Alc/mixer.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.,
  17  *  51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 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 "alListener.h"
  35 #include "alAuxEffectSlot.h"
  36 #include "alu.h"
  37
  38 #include "mixer_defs.h"
  39
  40
  41 static_assert((INT_MAX>>FRACTIONBITS)/MAX_PITCH > BUFFERSIZE,
  42               "MAX_PITCH and/or BUFFERSIZE are too large for FRACTIONBITS!");
  43
  44 extern inline void InitiatePositionArrays(ALsizei frac, ALint increment, ALsizei *restrict frac_arr, ALint *restrict pos_arr, ALsizei size);
  45
  46
  47 /* BSinc requires up to 11 extra samples before the current position, and 12 after. */
  48 static_assert(MAX_PRE_SAMPLES >= 11, "MAX_PRE_SAMPLES must be at least 11!");
  49 static_assert(MAX_POST_SAMPLES >= 12, "MAX_POST_SAMPLES must be at least 12!");
  50
  51
  52 enum Resampler ResamplerDefault = LinearResampler;
  53
  54 static MixerFunc MixSamples = Mix_C;
  55 static HrtfMixerFunc MixHrtfSamples = MixHrtf_C;
  56 HrtfMixerBlendFunc MixHrtfBlendSamples = MixHrtfBlend_C;
  57
  58 MixerFunc SelectMixer(void)
  59 {
  60 #ifdef HAVE_NEON
  61     if((CPUCapFlags&CPU_CAP_NEON))
  62         return Mix_Neon;
  63 #endif
  64 #ifdef HAVE_SSE
  65     if((CPUCapFlags&CPU_CAP_SSE))
  66         return Mix_SSE;
  67 #endif
  68     return Mix_C;
  69 }
  70
  71 RowMixerFunc SelectRowMixer(void)
  72 {
  73 #ifdef HAVE_NEON
  74     if((CPUCapFlags&CPU_CAP_NEON))
  75         return MixRow_Neon;
  76 #endif
  77 #ifdef HAVE_SSE
  78     if((CPUCapFlags&CPU_CAP_SSE))
  79         return MixRow_SSE;
  80 #endif
  81     return MixRow_C;
  82 }
  83
  84 static inline HrtfMixerFunc SelectHrtfMixer(void)
  85 {
  86 #ifdef HAVE_NEON
  87     if((CPUCapFlags&CPU_CAP_NEON))
  88         return MixHrtf_Neon;
  89 #endif
  90 #ifdef HAVE_SSE
  91     if((CPUCapFlags&CPU_CAP_SSE))
  92         return MixHrtf_SSE;
  93 #endif
  94     return MixHrtf_C;
  95 }
  96
  97 static inline HrtfMixerBlendFunc SelectHrtfBlendMixer(void)
  98 {
  99 #ifdef HAVE_NEON
 100     if((CPUCapFlags&CPU_CAP_NEON))
 101         return MixHrtfBlend_Neon;
 102 #endif
 103 #ifdef HAVE_SSE
 104     if((CPUCapFlags&CPU_CAP_SSE))
 105         return MixHrtfBlend_SSE;
 106 #endif
 107     return MixHrtfBlend_C;
 108 }
 109
 110 ResamplerFunc SelectResampler(enum Resampler resampler)
 111 {
 112     switch(resampler)
 113     {
 114         case PointResampler:
 115             return Resample_point_C;
 116         case LinearResampler:
 117 #ifdef HAVE_NEON
 118             if((CPUCapFlags&CPU_CAP_NEON))
 119                 return Resample_lerp_Neon;
 120 #endif
 121 #ifdef HAVE_SSE4_1
 122             if((CPUCapFlags&CPU_CAP_SSE4_1))
 123                 return Resample_lerp_SSE41;
 124 #endif
 125 #ifdef HAVE_SSE2
 126             if((CPUCapFlags&CPU_CAP_SSE2))
 127                 return Resample_lerp_SSE2;
 128 #endif
 129             return Resample_lerp_C;
 130         case FIR4Resampler:
 131 #ifdef HAVE_NEON
 132             if((CPUCapFlags&CPU_CAP_NEON))
 133                 return Resample_fir4_Neon;
 134 #endif
 135 #ifdef HAVE_SSE4_1
 136             if((CPUCapFlags&CPU_CAP_SSE4_1))
 137                 return Resample_fir4_SSE41;
 138 #endif
 139 #ifdef HAVE_SSE3
