/** * 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., 59 Temple Place - Suite 330, * Boston, MA 02111-1307, USA. * Or go to http://www.gnu.org/copyleft/lgpl.html */ #include "config.h" #include #include #include #include #include #include "alMain.h" #include "AL/al.h" #include "AL/alc.h" #include "alSource.h" #include "alBuffer.h" #include "alListener.h" #include "alAuxEffectSlot.h" #include "alu.h" #include "bs2b.h" struct ChanMap { enum Channel channel; ALfloat angle; }; /* Cone scalar */ ALfloat ConeScale = 0.5f; /* Localized Z scalar for mono sources */ ALfloat ZScale = 1.0f; static __inline ALvoid aluMatrixVector(ALfloat *vector,ALfloat w,ALfloat matrix[4][4]) { ALfloat temp[4] = { vector[0], vector[1], vector[2], w }; vector[0] = temp[0]*matrix[0][0] + temp[1]*matrix[1][0] + temp[2]*matrix[2][0] + temp[3]*matrix[3][0]; vector[1] = temp[0]*matrix[0][1] + temp[1]*matrix[1][1] + temp[2]*matrix[2][1] + temp[3]*matrix[3][1]; vector[2] = temp[0]*matrix[0][2] + temp[1]*matrix[1][2] + temp[2]*matrix[2][2] + temp[3]*matrix[3][2]; } ALvoid CalcNonAttnSourceParams(ALsource *ALSource, const ALCcontext *ALContext) { static const struct ChanMap MonoMap[1] = { { FrontCenter, 0.0f } }; static const struct ChanMap StereoMap[2] = { { FrontLeft, -30.0f * F_PI/180.0f }, { FrontRight, 30.0f * F_PI/180.0f } }; static const struct ChanMap RearMap[2] = { { BackLeft, -150.0f * F_PI/180.0f }, { BackRight, 150.0f * F_PI/180.0f } }; static const struct ChanMap QuadMap[4] = { { FrontLeft, -45.0f * F_PI/180.0f }, { FrontRight, 45.0f * F_PI/180.0f }, { BackLeft, -135.0f * F_PI/180.0f }, { BackRight, 135.0f * F_PI/180.0f } }; static const struct ChanMap X51Map[6] = { { FrontLeft, -30.0f * F_PI/180.0f }, { FrontRight, 30.0f * F_PI/180.0f }, { FrontCenter, 0.0f * F_PI/180.0f }, { LFE, 0.0f }, { BackLeft, -110.0f * F_PI/180.0f }, { BackRight, 110.0f * F_PI/180.0f } }; static const struct ChanMap X61Map[7] = { { FrontLeft, -30.0f * F_PI/180.0f }, { FrontRight, 30.0f * F_PI/180.0f }, { FrontCenter, 0.0f * F_PI/180.0f }, { LFE, 0.0f }, { BackCenter, 180.0f * F_PI/180.0f }, { SideLeft, -90.0f * F_PI/180.0f }, { SideRight, 90.0f * F_PI/180.0f } }; static const struct ChanMap X71Map[8] = { { FrontLeft, -30.0f * F_PI/180.0f }, { FrontRight, 30.0f * F_PI/180.0f }, { FrontCenter, 0.0f * F_PI/180.0f }, { LFE, 0.0f }, { BackLeft, -150.0f * F_PI/180.0f }, { BackRight, 150.0f * F_PI/180.0f }, { SideLeft, -90.0f * F_PI/180.0f }, { SideRight, 90.0f * F_PI/180.0f } }; ALCdevice *Device = ALContext->Device; ALfloat SourceVolume,ListenerGain,MinVolume,MaxVolume; ALbufferlistitem *BufferListItem; enum FmtChannels Channels; ALfloat (*SrcMatrix)[MaxChannels]; ALfloat DryGain, DryGainHF; ALfloat WetGain[MAX_SENDS]; ALfloat WetGainHF[MAX_SENDS]; ALint NumSends, Frequency; const struct ChanMap *chans = NULL; enum Resampler Resampler; ALint num_channels = 0; ALboolean DirectChannels; ALfloat Pitch; ALfloat cw; ALint i, c; /* Get device properties */ NumSends = Device->NumAuxSends; Frequency = Device->Frequency; /* Get listener properties */ ListenerGain = ALContext->Listener.Gain; /* Get source properties */ SourceVolume = ALSource->Gain; MinVolume = ALSource->MinGain; MaxVolume = ALSource->MaxGain; Pitch = ALSource->Pitch; Resampler = ALSource->Resampler; DirectChannels = ALSource->DirectChannels; /* Calculate the stepping value */ Channels = FmtMono; BufferListItem = ALSource->queue; while(BufferListItem != NULL) { ALbuffer *ALBuffer; if((ALBuffer=BufferListItem->buffer) != NULL) { ALsizei maxstep = STACK_DATA_SIZE/sizeof(ALfloat) / ALSource->NumChannels; maxstep -= ResamplerPadding[Resampler] + ResamplerPrePadding[Resampler] + 1; maxstep = mini(maxstep, INT_MAX>>FRACTIONBITS); Pitch = Pitch * ALBuffer->Frequency / Frequency; if(Pitch > (ALfloat)maxstep) ALSource->Params.Step = maxstep<Params.Step = fastf2i(Pitch*FRACTIONONE); if(ALSource->Params.Step == 0) ALSource->Params.Step = 1; } if(ALSource->Params.Step == FRACTIONONE) Resampler = PointResampler; Channels = ALBuffer->FmtChannels; break; } BufferListItem = BufferListItem->next; } if(!DirectChannels && Device->Hrtf) ALSource->Params.DryMix = SelectHrtfMixer(Resampler); else ALSource->Params.DryMix = SelectDirectMixer(Resampler); ALSource->Params.WetMix = SelectSendMixer(Resampler); /* Calculate gains */ DryGain = clampf(SourceVolume, MinVolume, MaxVolume); DryGain *= ALSource->DirectGain * ListenerGain; DryGainHF = ALSource->DirectGainHF; for(i = 0;i < NumSends;i++) { WetGain[i] = clampf(SourceVolume, MinVolume, MaxVolume); WetGain[i] *= ALSource->Send[i].Gain * ListenerGain; WetGainHF[i] = ALSource->Send[i].GainHF; } SrcMatrix = ALSource->Params.Direct.Gains; for(i = 0;i < MaxChannels;i++) { for(c = 0;c < MaxChannels;c++) SrcMatrix[i][c] = 0.0f; } switch(Channels) { case FmtMono: chans = MonoMap; num_channels = 1; break; case FmtStereo: chans = StereoMap; num_channels = 2; break; case FmtRear: chans = RearMap; num_channels = 2; break; case FmtQuad: chans = QuadMap; num_channels = 4; break; case FmtX51: chans = X51Map; num_channels = 6; break; case FmtX61: chans = X61Map; num_channels = 7; break; case FmtX71: chans = X71Map; num_channels = 8; break; } if(DirectChannels != AL_FALSE) { for(c = 0;c < num_channels;c++) { for(i = 0;i < (ALint)Device->NumChan;i++) { enum Channel chan = Device->Speaker2Chan[i]; if(chan == chans[c].channel) { SrcMatrix[c][chan] += DryGain; break; } } } } else if(Device->Hrtf) { for(c = 0;c < num_channels;c++) { if(chans[c].channel == LFE) { /* Skip LFE */ ALSource->Params.Direct.Hrtf.Delay[c][0] = 0; ALSource->Params.Direct.Hrtf.Delay[c][1] = 0; for(i = 0;i < HRIR_LENGTH;i++) { ALSource->Params.Direct.Hrtf.Coeffs[c][i][0] = 0.0f; ALSource->Params.Direct.Hrtf.Coeffs[c][i][1] = 0.0f; } } else { /* Get the static HRIR coefficients and delays for this * channel. */ GetLerpedHrtfCoeffs(Device->Hrtf, 0.0f, chans[c].angle, DryGain, ALSource->Params.Direct.Hrtf.Coeffs[c], ALSource->Params.Direct.Hrtf.Delay[c]); } } ALSource->Hrtf.Counter = 0; } else { for(c = 0;c < num_channels;c++) { /* Special-case LFE */ if(chans[c].channel == LFE) { SrcMatrix[c][chans[c].channel] = DryGain; continue; } ComputeAngleGains(Device, chans[c].angle, 0.0f, DryGain, SrcMatrix[c]); } } for(i = 0;i < NumSends;i++) { ALeffectslot *Slot = ALSource->Send[i].Slot; if(!Slot && i == 0) Slot = Device->DefaultSlot; if(Slot && Slot->effect.type == AL_EFFECT_NULL) Slot = NULL; ALSource->Params.Slot[i] = Slot; ALSource->Params.Send[i].Gain = WetGain[i]; } /* Update filter coefficients. Calculations based on the I3DL2 * spec. */ cw = cosf(F_PI*2.0f * LOWPASSFREQREF / Frequency); /* We use two chained one-pole filters, so we need to take the * square root of the squared gain, which is the same as the base * gain. */ ALSource->Params.Direct.iirFilter.coeff = lpCoeffCalc(DryGainHF, cw); for(i = 0;i < NumSends;i++) { ALfloat