/** * 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 */ #define _CRT_SECURE_NO_DEPRECATE // get rid of sprintf security warnings on VS2005 #include "config.h" #include #include "alMain.h" #include "AL/al.h" #include "AL/alc.h" #include "alSource.h" #include "alBuffer.h" #include "alThunk.h" #include "alListener.h" #include "alAuxEffectSlot.h" #include "alu.h" #include "bs2b.h" #if defined(HAVE_STDINT_H) #include typedef int64_t ALint64; #elif defined(HAVE___INT64) typedef __int64 ALint64; #elif (SIZEOF_LONG == 8) typedef long ALint64; #elif (SIZEOF_LONG_LONG == 8) typedef long long ALint64; #endif #ifdef HAVE_SQRTF #define aluSqrt(x) ((ALfloat)sqrtf((float)(x))) #else #define aluSqrt(x) ((ALfloat)sqrt((double)(x))) #endif #ifdef HAVE_ACOSF #define aluAcos(x) ((ALfloat)acosf((float)(x))) #else #define aluAcos(x) ((ALfloat)acos((double)(x))) #endif // fixes for mingw32. #if defined(max) && !defined(__max) #define __max max #endif #if defined(min) && !defined(__min) #define __min min #endif #define BUFFERSIZE 24000 #define FRACTIONBITS 14 #define FRACTIONMASK ((1L<history; ALfloat a = iir->coeff; ALfloat output = input; output = output + (history[0]-output)*a; history[0] = output; output = output + (history[1]-output)*a; history[1] = output; output = output + (history[2]-output)*a; history[2] = output; output = output + (history[3]-output)*a; history[3] = output; return output; } static __inline ALshort aluF2S(ALfloat Value) { ALint i; i = (ALint)Value; i = __min( 32767, i); i = __max(-32768, i); return ((ALshort)i); } static __inline ALvoid aluCrossproduct(ALfloat *inVector1,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(ALfloat *inVector1,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 matrix[3][3]) { ALfloat result[3]; result[0] = vector[0]*matrix[0][0] + vector[1]*matrix[1][0] + vector[2]*matrix[2][0]; result[1] = vector[0]*matrix[0][1] + vector[1]*matrix[1][1] + vector[2]*matrix[2][1]; result[2] = vector[0]*matrix[0][2] + vector[1]*matrix[1][2] + vector[2]*matrix[2][2]; memcpy(vector, result, sizeof(result)); } static ALvoid CalcSourceParams(ALCcontext *ALContext, ALsource *ALSource, ALenum isMono, ALenum OutputFormat, ALfloat *drysend, ALfloat *wetsend, ALfloat *pitch, ALfloat *drygainhf, ALfloat *wetgainhf) { ALfloat InnerAngle,OuterAngle,Angle,Distance,DryMix,WetMix=0.0f; ALfloat Direction[3],Position[3],SourceToListener[3]; ALfloat MinVolume,MaxVolume,MinDist,MaxDist,Rolloff,OuterGainHF; ALfloat ConeVolume,SourceVolume,PanningFB,PanningLR,ListenerGain; ALfloat U[3],V[3],N[3]; ALfloat DopplerFactor, DopplerVelocity, flSpeedOfSound, flMaxVelocity; ALfloat Matrix[3][3]; ALfloat flAttenuation; ALfloat RoomAttenuation; ALfloat MetersPerUnit; ALfloat RoomRolloff; ALfloat DryGainHF = 1.0f; ALfloat WetGainHF = 1.0f; ALfloat cw, a, g; //Get context properties DopplerFactor = ALContext->DopplerFactor * ALSource->DopplerFactor; DopplerVelocity = ALContext->DopplerVelocity; flSpeedOfSound = ALContext->flSpeedOfSound; //Get listener properties ListenerGain = ALContext->Listener.Gain; MetersPerUnit = ALContext->Listener.MetersPerUnit; //Get source properties SourceVolume = ALSource->flGain; memcpy(Position, ALSource->vPosition, sizeof(ALSource->vPosition)); memcpy(Direction, ALSource->vOrientation, sizeof(ALSource->vOrientation)); MinVolume = ALSource->flMinGain; MaxVolume = ALSource->flMaxGain; MinDist = ALSource->flRefDistance; MaxDist = ALSource->flMaxDistance; Rolloff = ALSource->flRollOffFactor; InnerAngle = ALSource->flInnerAngle; OuterAngle = ALSource->flOuterAngle; OuterGainHF = ALSource->OuterGainHF; RoomRolloff = ALSource->RoomRolloffFactor; //Only apply 3D calculations for mono buffers if(isMono != AL_FALSE) { //1. Translate Listener to origin (convert to head relative) // Note that