10 files changed, 909 insertions, 56 deletions

diff --git a/src/audio_core/hle/common.h b/src/audio_core/hle/common.h
index 37d441eb2..596b67eaf 100644
--- a/src/audio_core/hle/common.h
+++ b/src/audio_core/hle/common.h
@@ -7,18 +7,19 @@
 #include <algorithm>
 #include <array>
 
-#include "audio_core/audio_core.h"
-
 #include "common/common_types.h"
 
 namespace DSP {
 namespace HLE {
 
+constexpr int num_sources = 24;
+constexpr int samples_per_frame = 160;     ///< Samples per audio frame at native sample rate
+
 /// The final output to the speakers is stereo. Preprocessing output in Source is also stereo.
-using StereoFrame16 = std::array<std::array<s16, 2>, AudioCore::samples_per_frame>;
+using StereoFrame16 = std::array<std::array<s16, 2>, samples_per_frame>;
 
 /// The DSP is quadraphonic internally.
-using QuadFrame32   = std::array<std::array<s32, 4>, AudioCore::samples_per_frame>;
+using QuadFrame32   = std::array<std::array<s32, 4>, samples_per_frame>;
 
 /**
  * This performs the filter operation defined by FilterT::ProcessSample on the frame in-place.
@@ -26,7 +27,7 @@ using QuadFrame32   = std::array<std::array<s32, 4>, AudioCore::samples_per_fram
  */
 template<typename FrameT, typename FilterT>
 void FilterFrame(FrameT& frame, FilterT& filter) {
-    std::transform(frame.begin(), frame.end(), frame.begin(), [&filter](const typename FrameT::value_type& sample) {
+    std::transform(frame.begin(), frame.end(), frame.begin(), [&filter](const auto& sample) {
         return filter.ProcessSample(sample);
     });
 }
diff --git a/src/audio_core/hle/dsp.cpp b/src/audio_core/hle/dsp.cpp
index c89356edc..0640e1eff 100644
--- a/src/audio_core/hle/dsp.cpp
+++ b/src/audio_core/hle/dsp.cpp
@@ -2,40 +2,138 @@
 // 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 {
 
-SharedMemory g_region0;
-SharedMemory g_region1;
+// Region management
+
+std::array<SharedMemory, 2> g_regions;
+
+static size_t CurrentRegionIndex() {
+    // The region with the higher frame counter is chosen unless there is wraparound.
+    // This function only returns a 0 or 1.
+
+    if (g_regions[0].frame_counter == 0xFFFFu && g_regions[1].frame_counter != 0xFFFEu) {
+        // Wraparound has occured.
+        return 1;
+    }
+
+    if (g_regions[1].frame_counter == 0xFFFFu && g_regions[0].frame_counter != 0xFFFEu) {
+        // Wraparound has occured.
+        return 0;
+    }
+
+    return (g_regions[0].frame_counter > g_regions[1].frame_counter) ? 0 : 1;
+}
+
+static SharedMemory& ReadRegion() {
+    return g_regions[CurrentRegionIndex()];
+}
+
+static SharedMemory& WriteRegion() {
+    return g_regions[1 - CurrentRegionIndex()];
+}
+
+// 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 std::unique_ptr<AudioCore::Sink> sink;
+static AudioCore::TimeStretcher time_stretcher;
+
+static void OutputCurrentFrame(const StereoFrame16& frame) {
+    time_stretcher.AddSamples(&frame[0][0], frame.size());
+    sink->EnqueueSamples(time_stretcher.Process(sink->SamplesInQueue()));
+}
+
+// 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() {
+    time_stretcher.Flush();
+    while (true) {
+        std::vector<s16> residual_audio = time_stretcher.Process(sink->SamplesInQueue());
+        if (residual_audio.empty())
+            break;
+        sink->EnqueueSamples(residual_audio);
+    }
 }
 
 bool Tick() {
-    return true;
-}
+    StereoFrame16 current_frame = {};
 
-SharedMemory& CurrentRegion() {
-    // The region with the higher frame counter is chosen unless there is wraparound.
+    // TODO: Check dsp::DSP semaphore (which indicates emulated application has finished writing to shared memory region)
+    current_frame = GenerateCurrentFrame();
 
-    if (g_region0.frame_counter == 0xFFFFu && g_region1.frame_counter != 0xFFFEu) {
-        // Wraparound has occured.
-        return g_region1;
-    }
+    OutputCurrentFrame(current_frame);
 
