Rename LOWPASSFREQCUTOFF to LOWPASSFREQREF
[openal-soft/android.git] / Alc / ALu.c
blob433e1d2e78d133ec956e567147083903e8f13b26
1 /**
2 * OpenAL cross platform audio library
3 * Copyright (C) 1999-2007 by authors.
4 * This library is free software; you can redistribute it and/or
5 * modify it under the terms of the GNU Library General Public
6 * License as published by the Free Software Foundation; either
7 * version 2 of the License, or (at your option) any later version.
9 * This library is distributed in the hope that it will be useful,
10 * but WITHOUT ANY WARRANTY; without even the implied warranty of
11 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
12 * Library General Public License for more details.
14 * You should have received a copy of the GNU Library General Public
15 * License along with this library; if not, write to the
16 * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
17 * Boston, MA 02111-1307, USA.
18 * Or go to http://www.gnu.org/copyleft/lgpl.html
21 #include "config.h"
23 #include <math.h>
24 #include <stdlib.h>
25 #include <string.h>
26 #include <ctype.h>
27 #include <assert.h>
29 #include "alMain.h"
30 #include "AL/al.h"
31 #include "AL/alc.h"
32 #include "alSource.h"
33 #include "alBuffer.h"
34 #include "alListener.h"
35 #include "alAuxEffectSlot.h"
36 #include "alu.h"
37 #include "bs2b.h"
40 /* Cone scalar */
41 ALfloat ConeScale = 0.5f;
43 /* Localized Z scalar for mono sources */
44 ALfloat ZScale = 1.0f;
47 static __inline ALvoid aluCrossproduct(const ALfloat *inVector1, const ALfloat *inVector2, ALfloat *outVector)
49 outVector[0] = inVector1[1]*inVector2[2] - inVector1[2]*inVector2[1];
50 outVector[1] = inVector1[2]*inVector2[0] - inVector1[0]*inVector2[2];
51 outVector[2] = inVector1[0]*inVector2[1] - inVector1[1]*inVector2[0];
54 static __inline ALfloat aluDotproduct(const ALfloat *inVector1, const ALfloat *inVector2)
56 return inVector1[0]*inVector2[0] + inVector1[1]*inVector2[1] +
57 inVector1[2]*inVector2[2];
60 static __inline ALvoid aluNormalize(ALfloat *inVector)
62 ALfloat length, inverse_length;
64 length = aluSqrt(aluDotproduct(inVector, inVector));
65 if(length != 0.0f)
67 inverse_length = 1.0f/length;
68 inVector[0] *= inverse_length;
69 inVector[1] *= inverse_length;
70 inVector[2] *= inverse_length;
74 static __inline ALvoid aluMatrixVector(ALfloat *vector,ALfloat w,ALfloat matrix[4][4])
76 ALfloat temp[4] = {
77 vector[0], vector[1], vector[2], w
80 vector[0] = temp[0]*matrix[0][0] + temp[1]*matrix[1][0] + temp[2]*matrix[2][0] + temp[3]*matrix[3][0];
81 vector[1] = temp[0]*matrix[0][1] + temp[1]*matrix[1][1] + temp[2]*matrix[2][1] + temp[3]*matrix[3][1];
82 vector[2] = temp[0]*matrix[0][2] + temp[1]*matrix[1][2] + temp[2]*matrix[2][2] + temp[3]*matrix[3][2];
86 ALvoid CalcNonAttnSourceParams(ALsource *ALSource, const ALCcontext *ALContext)
88 static const ALfloat angles_Mono[1] = { 0.0f };
89 static const ALfloat angles_Stereo[2] = { -30.0f, 30.0f };
90 static const ALfloat angles_Rear[2] = { -150.0f, 150.0f };
91 static const ALfloat angles_Quad[4] = { -45.0f, 45.0f, -135.0f, 135.0f };
92 static const ALfloat angles_X51[6] = { -30.0f, 30.0f, 0.0f, 0.0f,
93 -110.0f, 110.0f };
94 static const ALfloat angles_X61[7] = { -30.0f, 30.0f, 0.0f, 0.0f,
95 180.0f, -90.0f, 90.0f };
96 static const ALfloat angles_X71[8] = { -30.0f, 30.0f, 0.0f, 0.0f,
97 -110.0f, 110.0f, -90.0f, 90.0f };
99 static const enum Channel chans_Mono[1] = { FRONT_CENTER };
100 static const enum Channel chans_Stereo[2] = { FRONT_LEFT, FRONT_RIGHT };
101 static const enum Channel chans_Rear[2] = { BACK_LEFT, BACK_RIGHT };
102 static const enum Channel chans_Quad[4] = { FRONT_LEFT, FRONT_RIGHT,
103 BACK_LEFT, BACK_RIGHT };
104 static const enum Channel chans_X51[6] = { FRONT_LEFT, FRONT_RIGHT,
105 FRONT_CENTER, LFE,
106 BACK_LEFT, BACK_RIGHT };
107 static const enum Channel chans_X61[7] = { FRONT_LEFT, FRONT_RIGHT,
108 FRONT_CENTER, LFE, BACK_CENTER,
109 SIDE_LEFT, SIDE_RIGHT };
110 static const enum Channel chans_X71[8] = { FRONT_LEFT, FRONT_RIGHT,
111 FRONT_CENTER, LFE,
112 BACK_LEFT, BACK_RIGHT,
113 SIDE_LEFT, SIDE_RIGHT };
115 ALCdevice *Device = ALContext->Device;
116 ALfloat SourceVolume,ListenerGain,MinVolume,MaxVolume;
117 ALbufferlistitem *BufferListItem;
118 enum DevFmtChannels DevChans;
119 enum FmtChannels Channels;