a = lpCoeffCalc(WetGainHF[i], cw); ALSource->Params.Send[i].iirFilter.coeff = a; } } ALvoid CalcSourceParams(ALsource *ALSource, const ALCcontext *ALContext) { const ALCdevice *Device = ALContext->Device; ALfloat InnerAngle,OuterAngle,Angle,Distance,ClampedDist; ALfloat Direction[3],Position[3],SourceToListener[3]; ALfloat Velocity[3],ListenerVel[3]; ALfloat MinVolume,MaxVolume,MinDist,MaxDist,Rolloff; ALfloat ConeVolume,ConeHF,SourceVolume,ListenerGain; ALfloat DopplerFactor, SpeedOfSound; ALfloat AirAbsorptionFactor; ALfloat RoomAirAbsorption[MAX_SENDS]; ALbufferlistitem *BufferListItem; ALfloat Attenuation; ALfloat RoomAttenuation[MAX_SENDS]; ALfloat MetersPerUnit; ALfloat RoomRolloffBase; ALfloat RoomRolloff[MAX_SENDS]; ALfloat DecayDistance[MAX_SENDS]; ALfloat DryGain; ALfloat DryGainHF; ALboolean DryGainHFAuto; ALfloat WetGain[MAX_SENDS]; ALfloat WetGainHF[MAX_SENDS]; ALboolean WetGainAuto; ALboolean WetGainHFAuto; enum Resampler Resampler; ALfloat Matrix[4][4]; ALfloat Pitch; ALuint Frequency; ALint NumSends; ALfloat cw; ALint i, j; DryGainHF = 1.0f; for(i = 0;i < MAX_SENDS;i++) WetGainHF[i] = 1.0f; /* Get context/device properties */ DopplerFactor = ALContext->DopplerFactor * ALSource->DopplerFactor; SpeedOfSound = ALContext->SpeedOfSound * ALContext->DopplerVelocity; NumSends = Device->NumAuxSends; Frequency = Device->Frequency; /* Get listener properties */ ListenerGain = ALContext->Listener.Gain; MetersPerUnit = ALContext->Listener.MetersPerUnit; ListenerVel[0] = ALContext->Listener.Velocity[0]; ListenerVel[1] = ALContext->Listener.Velocity[1]; ListenerVel[2] = ALContext->Listener.Velocity[2]; for(i = 0;i < 4;i++) { for(j = 0;j < 4;j++) Matrix[i][j] = ALContext->Listener.Matrix[i][j]; } /* Get source properties */ SourceVolume = ALSource->Gain; MinVolume = ALSource->MinGain; MaxVolume = ALSource->MaxGain; Pitch = ALSource->Pitch; Resampler = ALSource->Resampler; Position[0] = ALSource->Position[0]; Position[1] = ALSource->Position[1]; Position[2] = ALSource->Position[2]; Direction[0] = ALSource->Orientation[0]; Direction[1] = ALSource->Orientation[1]; Direction[2] = ALSource->Orientation[2]; Velocity[0] = ALSource->Velocity[0]; Velocity[1] = ALSource->Velocity[1]; Velocity[2] = ALSource->Velocity[2]; MinDist = ALSource->RefDistance; MaxDist = ALSource->MaxDistance; Rolloff = ALSource->RollOffFactor; InnerAngle = ALSource->InnerAngle * ConeScale; OuterAngle = ALSource->OuterAngle * ConeScale; AirAbsorptionFactor = ALSource->AirAbsorptionFactor; DryGainHFAuto = ALSource->DryGainHFAuto; WetGainAuto = ALSource->WetGainAuto; WetGainHFAuto = ALSource->WetGainHFAuto; RoomRolloffBase = ALSource->RoomRolloffFactor; for(i = 0;i < NumSends;i++) { ALeffectslot *Slot = ALSource->Send[i].Slot; if(!Slot && i == 0) Slot = Device->DefaultSlot; if(!Slot || Slot->effect.type == AL_EFFECT_NULL) { Slot = NULL; RoomRolloff[i] = 0.0f; DecayDistance[i] = 0.0f; RoomAirAbsorption[i] = 1.0f; } else if(Slot->AuxSendAuto) { RoomRolloff[i] = RoomRolloffBase; if(IsReverbEffect(Slot->effect.type)) { RoomRolloff[i] += Slot->effect.Reverb.RoomRolloffFactor; DecayDistance[i] = Slot->effect.Reverb.DecayTime * SPEEDOFSOUNDMETRESPERSEC; RoomAirAbsorption[i] = Slot->effect.Reverb.AirAbsorptionGainHF; } else { DecayDistance[i] = 0.0f; RoomAirAbsorption[i] = 1.0f; } } 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] = Rolloff; DecayDistance[i] = 