Direction and SourceToListener are *not* transformed. // SourceToListener is used with the source and listener velocities, // which are untransformed, and Direction is used with SourceToListener // for the sound cone if(ALSource->bHeadRelative==AL_FALSE) { // Build transform matrix aluCrossproduct(ALContext->Listener.Forward, ALContext->Listener.Up, U); // Right-vector aluNormalize(U); // Normalized Right-vector memcpy(V, ALContext->Listener.Up, sizeof(V)); // Up-vector aluNormalize(V); // Normalized Up-vector memcpy(N, ALContext->Listener.Forward, sizeof(N)); // At-vector aluNormalize(N); // Normalized At-vector Matrix[0][0] = U[0]; Matrix[0][1] = V[0]; Matrix[0][2] = -N[0]; Matrix[1][0] = U[1]; Matrix[1][1] = V[1]; Matrix[1][2] = -N[1]; Matrix[2][0] = U[2]; Matrix[2][1] = V[2]; Matrix[2][2] = -N[2]; // Translate source position into listener space Position[0] -= ALContext->Listener.Position[0]; Position[1] -= ALContext->Listener.Position[1]; Position[2] -= ALContext->Listener.Position[2]; SourceToListener[0] = -Position[0]; SourceToListener[1] = -Position[1]; SourceToListener[2] = -Position[2]; // Transform source position and direction into listener space aluMatrixVector(Position, Matrix); } else { 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)); if(ALSource->Send[0].Slot) { if(ALSource->Send[0].Slot->effect.type == AL_EFFECT_REVERB) RoomRolloff += ALSource->Send[0].Slot->effect.Reverb.RoomRolloffFactor; } flAttenuation = 1.0f; RoomAttenuation = 1.0f; switch (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))); if ((MinDist + (RoomRolloff * (Distance - MinDist))) > 0.0f) RoomAttenuation = MinDist / (MinDist + (RoomRolloff * (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)); RoomAttenuation = 1.0f - (RoomRolloff*(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 = (ALfloat)pow(Distance/MinDist, -Rolloff); RoomAttenuation = (ALfloat)pow(Distance/MinDist, -RoomRolloff); } break; case AL_NONE: flAttenuation = 1.0f; RoomAttenuation = 1.0f; break; } // Distance-based air absorption if(ALSource->AirAbsorptionFactor > 0.0f && ALContext->DistanceModel != AL_NONE) { ALfloat dist = Distance-MinDist; ALfloat absorb; if(dist < 0.0f) dist = 0.0f; // Absorption calculation is done in dB absorb = (ALSource->AirAbsorptionFactor*AIRABSORBGAINDBHF) * (Distance*MetersPerUnit); // Convert dB to linear gain before applying absorb = pow(0.5, absorb/-6.0); DryGainHF *= absorb; WetGainHF *= absorb; } // Source Gain + Attenuation and clamp to Min/Max Gain DryMix = SourceVolume * flAttenuation; DryMix = __min(DryMix,MaxVolume); DryMix = __max(DryMix,MinVolume); WetMix = SourceVolume * RoomAttenuation; WetMix = __min(WetMix,MaxVolume); WetMix = __max(WetMix,MinVolume); //3. Apply directional soundcones Angle = aluAcos(aluDotproduct(Direction,SourceToListener)) * 180.0f / 3.141592654f; if(Angle >= InnerAngle && Angle <= OuterAngle) { ALfloat scale = (Angle-InnerAngle) / (OuterAngle-InnerAngle); ConeVolume = (1.0f+(ALSource->flOuterGain-1.0f)*scale); DryMix *= ConeVolume; if(ALSource->WetGainAuto) WetMix *= ConeVolume; if(ALSource->DryGainHFAuto) DryGainHF *= (1.0f+(OuterGainHF-1.0f)*scale); if(ALSource->WetGainHFAuto) WetGainHF *= (1.0f+(OuterGainHF-1.0f)*scale); } else if(Angle > OuterAngle) { ConeVolume = (1.0f+(ALSource->flOuterGain-1.0f)); DryMix *= ConeVolume; if(ALSource->WetGainAuto) WetMix *= ConeVolume; if(ALSource->DryGainHFAuto) DryGainHF *= (1.0f+(OuterGainHF-1.0f)); if(ALSource->WetGainHFAuto) WetGainHF *= (1.0f+(OuterGainHF-1.0f)); } //4. Calculate Velocity if(DopplerFactor != 0.0f) { ALfloat flVSS, flVLS = 0.0f; if(ALSource->bHeadRelative==AL_FALSE) flVLS = aluDotproduct(ALContext->Listener.Velocity, SourceToListener); flVSS = aluDotproduct(ALSource->vVelocity, SourceToListener); flMaxVelocity = (DopplerVelocity * flSpeedOfSound) / DopplerFactor; if (flVSS >= flMaxVelocity) flVSS = (flMaxVelocity - 1.0f); else if (flVSS <= -flMaxVelocity) flVSS = -flMaxVelocity + 1.0f; if (flVLS >= flMaxVelocity) flVLS = (flMaxVelocity - 1.0f); else if (flVLS <= -flMaxVelocity) flVLS = -flMaxVelocity + 1.0f; pitch[0] = ALSource->flPitch * ((flSpeedOfSound * DopplerVelocity) - (DopplerFactor * flVLS)) / ((flSpeedOfSound * DopplerVelocity) - (DopplerFactor * flVSS)); } else pitch[0] = ALSource->flPitch; if(ALSource->Send[0].Slot && ALSource->Send[0].Slot->effect.type != AL_EFFECT_NULL) { // If the slot's auxilliary send auto is off, the data sent to the // effect slot is the same as the dry path, sans filter effects if(!ALSource->Send[0].Slot->AuxSendAuto) { WetMix = DryMix; WetGainHF = DryGainHF; } // Note that these are really applied by the effect slot. However, // it's easier to handle them here (particularly the lowpass // filter). Applying the gain to the individual sources going to // the effect slot should have the same effect as applying the gain // to the accumulated sources in the effect slot. // vol1*g + vol2*g + ... voln*g = (vol1+vol2+...voln)*g WetMix *= ALSource->Send[0].Slot->Gain; if(ALSource->Send[0].Slot->effect.type == AL_EFFECT_REVERB) { WetMix *= ALSource->Send[0].Slot->effect.Reverb.Gain; WetGainHF *= ALSource->Send[0].Slot->effect.Reverb.GainHF; WetGainHF *= pow(ALSource->Send[0].Slot->effect.Reverb.AirAbsorptionGainHF, Distance * MetersPerUnit); } } else { WetMix = 0.0f; WetGainHF = 1.0f; } //5. Apply filter gains and filters switch(ALSource->DirectFilter.type) { case AL_FILTER_LOWPASS: DryMix *= ALSource->DirectFilter.Gain; DryGainHF *= ALSource->DirectFilter.GainHF; break; } switch(ALSource->Send[0].WetFilter.type) { case AL_FILTER_LOWPASS: WetMix *= ALSource->Send[0].WetFilter.Gain; WetGainHF *= ALSource->Send[0].WetFilter.GainHF; break; } DryMix *= ListenerGain; WetMix *= ListenerGain; //6. Convert normalized position into pannings, then into channel volumes aluNormalize(Position); switch(aluChannelsFromFormat(OutputFormat)) { case 1: case 2: PanningLR = 0.5f + 0.5f*Position[0]; drysend[FRONT_LEFT] = DryMix * aluSqrt(1.0f-PanningLR); //L Direct drysend[FRONT_RIGHT] = DryMix * aluSqrt( PanningLR); //R Direct drysend[BACK_LEFT] = 0.0f; drysend[BACK_RIGHT] = 0.0f; drysend[SIDE_LEFT] = 0.0f; drysend[SIDE_RIGHT] = 0.0f; break; case 4: /* TODO: Add center/lfe channel in spatial calculations? */ case 6: // Apply a scalar so each individual speaker has more weight PanningLR = 0.5f + (0.5f*Position[0]*1.41421356f); PanningLR = __min(1.0f, PanningLR); PanningLR = __max(0.0f, PanningLR); PanningFB = 0.5f + (0.5f*Position[2]*1.41421356f); PanningFB = __min(1.0f, PanningFB); PanningFB = __max(0.0f, PanningFB); drysend[FRONT_LEFT] = DryMix * aluSqrt((1.0f-PanningLR)*(1.0f-PanningFB)); drysend[FRONT_RIGHT] = DryMix * aluSqrt(( PanningLR)*(1.0f-PanningFB)); drysend[BACK_LEFT] = DryMix * aluSqrt((1.0f-PanningLR)*( PanningFB)); drysend[BACK_RIGHT] = DryMix * aluSqrt(( PanningLR)*( PanningFB)); drysend[SIDE_LEFT] = 0.0f; drysend[SIDE_RIGHT] = 0.0f; break; case 7: case 8: PanningFB = 1.0f - fabs(Position[2]*1.15470054f); PanningFB = __min(1.0f, PanningFB); PanningFB = __max(0.0f, PanningFB); PanningLR = 0.5f + (0.5*Position[0]*((1.0f-PanningFB)*2.0f)); PanningLR = __min(1.0f, PanningLR); PanningLR = __max(0.0f, PanningLR); if(Position[2] > 0.0f) { drysend[BACK_LEFT] = DryMix * aluSqrt((1.0f-PanningLR)*(1.0f-PanningFB)); drysend[BACK_RIGHT] = DryMix * aluSqrt(( PanningLR)*(1.0f-PanningFB)); drysend[SIDE_LEFT] = DryMix * aluSqrt((1.0f-PanningLR)*( PanningFB)); drysend[SIDE_RIGHT] = DryMix * aluSqrt(( PanningLR)*( PanningFB)); drysend[FRONT_LEFT] = 0.0f; drysend[FRONT_RIGHT] = 0.0f; } else { drysend[FRONT_LEFT] = DryMix * aluSqrt((1.0f-PanningLR)*(1.0f-PanningFB)); drysend[FRONT_RIGHT] = DryMix * aluSqrt(( PanningLR)*(1.0f-PanningFB)); drysend[SIDE_LEFT] = DryMix * aluSqrt((1.0f-PanningLR)*( PanningFB)); drysend[SIDE_RIGHT] = DryMix * aluSqrt(( PanningLR)*( PanningFB)); drysend[BACK_LEFT] = 0.0f; drysend[BACK_RIGHT] = 0.0f; } default: break; } *wetsend = WetMix; // Update filter coefficients. Calculations based on the I3DL2 spec. cw = cos(2.0f*3.141592654f * LOWPASSFREQCUTOFF / ALContext->Frequency); // We 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. // Be careful with gains < 0.0001, as that causes the coefficient to // head towards 1, which will flatten the signal g = aluSqrt(__max(DryGainHF, 0.0001f)); a = 0.0f; if(g < 0.9999f) // 1-epsilon a = (1 - g*cw - aluSqrt(2*g*(1-cw) - g*g*(1 - cw*cw))) / (1 - g); ALSource->iirFilter.coeff = a; g = aluSqrt(__max(WetGainHF, 0.0001f)); a = 0.0f; if(g < 0.9999f) // 1-epsilon a = (1 - g*cw - aluSqrt(2*g*(1-cw) - g*g*(1 - cw*cw))) / (1 - g); ALSource->Send[0].iirFilter.coeff = a; *drygainhf = DryGainHF; *wetgainhf = WetGainHF; } else { //1. Multi-channel buffers always play "normal" pitch[0] = ALSource->flPitch; drysend[FRONT_LEFT] = SourceVolume * ListenerGain; drysend[FRONT_RIGHT] = SourceVolume * ListenerGain; drysend[SIDE_LEFT] = SourceVolume * ListenerGain; drysend[SIDE_RIGHT] = SourceVolume * ListenerGain; drysend[BACK_LEFT] = SourceVolume * ListenerGain; drysend[BACK_RIGHT] = SourceVolume * ListenerGain; drysend[CENTER] = SourceVolume * ListenerGain; drysend[LFE] = SourceVolume * ListenerGain; *wetsend = 0.0f; WetGainHF = 1.0f; *drygainhf = DryGainHF; *wetgainhf = WetGainHF; } } static __inline ALshort lerp(ALshort val1, ALshort val2, ALint frac) { return val1 + (((val2-val1)*frac)>>FRACTIONBITS); } ALvoid aluMixData(ALCcontext *ALContext,ALvoid *buffer,ALsizei size,ALenum format) { static float DryBuffer[BUFFERSIZE][OUTPUTCHANNELS]; static float WetBuffer[BUFFERSIZE]; ALfloat newDrySend[OUTPUTCHANNELS] = { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f }; ALfloat newWetSend = 0.0f; ALfloat DryGainHF = 0.0f; ALfloat WetGainHF = 0.0f; ALfloat *DrySend; ALfloat *WetSend; ALuint rampLength; ALfloat dryGainStep[OUTPUTCHANNELS]; ALfloat wetGainStep; ALuint BlockAlign,BufferSize; ALuint DataSize=0,DataPosInt=0,DataPosFrac=0; ALuint Channels,Frequency,ulExtraSamples; ALfloat Pitch; ALint Looping,State; ALint increment; ALuint Buffer; ALuint SamplesToDo; ALsource *ALSource; ALbuffer *ALBuffer; ALeffectslot *ALEffectSlot; ALfloat value; ALshort *Data; ALuint i,j,k; ALbufferlistitem *BufferListItem; ALuint loop; ALint64 DataSize64,DataPos64; FILTER *DryFilter, *WetFilter; int fpuState; SuspendContext(ALContext); #if defined(HAVE_FESETROUND) fpuState = fegetround(); fesetround(FE_TOWARDZERO); #elif defined(HAVE__CONTROLFP) fpuState = _controlfp(0, 0); _controlfp(_RC_CHOP, _MCW_RC); #else (void)fpuState; #endif //Figure output format variables BlockAlign = aluChannelsFromFormat(format); BlockAlign *= aluBytesFromFormat(format); size /= BlockAlign; while(size > 0) { //Setup variables SamplesToDo = min(size, BUFFERSIZE); if(ALContext) { ALEffectSlot = ALContext->AuxiliaryEffectSlot; ALSource = ALContext->Source; rampLength = ALContext->Frequency * MIN_RAMP_LENGTH / 1000; } else { ALEffectSlot = NULL; ALSource = NULL; rampLength = 0; } rampLength = max(rampLength, SamplesToDo); //Clear mixing buffer memset(WetBuffer, 0, SamplesToDo*sizeof(ALfloat)); memset(DryBuffer, 0, SamplesToDo*OUTPUTCHANNELS*sizeof(ALfloat)); //Actual mixing loop while(ALSource) { j = 0; State = ALSource->state; while(State == AL_PLAYING && j < SamplesToDo) { DataSize = 0; DataPosInt = 0; DataPosFrac = 0; //Get buffer info if((Buffer = ALSource->ulBufferID)) { ALBuffer = (ALbuffer*)ALTHUNK_LOOKUPENTRY(Buffer); Data = ALBuffer->data; Channels = aluChannelsFromFormat(ALBuffer->format); DataSize = ALBuffer->size; DataSize /= Channels * aluBytesFromFormat(ALBuffer->format); Frequency = ALBuffer->frequency; DataPosInt = ALSource->position; DataPosFrac = ALSource->position_fraction; if(DataPosInt >= DataSize) goto skipmix; CalcSourceParams(ALContext, ALSource, (Channels==1) ? AL_TRUE : AL_FALSE, format, newDrySend, &newWetSend, &Pitch, &DryGainHF, &WetGainHF); Pitch = (Pitch*Frequency) / ALContext->Frequency; //Get source info DryFilter = &ALSource->iirFilter; WetFilter = &ALSource->Send[0].iirFilter; DrySend = ALSource->DryGains; WetSend = &ALSource->WetGain; //Compute the gain steps for each output channel for(i = 0;i < OUTPUTCHANNELS;i++) dryGainStep[i] = (newDrySend[i]-DrySend[i]) / rampLength; wetGainStep = (newWetSend-(*WetSend)) / rampLength; //Compute 18.14 fixed point step if(Pitch > (float)MAX_PITCH) Pitch = (float)MAX_PITCH; increment = (ALint)(Pitch*(ALfloat)(1L<queue; for(loop = 0; loop < ALSource->BuffersPlayed; loop++) { if(BufferListItem) BufferListItem = BufferListItem->next; } if (BufferListItem) { if (BufferListItem->next) { ALbuffer *NextBuf = (ALbuffer*)ALTHUNK_LOOKUPENTRY(BufferListItem->next->buffer); if(NextBuf && NextBuf->data) { ulExtraSamples = min(NextBuf->size, (ALint)(ALBuffer->padding*Channels*2)); memcpy(&Data[DataSize*Channels], NextBuf->data, ulExtraSamples); } } else if (ALSource->bLooping) { ALbuffer *NextBuf = (ALbuffer*)ALTHUNK_LOOKUPENTRY(ALSource->queue->buffer); if (NextBuf && NextBuf->data) { ulExtraSamples = min(NextBuf->size, (ALint)(ALBuffer->padding*Channels*2)); memcpy(&Data[DataSize*Channels], NextBuf->data, ulExtraSamples); } } else memset(&Data[DataSize*Channels], 0, (ALBuffer->padding*Channels*2)); } BufferSize = min(BufferSize, (SamplesToDo-j)); //Actual sample mixing loop k = 0; Data += DataPosInt*Channels; while(BufferSize--) { for(i = 0;i < OUTPUTCHANNELS;i++) DrySend[i] += dryGainStep[i]; *WetSend += wetGainStep; if(Channels==1) { ALfloat sample, outsamp; //First order interpolator sample = lerp(Data[k], Data[k+1], DataPosFrac); //Direct path final mix buffer and panning outsamp = lpFilter(DryFilter, sample); DryBuffer[j][FRONT_LEFT] += outsamp*DrySend[FRONT_LEFT]; DryBuffer[j][FRONT_RIGHT] += outsamp*DrySend[FRONT_RIGHT]; DryBuffer[j][SIDE_LEFT] += outsamp*DrySend[SIDE_LEFT]; DryBuffer[j][SIDE_RIGHT] += outsamp*DrySend[SIDE_RIGHT]; DryBuffer[j][BACK_LEFT] += outsamp*DrySend[BACK_LEFT]; DryBuffer[j][BACK_RIGHT] += outsamp*DrySend[BACK_RIGHT]; //Room path final mix buffer and panning outsamp = lpFilter(WetFilter, sample); WetBuffer[j] += outsamp*(*WetSend); } else { ALfloat samp1, samp2; //First order interpolator (front left) samp1 = lerp(Data[k*Channels], Data[(k+1)*Channels], DataPosFrac); DryBuffer[j][FRONT_LEFT] += samp1*DrySend[FRONT_LEFT]; //First order interpolator (front right) samp2 = lerp(Data[k*Channels+1], Data[(k+1)*Channels+1], DataPosFrac); DryBuffer[j][FRONT_RIGHT] += samp2*DrySend[FRONT_RIGHT]; if(Channels >= 4) { int i = 2; if(Channels >= 6) { if(Channels != 7) { //First order interpolator (center) value = lerp(Data[k*Channels+i], Data[(k+1)*Channels+i], DataPosFrac); DryBuffer[j][CENTER] += value*DrySend[CENTER]; i++; } //First order interpolator (lfe) value = lerp(Data[k*Channels+i], Data[(k+1)*Channels+i], DataPosFrac); DryBuffer[j][LFE] += value*DrySend[LFE]; i++; } //First order interpolator (back left) value = lerp(Data[k*Channels+i], Data[(k+1)*Channels+i], DataPosFrac); DryBuffer[j][BACK_LEFT] += value*DrySend[BACK_LEFT]; i++; //First order interpolator (back right) value = lerp(Data[k*Channels+i], Data[(k+1)*Channels+i], DataPosFrac); DryBuffer[j][BACK_RIGHT] += value*DrySend[BACK_RIGHT]; i++; if(Channels >= 7) { //First