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_point32_C;
 116         case LinearResampler:
 117 #ifdef HAVE_NEON
 118             if((CPUCapFlags&CPU_CAP_NEON))
 119                 return Resample_lerp32_Neon;
 120 #endif
 121 #ifdef HAVE_SSE4_1
 122             if((CPUCapFlags&CPU_CAP_SSE4_1))
 123                 return Resample_lerp32_SSE41;
 124 #endif
 125 #ifdef HAVE_SSE2
 126             if((CPUCapFlags&CPU_CAP_SSE2))
 127                 return Resample_lerp32_SSE2;
 128 #endif
 129             return Resample_lerp32_C;
 130         case FIR4Resampler:
 131 #ifdef HAVE_NEON
 132             if((CPUCapFlags&CPU_CAP_NEON))
 133                 return Resample_fir4_32_Neon;
 134 #endif
 135 #ifdef HAVE_SSE4_1
 136             if((CPUCapFlags&CPU_CAP_SSE4_1))
 137                 return Resample_fir4_32_SSE41;
 138 #endif
 139 #ifdef HAVE_SSE3
 140             if((CPUCapFlags&CPU_CAP_SSE3))
 141                 return Resample_fir4_32_SSE3;
 142 #endif
 143             return Resample_fir4_32_C;
 144         case BSincResampler:
 145 #ifdef HAVE_NEON
 146             if((CPUCapFlags&CPU_CAP_NEON))
 147                 return Resample_bsinc32_Neon;
 148 #endif
 149 #ifdef HAVE_SSE
 150             if((CPUCapFlags&CPU_CAP_SSE))
 151                 return Resample_bsinc32_SSE;
 152 #endif
 153             return Resample_bsinc32_C;
 154     }
 155
 156     return Resample_point32_C;
 157 }
 158
 159
 160 void aluInitMixer(void)
 161 {
 162     const char *str;
 163
 164     if(ConfigValueStr(NULL, NULL, "resampler", &str))
 165     {
 166         if(strcasecmp(str, "point") == 0 || strcasecmp(str, "none") == 0)
 167             ResamplerDefault = PointResampler;
 168         else if(strcasecmp(str, "linear") == 0)
 169             ResamplerDefault = LinearResampler;
 170         else if(strcasecmp(str, "sinc4") == 0)
 171             ResamplerDefault = FIR4Resampler;
 172         else if(strcasecmp(str, "bsinc") == 0)
 173             ResamplerDefault = BSincResampler;
 174         else if(strcasecmp(str, "cubic") == 0 || strcasecmp(str, "sinc8") == 0)
 175         {
 176             WARN("Resampler option \"%s\" is deprecated, using sinc4\n", str);
 177             ResamplerDefault = FIR4Resampler;
 178         }
 179         else
 180         {
 181             char *end;
 182             long n = strtol(str, &end, 0);
 183             if(*end == '\0' && (n == PointResampler || n == LinearResampler || n == FIR4Resampler))
 184                 ResamplerDefault = n;
 185             else
 186                 WARN("Invalid resampler: %s\n", str);
 187         }
 188     }
 189
 190     MixHrtfBlendSamples = SelectHrtfBlendMixer();
 191     MixHrtfSamples = SelectHrtfMixer();
 192     MixSamples = SelectMixer();
 193 }
 194
 195
 196 static inline ALfloat Sample_ALbyte(ALbyte val)
 197 { return val * (1.0f/128.0f); }
 198
 199 static inline ALfloat Sample_ALshort(ALshort val)
 200 { return val * (1.0f/32768.0f); }
 201
 202 static inline ALfloat Sample_ALfloat(ALfloat val)
 203 { return val; }
 204
 205 #define DECL_TEMPLATE(T)                                                      \
 206 static inline void Load_##T(ALfloat *dst, const T *src, ALint srcstep, ALsizei samples)\
 207 {                                                                             \
 208     ALsizei i;                                                                \
 209     for(i = 0;i < samples;i++)                                                \
