19 files changed, 1014 insertions, 69 deletions

diff --git a/src/audio_core/CMakeLists.txt b/src/audio_core/CMakeLists.txt
index 869da5e83..13b5e400e 100644
--- a/src/audio_core/CMakeLists.txt
+++ b/src/audio_core/CMakeLists.txt
@@ -4,6 +4,9 @@ set(SRCS
             hle/dsp.cpp
             hle/filter.cpp
             hle/pipe.cpp
+            hle/source.cpp
+            interpolate.cpp
+            sink_details.cpp
             )
 
 set(HEADERS
@@ -13,9 +16,27 @@ set(HEADERS
             hle/dsp.h
             hle/filter.h
             hle/pipe.h
+            hle/source.h
+            interpolate.h
+            null_sink.h
             sink.h
+            sink_details.h
             )
 
+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})
\ No newline at end of file
+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/audio_core.cpp b/src/audio_core/audio_core.cpp
index 894f46990..d42249ebd 100644
--- a/src/audio_core/audio_core.cpp
+++ b/src/audio_core/audio_core.cpp
@@ -2,8 +2,15 @@
 // Licensed under GPLv2 or any later version
 // Refer to the license.txt file included.
 
+#include <memory>
+#include <string>
+
 #include "audio_core/audio_core.h"
 #include "audio_core/hle/dsp.h"
+#include "audio_core/hle/pipe.h"
+#include "audio_core/null_sink.h"
+#include "audio_core/sink.h"
+#include "audio_core/sink_details.h"
 
 #include "core/core_timing.h"
 #include "core/hle/kernel/vm_manager.h"
@@ -17,17 +24,16 @@ static constexpr u64 audio_frame_ticks = 1310252ull; ///< Units: ARM11 cycles
 
 static void AudioTickCallback(u64 /*userdata*/, int cycles_late) {
     if (DSP::HLE::Tick()) {
-        // HACK: We're not signaling the interrups when they should be, but just firing them all off together.
-        // It should be only (interrupt_id = 2, channel_id = 2) that's signalled here.
-        // TODO(merry): Understand when the other interrupts are fired.
-        DSP_DSP::SignalAllInterrupts();
+        // TODO(merry): Signal all the other interrupts as appropriate.
+        DSP_DSP::SignalPipeInterrupt(DSP::HLE::DspPipe::Audio);
+        // HACK(merry): Added to prevent regressions. Will remove soon.
+        DSP_DSP::SignalPipeInterrupt(DSP::HLE::DspPipe::Binary);
     }
 
     // Reschedule recurrent event
     CoreTiming::ScheduleEvent(audio_frame_ticks - cycles_late, tick_event);
 }
 
-/// Initialise Audio
 void Init() {
     DSP::HLE::Init();
 
@@ -35,19 +41,39 @@ void Init() {
     CoreTiming::ScheduleEvent(audio_frame_ticks, tick_event);
 }
 
-/// Add DSP address spaces to Process's address space.
 void AddAddressSpace(Kernel::VMManager& address_space) {
-    auto r0_vma = address_space.MapBackingMemory(DSP::HLE::region0_base, reinterpret_cast<u8*>(&DSP::HLE::g_region0), sizeof(DSP::HLE::SharedMemory), Kernel::MemoryState::IO).MoveFrom();
+    auto r0_vma = address_space.MapBackingMemory(DSP::HLE::region0_base, reinterpret_cast<u8*>(&DSP::HLE::g_regions[0]), sizeof(DSP::HLE::SharedMemory), Kernel::MemoryState::IO).MoveFrom();
     address_space.Reprotect(r0_vma, Kernel::VMAPermission::ReadWrite);
 
-    auto r1_vma = address_space.MapBackingMemory(DSP::HLE::region1_base, reinterpret_cast<u8*>(&DSP::HLE::g_region1), sizeof(DSP::HLE::SharedMemory), Kernel::MemoryState::IO).MoveFrom();
+    auto r1_vma = address_space.MapBackingMemory(DSP::HLE::region1_base, reinterpret_cast<u8*>(&DSP::HLE::g_regions[1]), sizeof(DSP::HLE::SharedMemory), Kernel::MemoryState::IO).MoveFrom();
     address_space.Reprotect(r1_vma, Kernel::VMAPermission::ReadWrite);
 }
 
