From b242bdf9458642201bab4f1f884556ef73051554 Mon Sep 17 00:00:00 2001
From: MerryMage <MerryMage@users.noreply.github.com>
Date: Mon, 25 Apr 2016 08:54:57 +0100
Subject: DSP/HLE: Implement Source processing

---
 src/audio_core/hle/source.h | 144 ++++++++++++++++++++++++++++++++++++++++++++
 1 file changed, 144 insertions(+)
 create mode 100644 src/audio_core/hle/source.h

(limited to 'src/audio_core/hle/source.h')

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
-- 
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