 210         dst[i] = Sample_##T(src[i*srcstep]);                                  \
 211 }
 212
 213 DECL_TEMPLATE(ALbyte)
 214 DECL_TEMPLATE(ALshort)
 215 DECL_TEMPLATE(ALfloat)
 216
 217 #undef DECL_TEMPLATE
 218
 219 static void LoadSamples(ALfloat *dst, const ALvoid *src, ALint srcstep, enum FmtType srctype, ALsizei samples)
 220 {
 221     switch(srctype)
 222     {
 223         case FmtByte:
 224             Load_ALbyte(dst, src, srcstep, samples);
 225             break;
 226         case FmtShort:
 227             Load_ALshort(dst, src, srcstep, samples);
 228             break;
 229         case FmtFloat:
 230             Load_ALfloat(dst, src, srcstep, samples);
 231             break;
 232     }
 233 }
 234
 235 static inline void SilenceSamples(ALfloat *dst, ALsizei samples)
 236 {
 237     ALsizei i;
 238     for(i = 0;i < samples;i++)
 239         dst[i] = 0.0f;
 240 }
 241
 242
 243 static const ALfloat *DoFilters(ALfilterState *lpfilter, ALfilterState *hpfilter,
 244                                 ALfloat *restrict dst, const ALfloat *restrict src,
 245                                 ALsizei numsamples, enum ActiveFilters type)
 246 {
 247     ALsizei i;
 248     switch(type)
 249     {
 250         case AF_None:
 251             ALfilterState_processPassthru(lpfilter, src, numsamples);
 252             ALfilterState_processPassthru(hpfilter, src, numsamples);
 253             break;
 254
 255         case AF_LowPass:
 256             ALfilterState_process(lpfilter, dst, src, numsamples);
 257             ALfilterState_processPassthru(hpfilter, dst, numsamples);
 258             return dst;
 259         case AF_HighPass:
 260             ALfilterState_processPassthru(lpfilter, src, numsamples);
 261             ALfilterState_process(hpfilter, dst, src, numsamples);
 262             return dst;
 263
 264         case AF_BandPass:
 265             for(i = 0;i < numsamples;)
 266             {
 267                 ALfloat temp[256];
 268                 ALsizei todo = mini(256, numsamples-i);
 269
 270                 ALfilterState_process(lpfilter, temp, src+i, todo);
 271                 ALfilterState_process(hpfilter, dst+i, temp, todo);
 272                 i += todo;
 273             }
 274             return dst;
 275     }
 276     return src;
 277 }
 278
 279
 280 ALboolean MixSource(ALvoice *voice, ALsource *Source, ALCdevice *Device, ALsizei SamplesToDo)
 281 {
 282     ALbufferlistitem *BufferListItem;
 283     ALbufferlistitem *BufferLoopItem;
 284     ALsizei NumChannels, SampleSize;
 285     ResamplerFunc Resample;
 286     ALsizei DataPosInt;
 287     ALsizei DataPosFrac;
 288     ALint64 DataSize64;
 289     ALint increment;
 290     ALsizei Counter;
 291     ALsizei OutPos;
 292     ALsizei IrSize;
 293     bool isplaying;
 294     bool firstpass;
 295     ALsizei chan;
 296     ALsizei send;
 297
 298     /* Get source info */
 299     isplaying      = true; /* Will only be called while playing. */
 300     DataPosInt     = ATOMIC_LOAD(&voice->position, almemory_order_acquire);
 301     DataPosFrac    = ATOMIC_LOAD(&voice->position_fraction, almemory_order_relaxed);
 302     BufferListItem = ATOMIC_LOAD(&voice->current_buffer, almemory_order_relaxed);
 303     BufferLoopItem = ATOMIC_LOAD(&voice->loop_buffer, almemory_order_relaxed);
 304     NumChannels    = voice->NumChannels;
 305     SampleSize     = voice->SampleSize;
 306     increment      = voice->Step;
