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 #include "alconfig.h"
  38
  39 #include "cpu_caps.h"
  40 #include "mixer_defs.h"
  41
  42
  43 static_assert((INT_MAX>>FRACTIONBITS)/MAX_PITCH > BUFFERSIZE,
  44               "MAX_PITCH and/or BUFFERSIZE are too large for FRACTIONBITS!");
  45
  46 extern inline void InitiatePositionArrays(ALsizei frac, ALint increment, ALsizei *restrict frac_arr, ALint *restrict pos_arr, ALsizei size);
  47
  48
  49 /* BSinc24 requires up to 23 extra samples before the current position, and 24 after. */
  50 static_assert(MAX_RESAMPLE_PADDING >= 24, "MAX_RESAMPLE_PADDING must be at least 24!");
  51
  52
  53 enum Resampler ResamplerDefault = LinearResampler;
  54
  55 MixerFunc MixSamples = Mix_C;
  56 static HrtfMixerFunc MixHrtfSamples = MixHrtf_C;
  57 static HrtfMixerBlendFunc MixHrtfBlendSamples = MixHrtfBlend_C;
  58
  59 static MixerFunc SelectMixer(void)
  60 {
  61 #ifdef HAVE_NEON
  62     if((CPUCapFlags&CPU_CAP_NEON))
  63         return Mix_Neon;
  64 #endif
  65 #ifdef HAVE_SSE
  66     if((CPUCapFlags&CPU_CAP_SSE))
  67         return Mix_SSE;
  68 #endif
  69     return Mix_C;
  70 }
  71
  72 RowMixerFunc SelectRowMixer(void)
  73 {
  74 #ifdef HAVE_NEON
  75     if((CPUCapFlags&CPU_CAP_NEON))
  76         return MixRow_Neon;
  77 #endif
  78 #ifdef HAVE_SSE
  79     if((CPUCapFlags&CPU_CAP_SSE))
  80         return MixRow_SSE;
  81 #endif
  82     return MixRow_C;
  83 }
  84
  85 static inline HrtfMixerFunc SelectHrtfMixer(void)
  86 {
  87 #ifdef HAVE_NEON
  88     if((CPUCapFlags&CPU_CAP_NEON))
  89         return MixHrtf_Neon;
  90 #endif
  91 #ifdef HAVE_SSE
  92     if((CPUCapFlags&CPU_CAP_SSE))
  93         return MixHrtf_SSE;
  94 #endif
  95     return MixHrtf_C;
  96 }
  97
  98 static inline HrtfMixerBlendFunc SelectHrtfBlendMixer(void)
  99 {
 100 #ifdef HAVE_NEON
 101     if((CPUCapFlags&CPU_CAP_NEON))
 102         return MixHrtfBlend_Neon;
 103 #endif
 104 #ifdef HAVE_SSE
 105     if((CPUCapFlags&CPU_CAP_SSE))
 106         return MixHrtfBlend_SSE;
 107 #endif
 108     return MixHrtfBlend_C;
 109 }
 110
 111 ResamplerFunc SelectResampler(enum Resampler resampler)
 112 {
 113     switch(resampler)
 114     {
 115         case PointResampler:
 116             return Resample_point_C;
 117         case LinearResampler:
 118 #ifdef HAVE_NEON
 119             if((CPUCapFlags&CPU_CAP_NEON))
 120                 return Resample_lerp_Neon;
 121 #endif
 122 #ifdef HAVE_SSE4_1
 123             if((CPUCapFlags&CPU_CAP_SSE4_1))
 124                 return Resample_lerp_SSE41;
 125 #endif
 126 #ifdef HAVE_SSE2
 127             if((CPUCapFlags&CPU_CAP_SSE2))
 128                 return Resample_lerp_SSE2;
 129 #endif
 130             return Resample_lerp_C;
 131         case FIR4Resampler:
 132             return Resample_cubic_C;
 133         case BSinc12Resampler:
 134         case BSinc24Resampler:
 135 #ifdef HAVE_NEON
 136             if((CPUCapFlags&CPU_CAP_NEON))
 137                 return Resample_bsinc_Neon;
 138 #endif
 139 #ifdef HAVE_SSE
 140             if((CPUCapFlags&CPU_CAP_SSE))
 141                 return Resample_bsinc_SSE;
 142 #endif
 143             return Resample_bsinc_C;
 144     }
 145
 146     return Resample_point_C;
 147 }
 148
 149
 150 void aluInitMixer(void)
 151 {
 152     const char *str;
 153
 154     if(ConfigValueStr(NULL, NULL, "resampler", &str))
 155     {
 156         if(strcasecmp(str, "point") == 0 || strcasecmp(str, "none") == 0)
 157             ResamplerDefault = PointResampler;
 158         else if(strcasecmp(str, "linear") == 0)
 159             ResamplerDefault = LinearResampler;
 160         else if(strcasecmp(str, "cubic") == 0)
 161             ResamplerDefault = FIR4Resampler;
 162         else if(strcasecmp(str, "bsinc12") == 0)
 163             ResamplerDefault = BSinc12Resampler;
 164         else if(strcasecmp(str, "bsinc24") == 0)
