Alc/ALu.c

   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.
   8  *
   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.
  13  *
  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
  19  */
  20
  21 #include "config.h"
  22
  23 #include <math.h>
  24 #include <stdlib.h>
  25 #include <string.h>
  26 #include <ctype.h>
  27 #include <assert.h>
  28
  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"
  39
  40 #define FRACTIONBITS 14
  41 #define FRACTIONMASK ((1L<<FRACTIONBITS)-1)
  42 #define MAX_PITCH 65536
  43
  44 /* Minimum ramp length in milliseconds. The value below was chosen to
  45  * adequately reduce clicks and pops from harsh gain changes. */
  46 #define MIN_RAMP_LENGTH  16
  47
  48 ALboolean DuplicateStereo = AL_FALSE;
  49
  50
  51 static __inline ALfloat aluF2F(ALfloat Value)
  52 {
  53     return Value;
  54 }
  55
  56 static __inline ALshort aluF2S(ALfloat Value)
  57 {
  58     ALint i;
  59
  60     if(Value < 0.0f)
  61     {
  62         i = (ALint)(Value*32768.0f);
  63         i = max(-32768, i);
  64     }
  65     else
  66     {
  67         i = (ALint)(Value*32767.0f);
  68         i = min( 32767, i);
  69     }
  70     return ((ALshort)i);
  71 }
  72
  73 static __inline ALubyte aluF2UB(ALfloat Value)
  74 {
  75     ALshort i = aluF2S(Value);
  76     return (i>>8)+128;
  77 }
  78
  79
  80 static __inline ALvoid aluCrossproduct(const ALfloat *inVector1, const ALfloat *inVector2, ALfloat *outVector)
  81 {
  82     outVector[0] = inVector1[1]*inVector2[2] - inVector1[2]*inVector2[1];
  83     outVector[1] = inVector1[2]*inVector2[0] - inVector1[0]*inVector2[2];
  84     outVector[2] = inVector1[0]*inVector2[1] - inVector1[1]*inVector2[0];
  85 }
  86
  87 static __inline ALfloat aluDotproduct(const ALfloat *inVector1, const ALfloat *inVector2)
  88 {
  89     return inVector1[0]*inVector2[0] + inVector1[1]*inVector2[1] +
  90            inVector1[2]*inVector2[2];
  91 }
  92
  93 static __inline ALvoid aluNormalize(ALfloat *inVector)
  94 {
  95     ALfloat length, inverse_length;
  96
  97     length = aluSqrt(aluDotproduct(inVector, inVector));
  98     if(length != 0.0f)
  99     {
 100         inverse_length = 1.0f/length;
 101         inVector[0] *= inverse_length;
 102         inVector[1] *= inverse_length;
 103         inVector[2] *= inverse_length;
 104     }
 105 }
 106
 107 static __inline ALvoid aluMatrixVector(ALfloat *vector,ALfloat w,ALfloat matrix[4][4])
 108 {
 109     ALfloat temp[4] = {
 110         vector[0], vector[1], vector[2], w
 111     };
 112
 113     vector[0] = temp[0]*matrix[0][0] + temp[1]*matrix[1][0] + temp[2]*matrix[2][0] + temp[3]*matrix[3][0];
 114     vector[1] = temp[0]*matrix[0][1] + temp[1]*matrix[1][1] + temp[2]*matrix[2][1] + temp[3]*matrix[3][1];
 115     vector[2] = temp[0]*matrix[0][2] + temp[1]*matrix[1][2] + temp[2]*matrix[2][2] + temp[3]*matrix[3][2];
 116 }
 117
 118 static ALvoid SetSpeakerArrangement(const char *name, ALfloat SpeakerAngle[OUTPUTCHANNELS],
 119                                     Channel Speaker2Chan[OUTPUTCHANNELS], ALint chans)
 120 {
 121     char layout_str[256];
 122     char *confkey, *next;
 123     char *sep, *end;
 124     Channel val;
 125     int i;
 126
 127     strncpy(layout_str, GetConfigValue(NULL, name, ""), sizeof(layout_str));
 128     layout_str[255] = 0;
 129
 130     if(!layout_str[0])
 131         return;
 132
 133     next = confkey = layout_str;
 134     while(next && *next)
 135     {
 136         confkey = next;
 137         next = strchr(confkey, ',');
 138         if(next)
 139         {
 140             *next = 0;
 141             do {
 142                 next++;
 143             } while(isspace(*next) || *next == ',');
 144         }
 145
 146         sep = strchr(confkey, '=');
 147         if(!sep || confkey == sep)
 148             continue;
 149
 150         end = sep - 1;
 151         while(isspace(*end) && end != confkey)
 152             end--;
 153         *(++end) = 0;
 154
 155         if(strcmp(confkey, "fl") == 0 || strcmp(confkey, "front-left") == 0)
 156             val = FRONT_LEFT;
 157         else if(strcmp(confkey, "fr") == 0 || strcmp(confkey, "front-right") == 0)
 158             val = FRONT_RIGHT;
 159         else if(strcmp(confkey, "fc") == 0 || strcmp(confkey, "front-center") == 0)
 160             val = FRONT_CENTER;
 161         else if(strcmp(confkey, "bl") == 0 || strcmp(confkey, "back-left") == 0)
 162             val = BACK_LEFT;
 163         else if(strcmp(confkey, "br") == 0 || strcmp(confkey, "back-right") == 0)
 164             val = BACK_RIGHT;
 165         else if(strcmp(confkey, "bc") == 0 || strcmp(confkey, "back-center") == 0)
 166             val = BACK_CENTER;
 167         else if(strcmp(confkey, "sl") == 0 || strcmp(confkey, "side-left") == 0)
 168             val = SIDE_LEFT;
 169         else if(strcmp(confkey, "sr") == 0 || strcmp(confkey, "side-right") == 0)
 170             val = SIDE_RIGHT;
 171         else
 172         {
 173             AL_PRINT("Unknown speaker for %s: \"%s\"\n", name, confkey);
 174             continue;
 175         }
 176
 177         *(sep++) = 0;
 178         while(isspace(*sep))
 179             sep++;
 180
 181         for(i = 0;i < chans;i++)
 182         {
 183             if(Speaker2Chan[i] == val)
 184             {
 185                 long angle = strtol(sep, NULL, 10);
 186                 if(angle >= -180 && angle <= 180)
 187                     SpeakerAngle[i] = angle * M_PI/180.0f;
 188                 else
 189                     AL_PRINT("Invalid angle for speaker \"%s\": %ld\n", confkey, angle);
 190                 break;
 191             }
 192         }
 193     }
 194
 195     for(i = 0;i < chans;i++)
 196     {
 197         int min = i;
 198         int i2;
 199
 200         for(i2 = i+1;i2 < chans;i2++)
 201         {
 202             if(SpeakerAngle[i2] < SpeakerAngle[min])
 203                 min = i2;
 204         }
 205
 206         if(min != i)
 207         {
 208             ALfloat tmpf;
 209             Channel tmpc;
 210
 211             tmpf = SpeakerAngle[i];
 212             SpeakerAngle[i] = SpeakerAngle[min];
 213             SpeakerAngle[min] = tmpf;
 214
 215             tmpc = Speaker2Chan[i];
 216             Speaker2Chan[i] = Speaker2Chan[min];
 217             Speaker2Chan[min] = tmpc;
 218         }
 219     }
 220 }
 221
 222 static __inline ALfloat aluLUTpos2Angle(ALint pos)
 223 {
 224     if(pos < QUADRANT_NUM)
 225         return aluAtan((ALfloat)pos / (ALfloat)(QUADRANT_NUM - pos));
 226     if(pos < 2 * QUADRANT_NUM)
 227         return M_PI_2 + aluAtan((ALfloat)(pos - QUADRANT_NUM) / (ALfloat)(2 * QUADRANT_NUM - pos));
 228     if(pos < 3 * QUADRANT_NUM)
 229         return aluAtan((ALfloat)(pos - 2 * QUADRANT_NUM) / (ALfloat)(3 * QUADRANT_NUM - pos)) - M_PI;
 230     return aluAtan((ALfloat)(pos - 3 * QUADRANT_NUM) / (ALfloat)(4 * QUADRANT_NUM - pos)) - M_PI_2;
 231 }
 232
 233 ALvoid aluInitPanning(ALCdevice *Device)
 234 {
 235     ALfloat SpeakerAngle[OUTPUTCHANNELS];
 236     Channel Speaker2Chan[OUTPUTCHANNELS];
 237     ALfloat Alpha, Theta;
 238     ALint pos, offset;
 239     ALfloat maxout;
 240     ALuint s, s2;
 241
 242     for(s = 0;s < OUTPUTCHANNELS;s++)
 243     {
 244         for(s2 = 0;s2 < OUTPUTCHANNELS;s2++)
 245             Device->ChannelMatrix[s][s2] = ((s==s2) ? 1.0f : 0.0f);
 246     }
 247
 248     switch(Device->Format)
 249     {
 250         case AL_FORMAT_MONO8:
 251         case AL_FORMAT_MONO16:
 252         case AL_FORMAT_MONO_FLOAT32:
 253             Device->ChannelMatrix[FRONT_LEFT][FRONT_CENTER]  = aluSqrt(0.5);
