rlm@0: /** rlm@0: * OpenAL cross platform audio library rlm@0: * Copyright (C) 1999-2007 by authors. rlm@0: * This library is free software; you can redistribute it and/or rlm@0: * modify it under the terms of the GNU Library General Public rlm@0: * License as published by the Free Software Foundation; either rlm@0: * version 2 of the License, or (at your option) any later version. rlm@0: * rlm@0: * This library is distributed in the hope that it will be useful, rlm@0: * but WITHOUT ANY WARRANTY; without even the implied warranty of rlm@0: * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU rlm@0: * Library General Public License for more details. rlm@0: * rlm@0: * You should have received a copy of the GNU Library General Public rlm@0: * License along with this library; if not, write to the rlm@0: * Free Software Foundation, Inc., 59 Temple Place - Suite 330, rlm@0: * Boston, MA 02111-1307, USA. rlm@0: * Or go to http://www.gnu.org/copyleft/lgpl.html rlm@0: */ rlm@0: rlm@0: #include "config.h" rlm@0: rlm@0: #include rlm@0: #include rlm@0: #include rlm@0: #include rlm@0: #include rlm@0: rlm@0: #include "alMain.h" rlm@0: #include "AL/al.h" rlm@0: #include "AL/alc.h" rlm@0: #include "alSource.h" rlm@0: #include "alBuffer.h" rlm@0: #include "alListener.h" rlm@0: #include "alAuxEffectSlot.h" rlm@0: #include "alu.h" rlm@0: #include "bs2b.h" rlm@0: rlm@0: rlm@0: static __inline ALvoid aluCrossproduct(const ALfloat *inVector1, const ALfloat *inVector2, ALfloat *outVector) rlm@0: { rlm@0: outVector[0] = inVector1[1]*inVector2[2] - inVector1[2]*inVector2[1]; rlm@0: outVector[1] = inVector1[2]*inVector2[0] - inVector1[0]*inVector2[2]; rlm@0: outVector[2] = inVector1[0]*inVector2[1] - inVector1[1]*inVector2[0]; rlm@0: } rlm@0: rlm@0: static __inline ALfloat aluDotproduct(const ALfloat *inVector1, const ALfloat *inVector2) rlm@0: { rlm@0: return inVector1[0]*inVector2[0] + inVector1[1]*inVector2[1] + rlm@0: inVector1[2]*inVector2[2]; rlm@0: } rlm@0: rlm@0: static __inline ALvoid aluNormalize(ALfloat *inVector) rlm@0: { rlm@0: ALfloat length, inverse_length; rlm@0: rlm@0: length = aluSqrt(aluDotproduct(inVector, inVector)); rlm@0: if(length != 0.0f) rlm@0: { rlm@0: inverse_length = 1.0f/length; rlm@0: inVector[0] *= inverse_length; rlm@0: inVector[1] *= inverse_length; rlm@0: inVector[2] *= inverse_length; rlm@0: } rlm@0: } rlm@0: rlm@0: static __inline ALvoid aluMatrixVector(ALfloat *vector,ALfloat w,ALfloat matrix[4][4]) rlm@0: { rlm@0: ALfloat temp[4] = { rlm@0: vector[0], vector[1], vector[2], w rlm@0: }; rlm@0: rlm@0: vector[0] = temp[0]*matrix[0][0] + temp[1]*matrix[1][0] + temp[2]*matrix[2][0] + temp[3]*matrix[3][0]; rlm@0: vector[1] = temp[0]*matrix[0][1] + temp[1]*matrix[1][1] + temp[2]*matrix[2][1] + temp[3]*matrix[3][1]; rlm@0: vector[2] = temp[0]*matrix[0][2] + temp[1]*matrix[1][2] + temp[2]*matrix[2][2] + temp[3]*matrix[3][2]; rlm@0: } rlm@0: rlm@0: rlm@0: ALvoid CalcNonAttnSourceParams(ALsource *ALSource, const ALCcontext *ALContext) rlm@0: { rlm@0: static const ALfloat angles_Mono[1] = { 0.0f }; rlm@0: static const ALfloat angles_Stereo[2] = { -30.0f, 30.0f }; rlm@0: static const ALfloat angles_Rear[2] = { -150.0f, 150.0f }; rlm@0: static const ALfloat angles_Quad[4] = { -45.0f, 45.0f, -135.0f, 135.0f }; rlm@0: static const ALfloat angles_X51[6] = { -30.0f, 30.0f, 0.0f, 0.0f, rlm@0: -110.0f, 110.0f }; rlm@0: static const ALfloat angles_X61[7] = { -30.0f, 30.0f, 0.0f, 0.0f, rlm@0: 180.0f, -90.0f, 90.0f }; rlm@0: static const ALfloat angles_X71[8] = { -30.0f, 30.0f, 0.0f, 0.0f, rlm@0: -110.0f, 110.0f, -90.0f, 90.0f }; rlm@0: rlm@0: static const enum Channel chans_Mono[1] = { FRONT_CENTER }; rlm@0: static const enum Channel chans_Stereo[2] = { FRONT_LEFT, FRONT_RIGHT }; rlm@0: static const enum Channel chans_Rear[2] = { BACK_LEFT, BACK_RIGHT }; rlm@0: static const enum Channel chans_Quad[4] = { FRONT_LEFT, FRONT_RIGHT, rlm@0: BACK_LEFT, BACK_RIGHT }; rlm@0: static const enum Channel chans_X51[6] = { FRONT_LEFT, FRONT_RIGHT, rlm@0: FRONT_CENTER, LFE, rlm@0: BACK_LEFT, BACK_RIGHT }; rlm@0: static const enum Channel chans_X61[7] = { FRONT_LEFT, FRONT_RIGHT, rlm@0: FRONT_CENTER, LFE, BACK_CENTER, rlm@0: SIDE_LEFT, SIDE_RIGHT }; rlm@0: static const enum Channel chans_X71[8] = { FRONT_LEFT, FRONT_RIGHT, rlm@0: FRONT_CENTER, LFE, rlm@0: BACK_LEFT, BACK_RIGHT, rlm@0: SIDE_LEFT, SIDE_RIGHT }; rlm@0: rlm@0: ALCdevice *Device = ALContext->Device; rlm@0: ALfloat SourceVolume,ListenerGain,MinVolume,MaxVolume; rlm@0: ALbufferlistitem *BufferListItem; rlm@0: enum DevFmtChannels DevChans; rlm@0: enum FmtChannels Channels; rlm@0: ALfloat (*SrcMatrix)[MAXCHANNELS]; rlm@0: ALfloat DryGain, DryGainHF; rlm@0: ALfloat WetGain[MAX_SENDS]; rlm@0: ALfloat WetGainHF[MAX_SENDS]; rlm@0: ALint