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
  49 static __inline ALfloat aluF2F(ALfloat Value)
  50 {
  51     return Value;
  52 }
  53
  54 static __inline ALshort aluF2S(ALfloat Value)
  55 {
  56     ALint i;
  57
  58     if(Value < 0.0f)
  59     {
  60         i = (ALint)(Value*32768.0f);
  61         i = max(-32768, i);
  62     }
  63     else
  64     {
  65         i = (ALint)(Value*32767.0f);
  66         i = min( 32767, i);
  67     }
  68     return ((ALshort)i);
  69 }
  70
  71 static __inline ALubyte aluF2UB(ALfloat Value)
  72 {
  73     ALshort i = aluF2S(Value);
  74     return (i>>8)+128;
  75 }
  76
  77
  78 static __inline ALvoid aluCrossproduct(const ALfloat *inVector1, const ALfloat *inVector2, ALfloat *outVector)
  79 {
  80     outVector[0] = inVector1[1]*inVector2[2] - inVector1[2]*inVector2[1];
  81     outVector[1] = inVector1[2]*inVector2[0] - inVector1[0]*inVector2[2];
  82     outVector[2] = inVector1[0]*inVector2[1] - inVector1[1]*inVector2[0];
  83 }
  84
  85 static __inline ALfloat aluDotproduct(const ALfloat *inVector1, const ALfloat *inVector2)
  86 {
  87     return inVector1[0]*inVector2[0] + inVector1[1]*inVector2[1] +
  88            inVector1[2]*inVector2[2];
  89 }
  90
  91 static __inline ALvoid aluNormalize(ALfloat *inVector)
  92 {
  93     ALfloat length, inverse_length;
  94
  95     length = aluSqrt(aluDotproduct(inVector, inVector));
  96     if(length != 0.0f)
  97     {
  98         inverse_length = 1.0f/length;
  99         inVector[0] *= inverse_length;
 100         inVector[1] *= inverse_length;
 101         inVector[2] *= inverse_length;
 102     }
 103 }
 104
 105 static __inline ALvoid aluMatrixVector(ALfloat *vector,ALfloat w,ALfloat matrix[4][4])
 106 {
 107     ALfloat temp[4] = {
 108         vector[0], vector[1], vector[2], w
 109     };
 110
 111     vector[0] = temp[0]*matrix[0][0] + temp[1]*matrix[1][0] + temp[2]*matrix[2][0] + temp[3]*matrix[3][0];
 112     vector[1] = temp[0]*matrix[0][1] + temp[1]*matrix[1][1] + temp[2]*matrix[2][1] + temp[3]*matrix[3][1];
 113     vector[2] = temp[0]*matrix[0][2] + temp[1]*matrix[1][2] + temp[2]*matrix[2][2] + temp[3]*matrix[3][2];
 114 }
 115
 116 static ALvoid SetSpeakerArrangement(const char *name, ALfloat SpeakerAngle[OUTPUTCHANNELS],
 117                                     Channel Speaker2Chan[OUTPUTCHANNELS], ALint chans)
 118 {
 119     char layout_str[256];
 120     char *confkey, *next;
 121     char *sep, *end;
 122     Channel val;
 123     int i;
 124
 125     strncpy(layout_str, GetConfigValue(NULL, name, ""), sizeof(layout_str));
 126     layout_str[255] = 0;
 127
 128     if(!layout_str[0])
 129         return;
 130
 131     next = confkey = layout_str;
 132     while(next && *next)
 133     {
 134         confkey = next;
 135         next = strchr(confkey, ',');
 136         if(next)
 137         {
 138             *next = 0;
 139             do {
 140                 next++;
 141             } while(isspace(*next) || *next == ',');
 142         }
 143
 144         sep = strchr(confkey, '=');
 145         if(!sep || confkey == sep)
 146             continue;
 147
 148         end = sep - 1;
 149         while(isspace(*end) && end != confkey)
 150             end--;
 151         *(++end) = 0;
 152
 153         if(strcmp(confkey, "fl") == 0 || strcmp(confkey, "front-left") == 0)
 154             val = FRONT_LEFT;
 155         else if(strcmp(confkey, "fr") == 0 || strcmp(confkey, "front-right") == 0)
 156             val = FRONT_RIGHT;
 157         else if(strcmp(confkey, "fc") == 0 || strcmp(confkey, "front-center") == 0)
 158             val = FRONT_CENTER;
 159         else if(strcmp(confkey, "bl") == 0 || strcmp(confkey, "back-left") == 0)
 160             val = BACK_LEFT;
 161         else if(strcmp(confkey, "br") == 0 || strcmp(confkey, "back-right") == 0)
 162             val = BACK_RIGHT;
 163         else if(strcmp(confkey, "bc") == 0 || strcmp(confkey, "back-center") == 0)
 164             val = BACK_CENTER;
 165         else if(strcmp(confkey, "sl") == 0 || strcmp(confkey, "side-left") == 0)
 166             val = SIDE_LEFT;
 167         else if(strcmp(confkey, "sr") == 0 || strcmp(confkey, "side-right") == 0)
 168             val = SIDE_RIGHT;
 169         else
 170         {
 171             AL_PRINT("Unknown speaker for %s: \"%s\"\n", name, confkey);
 172             continue;
 173         }
 174
 175         *(sep++) = 0;
 176         while(isspace(*sep))
 177             sep++;
 178
 179         for(i = 0;i < chans;i++)
 180         {
 181             if(Speaker2Chan[i] == val)
 182             {
 183                 long angle = strtol(sep, NULL, 10);
 184                 if(angle >= -180 && angle <= 180)
 185                     SpeakerAngle[i] = angle * M_PI/180.0f;
 186                 else
 187                     AL_PRINT("Invalid angle for speaker \"%s\": %ld\n", confkey, angle);
 188                 break;
 189             }
 190         }
 191     }
 192
 193     for(i = 0;i < chans;i++)
 194     {
 195         int min = i;
 196         int i2;
 197
 198         for(i2 = i+1;i2 < chans;i2++)
 199         {
 200             if(SpeakerAngle[i2] < SpeakerAngle[min])
 201                 min = i2;
 202         }
 203
 204         if(min != i)
 205         {
 206             ALfloat tmpf;
 207             Channel tmpc;
 208
 209             tmpf = SpeakerAngle[i];
 210             SpeakerAngle[i] = SpeakerAngle[min];
 211             SpeakerAngle[min] = tmpf;
 212
 213             tmpc = Speaker2Chan[i];
 214             Speaker2Chan[i] = Speaker2Chan[min];
 215             Speaker2Chan[min] = tmpc;
 216         }
 217     }
 218 }
 219
 220 static __inline ALfloat aluLUTpos2Angle(ALint pos)
 221 {
 222     if(pos < QUADRANT_NUM)
 223         return aluAtan((ALfloat)pos / (ALfloat)(QUADRANT_NUM - pos));
 224     if(pos < 2 * QUADRANT_NUM)
 225         return M_PI_2 + aluAtan((ALfloat)(pos - QUADRANT_NUM) / (ALfloat)(2 * QUADRANT_NUM - pos));
 226     if(pos < 3 * QUADRANT_NUM)
