Use macros for float-typed PI values, to avoid manual casts everywhere
[openal-soft/android.git] / Alc / ALu.c
blob364c6093ea6d936691239a36b32b6d5c7f5b3e0a
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 static __inline ALvoid aluCrossproduct(const ALfloat *inVector1, const ALfloat *inVector2, ALfloat *outVector)
42 outVector[0] = inVector1[1]*inVector2[2] - inVector1[2]*inVector2[1];
43 outVector[1] = inVector1[2]*inVector2[0] - inVector1[0]*inVector2[2];
44 outVector[2] = inVector1[0]*inVector2[1] - inVector1[1]*inVector2[0];
47 static __inline ALfloat aluDotproduct(const ALfloat *inVector1, const ALfloat *inVector2)
49 return inVector1[0]*inVector2[0] + inVector1[1]*inVector2[1] +
50 inVector1[2]*inVector2[2];
53 static __inline ALvoid aluNormalize(ALfloat *inVector)
55 ALfloat length, inverse_length;
57 length = aluSqrt(aluDotproduct(inVector, inVector));
58 if(length != 0.0f)
60 inverse_length = 1.0f/length;
61 inVector[0] *= inverse_length;
62 inVector[1] *= inverse_length;
63 inVector[2] *= inverse_length;
67 static __inline ALvoid aluMatrixVector(ALfloat *vector,ALfloat w,ALfloat matrix[4][4])
69 ALfloat temp[4] = {
70 vector[0], vector[1], vector[2], w
73 vector[0] = temp[0]*matrix[0][0] + temp[1]*matrix[1][0] + temp[2]*matrix[2][0] + temp[3]*matrix[3][0];
74 vector[1] = temp[0]*matrix[0][1] + temp[1]*matrix[1][1] + temp[2]*matrix[2][1] + temp[3]*matrix[3][1];
75 vector[2] = temp[0]*matrix[0][2] + temp[1]*matrix[1][2] + temp[2]*matrix[2][2] + temp[3]*matrix[3][2];
79 ALvoid CalcNonAttnSourceParams(ALsource *ALSource, const ALCcontext *ALContext)
81 static const ALfloat angles_Mono[1] = { 0.0f };
82 static const ALfloat angles_Stereo[2] = { -30.0f, 30.0f };
83 static const ALfloat angles_Rear[2] = { -150.0f, 150.0f };
84 static const ALfloat angles_Quad[4] = { -45.0f, 45.0f, -135.0f, 135.0f };
85 static const ALfloat angles_X51[6] = { -30.0f, 30.0f, 0.0f, 0.0f,
86 -110.0f, 110.0f };
87 static const ALfloat angles_X61[7] = { -30.0f, 30.0f, 0.0f, 0.0f,
88 180.0f, -90.0f, 90.0f };
89 static const ALfloat angles_X71[8] = { -30.0f, 30.0f, 0.0f, 0.0f,
90 -110.0f, 110.0f, -90.0f, 90.0f };
92 static const enum Channel chans_Mono[1] = { FRONT_CENTER };
93 static const enum Channel chans_Stereo[2] = { FRONT_LEFT, FRONT_RIGHT };
94 static const enum Channel chans_Rear[2] = { BACK_LEFT, BACK_RIGHT };
95 static const enum Channel chans_Quad[4] = { FRONT_LEFT, FRONT_RIGHT,
96 BACK_LEFT, BACK_RIGHT };
97 static const enum Channel chans_X51[6] = { FRONT_LEFT, FRONT_RIGHT,
98 FRONT_CENTER, LFE,
99 BACK_LEFT, BACK_RIGHT };
100 static const enum Channel chans_X61[7] = { FRONT_LEFT, FRONT_RIGHT,
101 FRONT_CENTER, LFE, BACK_CENTER,
102 SIDE_LEFT, SIDE_RIGHT };
103 static const enum Channel chans_X71[8] = { FRONT_LEFT, FRONT_RIGHT,
104 FRONT_CENTER, LFE,
105 BACK_LEFT, BACK_RIGHT,
106 SIDE_LEFT, SIDE_RIGHT };
108 ALCdevice *Device = ALContext->Device;
109 ALfloat SourceVolume,ListenerGain,MinVolume,MaxVolume;
110 ALbufferlistitem *BufferListItem;
111 enum DevFmtChannels DevChans;
112 enum FmtChannels Channels;
113 ALfloat (*SrcMatrix)[MAXCHANNELS];
114 ALfloat DryGain, DryGainHF;
115 ALfloat WetGain[MAX_SENDS];
116 ALfloat WetGainHF[MAX_SENDS];
117 ALint NumSends, Frequency;
118 const ALfloat *SpeakerGain;
119 const ALfloat *angles = NULL;
120 const enum Channel *chans = NULL;
121 enum Resampler Resampler;
122 ALint num_channels = 0;
123 ALboolean VirtualChannels;
124 ALfloat Pitch;
125 ALfloat cw;
126 ALuint pos;
127 ALint i, c;
129 /* Get device properties */
130 DevChans = ALContext->Device->FmtChans;
131 NumSends = ALContext->Device->NumAuxSends;
132 Frequency = ALContext->Device->Frequency;
134 /* Get listener properties */
135 ListenerGain = ALContext->Listener.Gain;
137 /* Get source properties */
138 SourceVolume = ALSource->flGain;
139 MinVolume = ALSource->flMinGain;
140 MaxVolume = ALSource->flMaxGain;
141 Pitch = ALSource->flPitch;
142 Resampler = ALSource->Resampler;
143 VirtualChannels = ALSource->VirtualChannels;
145 /* Calculate the stepping value */
146 Channels = FmtMono;