NumSends, Frequency; rlm@0: const ALfloat *SpeakerGain; rlm@0: const ALfloat *angles = NULL; rlm@0: const enum Channel *chans = NULL; rlm@0: enum Resampler Resampler; rlm@0: ALint num_channels = 0; rlm@0: ALboolean VirtualChannels; rlm@0: ALfloat Pitch; rlm@0: ALfloat cw; rlm@0: ALuint pos; rlm@0: ALint i, c; rlm@0: rlm@0: /* Get device properties */ rlm@0: DevChans = ALContext->Device->FmtChans; rlm@0: NumSends = ALContext->Device->NumAuxSends; rlm@0: Frequency = ALContext->Device->Frequency; rlm@0: rlm@0: /* Get listener properties */ rlm@0: ListenerGain = ALContext->Listener.Gain; rlm@0: rlm@0: /* Get source properties */ rlm@0: SourceVolume = ALSource->flGain; rlm@0: MinVolume = ALSource->flMinGain; rlm@0: MaxVolume = ALSource->flMaxGain; rlm@0: Pitch = ALSource->flPitch; rlm@0: Resampler = ALSource->Resampler; rlm@0: VirtualChannels = ALSource->VirtualChannels; rlm@0: rlm@0: /* Calculate the stepping value */ rlm@0: Channels = FmtMono; rlm@0: BufferListItem = ALSource->queue; rlm@0: while(BufferListItem != NULL) rlm@0: { rlm@0: ALbuffer *ALBuffer; rlm@0: if((ALBuffer=BufferListItem->buffer) != NULL) rlm@0: { rlm@0: ALint maxstep = STACK_DATA_SIZE / ALSource->NumChannels / rlm@0: ALSource->SampleSize; rlm@0: maxstep -= ResamplerPadding[Resampler] + rlm@0: ResamplerPrePadding[Resampler] + 1; rlm@0: maxstep = mini(maxstep, INT_MAX>>FRACTIONBITS); rlm@0: rlm@0: Pitch = Pitch * ALBuffer->Frequency / Frequency; rlm@0: if(Pitch > (ALfloat)maxstep) rlm@0: ALSource->Params.Step = maxstep<Params.Step = Pitch*FRACTIONONE; rlm@0: if(ALSource->Params.Step == 0) rlm@0: ALSource->Params.Step = 1; rlm@0: } rlm@0: rlm@0: Channels = ALBuffer->FmtChannels; rlm@0: rlm@0: if(ALSource->VirtualChannels && (Device->Flags&DEVICE_USE_HRTF)) rlm@0: ALSource->Params.DoMix = SelectHrtfMixer(ALBuffer, rlm@0: (ALSource->Params.Step==FRACTIONONE) ? POINT_RESAMPLER : rlm@0: Resampler); rlm@0: else rlm@0: ALSource->Params.DoMix = SelectMixer(ALBuffer, rlm@0: (ALSource->Params.Step==FRACTIONONE) ? POINT_RESAMPLER : rlm@0: Resampler); rlm@0: break; rlm@0: } rlm@0: BufferListItem = BufferListItem->next; rlm@0: } rlm@0: rlm@0: /* Calculate gains */ rlm@0: DryGain = clampf(SourceVolume, MinVolume, MaxVolume); rlm@0: DryGainHF = 1.0f; rlm@0: switch(ALSource->DirectFilter.type) rlm@0: { rlm@0: case AL_FILTER_LOWPASS: rlm@0: DryGain *= ALSource->DirectFilter.Gain; rlm@0: DryGainHF *= ALSource->DirectFilter.GainHF; rlm@0: break; rlm@0: } rlm@0: for(i = 0;i < NumSends;i++) rlm@0: { rlm@0: WetGain[i] = clampf(SourceVolume, MinVolume, MaxVolume); rlm@0: WetGainHF[i] = 1.0f; rlm@0: switch(ALSource->Send[i].WetFilter.type) rlm@0: { rlm@0: case AL_FILTER_LOWPASS: rlm@0: WetGain[i] *= ALSource->Send[i].WetFilter.Gain; rlm@0: WetGainHF[i] *= ALSource->Send[i].WetFilter.GainHF; rlm@0: break; rlm@0: } rlm@0: } rlm@0: rlm@0: SrcMatrix = ALSource->Params.DryGains; rlm@0: for(i = 0;i < MAXCHANNELS;i++) rlm@0: { rlm@0: for(c = 0;c < MAXCHANNELS;c++) rlm@0: SrcMatrix[i][c] = 0.0f; rlm@0: } rlm@0: switch(Channels) rlm@0: { rlm@0: case FmtMono: rlm@0: angles = angles_Mono; rlm@0: chans = chans_Mono; rlm@0: num_channels = 1; rlm@0: break; rlm@0: case FmtStereo: rlm@0: if(VirtualChannels && (ALContext->Device->Flags&DEVICE_DUPLICATE_STEREO)) rlm@0: { rlm@0: DryGain *= aluSqrt(2.0f/4.0f); rlm@0: for(c = 0;c < 2;c++) rlm@0: { rlm@0: pos = aluCart2LUTpos(cos(angles_Rear[c] * (M_PI/180.0)), rlm@0: sin(angles_Rear[c] * (M_PI/180.0))); rlm@0: SpeakerGain = Device->PanningLUT[pos]; rlm@0: rlm@0: for(i = 0;i < (ALint)Device->NumChan;i++) rlm@0: { rlm@0: enum Channel chan = Device->Speaker2Chan[i]; rlm@0: SrcMatrix[c][chan] += DryGain * ListenerGain * rlm@0: SpeakerGain[chan]; rlm@0: } rlm@0: } rlm@0: } rlm@0: angles = angles_Stereo; rlm@0: chans = chans_Stereo; rlm@0: num_channels = 2; rlm@0: break; rlm@0: rlm@0: case FmtRear: rlm@0: angles = angles_Rear; rlm@0: chans = chans_Rear; rlm@0: num_channels = 2; rlm@0: break; rlm@0: rlm@0: case FmtQuad: rlm@0: angles = angles_Quad; rlm@0: chans = chans_Quad; rlm@0: num_channels = 4; rlm@0: break; rlm@0: rlm@0: case FmtX51: rlm@0: angles = angles_X51; rlm@0: chans = chans_X51; rlm@0: num_channels = 6; rlm@0: break; rlm@0: rlm@0: case FmtX61: rlm@0: angles = angles_X61; rlm@0: chans = chans_X61; rlm@0: num_channels = 7; rlm@0: break; rlm@0: rlm@0: case FmtX71: rlm@0: angles = angles_X71; rlm@0: chans = chans_X71; rlm@0: num_channels = 8; rlm@0: break; rlm@0: } rlm@0: rlm@0: if(VirtualChannels == AL_FALSE) rlm@0: { rlm@0: for(c = 0;c < num_channels;c++) rlm@0: SrcMatrix[c][chans[c]] += DryGain * ListenerGain; rlm@0: } rlm@0: else if((Device->Flags&DEVICE_USE_HRTF)) rlm@0: { rlm@0: for(c = 0;c < num_channels;c++) rlm@0: { rlm@0: if(chans[c] == LFE) rlm@0: { rlm@0: /* Skip LFE */ rlm@0: ALSource->Params.HrtfDelay[c][0] = 