 227         return aluAtan((ALfloat)(pos - 2 * QUADRANT_NUM) / (ALfloat)(3 * QUADRANT_NUM - pos)) - M_PI;
 228     return aluAtan((ALfloat)(pos - 3 * QUADRANT_NUM) / (ALfloat)(4 * QUADRANT_NUM - pos)) - M_PI_2;
 229 }
 230
 231 ALvoid aluInitPanning(ALCdevice *Device)
 232 {
 233     ALfloat SpeakerAngle[OUTPUTCHANNELS];
 234     Channel *Speaker2Chan;
 235     ALfloat Alpha, Theta;
 236     ALint pos, offset;
 237     ALuint s, s2;
 238
 239     for(s = 0;s < OUTPUTCHANNELS;s++)
 240     {
 241         for(s2 = 0;s2 < OUTPUTCHANNELS;s2++)
 242             Device->ChannelMatrix[s][s2] = ((s==s2) ? 1.0f : 0.0f);
 243     }
 244
 245     Speaker2Chan = Device->Speaker2Chan;
 246     switch(Device->Format)
 247     {
 248         case AL_FORMAT_MONO8:
 249         case AL_FORMAT_MONO16:
 250         case AL_FORMAT_MONO_FLOAT32:
 251             Device->DuplicateStereo = AL_FALSE;
 252             Device->ChannelMatrix[FRONT_LEFT][FRONT_CENTER]  = aluSqrt(0.5);
 253             Device->ChannelMatrix[FRONT_RIGHT][FRONT_CENTER] = aluSqrt(0.5);
 254             Device->ChannelMatrix[SIDE_LEFT][FRONT_CENTER]   = aluSqrt(0.5);
 255             Device->ChannelMatrix[SIDE_RIGHT][FRONT_CENTER]  = aluSqrt(0.5);
 256             Device->ChannelMatrix[BACK_LEFT][FRONT_CENTER]   = aluSqrt(0.5);
 257             Device->ChannelMatrix[BACK_RIGHT][FRONT_CENTER]  = aluSqrt(0.5);
 258             Device->ChannelMatrix[BACK_CENTER][FRONT_CENTER] = 1.0f;
 259             Device->NumChan = 1;
 260             Speaker2Chan[0] = FRONT_CENTER;
 261             SpeakerAngle[0] = 0.0f * M_PI/180.0f;
 262             break;
 263
 264         case AL_FORMAT_STEREO8:
 265         case AL_FORMAT_STEREO16:
 266         case AL_FORMAT_STEREO_FLOAT32:
 267             Device->DuplicateStereo = AL_FALSE;
 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->DuplicateStereo = GetConfigValueBool(NULL, "stereodup", 0);
 288             Device->ChannelMatrix[FRONT_CENTER][FRONT_LEFT]  = aluSqrt(0.5);
 289             Device->ChannelMatrix[FRONT_CENTER][FRONT_RIGHT] = aluSqrt(0.5);
 290             Device->ChannelMatrix[SIDE_LEFT][FRONT_LEFT]     = aluSqrt(0.5);
 291             Device->ChannelMatrix[SIDE_LEFT][BACK_LEFT]      = aluSqrt(0.5);
 292             Device->ChannelMatrix[SIDE_RIGHT][FRONT_RIGHT]   = aluSqrt(0.5);
 293             Device->ChannelMatrix[SIDE_RIGHT][BACK_RIGHT]    = aluSqrt(0.5);
 294             Device->ChannelMatrix[BACK_CENTER][BACK_LEFT]    = aluSqrt(0.5);
 295             Device->ChannelMatrix[BACK_CENTER][BACK_RIGHT]   = aluSqrt(0.5);
 296             Device->NumChan = 4;
 297             Speaker2Chan[0] = BACK_LEFT;
 298             Speaker2Chan[1] = FRONT_LEFT;
 299             Speaker2Chan[2] = FRONT_RIGHT;
 300             Speaker2Chan[3] = BACK_RIGHT;
 301             SpeakerAngle[0] = -135.0f * M_PI/180.0f;
 302             SpeakerAngle[1] =  -45.0f * M_PI/180.0f;
 303             SpeakerAngle[2] =   45.0f * M_PI/180.0f;
 304             SpeakerAngle[3] =  135.0f * M_PI/180.0f;
 305             SetSpeakerArrangement("layout", SpeakerAngle, Speaker2Chan, Device->NumChan);
 306             break;
 307
 308         case AL_FORMAT_51CHN8:
 309         case AL_FORMAT_51CHN16:
 310         case AL_FORMAT_51CHN32:
 311             Device->DuplicateStereo = GetConfigValueBool(NULL, "stereodup", 0);
 312             Device->ChannelMatrix[SIDE_LEFT][FRONT_LEFT]   = aluSqrt(0.5);
 313             Device->ChannelMatrix[SIDE_LEFT][BACK_LEFT]    = aluSqrt(0.5);
 314             Device->ChannelMatrix[SIDE_RIGHT][FRONT_RIGHT] = aluSqrt(0.5);
 315             Device->ChannelMatrix[SIDE_RIGHT][BACK_RIGHT]  = aluSqrt(0.5);
 316             Device->ChannelMatrix[BACK_CENTER][BACK_LEFT]  = aluSqrt(0.5);
 317             Device->ChannelMatrix[BACK_CENTER][BACK_RIGHT] = aluSqrt(0.5);
 318             Device->NumChan = 5;
 319             Speaker2Chan[0] = BACK_LEFT;
 320             Speaker2Chan[1] = FRONT_LEFT;
 321             Speaker2Chan[2] = FRONT_CENTER;
 322             Speaker2Chan[3] = FRONT_RIGHT;
 323             Speaker2Chan[4] = BACK_RIGHT;
 324             SpeakerAngle[0] = -110.0f * M_PI/180.0f;
 325             SpeakerAngle[1] =  -30.0f * M_PI/180.0f;
 326             SpeakerAngle[2] =    0.0f * M_PI/180.0f;
 327             SpeakerAngle[3] =   30.0f * M_PI/180.0f;
 328             SpeakerAngle[4] =  110.0f * M_PI/180.0f;
 329             SetSpeakerArrangement("layout", SpeakerAngle, Speaker2Chan, Device->NumChan);
 330             break;
 331
 332         case AL_FORMAT_61CHN8:
 333         case AL_FORMAT_61CHN16:
 334         case AL_FORMAT_61CHN32:
 335             Device->DuplicateStereo = GetConfigValueBool(NULL, "stereodup", 0);
 336             Device->ChannelMatrix[BACK_LEFT][BACK_CENTER]  = aluSqrt(0.5);
 337             Device->ChannelMatrix[BACK_LEFT][SIDE_LEFT]    = aluSqrt(0.5);
 338             Device->ChannelMatrix[BACK_RIGHT][BACK_CENTER] = aluSqrt(0.5);
 339             Device->ChannelMatrix[BACK_RIGHT][SIDE_RIGHT]  = aluSqrt(0.5);
 340             Device->NumChan = 6;
 341             Speaker2Chan[0] = SIDE_LEFT;
 342             Speaker2Chan[1] = FRONT_LEFT;
 343             Speaker2Chan[2] = FRONT_CENTER;
 344             Speaker2Chan[3] = FRONT_RIGHT;
 345             Speaker2Chan[4] = SIDE_RIGHT;
 346             Speaker2Chan[5] = BACK_CENTER;
 347             SpeakerAngle[0] = -90.0f * M_PI/180.0f;
 348             SpeakerAngle[1] = -30.0f * M_PI/180.0f;
 349             SpeakerAngle[2] =   0.0f * M_PI/180.0f;
 350             SpeakerAngle[3] =  30.0f * M_PI/180.0f;
 351             SpeakerAngle[4] =  90.0f * M_PI/180.0f;
 352             SpeakerAngle[5] = 180.0f * M_PI/180.0f;
 353             SetSpeakerArrangement("layout", SpeakerAngle, Speaker2Chan, Device->NumChan);
 354             break;
 355
 356         case AL_FORMAT_71CHN8:
 357         case AL_FORMAT_71CHN16:
 358         case AL_FORMAT_71CHN32:
 359             Device->DuplicateStereo = GetConfigValueBool(NULL, "stereodup", 0);