 140             if((CPUCapFlags&CPU_CAP_SSE3))
 141                 return Resample_fir4_SSE3;
 142 #endif
 143             return Resample_fir4_C;
 144         case BSinc12Resampler:
 145         case BSinc24Resampler:
 146 #ifdef HAVE_NEON
 147             if((CPUCapFlags&CPU_CAP_NEON))
 148                 return Resample_bsinc_Neon;
 149 #endif
 150 #ifdef HAVE_SSE
 151             if((CPUCapFlags&CPU_CAP_SSE))
 152                 return Resample_bsinc_SSE;
 153 #endif
 154             return Resample_bsinc_C;
 155     }
 156
 157     return Resample_point_C;
 158 }
 159
 160
 161 void aluInitMixer(void)
 162 {
 163     const char *str;
 164
 165     if(ConfigValueStr(NULL, NULL, "resampler", &str))
 166     {
 167         if(strcasecmp(str, "point") == 0 || strcasecmp(str, "none") == 0)
 168             ResamplerDefault = PointResampler;
 169         else if(strcasecmp(str, "linear") == 0)
 170             ResamplerDefault = LinearResampler;
 171         else if(strcasecmp(str, "sinc4") == 0)
 172             ResamplerDefault = FIR4Resampler;
 173         else if(strcasecmp(str, "bsinc12") == 0)
 174             ResamplerDefault = BSinc12Resampler;
 175         else if(strcasecmp(str, "bsinc24") == 0)
 176             ResamplerDefault = BSinc24Resampler;
 177         else if(strcasecmp(str, "bsinc") == 0)
 178         {
 179             WARN("Resampler option \"%s\" is deprecated, using bsinc12\n", str);
 180             ResamplerDefault = BSinc12Resampler;
 181         }
 182         else if(strcasecmp(str, "cubic") == 0 || strcasecmp(str, "sinc8") == 0)
 183         {
 184             WARN("Resampler option \"%s\" is deprecated, using sinc4\n", str);
 185             ResamplerDefault = FIR4Resampler;
 186         }
 187         else
 188         {
 189             char *end;
 190             long n = strtol(str, &end, 0);
 191             if(*end == '\0' && (n == PointResampler || n == LinearResampler || n == FIR4Resampler))
 192                 ResamplerDefault = n;
 193             else
 194                 WARN("Invalid resampler: %s\n", str);
 195         }
 196     }
 197
 198     MixHrtfBlendSamples = SelectHrtfBlendMixer();
 199     MixHrtfSamples = SelectHrtfMixer();
 200     MixSamples = SelectMixer();
 201 }
 202
 203
 204 static inline ALfloat Sample_ALbyte(ALbyte val)
 205 { return val * (1.0f/128.0f); }
 206
 207 static inline ALfloat Sample_ALshort(ALshort val)
 208 { return val * (1.0f/32768.0f); }
 209
 210 static inline ALfloat Sample_ALfloat(ALfloat val)
 211 { return val; }
 212
 213 #define DECL_TEMPLATE(T)                                                      \
 214 static inline void Load_##T(ALfloat *restrict dst, const T *restrict src,     \
 215                             ALint srcstep, ALsizei samples)                   \
 216 {                                                                             \
 217     ALsizei i;                                                                \
 218     for(i = 0;i < samples;i++)                                                \
 219         dst[i] = Sample_##T(src[i*srcstep]);                                  \
 220 }
 221
 222 DECL_TEMPLATE(ALbyte)
 223 DECL_TEMPLATE(ALshort)
 224 DECL_TEMPLATE(ALfloat)
 225
 226 #undef DECL_TEMPLATE
 227
 228 static void LoadSamples(ALfloat *restrict dst, const ALvoid *restrict src, ALint srcstep,
