audio_core: Remove global state

src/audio_core/CMakeLists.txt

@@ -1,17 +1,17 @@
 add_library(audio_core STATIC
-    audio_core.cpp
-    audio_core.h
+    audio_types.h
     codec.cpp
     codec.h
+    dsp_interface.cpp
+    dsp_interface.h
     hle/common.h
-    hle/dsp.cpp
-    hle/dsp.h
     hle/filter.cpp
     hle/filter.h
+    hle/hle.cpp
+    hle/hle.h
     hle/mixers.cpp
     hle/mixers.h
-    hle/pipe.cpp
-    hle/pipe.h
+    hle/shared_memory.h
     hle/source.cpp
     hle/source.h
     interpolate.cpp

src/audio_core/audio_core.cpp

@@ -1,61 +0,0 @@
-// Copyright 2016 Citra Emulator Project
-// Licensed under GPLv2 or any later version
-// Refer to the license.txt file included.
-
-#include <array>
-#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 "common/common_types.h"
-#include "core/core_timing.h"
-#include "core/hle/service/dsp_dsp.h"
-
-namespace AudioCore {
-
-// Audio Ticks occur about every 5 miliseconds.
-static CoreTiming::EventType* tick_event;            ///< CoreTiming event
-static constexpr u64 audio_frame_ticks = 1310252ull; ///< Units: ARM11 cycles
-
-static void AudioTickCallback(u64 /*userdata*/, int cycles_late) {
-    if (DSP::HLE::Tick()) {
-        // TODO(merry): Signal all the other interrupts as appropriate.
-        Service::DSP_DSP::SignalPipeInterrupt(DSP::HLE::DspPipe::Audio);
-        // HACK(merry): Added to prevent regressions. Will remove soon.
-        Service::DSP_DSP::SignalPipeInterrupt(DSP::HLE::DspPipe::Binary);
-    }
-
-    // Reschedule recurrent event
-    CoreTiming::ScheduleEvent(audio_frame_ticks - cycles_late, tick_event);
-}
-
-void Init() {
-    DSP::HLE::Init();
-
-    tick_event = CoreTiming::RegisterEvent("AudioCore::tick_event", AudioTickCallback);
-    CoreTiming::ScheduleEvent(audio_frame_ticks, tick_event);
-}
-
-std::array<u8, Memory::DSP_RAM_SIZE>& GetDspMemory() {
-    return DSP::HLE::g_dsp_memory.raw_memory;
-}
-
-void SelectSink(std::string sink_id) {
-    const SinkDetails& sink_details = GetSinkDetails(sink_id);
-    DSP::HLE::SetSink(sink_details.factory());
-}
-
-void EnableStretching(bool enable) {
-    DSP::HLE::EnableStretching(enable);
-}
-
-void Shutdown() {
-    CoreTiming::UnscheduleEvent(tick_event, 0);
-    DSP::HLE::Shutdown();
-}
-
-} // namespace AudioCore

src/audio_core/audio_core.h

@@ -1,31 +0,0 @@
-// Copyright 2016 Citra Emulator Project
-// Licensed under GPLv2 or any later version
-// Refer to the license.txt file included.
-
-#pragma once
-
-#include <array>
-#include <string>
-#include "common/common_types.h"
-#include "core/memory.h"
-
-namespace AudioCore {
-
-constexpr int native_sample_rate = 32728; ///< 32kHz
-
-/// Initialise Audio Core
-void Init();
-
-/// Returns a reference to the array backing DSP memory
-std::array<u8, Memory::DSP_RAM_SIZE>& GetDspMemory();
-
-/// Select the sink to use based on sink id.
-void SelectSink(std::string sink_id);
-
-/// Enable/Disable stretching.
-void EnableStretching(bool enable);
-
-/// Shutdown Audio Core
-void Shutdown();
-
-} // namespace AudioCore

src/audio_core/audio_types.h

@@ -0,0 +1,43 @@
+// Copyright 2017 Citra Emulator Project
+// Licensed under GPLv2 or any later version
+// Refer to the license.txt file included.
+
+#pragma once
+
+#include <array>
+#include <cstddef>
+#include <deque>
+#include "common/common_types.h"
+
+namespace AudioCore {
+
+/// Samples per second which the 3DS's audio hardware natively outputs at
+constexpr int native_sample_rate = 32728; // Hz
+
+/// Samples per audio frame at native sample rate
+constexpr int samples_per_frame = 160;
+
+/// The final output to the speakers is stereo. Preprocessing output in Source is also stereo.
+using StereoFrame16 = std::array<std::array<s16, 2>, samples_per_frame>;
+
+/// The DSP is quadraphonic internally.
+using QuadFrame32 = std::array<std::array<s32, 4>, samples_per_frame>;
+
+/// A variable length buffer of signed PCM16 stereo samples.
+using StereoBuffer16 = std::deque<std::array<s16, 2>>;
+
+constexpr size_t num_dsp_pipe = 8;
+enum class DspPipe {
+    Debug = 0,
+    Dma = 1,
+    Audio = 2,
+    Binary = 3,
+};
+
+enum class DspState {
+    Off,
+    On,
+    Sleeping,
+};
+
+} // namespace AudioCore

src/audio_core/codec.cpp

@@ -5,12 +5,13 @@
 #include <array>
 #include <cstddef>
 #include <cstring>
-#include <vector>
+#include "audio_core/audio_types.h"
 #include "audio_core/codec.h"
 #include "common/assert.h"
 #include "common/common_types.h"
 #include "common/math_util.h"
 
+namespace AudioCore {
 namespace Codec {
 
 StereoBuffer16 DecodeADPCM(const u8* const data, const size_t sample_count,
@@ -124,4 +125,5 @@ StereoBuffer16 DecodePCM16(const unsigned num_channels, const u8* const data,
 
     return ret;
 }
-};
+} // namespace Codec
+} // namespace AudioCore

src/audio_core/codec.h

@@ -5,14 +5,12 @@
 #pragma once
 
 #include <array>
-#include <deque>
+#include "audio_core/audio_types.h"
 #include "common/common_types.h"
 