-/// Shutdown Audio
+void SelectSink(std::string sink_id) {
+    if (sink_id == "auto") {
+        // Auto-select.
+        // g_sink_details is ordered in terms of desirability, with the best choice at the front.
+        const auto& sink_detail = g_sink_details.front();
+        DSP::HLE::SetSink(sink_detail.factory());
+        return;
+    }
+
+    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 (iter == g_sink_details.end()) {
+        LOG_ERROR(Audio, "AudioCore::SelectSink given invalid sink_id");
+        DSP::HLE::SetSink(std::make_unique<NullSink>());
+        return;
+    }
+
+    DSP::HLE::SetSink(iter->factory());
+}
+
 void Shutdown() {
     CoreTiming::UnscheduleEvent(tick_event, 0);
     DSP::HLE::Shutdown();
 }
 
-} //namespace
+} // namespace AudioCore
diff --git a/src/audio_core/audio_core.h b/src/audio_core/audio_core.h
index 64c330914..f618361f3 100644
--- a/src/audio_core/audio_core.h
+++ b/src/audio_core/audio_core.h
@@ -4,14 +4,14 @@
 
 #pragma once
 
+#include <string>
+
 namespace Kernel {
 class VMManager;
 }
 
 namespace AudioCore {
 
-constexpr int num_sources = 24;
-constexpr int samples_per_frame = 160;     ///< Samples per audio frame at native sample rate
 constexpr int native_sample_rate = 32728;  ///< 32kHz
 
 /// Initialise Audio Core
@@ -20,6 +20,9 @@ void Init();
 /// Add DSP address spaces to a Process.
 void AddAddressSpace(Kernel::VMManager& vm_manager);
 
+/// Select the sink to use based on sink id.
+void SelectSink(std::string sink_id);
+
 /// Shutdown Audio Core
 void Shutdown();
 
diff --git a/src/audio_core/hle/common.h b/src/audio_core/hle/common.h
index 37d441eb2..596b67eaf 100644
--- a/src/audio_core/hle/common.h
+++ b/src/audio_core/hle/common.h
@@ -7,18 +7,19 @@
 #include <algorithm>
 #include <array>
 
-#include "audio_core/audio_core.h"
-
 #include "common/common_types.h"
 
 namespace DSP {
 namespace HLE {
 
+constexpr int num_sources = 24;
+constexpr int samples_per_frame = 160;     ///< Samples per audio frame at native sample rate
+
 /// The final output to the speakers is stereo. Preprocessing output in Source is also stereo.
-using StereoFrame16 = std::array<std::array<s16, 2>, AudioCore::samples_per_frame>;
+using StereoFrame16 = std::array<std::array<s16, 2>, samples_per_frame>;
 
 /// The DSP is quadraphonic internally.
-using QuadFrame32   = std::array<std::array<s32, 4>, AudioCore::samples_per_frame>;
+using QuadFrame32   = std::array<std::array<s32, 4>, samples_per_frame>;
 
 /**
  * This performs the filter operation defined by FilterT::ProcessSample on the frame in-place.
@@ -26,7 +27,7 @@ using QuadFrame32   = std::array<std::array<s32, 4>, AudioCore::samples_per_fram
  */
 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 c89356edc..0cdbdb06a 100644
--- a/src/audio_core/hle/dsp.cpp
+++ b/src/audio_core/hle/dsp.cpp
@@ -2,40 +2,81 @@
 // 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 {
 namespace HLE {
 
-SharedMemory g_region0;
-SharedMemory g_region1;
+std::array<SharedMemory, 2> g_regions;
+
+static size_t CurrentRegionIndex() {
+    // The region with the higher frame counter is chosen unless there is wraparound.
+    // This function only returns a 0 or 1.
+
+    if (g_regions[0].frame_counter == 0xFFFFu && g_regions[1].frame_counter != 0xFFFEu) {
+        // Wraparound has occured.
+        return 1;
+    }
+
+    if (g_regions[1].frame_counter == 0xFFFFu && g_regions[0].frame_counter != 0xFFFEu) {
+        // Wraparound has occured.
+        return 0;
+    }
+
+    return (g_regions[0].frame_counter > g_regions[1].frame_counter) ? 0 : 1;
+}
+
+static SharedMemory& ReadRegion() {
+    return g_regions[CurrentRegionIndex()];
+}
+
+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() {
-    return true;
-}
+    SharedMemory& read = ReadRegion();
+    SharedMemory& write = WriteRegion();
 
-SharedMemory& CurrentRegion() {
-    // The region with the higher frame counter is chosen unless there is wraparound.
+    std::array<QuadFrame32, 3> intermediate_mixes = {};
 