 165             ResamplerDefault = BSinc24Resampler;
 166         else if(strcasecmp(str, "bsinc") == 0)
 167         {
 168             WARN("Resampler option \"%s\" is deprecated, using bsinc12\n", str);
 169             ResamplerDefault = BSinc12Resampler;
 170         }
 171         else if(strcasecmp(str, "sinc4") == 0 || strcasecmp(str, "sinc8") == 0)
 172         {
 173             WARN("Resampler option \"%s\" is deprecated, using cubic\n", str);
 174             ResamplerDefault = FIR4Resampler;
 175         }
 176         else
 177         {
 178             char *end;
 179             long n = strtol(str, &end, 0);
 180             if(*end == '\0' && (n == PointResampler || n == LinearResampler || n == FIR4Resampler))
 181                 ResamplerDefault = n;
 182             else
 183                 WARN("Invalid resampler: %s\n", str);
 184         }
 185     }
 186
 187     MixHrtfBlendSamples = SelectHrtfBlendMixer();
 188     MixHrtfSamples = SelectHrtfMixer();
 189     MixSamples = SelectMixer();
 190 }
 191
 192
 193 static inline ALfloat Sample_ALbyte(ALbyte val)
 194 { return val * (1.0f/128.0f); }
 195
 196 static inline ALfloat Sample_ALshort(ALshort val)
 197 { return val * (1.0f/32768.0f); }
 198
 199 static inline ALfloat Sample_ALfloat(ALfloat val)
 200 { return val; }
 201
 202 #define DECL_TEMPLATE(T)                                                      \
 203 static inline void Load_##T(ALfloat *restrict dst, const T *restrict src,     \
 204                             ALint srcstep, ALsizei samples)                   \
 205 {                                                                             \
 206     ALsizei i;                                                                \
 207     for(i = 0;i < samples;i++)                                                \
 208         dst[i] += Sample_##T(src[i*srcstep]);                                 \
 209 }
 210
 211 DECL_TEMPLATE(ALbyte)
 212 DECL_TEMPLATE(ALshort)
 213 DECL_TEMPLATE(ALfloat)
 214
 215 #undef DECL_TEMPLATE
 216
 217 static void LoadSamples(ALfloat *restrict dst, const ALvoid *restrict src, ALint srcstep,
 218                         enum FmtType srctype, ALsizei samples)
 219 {
 220     switch(srctype)
 221     {
 222         case FmtByte:
 223             Load_ALbyte(dst, src, srcstep, samples);
 224             break;
 225         case FmtShort:
 226             Load_ALshort(dst, src, srcstep, samples);
 227             break;
 228         case FmtFloat:
 229             Load_ALfloat(dst, src, srcstep, samples);
 230             break;
 231     }
 232 }
 233
 234
 235 static const ALfloat *DoFilters(ALfilterState *lpfilter, ALfilterState *hpfilter,
 236                                 ALfloat *restrict dst, const ALfloat *restrict src,
 237                                 ALsizei numsamples, enum ActiveFilters type)
 238 {
 239     ALsizei i;
 240     switch(type)
 241     {
 242         case AF_None:
 243             ALfilterState_processPassthru(lpfilter, src, numsamples);
 244             ALfilterState_processPassthru(hpfilter, src, numsamples);
 245             break;
 246
 247         case AF_LowPass:
 248             ALfilterState_process(lpfilter, dst, src, numsamples);
 249             ALfilterState_processPassthru(hpfilter, dst, numsamples);
 250             return dst;
 251         case AF_HighPass:
 252             ALfilterState_processPassthru(lpfilter, src, numsamples);
 253             ALfilterState_process(hpfilter, dst, src, numsamples);
 254             return dst;
 255
 256         case AF_BandPass:
 257             for(i = 0;i < numsamples;)
 258             {
 259                 ALfloat temp[256];
 260                 ALsizei todo = mini(256, numsamples-i);
 261
 262                 ALfilterState_process(lpfilter, temp, src+i, todo);
 263                 ALfilterState_process(hpfilter, dst+i, temp, todo);
 264                 i += todo;
 265             }
