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Diffstat (limited to 'Alc/alu.cpp')
| -rw-r--r-- | Alc/alu.cpp | 1884 |
1 files changed, 1884 insertions, 0 deletions
diff --git a/Alc/alu.cpp b/Alc/alu.cpp new file mode 100644 index 00000000..df857b80 --- /dev/null +++ b/Alc/alu.cpp @@ -0,0 +1,1884 @@ +/** + * OpenAL cross platform audio library + * Copyright (C) 1999-2007 by authors. + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Library General Public + * License as published by the Free Software Foundation; either + * version 2 of the License, or (at your option) any later version. + * + * This library is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Library General Public License for more details. + * + * You should have received a copy of the GNU Library General Public + * License along with this library; if not, write to the + * Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + * Or go to http://www.gnu.org/copyleft/lgpl.html + */ + +#include "config.h" + +#include <math.h> +#include <stdlib.h> +#include <string.h> +#include <ctype.h> +#include <assert.h> + +#include "alMain.h" +#include "alSource.h" +#include "alBuffer.h" +#include "alListener.h" +#include "alAuxEffectSlot.h" +#include "alu.h" +#include "bs2b.h" +#include "hrtf.h" +#include "mastering.h" +#include "uhjfilter.h" +#include "bformatdec.h" +#include "static_assert.h" +#include "ringbuffer.h" +#include "filters/splitter.h" + +#include "mixer/defs.h" +#include "fpu_modes.h" +#include "cpu_caps.h" +#include "bsinc_inc.h" + + +/* Cone scalar */ +ALfloat ConeScale = 1.0f; + +/* Localized Z scalar for mono sources */ +ALfloat ZScale = 1.0f; + +/* Force default speed of sound for distance-related reverb decay. */ +ALboolean OverrideReverbSpeedOfSound = AL_FALSE; + +const aluMatrixf IdentityMatrixf = {{ + { 1.0f, 0.0f, 0.0f, 0.0f }, + { 0.0f, 1.0f, 0.0f, 0.0f }, + { 0.0f, 0.0f, 1.0f, 0.0f }, + { 0.0f, 0.0f, 0.0f, 1.0f }, +}}; + + +static void ClearArray(ALfloat f[MAX_OUTPUT_CHANNELS]) +{ + size_t i; + for(i = 0;i < MAX_OUTPUT_CHANNELS;i++) + f[i] = 0.0f; +} + +struct ChanMap { + enum Channel channel; + ALfloat angle; + ALfloat elevation; +}; + +static HrtfDirectMixerFunc MixDirectHrtf = MixDirectHrtf_C; + + +void DeinitVoice(ALvoice *voice) +{ + al_free(ATOMIC_EXCHANGE_PTR_SEQ(&voice->Update, static_cast<ALvoiceProps*>(nullptr))); +} + + +static inline HrtfDirectMixerFunc SelectHrtfMixer(void) +{ +#ifdef HAVE_NEON + if((CPUCapFlags&CPU_CAP_NEON)) + return MixDirectHrtf_Neon; +#endif +#ifdef HAVE_SSE + if((CPUCapFlags&CPU_CAP_SSE)) + return MixDirectHrtf_SSE; +#endif + + return MixDirectHrtf_C; +} + + +/* This RNG method was created based on the math found in opusdec. It's quick, + * and starting with a seed value of 22222, is suitable for generating + * whitenoise. + */ +static inline ALuint dither_rng(ALuint *seed) +{ + *seed = (*seed * 96314165) + 907633515; + return *seed; +} + + +static inline void aluCrossproduct(const ALfloat *inVector1, const ALfloat *inVector2, ALfloat *outVector) +{ + outVector[0] = inVector1[1]*inVector2[2] - inVector1[2]*inVector2[1]; + outVector[1] = inVector1[2]*inVector2[0] - inVector1[0]*inVector2[2]; + outVector[2] = inVector1[0]*inVector2[1] - inVector1[1]*inVector2[0]; +} + +static inline ALfloat aluDotproduct(const aluVector *vec1, const aluVector *vec2) +{ + return vec1->v[0]*vec2->v[0] + vec1->v[1]*vec2->v[1] + vec1->v[2]*vec2->v[2]; +} + +static ALfloat aluNormalize(ALfloat *vec) +{ + ALfloat length = sqrtf(vec[0]*vec[0] + vec[1]*vec[1] + vec[2]*vec[2]); + if(length > FLT_EPSILON) + { + ALfloat inv_length = 1.0f/length; + vec[0] *= inv_length; + vec[1] *= inv_length; + vec[2] *= inv_length; + return length; + } + vec[0] = vec[1] = vec[2] = 0.0f; + return 0.0f; +} + +static void aluMatrixfFloat3(ALfloat *vec, ALfloat w, const aluMatrixf *mtx) +{ + ALfloat v[4] = { vec[0], vec[1], vec[2], w }; + + vec[0] = v[0]*mtx->m[0][0] + v[1]*mtx->m[1][0] + v[2]*mtx->m[2][0] + v[3]*mtx->m[3][0]; + vec[1] = v[0]*mtx->m[0][1] + v[1]*mtx->m[1][1] + v[2]*mtx->m[2][1] + v[3]*mtx->m[3][1]; + vec[2] = v[0]*mtx->m[0][2] + v[1]*mtx->m[1][2] + v[2]*mtx->m[2][2] + v[3]*mtx->m[3][2]; +} + +static aluVector aluMatrixfVector(const aluMatrixf *mtx, const aluVector *vec) +{ + aluVector v; + v.v[0] = vec->v[0]*mtx->m[0][0] + vec->v[1]*mtx->m[1][0] + vec->v[2]*mtx->m[2][0] + vec->v[3]*mtx->m[3][0]; + v.v[1] = vec->v[0]*mtx->m[0][1] + vec->v[1]*mtx->m[1][1] + vec->v[2]*mtx->m[2][1] + vec->v[3]*mtx->m[3][1]; + v.v[2] = vec->v[0]*mtx->m[0][2] + vec->v[1]*mtx->m[1][2] + vec->v[2]*mtx->m[2][2] + vec->v[3]*mtx->m[3][2]; + v.v[3] = vec->v[0]*mtx->m[0][3] + vec->v[1]*mtx->m[1][3] + vec->v[2]*mtx->m[2][3] + vec->v[3]*mtx->m[3][3]; + return v; +} + + +void aluInit(void) +{ + MixDirectHrtf = SelectHrtfMixer(); +} + + +static void SendSourceStoppedEvent(ALCcontext *context, ALuint id) +{ + AsyncEvent evt = ASYNC_EVENT(EventType_SourceStateChange); + ALbitfieldSOFT enabledevt; + size_t strpos; + ALuint scale; + + enabledevt = ATOMIC_LOAD(&context->EnabledEvts, almemory_order_acquire); + if(!(enabledevt&EventType_SourceStateChange)) return; + + evt.u.user.type = AL_EVENT_TYPE_SOURCE_STATE_CHANGED_SOFT; + evt.u.user.id = id; + evt.u.user.param = AL_STOPPED; + + /* Normally snprintf would be used, but this is called from the mixer and + * that function's not real-time safe, so we have to construct it manually. + */ + strcpy(evt.u.user.msg, "Source ID "); strpos = 10; + scale = 1000000000; + while(scale > 0 && scale > id) + scale /= 10; + while(scale > 0) + { + evt.u.user.msg[strpos++] = '0' + ((id/scale)%10); + scale /= 10; + } + strcpy(evt.u.user.msg+strpos, " state changed to AL_STOPPED"); + + if(ll_ringbuffer_write(context->AsyncEvents, (const char*)&evt, 1) == 1) + alsem_post(&context->EventSem); +} + + +static void ProcessHrtf(ALCdevice *device, ALsizei SamplesToDo) +{ + DirectHrtfState *state; + int lidx, ridx; + ALsizei c; + + if(device->AmbiUp) + ambiup_process(device->AmbiUp, + device->Dry.Buffer, device->Dry.NumChannels, device->FOAOut.Buffer, + SamplesToDo + ); + + lidx = GetChannelIdxByName(&device->RealOut, FrontLeft); + ridx = GetChannelIdxByName(&device->RealOut, FrontRight); + assert(lidx != -1 && ridx != -1); + + state = device->Hrtf; + for(c = 0;c < device->Dry.NumChannels;c++) + { + MixDirectHrtf(device->RealOut.Buffer[lidx], device->RealOut.Buffer[ridx], + device->Dry.Buffer[c], state->Offset, state->IrSize, + state->Chan[c].Coeffs, state->Chan[c].Values, SamplesToDo + ); + } + state->Offset += SamplesToDo; +} + +static void ProcessAmbiDec(ALCdevice *device, ALsizei SamplesToDo) +{ + if(device->Dry.Buffer != device->FOAOut.Buffer) + bformatdec_upSample(device->AmbiDecoder, + device->Dry.Buffer, device->FOAOut.Buffer, device->FOAOut.NumChannels, + SamplesToDo + ); + bformatdec_process(device->AmbiDecoder, + device->RealOut.Buffer, device->RealOut.NumChannels, device->Dry.Buffer, + SamplesToDo + ); +} + +static void ProcessAmbiUp(ALCdevice *device, ALsizei SamplesToDo) +{ + ambiup_process(device->AmbiUp, + device->RealOut.Buffer, device->RealOut.NumChannels, device->FOAOut.Buffer, + SamplesToDo + ); +} + +static void ProcessUhj(ALCdevice *device, ALsizei SamplesToDo) +{ + int lidx = GetChannelIdxByName(&device->RealOut, FrontLeft); + int ridx = GetChannelIdxByName(&device->RealOut, FrontRight); + assert(lidx != -1 && ridx != -1); + + /* Encode to stereo-compatible 2-channel UHJ output. */ + EncodeUhj2(device->Uhj_Encoder, + device->RealOut.Buffer[lidx], device->RealOut.Buffer[ridx], + device->Dry.Buffer, SamplesToDo + ); +} + +static void ProcessBs2b(ALCdevice *device, ALsizei SamplesToDo) +{ + int lidx = GetChannelIdxByName(&device->RealOut, FrontLeft); + int ridx = GetChannelIdxByName(&device->RealOut, FrontRight); + assert(lidx != -1 && ridx != -1); + + /* Apply binaural/crossfeed filter */ + bs2b_cross_feed(device->Bs2b, device->RealOut.Buffer[lidx], + device->RealOut.Buffer[ridx], SamplesToDo); +} + +void aluSelectPostProcess(ALCdevice *device) +{ + if(device->HrtfHandle) + device->PostProcess = ProcessHrtf; + else if(device->AmbiDecoder) + device->PostProcess = ProcessAmbiDec; + else if(device->AmbiUp) + device->PostProcess = ProcessAmbiUp; + else if(device->Uhj_Encoder) + device->PostProcess = ProcessUhj; + else if(device->Bs2b) + device->PostProcess = ProcessBs2b; + else + device->PostProcess = NULL; +} + + +/* Prepares the interpolator for a given rate (determined by increment). + * + * With a bit of work, and a trade of memory for CPU cost, this could be + * modified for use with an interpolated increment for buttery-smooth pitch + * changes. + */ +void BsincPrepare(const ALuint increment, BsincState *state, const BSincTable *table) +{ + ALfloat sf = 0.0f; + ALsizei si = BSINC_SCALE_COUNT-1; + + if(increment > FRACTIONONE) + { + sf = (ALfloat)FRACTIONONE / increment; + sf = maxf(0.0f, (BSINC_SCALE_COUNT-1) * (sf-table->scaleBase) * table->scaleRange); + si = float2int(sf); + /* The interpolation factor is fit to this diagonally-symmetric curve + * to reduce the transition ripple caused by interpolating different + * scales of the sinc function. + */ + sf = 1.0f - cosf(asinf(sf - si)); + } + + state->sf = sf; + state->m = table->m[si]; + state->l = (state->m/2) - 1; + state->filter = table->Tab + table->filterOffset[si]; +} + + +static bool CalcContextParams(ALCcontext *Context) +{ + ALlistener *Listener = Context->Listener; + struct ALcontextProps *props; + + props = static_cast<ALcontextProps*>(ATOMIC_EXCHANGE_PTR(&Context->Update, + static_cast<ALcontextProps*>(nullptr), almemory_order_acq_rel)); + if(!props) return false; + + Listener->Params.MetersPerUnit = props->MetersPerUnit; + + Listener->Params.DopplerFactor = props->DopplerFactor; + Listener->Params.SpeedOfSound = props->SpeedOfSound * props->DopplerVelocity; + if(!OverrideReverbSpeedOfSound) + Listener->Params.ReverbSpeedOfSound = Listener->Params.SpeedOfSound * + Listener->Params.MetersPerUnit; + + Listener->Params.SourceDistanceModel = props->SourceDistanceModel; + Listener->Params.DistanceModel = props->DistanceModel; + + ATOMIC_REPLACE_HEAD(struct ALcontextProps*, &Context->FreeContextProps, props); + return true; +} + +static bool CalcListenerParams(ALCcontext *Context) +{ + ALlistener *Listener = Context->Listener; + ALfloat N[3], V[3], U[3], P[3]; + struct ALlistenerProps *props; + aluVector vel; + + props = static_cast<ALlistenerProps*>(ATOMIC_EXCHANGE_PTR(&Listener->Update, + static_cast<ALlistenerProps*>(nullptr), almemory_order_acq_rel)); + if(!props) return false; + + /* AT then UP */ + N[0] = props->Forward[0]; + N[1] = props->Forward[1]; + N[2] = props->Forward[2]; + aluNormalize(N); + V[0] = props->Up[0]; + V[1] = props->Up[1]; + V[2] = props->Up[2]; + aluNormalize(V); + /* Build and normalize right-vector */ + aluCrossproduct(N, V, U); + aluNormalize(U); + + aluMatrixfSet(&Listener->Params.Matrix, + U[0], V[0], -N[0], 0.0, + U[1], V[1], -N[1], 0.0, + U[2], V[2], -N[2], 0.0, + 0.0, 0.0, 0.0, 1.0 + ); + + P[0] = props->Position[0]; + P[1] = props->Position[1]; + P[2] = props->Position[2]; + aluMatrixfFloat3(P, 1.0, &Listener->Params.Matrix); + aluMatrixfSetRow(&Listener->Params.Matrix, 3, -P[0], -P[1], -P[2], 1.0f); + + aluVectorSet(&vel, props->Velocity[0], props->Velocity[1], props->Velocity[2], 0.0f); + Listener->Params.Velocity = aluMatrixfVector(&Listener->Params.Matrix, &vel); + + Listener->Params.Gain = props->Gain * Context->GainBoost; + + ATOMIC_REPLACE_HEAD(struct ALlistenerProps*, &Context->FreeListenerProps, props); + return true; +} + +static bool CalcEffectSlotParams(ALeffectslot *slot, ALCcontext *context, bool force) +{ + struct ALeffectslotProps *props; + ALeffectState *state; + + props = static_cast<ALeffectslotProps*>(ATOMIC_EXCHANGE_PTR(&slot->Update, + static_cast<ALeffectslotProps*>(nullptr), almemory_order_acq_rel)); + if(!props && !force) return false; + + if(props) + { + slot->Params.Gain = props->Gain; + slot->Params.AuxSendAuto = props->AuxSendAuto; + slot->Params.EffectType = props->Type; + slot->Params.EffectProps = props->Props; + if(IsReverbEffect(props->Type)) + { + slot->Params.RoomRolloff = props->Props.Reverb.RoomRolloffFactor; + slot->Params.DecayTime = props->Props.Reverb.DecayTime; + slot->Params.DecayLFRatio = props->Props.Reverb.DecayLFRatio; + slot->Params.DecayHFRatio = props->Props.Reverb.DecayHFRatio; + slot->Params.DecayHFLimit = props->Props.Reverb.DecayHFLimit; + slot->Params.AirAbsorptionGainHF = props->Props.Reverb.AirAbsorptionGainHF; + } + else + { + slot->Params.RoomRolloff = 0.0f; + slot->Params.DecayTime = 0.0f; + slot->Params.DecayLFRatio = 0.0f; + slot->Params.DecayHFRatio = 0.0f; + slot->Params.DecayHFLimit = AL_FALSE; + slot->Params.AirAbsorptionGainHF = 1.0f; + } + + state = props->State; + + if(state == slot->Params.EffectState) + { + /* If the effect state is the same as current, we can decrement its + * count safely to remove it from the update object (it can't reach + * 0 refs since the current params also hold a reference). + */ + DecrementRef(&state->Ref); + props->State = NULL; + } + else + { + /* Otherwise, replace it and send off the old one with a release + * event. + */ + AsyncEvent evt = ASYNC_EVENT(EventType_ReleaseEffectState); + evt.u.EffectState = slot->Params.EffectState; + + slot->Params.EffectState = state; + props->State = NULL; + + if(LIKELY(ll_ringbuffer_write(context->AsyncEvents, (const char*)&evt, 1) != 0)) + alsem_post(&context->EventSem); + else + { + /* If writing the event failed, the queue was probably full. + * Store the old state in the property object where it can + * eventually be cleaned up sometime later (not ideal, but + * better than blocking or leaking). + */ + props->State = evt.u.EffectState; + } + } + + ATOMIC_REPLACE_HEAD(struct ALeffectslotProps*, &context->FreeEffectslotProps, props); + } + else + state = slot->Params.EffectState; + + V(state,update)(context, slot, &slot->Params.EffectProps); + return true; +} + + +static const struct ChanMap MonoMap[1] = { + { FrontCenter, 0.0f, 0.0f } +}, RearMap[2] = { + { BackLeft, DEG2RAD(-150.0f), DEG2RAD(0.0f) }, + { BackRight, DEG2RAD( 