 254             Device->ChannelMatrix[FRONT_RIGHT][FRONT_CENTER] = aluSqrt(0.5);
 255             Device->ChannelMatrix[SIDE_LEFT][FRONT_CENTER]   = aluSqrt(0.5);
 256             Device->ChannelMatrix[SIDE_RIGHT][FRONT_CENTER]  = aluSqrt(0.5);
 257             Device->ChannelMatrix[BACK_LEFT][FRONT_CENTER]   = aluSqrt(0.5);
 258             Device->ChannelMatrix[BACK_RIGHT][FRONT_CENTER]  = aluSqrt(0.5);
 259             Device->ChannelMatrix[BACK_CENTER][FRONT_CENTER] = 1.0f;
 260             Device->NumChan = 1;
 261             Speaker2Chan[0] = FRONT_CENTER;
 262             SpeakerAngle[0] = 0.0f * M_PI/180.0f;
 263             break;
 264
 265         case AL_FORMAT_STEREO8:
 266         case AL_FORMAT_STEREO16:
 267         case AL_FORMAT_STEREO_FLOAT32:
 268             Device->ChannelMatrix[FRONT_CENTER][FRONT_LEFT]  = aluSqrt(0.5);
 269             Device->ChannelMatrix[FRONT_CENTER][FRONT_RIGHT] = aluSqrt(0.5);
 270             Device->ChannelMatrix[SIDE_LEFT][FRONT_LEFT]     = 1.0f;
 271             Device->ChannelMatrix[SIDE_RIGHT][FRONT_RIGHT]   = 1.0f;
 272             Device->ChannelMatrix[BACK_LEFT][FRONT_LEFT]     = 1.0f;
 273             Device->ChannelMatrix[BACK_RIGHT][FRONT_RIGHT]   = 1.0f;
 274             Device->ChannelMatrix[BACK_CENTER][FRONT_LEFT]   = aluSqrt(0.5);
 275             Device->ChannelMatrix[BACK_CENTER][FRONT_RIGHT]  = aluSqrt(0.5);
 276             Device->NumChan = 2;
 277             Speaker2Chan[0] = FRONT_LEFT;
 278             Speaker2Chan[1] = FRONT_RIGHT;
 279             SpeakerAngle[0] = -90.0f * M_PI/180.0f;
 280             SpeakerAngle[1] =  90.0f * M_PI/180.0f;
 281             SetSpeakerArrangement("layout", SpeakerAngle, Speaker2Chan, Device->NumChan);
 282             break;
 283
 284         case AL_FORMAT_QUAD8:
 285         case AL_FORMAT_QUAD16:
 286         case AL_FORMAT_QUAD32:
 287             Device->ChannelMatrix[FRONT_CENTER][FRONT_LEFT]  = aluSqrt(0.5);
 288             Device->ChannelMatrix[FRONT_CENTER][FRONT_RIGHT] = aluSqrt(0.5);
 289             Device->ChannelMatrix[SIDE_LEFT][FRONT_LEFT]     = aluSqrt(0.5);
 290             Device->ChannelMatrix[SIDE_LEFT][BACK_LEFT]      = aluSqrt(0.5);
 291             Device->ChannelMatrix[SIDE_RIGHT][FRONT_RIGHT]   = aluSqrt(0.5);
 292             Device->ChannelMatrix[SIDE_RIGHT][BACK_RIGHT]    = aluSqrt(0.5);
 293             Device->ChannelMatrix[BACK_CENTER][BACK_LEFT]    = aluSqrt(0.5);
 294             Device->ChannelMatrix[BACK_CENTER][BACK_RIGHT]   = aluSqrt(0.5);
 295             Device->NumChan = 4;
 296             Speaker2Chan[0] = BACK_LEFT;
 297             Speaker2Chan[1] = FRONT_LEFT;
 298             Speaker2Chan[2] = FRONT_RIGHT;
 299             Speaker2Chan[3] = BACK_RIGHT;
 300             SpeakerAngle[0] = -135.0f * M_PI/180.0f;
 301             SpeakerAngle[1] =  -45.0f * M_PI/180.0f;
 302             SpeakerAngle[2] =   45.0f * M_PI/180.0f;
 303             SpeakerAngle[3] =  135.0f * M_PI/180.0f;
 304             SetSpeakerArrangement("layout", SpeakerAngle, Speaker2Chan, Device->NumChan);
 305             break;
 306
 307         case AL_FORMAT_51CHN8:
 308         case AL_FORMAT_51CHN16:
 309         case AL_FORMAT_51CHN32:
 310             Device->ChannelMatrix[SIDE_LEFT][FRONT_LEFT]   = aluSqrt(0.5);
 311             Device->ChannelMatrix[SIDE_LEFT][BACK_LEFT]    = aluSqrt(0.5);
 312             Device->ChannelMatrix[SIDE_RIGHT][FRONT_RIGHT] = aluSqrt(0.5);
 313             Device->ChannelMatrix[SIDE_RIGHT][BACK_RIGHT]  = aluSqrt(0.5);
 314             Device->ChannelMatrix[BACK_CENTER][BACK_LEFT]  = aluSqrt(0.5);
 315             Device->ChannelMatrix[BACK_CENTER][BACK_RIGHT] = aluSqrt(0.5);
 316             Device->NumChan = 5;
 317             Speaker2Chan[0] = BACK_LEFT;
 318             Speaker2Chan[1] = FRONT_LEFT;
 319             Speaker2Chan[2] = FRONT_CENTER;
 320             Speaker2Chan[3] = FRONT_RIGHT;
 321             Speaker2Chan[4] = BACK_RIGHT;
 322             SpeakerAngle[0] = -110.0f * M_PI/180.0f;
 323             SpeakerAngle[1] =  -30.0f * M_PI/180.0f;
 324             SpeakerAngle[2] =    0.0f * M_PI/180.0f;
 325             SpeakerAngle[3] =   30.0f * M_PI/180.0f;
 326             SpeakerAngle[4] =  110.0f * M_PI/180.0f;
 327             SetSpeakerArrangement("layout", SpeakerAngle, Speaker2Chan, Device->NumChan);
 328             break;
 329
 330         case AL_FORMAT_61CHN8:
 331         case AL_FORMAT_61CHN16:
 332         case AL_FORMAT_61CHN32:
 333             Device->ChannelMatrix[BACK_LEFT][BACK_CENTER]  = aluSqrt(0.5);
 334             Device->ChannelMatrix[BACK_LEFT][SIDE_LEFT]    = aluSqrt(0.5);
 335             Device->ChannelMatrix[BACK_RIGHT][BACK_CENTER] = aluSqrt(0.5);
 336             Device->ChannelMatrix[BACK_RIGHT][SIDE_RIGHT]  = aluSqrt(0.5);
 337             Device->NumChan = 6;
 338             Speaker2Chan[0] = SIDE_LEFT;
 339             Speaker2Chan[1] = FRONT_LEFT;
 340             Speaker2Chan[2] = FRONT_CENTER;
 341             Speaker2Chan[3] = FRONT_RIGHT;
 342             Speaker2Chan[4] = SIDE_RIGHT;
 343             Speaker2Chan[5] = BACK_CENTER;
 344             SpeakerAngle[0] = -90.0f * M_PI/180.0f;
 345             SpeakerAngle[1] = -30.0f * M_PI/180.0f;
 346             SpeakerAngle[2] =   0.0f * M_PI/180.0f;
 347             SpeakerAngle[3] =  30.0f * M_PI/180.0f;
 348             SpeakerAngle[4] =  90.0f * M_PI/180.0f;
 349             SpeakerAngle[5] = 180.0f * M_PI/180.0f;
 350             SetSpeakerArrangement("layout", SpeakerAngle, Speaker2Chan, Device->NumChan);
 351             break;
 352
 353         case AL_FORMAT_71CHN8:
 354         case AL_FORMAT_71CHN16:
 355         case AL_FORMAT_71CHN32:
 356             Device->ChannelMatrix[BACK_CENTER][BACK_LEFT]  = aluSqrt(0.5);
 357             Device->ChannelMatrix[BACK_CENTER][BACK_RIGHT] = aluSqrt(0.5);
 358             Device->NumChan = 7;
 359             Speaker2Chan[0] = BACK_LEFT;
 360             Speaker2Chan[1] = SIDE_LEFT;
 361             Speaker2Chan[2] = FRONT_LEFT;
 362             Speaker2Chan[3] = FRONT_CENTER;
 363             Speaker2Chan[4] = FRONT_RIGHT;
 364             Speaker2Chan[5] = SIDE_RIGHT;
 365             Speaker2Chan[6] = BACK_RIGHT;
 366             SpeakerAngle[0] = -150.0f * M_PI/180.0f;
 367             SpeakerAngle[1] =  -90.0f * M_PI/180.0f;
 368             SpeakerAngle[2] =  -30.0f * M_PI/180.0f;
 369             SpeakerAngle[3] =    0.0f * M_PI/180.0f;
 370             SpeakerAngle[4] =   30.0f * M_PI/180.0f;
 371             SpeakerAngle[5] =   90.0f * M_PI/180.0f;
 372             SpeakerAngle[6] =  150.0f * M_PI/180.0f;
 373             SetSpeakerArrangement("layout", SpeakerAngle, Speaker2Chan, Device->NumChan);
 374             break;
 375
 376         default:
 377             assert(0);
 378     }
 379
 380     maxout = 1.0f;
 381     for(s = 0;s < OUTPUTCHANNELS;s++)
 382     {
 383         ALfloat out = 0.0f;
 384         for(s2 = 0;s2 < OUTPUTCHANNELS;s2++)
 385             out += Device->ChannelMatrix[s2][s];
 386         maxout = __max(maxout, out);
 387     }
 388
 389     maxout = 1.0f/maxout;
 390     for(s = 0;s < OUTPUTCHANNELS;s++)
 391     {
 392         for(s2 = 0;s2 < OUTPUTCHANNELS;s2++)
 393             Device->ChannelMatrix[s2][s] *= maxout;
 394     }
 395
 396
 397     for(pos = 0; pos < LUT_NUM; pos++)
 398     {
 399         /* clear all values */
 400         offset = OUTPUTCHANNELS * pos;
 401         for(s = 0; s < OUTPUTCHANNELS; s++)
 402             Device->PanningLUT[offset+s] = 0.0f;
 403
 404         if(Device->NumChan == 1)
 405         {
 406             Device->PanningLUT[offset + Speaker2Chan[0]] = 1.0f;
 407             continue;
 408         }
