Calculate the listener matrix when a new orientation is specified
[openal-soft/android.git] / Alc / ALu.c
blobc1388c4af1eada1db2c8abc30a9e1dcbad7b7f64
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 aluMatrixVector(ALfloat *vector,ALfloat w,ALfloat matrix[4][4])
49 ALfloat temp[4] = {
50 vector[0], vector[1], vector[2], w
53 vector[0] = temp[0]*matrix[0][0] + temp[1]*matrix[1][0] + temp[2]*matrix[2][0] + temp[3]*matrix[3][0];
54 vector[1] = temp[0]*matrix[0][1] + temp[1]*matrix[1][1] + temp[2]*matrix[2][1] + temp[3]*matrix[3][1];
55 vector[2] = temp[0]*matrix[0][2] + temp[1]*matrix[1][2] + temp[2]*matrix[2][2] + temp[3]*matrix[3][2];
59 ALvoid CalcNonAttnSourceParams(ALsource *ALSource, const ALCcontext *ALContext)
61 static const ALfloat angles_Mono[1] = { 0.0f };
62 static const ALfloat angles_Stereo[2] = { -30.0f, 30.0f };
63 static const ALfloat angles_Rear[2] = { -150.0f, 150.0f };
64 static const ALfloat angles_Quad[4] = { -45.0f, 45.0f, -135.0f, 135.0f };
65 static const ALfloat angles_X51[6] = { -30.0f, 30.0f, 0.0f, 0.0f,
66 -110.0f, 110.0f };
67 static const ALfloat angles_X61[7] = { -30.0f, 30.0f, 0.0f, 0.0f,
68 180.0f, -90.0f, 90.0f };
69 static const ALfloat angles_X71[8] = { -30.0f, 30.0f, 0.0f, 0.0f,
70 -110.0f, 110.0f, -90.0f, 90.0f };
72 static const enum Channel chans_Mono[1] = { FRONT_CENTER };
73 static const enum Channel chans_Stereo[2] = { FRONT_LEFT, FRONT_RIGHT };
74 static const enum Channel chans_Rear[2] = { BACK_LEFT, BACK_RIGHT };
75 static const enum Channel chans_Quad[4] = { FRONT_LEFT, FRONT_RIGHT,
76 BACK_LEFT, BACK_RIGHT };
77 static const enum Channel chans_X51[6] = { FRONT_LEFT, FRONT_RIGHT,
78 FRONT_CENTER, LFE,
79 BACK_LEFT, BACK_RIGHT };
80 static const enum Channel chans_X61[7] = { FRONT_LEFT, FRONT_RIGHT,
81 FRONT_CENTER, LFE, BACK_CENTER,
82 SIDE_LEFT, SIDE_RIGHT };
83 static const enum Channel chans_X71[8] = { FRONT_LEFT, FRONT_RIGHT,
84 FRONT_CENTER, LFE,
85 BACK_LEFT, BACK_RIGHT,
86 SIDE_LEFT, SIDE_RIGHT };
88 ALCdevice *Device = ALContext->Device;
89 ALfloat SourceVolume,ListenerGain,MinVolume,MaxVolume;
90 ALbufferlistitem *BufferListItem;
91 enum DevFmtChannels DevChans;
92 enum FmtChannels Channels;
93 ALfloat (*SrcMatrix)[MAXCHANNELS];
94 ALfloat DryGain, DryGainHF;
95 ALfloat WetGain[MAX_SENDS];
96 ALfloat WetGainHF[MAX_SENDS];
97 ALint NumSends, Frequency;
98 const ALfloat *SpeakerGain;
99 const ALfloat *angles = NULL;
100 const enum Channel *chans = NULL;
101 enum Resampler Resampler;
102 ALint num_channels = 0;
103 ALboolean VirtualChannels;
104 ALfloat Pitch;
105 ALfloat cw;
106 ALuint pos;
107 ALint i, c;
109 /* Get device properties */
110 DevChans = Device->FmtChans;
111 NumSends = Device->NumAuxSends;
112 Frequency = Device->Frequency;
114 /* Get listener properties */
115 ListenerGain = ALContext->Listener.Gain;
117 /* Get source properties */
118 SourceVolume = ALSource->flGain;
119 MinVolume = ALSource->flMinGain;
120 MaxVolume = ALSource->flMaxGain;
121 Pitch = ALSource->flPitch;
122 Resampler = ALSource->Resampler;
123 VirtualChannels = ALSource->VirtualChannels;
125 /* Calculate the stepping value */
126 Channels = FmtMono;
127 BufferListItem = ALSource->queue;
128 while(BufferListItem != NULL)
130 ALbuffer *ALBuffer;
131 if((ALBuffer=BufferListItem->buffer) != NULL)
133 ALint maxstep = STACK_DATA_SIZE / ALSource->NumChannels /
134 ALSource->SampleSize;
135 maxstep -= ResamplerPadding[Resampler] +
136 ResamplerPrePadding[Resampler] + 1;
137 maxstep = mini(maxstep, INT_MAX>>FRACTIONBITS);
139 Pitch = Pitch * ALBuffer->Frequency / Frequency;
140 if(Pitch > (ALfloat)maxstep)
141 ALSource->Params.Step = maxstep<<FRACTIONBITS;
142 else
144 ALSource->Params.Step = fastf2i(Pitch*FRACTIONONE);
145 if(ALSource->Params.Step == 0)
146 ALSource->Params.Step = 1;
149 Channels = ALBuffer->FmtChannels;
150 break;
152 BufferListItem = BufferListItem->next;
154 if(VirtualChannels && Device->Hrtf)
