty) {
+        config.interpolation_dirty.Assign(0);
+        state.interpolation_mode = config.interpolation_mode;
+        LOG_TRACE(Audio_DSP, "source_id=%zu interpolation_mode=%zu", source_id, static_cast<size_t>(state.interpolation_mode));
+    }
+
+    if (config.format_dirty || config.embedded_buffer_dirty) {
+        config.format_dirty.Assign(0);
+        state.format = config.format;
+        LOG_TRACE(Audio_DSP, "source_id=%zu format=%zu", source_id, static_cast<size_t>(state.format));
+    }
+
+    if (config.mono_or_stereo_dirty || config.embedded_buffer_dirty) {
+        config.mono_or_stereo_dirty.Assign(0);
+        state.mono_or_stereo = config.mono_or_stereo;
+        LOG_TRACE(Audio_DSP, "source_id=%zu mono_or_stereo=%zu", source_id, static_cast<size_t>(state.mono_or_stereo));
+    }
+
+    if (config.embedded_buffer_dirty) {
+        config.embedded_buffer_dirty.Assign(0);
+        state.input_queue.emplace(Buffer{
+            config.physical_address,
+            config.length,
+            static_cast<u8>(config.adpcm_ps),
+            { config.adpcm_yn[0], config.adpcm_yn[1] },
+            config.adpcm_dirty.ToBool(),
+            config.is_looping.ToBool(),
+            config.buffer_id,
+            state.mono_or_stereo,
+            state.format,
+            false
+        });
+        LOG_TRACE(Audio_DSP, "enqueuing embedded addr=0x%08x len=%u id=%hu", config.physical_address, config.length, config.buffer_id);
+    }
+
+    if (config.buffer_queue_dirty) {
+        config.buffer_queue_dirty.Assign(0);
+        for (size_t i = 0; i < 4; i++) {
+            if (config.buffers_dirty & (1 << i)) {
+                const auto& b = config.buffers[i];
+                state.input_queue.emplace(Buffer{
+                    b.physical_address,
+                    b.length,
+                    static_cast<u8>(b.adpcm_ps),
+                    { b.adpcm_yn[0], b.adpcm_yn[1] },
+                    b.adpcm_dirty != 0,
+                    b.is_looping != 0,
+                    b.buffer_id,
+                    state.mono_or_stereo,
+                    state.format,
+                    true
+                });
+                LOG_TRACE(Audio_DSP, "enqueuing queued %zu addr=0x%08x len=%u id=%hu", i, b.physical_address, b.length, b.buffer_id);
+            }
+        }
+        config.buffers_dirty = 0;
+    }
+
+    if (config.dirty_raw) {
+        LOG_DEBUG(Audio_DSP, "source_id=%zu remaining_dirty=%x", source_id, config.dirty_raw);
+    }
+
+    config.dirty_raw = 0;
+}
+
+void Source::GenerateFrame() {
+    current_frame.fill({});
+
+    if (state.current_buffer.empty() && !DequeueBuffer()) {
+        state.enabled = false;
+        state.buffer_update = true;
+        state.current_buffer_id = 0;
+        return;
+    }
+
+    size_t frame_position = 0;
+
+    state.current_sample_number = state.next_sample_number;
+    while (frame_position < current_frame.size()) {
+        if (state.current_buffer.empty() && !DequeueBuffer()) {
+            break;
+        }
+
+        const size_t size_to_copy = std::min(state.current_buffer.size(), current_frame.size() - frame_position);
+
+        std::copy(state.current_buffer.begin(), state.current_buffer.begin() + size_to_copy, current_frame.begin() + frame_position);
+        state.current_buffer.erase(state.current_buffer.begin(), state.current_buffer.begin() + size_to_copy);
+
+        frame_position += size_to_copy;
+        state.next_sample_number += static_cast<u32>(size_to_copy);
+    }
+
+    state.filters.ProcessFrame(current_frame);