120 ALfloat (*SrcMatrix)[MAXCHANNELS];
121 ALfloat DryGain, DryGainHF;
122 ALfloat WetGain[MAX_SENDS];
123 ALfloat WetGainHF[MAX_SENDS];
124 ALint NumSends, Frequency;
125 const ALfloat *SpeakerGain;
126 const ALfloat *angles = NULL;
127 const enum Channel *chans = NULL;
128 enum Resampler Resampler;
129 ALint num_channels = 0;
130 ALboolean VirtualChannels;
131 ALfloat Pitch;
132 ALfloat cw;
133 ALuint pos;
134 ALint i, c;
136 /* Get device properties */
137 DevChans = ALContext->Device->FmtChans;
138 NumSends = ALContext->Device->NumAuxSends;
139 Frequency = ALContext->Device->Frequency;
141 /* Get listener properties */
142 ListenerGain = ALContext->Listener.Gain;
144 /* Get source properties */
145 SourceVolume = ALSource->flGain;
146 MinVolume = ALSource->flMinGain;
147 MaxVolume = ALSource->flMaxGain;
148 Pitch = ALSource->flPitch;
149 Resampler = ALSource->Resampler;
150 VirtualChannels = ALSource->VirtualChannels;
152 /* Calculate the stepping value */
153 Channels = FmtMono;
154 BufferListItem = ALSource->queue;
155 while(BufferListItem != NULL)
157 ALbuffer *ALBuffer;
158 if((ALBuffer=BufferListItem->buffer) != NULL)
160 ALint maxstep = STACK_DATA_SIZE / ALSource->NumChannels /
161 ALSource->SampleSize;
162 maxstep -= ResamplerPadding[Resampler] +
163 ResamplerPrePadding[Resampler] + 1;
164 maxstep = mini(maxstep, INT_MAX>>FRACTIONBITS);
166 Pitch = Pitch * ALBuffer->Frequency / Frequency;
167 if(Pitch > (ALfloat)maxstep)
168 ALSource->Params.Step = maxstep<<FRACTIONBITS;
169 else
171 ALSource->Params.Step = fastf2i(Pitch*FRACTIONONE);
172 if(ALSource->Params.Step == 0)
173 ALSource->Params.Step = 1;
176 Channels = ALBuffer->FmtChannels;
178 if(ALSource->VirtualChannels && (Device->Flags&DEVICE_USE_HRTF))
179 ALSource->Params.DoMix = SelectHrtfMixer(ALBuffer,
180 (ALSource->Params.Step==FRACTIONONE) ? POINT_RESAMPLER :
181 Resampler);
182 else
183 ALSource->Params.DoMix = SelectMixer(ALBuffer,
184 (ALSource->Params.Step==FRACTIONONE) ? POINT_RESAMPLER :
185 Resampler);
186 break;
188 BufferListItem = BufferListItem->next;
191 /* Calculate gains */
192 DryGain = clampf(SourceVolume, MinVolume, MaxVolume);
193 DryGain *= ALSource->DirectGain;
194 DryGainHF = ALSource->DirectGainHF;
195 for(i = 0;i < NumSends;i++)
197 WetGain[i] = clampf(SourceVolume, MinVolume, MaxVolume);
198 WetGain[i] *= ALSource->Send[i].WetGain;
199 WetGainHF[i] = ALSource->Send[i].WetGainHF;
202 SrcMatrix = ALSource->Params.DryGains;
203 for(i = 0;i < MAXCHANNELS;i++)
205 for(c = 0;c < MAXCHANNELS;c++)
206 SrcMatrix[i][c] = 0.0f;
208 switch(Channels)
210 case FmtMono:
211 angles = angles_Mono;
212 chans = chans_Mono;
213 num_channels = 1;
214 break;
215 case FmtStereo:
216 if(VirtualChannels && (ALContext->Device->Flags&DEVICE_DUPLICATE_STEREO))
218 DryGain *= aluSqrt(2.0f/4.0f);
219 for(c = 0;c < 2;c++)
221 pos = aluCart2LUTpos(aluCos(F_PI/180.0f * angles_Rear[c]),
222 aluSin(F_PI/180.0f * angles_Rear[c]));
223 SpeakerGain = Device->PanningLUT[pos];
225 for(i = 0;i < (ALint)Device->NumChan;i++)
227 enum Channel chan = Device->Speaker2Chan[i];
228 SrcMatrix[c][chan] += DryGain * ListenerGain *
229 SpeakerGain[chan];
233 angles = angles_Stereo;
234 chans = chans_Stereo;
235 num_channels = 2;
236 break;
238 case FmtRear:
239 angles = angles_Rear;
240 chans = chans_Rear;
241 num_channels = 2;
242 break;
244 case FmtQuad:
245 angles = angles_Quad;
246 chans = chans_Quad;
247 num_channels = 4;
248 break;
250 case FmtX51:
251 angles = angles_X51;
252 chans = chans_X51;
253 num_channels = 6;
254 break;
256 case FmtX61:
257 angles = angles_X61;
258 chans = chans_X61;
259 num_channels = 7;
260 break;
262 case FmtX71:
263 angles = angles_X71;
264 chans = chans_X71;
265 num_channels = 8;
266 break;
269 if(VirtualChannels == AL_FALSE)
271 for(c = 0;c < num_channels;c++)
272 SrcMatrix[c][chans[c]] += DryGain * ListenerGain;
274 else if((Device->Flags&DEVICE_USE_HRTF))
276 for(c = 0;c < num_channels;c++)
278 if(chans[c] == LFE)
280 /* Skip LFE */
281 ALSource->Params.HrtfDelay[c][0] = 0;
282 ALSource->Params.HrtfDelay[c][1] = 0;
283 for(i = 0;i < HRIR_LENGTH;i++)
285 ALSource->Params.HrtfCoeffs[c][i][0] = 0.0f;
286 ALSource->Params.HrtfCoeffs[c][i][1] = 0.0f;
289 else
291 /* Get the static HRIR coefficients and delays for this
292 * channel. */