147 BufferListItem = ALSource->queue;
148 while(BufferListItem != NULL)
150 ALbuffer *ALBuffer;
151 if((ALBuffer=BufferListItem->buffer) != NULL)
153 ALint maxstep = STACK_DATA_SIZE / ALSource->NumChannels /
154 ALSource->SampleSize;
155 maxstep -= ResamplerPadding[Resampler] +
156 ResamplerPrePadding[Resampler] + 1;
157 maxstep = mini(maxstep, INT_MAX>>FRACTIONBITS);
159 Pitch = Pitch * ALBuffer->Frequency / Frequency;
160 if(Pitch > (ALfloat)maxstep)
161 ALSource->Params.Step = maxstep<<FRACTIONBITS;
162 else
164 ALSource->Params.Step = (ALint)(Pitch*FRACTIONONE);
165 if(ALSource->Params.Step == 0)
166 ALSource->Params.Step = 1;
169 Channels = ALBuffer->FmtChannels;
171 if(ALSource->VirtualChannels && (Device->Flags&DEVICE_USE_HRTF))
172 ALSource->Params.DoMix = SelectHrtfMixer(ALBuffer,
173 (ALSource->Params.Step==FRACTIONONE) ? POINT_RESAMPLER :
174 Resampler);
175 else
176 ALSource->Params.DoMix = SelectMixer(ALBuffer,
177 (ALSource->Params.Step==FRACTIONONE) ? POINT_RESAMPLER :
178 Resampler);
179 break;
181 BufferListItem = BufferListItem->next;
184 /* Calculate gains */
185 DryGain = clampf(SourceVolume, MinVolume, MaxVolume);
186 DryGain *= ALSource->DirectGain;
187 DryGainHF = ALSource->DirectGainHF;
188 for(i = 0;i < NumSends;i++)
190 WetGain[i] = clampf(SourceVolume, MinVolume, MaxVolume);
191 WetGain[i] *= ALSource->Send[i].WetGain;
192 WetGainHF[i] = ALSource->Send[i].WetGainHF;
195 SrcMatrix = ALSource->Params.DryGains;
196 for(i = 0;i < MAXCHANNELS;i++)
198 for(c = 0;c < MAXCHANNELS;c++)
199 SrcMatrix[i][c] = 0.0f;
201 switch(Channels)
203 case FmtMono:
204 angles = angles_Mono;
205 chans = chans_Mono;
206 num_channels = 1;
207 break;
208 case FmtStereo:
209 if(VirtualChannels && (ALContext->Device->Flags&DEVICE_DUPLICATE_STEREO))
211 DryGain *= aluSqrt(2.0f/4.0f);
212 for(c = 0;c < 2;c++)
214 pos = aluCart2LUTpos(aluCos(F_PI/180.0f * angles_Rear[c]),
215 aluSin(F_PI/180.0f * angles_Rear[c]));
216 SpeakerGain = Device->PanningLUT[pos];
218 for(i = 0;i < (ALint)Device->NumChan;i++)
220 enum Channel chan = Device->Speaker2Chan[i];
221 SrcMatrix[c][chan] += DryGain * ListenerGain *
222 SpeakerGain[chan];
226 angles = angles_Stereo;
227 chans = chans_Stereo;
228 num_channels = 2;
229 break;
231 case FmtRear:
232 angles = angles_Rear;
233 chans = chans_Rear;
234 num_channels = 2;
235 break;
237 case FmtQuad:
238 angles = angles_Quad;
239 chans = chans_Quad;
240 num_channels = 4;
241 break;
243 case FmtX51:
244 angles = angles_X51;
245 chans = chans_X51;
246 num_channels = 6;
247 break;
249 case FmtX61:
250 angles = angles_X61;
251 chans = chans_X61;
252 num_channels = 7;
253 break;
255 case FmtX71:
256 angles = angles_X71;
257 chans = chans_X71;
258 num_channels = 8;
259 break;
262 if(VirtualChannels == AL_FALSE)
264 for(c = 0;c < num_channels;c++)
265 SrcMatrix[c][chans[c]] += DryGain * ListenerGain;
267 else if((Device->Flags&DEVICE_USE_HRTF))
269 for(c = 0;c < num_channels;c++)
271 if(chans[c] == LFE)
273 /* Skip LFE */
274 ALSource->Params.HrtfDelay[c][0] = 0;
275 ALSource->Params.HrtfDelay[c][1] = 0;
276 for(i = 0;i < HRIR_LENGTH;i++)
278 ALSource->Params.HrtfCoeffs[c][i][0] = 0.0f;
279 ALSource->Params.HrtfCoeffs[c][i][1] = 0.0f;
282 else
284 /* Get the static HRIR coefficients and delays for this
285 * channel. */
286 GetLerpedHrtfCoeffs(ALContext->Device->Hrtf,
287 0.0f, F_PI/180.0f * angles[c],
288 DryGain*ListenerGain,
289 ALSource->Params.HrtfCoeffs[c],
290 ALSource->Params.HrtfDelay[c]);
292 ALSource->HrtfCounter = 0;
295 else
297 for(c = 0;c < num_channels;c++)
299 if(chans[c] == LFE) /* Special-case LFE */
301 SrcMatrix[c][LFE] += DryGain * ListenerGain;
302 continue;
304 pos = aluCart2LUTpos(aluCos(F_PI/180.0f * angles[c]),
305 aluSin(F_PI/180.0f * angles[c]));
306 SpeakerGain = Device->PanningLUT[pos];
308 for(i = 0;i < (ALint)Device->NumChan;i++)
310 enum Channel chan = Device->Speaker2Chan[i];
311 SrcMatrix[c][chan] += DryGain * ListenerGain *
312 SpeakerGain[chan];
316 for(i = 0;i < NumSends;i++)
318 ALSource->Params.Send[i].Slot = ALSource->Send[i].Slot;
