11 files changed, 0 insertions, 2044 deletions

diff --git a/src/audio_core/hle/common.h b/src/audio_core/hle/common.h
deleted file mode 100644
index 7fbc3ad9a..000000000
--- a/src/audio_core/hle/common.h
+++ /dev/null
@@ -1,34 +0,0 @@
-// Copyright 2016 Citra Emulator Project
-// Licensed under GPLv2 or any later version
-// Refer to the license.txt file included.
-
-#pragma once
-
-#include <algorithm>
-#include <array>
-#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>, samples_per_frame>;
-
-/// The DSP is quadraphonic internally.
-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.
- * FilterT::ProcessSample is called sequentially on the samples.
- */
-template <typename FrameT, typename FilterT>
-void FilterFrame(FrameT& frame, FilterT& filter) {
-    std::transform(frame.begin(), frame.end(), frame.begin(),
-                   [&filter](const auto& sample) { return filter.ProcessSample(sample); });
-}
-
-} // namespace HLE
-} // namespace DSP
diff --git a/src/audio_core/hle/dsp.cpp b/src/audio_core/hle/dsp.cpp
deleted file mode 100644
index 260b182ed..000000000
--- a/src/audio_core/hle/dsp.cpp
+++ /dev/null
@@ -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
diff --git a/src/audio_core/hle/dsp.h b/src/audio_core/hle/dsp.h
deleted file mode 100644
index 94ce48863..000000000
--- a/src/audio_core/hle/dsp.h
+++ /dev/null
@@ -1,595 +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 <cstddef>
-#include <memory>
-#include <type_traits>
-#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 {
-
-// The application-accessible region of DSP memory consists of two parts. Both are marked as IO and
-// have Read/Write permissions.
-//
-// First Region:  0x1FF50000 (Size: 0x8000)
-// Second Region: 0x1FF70000 (Size: 0x8000)
-//
-// The DSP reads from each region alternately based on the frame counter for each region much like a
-// double-buffer. The frame counter is located as the very last u16 of each region and is
-// incremented each audio tick.
-
-constexpr u32 region0_offset = 0x50000;
-constexpr u32 region1_offset = 0x70000;
-
-/**
- * The DSP is native 16-bit. The DSP also appears to be big-endian. When reading 32-bit numbers from
- * its memory regions, the higher and lower 16-bit halves are swapped compared to the little-endian
- * layout of the ARM11. Hence from the ARM11's point of view the memory space appears to be
- * middle-endian.
- *
- * Unusually this does not appear to be an issue for floating point numbers. The DSP makes the more
- * sensible choice of keeping that little-endian. There are also some exceptions such as the
- * IntermediateMixSamples structure, which is little-endian.
- *
- * This struct implements the conversion to and from this middle-endianness.
- */
-struct u32_dsp {
-    u32_dsp() = default;
-    operator u32() const {
-        return Convert(storage);
-    }
-    void operator=(u32 new_value) {
-        storage = Convert(new_value);
-    }
-
-private:
-    static constexpr u32 Convert(u32 value) {
-        return (value << 16) | (value >> 16);
-    }
-    u32_le storage;
-};
-#if (__GNUC__ >= 5) || defined(__clang__) || defined(_MSC_VER)
-static_assert(std::is_trivially_copyable<u32_dsp>::value, "u32_dsp isn't trivially copyable");
-#endif
-
-// There are 15 structures in each memory region. A table of them in the order they appear in memory
-// is presented below:
-//
-//       #           First Region DSP Address   Purpose                               Control
-//       5           0x8400                     DSP Status                            DSP
-//       9           0x8410                     DSP Debug Info                        DSP
-//       6           0x8540                     Final Mix Samples                     DSP
-//       2           0x8680                     Source Status [24]                    DSP
-//       8           0x8710                     Compressor Table                      Application
-//       4           0x9430                     DSP Configuration                     Application
-//       7           0x9492                     Intermediate Mix Samples              DSP + App
-//       1           0x9E92                     Source Configuration [24]             Application
-//       3           0xA792                     Source ADPCM Coefficients [24]        Application
-//       10          0xA912                     Surround Sound Related
-//       11          0xAA12                     Surround Sound Related
-//       12          0xAAD2                     Surround Sound Related
-//       13          0xAC52                     Surround Sound Related
-//       14          0xAC5C                     Surround Sound Related
-//       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.
-//
-// 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
-// implementation does and provides to applications.
-//
-// Application requests the DSP service to convert DSP addresses into ARM11 virtual addresses using
-// the ConvertProcessAddressFromDspDram service call. Applications seem to derive the addresses for
-// the second region via:
-//     second_region_dsp_addr = first_region_dsp_addr | 0x10000
-//
-// Applications maintain most of its own audio state, the memory region is used mainly for
-// communication and not storage of state.
-//
-// In the documentation below, filter and effect transfer functions are specified in the z domain.
-// (If you are more familiar with the Laplace transform, z = exp(sT). The z domain is the digital
-// frequency domain, just like how the s domain is the analog frequency domain.)
-
-#define INSERT_PADDING_DSPWORDS(num_words) INSERT_PADDING_BYTES(2 * (num_words))
-
-// GCC versions < 5.0 do not implement std::is_trivially_copyable.
-// Excluding MSVC because it has weird behaviour for std::is_trivially_copyable.