0; rlm@0: ALSource->Params.HrtfDelay[c][1] = 0; rlm@0: for(i = 0;i < HRIR_LENGTH;i++) rlm@0: { rlm@0: ALSource->Params.HrtfCoeffs[c][i][0] = 0.0f; rlm@0: ALSource->Params.HrtfCoeffs[c][i][1] = 0.0f; rlm@0: } rlm@0: } rlm@0: else rlm@0: { rlm@0: /* Get the static HRIR coefficients and delays for this rlm@0: * channel. */ rlm@0: GetLerpedHrtfCoeffs(0.0, angles[c] * (M_PI/180.0), rlm@0: DryGain*ListenerGain, rlm@0: ALSource->Params.HrtfCoeffs[c], rlm@0: ALSource->Params.HrtfDelay[c]); rlm@0: } rlm@0: ALSource->HrtfCounter = 0; rlm@0: } rlm@0: } rlm@0: else rlm@0: { rlm@0: for(c = 0;c < num_channels;c++) rlm@0: { rlm@0: if(chans[c] == LFE) /* Special-case LFE */ rlm@0: { rlm@0: SrcMatrix[c][LFE] += DryGain * ListenerGain; rlm@0: continue; rlm@0: } rlm@0: pos = aluCart2LUTpos(cos(angles[c] * (M_PI/180.0)), rlm@0: sin(angles[c] * (M_PI/180.0))); rlm@0: SpeakerGain = Device->PanningLUT[pos]; rlm@0: rlm@0: for(i = 0;i < (ALint)Device->NumChan;i++) rlm@0: { rlm@0: enum Channel chan = Device->Speaker2Chan[i]; rlm@0: SrcMatrix[c][chan] += DryGain * ListenerGain * rlm@0: SpeakerGain[chan]; rlm@0: } rlm@0: } rlm@0: } rlm@0: for(i = 0;i < NumSends;i++) rlm@0: { rlm@0: ALSource->Params.Send[i].Slot = ALSource->Send[i].Slot; rlm@0: ALSource->Params.Send[i].WetGain = WetGain[i] * ListenerGain; rlm@0: } rlm@0: rlm@0: /* Update filter coefficients. Calculations based on the I3DL2 rlm@0: * spec. */ rlm@0: cw = cos(2.0*M_PI * LOWPASSFREQCUTOFF / Frequency); rlm@0: rlm@0: /* We use two chained one-pole filters, so we need to take the rlm@0: * square root of the squared gain, which is the same as the base rlm@0: * gain. */ rlm@0: ALSource->Params.iirFilter.coeff = lpCoeffCalc(DryGainHF, cw); rlm@0: for(i = 0;i < NumSends;i++) rlm@0: { rlm@0: /* We use a one-pole filter, so we need to take the squared gain */ rlm@0: ALfloat a = lpCoeffCalc(WetGainHF[i]*WetGainHF[i], cw); rlm@0: ALSource->Params.Send[i].iirFilter.coeff = a; rlm@0: } rlm@0: } rlm@0: rlm@0: ALvoid CalcSourceParams(ALsource *ALSource, const ALCcontext *ALContext) rlm@0: { rlm@0: const ALCdevice *Device = ALContext->Device; rlm@0: ALfloat InnerAngle,OuterAngle,Angle,Distance,ClampedDist; rlm@0: ALfloat Direction[3],Position[3],SourceToListener[3]; rlm@0: ALfloat Velocity[3],ListenerVel[3]; rlm@0: ALfloat MinVolume,MaxVolume,MinDist,MaxDist,Rolloff; rlm@0: ALfloat ConeVolume,ConeHF,SourceVolume,ListenerGain; rlm@0: ALfloat DopplerFactor, DopplerVelocity, SpeedOfSound; rlm@0: ALfloat AirAbsorptionFactor; rlm@0: ALfloat RoomAirAbsorption[MAX_SENDS]; rlm@0: ALbufferlistitem *BufferListItem; rlm@0: ALfloat Attenuation, EffectiveDist; rlm@0: ALfloat RoomAttenuation[MAX_SENDS]; rlm@0: ALfloat MetersPerUnit; rlm@0: ALfloat RoomRolloffBase; rlm@0: ALfloat RoomRolloff[MAX_SENDS]; rlm@0: ALfloat DecayDistance[MAX_SENDS]; rlm@0: ALfloat DryGain; rlm@0: ALfloat DryGainHF; rlm@0: ALboolean DryGainHFAuto; rlm@0: ALfloat WetGain[MAX_SENDS]; rlm@0: ALfloat WetGainHF[MAX_SENDS]; rlm@0: ALboolean WetGainAuto; rlm@0: ALboolean WetGainHFAuto; rlm@0: enum Resampler Resampler; rlm@0: ALfloat Pitch; rlm@0: ALuint Frequency; rlm@0: ALint NumSends; rlm@0: ALfloat cw; rlm@0: ALint i; rlm@0: rlm@0: DryGainHF = 1.0f; rlm@0: for(i = 0;i < MAX_SENDS;i++) rlm@0: WetGainHF[i] = 1.0f; rlm@0: rlm@0: //Get context properties rlm@0: DopplerFactor = ALContext->DopplerFactor * ALSource->DopplerFactor; rlm@0: DopplerVelocity = ALContext->DopplerVelocity; rlm@0: SpeedOfSound = ALContext->flSpeedOfSound; rlm@0: NumSends = Device->NumAuxSends; rlm@0: Frequency = Device->Frequency; rlm@0: rlm@0: //Get listener properties rlm@0: ListenerGain = ALContext->Listener.Gain; rlm@0: MetersPerUnit = ALContext->Listener.MetersPerUnit; rlm@0: memcpy(ListenerVel, ALContext->Listener.Velocity, sizeof(ALContext->Listener.Velocity)); rlm@0: rlm@0: //Get source properties rlm@0: SourceVolume = ALSource->flGain; rlm@0: MinVolume = ALSource->flMinGain; rlm@0: MaxVolume = ALSource->flMaxGain; rlm@0: Pitch = ALSource->flPitch; rlm@0: Resampler = ALSource->Resampler; rlm@0: memcpy(Position, ALSource->vPosition, sizeof(ALSource->vPosition)); rlm@0: memcpy(Direction, ALSource->vOrientation, sizeof(ALSource->vOrientation)); rlm@0: memcpy(Velocity, ALSource->vVelocity, sizeof(ALSource->vVelocity)); rlm@0: MinDist = ALSource->flRefDistance; rlm@0: MaxDist = ALSource->flMaxDistance; rlm@0: Rolloff = ALSource->flRollOffFactor; rlm@0: InnerAngle = ALSource->flInnerAngle * ConeScale; rlm@0: OuterAngle = ALSource->flOuterAngle * ConeScale; rlm@0: AirAbsorptionFactor = ALSource->AirAbsorptionFactor; rlm@0: DryGainHFAuto = ALSource->DryGainHFAuto; rlm@0: WetGainAuto = ALSource->WetGainAuto; rlm@0: WetGainHFAuto = ALSource->WetGainHFAuto; rlm@0: RoomRolloffBase = ALSource->RoomRolloffFactor; rlm@0: for(i = 0;i < NumSends;i++) rlm@0: { rlm@0: ALeffectslot *Slot = ALSource->Send[i].Slot; rlm@0: rlm@0: if(!Slot || Slot->effect.type == AL_EFFECT_NULL) rlm@0: { rlm@0: RoomRolloff[i] = 0.0f; rlm@0: DecayDistance[i] = 0.0f; rlm@0: RoomAirAbsorption[i] = 1.0f; rlm@0: } rlm@0: else if(Slot->AuxSendAuto) rlm@0: { rlm@0: RoomRolloff[i] = RoomRolloffBase; rlm@0: if(IsReverbEffect(Slot->effect.type)) rlm@0: { rlm@0: RoomRolloff[i] += Slot->effect.Params.Reverb.RoomRolloffFactor; rlm@0: DecayDistance[i] = Slot->effect.Params.Reverb.DecayTime * rlm@0: SPEEDOFSOUNDMETRESPERSEC; rlm@0: RoomAirAbsorption[i] = Slot->effect.Params.Reverb.AirAbsorptionGainHF; rlm@0: } rlm@0: else rlm@0: { rlm@0: DecayDistance[i] = 0.0f; rlm@0: RoomAirAbsorption[i] = 1.0f; rlm@0: } rlm@0: } rlm@0: else rlm@0: { rlm@0: /* If the slot's auxiliary send auto is off, the data sent to the rlm@0: * effect slot is the same as the dry path, sans filter effects */ rlm@0: RoomRolloff[i] = Rolloff; rlm@0: DecayDistance[i] = 0.0f; rlm@0: RoomAirAbsorption[i] = AIRABSORBGAINHF; rlm@0: } rlm@0: rlm@0: ALSource->Params.Send[i].Slot = Slot; rlm@0: } rlm@0: rlm@0: //1. Translate Listener to origin (convert to head relative) rlm@0: if(ALSource->bHeadRelative == AL_FALSE) rlm@0: { rlm@0: ALfloat U[3],V[3],N[3]; rlm@0: ALfloat Matrix[4][4]; rlm@0: rlm@0: // Build transform matrix rlm@0: memcpy(N, ALContext->Listener.Forward, sizeof(N)); // At-vector rlm@0: aluNormalize(N); // Normalized At-vector rlm@0: memcpy(V, ALContext->Listener.Up, sizeof(V)); // Up-vector rlm@0: aluNormalize(V); // Normalized Up-vector rlm@0: aluCrossproduct(N, V, U); // Right-vector rlm@0: aluNormalize(U); // Normalized Right-vector rlm@0: Matrix[0][0] = U[0]; Matrix[0][1] = V[0]; Matrix[0][2] = -N[0]; Matrix[0][3] = 0.0f; rlm@0: Matrix[1][0] = U[1]; Matrix[1][1] = V[1]; Matrix[1][2] = -N[1]; Matrix[1][3] = 0.0f; rlm@0: Matrix[2][0] = U[2]; Matrix[2][1] = V[2]; Matrix[2][2] = -N[2]; Matrix[2][3] = 0.0f; rlm@0: Matrix[3][0] = 0.0f; Matrix[3][1] = 0.0f; Matrix[3][2] = 0.0f; Matrix[3][3] = 1.0f; rlm@0: rlm@0: // Translate position rlm@0: Position[0] -= ALContext->Listener.Position[0]; rlm@0: Position[1] -= ALContext->Listener.Position[1]; rlm@0: Position[2] -= ALContext->Listener.Position[2]; rlm@0: rlm@0: // Transform source position and direction into listener space rlm@0: aluMatrixVector(Position, 1.0f, Matrix); rlm@0: aluMatrixVector(Direction, 0.0f, Matrix); rlm@0: // Transform source and listener velocity into listener space rlm@0: aluMatrixVector(Velocity, 0.0f, Matrix); rlm@0: aluMatrixVector(ListenerVel, 0.0f, Matrix); rlm@0: } rlm@0: else rlm@0: ListenerVel[0] = ListenerVel[1] = ListenerVel[2] = 0.0f; rlm@0: rlm@0: SourceToListener[0] = -Position[0]; rlm@0: SourceToListener[1] = -Position[1]; rlm@0: SourceToListener[2] = -Position[2]; rlm@0: aluNormalize(SourceToListener); rlm@0: aluNormalize(Direction); rlm@0: rlm@0: //2. Calculate distance attenuation rlm@0: Distance = aluSqrt(aluDotproduct(Position, Position)); rlm@0: ClampedDist = Distance; rlm@0: rlm@0: Attenuation = 1.0f; rlm@0: for(i = 0;i < NumSends;i++) rlm@0: RoomAttenuation[i] = 1.0f; rlm@0: switch(ALContext->SourceDistanceModel ? ALSource->DistanceModel : rlm@0: ALContext->DistanceModel) rlm@0: { rlm@0: case InverseDistanceClamped: rlm@0: ClampedDist = clampf(ClampedDist, MinDist, MaxDist); rlm@0: if(MaxDist < MinDist) rlm@0: break; rlm@0: //fall-through rlm@0: case InverseDistance: rlm@0: if(MinDist > 0.0f) rlm@0: { rlm@0: if((MinDist + (Rolloff * (ClampedDist - MinDist))) > 0.0f) rlm@0: Attenuation = MinDist / (MinDist + (Rolloff * (ClampedDist - MinDist))); rlm@0: for(i = 0;i < NumSends;i++) rlm@0: { rlm@0: if((MinDist + (RoomRolloff[i] * (ClampedDist - MinDist))) > 0.0f) rlm@0: RoomAttenuation[i] = MinDist / (MinDist + (RoomRolloff[i] * (ClampedDist - MinDist))); rlm@0: } rlm@0: } rlm@0: break; rlm@0: rlm@0: case LinearDistanceClamped: rlm@0: ClampedDist = clampf(ClampedDist, MinDist, MaxDist); rlm@0: if(MaxDist < MinDist) rlm@0: break; rlm@0: //fall-through rlm@0: case LinearDistance: rlm@0: if(MaxDist != MinDist) rlm@0: { rlm@0: Attenuation = 1.0f - (Rolloff*(ClampedDist-MinDist)/(MaxDist - MinDist)); rlm@0: Attenuation = maxf(Attenuation, 0.0f); rlm@0: for(i = 0;i < NumSends;i++) rlm@0: { rlm@0: RoomAttenuation[i] = 1.0f - (RoomRolloff[i]*(ClampedDist-MinDist)/(MaxDist - MinDist)); rlm@0: RoomAttenuation[i] = maxf(RoomAttenuation[i], 0.0f); rlm@0: } rlm@0: } rlm@0: break; rlm@0: rlm@0: case ExponentDistanceClamped: rlm@0: ClampedDist = clampf(ClampedDist, MinDist, MaxDist); rlm@0: if(MaxDist < MinDist) rlm@0: break; rlm@0: //fall-through rlm@0: case ExponentDistance: rlm@0: if(ClampedDist > 0.0f && MinDist > 0.0f) rlm@0: { rlm@0: Attenuation = aluPow(ClampedDist/MinDist, -Rolloff); rlm@0: for(i = 0;i < NumSends;i++) rlm@0: RoomAttenuation[i] = aluPow(ClampedDist/MinDist, -RoomRolloff[i]); rlm@0: } rlm@0: break; rlm@0: rlm@0: case DisableDistance: rlm@0: break; rlm@0: } rlm@0: rlm@0: // Source Gain + Attenuation rlm@0: DryGain = SourceVolume * Attenuation; rlm@0: for(i = 0;i < NumSends;i++) rlm@0: WetGain[i] = SourceVolume * RoomAttenuation[i]; rlm@0: rlm@0: // Distance-based air absorption rlm@0: EffectiveDist = 0.0f; rlm@0: if(MinDist > 0.0f && Attenuation < 1.0f) rlm@0: EffectiveDist = (MinDist/Attenuation - MinDist)*MetersPerUnit; rlm@0: if(AirAbsorptionFactor > 0.0f && EffectiveDist > 0.0f) rlm@0: { rlm@0: DryGainHF *= aluPow(AIRABSORBGAINHF, AirAbsorptionFactor*EffectiveDist); rlm@0: for(i = 0;i < NumSends;i++) rlm@0: WetGainHF[i] *= aluPow(RoomAirAbsorption[i], rlm@0: AirAbsorptionFactor*EffectiveDist); rlm@0: } rlm@0: rlm@0: //3. Apply directional soundcones rlm@0: Angle = aluAcos(aluDotproduct(Direction,SourceToListener)) * (180.0/M_PI); rlm@0: if(Angle >= InnerAngle && Angle <= OuterAngle) rlm@0: { rlm@0: ALfloat scale = (Angle-InnerAngle) / (OuterAngle-InnerAngle); rlm@0: ConeVolume = lerp(1.0, ALSource->flOuterGain, scale); rlm@0: ConeHF = lerp(1.0, ALSource->OuterGainHF, scale); rlm@0: } rlm@0: else if(Angle > OuterAngle) rlm@0: { rlm@0: ConeVolume = ALSource->flOuterGain; rlm@0: ConeHF = ALSource->OuterGainHF; rlm@0: } rlm@0: else rlm@0: { rlm@0: ConeVolume = 1.0f; rlm@0: ConeHF = 1.0f; rlm@0: } rlm@0: rlm@0: DryGain *= ConeVolume; rlm@0: if(WetGainAuto) rlm@0: { rlm@0: for(i = 0;i < NumSends;i++) rlm@0: WetGain[i] *= ConeVolume; rlm@0: } rlm@0: if(DryGainHFAuto) rlm@0: DryGainHF *= ConeHF; rlm@0: if(WetGainHFAuto) rlm@0: { rlm@0: for(i = 0;i < NumSends;i++) rlm@0: WetGainHF[i] *= ConeHF; rlm@0: } rlm@0: rlm@0: // Clamp to Min/Max Gain rlm@0: DryGain = clampf(DryGain, MinVolume, MaxVolume); rlm@0: for(i = 0;i < NumSends;i++) rlm@0: WetGain[i] = clampf(WetGain[i], MinVolume, MaxVolume); rlm@0: rlm@0: // Apply filter gains and filters rlm@0: switch(ALSource->DirectFilter.type) rlm@0: { rlm@0: case AL_FILTER_LOWPASS: rlm@0: DryGain *= ALSource->DirectFilter.Gain; rlm@0: DryGainHF *= ALSource->DirectFilter.GainHF; rlm@0: break; rlm@0: } rlm@0: DryGain *= ListenerGain; rlm@0: for(i = 0;i < NumSends;i++) rlm@0: { rlm@0: switch(ALSource->Send[i].WetFilter.type) rlm@0: { rlm@0: case AL_FILTER_LOWPASS: rlm@0: WetGain[i] *= ALSource->Send[i].WetFilter.Gain; rlm@0: WetGainHF[i] *= ALSource->Send[i].WetFilter.GainHF; rlm@0: break; rlm@0: } rlm@0: WetGain[i] *= ListenerGain; rlm@0: } rlm@0: rlm@0: if(WetGainAuto) rlm@0: { rlm@0: /* Apply a decay-time transformation to the wet path, based on the rlm@0: * attenuation of the dry path. rlm@0: * rlm@0: * Using the approximate (effective) source to listener distance, the rlm@0: * initial decay of the reverb effect is calculated and applied to the rlm@0: * wet path. rlm@0: */ rlm@0: for(i = 0;i < NumSends;i++) rlm@0: { rlm@0: if(DecayDistance[i] > 0.0f) rlm@0: WetGain[i] *= aluPow(0.001f /* -60dB */, rlm@0: EffectiveDist / DecayDistance[i]); rlm@0: } rlm@0: } rlm@0: rlm@0: // Calculate Velocity rlm@0: if(DopplerFactor != 0.0f) rlm@0: { rlm@0: ALfloat VSS, VLS; rlm@0: ALfloat MaxVelocity = (SpeedOfSound*DopplerVelocity) / rlm@0: DopplerFactor; rlm@0: rlm@0: VSS = aluDotproduct(Velocity, SourceToListener); rlm@0: if(VSS >= MaxVelocity) rlm@0: VSS = (MaxVelocity - 1.0f); rlm@0: else if(VSS <= -MaxVelocity) rlm@0: VSS = -MaxVelocity + 1.0f; rlm@0: rlm@0: VLS = aluDotproduct(ListenerVel, SourceToListener); rlm@0: if(VLS >= MaxVelocity) rlm@0: VLS = (MaxVelocity - 1.0f); rlm@0: else if(VLS <= -MaxVelocity) rlm@0: VLS = -MaxVelocity + 1.0f; rlm@0: rlm@0: Pitch *= ((SpeedOfSound*DopplerVelocity) - (DopplerFactor*VLS)) / rlm@0: ((SpeedOfSound*DopplerVelocity) - (DopplerFactor*VSS)); rlm@0: } rlm@0: rlm@0: BufferListItem = ALSource->queue; rlm@0: while(BufferListItem != NULL) rlm@0: { rlm@0: ALbuffer *ALBuffer; rlm@0: if((ALBuffer=BufferListItem->buffer) != NULL) rlm@0: { rlm@0: ALint maxstep = STACK_DATA_SIZE / ALSource->NumChannels / rlm@0: ALSource->SampleSize; rlm@0: maxstep -= ResamplerPadding[Resampler] + rlm@0: ResamplerPrePadding[Resampler] + 1; rlm@0: maxstep = mini(maxstep, INT_MAX>>FRACTIONBITS); rlm@0: rlm@0: Pitch = Pitch * ALBuffer->Frequency / Frequency; rlm@0: if(Pitch > (ALfloat)maxstep) rlm@0: ALSource->Params.Step = maxstep<Params.Step = Pitch*FRACTIONONE; rlm@0: if(ALSource->Params.Step == 0) rlm@0: ALSource->Params.Step = 1; rlm@0: } rlm@0: rlm@0: if((Device->Flags&DEVICE_USE_HRTF)) rlm@0: ALSource->Params.DoMix = SelectHrtfMixer(ALBuffer, rlm@0: (ALSource->Params.Step==FRACTIONONE) ? POINT_RESAMPLER : rlm@0: Resampler); rlm@0: else rlm@0: ALSource->Params.DoMix = SelectMixer(ALBuffer, rlm@0: (ALSource->Params.Step==FRACTIONONE) ? POINT_RESAMPLER : rlm@0: Resampler); rlm@0: break; rlm@0: } rlm@0: BufferListItem = BufferListItem->next; rlm@0: } rlm@0: rlm@0: if((Device->Flags&DEVICE_USE_HRTF)) rlm@0: { rlm@0: // Use a binaural HRTF algorithm for stereo headphone playback rlm@0: ALfloat delta, ev = 0.0f, az = 0.0f; rlm@0: rlm@0: if(Distance > 0.0f) rlm@0: { rlm@0: ALfloat invlen = 1.0f/Distance; rlm@0: Position[0] *= invlen; rlm@0: Position[1] *= invlen; rlm@0: Position[2] *= invlen; rlm@0: rlm@0: // Calculate elevation and azimuth only when the source is not at rlm@0: // the listener. This prevents +0 and -0 Z from producing rlm@0: // inconsistent panning. rlm@0: ev = asin(Position[1]); rlm@0: az = atan2(Position[0], -Position[2]*ZScale); rlm@0: } rlm@0: rlm@0: // Check to see if the HRIR is already moving. rlm@0: if(ALSource->HrtfMoving) rlm@0: { rlm@0: // Calculate the normalized HRTF transition factor (delta). rlm@0: delta = CalcHrtfDelta(ALSource->Params.HrtfGain, DryGain, rlm@0: ALSource->Params.HrtfDir, Position); rlm@0: // If the delta is large enough, get the moving HRIR target rlm@0: // coefficients, target delays, steppping values, and counter. rlm@0: if(delta > 0.001f) rlm@0: { rlm@0: ALSource->HrtfCounter = GetMovingHrtfCoeffs(ev, az, DryGain, rlm@0: delta, ALSource->HrtfCounter, rlm@0: ALSource->Params.HrtfCoeffs[0], rlm@0: ALSource->Params.HrtfDelay[0], rlm@0: ALSource->Params.HrtfCoeffStep, rlm@0: ALSource->Params.HrtfDelayStep); rlm@0: ALSource->Params.HrtfGain = DryGain; rlm@0: ALSource->Params.HrtfDir[0] = Position[0]; rlm@0: ALSource->Params.HrtfDir[1] = Position[1]; rlm@0: ALSource->Params.HrtfDir[2] = Position[2]; rlm@0: } rlm@0: } rlm@0: else rlm@0: { rlm@0: // Get the initial (static) HRIR coefficients and delays. rlm@0: GetLerpedHrtfCoeffs(ev, az, DryGain, rlm@0: ALSource->Params.HrtfCoeffs[0], rlm@0: ALSource->Params.HrtfDelay[0]); rlm@0: ALSource->HrtfCounter = 0; rlm@0: ALSource->Params.HrtfGain = DryGain; rlm@0: ALSource->Params.HrtfDir[0] = Position[0]; rlm@0: ALSource->Params.HrtfDir[1] = Position[1]; rlm@0: ALSource->Params.HrtfDir[2] = Position[2]; rlm@0: } rlm@0: } rlm@0: else rlm@0: { rlm@0: // Use energy-preserving panning algorithm for multi-speaker playback rlm@0: ALfloat DirGain, AmbientGain; rlm@0: const ALfloat *SpeakerGain; rlm@0: ALfloat length; rlm@0: ALint pos; rlm@0: rlm@0: length = maxf(Distance, MinDist); rlm@0: if(length > 0.0f) rlm@0: { rlm@0: ALfloat invlen = 1.0f/length; rlm@0: Position[0] *= invlen; rlm@0: Position[1] *= invlen; rlm@0: Position[2] *= invlen; rlm@0: } rlm@0: rlm@0: pos = aluCart2LUTpos(-Position[2]*ZScale, Position[0]); rlm@0: SpeakerGain = Device->PanningLUT[pos]; rlm@0: rlm@0: DirGain = aluSqrt(Position[0]*Position[0] + Position[2]*Position[2]); rlm@0: // elevation adjustment for directional gain. this sucks, but rlm@0: // has low complexity rlm@0: AmbientGain = aluSqrt(1.0/Device->NumChan); rlm@0: for(i = 0;i < MAXCHANNELS;i++) rlm@0: { rlm@0: ALuint i2; rlm@0: for(i2 = 0;i2 < MAXCHANNELS;i2++) rlm@0: ALSource->Params.DryGains[i][i2] = 0.0f; rlm@0: } rlm@0: for(i = 0;i < (ALint)Device->NumChan;i++) rlm@0: { rlm@0: enum Channel chan = Device->Speaker2Chan[i]; rlm@0: ALfloat gain = lerp(AmbientGain, SpeakerGain[chan], DirGain); rlm@0: ALSource->Params.DryGains[0][chan] = DryGain * gain; rlm@0: } rlm@0: } rlm@0: for(i = 0;i < NumSends;i++) rlm@0: ALSource->Params.Send[i].WetGain = WetGain[i]; rlm@0: rlm@0: /* Update filter coefficients. */ rlm@0: cw = cos(2.0*M_PI * LOWPASSFREQCUTOFF / Frequency); rlm@0: rlm@0: ALSource->Params.iirFilter.coeff = lpCoeffCalc(DryGainHF, cw); rlm@0: for(i = 0;i < NumSends;i++) rlm@0: { rlm@0: ALfloat a = lpCoeffCalc(WetGainHF[i]*WetGainHF[i], cw); rlm@0: ALSource->Params.Send[i].iirFilter.coeff = a; rlm@0: } rlm@0: } rlm@0: rlm@0: rlm@0: static __inline ALfloat aluF2F(ALfloat val) rlm@0: { return val; } rlm@0: static __inline ALshort aluF2S(ALfloat val) rlm@0: { rlm@0: if(val > 1.0f) return 32767; rlm@0: if(val < -1.0f) return -32768; rlm@0: return (ALint)(val*32767.0f); rlm@0: } rlm@0: static __inline ALushort aluF2US(ALfloat val) rlm@0: { return aluF2S(val)+32768; } rlm@0: static __inline ALbyte aluF2B(ALfloat val) rlm@0: { return aluF2S(val)>>8; } rlm@0: static __inline ALubyte aluF2UB(ALfloat val) rlm@0: { return aluF2US(val)>>8; } rlm@0: rlm@0: #define DECL_TEMPLATE(T, N, func) \ rlm@0: static void Write_##T##_##N(ALCdevice *device, T *RESTRICT buffer, \ rlm@0: ALuint SamplesToDo) \ rlm@0: { \ rlm@0: ALfloat (*RESTRICT DryBuffer)[MAXCHANNELS] = device->DryBuffer; \ rlm@0: const enum Channel *ChanMap = device->DevChannels; \ rlm@0: ALuint i, j; \ rlm@0: \ rlm@0: for(i = 0;i < SamplesToDo;i++) \ rlm@0: { \ rlm@0: for(j = 0;j < N;j++) \ rlm@0: *(buffer++) = func(DryBuffer[i][ChanMap[j]]); \ rlm@0: } \ rlm@0: } rlm@0: rlm@0: DECL_TEMPLATE(ALfloat, 1, aluF2F) rlm@0: DECL_TEMPLATE(ALfloat, 4, aluF2F) rlm@0: DECL_TEMPLATE(ALfloat, 6, aluF2F) rlm@0: DECL_TEMPLATE(ALfloat, 7, aluF2F) rlm@0: DECL_TEMPLATE(ALfloat, 8, aluF2F) rlm@0: rlm@0: DECL_TEMPLATE(ALushort, 1, aluF2US) rlm@0: DECL_TEMPLATE(ALushort, 4, aluF2US) rlm@0: DECL_TEMPLATE(ALushort, 6, aluF2US) rlm@0: DECL_TEMPLATE(ALushort, 7, aluF2US) rlm@0: DECL_TEMPLATE(ALushort, 8, aluF2US) rlm@0: rlm@0: DECL_TEMPLATE(ALshort, 1, aluF2S) rlm@0: DECL_TEMPLATE(ALshort, 4, aluF2S) rlm@0: DECL_TEMPLATE(ALshort, 6, aluF2S) rlm@0: DECL_TEMPLATE(ALshort, 7, aluF2S) rlm@0: DECL_TEMPLATE(ALshort, 8, aluF2S) rlm@0: rlm@0: DECL_TEMPLATE(ALubyte, 1, aluF2UB) rlm@0: DECL_TEMPLATE(ALubyte, 4, aluF2UB) rlm@0: DECL_TEMPLATE(ALubyte, 6, aluF2UB) rlm@0: DECL_TEMPLATE(ALubyte, 7, aluF2UB) rlm@0: DECL_TEMPLATE(ALubyte, 8, aluF2UB) rlm@0: rlm@0: DECL_TEMPLATE(ALbyte, 1, aluF2B) rlm@0: DECL_TEMPLATE(ALbyte, 4, aluF2B) rlm@0: DECL_TEMPLATE(ALbyte, 6, aluF2B) rlm@0: DECL_TEMPLATE(ALbyte, 7, aluF2B) rlm@0: DECL_TEMPLATE(ALbyte, 8, aluF2B) rlm@0: rlm@0: #undef DECL_TEMPLATE rlm@0: rlm@0: #define DECL_TEMPLATE(T, N, func) \ rlm@0: static void Write_##T##_##N(ALCdevice *device, T *RESTRICT buffer, \ rlm@0: ALuint SamplesToDo) \ rlm@0: { \ rlm@0: ALfloat (*RESTRICT DryBuffer)[MAXCHANNELS] = device->DryBuffer; \ rlm@0: const enum Channel *ChanMap = device->DevChannels; \ rlm@0: ALuint i, j; \ rlm@0: \ rlm@0: if(device->Bs2b) \ rlm@0: { \ rlm@0: for(i = 0;i < SamplesToDo;i++) \ rlm@0: { \ rlm@0: float samples[2]; \ rlm@0: samples[0] = DryBuffer[i][ChanMap[0]]; \ rlm@0: samples[1] = DryBuffer[i][ChanMap[1]]; \ rlm@0: bs2b_cross_feed(device->Bs2b, samples); \ rlm@0: *(buffer++) = func(samples[0]); \ rlm@0: *(buffer++) = func(samples[1]); \ rlm@0: } \ rlm@0: } \ rlm@0: else \ rlm@0: { \ rlm@0: for(i = 0;i < SamplesToDo;i++) \ rlm@0: { \ rlm@0: for(j = 0;j < N;j++) \ rlm@0: *(buffer++) = func(DryBuffer[i][ChanMap[j]]); \ rlm@0: } \ rlm@0: } \ rlm@0: } rlm@0: rlm@0: DECL_TEMPLATE(ALfloat, 2, aluF2F) rlm@0: DECL_TEMPLATE(ALushort, 2, aluF2US) rlm@0: DECL_TEMPLATE(ALshort, 2, aluF2S) rlm@0: DECL_TEMPLATE(ALubyte, 2, aluF2UB) rlm@0: DECL_TEMPLATE(ALbyte, 2, aluF2B) rlm@0: rlm@0: #undef DECL_TEMPLATE rlm@0: rlm@0: #define DECL_TEMPLATE(T) \ rlm@0: static void Write_##T(ALCdevice *device, T *buffer, ALuint SamplesToDo) \ rlm@0: { \ rlm@0: switch(device->FmtChans) \ rlm@0: { \ rlm@0: case DevFmtMono: \ rlm@0: Write_##T##_1(device, buffer, SamplesToDo); \ rlm@0: break; \ rlm@0: case DevFmtStereo: \ rlm@0: Write_##T##_2(device, buffer, SamplesToDo); \ rlm@0: break; \ rlm@0: case DevFmtQuad: \ rlm@0: Write_##T##_4(device, buffer, SamplesToDo); \ rlm@0: break; \ rlm@0: case DevFmtX51: \ rlm@0: case DevFmtX51Side: \ rlm@0: Write_##T##_6(device, buffer, SamplesToDo); \ rlm@0: break; \ rlm@0: case DevFmtX61: \ rlm@0: Write_##T##_7(device, buffer, SamplesToDo); \ rlm@0: break; \ rlm@0: case DevFmtX71: \ rlm@0: Write_##T##_8(device, buffer, SamplesToDo); \ rlm@0: break; \ rlm@0: } \ rlm@0: } rlm@0: rlm@0: DECL_TEMPLATE(ALfloat) rlm@0: DECL_TEMPLATE(ALushort) rlm@0: DECL_TEMPLATE(ALshort) rlm@0: DECL_TEMPLATE(ALubyte) rlm@0: DECL_TEMPLATE(ALbyte) rlm@0: rlm@0: #undef DECL_TEMPLATE rlm@0: rlm@0: ALvoid aluMixData(ALCdevice *device, ALvoid *buffer, ALsizei size) rlm@0: { rlm@0: ALuint SamplesToDo; rlm@0: ALeffectslot *ALEffectSlot; rlm@0: ALCcontext **ctx, **ctx_end; rlm@0: ALsource **src, **src_end; rlm@0: int fpuState; rlm@0: ALuint i, c; rlm@0: ALsizei e; rlm@0: rlm@0: #if defined(HAVE_FESETROUND) rlm@0: fpuState = fegetround(); rlm@0: fesetround(FE_TOWARDZERO); rlm@0: #elif defined(HAVE__CONTROLFP) rlm@0: fpuState = _controlfp(0, 0); rlm@0: (void)_controlfp(_RC_CHOP, _MCW_RC); rlm@0: #else rlm@0: (void)fpuState; rlm@0: #endif rlm@0: rlm@0: while(size > 0) rlm@0: { rlm@0: /* Setup variables */ rlm@0: SamplesToDo = minu(size, BUFFERSIZE); rlm@0: rlm@0: /* Clear mixing buffer */ rlm@0: memset(device->DryBuffer, 0, SamplesToDo*MAXCHANNELS*sizeof(ALfloat)); rlm@0: rlm@0: LockDevice(device); rlm@0: ctx = device->Contexts; rlm@0: ctx_end = ctx + device->NumContexts; rlm@0: //printf("Contexts: %d\n", device->NumContexts); rlm@0: int context_number = 0; rlm@0: while(ctx != ctx_end) rlm@0: { rlm@0: //printf("Context %d:\n", context_number++); rlm@0: ALboolean DeferUpdates = (*ctx)->DeferUpdates; rlm@0: ALboolean UpdateSources = AL_FALSE; rlm@0: rlm@0: if(!DeferUpdates) rlm@0: { rlm@0: //printf("NOT deferring updates, whatever that means\n"); rlm@0: UpdateSources = (*ctx)->UpdateSources; rlm@0: //printf("update sources is set to %d\n", UpdateSources); rlm@0: (*ctx)->UpdateSources = AL_FALSE; rlm@0: } rlm@0: rlm@0: src = (*ctx)->ActiveSources; rlm@0: src_end = src + (*ctx)->ActiveSourceCount; rlm@0: //printf("number of active sources are %d\n", (*ctx)->ActiveSourceCount); rlm@0: while(src != src_end) rlm@0: { rlm@0: rlm@0: if((*src)->state != AL_PLAYING) rlm@0: { rlm@0: --((*ctx)->ActiveSourceCount); rlm@0: *src = *(--src_end); rlm@0: continue; rlm@0: } rlm@0: rlm@0: if(!DeferUpdates && ((*src)->NeedsUpdate || UpdateSources)) rlm@0: { rlm@0: (*src)->NeedsUpdate = AL_FALSE; rlm@0: ALsource_Update(*src, *ctx); rlm@0: } rlm@0: //printf("calling MixSource!