Base air absorption on distance attenuation, try 2
[openal-soft.git] / Alc / ALu.c
blob01621f6f76f261e5943f41bb705d5398fbf13d6c
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 "alThunk.h"
35 #include "alListener.h"
36 #include "alAuxEffectSlot.h"
37 #include "alu.h"
38 #include "bs2b.h"
40 #if defined(HAVE_STDINT_H)
41 #include <stdint.h>
42 typedef int64_t ALint64;
43 #elif defined(HAVE___INT64)
44 typedef __int64 ALint64;
45 #elif (SIZEOF_LONG == 8)
46 typedef long ALint64;
47 #elif (SIZEOF_LONG_LONG == 8)
48 typedef long long ALint64;
49 #endif
51 #define FRACTIONBITS 14
52 #define FRACTIONMASK ((1L<<FRACTIONBITS)-1)
53 #define MAX_PITCH 65536
55 /* Minimum ramp length in milliseconds. The value below was chosen to
56 * adequately reduce clicks and pops from harsh gain changes. */
57 #define MIN_RAMP_LENGTH 16
59 ALboolean DuplicateStereo = AL_FALSE;
62 static __inline ALfloat aluF2F(ALfloat Value)
64 if(Value < 0.f) return Value/32768.f;
65 if(Value > 0.f) return Value/32767.f;
66 return 0.f;
69 static __inline ALshort aluF2S(ALfloat Value)
71 ALint i;
73 i = (ALint)Value;
74 i = __min( 32767, i);
75 i = __max(-32768, i);
76 return ((ALshort)i);
79 static __inline ALubyte aluF2UB(ALfloat Value)
81 ALshort i = aluF2S(Value);
82 return (i>>8)+128;
86 static __inline ALvoid aluCrossproduct(const ALfloat *inVector1, const ALfloat *inVector2, ALfloat *outVector)
88 outVector[0] = inVector1[1]*inVector2[2] - inVector1[2]*inVector2[1];
89 outVector[1] = inVector1[2]*inVector2[0] - inVector1[0]*inVector2[2];
90 outVector[2] = inVector1[0]*inVector2[1] - inVector1[1]*inVector2[0];
93 static __inline ALfloat aluDotproduct(const ALfloat *inVector1, const ALfloat *inVector2)
95 return inVector1[0]*inVector2[0] + inVector1[1]*inVector2[1] +
96 inVector1[2]*inVector2[2];
99 static __inline ALvoid aluNormalize(ALfloat *inVector)
101 ALfloat length, inverse_length;
103 length = aluSqrt(aluDotproduct(inVector, inVector));
104 if(length != 0.0f)
106 inverse_length = 1.0f/length;
107 inVector[0] *= inverse_length;
108 inVector[1] *= inverse_length;
109 inVector[2] *= inverse_length;
113 static __inline ALvoid aluMatrixVector(ALfloat *vector,ALfloat w,ALfloat matrix[4][4])
115 ALfloat temp[4] = {
116 vector[0], vector[1], vector[2], w
119 vector[0] = temp[0]*matrix[0][0] + temp[1]*matrix[1][0] + temp[2]*matrix[2][0] + temp[3]*matrix[3][0];
120 vector[1] = temp[0]*matrix[0][1] + temp[1]*matrix[1][1] + temp[2]*matrix[2][1] + temp[3]*matrix[3][1];
121 vector[2] = temp[0]*matrix[0][2] + temp[1]*matrix[1][2] + temp[2]*matrix[2][2] + temp[3]*matrix[3][2];
124 static ALvoid SetSpeakerArrangement(const char *name, ALfloat SpeakerAngle[OUTPUTCHANNELS],
125 ALint Speaker2Chan[OUTPUTCHANNELS], ALint chans)
127 const char *confkey;
128 const char *next;
129 const char *sep;
130 const char *end;
131 int i, val;
133 confkey = GetConfigValue(NULL, name, "");
134 next = confkey;
135 while(next && *next)
137 confkey = next;
138 next = strchr(confkey, ',');
139 if(next)
141 do {
142 next++;
143 } while(isspace(*next));
146 sep = strchr(confkey, '=');
147 if(!sep || confkey == sep)
148 continue;
150 end = sep - 1;
151 while(isspace(*end) && end != confkey)
152 end--;
153 end++;
155 if(strncmp(confkey, "fl", end-confkey) == 0)
156 val = FRONT_LEFT;
157 else if(strncmp(confkey, "fr", end-confkey) == 0)
158 val = FRONT_RIGHT;
159 else if(strncmp(confkey, "fc", end-confkey) == 0)
160 val = FRONT_CENTER;
161 else if(strncmp(confkey, "bl", end-confkey) == 0)
162 val = BACK_LEFT;
163 else if(strncmp(confkey, "br", end-confkey) == 0)
164 val = BACK_RIGHT;
165 else if(strncmp(confkey, "bc", end-confkey) == 0)
166 val = BACK_CENTER;
167 else if(strncmp(confkey, "sl", end-confkey) == 0)
168 val = SIDE_LEFT;
169 else if(strncmp(confkey, "sr", end-confkey) == 0)
170 val = SIDE_RIGHT;
171 else
173 AL_PRINT("Unknown speaker for %s: \"%c%c\"\n", name, confkey[0], confkey[1]);
174 continue;
177 sep++;
178 while(isspace(*sep))
179 sep++;
181 for(i = 0;i < chans;i++)
183 if(Speaker2Chan[i] == val)
185 val = strtol(sep, NULL, 10);
186 if(val >= -180 && val <= 180)
187 SpeakerAngle[i] = val * M_PI/180.0f;
188 else
189 AL_PRINT("Invalid angle for speaker \"%c%c\": %d\n", confkey[0], confkey[1], val);
190 break;
195 for(i = 1;i < chans;i++)
197 if(SpeakerAngle[i] <= SpeakerAngle[i-1])
199 AL_PRINT("Speaker %d of %d does not follow previous: %f > %f\n", i, chans,
200 SpeakerAngle[i-1] * 180.0f/M_PI, SpeakerAngle[i] * 180.0f/M_PI);
201 SpeakerAngle[i] = SpeakerAngle[i-1] + 1 * 180.0f/M_PI;
206 static __inline ALfloat aluLUTpos2Angle(ALint pos)
208 if(pos < QUADRANT_NUM)
209 return aluAtan((ALfloat)pos / (ALfloat)(QUADRANT_NUM - pos));
210 if(pos < 2 * QUADRANT_NUM)
211 return M_PI_2 + aluAtan((ALfloat)(pos - QUADRANT_NUM) / (ALfloat)(2 * QUADRANT_NUM - pos));
212 if(pos < 3 * QUADRANT_NUM)
213 return aluAtan((ALfloat)(pos - 2 * QUADRANT_NUM) / (ALfloat)(3 * QUADRANT_NUM - pos)) - M_PI;
214 return aluAtan((ALfloat)(pos - 3 * QUADRANT_NUM) / (ALfloat)(4 * QUADRANT_NUM - pos)) - M_PI_2;
217 ALvoid aluInitPanning(ALCcontext *Context)
219 ALint pos, offset, s;
220 ALfloat Alpha, Theta;
221 ALfloat SpeakerAngle[OUTPUTCHANNELS];