-    if (g_region0.frame_counter == 0xFFFFu && g_region1.frame_counter != 0xFFFEu) {
-        // Wraparound has occured.
-        return g_region1;
+    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);
+        }
     }
 
-    if (g_region1.frame_counter == 0xFFFFu && g_region0.frame_counter != 0xFFFEu) {
-        // Wraparound has occured.
-        return g_region0;
-    }
+    return true;
+}
 
-    return (g_region0.frame_counter > g_region1.frame_counter) ? g_region0 : g_region1;
+void SetSink(std::unique_ptr<AudioCore::Sink> sink_) {
+    sink = std::move(sink_);
 }
 
 } // namespace HLE
diff --git a/src/audio_core/hle/dsp.h b/src/audio_core/hle/dsp.h
index c15ef0b7a..f6e53f68f 100644
--- a/src/audio_core/hle/dsp.h
+++ b/src/audio_core/hle/dsp.h
@@ -4,16 +4,22 @@
 
 #pragma once
 
+#include <array>
 #include <cstddef>
+#include <memory>
 #include <type_traits>
 
-#include "audio_core/audio_core.h"
+#include "audio_core/hle/common.h"
 
 #include "common/bit_field.h"
 #include "common/common_funcs.h"
 #include "common/common_types.h"
 #include "common/swap.h"
 
+namespace AudioCore {
+class Sink;
+}
+
 namespace DSP {
 namespace HLE {
 
@@ -27,13 +33,8 @@ 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;
-extern SharedMemory g_region0;
-
 constexpr VAddr region1_base = 0x1FF70000;
-extern SharedMemory g_region1;
 
 /**
  * The DSP is native 16-bit. The DSP also appears to be big-endian. When reading 32-bit numbers from
@@ -164,9 +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;
@@ -305,7 +306,7 @@ struct SourceConfiguration {
         u16_le buffer_id;
     };
 
-    Configuration config[AudioCore::num_sources];
+    Configuration config[num_sources];
 };
 ASSERT_DSP_STRUCT(SourceConfiguration::Configuration, 192);
 ASSERT_DSP_STRUCT(SourceConfiguration::Configuration::Buffer, 20);
@@ -313,14 +314,14 @@ 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);
     };
 
-    Status status[AudioCore::num_sources];
+    Status status[num_sources];
 };
 ASSERT_DSP_STRUCT(SourceStatus::Status, 12);
 
@@ -413,7 +414,7 @@ ASSERT_DSP_STRUCT(DspConfiguration::ReverbEffect, 52);
 struct AdpcmCoefficients {
     /// Coefficients are signed fixed point with 11 fractional bits.
     /// Each source has 16 coefficients associated with it.
-    s16_le coeff[AudioCore::num_sources][16];
+    s16_le coeff[num_sources][16];
 };
 ASSERT_DSP_STRUCT(AdpcmCoefficients, 768);
 
@@ -427,7 +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 * AudioCore::samples_per_frame];
+    s16_le pcm16[2 * samples_per_frame];
 };
 ASSERT_DSP_STRUCT(FinalMixSamples, 640);
 
@@ -437,7 +438,7 @@ ASSERT_DSP_STRUCT(FinalMixSamples, 640);
 /// Values that exceed s16 range will be clipped by the DSP after further processing.
 struct IntermediateMixSamples {
     struct Samples {
-        s32_le pcm32[4][AudioCore::samples_per_frame]; ///< Little-endian as opposed to DSP middle-endian.
+        s32_le pcm32[4][samples_per_frame]; ///< Little-endian as opposed to DSP middle-endian.
     };
 
     Samples mix1;
@@ -502,6 +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");
@@ -535,8 +538,11 @@ void Shutdown();
  */
 bool Tick();
 
-/// Returns a mutable reference to the current region. Current region is selected based on the frame counter.
-SharedMemory& CurrentRegion();
+/**
+ * Set the output sink. This must be called before calling Tick().
+ * @param sink The sink to which audio will be output to.
+ */
+void SetSink(std::unique_ptr<AudioCore::Sink> sink);
 
 } // namespace HLE
 } // namespace DSP
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/pipe.cpp b/src/audio_core/hle/pipe.cpp
index 9381883b4..44dff1345 100644
--- a/src/audio_core/hle/pipe.cpp
+++ b/src/audio_core/hle/pipe.cpp
@@ -12,12 +12,14 @@
 #include "common/common_types.h"
 #include "common/logging/log.h"
 