 409
 410         /* source angle */
 411         Theta = aluLUTpos2Angle(pos);
 412
 413         /* set panning values */
 414         for(s = 0; s < Device->NumChan - 1; s++)
 415         {
 416             if(Theta >= SpeakerAngle[s] && Theta < SpeakerAngle[s+1])
 417             {
 418                 /* source between speaker s and speaker s+1 */
 419                 Alpha = M_PI_2 * (Theta-SpeakerAngle[s]) /
 420                                  (SpeakerAngle[s+1]-SpeakerAngle[s]);
 421                 Device->PanningLUT[offset + Speaker2Chan[s]]   = cos(Alpha);
 422                 Device->PanningLUT[offset + Speaker2Chan[s+1]] = sin(Alpha);
 423                 break;
 424             }
 425         }
 426         if(s == Device->NumChan - 1)
 427         {
 428             /* source between last and first speaker */
 429             if(Theta < SpeakerAngle[0])
 430                 Theta += 2.0f * M_PI;
 431             Alpha = M_PI_2 * (Theta-SpeakerAngle[s]) /
 432                              (2.0f * M_PI + SpeakerAngle[0]-SpeakerAngle[s]);
 433             Device->PanningLUT[offset + Speaker2Chan[s]] = cos(Alpha);
 434             Device->PanningLUT[offset + Speaker2Chan[0]] = sin(Alpha);
 435         }
 436     }
 437 }
 438
 439 static ALvoid CalcNonAttnSourceParams(const ALCcontext *ALContext, ALsource *ALSource)
 440 {
 441     ALfloat SourceVolume,ListenerGain,MinVolume,MaxVolume;
 442     ALfloat DryGain, DryGainHF;
 443     ALfloat WetGain[MAX_SENDS];
 444     ALfloat WetGainHF[MAX_SENDS];
 445     ALint NumSends, Frequency;
 446     ALfloat cw;
 447     ALint i;
 448
 449     //Get context properties
 450     NumSends  = ALContext->Device->NumAuxSends;
 451     Frequency = ALContext->Device->Frequency;
 452
 453     //Get listener properties
 454     ListenerGain = ALContext->Listener.Gain;
 455
 456     //Get source properties
 457     SourceVolume = ALSource->flGain;
 458     MinVolume    = ALSource->flMinGain;
 459     MaxVolume    = ALSource->flMaxGain;
 460
 461     //1. Multi-channel buffers always play "normal"
 462     ALSource->Params.Pitch = ALSource->flPitch;
 463
 464     DryGain = SourceVolume;
 465     DryGain = __min(DryGain,MaxVolume);
 466     DryGain = __max(DryGain,MinVolume);
 467     DryGainHF = 1.0f;
 468
 469     switch(ALSource->DirectFilter.type)
 470     {
 471         case AL_FILTER_LOWPASS:
 472             DryGain *= ALSource->DirectFilter.Gain;
 473             DryGainHF *= ALSource->DirectFilter.GainHF;
 474             break;
 475     }
 476
 477     ALSource->Params.DryGains[FRONT_LEFT]   = DryGain * ListenerGain;
 478     ALSource->Params.DryGains[FRONT_RIGHT]  = DryGain * ListenerGain;
 479     ALSource->Params.DryGains[SIDE_LEFT]    = DryGain * ListenerGain;
 480     ALSource->Params.DryGains[SIDE_RIGHT]   = DryGain * ListenerGain;
 481     ALSource->Params.DryGains[BACK_LEFT]    = DryGain * ListenerGain;
 482     ALSource->Params.DryGains[BACK_RIGHT]   = DryGain * ListenerGain;
 483     ALSource->Params.DryGains[FRONT_CENTER] = DryGain * ListenerGain;
 484     ALSource->Params.DryGains[BACK_CENTER]  = DryGain * ListenerGain;
 485     ALSource->Params.DryGains[LFE]          = DryGain * ListenerGain;
 486
 487     for(i = 0;i < NumSends;i++)
 488     {
 489         WetGain[i] = SourceVolume;
 490         WetGain[i] = __min(WetGain[i],MaxVolume);
 491         WetGain[i] = __max(WetGain[i],MinVolume);
 492         WetGainHF[i] = 1.0f;
 493
 494         switch(ALSource->Send[i].WetFilter.type)
 495         {
 496             case AL_FILTER_LOWPASS:
 497                 WetGain[i] *= ALSource->Send[i].WetFilter.Gain;
 498                 WetGainHF[i] *= ALSource->Send[i].WetFilter.GainHF;
 499                 break;
 500         }
 501
 502         ALSource->Params.WetGains[i] = WetGain[i] * ListenerGain;
 503     }
 504     for(i = NumSends;i < MAX_SENDS;i++)
 505     {
 506         ALSource->Params.WetGains[i] = 0.0f;
 507         WetGainHF[i] = 1.0f;
 508     }
 509
 510     /* Update filter coefficients. Calculations based on the I3DL2
 511      * spec. */
 512     cw = cos(2.0*M_PI * LOWPASSFREQCUTOFF / Frequency);
 513
 514     /* We use two chained one-pole filters, so we need to take the
 515      * square root of the squared gain, which is the same as the base
 516      * gain. */
 517     ALSource->Params.iirFilter.coeff = lpCoeffCalc(DryGainHF, cw);
 518
 519     for(i = 0;i < NumSends;i++)
 520     {
 521         /* We use a one-pole filter, so we need to take the squared gain */
 522         ALfloat a = lpCoeffCalc(WetGainHF[i]*WetGainHF[i], cw);
 523         ALSource->Params.Send[i].iirFilter.coeff = a;
 524     }
 525 }
 526
 527 static ALvoid CalcSourceParams(const ALCcontext *ALContext, ALsource *ALSource)
 528 {
 529     ALfloat InnerAngle,OuterAngle,Angle,Distance,DryMix,OrigDist;
 530     ALfloat Direction[3],Position[3],SourceToListener[3];
 531     ALfloat Velocity[3],ListenerVel[3];
 532     ALfloat MinVolume,MaxVolume,MinDist,MaxDist,Rolloff,OuterGainHF;
 533     ALfloat ConeVolume,ConeHF,SourceVolume,ListenerGain;
 534     ALfloat DopplerFactor, DopplerVelocity, flSpeedOfSound;
 535     ALfloat Matrix[4][4];
 536     ALfloat flAttenuation, effectiveDist;
 537     ALfloat RoomAttenuation[MAX_SENDS];
 538     ALfloat MetersPerUnit;
 539     ALfloat RoomRolloff[MAX_SENDS];
 540     ALfloat DryGainHF = 1.0f;
 541     ALfloat WetGain[MAX_SENDS];
 542     ALfloat WetGainHF[MAX_SENDS];
 543     ALfloat DirGain, AmbientGain;
 544     ALfloat length;
 545     const ALfloat *SpeakerGain;
 546     ALuint Frequency;
 547     ALint NumSends;
 548     ALint pos, s, i;
 549     ALfloat cw;
 550
 551     for(i = 0;i < MAX_SENDS;i++)
 552         WetGainHF[i] = 1.0f;
 553
 554     //Get context properties
 555     DopplerFactor   = ALContext->DopplerFactor * ALSource->DopplerFactor;
 556     DopplerVelocity = ALContext->DopplerVelocity;
 557     flSpeedOfSound  = ALContext->flSpeedOfSound;
 558     NumSends        = ALContext->Device->NumAuxSends;
 559     Frequency       = ALContext->Device->Frequency;
 560
 561     //Get listener properties
 562     ListenerGain = ALContext->Listener.Gain;
 563     MetersPerUnit = ALContext->Listener.MetersPerUnit;
 564     memcpy(ListenerVel, ALContext->Listener.Velocity, sizeof(ALContext->Listener.Velocity));
 565
 566     //Get source properties
 567     SourceVolume = ALSource->flGain;
 568     memcpy(Position,  ALSource->vPosition,    sizeof(ALSource->vPosition));
 569     memcpy(Direction, ALSource->vOrientation, sizeof(ALSource->vOrientation));
 570     memcpy(Velocity,  ALSource->vVelocity,    sizeof(ALSource->vVelocity));
 571     MinVolume    = ALSource->flMinGain;
 572     MaxVolume    = ALSource->flMaxGain;
 573     MinDist      = ALSource->flRefDistance;
 574     MaxDist      = ALSource->flMaxDistance;
 575     Rolloff      = ALSource->flRollOffFactor;
 576     InnerAngle   = ALSource->flInnerAngle;
 577     OuterAngle   = ALSource->flOuterAngle;
 578     OuterGainHF  = ALSource->OuterGainHF;
 579
 580     //1. Translate Listener to origin (convert to head relative)
 581     if(ALSource->bHeadRelative==AL_FALSE)
 582     {
 583         ALfloat U[3],V[3],N[3];
 584
 585         // Build transform matrix
 586         memcpy(N, ALContext->Listener.Forward, sizeof(N));  // At-vector
 587         aluNormalize(N);  // Normalized At-vector
 588         memcpy(V, ALContext->Listener.Up, sizeof(V));  // Up-vector
 589         aluNormalize(V);  // Normalized Up-vector
 590         aluCrossproduct(N, V, U); // Right-vector
 591         aluNormalize(U);  // Normalized Right-vector
 592         Matrix[0][0] = U[0]; Matrix[0][1] = V[0]; Matrix[0][2] = -N[0]; Matrix[0][3] = 0.0f;