222 ALint Speaker2Chan[OUTPUTCHANNELS];
224 for(s = 0;s < OUTPUTCHANNELS;s++)
226 int s2;
227 for(s2 = 0;s2 < OUTPUTCHANNELS;s2++)
228 Context->ChannelMatrix[s][s2] = ((s==s2) ? 1.0f : 0.0f);
231 switch(Context->Device->Format)
233 /* Mono is rendered as stereo, then downmixed during post-process */
234 case AL_FORMAT_MONO8:
235 case AL_FORMAT_MONO16:
236 case AL_FORMAT_MONO_FLOAT32:
237 Context->ChannelMatrix[FRONT_CENTER][FRONT_LEFT] = aluSqrt(0.5);
238 Context->ChannelMatrix[FRONT_CENTER][FRONT_RIGHT] = aluSqrt(0.5);
239 Context->ChannelMatrix[SIDE_LEFT][FRONT_LEFT] = 1.0f;
240 Context->ChannelMatrix[SIDE_RIGHT][FRONT_RIGHT] = 1.0f;
241 Context->ChannelMatrix[BACK_LEFT][FRONT_LEFT] = 1.0f;
242 Context->ChannelMatrix[BACK_RIGHT][FRONT_RIGHT] = 1.0f;
243 Context->ChannelMatrix[BACK_CENTER][FRONT_LEFT] = aluSqrt(0.5);
244 Context->ChannelMatrix[BACK_CENTER][FRONT_RIGHT] = aluSqrt(0.5);
245 Context->NumChan = 2;
246 Speaker2Chan[0] = FRONT_LEFT;
247 Speaker2Chan[1] = FRONT_RIGHT;
248 SpeakerAngle[0] = -90.0f * M_PI/180.0f;
249 SpeakerAngle[1] = 90.0f * M_PI/180.0f;
250 break;
252 case AL_FORMAT_STEREO8:
253 case AL_FORMAT_STEREO16:
254 case AL_FORMAT_STEREO_FLOAT32:
255 Context->ChannelMatrix[FRONT_CENTER][FRONT_LEFT] = aluSqrt(0.5);
256 Context->ChannelMatrix[FRONT_CENTER][FRONT_RIGHT] = aluSqrt(0.5);
257 Context->ChannelMatrix[SIDE_LEFT][FRONT_LEFT] = 1.0f;
258 Context->ChannelMatrix[SIDE_RIGHT][FRONT_RIGHT] = 1.0f;
259 Context->ChannelMatrix[BACK_LEFT][FRONT_LEFT] = 1.0f;
260 Context->ChannelMatrix[BACK_RIGHT][FRONT_RIGHT] = 1.0f;
261 Context->ChannelMatrix[BACK_CENTER][FRONT_LEFT] = aluSqrt(0.5);
262 Context->ChannelMatrix[BACK_CENTER][FRONT_RIGHT] = aluSqrt(0.5);
263 Context->NumChan = 2;
264 Speaker2Chan[0] = FRONT_LEFT;
265 Speaker2Chan[1] = FRONT_RIGHT;
266 SpeakerAngle[0] = -90.0f * M_PI/180.0f;
267 SpeakerAngle[1] = 90.0f * M_PI/180.0f;
268 SetSpeakerArrangement("layout_STEREO", SpeakerAngle, Speaker2Chan, Context->NumChan);
269 break;
271 case AL_FORMAT_QUAD8:
272 case AL_FORMAT_QUAD16:
273 case AL_FORMAT_QUAD32:
274 Context->ChannelMatrix[FRONT_CENTER][FRONT_LEFT] = aluSqrt(0.5);
275 Context->ChannelMatrix[FRONT_CENTER][FRONT_RIGHT] = aluSqrt(0.5);
276 Context->ChannelMatrix[SIDE_LEFT][FRONT_LEFT] = aluSqrt(0.5);
277 Context->ChannelMatrix[SIDE_LEFT][BACK_LEFT] = aluSqrt(0.5);
278 Context->ChannelMatrix[SIDE_RIGHT][FRONT_RIGHT] = aluSqrt(0.5);
279 Context->ChannelMatrix[SIDE_RIGHT][BACK_RIGHT] = aluSqrt(0.5);
280 Context->ChannelMatrix[BACK_CENTER][BACK_LEFT] = aluSqrt(0.5);
281 Context->ChannelMatrix[BACK_CENTER][BACK_RIGHT] = aluSqrt(0.5);
282 Context->NumChan = 4;
283 Speaker2Chan[0] = BACK_LEFT;
284 Speaker2Chan[1] = FRONT_LEFT;
285 Speaker2Chan[2] = FRONT_RIGHT;
286 Speaker2Chan[3] = BACK_RIGHT;
287 SpeakerAngle[0] = -135.0f * M_PI/180.0f;
288 SpeakerAngle[1] = -45.0f * M_PI/180.0f;
289 SpeakerAngle[2] = 45.0f * M_PI/180.0f;
290 SpeakerAngle[3] = 135.0f * M_PI/180.0f;
291 SetSpeakerArrangement("layout_QUAD", SpeakerAngle, Speaker2Chan, Context->NumChan);
292 break;
294 case AL_FORMAT_51CHN8:
295 case AL_FORMAT_51CHN16:
296 case AL_FORMAT_51CHN32:
297 Context->ChannelMatrix[SIDE_LEFT][FRONT_LEFT] = aluSqrt(0.5);
298 Context->ChannelMatrix[SIDE_LEFT][BACK_LEFT] = aluSqrt(0.5);
299 Context->ChannelMatrix[SIDE_RIGHT][FRONT_RIGHT] = aluSqrt(0.5);
300 Context->ChannelMatrix[SIDE_RIGHT][BACK_RIGHT] = aluSqrt(0.5);
301 Context->ChannelMatrix[BACK_CENTER][BACK_LEFT] = aluSqrt(0.5);
302 Context->ChannelMatrix[BACK_CENTER][BACK_RIGHT] = aluSqrt(0.5);
303 Context->NumChan = 5;
304 Speaker2Chan[0] = BACK_LEFT;
305 Speaker2Chan[1] = FRONT_LEFT;
306 Speaker2Chan[2] = FRONT_CENTER;
307 Speaker2Chan[3] = FRONT_RIGHT;
308 Speaker2Chan[4] = BACK_RIGHT;
309 SpeakerAngle[0] = -110.0f * M_PI/180.0f;
310 SpeakerAngle[1] = -30.0f * M_PI/180.0f;
311 SpeakerAngle[2] = 0.0f * M_PI/180.0f;
312 SpeakerAngle[3] = 30.0f * M_PI/180.0f;
313 SpeakerAngle[4] = 110.0f * M_PI/180.0f;
314 SetSpeakerArrangement("layout_51CHN", SpeakerAngle, Speaker2Chan, Context->NumChan);
315 break;
317 case AL_FORMAT_61CHN8:
318 case AL_FORMAT_61CHN16:
319 case AL_FORMAT_61CHN32:
320 Context->ChannelMatrix[BACK_LEFT][BACK_CENTER] = aluSqrt(0.5);
321 Context->ChannelMatrix[BACK_LEFT][SIDE_LEFT] = aluSqrt(0.5);
322 Context->ChannelMatrix[BACK_RIGHT][BACK_CENTER] = aluSqrt(0.5);
323 Context->ChannelMatrix[BACK_RIGHT][SIDE_RIGHT] = aluSqrt(0.5);
324 Context->NumChan = 6;
325 Speaker2Chan[0] = SIDE_LEFT;
326 Speaker2Chan[1] = FRONT_LEFT;
327 Speaker2Chan[2] = FRONT_CENTER;
328 Speaker2Chan[3] = FRONT_RIGHT;
329 Speaker2Chan[4] = SIDE_RIGHT;
330 Speaker2Chan[5] = BACK_CENTER;
331 SpeakerAngle[0] = -90.0f * M_PI/180.0f;
332 SpeakerAngle[1] = -30.0f * M_PI/180.0f;
333 SpeakerAngle[2] = 0.0f * M_PI/180.0f;
334 SpeakerAngle[3] = 30.0f * M_PI/180.0f;
335 SpeakerAngle[4] = 90.0f * M_PI/180.0f;
336 SpeakerAngle[5] = 180.0f * M_PI/180.0f;
337 SetSpeakerArrangement("layout_61CHN", SpeakerAngle, Speaker2Chan, Context->NumChan);
338 break;
340 case AL_FORMAT_71CHN8:
341 case AL_FORMAT_71CHN16:
342 case AL_FORMAT_71CHN32:
343 Context->ChannelMatrix[BACK_CENTER][BACK_LEFT] = aluSqrt(0.5);
344 Context->ChannelMatrix[BACK_CENTER][BACK_RIGHT] = aluSqrt(0.5);