 360             Device->ChannelMatrix[BACK_CENTER][BACK_LEFT]  = aluSqrt(0.5);
 361             Device->ChannelMatrix[BACK_CENTER][BACK_RIGHT] = aluSqrt(0.5);
 362             Device->NumChan = 7;
 363             Speaker2Chan[0] = BACK_LEFT;
 364             Speaker2Chan[1] = SIDE_LEFT;
 365             Speaker2Chan[2] = FRONT_LEFT;
 366             Speaker2Chan[3] = FRONT_CENTER;
 367             Speaker2Chan[4] = FRONT_RIGHT;
 368             Speaker2Chan[5] = SIDE_RIGHT;
 369             Speaker2Chan[6] = BACK_RIGHT;
 370             SpeakerAngle[0] = -150.0f * M_PI/180.0f;
 371             SpeakerAngle[1] =  -90.0f * M_PI/180.0f;
 372             SpeakerAngle[2] =  -30.0f * M_PI/180.0f;
 373             SpeakerAngle[3] =    0.0f * M_PI/180.0f;
 374             SpeakerAngle[4] =   30.0f * M_PI/180.0f;
 375             SpeakerAngle[5] =   90.0f * M_PI/180.0f;
 376             SpeakerAngle[6] =  150.0f * M_PI/180.0f;
 377             SetSpeakerArrangement("layout", SpeakerAngle, Speaker2Chan, Device->NumChan);
 378             break;
 379
 380         default:
 381             assert(0);
 382     }
 383
 384     if(GetConfigValueBool(NULL, "scalemix", 0))
 385     {
 386         ALfloat maxout = 1.0f;
 387         for(s = 0;s < OUTPUTCHANNELS;s++)
 388         {
 389             ALfloat out = 0.0f;
 390             for(s2 = 0;s2 < OUTPUTCHANNELS;s2++)
 391                 out += Device->ChannelMatrix[s2][s];
 392             maxout = __max(maxout, out);
 393         }
 394
 395         maxout = 1.0f/maxout;
 396         for(s = 0;s < OUTPUTCHANNELS;s++)
 397         {
 398             for(s2 = 0;s2 < OUTPUTCHANNELS;s2++)
 399                 Device->ChannelMatrix[s2][s] *= maxout;
 400         }
 401     }
 402
 403     for(pos = 0; pos < LUT_NUM; pos++)
 404     {
 405         /* clear all values */
 406         offset = OUTPUTCHANNELS * pos;
 407         for(s = 0; s < OUTPUTCHANNELS; s++)
 408             Device->PanningLUT[offset+s] = 0.0f;
 409
 410         if(Device->NumChan == 1)
 411         {
 412             Device->PanningLUT[offset + Speaker2Chan[0]] = 1.0f;
 413             continue;
 414         }
 415
 416         /* source angle */
 417         Theta = aluLUTpos2Angle(pos);
 418
 419         /* set panning values */
 420         for(s = 0; s < Device->NumChan - 1; s++)
 421         {
 422             if(Theta >= SpeakerAngle[s] && Theta < SpeakerAngle[s+1])
 423             {
 424                 /* source between speaker s and speaker s+1 */
 425                 Alpha = M_PI_2 * (Theta-SpeakerAngle[s]) /
 426                                  (SpeakerAngle[s+1]-SpeakerAngle[s]);
 427                 Device->PanningLUT[offset + Speaker2Chan[s]]   = cos(Alpha);
 428                 Device->PanningLUT[offset + Speaker2Chan[s+1]] = sin(Alpha);
 429                 break;
 430             }
 431         }
 432         if(s == Device->NumChan - 1)
 433         {
 434             /* source between last and first speaker */
 435             if(Theta < SpeakerAngle[0])
 436                 Theta += 2.0f * M_PI;
 437             Alpha = M_PI_2 * (Theta-SpeakerAngle[s]) /
 438                              (2.0f * M_PI + SpeakerAngle[0]-SpeakerAngle[s]);
 439             Device->PanningLUT[offset + Speaker2Chan[s]] = cos(Alpha);
 440             Device->PanningLUT[offset + Speaker2Chan[0]] = sin(Alpha);
 441         }
 442     }
 443 }
 444
 445 static ALvoid CalcNonAttnSourceParams(const ALCcontext *ALContext, ALsource *ALSource)
 446 {
 447     ALfloat SourceVolume,ListenerGain,MinVolume,MaxVolume;
 448     ALfloat DryGain, DryGainHF;
 449     ALfloat WetGain[MAX_SENDS];
 450     ALfloat WetGainHF[MAX_SENDS];
 451     ALint NumSends, Frequency;
 452     ALfloat cw;
 453     ALint i;
 454
 455     //Get context properties
 456     NumSends  = ALContext->Device->NumAuxSends;
 457     Frequency = ALContext->Device->Frequency;
 458
 459     //Get listener properties
 460     ListenerGain = ALContext->Listener.Gain;
 461
 462     //Get source properties
 463     SourceVolume = ALSource->flGain;
 464     MinVolume    = ALSource->flMinGain;
 465     MaxVolume    = ALSource->flMaxGain;
 466
 467     //1. Multi-channel buffers always play "normal"
 468     ALSource->Params.Pitch = ALSource->flPitch;
 469
 470     DryGain = SourceVolume;
 471     DryGain = __min(DryGain,MaxVolume);
 472     DryGain = __max(DryGain,MinVolume);
 473     DryGainHF = 1.0f;
 474
 475     switch(ALSource->DirectFilter.type)
 476     {
 477         case AL_FILTER_LOWPASS:
 478             DryGain *= ALSource->DirectFilter.Gain;
 479             DryGainHF *= ALSource->DirectFilter.GainHF;
 480             break;
 481     }
 482
 483     for(i = 0;i < OUTPUTCHANNELS;i++)
 484         ALSource->Params.DryGains[i] = DryGain * ListenerGain;
 485
 486     for(i = 0;i < NumSends;i++)
 487     {
 488         WetGain[i] = SourceVolume;
 489         WetGain[i] = __min(WetGain[i],MaxVolume);
 490         WetGain[i] = __max(WetGain[i],MinVolume);
 491         WetGainHF[i] = 1.0f;
 492
 493         switch(ALSource->Send[i].WetFilter.type)
 494         {
 495             case AL_FILTER_LOWPASS:
 496                 WetGain[i] *= ALSource->Send[i].WetFilter.Gain;
 497                 WetGainHF[i] *= ALSource->Send[i].WetFilter.GainHF;
 498                 break;
 499         }
 500
 501         ALSource->Params.WetGains[i] = WetGain[i] * ListenerGain;
 502     }
 503     for(i = NumSends;i < MAX_SENDS;i++)
 504     {
 505         ALSource->Params.WetGains[i] = 0.0f;
 506         WetGainHF[i] = 1.0f;
 507     }
 508
 509     /* Update filter coefficients. Calculations based on the I3DL2
 510      * spec. */
 511     cw = cos(2.0*M_PI * LOWPASSFREQCUTOFF / Frequency);
 512
 513     /* We use two chained one-pole filters, so we need to take the
 514      * square root of the squared gain, which is the same as the base
 515      * gain. */
 516     ALSource->Params.iirFilter.coeff = lpCoeffCalc(DryGainHF, cw);
 517
 518     for(i = 0;i < NumSends;i++)