319 ALSource->Params.Send[i].WetGain = WetGain[i] * ListenerGain;
322 /* Update filter coefficients. Calculations based on the I3DL2
323 * spec. */
324 cw = aluCos(F_PI*2.0f * LOWPASSFREQCUTOFF / Frequency);
326 /* We use two chained one-pole filters, so we need to take the
327 * square root of the squared gain, which is the same as the base
328 * gain. */
329 ALSource->Params.iirFilter.coeff = lpCoeffCalc(DryGainHF, cw);
330 for(i = 0;i < NumSends;i++)
332 /* We use a one-pole filter, so we need to take the squared gain */
333 ALfloat a = lpCoeffCalc(WetGainHF[i]*WetGainHF[i], cw);
334 ALSource->Params.Send[i].iirFilter.coeff = a;
338 ALvoid CalcSourceParams(ALsource *ALSource, const ALCcontext *ALContext)
340 const ALCdevice *Device = ALContext->Device;
341 ALfloat InnerAngle,OuterAngle,Angle,Distance,ClampedDist;
342 ALfloat Direction[3],Position[3],SourceToListener[3];
343 ALfloat Velocity[3],ListenerVel[3];
344 ALfloat MinVolume,MaxVolume,MinDist,MaxDist,Rolloff;
345 ALfloat ConeVolume,ConeHF,SourceVolume,ListenerGain;
346 ALfloat DopplerFactor, DopplerVelocity, SpeedOfSound;
347 ALfloat AirAbsorptionFactor;
348 ALfloat RoomAirAbsorption[MAX_SENDS];
349 ALbufferlistitem *BufferListItem;
350 ALfloat Attenuation, EffectiveDist;
351 ALfloat RoomAttenuation[MAX_SENDS];
352 ALfloat MetersPerUnit;
353 ALfloat RoomRolloffBase;
354 ALfloat RoomRolloff[MAX_SENDS];
355 ALfloat DecayDistance[MAX_SENDS];
356 ALfloat DryGain;
357 ALfloat DryGainHF;
358 ALboolean DryGainHFAuto;
359 ALfloat WetGain[MAX_SENDS];
360 ALfloat WetGainHF[MAX_SENDS];
361 ALboolean WetGainAuto;
362 ALboolean WetGainHFAuto;
363 enum Resampler Resampler;
364 ALfloat Pitch;
365 ALuint Frequency;
366 ALint NumSends;
367 ALfloat cw;
368 ALint i;
370 DryGainHF = 1.0f;
371 for(i = 0;i < MAX_SENDS;i++)
372 WetGainHF[i] = 1.0f;
374 //Get context properties
375 DopplerFactor = ALContext->DopplerFactor * ALSource->DopplerFactor;
376 DopplerVelocity = ALContext->DopplerVelocity;
377 SpeedOfSound = ALContext->flSpeedOfSound;
378 NumSends = Device->NumAuxSends;
379 Frequency = Device->Frequency;
381 //Get listener properties
382 ListenerGain = ALContext->Listener.Gain;
383 MetersPerUnit = ALContext->Listener.MetersPerUnit;
384 ListenerVel[0] = ALContext->Listener.Velocity[0];
385 ListenerVel[1] = ALContext->Listener.Velocity[1];
386 ListenerVel[2] = ALContext->Listener.Velocity[2];
388 //Get source properties
389 SourceVolume = ALSource->flGain;
390 MinVolume = ALSource->flMinGain;
391 MaxVolume = ALSource->flMaxGain;
392 Pitch = ALSource->flPitch;
393 Resampler = ALSource->Resampler;
394 Position[0] = ALSource->vPosition[0];
395 Position[1] = ALSource->vPosition[1];
396 Position[2] = ALSource->vPosition[2];
397 Direction[0] = ALSource->vOrientation[0];
398 Direction[1] = ALSource->vOrientation[1];
399 Direction[2] = ALSource->vOrientation[2];
400 Velocity[0] = ALSource->vVelocity[0];
401 Velocity[1] = ALSource->vVelocity[1];
402 Velocity[2] = ALSource->vVelocity[2];
403 MinDist = ALSource->flRefDistance;
404 MaxDist = ALSource->flMaxDistance;
405 Rolloff = ALSource->flRollOffFactor;
406 InnerAngle = ALSource->flInnerAngle * ConeScale;
407 OuterAngle = ALSource->flOuterAngle * ConeScale;
408 AirAbsorptionFactor = ALSource->AirAbsorptionFactor;
409 DryGainHFAuto = ALSource->DryGainHFAuto;
410 WetGainAuto = ALSource->WetGainAuto;
411 WetGainHFAuto = ALSource->WetGainHFAuto;
412 RoomRolloffBase = ALSource->RoomRolloffFactor;
413 for(i = 0;i < NumSends;i++)
415 ALeffectslot *Slot = ALSource->Send[i].Slot;
417 if(!Slot || Slot->effect.type == AL_EFFECT_NULL)
419 RoomRolloff[i] = 0.0f;
420 DecayDistance[i] = 0.0f;
421 RoomAirAbsorption[i] = 1.0f;
423 else if(Slot->AuxSendAuto)
425 RoomRolloff[i] = RoomRolloffBase;
426 if(IsReverbEffect(Slot->effect.type))
428 RoomRolloff[i] += Slot->effect.Reverb.RoomRolloffFactor;
429 DecayDistance[i] = Slot->effect.Reverb.DecayTime *
430 SPEEDOFSOUNDMETRESPERSEC;
431 RoomAirAbsorption[i] = Slot->effect.Reverb.AirAbsorptionGainHF;
433 else
435 DecayDistance[i] = 0.0f;
436 RoomAirAbsorption[i] = 1.0f;
439 else