+}
+
+
+bool Source::DequeueBuffer() {
+    ASSERT_MSG(state.current_buffer.empty(), "Shouldn't dequeue; we still have data in current_buffer");
+
+    if (state.input_queue.empty())
+        return false;
+
+    const Buffer buf = state.input_queue.top();
+    state.input_queue.pop();
+
+    if (buf.adpcm_dirty) {
+        state.adpcm_state.yn1 = buf.adpcm_yn[0];
+        state.adpcm_state.yn2 = buf.adpcm_yn[1];
+    }
+
+    if (buf.is_looping) {
+        LOG_ERROR(Audio_DSP, "Looped buffers are unimplemented at the moment");
+    }
+
+    const u8* const memory = Memory::GetPhysicalPointer(buf.physical_address);
+    if (memory) {
+        const unsigned num_channels = buf.mono_or_stereo == MonoOrStereo::Stereo ? 2 : 1;
+        switch (buf.format) {
+        case Format::PCM8:
+            state.current_buffer = Codec::DecodePCM8(num_channels, memory, buf.length);
+            break;
+        case Format::PCM16:
+            state.current_buffer = Codec::DecodePCM16(num_channels, memory, buf.length);
+            break;
+        case Format::ADPCM:
+            DEBUG_ASSERT(num_channels == 1);
+            state.current_buffer = Codec::DecodeADPCM(memory, buf.length, state.adpcm_coeffs, state.adpcm_state);
+            break;
+        default:
+            UNIMPLEMENTED();
+            break;
+        }
+    } else {
+        LOG_WARNING(Audio_DSP, "source_id=%zu buffer_id=%hu length=%u: Invalid physical address 0x%08X",
+                               source_id, buf.buffer_id, buf.length, buf.physical_address);
+        state.current_buffer.clear();
+        return true;
+    }
+
+    switch (state.interpolation_mode) {
+    case InterpolationMode::None:
+        state.current_buffer = AudioInterp::None(state.interp_state, state.current_buffer, state.rate_multiplier);
+        break;
+    case InterpolationMode::Linear:
+        state.current_buffer = AudioInterp::Linear(state.interp_state, state.current_buffer, state.rate_multiplier);
+        break;
+    case InterpolationMode::Polyphase:
+        // TODO(merry): Implement polyphase interpolation
+        state.current_buffer = AudioInterp::Linear(state.interp_state, state.current_buffer, state.rate_multiplier);
+        break;
+    default:
+        UNIMPLEMENTED();
+        break;
+    }
+
+    state.current_sample_number = 0;
+    state.next_sample_number = 0;
+    state.current_buffer_id = buf.buffer_id;
+    state.buffer_update = buf.from_queue;
+
+    LOG_TRACE(Audio_DSP, "source_id=%zu buffer_id=%hu from_queue=%s current_buffer.size()=%zu",
+                         source_id, buf.buffer_id, buf.from_queue ? "true" : "false", state.current_buffer.size());
+    return true;
+}
+
+SourceStatus::Status Source::GetCurrentStatus() {
+    SourceStatus::Status ret;
+
+    // Applications depend on the correct emulation of
+    // current_buffer_id_dirty and current_buffer_id to synchronise
+    // audio with video.
+    ret.is_enabled = state.enabled;
+    ret.current_buffer_id_dirty = state.buffer_update ? 1 : 0;
+    state.buffer_update = false;
+    ret.current_buffer_id = state.current_buffer_id;
+    ret.buffer_position = state.current_sample_number;
+    ret.sync = state.sync;
+
+    return ret;
+}
+
+} // namespace HLE
+} // namespace DSP
diff --git a/src/audio_core/hle/source.h b/src/audio_core/hle/source.h
new file mode 100644
index 000000000..7ee08d424
--- /dev/null
+++ b/src/audio_core/hle/source.h
@@ -0,0 +1,144 @@
+// Copyright 2016 Citra Emulator Project
+// Licensed under GPLv2 or any later version
+// Refer to the license.txt file included.