155 ALSource->Params.DoMix = SelectHrtfMixer((ALSource->Params.Step==FRACTIONONE) ?
156 POINT_RESAMPLER : Resampler);
157 else
158 ALSource->Params.DoMix = SelectMixer((ALSource->Params.Step==FRACTIONONE) ?
159 POINT_RESAMPLER : Resampler);
161 /* Calculate gains */
162 DryGain = clampf(SourceVolume, MinVolume, MaxVolume);
163 DryGain *= ALSource->DirectGain;
164 DryGainHF = ALSource->DirectGainHF;
165 for(i = 0;i < NumSends;i++)
167 WetGain[i] = clampf(SourceVolume, MinVolume, MaxVolume);
168 WetGain[i] *= ALSource->Send[i].WetGain;
169 WetGainHF[i] = ALSource->Send[i].WetGainHF;
172 SrcMatrix = ALSource->Params.DryGains;
173 for(i = 0;i < MAXCHANNELS;i++)
175 for(c = 0;c < MAXCHANNELS;c++)
176 SrcMatrix[i][c] = 0.0f;
178 switch(Channels)
180 case FmtMono:
181 angles = angles_Mono;
182 chans = chans_Mono;
183 num_channels = 1;
184 break;
185 case FmtStereo:
186 if(VirtualChannels && (Device->Flags&DEVICE_DUPLICATE_STEREO))
188 DryGain *= aluSqrt(2.0f/4.0f);
189 for(c = 0;c < 2;c++)
191 pos = aluCart2LUTpos(aluCos(F_PI/180.0f * angles_Rear[c]),
192 aluSin(F_PI/180.0f * angles_Rear[c]));
193 SpeakerGain = Device->PanningLUT[pos];
195 for(i = 0;i < (ALint)Device->NumChan;i++)
197 enum Channel chan = Device->Speaker2Chan[i];
198 SrcMatrix[c][chan] += DryGain * ListenerGain *
199 SpeakerGain[chan];
203 angles = angles_Stereo;
204 chans = chans_Stereo;
205 num_channels = 2;
206 break;
208 case FmtRear:
209 angles = angles_Rear;
210 chans = chans_Rear;
211 num_channels = 2;
212 break;
214 case FmtQuad:
215 angles = angles_Quad;
216 chans = chans_Quad;
217 num_channels = 4;
218 break;
220 case FmtX51:
221 angles = angles_X51;
222 chans = chans_X51;
223 num_channels = 6;
224 break;
226 case FmtX61:
227 angles = angles_X61;
228 chans = chans_X61;
229 num_channels = 7;
230 break;
232 case FmtX71:
233 angles = angles_X71;
234 chans = chans_X71;
235 num_channels = 8;
236 break;
239 if(VirtualChannels == AL_FALSE)
241 for(c = 0;c < num_channels;c++)
242 SrcMatrix[c][chans[c]] += DryGain * ListenerGain;
244 else if(Device->Hrtf)
246 for(c = 0;c < num_channels;c++)
248 if(chans[c] == LFE)
250 /* Skip LFE */
251 ALSource->Params.HrtfDelay[c][0] = 0;
252 ALSource->Params.HrtfDelay[c][1] = 0;
253 for(i = 0;i < HRIR_LENGTH;i++)
255 ALSource->Params.HrtfCoeffs[c][i][0] = 0.0f;
256 ALSource->Params.HrtfCoeffs[c][i][1] = 0.0f;
259 else
261 /* Get the static HRIR coefficients and delays for this
262 * channel. */
263 GetLerpedHrtfCoeffs(Device->Hrtf,
264 0.0f, F_PI/180.0f * angles[c],
265 DryGain*ListenerGain,
266 ALSource->Params.HrtfCoeffs[c],
267 ALSource->Params.HrtfDelay[c]);
269 ALSource->HrtfCounter = 0;
272 else
274 for(c = 0;c < num_channels;c++)
276 if(chans[c] == LFE) /* Special-case LFE */
278 SrcMatrix[c][LFE] += DryGain * ListenerGain;
279 continue;
281 pos = aluCart2LUTpos(aluCos(F_PI/180.0f * angles[c]),
282 aluSin(F_PI/180.0f * angles[c]));
283 SpeakerGain = Device->PanningLUT[pos];
285 for(i = 0;i < (ALint)Device->NumChan;i++)
287 enum Channel chan = Device->Speaker2Chan[i];
288 SrcMatrix[c][chan] += DryGain * ListenerGain *
289 SpeakerGain[chan];
293 for(i = 0;i < NumSends;i++)
295 ALSource->Params.Send[i].Slot = ALSource->Send[i].Slot;
296 ALSource->Params.Send[i].WetGain = WetGain[i] * ListenerGain;
299 /* Update filter coefficients. Calculations based on the I3DL2
300 * spec. */
301 cw = aluCos(F_PI*2.0f * LOWPASSFREQREF / Frequency);
303 /* We use two chained one-pole filters, so we need to take the
304 * square root of the squared gain, which is the same as the base
305 * gain. */
306 ALSource->Params.iirFilter.coeff = lpCoeffCalc(DryGainHF, cw);
307 for(i = 0;i < NumSends;i++)