150.0f), DEG2RAD(0.0f) } +}, QuadMap[4] = { + { FrontLeft, DEG2RAD( -45.0f), DEG2RAD(0.0f) }, + { FrontRight, DEG2RAD( 45.0f), DEG2RAD(0.0f) }, + { BackLeft, DEG2RAD(-135.0f), DEG2RAD(0.0f) }, + { BackRight, DEG2RAD( 135.0f), DEG2RAD(0.0f) } +}, X51Map[6] = { + { FrontLeft, DEG2RAD( -30.0f), DEG2RAD(0.0f) }, + { FrontRight, DEG2RAD( 30.0f), DEG2RAD(0.0f) }, + { FrontCenter, DEG2RAD( 0.0f), DEG2RAD(0.0f) }, + { LFE, 0.0f, 0.0f }, + { SideLeft, DEG2RAD(-110.0f), DEG2RAD(0.0f) }, + { SideRight, DEG2RAD( 110.0f), DEG2RAD(0.0f) } +}, X61Map[7] = { + { FrontLeft, DEG2RAD(-30.0f), DEG2RAD(0.0f) }, + { FrontRight, DEG2RAD( 30.0f), DEG2RAD(0.0f) }, + { FrontCenter, DEG2RAD( 0.0f), DEG2RAD(0.0f) }, + { LFE, 0.0f, 0.0f }, + { BackCenter, DEG2RAD(180.0f), DEG2RAD(0.0f) }, + { SideLeft, DEG2RAD(-90.0f), DEG2RAD(0.0f) }, + { SideRight, DEG2RAD( 90.0f), DEG2RAD(0.0f) } +}, X71Map[8] = { + { FrontLeft, DEG2RAD( -30.0f), DEG2RAD(0.0f) }, + { FrontRight, DEG2RAD( 30.0f), DEG2RAD(0.0f) }, + { FrontCenter, DEG2RAD( 0.0f), DEG2RAD(0.0f) }, + { LFE, 0.0f, 0.0f }, + { BackLeft, DEG2RAD(-150.0f), DEG2RAD(0.0f) }, + { BackRight, DEG2RAD( 150.0f), DEG2RAD(0.0f) }, + { SideLeft, DEG2RAD( -90.0f), DEG2RAD(0.0f) }, + { SideRight, DEG2RAD( 90.0f), DEG2RAD(0.0f) } +}; + +static void CalcPanningAndFilters(ALvoice *voice, const ALfloat Azi, const ALfloat Elev, + const ALfloat Distance, const ALfloat Spread, + const ALfloat DryGain, const ALfloat DryGainHF, + const ALfloat DryGainLF, const ALfloat *WetGain, + const ALfloat *WetGainLF, const ALfloat *WetGainHF, + ALeffectslot **SendSlots, const ALbuffer *Buffer, + const struct ALvoiceProps *props, const ALlistener *Listener, + const ALCdevice *Device) +{ + struct ChanMap StereoMap[2] = { + { FrontLeft, DEG2RAD(-30.0f), DEG2RAD(0.0f) }, + { FrontRight, DEG2RAD( 30.0f), DEG2RAD(0.0f) } + }; + bool DirectChannels = props->DirectChannels; + const ALsizei NumSends = Device->NumAuxSends; + const ALuint Frequency = Device->Frequency; + const struct ChanMap *chans = NULL; + ALsizei num_channels = 0; + bool isbformat = false; + ALfloat downmix_gain = 1.0f; + ALsizei c, i; + + switch(Buffer->FmtChannels) + { + case FmtMono: + chans = MonoMap; + num_channels = 1; + /* Mono buffers are never played direct. */ + DirectChannels = false; + break; + + case FmtStereo: + /* Convert counter-clockwise to clockwise. */ + StereoMap[0].angle = -props->StereoPan[0]; + StereoMap[1].angle = -props->StereoPan[1]; + + chans = StereoMap; + num_channels = 2; + downmix_gain = 1.0f / 2.0f; + break; + + case FmtRear: + chans = RearMap; + num_channels = 2; + downmix_gain = 1.0f / 2.0f; + break; + + case FmtQuad: + chans = QuadMap; + num_channels = 4; + downmix_gain = 1.0f / 4.0f; + break; + + case FmtX51: + chans = X51Map; + num_channels = 6; + /* NOTE: Excludes LFE. */ + downmix_gain = 1.0f / 5.0f; + break; + + case FmtX61: + chans = X61Map; + num_channels = 7; + /* NOTE: Excludes LFE. */ + downmix_gain = 1.0f / 6.0f; + break; + + case FmtX71: + chans = X71Map; + num_channels = 8; + /* NOTE: Excludes LFE. */ + downmix_gain = 1.0f / 7.0f; + break; + + case FmtBFormat2D: + num_channels = 3; + isbformat = true; + DirectChannels = false; + break; + + case FmtBFormat3D: + num_channels = 4; + isbformat = true; + DirectChannels = false; + break; + } + + for(c = 0;c < num_channels;c++) + { + memset(&voice->Direct.Params[c].Hrtf.Target, 0, + sizeof(voice->Direct.Params[c].Hrtf.Target)); + ClearArray(voice->Direct.Params[c].Gains.Target); + } + for(i = 0;i < NumSends;i++) + { + for(c = 0;c < num_channels;c++) + ClearArray(voice->Send[i].Params[c].Gains.Target); + } + + voice->Flags &= ~(VOICE_HAS_HRTF | VOICE_HAS_NFC); + if(isbformat) + { + /* Special handling for B-Format sources. */ + + if(Distance > FLT_EPSILON) + { + /* Panning a B-Format sound toward some direction is easy. Just pan + * the first (W) channel as a normal mono sound and silence the + * others. + */ + ALfloat coeffs[MAX_AMBI_COEFFS]; + + if(Device->AvgSpeakerDist > 0.0f) + { + ALfloat mdist = Distance * Listener->Params.MetersPerUnit; + ALfloat w0 = SPEEDOFSOUNDMETRESPERSEC / + (mdist * (ALfloat)Device->Frequency); + ALfloat w1 = SPEEDOFSOUNDMETRESPERSEC / + (Device->AvgSpeakerDist * (ALfloat)Device->Frequency); + /* Clamp w0 for really close distances, to prevent excessive + * bass. + */ + w0 = minf(w0, w1*4.0f); + + /* Only need to adjust the first channel of a B-Format source. */ + NfcFilterAdjust(&voice->Direct.Params[0].NFCtrlFilter, w0); + + for(i = 0;i < MAX_AMBI_ORDER+1;i++) + voice->Direct.ChannelsPerOrder[i] = Device->NumChannelsPerOrder[i]; + voice->Flags |= VOICE_HAS_NFC; + } + + /* A scalar of 1.5 for plain stereo results in +/-60 degrees being + * moved to +/-90 degrees for direct right and left speaker + * responses. + */ + CalcAngleCoeffs((Device->Render_Mode==StereoPair) ? ScaleAzimuthFront(Azi, 1.5f) : Azi, + Elev, Spread, coeffs); + + /* NOTE: W needs to be scaled by sqrt(2) due to FuMa normalization. */ + ComputePanGains(&Device->Dry, coeffs, DryGain*SQRTF_2, + voice->Direct.Params[0].Gains.Target); + for(i = 0;i < NumSends;i++) + { + const ALeffectslot *Slot = SendSlots[i]; + if(Slot) + ComputePanningGainsBF(Slot->ChanMap, Slot->NumChannels, coeffs, + WetGain[i]*SQRTF_2, voice->Send[i].Params[0].Gains.Target + ); + } + } + else + { + /* Local B-Format sources have their XYZ channels rotated according + * to the orientation. + */ + ALfloat N[3], V[3], U[3]; + aluMatrixf matrix; + + if(Device->AvgSpeakerDist > 0.0f) + { + /* NOTE: The NFCtrlFilters were created with a w0 of 0, which + * is what we want for FOA input. The first channel may have + * been previously re-adjusted if panned, so reset it. + */ + NfcFilterAdjust(&voice->Direct.Params[0].NFCtrlFilter, 0.0f); + + voice->Direct.ChannelsPerOrder[0] = 1; + voice->Direct.ChannelsPerOrder[1] = mini(voice->Direct.Channels-1, 3); + for(i = 2;i < MAX_AMBI_ORDER+1;i++) + voice->Direct.ChannelsPerOrder[i] = 0; + voice->Flags |= VOICE_HAS_NFC; + } + + /* AT then UP */ + N[0] = props->Orientation[0][0]; + N[1] = props->Orientation[0][1]; + N[2] = props->Orientation[0][2]; + aluNormalize(N); + V[0] = props->Orientation[1][0]; + V[1] = props->Orientation[1][1]; + V[2] = props->Orientation[1][2]; + aluNormalize(V); + if(!props->HeadRelative) + { + const aluMatrixf *lmatrix = &Listener->Params.Matrix; + aluMatrixfFloat3(N, 0.0f, lmatrix); + aluMatrixfFloat3(V, 0.0f, lmatrix); + } + /* Build and normalize right-vector */ + aluCrossproduct(N, V, U); + aluNormalize(U); + + /* Build a rotate + conversion matrix (FuMa -> ACN+N3D). NOTE: This + * matrix is transposed, for the inputs to align on the rows and + * outputs on the columns. + */ + aluMatrixfSet(&matrix, + // ACN0 ACN1 ACN2 ACN3 + SQRTF_2, 0.0f, 0.0f, 0.0f, // Ambi W + 0.0f, -N[0]*SQRTF_3, N[1]*SQRTF_3, -N[2]*SQRTF_3, // Ambi X + 0.0f, U[0]*SQRTF_3, -U[1]*SQRTF_3, U[2]*SQRTF_3, // Ambi Y + 0.0f, -V[0]*SQRTF_3, V[1]*SQRTF_3, -V[2]*SQRTF_3 // Ambi Z + ); + + voice->Direct.Buffer = Device->FOAOut.Buffer; + voice->Direct.Channels = Device->FOAOut.NumChannels; + for(c = 0;c < num_channels;c++) + ComputePanGains(&Device->FOAOut, matrix.m[c], DryGain, + voice->Direct.Params[c].Gains.Target); + for(i = 0;i < NumSends;i++) + { + const ALeffectslot *Slot = SendSlots[i]; + if(Slot) + { + for(c = 0;c < num_channels;c++) + ComputePanningGainsBF(Slot->ChanMap, Slot->NumChannels, + matrix.m[c], WetGain[i], voice->Send[i].Params[c].Gains.Target + ); + } + } + } + } + else if(DirectChannels) + { + /* Direct source channels always play local. Skip the virtual channels + * and write inputs to the matching real outputs. + */ + voice->Direct.Buffer = Device->RealOut.Buffer; + voice->Direct.Channels = Device->RealOut.NumChannels; + + for(c = 0;c < num_channels;c++) + { + int idx = GetChannelIdxByName(&Device->RealOut, chans[c].channel); + if(idx != -1) voice->Direct.Params[c].Gains.Target[idx] = DryGain; + } + + /* Auxiliary sends still use normal channel panning since they mix to + * B-Format, which can't channel-match. + */ + for(c = 0;c < num_channels;c++) + { + ALfloat coeffs[MAX_AMBI_COEFFS]; + CalcAngleCoeffs(chans[c].angle, chans[c].elevation, 0.0f, coeffs); + + for(i = 0;i < NumSends;i++) + { + const ALeffectslot *Slot = SendSlots[i]; + if(Slot) + ComputePanningGainsBF(Slot->ChanMap, Slot->NumChannels, + coeffs, WetGain[i], voice->Send[i].Params[c].Gains.Target + ); + } + } + } + else if(Device->Render_Mode == HrtfRender) + { + /* Full HRTF rendering. Skip the virtual channels and render to the + * real outputs. + */ + voice->Direct.Buffer = Device->RealOut.Buffer; + voice->Direct.Channels = Device->RealOut.NumChannels; + + if(Distance > FLT_EPSILON) + { + ALfloat coeffs[MAX_AMBI_COEFFS]; + + /* Get the HRIR coefficients and delays just once, for the given + * source direction. + */ + GetHrtfCoeffs(Device->HrtfHandle, Elev, Azi, Spread, + voice->Direct.Params[0].Hrtf.Target.Coeffs, + voice->Direct.Params[0].Hrtf.Target.Delay); + voice->Direct.Params[0].Hrtf.Target.Gain = DryGain * downmix_gain; + + /* Remaining channels use the same results as the first. */ + for(c = 1;c < num_channels;c++) + { + /* Skip LFE */ + if(chans[c].channel != LFE) + voice->Direct.Params[c].Hrtf.Target = voice->Direct.Params[0].Hrtf.Target; + } + + /* Calculate the directional coefficients once, which apply to all + * input channels of the source sends. + */ + CalcAngleCoeffs(Azi, Elev, Spread, coeffs); + + for(i = 0;i < NumSends;i++) + { + const ALeffectslot *Slot = SendSlots[i]; + if(Slot) + for(c = 0;c < num_channels;c++) + { + /* Skip LFE */ + if(chans[c].channel != LFE) + ComputePanningGainsBF(Slot->ChanMap, + Slot->NumChannels, coeffs, WetGain[i] * downmix_gain, + voice->Send[i].Params[c].Gains.Target + ); + } + } + } + else + { + /* Local sources on HRTF play with each channel panned to its + * relative location around the listener, providing "virtual + * speaker" responses. + */ + for(c = 0;c < num_channels;c++) + { + ALfloat coeffs[MAX_AMBI_COEFFS]; + + if(chans[c].channel == LFE) + { + /* Skip LFE */ + continue; + } + + /* Get the HRIR coefficients and delays for this channel + * position. + */ + GetHrtfCoeffs(Device->HrtfHandle, + chans[c].elevation, chans[c].angle, Spread, + voice->Direct.Params[c].Hrtf.Target.Coeffs, + voice->Direct.Params[c].Hrtf.Target.Delay + ); + voice->Direct.Params[c].Hrtf.Target.Gain = DryGain; + + /* Normal panning for auxiliary sends. */ + CalcAngleCoeffs(chans[c].angle, chans[c].elevation, Spread, coeffs); + + for(i = 0;i < NumSends;i++) + { + const ALeffectslot *Slot = SendSlots[i]; + if(Slot) + ComputePanningGainsBF(Slot->ChanMap, Slot->NumChannels, + coeffs, WetGain[i], voice->Send[i].Params[c].Gains.Target + ); + } + } + } + + voice->Flags |= VOICE_HAS_HRTF; + } + else + { + /* Non-HRTF rendering. Use normal panning to the output. */ + + if(Distance > FLT_EPSILON) + { + ALfloat coeffs[MAX_AMBI_COEFFS]; + ALfloat w0 = 0.0f; + + /* Calculate NFC filter coefficient if needed. */ + if(Device->AvgSpeakerDist > 0.0f) + { + ALfloat mdist = Distance * Listener->Params.MetersPerUnit; + ALfloat w1 = SPEEDOFSOUNDMETRESPERSEC / + (Device->AvgSpeakerDist * (ALfloat)Device->Frequency); + w0 = SPEEDOFSOUNDMETRESPERSEC / + (mdist * (ALfloat)Device->Frequency); + /* Clamp w0 for really close distances, to prevent excessive + * bass. + */ + w0 = minf(w0, w1*4.0f); + + /* Adjust NFC filters. */ + for(c = 0;c < num_channels;c++) + NfcFilterAdjust(&voice->Direct.Params[c].NFCtrlFilter, w0); + + for(i = 0;i < MAX_AMBI_ORDER+1;i++) + voice->Direct.ChannelsPerOrder[i] = Device->NumChannelsPerOrder[i]; + voice->Flags |= VOICE_HAS_NFC; + } + + /* Calculate the directional coefficients once, which apply to all + * input channels. + */ + CalcAngleCoeffs((Device->Render_Mode==StereoPair) ? ScaleAzimuthFront(Azi, 1.5f) : Azi, + Elev, Spread, coeffs); + + for(c = 0;c < num_channels;c++) + { + /* Special-case LFE */ + if(chans[c].channel == LFE) + { + if(Device->Dry.Buffer == Device->RealOut.Buffer) + { + int idx = GetChannelIdxByName(&Device->RealOut, chans[c].channel); + if(idx != -1) voice->Direct.Params[c].Gains.Target[idx] = DryGain; + } + continue; + } + + ComputePanGains(&Device->Dry, coeffs, DryGain * downmix_gain, + voice->Direct.Params[c].Gains.Target); + } + + for(i = 0;i < NumSends;i++) + { + const ALeffectslot *Slot = SendSlots[i]; + if(Slot) + for(c = 0;c < num_channels;c++) + { + /* Skip LFE */ + if(chans[c].channel != LFE) + ComputePanningGainsBF(Slot->ChanMap, + Slot->NumChannels, coeffs, WetGain[i] * downmix_gain, + voice->Send[i].Params[c].Gains.Target + ); + } + } + } + else + { + ALfloat w0 = 0.0f; + + if(Device->AvgSpeakerDist > 0.0f) + { + /* If the source distance is 0, set w0 to w1 to act as a pass- + * through. We still want to pass the signal through the + * filters so they keep an appropriate history, in case the + * source moves away from the listener. + */ + w0 = SPEEDOFSOUNDMETRESPERSEC / + (Device->AvgSpeakerDist * (ALfloat)Device->Frequency); + + for(c = 0;c < num_channels;c++) + NfcFilterAdjust(&voice->Direct.Params[c].NFCtrlFilter, w0); + + for(i = 0;i < MAX_AMBI_ORDER+1;i++) + voice->Direct.ChannelsPerOrder[i] = Device->NumChannelsPerOrder[i]; + voice->Flags |= VOICE_HAS_NFC; + } + + for(c = 0;c < num_channels;c++) + { + ALfloat coeffs[MAX_AMBI_COEFFS]; + + /* Special-case LFE */ + if(chans[c].channel == LFE) + { + if(Device->Dry.Buffer == Device->RealOut.Buffer) + { + int idx = GetChannelIdxByName(&Device->RealOut, chans[c].channel); + if(idx != -1) voice->Direct.Params[c].Gains.Target[idx] = DryGain; + } + continue; + } + + CalcAngleCoeffs( + (Device->Render_Mode==StereoPair) ? ScaleAzimuthFront(chans[c].angle, 3.0f) + : chans[c].angle, + chans[c].elevation, Spread, coeffs + ); + + ComputePanGains(&Device->Dry, coeffs, DryGain, + voice->Direct.Params[c].Gains.Target); + for(i = 0;i < NumSends;i++) + { + const ALeffectslot *Slot = SendSlots[i]; + if(Slot) + ComputePanningGainsBF(Slot->ChanMap, Slot->NumChannels, + coeffs, WetGain[i], voice->Send[i].Params[c].Gains.Target + ); + } + } + } + } + + { + ALfloat hfScale = props->Direct.HFReference / Frequency; + ALfloat lfScale = props->Direct.LFReference / Frequency; + ALfloat gainHF = maxf(DryGainHF, 0.001f); /* Limit -60dB */ + ALfloat gainLF = maxf(DryGainLF, 0.001f); + + voice->Direct.FilterType = AF_None; + if(gainHF != 1.0f) voice->Direct.FilterType |= AF_LowPass; + if(gainLF != 1.0f) voice->Direct.FilterType |= AF_HighPass; + BiquadFilter_setParams( + &voice->Direct.Params[0].LowPass, BiquadType_HighShelf, + gainHF, hfScale, calc_rcpQ_from_slope(gainHF, 1.0f) + ); + BiquadFilter_setParams( + &voice->Direct.Params[0].HighPass, BiquadType_LowShelf, + gainLF, lfScale, calc_rcpQ_from_slope(gainLF, 1.0f) + ); + for(c = 1;c < num_channels;c++) + { + BiquadFilter_copyParams(&voice->Direct.Params[c].LowPass, + &voice->Direct.Params[0].LowPass); + BiquadFilter_copyParams(&voice->Direct.Params[c].HighPass, + &voice->Direct.Params[0].HighPass); + } + } + for(i = 0;i < NumSends;i++) + { + ALfloat hfScale = props->Send[i].HFReference / Frequency; + ALfloat lfScale = props->Send[i].LFReference / Frequency; + ALfloat gainHF = maxf(WetGainHF[i], 0.001f); + ALfloat gainLF = maxf(WetGainLF[i], 0.001f); + + voice->Send[i].FilterType = AF_None; + if(gainHF != 1.0f) voice->Send[i].FilterType |= AF_LowPass; + if(gainLF != 1.0f) voice->Send[i].FilterType |= AF_HighPass; + BiquadFilter_setParams( + &voice->Send[i].Params[0].LowPass, BiquadType_HighShelf, + gainHF, hfScale, calc_rcpQ_from_slope(gainHF, 1.0f) + ); + BiquadFilter_setParams( + &voice->Send[i].Params[0].HighPass, BiquadType_LowShelf, + gainLF, lfScale, calc_rcpQ_from_slope(gainLF, 1.0f) + ); + for(c = 1;c < num_channels;c++) + { + BiquadFilter_copyParams(&voice->Send[i].Params[c].LowPass, + &voice->Send[i].Params[0].LowPass); + BiquadFilter_copyParams(&voice->Send[i].Params[c].HighPass, + &voice->Send[i].Params[0].HighPass); + } + } +} + +static void CalcNonAttnSourceParams(ALvoice *voice, const struct ALvoiceProps *props, const ALbuffer *ALBuffer, const ALCcontext *ALContext) +{ + const ALCdevice *Device = ALContext->Device; + const ALlistener *Listener = ALContext->Listener; + ALfloat DryGain, DryGainHF, DryGainLF; + ALfloat WetGain[MAX_SENDS]; + ALfloat WetGainHF[MAX_SENDS]; + ALfloat WetGainLF[MAX_SENDS]; + ALeffectslot *SendSlots[MAX_SENDS]; + ALfloat Pitch; + ALsizei i; + + voice->Direct.Buffer = Device->Dry.Buffer; + voice->Direct.Channels = Device->Dry.NumChannels; + for(i = 0;i < Device->NumAuxSends;i++) + { + SendSlots[i] = props->Send[i].Slot; + if(!SendSlots[i] && i == 0) + SendSlots[i] = ALContext->DefaultSlot; + if(!SendSlots[i] || SendSlots[i]->Params.EffectType == AL_EFFECT_NULL) + { + SendSlots[i] = NULL; + voice->Send[i].Buffer = NULL; + voice->Send[i].Channels = 0; + } + else + { + voice->Send[i].Buffer = SendSlots[i]->WetBuffer; + voice->Send[i].Channels = SendSlots[i]->NumChannels; + } + } + + /* Calculate the stepping value */ + Pitch = (ALfloat)ALBuffer->Frequency/(ALfloat)Device->Frequency * props->Pitch; + if(Pitch > (ALfloat)MAX_PITCH) + voice->Step = MAX_PITCH<<FRACTIONBITS; + else + voice->Step = maxi(fastf2i(Pitch * FRACTIONONE), 1); + if(props->Resampler == BSinc24Resampler) + BsincPrepare(voice->Step, &voice->ResampleState.bsinc, &bsinc24); + else if(props->Resampler == BSinc12Resampler) + BsincPrepare(voice->Step, &voice->ResampleState.bsinc, &bsinc12); + voice->Resampler = SelectResampler(props->Resampler); + + /* Calculate gains */ + DryGain = clampf(props->Gain, props->MinGain, props->MaxGain); + DryGain *= props->Direct.Gain * Listener->Params.Gain; + DryGain = minf(DryGain, GAIN_MIX_MAX); + DryGainHF = props->Direct.GainHF; + DryGainLF = props->Direct.GainLF; + for(i = 0;i < Device->NumAuxSends;i++) + { + WetGain[i] = clampf(props->Gain, props->MinGain, props->MaxGain); + WetGain[i] *= props->Send[i].Gain * Listener->Params.Gain; + WetGain[i] = minf(WetGain[i], GAIN_MIX_MAX); + WetGainHF[i] = props->Send[i].GainHF; + WetGainLF[i] = props->Send[i].GainLF; + } + + CalcPanningAndFilters(voice, 0.0f, 0.0f, 0.0f, 0.0f, DryGain, DryGainHF, DryGainLF, WetGain, + WetGainLF, WetGainHF, SendSlots, ALBuffer, props, Listener, Device); +} + +static void CalcAttnSourceParams(ALvoice *voice, const struct ALvoiceProps *props, const ALbuffer *ALBuffer, const ALCcontext *ALContext) +{ + const ALCdevice *Device = ALContext->Device; + const ALlistener *Listener = ALContext->Listener; + const ALsizei NumSends = Device->NumAuxSends; + aluVector Position, Velocity, Direction, SourceToListener; + ALfloat Distance, ClampedDist, DopplerFactor; + ALeffectslot *SendSlots[MAX_SENDS]; + ALfloat RoomRolloff[MAX_SENDS]; + ALfloat DecayDistance[MAX_SENDS]; + ALfloat DecayLFDistance[MAX_SENDS]; + ALfloat DecayHFDistance[MAX_SENDS]; + ALfloat DryGain, DryGainHF, DryGainLF; + ALfloat WetGain[MAX_SENDS]; + ALfloat WetGainHF[MAX_SENDS]; + ALfloat WetGainLF[MAX_SENDS]; + bool directional; + ALfloat ev, az; + ALfloat spread; + ALfloat Pitch; + ALint i; + + /* Set mixing buffers and get send parameters. */ + voice->Direct.Buffer = Device->Dry.Buffer; + voice->Direct.Channels = Device->Dry.NumChannels; + for(i = 0;i < NumSends;i++) + { + SendSlots[i] = props->Send[i].Slot; + if(!SendSlots[i] && i == 0) + SendSlots[i] = ALContext->DefaultSlot; + if(!SendSlots[i] || SendSlots[i]->Params.EffectType == AL_EFFECT_NULL) + { + SendSlots[i] = NULL; + RoomRolloff[i] = 0.0f; + DecayDistance[i] = 0.0f; + DecayLFDistance[i] = 0.0f; + DecayHFDistance[i] = 0.0f; + } + else if(SendSlots[i]->Params.AuxSendAuto) + { + RoomRolloff[i] = SendSlots[i]->Params.RoomRolloff + props->RoomRolloffFactor; + /* Calculate the distances to where this effect's decay reaches + * -60dB. + */ + DecayDistance[i] = SendSlots[i]->Params.DecayTime * + Listener->Params.ReverbSpeedOfSound; + DecayLFDistance[i] = DecayDistance[i] * SendSlots[i]->Params.DecayLFRatio; + DecayHFDistance[i] = DecayDistance[i] * SendSlots[i]->Params.DecayHFRatio; + if(SendSlots[i]->Params.DecayHFLimit) + { + ALfloat airAbsorption = SendSlots[i]->Params.AirAbsorptionGainHF; + if(airAbsorption < 1.0f) + { + /* Calculate the distance to where this effect's air + * absorption reaches -60dB, and limit the effect's HF + * decay distance (so it doesn't take any longer to decay + * than the air would allow). + */ + ALfloat absorb_dist = log10f(REVERB_DECAY_GAIN) / log10f(airAbsorption); + DecayHFDistance[i] = minf(absorb_dist, DecayHFDistance[i]); + } + } + } + else + { + /* If the slot's auxiliary send auto is off, the data sent to the + * effect slot is the same as the dry path, sans filter effects */ + RoomRolloff[i] = props->RolloffFactor; + DecayDistance[i] = 0.0f; + DecayLFDistance[i] = 0.0f; + DecayHFDistance[i] = 0.0f; + } + + if(!SendSlots[i]) + { + voice->Send[i].Buffer = NULL; + voice->Send[i].Channels = 0; + } + else + { + voice->Send[i].Buffer = SendSlots[i]->WetBuffer; + voice->Send[i].Channels = SendSlots[i]->NumChannels; + } + } + + /* Transform source to listener space (convert to head relative) */ + aluVectorSet(&Position, props->Position[0], props->Position[1], props->Position[2], 