From 93e60919c8f387c36c267ca9faa1ac653254aea6 Mon Sep 17 00:00:00 2001
From: Chris Robinson <chris.kcat@gmail.com>
Date: Sun, 28 Jul 2019 18:33:29 -0700
Subject: Rename alMain.h to alcmain.h

And move it and alu.h to Alc/.
---
 Alc/alu.h | 466 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
 1 file changed, 466 insertions(+)
 create mode 100644 Alc/alu.h

(limited to 'Alc/alu.h')

diff --git a/Alc/alu.h b/Alc/alu.h
new file mode 100644
index 00000000..9acf904a
--- /dev/null
+++ b/Alc/alu.h
@@ -0,0 +1,466 @@
+#ifndef _ALU_H_
+#define _ALU_H_
+
+#include <array>
+#include <atomic>
+#include <cmath>
+#include <cstddef>
+
+#include "AL/al.h"
+#include "AL/alc.h"
+#include "AL/alext.h"
+
+#include "alBuffer.h"
+#include "alcmain.h"
+#include "almalloc.h"
+#include "alspan.h"
+#include "ambidefs.h"
+#include "filters/biquad.h"
+#include "filters/nfc.h"
+#include "filters/splitter.h"
+#include "hrtf.h"
+#include "logging.h"
+
+struct ALbufferlistitem;
+struct ALeffectslot;
+struct BSincTable;
+
+
+enum class DistanceModel;
+
+#define MAX_PITCH  255
+#define MAX_SENDS  16
+
+
+#define DITHER_RNG_SEED 22222
+
+
+enum SpatializeMode {
+    SpatializeOff = AL_FALSE,
+    SpatializeOn = AL_TRUE,
+    SpatializeAuto = AL_AUTO_SOFT
+};
+
+enum Resampler {
+    PointResampler,
+    LinearResampler,
+    FIR4Resampler,
+    BSinc12Resampler,
+    BSinc24Resampler,
+
+    ResamplerMax = BSinc24Resampler
+};
+extern Resampler ResamplerDefault;
+
+/* The number of distinct scale and phase intervals within the bsinc filter
+ * table.
+ */
+#define BSINC_SCALE_BITS  4
+#define BSINC_SCALE_COUNT (1<<BSINC_SCALE_BITS)
+#define BSINC_PHASE_BITS  4
+#define BSINC_PHASE_COUNT (1<<BSINC_PHASE_BITS)
+
+/* Interpolator state.  Kind of a misnomer since the interpolator itself is
+ * stateless.  This just keeps it from having to recompute scale-related
+ * mappings for every sample.
+ */
+struct BsincState {
+    ALfloat sf; /* Scale interpolation factor. */
+    ALsizei m;  /* Coefficient count. */
+    ALsizei l;  /* Left coefficient offset. */
+    /* Filter coefficients, followed by the scale, phase, and scale-phase
+     * delta coefficients. Starting at phase index 0, each subsequent phase
+     * index follows contiguously.
+     */
+    const ALfloat *filter;
+};
+
+union InterpState {
+    BsincState bsinc;
+};
+
+using ResamplerFunc = const ALfloat*(*)(const InterpState *state,
+    const ALfloat *RESTRICT src, ALsizei frac, ALint increment,
+    ALfloat *RESTRICT dst, ALsizei dstlen);
+
+void BsincPrepare(const ALuint increment, BsincState *state, const BSincTable *table);
+
+extern const BSincTable bsinc12;
+extern const BSincTable bsinc24;
+
+
+enum {
+    AF_None = 0,
+    AF_LowPass = 1,
+    AF_HighPass = 2,
+    AF_BandPass = AF_LowPass | AF_HighPass
+};
+
+
+struct MixHrtfFilter {
+    const HrirArray<ALfloat> *Coeffs;
+    ALsizei Delay[2];
+    ALfloat Gain;
+    ALfloat GainStep;
+};
+
+
+struct DirectParams {
+    BiquadFilter LowPass;
+    BiquadFilter HighPass;
+
+    NfcFilter NFCtrlFilter;
+
+    struct {
+        HrtfFilter Old;
+        HrtfFilter Target;
+        HrtfState State;
+    } Hrtf;
+
+    struct {
+        ALfloat Current[MAX_OUTPUT_CHANNELS];
+        ALfloat Target[MAX_OUTPUT_CHANNELS];
+    } Gains;
+};
+
+struct SendParams {
+    BiquadFilter LowPass;
+    BiquadFilter HighPass;
+
+    struct {
+        ALfloat Current[MAX_OUTPUT_CHANNELS];
+        ALfloat Target[MAX_OUTPUT_CHANNELS];
+    } Gains;
+};
+
+
+struct ALvoicePropsBase {
+    ALfloat Pitch;
+    ALfloat Gain;
+    ALfloat OuterGain;
+    ALfloat MinGain;
+    ALfloat MaxGain;
+    ALfloat InnerAngle;
+    ALfloat OuterAngle;
