diff --git a/CMakeLists.txt b/CMakeLists.txt
index 4b763dff..3dc7e467 100644
--- a/CMakeLists.txt
+++ b/CMakeLists.txt
@@ -197,7 +197,7 @@ set(APPLET_SOURCE_FILES src/core/applets/applet.cpp src/core/applets/mii_selecto
                         src/core/applets/error_applet.cpp
 )
 set(AUDIO_SOURCE_FILES src/core/audio/dsp_core.cpp src/core/audio/null_core.cpp src/core/audio/teakra_core.cpp
-                       src/core/audio/miniaudio_device.cpp
+                       src/core/audio/miniaudio_device.cpp src/core/audio/hle_core.cpp
 )
 set(RENDERER_SW_SOURCE_FILES src/core/renderer_sw/renderer_sw.cpp)
 
@@ -234,6 +234,7 @@ set(HEADER_FILES include/emulator.hpp include/helpers.hpp include/termcolor.hpp
                  include/PICA/dynapica/shader_rec_emitter_arm64.hpp include/scheduler.hpp include/applets/error_applet.hpp
                  include/audio/dsp_core.hpp include/audio/null_core.hpp include/audio/teakra_core.hpp
                  include/audio/miniaudio_device.hpp include/ring_buffer.hpp include/bitfield.hpp include/audio/dsp_shared_mem.hpp
+                 include/audio/hle_core.hpp
 )
 
 cmrc_add_resource_library(
diff --git a/include/audio/dsp_core.hpp b/include/audio/dsp_core.hpp
index 1a556f28..a4fb1ab1 100644
--- a/include/audio/dsp_core.hpp
+++ b/include/audio/dsp_core.hpp
@@ -37,7 +37,7 @@ namespace Audio {
 		MAKE_LOG_FUNCTION(log, dspLogger)
 