441 /* If the slot's auxiliary send auto is off, the data sent to the
442 * effect slot is the same as the dry path, sans filter effects */
443 RoomRolloff[i] = Rolloff;
444 DecayDistance[i] = 0.0f;
445 RoomAirAbsorption[i] = AIRABSORBGAINHF;
448 ALSource->Params.Send[i].Slot = Slot;
451 //1. Translate Listener to origin (convert to head relative)
452 if(ALSource->bHeadRelative == AL_FALSE)
454 ALfloat U[3],V[3],N[3];
455 ALfloat Matrix[4][4];
457 // Build transform matrix
458 N[0] = ALContext->Listener.Forward[0]; // At-vector
459 N[1] = ALContext->Listener.Forward[1];
460 N[2] = ALContext->Listener.Forward[2];
461 aluNormalize(N); // Normalized At-vector
462 V[0] = ALContext->Listener.Up[0]; // Up-vector
463 V[1] = ALContext->Listener.Up[1];
464 V[2] = ALContext->Listener.Up[2];
465 aluNormalize(V); // Normalized Up-vector
466 aluCrossproduct(N, V, U); // Right-vector
467 aluNormalize(U); // Normalized Right-vector
468 Matrix[0][0] = U[0]; Matrix[0][1] = V[0]; Matrix[0][2] = -N[0]; Matrix[0][3] = 0.0f;
469 Matrix[1][0] = U[1]; Matrix[1][1] = V[1]; Matrix[1][2] = -N[1]; Matrix[1][3] = 0.0f;
470 Matrix[2][0] = U[2]; Matrix[2][1] = V[2]; Matrix[2][2] = -N[2]; Matrix[2][3] = 0.0f;
471 Matrix[3][0] = 0.0f; Matrix[3][1] = 0.0f; Matrix[3][2] = 0.0f; Matrix[3][3] = 1.0f;
473 // Translate position
474 Position[0] -= ALContext->Listener.Position[0];
475 Position[1] -= ALContext->Listener.Position[1];
476 Position[2] -= ALContext->Listener.Position[2];
478 // Transform source position and direction into listener space
479 aluMatrixVector(Position, 1.0f, Matrix);
480 aluMatrixVector(Direction, 0.0f, Matrix);
481 // Transform source and listener velocity into listener space
482 aluMatrixVector(Velocity, 0.0f, Matrix);
483 aluMatrixVector(ListenerVel, 0.0f, Matrix);
485 else
486 ListenerVel[0] = ListenerVel[1] = ListenerVel[2] = 0.0f;
488 SourceToListener[0] = -Position[0];
489 SourceToListener[1] = -Position[1];
490 SourceToListener[2] = -Position[2];
491 aluNormalize(SourceToListener);
492 aluNormalize(Direction);
494 //2. Calculate distance attenuation
495 Distance = aluSqrt(aluDotproduct(Position, Position));
496 ClampedDist = Distance;
498 Attenuation = 1.0f;
499 for(i = 0;i < NumSends;i++)
500 RoomAttenuation[i] = 1.0f;
501 switch(ALContext->SourceDistanceModel ? ALSource->DistanceModel :
502 ALContext->DistanceModel)
504 case InverseDistanceClamped:
505 ClampedDist = clampf(ClampedDist, MinDist, MaxDist);
506 if(MaxDist < MinDist)
507 break;
508 //fall-through
509 case InverseDistance:
510 if(MinDist > 0.0f)
512 if((MinDist + (Rolloff * (ClampedDist - MinDist))) > 0.0f)
513 Attenuation = MinDist / (MinDist + (Rolloff * (ClampedDist - MinDist)));
514 for(i = 0;i < NumSends;i++)
516 if((MinDist + (RoomRolloff[i] * (ClampedDist - MinDist))) > 0.0f)
517 RoomAttenuation[i] = MinDist / (MinDist + (RoomRolloff[i] * (ClampedDist - MinDist)));
520 break;
522 case LinearDistanceClamped:
523 ClampedDist = clampf(ClampedDist, MinDist, MaxDist);
524 if(MaxDist < MinDist)
525 break;
526 //fall-through
527 case LinearDistance:
528 if(MaxDist != MinDist)
530 Attenuation = 1.0f - (Rolloff*(ClampedDist-MinDist)/(MaxDist - MinDist));
531 Attenuation = maxf(Attenuation, 0.0f);
532 for(i = 0;i < NumSends;i++)
534 RoomAttenuation[i] = 1.0f - (RoomRolloff[i]*(ClampedDist-MinDist)/(MaxDist - MinDist));
535 RoomAttenuation[i] = maxf(RoomAttenuation[i], 0.0f);
538 break;
540 case ExponentDistanceClamped:
541 ClampedDist = clampf(ClampedDist, MinDist, MaxDist);
542 if(MaxDist < MinDist)
543 break;
544 //fall-through
545 case ExponentDistance:
546 if(ClampedDist > 0.0f && MinDist > 0.0f)
548 Attenuation = aluPow(ClampedDist/MinDist, -Rolloff);
549 for(i = 0;i < NumSends;i++)
550 RoomAttenuation[i] = aluPow(ClampedDist/MinDist, -RoomRolloff[i]);
552 break;
554 case DisableDistance:
555 break;
558 // Source Gain + Attenuation
559 DryGain = SourceVolume * Attenuation;
560 for(i = 0;i < NumSends;i++)
561 WetGain[i] = SourceVolume * RoomAttenuation[i];
563 // Distance-based air absorption
564 EffectiveDist = 0.0f;
565 if(MinDist > 0.0f && Attenuation < 1.0f)