order interpolator (side left) value = lerp(Data[k*Channels+i], Data[(k+1)*Channels+i], DataPosFrac); DryBuffer[j][SIDE_LEFT] += value*DrySend[SIDE_LEFT]; i++; //First order interpolator (side right) value = lerp(Data[k*Channels+i], Data[(k+1)*Channels+i], DataPosFrac); DryBuffer[j][SIDE_RIGHT] += value*DrySend[SIDE_RIGHT]; i++; } } else if(DuplicateStereo) { //Duplicate stereo channels on the back speakers DryBuffer[j][BACK_LEFT] += samp1*DrySend[BACK_LEFT]; DryBuffer[j][BACK_RIGHT] += samp2*DrySend[BACK_RIGHT]; } } DataPosFrac += increment; k += DataPosFrac>>FRACTIONBITS; DataPosFrac &= FRACTIONMASK; j++; } DataPosInt += k; //Update source info ALSource->position = DataPosInt; ALSource->position_fraction = DataPosFrac; skipmix: ; } //Handle looping sources if(!Buffer || DataPosInt >= DataSize) { //queueing if(ALSource->queue) { Looping = ALSource->bLooping; if(ALSource->BuffersPlayed < (ALSource->BuffersInQueue-1)) { BufferListItem = ALSource->queue; for(loop = 0; loop <= ALSource->BuffersPlayed; loop++) { if(BufferListItem) { if(!Looping) BufferListItem->bufferstate = PROCESSED; BufferListItem = BufferListItem->next; } } if(BufferListItem) ALSource->ulBufferID = BufferListItem->buffer; ALSource->position = DataPosInt-DataSize; ALSource->position_fraction = DataPosFrac; ALSource->BuffersPlayed++; } else { if(!Looping) { /* alSourceStop */ ALSource->state = AL_STOPPED; ALSource->inuse = AL_FALSE; ALSource->BuffersPlayed = ALSource->BuffersInQueue; BufferListItem = ALSource->queue; while(BufferListItem != NULL) { BufferListItem->bufferstate = PROCESSED; BufferListItem = BufferListItem->next; } ALSource->position = DataSize; ALSource->position_fraction = 0; } else { /* alSourceRewind */ /* alSourcePlay */ ALSource->state = AL_PLAYING; ALSource->inuse = AL_TRUE; ALSource->play = AL_TRUE; ALSource->BuffersPlayed = 0; BufferListItem = ALSource->queue; while(BufferListItem != NULL) { BufferListItem->bufferstate = PENDING; BufferListItem = BufferListItem->next; } ALSource->ulBufferID = ALSource->queue->buffer; if(ALSource->BuffersInQueue == 1) ALSource->position = DataPosInt%DataSize; else ALSource->position = DataPosInt-DataSize; ALSource->position_fraction = DataPosFrac; } } } } //Get source state State = ALSource->state; } ALSource = ALSource->next; } // effect slot processing while(ALEffectSlot) { if(ALEffectSlot->effect.type == AL_EFFECT_REVERB) { ALfloat *DelayBuffer = ALEffectSlot->ReverbBuffer; ALuint Pos = ALEffectSlot->ReverbPos; ALuint LatePos = ALEffectSlot->ReverbLatePos; ALuint ReflectPos = ALEffectSlot->ReverbReflectPos; ALuint Length = ALEffectSlot->ReverbLength; ALfloat DecayGain = ALEffectSlot->ReverbDecayGain; ALfloat DecayHFRatio = ALEffectSlot->effect.Reverb.DecayHFRatio; ALfloat ReflectGain = ALEffectSlot->effect.Reverb.ReflectionsGain; ALfloat LateReverbGain = ALEffectSlot->effect.Reverb.LateReverbGain; ALfloat sample, lowsample; WetFilter = &ALEffectSlot->iirFilter; for(i = 0;i < SamplesToDo;i++) { DelayBuffer[Pos] = WetBuffer[i]; sample = DelayBuffer[ReflectPos] * ReflectGain; DelayBuffer[LatePos] *= LateReverbGain; Pos = (Pos+1) % Length; lowsample = lpFilter(WetFilter, DelayBuffer[Pos]); lowsample += (DelayBuffer[Pos]-lowsample) * DecayHFRatio; DelayBuffer[LatePos] += lowsample * DecayGain; sample += DelayBuffer[LatePos]; DryBuffer[i][FRONT_LEFT] += sample; DryBuffer[i][FRONT_RIGHT] += sample; DryBuffer[i][SIDE_LEFT] += sample; DryBuffer[i][SIDE_RIGHT] += sample; DryBuffer[i][BACK_LEFT] += sample; DryBuffer[i][BACK_RIGHT] += sample; LatePos = (LatePos+1) % Length; ReflectPos = (ReflectPos+1) % Length; } ALEffectSlot->ReverbPos = Pos; ALEffectSlot->ReverbLatePos = LatePos; ALEffectSlot->ReverbReflectPos = ReflectPos; } ALEffectSlot = ALEffectSlot->next; } //Post processing loop