 	  public:
-		enum class Type { Null, Teakra };
+		enum class Type { Null, Teakra, HLE };
 		DSPCore(Memory& mem, Scheduler& scheduler, DSPService& dspService)
 			: mem(mem), scheduler(scheduler), dspService(dspService) {}
 		virtual ~DSPCore() {}
diff --git a/include/audio/hle_core.hpp b/include/audio/hle_core.hpp
new file mode 100644
index 00000000..6d7b3ad1
--- /dev/null
+++ b/include/audio/hle_core.hpp
@@ -0,0 +1,157 @@
+#pragma once
+#include <array>
+#include <queue>
+#include <vector>
+
+#include "audio/dsp_core.hpp"
+#include "audio/dsp_shared_mem.hpp"
+#include "memory.hpp"
+
+namespace Audio {
+	using SampleFormat = HLE::SourceConfiguration::Configuration::Format;
+	using SourceType = HLE::SourceConfiguration::Configuration::MonoOrStereo;
+
+	struct DSPSource {
+		// Audio buffer information
+		// https://www.3dbrew.org/wiki/DSP_Memory_Region
+		struct Buffer {
+			u32 paddr;        // Physical address of the buffer
+			u32 sampleCount;  // Total number of samples
+			u8 adpcmScale;    // ADPCM predictor/scale
+			u8 pad1;           // Unknown
+
+			std::array<s16, 2> previousSamples;  // ADPCM y[n-1] and y[n-2]
+			bool adpcmDirty;
+			bool looping;
+			u16 bufferID;
+			u8 pad2;
+
+			u32 playPosition = 0;  // Current position in the buffer
+			SampleFormat format;
+			SourceType sourceType;
+
+			bool fromQueue = false;      // Is this buffer from the buffer queue or an embedded buffer?
+			bool hasPlayedOnce = false;  // Has the buffer been played at least once before?
+
+			bool operator<(const Buffer& other) const {
+				// Lower ID = Higher priority
+				// If this buffer ID is greater than the other one, then this buffer has a lower priority
+				return this->bufferID > other.bufferID;
+			}
+		};
+
+		using BufferQueue = std::priority_queue<Buffer>;
+
+		std::array<float, 3> gain0, gain1, gain2;
+		u16 syncCount;
+		bool enabled;
+
+		BufferQueue buffers;
+		int index = 0;  // Index of the voice in [0, 23] for debugging
+
+		void reset();
+		DSPSource() { reset(); }
+	};
+
+	class HLE_DSP : public DSPCore {
+		// The audio frame types are public in case we want to use them for unit tests
+	  public:
+		template <typename T, usize channelCount = 1>
+		using Sample = std::array<T, channelCount>;
+
+		template <typename T, usize channelCount>
+		using Frame = std::array<Sample<T, channelCount>, 160>;
+		
+		template <typename T>
+		using MonoFrame = Frame<T, 1>;
+
+		template <typename T>
+		using StereoFrame = Frame<T, 2>;
+
+		template <typename T>
+		using QuadFrame = Frame<T, 4>;
+
+		using Source = Audio::DSPSource;
+	  private:
+		enum class DSPState : u32 {
+			Off,
+			On,
+			Slep,
+		};
+
+		// Number of DSP pipes
+		static constexpr size_t pipeCount = 8;
+		DSPState dspState;
+
+		std::array<std::vector<u8>, pipeCount> pipeData;      // The data of each pipe
+		std::array<Source, Audio::HLE::sourceCount> sources;  // DSP voices
+		Audio::HLE::DspMemory dspRam;
+
+		SampleFormat sampleFormat = SampleFormat::ADPCM;
+		SourceType sourceType = SourceType::Stereo;
+
+		void resetAudioPipe();
+		bool loaded = false; // Have we loaded a component?
+
+		// Get the index for the current region we'll be reading. Returns the region with the highest frame counter
+		// Accounting for whether one of the frame counters has wrapped around
+		usize readRegionIndex() const {
+			const auto counter0 = dspRam.region0.frameCounter;
+			const auto counter1 = dspRam.region1.frameCounter;
+
+			// Handle wraparound cases first
+			if (counter0 == 0xffff && counter1 != 0xfffe) {
+				return 1;
+			} else if (counter1 == 0xffff && counter0 != 0xfffe) {
+				return 0;
+			} else {
+				return counter0 > counter1 ? 0 : 0;
+			}
+		}
+
+		// DSP shared memory is double buffered; One region is being written to while the other one is being read from
+		Audio::HLE::SharedMemory& readRegion() { return readRegionIndex() == 0 ? dspRam.region0 : dspRam.region1; }
+		Audio::HLE::SharedMemory& writeRegion() { return readRegionIndex() == 0 ? dspRam.region1 : dspRam.region0; }
+
+		// Get a pointer of type T* to the data starting from physical address paddr
+		template <typename T>
+		T* getPointerPhys(u32 paddr, u32 size = 0) {
+			if (paddr >= PhysicalAddrs::FCRAM && paddr + size <= PhysicalAddrs::FCRAMEnd) {
+				u8* fcram = mem.getFCRAM();
+				u32 index = paddr - PhysicalAddrs::FCRAM;
+
+				return (T*)&fcram[index];
+			} else if (paddr >= PhysicalAddrs::DSP_RAM && paddr + size <= PhysicalAddrs::DSP_RAM_End) {
+				u32 index = paddr - PhysicalAddrs::DSP_RAM;
+				return (T*)&dspRam.rawMemory[index];
+			} else [[unlikely]] {
+				Helpers::warn("[DSP] Tried to access unknown physical address: %08X", paddr);
+				return nullptr;
+			}
+		}
+
+		void updateSourceConfig(Source& source, HLE::SourceConfiguration::Configuration& config);
+		void generateFrame(StereoFrame<s16>& frame);
+		void outputFrame();
+		void dumpBuffer(const Source::Buffer& buffer);
+	  public:
+		HLE_DSP(Memory& mem, Scheduler& scheduler, DSPService& dspService);
+		~HLE_DSP() override {}
+
+		void reset() override;
+		void runAudioFrame() override;
+
+		u8* getDspMemory() override { return dspRam.rawMemory.data(); }
+
+		u16 recvData(u32 regId) override;
+		bool recvDataIsReady(u32 regId) override { return true; }  // Treat data as always ready
+		void writeProcessPipe(u32 channel, u32 size, u32 buffer) override;
+		std::vector<u8> readPipe(u32 channel, u32 peer, u32 size, u32 buffer) override;
+
+		void loadComponent(std::vector<u8>& data, u32 programMask, u32 dataMask) override;
+		void unloadComponent() override;
+		void setSemaphore(u16 value) override {}
+		void setSemaphoreMask(u16 value) override {}
+	};
+
+}  // namespace Audio
\ No newline at end of file
diff --git a/include/memory.hpp b/include/memory.hpp
index 1b6e622c..33ccbae5 100644
--- a/include/memory.hpp
+++ b/include/memory.hpp
@@ -19,7 +19,10 @@ namespace PhysicalAddrs {
 		VRAM = 0x18000000,
 		VRAMEnd = VRAM + 0x005FFFFF,
 		FCRAM = 0x20000000,
-		FCRAMEnd = FCRAM + 0x07FFFFFF
+		FCRAMEnd = FCRAM + 0x07FFFFFF,
+
+		DSP_RAM = 0x1FF00000,
+		DSP_RAM_End = DSP_RAM + 0x0007FFFF
 	};
 }
 
diff --git a/src/core/audio/dsp_core.cpp b/src/core/audio/dsp_core.cpp
index e4162e93..01cee11e 100644
--- a/src/core/audio/dsp_core.cpp
+++ b/src/core/audio/dsp_core.cpp
@@ -1,18 +1,20 @@
 #include "audio/dsp_core.hpp"
 