 266             return dst;
 267     }
 268     return src;
 269 }
 270
 271
 272 /* This function uses these device temp buffers. */
 273 #define SOURCE_DATA_BUF 0
 274 #define RESAMPLED_BUF 1
 275 #define FILTERED_BUF 2
 276 #define NFC_DATA_BUF 3
 277 ALboolean MixSource(ALvoice *voice, ALsource *Source, ALCdevice *Device, ALsizei SamplesToDo)
 278 {
 279     ALbufferlistitem *BufferListItem;
 280     ALbufferlistitem *BufferLoopItem;
 281     ALsizei NumChannels, SampleSize;
 282     ResamplerFunc Resample;
 283     ALsizei DataPosInt;
 284     ALsizei DataPosFrac;
 285     ALint64 DataSize64;
 286     ALint increment;
 287     ALsizei Counter;
 288     ALsizei OutPos;
 289     ALsizei IrSize;
 290     bool isplaying;
 291     bool firstpass;
 292     bool isstatic;
 293     ALsizei chan;
 294     ALsizei send;
 295
 296     /* Get source info */
 297     isplaying      = true; /* Will only be called while playing. */
 298     isstatic       = Source->SourceType == AL_STATIC;
 299     DataPosInt     = ATOMIC_LOAD(&voice->position, almemory_order_acquire);
 300     DataPosFrac    = ATOMIC_LOAD(&voice->position_fraction, almemory_order_relaxed);
 301     BufferListItem = ATOMIC_LOAD(&voice->current_buffer, almemory_order_relaxed);
 302     BufferLoopItem = ATOMIC_LOAD(&voice->loop_buffer, almemory_order_relaxed);
 303     NumChannels    = voice->NumChannels;
 304     SampleSize     = voice->SampleSize;
 305     increment      = voice->Step;
 306
 307     IrSize = (Device->HrtfHandle ? Device->HrtfHandle->irSize : 0);
 308
 309     Resample = ((increment == FRACTIONONE && DataPosFrac == 0) ?
 310                 Resample_copy_C : voice->Resampler);
 311
 312     Counter = (voice->Flags&VOICE_IS_FADING) ? SamplesToDo : 0;
 313     firstpass = true;
 314     OutPos = 0;
 315
 316     do {
 317         ALsizei SrcBufferSize, DstBufferSize;
 318
 319         /* Figure out how many buffer samples will be needed */
 320         DataSize64  = SamplesToDo-OutPos;
 321         DataSize64 *= increment;
 322         DataSize64 += DataPosFrac+FRACTIONMASK;
 323         DataSize64 >>= FRACTIONBITS;
 324         DataSize64 += MAX_RESAMPLE_PADDING*2;
 325         SrcBufferSize = (ALsizei)mini64(DataSize64, BUFFERSIZE);
 326
 327         /* Figure out how many samples we can actually mix from this. */
 328         DataSize64  = SrcBufferSize;
 329         DataSize64 -= MAX_RESAMPLE_PADDING*2;
 330         DataSize64 <<= FRACTIONBITS;
 331         DataSize64 -= DataPosFrac;
 332         DstBufferSize = (ALsizei)mini64((DataSize64+(increment-1)) / increment,
 333                                         SamplesToDo - OutPos);
 334
 335         /* Some mixers like having a multiple of 4, so try to give that unless
 336          * this is the last update. */
 337         if(DstBufferSize < SamplesToDo-OutPos)
 338             DstBufferSize &= ~3;
 339
 340         /* It's impossible to have a buffer list item with no entries. */
 341         assert(BufferListItem->num_buffers > 0);
 342
 343         for(chan = 0;chan < NumChannels;chan++)
 344         {
 345             const ALfloat *ResampledData;
 346             ALfloat *SrcData = Device->TempBuffer[SOURCE_DATA_BUF];
 347             ALsizei FilledAmt;
 348
 349             /* Load the previous samples into the source data first, and clear the rest. */
 350             memcpy(SrcData, voice->PrevSamples[chan], MAX_RESAMPLE_PADDING*sizeof(ALfloat));
 351             memset(SrcData+MAX_RESAMPLE_PADDING, 0, (BUFFERSIZE-MAX_RESAMPLE_PADDING)*
 352                                                     sizeof(ALfloat));
 353             FilledAmt = MAX_RESAMPLE_PADDING;
 354
 355             if(isstatic)
 356             {
 357                 /* TODO: For static sources, loop points are taken from the
 358                  * first buffer (should be adjusted by any buffer offset, to
 359                  * possibly be added later).