 307
 308     IrSize = (Device->HrtfHandle ? Device->HrtfHandle->irSize : 0);
 309
 310     Resample = ((increment == FRACTIONONE && DataPosFrac == 0) ?
 311                 Resample_copy32_C : voice->Resampler);
 312
 313     Counter = (voice->Flags&VOICE_IS_FADING) ? SamplesToDo : 0;
 314     firstpass = true;
 315     OutPos = 0;
 316
 317     do {
 318         ALsizei SrcBufferSize, DstBufferSize;
 319
 320         /* Figure out how many buffer samples will be needed */
 321         DataSize64  = SamplesToDo-OutPos;
 322         DataSize64 *= increment;
 323         DataSize64 += DataPosFrac+FRACTIONMASK;
 324         DataSize64 >>= FRACTIONBITS;
 325         DataSize64 += MAX_POST_SAMPLES+MAX_PRE_SAMPLES;
 326
 327         SrcBufferSize = (ALsizei)mini64(DataSize64, BUFFERSIZE);
 328
 329         /* Figure out how many samples we can actually mix from this. */
 330         DataSize64  = SrcBufferSize;
 331         DataSize64 -= MAX_POST_SAMPLES+MAX_PRE_SAMPLES;
 332         DataSize64 <<= FRACTIONBITS;
 333         DataSize64 -= DataPosFrac;
 334
 335         DstBufferSize = (ALsizei)((DataSize64+(increment-1)) / increment);
 336         DstBufferSize = mini(DstBufferSize, (SamplesToDo-OutPos));
 337
 338         /* Some mixers like having a multiple of 4, so try to give that unless
 339          * this is the last update. */
 340         if(OutPos+DstBufferSize < SamplesToDo)
 341             DstBufferSize &= ~3;
 342
 343         for(chan = 0;chan < NumChannels;chan++)
 344         {
 345             const ALfloat *ResampledData;
 346             ALfloat *SrcData = Device->SourceData;
 347             ALsizei SrcDataSize;
 348
 349             /* Load the previous samples into the source data first. */
 350             memcpy(SrcData, voice->PrevSamples[chan], MAX_PRE_SAMPLES*sizeof(ALfloat));
 351             SrcDataSize = MAX_PRE_SAMPLES;
 352
 353             if(Source->SourceType == AL_STATIC)
 354             {
 355                 const ALbuffer *ALBuffer = BufferListItem->buffer;
 356                 const ALubyte *Data = ALBuffer->data;
 357                 ALsizei DataSize;
 358
 359                 /* Offset buffer data to current channel */
 360                 Data += chan*SampleSize;
 361
 362                 /* If current pos is beyond the loop range, do not loop */
 363                 if(!BufferLoopItem || DataPosInt >= ALBuffer->LoopEnd)
 364                 {
 365                     BufferLoopItem = NULL;
 366
 367                     /* Load what's left to play from the source buffer, and
 368                      * clear the rest of the temp buffer */
 369                     DataSize = minu(SrcBufferSize - SrcDataSize,
 370                                     ALBuffer->SampleLen - DataPosInt);
 371
 372                     LoadSamples(&SrcData[SrcDataSize], &Data[DataPosInt * NumChannels*SampleSize],
 373                                 NumChannels, ALBuffer->FmtType, DataSize);
 374                     SrcDataSize += DataSize;
 375
 376                     SilenceSamples(&SrcData[SrcDataSize], SrcBufferSize - SrcDataSize);
 377                     SrcDataSize += SrcBufferSize - SrcDataSize;
 378                 }
 379                 else
 380                 {
 381                     ALsizei LoopStart = ALBuffer->LoopStart;
 382                     ALsizei LoopEnd   = ALBuffer->LoopEnd;
 383
 384                     /* Load what's left of this loop iteration, then load
 385                      * repeats of the loop section */
 386                     DataSize = minu(SrcBufferSize - SrcDataSize, LoopEnd - DataPosInt);
 387
 388                     LoadSamples(&SrcData[SrcDataSize], &Data[DataPosInt * NumChannels*SampleSize],
 389                                 NumChannels, ALBuffer->FmtType, DataSize);
 390                     SrcDataSize += DataSize;
 391
 392                     DataSize = LoopEnd-LoopStart;
 393                     while(SrcBufferSize > SrcDataSize)
 394                     {
 395                         DataSize = mini(SrcBufferSize - SrcDataSize, DataSize);
 396
 397                         LoadSamples(&SrcData[SrcDataSize], &Data[LoopStart * NumChannels*SampleSize],
 398                                     NumChannels, ALBuffer->FmtType, DataSize);
 399                         SrcDataSize += DataSize;