-    if (g_region1.frame_counter == 0xFFFFu && g_region0.frame_counter != 0xFFFEu) {
-        // Wraparound has occured.
-        return g_region0;
-    }
+    return true;
+}
 
-    return (g_region0.frame_counter > g_region1.frame_counter) ? g_region0 : g_region1;
+void SetSink(std::unique_ptr<AudioCore::Sink> sink_) {
+    sink = std::move(sink_);
+    time_stretcher.SetOutputSampleRate(sink->GetNativeSampleRate());
 }
 
 } // namespace HLE
diff --git a/src/audio_core/hle/dsp.h b/src/audio_core/hle/dsp.h
index c15ef0b7a..9275cd7de 100644
--- a/src/audio_core/hle/dsp.h
+++ b/src/audio_core/hle/dsp.h
@@ -4,16 +4,22 @@
 
 #pragma once
 
+#include <array>
 #include <cstddef>
+#include <memory>
 #include <type_traits>
 
-#include "audio_core/audio_core.h"
+#include "audio_core/hle/common.h"
 
 #include "common/bit_field.h"
 #include "common/common_funcs.h"
 #include "common/common_types.h"
 #include "common/swap.h"
 
+namespace AudioCore {
+class Sink;
+}
+
 namespace DSP {
 namespace HLE {
 
@@ -27,13 +33,8 @@ namespace HLE {
 // double-buffer. The frame counter is located as the very last u16 of each region and is incremented
 // each audio tick.
 
-struct SharedMemory;
-
 constexpr VAddr region0_base = 0x1FF50000;
-extern SharedMemory g_region0;
-
 constexpr VAddr region1_base = 0x1FF70000;
-extern SharedMemory g_region1;
 
 /**
  * The DSP is native 16-bit. The DSP also appears to be big-endian. When reading 32-bit numbers from
@@ -164,9 +165,9 @@ struct SourceConfiguration {
         float_le rate_multiplier;
 
         enum class InterpolationMode : u8 {
-            None = 0,
+            Polyphase = 0,
             Linear = 1,
-            Polyphase = 2
+            None = 2
         };
 
         InterpolationMode interpolation_mode;
@@ -305,7 +306,7 @@ struct SourceConfiguration {
         u16_le buffer_id;
     };
 
-    Configuration config[AudioCore::num_sources];
+    Configuration config[num_sources];
 };
 ASSERT_DSP_STRUCT(SourceConfiguration::Configuration, 192);
 ASSERT_DSP_STRUCT(SourceConfiguration::Configuration::Buffer, 20);
@@ -313,14 +314,14 @@ ASSERT_DSP_STRUCT(SourceConfiguration::Configuration::Buffer, 20);
 struct SourceStatus {
     struct Status {
         u8 is_enabled;               ///< Is this channel enabled? (Doesn't have to be playing anything.)
-        u8 previous_buffer_id_dirty; ///< Non-zero when previous_buffer_id changes
+        u8 current_buffer_id_dirty;  ///< Non-zero when current_buffer_id changes
         u16_le sync;                 ///< Is set by the DSP to the value of SourceConfiguration::sync
         u32_dsp buffer_position;     ///< Number of samples into the current buffer
-        u16_le previous_buffer_id;   ///< Updated when a buffer finishes playing
+        u16_le current_buffer_id;    ///< Updated when a buffer finishes playing
         INSERT_PADDING_DSPWORDS(1);
     };
 
-    Status status[AudioCore::num_sources];
+    Status status[num_sources];
 };
 ASSERT_DSP_STRUCT(SourceStatus::Status, 12);
 
@@ -413,7 +414,7 @@ ASSERT_DSP_STRUCT(DspConfiguration::ReverbEffect, 52);
 struct AdpcmCoefficients {
     /// Coefficients are signed fixed point with 11 fractional bits.
     /// Each source has 16 coefficients associated with it.
-    s16_le coeff[AudioCore::num_sources][16];
+    s16_le coeff[num_sources][16];
 };
 ASSERT_DSP_STRUCT(AdpcmCoefficients, 768);
 
@@ -427,7 +428,7 @@ ASSERT_DSP_STRUCT(DspStatus, 32);
 /// Final mixed output in PCM16 stereo format, what you hear out of the speakers.
 /// When the application writes to this region it has no effect.
 struct FinalMixSamples {
-    s16_le pcm16[2 * AudioCore::samples_per_frame];
+    s16_le pcm16[samples_per_frame][2];
 };
 ASSERT_DSP_STRUCT(FinalMixSamples, 640);
 