-#if (__GNUC__ >= 5) || defined(__clang__)
-#define ASSERT_DSP_STRUCT(name, size)                                                              \
-    static_assert(std::is_standard_layout<name>::value,                                            \
-                  "DSP structure " #name " doesn't use standard layout");                          \
-    static_assert(std::is_trivially_copyable<name>::value,                                         \
-                  "DSP structure " #name " isn't trivially copyable");                             \
-    static_assert(sizeof(name) == (size), "Unexpected struct size for DSP structure " #name)
-#else
-#define ASSERT_DSP_STRUCT(name, size)                                                              \
-    static_assert(std::is_standard_layout<name>::value,                                            \
-                  "DSP structure " #name " doesn't use standard layout");                          \
-    static_assert(sizeof(name) == (size), "Unexpected struct size for DSP structure " #name)
-#endif
-
-struct SourceConfiguration {
-    struct Configuration {
-        /// These dirty flags are set by the application when it updates the fields in this struct.
-        /// The DSP clears these each audio frame.
-        union {
-            u32_le dirty_raw;
-
-            BitField<0, 1, u32_le> format_dirty;
-            BitField<1, 1, u32_le> mono_or_stereo_dirty;
-            BitField<2, 1, u32_le> adpcm_coefficients_dirty;
-            /// Tends to be set when a looped buffer is queued.
-            BitField<3, 1, u32_le> partial_embedded_buffer_dirty;
-            BitField<4, 1, u32_le> partial_reset_flag;
-
-            BitField<16, 1, u32_le> enable_dirty;
-            BitField<17, 1, u32_le> interpolation_dirty;
-            BitField<18, 1, u32_le> rate_multiplier_dirty;
-            BitField<19, 1, u32_le> buffer_queue_dirty;
-            BitField<20, 1, u32_le> loop_related_dirty;
-            /// Tends to also be set when embedded buffer is updated.
-            BitField<21, 1, u32_le> play_position_dirty;
-            BitField<22, 1, u32_le> filters_enabled_dirty;
-            BitField<23, 1, u32_le> simple_filter_dirty;
-            BitField<24, 1, u32_le> biquad_filter_dirty;
-            BitField<25, 1, u32_le> gain_0_dirty;
-            BitField<26, 1, u32_le> gain_1_dirty;
-            BitField<27, 1, u32_le> gain_2_dirty;
-            BitField<28, 1, u32_le> sync_dirty;
-            BitField<29, 1, u32_le> reset_flag;
-            BitField<30, 1, u32_le> embedded_buffer_dirty;
-        };
-
-        // Gain control
-
-        /**
-         * Gain is between 0.0-1.0. This determines how much will this source appear on each of the
-         * 12 channels that feed into the intermediate mixers. Each of the three intermediate mixers
-         * is fed two left and two right channels.
-         */
-        float_le gain[3][4];
-
-        // Interpolation
-
-        /// Multiplier for sample rate. Resampling occurs with the selected interpolation method.
-        float_le rate_multiplier;
-
-        enum class InterpolationMode : u8 {
-            Polyphase = 0,
-            Linear = 1,
-            None = 2,
-        };
-
-        InterpolationMode interpolation_mode;
-        INSERT_PADDING_BYTES(1); ///< Interpolation related
-
-        // Filters
-
-        /**
-         * This is the simplest normalized first-order digital recursive filter.
-         * The transfer function of this filter is:
-         *     H(z) = b0 / (1 - a1 z^-1)
-         * Note the feedbackward coefficient is negated.
-         * Values are signed fixed point with 15 fractional bits.
-         */
-        struct SimpleFilter {
-            s16_le b0;
-            s16_le a1;
-        };
-
-        /**
-         * This is a normalised biquad filter (second-order).
-         * The transfer function of this filter is:
-         *     H(z) = (b0 + b1 z^-1 + b2 z^-2) / (1 - a1 z^-1 - a2 z^-2)
-         * Nintendo chose to negate the feedbackward coefficients. This differs from standard
-         * notation as in: https://ccrma.stanford.edu/~jos/filters/Direct_Form_I.html
-         * Values are signed fixed point with 14 fractional bits.
-         */
-        struct BiquadFilter {
-            s16_le a2;
-            s16_le a1;
-            s16_le b2;
-            s16_le b1;
-            s16_le b0;
-        };
-
-        union {
-            u16_le filters_enabled;
-            BitField<0, 1, u16_le> simple_filter_enabled;
-            BitField<1, 1, u16_le> biquad_filter_enabled;
-        };
-
-        SimpleFilter simple_filter;
-        BiquadFilter biquad_filter;
-
-        // Buffer Queue
-
-        /// A buffer of audio data from the application, along with metadata about it.
-        struct Buffer {
-            /// Physical memory address of the start of the buffer
-            u32_dsp physical_address;
-
-            /// This is length in terms of samples.
-            /// Note that in different buffer formats a sample takes up different number of bytes.
-            u32_dsp length;
-
-            /// ADPCM Predictor (4 bits) and Scale (4 bits)
-            union {
-                u16_le adpcm_ps;
-                BitField<0, 4, u16_le> adpcm_scale;
-                BitField<4, 4, u16_le> adpcm_predictor;
-            };
-
-            /// ADPCM Historical Samples (y[n-1] and y[n-2])
-            u16_le adpcm_yn[2];
-
-            /// This is non-zero when the ADPCM values above are to be updated.
-            u8 adpcm_dirty;
-
-            /// Is a looping buffer.