 400                     }
 401                 }
 402             }
 403             else
 404             {
 405                 /* Crawl the buffer queue to fill in the temp buffer */
 406                 ALbufferlistitem *tmpiter = BufferListItem;
 407                 ALsizei pos = DataPosInt;
 408
 409                 while(tmpiter && SrcBufferSize > SrcDataSize)
 410                 {
 411                     const ALbuffer *ALBuffer;
 412                     if((ALBuffer=tmpiter->buffer) != NULL)
 413                     {
 414                         const ALubyte *Data = ALBuffer->data;
 415                         ALsizei DataSize = ALBuffer->SampleLen;
 416
 417                         /* Skip the data already played */
 418                         if(DataSize <= pos)
 419                             pos -= DataSize;
 420                         else
 421                         {
 422                             Data += (pos*NumChannels + chan)*SampleSize;
 423                             DataSize -= pos;
 424                             pos -= pos;
 425
 426                             DataSize = minu(SrcBufferSize - SrcDataSize, DataSize);
 427                             LoadSamples(&SrcData[SrcDataSize], Data, NumChannels,
 428                                         ALBuffer->FmtType, DataSize);
 429                             SrcDataSize += DataSize;
 430                         }
 431                     }
 432                     tmpiter = ATOMIC_LOAD(&tmpiter->next, almemory_order_acquire);
 433                     if(!tmpiter && BufferLoopItem)
 434                         tmpiter = BufferLoopItem;
 435                     else if(!tmpiter)
 436                     {
 437                         SilenceSamples(&SrcData[SrcDataSize], SrcBufferSize - SrcDataSize);
 438                         SrcDataSize += SrcBufferSize - SrcDataSize;
 439                     }
 440                 }
 441             }
 442
 443             /* Store the last source samples used for next time. */
 444             memcpy(voice->PrevSamples[chan],
 445                 &SrcData[(increment*DstBufferSize + DataPosFrac)>>FRACTIONBITS],
 446                 MAX_PRE_SAMPLES*sizeof(ALfloat)
 447             );
 448
 449             /* Now resample, then filter and mix to the appropriate outputs. */
 450             ResampledData = Resample(&voice->ResampleState,
 451                 &SrcData[MAX_PRE_SAMPLES], DataPosFrac, increment,
 452                 Device->ResampledData, DstBufferSize
 453             );
 454             {
 455                 DirectParams *parms = &voice->Direct.Params[chan];
 456                 const ALfloat *samples;
 457
 458                 samples = DoFilters(
 459                     &parms->LowPass, &parms->HighPass, Device->FilteredData,
 460                     ResampledData, DstBufferSize, voice->Direct.FilterType
 461                 );
 462                 if(!(voice->Flags&VOICE_HAS_HRTF))
 463                 {
 464                     if(!Counter)
 465                         memcpy(parms->Gains.Current, parms->Gains.Target,
 466                                sizeof(parms->Gains.Current));
 467                     if(!(voice->Flags&VOICE_HAS_NFC))
 468                         MixSamples(samples, voice->Direct.Channels, voice->Direct.Buffer,
 469                             parms->Gains.Current, parms->Gains.Target, Counter, OutPos,
 470                             DstBufferSize
 471                         );
 472                     else
 473                     {
 474                         ALfloat *nfcsamples = Device->NFCtrlData;
 475                         ALsizei chanoffset = 0;
 476
 477                         MixSamples(samples,
 478                             voice->Direct.ChannelsPerOrder[0], voice->Direct.Buffer,
 479                             parms->Gains.Current, parms->Gains.Target, Counter, OutPos,
 480                             DstBufferSize
 481                         );
 482                         chanoffset += voice->Direct.ChannelsPerOrder[0];
 483 #define APPLY_NFC_MIX(order)                                                  \
 484     if(voice->Direct.ChannelsPerOrder[order] > 0)                             \