+namespace AudioCore {
 namespace Codec {
 
-/// A variable length buffer of signed PCM16 stereo samples.
-using StereoBuffer16 = std::deque<std::array<s16, 2>>;
-
 /// See: Codec::DecodeADPCM
 struct ADPCMState {
     // Two historical samples from previous processed buffer,
@@ -48,4 +46,5 @@ StereoBuffer16 DecodePCM8(const unsigned num_channels, const u8* const data,
  */
 StereoBuffer16 DecodePCM16(const unsigned num_channels, const u8* const data,
                            const size_t sample_count);
-};
+} // namespace Codec
+} // namespace AudioCore

src/audio_core/dsp_interface.cpp

@@ -0,0 +1,75 @@
+// Copyright 2017 Citra Emulator Project
+// Licensed under GPLv2 or any later version
+// Refer to the license.txt file included.
+
+#include <cstddef>
+#include "audio_core/dsp_interface.h"
+#include "audio_core/sink.h"
+#include "audio_core/sink_details.h"
+#include "common/assert.h"
+
+namespace AudioCore {
+
+DspInterface::DspInterface() = default;
+
+DspInterface::~DspInterface() {
+    if (perform_time_stretching) {
+        FlushResidualStretcherAudio();
+    }
+}
+
+void DspInterface::SetSink(const std::string& sink_id) {
+    const SinkDetails& sink_details = GetSinkDetails(sink_id);
+    sink = sink_details.factory();
+    time_stretcher.SetOutputSampleRate(sink->GetNativeSampleRate());
+}
+
+Sink& DspInterface::GetSink() {
+    ASSERT(sink);
+    return *sink.get();
+}
+
+void DspInterface::EnableStretching(bool enable) {
+    if (perform_time_stretching == enable)
+        return;
+
+    if (!enable) {
+        FlushResidualStretcherAudio();
+    }
+    perform_time_stretching = enable;
+}
+
+void DspInterface::OutputFrame(const StereoFrame16& frame) {
+    if (!sink)
+        return;
+
+    if (perform_time_stretching) {
+        time_stretcher.AddSamples(&frame[0][0], frame.size());
+        std::vector<s16> stretched_samples = time_stretcher.Process(sink->SamplesInQueue());
+        sink->EnqueueSamples(stretched_samples.data(), stretched_samples.size() / 2);
+    } else {
+        constexpr size_t maximum_sample_latency = 2048; // about 64 miliseconds
+        if (sink->SamplesInQueue() > maximum_sample_latency) {
+            // This can occur if we're running too fast and samples are starting to back up.
+            // Just drop the samples.
+            return;
+        }
+
+        sink->EnqueueSamples(&frame[0][0], frame.size());
+    }
+}
+
+void DspInterface::FlushResidualStretcherAudio() {
+    if (!sink)
+        return;
+
+    time_stretcher.Flush();
+    while (true) {
+        std::vector<s16> residual_audio = time_stretcher.Process(sink->SamplesInQueue());
+        if (residual_audio.empty())
+            break;
+        sink->EnqueueSamples(residual_audio.data(), residual_audio.size() / 2);
+    }
+}
+
+} // namespace AudioCore

src/audio_core/dsp_interface.h

@@ -0,0 +1,81 @@
+// Copyright 2017 Citra Emulator Project
+// Licensed under GPLv2 or any later version
+// Refer to the license.txt file included.
+
+#pragma once
+
+#include <memory>
+#include <vector>
+#include "audio_core/audio_types.h"
+#include "audio_core/time_stretch.h"
+#include "common/common_types.h"
+#include "core/memory.h"
+
+namespace AudioCore {
+
+class Sink;
+
+class DspInterface {
+public:
+    DspInterface();
+    virtual ~DspInterface();
+
+    DspInterface(const DspInterface&) = delete;
+    DspInterface(DspInterface&&) = delete;
+    DspInterface& operator=(const DspInterface&) = delete;
+    DspInterface& operator=(DspInterface&&) = delete;
+
+    /// Get the state of the DSP
+    virtual DspState GetDspState() const = 0;
+
+    /**
+     * 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.
+     */
+    virtual std::vector<u8> PipeRead(DspPipe pipe_number, u32 length) = 0;
+
+    /**
+     * How much data is left in pipe
+     * @param pipe_number The Pipe ID
+     * @return The amount of data remaning in the pipe. This is the maximum length PipeRead will
+     * return.
+     */
+    virtual size_t GetPipeReadableSize(DspPipe pipe_number) const = 0;
+
+    /**
+     * Write to a DSP pipe.
+     * @param pipe_number The Pipe ID
+     * @param buffer The data to write to the pipe.
+     */
+    virtual void PipeWrite(DspPipe pipe_number, const std::vector<u8>& buffer) = 0;
+
+    /// Returns a reference to the array backing DSP memory
+    virtual std::array<u8, Memory::DSP_RAM_SIZE>& GetDspMemory() = 0;
+
+    /// Select the sink to use based on sink id.
+    void SetSink(const std::string& sink_id);
+    /// Get the current sink
+    Sink& GetSink();
+    /// Enable/Disable audio stretching.
+    void EnableStretching(bool enable);
+
+protected:
+    void OutputFrame(const StereoFrame16& frame);
+
+private:
+    void FlushResidualStretcherAudio();
+
+    std::unique_ptr<Sink> sink;
+    bool perform_time_stretching = false;
+    TimeStretcher time_stretcher;
+};
+
+} // namespace AudioCore

src/audio_core/hle/common.h

@@ -5,20 +5,12 @@
 #pragma once
 
 #include <algorithm>
-#include <array>
-#include "common/common_types.h"
+#include <cstddef>
 
-namespace DSP {
+namespace AudioCore {
 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>, samples_per_frame>;
-
-/// The DSP is quadraphonic internally.
-using QuadFrame32 = std::array<std::array<s32, 4>, samples_per_frame>;
+constexpr size_t num_sources = 24;
 
 /**
  * This performs the filter operation defined by FilterT::ProcessSample on the frame in-place.
@@ -31,4 +23,4 @@ void FilterFrame(FrameT& frame, FilterT& filter) {
 }
 