345 Context->NumChan = 7;
346 Speaker2Chan[0] = BACK_LEFT;
347 Speaker2Chan[1] = SIDE_LEFT;
348 Speaker2Chan[2] = FRONT_LEFT;
349 Speaker2Chan[3] = FRONT_CENTER;
350 Speaker2Chan[4] = FRONT_RIGHT;
351 Speaker2Chan[5] = SIDE_RIGHT;
352 Speaker2Chan[6] = BACK_RIGHT;
353 SpeakerAngle[0] = -150.0f * M_PI/180.0f;
354 SpeakerAngle[1] = -90.0f * M_PI/180.0f;
355 SpeakerAngle[2] = -30.0f * M_PI/180.0f;
356 SpeakerAngle[3] = 0.0f * M_PI/180.0f;
357 SpeakerAngle[4] = 30.0f * M_PI/180.0f;
358 SpeakerAngle[5] = 90.0f * M_PI/180.0f;
359 SpeakerAngle[6] = 150.0f * M_PI/180.0f;
360 SetSpeakerArrangement("layout_71CHN", SpeakerAngle, Speaker2Chan, Context->NumChan);
361 break;
363 default:
364 assert(0);
367 for(pos = 0; pos < LUT_NUM; pos++)
369 /* source angle */
370 Theta = aluLUTpos2Angle(pos);
372 /* clear all values */
373 offset = OUTPUTCHANNELS * pos;
374 for(s = 0; s < OUTPUTCHANNELS; s++)
375 Context->PanningLUT[offset+s] = 0.0f;
377 /* set panning values */
378 for(s = 0; s < Context->NumChan - 1; s++)
380 if(Theta >= SpeakerAngle[s] && Theta < SpeakerAngle[s+1])
382 /* source between speaker s and speaker s+1 */
383 Alpha = M_PI_2 * (Theta-SpeakerAngle[s]) /
384 (SpeakerAngle[s+1]-SpeakerAngle[s]);
385 Context->PanningLUT[offset + Speaker2Chan[s]] = cos(Alpha);
386 Context->PanningLUT[offset + Speaker2Chan[s+1]] = sin(Alpha);
387 break;
390 if(s == Context->NumChan - 1)
392 /* source between last and first speaker */
393 if(Theta < SpeakerAngle[0])
394 Theta += 2.0f * M_PI;
395 Alpha = M_PI_2 * (Theta-SpeakerAngle[s]) /
396 (2.0f * M_PI + SpeakerAngle[0]-SpeakerAngle[s]);
397 Context->PanningLUT[offset + Speaker2Chan[s]] = cos(Alpha);
398 Context->PanningLUT[offset + Speaker2Chan[0]] = sin(Alpha);
403 static ALvoid CalcSourceParams(const ALCcontext *ALContext, ALsource *ALSource,
404 ALboolean isMono)
406 ALfloat InnerAngle,OuterAngle,Angle,Distance,DryMix;
407 ALfloat Direction[3],Position[3],SourceToListener[3];
408 ALfloat Velocity[3],ListenerVel[3];
409 ALfloat MinVolume,MaxVolume,MinDist,MaxDist,Rolloff,OuterGainHF;
410 ALfloat ConeVolume,ConeHF,SourceVolume,ListenerGain;
411 ALfloat DopplerFactor, DopplerVelocity, flSpeedOfSound;
412 ALfloat Matrix[4][4];
413 ALfloat flAttenuation;
414 ALfloat RoomAttenuation[MAX_SENDS];
415 ALfloat MetersPerUnit;
416 ALfloat RoomRolloff[MAX_SENDS];
417 ALfloat DryGainHF = 1.0f;
418 ALfloat WetGain[MAX_SENDS];
419 ALfloat WetGainHF[MAX_SENDS];
420 ALfloat DirGain, AmbientGain;
421 ALfloat length;
422 const ALfloat *SpeakerGain;
423 ALuint Frequency;
424 ALint NumSends;
425 ALint pos, s, i;
426 ALfloat cw, a, g;
428 //Get context properties
429 DopplerFactor = ALContext->DopplerFactor * ALSource->DopplerFactor;
430 DopplerVelocity = ALContext->DopplerVelocity;
431 flSpeedOfSound = ALContext->flSpeedOfSound;
432 NumSends = ALContext->Device->NumAuxSends;
433 Frequency = ALContext->Device->Frequency;
435 //Get listener properties
436 ListenerGain = ALContext->Listener.Gain;
437 MetersPerUnit = ALContext->Listener.MetersPerUnit;
438 memcpy(ListenerVel, ALContext->Listener.Velocity, sizeof(ALContext->Listener.Velocity));
440 //Get source properties
441 SourceVolume = ALSource->flGain;
442 memcpy(Position, ALSource->vPosition, sizeof(ALSource->vPosition));
443 memcpy(Direction, ALSource->vOrientation, sizeof(ALSource->vOrientation));
444 memcpy(Velocity, ALSource->vVelocity, sizeof(ALSource->vVelocity));
445 MinVolume = ALSource->flMinGain;
446 MaxVolume = ALSource->flMaxGain;
447 MinDist = ALSource->flRefDistance;
448 MaxDist = ALSource->flMaxDistance;
449 Rolloff = ALSource->flRollOffFactor;
450 InnerAngle = ALSource->flInnerAngle;
451 OuterAngle = ALSource->flOuterAngle;
452 OuterGainHF = ALSource->OuterGainHF;
454 //Only apply 3D calculations for mono buffers
455 if(isMono == AL_FALSE)
457 //1. Multi-channel buffers always play "normal"
458 ALSource->Params.Pitch = ALSource->flPitch;
460 DryMix = SourceVolume;
461 DryMix = __min(DryMix,MaxVolume);
462 DryMix = __max(DryMix,MinVolume);
464 switch(ALSource->DirectFilter.type)
466 case AL_FILTER_LOWPASS:
467 DryMix *= ALSource->DirectFilter.Gain;
468 DryGainHF *= ALSource->DirectFilter.GainHF;
469 break;
472 ALSource->Params.DryGains[FRONT_LEFT] = DryMix * ListenerGain;
473 ALSource->Params.DryGains[FRONT_RIGHT] = DryMix * ListenerGain;
474 ALSource->Params.DryGains[SIDE_LEFT] = DryMix * ListenerGain;
475 ALSource->Params.DryGains[SIDE_RIGHT] = DryMix * ListenerGain;
476 ALSource->Params.DryGains[BACK_LEFT] = DryMix * ListenerGain;
477 ALSource->Params.DryGains[BACK_RIGHT] = DryMix * ListenerGain;
478 ALSource->Params.DryGains[FRONT_CENTER] = DryMix * ListenerGain;
479 ALSource->Params.DryGains[BACK_CENTER] = DryMix * ListenerGain;
480 ALSource->Params.DryGains[LFE] = DryMix * ListenerGain;
481 for(i = 0;i < MAX_SENDS;i++)
482 ALSource->Params.WetGains[i] = 0.0f;
484 /* Update filter coefficients. Calculations based on the I3DL2
485 * spec. */
486 cw = cos(2.0*M_PI * LOWPASSFREQCUTOFF / Frequency);
487 /* We use two chained one-pole filters, so we need to take the
488 * square root of the squared gain, which is the same as the base
489 * gain. */