 519     {
 520         /* We use a one-pole filter, so we need to take the squared gain */
 521         ALfloat a = lpCoeffCalc(WetGainHF[i]*WetGainHF[i], cw);
 522         ALSource->Params.Send[i].iirFilter.coeff = a;
 523     }
 524 }
 525
 526 static ALvoid CalcSourceParams(const ALCcontext *ALContext, ALsource *ALSource)
 527 {
 528     const ALCdevice *Device = ALContext->Device;
 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     const ALfloat *SpeakerGain;
 545     ALfloat length;
 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        = Device->NumAuxSends;
 559     Frequency       = 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 - (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 = &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(Device->NumChan) * (1.0-DirGain);
 873     for(s = 0;s < OUTPUTCHANNELS;s++)
 874         ALSource->Params.DryGains[s] = 0.0f;
 875     for(s = 0;s < (ALsizei)Device->NumChan;s++)
 876     {
 877         Channel chan = Device->Speaker2Chan[s];
 878         ALfloat gain = SpeakerGain[chan]*DirGain + AmbientGain;
 879         ALSource->Params.DryGains[chan] = DryMix * gain;
 880     }
 881
 882     /* Update filter coefficients. */
 883     cw = cos(2.0*M_PI * LOWPASSFREQCUTOFF / Frequency);
 884
 885     /* Spatialized sources use four chained one-pole filters, so we need to
 886      * take the fourth root of the squared gain, which is the same as the
 887      * square root of the base gain. */
 888     ALSource->Params.iirFilter.coeff = lpCoeffCalc(aluSqrt(DryGainHF), cw);
 889
 890     for(i = 0;i < NumSends;i++)
 891     {
 892         /* The wet path uses two chained one-pole filters, so take the
 893          * base gain (square root of the squared gain) */
 894         ALSource->Params.Send[i].iirFilter.coeff = lpCoeffCalc(WetGainHF[i], cw);
 895     }
 896 }
 897
 898 static __inline ALfloat point(ALfloat val1, ALfloat val2, ALint frac)
 899 {
 900     return val1;
 901     (void)val2;
 902     (void)frac;
 903 }
 904 static __inline ALfloat lerp(ALfloat val1, ALfloat val2, ALint frac)
 905 {
 906     return val1 + ((val2-val1)*(frac * (1.0f/(1<<FRACTIONBITS))));
 907 }
 908 static __inline ALfloat cos_lerp(ALfloat val1, ALfloat val2, ALint frac)
 909 {
 910     ALfloat mult = (1.0f-cos(frac * (1.0f/(1<<FRACTIONBITS)) * M_PI)) * 0.5f;
 911     return val1 + ((val2-val1)*mult);
 912 }
 913
 914 static void MixSomeSources(ALCcontext *ALContext, float (*DryBuffer)[OUTPUTCHANNELS], ALuint SamplesToDo)
 915 {
 916     static float DummyBuffer[BUFFERSIZE];
 917     ALfloat *WetBuffer[MAX_SENDS];
 918     ALfloat DrySend[OUTPUTCHANNELS];
 919     ALfloat dryGainStep[OUTPUTCHANNELS];
 920     ALfloat wetGainStep[MAX_SENDS];
 921     ALuint i, j, k, out;
 922     ALsource *ALSource;
 923     ALfloat value, outsamp;
 924     ALbufferlistitem *BufferListItem;
 925     ALint64 DataSize64,DataPos64;
 926     FILTER *DryFilter, *WetFilter[MAX_SENDS];
 927     ALfloat WetSend[MAX_SENDS];
 928     ALuint rampLength;
 929     ALboolean DuplicateStereo;
 930     ALuint DeviceFreq;
 931     ALint increment;
 932     ALuint DataPosInt, DataPosFrac;
 933     ALuint Channels, Bytes;
 934     ALuint Frequency;
 935     resampler_t Resampler;
 936     ALuint BuffersPlayed;
 937     ALfloat Pitch;
 938     ALenum State;
 939     ALsizei pos;
 940
 941     if(ALContext->SourceMap.size <= 0)
 942         return;
 943
 944     DuplicateStereo = ALContext->Device->DuplicateStereo;
 945     DeviceFreq = ALContext->Device->Frequency;
 946
 947     rampLength = DeviceFreq * MIN_RAMP_LENGTH / 1000;
 948     rampLength = max(rampLength, SamplesToDo);
 949
 950     pos = ALContext->SourceMap.size;
 951 next_source:
 952     do {
 953         if(pos-- <= 0)
 954             return;
 955         ALSource = ALContext->SourceMap.array[pos].value;
 956     } while(ALSource->state != AL_PLAYING);
 957     j = 0;
 958
 959     /* Find buffer format */
 960     Frequency = 0;
 961     Channels = 0;
 962     Bytes = 0;
 963     BufferListItem = ALSource->queue;
 964     while(BufferListItem != NULL)
 965     {
 966         ALbuffer *ALBuffer;
 967         if((ALBuffer=BufferListItem->buffer) != NULL)
 968         {
 969             Channels  = aluChannelsFromFormat(ALBuffer->format);
 970             Bytes     = aluBytesFromFormat(ALBuffer->format);
 971             Frequency = ALBuffer->frequency;
 972             break;
 973         }
 974         BufferListItem = BufferListItem->next;
 975     }
 976
 977     if(ALSource->NeedsUpdate)
 978     {
 979         //Only apply 3D calculations for mono buffers
 980         if(Channels == 1)
 981             CalcSourceParams(ALContext, ALSource);
 982         else
 983             CalcNonAttnSourceParams(ALContext, ALSource);
 984         ALSource->NeedsUpdate = AL_FALSE;
 985     }
 986
 987     /* Get source info */
 988     Resampler     = ALSource->Resampler;
 989     State         = ALSource->state;
 990     BuffersPlayed = ALSource->BuffersPlayed;
 991     DataPosInt    = ALSource->position;
 992     DataPosFrac   = ALSource->position_fraction;
 993
 994     /* Compute 18.14 fixed point step */
 995     Pitch = (ALSource->Params.Pitch*Frequency) / DeviceFreq;
 996     if(Pitch > (float)MAX_PITCH)  Pitch = (float)MAX_PITCH;
 997     increment = (ALint)(Pitch*(ALfloat)(1L<<FRACTIONBITS));
 998     if(increment <= 0)  increment = (1<<FRACTIONBITS);
 999
1000     if(ALSource->FirstStart)
1001     {
1002         for(i = 0;i < OUTPUTCHANNELS;i++)
1003             DrySend[i] = ALSource->Params.DryGains[i];
1004         for(i = 0;i < MAX_SENDS;i++)
1005             WetSend[i] = ALSource->Params.WetGains[i];
1006     }
1007     else
1008     {
1009         for(i = 0;i < OUTPUTCHANNELS;i++)
1010             DrySend[i] = ALSource->DryGains[i];
1011         for(i = 0;i < MAX_SENDS;i++)
1012             WetSend[i] = ALSource->WetGains[i];
1013     }
1014
1015     DryFilter = &ALSource->Params.iirFilter;
1016     for(i = 0;i < MAX_SENDS;i++)