 360                  */
 361                 const ALbuffer *Buffer0 = BufferListItem->buffers[0];
 362                 const ALsizei LoopStart = Buffer0->LoopStart;
 363                 const ALsizei LoopEnd   = Buffer0->LoopEnd;
 364                 const ALsizei LoopSize  = LoopEnd - LoopStart;
 365
 366                 /* If current pos is beyond the loop range, do not loop */
 367                 if(!BufferLoopItem || DataPosInt >= LoopEnd)
 368                 {
 369                     ALsizei SizeToDo = SrcBufferSize - FilledAmt;
 370                     ALsizei CompLen = 0;
 371                     ALsizei i;
 372
 373                     BufferLoopItem = NULL;
 374
 375                     for(i = 0;i < BufferListItem->num_buffers;i++)
 376                     {
 377                         const ALbuffer *buffer = BufferListItem->buffers[i];
 378                         const ALubyte *Data = buffer->data;
 379                         ALsizei DataSize;
 380
 381                         if(DataPosInt >= buffer->SampleLen)
 382                             continue;
 383
 384                         /* Load what's left to play from the buffer */
 385                         DataSize = mini(SizeToDo, buffer->SampleLen - DataPosInt);
 386                         CompLen = maxi(CompLen, DataSize);
 387
 388                         LoadSamples(&SrcData[FilledAmt],
 389                             &Data[(DataPosInt*NumChannels + chan)*SampleSize],
 390                             NumChannels, buffer->FmtType, DataSize
 391                         );
 392                     }
 393                     FilledAmt += CompLen;
 394                 }
 395                 else
 396                 {
 397                     ALsizei SizeToDo = mini(SrcBufferSize - FilledAmt, LoopEnd - DataPosInt);
 398                     ALsizei CompLen = 0;
 399                     ALsizei i;
 400
 401                     for(i = 0;i < BufferListItem->num_buffers;i++)
 402                     {
 403                         const ALbuffer *buffer = BufferListItem->buffers[i];
 404                         const ALubyte *Data = buffer->data;
 405                         ALsizei DataSize;
 406
 407                         if(DataPosInt >= buffer->SampleLen)
 408                             continue;
 409
 410                         /* Load what's left of this loop iteration */
 411                         DataSize = mini(SizeToDo, buffer->SampleLen - DataPosInt);
 412                         CompLen = maxi(CompLen, DataSize);
 413
 414                         LoadSamples(&SrcData[FilledAmt],
 415                             &Data[(DataPosInt*NumChannels + chan)*SampleSize],
 416                             NumChannels, buffer->FmtType, DataSize
 417                         );
 418                     }
 419                     FilledAmt += CompLen;
 420
 421                     while(SrcBufferSize > FilledAmt)
 422                     {
 423                         const ALsizei SizeToDo = mini(SrcBufferSize - FilledAmt, LoopSize);
 424
 425                         CompLen = 0;
 426                         for(i = 0;i < BufferListItem->num_buffers;i++)
 427                         {
 428                             const ALbuffer *buffer = BufferListItem->buffers[i];
 429                             const ALubyte *Data = buffer->data;
 430                             ALsizei DataSize;
 431
 432                             if(LoopStart >= buffer->SampleLen)
 433                                 continue;
 434
 435                             DataSize = mini(SizeToDo, buffer->SampleLen - LoopStart);
 436                             CompLen = maxi(CompLen, DataSize);
 437
 438                             LoadSamples(&SrcData[FilledAmt],
 439                                 &Data[(LoopStart*NumChannels + chan)*SampleSize],
 440                                 NumChannels, buffer->FmtType, DataSize
 441                             );
 442                         }
 443                         FilledAmt += CompLen;
 444                     }
 445                 }
 446             }
 447             else
 448             {
 449                 /* Crawl the buffer queue to fill in the temp buffer */