0.0f; RoomAirAbsorption[i] = AIRABSORBGAINHF; } ALSource->Params.Slot[i] = Slot; } /* Transform source to listener space (convert to head relative) */ if(ALSource->HeadRelative == AL_FALSE) { /* Translate position */ Position[0] -= ALContext->Listener.Position[0]; Position[1] -= ALContext->Listener.Position[1]; Position[2] -= ALContext->Listener.Position[2]; /* Transform source vectors */ aluMatrixVector(Position, 1.0f, Matrix); aluMatrixVector(Direction, 0.0f, Matrix); aluMatrixVector(Velocity, 0.0f, Matrix); /* Transform listener velocity */ aluMatrixVector(ListenerVel, 0.0f, Matrix); } else { /* Transform listener velocity from world space to listener space */ aluMatrixVector(ListenerVel, 0.0f, Matrix); /* Offset the source velocity to be relative of the listener velocity */ Velocity[0] += ListenerVel[0]; Velocity[1] += ListenerVel[1]; Velocity[2] += ListenerVel[2]; } SourceToListener[0] = -Position[0]; SourceToListener[1] = -Position[1]; SourceToListener[2] = -Position[2]; aluNormalize(SourceToListener); aluNormalize(Direction); /* Calculate distance attenuation */ Distance = sqrtf(aluDotproduct(Position, Position)); ClampedDist = Distance; Attenuation = 1.0f; for(i = 0;i < NumSends;i++) RoomAttenuation[i] = 1.0f; switch(ALContext->SourceDistanceModel ? ALSource->DistanceModel : ALContext->DistanceModel) { case InverseDistanceClamped: ClampedDist = clampf(ClampedDist, MinDist, MaxDist); if(MaxDist < MinDist) break; /*fall-through*/ case InverseDistance: if(MinDist > 0.0f) { if((MinDist + (Rolloff * (ClampedDist - MinDist))) > 0.0f) Attenuation = MinDist / (MinDist + (Rolloff * (ClampedDist - MinDist))); for(i = 0;i < NumSends;i++) { if((MinDist + (RoomRolloff[i] * (ClampedDist - MinDist))) > 0.0f) RoomAttenuation[i] = MinDist / (MinDist + (RoomRolloff[i] * (ClampedDist - MinDist))); } } break; case LinearDistanceClamped: ClampedDist = clampf(ClampedDist, MinDist, MaxDist); if(MaxDist < MinDist) break; /*fall-through*/ case LinearDistance: if(MaxDist != MinDist) { Attenuation = 1.0f - (Rolloff*(ClampedDist-MinDist)/(MaxDist - MinDist)); Attenuation = maxf(Attenuation, 0.0f); for(i = 0;i < NumSends;i++) { RoomAttenuation[i] = 1.0f - (RoomRolloff[i]*(ClampedDist-MinDist)/(MaxDist - MinDist)); RoomAttenuation[i] = maxf(RoomAttenuation[i], 0.0f); } } break; case ExponentDistanceClamped: ClampedDist = clampf(ClampedDist, MinDist, MaxDist); if(MaxDist < MinDist) break; /*fall-through*/ case ExponentDistance: if(ClampedDist > 0.0f && MinDist > 0.0f) { Attenuation = powf(ClampedDist/MinDist, -Rolloff); for(i = 0;i < NumSends;i++) RoomAttenuation[i] = powf(ClampedDist/MinDist, -RoomRolloff[i]); } break; case DisableDistance: ClampedDist = MinDist; break; } /* Source Gain + Attenuation */ DryGain = SourceVolume * Attenuation; for(i = 0;i < NumSends;i++) WetGain[i] = SourceVolume * RoomAttenuation[i]; /* Distance-based air absorption */ if(AirAbsorptionFactor > 0.0f && ClampedDist > MinDist) { ALfloat meters = maxf(ClampedDist-MinDist, 0.0f) * MetersPerUnit; DryGainHF *= powf(AIRABSORBGAINHF, AirAbsorptionFactor*meters); for(i = 0;i < NumSends;i++) WetGainHF[i] *= powf(RoomAirAbsorption[i], AirAbsorptionFactor*meters); } if(WetGainAuto) { ALfloat ApparentDist = 1.0f/maxf(Attenuation, 0.00001f) - 1.0f; /* Apply a decay-time transformation to the wet path, based on the * attenuation of the dry path. * * Using the apparent distance, based on the distance attenuation, the * initial decay of the reverb effect is