@@ -437,7 +438,7 @@ ASSERT_DSP_STRUCT(FinalMixSamples, 640);
 /// Values that exceed s16 range will be clipped by the DSP after further processing.
 struct IntermediateMixSamples {
     struct Samples {
-        s32_le pcm32[4][AudioCore::samples_per_frame]; ///< Little-endian as opposed to DSP middle-endian.
+        s32_le pcm32[4][samples_per_frame]; ///< Little-endian as opposed to DSP middle-endian.
     };
 
     Samples mix1;
@@ -502,6 +503,8 @@ struct SharedMemory {
 };
 ASSERT_DSP_STRUCT(SharedMemory, 0x8000);
 
+extern std::array<SharedMemory, 2> g_regions;
+
 // Structures must have an offset that is a multiple of two.
 static_assert(offsetof(SharedMemory, frame_counter) % 2 == 0, "Structures in DSP::HLE::SharedMemory must be 2-byte aligned");
 static_assert(offsetof(SharedMemory, source_configurations) % 2 == 0, "Structures in DSP::HLE::SharedMemory must be 2-byte aligned");
@@ -535,8 +538,11 @@ void Shutdown();
  */
 bool Tick();
 
-/// Returns a mutable reference to the current region. Current region is selected based on the frame counter.
-SharedMemory& CurrentRegion();
+/**
+ * Set the output sink. This must be called before calling Tick().
+ * @param sink The sink to which audio will be output to.
+ */
+void SetSink(std::unique_ptr<AudioCore::Sink> sink);
 
 } // namespace HLE
 } // namespace DSP
diff --git a/src/audio_core/hle/filter.h b/src/audio_core/hle/filter.h
index 75738f600..43d2035cd 100644
--- a/src/audio_core/hle/filter.h
+++ b/src/audio_core/hle/filter.h
@@ -16,6 +16,7 @@ namespace HLE {
 
 /// Preprocessing filters. There is an independent set of filters for each Source.
 class SourceFilters final {
+public:
     SourceFilters() { Reset(); }
 
     /// Reset internal state.
diff --git a/src/audio_core/hle/mixers.cpp b/src/audio_core/hle/mixers.cpp
new file mode 100644
index 000000000..18335f7f0
--- /dev/null
+++ b/src/audio_core/hle/mixers.cpp
@@ -0,0 +1,201 @@
+// 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
new file mode 100644
index 000000000..b52952eb5
--- /dev/null
+++ b/src/audio_core/hle/mixers.h
@@ -0,0 +1,63 @@
+// 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
index 9381883b4..44dff1345 100644
--- a/src/audio_core/hle/pipe.cpp
+++ b/src/audio_core/hle/pipe.cpp
@@ -12,12 +12,14 @@
 #include "common/common_types.h"
 #include "common/logging/log.h"
 
+#include "core/hle/service/dsp_dsp.h"
+
 namespace DSP {
 namespace HLE {
 
 static DspState dsp_state = DspState::Off;
 
-static std::array<std::vector<u8>, static_cast<size_t>(DspPipe::DspPipe_MAX)> pipe_data;
+static std::array<std::vector<u8>, NUM_DSP_PIPE> pipe_data;
 
 void ResetPipes() {
     for (auto& data : pipe_data) {
@@ -27,17 +29,24 @@ void ResetPipes() {
 }
 
 std::vector<u8> PipeRead(DspPipe pipe_number, u32 length) {
-    if (pipe_number >= DspPipe::DspPipe_MAX) {
-        LOG_ERROR(Audio_DSP, "pipe_number = %u invalid", pipe_number);
+    const size_t pipe_index = static_cast<size_t>(pipe_number);
+
+    if (pipe_index >= NUM_DSP_PIPE) {
+        LOG_ERROR(Audio_DSP, "pipe_number = %zu invalid", pipe_index);
         return {};
     }
 
-    std::vector<u8>& data = pipe_data[static_cast<size_t>(pipe_number)];
+    if (length > UINT16_MAX) { // Can only read at most UINT16_MAX from the pipe
+        LOG_ERROR(Audio_DSP, "length of %u greater than max of %u", length, UINT16_MAX);
+        return {};
+    }
+
+    std::vector<u8>& data = pipe_data[pipe_index];
 
     if (length > data.size()) {
-        LOG_WARNING(Audio_DSP, "pipe_number = %u is out of data, application requested read of %u but %zu remain",
-                    pipe_number, length, data.size());
-        length = data.size();
+        LOG_WARNING(Audio_DSP, "pipe_number = %zu is out of data, application requested read of %u but %zu remain",
+                    pipe_index, length, data.size());
+        length = static_cast<u32>(data.size());
     }
 