293 GetLerpedHrtfCoeffs(ALContext->Device->Hrtf,
294 0.0f, F_PI/180.0f * angles[c],
295 DryGain*ListenerGain,
296 ALSource->Params.HrtfCoeffs[c],
297 ALSource->Params.HrtfDelay[c]);
299 ALSource->HrtfCounter = 0;
302 else
304 for(c = 0;c < num_channels;c++)
306 if(chans[c] == LFE) /* Special-case LFE */
308 SrcMatrix[c][LFE] += DryGain * ListenerGain;
309 continue;
311 pos = aluCart2LUTpos(aluCos(F_PI/180.0f * angles[c]),
312 aluSin(F_PI/180.0f * angles[c]));
313 SpeakerGain = Device->PanningLUT[pos];
315 for(i = 0;i < (ALint)Device->NumChan;i++)
317 enum Channel chan = Device->Speaker2Chan[i];
318 SrcMatrix[c][chan] += DryGain * ListenerGain *
319 SpeakerGain[chan];
323 for(i = 0;i < NumSends;i++)
325 ALSource->Params.Send[i].Slot = ALSource->Send[i].Slot;
326 ALSource->Params.Send[i].WetGain = WetGain[i] * ListenerGain;
329 /* Update filter coefficients. Calculations based on the I3DL2
330 * spec. */
331 cw = aluCos(F_PI*2.0f * LOWPASSFREQREF / Frequency);
333 /* We use two chained one-pole filters, so we need to take the
334 * square root of the squared gain, which is the same as the base
335 * gain. */
336 ALSource->Params.iirFilter.coeff = lpCoeffCalc(DryGainHF, cw);
337 for(i = 0;i < NumSends;i++)
339 /* We use a one-pole filter, so we need to take the squared gain */
340 ALfloat a = lpCoeffCalc(WetGainHF[i]*WetGainHF[i], cw);
341 ALSource->Params.Send[i].iirFilter.coeff = a;
345 ALvoid CalcSourceParams(ALsource *ALSource, const ALCcontext *ALContext)
347 const ALCdevice *Device = ALContext->Device;
348 ALfloat InnerAngle,OuterAngle,Angle,Distance,ClampedDist;
349 ALfloat Direction[3],Position[3],SourceToListener[3];
350 ALfloat Velocity[3],ListenerVel[3];
351 ALfloat MinVolume,MaxVolume,MinDist,MaxDist,Rolloff;
352 ALfloat ConeVolume,ConeHF,SourceVolume,ListenerGain;
353 ALfloat DopplerFactor, DopplerVelocity, SpeedOfSound;
354 ALfloat AirAbsorptionFactor;
355 ALfloat RoomAirAbsorption[MAX_SENDS];
356 ALbufferlistitem *BufferListItem;
357 ALfloat Attenuation, EffectiveDist;
358 ALfloat RoomAttenuation[MAX_SENDS];
359 ALfloat MetersPerUnit;
360 ALfloat RoomRolloffBase;
361 ALfloat RoomRolloff[MAX_SENDS];
362 ALfloat DecayDistance[MAX_SENDS];
363 ALfloat DryGain;
364 ALfloat DryGainHF;
365 ALboolean DryGainHFAuto;
366 ALfloat WetGain[MAX_SENDS];
367 ALfloat WetGainHF[MAX_SENDS];
368 ALboolean WetGainAuto;
369 ALboolean WetGainHFAuto;
370 enum Resampler Resampler;
371 ALfloat Pitch;
372 ALuint Frequency;
373 ALint NumSends;
374 ALfloat cw;
375 ALint i;
377 DryGainHF = 1.0f;
378 for(i = 0;i < MAX_SENDS;i++)
379 WetGainHF[i] = 1.0f;
381 //Get context properties
382 DopplerFactor = ALContext->DopplerFactor * ALSource->DopplerFactor;
383 DopplerVelocity = ALContext->DopplerVelocity;
384 SpeedOfSound = ALContext->flSpeedOfSound;
385 NumSends = Device->NumAuxSends;
386 Frequency = Device->Frequency;
388 //Get listener properties
389 ListenerGain = ALContext->Listener.Gain;
390 MetersPerUnit = ALContext->Listener.MetersPerUnit;
391 ListenerVel[0] = ALContext->Listener.Velocity[0];
392 ListenerVel[1] = ALContext->Listener.Velocity[1];
393 ListenerVel[2] = ALContext->Listener.Velocity[2];
395 //Get source properties
396 SourceVolume = ALSource->flGain;
397 MinVolume = ALSource->flMinGain;
398 MaxVolume = ALSource->flMaxGain;
399 Pitch = ALSource->flPitch;
400 Resampler = ALSource->Resampler;
401 Position[0] = ALSource->vPosition[0];
402 Position[1] = ALSource->vPosition[1];
403 Position[2] = ALSource->vPosition[2];
404 Direction[0] = ALSource->vOrientation[0];
405 Direction[1] = ALSource->vOrientation[1];
406 Direction[2] = ALSource->vOrientation[2];
407 Velocity[0] = ALSource->vVelocity[0];
408 Velocity[1] = ALSource->vVelocity[1];
409 Velocity[2] = ALSource->vVelocity[2];
410 MinDist = ALSource->flRefDistance;
411 MaxDist = ALSource->flMaxDistance;
412 Rolloff = ALSource->flRollOffFactor;
413 InnerAngle = ALSource->flInnerAngle * ConeScale;
414 OuterAngle = ALSource->flOuterAngle * ConeScale;
415 AirAbsorptionFactor = ALSource->AirAbsorptionFactor;
416 DryGainHFAuto = ALSource->DryGainHFAuto;
417 WetGainAuto = ALSource->WetGainAuto;
418 WetGainHFAuto = ALSource->WetGainHFAuto;
419 RoomRolloffBase = ALSource->RoomRolloffFactor;
420 for(i = 0;i < NumSends;i++)
422 ALeffectslot *Slot = ALSource->Send[i].Slot;
424 if(!Slot || Slot->effect.type == AL_EFFECT_NULL)