490 g = __max(DryGainHF, 0.01f);
491 a = 0.0f;
492 /* Be careful with gains < 0.0001, as that causes the coefficient
493 * head towards 1, which will flatten the signal */
494 if(g < 0.9999f) /* 1-epsilon */
495 a = (1 - g*cw - aluSqrt(2*g*(1-cw) - g*g*(1 - cw*cw))) /
496 (1 - g);
497 ALSource->Params.iirFilter.coeff = a;
498 for(i = 0;i < MAX_SENDS;i++)
499 ALSource->Params.Send[i].iirFilter.coeff = 0.0f;
501 return;
504 //1. Translate Listener to origin (convert to head relative)
505 if(ALSource->bHeadRelative==AL_FALSE)
507 ALfloat U[3],V[3],N[3],P[3];
509 // Build transform matrix
510 memcpy(N, ALContext->Listener.Forward, sizeof(N)); // At-vector
511 aluNormalize(N); // Normalized At-vector
512 memcpy(V, ALContext->Listener.Up, sizeof(V)); // Up-vector
513 aluNormalize(V); // Normalized Up-vector
514 aluCrossproduct(N, V, U); // Right-vector
515 aluNormalize(U); // Normalized Right-vector
516 P[0] = -(ALContext->Listener.Position[0]*U[0] + // Translation
517 ALContext->Listener.Position[1]*U[1] +
518 ALContext->Listener.Position[2]*U[2]);
519 P[1] = -(ALContext->Listener.Position[0]*V[0] +
520 ALContext->Listener.Position[1]*V[1] +
521 ALContext->Listener.Position[2]*V[2]);
522 P[2] = -(ALContext->Listener.Position[0]*-N[0] +
523 ALContext->Listener.Position[1]*-N[1] +
524 ALContext->Listener.Position[2]*-N[2]);
525 Matrix[0][0] = U[0]; Matrix[0][1] = V[0]; Matrix[0][2] = -N[0]; Matrix[0][3] = 0.0f;
526 Matrix[1][0] = U[1]; Matrix[1][1] = V[1]; Matrix[1][2] = -N[1]; Matrix[1][3] = 0.0f;
527 Matrix[2][0] = U[2]; Matrix[2][1] = V[2]; Matrix[2][2] = -N[2]; Matrix[2][3] = 0.0f;
528 Matrix[3][0] = P[0]; Matrix[3][1] = P[1]; Matrix[3][2] = P[2]; Matrix[3][3] = 1.0f;
530 // Transform source position and direction into listener space
531 aluMatrixVector(Position, 1.0f, Matrix);
532 aluMatrixVector(Direction, 0.0f, Matrix);
533 // Transform source and listener velocity into listener space
534 aluMatrixVector(Velocity, 0.0f, Matrix);
535 aluMatrixVector(ListenerVel, 0.0f, Matrix);
537 else
538 ListenerVel[0] = ListenerVel[1] = ListenerVel[2] = 0.0f;
540 SourceToListener[0] = -Position[0];
541 SourceToListener[1] = -Position[1];
542 SourceToListener[2] = -Position[2];
543 aluNormalize(SourceToListener);
544 aluNormalize(Direction);
546 //2. Calculate distance attenuation
547 Distance = aluSqrt(aluDotproduct(Position, Position));
549 flAttenuation = 1.0f;
550 for(i = 0;i < MAX_SENDS;i++)
552 RoomAttenuation[i] = 1.0f;
554 RoomRolloff[i] = ALSource->RoomRolloffFactor;
555 if(ALSource->Send[i].Slot &&
556 (ALSource->Send[i].Slot->effect.type == AL_EFFECT_REVERB ||
557 ALSource->Send[i].Slot->effect.type == AL_EFFECT_EAXREVERB))
558 RoomRolloff[i] += ALSource->Send[i].Slot->effect.Reverb.RoomRolloffFactor;
561 switch(ALSource->DistanceModel)
563 case AL_INVERSE_DISTANCE_CLAMPED:
564 Distance=__max(Distance,MinDist);
565 Distance=__min(Distance,MaxDist);
566 if(MaxDist < MinDist)
567 break;
568 //fall-through
569 case AL_INVERSE_DISTANCE:
570 if(MinDist > 0.0f)
572 if((MinDist + (Rolloff * (Distance - MinDist))) > 0.0f)
573 flAttenuation = MinDist / (MinDist + (Rolloff * (Distance - MinDist)));
574 for(i = 0;i < NumSends;i++)
576 if((MinDist + (RoomRolloff[i] * (Distance - MinDist))) > 0.0f)
577 RoomAttenuation[i] = MinDist / (MinDist + (RoomRolloff[i] * (Distance - MinDist)));
580 break;
582 case AL_LINEAR_DISTANCE_CLAMPED:
583 Distance=__max(Distance,MinDist);
584 Distance=__min(Distance,MaxDist);
585 if(MaxDist < MinDist)
586 break;
587 //fall-through
588 case AL_LINEAR_DISTANCE:
589 Distance=__min(Distance,MaxDist);
590 if(MaxDist != MinDist)
592 flAttenuation = 1.0f - (Rolloff*(Distance-MinDist)/(MaxDist - MinDist));
593 for(i = 0;i < NumSends;i++)
594 RoomAttenuation[i] = 1.0f - (RoomRolloff[i]*(Distance-MinDist)/(MaxDist - MinDist));
596 break;
598 case AL_EXPONENT_DISTANCE_CLAMPED:
599 Distance=__max(Distance,MinDist);
600 Distance=__min(Distance,MaxDist);
601 if(MaxDist < MinDist)
602 break;
603 //fall-through
604 case AL_EXPONENT_DISTANCE:
605 if(Distance > 0.0f && MinDist > 0.0f)
607 flAttenuation = (ALfloat)pow(Distance/MinDist, -Rolloff);
608 for(i = 0;i < NumSends;i++)
609 RoomAttenuation[i] = (ALfloat)pow(Distance/MinDist, -RoomRolloff[i]);
611 break;
613 case AL_NONE:
614 break;
617 // Source Gain + Attenuation and clamp to Min/Max Gain
618 DryMix = SourceVolume * flAttenuation;
619 DryMix = __min(DryMix,MaxVolume);
620 DryMix = __max(DryMix,MinVolume);
622 for(i = 0;i < NumSends;i++)
624 ALfloat WetMix = SourceVolume * RoomAttenuation[i];
625 WetMix = __min(WetMix,MaxVolume);
626 WetGain[i] = __max(WetMix,MinVolume);
627 WetGainHF[i] = 1.0f;
630 // Distance-based air absorption
631 if(ALSource->AirAbsorptionFactor > 0.0f && flAttenuation < 1.0f)
633 ALfloat absorb = 0.0f;
635 // Absorption calculation is done in dB
636 if(flAttenuation > 0.0f)
638 absorb = (MinDist/flAttenuation - MinDist)*MetersPerUnit *
639 (ALSource->AirAbsorptionFactor*AIRABSORBGAINDBHF);
640 // Convert dB to linear gain before applying
641 absorb = pow(10.0, absorb/20.0);
643 DryGainHF *= absorb;
644 for(i = 0;i < MAX_SENDS;i++)
645 WetGainHF[i] *= absorb;
648 //3. Apply directional soundcones