 229                         enum FmtType srctype, ALsizei samples)
 230 {
 231     switch(srctype)
 232     {
 233         case FmtByte:
 234             Load_ALbyte(dst, src, srcstep, samples);
 235             break;
 236         case FmtShort:
 237             Load_ALshort(dst, src, srcstep, samples);
 238             break;
 239         case FmtFloat:
 240             Load_ALfloat(dst, src, srcstep, samples);
 241             break;
 242     }
 243 }
 244
 245 static inline void SilenceSamples(ALfloat *dst, ALsizei samples)
 246 {
 247     ALsizei i;
 248     for(i = 0;i < samples;i++)
 249         dst[i] = 0.0f;
 250 }
 251
 252
 253 static const ALfloat *DoFilters(ALfilterState *lpfilter, ALfilterState *hpfilter,
 254                                 ALfloat *restrict dst, const ALfloat *restrict src,
 255                                 ALsizei numsamples, enum ActiveFilters type)
 256 {
 257     ALsizei i;
 258     switch(type)
 259     {
 260         case AF_None:
 261             ALfilterState_processPassthru(lpfilter, src, numsamples);
 262             ALfilterState_processPassthru(hpfilter, src, numsamples);
 263             break;
 264
 265         case AF_LowPass:
 266             ALfilterState_process(lpfilter, dst, src, numsamples);
 267             ALfilterState_processPassthru(hpfilter, dst, numsamples);
 268             return dst;
 269         case AF_HighPass:
 270             ALfilterState_processPassthru(lpfilter, src, numsamples);
 271             ALfilterState_process(hpfilter, dst, src, numsamples);
 272             return dst;
 273
 274         case AF_BandPass:
 275             for(i = 0;i < numsamples;)
 276             {
 277                 ALfloat temp[256];
 278                 ALsizei todo = mini(256, numsamples-i);
 279
 280                 ALfilterState_process(lpfilter, temp, src+i, todo);
 281                 ALfilterState_process(hpfilter, dst+i, temp, todo);
 282                 i += todo;
 283             }
 284             return dst;
 285     }
 286     return src;
 287 }
 288
 289
 290 ALboolean MixSource(ALvoice *voice, ALsource *Source, ALCdevice *Device, ALsizei SamplesToDo)
 291 {
 292     ALbufferlistitem *BufferListItem;
 293     ALbufferlistitem *BufferLoopItem;
 294     ALsizei NumChannels, SampleSize;
 295     ResamplerFunc Resample;
 296     ALsizei DataPosInt;
 297     ALsizei DataPosFrac;
 298     ALint64 DataSize64;
 299     ALint increment;
 300     ALsizei Counter;
 301     ALsizei OutPos;
 302     ALsizei IrSize;
 303     bool isplaying;
 304     bool firstpass;
 305     bool isstatic;
 306     ALsizei chan;
 307     ALsizei send;
 308
 309     /* Get source info */
 310     isplaying      = true; /* Will only be called while playing. */
 311     isstatic       = Source->SourceType == AL_STATIC;
 312     DataPosInt     = ATOMIC_LOAD(&voice->position, almemory_order_acquire);
 313     DataPosFrac    = ATOMIC_LOAD(&voice->position_fraction, almemory_order_relaxed);
 314     BufferListItem = ATOMIC_LOAD(&voice->current_buffer, almemory_order_relaxed);
 315     BufferLoopItem = ATOMIC_LOAD(&voice->loop_buffer, almemory_order_relaxed);
 316     NumChannels    = voice->NumChannels;
 317     SampleSize     = voice->SampleSize;
 318     increment      = voice->Step;