+    ALfloat RefDistance;
+    ALfloat MaxDistance;
+    ALfloat RolloffFactor;
+    std::array<ALfloat,3> Position;
+    std::array<ALfloat,3> Velocity;
+    std::array<ALfloat,3> Direction;
+    std::array<ALfloat,3> OrientAt;
+    std::array<ALfloat,3> OrientUp;
+    ALboolean HeadRelative;
+    DistanceModel mDistanceModel;
+    Resampler mResampler;
+    ALboolean DirectChannels;
+    SpatializeMode mSpatializeMode;
+
+    ALboolean DryGainHFAuto;
+    ALboolean WetGainAuto;
+    ALboolean WetGainHFAuto;
+    ALfloat   OuterGainHF;
+
+    ALfloat AirAbsorptionFactor;
+    ALfloat RoomRolloffFactor;
+    ALfloat DopplerFactor;
+
+    std::array<ALfloat,2> StereoPan;
+
+    ALfloat Radius;
+
+    /** Direct filter and auxiliary send info. */
+    struct {
+        ALfloat Gain;
+        ALfloat GainHF;
+        ALfloat HFReference;
+        ALfloat GainLF;
+        ALfloat LFReference;
+    } Direct;
+    struct SendData {
+        ALeffectslot *Slot;
+        ALfloat Gain;
+        ALfloat GainHF;
+        ALfloat HFReference;
+        ALfloat GainLF;
+        ALfloat LFReference;
+    } Send[MAX_SENDS];
+};
+
+struct ALvoiceProps : public ALvoicePropsBase {
+    std::atomic<ALvoiceProps*> next{nullptr};
+
+    DEF_NEWDEL(ALvoiceProps)
+};
+
+#define VOICE_IS_STATIC    (1u<<0)
+#define VOICE_IS_FADING    (1u<<1) /* Fading sources use gain stepping for smooth transitions. */
+#define VOICE_IS_AMBISONIC (1u<<2) /* Voice needs HF scaling for ambisonic upsampling. */
+#define VOICE_HAS_HRTF     (1u<<3)
+#define VOICE_HAS_NFC      (1u<<4)
+
+struct ALvoice {
+    enum State {
+        Stopped = 0,
+        Playing = 1,
+        Stopping = 2
+    };
+
+    std::atomic<ALvoiceProps*> mUpdate{nullptr};
+
+    std::atomic<ALuint> mSourceID{0u};
+    std::atomic<State> mPlayState{Stopped};
+
+    ALvoicePropsBase mProps;
+
+    /**
+     * Source offset in samples, relative to the currently playing buffer, NOT
+     * the whole queue.
+     */
+    std::atomic<ALuint> mPosition;
+    /** Fractional (fixed-point) offset to the next sample. */
+    std::atomic<ALsizei> mPositionFrac;
+
+    /* Current buffer queue item being played. */
+    std::atomic<ALbufferlistitem*> mCurrentBuffer;
+
+    /* Buffer queue item to loop to at end of queue (will be NULL for non-
+     * looping voices).
+     */
+    std::atomic<ALbufferlistitem*> mLoopBuffer;
+
+    /* Properties for the attached buffer(s). */
+    FmtChannels mFmtChannels;
+    ALuint mFrequency;
+    ALsizei mNumChannels;
+    ALsizei mSampleSize;
+
+    /** Current target parameters used for mixing. */
+    ALint mStep;
+
+    ResamplerFunc mResampler;
+
+    InterpState mResampleState;
+
+    ALuint mFlags;
+
+    struct DirectData {
+        int FilterType;
+        al::span<FloatBufferLine> Buffer;
+    };
+    DirectData mDirect;
+
+    struct SendData {
+        int FilterType;
+        al::span<FloatBufferLine> Buffer;
+    };
+    std::array<SendData,MAX_SENDS> mSend;
+
+    struct ChannelData {
+        alignas(16) std::array<ALfloat,MAX_RESAMPLE_PADDING*2> mPrevSamples;
+
+        ALfloat mAmbiScale;
+        BandSplitter mAmbiSplitter;
+
+        DirectParams mDryParams;
+        std::array<SendParams,MAX_SENDS> mWetParams;
+    };
+    std::array<ChannelData,MAX_INPUT_CHANNELS> mChans;
+
+    ALvoice() = default;
+    ALvoice(const ALvoice&) = delete;
+    ~ALvoice() { delete mUpdate.exchange(nullptr, std::memory_order_acq_rel); }
+    ALvoice& operator=(const ALvoice&) = delete;
+    ALvoice& operator=(ALvoice&& rhs) noexcept
+    {
+        ALvoiceProps *old_update{mUpdate.load(std::memory_order_relaxed)};