649 Angle = aluAcos(aluDotproduct(Direction,SourceToListener)) * 180.0f/M_PI;
650 if(Angle >= InnerAngle && Angle <= OuterAngle)
652 ALfloat scale = (Angle-InnerAngle) / (OuterAngle-InnerAngle);
653 ConeVolume = (1.0f+(ALSource->flOuterGain-1.0f)*scale);
654 ConeHF = (1.0f+(OuterGainHF-1.0f)*scale);
655 DryMix *= ConeVolume;
656 if(ALSource->DryGainHFAuto)
657 DryGainHF *= ConeHF;
659 else if(Angle > OuterAngle)
661 ConeVolume = (1.0f+(ALSource->flOuterGain-1.0f));
662 ConeHF = (1.0f+(OuterGainHF-1.0f));
663 DryMix *= ConeVolume;
664 if(ALSource->DryGainHFAuto)
665 DryGainHF *= ConeHF;
667 else
669 ConeVolume = 1.0f;
670 ConeHF = 1.0f;
673 //4. Calculate Velocity
674 if(DopplerFactor != 0.0f)
676 ALfloat flVSS, flVLS;
677 ALfloat flMaxVelocity = (DopplerVelocity * flSpeedOfSound) /
678 DopplerFactor;
680 flVSS = aluDotproduct(Velocity, SourceToListener);
681 if(flVSS >= flMaxVelocity)
682 flVSS = (flMaxVelocity - 1.0f);
683 else if(flVSS <= -flMaxVelocity)
684 flVSS = -flMaxVelocity + 1.0f;
686 flVLS = aluDotproduct(ListenerVel, SourceToListener);
687 if(flVLS >= flMaxVelocity)
688 flVLS = (flMaxVelocity - 1.0f);
689 else if(flVLS <= -flMaxVelocity)
690 flVLS = -flMaxVelocity + 1.0f;
692 ALSource->Params.Pitch = ALSource->flPitch *
693 ((flSpeedOfSound * DopplerVelocity) - (DopplerFactor * flVLS)) /
694 ((flSpeedOfSound * DopplerVelocity) - (DopplerFactor * flVSS));
696 else
697 ALSource->Params.Pitch = ALSource->flPitch;
699 for(i = 0;i < NumSends;i++)
701 if(ALSource->Send[i].Slot &&
702 ALSource->Send[i].Slot->effect.type != AL_EFFECT_NULL)
704 if(ALSource->Send[i].Slot->AuxSendAuto)
706 if(ALSource->WetGainAuto)
707 WetGain[i] *= ConeVolume;
708 if(ALSource->WetGainHFAuto)
709 WetGainHF[i] *= ConeHF;
711 if(ALSource->Send[i].Slot->effect.type == AL_EFFECT_REVERB ||
712 ALSource->Send[i].Slot->effect.type == AL_EFFECT_EAXREVERB)
714 /* Apply a decay-time transformation to the wet path,
715 * based on the attenuation of the dry path. This should
716 * better approximate the statistical attenuation model
717 * for the reverb effect.
719 * This simple equation converts the distance attenuation
720 * into the time it would take to reach -60 dB. From
721 * there it establishes an origin (0.333s; the decay time
722 * that will produce equal attenuation) and applies the
723 * current decay time. Finally, it converts the result
724 * back to an attenuation for the reverb path.
726 WetGain[i] *= pow(10.0f, log10(flAttenuation) * 0.333f /
727 ALSource->Send[i].Slot->effect.Reverb.DecayTime);
730 else
732 // If the slot's auxiliary send auto is off, the data sent to
733 // the effect slot is the same as the dry path, sans filter
734 // effects
735 WetGain[i] = DryMix;
736 WetGainHF[i] = DryGainHF;
739 switch(ALSource->Send[i].WetFilter.type)
741 case AL_FILTER_LOWPASS:
742 WetGain[i] *= ALSource->Send[i].WetFilter.Gain;
743 WetGainHF[i] *= ALSource->Send[i].WetFilter.GainHF;
744 break;
746 ALSource->Params.WetGains[i] = WetGain[i] * ListenerGain;
748 else
750 ALSource->Params.WetGains[i] = 0.0f;
751 WetGainHF[i] = 1.0f;
754 for(i = NumSends;i < MAX_SENDS;i++)
756 ALSource->Params.WetGains[i] = 0.0f;
757 WetGainHF[i] = 1.0f;
760 //5. Apply filter gains and filters
761 switch(ALSource->DirectFilter.type)
763 case AL_FILTER_LOWPASS:
764 DryMix *= ALSource->DirectFilter.Gain;
765 DryGainHF *= ALSource->DirectFilter.GainHF;
766 break;
768 DryMix *= ListenerGain;
770 // Use energy-preserving panning algorithm for multi-speaker playback
771 length = aluSqrt(Position[0]*Position[0] + Position[1]*Position[1] +
772 Position[2]*Position[2]);
773 length = __max(length, MinDist);
774 if(length > 0.0f)
776 ALfloat invlen = 1.0f/length;
777 Position[0] *= invlen;
778 Position[1] *= invlen;
779 Position[2] *= invlen;
782 pos = aluCart2LUTpos(-Position[2], Position[0]);
783 SpeakerGain = &ALContext->PanningLUT[OUTPUTCHANNELS * pos];
785 DirGain = aluSqrt(Position[0]*Position[0] + Position[2]*Position[2]);
786 // elevation adjustment for directional gain. this sucks, but
787 // has low complexity
788 AmbientGain = 1.0/aluSqrt(ALContext->NumChan) * (1.0-DirGain);
789 for(s = 0; s < OUTPUTCHANNELS; s++)
791 ALfloat gain = SpeakerGain[s]*DirGain + AmbientGain;
792 ALSource->Params.DryGains[s] = DryMix * gain;
795 /* Update filter coefficients. */
796 cw = cos(2.0*M_PI * LOWPASSFREQCUTOFF / Frequency);
797 /* Spatialized sources use four chained one-pole filters, so we need to
798 * take the fourth root of the squared gain, which is the same as the
799 * square root of the base gain. */
800 g = aluSqrt(__max(DryGainHF, 0.0001f));
801 a = 0.0f;
802 if(g < 0.9999f) /* 1-epsilon */
803 a = (1 - g*cw - aluSqrt(2*g*(1-cw) - g*g*(1 - cw*cw))) /
804 (1 - g);
805 ALSource->Params.iirFilter.coeff = a;
807 for(i = 0;i < NumSends;i++)
809 /* The wet path uses two chained one-pole filters, so take the
810 * base gain (square root of the squared gain) */
811 g = __max(WetGainHF[i], 0.01f);
812 a = 0.0f;
813 if(g < 0.9999f) /* 1-epsilon */
814 a = (1 - g*cw - aluSqrt(2*g*(1-cw) - g*g*(1 - cw*cw))) /
815 (1 - g);
816 ALSource->Params.Send[i].iirFilter.coeff = a;
820 static __inline ALshort lerp(ALshort val1, ALshort val2, ALint frac)