426 RoomRolloff[i] = 0.0f;
427 DecayDistance[i] = 0.0f;
428 RoomAirAbsorption[i] = 1.0f;
430 else if(Slot->AuxSendAuto)
432 RoomRolloff[i] = RoomRolloffBase;
433 if(IsReverbEffect(Slot->effect.type))
435 RoomRolloff[i] += Slot->effect.Reverb.RoomRolloffFactor;
436 DecayDistance[i] = Slot->effect.Reverb.DecayTime *
437 SPEEDOFSOUNDMETRESPERSEC;
438 RoomAirAbsorption[i] = Slot->effect.Reverb.AirAbsorptionGainHF;
440 else
442 DecayDistance[i] = 0.0f;
443 RoomAirAbsorption[i] = 1.0f;
446 else
448 /* If the slot's auxiliary send auto is off, the data sent to the
449 * effect slot is the same as the dry path, sans filter effects */
450 RoomRolloff[i] = Rolloff;
451 DecayDistance[i] = 0.0f;
452 RoomAirAbsorption[i] = AIRABSORBGAINHF;
455 ALSource->Params.Send[i].Slot = Slot;
458 //1. Translate Listener to origin (convert to head relative)
459 if(ALSource->bHeadRelative == AL_FALSE)
461 ALfloat U[3],V[3],N[3];
462 ALfloat Matrix[4][4];
464 // Build transform matrix
465 N[0] = ALContext->Listener.Forward[0]; // At-vector
466 N[1] = ALContext->Listener.Forward[1];
467 N[2] = ALContext->Listener.Forward[2];
468 aluNormalize(N); // Normalized At-vector
469 V[0] = ALContext->Listener.Up[0]; // Up-vector
470 V[1] = ALContext->Listener.Up[1];
471 V[2] = ALContext->Listener.Up[2];
472 aluNormalize(V); // Normalized Up-vector
473 aluCrossproduct(N, V, U); // Right-vector
474 aluNormalize(U); // Normalized Right-vector
475 Matrix[0][0] = U[0]; Matrix[0][1] = V[0]; Matrix[0][2] = -N[0]; Matrix[0][3] = 0.0f;
476 Matrix[1][0] = U[1]; Matrix[1][1] = V[1]; Matrix[1][2] = -N[1]; Matrix[1][3] = 0.0f;
477 Matrix[2][0] = U[2]; Matrix[2][1] = V[2]; Matrix[2][2] = -N[2]; Matrix[2][3] = 0.0f;
478 Matrix[3][0] = 0.0f; Matrix[3][1] = 0.0f; Matrix[3][2] = 0.0f; Matrix[3][3] = 1.0f;
480 // Translate position
481 Position[0] -= ALContext->Listener.Position[0];
482 Position[1] -= ALContext->Listener.Position[1];
483 Position[2] -= ALContext->Listener.Position[2];
485 // Transform source position and direction into listener space
486 aluMatrixVector(Position, 1.0f, Matrix);
487 aluMatrixVector(Direction, 0.0f, Matrix);
488 // Transform source and listener velocity into listener space
489 aluMatrixVector(Velocity, 0.0f, Matrix);
490 aluMatrixVector(ListenerVel, 0.0f, Matrix);
492 else
493 ListenerVel[0] = ListenerVel[1] = ListenerVel[2] = 0.0f;
495 SourceToListener[0] = -Position[0];
496 SourceToListener[1] = -Position[1];
497 SourceToListener[2] = -Position[2];
498 aluNormalize(SourceToListener);
499 aluNormalize(Direction);
501 //2. Calculate distance attenuation
502 Distance = aluSqrt(aluDotproduct(Position, Position));
503 ClampedDist = Distance;
505 Attenuation = 1.0f;
506 for(i = 0;i < NumSends;i++)
507 RoomAttenuation[i] = 1.0f;
508 switch(ALContext->SourceDistanceModel ? ALSource->DistanceModel :
509 ALContext->DistanceModel)
511 case InverseDistanceClamped:
512 ClampedDist = clampf(ClampedDist, MinDist, MaxDist);
513 if(MaxDist < MinDist)
514 break;
515 //fall-through
516 case InverseDistance:
517 if(MinDist > 0.0f)
519 if((MinDist + (Rolloff * (ClampedDist - MinDist))) > 0.0f)
520 Attenuation = MinDist / (MinDist + (Rolloff * (ClampedDist - MinDist)));
521 for(i = 0;i < NumSends;i++)
523 if((MinDist + (RoomRolloff[i] * (ClampedDist - MinDist))) > 0.0f)
524 RoomAttenuation[i] = MinDist / (MinDist + (RoomRolloff[i] * (ClampedDist - MinDist)));
527 break;
529 case LinearDistanceClamped:
530 ClampedDist = clampf(ClampedDist, MinDist, MaxDist);
531 if(MaxDist < MinDist)
532 break;
533 //fall-through
534 case LinearDistance:
535 if(MaxDist != MinDist)
537 Attenuation = 1.0f - (Rolloff*(ClampedDist-MinDist)/(MaxDist - MinDist));
538 Attenuation = maxf(Attenuation, 0.0f);
539 for(i = 0;i < NumSends;i++)
541 RoomAttenuation[i] = 1.0f - (RoomRolloff[i]*(ClampedDist-MinDist)/(MaxDist - MinDist));
542 RoomAttenuation[i] = maxf(RoomAttenuation[i], 0.0f);
545 break;
547 case ExponentDistanceClamped:
548 ClampedDist = clampf(ClampedDist, MinDist, MaxDist);
549 if(MaxDist < MinDist)
550 break;
551 //fall-through
552 case ExponentDistance:
553 if(ClampedDist > 0.0f && MinDist > 0.0f)
555 Attenuation = aluPow(ClampedDist/MinDist, -Rolloff);
556 for(i = 0;i < NumSends;i++)
557 RoomAttenuation[i] = aluPow(ClampedDist/MinDist, -RoomRolloff[i]);
559 break;
561 case DisableDistance:
562 break;
