#ifndef VOICE_H #define VOICE_H #include #include "AL/al.h" #include "AL/alext.h" #include "al/buffer.h" #include "almalloc.h" #include "alspan.h" #include "alu.h" #include "devformat.h" #include "filters/biquad.h" #include "filters/nfc.h" #include "filters/splitter.h" #include "hrtf.h" enum class DistanceModel; enum class SpatializeMode : unsigned char { Off = AL_FALSE, On = AL_TRUE, Auto = AL_AUTO_SOFT }; enum class DirectMode : unsigned char { Off = AL_FALSE, DropMismatch = AL_DROP_UNMATCHED_SOFT, RemixMismatch = AL_REMIX_UNMATCHED_SOFT }; enum class Resampler { Point, Linear, Cubic, FastBSinc12, BSinc12, FastBSinc24, BSinc24, Max = BSinc24 }; extern Resampler ResamplerDefault; /* 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 { float sf; /* Scale interpolation factor. */ ALuint m; /* Coefficient count. */ ALuint l; /* Left coefficient offset. */ /* Filter coefficients, followed by the phase, scale, and scale-phase * delta coefficients. Starting at phase index 0, each subsequent phase * index follows contiguously. */ const float *filter; }; union InterpState { BsincState bsinc; }; using ResamplerFunc = const float*(*)(const InterpState *state, const float *RESTRICT src, ALuint frac, ALuint increment, const al::span dst); ResamplerFunc PrepareResampler(Resampler resampler, ALuint increment, InterpState *state); enum { AF_None = 0, AF_LowPass = 1, AF_HighPass = 2, AF_BandPass = AF_LowPass | AF_HighPass }; struct MixHrtfFilter { const HrirArray *Coeffs; std::array Delay; float Gain; float GainStep; }; struct DirectParams { BiquadFilter LowPass; BiquadFilter HighPass; NfcFilter NFCtrlFilter; struct { HrtfFilter Old; HrtfFilter Target; alignas(16) std::array History; } Hrtf; struct { std::array Current; std::array Target; } Gains; }; struct SendParams { BiquadFilter LowPass; BiquadFilter HighPass; struct { std::array Current; std::array Target; } Gains; }; struct VoiceProps { float Pitch; float Gain; float OuterGain; float MinGain; float MaxGain; float InnerAngle; float OuterAngle; float RefDistance; float MaxDistance; float RolloffFactor; std::array Position; std::array Velocity; std::array Direction; std::array OrientAt; std::array OrientUp; bool HeadRelative; DistanceModel mDistanceModel; Resampler mResampler; DirectMode DirectChannels; SpatializeMode mSpatializeMode; bool DryGainHFAuto; bool WetGainAuto; bool WetGainHFAuto; float OuterGainHF; float AirAbsorptionFactor; float RoomRolloffFactor; float DopplerFactor; std::array StereoPan; float Radius; /** Direct filter and auxiliary send info. */ struct { float Gain; float GainHF; float HFReference; float GainLF; float LFReference; } Direct; struct SendData { ALeffectslot *Slot; float Gain; float GainHF; float HFReference; float GainLF; float LFReference; } Send[MAX_SENDS]; }; struct VoicePropsItem : public VoiceProps { std::atomic next{nullptr}; DEF_NEWDEL(VoicePropsItem) }; #define VOICE_IS_STATIC (1u<<0) #define VOICE_IS_CALLBACK (1u<<1) #define VOICE_IS_AMBISONIC (1u<<2) /* Voice needs HF scaling for ambisonic upsampling. */ #define VOICE_CALLBACK_STOPPED (1u<<3) #define VOICE_IS_FADING (1u<<4) /* Fading sources use gain stepping for smooth transitions. */ #define VOICE_HAS_HRTF (1u<<5) #define VOICE_HAS_NFC (1u<<6) #define VOICE_TYPE_MASK (VOICE_IS_STATIC | VOICE_IS_CALLBACK) struct Voice { enum State { Stopped, Playing, Stopping, Pending }; std::atomic mUpdate{nullptr}; VoiceProps mProps; std::atomic mSourceID{0u}; std::atomic mPlayState{Stopped}; std::atomic mPendingChange{false}; /** * Source offset in samples, relative to the currently playing buffer, NOT * the whole queue. */ std::atomic mPosition; /** Fractional (fixed-point) offset to the next sample. */ std::atomic mPositionFrac; /* Current buffer queue item being played. */ std::atomic mCurrentBuffer; /* Buffer queue item to loop to at end of queue (will be NULL for non- * looping voices). */ std::atomic mLoopBuffer; /* Properties for the attached buffer(s). */ FmtChannels mFmtChannels; ALuint mFrequency; ALuint mSampleSize; AmbiLayout mAmbiLayout; AmbiNorm mAmbiScaling; ALuint mAmbiOrder; /** Current target parameters used for mixing. */ ALuint mStep{0}; ResamplerFunc mResampler; InterpState mResampleState; ALuint mFlags{}; ALuint mNumCallbackSamples{0}; struct TargetData { int FilterType; al::span Buffer; }; TargetData mDirect; std::array mSend; struct ChannelData { alignas(16) std::array mPrevSamples; float mAmbiScale; BandSplitter mAmbiSplitter; DirectParams mDryParams; std::array mWetParams; }; al::vector mChans{2}; Voice() = default; Voice(const Voice&) = delete; ~Voice() { delete mUpdate.exchange(nullptr, std::memory_order_acq_rel); } Voice& operator=(const Voice&) = delete; void mix(const State vstate, ALCcontext *Context, const ALuint SamplesToDo); DEF_NEWDEL(Voice) }; #endif /* VOICE_H */