     if (length == 0)
@@ -49,16 +58,20 @@ std::vector<u8> PipeRead(DspPipe pipe_number, u32 length) {
 }
 
 size_t GetPipeReadableSize(DspPipe pipe_number) {
-    if (pipe_number >= DspPipe::DspPipe_MAX) {
-        LOG_ERROR(Audio_DSP, "pipe_number = %u invalid", pipe_number);
+    const size_t pipe_index = static_cast<size_t>(pipe_number);
+
+    if (pipe_index >= NUM_DSP_PIPE) {
+        LOG_ERROR(Audio_DSP, "pipe_number = %zu invalid", pipe_index);
         return 0;
     }
 
-    return pipe_data[static_cast<size_t>(pipe_number)].size();
+    return pipe_data[pipe_index].size();
 }
 
 static void WriteU16(DspPipe pipe_number, u16 value) {
-    std::vector<u8>& data = pipe_data[static_cast<size_t>(pipe_number)];
+    const size_t pipe_index = static_cast<size_t>(pipe_number);
+
+    std::vector<u8>& data = pipe_data.at(pipe_index);
     // Little endian
     data.emplace_back(value & 0xFF);
     data.emplace_back(value >> 8);
@@ -86,11 +99,13 @@ static void AudioPipeWriteStructAddresses() {
     };
 
     // Begin with a u16 denoting the number of structs.
-    WriteU16(DspPipe::Audio, struct_addresses.size());
+    WriteU16(DspPipe::Audio, static_cast<u16>(struct_addresses.size()));
     // Then write the struct addresses.
     for (u16 addr : struct_addresses) {
         WriteU16(DspPipe::Audio, addr);
     }
+    // Signal that we have data on this pipe.
+    DSP_DSP::SignalPipeInterrupt(DspPipe::Audio);
 }
 
 void PipeWrite(DspPipe pipe_number, const std::vector<u8>& buffer) {
@@ -145,7 +160,7 @@ void PipeWrite(DspPipe pipe_number, const std::vector<u8>& buffer) {
         return;
     }
     default:
-        LOG_CRITICAL(Audio_DSP, "pipe_number = %u unimplemented", pipe_number);
+        LOG_CRITICAL(Audio_DSP, "pipe_number = %zu unimplemented", static_cast<size_t>(pipe_number));
         UNIMPLEMENTED();
         return;
     }
diff --git a/src/audio_core/hle/pipe.h b/src/audio_core/hle/pipe.h
index 382d35e87..b714c0496 100644
--- a/src/audio_core/hle/pipe.h
+++ b/src/audio_core/hle/pipe.h
@@ -19,15 +19,19 @@ enum class DspPipe {
     Debug = 0,
     Dma = 1,
     Audio = 2,
-    Binary = 3,
-    DspPipe_MAX
+    Binary = 3
 };
+constexpr size_t NUM_DSP_PIPE = 8;
 
 /**
- * Read a DSP pipe.
- * @param pipe_number The Pipe ID
- * @param length How much data to request.
- * @return The data read from the pipe. The size of this vector can be less than the length requested.
+ * Reads `length` bytes from the DSP pipe identified with `pipe_number`.
+ * @note Can read up to the maximum value of a u16 in bytes (65,535).
+ * @note IF an error is encoutered with either an invalid `pipe_number` or `length` value, an empty vector will be returned.
+ * @note IF `length` is set to 0, an empty vector will be returned.
+ * @note IF `length` is greater than the amount of data available, this function will only read the available amount.
+ * @param pipe_number a `DspPipe`
+ * @param length the number of bytes to read. The max is 65,535 (max of u16).
+ * @returns a vector of bytes from the specified pipe. On error, will be empty.
  */
 std::vector<u8> PipeRead(DspPipe pipe_number, u32 length);
 