822 return val1 + (((val2-val1)*frac)>>FRACTIONBITS);
825 static void MixSomeSources(ALCcontext *ALContext, float (*DryBuffer)[OUTPUTCHANNELS], ALuint SamplesToDo)
827 static float DummyBuffer[BUFFERSIZE];
828 ALfloat *WetBuffer[MAX_SENDS];
829 ALfloat (*Matrix)[OUTPUTCHANNELS] = ALContext->ChannelMatrix;
830 ALfloat DrySend[OUTPUTCHANNELS];
831 ALfloat dryGainStep[OUTPUTCHANNELS];
832 ALfloat wetGainStep[MAX_SENDS];
833 ALuint i, j, k, out;
834 ALsource *ALSource;
835 ALfloat value;
836 ALbufferlistitem *BufferListItem;
837 ALint64 DataSize64,DataPos64;
838 FILTER *DryFilter, *WetFilter[MAX_SENDS];
839 ALfloat WetSend[MAX_SENDS];
840 ALuint rampLength;
841 ALuint DeviceFreq;
842 ALint increment;
843 ALuint DataPosInt, DataPosFrac;
844 ALuint Channels, Bytes;
845 ALuint Frequency;
846 ALuint BuffersPlayed;
847 ALfloat Pitch;
848 ALenum State;
850 if(!(ALSource=ALContext->Source))
851 return;
853 DeviceFreq = ALContext->Device->Frequency;
855 rampLength = DeviceFreq * MIN_RAMP_LENGTH / 1000;
856 rampLength = max(rampLength, SamplesToDo);
858 another_source:
859 State = ALSource->state;
860 if(State != AL_PLAYING)
862 if((ALSource=ALSource->next) != NULL)
863 goto another_source;
864 return;
866 j = 0;
868 /* Find buffer format */
869 Frequency = 0;
870 Channels = 0;
871 Bytes = 0;
872 BufferListItem = ALSource->queue;
873 while(BufferListItem != NULL)
875 ALbuffer *ALBuffer;
876 if((ALBuffer=BufferListItem->buffer) != NULL)
878 Channels = aluChannelsFromFormat(ALBuffer->format);
879 Bytes = aluBytesFromFormat(ALBuffer->format);
880 Frequency = ALBuffer->frequency;
881 break;
883 BufferListItem = BufferListItem->next;
886 /* Get source info */
887 BuffersPlayed = ALSource->BuffersPlayed;
888 DataPosInt = ALSource->position;
889 DataPosFrac = ALSource->position_fraction;
891 CalcSourceParams(ALContext, ALSource, (Channels==1)?AL_TRUE:AL_FALSE);
893 /* Compute 18.14 fixed point step */
894 Pitch = (ALSource->Params.Pitch*Frequency) / DeviceFreq;
895 if(Pitch > (float)MAX_PITCH) Pitch = (float)MAX_PITCH;
896 increment = (ALint)(Pitch*(ALfloat)(1L<<FRACTIONBITS));
897 if(increment <= 0) increment = (1<<FRACTIONBITS);
899 /* Compute the gain steps for each output channel */
900 if(ALSource->FirstStart)
902 for(i = 0;i < OUTPUTCHANNELS;i++)
903 DrySend[i] = ALSource->Params.DryGains[i];
904 for(i = 0;i < MAX_SENDS;i++)
905 WetSend[i] = ALSource->Params.WetGains[i];
907 else
909 for(i = 0;i < OUTPUTCHANNELS;i++)
910 DrySend[i] = ALSource->DryGains[i];
911 for(i = 0;i < MAX_SENDS;i++)
912 WetSend[i] = ALSource->WetGains[i];
915 DryFilter = &ALSource->Params.iirFilter;
916 for(i = 0;i < MAX_SENDS;i++)
918 WetFilter[i] = &ALSource->Params.Send[i].iirFilter;
919 WetBuffer[i] = (ALSource->Send[i].Slot ?
920 ALSource->Send[i].Slot->WetBuffer :
921 DummyBuffer);
924 if(DuplicateStereo && Channels == 2)
926 Matrix[FRONT_LEFT][SIDE_LEFT] = 1.0f;
927 Matrix[FRONT_RIGHT][SIDE_RIGHT] = 1.0f;
928 Matrix[FRONT_LEFT][BACK_LEFT] = 1.0f;
929 Matrix[FRONT_RIGHT][BACK_RIGHT] = 1.0f;
931 else if(DuplicateStereo)
933 Matrix[FRONT_LEFT][SIDE_LEFT] = 0.0f;
934 Matrix[FRONT_RIGHT][SIDE_RIGHT] = 0.0f;
935 Matrix[FRONT_LEFT][BACK_LEFT] = 0.0f;
936 Matrix[FRONT_RIGHT][BACK_RIGHT] = 0.0f;
939 /* Get current buffer queue item */
940 BufferListItem = ALSource->queue;
941 for(i = 0;i < BuffersPlayed && BufferListItem;i++)
942 BufferListItem = BufferListItem->next;
944 while(State == AL_PLAYING && j < SamplesToDo)
946 ALuint DataSize = 0;
947 ALbuffer *ALBuffer;
948 ALshort *Data;
949 ALuint BufferSize;
951 /* Get buffer info */
952 if((ALBuffer=BufferListItem->buffer) != NULL)
954 Data = ALBuffer->data;
955 DataSize = ALBuffer->size;
956 DataSize /= Channels * Bytes;
958 if(DataPosInt >= DataSize)
959 goto skipmix;
961 if(BufferListItem->next)
963 ALbuffer *NextBuf = BufferListItem->next->buffer;
964 if(NextBuf && NextBuf->data)
966 ALint ulExtraSamples = BUFFER_PADDING*Channels*Bytes;
967 ulExtraSamples = min(NextBuf->size, ulExtraSamples);
968 memcpy(&Data[DataSize*Channels], NextBuf->data, ulExtraSamples);
971 else if(ALSource->bLooping)
973 ALbuffer *NextBuf = ALSource->queue->buffer;
974 if(NextBuf && NextBuf->data)
976 ALint ulExtraSamples = BUFFER_PADDING*Channels*Bytes;
977 ulExtraSamples = min(NextBuf->size, ulExtraSamples);
978 memcpy(&Data[DataSize*Channels], NextBuf->data, ulExtraSamples);
981 else
982 memset(&Data[DataSize*Channels], 0, (BUFFER_PADDING*Channels*Bytes));
984 /* Compute the gain steps for each output channel */
985 for(i = 0;i < OUTPUTCHANNELS;i++)
986 dryGainStep[i] = (ALSource->Params.DryGains[i]-
987 DrySend[i]) / rampLength;
988 for(i = 0;i < MAX_SENDS;i++)
989 wetGainStep[i] = (ALSource->Params.WetGains[i]-
990 WetSend[i]) / rampLength;
992 /* Figure out how many samples we can mix. */
993 DataSize64 = DataSize;
994 DataSize64 <<= FRACTIONBITS;
995 DataPos64 = DataPosInt;
996 DataPos64 <<= FRACTIONBITS;
997 DataPos64 += DataPosFrac;
998 BufferSize = (ALuint)((DataSize64-DataPos64+(increment-1)) / increment);
1000 BufferSize = min(BufferSize, (SamplesToDo-j));
1002 /* Actual sample mixing loop */
1003 k = 0;
1004 Data += DataPosInt*Channels;
1006 if(Channels == 1) /* Mono */