calculated and applied to the * wet path. */ for(i = 0;i < NumSends;i++) { if(DecayDistance[i] > 0.0f) WetGain[i] *= powf(0.001f/*-60dB*/, ApparentDist/DecayDistance[i]); } } /* Calculate directional soundcones */ Angle = acosf(aluDotproduct(Direction,SourceToListener)) * (180.0f/F_PI); if(Angle > InnerAngle && Angle <= OuterAngle) { ALfloat scale = (Angle-InnerAngle) / (OuterAngle-InnerAngle); ConeVolume = lerp(1.0f, ALSource->OuterGain, scale); ConeHF = lerp(1.0f, ALSource->OuterGainHF, scale); } else if(Angle > OuterAngle) { ConeVolume = ALSource->OuterGain; ConeHF = ALSource->OuterGainHF; } else { ConeVolume = 1.0f; ConeHF = 1.0f; } DryGain *= ConeVolume; if(WetGainAuto) { for(i = 0;i < NumSends;i++) WetGain[i] *= ConeVolume; } if(DryGainHFAuto) DryGainHF *= ConeHF; if(WetGainHFAuto) { for(i = 0;i < NumSends;i++) WetGainHF[i] *= ConeHF; } /* Clamp to Min/Max Gain */ DryGain = clampf(DryGain, MinVolume, MaxVolume); for(i = 0;i < NumSends;i++) WetGain[i] = clampf(WetGain[i], MinVolume, MaxVolume); /* Apply gain and frequency filters */ DryGain *= ALSource->DirectGain * ListenerGain; DryGainHF *= ALSource->DirectGainHF; for(i = 0;i < NumSends;i++) { WetGain[i] *= ALSource->Send[i].Gain * ListenerGain; WetGainHF[i] *= ALSource->Send[i].GainHF; } /* Calculate velocity-based doppler effect */ if(DopplerFactor > 0.0f) { ALfloat VSS, VLS; if(SpeedOfSound < 1.0f) { DopplerFactor *= 1.0f/SpeedOfSound; SpeedOfSound = 1.0f; } VSS = aluDotproduct(Velocity, SourceToListener) * DopplerFactor; VLS = aluDotproduct(ListenerVel, SourceToListener) * DopplerFactor; Pitch *= clampf(SpeedOfSound-VLS, 1.0f, SpeedOfSound*2.0f - 1.0f) / clampf(SpeedOfSound-VSS, 1.0f, SpeedOfSound*2.0f - 1.0f); } BufferListItem = ALSource->queue; while(BufferListItem != NULL) { ALbuffer *ALBuffer; if((ALBuffer=BufferListItem->buffer) != NULL) { /* Calculate fixed-point stepping value, based on the pitch, buffer * frequency, and output frequency. */ ALsizei maxstep = STACK_DATA_SIZE/sizeof(ALfloat) / ALSource->NumChannels; maxstep -= ResamplerPadding[Resampler] + ResamplerPrePadding[Resampler] + 1; maxstep = mini(maxstep, INT_MAX>>FRACTIONBITS); Pitch = Pitch * ALBuffer->Frequency / Frequency; if(Pitch > (ALfloat)maxstep) ALSource->Params.Step = maxstep<Params.Step = fastf2i(Pitch*FRACTIONONE); if(ALSource->Params.Step == 0) ALSource->Params.Step = 1; } if(ALSource->Params.Step == FRACTIONONE) Resampler = PointResampler; break; } BufferListItem = BufferListItem->next; } if(Device->Hrtf) ALSource->Params.DryMix = SelectHrtfMixer(Resampler); else ALSource->Params.DryMix = SelectDirectMixer(Resampler); ALSource->Params.WetMix = SelectSendMixer(Resampler); if(Device->Hrtf) { /* Use a binaural HRTF algorithm for stereo headphone playback */ ALfloat delta, ev = 0.0f, az = 0.0f; if(Distance > 0.0f) { ALfloat invlen = 1.0f/Distance; Position[0] *= invlen; Position[1] *= invlen; Position[2] *= invlen; /* Calculate elevation and azimuth only when the source is not at * the listener. This prevents +0 and -0 Z from producing * inconsistent panning. Also, clamp Y in case FP precision errors * cause it to land outside of -1..