diff --git a/src/audio_core/hle/source.cpp b/src/audio_core/hle/source.cpp
new file mode 100644
index 000000000..30552fe26
--- /dev/null
+++ b/src/audio_core/hle/source.cpp
@@ -0,0 +1,320 @@
+// Copyright 2016 Citra Emulator Project
+// Licensed under GPLv2 or any later version
+// Refer to the license.txt file included.
+
+#include <algorithm>
+#include <array>
+
+#include "audio_core/codec.h"
+#include "audio_core/hle/common.h"
+#include "audio_core/hle/source.h"
+#include "audio_core/interpolate.h"
+
+#include "common/assert.h"
+#include "common/logging/log.h"
+
+#include "core/memory.h"
+
+namespace DSP {
+namespace HLE {
+
+SourceStatus::Status Source::Tick(SourceConfiguration::Configuration& config, const s16_le (&adpcm_coeffs)[16]) {
+    ParseConfig(config, adpcm_coeffs);
+
+    if (state.enabled) {
+        GenerateFrame();
+    }
+
+    return GetCurrentStatus();
+}
+
+void Source::MixInto(QuadFrame32& dest, size_t intermediate_mix_id) const {
+    if (!state.enabled)
+        return;
+
+    const std::array<float, 4>& gains = state.gain.at(intermediate_mix_id);
+    for (size_t samplei = 0; samplei < samples_per_frame; samplei++) {
+        // Conversion from stereo (current_frame) to quadraphonic (dest) occurs here.
+        dest[samplei][0] += static_cast<s32>(gains[0] * current_frame[samplei][0]);
+        dest[samplei][1] += static_cast<s32>(gains[1] * current_frame[samplei][1]);
+        dest[samplei][2] += static_cast<s32>(gains[2] * current_frame[samplei][0]);
+        dest[samplei][3] += static_cast<s32>(gains[3] * current_frame[samplei][1]);
+    }
+}
+
+void Source::Reset() {
+    current_frame.fill({});
+    state = {};
+}
+
+void Source::ParseConfig(SourceConfiguration::Configuration& config, const s16_le (&adpcm_coeffs)[16]) {
+    if (!config.dirty_raw) {
+        return;
+    }
+
+    if (config.reset_flag) {
+        config.reset_flag.Assign(0);
+        Reset();
+        LOG_TRACE(Audio_DSP, "source_id=%zu reset", source_id);
+    }
+
+    if (config.partial_reset_flag) {
+        config.partial_reset_flag.Assign(0);
+        state.input_queue = std::priority_queue<Buffer, std::vector<Buffer>, BufferOrder>{};
+        LOG_TRACE(Audio_DSP, "source_id=%zu partial_reset", source_id);
+    }
+
+    if (config.enable_dirty) {
+        config.enable_dirty.Assign(0);
+        state.enabled = config.enable != 0;
+        LOG_TRACE(Audio_DSP, "source_id=%zu enable=%d", source_id, state.enabled);
+    }
+
+    if (config.sync_dirty) {
+        config.sync_dirty.Assign(0);
+        state.sync = config.sync;
+        LOG_TRACE(Audio_DSP, "source_id=%zu sync=%u", source_id, state.sync);
+    }
+
+    if (config.rate_multiplier_dirty) {
+        config.rate_multiplier_dirty.Assign(0);
+        state.rate_multiplier = config.rate_multiplier;
+        LOG_TRACE(Audio_DSP, "source_id=%zu rate=%f", source_id, state.rate_multiplier);
+
+        if (state.rate_multiplier <= 0) {
+            LOG_ERROR(Audio_DSP, "Was given an invalid rate multiplier: source_id=%zu rate=%f", source_id, state.rate_multiplier);
+            state.rate_multiplier = 1.0f;
+            // Note: Actual firmware starts producing garbage if this occurs.
+        }
+    }
+
+    if (config.adpcm_coefficients_dirty) {
+        config.adpcm_coefficients_dirty.Assign(0);
+        std::transform(adpcm_coeffs, adpcm_coeffs + state.adpcm_coeffs.size(), state.adpcm_coeffs.begin(),
+            [](const auto& coeff) { return static_cast<s16>(coeff); });
+        LOG_TRACE(Audio_DSP, "source_id=%zu adpcm update", source_id);
+    }
+
+    if (config.gain_0_dirty) {
+        config.gain_0_dirty.Assign(0);
+        std::transform(config.gain[0], config.gain[0] + state.gain[0].size(), state.gain[0].begin(),
+            [](const auto& coeff) { return static_cast<float>(coeff); });
+        LOG_TRACE(Audio_DSP, "source_id=%zu gain 0 update", source_id);
+    }
+
+    if (config.gain_1_dirty) {
+        config.gain_1_dirty.Assign(0);
+        std::transform(config.gain[1], config.gain[1] + state.gain[1].size(), state.gain[1].begin(),
+            [](const auto& coeff) { return static_cast<float>(coeff); });
+        LOG_TRACE(Audio_DSP, "source_id=%zu gain 1 update", source_id);