 593         Matrix[1][0] = U[1]; Matrix[1][1] = V[1]; Matrix[1][2] = -N[1]; Matrix[1][3] = 0.0f;
 594         Matrix[2][0] = U[2]; Matrix[2][1] = V[2]; Matrix[2][2] = -N[2]; Matrix[2][3] = 0.0f;
 595         Matrix[3][0] = 0.0f; Matrix[3][1] = 0.0f; Matrix[3][2] =  0.0f; Matrix[3][3] = 1.0f;
 596
 597         // Translate position
 598         Position[0] -= ALContext->Listener.Position[0];
 599         Position[1] -= ALContext->Listener.Position[1];
 600         Position[2] -= ALContext->Listener.Position[2];
 601
 602         // Transform source position and direction into listener space
 603         aluMatrixVector(Position, 1.0f, Matrix);
 604         aluMatrixVector(Direction, 0.0f, Matrix);
 605         // Transform source and listener velocity into listener space
 606         aluMatrixVector(Velocity, 0.0f, Matrix);
 607         aluMatrixVector(ListenerVel, 0.0f, Matrix);
 608     }
 609     else
 610         ListenerVel[0] = ListenerVel[1] = ListenerVel[2] = 0.0f;
 611
 612     SourceToListener[0] = -Position[0];
 613     SourceToListener[1] = -Position[1];
 614     SourceToListener[2] = -Position[2];
 615     aluNormalize(SourceToListener);
 616     aluNormalize(Direction);
 617
 618     //2. Calculate distance attenuation
 619     Distance = aluSqrt(aluDotproduct(Position, Position));
 620     OrigDist = Distance;
 621
 622     flAttenuation = 1.0f;
 623     for(i = 0;i < NumSends;i++)
 624     {
 625         RoomAttenuation[i] = 1.0f;
 626
 627         RoomRolloff[i] = ALSource->RoomRolloffFactor;
 628         if(ALSource->Send[i].Slot &&
 629            (ALSource->Send[i].Slot->effect.type == AL_EFFECT_REVERB ||
 630             ALSource->Send[i].Slot->effect.type == AL_EFFECT_EAXREVERB))
 631             RoomRolloff[i] += ALSource->Send[i].Slot->effect.Reverb.RoomRolloffFactor;
 632     }
 633
 634     switch(ALContext->SourceDistanceModel ? ALSource->DistanceModel :
 635                                             ALContext->DistanceModel)
 636     {
 637         case AL_INVERSE_DISTANCE_CLAMPED:
 638             Distance=__max(Distance,MinDist);
 639             Distance=__min(Distance,MaxDist);
 640             if(MaxDist < MinDist)
 641                 break;
 642             //fall-through
 643         case AL_INVERSE_DISTANCE:
 644             if(MinDist > 0.0f)
 645             {
 646                 if((MinDist + (Rolloff * (Distance - MinDist))) > 0.0f)
 647                     flAttenuation = MinDist / (MinDist + (Rolloff * (Distance - MinDist)));
 648                 for(i = 0;i < NumSends;i++)
 649                 {
 650                     if((MinDist + (RoomRolloff[i] * (Distance - MinDist))) > 0.0f)
 651                         RoomAttenuation[i] = MinDist / (MinDist + (RoomRolloff[i] * (Distance - MinDist)));
 652                 }
 653             }
 654             break;
 655
 656         case AL_LINEAR_DISTANCE_CLAMPED:
 657             Distance=__max(Distance,MinDist);
 658             Distance=__min(Distance,MaxDist);
 659             if(MaxDist < MinDist)
 660                 break;
 661             //fall-through
 662         case AL_LINEAR_DISTANCE:
 663             Distance=__min(Distance,MaxDist);
 664             if(MaxDist != MinDist)
 665             {
 666                 flAttenuation = 1.0f - (Rolloff*(Distance-MinDist)/(MaxDist - MinDist));
 667                 for(i = 0;i < NumSends;i++)
 668                     RoomAttenuation[i] = 1.0f - (RoomRolloff[i]*(Distance-MinDist)/(MaxDist - MinDist));
 669             }
 670             break;
 671
 672         case AL_EXPONENT_DISTANCE_CLAMPED:
 673             Distance=__max(Distance,MinDist);
 674             Distance=__min(Distance,MaxDist);
 675             if(MaxDist < MinDist)
 676                 break;
 677             //fall-through
 678         case AL_EXPONENT_DISTANCE:
 679             if(Distance > 0.0f && MinDist > 0.0f)
 680             {
 681                 flAttenuation = aluPow(Distance/MinDist, -Rolloff);
 682                 for(i = 0;i < NumSends;i++)
 683                     RoomAttenuation[i] = aluPow(Distance/MinDist, -RoomRolloff[i]);
 684             }
 685             break;
 686
 687         case AL_NONE:
 688             break;
 689     }
 690
 691     // Source Gain + Attenuation
 692     DryMix = SourceVolume * flAttenuation;
 693     for(i = 0;i < NumSends;i++)
 694         WetGain[i] = SourceVolume * RoomAttenuation[i];
 695
 696     effectiveDist = 0.0f;
 697     if(MinDist > 0.0f)
 698         effectiveDist = (MinDist/flAttenuation - MinDist)*MetersPerUnit;
 699
 700     // Distance-based air absorption
 701     if(ALSource->AirAbsorptionFactor > 0.0f && effectiveDist > 0.0f)
 702     {
 703         ALfloat absorb;
 704
 705         // Absorption calculation is done in dB
 706         absorb = (ALSource->AirAbsorptionFactor*AIRABSORBGAINDBHF) *
 707                  effectiveDist;
 708         // Convert dB to linear gain before applying
 709         absorb = aluPow(10.0f, absorb/20.0f);
 710
 711         DryGainHF *= absorb;
 712     }
 713
 714     //3. Apply directional soundcones
 715     Angle = aluAcos(aluDotproduct(Direction,SourceToListener)) * 180.0f/M_PI;
 716     if(Angle >= InnerAngle && Angle <= OuterAngle)
 717     {
 718         ALfloat scale = (Angle-InnerAngle) / (OuterAngle-InnerAngle);
 719         ConeVolume = (1.0f+(ALSource->flOuterGain-1.0f)*scale);
 720         ConeHF = (1.0f+(OuterGainHF-1.0f)*scale);
 721     }
 722     else if(Angle > OuterAngle)
 723     {
 724         ConeVolume = (1.0f+(ALSource->flOuterGain-1.0f));
 725         ConeHF = (1.0f+(OuterGainHF-1.0f));
 726     }
 727     else
 728     {
 729         ConeVolume = 1.0f;
 730         ConeHF = 1.0f;
 731     }
 732
 733     // Apply some high-frequency attenuation for sources behind the listener
 734     // NOTE: This should be aluDotproduct({0,0,-1}, ListenerToSource), however
 735     // that is equivalent to aluDotproduct({0,0,1}, SourceToListener), which is
 736     // the same as SourceToListener[2]
 737     Angle = aluAcos(SourceToListener[2]) * 180.0f/M_PI;
 738     // Sources within the minimum distance attenuate less
 739     if(OrigDist < MinDist)
 740         Angle *= OrigDist/MinDist;
 741     if(Angle > 90.0f)
 742     {
 743         ALfloat scale = (Angle-90.0f) / (180.1f-90.0f); // .1 to account for fp errors
 744         ConeHF *= 1.0f - (ALContext->Device->HeadDampen*scale);
 745     }
 746
 747     DryMix *= ConeVolume;
 748     if(ALSource->DryGainHFAuto)
 749         DryGainHF *= ConeHF;
 750
 751     // Clamp to Min/Max Gain
 752     DryMix = __min(DryMix,MaxVolume);
 753     DryMix = __max(DryMix,MinVolume);
 754
 755     for(i = 0;i < NumSends;i++)
 756     {
 757         ALeffectslot *Slot = ALSource->Send[i].Slot;
 758
 759         if(!Slot || Slot->effect.type == AL_EFFECT_NULL)
 760         {
 761             ALSource->Params.WetGains[i] = 0.0f;
 762             WetGainHF[i] = 1.0f;
 763             continue;
 764         }
 765
 766         if(Slot->AuxSendAuto)
 767         {
 768             if(ALSource->WetGainAuto)
 769                 WetGain[i] *= ConeVolume;
 770             if(ALSource->WetGainHFAuto)
 771                 WetGainHF[i] *= ConeHF;
 772
 773             // Clamp to Min/Max Gain
 774             WetGain[i] = __min(WetGain[i],MaxVolume);
 775             WetGain[i] = __max(WetGain[i],MinVolume);
 776
 777             if(Slot->effect.type == AL_EFFECT_REVERB ||
 778                Slot->effect.type == AL_EFFECT_EAXREVERB)
 779             {
 780                 /* Apply a decay-time transformation to the wet path, based on
 781                  * the attenuation of the dry path.