+1. */ ev = asinf(clampf(Position[1], -1.0f, 1.0f)); az = atan2f(Position[0], -Position[2]*ZScale); } /* Check to see if the HRIR is already moving. */ if(ALSource->Hrtf.Moving) { /* Calculate the normalized HRTF transition factor (delta). */ delta = CalcHrtfDelta(ALSource->Params.Direct.Hrtf.Gain, DryGain, ALSource->Params.Direct.Hrtf.Dir, Position); /* If the delta is large enough, get the moving HRIR target * coefficients, target delays, steppping values, and counter. */ if(delta > 0.001f) { ALSource->Hrtf.Counter = GetMovingHrtfCoeffs(Device->Hrtf, ev, az, DryGain, delta, ALSource->Hrtf.Counter, ALSource->Params.Direct.Hrtf.Coeffs[0], ALSource->Params.Direct.Hrtf.Delay[0], ALSource->Params.Direct.Hrtf.CoeffStep, ALSource->Params.Direct.Hrtf.DelayStep); ALSource->Params.Direct.Hrtf.Gain = DryGain; ALSource->Params.Direct.Hrtf.Dir[0] = Position[0]; ALSource->Params.Direct.Hrtf.Dir[1] = Position[1]; ALSource->Params.Direct.Hrtf.Dir[2] = Position[2]; } } else { /* Get the initial (static) HRIR coefficients and delays. */ GetLerpedHrtfCoeffs(Device->Hrtf, ev, az, DryGain, ALSource->Params.Direct.Hrtf.Coeffs[0], ALSource->Params.Direct.Hrtf.Delay[0]); ALSource->Hrtf.Counter = 0; ALSource->Params.Direct.Hrtf.Gain = DryGain; ALSource->Params.Direct.Hrtf.Dir[0] = Position[0]; ALSource->Params.Direct.Hrtf.Dir[1] = Position[1]; ALSource->Params.Direct.Hrtf.Dir[2] = Position[2]; } } else { ALfloat (*Matrix)[MaxChannels] = ALSource->Params.Direct.Gains; ALfloat DirGain = 0.0f; ALfloat AmbientGain; for(i = 0;i < MaxChannels;i++) { for(j = 0;j < MaxChannels;j++) Matrix[i][j] = 0.0f; } /* Normalize the length, and compute panned gains. */ if(Distance > 0.0f) { ALfloat invlen = 1.0f/Distance; Position[0] *= invlen; Position[1] *= invlen; Position[2] *= invlen; DirGain = sqrtf(Position[0]*Position[0] + Position[2]*Position[2]); ComputeAngleGains(Device, atan2f(Position[0], -Position[2]*ZScale), 0.0f, DryGain*DirGain, Matrix[0]); } /* Adjustment for vertical offsets. Not the greatest, but simple * enough. */ AmbientGain = DryGain * sqrtf(1.0f/Device->NumChan) * (1.0f-DirGain); for(i = 0;i < (ALint)Device->NumChan;i++) { enum Channel chan = Device->Speaker2Chan[i]; Matrix[0][chan] = maxf(Matrix[0][chan], AmbientGain); } } for(i = 0;i < NumSends;i++) ALSource->Params.Send[i].Gain = WetGain[i]; /* Update filter coefficients. */ cw = cosf(F_PI*2.0f * LOWPASSFREQREF / Frequency); ALSource->Params.Direct.iirFilter.coeff = lpCoeffCalc(DryGainHF, cw); for(i = 0;i < NumSends;i++) { ALfloat a = lpCoeffCalc(WetGainHF[i], cw); ALSource->Params.Send[i].iirFilter.coeff = a; } } static __inline ALfloat aluF2F(ALfloat val) { return val; } static __inline ALint aluF2I(ALfloat val) { if(val > 1.0f) return 2147483647; if(val < -1.0f) return -2147483647-1; return fastf2i((ALfloat)(val*2147483647.0)); } static __inline ALuint aluF2UI(ALfloat val) { return aluF2I(val)+2147483648u; } static __inline ALshort aluF2S(ALfloat val) { return aluF2I(val)>>16; } static __inline ALushort aluF2US(ALfloat val) { return aluF2S(val)+32768; } static __inline ALbyte aluF2B(ALfloat val) { return aluF2I(val)>>24; } static __inline ALubyte aluF2UB(ALfloat val) { return aluF2B(val)+128; } #define DECL_TEMPLATE(T, N, func) \ static void Write_##T##_##N(ALCdevice *device, T *RESTRICT buffer, \ ALuint SamplesToDo) \ { \ ALfloat (*RESTRICT DryBuffer)[MaxChannels] = device->DryBuffer; \ const enum Channel *ChanMap = device->DevChannels; \ ALuint i, j; \ \ for(j = 0;j < N;j++) \ { \ T *RESTRICT out = buffer + j; \ enum Channel chan = ChanMap[j]; \ \ for(i = 0;i < SamplesToDo;i++) \ out[i*N] = func(DryBuffer[i][chan]); \ } \ } DECL_TEMPLATE(ALfloat, 1, aluF2F) DECL_TEMPLATE(ALfloat, 2, aluF2F) DECL_TEMPLATE(ALfloat, 