566 EffectiveDist = (MinDist/Attenuation - MinDist)*MetersPerUnit;
567 if(AirAbsorptionFactor > 0.0f && EffectiveDist > 0.0f)
569 DryGainHF *= aluPow(AIRABSORBGAINHF, AirAbsorptionFactor*EffectiveDist);
570 for(i = 0;i < NumSends;i++)
571 WetGainHF[i] *= aluPow(RoomAirAbsorption[i],
572 AirAbsorptionFactor*EffectiveDist);
575 //3. Apply directional soundcones
576 Angle = aluAcos(aluDotproduct(Direction,SourceToListener)) * (180.0f/F_PI);
577 if(Angle >= InnerAngle && Angle <= OuterAngle)
579 ALfloat scale = (Angle-InnerAngle) / (OuterAngle-InnerAngle);
580 ConeVolume = lerp(1.0, ALSource->flOuterGain, scale);
581 ConeHF = lerp(1.0, ALSource->OuterGainHF, scale);
583 else if(Angle > OuterAngle)
585 ConeVolume = ALSource->flOuterGain;
586 ConeHF = ALSource->OuterGainHF;
588 else
590 ConeVolume = 1.0f;
591 ConeHF = 1.0f;
594 DryGain *= ConeVolume;
595 if(WetGainAuto)
597 for(i = 0;i < NumSends;i++)
598 WetGain[i] *= ConeVolume;
600 if(DryGainHFAuto)
601 DryGainHF *= ConeHF;
602 if(WetGainHFAuto)
604 for(i = 0;i < NumSends;i++)
605 WetGainHF[i] *= ConeHF;
608 // Clamp to Min/Max Gain
609 DryGain = clampf(DryGain, MinVolume, MaxVolume);
610 for(i = 0;i < NumSends;i++)
611 WetGain[i] = clampf(WetGain[i], MinVolume, MaxVolume);
613 // Apply filter gains and filters
614 DryGain *= ALSource->DirectGain * ListenerGain;
615 DryGainHF *= ALSource->DirectGainHF;
616 for(i = 0;i < NumSends;i++)
618 WetGain[i] *= ALSource->Send[i].WetGain * ListenerGain;
619 WetGainHF[i] *= ALSource->Send[i].WetGainHF;
622 if(WetGainAuto)
624 /* Apply a decay-time transformation to the wet path, based on the
625 * attenuation of the dry path.
627 * Using the approximate (effective) source to listener distance, the
628 * initial decay of the reverb effect is calculated and applied to the
629 * wet path.
631 for(i = 0;i < NumSends;i++)
633 if(DecayDistance[i] > 0.0f)
634 WetGain[i] *= aluPow(0.001f /* -60dB */,
635 EffectiveDist / DecayDistance[i]);
639 // Calculate Velocity
640 if(DopplerFactor != 0.0f)
642 ALfloat VSS, VLS;
643 ALfloat MaxVelocity = (SpeedOfSound*DopplerVelocity) /
644 DopplerFactor;
646 VSS = aluDotproduct(Velocity, SourceToListener);
647 if(VSS >= MaxVelocity)
648 VSS = (MaxVelocity - 1.0f);
649 else if(VSS <= -MaxVelocity)
650 VSS = -MaxVelocity + 1.0f;
652 VLS = aluDotproduct(ListenerVel, SourceToListener);
653 if(VLS >= MaxVelocity)
654 VLS = (MaxVelocity - 1.0f);
655 else if(VLS <= -MaxVelocity)
656 VLS = -MaxVelocity + 1.0f;
658 Pitch *= ((SpeedOfSound*DopplerVelocity) - (DopplerFactor*VLS)) /
659 ((SpeedOfSound*DopplerVelocity) - (DopplerFactor*VSS));
662 BufferListItem = ALSource->queue;
663 while(BufferListItem != NULL)
665 ALbuffer *ALBuffer;
666 if((ALBuffer=BufferListItem->buffer) != NULL)
668 ALint maxstep = STACK_DATA_SIZE / ALSource->NumChannels /
669 ALSource->SampleSize;
670 maxstep -= ResamplerPadding[Resampler] +
671 ResamplerPrePadding[Resampler] + 1;
672 maxstep = mini(maxstep, INT_MAX>>FRACTIONBITS);
674 Pitch = Pitch * ALBuffer->Frequency / Frequency;
675 if(Pitch > (ALfloat)maxstep)
676 ALSource->Params.Step = maxstep<<FRACTIONBITS;
677 else
679 ALSource->Params.Step = (ALint)(Pitch*FRACTIONONE);
680 if(ALSource->Params.Step == 0)
681 ALSource->Params.Step = 1;
684 if((Device->Flags&DEVICE_USE_HRTF))
685 ALSource->Params.DoMix = SelectHrtfMixer(ALBuffer,
686 (ALSource->Params.Step==FRACTIONONE) ? POINT_RESAMPLER :
687 Resampler);
688 else
689 ALSource->Params.DoMix = SelectMixer(ALBuffer,
690 (ALSource->Params.Step==FRACTIONONE) ? POINT_RESAMPLER :
691 Resampler);
692 break;
694 BufferListItem = BufferListItem->next;
697 if((Device->Flags&DEVICE_USE_HRTF))
699 // Use a binaural HRTF algorithm for stereo headphone playback
700 ALfloat delta, ev = 0.0f, az = 0.0f;
702 if(Distance > 0.0f)
704 ALfloat invlen = 1.0f/Distance;
705 Position[0] *= invlen;
706 Position[1] *= invlen;
707 Position[2] *= invlen;
709 // Calculate elevation and azimuth only when the source is not at
710 // the listener. This prevents +0 and -0 Z from producing
711 // inconsistent panning.