 782                  *
 783                  * Using the approximate (effective) source to listener
 784                  * distance, the initial decay of the reverb effect is
 785                  * calculated and applied to the wet path.
 786                  */
 787                 WetGain[i] *= aluPow(10.0f, effectiveDist /
 788                                             (SPEEDOFSOUNDMETRESPERSEC *
 789                                              Slot->effect.Reverb.DecayTime) *
 790                                             -60.0 / 20.0);
 791
 792                 WetGainHF[i] *= aluPow(10.0f,
 793                                        log10(Slot->effect.Reverb.AirAbsorptionGainHF) *
 794                                        ALSource->AirAbsorptionFactor * effectiveDist);
 795             }
 796         }
 797         else
 798         {
 799             /* If the slot's auxiliary send auto is off, the data sent to the
 800              * effect slot is the same as the dry path, sans filter effects */
 801             WetGain[i] = DryMix;
 802             WetGainHF[i] = DryGainHF;
 803         }
 804
 805         switch(ALSource->Send[i].WetFilter.type)
 806         {
 807             case AL_FILTER_LOWPASS:
 808                 WetGain[i] *= ALSource->Send[i].WetFilter.Gain;
 809                 WetGainHF[i] *= ALSource->Send[i].WetFilter.GainHF;
 810                 break;
 811         }
 812         ALSource->Params.WetGains[i] = WetGain[i] * ListenerGain;
 813     }
 814     for(i = NumSends;i < MAX_SENDS;i++)
 815     {
 816         ALSource->Params.WetGains[i] = 0.0f;
 817         WetGainHF[i] = 1.0f;
 818     }
 819
 820     // Apply filter gains and filters
 821     switch(ALSource->DirectFilter.type)
 822     {
 823         case AL_FILTER_LOWPASS:
 824             DryMix *= ALSource->DirectFilter.Gain;
 825             DryGainHF *= ALSource->DirectFilter.GainHF;
 826             break;
 827     }
 828     DryMix *= ListenerGain;
 829
 830     // Calculate Velocity
 831     if(DopplerFactor != 0.0f)
 832     {
 833         ALfloat flVSS, flVLS;
 834         ALfloat flMaxVelocity = (DopplerVelocity * flSpeedOfSound) /
 835                                 DopplerFactor;
 836
 837         flVSS = aluDotproduct(Velocity, SourceToListener);
 838         if(flVSS >= flMaxVelocity)
 839             flVSS = (flMaxVelocity - 1.0f);
 840         else if(flVSS <= -flMaxVelocity)
 841             flVSS = -flMaxVelocity + 1.0f;
 842
 843         flVLS = aluDotproduct(ListenerVel, SourceToListener);
 844         if(flVLS >= flMaxVelocity)
 845             flVLS = (flMaxVelocity - 1.0f);
 846         else if(flVLS <= -flMaxVelocity)
 847             flVLS = -flMaxVelocity + 1.0f;
 848
 849         ALSource->Params.Pitch = ALSource->flPitch *
 850             ((flSpeedOfSound * DopplerVelocity) - (DopplerFactor * flVLS)) /
 851             ((flSpeedOfSound * DopplerVelocity) - (DopplerFactor * flVSS));
 852     }
 853     else
 854         ALSource->Params.Pitch = ALSource->flPitch;
 855
 856     // Use energy-preserving panning algorithm for multi-speaker playback
 857     length = __max(OrigDist, MinDist);
 858     if(length > 0.0f)
 859     {
 860         ALfloat invlen = 1.0f/length;
 861         Position[0] *= invlen;
 862         Position[1] *= invlen;
 863         Position[2] *= invlen;
 864     }
 865
 866     pos = aluCart2LUTpos(-Position[2], Position[0]);
 867     SpeakerGain = &ALContext->Device->PanningLUT[OUTPUTCHANNELS * pos];
 868
 869     DirGain = aluSqrt(Position[0]*Position[0] + Position[2]*Position[2]);
 870     // elevation adjustment for directional gain. this sucks, but
 871     // has low complexity
 872     AmbientGain = 1.0/aluSqrt(ALContext->Device->NumChan) * (1.0-DirGain);
 873     for(s = 0; s < OUTPUTCHANNELS; s++)
 874     {
 875         ALfloat gain = SpeakerGain[s]*DirGain + AmbientGain;
 876         ALSource->Params.DryGains[s] = DryMix * gain;
 877     }
 878
 879     /* Update filter coefficients. */
 880     cw = cos(2.0*M_PI * LOWPASSFREQCUTOFF / Frequency);
 881
 882     /* Spatialized sources use four chained one-pole filters, so we need to
 883      * take the fourth root of the squared gain, which is the same as the
 884      * square root of the base gain. */
 885     ALSource->Params.iirFilter.coeff = lpCoeffCalc(aluSqrt(DryGainHF), cw);
 886
 887     for(i = 0;i < NumSends;i++)
 888     {
 889         /* The wet path uses two chained one-pole filters, so take the
 890          * base gain (square root of the squared gain) */
 891         ALSource->Params.Send[i].iirFilter.coeff = lpCoeffCalc(WetGainHF[i], cw);
 892     }
 893 }
 894
 895 static __inline ALfloat point(ALfloat val1, ALfloat val2, ALint frac)
 896 {
 897     return val1;
 898     (void)val2;
 899     (void)frac;
 900 }
 901 static __inline ALfloat lerp(ALfloat val1, ALfloat val2, ALint frac)
 902 {
 903     return val1 + ((val2-val1)*(frac * (1.0f/(1<<FRACTIONBITS))));
 904 }
 905 static __inline ALfloat cos_lerp(ALfloat val1, ALfloat val2, ALint frac)
 906 {
 907     ALfloat mult = (1.0f-cos(frac * (1.0f/(1<<FRACTIONBITS)) * M_PI)) * 0.5f;
 908     return val1 + ((val2-val1)*mult);
 909 }
 910
 911 static void MixSomeSources(ALCcontext *ALContext, float (*DryBuffer)[OUTPUTCHANNELS], ALuint SamplesToDo)
 912 {
 913     static float DummyBuffer[BUFFERSIZE];
 914     ALfloat *WetBuffer[MAX_SENDS];
 915     ALfloat DrySend[OUTPUTCHANNELS];
 916     ALfloat dryGainStep[OUTPUTCHANNELS];
 917     ALfloat wetGainStep[MAX_SENDS];
 918     ALuint i, j, k, out;
 919     ALsource *ALSource;
 920     ALfloat value, outsamp;
 921     ALbufferlistitem *BufferListItem;
 922     ALint64 DataSize64,DataPos64;
 923     FILTER *DryFilter, *WetFilter[MAX_SENDS];
 924     ALfloat WetSend[MAX_SENDS];
 925     ALuint rampLength;
 926     ALuint DeviceFreq;
 927     ALint increment;
 928     ALuint DataPosInt, DataPosFrac;
 929     ALuint Channels, Bytes;
 930     ALuint Frequency;
 931     resampler_t Resampler;
 932     ALuint BuffersPlayed;
 933     ALfloat Pitch;
 934     ALenum State;
 935
 936     if(!(ALSource=ALContext->SourceList))
 937         return;
 938
 939     DeviceFreq = ALContext->Device->Frequency;
 940
 941     rampLength = DeviceFreq * MIN_RAMP_LENGTH / 1000;
 942     rampLength = max(rampLength, SamplesToDo);
 943
 944 another_source:
 945     if(ALSource->state != AL_PLAYING)
 946     {
 947         if((ALSource=ALSource->next) != NULL)
 948             goto another_source;
 949         return;
 950     }
 951     j = 0;
 952
 953     /* Find buffer format */
 954     Frequency = 0;
 955     Channels = 0;
 956     Bytes = 0;
 957     BufferListItem = ALSource->queue;
 958     while(BufferListItem != NULL)
 959     {
 960         ALbuffer *ALBuffer;
 961         if((ALBuffer=BufferListItem->buffer) != NULL)
 962         {
 963             Channels  = aluChannelsFromFormat(ALBuffer->format);
 964             Bytes     = aluBytesFromFormat(ALBuffer->format);
 965             Frequency = ALBuffer->frequency;
 966             break;
 967         }
 968         BufferListItem = BufferListItem->next;
 969     }
 970
 971     if(ALSource->NeedsUpdate)
 972     {
 973         //Only apply 3D calculations for mono buffers
 974         if(Channels == 1)
 975             CalcSourceParams(ALContext, ALSource);
 976         else
 977             CalcNonAttnSourceParams(ALContext, ALSource);
 978         ALSource->NeedsUpdate = AL_FALSE;
 979     }
 980
 981     /* Get source info */
 982     Resampler     = ALSource->Resampler;
 983     State         = ALSource->state;
 984     BuffersPlayed = ALSource->BuffersPlayed;
 985     DataPosInt    = ALSource->position;
 986     DataPosFrac   = ALSource->position_fraction;
 987
 988     /* Compute 18.14 fixed point step */
 989     Pitch = (ALSource->Params.Pitch*Frequency) / DeviceFreq;
 990     if(Pitch > (float)MAX_PITCH)  Pitch = (float)MAX_PITCH;
 991     increment = (ALint)(Pitch*(ALfloat)(1L<<FRACTIONBITS));
 992     if(increment <= 0)  increment = (1<<FRACTIONBITS);
 993
 994     if(ALSource->FirstStart)
 995     {
 996         for(i = 0;i < OUTPUTCHANNELS;i++)
 997             DrySend[i] = ALSource->Params.DryGains[i];
 998         for(i = 0;i < MAX_SENDS;i++)
 999             WetSend[i] = ALSource->Params.WetGains[i];
1000     }
1001     else
1002     {
1003         for(i = 0;i < OUTPUTCHANNELS;i++)
1004             DrySend[i] = ALSource->DryGains[i];
1005         for(i = 0;i < MAX_SENDS;i++)
1006             WetSend[i] = ALSource->WetGains[i];
1007     }
1008
1009     DryFilter = &ALSource->Params.iirFilter;
1010     for(i = 0;i < MAX_SENDS;i++)