4, aluF2F) DECL_TEMPLATE(ALfloat, 6, aluF2F) DECL_TEMPLATE(ALfloat, 7, aluF2F) DECL_TEMPLATE(ALfloat, 8, aluF2F) DECL_TEMPLATE(ALuint, 1, aluF2UI) DECL_TEMPLATE(ALuint, 2, aluF2UI) DECL_TEMPLATE(ALuint, 4, aluF2UI) DECL_TEMPLATE(ALuint, 6, aluF2UI) DECL_TEMPLATE(ALuint, 7, aluF2UI) DECL_TEMPLATE(ALuint, 8, aluF2UI) DECL_TEMPLATE(ALint, 1, aluF2I) DECL_TEMPLATE(ALint, 2, aluF2I) DECL_TEMPLATE(ALint, 4, aluF2I) DECL_TEMPLATE(ALint, 6, aluF2I) DECL_TEMPLATE(ALint, 7, aluF2I) DECL_TEMPLATE(ALint, 8, aluF2I) DECL_TEMPLATE(ALushort, 1, aluF2US) DECL_TEMPLATE(ALushort, 2, aluF2US) DECL_TEMPLATE(ALushort, 4, aluF2US) DECL_TEMPLATE(ALushort, 6, aluF2US) DECL_TEMPLATE(ALushort, 7, aluF2US) DECL_TEMPLATE(ALushort, 8, aluF2US) DECL_TEMPLATE(ALshort, 1, aluF2S) DECL_TEMPLATE(ALshort, 2, aluF2S) DECL_TEMPLATE(ALshort, 4, aluF2S) DECL_TEMPLATE(ALshort, 6, aluF2S) DECL_TEMPLATE(ALshort, 7, aluF2S) DECL_TEMPLATE(ALshort, 8, aluF2S) DECL_TEMPLATE(ALubyte, 1, aluF2UB) DECL_TEMPLATE(ALubyte, 2, aluF2UB) DECL_TEMPLATE(ALubyte, 4, aluF2UB) DECL_TEMPLATE(ALubyte, 6, aluF2UB) DECL_TEMPLATE(ALubyte, 7, aluF2UB) DECL_TEMPLATE(ALubyte, 8, aluF2UB) DECL_TEMPLATE(ALbyte, 1, aluF2B) DECL_TEMPLATE(ALbyte, 2, aluF2B) DECL_TEMPLATE(ALbyte, 4, aluF2B) DECL_TEMPLATE(ALbyte, 6, aluF2B) DECL_TEMPLATE(ALbyte, 7, aluF2B) DECL_TEMPLATE(ALbyte, 8, aluF2B) #undef DECL_TEMPLATE #define DECL_TEMPLATE(T) \ static void Write_##T(ALCdevice *device, T *buffer, ALuint SamplesToDo) \ { \ switch(device->FmtChans) \ { \ case DevFmtMono: \ Write_##T##_1(device, buffer, SamplesToDo); \ break; \ case DevFmtStereo: \ Write_##T##_2(device, buffer, SamplesToDo); \ break; \ case DevFmtQuad: \ Write_##T##_4(device, buffer, SamplesToDo); \ break; \ case DevFmtX51: \ case DevFmtX51Side: \ Write_##T##_6(device, buffer, SamplesToDo); \ break; \ case DevFmtX61: \ Write_##T##_7(device, buffer, SamplesToDo); \ break; \ case DevFmtX71: \ Write_##T##_8(device, buffer, SamplesToDo); \ break; \ } \ } DECL_TEMPLATE(ALfloat) DECL_TEMPLATE(ALuint) DECL_TEMPLATE(ALint) DECL_TEMPLATE(ALushort) DECL_TEMPLATE(ALshort) DECL_TEMPLATE(ALubyte) DECL_TEMPLATE(ALbyte) #undef DECL_TEMPLATE ALvoid aluMixData(ALCdevice *device, ALvoid *buffer, ALsizei size) { ALuint SamplesToDo; ALeffectslot **slot, **slot_end; ALsource **src, **src_end; ALCcontext *ctx; int fpuState; ALuint i, c; fpuState = SetMixerFPUMode(); while(size > 0) { SamplesToDo = minu(size, BUFFERSIZE); memset(device->DryBuffer, 0, SamplesToDo*MaxChannels*sizeof(ALfloat)); LockDevice(device); ctx = device->ContextList; while(ctx) { ALenum DeferUpdates = ctx->DeferUpdates; ALenum UpdateSources = AL_FALSE; if(!DeferUpdates) UpdateSources = ExchangeInt(&ctx->UpdateSources, AL_FALSE); /* source processing */ src = ctx->ActiveSources; src_end = src + ctx->ActiveSourceCount; while(src != src_end) { if((*src)->state != AL_PLAYING) { --(ctx->ActiveSourceCount); *src = *(--src_end); continue; } if(!DeferUpdates && (ExchangeInt(&(*src)->NeedsUpdate, AL_FALSE) || UpdateSources)) ALsource_Update(*src, ctx); MixSource(*src, device, SamplesToDo); src++; } /* effect slot processing */ slot = ctx->ActiveEffectSlots; slot_end = slot + ctx->ActiveEffectSlotCount; while(slot != slot_end) { for(c = 0;c < SamplesToDo;c++) { (*slot)->WetBuffer[c] += (*slot)->ClickRemoval[0]; (*slot)->ClickRemoval[0] -= (*slot)->ClickRemoval[0] * (1.0f/256.0f); } (*slot)->ClickRemoval[0] += (*slot)->PendingClicks[0]; (*slot)->PendingClicks[0] = 