+
+#pragma once
+
+#include <array>
+#include <queue>
+#include <vector>
+
+#include "audio_core/codec.h"
+#include "audio_core/hle/common.h"
+#include "audio_core/hle/dsp.h"
+#include "audio_core/hle/filter.h"
+#include "audio_core/interpolate.h"
+
+#include "common/common_types.h"
+
+namespace DSP {
+namespace HLE {
+
+/**
+ * This module performs:
+ * - Buffer management
+ * - Decoding of buffers
+ * - Buffer resampling and interpolation
+ * - Per-source filtering (SimpleFilter, BiquadFilter)
+ * - Per-source gain
+ * - Other per-source processing
+ */
+class Source final {
+public:
+    explicit Source(size_t source_id_) : source_id(source_id_) {
+        Reset();
+    }
+
+    /// Resets internal state.
+    void Reset();
+
+    /**
+     * This is called once every audio frame. This performs per-source processing every frame.
+     * @param config The new configuration we've got for this Source from the application.
+     * @param adpcm_coeffs ADPCM coefficients to use if config tells us to use them (may contain invalid values otherwise).
+     * @return The current status of this Source. This is given back to the emulated application via SharedMemory.
+     */
+    SourceStatus::Status Tick(SourceConfiguration::Configuration& config, const s16_le (&adpcm_coeffs)[16]);
+
+    /**
+     * Mix this source's output into dest, using the gains for the `intermediate_mix_id`-th intermediate mixer.
+     * @param dest The QuadFrame32 to mix into.
+     * @param intermediate_mix_id The id of the intermediate mix whose gains we are using.
+     */
+    void MixInto(QuadFrame32& dest, size_t intermediate_mix_id) const;
+
+private:
+    const size_t source_id;
+    StereoFrame16 current_frame;
+
+    using Format = SourceConfiguration::Configuration::Format;
+    using InterpolationMode = SourceConfiguration::Configuration::InterpolationMode;
+    using MonoOrStereo = SourceConfiguration::Configuration::MonoOrStereo;
+
+    /// Internal representation of a buffer for our buffer queue
+    struct Buffer {
+        PAddr physical_address;
+        u32 length;
+        u8 adpcm_ps;
+        std::array<u16, 2> adpcm_yn;
+        bool adpcm_dirty;
+        bool is_looping;
+        u16 buffer_id;
+
+        MonoOrStereo mono_or_stereo;
+        Format format;
+
+        bool from_queue;
+    };
+
+    struct BufferOrder {
+        bool operator() (const Buffer& a, const Buffer& b) const {
+            // Lower buffer_id comes first.
+            return a.buffer_id > b.buffer_id;
+        }
+    };
+
+    struct {
+
+        // State variables
+
+        bool enabled = false;
+        u16 sync = 0;
+
+        // Mixing
+
+        std::array<std::array<float, 4>, 3> gain = {};
+
+        // Buffer queue
+
+        std::priority_queue<Buffer, std::vector<Buffer>, BufferOrder> input_queue;
+        MonoOrStereo mono_or_stereo = MonoOrStereo::Mono;
+        Format format = Format::ADPCM;
+
+        // Current buffer
+
+        u32 current_sample_number = 0;
+        u32 next_sample_number = 0;
+        std::vector<std::array<s16, 2>> current_buffer;
+
+        // buffer_id state
+
+        bool buffer_update = false;
+        u32 current_buffer_id = 0;
+
+        // Decoding state
+
+        std::array<s16, 16> adpcm_coeffs = {};
+        Codec::ADPCMState adpcm_state = {};
+
+        // Resampling state
+
+        float rate_multiplier = 1.0;
+        InterpolationMode interpolation_mode = InterpolationMode::Polyphase;
+        AudioInterp::State interp_state = {};
+
+        // Filter state
+
+        SourceFilters filters;
+
+    } state;
+
+    // Internal functions
+
+    /// INTERNAL: Update our internal state based on the current config.