+    }
+
+    if (config.gain_2_dirty) {
+        config.gain_2_dirty.Assign(0);
+        std::transform(config.gain[2], config.gain[2] + state.gain[2].size(), state.gain[2].begin(),
+            [](const auto& coeff) { return static_cast<float>(coeff); });
+        LOG_TRACE(Audio_DSP, "source_id=%zu gain 2 update", source_id);
+    }
+
+    if (config.filters_enabled_dirty) {
+        config.filters_enabled_dirty.Assign(0);
+        state.filters.Enable(config.simple_filter_enabled.ToBool(), config.biquad_filter_enabled.ToBool());
+        LOG_TRACE(Audio_DSP, "source_id=%zu enable_simple=%hu enable_biquad=%hu",
+                  source_id, config.simple_filter_enabled.Value(), config.biquad_filter_enabled.Value());
+    }
+
+    if (config.simple_filter_dirty) {
+        config.simple_filter_dirty.Assign(0);
+        state.filters.Configure(config.simple_filter);
+        LOG_TRACE(Audio_DSP, "source_id=%zu simple filter update", source_id);
+    }
+
+    if (config.biquad_filter_dirty) {
+        config.biquad_filter_dirty.Assign(0);
+        state.filters.Configure(config.biquad_filter);
+        LOG_TRACE(Audio_DSP, "source_id=%zu biquad filter update", source_id);
+    }
+
+    if (config.interpolation_dirty) {
+        config.interpolation_dirty.Assign(0);
+        state.interpolation_mode = config.interpolation_mode;
+        LOG_TRACE(Audio_DSP, "source_id=%zu interpolation_mode=%zu", source_id, static_cast<size_t>(state.interpolation_mode));
+    }
+
+    if (config.format_dirty || config.embedded_buffer_dirty) {
+        config.format_dirty.Assign(0);
+        state.format = config.format;
+        LOG_TRACE(Audio_DSP, "source_id=%zu format=%zu", source_id, static_cast<size_t>(state.format));
+    }
+
+    if (config.mono_or_stereo_dirty || config.embedded_buffer_dirty) {
+        config.mono_or_stereo_dirty.Assign(0);
+        state.mono_or_stereo = config.mono_or_stereo;
+        LOG_TRACE(Audio_DSP, "source_id=%zu mono_or_stereo=%zu", source_id, static_cast<size_t>(state.mono_or_stereo));
+    }
+
+    if (config.embedded_buffer_dirty) {
+        config.embedded_buffer_dirty.Assign(0);
+        state.input_queue.emplace(Buffer{
+            config.physical_address,
+            config.length,
+            static_cast<u8>(config.adpcm_ps),
+            { config.adpcm_yn[0], config.adpcm_yn[1] },
+            config.adpcm_dirty.ToBool(),
+            config.is_looping.ToBool(),
+            config.buffer_id,
+            state.mono_or_stereo,
+            state.format,
+            false
+        });
+        LOG_TRACE(Audio_DSP, "enqueuing embedded addr=0x%08x len=%u id=%hu", config.physical_address, config.length, config.buffer_id);
+    }
+
+    if (config.buffer_queue_dirty) {
+        config.buffer_queue_dirty.Assign(0);
+        for (size_t i = 0; i < 4; i++) {
+            if (config.buffers_dirty & (1 << i)) {
+                const auto& b = config.buffers[i];
+                state.input_queue.emplace(Buffer{
+                    b.physical_address,
+                    b.length,
+                    static_cast<u8>(b.adpcm_ps),
+                    { b.adpcm_yn[0], b.adpcm_yn[1] },
+                    b.adpcm_dirty != 0,
+                    b.is_looping != 0,
+                    b.buffer_id,
+                    state.mono_or_stereo,
+                    state.format,
+                    true
+                });
+                LOG_TRACE(Audio_DSP, "enqueuing queued %zu addr=0x%08x len=%u id=%hu", i, b.physical_address, b.length, b.buffer_id);
+            }
+        }
+        config.buffers_dirty = 0;
+    }
+
+    if (config.dirty_raw) {
+        LOG_DEBUG(Audio_DSP, "source_id=%zu remaining_dirty=%x", source_id, config.dirty_raw);
+    }
+
+    config.dirty_raw = 0;
+}
+
+void Source::GenerateFrame() {
+    current_frame.fill({});
+
+    if (state.current_buffer.empty() && !DequeueBuffer()) {
+        state.enabled = false;
+        state.buffer_update = true;
+        state.current_buffer_id = 0;
+        return;
+    }
+
+    size_t frame_position = 0;
+
+    state.current_sample_number = state.next_sample_number;
+    while (frame_position < current_frame.size()) {