565 // Source Gain + Attenuation
566 DryGain = SourceVolume * Attenuation;
567 for(i = 0;i < NumSends;i++)
568 WetGain[i] = SourceVolume * RoomAttenuation[i];
570 // Distance-based air absorption
571 EffectiveDist = 0.0f;
572 if(MinDist > 0.0f && Attenuation < 1.0f)
573 EffectiveDist = (MinDist/Attenuation - MinDist)*MetersPerUnit;
574 if(AirAbsorptionFactor > 0.0f && EffectiveDist > 0.0f)
576 DryGainHF *= aluPow(AIRABSORBGAINHF, AirAbsorptionFactor*EffectiveDist);
577 for(i = 0;i < NumSends;i++)
578 WetGainHF[i] *= aluPow(RoomAirAbsorption[i],
579 AirAbsorptionFactor*EffectiveDist);
582 if(WetGainAuto)
584 /* Apply a decay-time transformation to the wet path, based on the
585 * attenuation of the dry path.
587 * Using the approximate (effective) source to listener distance, the
588 * initial decay of the reverb effect is calculated and applied to the
589 * wet path.
591 for(i = 0;i < NumSends;i++)
593 if(DecayDistance[i] > 0.0f)
594 WetGain[i] *= aluPow(0.001f /* -60dB */,
595 EffectiveDist / DecayDistance[i]);
599 /* Calculate directional soundcones */
600 Angle = aluAcos(aluDotproduct(Direction,SourceToListener)) * (180.0f/F_PI);
601 if(Angle >= InnerAngle && Angle <= OuterAngle)
603 ALfloat scale = (Angle-InnerAngle) / (OuterAngle-InnerAngle);
604 ConeVolume = lerp(1.0f, ALSource->flOuterGain, scale);
605 ConeHF = lerp(1.0f, ALSource->OuterGainHF, scale);
607 else if(Angle > OuterAngle)
609 ConeVolume = ALSource->flOuterGain;
610 ConeHF = ALSource->OuterGainHF;
612 else
614 ConeVolume = 1.0f;
615 ConeHF = 1.0f;
618 DryGain *= ConeVolume;
619 if(WetGainAuto)
621 for(i = 0;i < NumSends;i++)
622 WetGain[i] *= ConeVolume;
624 if(DryGainHFAuto)
625 DryGainHF *= ConeHF;
626 if(WetGainHFAuto)
628 for(i = 0;i < NumSends;i++)
629 WetGainHF[i] *= ConeHF;
632 // Clamp to Min/Max Gain
633 DryGain = clampf(DryGain, MinVolume, MaxVolume);
634 for(i = 0;i < NumSends;i++)
635 WetGain[i] = clampf(WetGain[i], MinVolume, MaxVolume);
637 // Apply filter gains and filters
638 DryGain *= ALSource->DirectGain * ListenerGain;
639 DryGainHF *= ALSource->DirectGainHF;
640 for(i = 0;i < NumSends;i++)
642 WetGain[i] *= ALSource->Send[i].WetGain * ListenerGain;
643 WetGainHF[i] *= ALSource->Send[i].WetGainHF;
646 // Calculate Velocity
647 if(DopplerFactor != 0.0f)
649 ALfloat VSS, VLS;
650 ALfloat MaxVelocity = (SpeedOfSound*DopplerVelocity) /
651 DopplerFactor;
653 VSS = aluDotproduct(Velocity, SourceToListener);
654 if(VSS >= MaxVelocity)
655 VSS = (MaxVelocity - 1.0f);
656 else if(VSS <= -MaxVelocity)
657 VSS = -MaxVelocity + 1.0f;
659 VLS = aluDotproduct(ListenerVel, SourceToListener);
660 if(VLS >= MaxVelocity)
661 VLS = (MaxVelocity - 1.0f);
662 else if(VLS <= -MaxVelocity)
663 VLS = -MaxVelocity + 1.0f;
665 Pitch *= ((SpeedOfSound*DopplerVelocity) - (DopplerFactor*VLS)) /
666 ((SpeedOfSound*DopplerVelocity) - (DopplerFactor*VSS));
669 BufferListItem = ALSource->queue;
670 while(BufferListItem != NULL)
672 ALbuffer *ALBuffer;
673 if((ALBuffer=BufferListItem->buffer) != NULL)
675 ALint maxstep = STACK_DATA_SIZE / ALSource->NumChannels /
676 ALSource->SampleSize;
677 maxstep -= ResamplerPadding[Resampler] +
678 ResamplerPrePadding[Resampler] + 1;
679 maxstep = mini(maxstep, INT_MAX>>FRACTIONBITS);
681 Pitch = Pitch * ALBuffer->Frequency / Frequency;
682 if(Pitch > (ALfloat)maxstep)
683 ALSource->Params.Step = maxstep<<FRACTIONBITS;
684 else
686 ALSource->Params.Step = fastf2i(Pitch*FRACTIONONE);
687 if(ALSource->Params.Step == 0)
688 ALSource->Params.Step = 1;
691 if((Device->Flags&DEVICE_USE_HRTF))
692 ALSource->Params.DoMix = SelectHrtfMixer(ALBuffer,
693 (ALSource->Params.Step==FRACTIONONE) ? POINT_RESAMPLER :
694 Resampler);
695 else
696 ALSource->Params.DoMix = SelectMixer(ALBuffer,
697 (ALSource->Params.Step==FRACTIONONE) ? POINT_RESAMPLER :
698 Resampler);
699 break;
701 BufferListItem = BufferListItem->next;
704 if((Device->Flags&DEVICE_USE_HRTF))
706 // Use a binaural HRTF algorithm for stereo headphone playback
707 ALfloat delta, ev = 0.0f, az = 0.0f;
709 if(Distance > 0.0f)
711 ALfloat invlen = 1.0f/Distance;
712 Position[0] *= invlen;
713 Position[1] *= invlen;
714 Position[2] *= invlen;
716 // Calculate elevation and azimuth only when the source is not at
717 // the listener. This prevents +0 and -0 Z from producing
718 // inconsistent panning.