309 /* We use a one-pole filter, so we need to take the squared gain */
310 ALfloat a = lpCoeffCalc(WetGainHF[i]*WetGainHF[i], cw);
311 ALSource->Params.Send[i].iirFilter.coeff = a;
315 ALvoid CalcSourceParams(ALsource *ALSource, const ALCcontext *ALContext)
317 const ALCdevice *Device = ALContext->Device;
318 ALfloat InnerAngle,OuterAngle,Angle,Distance,ClampedDist;
319 ALfloat Direction[3],Position[3],SourceToListener[3];
320 ALfloat Velocity[3],ListenerVel[3];
321 ALfloat MinVolume,MaxVolume,MinDist,MaxDist,Rolloff;
322 ALfloat ConeVolume,ConeHF,SourceVolume,ListenerGain;
323 ALfloat DopplerFactor, DopplerVelocity, SpeedOfSound;
324 ALfloat AirAbsorptionFactor;
325 ALfloat RoomAirAbsorption[MAX_SENDS];
326 ALbufferlistitem *BufferListItem;
327 ALfloat Attenuation, EffectiveDist;
328 ALfloat RoomAttenuation[MAX_SENDS];
329 ALfloat MetersPerUnit;
330 ALfloat RoomRolloffBase;
331 ALfloat RoomRolloff[MAX_SENDS];
332 ALfloat DecayDistance[MAX_SENDS];
333 ALfloat DryGain;
334 ALfloat DryGainHF;
335 ALboolean DryGainHFAuto;
336 ALfloat WetGain[MAX_SENDS];
337 ALfloat WetGainHF[MAX_SENDS];
338 ALboolean WetGainAuto;
339 ALboolean WetGainHFAuto;
340 enum Resampler Resampler;
341 ALfloat Pitch;
342 ALuint Frequency;
343 ALint NumSends;
344 ALfloat cw;
345 ALint i;
347 DryGainHF = 1.0f;
348 for(i = 0;i < MAX_SENDS;i++)
349 WetGainHF[i] = 1.0f;
351 //Get context properties
352 DopplerFactor = ALContext->DopplerFactor * ALSource->DopplerFactor;
353 DopplerVelocity = ALContext->DopplerVelocity;
354 SpeedOfSound = ALContext->flSpeedOfSound;
355 NumSends = Device->NumAuxSends;
356 Frequency = Device->Frequency;
358 //Get listener properties
359 ListenerGain = ALContext->Listener.Gain;
360 MetersPerUnit = ALContext->Listener.MetersPerUnit;
361 ListenerVel[0] = ALContext->Listener.Velocity[0];
362 ListenerVel[1] = ALContext->Listener.Velocity[1];
363 ListenerVel[2] = ALContext->Listener.Velocity[2];
365 //Get source properties
366 SourceVolume = ALSource->flGain;
367 MinVolume = ALSource->flMinGain;
368 MaxVolume = ALSource->flMaxGain;
369 Pitch = ALSource->flPitch;
370 Resampler = ALSource->Resampler;
371 Position[0] = ALSource->vPosition[0];
372 Position[1] = ALSource->vPosition[1];
373 Position[2] = ALSource->vPosition[2];
374 Direction[0] = ALSource->vOrientation[0];
375 Direction[1] = ALSource->vOrientation[1];
376 Direction[2] = ALSource->vOrientation[2];
377 Velocity[0] = ALSource->vVelocity[0];
378 Velocity[1] = ALSource->vVelocity[1];
379 Velocity[2] = ALSource->vVelocity[2];
380 MinDist = ALSource->flRefDistance;
381 MaxDist = ALSource->flMaxDistance;
382 Rolloff = ALSource->flRollOffFactor;
383 InnerAngle = ALSource->flInnerAngle * ConeScale;
384 OuterAngle = ALSource->flOuterAngle * ConeScale;
385 AirAbsorptionFactor = ALSource->AirAbsorptionFactor;
386 DryGainHFAuto = ALSource->DryGainHFAuto;
387 WetGainAuto = ALSource->WetGainAuto;
388 WetGainHFAuto = ALSource->WetGainHFAuto;
389 RoomRolloffBase = ALSource->RoomRolloffFactor;
390 for(i = 0;i < NumSends;i++)
392 ALeffectslot *Slot = ALSource->Send[i].Slot;
394 if(!Slot || Slot->effect.type == AL_EFFECT_NULL)
396 RoomRolloff[i] = 0.0f;
397 DecayDistance[i] = 0.0f;
398 RoomAirAbsorption[i] = 1.0f;
400 else if(Slot->AuxSendAuto)
402 RoomRolloff[i] = RoomRolloffBase;
403 if(IsReverbEffect(Slot->effect.type))
405 RoomRolloff[i] += Slot->effect.Reverb.RoomRolloffFactor;
406 DecayDistance[i] = Slot->effect.Reverb.DecayTime *
407 SPEEDOFSOUNDMETRESPERSEC;
408 RoomAirAbsorption[i] = Slot->effect.Reverb.AirAbsorptionGainHF;
410 else
412 DecayDistance[i] = 0.0f;
413 RoomAirAbsorption[i] = 1.0f;
416 else
418 /* If the slot's auxiliary send auto is off, the data sent to the