+        mUpdate.store(rhs.mUpdate.exchange(old_update, std::memory_order_relaxed),
+            std::memory_order_relaxed);
+
+        mSourceID.store(rhs.mSourceID.load(std::memory_order_relaxed), std::memory_order_relaxed);
+        mPlayState.store(rhs.mPlayState.load(std::memory_order_relaxed),
+            std::memory_order_relaxed);
+
+        mProps = rhs.mProps;
+
+        mPosition.store(rhs.mPosition.load(std::memory_order_relaxed), std::memory_order_relaxed);
+        mPositionFrac.store(rhs.mPositionFrac.load(std::memory_order_relaxed),
+            std::memory_order_relaxed);
+
+        mCurrentBuffer.store(rhs.mCurrentBuffer.load(std::memory_order_relaxed),
+            std::memory_order_relaxed);
+        mLoopBuffer.store(rhs.mLoopBuffer.load(std::memory_order_relaxed),
+            std::memory_order_relaxed);
+
+        mFmtChannels = rhs.mFmtChannels;
+        mFrequency = rhs.mFrequency;
+        mNumChannels = rhs.mNumChannels;
+        mSampleSize = rhs.mSampleSize;
+
+        mStep = rhs.mStep;
+        mResampler = rhs.mResampler;
+
+        mResampleState = rhs.mResampleState;
+
+        mFlags = rhs.mFlags;
+
+        mDirect = rhs.mDirect;
+        mSend = rhs.mSend;
+        mChans = rhs.mChans;
+
+        return *this;
+    }
+};
+
+
+using MixerFunc = void(*)(const ALfloat *data, const al::span<FloatBufferLine> OutBuffer,
+    ALfloat *CurrentGains, const ALfloat *TargetGains, const ALsizei Counter, const ALsizei OutPos,
+    const ALsizei BufferSize);
+using RowMixerFunc = void(*)(FloatBufferLine &OutBuffer, const ALfloat *gains,
+    const al::span<const FloatBufferLine> InSamples, const ALsizei InPos,
+    const ALsizei BufferSize);
+using HrtfMixerFunc = void(*)(FloatBufferLine &LeftOut, FloatBufferLine &RightOut,
+    const ALfloat *InSamples, float2 *AccumSamples, const ALsizei OutPos, const ALsizei IrSize,
+    MixHrtfFilter *hrtfparams, const ALsizei BufferSize);
+using HrtfMixerBlendFunc = void(*)(FloatBufferLine &LeftOut, FloatBufferLine &RightOut,
+    const ALfloat *InSamples, float2 *AccumSamples, const ALsizei OutPos, const ALsizei IrSize,
+    const HrtfFilter *oldparams, MixHrtfFilter *newparams, const ALsizei BufferSize);
+using HrtfDirectMixerFunc = void(*)(FloatBufferLine &LeftOut, FloatBufferLine &RightOut,
+    const al::span<const FloatBufferLine> InSamples, float2 *AccumSamples, DirectHrtfState *State,
+    const ALsizei BufferSize);
+
+
+#define GAIN_MIX_MAX  (1000.0f) /* +60dB */
+
+#define GAIN_SILENCE_THRESHOLD  (0.00001f) /* -100dB */
+
+#define SPEEDOFSOUNDMETRESPERSEC  (343.3f)
+#define AIRABSORBGAINHF           (0.99426f) /* -0.05dB */
+
+/* Target gain for the reverb decay feedback reaching the decay time. */
+#define REVERB_DECAY_GAIN  (0.001f) /* -60 dB */
+
+#define FRACTIONBITS (12)
+#define FRACTIONONE  (1<<FRACTIONBITS)
+#define FRACTIONMASK (FRACTIONONE-1)
+
+
+inline ALfloat lerp(ALfloat val1, ALfloat val2, ALfloat mu) noexcept
+{ return val1 + (val2-val1)*mu; }
+inline ALfloat cubic(ALfloat val1, ALfloat val2, ALfloat val3, ALfloat val4, ALfloat mu) noexcept
+{
+    ALfloat mu2 = mu*mu, mu3 = mu2*mu;
+    ALfloat a0 = -0.5f*mu3 +       mu2 + -0.5f*mu;
+    ALfloat a1 =  1.5f*mu3 + -2.5f*mu2            + 1.0f;
+    ALfloat a2 = -1.5f*mu3 +  2.0f*mu2 +  0.5f*mu;
+    ALfloat a3 =  0.5f*mu3 + -0.5f*mu2;
+    return val1*a0 + val2*a1 + val3*a2 + val4*a3;
+}
+
+
+enum HrtfRequestMode {
+    Hrtf_Default = 0,
+    Hrtf_Enable = 1,
+    Hrtf_Disable = 2,
+};
+
+void aluInit(void);
+
+void aluInitMixer(void);
+
+ResamplerFunc SelectResampler(Resampler resampler);
+
+/* aluInitRenderer
+ *
+ * Set up the appropriate panning method and mixing method given the device
+ * properties.