 450                 ALbufferlistitem *tmpiter = BufferListItem;
 451                 ALsizei pos = DataPosInt;
 452
 453                 while(tmpiter && SrcBufferSize > FilledAmt)
 454                 {
 455                     ALsizei SizeToDo = SrcBufferSize - FilledAmt;
 456                     ALsizei CompLen = 0;
 457                     ALsizei i;
 458
 459                     for(i = 0;i < tmpiter->num_buffers;i++)
 460                     {
 461                         const ALbuffer *ALBuffer = tmpiter->buffers[i];
 462                         ALsizei DataSize = ALBuffer ? ALBuffer->SampleLen : 0;
 463                         CompLen = maxi(CompLen, DataSize);
 464
 465                         if(DataSize > pos)
 466                         {
 467                             const ALubyte *Data = ALBuffer->data;
 468                             Data += (pos*NumChannels + chan)*SampleSize;
 469
 470                             DataSize = minu(SizeToDo, DataSize - pos);
 471                             LoadSamples(&SrcData[FilledAmt], Data, NumChannels,
 472                                         ALBuffer->FmtType, DataSize);
 473                         }
 474                     }
 475                     if(pos > CompLen)
 476                         pos -= CompLen;
 477                     else
 478                     {
 479                         FilledAmt += CompLen - pos;
 480                         pos = 0;
 481                     }
 482                     if(SrcBufferSize > FilledAmt)
 483                     {
 484                         tmpiter = ATOMIC_LOAD(&tmpiter->next, almemory_order_acquire);
 485                         if(!tmpiter) tmpiter = BufferLoopItem;
 486                     }
 487                 }
 488             }
 489
 490             /* Store the last source samples used for next time. */
 491             memcpy(voice->PrevSamples[chan],
 492                 &SrcData[(increment*DstBufferSize + DataPosFrac)>>FRACTIONBITS],
 493                 MAX_RESAMPLE_PADDING*sizeof(ALfloat)
 494             );
 495
 496             /* Now resample, then filter and mix to the appropriate outputs. */
 497             ResampledData = Resample(&voice->ResampleState,
 498                 &SrcData[MAX_RESAMPLE_PADDING], DataPosFrac, increment,
 499                 Device->TempBuffer[RESAMPLED_BUF], DstBufferSize
 500             );
 501             {
 502                 DirectParams *parms = &voice->Direct.Params[chan];
 503                 const ALfloat *samples;
 504
 505                 samples = DoFilters(
 506                     &parms->LowPass, &parms->HighPass, Device->TempBuffer[FILTERED_BUF],
 507                     ResampledData, DstBufferSize, voice->Direct.FilterType
 508                 );
 509                 if(!(voice->Flags&VOICE_HAS_HRTF))
 510                 {
 511                     if(!Counter)
 512                         memcpy(parms->Gains.Current, parms->Gains.Target,
 513                                sizeof(parms->Gains.Current));
 514                     if(!(voice->Flags&VOICE_HAS_NFC))
 515                         MixSamples(samples, voice->Direct.Channels, voice->Direct.Buffer,
 516                             parms->Gains.Current, parms->Gains.Target, Counter, OutPos,
 517                             DstBufferSize
 518                         );
 519                     else
 520                     {
 521                         ALfloat *nfcsamples = Device->TempBuffer[NFC_DATA_BUF];
 522                         ALsizei chanoffset = 0;
 523
 524                         MixSamples(samples,
 525                             voice->Direct.ChannelsPerOrder[0], voice->Direct.Buffer,
 526                             parms->Gains.Current, parms->Gains.Target, Counter, OutPos,
 527                             DstBufferSize
 528                         );
 529                         chanoffset += voice->Direct.ChannelsPerOrder[0];
 530 #define APPLY_NFC_MIX(order)                                                  \
 531     if(voice->Direct.ChannelsPerOrder[order] > 0)                             \
 532     {                                                                         \