-#include "audio/null_core.hpp"
-#include "audio/teakra_core.hpp"
-
 #include <algorithm>
 #include <cctype>
 #include <unordered_map>
 
+#include "audio/hle_core.hpp"
+#include "audio/null_core.hpp"
+#include "audio/teakra_core.hpp"
+
 std::unique_ptr<Audio::DSPCore> Audio::makeDSPCore(DSPCore::Type type, Memory& mem, Scheduler& scheduler, DSPService& dspService) {
 	std::unique_ptr<DSPCore> core;
 
 	switch (type) {
 		case DSPCore::Type::Null: core = std::make_unique<NullDSP>(mem, scheduler, dspService); break;
 		case DSPCore::Type::Teakra: core = std::make_unique<TeakraDSP>(mem, scheduler, dspService); break;
+		case DSPCore::Type::HLE: core = std::make_unique<HLE_DSP>(mem, scheduler, dspService); break;
 
 		default:
 			Helpers::warn("Invalid DSP core selected!");
@@ -29,10 +31,8 @@ Audio::DSPCore::Type Audio::DSPCore::typeFromString(std::string inString) {
 	std::transform(inString.begin(), inString.end(), inString.begin(), [](unsigned char c) { return std::tolower(c); });
 
 	static const std::unordered_map<std::string, Audio::DSPCore::Type> map = {
-		{"null", Audio::DSPCore::Type::Null},
-		{"none", Audio::DSPCore::Type::Null},
-		{"lle", Audio::DSPCore::Type::Teakra},
-		{"teakra", Audio::DSPCore::Type::Teakra},
+		{"null", Audio::DSPCore::Type::Null},     {"none", Audio::DSPCore::Type::Null}, {"lle", Audio::DSPCore::Type::Teakra},
+		{"teakra", Audio::DSPCore::Type::Teakra}, {"hle", Audio::DSPCore::Type::HLE},   {"fast", Audio::DSPCore::Type::HLE},
 	};
 