1017     {
1018         WetFilter[i] = &ALSource->Params.Send[i].iirFilter;
1019         WetBuffer[i] = (ALSource->Send[i].Slot ?
1020                         ALSource->Send[i].Slot->WetBuffer :
1021                         DummyBuffer);
1022     }
1023
1024     /* Get current buffer queue item */
1025     BufferListItem = ALSource->queue;
1026     for(i = 0;i < BuffersPlayed && BufferListItem;i++)
1027         BufferListItem = BufferListItem->next;
1028
1029     while(State == AL_PLAYING && j < SamplesToDo)
1030     {
1031         ALuint DataSize = 0;
1032         ALbuffer *ALBuffer;
1033         ALfloat *Data;
1034         ALuint BufferSize;
1035
1036         /* Get buffer info */
1037         if((ALBuffer=BufferListItem->buffer) != NULL)
1038         {
1039             Data      = ALBuffer->data;
1040             DataSize  = ALBuffer->size;
1041             DataSize /= Channels * Bytes;
1042         }
1043         if(DataPosInt >= DataSize)
1044             goto skipmix;
1045
1046         if(BufferListItem->next)
1047         {
1048             ALbuffer *NextBuf = BufferListItem->next->buffer;
1049             if(NextBuf && NextBuf->size)
1050             {
1051                 ALint ulExtraSamples = BUFFER_PADDING*Channels*Bytes;
1052                 ulExtraSamples = min(NextBuf->size, ulExtraSamples);
1053                 memcpy(&Data[DataSize*Channels], NextBuf->data, ulExtraSamples);
1054             }
1055         }
1056         else if(ALSource->bLooping)
1057         {
1058             ALbuffer *NextBuf = ALSource->queue->buffer;
1059             if(NextBuf && NextBuf->size)
1060             {
1061                 ALint ulExtraSamples = BUFFER_PADDING*Channels*Bytes;
1062                 ulExtraSamples = min(NextBuf->size, ulExtraSamples);
1063                 memcpy(&Data[DataSize*Channels], NextBuf->data, ulExtraSamples);
1064             }
1065         }
1066         else
1067             memset(&Data[DataSize*Channels], 0, (BUFFER_PADDING*Channels*Bytes));
1068
1069         /* Compute the gain steps for each output channel */
1070         for(i = 0;i < OUTPUTCHANNELS;i++)
1071             dryGainStep[i] = (ALSource->Params.DryGains[i]-DrySend[i]) /
1072                              rampLength;
1073         for(i = 0;i < MAX_SENDS;i++)
1074             wetGainStep[i] = (ALSource->Params.WetGains[i]-WetSend[i]) /
1075                              rampLength;
1076
1077         /* Figure out how many samples we can mix. */
1078         DataSize64 = DataSize;
1079         DataSize64 <<= FRACTIONBITS;
1080         DataPos64 = DataPosInt;
1081         DataPos64 <<= FRACTIONBITS;
1082         DataPos64 += DataPosFrac;
1083         BufferSize = (ALuint)((DataSize64-DataPos64+(increment-1)) / increment);
1084
1085         BufferSize = min(BufferSize, (SamplesToDo-j));
1086
1087         /* Actual sample mixing loop */
1088         k = 0;
1089         Data += DataPosInt*Channels;
1090
1091         if(Channels == 1) /* Mono */
1092         {
1093 #define DO_MIX(resampler) do { \
1094     while(BufferSize--) \
1095     { \
1096         for(i = 0;i < OUTPUTCHANNELS;i++) \
1097             DrySend[i] += dryGainStep[i]; \
1098         for(i = 0;i < MAX_SENDS;i++) \
1099             WetSend[i] += wetGainStep[i]; \
1100  \
1101         /* First order interpolator */ \
1102         value = (resampler)(Data[k], Data[k+1], DataPosFrac); \
1103  \
1104         /* Direct path final mix buffer and panning */ \
1105         outsamp = lpFilter4P(DryFilter, 0, value); \
1106         DryBuffer[j][FRONT_LEFT]   += outsamp*DrySend[FRONT_LEFT]; \
1107         DryBuffer[j][FRONT_RIGHT]  += outsamp*DrySend[FRONT_RIGHT]; \
1108         DryBuffer[j][SIDE_LEFT]    += outsamp*DrySend[SIDE_LEFT]; \
1109         DryBuffer[j][SIDE_RIGHT]   += outsamp*DrySend[SIDE_RIGHT]; \
1110         DryBuffer[j][BACK_LEFT]    += outsamp*DrySend[BACK_LEFT]; \
1111         DryBuffer[j][BACK_RIGHT]   += outsamp*DrySend[BACK_RIGHT]; \
1112         DryBuffer[j][FRONT_CENTER] += outsamp*DrySend[FRONT_CENTER]; \
1113         DryBuffer[j][BACK_CENTER]  += outsamp*DrySend[BACK_CENTER]; \
1114  \
1115         /* Room path final mix buffer and panning */ \
1116         for(i = 0;i < MAX_SENDS;i++) \
1117         { \
1118             outsamp = lpFilter2P(WetFilter[i], 0, value); \
1119             WetBuffer[i][j] += outsamp*WetSend[i]; \
1120         } \
1121  \
1122         DataPosFrac += increment; \
1123         k += DataPosFrac>>FRACTIONBITS; \
1124         DataPosFrac &= FRACTIONMASK; \
1125         j++; \
1126     } \
1127 } while(0)
1128
1129             switch(Resampler)
1130             {
1131                 case POINT_RESAMPLER:
1132                 DO_MIX(point); break;
1133                 case LINEAR_RESAMPLER:
1134                 DO_MIX(lerp); break;
1135                 case COSINE_RESAMPLER:
1136                 DO_MIX(cos_lerp); break;
1137                 case RESAMPLER_MIN:
1138                 case RESAMPLER_MAX:
1139                 break;
1140             }
1141 #undef DO_MIX
1142         }
1143         else if(Channels == 2 && DuplicateStereo) /* Stereo */
1144         {
1145             const int chans[] = {
1146                 FRONT_LEFT, FRONT_RIGHT
1147             };
1148             const int chans2[] = {
1149                 BACK_LEFT, SIDE_LEFT, BACK_RIGHT, SIDE_RIGHT
1150             };
1151             const ALfloat scaler = 1.0f/Channels;
1152             const ALfloat dupscaler = aluSqrt(1.0f/3.0f);
1153
1154 #define DO_MIX(resampler) do {                                                \
1155     while(BufferSize--)                                                       \
1156     {                                                                         \
1157         for(i = 0;i < OUTPUTCHANNELS;i++)                                     \
1158             DrySend[i] += dryGainStep[i];                                     \
1159         for(i = 0;i < MAX_SENDS;i++)                                          \
1160             WetSend[i] += wetGainStep[i];                                     \
1161                                                                               \
1162         for(i = 0;i < Channels;i++)                                           \
1163         {                                                                     \
1164             value = (resampler)(Data[k*Channels + i],Data[(k+1)*Channels + i],\
1165                                 DataPosFrac);                                 \
1166             outsamp = lpFilter2P(DryFilter, chans[i]*2, value) * dupscaler;   \
1167             DryBuffer[j][chans[i]] += outsamp*DrySend[chans[i]];              \
1168             DryBuffer[j][chans2[i*2+0]] += outsamp*DrySend[chans2[i*2+0]];    \
1169             DryBuffer[j][chans2[i*2+1]] += outsamp*DrySend[chans2[i*2+1]];    \
1170             for(out = 0;out < MAX_SENDS;out++)                                \
1171             {                                                                 \
1172                 outsamp = lpFilter1P(WetFilter[out], chans[i], value);        \
1173                 WetBuffer[out][j] += outsamp*WetSend[out]*scaler;             \
1174             }                                                                 \
1175         }                                                                     \
1176                                                                               \
1177         DataPosFrac += increment;                                             \