719 ev = aluAsin(Position[1]);
720 az = aluAtan2(Position[0], -Position[2]*ZScale);
723 // Check to see if the HRIR is already moving.
724 if(ALSource->HrtfMoving)
726 // Calculate the normalized HRTF transition factor (delta).
727 delta = CalcHrtfDelta(ALSource->Params.HrtfGain, DryGain,
728 ALSource->Params.HrtfDir, Position);
729 // If the delta is large enough, get the moving HRIR target
730 // coefficients, target delays, steppping values, and counter.
731 if(delta > 0.001f)
733 ALSource->HrtfCounter = GetMovingHrtfCoeffs(Device->Hrtf,
734 ev, az, DryGain, delta,
735 ALSource->HrtfCounter,
736 ALSource->Params.HrtfCoeffs[0],
737 ALSource->Params.HrtfDelay[0],
738 ALSource->Params.HrtfCoeffStep,
739 ALSource->Params.HrtfDelayStep);
740 ALSource->Params.HrtfGain = DryGain;
741 ALSource->Params.HrtfDir[0] = Position[0];
742 ALSource->Params.HrtfDir[1] = Position[1];
743 ALSource->Params.HrtfDir[2] = Position[2];
746 else
748 // Get the initial (static) HRIR coefficients and delays.
749 GetLerpedHrtfCoeffs(Device->Hrtf, ev, az, DryGain,
750 ALSource->Params.HrtfCoeffs[0],
751 ALSource->Params.HrtfDelay[0]);
752 ALSource->HrtfCounter = 0;
753 ALSource->Params.HrtfGain = DryGain;
754 ALSource->Params.HrtfDir[0] = Position[0];
755 ALSource->Params.HrtfDir[1] = Position[1];
756 ALSource->Params.HrtfDir[2] = Position[2];
759 else
761 // Use energy-preserving panning algorithm for multi-speaker playback
762 ALfloat DirGain, AmbientGain;
763 const ALfloat *SpeakerGain;
764 ALfloat length;
765 ALint pos;
767 length = maxf(Distance, MinDist);
768 if(length > 0.0f)
770 ALfloat invlen = 1.0f/length;
771 Position[0] *= invlen;
772 Position[1] *= invlen;
773 Position[2] *= invlen;
776 pos = aluCart2LUTpos(-Position[2]*ZScale, Position[0]);
777 SpeakerGain = Device->PanningLUT[pos];
779 DirGain = aluSqrt(Position[0]*Position[0] + Position[2]*Position[2]);
780 // elevation adjustment for directional gain. this sucks, but
781 // has low complexity
782 AmbientGain = aluSqrt(1.0f/Device->NumChan);
783 for(i = 0;i < MAXCHANNELS;i++)
785 ALuint i2;
786 for(i2 = 0;i2 < MAXCHANNELS;i2++)
787 ALSource->Params.DryGains[i][i2] = 0.0f;
789 for(i = 0;i < (ALint)Device->NumChan;i++)
791 enum Channel chan = Device->Speaker2Chan[i];
792 ALfloat gain = lerp(AmbientGain, SpeakerGain[chan], DirGain);
793 ALSource->Params.DryGains[0][chan] = DryGain * gain;
796 for(i = 0;i < NumSends;i++)
797 ALSource->Params.Send[i].WetGain = WetGain[i];
799 /* Update filter coefficients. */
800 cw = aluCos(F_PI*2.0f * LOWPASSFREQREF / Frequency);
802 ALSource->Params.iirFilter.coeff = lpCoeffCalc(DryGainHF, cw);
803 for(i = 0;i < NumSends;i++)
805 ALfloat a = lpCoeffCalc(WetGainHF[i]*WetGainHF[i], cw);
806 ALSource->Params.Send[i].iirFilter.coeff = a;
811 static __inline ALfloat aluF2F(ALfloat val)
812 { return val; }
813 static __inline ALshort aluF2S(ALfloat val)
815 if(val > 1.0f) return 32767;
816 if(val < -1.0f) return -32768;
817 return fastf2i(val*32767.0f);
819 static __inline ALushort aluF2US(ALfloat val)
820 { return aluF2S(val)+32768; }
821 static __inline ALbyte aluF2B(ALfloat val)
822 { return aluF2S(val)>>8; }
823 static __inline ALubyte aluF2UB(ALfloat val)
824 { return aluF2US(val)>>8; }
826 #define DECL_TEMPLATE(T, N, func) \
827 static void Write_##T##_##N(ALCdevice *device, T *RESTRICT buffer, \
828 ALuint SamplesToDo) \
830 ALfloat (*RESTRICT DryBuffer)[MAXCHANNELS] = device->DryBuffer; \
831 const enum Channel *ChanMap = device->DevChannels; \
832 ALuint i, j; \