switch(format) { case AL_FORMAT_MONO8: for(i = 0;i < SamplesToDo;i++) { ((ALubyte*)buffer)[0] = (ALubyte)((aluF2S(DryBuffer[i][FRONT_LEFT]+DryBuffer[i][FRONT_RIGHT])>>8)+128); buffer = ((ALubyte*)buffer) + 1; } break; case AL_FORMAT_STEREO8: if(ALContext && ALContext->bs2b) { for(i = 0;i < SamplesToDo;i++) { float samples[2]; samples[0] = DryBuffer[i][FRONT_LEFT]; samples[1] = DryBuffer[i][FRONT_RIGHT]; bs2b_cross_feed(ALContext->bs2b, samples); ((ALubyte*)buffer)[0] = (ALubyte)((aluF2S(samples[0])>>8)+128); ((ALubyte*)buffer)[1] = (ALubyte)((aluF2S(samples[1])>>8)+128); buffer = ((ALubyte*)buffer) + 2; } } else { for(i = 0;i < SamplesToDo;i++) { ((ALubyte*)buffer)[0] = (ALubyte)((aluF2S(DryBuffer[i][FRONT_LEFT])>>8)+128); ((ALubyte*)buffer)[1] = (ALubyte)((aluF2S(DryBuffer[i][FRONT_RIGHT])>>8)+128); buffer = ((ALubyte*)buffer) + 2; } } break; case AL_FORMAT_QUAD8: for(i = 0;i < SamplesToDo;i++) { ((ALubyte*)buffer)[0] = (ALubyte)((aluF2S(DryBuffer[i][FRONT_LEFT])>>8)+128); ((ALubyte*)buffer)[1] = (ALubyte)((aluF2S(DryBuffer[i][FRONT_RIGHT])>>8)+128); ((ALubyte*)buffer)[2] = (ALubyte)((aluF2S(DryBuffer[i][BACK_LEFT])>>8)+128); ((ALubyte*)buffer)[3] = (ALubyte)((aluF2S(DryBuffer[i][BACK_RIGHT])>>8)+128); buffer = ((ALubyte*)buffer) + 4; } break; case AL_FORMAT_51CHN8: for(i = 0;i < SamplesToDo;i++) { ((ALubyte*)buffer)[0] = (ALubyte)((aluF2S(DryBuffer[i][FRONT_LEFT])>>8)+128); ((ALubyte*)buffer)[1] = (ALubyte)((aluF2S(DryBuffer[i][FRONT_RIGHT])>>8)+128); #ifdef _WIN32 /* Of course, Windows can't use the same ordering... */ ((ALubyte*)buffer)[2] = (ALubyte)((aluF2S(DryBuffer[i][CENTER])>>8)+128); ((ALubyte*)buffer)[3] = (ALubyte)((aluF2S(DryBuffer[i][LFE])>>8)+128); ((ALubyte*)buffer)[4] = (ALubyte)((aluF2S(DryBuffer[i][BACK_LEFT])>>8)+128); ((ALubyte*)buffer)[5] = (ALubyte)((aluF2S(DryBuffer[i][BACK_RIGHT])>>8)+128); #else ((ALubyte*)buffer)[2] = (ALubyte)((aluF2S(DryBuffer[i][BACK_LEFT])>>8)+128); ((ALubyte*)buffer)[3] = (ALubyte)((aluF2S(DryBuffer[i][BACK_RIGHT])>>8)+128); ((ALubyte*)buffer)[4] = (ALubyte)((aluF2S(DryBuffer[i][CENTER])>>8)+128); ((ALubyte*)buffer)[5] = (ALubyte)((aluF2S(DryBuffer[i][LFE])>>8)+128); #endif buffer = ((ALubyte*)buffer) + 6; } break; case AL_FORMAT_61CHN8: for(i = 0;i < SamplesToDo;i++) { ((ALubyte*)buffer)[0] = (ALubyte)((aluF2S(DryBuffer[i][FRONT_LEFT])>>8)+128); ((ALubyte*)buffer)[1] = (ALubyte)((aluF2S(DryBuffer[i][FRONT_RIGHT])>>8)+128); #ifdef _WIN32 ((ALubyte*)buffer)[2] = (ALubyte)((aluF2S(DryBuffer[i][LFE])>>8)+128); ((ALubyte*)buffer)[3] = (ALubyte)((aluF2S(DryBuffer[i][BACK_LEFT])>>8)+128); ((ALubyte*)buffer)[4] = (ALubyte)((aluF2S(DryBuffer[i][BACK_RIGHT])>>8)+128); #else ((ALubyte*)buffer)[2] = (ALubyte)((aluF2S(DryBuffer[i][BACK_LEFT])>>8)+128); ((ALubyte*)buffer)[3] = (ALubyte)((aluF2S(DryBuffer[i][BACK_RIGHT])>>8)+128); ((ALubyte*)buffer)[4] = (ALubyte)((aluF2S(DryBuffer[i][LFE])>>8)+128); #endif ((ALubyte*)buffer)[5] = (ALubyte)((aluF2S(DryBuffer[i][SIDE_LEFT])>>8)+128); ((ALubyte*)buffer)[6] = (ALubyte)((aluF2S(DryBuffer[i][SIDE_RIGHT])>>8)+128); buffer = ((ALubyte*)buffer) + 7; } break; case AL_FORMAT_71CHN8: for(i = 0;i < SamplesToDo;i++) { ((ALubyte*)buffer)[0] = (ALubyte)((aluF2S(DryBuffer[i][FRONT_LEFT])>>8)+128); ((ALubyte*)buffer)[1] = (ALubyte)((aluF2S(DryBuffer[i][FRONT_RIGHT])>>8)+128); #ifdef _WIN32 ((ALubyte*)buffer)[2] = (ALubyte)((aluF2S(DryBuffer[i][CENTER])>>8)+128); ((ALubyte*)buffer)[3] = (ALubyte)((aluF2S(DryBuffer[i][LFE])>>8)+128); ((ALubyte*)buffer)[4] = (ALubyte)((aluF2S(DryBuffer[i][BACK_LEFT])>>8)+128); ((ALubyte*)buffer)[5] = (ALubyte)((aluF2S(DryBuffer[i][BACK_RIGHT])>>8)+128); #else ((ALubyte*)buffer)[2] = (ALubyte)((aluF2S(DryBuffer[i][BACK_LEFT])>>8)+128); ((ALubyte*)buffer)[3] = (ALubyte)((aluF2S(DryBuffer[i][BACK_RIGHT])>>8)+128); ((ALubyte*)buffer)[4] = (ALubyte)((aluF2S(DryBuffer[i][CENTER])>>8)+128); ((ALubyte*)buffer)[5] = (ALubyte)((aluF2S(DryBuffer[i][LFE])>>8)+128); #endif ((ALubyte*)buffer)[6] = (ALubyte)((aluF2S(DryBuffer[i][SIDE_LEFT])>>8)+128); ((ALubyte*)buffer)[7] = (ALubyte)((aluF2S(DryBuffer[i][SIDE_RIGHT])>>8)+128); buffer = ((ALubyte*)buffer) + 8; } break; case AL_FORMAT_MONO16: for(i = 0;i < SamplesToDo;i++) { ((ALshort*)buffer)[0] = aluF2S(DryBuffer[i][FRONT_LEFT]+DryBuffer[i][FRONT_RIGHT]); buffer = ((ALshort*)buffer) + 1; } break; case AL_FORMAT_STEREO16: if(ALContext && ALContext->bs2b) { for(i = 0;i < SamplesToDo;i++) { float samples[2]; samples[0] = DryBuffer[i][FRONT_LEFT]; samples[1] = DryBuffer[i][FRONT_RIGHT]; bs2b_cross_feed(ALContext->bs2b, samples); ((ALshort*)buffer)[0] = aluF2S(samples[0]); ((ALshort*)buffer)[1] = aluF2S(samples[1]); buffer = ((ALshort*)buffer) + 2; } } else { for(i = 0;i < SamplesToDo;i++) { ((ALshort*)buffer)[0] = aluF2S(DryBuffer[i][FRONT_LEFT]); ((ALshort*)buffer)[1] = aluF2S(DryBuffer[i][FRONT_RIGHT]); buffer = ((ALshort*)buffer) + 2; } } break; case AL_FORMAT_QUAD16: for(i = 0;i < SamplesToDo;i++) { ((ALshort*)buffer)[0] = aluF2S(DryBuffer[i][FRONT_LEFT]); ((ALshort*)buffer)[1] = aluF2S(DryBuffer[i][FRONT_RIGHT]); ((ALshort*)buffer)[2] = aluF2S(DryBuffer[i][BACK_LEFT]); ((ALshort*)buffer)[3] = aluF2S(DryBuffer[i][BACK_RIGHT]); buffer = ((ALshort*)buffer) + 4; } break; case AL_FORMAT_51CHN16: for(i = 0;i < SamplesToDo;i++) { ((ALshort*)buffer)[0] = aluF2S(DryBuffer[i][FRONT_LEFT]); ((ALshort*)buffer)[1] = aluF2S(DryBuffer[i][FRONT_RIGHT]); #ifdef _WIN32 ((ALshort*)buffer)[2] = aluF2S(DryBuffer[i][CENTER]); ((ALshort*)buffer)[3] = aluF2S(DryBuffer[i][LFE]); ((ALshort*)buffer)[4] = aluF2S(DryBuffer[i][BACK_LEFT]); ((ALshort*)buffer)[5] = aluF2S(DryBuffer[i][BACK_RIGHT]); #else ((ALshort*)buffer)[2] = aluF2S(DryBuffer[i][BACK_LEFT]); ((ALshort*)buffer)[3] = aluF2S(DryBuffer[i][BACK_RIGHT]); ((ALshort*)buffer)[4] = aluF2S(DryBuffer[i][CENTER]); ((ALshort*)buffer)[5] = aluF2S(DryBuffer[i][LFE]); #endif buffer = ((ALshort*)buffer) + 6; } break; case AL_FORMAT_61CHN16: for(i = 0;i < SamplesToDo;i++) { ((ALshort*)buffer)[0] = aluF2S(DryBuffer[i][FRONT_LEFT]); ((ALshort*)buffer)[1] = aluF2S(DryBuffer[i][FRONT_RIGHT]); #ifdef _WIN32 ((ALshort*)buffer)[2] = aluF2S(DryBuffer[i][LFE]); ((ALshort*)buffer)[3] = aluF2S(DryBuffer[i][BACK_LEFT]); ((ALshort*)buffer)[4] = aluF2S(DryBuffer[i][BACK_RIGHT]); #else ((ALshort*)buffer)[2] = aluF2S(DryBuffer[i][BACK_LEFT]); ((ALshort*)buffer)[3] = aluF2S(DryBuffer[i][BACK_RIGHT]); ((ALshort*)buffer)[4] = aluF2S(DryBuffer[i][LFE]); #endif ((ALshort*)buffer)[5] = aluF2S(DryBuffer[i][SIDE_LEFT]); ((ALshort*)buffer)[6] = aluF2S(DryBuffer[i][SIDE_RIGHT]); buffer = ((ALshort*)buffer) + 7; } break; case AL_FORMAT_71CHN16: for(i = 0;i < SamplesToDo;i++) { ((ALshort*)buffer)[0] = aluF2S(DryBuffer[i][FRONT_LEFT]); ((ALshort*)buffer)[1] = aluF2S(DryBuffer[i][FRONT_RIGHT]); #ifdef _WIN32 ((ALshort*)buffer)[2] = aluF2S(DryBuffer[i][CENTER]); ((ALshort*)buffer)[3] = aluF2S(DryBuffer[i][LFE]); ((ALshort*)buffer)[4] = aluF2S(DryBuffer[i][BACK_LEFT]); ((ALshort*)buffer)[5] = aluF2S(DryBuffer[i][BACK_RIGHT]); #else ((ALshort*)buffer)[2] = aluF2S(DryBuffer[i][BACK_LEFT]); ((ALshort*)buffer)[3] = aluF2S(DryBuffer[i][BACK_RIGHT]); ((ALshort*)buffer)[4] = aluF2S(DryBuffer[i][CENTER]); ((ALshort*)buffer)[5] = aluF2S(DryBuffer[i][LFE]); #endif ((ALshort*)buffer)[6] = aluF2S(DryBuffer[i][SIDE_LEFT]); ((ALshort*)buffer)[7] = aluF2S(DryBuffer[i][SIDE_RIGHT]); buffer = ((ALshort*)buffer) + 8; } break; default: break; } size -= SamplesToDo; } #if defined(HAVE_FESETROUND) fesetround(fpuState); #elif defined(HAVE__CONTROLFP) _controlfp(fpuState, 0xfffff); #endif ProcessContext(ALContext); }