 } // namespace HLE
-} // namespace DSP
+} // namespace AudioCore

src/audio_core/hle/dsp.cpp

@@ -1,172 +0,0 @@
-// Copyright 2016 Citra Emulator Project
-// 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/mixers.h"
-#include "audio_core/hle/pipe.h"
-#include "audio_core/hle/source.h"
-#include "audio_core/sink.h"
-#include "audio_core/time_stretch.h"
-
-namespace DSP {
-namespace HLE {
-
-// Region management
-
-DspMemory g_dsp_memory;
-
-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.
-    u16 frame_counter_0 = g_dsp_memory.region_0.frame_counter;
-    u16 frame_counter_1 = g_dsp_memory.region_1.frame_counter;
-
-    if (frame_counter_0 == 0xFFFFu && frame_counter_1 != 0xFFFEu) {
-        // Wraparound has occurred.
-        return 1;
-    }
-
-    if (frame_counter_1 == 0xFFFFu && frame_counter_0 != 0xFFFEu) {
-        // Wraparound has occurred.
-        return 0;
-    }
-
-    return (frame_counter_0 > frame_counter_1) ? 0 : 1;
-}
-
-static SharedMemory& ReadRegion() {
-    return CurrentRegionIndex() == 0 ? g_dsp_memory.region_0 : g_dsp_memory.region_1;
-}
-
-static SharedMemory& WriteRegion() {
-    return CurrentRegionIndex() != 0 ? g_dsp_memory.region_0 : g_dsp_memory.region_1;
-}
-
-// Audio processing and mixing
-
-static std::array<Source, num_sources> sources = {
-    Source(0),  Source(1),  Source(2),  Source(3),  Source(4),  Source(5),  Source(6),  Source(7),
-    Source(8),  Source(9),  Source(10), Source(11), Source(12), Source(13), Source(14), Source(15),
-    Source(16), Source(17), Source(18), Source(19), Source(20), Source(21), Source(22), Source(23),
-};
-static Mixers mixers;
-
-static StereoFrame16 GenerateCurrentFrame() {
-    SharedMemory& read = ReadRegion();
-    SharedMemory& write = WriteRegion();
-
-    std::array<QuadFrame32, 3> intermediate_mixes = {};
-
-    // Generate intermediate mixes
-    for (size_t i = 0; i < num_sources; i++) {
-        write.source_statuses.status[i] =
-            sources[i].Tick(read.source_configurations.config[i], read.adpcm_coefficients.coeff[i]);
-        for (size_t mix = 0; mix < 3; mix++) {
-            sources[i].MixInto(intermediate_mixes[mix], mix);
-        }
-    }
-
-    // Generate final mix
-    write.dsp_status = mixers.Tick(read.dsp_configuration, read.intermediate_mix_samples,
-                                   write.intermediate_mix_samples, intermediate_mixes);
-
-    StereoFrame16 output_frame = mixers.GetOutput();
-
-    // Write current output frame to the shared memory region
-    for (size_t samplei = 0; samplei < output_frame.size(); samplei++) {
-        for (size_t channeli = 0; channeli < output_frame[0].size(); channeli++) {
-            write.final_samples.pcm16[samplei][channeli] = s16_le(output_frame[samplei][channeli]);
-        }
-    }
-
-    return output_frame;
-}
-
-// Audio output
-
-static bool perform_time_stretching = true;
-static std::unique_ptr<AudioCore::Sink> sink;
-static AudioCore::TimeStretcher time_stretcher;
-
-static void FlushResidualStretcherAudio() {
-    time_stretcher.Flush();
-    while (true) {
-        std::vector<s16> residual_audio = time_stretcher.Process(sink->SamplesInQueue());
-        if (residual_audio.empty())
-            break;
-        sink->EnqueueSamples(residual_audio.data(), residual_audio.size() / 2);
-    }
-}
-
-static void OutputCurrentFrame(const StereoFrame16& frame) {
-    if (perform_time_stretching) {
-        time_stretcher.AddSamples(&frame[0][0], frame.size());
-        std::vector<s16> stretched_samples = time_stretcher.Process(sink->SamplesInQueue());
-        sink->EnqueueSamples(stretched_samples.data(), stretched_samples.size() / 2);
-    } else {
-        constexpr size_t maximum_sample_latency = 2048; // about 64 miliseconds
-        if (sink->SamplesInQueue() > maximum_sample_latency) {
-            // This can occur if we're running too fast and samples are starting to back up.
-            // Just drop the samples.
-            return;
-        }
-
-        sink->EnqueueSamples(&frame[0][0], frame.size());
-    }
-}
-
-void EnableStretching(bool enable) {
-    if (perform_time_stretching == enable)
-        return;
-
-    if (!enable) {
-        FlushResidualStretcherAudio();
-    }
-    perform_time_stretching = enable;
-}
-
-// Public Interface
-
-void Init() {
-    DSP::HLE::ResetPipes();
-
-    for (auto& source : sources) {
-        source.Reset();
-    }
-
-    mixers.Reset();
-
-    time_stretcher.Reset();
-    if (sink) {
-        time_stretcher.SetOutputSampleRate(sink->GetNativeSampleRate());
-    }
-}
-
-void Shutdown() {
-    if (perform_time_stretching) {
-        FlushResidualStretcherAudio();
-    }
-}
-
-bool Tick() {
-    StereoFrame16 current_frame = {};
-
-    // TODO: Check dsp::DSP semaphore (which indicates emulated application has finished writing to
-    // shared memory region)
-    current_frame = GenerateCurrentFrame();
-
-    OutputCurrentFrame(current_frame);
-
-    return true;
-}
-
-void SetSink(std::unique_ptr<AudioCore::Sink> sink_) {
-    sink = std::move(sink_);
-    time_stretcher.SetOutputSampleRate(sink->GetNativeSampleRate());
-}
-
-} // namespace HLE
-} // namespace DSP

src/audio_core/hle/filter.cpp

@@ -5,12 +5,12 @@
 #include <array>
 #include <cstddef>
 #include "audio_core/hle/common.h"
-#include "audio_core/hle/dsp.h"
 #include "audio_core/hle/filter.h"
+#include "audio_core/hle/shared_memory.h"
 #include "common/common_types.h"
 #include "common/math_util.h"
 
-namespace DSP {
+namespace AudioCore {
 namespace HLE {
 
 void SourceFilters::Reset() {
@@ -114,4 +114,4 @@ std::array<s16, 2> SourceFilters::BiquadFilter::ProcessSample(const std::array<s
 }
 
 } // namespace HLE
-} // namespace DSP
+} // namespace AudioCore

src/audio_core/hle/filter.h

@@ -5,11 +5,11 @@
 #pragma once
 
 #include <array>
-#include "audio_core/hle/common.h"
-#include "audio_core/hle/dsp.h"
+#include "audio_core/audio_types.h"
+#include "audio_core/hle/shared_memory.h"
 #include "common/common_types.h"
 
-namespace DSP {
+namespace AudioCore {
 namespace HLE {
 
 /// Preprocessing filters. There is an independent set of filters for each Source.
@@ -114,4 +114,4 @@ private:
 };
 