-            u8 is_looping;
-
-            /// This value is shown in SourceStatus::previous_buffer_id when this buffer has
-            /// finished. This allows the emulated application to tell what buffer is currently
-            /// playing.
-            u16_le buffer_id;
-
-            INSERT_PADDING_DSPWORDS(1);
-        };
-
-        u16_le buffers_dirty; ///< Bitmap indicating which buffers are dirty (bit i -> buffers[i])
-        Buffer buffers[4];    ///< Queued Buffers
-
-        // Playback controls
-
-        u32_dsp loop_related;
-        u8 enable;
-        INSERT_PADDING_BYTES(1);
-        u16_le sync;           ///< Application-side sync (See also: SourceStatus::sync)
-        u32_dsp play_position; ///< Position. (Units: number of samples)
-        INSERT_PADDING_DSPWORDS(2);
-
-        // Embedded Buffer
-        // This buffer is often the first buffer to be used when initiating audio playback,
-        // after which the buffer queue is used.
-
-        u32_dsp physical_address;
-
-        /// This is length in terms of samples.
-        /// Note a sample takes up different number of bytes in different buffer formats.
-        u32_dsp length;
-
-        enum class MonoOrStereo : u16_le {
-            Mono = 1,
-            Stereo = 2,
-        };
-
-        enum class Format : u16_le {
-            PCM8 = 0,
-            PCM16 = 1,
-            ADPCM = 2,
-        };
-
-        union {
-            u16_le flags1_raw;
-            BitField<0, 2, MonoOrStereo> mono_or_stereo;
-            BitField<2, 2, Format> format;
-            BitField<5, 1, u16_le> fade_in;
-        };
-
-        /// ADPCM Predictor (4 bit) and Scale (4 bit)
-        union {
-            u16_le adpcm_ps;
-            BitField<0, 4, u16_le> adpcm_scale;
-            BitField<4, 4, u16_le> adpcm_predictor;
-        };
-
-        /// ADPCM Historical Samples (y[n-1] and y[n-2])
-        u16_le adpcm_yn[2];
-
-        union {
-            u16_le flags2_raw;
-            BitField<0, 1, u16_le> adpcm_dirty; ///< Has the ADPCM info above been changed?
-            BitField<1, 1, u16_le> is_looping;  ///< Is this a looping buffer?
-        };
-
-        /// Buffer id of embedded buffer (used as a buffer id in SourceStatus to reference this
-        /// buffer).
-        u16_le buffer_id;
-    };
-
-    Configuration config[num_sources];
-};
-ASSERT_DSP_STRUCT(SourceConfiguration::Configuration, 192);
-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 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 current_buffer_id;   ///< Updated when a buffer finishes playing
-        INSERT_PADDING_DSPWORDS(1);
-    };
-
-    Status status[num_sources];
-};
-ASSERT_DSP_STRUCT(SourceStatus::Status, 12);
-
-struct DspConfiguration {
-    /// These dirty flags are set by the application when it updates the fields in this struct.
-    /// The DSP clears these each audio frame.
-    union {
-        u32_le dirty_raw;
-
-        BitField<8, 1, u32_le> mixer1_enabled_dirty;
-        BitField<9, 1, u32_le> mixer2_enabled_dirty;
-        BitField<10, 1, u32_le> delay_effect_0_dirty;
-        BitField<11, 1, u32_le> delay_effect_1_dirty;
-        BitField<12, 1, u32_le> reverb_effect_0_dirty;
-        BitField<13, 1, u32_le> reverb_effect_1_dirty;
-
-        BitField<16, 1, u32_le> volume_0_dirty;
-
-        BitField<24, 1, u32_le> volume_1_dirty;
-        BitField<25, 1, u32_le> volume_2_dirty;
-        BitField<26, 1, u32_le> output_format_dirty;
-        BitField<27, 1, u32_le> limiter_enabled_dirty;
-        BitField<28, 1, u32_le> headphones_connected_dirty;
-    };
-
-    /// The DSP has three intermediate audio mixers. This controls the volume level (0.0-1.0) for
-    /// each at the final mixer.
-    float_le volume[3];
-
-    INSERT_PADDING_DSPWORDS(3);
-
-    enum class OutputFormat : u16_le {
-        Mono = 0,
-        Stereo = 1,
-        Surround = 2,
-    };
-
-    OutputFormat output_format;
-
-    u16_le limiter_enabled;      ///< Not sure of the exact gain equation for the limiter.
-    u16_le headphones_connected; ///< Application updates the DSP on headphone status.
-    INSERT_PADDING_DSPWORDS(4);  ///< TODO: Surround sound related
-    INSERT_PADDING_DSPWORDS(2);  ///< TODO: Intermediate mixer 1/2 related
-    u16_le mixer1_enabled;
-    u16_le mixer2_enabled;
-
-    /**
-     * This is delay with feedback.
-     * Transfer function:
-     *     H(z) = a z^-N / (1 - b z^-1 + a g z^-N)
-     *   where
-     *     N = frame_count * samples_per_frame
-     * g, a and b are fixed point with 7 fractional bits
-     */
-    struct DelayEffect {
-        /// These dirty flags are set by the application when it updates the fields in this struct.
-        /// The DSP clears these each audio frame.
-        union {
-            u16_le dirty_raw;
-            BitField<0, 1, u16_le> enable_dirty;
-            BitField<1, 1, u16_le> work_buffer_address_dirty;
-            BitField<2, 1, u16_le> other_dirty; ///< Set when anything else has been changed
-        };
-
-        u16_le enable;
-        INSERT_PADDING_DSPWORDS(1);
-        u16_le outputs;
-        /// The application allocates a block of memory for the DSP to use as a work buffer.