1178         k += DataPosFrac>>FRACTIONBITS;                                       \
1179         DataPosFrac &= FRACTIONMASK;                                          \
1180         j++;                                                                  \
1181     }                                                                         \
1182 } while(0)
1183
1184             switch(Resampler)
1185             {
1186                 case POINT_RESAMPLER:
1187                 DO_MIX(point); break;
1188                 case LINEAR_RESAMPLER:
1189                 DO_MIX(lerp); break;
1190                 case COSINE_RESAMPLER:
1191                 DO_MIX(cos_lerp); break;
1192                 case RESAMPLER_MIN:
1193                 case RESAMPLER_MAX:
1194                 break;
1195             }
1196 #undef DO_MIX
1197         }
1198         else if(Channels == 2) /* Stereo */
1199         {
1200             const int chans[] = {
1201                 FRONT_LEFT, FRONT_RIGHT
1202             };
1203             const ALfloat scaler = 1.0f/Channels;
1204
1205 #define DO_MIX(resampler) do {                                                \
1206     while(BufferSize--)                                                       \
1207     {                                                                         \
1208         for(i = 0;i < OUTPUTCHANNELS;i++)                                     \
1209             DrySend[i] += dryGainStep[i];                                     \
1210         for(i = 0;i < MAX_SENDS;i++)                                          \
1211             WetSend[i] += wetGainStep[i];                                     \
1212                                                                               \
1213         for(i = 0;i < Channels;i++)                                           \
1214         {                                                                     \
1215             value = (resampler)(Data[k*Channels + i],Data[(k+1)*Channels + i],\
1216                                 DataPosFrac);                                 \
1217             outsamp = lpFilter2P(DryFilter, chans[i]*2, value);               \
1218             DryBuffer[j][chans[i]] += outsamp*DrySend[chans[i]];              \
1219             for(out = 0;out < MAX_SENDS;out++)                                \
1220             {                                                                 \
1221                 outsamp = lpFilter1P(WetFilter[out], chans[i], value);        \
1222                 WetBuffer[out][j] += outsamp*WetSend[out]*scaler;             \
1223             }                                                                 \
1224         }                                                                     \
1225                                                                               \
1226         DataPosFrac += increment;                                             \
1227         k += DataPosFrac>>FRACTIONBITS;                                       \
1228         DataPosFrac &= FRACTIONMASK;                                          \
1229         j++;                                                                  \
1230     }                                                                         \
1231 } while(0)
1232
1233             switch(Resampler)
1234             {
1235                 case POINT_RESAMPLER:
1236                 DO_MIX(point); break;
1237                 case LINEAR_RESAMPLER:
1238                 DO_MIX(lerp); break;
1239                 case COSINE_RESAMPLER:
1240                 DO_MIX(cos_lerp); break;
1241                 case RESAMPLER_MIN:
1242                 case RESAMPLER_MAX:
1243                 break;
1244             }
1245         }
1246         else if(Channels == 4) /* Quad */
1247         {
1248             const int chans[] = {
1249                 FRONT_LEFT, FRONT_RIGHT,
1250                 BACK_LEFT,  BACK_RIGHT
1251             };
1252             const ALfloat scaler = 1.0f/Channels;
1253
1254             switch(Resampler)
1255             {
1256                 case POINT_RESAMPLER:
1257                 DO_MIX(point); break;
1258                 case LINEAR_RESAMPLER:
1259                 DO_MIX(lerp); break;
1260                 case COSINE_RESAMPLER:
1261                 DO_MIX(cos_lerp); break;
1262                 case RESAMPLER_MIN:
1263                 case RESAMPLER_MAX:
1264                 break;
1265             }
1266         }
1267         else if(Channels == 6) /* 5.1 */
1268         {
1269             const int chans[] = {
1270                 FRONT_LEFT,   FRONT_RIGHT,
1271                 FRONT_CENTER, LFE,
1272                 BACK_LEFT,    BACK_RIGHT
1273             };
1274             const ALfloat scaler = 1.0f/Channels;
1275
1276             switch(Resampler)
1277             {
1278                 case POINT_RESAMPLER:
1279                 DO_MIX(point); break;
1280                 case LINEAR_RESAMPLER:
1281                 DO_MIX(lerp); break;
1282                 case COSINE_RESAMPLER:
1283                 DO_MIX(cos_lerp); break;
1284                 case RESAMPLER_MIN:
1285                 case RESAMPLER_MAX:
1286                 break;
1287             }
1288         }
1289         else if(Channels == 7) /* 6.1 */
1290         {
1291             const int chans[] = {
1292                 FRONT_LEFT,   FRONT_RIGHT,
1293                 FRONT_CENTER, LFE,
1294                 BACK_CENTER,
1295                 SIDE_LEFT,    SIDE_RIGHT
1296             };
1297             const ALfloat scaler = 1.0f/Channels;
1298
1299             switch(Resampler)
1300             {
1301                 case POINT_RESAMPLER:
1302                 DO_MIX(point); break;
1303                 case LINEAR_RESAMPLER:
1304                 DO_MIX(lerp); break;
1305                 case COSINE_RESAMPLER:
1306                 DO_MIX(cos_lerp); break;
1307                 case RESAMPLER_MIN:
1308                 case RESAMPLER_MAX:
1309                 break;
1310             }
1311         }
1312         else if(Channels == 8) /* 7.1 */
1313         {
1314             const int chans[] = {
1315                 FRONT_LEFT,   FRONT_RIGHT,
1316                 FRONT_CENTER, LFE,
1317                 BACK_LEFT,    BACK_RIGHT,
1318                 SIDE_LEFT,    SIDE_RIGHT
1319             };
1320             const ALfloat scaler = 1.0f/Channels;
1321
1322             switch(Resampler)
1323             {
1324                 case POINT_RESAMPLER:
1325                 DO_MIX(point); break;
1326                 case LINEAR_RESAMPLER:
1327                 DO_MIX(lerp); break;
1328                 case COSINE_RESAMPLER:
1329                 DO_MIX(cos_lerp); break;
1330                 case RESAMPLER_MIN:
1331                 case RESAMPLER_MAX:
1332                 break;
1333             }
1334 #undef DO_MIX
1335         }
1336         else /* Unknown? */
1337         {
1338             for(i = 0;i < OUTPUTCHANNELS;i++)
1339                 DrySend[i] += dryGainStep[i]*BufferSize;
1340             for(i = 0;i < MAX_SENDS;i++)
1341                 WetSend[i] += wetGainStep[i]*BufferSize;
1342             while(BufferSize--)
1343             {
1344                 DataPosFrac += increment;
1345                 k += DataPosFrac>>FRACTIONBITS;
1346                 DataPosFrac &= FRACTIONMASK;
1347                 j++;
1348             }
1349         }
1350         DataPosInt += k;
1351
1352     skipmix:
1353         /* Handle looping sources */
1354         if(DataPosInt >= DataSize)