 319
 320     IrSize = (Device->HrtfHandle ? Device->HrtfHandle->irSize : 0);
 321
 322     Resample = ((increment == FRACTIONONE && DataPosFrac == 0) ?
 323                 Resample_copy_C : voice->Resampler);
 324
 325     Counter = (voice->Flags&VOICE_IS_FADING) ? SamplesToDo : 0;
 326     firstpass = true;
 327     OutPos = 0;
 328
 329     do {
 330         ALsizei SrcBufferSize, DstBufferSize;
 331
 332         /* Figure out how many buffer samples will be needed */
 333         DataSize64  = SamplesToDo-OutPos;
 334         DataSize64 *= increment;
 335         DataSize64 += DataPosFrac+FRACTIONMASK;
 336         DataSize64 >>= FRACTIONBITS;
 337         DataSize64 += MAX_POST_SAMPLES+MAX_PRE_SAMPLES;
 338
 339         SrcBufferSize = (ALsizei)mini64(DataSize64, BUFFERSIZE);
 340
 341         /* Figure out how many samples we can actually mix from this. */
 342         DataSize64  = SrcBufferSize;
 343         DataSize64 -= MAX_POST_SAMPLES+MAX_PRE_SAMPLES;
 344         DataSize64 <<= FRACTIONBITS;
 345         DataSize64 -= DataPosFrac;
 346
 347         DstBufferSize = (ALsizei)((DataSize64+(increment-1)) / increment);
 348         DstBufferSize = mini(DstBufferSize, (SamplesToDo-OutPos));
 349
 350         /* Some mixers like having a multiple of 4, so try to give that unless
 351          * this is the last update. */
 352         if(OutPos+DstBufferSize < SamplesToDo)
 353             DstBufferSize &= ~3;
 354
 355         for(chan = 0;chan < NumChannels;chan++)
 356         {
 357             const ALfloat *ResampledData;
 358             ALfloat *SrcData = Device->SourceData;
 359             ALsizei SrcDataSize;
 360
 361             /* Load the previous samples into the source data first. */
 362             memcpy(SrcData, voice->PrevSamples[chan], MAX_PRE_SAMPLES*sizeof(ALfloat));
 363             SrcDataSize = MAX_PRE_SAMPLES;
 364
 365             if(isstatic)
 366             {
 367                 const ALbuffer *ALBuffer = BufferListItem->buffer;
 368                 const ALubyte *Data = ALBuffer->data;
 369                 ALsizei DataSize;
 370
 371                 /* Offset buffer data to current channel */
 372                 Data += chan*SampleSize;
 373
 374                 /* If current pos is beyond the loop range, do not loop */
 375                 if(!BufferLoopItem || DataPosInt >= ALBuffer->LoopEnd)
 376                 {
 377                     BufferLoopItem = NULL;
 378
 379                     /* Load what's left to play from the source buffer, and
 380                      * clear the rest of the temp buffer */
 381                     DataSize = minu(SrcBufferSize - SrcDataSize,
 382                                     ALBuffer->SampleLen - DataPosInt);
 383
 384                     LoadSamples(&SrcData[SrcDataSize], &Data[DataPosInt * NumChannels*SampleSize],
 385                                 NumChannels, ALBuffer->FmtType, DataSize);
 386                     SrcDataSize += DataSize;
 387
 388                     SilenceSamples(&SrcData[SrcDataSize], SrcBufferSize - SrcDataSize);
 389                     SrcDataSize += SrcBufferSize - SrcDataSize;
 390                 }
 391                 else
 392                 {
 393                     ALsizei LoopStart = ALBuffer->LoopStart;
 394                     ALsizei LoopEnd   = ALBuffer->LoopEnd;
 395
 396                     /* Load what's left of this loop iteration, then load
 397                      * repeats of the loop section */
 398                     DataSize = minu(SrcBufferSize - SrcDataSize, LoopEnd - DataPosInt);
 399
 400                     LoadSamples(&SrcData[SrcDataSize], &Data[DataPosInt * NumChannels*SampleSize],
 401                                 NumChannels, ALBuffer->FmtType, DataSize);
 402                     SrcDataSize += DataSize;
 403
 404                     DataSize = LoopEnd-LoopStart;
 405                     while(SrcBufferSize > SrcDataSize)
 406                     {
 407                         DataSize = mini(SrcBufferSize - SrcDataSize, DataSize);
 408
 409                         LoadSamples(&SrcData[SrcDataSize], &Data[LoopStart * NumChannels*SampleSize],
 410                                     NumChannels, ALBuffer->FmtType, DataSize);
 411                         SrcDataSize += DataSize;
 412                     }
 413                 }
 414             }
 415             else
 416             {
 417                 /* Crawl the buffer queue to fill in the temp buffer */
 418                 ALbufferlistitem *tmpiter = BufferListItem;
 419                 ALsizei pos = DataPosInt;
 420
 421                 while(tmpiter && SrcBufferSize > SrcDataSize)