-        u32_dsp work_buffer_address;
-        /// Frames to delay by
-        u16_le frame_count;
-
-        // Coefficients
-        s16_le g; ///< Fixed point with 7 fractional bits
-        s16_le a; ///< Fixed point with 7 fractional bits
-        s16_le b; ///< Fixed point with 7 fractional bits
-    };
-
-    DelayEffect delay_effect[2];
-
-    struct ReverbEffect {
-        INSERT_PADDING_DSPWORDS(26); ///< TODO
-    };
-
-    ReverbEffect reverb_effect[2];
-
-    INSERT_PADDING_DSPWORDS(4);
-};
-ASSERT_DSP_STRUCT(DspConfiguration, 196);
-ASSERT_DSP_STRUCT(DspConfiguration::DelayEffect, 20);
-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[num_sources][16];
-};
-ASSERT_DSP_STRUCT(AdpcmCoefficients, 768);
-
-struct DspStatus {
-    u16_le unknown;
-    u16_le dropped_frames;
-    INSERT_PADDING_DSPWORDS(0xE);
-};
-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[samples_per_frame][2];
-};
-ASSERT_DSP_STRUCT(FinalMixSamples, 640);
-
-/// DSP writes output of intermediate mixers 1 and 2 here.
-/// Writes to this region by the application edits the output of the intermediate mixers.
-/// This seems to be intended to allow the application to do custom effects on the ARM11.
-/// Values that exceed s16 range will be clipped by the DSP after further processing.
-struct IntermediateMixSamples {
-    struct Samples {
-        s32_le pcm32[4][samples_per_frame]; ///< Little-endian as opposed to DSP middle-endian.
-    };
-
-    Samples mix1;
-    Samples mix2;
-};
-ASSERT_DSP_STRUCT(IntermediateMixSamples, 5120);
-
-/// Compressor table
-struct Compressor {
-    INSERT_PADDING_DSPWORDS(0xD20); ///< TODO
-};
-
-/// There is no easy way to implement this in a HLE implementation.
-struct DspDebug {
-    INSERT_PADDING_DSPWORDS(0x130);
-};
-ASSERT_DSP_STRUCT(DspDebug, 0x260);
-
-struct SharedMemory {
-    /// Padding
-    INSERT_PADDING_DSPWORDS(0x400);
-
-    DspStatus dsp_status;
-
-    DspDebug dsp_debug;
-
-    FinalMixSamples final_samples;
-
-    SourceStatus source_statuses;
-
-    Compressor compressor;
-
-    DspConfiguration dsp_configuration;
-
-    IntermediateMixSamples intermediate_mix_samples;
-
-    SourceConfiguration source_configurations;
-
-    AdpcmCoefficients adpcm_coefficients;
-
-    struct {
-        INSERT_PADDING_DSPWORDS(0x100);
-    } unknown10;
-
-    struct {
-        INSERT_PADDING_DSPWORDS(0xC0);
-    } unknown11;
-
-    struct {
-        INSERT_PADDING_DSPWORDS(0x180);
-    } unknown12;
-
-    struct {
-        INSERT_PADDING_DSPWORDS(0xA);
-    } unknown13;
-
-    struct {
-        INSERT_PADDING_DSPWORDS(0x13A3);
-    } unknown14;
-
-    u16_le frame_counter;
-};
-ASSERT_DSP_STRUCT(SharedMemory, 0x8000);
-
-union DspMemory {
-    std::array<u8, 0x80000> raw_memory;
-    struct {
-        u8 unused_0[0x50000];
-        SharedMemory region_0;
-        u8 unused_1[0x18000];
-        SharedMemory region_1;
-        u8 unused_2[0x8000];
-    };
-};
-static_assert(offsetof(DspMemory, region_0) == region0_offset,
-              "DSP region 0 is at the wrong 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");
-static_assert(offsetof(SharedMemory, source_configurations) % 2 == 0,
-              "Structures in DSP::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");
-static_assert(offsetof(SharedMemory, adpcm_coefficients) % 2 == 0,
-              "Structures in DSP::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");
-static_assert(offsetof(SharedMemory, dsp_status) % 2 == 0,
-              "Structures in DSP::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");
-static_assert(offsetof(SharedMemory, intermediate_mix_samples) % 2 == 0,
-              "Structures in DSP::HLE::SharedMemory must be 2-byte aligned");
-static_assert(offsetof(SharedMemory, compressor) % 2 == 0,
-              "Structures in DSP::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");
-static_assert(offsetof(SharedMemory, unknown10) % 2 == 0,
-              "Structures in DSP::HLE::SharedMemory must be 2-byte aligned");
-static_assert(offsetof(SharedMemory, unknown11) % 2 == 0,
-              "Structures in DSP::HLE::SharedMemory must be 2-byte aligned");
-static_assert(offsetof(SharedMemory, unknown12) % 2 == 0,
-              "Structures in DSP::HLE::SharedMemory must be 2-byte aligned");
-static_assert(offsetof(SharedMemory, unknown13) % 2 == 0,
-              "Structures in DSP::HLE::SharedMemory must be 2-byte aligned");
-static_assert(offsetof(SharedMemory, unknown14) % 2 == 0,
-              "Structures in DSP::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
diff --git a/src/audio_core/hle/filter.cpp b/src/audio_core/hle/filter.cpp
deleted file mode 100644
index b24a79b89..000000000
--- a/src/audio_core/hle/filter.cpp
+++ /dev/null
@@ -1,117 +0,0 @@
-// Copyright 2016 Citra Emulator Project
-// Licensed under GPLv2 or any later version
-// Refer to the license.txt file included.