0.0f; if(!DeferUpdates && ExchangeInt(&(*slot)->NeedsUpdate, AL_FALSE)) ALeffectState_Update((*slot)->EffectState, device, *slot); ALeffectState_Process((*slot)->EffectState, SamplesToDo, (*slot)->WetBuffer, device->DryBuffer); for(i = 0;i < SamplesToDo;i++) (*slot)->WetBuffer[i] = 0.0f; slot++; } ctx = ctx->next; } slot = &device->DefaultSlot; if(*slot != NULL) { for(c = 0;c < SamplesToDo;c++) { (*slot)->WetBuffer[c] += (*slot)->ClickRemoval[0]; (*slot)->ClickRemoval[0] -= (*slot)->ClickRemoval[0] * (1.0f/256.0f); } (*slot)->ClickRemoval[0] += (*slot)->PendingClicks[0]; (*slot)->PendingClicks[0] = 0.0f; if(ExchangeInt(&(*slot)->NeedsUpdate, AL_FALSE)) ALeffectState_Update((*slot)->EffectState, device, *slot); ALeffectState_Process((*slot)->EffectState, SamplesToDo, (*slot)->WetBuffer, device->DryBuffer); for(i = 0;i < SamplesToDo;i++) (*slot)->WetBuffer[i] = 0.0f; } UnlockDevice(device); /* Click-removal. Could do better; this only really handles immediate * changes between updates where a predictive sample could be * generated. Delays caused by effects and HRTF aren't caught. */ if(device->FmtChans == DevFmtMono) { for(i = 0;i < SamplesToDo;i++) { device->DryBuffer[i][FrontCenter] += device->ClickRemoval[FrontCenter]; device->ClickRemoval[FrontCenter] -= device->ClickRemoval[FrontCenter] * (1.0f/256.0f); } device->ClickRemoval[FrontCenter] += device->PendingClicks[FrontCenter]; device->PendingClicks[FrontCenter] = 0.0f; } else if(device->FmtChans == DevFmtStereo) { /* Assumes the first two channels are FrontLeft and FrontRight */ for(i = 0;i < SamplesToDo;i++) { for(c = 0;c < 2;c++) { device->DryBuffer[i][c] += device->ClickRemoval[c]; device->ClickRemoval[c] -= device->ClickRemoval[c] * (1.0f/256.0f); } } for(c = 0;c < 2;c++) { device->ClickRemoval[c] += device->PendingClicks[c]; device->PendingClicks[c] = 0.0f; } if(device->Bs2b) { for(i = 0;i < SamplesToDo;i++) bs2b_cross_feed(device->Bs2b, &device->DryBuffer[i][0]); } } else { for(i = 0;i < SamplesToDo;i++) { for(c = 0;c < MaxChannels;c++) { device->DryBuffer[i][c] += device->ClickRemoval[c]; device->ClickRemoval[c] -= device->ClickRemoval[c] * (1.0f/256.0f); } } for(c = 0;c < MaxChannels;c++) { device->ClickRemoval[c] += device->PendingClicks[c]; device->PendingClicks[c] = 0.0f; } } if(buffer) { switch(device->FmtType) { case DevFmtByte: Write_ALbyte(device, buffer, SamplesToDo); break; case DevFmtUByte: Write_ALubyte(device, buffer, SamplesToDo); break; case DevFmtShort: Write_ALshort(device, buffer, SamplesToDo); break; case DevFmtUShort: Write_ALushort(device, buffer, SamplesToDo); break; case DevFmtInt: Write_ALint(device, buffer, SamplesToDo); break; case DevFmtUInt: Write_ALuint(device, buffer, SamplesToDo); break; case DevFmtFloat: Write_ALfloat(device, buffer, SamplesToDo); break; } } size -= SamplesToDo; } RestoreFPUMode(fpuState); } ALvoid aluHandleDisconnect(ALCdevice *device) { ALCcontext *Context; LockDevice(device); device->Connected = ALC_FALSE; Context = device->ContextList; while(Context) { ALsource **src, **src_end; src = Context->ActiveSources; src_end = src + Context->ActiveSourceCount; while(src != src_end) { if((*src)->state == AL_PLAYING) { (*src)->state = AL_STOPPED; (*src)->BuffersPlayed = (*src)->BuffersInQueue; (*src)->position = 0; (*src)->position_fraction = 0; } src++; } Context->ActiveSourceCount = 0; Context = Context->next; } UnlockDevice(device); }