419 * effect slot is the same as the dry path, sans filter effects */
420 RoomRolloff[i] = Rolloff;
421 DecayDistance[i] = 0.0f;
422 RoomAirAbsorption[i] = AIRABSORBGAINHF;
425 ALSource->Params.Send[i].Slot = Slot;
428 //1. Translate Listener to origin (convert to head relative)
429 if(ALSource->bHeadRelative == AL_FALSE)
431 ALfloat Matrix[4][4];
432 for(i = 0;i < 4;i++)
434 ALint i2;
435 for(i2 = 0;i2 < 4;i2++)
436 Matrix[i][i2] = ALContext->Listener.Matrix[i][i2];
439 /* Translate position */
440 Position[0] -= ALContext->Listener.Position[0];
441 Position[1] -= ALContext->Listener.Position[1];
442 Position[2] -= ALContext->Listener.Position[2];
444 /* Transform source vectors into listener space */
445 aluMatrixVector(Position, 1.0f, Matrix);
446 aluMatrixVector(Direction, 0.0f, Matrix);
447 aluMatrixVector(Velocity, 0.0f, Matrix);
449 else
451 ListenerVel[0] = 0.0f;
452 ListenerVel[1] = 0.0f;
453 ListenerVel[2] = 0.0f;
456 SourceToListener[0] = -Position[0];
457 SourceToListener[1] = -Position[1];
458 SourceToListener[2] = -Position[2];
459 aluNormalize(SourceToListener);
460 aluNormalize(Direction);
462 //2. Calculate distance attenuation
463 Distance = aluSqrt(aluDotproduct(Position, Position));
464 ClampedDist = Distance;
466 Attenuation = 1.0f;
467 for(i = 0;i < NumSends;i++)
468 RoomAttenuation[i] = 1.0f;
469 switch(ALContext->SourceDistanceModel ? ALSource->DistanceModel :
470 ALContext->DistanceModel)
472 case InverseDistanceClamped:
473 ClampedDist = clampf(ClampedDist, MinDist, MaxDist);
474 if(MaxDist < MinDist)
475 break;
476 //fall-through
477 case InverseDistance:
478 if(MinDist > 0.0f)
480 if((MinDist + (Rolloff * (ClampedDist - MinDist))) > 0.0f)
481 Attenuation = MinDist / (MinDist + (Rolloff * (ClampedDist - MinDist)));
482 for(i = 0;i < NumSends;i++)
484 if((MinDist + (RoomRolloff[i] * (ClampedDist - MinDist))) > 0.0f)
485 RoomAttenuation[i] = MinDist / (MinDist + (RoomRolloff[i] * (ClampedDist - MinDist)));
488 break;
490 case LinearDistanceClamped:
491 ClampedDist = clampf(ClampedDist, MinDist, MaxDist);
492 if(MaxDist < MinDist)
493 break;
494 //fall-through
495 case LinearDistance:
496 if(MaxDist != MinDist)
498 Attenuation = 1.0f - (Rolloff*(ClampedDist-MinDist)/(MaxDist - MinDist));
499 Attenuation = maxf(Attenuation, 0.0f);
500 for(i = 0;i < NumSends;i++)
502 RoomAttenuation[i] = 1.0f - (RoomRolloff[i]*(ClampedDist-MinDist)/(MaxDist - MinDist));
503 RoomAttenuation[i] = maxf(RoomAttenuation[i], 0.0f);
506 break;
508 case ExponentDistanceClamped:
509 ClampedDist = clampf(ClampedDist, MinDist, MaxDist);
510 if(MaxDist < MinDist)
511 break;
512 //fall-through
513 case ExponentDistance:
514 if(ClampedDist > 0.0f && MinDist > 0.0f)
516 Attenuation = aluPow(ClampedDist/MinDist, -Rolloff);
517 for(i = 0;i < NumSends;i++)
518 RoomAttenuation[i] = aluPow(ClampedDist/MinDist, -RoomRolloff[i]);
520 break;
522 case DisableDistance:
523 break;
526 // Source Gain + Attenuation
527 DryGain = SourceVolume * Attenuation;
528 for(i = 0;i < NumSends;i++)
529 WetGain[i] = SourceVolume * RoomAttenuation[i];
531 // Distance-based air absorption
532 EffectiveDist = 0.0f;
533 if(MinDist > 0.0f && Attenuation < 1.0f)
534 EffectiveDist = (MinDist/Attenuation - MinDist)*MetersPerUnit;
535 if(AirAbsorptionFactor > 0.0f && EffectiveDist > 0.0f)
537 DryGainHF *= aluPow(AIRABSORBGAINHF, AirAbsorptionFactor*EffectiveDist);
538 for(i = 0;i < NumSends;i++)
539 WetGainHF[i] *= aluPow(RoomAirAbsorption[i],
540 AirAbsorptionFactor*EffectiveDist);
543 if(WetGainAuto)
545 /* Apply a decay-time transformation to the wet path, based on the
546 * attenuation of the dry path.