 485     {                                                                         \
 486         NfcFilterUpdate##order(&parms->NFCtrlFilter[order-1], nfcsamples,     \
 487                                samples, DstBufferSize);                       \
 488         MixSamples(nfcsamples, voice->Direct.ChannelsPerOrder[order],         \
 489             voice->Direct.Buffer+chanoffset, parms->Gains.Current+chanoffset, \
 490             parms->Gains.Target+chanoffset, Counter, OutPos, DstBufferSize    \
 491         );                                                                    \
 492         chanoffset += voice->Direct.ChannelsPerOrder[order];                  \
 493     }
 494                         APPLY_NFC_MIX(1)
 495                         APPLY_NFC_MIX(2)
 496                         APPLY_NFC_MIX(3)
 497 #undef APPLY_NFC_MIX
 498                     }
 499                 }
 500                 else
 501                 {
 502                     MixHrtfParams hrtfparams;
 503                     ALsizei fademix = 0;
 504                     int lidx, ridx;
 505
 506                     lidx = GetChannelIdxByName(Device->RealOut, FrontLeft);
 507                     ridx = GetChannelIdxByName(Device->RealOut, FrontRight);
 508                     assert(lidx != -1 && ridx != -1);
 509
 510                     if(!Counter)
 511                     {
 512                         /* No fading, just overwrite the old HRTF params. */
 513                         parms->Hrtf.Old = parms->Hrtf.Target;
 514                     }
 515                     else if(!(parms->Hrtf.Old.Gain > GAIN_SILENCE_THRESHOLD))
 516                     {
 517                         /* The old HRTF params are silent, so overwrite the old
 518                          * coefficients with the new, and reset the old gain to
 519                          * 0. The future mix will then fade from silence.
 520                          */
 521                         parms->Hrtf.Old = parms->Hrtf.Target;
 522                         parms->Hrtf.Old.Gain = 0.0f;
 523                     }
 524                     else if(firstpass)
 525                     {
 526                         ALfloat gain;
 527
 528                         /* Fade between the coefficients over 128 samples. */
 529                         fademix = mini(DstBufferSize, 128);
 530
 531                         /* The new coefficients need to fade in completely
 532                          * since they're replacing the old ones. To keep the
 533                          * gain fading consistent, interpolate between the old
 534                          * and new target gains given how much of the fade time
 535                          * this mix handles.
 536                          */
 537                         gain = lerp(parms->Hrtf.Old.Gain, parms->Hrtf.Target.Gain,
 538                                     minf(1.0f, (ALfloat)fademix/Counter));
 539                         hrtfparams.Coeffs = SAFE_CONST(ALfloat2*,parms->Hrtf.Target.Coeffs);
 540                         hrtfparams.Delay[0] = parms->Hrtf.Target.Delay[0];
 541                         hrtfparams.Delay[1] = parms->Hrtf.Target.Delay[1];
 542                         hrtfparams.Gain = 0.0f;
 543                         hrtfparams.GainStep = gain / (ALfloat)fademix;
 544
 545                         MixHrtfBlendSamples(
 546                             voice->Direct.Buffer[lidx], voice->Direct.Buffer[ridx],
 547                             samples, voice->Offset, OutPos, IrSize, &parms->Hrtf.Old,
 548                             &hrtfparams, &parms->Hrtf.State, fademix
 549                         );
 550                         /* Update the old parameters with the result. */
 551                         parms->Hrtf.Old = parms->Hrtf.Target;
 552                         if(fademix < Counter)
 553                             parms->Hrtf.Old.Gain = hrtfparams.Gain;
 554                     }
 555
 556                     if(fademix < DstBufferSize)
 557                     {
 558                         ALsizei todo = DstBufferSize - fademix;
 559                         ALfloat gain = parms->Hrtf.Target.Gain;
 560
 561                         /* Interpolate the target gain if the gain fading lasts
 562                          * longer than this mix.