+ */
+void aluInitRenderer(ALCdevice *device, ALint hrtf_id, HrtfRequestMode hrtf_appreq, HrtfRequestMode hrtf_userreq);
+
+void aluInitEffectPanning(ALeffectslot *slot, ALCdevice *device);
+
+void ProcessHrtf(ALCdevice *device, const ALsizei SamplesToDo);
+void ProcessAmbiDec(ALCdevice *device, const ALsizei SamplesToDo);
+void ProcessUhj(ALCdevice *device, const ALsizei SamplesToDo);
+void ProcessBs2b(ALCdevice *device, const ALsizei SamplesToDo);
+
+/**
+ * Calculates ambisonic encoder coefficients using the X, Y, and Z direction
+ * components, which must represent a normalized (unit length) vector, and the
+ * spread is the angular width of the sound (0...tau).
+ *
+ * NOTE: The components use ambisonic coordinates. As a result:
+ *
+ * Ambisonic Y = OpenAL -X
+ * Ambisonic Z = OpenAL Y
+ * Ambisonic X = OpenAL -Z
+ *
+ * The components are ordered such that OpenAL's X, Y, and Z are the first,
+ * second, and third parameters respectively -- simply negate X and Z.
+ */
+void CalcAmbiCoeffs(const ALfloat y, const ALfloat z, const ALfloat x, const ALfloat spread,
+                    ALfloat (&coeffs)[MAX_AMBI_CHANNELS]);
+
+/**
+ * CalcDirectionCoeffs
+ *
+ * Calculates ambisonic coefficients based on an OpenAL direction vector. The
+ * vector must be normalized (unit length), and the spread is the angular width
+ * of the sound (0...tau).
+ */
+inline void CalcDirectionCoeffs(const ALfloat (&dir)[3], ALfloat spread, ALfloat (&coeffs)[MAX_AMBI_CHANNELS])
+{
+    /* Convert from OpenAL coords to Ambisonics. */
+    CalcAmbiCoeffs(-dir[0], dir[1], -dir[2], spread, coeffs);
+}
+
+/**
+ * CalcAngleCoeffs
+ *
+ * Calculates ambisonic coefficients based on azimuth and elevation. The
+ * azimuth and elevation parameters are in radians, going right and up
+ * respectively.
+ */
+inline void CalcAngleCoeffs(ALfloat azimuth, ALfloat elevation, ALfloat spread, ALfloat (&coeffs)[MAX_AMBI_CHANNELS])
+{
+    ALfloat x = -std::sin(azimuth) * std::cos(elevation);
+    ALfloat y = std::sin(elevation);
+    ALfloat z = std::cos(azimuth) * std::cos(elevation);
+
+    CalcAmbiCoeffs(x, y, z, spread, coeffs);
+}
+
+
+/**
+ * ComputePanGains
+ *
+ * Computes panning gains using the given channel decoder coefficients and the
+ * pre-calculated direction or angle coefficients. For B-Format sources, the
+ * coeffs are a 'slice' of a transform matrix for the input channel, used to
+ * scale and orient the sound samples.
+ */
+void ComputePanGains(const MixParams *mix, const ALfloat*RESTRICT coeffs, ALfloat ingain, ALfloat (&gains)[MAX_OUTPUT_CHANNELS]);
+
+
+inline std::array<ALfloat,MAX_AMBI_CHANNELS> GetAmbiIdentityRow(size_t i) noexcept
+{
+    std::array<ALfloat,MAX_AMBI_CHANNELS> ret{};
+    ret[i] = 1.0f;
+    return ret;
+}
+
+
+void MixVoice(ALvoice *voice, ALvoice::State vstate, const ALuint SourceID, ALCcontext *Context, const ALsizei SamplesToDo);
+
+void aluMixData(ALCdevice *device, ALvoid *OutBuffer, ALsizei NumSamples);
+/* Caller must lock the device state, and the mixer must not be running. */
+void aluHandleDisconnect(ALCdevice *device, const char *msg, ...) DECL_FORMAT(printf, 2, 3);
+
+extern MixerFunc MixSamples;
+extern RowMixerFunc MixRowSamples;
+
+extern const ALfloat ConeScale;
+extern const ALfloat ZScale;
+extern const ALboolean OverrideReverbSpeedOfSound;
+
+#endif
-- 
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