 533         NfcFilterUpdate##order(&parms->NFCtrlFilter[order-1], nfcsamples,     \
 534                                samples, DstBufferSize);                       \
 535         MixSamples(nfcsamples, voice->Direct.ChannelsPerOrder[order],         \
 536             voice->Direct.Buffer+chanoffset, parms->Gains.Current+chanoffset, \
 537             parms->Gains.Target+chanoffset, Counter, OutPos, DstBufferSize    \
 538         );                                                                    \
 539         chanoffset += voice->Direct.ChannelsPerOrder[order];                  \
 540     }
 541                         APPLY_NFC_MIX(1)
 542                         APPLY_NFC_MIX(2)
 543                         APPLY_NFC_MIX(3)
 544 #undef APPLY_NFC_MIX
 545                     }
 546                 }
 547                 else
 548                 {
 549                     MixHrtfParams hrtfparams;
 550                     ALsizei fademix = 0;
 551                     int lidx, ridx;
 552
 553                     lidx = GetChannelIdxByName(&Device->RealOut, FrontLeft);
 554                     ridx = GetChannelIdxByName(&Device->RealOut, FrontRight);
 555                     assert(lidx != -1 && ridx != -1);
 556
 557                     if(!Counter)
 558                     {
 559                         /* No fading, just overwrite the old HRTF params. */
 560                         parms->Hrtf.Old = parms->Hrtf.Target;
 561                     }
 562                     else if(!(parms->Hrtf.Old.Gain > GAIN_SILENCE_THRESHOLD))
 563                     {
 564                         /* The old HRTF params are silent, so overwrite the old
 565                          * coefficients with the new, and reset the old gain to
 566                          * 0. The future mix will then fade from silence.
 567                          */
 568                         parms->Hrtf.Old = parms->Hrtf.Target;
 569                         parms->Hrtf.Old.Gain = 0.0f;
 570                     }
 571                     else if(firstpass)
 572                     {
 573                         ALfloat gain;
 574
 575                         /* Fade between the coefficients over 128 samples. */
 576                         fademix = mini(DstBufferSize, 128);
 577
 578                         /* The new coefficients need to fade in completely
 579                          * since they're replacing the old ones. To keep the
 580                          * gain fading consistent, interpolate between the old
 581                          * and new target gains given how much of the fade time
 582                          * this mix handles.
 583                          */
 584                         gain = lerp(parms->Hrtf.Old.Gain, parms->Hrtf.Target.Gain,
 585                                     minf(1.0f, (ALfloat)fademix/Counter));
 586                         hrtfparams.Coeffs = parms->Hrtf.Target.Coeffs;
 587                         hrtfparams.Delay[0] = parms->Hrtf.Target.Delay[0];
 588                         hrtfparams.Delay[1] = parms->Hrtf.Target.Delay[1];
 589                         hrtfparams.Gain = 0.0f;
 590                         hrtfparams.GainStep = gain / (ALfloat)fademix;
 591
 592                         MixHrtfBlendSamples(
 593                             voice->Direct.Buffer[lidx], voice->Direct.Buffer[ridx],
 594                             samples, voice->Offset, OutPos, IrSize, &parms->Hrtf.Old,
 595                             &hrtfparams, &parms->Hrtf.State, fademix
 596                         );
 597                         /* Update the old parameters with the result. */
 598                         parms->Hrtf.Old = parms->Hrtf.Target;
 599                         if(fademix < Counter)
 600                             parms->Hrtf.Old.Gain = hrtfparams.Gain;
 601                     }
 602
 603                     if(fademix < DstBufferSize)
 604                     {
 605                         ALsizei todo = DstBufferSize - fademix;
 606                         ALfloat gain = parms->Hrtf.Target.Gain;
 607
 608                         /* Interpolate the target gain if the gain fading lasts
 609                          * longer than this mix.