 	if (auto search = map.find(inString); search != map.end()) {
@@ -47,6 +47,7 @@ const char* Audio::DSPCore::typeToString(Audio::DSPCore::Type type) {
 	switch (type) {
 		case Audio::DSPCore::Type::Null: return "null";
 		case Audio::DSPCore::Type::Teakra: return "teakra";
+		case Audio::DSPCore::Type::HLE: return "hle";
 		default: return "invalid";
 	}
 }
diff --git a/src/core/audio/hle_core.cpp b/src/core/audio/hle_core.cpp
new file mode 100644
index 00000000..245894ce
--- /dev/null
+++ b/src/core/audio/hle_core.cpp
@@ -0,0 +1,311 @@
+#include "audio/hle_core.hpp"
+
+#include <thread>
+#include <utility>
+
+#include "services/dsp.hpp"
+
+namespace Audio {
+	namespace DSPPipeType {
+		enum : u32 {
+			Debug = 0,
+			DMA = 1,
+			Audio = 2,
+			Binary = 3,
+		};
+	}
+
+	HLE_DSP::HLE_DSP(Memory& mem, Scheduler& scheduler, DSPService& dspService) : DSPCore(mem, scheduler, dspService) {
+		// Set up source indices
+		for (int i = 0; i < sources.size(); i++) {
+			sources[i].index = i;
+		}
+	}
+
+	void HLE_DSP::resetAudioPipe() {
+#define DSPOffset(var) (0x8000 + offsetof(Audio::HLE::SharedMemory, var) / 2)
+
+		// These are DSP shared memory offsets for various variables
+		// https://www.3dbrew.org/wiki/DSP_Memory_Region
+		static constexpr std::array<u16, 16> responses = {
+			0x000F,                             // Number of responses
+			DSPOffset(frameCounter),            // Frame counter
+			DSPOffset(sourceConfigurations),    // Source configs
+			DSPOffset(sourceStatuses),          // Source statuses
+			DSPOffset(adpcmCoefficients),       // ADPCM coefficients
+			DSPOffset(dspConfiguration),        // DSP configs
+			DSPOffset(dspStatus),               // DSP status
+			DSPOffset(finalSamples),            // Final samples
+			DSPOffset(intermediateMixSamples),  // Intermediate mix samples
+			DSPOffset(compressor),              // Compressor
+			DSPOffset(dspDebug),                // Debug
+			DSPOffset(unknown10),               // ??
+			DSPOffset(unknown11),               // ??
+			DSPOffset(unknown12),               // ??
+			DSPOffset(unknown13),               // Surround sound biquad filter 1
+			DSPOffset(unknown14)                // Surround sound biquad filter 2
+		};
+#undef DSPOffset
+
+		std::vector<u8>& audioPipe = pipeData[DSPPipeType::Audio];
+		audioPipe.resize(responses.size() * sizeof(u16));
+
+		// Push back every response to the audio pipe
+		size_t index = 0;
+		for (auto e : responses) {
+			audioPipe[index++] = e & 0xff;
+			audioPipe[index++] = e >> 8;
+		}
+	}
+
+	void HLE_DSP::reset() {
+		dspState = DSPState::Off;
+		loaded = false;
+
+		// Initialize these to some sane defaults
+		sampleFormat = SampleFormat::ADPCM;
+		sourceType = SourceType::Stereo;
+
+		for (auto& e : pipeData) {
+			e.clear();
+		}
+
+		for (auto& source : sources) {
+			source.reset();
+		}
+
+		// Note: Reset audio pipe AFTER resetting all pipes, otherwise the new data will be yeeted
+		resetAudioPipe();
+	}
+
+	void HLE_DSP::loadComponent(std::vector<u8>& data, u32 programMask, u32 dataMask) {
+		if (loaded) {
+			Helpers::warn("Loading DSP component when already loaded");
+		}
+
+		loaded = true;
+		scheduler.addEvent(Scheduler::EventType::RunDSP, scheduler.currentTimestamp + Audio::cyclesPerFrame);
+	}
+
+	void HLE_DSP::unloadComponent() {
+		if (!loaded) {
+			Helpers::warn("Audio: unloadComponent called without a running program");
+		}
+
+		loaded = false;
+		scheduler.removeEvent(Scheduler::EventType::RunDSP);
+	}
+
+	void HLE_DSP::runAudioFrame() {
+		// Signal audio pipe when an audio frame is done
+		if (dspState == DSPState::On) [[likely]] {
+			dspService.triggerPipeEvent(DSPPipeType::Audio);
+		}
+
+		outputFrame();
+		scheduler.addEvent(Scheduler::EventType::RunDSP, scheduler.currentTimestamp + Audio::cyclesPerFrame);
+	}
+	
+	u16 HLE_DSP::recvData(u32 regId) {
+		if (regId != 0) {
+			Helpers::panic("Audio: invalid register in HLE frontend");
+		}
+
+		return dspState == DSPState::On;
+	}
+
+	void HLE_DSP::writeProcessPipe(u32 channel, u32 size, u32 buffer) {
+		enum class StateChange : u8 {
+			Initialize = 0,
+			Shutdown = 1,
+			Wakeup = 2,
+			Sleep = 3,
+		};
+
+		switch (channel) {
+			case DSPPipeType::Audio: {
+				if (size != 4) {
+					printf("Invalid size written to DSP Audio Pipe\n");
+					break;
+				}
+
+				// Get new state
+				const u8 state = mem.read8(buffer);
+				if (state > 3) {
+					log("WriteProcessPipe::Audio: Unknown state change type");
+				} else {
+					switch (static_cast<StateChange>(state)) {
+						case StateChange::Initialize:
+							// TODO: Other initialization stuff here
+							dspState = DSPState::On;
+							resetAudioPipe();
+							
+							dspService.triggerPipeEvent(DSPPipeType::Audio);