-
-#include <array>
-#include <cstddef>
-#include "audio_core/hle/common.h"
-#include "audio_core/hle/dsp.h"
-#include "audio_core/hle/filter.h"
-#include "common/common_types.h"
-#include "common/math_util.h"
-
-namespace DSP {
-namespace HLE {
-
-void SourceFilters::Reset() {
-    Enable(false, false);
-}
-
-void SourceFilters::Enable(bool simple, bool biquad) {
-    simple_filter_enabled = simple;
-    biquad_filter_enabled = biquad;
-
-    if (!simple)
-        simple_filter.Reset();
-    if (!biquad)
-        biquad_filter.Reset();
-}
-
-void SourceFilters::Configure(SourceConfiguration::Configuration::SimpleFilter config) {
-    simple_filter.Configure(config);
-}
-
-void SourceFilters::Configure(SourceConfiguration::Configuration::BiquadFilter config) {
-    biquad_filter.Configure(config);
-}
-
-void SourceFilters::ProcessFrame(StereoFrame16& frame) {
-    if (!simple_filter_enabled && !biquad_filter_enabled)
-        return;
-
-    if (simple_filter_enabled) {
-        FilterFrame(frame, simple_filter);
-    }
-
-    if (biquad_filter_enabled) {
-        FilterFrame(frame, biquad_filter);
-    }
-}
-
-// SimpleFilter
-
-void SourceFilters::SimpleFilter::Reset() {
-    y1.fill(0);
-    // Configure as passthrough.
-    a1 = 0;
-    b0 = 1 << 15;
-}
-
-void SourceFilters::SimpleFilter::Configure(
-    SourceConfiguration::Configuration::SimpleFilter config) {
-
-    a1 = config.a1;
-    b0 = config.b0;
-}
-
-std::array<s16, 2> SourceFilters::SimpleFilter::ProcessSample(const std::array<s16, 2>& x0) {
-    std::array<s16, 2> y0;
-    for (size_t i = 0; i < 2; i++) {
-        const s32 tmp = (b0 * x0[i] + a1 * y1[i]) >> 15;
-        y0[i] = MathUtil::Clamp(tmp, -32768, 32767);
-    }
-
-    y1 = y0;
-
-    return y0;
-}
-
-// BiquadFilter
-
-void SourceFilters::BiquadFilter::Reset() {
-    x1.fill(0);
-    x2.fill(0);
-    y1.fill(0);
-    y2.fill(0);
-    // Configure as passthrough.
-    a1 = a2 = b1 = b2 = 0;
-    b0 = 1 << 14;
-}
-
-void SourceFilters::BiquadFilter::Configure(
-    SourceConfiguration::Configuration::BiquadFilter config) {
-
-    a1 = config.a1;
-    a2 = config.a2;
-    b0 = config.b0;
-    b1 = config.b1;
-    b2 = config.b2;
-}
-
-std::array<s16, 2> SourceFilters::BiquadFilter::ProcessSample(const std::array<s16, 2>& x0) {
-    std::array<s16, 2> y0;
-    for (size_t i = 0; i < 2; i++) {
-        const s32 tmp = (b0 * x0[i] + b1 * x1[i] + b2 * x2[i] + a1 * y1[i] + a2 * y2[i]) >> 14;
-        y0[i] = MathUtil::Clamp(tmp, -32768, 32767);
-    }
-
-    x2 = x1;
-    x1 = x0;
-    y2 = y1;
-    y1 = y0;
-
-    return y0;
-}
-
-} // namespace HLE
-} // namespace DSP
diff --git a/src/audio_core/hle/filter.h b/src/audio_core/hle/filter.h
deleted file mode 100644
index 5350e2857..000000000
--- a/src/audio_core/hle/filter.h
+++ /dev/null
@@ -1,117 +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 "audio_core/hle/common.h"
-#include "audio_core/hle/dsp.h"
-#include "common/common_types.h"
-
-namespace DSP {
-namespace HLE {
-
-/// Preprocessing filters. There is an independent set of filters for each Source.
-class SourceFilters final {
-public:
-    SourceFilters() {
-        Reset();
-    }
-
-    /// Reset internal state.
-    void Reset();
-
-    /**
-     * Enable/Disable filters
-     * See also: SourceConfiguration::Configuration::simple_filter_enabled,
-     *           SourceConfiguration::Configuration::biquad_filter_enabled.
-     * @param simple If true, enables the simple filter. If false, disables it.
-     * @param biquad If true, enables the biquad filter. If false, disables it.
-     */
-    void Enable(bool simple, bool biquad);
-
-    /**
-     * Configure simple filter.
-     * @param config Configuration from DSP shared memory.
-     */
-    void Configure(SourceConfiguration::Configuration::SimpleFilter config);
-
-    /**
-     * Configure biquad filter.
-     * @param config Configuration from DSP shared memory.
-     */
-    void Configure(SourceConfiguration::Configuration::BiquadFilter config);
-
-    /**
-     * Processes a frame in-place.
-     * @param frame Audio samples to process. Modified in-place.