548 * Using the approximate (effective) source to listener distance, the
549 * initial decay of the reverb effect is calculated and applied to the
550 * wet path.
552 for(i = 0;i < NumSends;i++)
554 if(DecayDistance[i] > 0.0f)
555 WetGain[i] *= aluPow(0.001f /* -60dB */,
556 EffectiveDist / DecayDistance[i]);
560 /* Calculate directional soundcones */
561 Angle = aluAcos(aluDotproduct(Direction,SourceToListener)) * (180.0f/F_PI);
562 if(Angle >= InnerAngle && Angle <= OuterAngle)
564 ALfloat scale = (Angle-InnerAngle) / (OuterAngle-InnerAngle);
565 ConeVolume = lerp(1.0f, ALSource->flOuterGain, scale);
566 ConeHF = lerp(1.0f, ALSource->OuterGainHF, scale);
568 else if(Angle > OuterAngle)
570 ConeVolume = ALSource->flOuterGain;
571 ConeHF = ALSource->OuterGainHF;
573 else
575 ConeVolume = 1.0f;
576 ConeHF = 1.0f;
579 DryGain *= ConeVolume;
580 if(WetGainAuto)
582 for(i = 0;i < NumSends;i++)
583 WetGain[i] *= ConeVolume;
585 if(DryGainHFAuto)
586 DryGainHF *= ConeHF;
587 if(WetGainHFAuto)
589 for(i = 0;i < NumSends;i++)
590 WetGainHF[i] *= ConeHF;
593 // Clamp to Min/Max Gain
594 DryGain = clampf(DryGain, MinVolume, MaxVolume);
595 for(i = 0;i < NumSends;i++)
596 WetGain[i] = clampf(WetGain[i], MinVolume, MaxVolume);
598 // Apply filter gains and filters
599 DryGain *= ALSource->DirectGain * ListenerGain;
600 DryGainHF *= ALSource->DirectGainHF;
601 for(i = 0;i < NumSends;i++)
603 WetGain[i] *= ALSource->Send[i].WetGain * ListenerGain;
604 WetGainHF[i] *= ALSource->Send[i].WetGainHF;
607 // Calculate Velocity
608 if(DopplerFactor != 0.0f)
610 ALfloat VSS, VLS;
611 ALfloat MaxVelocity = (SpeedOfSound*DopplerVelocity) /
612 DopplerFactor;
614 VSS = aluDotproduct(Velocity, SourceToListener);
615 if(VSS >= MaxVelocity)
616 VSS = (MaxVelocity - 1.0f);
617 else if(VSS <= -MaxVelocity)
618 VSS = -MaxVelocity + 1.0f;
620 VLS = aluDotproduct(ListenerVel, SourceToListener);
621 if(VLS >= MaxVelocity)
622 VLS = (MaxVelocity - 1.0f);
623 else if(VLS <= -MaxVelocity)
624 VLS = -MaxVelocity + 1.0f;
626 Pitch *= ((SpeedOfSound*DopplerVelocity) - (DopplerFactor*VLS)) /
627 ((SpeedOfSound*DopplerVelocity) - (DopplerFactor*VSS));
630 BufferListItem = ALSource->queue;
631 while(BufferListItem != NULL)
633 ALbuffer *ALBuffer;
634 if((ALBuffer=BufferListItem->buffer) != NULL)
636 ALint maxstep = STACK_DATA_SIZE / ALSource->NumChannels /
637 ALSource->SampleSize;
638 maxstep -= ResamplerPadding[Resampler] +
639 ResamplerPrePadding[Resampler] + 1;
640 maxstep = mini(maxstep, INT_MAX>>FRACTIONBITS);
642 Pitch = Pitch * ALBuffer->Frequency / Frequency;
643 if(Pitch > (ALfloat)maxstep)
644 ALSource->Params.Step = maxstep<<FRACTIONBITS;
645 else
647 ALSource->Params.Step = fastf2i(Pitch*FRACTIONONE);
648 if(ALSource->Params.Step == 0)
649 ALSource->Params.Step = 1;
652 break;
654 BufferListItem = BufferListItem->next;
656 if(Device->Hrtf)
657 ALSource->Params.DoMix = SelectHrtfMixer((ALSource->Params.Step==FRACTIONONE) ?
658 POINT_RESAMPLER : Resampler);
659 else
660 ALSource->Params.DoMix = SelectMixer((ALSource->Params.Step==FRACTIONONE) ?
661 POINT_RESAMPLER : Resampler);
663 if(Device->Hrtf)
665 // Use a binaural HRTF algorithm for stereo headphone playback
666 ALfloat delta, ev = 0.0f, az = 0.0f;
668 if(Distance > 0.0f)
670 ALfloat invlen = 1.0f/Distance;
671 Position[0] *= invlen;
672 Position[1] *= invlen;
673 Position[2] *= invlen;
675 // Calculate elevation and azimuth only when the source is not at
676 // the listener. This prevents +0 and -0 Z from producing
677 // inconsistent panning.