1.0f); + aluVectorSet(&Direction, props->Direction[0], props->Direction[1], props->Direction[2], 0.0f); + aluVectorSet(&Velocity, props->Velocity[0], props->Velocity[1], props->Velocity[2], 0.0f); + if(props->HeadRelative == AL_FALSE) + { + const aluMatrixf *Matrix = &Listener->Params.Matrix; + /* Transform source vectors */ + Position = aluMatrixfVector(Matrix, &Position); + Velocity = aluMatrixfVector(Matrix, &Velocity); + Direction = aluMatrixfVector(Matrix, &Direction); + } + else + { + const aluVector *lvelocity = &Listener->Params.Velocity; + /* Offset the source velocity to be relative of the listener velocity */ + Velocity.v[0] += lvelocity->v[0]; + Velocity.v[1] += lvelocity->v[1]; + Velocity.v[2] += lvelocity->v[2]; + } + + directional = aluNormalize(Direction.v) > 0.0f; + SourceToListener.v[0] = -Position.v[0]; + SourceToListener.v[1] = -Position.v[1]; + SourceToListener.v[2] = -Position.v[2]; + SourceToListener.v[3] = 0.0f; + Distance = aluNormalize(SourceToListener.v); + + /* Initial source gain */ + DryGain = props->Gain; + DryGainHF = 1.0f; + DryGainLF = 1.0f; + for(i = 0;i < NumSends;i++) + { + WetGain[i] = props->Gain; + WetGainHF[i] = 1.0f; + WetGainLF[i] = 1.0f; + } + + /* Calculate distance attenuation */ + ClampedDist = Distance; + + switch(Listener->Params.SourceDistanceModel ? + props->DistanceModel : Listener->Params.DistanceModel) + { + case InverseDistanceClamped: + ClampedDist = clampf(ClampedDist, props->RefDistance, props->MaxDistance); + if(props->MaxDistance < props->RefDistance) + break; + /*fall-through*/ + case InverseDistance: + if(!(props->RefDistance > 0.0f)) + ClampedDist = props->RefDistance; + else + { + ALfloat dist = lerp(props->RefDistance, ClampedDist, props->RolloffFactor); + if(dist > 0.0f) DryGain *= props->RefDistance / dist; + for(i = 0;i < NumSends;i++) + { + dist = lerp(props->RefDistance, ClampedDist, RoomRolloff[i]); + if(dist > 0.0f) WetGain[i] *= props->RefDistance / dist; + } + } + break; + + case LinearDistanceClamped: + ClampedDist = clampf(ClampedDist, props->RefDistance, props->MaxDistance); + if(props->MaxDistance < props->RefDistance) + break; + /*fall-through*/ + case LinearDistance: + if(!(props->MaxDistance != props->RefDistance)) + ClampedDist = props->RefDistance; + else + { + ALfloat attn = props->RolloffFactor * (ClampedDist-props->RefDistance) / + (props->MaxDistance-props->RefDistance); + DryGain *= maxf(1.0f - attn, 0.0f); + for(i = 0;i < NumSends;i++) + { + attn = RoomRolloff[i] * (ClampedDist-props->RefDistance) / + (props->MaxDistance-props->RefDistance); + WetGain[i] *= maxf(1.0f - attn, 0.0f); + } + } + break; + + case ExponentDistanceClamped: + ClampedDist = clampf(ClampedDist, props->RefDistance, props->MaxDistance); + if(props->MaxDistance < props->RefDistance) + break; + /*fall-through*/ + case ExponentDistance: + if(!(ClampedDist > 0.0f && props->RefDistance > 0.0f)) + ClampedDist = props->RefDistance; + else + { + DryGain *= powf(ClampedDist/props->RefDistance, -props->RolloffFactor); + for(i = 0;i < NumSends;i++) + WetGain[i] *= powf(ClampedDist/props->RefDistance, -RoomRolloff[i]); + } + break; + + case DisableDistance: + ClampedDist = props->RefDistance; + break; + } + + /* Calculate directional soundcones */ + if(directional && props->InnerAngle < 360.0f) + { + ALfloat ConeVolume; + ALfloat ConeHF; + ALfloat Angle; + + Angle = acosf(aluDotproduct(&Direction, &SourceToListener)); + Angle = RAD2DEG(Angle * ConeScale * 2.0f); + if(!(Angle > props->InnerAngle)) + { + ConeVolume = 1.0f; + ConeHF = 1.0f; + } + else if(Angle < props->OuterAngle) + { + ALfloat scale = ( Angle-props->InnerAngle) / + (props->OuterAngle-props->InnerAngle); + ConeVolume = lerp(1.0f, props->OuterGain, scale); + ConeHF = lerp(1.0f, props->OuterGainHF, scale); + } + else + { + ConeVolume = props->OuterGain; + ConeHF = props->OuterGainHF; + } + + DryGain *= ConeVolume; + if(props->DryGainHFAuto) + DryGainHF *= ConeHF; + if(props->WetGainAuto) + { + for(i = 0;i < NumSends;i++) + WetGain[i] *= ConeVolume; + } + if(props->WetGainHFAuto) + { + for(i = 0;i < NumSends;i++) + WetGainHF[i] *= ConeHF; + } + } + + /* Apply gain and frequency filters */ + DryGain = clampf(DryGain, props->MinGain, props->MaxGain); + DryGain = minf(DryGain*props->Direct.Gain*Listener->Params.Gain, GAIN_MIX_MAX); + DryGainHF *= props->Direct.GainHF; + DryGainLF *= props->Direct.GainLF; + for(i = 0;i < NumSends;i++) + { + WetGain[i] = clampf(WetGain[i], props->MinGain, props->MaxGain); + WetGain[i] = minf(WetGain[i]*props->Send[i].Gain*Listener->Params.Gain, GAIN_MIX_MAX); + WetGainHF[i] *= props->Send[i].GainHF; + WetGainLF[i] *= props->Send[i].GainLF; + } + + /* Distance-based air absorption and initial send decay. */ + if(ClampedDist > props->RefDistance && props->RolloffFactor > 0.0f) + { + ALfloat meters_base = (ClampedDist-props->RefDistance) * props->RolloffFactor * + Listener->Params.MetersPerUnit; + if(props->AirAbsorptionFactor > 0.0f) + { + ALfloat hfattn = powf(AIRABSORBGAINHF, meters_base * props->AirAbsorptionFactor); + DryGainHF *= hfattn; + for(i = 0;i < NumSends;i++) + WetGainHF[i] *= hfattn; + } + + if(props->WetGainAuto) + { + /* Apply a decay-time transformation to the wet path, based on the + * source distance in meters. The initial decay of the reverb + * effect is calculated and applied to the wet path. + */ + for(i = 0;i < NumSends;i++) + { + ALfloat gain, gainhf, gainlf; + + if(!(DecayDistance[i] > 0.0f)) + continue; + + gain = powf(REVERB_DECAY_GAIN, meters_base/DecayDistance[i]); + WetGain[i] *= gain; + /* Yes, the wet path's air absorption is applied with + * WetGainAuto on, rather than WetGainHFAuto. + */ + if(gain > 0.0f) + { + gainhf = powf(REVERB_DECAY_GAIN, meters_base/DecayHFDistance[i]); + WetGainHF[i] *= minf(gainhf / gain, 1.0f); + gainlf = powf(REVERB_DECAY_GAIN, meters_base/DecayLFDistance[i]); + WetGainLF[i] *= minf(gainlf / gain, 1.0f); + } + } + } + } + + + /* Initial source pitch */ + Pitch = props->Pitch; + + /* Calculate velocity-based doppler effect */ + DopplerFactor = props->DopplerFactor * Listener->Params.DopplerFactor; + if(DopplerFactor > 0.0f) + { + const aluVector *lvelocity = &Listener->Params.Velocity; + const ALfloat SpeedOfSound = Listener->Params.SpeedOfSound; + ALfloat vss, vls; + + vss = aluDotproduct(&Velocity, &SourceToListener) * DopplerFactor; + vls = aluDotproduct(lvelocity, &SourceToListener) * DopplerFactor; + + if(!(vls < SpeedOfSound)) + { + /* Listener moving away from the source at the speed of sound. + * Sound waves can't catch it. + */ + Pitch = 0.0f; + } + else if(!