1008 ALfloat outsamp;
1010 while(BufferSize--)
1012 for(i = 0;i < OUTPUTCHANNELS;i++)
1013 DrySend[i] += dryGainStep[i];
1014 for(i = 0;i < MAX_SENDS;i++)
1015 WetSend[i] += wetGainStep[i];
1017 /* First order interpolator */
1018 value = lerp(Data[k], Data[k+1], DataPosFrac);
1020 /* Direct path final mix buffer and panning */
1021 outsamp = lpFilter4P(DryFilter, 0, value);
1022 DryBuffer[j][FRONT_LEFT] += outsamp*DrySend[FRONT_LEFT];
1023 DryBuffer[j][FRONT_RIGHT] += outsamp*DrySend[FRONT_RIGHT];
1024 DryBuffer[j][SIDE_LEFT] += outsamp*DrySend[SIDE_LEFT];
1025 DryBuffer[j][SIDE_RIGHT] += outsamp*DrySend[SIDE_RIGHT];
1026 DryBuffer[j][BACK_LEFT] += outsamp*DrySend[BACK_LEFT];
1027 DryBuffer[j][BACK_RIGHT] += outsamp*DrySend[BACK_RIGHT];
1028 DryBuffer[j][FRONT_CENTER] += outsamp*DrySend[FRONT_CENTER];
1029 DryBuffer[j][BACK_CENTER] += outsamp*DrySend[BACK_CENTER];
1031 /* Room path final mix buffer and panning */
1032 for(i = 0;i < MAX_SENDS;i++)
1034 outsamp = lpFilter2P(WetFilter[i], 0, value);
1035 WetBuffer[i][j] += outsamp*WetSend[i];
1038 DataPosFrac += increment;
1039 k += DataPosFrac>>FRACTIONBITS;
1040 DataPosFrac &= FRACTIONMASK;
1041 j++;
1044 else if(Channels == 2) /* Stereo */
1046 const int chans[] = {
1047 FRONT_LEFT, FRONT_RIGHT
1050 #define DO_MIX() do { \
1051 for(i = 0;i < MAX_SENDS;i++) \
1052 WetSend[i] += wetGainStep[i]*BufferSize; \
1053 while(BufferSize--) \
1055 for(i = 0;i < OUTPUTCHANNELS;i++) \
1056 DrySend[i] += dryGainStep[i]; \
1058 for(i = 0;i < Channels;i++) \
1060 value = lerp(Data[k*Channels + i], Data[(k+1)*Channels + i], DataPosFrac); \
1061 value = lpFilter2P(DryFilter, chans[i]*2, value)*DrySend[chans[i]]; \
1062 for(out = 0;out < OUTPUTCHANNELS;out++) \
1063 DryBuffer[j][out] += value*Matrix[chans[i]][out]; \
1066 DataPosFrac += increment; \
1067 k += DataPosFrac>>FRACTIONBITS; \
1068 DataPosFrac &= FRACTIONMASK; \
1069 j++; \
1071 } while(0)
1073 DO_MIX();
1075 else if(Channels == 4) /* Quad */
1077 const int chans[] = {
1078 FRONT_LEFT, FRONT_RIGHT,
1079 BACK_LEFT, BACK_RIGHT
1082 DO_MIX();
1084 else if(Channels == 6) /* 5.1 */
1086 const int chans[] = {
1087 FRONT_LEFT, FRONT_RIGHT,
1088 FRONT_CENTER, LFE,
1089 BACK_LEFT, BACK_RIGHT
1092 DO_MIX();
1094 else if(Channels == 7) /* 6.1 */
1096 const int chans[] = {
1097 FRONT_LEFT, FRONT_RIGHT,
1098 FRONT_CENTER, LFE,
1099 BACK_CENTER,
1100 SIDE_LEFT, SIDE_RIGHT
1103 DO_MIX();
1105 else if(Channels == 8) /* 7.1 */
1107 const int chans[] = {
1108 FRONT_LEFT, FRONT_RIGHT,
1109 FRONT_CENTER, LFE,
1110 BACK_LEFT, BACK_RIGHT,
1111 SIDE_LEFT, SIDE_RIGHT
1114 DO_MIX();
1115 #undef DO_MIX
1117 else /* Unknown? */
1119 for(i = 0;i < OUTPUTCHANNELS;i++)
1120 DrySend[i] += dryGainStep[i]*BufferSize;
1121 for(i = 0;i < MAX_SENDS;i++)
1122 WetSend[i] += wetGainStep[i]*BufferSize;
1123 while(BufferSize--)
1125 DataPosFrac += increment;
1126 k += DataPosFrac>>FRACTIONBITS;
1127 DataPosFrac &= FRACTIONMASK;
1128 j++;
1131 DataPosInt += k;
1133 skipmix:
1134 /* Handle looping sources */
1135 if(DataPosInt >= DataSize)
1137 if(BuffersPlayed < (ALSource->BuffersInQueue-1))
1139 BufferListItem = BufferListItem->next;
1140 BuffersPlayed++;
1141 DataPosInt -= DataSize;
1143 else
1145 if(!ALSource->bLooping)
1147 State = AL_STOPPED;
1148 BufferListItem = ALSource->queue;
1149 BuffersPlayed = ALSource->BuffersInQueue;
1150 DataPosInt = 0;
1151 DataPosFrac = 0;
1153 else
1155 BufferListItem = ALSource->queue;
1156 BuffersPlayed = 0;
1157 if(ALSource->BuffersInQueue == 1)
1158 DataPosInt %= DataSize;
1159 else
1160 DataPosInt -= DataSize;
1166 /* Update source info */
1167 ALSource->state = State;
1168 ALSource->BuffersPlayed = BuffersPlayed;
1169 ALSource->position = DataPosInt;
1170 ALSource->position_fraction = DataPosFrac;
1171 ALSource->Buffer = BufferListItem->buffer;
1173 for(i = 0;i < OUTPUTCHANNELS;i++)
1174 ALSource->DryGains[i] = DrySend[i];
1175 for(i = 0;i < MAX_SENDS;i++)
1176 ALSource->WetGains[i] = WetSend[i];
1178 ALSource->FirstStart = AL_FALSE;
1180 if((ALSource=ALSource->next) != NULL)
1181 goto another_source;
1184 ALvoid aluMixData(ALCdevice *device, ALvoid *buffer, ALsizei size)
1186 float (*DryBuffer)[OUTPUTCHANNELS];
1187 ALuint SamplesToDo;
1188 ALeffectslot *ALEffectSlot;
1189 ALCcontext *ALContext;
1190 int fpuState;
1191 ALuint i, c;
1193 SuspendContext(NULL);
1195 #if defined(HAVE_FESETROUND)
1196 fpuState = fegetround();
1197 fesetround(FE_TOWARDZERO);
1198 #elif defined(HAVE__CONTROLFP)
1199 fpuState = _controlfp(0, 0);
1200 _controlfp(_RC_CHOP, _MCW_RC);
1201 #else
1202 (void)fpuState;
1203 #endif
1205 DryBuffer = device->DryBuffer;
1206 while(size > 0)
1208 /* Setup variables */
1209 SamplesToDo = min(size, BUFFERSIZE);
1211 /* Clear mixing buffer */
1212 memset(DryBuffer, 0, SamplesToDo*OUTPUTCHANNELS*sizeof(ALfloat));
1214 for(c = 0;c < device->NumContexts;c++)
1216 ALContext = device->Contexts[c];
1217 SuspendContext(ALContext);
1219 MixSomeSources(ALContext, DryBuffer, SamplesToDo);
1221 /* effect slot processing */
1222 ALEffectSlot = ALContext->AuxiliaryEffectSlot;
1223 while(ALEffectSlot)
1225 if(ALEffectSlot->EffectState)