712 ev = asin(Position[1]);
713 az = atan2(Position[0], -Position[2]*ZScale);
716 // Check to see if the HRIR is already moving.
717 if(ALSource->HrtfMoving)
719 // Calculate the normalized HRTF transition factor (delta).
720 delta = CalcHrtfDelta(ALSource->Params.HrtfGain, DryGain,
721 ALSource->Params.HrtfDir, Position);
722 // If the delta is large enough, get the moving HRIR target
723 // coefficients, target delays, steppping values, and counter.
724 if(delta > 0.001f)
726 ALSource->HrtfCounter = GetMovingHrtfCoeffs(Device->Hrtf,
727 ev, az, DryGain, delta,
728 ALSource->HrtfCounter,
729 ALSource->Params.HrtfCoeffs[0],
730 ALSource->Params.HrtfDelay[0],
731 ALSource->Params.HrtfCoeffStep,
732 ALSource->Params.HrtfDelayStep);
733 ALSource->Params.HrtfGain = DryGain;
734 ALSource->Params.HrtfDir[0] = Position[0];
735 ALSource->Params.HrtfDir[1] = Position[1];
736 ALSource->Params.HrtfDir[2] = Position[2];
739 else
741 // Get the initial (static) HRIR coefficients and delays.
742 GetLerpedHrtfCoeffs(Device->Hrtf, ev, az, DryGain,
743 ALSource->Params.HrtfCoeffs[0],
744 ALSource->Params.HrtfDelay[0]);
745 ALSource->HrtfCounter = 0;
746 ALSource->Params.HrtfGain = DryGain;
747 ALSource->Params.HrtfDir[0] = Position[0];
748 ALSource->Params.HrtfDir[1] = Position[1];
749 ALSource->Params.HrtfDir[2] = Position[2];
752 else
754 // Use energy-preserving panning algorithm for multi-speaker playback
755 ALfloat DirGain, AmbientGain;
756 const ALfloat *SpeakerGain;
757 ALfloat length;
758 ALint pos;
760 length = maxf(Distance, MinDist);
761 if(length > 0.0f)
763 ALfloat invlen = 1.0f/length;
764 Position[0] *= invlen;
765 Position[1] *= invlen;
766 Position[2] *= invlen;
769 pos = aluCart2LUTpos(-Position[2]*ZScale, Position[0]);
770 SpeakerGain = Device->PanningLUT[pos];
772 DirGain = aluSqrt(Position[0]*Position[0] + Position[2]*Position[2]);
773 // elevation adjustment for directional gain. this sucks, but
774 // has low complexity
775 AmbientGain = aluSqrt(1.0/Device->NumChan);
776 for(i = 0;i < MAXCHANNELS;i++)
778 ALuint i2;
779 for(i2 = 0;i2 < MAXCHANNELS;i2++)
780 ALSource->Params.DryGains[i][i2] = 0.0f;
782 for(i = 0;i < (ALint)Device->NumChan;i++)
784 enum Channel chan = Device->Speaker2Chan[i];
785 ALfloat gain = lerp(AmbientGain, SpeakerGain[chan], DirGain);
786 ALSource->Params.DryGains[0][chan] = DryGain * gain;
789 for(i = 0;i < NumSends;i++)
790 ALSource->Params.Send[i].WetGain = WetGain[i];
792 /* Update filter coefficients. */
793 cw = aluCos(F_PI*2.0f * LOWPASSFREQCUTOFF / Frequency);
795 ALSource->Params.iirFilter.coeff = lpCoeffCalc(DryGainHF, cw);
796 for(i = 0;i < NumSends;i++)
798 ALfloat a = lpCoeffCalc(WetGainHF[i]*WetGainHF[i], cw);
799 ALSource->Params.Send[i].iirFilter.coeff = a;
804 static __inline ALfloat aluF2F(ALfloat val)
805 { return val; }
806 static __inline ALshort aluF2S(ALfloat val)
808 if(val > 1.0f) return 32767;
809 if(val < -1.0f) return -32768;
810 return (ALint)(val*32767.0f);
812 static __inline ALushort aluF2US(ALfloat val)
813 { return aluF2S(val)+32768; }
814 static __inline ALbyte aluF2B(ALfloat val)
815 { return aluF2S(val)>>8; }
816 static __inline ALubyte aluF2UB(ALfloat val)
817 { return aluF2US(val)>>8; }
819 #define DECL_TEMPLATE(T, N, func) \
820 static void Write_##T##_##N(ALCdevice *device, T *RESTRICT buffer, \
821 ALuint SamplesToDo) \
823 ALfloat (*RESTRICT DryBuffer)[MAXCHANNELS] = device->DryBuffer; \
824 const enum Channel *ChanMap = device->DevChannels; \
825 ALuint i, j; \
827 for(i = 0;i < SamplesToDo;i++) \
829 for(j = 0;j < N;j++) \
830 *(buffer++) = func(DryBuffer[i][ChanMap[j]]); \