834 for(i = 0;i < SamplesToDo;i++) \
836 for(j = 0;j < N;j++) \
837 *(buffer++) = func(DryBuffer[i][ChanMap[j]]); \
841 DECL_TEMPLATE(ALfloat, 1, aluF2F)
842 DECL_TEMPLATE(ALfloat, 4, aluF2F)
843 DECL_TEMPLATE(ALfloat, 6, aluF2F)
844 DECL_TEMPLATE(ALfloat, 7, aluF2F)
845 DECL_TEMPLATE(ALfloat, 8, aluF2F)
847 DECL_TEMPLATE(ALushort, 1, aluF2US)
848 DECL_TEMPLATE(ALushort, 4, aluF2US)
849 DECL_TEMPLATE(ALushort, 6, aluF2US)
850 DECL_TEMPLATE(ALushort, 7, aluF2US)
851 DECL_TEMPLATE(ALushort, 8, aluF2US)
853 DECL_TEMPLATE(ALshort, 1, aluF2S)
854 DECL_TEMPLATE(ALshort, 4, aluF2S)
855 DECL_TEMPLATE(ALshort, 6, aluF2S)
856 DECL_TEMPLATE(ALshort, 7, aluF2S)
857 DECL_TEMPLATE(ALshort, 8, aluF2S)
859 DECL_TEMPLATE(ALubyte, 1, aluF2UB)
860 DECL_TEMPLATE(ALubyte, 4, aluF2UB)
861 DECL_TEMPLATE(ALubyte, 6, aluF2UB)
862 DECL_TEMPLATE(ALubyte, 7, aluF2UB)
863 DECL_TEMPLATE(ALubyte, 8, aluF2UB)
865 DECL_TEMPLATE(ALbyte, 1, aluF2B)
866 DECL_TEMPLATE(ALbyte, 4, aluF2B)
867 DECL_TEMPLATE(ALbyte, 6, aluF2B)
868 DECL_TEMPLATE(ALbyte, 7, aluF2B)
869 DECL_TEMPLATE(ALbyte, 8, aluF2B)
871 #undef DECL_TEMPLATE
873 #define DECL_TEMPLATE(T, N, func) \
874 static void Write_##T##_##N(ALCdevice *device, T *RESTRICT buffer, \
875 ALuint SamplesToDo) \
877 ALfloat (*RESTRICT DryBuffer)[MAXCHANNELS] = device->DryBuffer; \
878 const enum Channel *ChanMap = device->DevChannels; \
879 ALuint i, j; \
881 if(device->Bs2b) \
883 for(i = 0;i < SamplesToDo;i++) \
885 float samples[2]; \
886 samples[0] = DryBuffer[i][ChanMap[0]]; \
887 samples[1] = DryBuffer[i][ChanMap[1]]; \
888 bs2b_cross_feed(device->Bs2b, samples); \
889 *(buffer++) = func(samples[0]); \
890 *(buffer++) = func(samples[1]); \
893 else \
895 for(i = 0;i < SamplesToDo;i++) \
897 for(j = 0;j < N;j++) \
898 *(buffer++) = func(DryBuffer[i][ChanMap[j]]); \
903 DECL_TEMPLATE(ALfloat, 2, aluF2F)
904 DECL_TEMPLATE(ALushort, 2, aluF2US)
905 DECL_TEMPLATE(ALshort, 2, aluF2S)
906 DECL_TEMPLATE(ALubyte, 2, aluF2UB)
907 DECL_TEMPLATE(ALbyte, 2, aluF2B)
909 #undef DECL_TEMPLATE
911 #define DECL_TEMPLATE(T) \
912 static void Write_##T(ALCdevice *device, T *buffer, ALuint SamplesToDo) \
914 switch(device->FmtChans) \
916 case DevFmtMono: \
917 Write_##T##_1(device, buffer, SamplesToDo); \
918 break; \
919 case DevFmtStereo: \
920 Write_##T##_2(device, buffer, SamplesToDo); \
921 break; \
922 case DevFmtQuad: \
923 Write_##T##_4(device, buffer, SamplesToDo); \
924 break; \
925 case DevFmtX51: \
926 case DevFmtX51Side: \
927 Write_##T##_6(device, buffer, SamplesToDo); \
928 break; \
929 case DevFmtX61: \
930 Write_##T##_7(device, buffer, SamplesToDo); \
931 break; \
932 case DevFmtX71: \
933 Write_##T##_8(device, buffer, SamplesToDo); \
934 break; \
938 DECL_TEMPLATE(ALfloat)
939 DECL_TEMPLATE(ALushort)
940 DECL_TEMPLATE(ALshort)
941 DECL_TEMPLATE(ALubyte)
942 DECL_TEMPLATE(ALbyte)
944 #undef DECL_TEMPLATE
946 ALvoid aluMixData(ALCdevice *device, ALvoid *buffer, ALsizei size)
948 ALuint SamplesToDo;
949 ALeffectslot **slot, **slot_end;
950 ALsource **src, **src_end;
951 ALCcontext *ctx;
952 int fpuState;
953 ALuint i, c;
955 fpuState = SetMixerFPUMode();
957 while(size > 0)
959 /* Setup variables */
960 SamplesToDo = minu(size, BUFFERSIZE);
962 /* Clear mixing buffer */
963 memset(device->DryBuffer, 0, SamplesToDo*MAXCHANNELS*sizeof(ALfloat));
965 LockDevice(device);
966 ctx = device->ContextList;
967 while(ctx)
969 ALenum DeferUpdates = ctx->DeferUpdates;
970 ALenum UpdateSources = AL_FALSE;
972 if(!DeferUpdates)
973 UpdateSources = ExchangeInt(&ctx->UpdateSources, AL_FALSE);