 } // namespace HLE
-} // namespace DSP
+} // namespace AudioCore

src/audio_core/hle/hle.cpp

@@ -0,0 +1,341 @@
+// Copyright 2017 Citra Emulator Project
+// Licensed under GPLv2 or any later version
+// Refer to the license.txt file included.
+
+#include "audio_core/audio_types.h"
+#include "audio_core/hle/common.h"
+#include "audio_core/hle/hle.h"
+#include "audio_core/hle/mixers.h"
+#include "audio_core/hle/shared_memory.h"
+#include "audio_core/hle/source.h"
+#include "audio_core/sink.h"
+#include "common/assert.h"
+#include "common/common_types.h"
+#include "common/logging/log.h"
+#include "core/core_timing.h"
+#include "core/hle/service/dsp_dsp.h"
+
+namespace AudioCore {
+
+static constexpr u64 audio_frame_ticks = 1310252ull; ///< Units: ARM11 cycles
+
+struct DspHle::Impl final {
+public:
+    explicit Impl(DspHle& parent);
+    ~Impl();
+
+    DspState GetDspState() const;
+
+    std::vector<u8> PipeRead(DspPipe pipe_number, u32 length);
+    size_t GetPipeReadableSize(DspPipe pipe_number) const;
+    void PipeWrite(DspPipe pipe_number, const std::vector<u8>& buffer);
+
+    std::array<u8, Memory::DSP_RAM_SIZE>& GetDspMemory();
+
+private:
+    void ResetPipes();
+    void WriteU16(DspPipe pipe_number, u16 value);
+    void AudioPipeWriteStructAddresses();
+
+    size_t CurrentRegionIndex() const;
+    HLE::SharedMemory& ReadRegion();
+    HLE::SharedMemory& WriteRegion();
+
+    StereoFrame16 GenerateCurrentFrame();
+    bool Tick();
+    void AudioTickCallback(int cycles_late);
+
+    DspState dsp_state = DspState::Off;
+    std::array<std::vector<u8>, num_dsp_pipe> pipe_data;
+
+    HLE::DspMemory dsp_memory;
+    std::array<HLE::Source, HLE::num_sources> sources{{
+        HLE::Source(0),  HLE::Source(1),  HLE::Source(2),  HLE::Source(3),  HLE::Source(4),
+        HLE::Source(5),  HLE::Source(6),  HLE::Source(7),  HLE::Source(8),  HLE::Source(9),
+        HLE::Source(10), HLE::Source(11), HLE::Source(12), HLE::Source(13), HLE::Source(14),
+        HLE::Source(15), HLE::Source(16), HLE::Source(17), HLE::Source(18), HLE::Source(19),
+        HLE::Source(20), HLE::Source(21), HLE::Source(22), HLE::Source(23),
+    }};
+    HLE::Mixers mixers;
+
+    DspHle& parent;
+    CoreTiming::EventType* tick_event;
+};
+
+DspHle::Impl::Impl(DspHle& parent_) : parent(parent_) {
+    tick_event =
+        CoreTiming::RegisterEvent("AudioCore::DspHle::tick_event", [this](u64, int cycles_late) {
+            this->AudioTickCallback(cycles_late);
+        });
+    CoreTiming::ScheduleEvent(audio_frame_ticks, tick_event);
+}
+
+DspHle::Impl::~Impl() {
+    CoreTiming::UnscheduleEvent(tick_event, 0);
+}
+
+DspState DspHle::Impl::GetDspState() const {
+    return dsp_state;
+}
+
+std::vector<u8> DspHle::Impl::PipeRead(DspPipe pipe_number, u32 length) {
+    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 {};
+    }
+
+    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 = %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)
+        return {};
+
+    std::vector<u8> ret(data.begin(), data.begin() + length);
+    data.erase(data.begin(), data.begin() + length);
+    return ret;
+}
+
+size_t DspHle::Impl::GetPipeReadableSize(DspPipe pipe_number) const {
+    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[pipe_index].size();
+}
+
+void DspHle::Impl::PipeWrite(DspPipe pipe_number, const std::vector<u8>& buffer) {
+    switch (pipe_number) {
+    case DspPipe::Audio: {
+        if (buffer.size() != 4) {
+            LOG_ERROR(Audio_DSP, "DspPipe::Audio: Unexpected buffer length %zu was written",
+                      buffer.size());
+            return;
+        }
+
+        enum class StateChange {
+            Initialize = 0,
+            Shutdown = 1,
+            Wakeup = 2,
+            Sleep = 3,
+        };
+
+        // The difference between Initialize and Wakeup is that Input state is maintained
+        // when sleeping but isn't when turning it off and on again. (TODO: Implement this.)
+        // Waking up from sleep garbles some of the structs in the memory region. (TODO:
+        // Implement this.) Applications store away the state of these structs before
+        // sleeping and reset it back after wakeup on behalf of the DSP.
+
+        switch (static_cast<StateChange>(buffer[0])) {
+        case StateChange::Initialize:
+            LOG_INFO(Audio_DSP, "Application has requested initialization of DSP hardware");
+            ResetPipes();
+            AudioPipeWriteStructAddresses();
+            dsp_state = DspState::On;
+            break;
+        case StateChange::Shutdown:
+            LOG_INFO(Audio_DSP, "Application has requested shutdown of DSP hardware");
+            dsp_state = DspState::Off;
+            break;
+        case StateChange::Wakeup:
+            LOG_INFO(Audio_DSP, "Application has requested wakeup of DSP hardware");
+            ResetPipes();
+            AudioPipeWriteStructAddresses();
+            dsp_state = DspState::On;
+            break;
+        case StateChange::Sleep:
+            LOG_INFO(Audio_DSP, "Application has requested sleep of DSP hardware");
+            UNIMPLEMENTED();
+            dsp_state = DspState::Sleeping;
+            break;
+        default:
+            LOG_ERROR(Audio_DSP,
+                      "Application has requested unknown state transition of DSP hardware %hhu",
+                      buffer[0]);
+            dsp_state = DspState::Off;
+            break;
+        }
+
+        return;
+    }
+    default:
+        LOG_CRITICAL(Audio_DSP, "pipe_number = %zu unimplemented",
+                     static_cast<size_t>(pipe_number));
+        UNIMPLEMENTED();
+        return;
+    }
+}
+
+std::array<u8, Memory::DSP_RAM_SIZE>& DspHle::Impl::GetDspMemory() {
+    return dsp_memory.raw_memory;
+}
+
+void DspHle::Impl::ResetPipes() {
+    for (auto& data : pipe_data) {
+        data.clear();
+    }
+    dsp_state = DspState::Off;
+}
+
+void DspHle::Impl::WriteU16(DspPipe pipe_number, u16 value) {
+    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);
+}
+
+void DspHle::Impl::AudioPipeWriteStructAddresses() {
+    // These struct addresses are DSP dram addresses.
+    // See also: DSP_DSP::ConvertProcessAddressFromDspDram
+    static const std::array<u16, 15> struct_addresses = {
+        0x8000 + offsetof(HLE::SharedMemory, frame_counter) / 2,
+        0x8000 + offsetof(HLE::SharedMemory, source_configurations) / 2,
+        0x8000 + offsetof(HLE::SharedMemory, source_statuses) / 2,
+        0x8000 + offsetof(HLE::SharedMemory, adpcm_coefficients) / 2,
+        0x8000 + offsetof(HLE::SharedMemory, dsp_configuration) / 2,
+        0x8000 + offsetof(HLE::SharedMemory, dsp_status) / 2,
+        0x8000 + offsetof(HLE::SharedMemory, final_samples) / 2,
+        0x8000 + offsetof(HLE::SharedMemory, intermediate_mix_samples) / 2,
+        0x8000 + offsetof(HLE::SharedMemory, compressor) / 2,
+        0x8000 + offsetof(HLE::SharedMemory, dsp_debug) / 2,
+        0x8000 + offsetof(HLE::SharedMemory, unknown10) / 2,
+        0x8000 + offsetof(HLE::SharedMemory, unknown11) / 2,
+        0x8000 + offsetof(HLE::SharedMemory, unknown12) / 2,
+        0x8000 + offsetof(HLE::SharedMemory, unknown13) / 2,
+        0x8000 + offsetof(HLE::SharedMemory, unknown14) / 2,
+    };
+
+    // Begin with a u16 denoting the number of structs.
+    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.
+    Service::DSP_DSP::SignalPipeInterrupt(DspPipe::Audio);
+}
+
+size_t DspHle::Impl::CurrentRegionIndex() const {
+    // The region with the higher frame counter is chosen unless there is wraparound.
+    // This function only returns a 0 or 1.
+    const u16 frame_counter_0 = dsp_memory.region_0.frame_counter;
+    const u16 frame_counter_1 = dsp_memory.region_1.frame_counter;
+
+    if (frame_counter_0 == 0xFFFFu && frame_counter_1 != 0xFFFEu) {
+        // Wraparound has occurred.
+        return 1;
+    }
+
+    if (frame_counter_1 == 0xFFFFu && frame_counter_0 != 0xFFFEu) {
+        // Wraparound has occurred.
+        return 0;
+    }
+
+    return (frame_counter_0 > frame_counter_1) ? 0 : 1;
+}
+
+HLE::SharedMemory& DspHle::Impl::ReadRegion() {
+    return CurrentRegionIndex() == 0 ? dsp_memory.region_0 : dsp_memory.region_1;
+}
+
+HLE::SharedMemory& DspHle::Impl::WriteRegion() {
+    return CurrentRegionIndex() != 0 ? dsp_memory.region_0 : dsp_memory.region_1;
+}
+
+StereoFrame16 DspHle::Impl::GenerateCurrentFrame() {
+    HLE::SharedMemory& read = ReadRegion();
+    HLE::SharedMemory& write = WriteRegion();
+
+    std::array<QuadFrame32, 3> intermediate_mixes = {};
+
+    // Generate intermediate mixes
+    for (size_t i = 0; i < HLE::num_sources; i++) {
+        write.source_statuses.status[i] =
+            sources[i].Tick(read.source_configurations.config[i], read.adpcm_coefficients.coeff[i]);
+        for (size_t mix = 0; mix < 3; mix++) {
+            sources[i].MixInto(intermediate_mixes[mix], mix);
+        }
+    }
+
+    // Generate final mix
+    write.dsp_status = mixers.Tick(read.dsp_configuration, read.intermediate_mix_samples,
+                                   write.intermediate_mix_samples, intermediate_mixes);
+
+    StereoFrame16 output_frame = mixers.GetOutput();
+
+    // Write current output frame to the shared memory region
+    for (size_t samplei = 0; samplei < output_frame.size(); samplei++) {
+        for (size_t channeli = 0; channeli < output_frame[0].size(); channeli++) {
+            write.final_samples.pcm16[samplei][channeli] = s16_le(output_frame[samplei][channeli]);
+        }
+    }
+
+    return output_frame;
+}
+
+bool DspHle::Impl::Tick() {
+    StereoFrame16 current_frame = {};
+
+    // TODO: Check dsp::DSP semaphore (which indicates emulated application has finished writing to
+    // shared memory region)
+    current_frame = GenerateCurrentFrame();
+
+    parent.OutputFrame(current_frame);
+
+    return true;
+}
+
+void DspHle::Impl::AudioTickCallback(int cycles_late) {
+    if (Tick()) {
+        // TODO(merry): Signal all the other interrupts as appropriate.
+        Service::DSP_DSP::SignalPipeInterrupt(DspPipe::Audio);
+        // HACK(merry): Added to prevent regressions. Will remove soon.
+        Service::DSP_DSP::SignalPipeInterrupt(DspPipe::Binary);
+    }
+
+    // Reschedule recurrent event
+    CoreTiming::ScheduleEvent(audio_frame_ticks - cycles_late, tick_event);
+}
+
+DspHle::DspHle() : impl(std::make_unique<Impl>(*this)) {}
+DspHle::~DspHle() = default;
+
+DspState DspHle::GetDspState() const {
+    return impl->GetDspState();
+}
+
+std::vector<u8> DspHle::PipeRead(DspPipe pipe_number, u32 length) {
+    return impl->PipeRead(pipe_number, length);
+}
+
+size_t DspHle::GetPipeReadableSize(DspPipe pipe_number) const {
+    return impl->GetPipeReadableSize(pipe_number);
+}
+
+void DspHle::PipeWrite(DspPipe pipe_number, const std::vector<u8>& buffer) {
+    impl->PipeWrite(pipe_number, buffer);
+}
+
+std::array<u8, Memory::DSP_RAM_SIZE>& DspHle::GetDspMemory() {
+    return impl->GetDspMemory();
+}
+
+} // namespace AudioCore