+    void ParseConfig(SourceConfiguration::Configuration& config, const s16_le (&adpcm_coeffs)[16]);
+    /// INTERNAL: Generate the current audio output for this frame based on our internal state.
+    void GenerateFrame();
+    /// INTERNAL: Dequeues a buffer and does preprocessing on it (decoding, resampling). Puts it into current_buffer.
+    bool DequeueBuffer();
+    /// INTERNAL: Generates a SourceStatus::Status based on our internal state.
+    SourceStatus::Status GetCurrentStatus();
+};
+
+} // namespace HLE
+} // namespace DSP
diff --git a/src/audio_core/sdl2_sink.cpp b/src/audio_core/sdl2_sink.cpp
new file mode 100644
index 000000000..dc75c04ee
--- /dev/null
+++ b/src/audio_core/sdl2_sink.cpp
@@ -0,0 +1,126 @@
+// Copyright 2016 Citra Emulator Project
+// Licensed under GPLv2 or any later version
+// Refer to the license.txt file included.
+
+#include <list>
+#include <vector>
+
+#include <SDL.h>
+
+#include "audio_core/audio_core.h"
+#include "audio_core/sdl2_sink.h"
+
+#include "common/assert.h"
+#include "common/logging/log.h"
+#include <numeric>
+
+namespace AudioCore {
+
+struct SDL2Sink::Impl {
+    unsigned int sample_rate = 0;
+
+    SDL_AudioDeviceID audio_device_id = 0;
+
+    std::list<std::vector<s16>> queue;
+
+    static void Callback(void* impl_, u8* buffer, int buffer_size_in_bytes);
+};
+
+SDL2Sink::SDL2Sink() : impl(std::make_unique<Impl>()) {
+    if (SDL_Init(SDL_INIT_AUDIO) < 0) {
+        LOG_CRITICAL(Audio_Sink, "SDL_Init(SDL_INIT_AUDIO) failed");
+        impl->audio_device_id = 0;
+        return;
+    }
+
+    SDL_AudioSpec desired_audiospec;
+    SDL_zero(desired_audiospec);
+    desired_audiospec.format = AUDIO_S16;
+    desired_audiospec.channels = 2;
+    desired_audiospec.freq = native_sample_rate;
+    desired_audiospec.samples = 1024;
+    desired_audiospec.userdata = impl.get();
+    desired_audiospec.callback = &Impl::Callback;
+
+    SDL_AudioSpec obtained_audiospec;
+    SDL_zero(obtained_audiospec);
+
+    impl->audio_device_id = SDL_OpenAudioDevice(nullptr, false, &desired_audiospec, &obtained_audiospec, 0);
+    if (impl->audio_device_id <= 0) {
+        LOG_CRITICAL(Audio_Sink, "SDL_OpenAudioDevice failed");
+        return;
+    }
+
+    impl->sample_rate = obtained_audiospec.freq;
+
+    // SDL2 audio devices start out paused, unpause it:
+    SDL_PauseAudioDevice(impl->audio_device_id, 0);
+}
+
+SDL2Sink::~SDL2Sink() {
+    if (impl->audio_device_id <= 0)
+        return;
+
+    SDL_CloseAudioDevice(impl->audio_device_id);
+}
+
+unsigned int SDL2Sink::GetNativeSampleRate() const {
+    if (impl->audio_device_id <= 0)
+        return native_sample_rate;
+
+    return impl->sample_rate;
+}
+
+void SDL2Sink::EnqueueSamples(const std::vector<s16>& samples) {
+    if (impl->audio_device_id <= 0)
+        return;
+
+    ASSERT_MSG(samples.size() % 2 == 0, "Samples must be in interleaved stereo PCM16 format (size must be a multiple of two)");
+
+    SDL_LockAudioDevice(impl->audio_device_id);
+    impl->queue.emplace_back(samples);
+    SDL_UnlockAudioDevice(impl->audio_device_id);
+}
+
+size_t SDL2Sink::SamplesInQueue() const {
+    if (impl->audio_device_id <= 0)
+        return 0;
+
+    SDL_LockAudioDevice(impl->audio_device_id);
+
+    size_t total_size = std::accumulate(impl->queue.begin(), impl->queue.end(), static_cast<size_t>(0),
+        [](size_t sum, const auto& buffer) {
+            // Division by two because each stereo sample is made of two s16.