+#include "core/hle/service/dsp_dsp.h"
+
 namespace DSP {
 namespace HLE {
 
 static DspState dsp_state = DspState::Off;
 
-static std::array<std::vector<u8>, static_cast<size_t>(DspPipe::DspPipe_MAX)> pipe_data;
+static std::array<std::vector<u8>, NUM_DSP_PIPE> pipe_data;
 
 void ResetPipes() {
     for (auto& data : pipe_data) {
@@ -27,17 +29,24 @@ void ResetPipes() {
 }
 
 std::vector<u8> PipeRead(DspPipe pipe_number, u32 length) {
-    if (pipe_number >= DspPipe::DspPipe_MAX) {
-        LOG_ERROR(Audio_DSP, "pipe_number = %u invalid", pipe_number);
+    const size_t pipe_index = static_cast<size_t>(pipe_number);
+
+    if (pipe_index >= NUM_DSP_PIPE) {
+        LOG_ERROR(Audio_DSP, "pipe_number = %zu invalid", pipe_index);
         return {};
     }
 
-    std::vector<u8>& data = pipe_data[static_cast<size_t>(pipe_number)];
+    if (length > UINT16_MAX) { // Can only read at most UINT16_MAX from the pipe
+        LOG_ERROR(Audio_DSP, "length of %u greater than max of %u", length, UINT16_MAX);
+        return {};
+    }
+
+    std::vector<u8>& data = pipe_data[pipe_index];
 
     if (length > data.size()) {
-        LOG_WARNING(Audio_DSP, "pipe_number = %u is out of data, application requested read of %u but %zu remain",
-                    pipe_number, length, data.size());
-        length = data.size();
+        LOG_WARNING(Audio_DSP, "pipe_number = %zu is out of data, application requested read of %u but %zu remain",
+                    pipe_index, length, data.size());
+        length = static_cast<u32>(data.size());
     }
 
     if (length == 0)
@@ -49,16 +58,20 @@ std::vector<u8> PipeRead(DspPipe pipe_number, u32 length) {
 }
 
 size_t GetPipeReadableSize(DspPipe pipe_number) {
-    if (pipe_number >= DspPipe::DspPipe_MAX) {
-        LOG_ERROR(Audio_DSP, "pipe_number = %u invalid", pipe_number);
+    const size_t pipe_index = static_cast<size_t>(pipe_number);
+
+    if (pipe_index >= NUM_DSP_PIPE) {
+        LOG_ERROR(Audio_DSP, "pipe_number = %zu invalid", pipe_index);
         return 0;
     }
 
-    return pipe_data[static_cast<size_t>(pipe_number)].size();
+    return pipe_data[pipe_index].size();
 }
 
 static void WriteU16(DspPipe pipe_number, u16 value) {
-    std::vector<u8>& data = pipe_data[static_cast<size_t>(pipe_number)];
+    const size_t pipe_index = static_cast<size_t>(pipe_number);
+
+    std::vector<u8>& data = pipe_data.at(pipe_index);
     // Little endian
     data.emplace_back(value & 0xFF);
     data.emplace_back(value >> 8);
@@ -86,11 +99,13 @@ static void AudioPipeWriteStructAddresses() {
     };
 
     // Begin with a u16 denoting the number of structs.
-    WriteU16(DspPipe::Audio, struct_addresses.size());
+    WriteU16(DspPipe::Audio, static_cast<u16>(struct_addresses.size()));
     // Then write the struct addresses.
     for (u16 addr : struct_addresses) {
         WriteU16(DspPipe::Audio, addr);
     }
+    // Signal that we have data on this pipe.
+    DSP_DSP::SignalPipeInterrupt(DspPipe::Audio);
 }
 
 void PipeWrite(DspPipe pipe_number, const std::vector<u8>& buffer) {
@@ -145,7 +160,7 @@ void PipeWrite(DspPipe pipe_number, const std::vector<u8>& buffer) {
         return;
     }
     default:
-        LOG_CRITICAL(Audio_DSP, "pipe_number = %u unimplemented", pipe_number);
+        LOG_CRITICAL(Audio_DSP, "pipe_number = %zu unimplemented", static_cast<size_t>(pipe_number));
         UNIMPLEMENTED();
         return;
     }
diff --git a/src/audio_core/hle/pipe.h b/src/audio_core/hle/pipe.h
index 382d35e87..b714c0496 100644
--- a/src/audio_core/hle/pipe.h
+++ b/src/audio_core/hle/pipe.h
@@ -19,15 +19,19 @@ enum class DspPipe {
     Debug = 0,
     Dma = 1,
     Audio = 2,
-    Binary = 3,
-    DspPipe_MAX
+    Binary = 3
 };
+constexpr size_t NUM_DSP_PIPE = 8;
 