src/audio_core/hle/hle.h

@@ -0,0 +1,36 @@
+// Copyright 2017 Citra Emulator Project
+// Licensed under GPLv2 or any later version
+// Refer to the license.txt file included.
+
+#pragma once
+
+#include <array>
+#include <memory>
+#include <vector>
+#include "audio_core/audio_types.h"
+#include "audio_core/dsp_interface.h"
+#include "common/common_types.h"
+#include "core/memory.h"
+
+namespace AudioCore {
+
+class DspHle final : public DspInterface {
+public:
+    DspHle();
+    ~DspHle();
+
+    DspState GetDspState() const override;
+
+    std::vector<u8> PipeRead(DspPipe pipe_number, u32 length) override;
+    size_t GetPipeReadableSize(DspPipe pipe_number) const override;
+    void PipeWrite(DspPipe pipe_number, const std::vector<u8>& buffer) override;
+
+    std::array<u8, Memory::DSP_RAM_SIZE>& GetDspMemory() override;
+
+private:
+    struct Impl;
+    friend struct Impl;
+    std::unique_ptr<Impl> impl;
+};
+
+} // namespace AudioCore

src/audio_core/hle/mixers.cpp

@@ -4,14 +4,12 @@
 
 #include <cstddef>
 
-#include "audio_core/hle/common.h"
-#include "audio_core/hle/dsp.h"
 #include "audio_core/hle/mixers.h"
 #include "common/assert.h"
 #include "common/logging/log.h"
 #include "common/math_util.h"
 
-namespace DSP {
+namespace AudioCore {
 namespace HLE {
 
 void Mixers::Reset() {
@@ -207,4 +205,4 @@ DspStatus Mixers::GetCurrentStatus() const {
 }
 
 } // namespace HLE
-} // namespace DSP
+} // namespace AudioCore

src/audio_core/hle/mixers.h

@@ -5,10 +5,10 @@
 #pragma once
 
 #include <array>
-#include "audio_core/hle/common.h"
-#include "audio_core/hle/dsp.h"
+#include "audio_core/audio_types.h"
+#include "audio_core/hle/shared_memory.h"
 
-namespace DSP {
+namespace AudioCore {
 namespace HLE {
 
 class Mixers final {
@@ -58,4 +58,4 @@ private:
 };
 
 } // namespace HLE
-} // namespace DSP
+} // namespace AudioCore

src/audio_core/hle/pipe.cpp

@@ -1,177 +0,0 @@
-// Copyright 2016 Citra Emulator Project
-// Licensed under GPLv2 or any later version
-// Refer to the license.txt file included.
-
-#include <array>
-#include <vector>
-#include "audio_core/hle/dsp.h"
-#include "audio_core/hle/pipe.h"
-#include "common/assert.h"
-#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>, NUM_DSP_PIPE> pipe_data;
-
-void ResetPipes() {
-    for (auto& data : pipe_data) {
-        data.clear();
-    }
-    dsp_state = DspState::Off;
-}
-
-std::vector<u8> PipeRead(DspPipe pipe_number, u32 length) {
-    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 {};
-    }
-
-    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 = %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)
-        return {};
-
-    std::vector<u8> ret(data.begin(), data.begin() + length);
-    data.erase(data.begin(), data.begin() + length);
-    return ret;
-}
-
-size_t GetPipeReadableSize(DspPipe 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[pipe_index].size();
-}
-
-static void WriteU16(DspPipe pipe_number, u16 value) {
-    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);
-}
-
-static void AudioPipeWriteStructAddresses() {
-    // These struct addresses are DSP dram addresses.
-    // See also: DSP_DSP::ConvertProcessAddressFromDspDram
-    static const std::array<u16, 15> struct_addresses = {
-        0x8000 + offsetof(SharedMemory, frame_counter) / 2,
-        0x8000 + offsetof(SharedMemory, source_configurations) / 2,
-        0x8000 + offsetof(SharedMemory, source_statuses) / 2,
-        0x8000 + offsetof(SharedMemory, adpcm_coefficients) / 2,
-        0x8000 + offsetof(SharedMemory, dsp_configuration) / 2,
-        0x8000 + offsetof(SharedMemory, dsp_status) / 2,
-        0x8000 + offsetof(SharedMemory, final_samples) / 2,
-        0x8000 + offsetof(SharedMemory, intermediate_mix_samples) / 2,
-        0x8000 + offsetof(SharedMemory, compressor) / 2,
-        0x8000 + offsetof(SharedMemory, dsp_debug) / 2,
-        0x8000 + offsetof(SharedMemory, unknown10) / 2,
-        0x8000 + offsetof(SharedMemory, unknown11) / 2,
-        0x8000 + offsetof(SharedMemory, unknown12) / 2,
-        0x8000 + offsetof(SharedMemory, unknown13) / 2,
-        0x8000 + offsetof(SharedMemory, unknown14) / 2,
-    };
-
-    // Begin with a u16 denoting the number of structs.
-    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.
-    Service::DSP_DSP::SignalPipeInterrupt(DspPipe::Audio);
-}
-
-void PipeWrite(DspPipe pipe_number, const std::vector<u8>& buffer) {
-    switch (pipe_number) {
-    case DspPipe::Audio: {
-        if (buffer.size() != 4) {
-            LOG_ERROR(Audio_DSP, "DspPipe::Audio: Unexpected buffer length %zu was written",
-                      buffer.size());
-            return;
-        }
-
-        enum class StateChange {
-            Initialize = 0,
-            Shutdown = 1,
-            Wakeup = 2,
-            Sleep = 3,
-        };
-
-        // The difference between Initialize and Wakeup is that Input state is maintained
-        // when sleeping but isn't when turning it off and on again. (TODO: Implement this.)
-        // Waking up from sleep garbles some of the structs in the memory region. (TODO:
-        // Implement this.) Applications store away the state of these structs before
-        // sleeping and reset it back after wakeup on behalf of the DSP.
-
-        switch (static_cast<StateChange>(buffer[0])) {
-        case StateChange::Initialize:
-            LOG_INFO(Audio_DSP, "Application has requested initialization of DSP hardware");
-            ResetPipes();
-            AudioPipeWriteStructAddresses();
-            dsp_state = DspState::On;
-            break;
-        case StateChange::Shutdown:
-            LOG_INFO(Audio_DSP, "Application has requested shutdown of DSP hardware");
-            dsp_state = DspState::Off;
-            break;
-        case StateChange::Wakeup:
-            LOG_INFO(Audio_DSP, "Application has requested wakeup of DSP hardware");
-            ResetPipes();
-            AudioPipeWriteStructAddresses();
-            dsp_state = DspState::On;
-            break;
-        case StateChange::Sleep:
-            LOG_INFO(Audio_DSP, "Application has requested sleep of DSP hardware");
-            UNIMPLEMENTED();
-            dsp_state = DspState::Sleeping;
-            break;
-        default:
-            LOG_ERROR(Audio_DSP,
-                      "Application has requested unknown state transition of DSP hardware %hhu",
-                      buffer[0]);
-            dsp_state = DspState::Off;
-            break;
-        }
-
-        return;
-    }
-    default:
-        LOG_CRITICAL(Audio_DSP, "pipe_number = %zu unimplemented",
-                     static_cast<size_t>(pipe_number));
-        UNIMPLEMENTED();
-        return;
-    }
-}
-
-DspState GetDspState() {
-    return dsp_state;
-}
-
-} // namespace HLE
-} // namespace DSP