834 DECL_TEMPLATE(ALfloat, 1, aluF2F)
835 DECL_TEMPLATE(ALfloat, 4, aluF2F)
836 DECL_TEMPLATE(ALfloat, 6, aluF2F)
837 DECL_TEMPLATE(ALfloat, 7, aluF2F)
838 DECL_TEMPLATE(ALfloat, 8, aluF2F)
840 DECL_TEMPLATE(ALushort, 1, aluF2US)
841 DECL_TEMPLATE(ALushort, 4, aluF2US)
842 DECL_TEMPLATE(ALushort, 6, aluF2US)
843 DECL_TEMPLATE(ALushort, 7, aluF2US)
844 DECL_TEMPLATE(ALushort, 8, aluF2US)
846 DECL_TEMPLATE(ALshort, 1, aluF2S)
847 DECL_TEMPLATE(ALshort, 4, aluF2S)
848 DECL_TEMPLATE(ALshort, 6, aluF2S)
849 DECL_TEMPLATE(ALshort, 7, aluF2S)
850 DECL_TEMPLATE(ALshort, 8, aluF2S)
852 DECL_TEMPLATE(ALubyte, 1, aluF2UB)
853 DECL_TEMPLATE(ALubyte, 4, aluF2UB)
854 DECL_TEMPLATE(ALubyte, 6, aluF2UB)
855 DECL_TEMPLATE(ALubyte, 7, aluF2UB)
856 DECL_TEMPLATE(ALubyte, 8, aluF2UB)
858 DECL_TEMPLATE(ALbyte, 1, aluF2B)
859 DECL_TEMPLATE(ALbyte, 4, aluF2B)
860 DECL_TEMPLATE(ALbyte, 6, aluF2B)
861 DECL_TEMPLATE(ALbyte, 7, aluF2B)
862 DECL_TEMPLATE(ALbyte, 8, aluF2B)
864 #undef DECL_TEMPLATE
866 #define DECL_TEMPLATE(T, N, func) \
867 static void Write_##T##_##N(ALCdevice *device, T *RESTRICT buffer, \
868 ALuint SamplesToDo) \
870 ALfloat (*RESTRICT DryBuffer)[MAXCHANNELS] = device->DryBuffer; \
871 const enum Channel *ChanMap = device->DevChannels; \
872 ALuint i, j; \
874 if(device->Bs2b) \
876 for(i = 0;i < SamplesToDo;i++) \
878 float samples[2]; \
879 samples[0] = DryBuffer[i][ChanMap[0]]; \
880 samples[1] = DryBuffer[i][ChanMap[1]]; \
881 bs2b_cross_feed(device->Bs2b, samples); \
882 *(buffer++) = func(samples[0]); \
883 *(buffer++) = func(samples[1]); \
886 else \
888 for(i = 0;i < SamplesToDo;i++) \
890 for(j = 0;j < N;j++) \
891 *(buffer++) = func(DryBuffer[i][ChanMap[j]]); \
896 DECL_TEMPLATE(ALfloat, 2, aluF2F)
897 DECL_TEMPLATE(ALushort, 2, aluF2US)
898 DECL_TEMPLATE(ALshort, 2, aluF2S)
899 DECL_TEMPLATE(ALubyte, 2, aluF2UB)
900 DECL_TEMPLATE(ALbyte, 2, aluF2B)
902 #undef DECL_TEMPLATE
904 #define DECL_TEMPLATE(T) \
905 static void Write_##T(ALCdevice *device, T *buffer, ALuint SamplesToDo) \
907 switch(device->FmtChans) \
909 case DevFmtMono: \
910 Write_##T##_1(device, buffer, SamplesToDo); \
911 break; \
912 case DevFmtStereo: \
913 Write_##T##_2(device, buffer, SamplesToDo); \
914 break; \
915 case DevFmtQuad: \
916 Write_##T##_4(device, buffer, SamplesToDo); \
917 break; \
918 case DevFmtX51: \
919 case DevFmtX51Side: \
920 Write_##T##_6(device, buffer, SamplesToDo); \
921 break; \
922 case DevFmtX61: \
923 Write_##T##_7(device, buffer, SamplesToDo); \
924 break; \
925 case DevFmtX71: \
926 Write_##T##_8(device, buffer, SamplesToDo); \
927 break; \
931 DECL_TEMPLATE(ALfloat)
932 DECL_TEMPLATE(ALushort)
933 DECL_TEMPLATE(ALshort)
934 DECL_TEMPLATE(ALubyte)
935 DECL_TEMPLATE(ALbyte)
937 #undef DECL_TEMPLATE
939 ALvoid aluMixData(ALCdevice *device, ALvoid *buffer, ALsizei size)
941 ALuint SamplesToDo;
942 ALeffectslot **slot, **slot_end;
943 ALsource **src, **src_end;
944 ALCcontext *ctx;
945 int fpuState;
946 ALuint i, c;
948 #if defined(HAVE_FESETROUND)
949 fpuState = fegetround();
950 fesetround(FE_TOWARDZERO);
951 #elif defined(HAVE__CONTROLFP)
952 fpuState = _controlfp(0, 0);
953 (void)_controlfp(_RC_CHOP, _MCW_RC);
954 #else
955 (void)fpuState;
956 #endif
958 while(size > 0)
960 /* Setup variables */
961 SamplesToDo = minu(size, BUFFERSIZE);
963 /* Clear mixing buffer */
964 memset(device->DryBuffer, 0, SamplesToDo*MAXCHANNELS*sizeof(ALfloat));
966 LockDevice(device);
967 ctx = device->ContextList;
968 while(ctx)
970 ALenum DeferUpdates = ctx->DeferUpdates;
971 ALenum UpdateSources = AL_FALSE;
973 if(!DeferUpdates)
974 UpdateSources = ExchangeInt(&ctx->UpdateSources, AL_FALSE);