\n"); rlm@0: MixSource(*src, device, SamplesToDo); rlm@0: src++; rlm@0: } rlm@0: rlm@0: /* effect slot processing */ rlm@0: for(e = 0;e < (*ctx)->EffectSlotMap.size;e++) rlm@0: { rlm@0: ALEffectSlot = (*ctx)->EffectSlotMap.array[e].value; rlm@0: rlm@0: for(i = 0;i < SamplesToDo;i++) rlm@0: { rlm@0: // RLM: remove click-removal rlm@0: ALEffectSlot->WetBuffer[i] += ALEffectSlot->ClickRemoval[0]; rlm@0: ALEffectSlot->ClickRemoval[0] -= ALEffectSlot->ClickRemoval[0] / 256.0f; rlm@0: } rlm@0: for(i = 0;i < 1;i++) rlm@0: { rlm@0: // RLM: remove click-removal rlm@0: ALEffectSlot->ClickRemoval[i] += ALEffectSlot->PendingClicks[i]; rlm@0: ALEffectSlot->PendingClicks[i] = 0.0f; rlm@0: } rlm@0: rlm@0: if(!DeferUpdates && ALEffectSlot->NeedsUpdate) rlm@0: { rlm@0: ALEffectSlot->NeedsUpdate = AL_FALSE; rlm@0: ALEffect_Update(ALEffectSlot->EffectState, *ctx, ALEffectSlot); rlm@0: } rlm@0: rlm@0: ALEffect_Process(ALEffectSlot->EffectState, ALEffectSlot, rlm@0: SamplesToDo, ALEffectSlot->WetBuffer, rlm@0: device->DryBuffer); rlm@0: rlm@0: for(i = 0;i < SamplesToDo;i++) rlm@0: ALEffectSlot->WetBuffer[i] = 0.0f; rlm@0: } rlm@0: rlm@0: ctx++; rlm@0: } rlm@0: UnlockDevice(device); rlm@0: rlm@0: //Post processing loop rlm@0: if(device->FmtChans == DevFmtMono) rlm@0: { rlm@0: for(i = 0;i < SamplesToDo;i++) rlm@0: { rlm@0: // RLM: remove click-removal rlm@0: device->DryBuffer[i][FRONT_CENTER] += device->ClickRemoval[FRONT_CENTER]; rlm@0: device->ClickRemoval[FRONT_CENTER] -= device->ClickRemoval[FRONT_CENTER] / 256.0f; rlm@0: } rlm@0: // RLM: remove click-removal rlm@0: device->ClickRemoval[FRONT_CENTER] += device->PendingClicks[FRONT_CENTER]; rlm@0: device->PendingClicks[FRONT_CENTER] = 0.0f; rlm@0: } rlm@0: else if(device->FmtChans == DevFmtStereo) rlm@0: { rlm@0: /* Assumes the first two channels are FRONT_LEFT and FRONT_RIGHT */ rlm@0: for(i = 0;i < SamplesToDo;i++) rlm@0: { rlm@0: for(c = 0;c < 2;c++) rlm@0: { rlm@0: // RLM: remove click-removal rlm@0: device->DryBuffer[i][c] += device->ClickRemoval[c]; rlm@0: device->ClickRemoval[c] -= device->ClickRemoval[c] / 256.0f; rlm@0: } rlm@0: } rlm@0: for(c = 0;c < 2;c++) rlm@0: { rlm@0: // RLM: remove click-removal rlm@0: device->ClickRemoval[c] += device->PendingClicks[c]; rlm@0: device->PendingClicks[c] = 0.0f; rlm@0: } rlm@0: } rlm@0: else rlm@0: { rlm@0: for(i = 0;i < SamplesToDo;i++) rlm@0: { rlm@0: for(c = 0;c < MAXCHANNELS;c++) rlm@0: { rlm@0: // RLM: remove click-removal rlm@0: device->DryBuffer[i][c] += device->ClickRemoval[c]; rlm@0: device->ClickRemoval[c] -= device->ClickRemoval[c] / 256.0f; rlm@0: } rlm@0: } rlm@0: for(c = 0;c < MAXCHANNELS;c++) rlm@0: { rlm@0: // RLM: remove click-removal rlm@0: device->ClickRemoval[c] += device->PendingClicks[c]; rlm@0: device->PendingClicks[c] = 0.0f; rlm@0: } rlm@0: } rlm@0: rlm@0: if(buffer) rlm@0: { rlm@0: switch(device->FmtType) rlm@0: { rlm@0: case DevFmtByte: rlm@0: Write_ALbyte(device, buffer, SamplesToDo); rlm@0: break; rlm@0: case DevFmtUByte: rlm@0: Write_ALubyte(device, buffer, SamplesToDo); rlm@0: break; rlm@0: case DevFmtShort: rlm@0: Write_ALshort(device, buffer, SamplesToDo); rlm@0: break; rlm@0: case DevFmtUShort: rlm@0: Write_ALushort(device, buffer, SamplesToDo); rlm@0: break; rlm@0: case DevFmtFloat: rlm@0: Write_ALfloat(device, buffer, SamplesToDo); rlm@0: break; rlm@0: } rlm@0: } rlm@0: rlm@0: size -= SamplesToDo; rlm@0: } rlm@0: rlm@0: #if defined(HAVE_FESETROUND) rlm@0: fesetround(fpuState); rlm@0: #elif defined(HAVE__CONTROLFP) rlm@0: _controlfp(fpuState, _MCW_RC); rlm@0: #endif rlm@0: } rlm@0: rlm@0: rlm@0: rlm@0: rlm@0: rlm@0: ALvoid aluHandleDisconnect(ALCdevice *device) rlm@0: { rlm@0: ALuint i; rlm@0: rlm@0: LockDevice(device); rlm@0: for(i = 0;i < device->NumContexts;i++) rlm@0: { rlm@0: ALCcontext *Context = device->Contexts[i]; rlm@0: ALsource *source; rlm@0: ALsizei pos; rlm@0: rlm@0: for(pos = 0;pos < Context->SourceMap.size;pos++) rlm@0: { rlm@0: source = Context->SourceMap.array[pos].value; rlm@0: if(source->state == AL_PLAYING) rlm@0: { rlm@0: source->state = AL_STOPPED; rlm@0: source->BuffersPlayed = source->BuffersInQueue; rlm@0: source->position = 0; rlm@0: source->position_fraction = 0; rlm@0: } rlm@0: } rlm@0: } rlm@0: rlm@0: device->Connected = ALC_FALSE; rlm@0: UnlockDevice(device); rlm@0: }