 422                 {
 423                     const ALbuffer *ALBuffer;
 424                     if((ALBuffer=tmpiter->buffer) != NULL)
 425                     {
 426                         const ALubyte *Data = ALBuffer->data;
 427                         ALsizei DataSize = ALBuffer->SampleLen;
 428
 429                         /* Skip the data already played */
 430                         if(DataSize <= pos)
 431                             pos -= DataSize;
 432                         else
 433                         {
 434                             Data += (pos*NumChannels + chan)*SampleSize;
 435                             DataSize -= pos;
 436                             pos -= pos;
 437
 438                             DataSize = minu(SrcBufferSize - SrcDataSize, DataSize);
 439                             LoadSamples(&SrcData[SrcDataSize], Data, NumChannels,
 440                                         ALBuffer->FmtType, DataSize);
 441                             SrcDataSize += DataSize;
 442                         }
 443                     }
 444                     tmpiter = ATOMIC_LOAD(&tmpiter->next, almemory_order_acquire);
 445                     if(!tmpiter && BufferLoopItem)
 446                         tmpiter = BufferLoopItem;
 447                     else if(!tmpiter)
 448                     {
 449                         SilenceSamples(&SrcData[SrcDataSize], SrcBufferSize - SrcDataSize);
 450                         SrcDataSize += SrcBufferSize - SrcDataSize;
 451                     }
 452                 }
 453             }
 454
 455             /* Store the last source samples used for next time. */
 456             memcpy(voice->PrevSamples[chan],
 457                 &SrcData[(increment*DstBufferSize + DataPosFrac)>>FRACTIONBITS],
 458                 MAX_PRE_SAMPLES*sizeof(ALfloat)
 459             );
 460
 461             /* Now resample, then filter and mix to the appropriate outputs. */
 462             ResampledData = Resample(&voice->ResampleState,
 463                 &SrcData[MAX_PRE_SAMPLES], DataPosFrac, increment,
 464                 Device->ResampledData, DstBufferSize
 465             );
 466             {
 467                 DirectParams *parms = &voice->Direct.Params[chan];
 468                 const ALfloat *samples;
 469
 470                 samples = DoFilters(
 471                     &parms->LowPass, &parms->HighPass, Device->FilteredData,
 472                     ResampledData, DstBufferSize, voice->Direct.FilterType
 473                 );
 474                 if(!(voice->Flags&VOICE_HAS_HRTF))
 475                 {
 476                     if(!Counter)
 477                         memcpy(parms->Gains.Current, parms->Gains.Target,
 478                                sizeof(parms->Gains.Current));
 479                     if(!(voice->Flags&VOICE_HAS_NFC))
 480                         MixSamples(samples, voice->Direct.Channels, voice->Direct.Buffer,
 481                             parms->Gains.Current, parms->Gains.Target, Counter, OutPos,
 482                             DstBufferSize
 483                         );
 484                     else
 485                     {
 486                         ALfloat *nfcsamples = Device->NFCtrlData;
 487                         ALsizei chanoffset = 0;
 488
 489                         MixSamples(samples,
 490                             voice->Direct.ChannelsPerOrder[0], voice->Direct.Buffer,
 491                             parms->Gains.Current, parms->Gains.Target, Counter, OutPos,
 492                             DstBufferSize
 493                         );
 494                         chanoffset += voice->Direct.ChannelsPerOrder[0];
 495 #define APPLY_NFC_MIX(order)                                                  \
 496     if(voice->Direct.ChannelsPerOrder[order] > 0)                             \
 497     {                                                                         \
 498         NfcFilterUpdate##order(&parms->NFCtrlFilter[order-1], nfcsamples,     \
 499                                samples, DstBufferSize);                       \
 500         MixSamples(nfcsamples, voice->Direct.ChannelsPerOrder[order],         \
 501             voice->Direct.Buffer+chanoffset, parms->Gains.Current+chanoffset, \
 502             parms->Gains.Target+chanoffset, Counter, OutPos, DstBufferSize    \
 503         );                                                                    \
 504         chanoffset += voice->Direct.ChannelsPerOrder[order];                  \
 505     }
 506                         APPLY_NFC_MIX(1)
 507                         APPLY_NFC_MIX(2)
 508                         APPLY_NFC_MIX(3)