1011     {
1012         WetFilter[i] = &ALSource->Params.Send[i].iirFilter;
1013         WetBuffer[i] = (ALSource->Send[i].Slot ?
1014                         ALSource->Send[i].Slot->WetBuffer :
1015                         DummyBuffer);
1016     }
1017
1018     /* Get current buffer queue item */
1019     BufferListItem = ALSource->queue;
1020     for(i = 0;i < BuffersPlayed && BufferListItem;i++)
1021         BufferListItem = BufferListItem->next;
1022
1023     while(State == AL_PLAYING && j < SamplesToDo)
1024     {
1025         ALuint DataSize = 0;
1026         ALbuffer *ALBuffer;
1027         ALfloat *Data;
1028         ALuint BufferSize;
1029
1030         /* Get buffer info */
1031         if((ALBuffer=BufferListItem->buffer) != NULL)
1032         {
1033             Data      = ALBuffer->data;
1034             DataSize  = ALBuffer->size;
1035             DataSize /= Channels * Bytes;
1036         }
1037         if(DataPosInt >= DataSize)
1038             goto skipmix;
1039
1040         if(BufferListItem->next)
1041         {
1042             ALbuffer *NextBuf = BufferListItem->next->buffer;
1043             if(NextBuf && NextBuf->size)
1044             {
1045                 ALint ulExtraSamples = BUFFER_PADDING*Channels*Bytes;
1046                 ulExtraSamples = min(NextBuf->size, ulExtraSamples);
1047                 memcpy(&Data[DataSize*Channels], NextBuf->data, ulExtraSamples);
1048             }
1049         }
1050         else if(ALSource->bLooping)
1051         {
1052             ALbuffer *NextBuf = ALSource->queue->buffer;
1053             if(NextBuf && NextBuf->size)
1054             {
1055                 ALint ulExtraSamples = BUFFER_PADDING*Channels*Bytes;
1056                 ulExtraSamples = min(NextBuf->size, ulExtraSamples);
1057                 memcpy(&Data[DataSize*Channels], NextBuf->data, ulExtraSamples);
1058             }
1059         }
1060         else
1061             memset(&Data[DataSize*Channels], 0, (BUFFER_PADDING*Channels*Bytes));
1062
1063         /* Compute the gain steps for each output channel */
1064         for(i = 0;i < OUTPUTCHANNELS;i++)
1065             dryGainStep[i] = (ALSource->Params.DryGains[i]-DrySend[i]) /
1066                              rampLength;
1067         for(i = 0;i < MAX_SENDS;i++)
1068             wetGainStep[i] = (ALSource->Params.WetGains[i]-WetSend[i]) /
1069                              rampLength;
1070
1071         /* Figure out how many samples we can mix. */
1072         DataSize64 = DataSize;
1073         DataSize64 <<= FRACTIONBITS;
1074         DataPos64 = DataPosInt;
1075         DataPos64 <<= FRACTIONBITS;
1076         DataPos64 += DataPosFrac;
1077         BufferSize = (ALuint)((DataSize64-DataPos64+(increment-1)) / increment);
1078
1079         BufferSize = min(BufferSize, (SamplesToDo-j));
1080
1081         /* Actual sample mixing loop */
1082         k = 0;
1083         Data += DataPosInt*Channels;
1084
1085         if(Channels == 1) /* Mono */
1086         {
1087 #define DO_MIX(resampler) do { \
1088     while(BufferSize--) \
1089     { \
1090         for(i = 0;i < OUTPUTCHANNELS;i++) \
1091             DrySend[i] += dryGainStep[i]; \
1092         for(i = 0;i < MAX_SENDS;i++) \
1093             WetSend[i] += wetGainStep[i]; \
1094  \
1095         /* First order interpolator */ \
1096         value = (resampler)(Data[k], Data[k+1], DataPosFrac); \
1097  \
1098         /* Direct path final mix buffer and panning */ \
1099         outsamp = lpFilter4P(DryFilter, 0, value); \
1100         DryBuffer[j][FRONT_LEFT]   += outsamp*DrySend[FRONT_LEFT]; \
1101         DryBuffer[j][FRONT_RIGHT]  += outsamp*DrySend[FRONT_RIGHT]; \
1102         DryBuffer[j][SIDE_LEFT]    += outsamp*DrySend[SIDE_LEFT]; \
1103         DryBuffer[j][SIDE_RIGHT]   += outsamp*DrySend[SIDE_RIGHT]; \
1104         DryBuffer[j][BACK_LEFT]    += outsamp*DrySend[BACK_LEFT]; \
1105         DryBuffer[j][BACK_RIGHT]   += outsamp*DrySend[BACK_RIGHT]; \
1106         DryBuffer[j][FRONT_CENTER] += outsamp*DrySend[FRONT_CENTER]; \
1107         DryBuffer[j][BACK_CENTER]  += outsamp*DrySend[BACK_CENTER]; \
1108  \
1109         /* Room path final mix buffer and panning */ \
1110         for(i = 0;i < MAX_SENDS;i++) \
1111         { \
1112             outsamp = lpFilter2P(WetFilter[i], 0, value); \
1113             WetBuffer[i][j] += outsamp*WetSend[i]; \
1114         } \
1115  \
1116         DataPosFrac += increment; \
1117         k += DataPosFrac>>FRACTIONBITS; \
1118         DataPosFrac &= FRACTIONMASK; \
1119         j++; \
1120     } \
1121 } while(0)
1122
1123             switch(Resampler)
1124             {
1125                 case POINT_RESAMPLER:
1126                 DO_MIX(point); break;
1127                 case LINEAR_RESAMPLER:
1128                 DO_MIX(lerp); break;
1129                 case COSINE_RESAMPLER:
1130                 DO_MIX(cos_lerp); break;
1131                 case RESAMPLER_MIN:
1132                 case RESAMPLER_MAX:
1133                 break;
1134             }
1135 #undef DO_MIX
1136         }
1137         else if(Channels == 2 && DuplicateStereo) /* Stereo */
1138         {
1139             const int chans[] = {
1140                 FRONT_LEFT, FRONT_RIGHT
1141             };
1142             const int chans2[] = {
1143                 BACK_LEFT, SIDE_LEFT, BACK_RIGHT, SIDE_RIGHT
1144             };
1145             const ALfloat scaler = 1.0f/Channels;
1146             const ALfloat dupscaler = aluSqrt(1.0f/3.0f);
1147
1148 #define DO_MIX(resampler) do {                                                \
1149     while(BufferSize--)                                                       \
1150     {                                                                         \
1151         for(i = 0;i < OUTPUTCHANNELS;i++)                                     \
1152             DrySend[i] += dryGainStep[i];                                     \
1153         for(i = 0;i < MAX_SENDS;i++)                                          \
1154             WetSend[i] += wetGainStep[i];                                     \
1155                                                                               \
1156         for(i = 0;i < Channels;i++)                                           \
1157         {                                                                     \
1158             value = (resampler)(Data[k*Channels + i],Data[(k+1)*Channels + i],\
1159                                 DataPosFrac);                                 \
1160             outsamp = lpFilter2P(DryFilter, chans[i]*2, value) * dupscaler;   \
1161             DryBuffer[j][chans[i]] += outsamp*DrySend[chans[i]];              \
1162             DryBuffer[j][chans2[i*2+0]] += outsamp*DrySend[chans2[i*2+0]];    \
1163             DryBuffer[j][chans2[i*2+1]] += outsamp*DrySend[chans2[i*2+1]];    \
1164             for(out = 0;out < MAX_SENDS;out++)                                \
1165             {                                                                 \
1166                 outsamp = lpFilter1P(WetFilter[out], chans[i], value);        \
1167                 WetBuffer[out][j] += outsamp*WetSend[out]*scaler;             \
1168             }                                                                 \
1169         }                                                                     \
1170                                                                               \
1171         DataPosFrac += increment;                                             \
1172         k += DataPosFrac>>FRACTIONBITS;                                       \
1173         DataPosFrac &= FRACTIONMASK;                                          \
1174         j++;                                                                  \
1175     }                                                                         \
1176 } while(0)
1177
1178             switch(Resampler)
1179             {
1180                 case POINT_RESAMPLER:
1181                 DO_MIX(point); break;
1182                 case LINEAR_RESAMPLER:
1183                 DO_MIX(lerp); break;
1184                 case COSINE_RESAMPLER:
1185                 DO_MIX(cos_lerp); break;
1186                 case RESAMPLER_MIN:
1187                 case RESAMPLER_MAX:
1188                 break;
1189             }
1190 #undef DO_MIX
1191         }
1192         else if(Channels == 2) /* Stereo */
1193         {
1194             const int chans[] = {
1195                 FRONT_LEFT, FRONT_RIGHT
1196             };
1197             const ALfloat scaler = 1.0f/Channels;
1198
1199 #define DO_MIX(resampler) do {                                                \
1200     while(BufferSize--)                                                       \
1201     {                                                                         \
1202         for(i = 0;i < OUTPUTCHANNELS;i++)                                     \
1203             DrySend[i] += dryGainStep[i];                                     \
1204         for(i = 0;i < MAX_SENDS;i++)                                          \
1205             WetSend[i] += wetGainStep[i];                                     \
1206                                                                               \
1207         for(i = 0;i < Channels;i++)                                           \
1208         {                                                                     \
1209             value = (resampler)(Data[k*Channels + i],Data[(k+1)*Channels + i],\
1210                                 DataPosFrac);                                 \
1211             outsamp = lpFilter2P(DryFilter, chans[i]*2, value);               \
1212             DryBuffer[j][chans[i]] += outsamp*DrySend[chans[i]];              \
1213             for(out = 0;out < MAX_SENDS;out++)                                \
1214             {                                                                 \
1215                 outsamp = lpFilter1P(WetFilter[out], chans[i], value);        \