976 src = ctx->ActiveSources;
977 src_end = src + ctx->ActiveSourceCount;
978 while(src != src_end)
980 if((*src)->state != AL_PLAYING)
982 --(ctx->ActiveSourceCount);
983 *src = *(--src_end);
984 continue;
987 if(!DeferUpdates && (ExchangeInt(&(*src)->NeedsUpdate, AL_FALSE) ||
988 UpdateSources))
989 ALsource_Update(*src, ctx);
991 MixSource(*src, device, SamplesToDo);
992 src++;
995 /* effect slot processing */
996 slot = ctx->ActiveEffectSlots;
997 slot_end = slot + ctx->ActiveEffectSlotCount;
998 while(slot != slot_end)
1000 for(c = 0;c < SamplesToDo;c++)
1002 (*slot)->WetBuffer[c] += (*slot)->ClickRemoval[0];
1003 (*slot)->ClickRemoval[0] -= (*slot)->ClickRemoval[0] / 256.0f;
1005 (*slot)->ClickRemoval[0] += (*slot)->PendingClicks[0];
1006 (*slot)->PendingClicks[0] = 0.0f;
1008 if(!DeferUpdates && ExchangeInt(&(*slot)->NeedsUpdate, AL_FALSE))
1009 ALeffectState_Update((*slot)->EffectState, ctx, *slot);
1011 ALeffectState_Process((*slot)->EffectState, SamplesToDo,
1012 (*slot)->WetBuffer, device->DryBuffer);
1014 for(i = 0;i < SamplesToDo;i++)
1015 (*slot)->WetBuffer[i] = 0.0f;
1017 slot++;
1020 ctx = ctx->next;
1022 UnlockDevice(device);
1024 //Post processing loop
1025 if(device->FmtChans == DevFmtMono)
1027 for(i = 0;i < SamplesToDo;i++)
1029 device->DryBuffer[i][FRONT_CENTER] += device->ClickRemoval[FRONT_CENTER];
1030 device->ClickRemoval[FRONT_CENTER] -= device->ClickRemoval[FRONT_CENTER] / 256.0f;
1032 device->ClickRemoval[FRONT_CENTER] += device->PendingClicks[FRONT_CENTER];
1033 device->PendingClicks[FRONT_CENTER] = 0.0f;
1035 else if(device->FmtChans == DevFmtStereo)
1037 /* Assumes the first two channels are FRONT_LEFT and FRONT_RIGHT */
1038 for(i = 0;i < SamplesToDo;i++)
1040 for(c = 0;c < 2;c++)
1042 device->DryBuffer[i][c] += device->ClickRemoval[c];
1043 device->ClickRemoval[c] -= device->ClickRemoval[c] / 256.0f;
1046 for(c = 0;c < 2;c++)
1048 device->ClickRemoval[c] += device->PendingClicks[c];
1049 device->PendingClicks[c] = 0.0f;
1052 else
1054 for(i = 0;i < SamplesToDo;i++)
1056 for(c = 0;c < MAXCHANNELS;c++)
1058 device->DryBuffer[i][c] += device->ClickRemoval[c];
1059 device->ClickRemoval[c] -= device->ClickRemoval[c] / 256.0f;
1062 for(c = 0;c < MAXCHANNELS;c++)
1064 device->ClickRemoval[c] += device->PendingClicks[c];
1065 device->PendingClicks[c] = 0.0f;
1069 if(buffer)
1071 switch(device->FmtType)
1073 case DevFmtByte:
1074 Write_ALbyte(device, buffer, SamplesToDo);
1075 break;
1076 case DevFmtUByte:
1077 Write_ALubyte(device, buffer, SamplesToDo);
1078 break;
1079 case DevFmtShort:
1080 Write_ALshort(device, buffer, SamplesToDo);
1081 break;
1082 case DevFmtUShort:
1083 Write_ALushort(device, buffer, SamplesToDo);
1084 break;
1085 case DevFmtFloat:
1086 Write_ALfloat(device, buffer, SamplesToDo);
1087 break;
1091 size -= SamplesToDo;
1094 #if defined(HAVE_FESETROUND)
1095 fesetround(fpuState);
1096 #elif defined(HAVE__CONTROLFP)
1097 _controlfp(fpuState, _MCW_RC);
1098 #endif
1102 ALvoid aluHandleDisconnect(ALCdevice *device)
1104 ALCcontext *Context;
1106 LockDevice(device);
1107 device->Connected = ALC_FALSE;
1109 Context = device->ContextList;
1110 while(Context)
1112 ALsource **src, **src_end;
1114 src = Context->ActiveSources;
1115 src_end = src + Context->ActiveSourceCount;
1116 while(src != src_end)
1118 if((*src)->state == AL_PLAYING)
1120 (*src)->state = AL_STOPPED;
1121 (*src)->BuffersPlayed = (*src)->BuffersInQueue;
1122 (*src)->position = 0;
1123 (*src)->position_fraction = 0;
1125 src++;
1127 Context->ActiveSourceCount = 0;
1129 Context = Context->next;
1131 UnlockDevice(device);