src/audio_core/hle/pipe.h

@@ -1,63 +0,0 @@
-// Copyright 2016 Citra Emulator Project
-// Licensed under GPLv2 or any later version
-// Refer to the license.txt file included.
-
-#pragma once
-
-#include <cstddef>
-#include <vector>
-#include "common/common_types.h"
-
-namespace DSP {
-namespace HLE {
-
-/// Reset the pipes by setting pipe positions back to the beginning.
-void ResetPipes();
-
-enum class DspPipe {
-    Debug = 0,
-    Dma = 1,
-    Audio = 2,
-    Binary = 3,
-};
-constexpr size_t NUM_DSP_PIPE = 8;
-
-/**
- * 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);
-
-/**
- * How much data is left in pipe
- * @param pipe_number The Pipe ID
- * @return The amount of data remaning in the pipe. This is the maximum length PipeRead will return.
- */
-size_t GetPipeReadableSize(DspPipe pipe_number);
-
-/**
- * Write to a DSP pipe.
- * @param pipe_number The Pipe ID
- * @param buffer The data to write to the pipe.
- */
-void PipeWrite(DspPipe pipe_number, const std::vector<u8>& buffer);
-
-enum class DspState {
-    Off,
-    On,
-    Sleeping,
-};
-
-/// Get the state of the DSP
-DspState GetDspState();
-
-} // namespace HLE
-} // namespace DSP

src/audio_core/hle/shared_memory.h

@@ -8,6 +8,7 @@
 #include <cstddef>
 #include <memory>
 #include <type_traits>
+#include "audio_core/audio_types.h"
 #include "audio_core/hle/common.h"
 #include "common/bit_field.h"
 #include "common/common_funcs.h"
@@ -15,10 +16,6 @@
 #include "common/swap.h"
 
 namespace AudioCore {
-class Sink;
-}
-
-namespace DSP {
 namespace HLE {
 
 // The application-accessible region of DSP memory consists of two parts. Both are marked as IO and
@@ -86,7 +83,7 @@ static_assert(std::is_trivially_copyable<u32_dsp>::value, "u32_dsp isn't trivial
 //       0           0xBFFF                     Frame Counter                         Application
 //
 // #: This refers to the order in which they appear in the DspPipe::Audio DSP pipe.
-//    See also: DSP::HLE::PipeRead.
+//    See also: HLE::PipeRead.
 //
 // Note that the above addresses do vary slightly between audio firmwares observed; the addresses
 // are not fixed in stone. The addresses above are only an examplar; they're what this
@@ -527,69 +524,40 @@ static_assert(offsetof(DspMemory, region_0) == region0_offset,
 static_assert(offsetof(DspMemory, region_1) == region1_offset,
               "DSP region 1 is at the wrong offset");
 
-extern DspMemory g_dsp_memory;
-
 // 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");
+              "Structures in 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");
+              "Structures in HLE::SharedMemory must be 2-byte aligned");
 static_assert(offsetof(SharedMemory, source_statuses) % 2 == 0,
-              "Structures in DSP::HLE::SharedMemory must be 2-byte aligned");
+              "Structures in HLE::SharedMemory must be 2-byte aligned");
 static_assert(offsetof(SharedMemory, adpcm_coefficients) % 2 == 0,
-              "Structures in DSP::HLE::SharedMemory must be 2-byte aligned");
+              "Structures in HLE::SharedMemory must be 2-byte aligned");
 static_assert(offsetof(SharedMemory, dsp_configuration) % 2 == 0,
-              "Structures in DSP::HLE::SharedMemory must be 2-byte aligned");
+              "Structures in HLE::SharedMemory must be 2-byte aligned");
 static_assert(offsetof(SharedMemory, dsp_status) % 2 == 0,
-              "Structures in DSP::HLE::SharedMemory must be 2-byte aligned");
+              "Structures in HLE::SharedMemory must be 2-byte aligned");
 static_assert(offsetof(SharedMemory, final_samples) % 2 == 0,
-              "Structures in DSP::HLE::SharedMemory must be 2-byte aligned");
+              "Structures in HLE::SharedMemory must be 2-byte aligned");
 static_assert(offsetof(SharedMemory, intermediate_mix_samples) % 2 == 0,
-              "Structures in DSP::HLE::SharedMemory must be 2-byte aligned");
+              "Structures in HLE::SharedMemory must be 2-byte aligned");
 static_assert(offsetof(SharedMemory, compressor) % 2 == 0,
-              "Structures in DSP::HLE::SharedMemory must be 2-byte aligned");
+              "Structures in HLE::SharedMemory must be 2-byte aligned");
 static_assert(offsetof(SharedMemory, dsp_debug) % 2 == 0,
-              "Structures in DSP::HLE::SharedMemory must be 2-byte aligned");
+              "Structures in HLE::SharedMemory must be 2-byte aligned");
 static_assert(offsetof(SharedMemory, unknown10) % 2 == 0,
-              "Structures in DSP::HLE::SharedMemory must be 2-byte aligned");
+              "Structures in HLE::SharedMemory must be 2-byte aligned");
 static_assert(offsetof(SharedMemory, unknown11) % 2 == 0,
-              "Structures in DSP::HLE::SharedMemory must be 2-byte aligned");
+              "Structures in HLE::SharedMemory must be 2-byte aligned");
 static_assert(offsetof(SharedMemory, unknown12) % 2 == 0,
-              "Structures in DSP::HLE::SharedMemory must be 2-byte aligned");
+              "Structures in HLE::SharedMemory must be 2-byte aligned");
 static_assert(offsetof(SharedMemory, unknown13) % 2 == 0,
-              "Structures in DSP::HLE::SharedMemory must be 2-byte aligned");
+              "Structures in HLE::SharedMemory must be 2-byte aligned");
 static_assert(offsetof(SharedMemory, unknown14) % 2 == 0,
-              "Structures in DSP::HLE::SharedMemory must be 2-byte aligned");
+              "Structures in HLE::SharedMemory must be 2-byte aligned");
 