+            return sum + buffer.size() / 2;
+        });
+
+    SDL_UnlockAudioDevice(impl->audio_device_id);
+
+    return total_size;
+}
+
+void SDL2Sink::Impl::Callback(void* impl_, u8* buffer, int buffer_size_in_bytes) {
+    Impl* impl = reinterpret_cast<Impl*>(impl_);
+
+    size_t remaining_size = static_cast<size_t>(buffer_size_in_bytes) / sizeof(s16); // Keep track of size in 16-bit increments.
+
+    while (remaining_size > 0 && !impl->queue.empty()) {
+        if (impl->queue.front().size() <= remaining_size) {
+            memcpy(buffer, impl->queue.front().data(), impl->queue.front().size() * sizeof(s16));
+            buffer += impl->queue.front().size() * sizeof(s16);
+            remaining_size -= impl->queue.front().size();
+            impl->queue.pop_front();
+        } else {
+            memcpy(buffer, impl->queue.front().data(), remaining_size * sizeof(s16));
+            buffer += remaining_size * sizeof(s16);
+            impl->queue.front().erase(impl->queue.front().begin(), impl->queue.front().begin() + remaining_size);
+            remaining_size = 0;
+        }
+    }
+
+    if (remaining_size > 0) {
+        memset(buffer, 0, remaining_size * sizeof(s16));
+    }
+}
+
+} // namespace AudioCore
diff --git a/src/audio_core/sdl2_sink.h b/src/audio_core/sdl2_sink.h
new file mode 100644
index 000000000..0f296b673
--- /dev/null
+++ b/src/audio_core/sdl2_sink.h
@@ -0,0 +1,30 @@
+// Copyright 2016 Citra Emulator Project
+// Licensed under GPLv2 or any later version
+// Refer to the license.txt file included.
+
+#pragma once
+
+#include <cstddef>
+#include <memory>
+
+#include "audio_core/sink.h"
+
+namespace AudioCore {
+
+class SDL2Sink final : public Sink {
+public:
+    SDL2Sink();
+    ~SDL2Sink() override;
+
+    unsigned int GetNativeSampleRate() const override;
+
+    void EnqueueSamples(const std::vector<s16>& samples) override;
+
+    size_t SamplesInQueue() const override;
+
+private:
+    struct Impl;
+    std::unique_ptr<Impl> impl;
+};
+
+} // namespace AudioCore
diff --git a/src/audio_core/sink.h b/src/audio_core/sink.h
index cad21a85e..1c881c3d2 100644
--- a/src/audio_core/sink.h
+++ b/src/audio_core/sink.h
@@ -19,7 +19,7 @@ public:
     virtual ~Sink() = default;
 
     /// The native rate of this sink. The sink expects to be fed samples that respect this. (Units: samples/sec)
-    virtual unsigned GetNativeSampleRate() const = 0;
+    virtual unsigned int GetNativeSampleRate() const = 0;
 
     /**
      * Feed stereo samples to sink.
diff --git a/src/audio_core/sink_details.cpp b/src/audio_core/sink_details.cpp
index d2cc74103..ba5e83d17 100644
--- a/src/audio_core/sink_details.cpp
+++ b/src/audio_core/sink_details.cpp
@@ -8,10 +8,17 @@
 #include "audio_core/null_sink.h"
 #include "audio_core/sink_details.h"
 
+#ifdef HAVE_SDL2
+#include "audio_core/sdl2_sink.h"
+#endif
+
 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_SDL2
+    { "sdl2", []() { return std::make_unique<SDL2Sink>(); } },
+#endif
     { "null", []() { return std::make_unique<NullSink>(); } },
 };