+        if (state.current_buffer.empty() && !DequeueBuffer()) {
+            break;
+        }
+
+        const size_t size_to_copy = std::min(state.current_buffer.size(), current_frame.size() - frame_position);
+
+        std::copy(state.current_buffer.begin(), state.current_buffer.begin() + size_to_copy, current_frame.begin() + frame_position);
+        state.current_buffer.erase(state.current_buffer.begin(), state.current_buffer.begin() + size_to_copy);
+
+        frame_position += size_to_copy;
+        state.next_sample_number += static_cast<u32>(size_to_copy);
+    }
+
+    state.filters.ProcessFrame(current_frame);
+}
+
+
+bool Source::DequeueBuffer() {
+    ASSERT_MSG(state.current_buffer.empty(), "Shouldn't dequeue; we still have data in current_buffer");
+
+    if (state.input_queue.empty())
+        return false;
+
+    const Buffer buf = state.input_queue.top();
+    state.input_queue.pop();
+
+    if (buf.adpcm_dirty) {
+        state.adpcm_state.yn1 = buf.adpcm_yn[0];
+        state.adpcm_state.yn2 = buf.adpcm_yn[1];
+    }
+
+    if (buf.is_looping) {
+        LOG_ERROR(Audio_DSP, "Looped buffers are unimplemented at the moment");
+    }
+
+    const u8* const memory = Memory::GetPhysicalPointer(buf.physical_address);
+    if (memory) {
+        const unsigned num_channels = buf.mono_or_stereo == MonoOrStereo::Stereo ? 2 : 1;
+        switch (buf.format) {
+        case Format::PCM8:
+            state.current_buffer = Codec::DecodePCM8(num_channels, memory, buf.length);
+            break;
+        case Format::PCM16:
+            state.current_buffer = Codec::DecodePCM16(num_channels, memory, buf.length);
+            break;
+        case Format::ADPCM:
+            DEBUG_ASSERT(num_channels == 1);
+            state.current_buffer = Codec::DecodeADPCM(memory, buf.length, state.adpcm_coeffs, state.adpcm_state);
+            break;
+        default:
+            UNIMPLEMENTED();
+            break;
+        }
+    } else {
+        LOG_WARNING(Audio_DSP, "source_id=%zu buffer_id=%hu length=%u: Invalid physical address 0x%08X",
+                               source_id, buf.buffer_id, buf.length, buf.physical_address);
+        state.current_buffer.clear();
+        return true;
+    }
+
+    switch (state.interpolation_mode) {
+    case InterpolationMode::None:
+        state.current_buffer = AudioInterp::None(state.interp_state, state.current_buffer, state.rate_multiplier);
+        break;
+    case InterpolationMode::Linear:
+        state.current_buffer = AudioInterp::Linear(state.interp_state, state.current_buffer, state.rate_multiplier);
+        break;
+    case InterpolationMode::Polyphase:
+        // TODO(merry): Implement polyphase interpolation
+        state.current_buffer = AudioInterp::Linear(state.interp_state, state.current_buffer, state.rate_multiplier);
+        break;
+    default:
+        UNIMPLEMENTED();
+        break;
+    }
+
+    state.current_sample_number = 0;
+    state.next_sample_number = 0;
+    state.current_buffer_id = buf.buffer_id;
+    state.buffer_update = buf.from_queue;
+
+    LOG_TRACE(Audio_DSP, "source_id=%zu buffer_id=%hu from_queue=%s current_buffer.size()=%zu",
+                         source_id, buf.buffer_id, buf.from_queue ? "true" : "false", state.current_buffer.size());
+    return true;
+}
+
+SourceStatus::Status Source::GetCurrentStatus() {
+    SourceStatus::Status ret;
+
+    // Applications depend on the correct emulation of
+    // current_buffer_id_dirty and current_buffer_id to synchronise
+    // audio with video.
+    ret.is_enabled = state.enabled;
+    ret.current_buffer_id_dirty = state.buffer_update ? 1 : 0;
+    state.buffer_update = false;
+    ret.current_buffer_id = state.current_buffer_id;
+    ret.buffer_position = state.current_sample_number;
+    ret.sync = state.sync;
+
+    return ret;
+}
+
+} // namespace HLE
+} // namespace DSP
diff --git a/src/audio_core/hle/source.h b/src/audio_core/hle/source.h
new file mode 100644
index 000000000..7ee08d424
--- /dev/null
+++ b/src/audio_core/hle/source.h
@@ -0,0 +1,144 @@
+// Copyright 2016 Citra Emulator Project
+// Licensed under GPLv2 or any later version
+// Refer to the license.txt file included.
+
+#pragma once
+