1355         {
1356             if(BuffersPlayed < (ALSource->BuffersInQueue-1))
1357             {
1358                 BufferListItem = BufferListItem->next;
1359                 BuffersPlayed++;
1360                 DataPosInt -= DataSize;
1361             }
1362             else if(ALSource->bLooping)
1363             {
1364                 BufferListItem = ALSource->queue;
1365                 BuffersPlayed = 0;
1366                 if(ALSource->BuffersInQueue == 1)
1367                     DataPosInt %= DataSize;
1368                 else
1369                     DataPosInt -= DataSize;
1370             }
1371             else
1372             {
1373                 State = AL_STOPPED;
1374                 BufferListItem = ALSource->queue;
1375                 BuffersPlayed = ALSource->BuffersInQueue;
1376                 DataPosInt = 0;
1377                 DataPosFrac = 0;
1378             }
1379         }
1380     }
1381
1382     /* Update source info */
1383     ALSource->state             = State;
1384     ALSource->BuffersPlayed     = BuffersPlayed;
1385     ALSource->position          = DataPosInt;
1386     ALSource->position_fraction = DataPosFrac;
1387     ALSource->Buffer            = BufferListItem->buffer;
1388
1389     for(i = 0;i < OUTPUTCHANNELS;i++)
1390         ALSource->DryGains[i] = DrySend[i];
1391     for(i = 0;i < MAX_SENDS;i++)
1392         ALSource->WetGains[i] = WetSend[i];
1393
1394     ALSource->FirstStart = AL_FALSE;
1395
1396     goto next_source;
1397 }
1398
1399 ALvoid aluMixData(ALCdevice *device, ALvoid *buffer, ALsizei size)
1400 {
1401     float (*DryBuffer)[OUTPUTCHANNELS];
1402     ALfloat (*Matrix)[OUTPUTCHANNELS];
1403     const ALuint *ChanMap;
1404     ALuint SamplesToDo;
1405     ALeffectslot *ALEffectSlot;
1406     ALCcontext *ALContext;
1407     ALfloat samp;
1408     int fpuState;
1409     ALuint i, j, c;
1410
1411 #if defined(HAVE_FESETROUND)
1412     fpuState = fegetround();
1413     fesetround(FE_TOWARDZERO);
1414 #elif defined(HAVE__CONTROLFP)
1415     fpuState = _controlfp(0, 0);
1416     _controlfp(_RC_CHOP, _MCW_RC);
1417 #else
1418     (void)fpuState;
1419 #endif
1420
1421     DryBuffer = device->DryBuffer;
1422     while(size > 0)
1423     {
1424         /* Setup variables */
1425         SamplesToDo = min(size, BUFFERSIZE);
1426
1427         /* Clear mixing buffer */
1428         memset(DryBuffer, 0, SamplesToDo*OUTPUTCHANNELS*sizeof(ALfloat));
1429
1430         SuspendContext(NULL);
1431         for(c = 0;c < device->NumContexts;c++)
1432         {
1433             ALContext = device->Contexts[c];
1434             SuspendContext(ALContext);
1435
1436             MixSomeSources(ALContext, DryBuffer, SamplesToDo);
1437
1438             /* effect slot processing */
1439             ALEffectSlot = ALContext->EffectSlotList;
1440             while(ALEffectSlot)
1441             {
1442                 if(ALEffectSlot->EffectState)
1443                     ALEffect_Process(ALEffectSlot->EffectState, ALEffectSlot, SamplesToDo, ALEffectSlot->WetBuffer, DryBuffer);
1444
1445                 for(i = 0;i < SamplesToDo;i++)
1446                     ALEffectSlot->WetBuffer[i] = 0.0f;
1447                 ALEffectSlot = ALEffectSlot->next;
1448             }
1449             ProcessContext(ALContext);
1450         }
1451         ProcessContext(NULL);
1452
1453         //Post processing loop
1454         ChanMap = device->DevChannels;
1455         Matrix = device->ChannelMatrix;
1456         switch(device->Format)
1457         {
1458 #define CHECK_WRITE_FORMAT(bits, type, func)                                  \
1459         case AL_FORMAT_MONO##bits:                                            \
1460             for(i = 0;i < SamplesToDo;i++)                                    \
1461             {                                                                 \
1462                 samp = 0.0f;                                                  \
1463                 for(c = 0;c < OUTPUTCHANNELS;c++)                             \
1464                     samp += DryBuffer[i][c] * Matrix[c][FRONT_CENTER];        \
1465                 ((type*)buffer)[ChanMap[FRONT_CENTER]] = (func)(samp);        \
1466                 buffer = ((type*)buffer) + 1;                                 \
1467             }                                                                 \
1468             break;                                                            \
1469         case AL_FORMAT_STEREO##bits:                                          \
1470             if(device->Bs2b)                                                  \
1471             {                                                                 \
1472                 for(i = 0;i < SamplesToDo;i++)                                \
1473                 {                                                             \
1474                     float samples[2] = { 0.0f, 0.0f };                        \
1475                     for(c = 0;c < OUTPUTCHANNELS;c++)                         \
1476                     {                                                         \
1477                         samples[0] += DryBuffer[i][c]*Matrix[c][FRONT_LEFT];  \
1478                         samples[1] += DryBuffer[i][c]*Matrix[c][FRONT_RIGHT]; \
1479                     }                                                         \
1480                     bs2b_cross_feed(device->Bs2b, samples);                   \
1481                     ((type*)buffer)[ChanMap[FRONT_LEFT]] = (func)(samples[0]);\
1482                     ((type*)buffer)[ChanMap[FRONT_RIGHT]]= (func)(samples[1]);\
1483                     buffer = ((type*)buffer) + 2;                             \
1484                 }                                                             \
1485             }                                                                 \
1486             else                                                              \
1487             {                                                                 \
1488                 for(i = 0;i < SamplesToDo;i++)                                \
1489                 {                                                             \
1490                     static const Channel chans[] = {                          \
1491                         FRONT_LEFT, FRONT_RIGHT                               \
1492                     };                                                        \
1493                     for(j = 0;j < 2;j++)                                      \
1494                     {                                                         \
1495                         samp = 0.0f;                                          \
1496                         for(c = 0;c < OUTPUTCHANNELS;c++)                     \
1497                             samp += DryBuffer[i][c] * Matrix[c][chans[j]];    \
1498                         ((type*)buffer)[ChanMap[chans[j]]] = (func)(samp);    \
1499                     }                                                         \
1500                     buffer = ((type*)buffer) + 2;                             \
1501                 }                                                             \
1502             }                                                                 \
1503             break;                                                            \
1504         case AL_FORMAT_QUAD##bits:                                            \