-     */
-    void ProcessFrame(StereoFrame16& frame);
-
-private:
-    bool simple_filter_enabled;
-    bool biquad_filter_enabled;
-
-    struct SimpleFilter {
-        SimpleFilter() {
-            Reset();
-        }
-
-        /// Resets internal state.
-        void Reset();
-
-        /**
-         * Configures this filter with application settings.
-         * @param config Configuration from DSP shared memory.
-         */
-        void Configure(SourceConfiguration::Configuration::SimpleFilter config);
-
-        /**
-         * Processes a single stereo PCM16 sample.
-         * @param x0 Input sample
-         * @return Output sample
-         */
-        std::array<s16, 2> ProcessSample(const std::array<s16, 2>& x0);
-
-    private:
-        // Configuration
-        s32 a1, b0;
-        // Internal state
-        std::array<s16, 2> y1;
-    } simple_filter;
-
-    struct BiquadFilter {
-        BiquadFilter() {
-            Reset();
-        }
-
-        /// Resets internal state.
-        void Reset();
-
-        /**
-         * Configures this filter with application settings.
-         * @param config Configuration from DSP shared memory.
-         */
-        void Configure(SourceConfiguration::Configuration::BiquadFilter config);
-
-        /**
-         * Processes a single stereo PCM16 sample.
-         * @param x0 Input sample
-         * @return Output sample
-         */
-        std::array<s16, 2> ProcessSample(const std::array<s16, 2>& x0);
-
-    private:
-        // Configuration
-        s32 a1, a2, b0, b1, b2;
-        // Internal state
-        std::array<s16, 2> x1;
-        std::array<s16, 2> x2;
-        std::array<s16, 2> y1;
-        std::array<s16, 2> y2;
-    } biquad_filter;
-};
-
-} // namespace HLE
-} // namespace DSP
diff --git a/src/audio_core/hle/mixers.cpp b/src/audio_core/hle/mixers.cpp
deleted file mode 100644
index 6cc81dfca..000000000
--- a/src/audio_core/hle/mixers.cpp
+++ /dev/null
@@ -1,210 +0,0 @@
-// Copyright 2016 Citra Emulator Project
-// Licensed under GPLv2 or any later version
-// Refer to the license.txt file included.
-
-#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 HLE {
-
-void Mixers::Reset() {
-    current_frame.fill({});
-    state = {};
-}
-
-DspStatus Mixers::Tick(DspConfiguration& config, const IntermediateMixSamples& read_samples,
-                       IntermediateMixSamples& write_samples,
-                       const std::array<QuadFrame32, 3>& input) {
-    ParseConfig(config);
-
-    AuxReturn(read_samples);
-    AuxSend(write_samples, input);
-
-    MixCurrentFrame();
-
-    return GetCurrentStatus();
-}
-
-void Mixers::ParseConfig(DspConfiguration& config) {
-    if (!config.dirty_raw) {
-        return;
-    }
-
-    if (config.mixer1_enabled_dirty) {
-        config.mixer1_enabled_dirty.Assign(0);
-        state.mixer1_enabled = config.mixer1_enabled != 0;
-        LOG_TRACE(Audio_DSP, "mixers mixer1_enabled = %hu", config.mixer1_enabled);
-    }
-
-    if (config.mixer2_enabled_dirty) {
-        config.mixer2_enabled_dirty.Assign(0);
-        state.mixer2_enabled = config.mixer2_enabled != 0;
-        LOG_TRACE(Audio_DSP, "mixers mixer2_enabled = %hu", config.mixer2_enabled);
-    }
-
-    if (config.volume_0_dirty) {
-        config.volume_0_dirty.Assign(0);
-        state.intermediate_mixer_volume[0] = config.volume[0];
-        LOG_TRACE(Audio_DSP, "mixers volume[0] = %f", config.volume[0]);
-    }
-
-    if (config.volume_1_dirty) {
-        config.volume_1_dirty.Assign(0);
-        state.intermediate_mixer_volume[1] = config.volume[1];
-        LOG_TRACE(Audio_DSP, "mixers volume[1] = %f", config.volume[1]);
-    }
-
-    if (config.volume_2_dirty) {
-        config.volume_2_dirty.Assign(0);
-        state.intermediate_mixer_volume[2] = config.volume[2];
-        LOG_TRACE(Audio_DSP, "mixers volume[2] = %f", config.volume[2]);
-    }
-
-    if (config.output_format_dirty) {
-        config.output_format_dirty.Assign(0);
-        state.output_format = config.output_format;
-        LOG_TRACE(Audio_DSP, "mixers output_format = %zu",
-                  static_cast<size_t>(config.output_format));
-    }
-
-    if (config.headphones_connected_dirty) {
-        config.headphones_connected_dirty.Assign(0);
-        // Do nothing. (Note: Whether headphones are connected does affect coefficients used for
-        // surround sound.)
-        LOG_TRACE(Audio_DSP, "mixers headphones_connected=%hu", config.headphones_connected);
-    }
-
-    if (config.dirty_raw) {
-        LOG_DEBUG(Audio_DSP, "mixers remaining_dirty=%x", config.dirty_raw);
-    }
-
-    config.dirty_raw = 0;
-}
-
-static s16 ClampToS16(s32 value) {
-    return static_cast<s16>(MathUtil::Clamp(value, -32768, 32767));
-}
-
-static std::array<s16, 2> AddAndClampToS16(const std::array<s16, 2>& a,
-                                           const std::array<s16, 2>& b) {
-    return {ClampToS16(static_cast<s32>(a[0]) + static_cast<s32>(b[0])),
-            ClampToS16(static_cast<s32>(a[1]) + static_cast<s32>(b[1]))};
-}
-
-void Mixers::DownmixAndMixIntoCurrentFrame(float gain, const QuadFrame32& samples) {
-    // TODO(merry): Limiter. (Currently we're performing final mixing assuming a disabled limiter.)