+#include <array>
+#include <queue>
+#include <vector>
+
+#include "audio_core/codec.h"
+#include "audio_core/hle/common.h"
+#include "audio_core/hle/dsp.h"
+#include "audio_core/hle/filter.h"
+#include "audio_core/interpolate.h"
+
+#include "common/common_types.h"
+
+namespace DSP {
+namespace HLE {
+
+/**
+ * This module performs:
+ * - Buffer management
+ * - Decoding of buffers
+ * - Buffer resampling and interpolation
+ * - Per-source filtering (SimpleFilter, BiquadFilter)
+ * - Per-source gain
+ * - Other per-source processing
+ */
+class Source final {
+public:
+    explicit Source(size_t source_id_) : source_id(source_id_) {
+        Reset();
+    }
+
+    /// Resets internal state.
+    void Reset();
+
+    /**
+     * This is called once every audio frame. This performs per-source processing every frame.
+     * @param config The new configuration we've got for this Source from the application.
+     * @param adpcm_coeffs ADPCM coefficients to use if config tells us to use them (may contain invalid values otherwise).
+     * @return The current status of this Source. This is given back to the emulated application via SharedMemory.
+     */
+    SourceStatus::Status Tick(SourceConfiguration::Configuration& config, const s16_le (&adpcm_coeffs)[16]);
+
+    /**
+     * Mix this source's output into dest, using the gains for the `intermediate_mix_id`-th intermediate mixer.
+     * @param dest The QuadFrame32 to mix into.
+     * @param intermediate_mix_id The id of the intermediate mix whose gains we are using.
+     */
+    void MixInto(QuadFrame32& dest, size_t intermediate_mix_id) const;
+
+private:
+    const size_t source_id;
+    StereoFrame16 current_frame;
+
+    using Format = SourceConfiguration::Configuration::Format;
+    using InterpolationMode = SourceConfiguration::Configuration::InterpolationMode;
+    using MonoOrStereo = SourceConfiguration::Configuration::MonoOrStereo;
+
+    /// Internal representation of a buffer for our buffer queue
+    struct Buffer {
+        PAddr physical_address;
+        u32 length;
+        u8 adpcm_ps;
+        std::array<u16, 2> adpcm_yn;
+        bool adpcm_dirty;
+        bool is_looping;
+        u16 buffer_id;
+
+        MonoOrStereo mono_or_stereo;
+        Format format;
+
+        bool from_queue;
+    };
+
+    struct BufferOrder {
+        bool operator() (const Buffer& a, const Buffer& b) const {
+            // Lower buffer_id comes first.
+            return a.buffer_id > b.buffer_id;
+        }
+    };
+
+    struct {
+
+        // State variables
+
+        bool enabled = false;
+        u16 sync = 0;
+
+        // Mixing
+
+        std::array<std::array<float, 4>, 3> gain = {};
+
+        // Buffer queue
+
+        std::priority_queue<Buffer, std::vector<Buffer>, BufferOrder> input_queue;
+        MonoOrStereo mono_or_stereo = MonoOrStereo::Mono;
+        Format format = Format::ADPCM;
+
+        // Current buffer
+
+        u32 current_sample_number = 0;
+        u32 next_sample_number = 0;
+        std::vector<std::array<s16, 2>> current_buffer;
+
+        // buffer_id state
+
+        bool buffer_update = false;
+        u32 current_buffer_id = 0;
+
+        // Decoding state
+
+        std::array<s16, 16> adpcm_coeffs = {};
+        Codec::ADPCMState adpcm_state = {};
+
+        // Resampling state
+
+        float rate_multiplier = 1.0;
+        InterpolationMode interpolation_mode = InterpolationMode::Polyphase;
+        AudioInterp::State interp_state = {};
+
+        // Filter state
+
+        SourceFilters filters;
+
+    } state;
+
+    // Internal functions
+
+    /// INTERNAL: Update our internal state based on the current config.
+    void ParseConfig(SourceConfiguration::Configuration& config, const s16_le (&adpcm_coeffs)[16]);
+    /// INTERNAL: Generate the current audio output for this frame based on our internal state.
+    void GenerateFrame();
+    /// INTERNAL: Dequeues a buffer and does preprocessing on it (decoding, resampling). Puts it into current_buffer.
+    bool DequeueBuffer();
+    /// INTERNAL: Generates a SourceStatus::Status based on our internal state.
+    SourceStatus::Status GetCurrentStatus();
+};
+
+} // namespace HLE
+} // namespace DSP