(vss < SpeedOfSound)) + { + /* Source moving toward the listener at the speed of sound. Sound + * waves bunch up to extreme frequencies. + */ + Pitch = HUGE_VALF; + } + else + { + /* Source and listener movement is nominal. Calculate the proper + * doppler shift. + */ + Pitch *= (SpeedOfSound-vls) / (SpeedOfSound-vss); + } + } + + /* Adjust pitch based on the buffer and output frequencies, and calculate + * fixed-point stepping value. + */ + Pitch *= (ALfloat)ALBuffer->Frequency/(ALfloat)Device->Frequency; + if(Pitch > (ALfloat)MAX_PITCH) + voice->Step = MAX_PITCH<<FRACTIONBITS; + else + voice->Step = maxi(fastf2i(Pitch * FRACTIONONE), 1); + if(props->Resampler == BSinc24Resampler) + BsincPrepare(voice->Step, &voice->ResampleState.bsinc, &bsinc24); + else if(props->Resampler == BSinc12Resampler) + BsincPrepare(voice->Step, &voice->ResampleState.bsinc, &bsinc12); + voice->Resampler = SelectResampler(props->Resampler); + + if(Distance > 0.0f) + { + /* Clamp Y, in case rounding errors caused it to end up outside of + * -1...+1. + */ + ev = asinf(clampf(-SourceToListener.v[1], -1.0f, 1.0f)); + /* Double negation on Z cancels out; negate once for changing source- + * to-listener to listener-to-source, and again for right-handed coords + * with -Z in front. + */ + az = atan2f(-SourceToListener.v[0], SourceToListener.v[2]*ZScale); + } + else + ev = az = 0.0f; + + if(props->Radius > Distance) + spread = F_TAU - Distance/props->Radius*F_PI; + else if(Distance > 0.0f) + spread = asinf(props->Radius / Distance) * 2.0f; + else + spread = 0.0f; + + CalcPanningAndFilters(voice, az, ev, Distance, spread, DryGain, DryGainHF, DryGainLF, WetGain, + WetGainLF, WetGainHF, SendSlots, ALBuffer, props, Listener, Device); +} + +static void CalcSourceParams(ALvoice *voice, ALCcontext *context, bool force) +{ + ALbufferlistitem *BufferListItem; + struct ALvoiceProps *props; + + props = static_cast<ALvoiceProps*>(ATOMIC_EXCHANGE_PTR(&voice->Update, + static_cast<ALvoiceProps*>(nullptr), almemory_order_acq_rel)); + if(!props && !force) return; + + if(props) + { + memcpy(voice->Props, props, + FAM_SIZE(struct ALvoiceProps, Send, context->Device->NumAuxSends) + ); + + ATOMIC_REPLACE_HEAD(struct ALvoiceProps*, &context->FreeVoiceProps, props); + } + props = voice->Props; + + BufferListItem = ATOMIC_LOAD(&voice->current_buffer, almemory_order_relaxed); + while(BufferListItem != NULL) + { + const ALbuffer *buffer = NULL; + ALsizei i = 0; + while(!buffer && i < BufferListItem->num_buffers) + buffer = BufferListItem->buffers[i]; + if(LIKELY(buffer)) + { + if(props->SpatializeMode == SpatializeOn || + (props->SpatializeMode == SpatializeAuto && buffer->FmtChannels == FmtMono)) + CalcAttnSourceParams(voice, props, buffer, context); + else + CalcNonAttnSourceParams(voice, props, buffer, context); + break; + } + BufferListItem = ATOMIC_LOAD(&BufferListItem->next, almemory_order_acquire); + } +} + + +static void ProcessParamUpdates(ALCcontext *ctx, const struct ALeffectslotArray *slots) +{ + ALvoice **voice, **voice_end; + ALsource *source; + ALsizei i; + + IncrementRef(&ctx->UpdateCount); + if(!ATOMIC_LOAD(&ctx->HoldUpdates, almemory_order_acquire)) + { + bool cforce = CalcContextParams(ctx); + bool force = CalcListenerParams(ctx) | cforce; + for(i = 0;i < slots->count;i++) + force |= CalcEffectSlotParams(slots->slot[i], ctx, cforce); + + voice = ctx->Voices; + voice_end = voice + ctx->VoiceCount; + for(;voice != voice_end;++voice) + { + source = ATOMIC_LOAD(&(*voice)->Source, almemory_order_acquire); + if(source) CalcSourceParams(*voice, ctx, force); + } + } + IncrementRef(&ctx->UpdateCount); +} + + +static void ApplyStablizer(FrontStablizer *Stablizer, ALfloat (*RESTRICT Buffer)[BUFFERSIZE], + int lidx, int ridx, int cidx, ALsizei SamplesToDo, + ALsizei NumChannels) +{ + ALfloat (*RESTRICT lsplit)[BUFFERSIZE] = Stablizer->LSplit; + ALfloat (*RESTRICT rsplit)[BUFFERSIZE] = Stablizer->RSplit; + ALsizei i; + + /* Apply an all-pass to all channels, except the front-left and front- + * right, so they maintain the same relative phase. + */ + for(i = 0;i < NumChannels;i++) + { + if(i == lidx || i == ridx) + continue; + splitterap_process(&Stablizer->APFilter[i], Buffer[i], SamplesToDo); + } + + bandsplit_process(&Stablizer->LFilter, lsplit[1], lsplit[0], Buffer[lidx], SamplesToDo); + bandsplit_process(&Stablizer->RFilter, rsplit[1], rsplit[0], Buffer[ridx], SamplesToDo); + + for(i = 0;i < SamplesToDo;i++) + { + ALfloat lfsum, hfsum; + ALfloat m, s, c; + + lfsum = lsplit[0][i] + rsplit[0][i]; + hfsum = lsplit[1][i] + rsplit[1][i]; + s = lsplit[0][i] + lsplit[1][i] - rsplit[0][i] - rsplit[1][i]; + + /* This pans the separate low- and high-frequency sums between being on + * the center channel and the left/right channels. The low-frequency + * sum is 1/3rd toward center (2/3rds on left/right) and the high- + * frequency sum is 1/4th toward center (3/4ths on left/right). These + * values can be tweaked. + */ + m = lfsum*cosf(1.0f/3.0f * F_PI_2) + hfsum*cosf(1.0f/4.0f * F_PI_2); + c = lfsum*sinf(1.0f/3.0f * F_PI_2) + hfsum*sinf(1.0f/4.0f * F_PI_2); + + /* The generated center channel signal adds to the existing signal, + * while the modified left and right channels replace. + */ + Buffer[lidx][i] = (m + s) * 0.5f; + Buffer[ridx][i] = (m - s) * 0.5f; + Buffer[cidx][i] += c * 0.5f; + } +} + +static void ApplyDistanceComp(ALfloat (*RESTRICT Samples)[BUFFERSIZE], DistanceComp *distcomp, + ALfloat *RESTRICT Values, ALsizei SamplesToDo, ALsizei numchans) +{ + ALsizei i, c; + + for(c = 0;c < numchans;c++) + { + ALfloat *RESTRICT inout = Samples[c]; + const ALfloat gain = distcomp[c].Gain; + const ALsizei base = distcomp[c].Length; + ALfloat *RESTRICT distbuf = distcomp[c].Buffer; + + if(base == 0) + { + if(gain < 1.0f) + { + for(i = 0;i < SamplesToDo;i++) + inout[i] *= gain; + } + continue; + } + + if(LIKELY(SamplesToDo >= base)) + { + for(i = 0;i < base;i++) + Values[i] = distbuf[i]; + for(;i < SamplesToDo;i++) + Values[i] = inout[i-base]; + memcpy(distbuf, &inout[SamplesToDo-base], base*sizeof(ALfloat)); + } + else + { + for(i = 0;i < SamplesToDo;i++) + Values[i] = distbuf[i]; + memmove(distbuf, distbuf+SamplesToDo, (base-SamplesToDo)*sizeof(ALfloat)); + memcpy(distbuf+base-SamplesToDo, inout, SamplesToDo*sizeof(ALfloat)); + } + for(i = 0;i < SamplesToDo;i++) + inout[i] = Values[i]*gain; + } +} + +static void ApplyDither(ALfloat (*RESTRICT Samples)[BUFFERSIZE], ALuint *dither_seed, + const ALfloat quant_scale, const ALsizei SamplesToDo, + const ALsizei numchans) +{ + const ALfloat invscale = 1.0f / quant_scale; + ALuint seed = *dither_seed; + ALsizei c, i; + + ASSUME(numchans > 0); + ASSUME(SamplesToDo > 0); + + /* Dithering. Step 1, generate whitenoise (uniform distribution of random + * values between -1 and +1). Step 2 is to add the noise to the samples, + * before rounding and after scaling up to the desired quantization depth. + */ + for(c = 0;c < numchans;c++) + { + ALfloat *RESTRICT samples = Samples[c]; + for(i = 0;i < SamplesToDo;i++) + { + ALfloat val = samples[i] * quant_scale; + ALuint rng0 = dither_rng(&seed); + ALuint rng1 = dither_rng(&seed); + val += (ALfloat)(rng0*(1.0/UINT_MAX) - rng1*(1.0/UINT_MAX)); + samples[i] = fast_roundf(val) * invscale; + } + } + *dither_seed = seed; +} + + +static inline ALfloat Conv_ALfloat(ALfloat val) +{ return val; } +static inline ALint Conv_ALint(ALfloat val) +{ + /* Floats have a 23-bit mantissa. There is an implied 1 bit in the mantissa + * along with the sign bit, giving 25 bits total, so [-16777216, +16777216] + * is