1505             for(i = 0;i < SamplesToDo;i++)                                    \
1506             {                                                                 \
1507                 static const Channel chans[] = {                              \
1508                     FRONT_LEFT, FRONT_RIGHT,                                  \
1509                     BACK_LEFT,  BACK_RIGHT,                                   \
1510                 };                                                            \
1511                 for(j = 0;j < 4;j++)                                          \
1512                 {                                                             \
1513                     samp = 0.0f;                                              \
1514                     for(c = 0;c < OUTPUTCHANNELS;c++)                         \
1515                         samp += DryBuffer[i][c] * Matrix[c][chans[j]];        \
1516                     ((type*)buffer)[ChanMap[chans[j]]] = (func)(samp);        \
1517                 }                                                             \
1518                 buffer = ((type*)buffer) + 4;                                 \
1519             }                                                                 \
1520             break;                                                            \
1521         case AL_FORMAT_51CHN##bits:                                           \
1522             for(i = 0;i < SamplesToDo;i++)                                    \
1523             {                                                                 \
1524                 static const Channel chans[] = {                              \
1525                     FRONT_LEFT, FRONT_RIGHT,                                  \
1526                     FRONT_CENTER, LFE,                                        \
1527                     BACK_LEFT,  BACK_RIGHT,                                   \
1528                 };                                                            \
1529                 for(j = 0;j < 6;j++)                                          \
1530                 {                                                             \
1531                     samp = 0.0f;                                              \
1532                     for(c = 0;c < OUTPUTCHANNELS;c++)                         \
1533                         samp += DryBuffer[i][c] * Matrix[c][chans[j]];        \
1534                     ((type*)buffer)[ChanMap[chans[j]]] = (func)(samp);        \
1535                 }                                                             \
1536                 buffer = ((type*)buffer) + 6;                                 \
1537             }                                                                 \
1538             break;                                                            \
1539         case AL_FORMAT_61CHN##bits:                                           \
1540             for(i = 0;i < SamplesToDo;i++)                                    \
1541             {                                                                 \
1542                 static const Channel chans[] = {                              \
1543                     FRONT_LEFT, FRONT_RIGHT,                                  \
1544                     FRONT_CENTER, LFE, BACK_CENTER,                           \
1545                     SIDE_LEFT,  SIDE_RIGHT,                                   \
1546                 };                                                            \
1547                 for(j = 0;j < 7;j++)                                          \
1548                 {                                                             \
1549                     samp = 0.0f;                                              \
1550                     for(c = 0;c < OUTPUTCHANNELS;c++)                         \
1551                         samp += DryBuffer[i][c] * Matrix[c][chans[j]];        \
1552                     ((type*)buffer)[ChanMap[chans[j]]] = (func)(samp);        \
1553                 }                                                             \
1554                 buffer = ((type*)buffer) + 7;                                 \
1555             }                                                                 \
1556             break;                                                            \
1557         case AL_FORMAT_71CHN##bits:                                           \
1558             for(i = 0;i < SamplesToDo;i++)                                    \
1559             {                                                                 \
1560                 static const Channel chans[] = {                              \
1561                     FRONT_LEFT, FRONT_RIGHT,                                  \
1562                     FRONT_CENTER, LFE,                                        \
1563                     BACK_LEFT,  BACK_RIGHT,                                   \
1564                     SIDE_LEFT,  SIDE_RIGHT                                    \
1565                 };                                                            \
1566                 for(j = 0;j < 8;j++)                                          \
1567                 {                                                             \
1568                     samp = 0.0f;                                              \
1569                     for(c = 0;c < OUTPUTCHANNELS;c++)                         \
1570                         samp += DryBuffer[i][c] * Matrix[c][chans[j]];        \
1571                     ((type*)buffer)[ChanMap[chans[j]]] = (func)(samp);        \
1572                 }                                                             \
1573                 buffer = ((type*)buffer) + 8;                                 \
1574             }                                                                 \
1575             break;
1576
1577 #define AL_FORMAT_MONO32 AL_FORMAT_MONO_FLOAT32
1578 #define AL_FORMAT_STEREO32 AL_FORMAT_STEREO_FLOAT32
1579             CHECK_WRITE_FORMAT(8,  ALubyte, aluF2UB)
1580             CHECK_WRITE_FORMAT(16, ALshort, aluF2S)
1581             CHECK_WRITE_FORMAT(32, ALfloat, aluF2F)
1582 #undef AL_FORMAT_STEREO32
1583 #undef AL_FORMAT_MONO32
1584 #undef CHECK_WRITE_FORMAT
1585
1586             default:
1587                 break;
1588         }
1589
1590         size -= SamplesToDo;
1591     }
1592
1593 #if defined(HAVE_FESETROUND)
1594     fesetround(fpuState);
1595 #elif defined(HAVE__CONTROLFP)
1596     _controlfp(fpuState, 0xfffff);
1597 #endif
1598 }
1599
1600 ALvoid aluHandleDisconnect(ALCdevice *device)
1601 {
1602     ALuint i;
1603
1604     SuspendContext(NULL);
1605     for(i = 0;i < device->NumContexts;i++)
1606     {
1607         ALCcontext *Context = device->Contexts[i];
1608         ALsource *source;
1609         ALsizei pos;
1610
1611         SuspendContext(Context);
1612
1613         for(pos = 0;pos < Context->SourceMap.size;pos++)
1614         {
1615             source = Context->SourceMap.array[pos].value;
1616             if(source->state == AL_PLAYING)
1617             {
1618                 source->state = AL_STOPPED;
1619                 source->BuffersPlayed = source->BuffersInQueue;
1620                 source->position = 0;
1621                 source->position_fraction = 0;
1622             }
1623         }
1624         ProcessContext(Context);
1625     }
1626
1627     device->Connected = ALC_FALSE;
1628     ProcessContext(NULL);
1629 }