-
-    switch (state.output_format) {
-    case OutputFormat::Mono:
-        std::transform(
-            current_frame.begin(), current_frame.end(), samples.begin(), current_frame.begin(),
-            [gain](const std::array<s16, 2>& accumulator,
-                   const std::array<s32, 4>& sample) -> std::array<s16, 2> {
-                // Downmix to mono
-                s16 mono = ClampToS16(static_cast<s32>(
-                    (gain * sample[0] + gain * sample[1] + gain * sample[2] + gain * sample[3]) /
-                    2));
-                // Mix into current frame
-                return AddAndClampToS16(accumulator, {mono, mono});
-            });
-        return;
-
-    case OutputFormat::Surround:
-    // TODO(merry): Implement surround sound.
-    // fallthrough
-
-    case OutputFormat::Stereo:
-        std::transform(
-            current_frame.begin(), current_frame.end(), samples.begin(), current_frame.begin(),
-            [gain](const std::array<s16, 2>& accumulator,
-                   const std::array<s32, 4>& sample) -> std::array<s16, 2> {
-                // Downmix to stereo
-                s16 left = ClampToS16(static_cast<s32>(gain * sample[0] + gain * sample[2]));
-                s16 right = ClampToS16(static_cast<s32>(gain * sample[1] + gain * sample[3]));
-                // Mix into current frame
-                return AddAndClampToS16(accumulator, {left, right});
-            });
-        return;
-    }
-
-    UNREACHABLE_MSG("Invalid output_format %zu", static_cast<size_t>(state.output_format));
-}
-
-void Mixers::AuxReturn(const IntermediateMixSamples& read_samples) {
-    // NOTE: read_samples.mix{1,2}.pcm32 annoyingly have their dimensions in reverse order to
-    // QuadFrame32.
-
-    if (state.mixer1_enabled) {
-        for (size_t sample = 0; sample < samples_per_frame; sample++) {
-            for (size_t channel = 0; channel < 4; channel++) {
-                state.intermediate_mix_buffer[1][sample][channel] =
-                    read_samples.mix1.pcm32[channel][sample];
-            }
-        }
-    }
-
-    if (state.mixer2_enabled) {
-        for (size_t sample = 0; sample < samples_per_frame; sample++) {
-            for (size_t channel = 0; channel < 4; channel++) {
-                state.intermediate_mix_buffer[2][sample][channel] =
-                    read_samples.mix2.pcm32[channel][sample];
-            }
-        }
-    }
-}
-
-void Mixers::AuxSend(IntermediateMixSamples& write_samples,
-                     const std::array<QuadFrame32, 3>& input) {
-    // NOTE: read_samples.mix{1,2}.pcm32 annoyingly have their dimensions in reverse order to
-    // QuadFrame32.
-
-    state.intermediate_mix_buffer[0] = input[0];
-
-    if (state.mixer1_enabled) {
-        for (size_t sample = 0; sample < samples_per_frame; sample++) {
-            for (size_t channel = 0; channel < 4; channel++) {
-                write_samples.mix1.pcm32[channel][sample] = input[1][sample][channel];
-            }
-        }
-    } else {
-        state.intermediate_mix_buffer[1] = input[1];
-    }
-
-    if (state.mixer2_enabled) {
-        for (size_t sample = 0; sample < samples_per_frame; sample++) {
-            for (size_t channel = 0; channel < 4; channel++) {
-                write_samples.mix2.pcm32[channel][sample] = input[2][sample][channel];
-            }
-        }
-    } else {
-        state.intermediate_mix_buffer[2] = input[2];
-    }
-}
-
-void Mixers::MixCurrentFrame() {
-    current_frame.fill({});
-
-    for (size_t mix = 0; mix < 3; mix++) {
-        DownmixAndMixIntoCurrentFrame(state.intermediate_mixer_volume[mix],
-                                      state.intermediate_mix_buffer[mix]);
-    }
-
-    // TODO(merry): Compressor. (We currently assume a disabled compressor.)
-}
-
-DspStatus Mixers::GetCurrentStatus() const {
-    DspStatus status;
-    status.unknown = 0;
-    status.dropped_frames = 0;
-    return status;
-}
-
-} // namespace HLE
-} // namespace DSP
diff --git a/src/audio_core/hle/mixers.h b/src/audio_core/hle/mixers.h
deleted file mode 100644
index bf4e865ae..000000000
--- a/src/audio_core/hle/mixers.h
+++ /dev/null
@@ -1,61 +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 "audio_core/hle/common.h"
-#include "audio_core/hle/dsp.h"
-
-namespace DSP {
-namespace HLE {
-
-class Mixers final {
-public:
-    Mixers() {
-        Reset();
-    }
-
-    void Reset();
-
-    DspStatus Tick(DspConfiguration& config, const IntermediateMixSamples& read_samples,
-                   IntermediateMixSamples& write_samples, const std::array<QuadFrame32, 3>& input);
-
-    StereoFrame16 GetOutput() const {
-        return current_frame;
-    }
-
-private:
-    StereoFrame16 current_frame = {};
-
-    using OutputFormat = DspConfiguration::OutputFormat;
-
-    struct {
-        std::array<float, 3> intermediate_mixer_volume = {};
-
-        bool mixer1_enabled = false;
-        bool mixer2_enabled = false;
-        std::array<QuadFrame32, 3> intermediate_mix_buffer = {};
-
-        OutputFormat output_format = OutputFormat::Stereo;
-
-    } state;
-
-    /// INTERNAL: Update our internal state based on the current config.
