/** * 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" static __inline ALvoid 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 ALfloat *inVector1, const ALfloat *inVector2) { return inVector1[0]*inVector2[0] + inVector1[1]*inVector2[1] + inVector1[2]*inVector2[2]; } static __inline ALvoid aluNormalize(ALfloat *inVector) { ALfloat length, inverse_length; length = aluSqrt(aluDotproduct(inVector, inVector)); if(length != 0.0f) { inverse_length = 1.0f/length; inVector[0] *= inverse_length; inVector[1] *= inverse_length; inVector[2] *= inverse_length; } } 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) { ALfloat SourceVolume,ListenerGain,MinVolume,MaxVolume; ALbufferlistitem *BufferListItem; ALfloat DryGain, DryGainHF; ALfloat WetGain[MAX_SENDS]; ALfloat WetGainHF[MAX_SENDS]; ALint NumSends, Frequency; ALboolean DupStereo; ALint Channels; ALfloat Pitch; ALenum Format; ALfloat cw; ALint i; //Get context properties Format = ALContext->Device->Format; DupStereo = ALContext->Device->DuplicateStereo; NumSends = ALContext->Device->NumAuxSends; Frequency = ALContext->Device->Frequency; //Get listener properties ListenerGain = ALContext->Listener.Gain; //Get source properties SourceVolume = ALSource->flGain; MinVolume = ALSource->flMinGain; MaxVolume = ALSource->flMaxGain; //1. Multi-channel buffers always play "normal" Channels = 0; Pitch = ALSource->flPitch; BufferListItem = ALSource->queue; while(BufferListItem != NULL) { ALbuffer *ALBuffer; if((ALBuffer=BufferListItem->buffer) != NULL) { Channels = aluChannelsFromFormat(ALBuffer->format); Pitch = Pitch * ALBuffer->frequency / Frequency; break; } BufferListItem = BufferListItem->next; } if(Pitch > (float)MAX_PITCH) ALSource->Params.Step = MAX_PITCH< 0.0f)) ALSource->Params.Step = 1<Params.Step = Pitch*(1<Params.Step == 0) ALSource->Params.Step = 1; } DryGain = SourceVolume; DryGain = __min(DryGain,MaxVolume); DryGain = __max(DryGain,MinVolume); DryGainHF = 1.0f; switch(ALSource->DirectFilter.type) { case AL_FILTER_LOWPASS: DryGain *= ALSource->DirectFilter.Gain; DryGainHF *= ALSource->DirectFilter.GainHF; break; } if(Channels == 2) { for(i = 0;i < OUTPUTCHANNELS;i++) ALSource->Params.DryGains[i] = 0.0f; if(DupStereo == AL_FALSE) { ALSource->Params.DryGains[FRONT_LEFT] = DryGain * ListenerGain; ALSource->Params.DryGains[FRONT_RIGHT] = DryGain * ListenerGain; } else { switch(Format) { case AL_FORMAT_MONO8: case AL_FORMAT_MONO16: case AL_FORMAT_MONO_FLOAT32: case AL_FORMAT_STEREO8: case AL_FORMAT_STEREO16: case AL_FORMAT_STEREO_FLOAT32: ALSource->Params.DryGains[FRONT_LEFT] = DryGain * ListenerGain; ALSource->Params.DryGains[FRONT_RIGHT] = DryGain * ListenerGain; break; case AL_FORMAT_QUAD8: case AL_FORMAT_QUAD16: case AL_FORMAT_QUAD32: case AL_FORMAT_51CHN8: case AL_FORMAT_51CHN16: case AL_FORMAT_51CHN32: DryGain *= aluSqrt(2.0f/4.0f); ALSource->Params.DryGains[FRONT_LEFT] = DryGain * ListenerGain; ALSource->Params.DryGains[FRONT_RIGHT] = DryGain * ListenerGain; ALSource->Params.DryGains[BACK_LEFT] = DryGain * ListenerGain; ALSource->Params.DryGains[BACK_RIGHT] = DryGain * ListenerGain; break; case AL_FORMAT_61CHN8: case AL_FORMAT_61CHN16: case AL_FORMAT_61CHN32: DryGain *= aluSqrt(2.0f/4.0f); ALSource->Params.DryGains[FRONT_LEFT] = DryGain * ListenerGain; ALSource->Params.DryGains[FRONT_RIGHT] = DryGain * ListenerGain; ALSource->Params.DryGains[SIDE_LEFT] = DryGain * ListenerGain; ALSource->Params.DryGains[SIDE_RIGHT] = DryGain * ListenerGain; break; case AL_FORMAT_71CHN8: case AL_FORMAT_71CHN16: case AL_FORMAT_71CHN32: DryGain *= aluSqrt(2.0f/6.0f); ALSource->Params.DryGains[FRONT_LEFT] = DryGain * ListenerGain; ALSource->Params.DryGains[FRONT_RIGHT] = DryGain * ListenerGain; ALSource->Params.DryGains[BACK_LEFT] = DryGain * ListenerGain; ALSource->Params.DryGains[BACK_RIGHT] = DryGain * ListenerGain; ALSource->Params.DryGains[SIDE_LEFT] = DryGain * ListenerGain; ALSource->Params.DryGains[SIDE_RIGHT] = DryGain * ListenerGain; break; default: break; } } } else { for(i = 0;i < OUTPUTCHANNELS;i++) ALSource->Params.DryGains[i] = DryGain * ListenerGain; } for(i = 0;i < NumSends;i++) { WetGain[i] = SourceVolume; WetGain[i] = __min(WetGain[i],MaxVolume); WetGain[i] = __max(WetGain[i],MinVolume); WetGainHF[i] = 1.0f; switch(ALSource->Send[i].WetFilter.type) { case AL_FILTER_LOWPASS: WetGain[i] *= ALSource->Send[i].WetFilter.Gain; WetGainHF[i] *= ALSource->Send[i].WetFilter.GainHF; break; } ALSource->Params.WetGains[i] = WetGain[i] * ListenerGain; } for(i = NumSends;i < MAX_SENDS;i++) { ALSource->Params.WetGains[i] = 0.0f; WetGainHF[i] = 1.0f; } /* Update filter coefficients. Calculations based on the I3DL2 * spec. */ cw = cos(2.0*M_PI * LOWPASSFREQCUTOFF / 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.iirFilter.coeff = lpCoeffCalc(DryGainHF, cw); for(i = 0;i < NumSends;i++) { /* We use a one-pole filter, so we need to take the squared gain */ ALfloat a = lpCoeffCalc(WetGainHF[i]*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,DryMix,OrigDist; ALfloat Direction[3],Position[3],SourceToListener[3]; ALfloat Velocity[3],ListenerVel[3]; ALfloat MinVolume,MaxVolume,MinDist,MaxDist,Rolloff,OuterGainHF; ALfloat ConeVolume,ConeHF,SourceVolume,ListenerGain; ALfloat DopplerFactor, DopplerVelocity, flSpeedOfSound; ALbufferlistitem *BufferListItem; ALfloat Matrix[4][4]; ALfloat flAttenuation, effectiveDist; ALfloat RoomAttenuation[MAX_SENDS]; ALfloat MetersPerUnit; ALfloat RoomRolloff[MAX_SENDS]; ALfloat DryGainHF = 1.0f; ALfloat WetGain[MAX_SENDS]; ALfloat WetGainHF[MAX_SENDS]; ALfloat DirGain, AmbientGain; const ALfloat *SpeakerGain; ALfloat Pitch; ALfloat length; ALuint Frequency; ALint NumSends; ALint pos, s, i; ALfloat cw; for(i = 0;i < MAX_SENDS;i++) WetGainHF[i] = 1.0f; //Get context properties DopplerFactor = ALContext->DopplerFactor * ALSource->DopplerFactor; DopplerVelocity = ALContext->DopplerVelocity; flSpeedOfSound = ALContext->flSpeedOfSound; NumSends = Device->NumAuxSends; Frequency = Device->Frequency; //Get listener properties ListenerGain = ALContext->Listener.Gain; MetersPerUnit = ALContext->Listener.MetersPerUnit; memcpy(ListenerVel, ALContext->Listener.Velocity, sizeof(ALContext->Listener.Velocity)); //Get source properties SourceVolume = ALSource->flGain; memcpy(Position, ALSource->vPosition, sizeof(ALSource->vPosition)); memcpy(Direction, ALSource->vOrientation, sizeof(ALSource->vOrientation)); memcpy(Velocity, ALSource->vVelocity, sizeof(ALSource->vVelocity)); MinVolume = ALSource->flMinGain; MaxVolume = ALSource->flMaxGain; MinDist = ALSource->flRefDistance; MaxDist = ALSource->flMaxDistance; Rolloff = ALSource->flRollOffFactor; InnerAngle = ALSource->flInnerAngle; OuterAngle = ALSource->flOuterAngle; OuterGainHF = ALSource->OuterGainHF; //1. Translate Listener to origin (convert to head relative) if(ALSource->bHeadRelative==AL_FALSE) { ALfloat U[3],V[3],N[3]; // Build transform matrix memcpy(N, ALContext->Listener.Forward, sizeof(N)); // At-vector aluNormalize(N); // Normalized At-vector memcpy(V, ALContext->Listener.Up, sizeof(V)); // Up-vector aluNormalize(V); // Normalized Up-vector aluCrossproduct(N, V, U); // Right-vector aluNormalize(U); // Normalized Right-vector Matrix[0][0] = U[0]; Matrix[0][1] = V[0]; Matrix[0][2] = -N[0]; Matrix[0][3] = 0.0f; Matrix[1][0] = U[1]; Matrix[1][1] = V[1]; Matrix[1][2] = -N[1]; Matrix[1][3] = 0.0f; Matrix[2][0] = U[2]; Matrix[2][1] = V[2]; Matrix[2][2] = -N[2]; Matrix[2][3] = 0.0f; Matrix[3][0] = 0.0f; Matrix[3][1] = 0.0f; Matrix[3][2] = 0.0f; Matrix[3][3] = 1.0f; // Translate position Position[0] -= ALContext->Listener.Position[0]; Position[1] -= ALContext->Listener.Position[1]; Position[2] -= ALContext->Listener.Position[2]; // Transform source position and direction into listener space aluMatrixVector(Position, 1.0f, Matrix); aluMatrixVector(Direction, 0.0f, Matrix); // Transform source and listener velocity into listener space aluMatrixVector(Velocity, 0.0f, Matrix); aluMatrixVector(ListenerVel, 0.0f, Matrix); } else ListenerVel[0] = ListenerVel[1] = ListenerVel[2] = 0.0f; SourceToListener[0] = -Position[0]; SourceToListener[1] = -Position[1]; SourceToListener[2] = -Position[2]; aluNormalize(SourceToListener); aluNormalize(Direction); //2. Calculate distance attenuation Distance = aluSqrt(aluDotproduct(Position, Position)); OrigDist = Distance; flAttenuation = 1.0f; for(i = 0;i < NumSends;i++) { RoomAttenuation[i] = 1.0f; RoomRolloff[i] = ALSource->RoomRolloffFactor; if(ALSource->Send[i].Slot && (ALSource->Send[i].Slot->effect.type == AL_EFFECT_REVERB || ALSource->Send[i].Slot->effect.type == AL_EFFECT_EAXREVERB)) RoomRolloff[i] += ALSource->Send[i].Slot->effect.Reverb.RoomRolloffFactor; } switch(ALContext->SourceDistanceModel ? ALSource->DistanceModel : ALContext->DistanceModel) { case AL_INVERSE_DISTANCE_CLAMPED: Distance=__max(Distance,MinDist); Distance=__min(Distance,MaxDist); if(MaxDist < MinDist) break; //fall-through case AL_INVERSE_DISTANCE: if(MinDist > 0.0f) { if((MinDist + (Rolloff * (Distance - MinDist))) > 0.0f) flAttenuation = MinDist / (MinDist + (Rolloff * (Distance - MinDist))); for(i = 0;i < NumSends;i++) { if((MinDist + (RoomRolloff[i] * (Distance - MinDist))) > 0.0f) RoomAttenuation[i] = MinDist / (MinDist + (RoomRolloff[i] * (Distance - MinDist))); } } break; case AL_LINEAR_DISTANCE_CLAMPED: Distance=__max(Distance,MinDist); Distance=__min(Distance,MaxDist); if(MaxDist < MinDist) break; //fall-through case AL_LINEAR_DISTANCE: Distance=__min(Distance,MaxDist); if(MaxDist != MinDist) { flAttenuation = 1.0f - (Rolloff*(Distance-MinDist)/(MaxDist - MinDist)); for(i = 0;i < NumSends;i++) RoomAttenuation[i] = 1.0f - (RoomRolloff[i]*(Distance-MinDist)/(MaxDist - MinDist)); } break; case AL_EXPONENT_DISTANCE_CLAMPED: Distance=__max(Distance,MinDist); Distance=__min(Distance,MaxDist); if(MaxDist < MinDist) break; //fall-through case AL_EXPONENT_DISTANCE: if(Distance > 0.0f && MinDist > 0.0f) { flAttenuation = aluPow(Distance/MinDist, -Rolloff); for(i = 0;i < NumSends;i++) RoomAttenuation[i] = aluPow(Distance/MinDist, -RoomRolloff[i]); } break; case AL_NONE: break; } // Source Gain + Attenuation DryMix = SourceVolume * flAttenuation; for(i = 0;i < NumSends;i++) WetGain[i] = SourceVolume * RoomAttenuation[i]; effectiveDist = 0.0f; if(MinDist > 0.0f && flAttenuation < 1.0f) effectiveDist = (MinDist/flAttenuation - MinDist)*MetersPerUnit; // Distance-based air absorption if(ALSource->AirAbsorptionFactor > 0.0f && effectiveDist > 0.0f) { ALfloat absorb; // Absorption calculation is done in dB absorb = (ALSource->AirAbsorptionFactor*AIRABSORBGAINDBHF) * effectiveDist; // Convert dB to linear gain before applying absorb = aluPow(10.0f, absorb/20.0f); DryGainHF *= absorb; } //3. Apply directional soundcones Angle = aluAcos(aluDotproduct(Direction,SourceToListener)) * 180.0f/M_PI; if(Angle >= InnerAngle && Angle <= OuterAngle) { ALfloat scale = (Angle-InnerAngle) / (OuterAngle-InnerAngle); ConeVolume = (1.0f+(ALSource->flOuterGain-1.0f)*scale); ConeHF = (1.0f+(OuterGainHF-1.0f)*scale); } else if(Angle > OuterAngle) { ConeVolume = (1.0f+(ALSource->flOuterGain-1.0f)); ConeHF = (1.0f+(OuterGainHF-1.0f)); } else { ConeVolume = 1.0f; ConeHF = 1.0f; } // Apply some high-frequency attenuation for sources behind the listener // NOTE: This should be aluDotproduct({0,0,-1}, ListenerToSource), however // that is equivalent to aluDotproduct({0,0,1}, SourceToListener), which is // the same as SourceToListener[2] Angle = aluAcos(SourceToListener[2]) * 180.0f/M_PI; // Sources within the minimum distance attenuate less if(OrigDist < MinDist) Angle *= OrigDist/MinDist; if(Angle > 90.0f) { ALfloat scale = (Angle-90.0f) / (180.1f-90.0f); // .1 to account for fp errors ConeHF *= 1.0f - (Device->HeadDampen*scale); } DryMix *= ConeVolume; if(ALSource->DryGainHFAuto) DryGainHF *= ConeHF; // Clamp to Min/Max Gain DryMix = __min(DryMix,MaxVolume); DryMix = __max(DryMix,MinVolume); for(i = 0;i < NumSends;i++) { ALeffectslot *Slot = ALSource->Send[i].Slot; if(!Slot || Slot->effect.type == AL_EFFECT_NULL) { ALSource->Params.WetGains[i] = 0.0f; WetGainHF[i] = 1.0f; continue; } if(Slot->AuxSendAuto) { if(ALSource->WetGainAuto) WetGain[i] *= ConeVolume; if(ALSource->WetGainHFAuto) WetGainHF[i] *= ConeHF; // Clamp to Min/Max Gain WetGain[i] = __min(WetGain[i],MaxVolume); WetGain[i] = __max(WetGain[i],MinVolume); if(Slot->effect.type == AL_EFFECT_REVERB || Slot->effect.type == AL_EFFECT_EAXREVERB) { /* Apply a decay-time transformation to the wet path, based on * the attenuation of the dry path. * * Using the approximate (effective) source to listener * distance, the initial decay of the reverb effect is * calculated and applied to the wet path. */ WetGain[i] *= aluPow(10.0f, effectiveDist / (SPEEDOFSOUNDMETRESPERSEC * Slot->effect.Reverb.DecayTime) * -60.0 / 20.0); WetGainHF[i] *= aluPow(10.0f, log10(Slot->effect.Reverb.AirAbsorptionGainHF) * ALSource->AirAbsorptionFactor * effectiveDist); } } 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 */ WetGain[i] = DryMix; WetGainHF[i] = DryGainHF; } switch(ALSource->Send[i].WetFilter.type) { case AL_FILTER_LOWPASS: WetGain[i] *= ALSource->Send[i].WetFilter.Gain; WetGainHF[i] *= ALSource->Send[i].WetFilter.GainHF; break; } ALSource->Params.WetGains[i] = WetGain[i] * ListenerGain; } for(i = NumSends;i < MAX_SENDS;i++) { ALSource->Params.WetGains[i] = 0.0f; WetGainHF[i] = 1.0f; } // Apply filter gains and filters switch(ALSource->DirectFilter.type) { case AL_FILTER_LOWPASS: DryMix *= ALSource->DirectFilter.Gain; DryGainHF *= ALSource->DirectFilter.GainHF; break; } DryMix *= ListenerGain; // Calculate Velocity if(DopplerFactor != 0.0f) { ALfloat flVSS, flVLS; ALfloat flMaxVelocity = (DopplerVelocity * flSpeedOfSound) / DopplerFactor; flVSS = aluDotproduct(Velocity, SourceToListener); if(flVSS >= flMaxVelocity) flVSS = (flMaxVelocity - 1.0f); else if(flVSS <= -flMaxVelocity) flVSS = -flMaxVelocity + 1.0f; flVLS = aluDotproduct(ListenerVel, SourceToListener); if(flVLS >= flMaxVelocity) flVLS = (flMaxVelocity - 1.0f); else if(flVLS <= -flMaxVelocity) flVLS = -flMaxVelocity + 1.0f; Pitch = ALSource->flPitch * ((flSpeedOfSound * DopplerVelocity) - (DopplerFactor * flVLS)) / ((flSpeedOfSound * DopplerVelocity) - (DopplerFactor * flVSS)); } else Pitch = ALSource->flPitch; BufferListItem = ALSource->queue; while(BufferListItem != NULL) { ALbuffer *ALBuffer; if((ALBuffer=BufferListItem->buffer) != NULL) { Pitch = Pitch * ALBuffer->frequency / Frequency; break; } BufferListItem = BufferListItem->next; } if(Pitch > (float)MAX_PITCH) ALSource->Params.Step = MAX_PITCH< 0.0f)) ALSource->Params.Step = 1<Params.Step = Pitch*(1<Params.Step == 0) ALSource->Params.Step = 1; } // Use energy-preserving panning algorithm for multi-speaker playback length = __max(OrigDist, MinDist); if(length > 0.0f) { ALfloat invlen = 1.0f/length; Position[0] *= invlen; Position[1] *= invlen; Position[2] *= invlen; } pos = aluCart2LUTpos(-Position[2], Position[0]); SpeakerGain = &Device->PanningLUT[OUTPUTCHANNELS * pos]; DirGain = aluSqrt(Position[0]*Position[0] + Position[2]*Position[2]); // elevation adjustment for directional gain. this sucks, but // has low complexity AmbientGain = 1.0/aluSqrt(Device->NumChan) * (1.0-DirGain); for(s = 0;s < OUTPUTCHANNELS;s++) ALSource->Params.DryGains[s] = 0.0f; for(s = 0;s < (ALsizei)Device->NumChan;s++) { Channel chan = Device->Speaker2Chan[s]; ALfloat gain = SpeakerGain[chan]*DirGain + AmbientGain; ALSource->Params.DryGains[chan] = DryMix * gain; } /* Update filter coefficients. */ cw = cos(2.0*M_PI * LOWPASSFREQCUTOFF / Frequency); /* Spatialized sources use four chained one-pole filters, so we need to * take the fourth root of the squared gain, which is the same as the * square root of the base gain. */ ALSource->Params.iirFilter.coeff = lpCoeffCalc(aluSqrt(DryGainHF), cw); for(i = 0;i < NumSends;i++) { /* The wet path uses two chained one-pole filters, so take the * base gain (square root of the squared gain) */ ALSource->Params.Send[i].iirFilter.coeff = lpCoeffCalc(WetGainHF[i], cw); } } ALvoid aluHandleDisconnect(ALCdevice *device) { ALuint i; SuspendContext(NULL); for(i = 0;i < device->NumContexts;i++) { ALCcontext *Context = device->Contexts[i]; ALsource *source; ALsizei pos; SuspendContext(Context); for(pos = 0;pos < Context->SourceMap.size;pos++) { source = Context->SourceMap.array[pos].value; if(source->state == AL_PLAYING) { source->state = AL_STOPPED; source->BuffersPlayed = source->BuffersInQueue; source->position = 0; source->position_fraction = 0; } } ProcessContext(Context); } device->Connected = ALC_FALSE; ProcessContext(NULL); }