678 ev = aluAsin(Position[1]);
679 az = aluAtan2(Position[0], -Position[2]*ZScale);
682 // Check to see if the HRIR is already moving.
683 if(ALSource->HrtfMoving)
685 // Calculate the normalized HRTF transition factor (delta).
686 delta = CalcHrtfDelta(ALSource->Params.HrtfGain, DryGain,
687 ALSource->Params.HrtfDir, Position);
688 // If the delta is large enough, get the moving HRIR target
689 // coefficients, target delays, steppping values, and counter.
690 if(delta > 0.001f)
692 ALSource->HrtfCounter = GetMovingHrtfCoeffs(Device->Hrtf,
693 ev, az, DryGain, delta,
694 ALSource->HrtfCounter,
695 ALSource->Params.HrtfCoeffs[0],
696 ALSource->Params.HrtfDelay[0],
697 ALSource->Params.HrtfCoeffStep,
698 ALSource->Params.HrtfDelayStep);
699 ALSource->Params.HrtfGain = DryGain;
700 ALSource->Params.HrtfDir[0] = Position[0];
701 ALSource->Params.HrtfDir[1] = Position[1];
702 ALSource->Params.HrtfDir[2] = Position[2];
705 else
707 // Get the initial (static) HRIR coefficients and delays.
708 GetLerpedHrtfCoeffs(Device->Hrtf, ev, az, DryGain,
709 ALSource->Params.HrtfCoeffs[0],
710 ALSource->Params.HrtfDelay[0]);
711 ALSource->HrtfCounter = 0;
712 ALSource->Params.HrtfGain = DryGain;
713 ALSource->Params.HrtfDir[0] = Position[0];
714 ALSource->Params.HrtfDir[1] = Position[1];
715 ALSource->Params.HrtfDir[2] = Position[2];
718 else
720 // Use energy-preserving panning algorithm for multi-speaker playback
721 ALfloat DirGain, AmbientGain;
722 const ALfloat *SpeakerGain;
723 ALfloat length;
724 ALint pos;
726 length = maxf(Distance, MinDist);
727 if(length > 0.0f)
729 ALfloat invlen = 1.0f/length;
730 Position[0] *= invlen;
731 Position[1] *= invlen;
732 Position[2] *= invlen;
735 pos = aluCart2LUTpos(-Position[2]*ZScale, Position[0]);
736 SpeakerGain = Device->PanningLUT[pos];
738 DirGain = aluSqrt(Position[0]*Position[0] + Position[2]*Position[2]);
739 // elevation adjustment for directional gain. this sucks, but
740 // has low complexity
741 AmbientGain = aluSqrt(1.0f/Device->NumChan);
742 for(i = 0;i < MAXCHANNELS;i++)
744 ALuint i2;
745 for(i2 = 0;i2 < MAXCHANNELS;i2++)
746 ALSource->Params.DryGains[i][i2] = 0.0f;
748 for(i = 0;i < (ALint)Device->NumChan;i++)
750 enum Channel chan = Device->Speaker2Chan[i];
751 ALfloat gain = lerp(AmbientGain, SpeakerGain[chan], DirGain);
752 ALSource->Params.DryGains[0][chan] = DryGain * gain;
755 for(i = 0;i < NumSends;i++)
756 ALSource->Params.Send[i].WetGain = WetGain[i];
758 /* Update filter coefficients. */
759 cw = aluCos(F_PI*2.0f * LOWPASSFREQREF / Frequency);
761 ALSource->Params.iirFilter.coeff = lpCoeffCalc(DryGainHF, cw);
762 for(i = 0;i < NumSends;i++)
764 ALfloat a = lpCoeffCalc(WetGainHF[i]*WetGainHF[i], cw);
765 ALSource->Params.Send[i].iirFilter.coeff = a;
770 static __inline ALfloat aluF2F(ALfloat val)
771 { return val; }
772 static __inline ALshort aluF2S(ALfloat val)
774 if(val > 1.0f) return 32767;
775 if(val < -1.0f) return -32768;
776 return fastf2i(val*32767.0f);
778 static __inline ALushort aluF2US(ALfloat val)
779 { return aluF2S(val)+32768; }
780 static __inline ALbyte aluF2B(ALfloat val)
781 { return aluF2S(val)>>8; }
782 static __inline ALubyte aluF2UB(ALfloat val)
783 { return aluF2US(val)>>8; }
785 #define DECL_TEMPLATE(T, N, func) \
786 static void Write_##T##_##N(ALCdevice *device, T *RESTRICT buffer, \
787 ALuint SamplesToDo) \
789 ALfloat (*RESTRICT DryBuffer)[MAXCHANNELS] = device->DryBuffer; \
790 const enum Channel *ChanMap = device->DevChannels; \
791 ALuint i, j; \