-    void ParseConfig(DspConfiguration& config);
-    /// INTERNAL: Read samples from shared memory that have been modified by the ARM11.
-    void AuxReturn(const IntermediateMixSamples& read_samples);
-    /// INTERNAL: Write samples to shared memory for the ARM11 to modify.
-    void AuxSend(IntermediateMixSamples& write_samples, const std::array<QuadFrame32, 3>& input);
-    /// INTERNAL: Mix current_frame.
-    void MixCurrentFrame();
-    /// INTERNAL: Downmix from quadraphonic to stereo based on status.output_format and accumulate
-    /// into current_frame.
-    void DownmixAndMixIntoCurrentFrame(float gain, const QuadFrame32& samples);
-    /// INTERNAL: Generate DspStatus based on internal state.
-    DspStatus GetCurrentStatus() const;
-};
-
-} // namespace HLE
-} // namespace DSP
diff --git a/src/audio_core/hle/pipe.cpp b/src/audio_core/hle/pipe.cpp
deleted file mode 100644
index 24074a514..000000000
--- a/src/audio_core/hle/pipe.cpp
+++ /dev/null
@@ -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
diff --git a/src/audio_core/hle/pipe.h b/src/audio_core/hle/pipe.h
deleted file mode 100644
index ac053c029..000000000
--- a/src/audio_core/hle/pipe.h
+++ /dev/null
@@ -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
diff --git a/src/audio_core/hle/source.cpp b/src/audio_core/hle/source.cpp
deleted file mode 100644
index c12287700..000000000
--- a/src/audio_core/hle/source.cpp
+++ /dev/null
@@ -1,349 +0,0 @@
-// 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));
-    }
-
-    u32_dsp play_position = {};
-    if (config.play_position_dirty && config.play_position != 0) {
-        config.play_position_dirty.Assign(0);
-        play_position = config.play_position;
-        // play_position applies only to the embedded buffer, and defaults to 0 w/o a dirty bit
-        // This will be the starting sample for the first time the buffer is played.
-    }
-
-    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,
-            play_position,
-            false,
-        });
-        LOG_TRACE(Audio_DSP, "enqueuing embedded addr=0x%08x len=%u id=%hu start=%u",
-                  config.physical_address, config.length, config.buffer_id,
-                  static_cast<u32>(config.play_position));
-    }
-
-    if (config.loop_related_dirty && config.loop_related != 0) {
-        config.loop_related_dirty.Assign(0);
-        LOG_WARNING(Audio_DSP, "Unhandled complex loop with loop_related=0x%08x",
-                    static_cast<u32>(config.loop_related));
-    }
-
-    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,
-                    {}, // 0 in u32_dsp
-                    false,
-                });
-                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;
-        }
-
-        switch (state.interpolation_mode) {
-        case InterpolationMode::None:
-            AudioInterp::None(state.interp_state, state.current_buffer, state.rate_multiplier,
-                              current_frame, frame_position);
-            break;
-        case InterpolationMode::Linear:
-            AudioInterp::Linear(state.interp_state, state.current_buffer, state.rate_multiplier,
-                                current_frame, frame_position);
-            break;
-        case InterpolationMode::Polyphase:
-            // TODO(merry): Implement polyphase interpolation
-            LOG_DEBUG(Audio_DSP, "Polyphase interpolation unimplemented; falling back to linear");
-            AudioInterp::Linear(state.interp_state, state.current_buffer, state.rate_multiplier,
-                                current_frame, frame_position);
-            break;
-        default:
-            UNIMPLEMENTED();
-            break;
-        }
-    }
-    state.next_sample_number += static_cast<u32>(frame_position);
-
-    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;
-
-    Buffer buf = state.input_queue.top();
-
-    // if we're in a loop, the current sound keeps playing afterwards, so leave the queue alone
-    if (!buf.is_looping) {
-        state.input_queue.pop();
-    }
-
-    if (buf.adpcm_dirty) {
-        state.adpcm_state.yn1 = buf.adpcm_yn[0];
-        state.adpcm_state.yn2 = buf.adpcm_yn[1];
-    }
-
-    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;
-    }
-
-    // the first playthrough starts at play_position, loops start at the beginning of the buffer
-    state.current_sample_number = (!buf.has_played) ? buf.play_position : 0;
-    state.next_sample_number = state.current_sample_number;
-    state.current_buffer_id = buf.buffer_id;
-    state.buffer_update = buf.from_queue && !buf.has_played;
-
-    buf.has_played = true;
-
-    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
deleted file mode 100644
index c4d2debc2..000000000
--- a/src/audio_core/hle/source.h
+++ /dev/null
@@ -1,149 +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 <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;
-        u32_dsp play_position; // = 0;
-        bool has_played;       // = false;
-    };
-
-    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;
-        AudioInterp::StereoBuffer16 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