 /**
- * Read a DSP pipe.
- * @param pipe_number The Pipe ID
- * @param length How much data to request.
- * @return The data read from the pipe. The size of this vector can be less than the length requested.
+ * Reads `length` bytes from the DSP pipe identified with `pipe_number`.
+ * @note Can read up to the maximum value of a u16 in bytes (65,535).
+ * @note IF an error is encoutered with either an invalid `pipe_number` or `length` value, an empty vector will be returned.
+ * @note IF `length` is set to 0, an empty vector will be returned.
+ * @note IF `length` is greater than the amount of data available, this function will only read the available amount.
+ * @param pipe_number a `DspPipe`
+ * @param length the number of bytes to read. The max is 65,535 (max of u16).
+ * @returns a vector of bytes from the specified pipe. On error, will be empty.
  */
 std::vector<u8> PipeRead(DspPipe pipe_number, u32 length);
 
diff --git a/src/audio_core/hle/source.cpp b/src/audio_core/hle/source.cpp
new file mode 100644
index 000000000..30552fe26
--- /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", 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", source_id);
+    }
+
+    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/interpolate.cpp b/src/audio_core/interpolate.cpp
new file mode 100644
index 000000000..fcd3aa066
--- /dev/null
+++ b/src/audio_core/interpolate.cpp
@@ -0,0 +1,85 @@
+// Copyright 2016 Citra Emulator Project
+// Licensed under GPLv2 or any later version
+// Refer to the license.txt file included.
+
+#include "audio_core/interpolate.h"
+
+#include "common/assert.h"
+#include "common/math_util.h"
+
+namespace AudioInterp {
+
+// Calculations are done in fixed point with 24 fractional bits.
+// (This is not verified. This was chosen for minimal error.)
+constexpr u64 scale_factor = 1 << 24;
+constexpr u64 scale_mask = scale_factor - 1;
+
+/// Here we step over the input in steps of rate_multiplier, until we consume all of the input.
+/// Three adjacent samples are passed to fn each step.
+template <typename Function>
+static StereoBuffer16 StepOverSamples(State& state, const StereoBuffer16& input, float rate_multiplier, Function fn) {
+    ASSERT(rate_multiplier > 0);
+
+    if (input.size() < 2)
+        return {};
+
+    StereoBuffer16 output;
+    output.reserve(static_cast<size_t>(input.size() / rate_multiplier));
+
+    u64 step_size = static_cast<u64>(rate_multiplier * scale_factor);
+
+    u64 fposition = 0;
+    const u64 max_fposition = input.size() * scale_factor;
+
+    while (fposition < 1 * scale_factor) {
+        u64 fraction = fposition & scale_mask;
+
+        output.push_back(fn(fraction, state.xn2, state.xn1, input[0]));
+
+        fposition += step_size;
+    }
+
+    while (fposition < 2 * scale_factor) {
+        u64 fraction = fposition & scale_mask;
+
+        output.push_back(fn(fraction, state.xn1, input[0], input[1]));
+
+        fposition += step_size;
+    }
+
+    while (fposition < max_fposition) {
+        u64 fraction = fposition & scale_mask;
+
+        size_t index = static_cast<size_t>(fposition / scale_factor);
+        output.push_back(fn(fraction, input[index - 2], input[index - 1], input[index]));
+
+        fposition += step_size;
+    }
+
+    state.xn2 = input[input.size() - 2];
+    state.xn1 = input[input.size() - 1];
+
+    return output;
+}
+
+StereoBuffer16 None(State& state, const StereoBuffer16& input, float rate_multiplier) {
+    return StepOverSamples(state, input, rate_multiplier, [](u64 fraction, const auto& x0, const auto& x1, const auto& x2) {
+        return x0;
+    });
+}
+
+StereoBuffer16 Linear(State& state, const StereoBuffer16& input, float rate_multiplier) {
+    // Note on accuracy: Some values that this produces are +/- 1 from the actual firmware.
+    return StepOverSamples(state, input, rate_multiplier, [](u64 fraction, const auto& x0, const auto& x1, const auto& x2) {
+        // This is a saturated subtraction. (Verified by black-box fuzzing.)