 563                          */
 564                         if(Counter > DstBufferSize)
 565                             gain = lerp(parms->Hrtf.Old.Gain, gain,
 566                                         (ALfloat)todo/(Counter-fademix));
 567
 568                         hrtfparams.Coeffs = SAFE_CONST(ALfloat2*,parms->Hrtf.Target.Coeffs);
 569                         hrtfparams.Delay[0] = parms->Hrtf.Target.Delay[0];
 570                         hrtfparams.Delay[1] = parms->Hrtf.Target.Delay[1];
 571                         hrtfparams.Gain = parms->Hrtf.Old.Gain;
 572                         hrtfparams.GainStep = (gain - parms->Hrtf.Old.Gain) / (ALfloat)todo;
 573                         MixHrtfSamples(
 574                             voice->Direct.Buffer[lidx], voice->Direct.Buffer[ridx],
 575                             samples+fademix, voice->Offset+fademix, OutPos+fademix, IrSize,
 576                             &hrtfparams, &parms->Hrtf.State, todo
 577                         );
 578                         /* Store the interpolated gain or the final target gain
 579                          * depending if the fade is done.
 580                          */
 581                         if(DstBufferSize < Counter)
 582                             parms->Hrtf.Old.Gain = gain;
 583                         else
 584                             parms->Hrtf.Old.Gain = parms->Hrtf.Target.Gain;
 585                     }
 586                 }
 587             }
 588
 589             for(send = 0;send < Device->NumAuxSends;send++)
 590             {
 591                 SendParams *parms = &voice->Send[send].Params[chan];
 592                 const ALfloat *samples;
 593
 594                 if(!voice->Send[send].Buffer)
 595                     continue;
 596
 597                 samples = DoFilters(
 598                     &parms->LowPass, &parms->HighPass, Device->FilteredData,
 599                     ResampledData, DstBufferSize, voice->Send[send].FilterType
 600                 );
 601
 602                 if(!Counter)
 603                     memcpy(parms->Gains.Current, parms->Gains.Target,
 604                            sizeof(parms->Gains.Current));
 605                 MixSamples(samples, voice->Send[send].Channels, voice->Send[send].Buffer,
 606                     parms->Gains.Current, parms->Gains.Target, Counter, OutPos, DstBufferSize
 607                 );
 608             }
 609         }
 610         /* Update positions */
 611         DataPosFrac += increment*DstBufferSize;
 612         DataPosInt  += DataPosFrac>>FRACTIONBITS;
 613         DataPosFrac &= FRACTIONMASK;
 614
 615         OutPos += DstBufferSize;
 616         voice->Offset += DstBufferSize;
 617         Counter = maxi(DstBufferSize, Counter) - DstBufferSize;
 618         firstpass = false;
 619
 620         /* Handle looping sources */
 621         while(1)
 622         {
 623             const ALbuffer *ALBuffer;
 624             ALsizei DataSize = 0;
 625             ALsizei LoopStart = 0;
 626             ALsizei LoopEnd = 0;
 627
 628             if((ALBuffer=BufferListItem->buffer) != NULL)
 629             {
 630                 DataSize = ALBuffer->SampleLen;
 631                 LoopStart = ALBuffer->LoopStart;
 632                 LoopEnd = ALBuffer->LoopEnd;
 633                 if(LoopEnd > DataPosInt)
 634                     break;
 635             }
 636
 637             if(BufferLoopItem && Source->SourceType == AL_STATIC)
 638             {
 639                 assert(LoopEnd > LoopStart);
 640                 DataPosInt = ((DataPosInt-LoopStart)%(LoopEnd-LoopStart)) + LoopStart;
 641                 break;
 642             }
 643
 644             if(DataSize > DataPosInt)
 645                 break;
 646
 647             BufferListItem = ATOMIC_LOAD(&BufferListItem->next, almemory_order_acquire);
 648             if(!BufferListItem)
 649             {
 650                 BufferListItem = BufferLoopItem;
 651                 if(!BufferListItem)
 652                 {
 653                     isplaying = false;
 654                     DataPosInt = 0;
 655                     DataPosFrac = 0;
 656                     break;
 657                 }
 658             }
 659
 660             DataPosInt -= DataSize;
 661         }
 662     } while(isplaying && OutPos < SamplesToDo);
 663
 664     voice->Flags |= VOICE_IS_FADING;
 665
 666     /* Update source info */
 667     ATOMIC_STORE(&voice->position,          DataPosInt, almemory_order_relaxed);
 668     ATOMIC_STORE(&voice->position_fraction, DataPosFrac, almemory_order_relaxed);
 669     ATOMIC_STORE(&voice->current_buffer,    BufferListItem, almemory_order_release);
 670     return isplaying;
 671 }