 #undef INSERT_PADDING_DSPWORDS
 #undef ASSERT_DSP_STRUCT
 
-/// Initialize DSP hardware
-void Init();
-
-/// Shutdown DSP hardware
-void Shutdown();
-
-/**
- * Perform processing and updates state of current shared memory buffer.
- * This function is called every audio tick before triggering the audio interrupt.
- * @return Whether an audio interrupt should be triggered this frame.
- */
-bool Tick();
-
-/**
- * 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);
-
-/**
- * Enables/Disables audio-stretching.
- * Audio stretching is an enhancement that stretches audio to match emulation
- * speed to prevent stuttering at the cost of some audio latency.
- * @param enable true to enable, false to disable.
- */
-void EnableStretching(bool enable);
-
 } // namespace HLE
-} // namespace DSP
+} // namespace AudioCore

src/audio_core/hle/source.cpp

@@ -12,7 +12,7 @@
 #include "common/logging/log.h"
 #include "core/memory.h"
 
-namespace DSP {
+namespace AudioCore {
 namespace HLE {
 
 SourceStatus::Status Source::Tick(SourceConfiguration::Configuration& config,
@@ -345,4 +345,4 @@ SourceStatus::Status Source::GetCurrentStatus() {
 }
 
 } // namespace HLE
-} // namespace DSP
+} // namespace AudioCore

src/audio_core/hle/source.h

@@ -7,14 +7,14 @@
 #include <array>
 #include <queue>
 #include <vector>
+#include "audio_core/audio_types.h"
 #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 AudioCore {
 namespace HLE {
 
 /**
@@ -146,4 +146,4 @@ private:
 };
 
 } // namespace HLE
-} // namespace DSP
+} // namespace AudioCore

src/audio_core/interpolate.cpp

@@ -6,6 +6,7 @@
 #include "common/assert.h"
 #include "common/math_util.h"
 
+namespace AudioCore {
 namespace AudioInterp {
 
 // Calculations are done in fixed point with 24 fractional bits.
@@ -16,8 +17,8 @@ constexpr u64 scale_mask = scale_factor - 1;
 /// Here we step over the input in steps of rate, until we consume all of the input.
 /// Three adjacent samples are passed to fn each step.
 template <typename Function>
-static void StepOverSamples(State& state, StereoBuffer16& input, float rate,
-                            DSP::HLE::StereoFrame16& output, size_t& outputi, Function fn) {
+static void StepOverSamples(State& state, StereoBuffer16& input, float rate, StereoFrame16& output,
+                            size_t& outputi, Function fn) {
     ASSERT(rate > 0);
 
     if (input.empty())
@@ -50,14 +51,13 @@ static void StepOverSamples(State& state, StereoBuffer16& input, float rate,
     input.erase(input.begin(), std::next(input.begin(), inputi + 2));
 }
 
-void None(State& state, StereoBuffer16& input, float rate, DSP::HLE::StereoFrame16& output,
-          size_t& outputi) {
+void None(State& state, StereoBuffer16& input, float rate, StereoFrame16& output, size_t& outputi) {
     StepOverSamples(
         state, input, rate, output, outputi,
         [](u64 fraction, const auto& x0, const auto& x1, const auto& x2) { return x0; });
 }
 
-void Linear(State& state, StereoBuffer16& input, float rate, DSP::HLE::StereoFrame16& output,
+void Linear(State& state, StereoBuffer16& input, float rate, StereoFrame16& output,
             size_t& outputi) {
     // Note on accuracy: Some values that this produces are +/- 1 from the actual firmware.
     StepOverSamples(state, input, rate, output, outputi,
@@ -74,3 +74,4 @@ void Linear(State& state, StereoBuffer16& input, float rate, DSP::HLE::StereoFra
 }
 
 } // namespace AudioInterp
+} // namespace AudioCore

src/audio_core/interpolate.h

@@ -6,9 +6,10 @@
 
 #include <array>
 #include <deque>
-#include "audio_core/hle/common.h"
+#include "audio_core/audio_types.h"
 #include "common/common_types.h"
 
+namespace AudioCore {
 namespace AudioInterp {
 
 /// A variable length buffer of signed PCM16 stereo samples.
@@ -31,8 +32,7 @@ struct State {
  * @param output The resampled audio buffer.
  * @param outputi The index of output to start writing to.
  */
-void None(State& state, StereoBuffer16& input, float rate, DSP::HLE::StereoFrame16& output,
-          size_t& outputi);
+void None(State& state, StereoBuffer16& input, float rate, StereoFrame16& output, size_t& outputi);
 
 /**
  * Linear interpolation. This is equivalent to a first-order hold. There is a two-sample predelay.
@@ -43,7 +43,8 @@ void None(State& state, StereoBuffer16& input, float rate, DSP::HLE::StereoFrame
  * @param output The resampled audio buffer.
  * @param outputi The index of output to start writing to.
  */
-void Linear(State& state, StereoBuffer16& input, float rate, DSP::HLE::StereoFrame16& output,
+void Linear(State& state, StereoBuffer16& input, float rate, StereoFrame16& output,
             size_t& outputi);
 
 } // namespace AudioInterp
+} // namespace AudioCore

src/audio_core/null_sink.h

@@ -5,7 +5,7 @@
 #pragma once
 
 #include <cstddef>
-#include "audio_core/audio_core.h"
+#include "audio_core/audio_types.h"
 #include "audio_core/sink.h"
 
 namespace AudioCore {

src/audio_core/sdl2_sink.cpp

@@ -5,7 +5,7 @@
 #include <list>
 #include <numeric>
 #include <SDL.h>
-#include "audio_core/audio_core.h"
+#include "audio_core/audio_types.h"
 #include "audio_core/sdl2_sink.h"
 #include "common/assert.h"
 #include "common/logging/log.h"

src/audio_core/sink_details.cpp

@@ -4,6 +4,7 @@
 
 #include <algorithm>
 #include <memory>
+#include <string>
 #include <vector>
 #include "audio_core/null_sink.h"
 #include "audio_core/sink_details.h"

src/audio_core/time_stretch.cpp

@@ -6,7 +6,7 @@
 #include <cmath>
 #include <vector>
 #include <SoundTouch.h>
-#include "audio_core/audio_core.h"
+#include "audio_core/audio_types.h"
 #include "audio_core/time_stretch.h"
 #include "common/common_types.h"
 #include "common/logging/log.h"