1226 ALEffect_Process(ALEffectSlot->EffectState, ALEffectSlot, SamplesToDo, ALEffectSlot->WetBuffer, DryBuffer);
1228 for(i = 0;i < SamplesToDo;i++)
1229 ALEffectSlot->WetBuffer[i] = 0.0f;
1230 ALEffectSlot = ALEffectSlot->next;
1232 ProcessContext(ALContext);
1235 //Post processing loop
1236 switch(device->Format)
1238 #define CHECK_WRITE_FORMAT(bits, type, func, isWin) \
1239 case AL_FORMAT_MONO##bits: \
1240 for(i = 0;i < SamplesToDo;i++) \
1242 ((type*)buffer)[0] = (func)(DryBuffer[i][FRONT_LEFT] + \
1243 DryBuffer[i][FRONT_RIGHT]); \
1244 buffer = ((type*)buffer) + 1; \
1246 break; \
1247 case AL_FORMAT_STEREO##bits: \
1248 if(device->Bs2b) \
1250 for(i = 0;i < SamplesToDo;i++) \
1252 float samples[2]; \
1253 samples[0] = DryBuffer[i][FRONT_LEFT]; \
1254 samples[1] = DryBuffer[i][FRONT_RIGHT]; \
1255 bs2b_cross_feed(device->Bs2b, samples); \
1256 ((type*)buffer)[0] = (func)(samples[0]); \
1257 ((type*)buffer)[1] = (func)(samples[1]); \
1258 buffer = ((type*)buffer) + 2; \
1261 else \
1263 for(i = 0;i < SamplesToDo;i++) \
1265 ((type*)buffer)[0] = (func)(DryBuffer[i][FRONT_LEFT]); \
1266 ((type*)buffer)[1] = (func)(DryBuffer[i][FRONT_RIGHT]); \
1267 buffer = ((type*)buffer) + 2; \
1270 break; \
1271 case AL_FORMAT_QUAD##bits: \
1272 for(i = 0;i < SamplesToDo;i++) \
1274 ((type*)buffer)[0] = (func)(DryBuffer[i][FRONT_LEFT]); \
1275 ((type*)buffer)[1] = (func)(DryBuffer[i][FRONT_RIGHT]); \
1276 ((type*)buffer)[2] = (func)(DryBuffer[i][BACK_LEFT]); \
1277 ((type*)buffer)[3] = (func)(DryBuffer[i][BACK_RIGHT]); \
1278 buffer = ((type*)buffer) + 4; \
1280 break; \
1281 case AL_FORMAT_51CHN##bits: \
1282 for(i = 0;i < SamplesToDo;i++) \
1284 ((type*)buffer)[0] = (func)(DryBuffer[i][FRONT_LEFT]); \
1285 ((type*)buffer)[1] = (func)(DryBuffer[i][FRONT_RIGHT]); \
1286 if(isWin) { \
1287 /* Of course, Windows can't use the same ordering... */ \
1288 ((type*)buffer)[2] = (func)(DryBuffer[i][FRONT_CENTER]); \
1289 ((type*)buffer)[3] = (func)(DryBuffer[i][LFE]); \
1290 ((type*)buffer)[4] = (func)(DryBuffer[i][BACK_LEFT]); \
1291 ((type*)buffer)[5] = (func)(DryBuffer[i][BACK_RIGHT]); \
1292 } else { \
1293 ((type*)buffer)[2] = (func)(DryBuffer[i][BACK_LEFT]); \
1294 ((type*)buffer)[3] = (func)(DryBuffer[i][BACK_RIGHT]); \
1295 ((type*)buffer)[4] = (func)(DryBuffer[i][FRONT_CENTER]); \
1296 ((type*)buffer)[5] = (func)(DryBuffer[i][LFE]); \
1298 buffer = ((type*)buffer) + 6; \
1300 break; \
1301 case AL_FORMAT_61CHN##bits: \
1302 for(i = 0;i < SamplesToDo;i++) \
1304 ((type*)buffer)[0] = (func)(DryBuffer[i][FRONT_LEFT]); \
1305 ((type*)buffer)[1] = (func)(DryBuffer[i][FRONT_RIGHT]); \
1306 ((type*)buffer)[2] = (func)(DryBuffer[i][FRONT_CENTER]); \
1307 ((type*)buffer)[3] = (func)(DryBuffer[i][LFE]); \
1308 ((type*)buffer)[4] = (func)(DryBuffer[i][BACK_CENTER]); \
1309 ((type*)buffer)[5] = (func)(DryBuffer[i][SIDE_LEFT]); \
1310 ((type*)buffer)[6] = (func)(DryBuffer[i][SIDE_RIGHT]); \
1311 buffer = ((type*)buffer) + 7; \
1313 break; \
1314 case AL_FORMAT_71CHN##bits: \
1315 for(i = 0;i < SamplesToDo;i++) \
1317 ((type*)buffer)[0] = (func)(DryBuffer[i][FRONT_LEFT]); \
1318 ((type*)buffer)[1] = (func)(DryBuffer[i][FRONT_RIGHT]); \
1319 if(isWin) { \
1320 ((type*)buffer)[2] = (func)(DryBuffer[i][FRONT_CENTER]); \
1321 ((type*)buffer)[3] = (func)(DryBuffer[i][LFE]); \
1322 ((type*)buffer)[4] = (func)(DryBuffer[i][BACK_LEFT]); \
1323 ((type*)buffer)[5] = (func)(DryBuffer[i][BACK_RIGHT]); \
1324 } else { \
1325 ((type*)buffer)[2] = (func)(DryBuffer[i][BACK_LEFT]); \
1326 ((type*)buffer)[3] = (func)(DryBuffer[i][BACK_RIGHT]); \
1327 ((type*)buffer)[4] = (func)(DryBuffer[i][FRONT_CENTER]); \
1328 ((type*)buffer)[5] = (func)(DryBuffer[i][LFE]); \
1330 ((type*)buffer)[6] = (func)(DryBuffer[i][SIDE_LEFT]); \
1331 ((type*)buffer)[7] = (func)(DryBuffer[i][SIDE_RIGHT]); \
1332 buffer = ((type*)buffer) + 8; \
1334 break;
1336 #define AL_FORMAT_MONO32 AL_FORMAT_MONO_FLOAT32
1337 #define AL_FORMAT_STEREO32 AL_FORMAT_STEREO_FLOAT32
1338 #ifdef _WIN32
1339 CHECK_WRITE_FORMAT(8, ALubyte, aluF2UB, 1)
1340 CHECK_WRITE_FORMAT(16, ALshort, aluF2S, 1)
1341 CHECK_WRITE_FORMAT(32, ALfloat, aluF2F, 1)
1342 #else
1343 CHECK_WRITE_FORMAT(8, ALubyte, aluF2UB, 0)
1344 CHECK_WRITE_FORMAT(16, ALshort, aluF2S, 0)
1345 CHECK_WRITE_FORMAT(32, ALfloat, aluF2F, 0)
1346 #endif
1347 #undef AL_FORMAT_STEREO32
1348 #undef AL_FORMAT_MONO32
1349 #undef CHECK_WRITE_FORMAT
1351 default:
1352 break;
1355 size -= SamplesToDo;
1358 #if defined(HAVE_FESETROUND)
1359 fesetround(fpuState);
1360 #elif defined(HAVE__CONTROLFP)
1361 _controlfp(fpuState, 0xfffff);
1362 #endif
1364 ProcessContext(NULL);
1367 ALvoid aluHandleDisconnect(ALCdevice *device)
1369 ALuint i;
1371 SuspendContext(NULL);
1372 for(i = 0;i < device->NumContexts;i++)
1374 ALsource *source;
1376 SuspendContext(device->Contexts[i]);
1378 source = device->Contexts[i]->Source;
1379 while(source)
1381 if(source->state == AL_PLAYING)
1383 source->state = AL_STOPPED;
1384 source->BuffersPlayed = source->BuffersInQueue;
1385 source->position = 0;
1386 source->position_fraction = 0;
1388 source = source->next;
1390 ProcessContext(device->Contexts[i]);
1393 device->Connected = ALC_FALSE;
1394 ProcessContext(NULL);