 509 #undef APPLY_NFC_MIX
 510                     }
 511                 }
 512                 else
 513                 {
 514                     MixHrtfParams hrtfparams;
 515                     ALsizei fademix = 0;
 516                     int lidx, ridx;
 517
 518                     lidx = GetChannelIdxByName(Device->RealOut, FrontLeft);
 519                     ridx = GetChannelIdxByName(Device->RealOut, FrontRight);
 520                     assert(lidx != -1 && ridx != -1);
 521
 522                     if(!Counter)
 523                     {
 524                         /* No fading, just overwrite the old HRTF params. */
 525                         parms->Hrtf.Old = parms->Hrtf.Target;
 526                     }
 527                     else if(!(parms->Hrtf.Old.Gain > GAIN_SILENCE_THRESHOLD))
 528                     {
 529                         /* The old HRTF params are silent, so overwrite the old
 530                          * coefficients with the new, and reset the old gain to
 531                          * 0. The future mix will then fade from silence.
 532                          */
 533                         parms->Hrtf.Old = parms->Hrtf.Target;
 534                         parms->Hrtf.Old.Gain = 0.0f;
 535                     }
 536                     else if(firstpass)
 537                     {
 538                         ALfloat gain;
 539
 540                         /* Fade between the coefficients over 128 samples. */
 541                         fademix = mini(DstBufferSize, 128);
 542
 543                         /* The new coefficients need to fade in completely
 544                          * since they're replacing the old ones. To keep the
 545                          * gain fading consistent, interpolate between the old
 546                          * and new target gains given how much of the fade time
 547                          * this mix handles.
 548                          */
 549                         gain = lerp(parms->Hrtf.Old.Gain, parms->Hrtf.Target.Gain,
 550                                     minf(1.0f, (ALfloat)fademix/Counter));
 551                         hrtfparams.Coeffs = SAFE_CONST(ALfloat2*,parms->Hrtf.Target.Coeffs);
 552                         hrtfparams.Delay[0] = parms->Hrtf.Target.Delay[0];
 553                         hrtfparams.Delay[1] = parms->Hrtf.Target.Delay[1];
 554                         hrtfparams.Gain = 0.0f;
 555                         hrtfparams.GainStep = gain / (ALfloat)fademix;
 556
 557                         MixHrtfBlendSamples(
 558                             voice->Direct.Buffer[lidx], voice->Direct.Buffer[ridx],
 559                             samples, voice->Offset, OutPos, IrSize, &parms->Hrtf.Old,
 560                             &hrtfparams, &parms->Hrtf.State, fademix
 561                         );
 562                         /* Update the old parameters with the result. */
 563                         parms->Hrtf.Old = parms->Hrtf.Target;
 564                         if(fademix < Counter)
 565                             parms->Hrtf.Old.Gain = hrtfparams.Gain;
 566                     }
 567
 568                     if(fademix < DstBufferSize)
 569                     {
 570                         ALsizei todo = DstBufferSize - fademix;
 571                         ALfloat gain = parms->Hrtf.Target.Gain;
 572
 573                         /* Interpolate the target gain if the gain fading lasts
 574                          * longer than this mix.
 575                          */
 576                         if(Counter > DstBufferSize)
 577                             gain = lerp(parms->Hrtf.Old.Gain, gain,
 578                                         (ALfloat)todo/(Counter-fademix));
 579
 580                         hrtfparams.Coeffs = SAFE_CONST(ALfloat2*,parms->Hrtf.Target.Coeffs);
 581                         hrtfparams.Delay[0] = parms->Hrtf.Target.Delay[0];
 582                         hrtfparams.Delay[1] = parms->Hrtf.Target.Delay[1];
 583                         hrtfparams.Gain = parms->Hrtf.Old.Gain;
 584                         hrtfparams.GainStep = (gain - parms->Hrtf.Old.Gain) / (ALfloat)todo;
 585                         MixHrtfSamples(
 586                             voice->Direct.Buffer[lidx], voice->Direct.Buffer[ridx],
 587                             samples+fademix, voice->Offset+fademix, OutPos+fademix, IrSize,
 588                             &hrtfparams, &parms->Hrtf.State, todo
 589                         );
 590                         /* Store the interpolated gain or the final target gain
 591                          * depending if the fade is done.