+        s64 delta0 = MathUtil::Clamp<s64>(x1[0] - x0[0], -32768, 32767);
+        s64 delta1 = MathUtil::Clamp<s64>(x1[1] - x0[1], -32768, 32767);
+
+        return std::array<s16, 2> {
+            static_cast<s16>(x0[0] + fraction * delta0 / scale_factor),
+            static_cast<s16>(x0[1] + fraction * delta1 / scale_factor)
+        };
+    });
+}
+
+} // namespace AudioInterp
diff --git a/src/audio_core/interpolate.h b/src/audio_core/interpolate.h
new file mode 100644
index 000000000..a4c0a453d
--- /dev/null
+++ b/src/audio_core/interpolate.h
@@ -0,0 +1,41 @@
+// Copyright 2016 Citra 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 AudioInterp {
+
+/// A variable length buffer of signed PCM16 stereo samples.
+using StereoBuffer16 = std::vector<std::array<s16, 2>>;
+
+struct State {
+    // Two historical samples.
+    std::array<s16, 2> xn1 = {}; ///< x[n-1]
+    std::array<s16, 2> xn2 = {}; ///< x[n-2]
+};
+
+/**
+ * No interpolation. This is equivalent to a zero-order hold. There is a two-sample predelay.
+ * @param input Input buffer.
+ * @param rate_multiplier Stretch factor. Must be a positive non-zero value.
+ *                        rate_multiplier > 1.0 performs decimation and rate_multipler < 1.0 performs upsampling.
+ * @return The resampled audio buffer.
+ */
+StereoBuffer16 None(State& state, const StereoBuffer16& input, float rate_multiplier);
+
+/**
+ * Linear interpolation. This is equivalent to a first-order hold. There is a two-sample predelay.
+ * @param input Input buffer.
+ * @param rate_multiplier Stretch factor. Must be a positive non-zero value.
+ *                        rate_multiplier > 1.0 performs decimation and rate_multipler < 1.0 performs upsampling.
+ * @return The resampled audio buffer.
+ */
+StereoBuffer16 Linear(State& state, const StereoBuffer16& input, float rate_multiplier);
+
+} // namespace AudioInterp
diff --git a/src/audio_core/null_sink.h b/src/audio_core/null_sink.h
new file mode 100644
index 000000000..faf0ee4e1
--- /dev/null
+++ b/src/audio_core/null_sink.h
@@ -0,0 +1,29 @@
+// Copyright 2016 Citra Emulator Project
+// Licensed under GPLv2 or any later version
+// Refer to the license.txt file included.
+
+#pragma once
+
+#include <cstddef>
+
+#include "audio_core/audio_core.h"
+#include "audio_core/sink.h"
+
+namespace AudioCore {
+
+class NullSink final : public Sink {
+public:
+    ~NullSink() override = default;
+
+    unsigned int GetNativeSampleRate() const override {
+        return native_sample_rate;
+    }
+
+    void EnqueueSamples(const std::vector<s16>&) override {}
+
+    size_t SamplesInQueue() const override {
+        return 0;
+    }
+};
+
+} // namespace AudioCore
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
new file mode 100644
index 000000000..ba5e83d17
--- /dev/null
+++ b/src/audio_core/sink_details.cpp
@@ -0,0 +1,25 @@
+// Copyright 2016 Citra Emulator Project
+// Licensed under GPLv2 or any later version
+// Refer to the license.txt file included.
+
+#include <memory>
+#include <vector>
+
+#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>(); } },
+};
+
+} // namespace AudioCore
diff --git a/src/audio_core/sink_details.h b/src/audio_core/sink_details.h
new file mode 100644
index 000000000..4b30cf835
--- /dev/null
+++ b/src/audio_core/sink_details.h
@@ -0,0 +1,27 @@
+// Copyright 2016 Citra Emulator Project
+// Licensed under GPLv2 or any later version
+// Refer to the license.txt file included.
+
+#pragma once
+
+#include <functional>
+#include <memory>
+#include <vector>
+
+namespace AudioCore {
+
+class Sink;
+
+struct SinkDetails {
+    SinkDetails(const char* id_, std::function<std::unique_ptr<Sink>()> factory_)
+        : id(id_), factory(factory_) {}
+
+    /// Name for this sink.
+    const char* id;
+    /// A method to call to construct an instance of this type of sink.
+    std::function<std::unique_ptr<Sink>()> factory;
+};
+
+extern const std::vector<SinkDetails> g_sink_details;
+
+} // namespace AudioCore