1216                 WetBuffer[out][j] += outsamp*WetSend[out]*scaler;             \
1217             }                                                                 \
1218         }                                                                     \
1219                                                                               \
1220         DataPosFrac += increment;                                             \
1221         k += DataPosFrac>>FRACTIONBITS;                                       \
1222         DataPosFrac &= FRACTIONMASK;                                          \
1223         j++;                                                                  \
1224     }                                                                         \
1225 } while(0)
1226
1227             switch(Resampler)
1228             {
1229                 case POINT_RESAMPLER:
1230                 DO_MIX(point); break;
1231                 case LINEAR_RESAMPLER:
1232                 DO_MIX(lerp); break;
1233                 case COSINE_RESAMPLER:
1234                 DO_MIX(cos_lerp); break;
1235                 case RESAMPLER_MIN:
1236                 case RESAMPLER_MAX:
1237                 break;
1238             }
1239         }
1240         else if(Channels == 4) /* Quad */
1241         {
1242             const int chans[] = {
1243                 FRONT_LEFT, FRONT_RIGHT,
1244                 BACK_LEFT,  BACK_RIGHT
1245             };
1246             const ALfloat scaler = 1.0f/Channels;
1247
1248             switch(Resampler)
1249             {
1250                 case POINT_RESAMPLER:
1251                 DO_MIX(point); break;
1252                 case LINEAR_RESAMPLER:
1253                 DO_MIX(lerp); break;
1254                 case COSINE_RESAMPLER:
1255                 DO_MIX(cos_lerp); break;
1256                 case RESAMPLER_MIN:
1257                 case RESAMPLER_MAX:
1258                 break;
1259             }
1260         }
1261         else if(Channels == 6) /* 5.1 */
1262         {
1263             const int chans[] = {
1264                 FRONT_LEFT,   FRONT_RIGHT,
1265                 FRONT_CENTER, LFE,
1266                 BACK_LEFT,    BACK_RIGHT
1267             };
1268             const ALfloat scaler = 1.0f/Channels;
1269
1270             switch(Resampler)
1271             {
1272                 case POINT_RESAMPLER:
1273                 DO_MIX(point); break;
1274                 case LINEAR_RESAMPLER:
1275                 DO_MIX(lerp); break;
1276                 case COSINE_RESAMPLER:
1277                 DO_MIX(cos_lerp); break;
1278                 case RESAMPLER_MIN:
1279                 case RESAMPLER_MAX:
1280                 break;
1281             }
1282         }
1283         else if(Channels == 7) /* 6.1 */
1284         {
1285             const int chans[] = {
1286                 FRONT_LEFT,   FRONT_RIGHT,
1287                 FRONT_CENTER, LFE,
1288                 BACK_CENTER,
1289                 SIDE_LEFT,    SIDE_RIGHT
1290             };
1291             const ALfloat scaler = 1.0f/Channels;
1292
1293             switch(Resampler)
1294             {
1295                 case POINT_RESAMPLER:
1296                 DO_MIX(point); break;
1297                 case LINEAR_RESAMPLER:
1298                 DO_MIX(lerp); break;
1299                 case COSINE_RESAMPLER:
1300                 DO_MIX(cos_lerp); break;
1301                 case RESAMPLER_MIN:
1302                 case RESAMPLER_MAX:
1303                 break;
1304             }
1305         }
1306         else if(Channels == 8) /* 7.1 */
1307         {
1308             const int chans[] = {
1309                 FRONT_LEFT,   FRONT_RIGHT,
1310                 FRONT_CENTER, LFE,
1311                 BACK_LEFT,    BACK_RIGHT,
1312                 SIDE_LEFT,    SIDE_RIGHT
1313             };
1314             const ALfloat scaler = 1.0f/Channels;
1315
1316             switch(Resampler)
1317             {
1318                 case POINT_RESAMPLER:
1319                 DO_MIX(point); break;
1320                 case LINEAR_RESAMPLER:
1321                 DO_MIX(lerp); break;
1322                 case COSINE_RESAMPLER:
1323                 DO_MIX(cos_lerp); break;
1324                 case RESAMPLER_MIN:
1325                 case RESAMPLER_MAX:
1326                 break;
1327             }
1328 #undef DO_MIX
1329         }
1330         else /* Unknown? */
1331         {
1332             for(i = 0;i < OUTPUTCHANNELS;i++)
1333                 DrySend[i] += dryGainStep[i]*BufferSize;
1334             for(i = 0;i < MAX_SENDS;i++)
1335                 WetSend[i] += wetGainStep[i]*BufferSize;
1336             while(BufferSize--)
1337             {
1338                 DataPosFrac += increment;
1339                 k += DataPosFrac>>FRACTIONBITS;
1340                 DataPosFrac &= FRACTIONMASK;
1341                 j++;
1342             }
1343         }
1344         DataPosInt += k;
1345
1346     skipmix:
1347         /* Handle looping sources */
1348         if(DataPosInt >= DataSize)
1349         {
1350             if(BuffersPlayed < (ALSource->BuffersInQueue-1))
1351             {
1352                 BufferListItem = BufferListItem->next;
1353                 BuffersPlayed++;
1354                 DataPosInt -= DataSize;
1355             }
1356             else if(ALSource->bLooping)
1357             {
1358                 BufferListItem = ALSource->queue;
1359                 BuffersPlayed = 0;
1360                 if(ALSource->BuffersInQueue == 1)
1361                     DataPosInt %= DataSize;
1362                 else
1363                     DataPosInt -= DataSize;
1364             }
1365             else
1366             {
1367                 State = AL_STOPPED;
1368                 BufferListItem = ALSource->queue;
1369                 BuffersPlayed = ALSource->BuffersInQueue;
1370                 DataPosInt = 0;
1371                 DataPosFrac = 0;
1372             }
1373         }
1374     }
1375
1376     /* Update source info */
1377     ALSource->state             = State;
1378     ALSource->BuffersPlayed     = BuffersPlayed;
1379     ALSource->position          = DataPosInt;
1380     ALSource->position_fraction = DataPosFrac;
1381     ALSource->Buffer            = BufferListItem->buffer;
1382
1383     for(i = 0;i < OUTPUTCHANNELS;i++)
1384         ALSource->DryGains[i] = DrySend[i];
1385     for(i = 0;i < MAX_SENDS;i++)
1386         ALSource->WetGains[i] = WetSend[i];
1387
1388     ALSource->FirstStart = AL_FALSE;
1389
1390     if((ALSource=ALSource->next) != NULL)
1391         goto another_source;
1392 }
1393
1394 ALvoid aluMixData(ALCdevice *device, ALvoid *buffer, ALsizei size)
1395 {
1396     float (*DryBuffer)[OUTPUTCHANNELS];
1397     ALfloat (*Matrix)[OUTPUTCHANNELS];
1398     const ALuint *ChanMap;
1399     ALuint SamplesToDo;
1400     ALeffectslot *ALEffectSlot;
1401     ALCcontext *ALContext;
1402     ALfloat samp;
1403     int fpuState;
1404     ALuint i, j, c;
1405
1406 #if defined(HAVE_FESETROUND)
1407     fpuState = fegetround();
1408     fesetround(FE_TOWARDZERO);
1409 #elif defined(HAVE__CONTROLFP)
1410     fpuState = _controlfp(0, 0);
1411     _controlfp(_RC_CHOP, _MCW_RC);
1412 #else
1413     (void)fpuState;
1414 #endif
1415
1416     DryBuffer = device->DryBuffer;
1417     while(size > 0)
1418     {
1419         /* Setup variables */
1420         SamplesToDo = min(size, BUFFERSIZE);
1421
1422         /* Clear mixing buffer */
1423         memset(DryBuffer, 0, SamplesToDo*OUTPUTCHANNELS*sizeof(ALfloat));
1424
1425         SuspendContext(NULL);
1426         for(c = 0;c < device->NumContexts;c++)
1427         {
1428             ALContext = device->Contexts[c];
1429             SuspendContext(ALContext);
1430
1431             MixSomeSources(ALContext, DryBuffer, SamplesToDo);
1432
1433             /* effect slot processing */
1434             ALEffectSlot = ALContext->EffectSlotList;
1435             while(ALEffectSlot)
1436             {
1437                 if(ALEffectSlot->EffectState)
1438                     ALEffect_Process(ALEffectSlot->EffectState, ALEffectSlot, SamplesToDo, ALEffectSlot->WetBuffer, DryBuffer);
1439
1440                 for(i = 0;i < SamplesToDo;i++)
1441                     ALEffectSlot->WetBuffer[i] = 0.0f;
1442                 ALEffectSlot = ALEffectSlot->next;
1443             }
1444             ProcessContext(ALContext);
1445         }
1446         ProcessContext(NULL);
1447
1448         //Post processing loop
1449         ChanMap = device->DevChannels;
1450         Matrix = device->ChannelMatrix;
1451         switch(device->Format)
1452         {
1453 #define CHECK_WRITE_FORMAT(bits, type, func)                                  \
1454         case AL_FORMAT_MONO##bits:                                            \
1455             for(i = 0;i < SamplesToDo;i++)                                    \
1456             {                                                                 \
1457                 samp = 0.0f;                                                  \
1458                 for(c = 0;c < OUTPUTCHANNELS;c++)                             \
1459                     samp += DryBuffer[i][c] * Matrix[c][FRONT_CENTER];        \
1460                 ((type*)buffer)[ChanMap[FRONT_CENTER]] = (func)(samp);        \
1461                 buffer = ((type*)buffer) + 1;                                 \
1462             }                                                                 \
1463             break;                                                            \
1464         case AL_FORMAT_STEREO##bits:                                          \
1465             if(device->Bs2b)                                                  \
1466             {                                                                 \
1467                 for(i = 0;i < SamplesToDo;i++)                                \
1468                 {                                                             \
1469                     float samples[2] = { 0.0f, 0.0f };                        \