 592                          */
 593                         if(DstBufferSize < Counter)
 594                             parms->Hrtf.Old.Gain = gain;
 595                         else
 596                             parms->Hrtf.Old.Gain = parms->Hrtf.Target.Gain;
 597                     }
 598                 }
 599             }
 600
 601             for(send = 0;send < Device->NumAuxSends;send++)
 602             {
 603                 SendParams *parms = &voice->Send[send].Params[chan];
 604                 const ALfloat *samples;
 605
 606                 if(!voice->Send[send].Buffer)
 607                     continue;
 608
 609                 samples = DoFilters(
 610                     &parms->LowPass, &parms->HighPass, Device->FilteredData,
 611                     ResampledData, DstBufferSize, voice->Send[send].FilterType
 612                 );
 613
 614                 if(!Counter)
 615                     memcpy(parms->Gains.Current, parms->Gains.Target,
 616                            sizeof(parms->Gains.Current));
 617                 MixSamples(samples, voice->Send[send].Channels, voice->Send[send].Buffer,
 618                     parms->Gains.Current, parms->Gains.Target, Counter, OutPos, DstBufferSize
 619                 );
 620             }
 621         }
 622         /* Update positions */
 623         DataPosFrac += increment*DstBufferSize;
 624         DataPosInt  += DataPosFrac>>FRACTIONBITS;
 625         DataPosFrac &= FRACTIONMASK;
 626
 627         OutPos += DstBufferSize;
 628         voice->Offset += DstBufferSize;
 629         Counter = maxi(DstBufferSize, Counter) - DstBufferSize;
 630         firstpass = false;
 631
 632         /* Handle looping sources */
 633         while(1)
 634         {
 635             const ALbuffer *ALBuffer;
 636             ALsizei DataSize = 0;
 637             ALsizei LoopStart = 0;
 638             ALsizei LoopEnd = 0;
 639
 640             if((ALBuffer=BufferListItem->buffer) != NULL)
 641             {
 642                 DataSize = ALBuffer->SampleLen;
 643                 LoopStart = ALBuffer->LoopStart;
 644                 LoopEnd = ALBuffer->LoopEnd;
 645                 if(LoopEnd > DataPosInt)
 646                     break;
 647             }
 648
 649             if(isstatic && BufferLoopItem)
 650             {
 651                 assert(LoopEnd > LoopStart);
 652                 DataPosInt = ((DataPosInt-LoopStart)%(LoopEnd-LoopStart)) + LoopStart;
 653                 break;
 654             }
 655
 656             if(DataSize > DataPosInt)
 657                 break;
 658
 659             BufferListItem = ATOMIC_LOAD(&BufferListItem->next, almemory_order_acquire);
 660             if(!BufferListItem)
 661             {
 662                 BufferListItem = BufferLoopItem;
 663                 if(!BufferListItem)
 664                 {
 665                     isplaying = false;
 666                     DataPosInt = 0;
 667                     DataPosFrac = 0;
 668                     break;
 669                 }
 670             }
 671
 672             DataPosInt -= DataSize;
 673         }
 674     } while(isplaying && OutPos < SamplesToDo);
 675
 676     voice->Flags |= VOICE_IS_FADING;
 677
 678     /* Update source info */
 679     ATOMIC_STORE(&voice->position,          DataPosInt, almemory_order_relaxed);
 680     ATOMIC_STORE(&voice->position_fraction, DataPosFrac, almemory_order_relaxed);
 681     ATOMIC_STORE(&voice->current_buffer,    BufferListItem, almemory_order_release);
 682     return isplaying;
 683 }