975 src = ctx->ActiveSources;
976 src_end = src + ctx->ActiveSourceCount;
977 while(src != src_end)
979 if((*src)->state != AL_PLAYING)
981 --(ctx->ActiveSourceCount);
982 *src = *(--src_end);
983 continue;
986 if(!DeferUpdates && (ExchangeInt(&(*src)->NeedsUpdate, AL_FALSE) ||
987 UpdateSources))
988 ALsource_Update(*src, ctx);
990 MixSource(*src, device, SamplesToDo);
991 src++;
994 /* effect slot processing */
995 slot = ctx->ActiveEffectSlots;
996 slot_end = slot + ctx->ActiveEffectSlotCount;
997 while(slot != slot_end)
999 for(c = 0;c < SamplesToDo;c++)
1001 (*slot)->WetBuffer[c] += (*slot)->ClickRemoval[0];
1002 (*slot)->ClickRemoval[0] -= (*slot)->ClickRemoval[0] * (1.0f/256.0f);
1004 (*slot)->ClickRemoval[0] += (*slot)->PendingClicks[0];
1005 (*slot)->PendingClicks[0] = 0.0f;
1007 if(!DeferUpdates && ExchangeInt(&(*slot)->NeedsUpdate, AL_FALSE))
1008 ALeffectState_Update((*slot)->EffectState, ctx, *slot);
1010 ALeffectState_Process((*slot)->EffectState, SamplesToDo,
1011 (*slot)->WetBuffer, device->DryBuffer);
1013 for(i = 0;i < SamplesToDo;i++)
1014 (*slot)->WetBuffer[i] = 0.0f;
1016 slot++;
1019 ctx = ctx->next;
1021 UnlockDevice(device);
1023 //Post processing loop
1024 if(device->FmtChans == DevFmtMono)
1026 for(i = 0;i < SamplesToDo;i++)
1028 device->DryBuffer[i][FRONT_CENTER] += device->ClickRemoval[FRONT_CENTER];
1029 device->ClickRemoval[FRONT_CENTER] -= device->ClickRemoval[FRONT_CENTER] * (1.0f/256.0f);
1031 device->ClickRemoval[FRONT_CENTER] += device->PendingClicks[FRONT_CENTER];
1032 device->PendingClicks[FRONT_CENTER] = 0.0f;
1034 else if(device->FmtChans == DevFmtStereo)
1036 /* Assumes the first two channels are FRONT_LEFT and FRONT_RIGHT */
1037 for(i = 0;i < SamplesToDo;i++)
1039 for(c = 0;c < 2;c++)
1041 device->DryBuffer[i][c] += device->ClickRemoval[c];
1042 device->ClickRemoval[c] -= device->ClickRemoval[c] * (1.0f/256.0f);
1045 for(c = 0;c < 2;c++)
1047 device->ClickRemoval[c] += device->PendingClicks[c];
1048 device->PendingClicks[c] = 0.0f;
1051 else
1053 for(i = 0;i < SamplesToDo;i++)
1055 for(c = 0;c < MAXCHANNELS;c++)
1057 device->DryBuffer[i][c] += device->ClickRemoval[c];
1058 device->ClickRemoval[c] -= device->ClickRemoval[c] * (1.0f/256.0f);
1061 for(c = 0;c < MAXCHANNELS;c++)
1063 device->ClickRemoval[c] += device->PendingClicks[c];
1064 device->PendingClicks[c] = 0.0f;
1068 if(buffer)
1070 switch(device->FmtType)
1072 case DevFmtByte:
1073 Write_ALbyte(device, buffer, SamplesToDo);
1074 break;
1075 case DevFmtUByte:
1076 Write_ALubyte(device, buffer, SamplesToDo);
1077 break;
1078 case DevFmtShort:
1079 Write_ALshort(device, buffer, SamplesToDo);
1080 break;
1081 case DevFmtUShort:
1082 Write_ALushort(device, buffer, SamplesToDo);
1083 break;
1084 case DevFmtFloat:
1085 Write_ALfloat(device, buffer, SamplesToDo);
1086 break;
1090 size -= SamplesToDo;
1093 RestoreFPUMode(fpuState);
1097 ALvoid aluHandleDisconnect(ALCdevice *device)
1099 ALCcontext *Context;
1101 LockDevice(device);
1102 device->Connected = ALC_FALSE;
1104 Context = device->ContextList;
1105 while(Context)
1107 ALsource **src, **src_end;
1109 src = Context->ActiveSources;
1110 src_end = src + Context->ActiveSourceCount;
1111 while(src != src_end)
1113 if((*src)->state == AL_PLAYING)
1115 (*src)->state = AL_STOPPED;
1116 (*src)->BuffersPlayed = (*src)->BuffersInQueue;
1117 (*src)->position = 0;
1118 (*src)->position_fraction = 0;
1120 src++;
1122 Context->ActiveSourceCount = 0;
1124 Context = Context->next;
1126 UnlockDevice(device);