793 for(i = 0;i < SamplesToDo;i++) \
795 for(j = 0;j < N;j++) \
796 *(buffer++) = func(DryBuffer[i][ChanMap[j]]); \
800 DECL_TEMPLATE(ALfloat, 1, aluF2F)
801 DECL_TEMPLATE(ALfloat, 4, aluF2F)
802 DECL_TEMPLATE(ALfloat, 6, aluF2F)
803 DECL_TEMPLATE(ALfloat, 7, aluF2F)
804 DECL_TEMPLATE(ALfloat, 8, aluF2F)
806 DECL_TEMPLATE(ALushort, 1, aluF2US)
807 DECL_TEMPLATE(ALushort, 4, aluF2US)
808 DECL_TEMPLATE(ALushort, 6, aluF2US)
809 DECL_TEMPLATE(ALushort, 7, aluF2US)
810 DECL_TEMPLATE(ALushort, 8, aluF2US)
812 DECL_TEMPLATE(ALshort, 1, aluF2S)
813 DECL_TEMPLATE(ALshort, 4, aluF2S)
814 DECL_TEMPLATE(ALshort, 6, aluF2S)
815 DECL_TEMPLATE(ALshort, 7, aluF2S)
816 DECL_TEMPLATE(ALshort, 8, aluF2S)
818 DECL_TEMPLATE(ALubyte, 1, aluF2UB)
819 DECL_TEMPLATE(ALubyte, 4, aluF2UB)
820 DECL_TEMPLATE(ALubyte, 6, aluF2UB)
821 DECL_TEMPLATE(ALubyte, 7, aluF2UB)
822 DECL_TEMPLATE(ALubyte, 8, aluF2UB)
824 DECL_TEMPLATE(ALbyte, 1, aluF2B)
825 DECL_TEMPLATE(ALbyte, 4, aluF2B)
826 DECL_TEMPLATE(ALbyte, 6, aluF2B)
827 DECL_TEMPLATE(ALbyte, 7, aluF2B)
828 DECL_TEMPLATE(ALbyte, 8, aluF2B)
830 #undef DECL_TEMPLATE
832 #define DECL_TEMPLATE(T, N, func) \
833 static void Write_##T##_##N(ALCdevice *device, T *RESTRICT buffer, \
834 ALuint SamplesToDo) \
836 ALfloat (*RESTRICT DryBuffer)[MAXCHANNELS] = device->DryBuffer; \
837 const enum Channel *ChanMap = device->DevChannels; \
838 ALuint i, j; \
840 if(device->Bs2b) \
842 for(i = 0;i < SamplesToDo;i++) \
844 float samples[2]; \
845 samples[0] = DryBuffer[i][ChanMap[0]]; \
846 samples[1] = DryBuffer[i][ChanMap[1]]; \
847 bs2b_cross_feed(device->Bs2b, samples); \
848 *(buffer++) = func(samples[0]); \
849 *(buffer++) = func(samples[1]); \
852 else \
854 for(i = 0;i < SamplesToDo;i++) \
856 for(j = 0;j < N;j++) \
857 *(buffer++) = func(DryBuffer[i][ChanMap[j]]); \
862 DECL_TEMPLATE(ALfloat, 2, aluF2F)
863 DECL_TEMPLATE(ALushort, 2, aluF2US)
864 DECL_TEMPLATE(ALshort, 2, aluF2S)
865 DECL_TEMPLATE(ALubyte, 2, aluF2UB)
866 DECL_TEMPLATE(ALbyte, 2, aluF2B)
868 #undef DECL_TEMPLATE
870 #define DECL_TEMPLATE(T) \
871 static void Write_##T(ALCdevice *device, T *buffer, ALuint SamplesToDo) \
873 switch(device->FmtChans) \
875 case DevFmtMono: \
876 Write_##T##_1(device, buffer, SamplesToDo); \
877 break; \
878 case DevFmtStereo: \
879 Write_##T##_2(device, buffer, SamplesToDo); \
880 break; \
881 case DevFmtQuad: \
882 Write_##T##_4(device, buffer, SamplesToDo); \
883 break; \
884 case DevFmtX51: \
885 case DevFmtX51Side: \
886 Write_##T##_6(device, buffer, SamplesToDo); \
887 break; \
888 case DevFmtX61: \
889 Write_##T##_7(device, buffer, SamplesToDo); \
890 break; \
891 case DevFmtX71: \
892 Write_##T##_8(device, buffer, SamplesToDo); \
893 break; \
897 DECL_TEMPLATE(ALfloat)
898 DECL_TEMPLATE(ALushort)
899 DECL_TEMPLATE(ALshort)
900 DECL_TEMPLATE(ALubyte)
901 DECL_TEMPLATE(ALbyte)
903 #undef DECL_TEMPLATE
905 ALvoid aluMixData(ALCdevice *device, ALvoid *buffer, ALsizei size)
907 ALuint SamplesToDo;
908 ALeffectslot **slot, **slot_end;
909 ALsource **src, **src_end;
910 ALCcontext *ctx;
911 int fpuState;
912 ALuint i, c;
914 fpuState = SetMixerFPUMode();
916 while(size > 0)
918 /* Setup variables */
919 SamplesToDo = minu(size, BUFFERSIZE);
921 /* Clear mixing buffer */
922 memset(device->DryBuffer, 0, SamplesToDo*MAXCHANNELS*sizeof(ALfloat));
924 LockDevice(device);
925 ctx = device->ContextList;
926 while(ctx)
928 ALenum DeferUpdates = ctx->DeferUpdates;
929 ALenum UpdateSources = AL_FALSE;
931 if(!DeferUpdates)
932 UpdateSources = ExchangeInt(&ctx->UpdateSources, AL_FALSE);