the max value a normalized float can be scaled to before losing + * precision. + */ + return fastf2i(clampf(val*16777216.0f, -16777216.0f, 16777215.0f))<<7; +} +static inline ALshort Conv_ALshort(ALfloat val) +{ return fastf2i(clampf(val*32768.0f, -32768.0f, 32767.0f)); } +static inline ALbyte Conv_ALbyte(ALfloat val) +{ return fastf2i(clampf(val*128.0f, -128.0f, 127.0f)); } + +/* Define unsigned output variations. */ +#define DECL_TEMPLATE(T, func, O) \ +static inline T Conv_##T(ALfloat val) { return func(val)+O; } + +DECL_TEMPLATE(ALubyte, Conv_ALbyte, 128) +DECL_TEMPLATE(ALushort, Conv_ALshort, 32768) +DECL_TEMPLATE(ALuint, Conv_ALint, 2147483648u) + +#undef DECL_TEMPLATE + +#define DECL_TEMPLATE(T, A) \ +static void Write##A(const ALfloat (*RESTRICT InBuffer)[BUFFERSIZE], \ + ALvoid *OutBuffer, ALsizei Offset, ALsizei SamplesToDo, \ + ALsizei numchans) \ +{ \ + ALsizei i, j; \ + \ + ASSUME(numchans > 0); \ + ASSUME(SamplesToDo > 0); \ + \ + for(j = 0;j < numchans;j++) \ + { \ + const ALfloat *RESTRICT in = InBuffer[j]; \ + T *RESTRICT out = (T*)OutBuffer + Offset*numchans + j; \ + \ + for(i = 0;i < SamplesToDo;i++) \ + out[i*numchans] = Conv_##T(in[i]); \ + } \ +} + +DECL_TEMPLATE(ALfloat, F32) +DECL_TEMPLATE(ALuint, UI32) +DECL_TEMPLATE(ALint, I32) +DECL_TEMPLATE(ALushort, UI16) +DECL_TEMPLATE(ALshort, I16) +DECL_TEMPLATE(ALubyte, UI8) +DECL_TEMPLATE(ALbyte, I8) + +#undef DECL_TEMPLATE + + +void aluMixData(ALCdevice *device, ALvoid *OutBuffer, ALsizei NumSamples) +{ + ALsizei SamplesToDo; + ALsizei SamplesDone; + ALCcontext *ctx; + ALsizei i, c; + + START_MIXER_MODE(); + for(SamplesDone = 0;SamplesDone < NumSamples;) + { + SamplesToDo = mini(NumSamples-SamplesDone, BUFFERSIZE); + for(c = 0;c < device->Dry.NumChannels;c++) + memset(device->Dry.Buffer[c], 0, SamplesToDo*sizeof(ALfloat)); + if(device->Dry.Buffer != device->FOAOut.Buffer) + for(c = 0;c < device->FOAOut.NumChannels;c++) + memset(device->FOAOut.Buffer[c], 0, SamplesToDo*sizeof(ALfloat)); + if(device->Dry.Buffer != device->RealOut.Buffer) + for(c = 0;c < device->RealOut.NumChannels;c++) + memset(device->RealOut.Buffer[c], 0, SamplesToDo*sizeof(ALfloat)); + + IncrementRef(&device->MixCount); + + ctx = ATOMIC_LOAD(&device->ContextList, almemory_order_acquire); + while(ctx) + { + const struct ALeffectslotArray *auxslots; + + auxslots = ATOMIC_LOAD(&ctx->ActiveAuxSlots, almemory_order_acquire); + ProcessParamUpdates(ctx, auxslots); + + for(i = 0;i < auxslots->count;i++) + { + ALeffectslot *slot = auxslots->slot[i]; + for(c = 0;c < slot->NumChannels;c++) + memset(slot->WetBuffer[c], 0, SamplesToDo*sizeof(ALfloat)); + } + + /* source processing */ + for(i = 0;i < ctx->VoiceCount;i++) + { + ALvoice *voice = ctx->Voices[i]; + ALsource *source = ATOMIC_LOAD(&voice->Source, almemory_order_acquire); + if(source && ATOMIC_LOAD(&voice->Playing, almemory_order_relaxed) && + voice->Step > 0) + { + if(!MixSource(voice, source->id, ctx, SamplesToDo)) + { + ATOMIC_STORE(&voice->Source, static_cast<ALsource*>(nullptr), + almemory_order_relaxed); + ATOMIC_STORE(&voice->Playing, false, almemory_order_release); + SendSourceStoppedEvent(ctx, source->id); + } + } + } + + /* effect slot processing */ + for(i = 0;i < auxslots->count;i++) + { + const ALeffectslot *slot = auxslots->slot[i]; + ALeffectState *state = slot->Params.EffectState; + V(state,process)(SamplesToDo, slot->WetBuffer, state->OutBuffer, + state->OutChannels); + } + + ctx = ATOMIC_LOAD(&ctx->next, almemory_order_relaxed); + } + + /* Increment the clock time. Every second's worth of samples is + * converted and added to clock base so that large sample counts don't + * overflow during conversion. This also guarantees an exact, stable + * conversion. */ + device->SamplesDone += SamplesToDo; + device->ClockBase += (device->SamplesDone/device->Frequency) * DEVICE_CLOCK_RES; + device->SamplesDone %= device->Frequency; + IncrementRef(&device->MixCount); + + /* Apply post-process for finalizing the Dry mix to the RealOut + * (Ambisonic decode, UHJ encode, etc). + */ + if(LIKELY(device->PostProcess)) + device->PostProcess(device, SamplesToDo); + + if(device->Stablizer) + { + int lidx = GetChannelIdxByName(&device->RealOut, FrontLeft); + int ridx = GetChannelIdxByName(&device->RealOut, FrontRight); + int cidx = GetChannelIdxByName(&device->RealOut, FrontCenter); + assert(lidx >= 0 && ridx >= 0 && cidx >= 0); + + ApplyStablizer(device->Stablizer, device->RealOut.Buffer, lidx, ridx, cidx, + SamplesToDo, device->RealOut.NumChannels); + } + + ApplyDistanceComp(device->RealOut.Buffer, device->ChannelDelay, device->TempBuffer[0], + SamplesToDo, device->RealOut.NumChannels); + + if(device->Limiter) + ApplyCompression(device->Limiter, SamplesToDo, device->RealOut.Buffer); + + if(device->DitherDepth > 0.0f) + ApplyDither(device->RealOut.Buffer, &device->DitherSeed, device->DitherDepth, + SamplesToDo, device->RealOut.NumChannels); + + if(LIKELY(OutBuffer)) + { + ALfloat (*Buffer)[BUFFERSIZE] = device->RealOut.Buffer; + ALsizei Channels = device->RealOut.NumChannels; + + switch(device->FmtType) + { +#define HANDLE_WRITE(T, S) case T: \ + Write##S(Buffer, OutBuffer, SamplesDone, SamplesToDo, Channels); break; + HANDLE_WRITE(DevFmtByte, I8) + HANDLE_WRITE(DevFmtUByte, UI8) + HANDLE_WRITE(DevFmtShort, I16) + HANDLE_WRITE(DevFmtUShort, UI16) + HANDLE_WRITE(DevFmtInt, I32) + HANDLE_WRITE(DevFmtUInt, UI32) + HANDLE_WRITE(DevFmtFloat, F32) +#undef HANDLE_WRITE + } + } + + SamplesDone += SamplesToDo; + } + END_MIXER_MODE(); +} + + +void aluHandleDisconnect(ALCdevice *device, const char *msg, ...) +{ + AsyncEvent evt = ASYNC_EVENT(EventType_Disconnected); + ALCcontext *ctx; + va_list args; + int msglen; + + if(!ATOMIC_EXCHANGE(&device->Connected, AL_FALSE, almemory_order_acq_rel)) + return; + + evt.u.user.type = AL_EVENT_TYPE_DISCONNECTED_SOFT; + evt.u.user.id = 0; + evt.u.user.param = 0; + + va_start(args, msg); + msglen = vsnprintf(evt.u.user.msg, sizeof(evt.u.user.msg), msg, args); + va_end(args); + + if(msglen < 0 || (size_t)msglen >= sizeof(evt.u.user.msg)) + evt.u.user.msg[sizeof(evt.u.user.msg)-1] = 0; + + ctx = ATOMIC_LOAD_SEQ(&device->ContextList); + while(ctx) + { + ALbitfieldSOFT enabledevt = ATOMIC_LOAD(&ctx->EnabledEvts, almemory_order_acquire); + ALsizei i; + + if((enabledevt&EventType_Disconnected) && + ll_ringbuffer_write(ctx->AsyncEvents, (const char*)&evt, 1) == 1) + alsem_post(&ctx->EventSem); + + for(i = 0;i < ctx->VoiceCount;i++) + { + ALvoice *voice = ctx->Voices[i]; + ALsource *source; + + source = static_cast<ALsource*>(ATOMIC_EXCHANGE_PTR(&voice->Source, + static_cast<ALsource*>(nullptr), almemory_order_relaxed)); + if(source && ATOMIC_LOAD(&voice->Playing, almemory_order_relaxed)) + { + /* If the source's voice was playing, it's now effectively + * stopped (the source state will be updated the next time it's + * checked). + */ + SendSourceStoppedEvent(ctx, source->id); + } + ATOMIC_STORE(&voice->Playing, false, almemory_order_release); + } + + ctx = ATOMIC_LOAD(&ctx->next, almemory_order_relaxed); + } +} |