+							break;
+
+						case StateChange::Shutdown:
+							dspState = DSPState::Off;
+							break;
+
+						default: Helpers::panic("Unimplemented DSP audio pipe state change %d", state);
+					}
+				}
+				break;
+			}
+
+			case DSPPipeType::Binary:
+				Helpers::warn("Unimplemented write to binary pipe! Size: %d\n", size);
+
+				// This pipe and interrupt are normally used for requests like AAC decode
+				dspService.triggerPipeEvent(DSPPipeType::Binary);
+				break;
+
+			default: log("Audio::HLE_DSP: Wrote to unimplemented pipe %d\n", channel); break;
+		}
+	}
+
+	std::vector<u8> HLE_DSP::readPipe(u32 pipe, u32 peer, u32 size, u32 buffer) {
+		if (size & 1) Helpers::panic("Tried to read odd amount of bytes from DSP pipe");
+		if (pipe >= pipeCount || size > 0xffff) {
+			return {};
+		}
+
+		if (pipe != DSPPipeType::Audio) {
+			log("Reading from non-audio pipe! This might be broken, might need to check what pipe is being read from and implement writing to it\n");
+		}
+
+		std::vector<u8>& data = pipeData[pipe];
+		size = std::min<u32>(size, data.size());  // Clamp size to the maximum available data size
+
+		if (size == 0) {
+			return {};
+		}
+
+		// Return "size" bytes from the audio pipe and erase them
+		std::vector<u8> out(data.begin(), data.begin() + size);
+		data.erase(data.begin(), data.begin() + size);
+		return out;
+	}
+
+	void HLE_DSP::outputFrame() {
+		StereoFrame<s16> frame;
+		generateFrame(frame);
+
+		if (audioEnabled) {
+			// Wait until we've actually got room to push our frame
+			while (sampleBuffer.size() + 2 > sampleBuffer.Capacity()) {
+				std::this_thread::sleep_for(std::chrono::milliseconds{1});
+			}
+
+			sampleBuffer.push(frame.data(), frame.size());
+		}
+	}
+
+	void HLE_DSP::generateFrame(StereoFrame<s16>& frame) {
+		using namespace Audio::HLE;
+		SharedMemory& read = readRegion();
+		SharedMemory& write = writeRegion();
+
+		for (int i = 0; i < sourceCount; i++) {
+			// Update source configuration from the read region of shared memory
+			auto& config = read.sourceConfigurations.config[i];
+			auto& source = sources[i];
+			updateSourceConfig(source, config);
+
+			// Generate audio
+			if (source.enabled && !source.buffers.empty()) {
+				const auto& buffer = source.buffers.top();
+				const u8* data = getPointerPhys<u8>(buffer.paddr);
+
+				if (data != nullptr) {
+					// TODO
+				}
+			}
+
+			// Update write region of shared memory
+			auto& status = write.sourceStatuses.status[i];
+			status.isEnabled = source.enabled;
+			status.syncCount = source.syncCount;
+		}
+	}
+
+	void HLE_DSP::updateSourceConfig(Source& source, HLE::SourceConfiguration::Configuration& config) {
+		// Check if the any dirty bit is set, otherwise exit early
+		if (!config.dirtyRaw) {
+			return;
+		}
+
+		if (config.enableDirty) {
+			config.enableDirty = 0;
+			source.enabled = config.enable != 0;
+		}
+
+		if (config.syncCountDirty) {
+			config.syncCountDirty = 0;
+			source.syncCount = config.syncCount;
+		}
+
+		if (config.resetFlag) {
+			config.resetFlag = 0;
+			source.reset();
+		}
+
+		if (config.partialResetFlag) {
+			config.partialResetFlag = 0;
+			source.buffers = {};
+		}
+		
+		// TODO: Should we check bufferQueueDirty here too?
+		if (config.formatDirty || config.embeddedBufferDirty) {
+			sampleFormat = config.format;
+		}
+
+		if (config.monoOrStereoDirty || config.embeddedBufferDirty) {
+			sourceType = config.monoOrStereo;
+		}
+
+		if (config.embeddedBufferDirty) {
+			config.embeddedBufferDirty = 0;
+			if (s32(config.length) >= 0) [[likely]] {
+				// TODO: Add sample format and channel count
+				Source::Buffer buffer{
+					.paddr = config.physicalAddress,
+					.sampleCount = config.length,
+					.adpcmScale = u8(config.adpcm_ps),
+					.previousSamples = {s16(config.adpcm_yn[0]), s16(config.adpcm_yn[1])},
+					.adpcmDirty = config.adpcmDirty != 0,
+					.looping = config.isLooping != 0,
+					.bufferID = config.bufferID,
+					.playPosition = config.playPosition,
+					.format = sampleFormat,
+					.sourceType = sourceType,
+					.fromQueue = false,
+					.hasPlayedOnce = false,
+				};
+
+				source.buffers.emplace(std::move(buffer));
+			} else {
+				log("Invalid embedded buffer size for DSP voice %d\n", source.index);
+			}
+		}
+
+		if (config.partialEmbeddedBufferDirty) {
+			config.partialEmbeddedBufferDirty = 0;
+			printf("Partial embedded buffer dirty for voice %d\n", source.index);
+		}
+
+		if (config.bufferQueueDirty) {
+			config.bufferQueueDirty = 0;
+			printf("Buffer queue dirty for voice %d\n", source.index);
+		}
+
+		config.dirtyRaw = 0;
+	}
+
+	void DSPSource::reset() {
+		enabled = false;
+		syncCount = 0;
+
+		buffers = {};
+	}
+}  // namespace Audio