934 src = ctx->ActiveSources;
935 src_end = src + ctx->ActiveSourceCount;
936 while(src != src_end)
938 if((*src)->state != AL_PLAYING)
940 --(ctx->ActiveSourceCount);
941 *src = *(--src_end);
942 continue;
945 if(!DeferUpdates && (ExchangeInt(&(*src)->NeedsUpdate, AL_FALSE) ||
946 UpdateSources))
947 ALsource_Update(*src, ctx);
949 MixSource(*src, device, SamplesToDo);
950 src++;
953 /* effect slot processing */
954 slot = ctx->ActiveEffectSlots;
955 slot_end = slot + ctx->ActiveEffectSlotCount;
956 while(slot != slot_end)
958 for(c = 0;c < SamplesToDo;c++)
960 (*slot)->WetBuffer[c] += (*slot)->ClickRemoval[0];
961 (*slot)->ClickRemoval[0] -= (*slot)->ClickRemoval[0] * (1.0f/256.0f);
963 (*slot)->ClickRemoval[0] += (*slot)->PendingClicks[0];
964 (*slot)->PendingClicks[0] = 0.0f;
966 if(!DeferUpdates && ExchangeInt(&(*slot)->NeedsUpdate, AL_FALSE))
967 ALeffectState_Update((*slot)->EffectState, ctx, *slot);
969 ALeffectState_Process((*slot)->EffectState, SamplesToDo,
970 (*slot)->WetBuffer, device->DryBuffer);
972 for(i = 0;i < SamplesToDo;i++)
973 (*slot)->WetBuffer[i] = 0.0f;
975 slot++;
978 ctx = ctx->next;
980 UnlockDevice(device);
982 //Post processing loop
983 if(device->FmtChans == DevFmtMono)
985 for(i = 0;i < SamplesToDo;i++)
987 device->DryBuffer[i][FRONT_CENTER] += device->ClickRemoval[FRONT_CENTER];
988 device->ClickRemoval[FRONT_CENTER] -= device->ClickRemoval[FRONT_CENTER] * (1.0f/256.0f);
990 device->ClickRemoval[FRONT_CENTER] += device->PendingClicks[FRONT_CENTER];
991 device->PendingClicks[FRONT_CENTER] = 0.0f;
993 else if(device->FmtChans == DevFmtStereo)
995 /* Assumes the first two channels are FRONT_LEFT and FRONT_RIGHT */
996 for(i = 0;i < SamplesToDo;i++)
998 for(c = 0;c < 2;c++)
1000 device->DryBuffer[i][c] += device->ClickRemoval[c];
1001 device->ClickRemoval[c] -= device->ClickRemoval[c] * (1.0f/256.0f);
1004 for(c = 0;c < 2;c++)
1006 device->ClickRemoval[c] += device->PendingClicks[c];
1007 device->PendingClicks[c] = 0.0f;
1010 else
1012 for(i = 0;i < SamplesToDo;i++)
1014 for(c = 0;c < MAXCHANNELS;c++)
1016 device->DryBuffer[i][c] += device->ClickRemoval[c];
1017 device->ClickRemoval[c] -= device->ClickRemoval[c] * (1.0f/256.0f);
1020 for(c = 0;c < MAXCHANNELS;c++)
1022 device->ClickRemoval[c] += device->PendingClicks[c];
1023 device->PendingClicks[c] = 0.0f;
1027 if(buffer)
1029 switch(device->FmtType)
1031 case DevFmtByte:
1032 Write_ALbyte(device, buffer, SamplesToDo);
1033 break;
1034 case DevFmtUByte:
1035 Write_ALubyte(device, buffer, SamplesToDo);
1036 break;
1037 case DevFmtShort:
1038 Write_ALshort(device, buffer, SamplesToDo);
1039 break;
1040 case DevFmtUShort:
1041 Write_ALushort(device, buffer, SamplesToDo);
1042 break;
1043 case DevFmtFloat:
1044 Write_ALfloat(device, buffer, SamplesToDo);
1045 break;
1049 size -= SamplesToDo;
1052 RestoreFPUMode(fpuState);
1056 ALvoid aluHandleDisconnect(ALCdevice *device)
1058 ALCcontext *Context;
1060 LockDevice(device);
1061 device->Connected = ALC_FALSE;
1063 Context = device->ContextList;
1064 while(Context)
1066 ALsource **src, **src_end;
1068 src = Context->ActiveSources;
1069 src_end = src + Context->ActiveSourceCount;
1070 while(src != src_end)
1072 if((*src)->state == AL_PLAYING)
1074 (*src)->state = AL_STOPPED;
1075 (*src)->BuffersPlayed = (*src)->BuffersInQueue;
1076 (*src)->position = 0;
1077 (*src)->position_fraction = 0;
1079 src++;
1081 Context->ActiveSourceCount = 0;
1083 Context = Context->next;
1085 UnlockDevice(device);