1470                     for(c = 0;c < OUTPUTCHANNELS;c++)                         \
1471                     {                                                         \
1472                         samples[0] += DryBuffer[i][c]*Matrix[c][FRONT_LEFT];  \
1473                         samples[1] += DryBuffer[i][c]*Matrix[c][FRONT_RIGHT]; \
1474                     }                                                         \
1475                     bs2b_cross_feed(device->Bs2b, samples);                   \
1476                     ((type*)buffer)[ChanMap[FRONT_LEFT]] = (func)(samples[0]);\
1477                     ((type*)buffer)[ChanMap[FRONT_RIGHT]]= (func)(samples[1]);\
1478                     buffer = ((type*)buffer) + 2;                             \
1479                 }                                                             \
1480             }                                                                 \
1481             else                                                              \
1482             {                                                                 \
1483                 for(i = 0;i < SamplesToDo;i++)                                \
1484                 {                                                             \
1485                     static const Channel chans[] = {                          \
1486                         FRONT_LEFT, FRONT_RIGHT                               \
1487                     };                                                        \
1488                     for(j = 0;j < 2;j++)                                      \
1489                     {                                                         \
1490                         samp = 0.0f;                                          \
1491                         for(c = 0;c < OUTPUTCHANNELS;c++)                     \
1492                             samp += DryBuffer[i][c] * Matrix[c][chans[j]];    \
1493                         ((type*)buffer)[ChanMap[chans[j]]] = (func)(samp);    \
1494                     }                                                         \
1495                     buffer = ((type*)buffer) + 2;                             \
1496                 }                                                             \
1497             }                                                                 \
1498             break;                                                            \
1499         case AL_FORMAT_QUAD##bits:                                            \
1500             for(i = 0;i < SamplesToDo;i++)                                    \
1501             {                                                                 \
1502                 static const Channel chans[] = {                              \
1503                     FRONT_LEFT, FRONT_RIGHT,                                  \
1504                     BACK_LEFT,  BACK_RIGHT,                                   \
1505                 };                                                            \
1506                 for(j = 0;j < 4;j++)                                          \
1507                 {                                                             \
1508                     samp = 0.0f;                                              \
1509                     for(c = 0;c < OUTPUTCHANNELS;c++)                         \
1510                         samp += DryBuffer[i][c] * Matrix[c][chans[j]];        \
1511                     ((type*)buffer)[ChanMap[chans[j]]] = (func)(samp);        \
1512                 }                                                             \
1513                 buffer = ((type*)buffer) + 4;                                 \
1514             }                                                                 \
1515             break;                                                            \
1516         case AL_FORMAT_51CHN##bits:                                           \
1517             for(i = 0;i < SamplesToDo;i++)                                    \
1518             {                                                                 \
1519                 static const Channel chans[] = {                              \
1520                     FRONT_LEFT, FRONT_RIGHT,                                  \
1521                     FRONT_CENTER, LFE,                                        \
1522                     BACK_LEFT,  BACK_RIGHT,                                   \
1523                 };                                                            \
1524                 for(j = 0;j < 6;j++)                                          \
1525                 {                                                             \
1526                     samp = 0.0f;                                              \
1527                     for(c = 0;c < OUTPUTCHANNELS;c++)                         \
1528                         samp += DryBuffer[i][c] * Matrix[c][chans[j]];        \
1529                     ((type*)buffer)[ChanMap[chans[j]]] = (func)(samp);        \
1530                 }                                                             \
1531                 buffer = ((type*)buffer) + 6;                                 \
1532             }                                                                 \
1533             break;                                                            \
1534         case AL_FORMAT_61CHN##bits:                                           \
1535             for(i = 0;i < SamplesToDo;i++)                                    \
1536             {                                                                 \
1537                 static const Channel chans[] = {                              \
1538                     FRONT_LEFT, FRONT_RIGHT,                                  \
1539                     FRONT_CENTER, LFE, BACK_CENTER,                           \
1540                     SIDE_LEFT,  SIDE_RIGHT,                                   \
1541                 };                                                            \
1542                 for(j = 0;j < 7;j++)                                          \
1543                 {                                                             \
1544                     samp = 0.0f;                                              \
1545                     for(c = 0;c < OUTPUTCHANNELS;c++)                         \
1546                         samp += DryBuffer[i][c] * Matrix[c][chans[j]];        \
1547                     ((type*)buffer)[ChanMap[chans[j]]] = (func)(samp);        \
1548                 }                                                             \
1549                 buffer = ((type*)buffer) + 7;                                 \
1550             }                                                                 \
1551             break;                                                            \
1552         case AL_FORMAT_71CHN##bits:                                           \
1553             for(i = 0;i < SamplesToDo;i++)                                    \
1554             {                                                                 \
1555                 static const Channel chans[] = {                              \
1556                     FRONT_LEFT, FRONT_RIGHT,                                  \
1557                     FRONT_CENTER, LFE,                                        \
1558                     BACK_LEFT,  BACK_RIGHT,                                   \
1559                     SIDE_LEFT,  SIDE_RIGHT                                    \
1560                 };                                                            \
1561                 for(j = 0;j < 8;j++)                                          \
1562                 {                                                             \
1563                     samp = 0.0f;                                              \
1564                     for(c = 0;c < OUTPUTCHANNELS;c++)                         \
1565                         samp += DryBuffer[i][c] * Matrix[c][chans[j]];        \
1566                     ((type*)buffer)[ChanMap[chans[j]]] = (func)(samp);        \
1567                 }                                                             \
1568                 buffer = ((type*)buffer) + 8;                                 \
1569             }                                                                 \
1570             break;
1571
1572 #define AL_FORMAT_MONO32 AL_FORMAT_MONO_FLOAT32
1573 #define AL_FORMAT_STEREO32 AL_FORMAT_STEREO_FLOAT32
1574             CHECK_WRITE_FORMAT(8,  ALubyte, aluF2UB)
1575             CHECK_WRITE_FORMAT(16, ALshort, aluF2S)
1576             CHECK_WRITE_FORMAT(32, ALfloat, aluF2F)
1577 #undef AL_FORMAT_STEREO32
1578 #undef AL_FORMAT_MONO32
1579 #undef CHECK_WRITE_FORMAT
1580
1581             default:
1582                 break;
1583         }
1584
1585         size -= SamplesToDo;
1586     }
1587
1588 #if defined(HAVE_FESETROUND)
1589     fesetround(fpuState);
1590 #elif defined(HAVE__CONTROLFP)
1591     _controlfp(fpuState, 0xfffff);
1592 #endif
1593 }
1594
1595 ALvoid aluHandleDisconnect(ALCdevice *device)
1596 {
1597     ALuint i;
1598
1599     SuspendContext(NULL);
1600     for(i = 0;i < device->NumContexts;i++)
1601     {
1602         ALsource *source;
1603
1604         SuspendContext(device->Contexts[i]);
1605
1606         source = device->Contexts[i]->SourceList;
1607         while(source)
1608         {
1609             if(source->state == AL_PLAYING)
1610             {
1611                 source->state = AL_STOPPED;
1612                 source->BuffersPlayed = source->BuffersInQueue;
1613                 source->position = 0;
1614                 source->position_fraction = 0;
1615             }
1616             source = source->next;
1617         }
1618         ProcessContext(device->Contexts[i]);
1619     }
1620
1621     device->Connected = ALC_FALSE;
1622     ProcessContext(NULL);
1623 }