 610                          */
 611                         if(Counter > DstBufferSize)
 612                             gain = lerp(parms->Hrtf.Old.Gain, gain,
 613                                         (ALfloat)todo/(Counter-fademix));
 614
 615                         hrtfparams.Coeffs = parms->Hrtf.Target.Coeffs;
 616                         hrtfparams.Delay[0] = parms->Hrtf.Target.Delay[0];
 617                         hrtfparams.Delay[1] = parms->Hrtf.Target.Delay[1];
 618                         hrtfparams.Gain = parms->Hrtf.Old.Gain;
 619                         hrtfparams.GainStep = (gain - parms->Hrtf.Old.Gain) / (ALfloat)todo;
 620                         MixHrtfSamples(
 621                             voice->Direct.Buffer[lidx], voice->Direct.Buffer[ridx],
 622                             samples+fademix, voice->Offset+fademix, OutPos+fademix, IrSize,
 623                             &hrtfparams, &parms->Hrtf.State, todo
 624                         );
 625                         /* Store the interpolated gain or the final target gain
 626                          * depending if the fade is done.
 627                          */
 628                         if(DstBufferSize < Counter)
 629                             parms->Hrtf.Old.Gain = gain;
 630                         else
 631                             parms->Hrtf.Old.Gain = parms->Hrtf.Target.Gain;
 632                     }
 633                 }
 634             }
 635
 636             for(send = 0;send < Device->NumAuxSends;send++)
 637             {
 638                 SendParams *parms = &voice->Send[send].Params[chan];
 639                 const ALfloat *samples;
 640
 641                 if(!voice->Send[send].Buffer)
 642                     continue;
 643
 644                 samples = DoFilters(
 645                     &parms->LowPass, &parms->HighPass, Device->TempBuffer[FILTERED_BUF],
 646                     ResampledData, DstBufferSize, voice->Send[send].FilterType
 647                 );
 648
 649                 if(!Counter)
 650                     memcpy(parms->Gains.Current, parms->Gains.Target,
 651                            sizeof(parms->Gains.Current));
 652                 MixSamples(samples, voice->Send[send].Channels, voice->Send[send].Buffer,
 653                     parms->Gains.Current, parms->Gains.Target, Counter, OutPos, DstBufferSize
 654                 );
 655             }
 656         }
 657         /* Update positions */
 658         DataPosFrac += increment*DstBufferSize;
 659         DataPosInt  += DataPosFrac>>FRACTIONBITS;
 660         DataPosFrac &= FRACTIONMASK;
 661
 662         OutPos += DstBufferSize;
 663         voice->Offset += DstBufferSize;
 664         Counter = maxi(DstBufferSize, Counter) - DstBufferSize;
 665         firstpass = false;
 666
 667         /* Handle looping source position */
 668         if(isstatic && BufferLoopItem)
 669         {
 670             const ALbuffer *Buffer = BufferListItem->buffers[0];
 671             ALsizei LoopStart = Buffer->LoopStart;
 672             ALsizei LoopEnd = Buffer->LoopEnd;
 673             if(DataPosInt >= LoopEnd)
 674             {
 675                 assert(LoopEnd > LoopStart);
 676                 DataPosInt = ((DataPosInt-LoopStart)%(LoopEnd-LoopStart)) + LoopStart;
 677             }
 678         }
 679         else while(1)
 680         {
 681             /* Handle non-looping or buffer queue source position */
 682             ALsizei CompLen = 0;
 683             ALsizei i;
 684
 685             for(i = 0;i < BufferListItem->num_buffers;i++)
 686             {
 687                 const ALbuffer *buffer = BufferListItem->buffers[i];
 688                 if(buffer) CompLen = maxi(CompLen, buffer->SampleLen);
 689             }
 690
 691             if(CompLen > DataPosInt)
 692                 break;
 693
 694             BufferListItem = ATOMIC_LOAD(&BufferListItem->next, almemory_order_acquire);
 695             if(!BufferListItem)
 696             {
 697                 BufferListItem = BufferLoopItem;
 698                 if(!BufferListItem)
 699                 {
 700                     isplaying = false;
 701                     DataPosInt = 0;
 702                     DataPosFrac = 0;
 703                     break;
 704                 }
 705             }
 706
 707             DataPosInt -= CompLen;
 708         }
 709     } while(isplaying && OutPos < SamplesToDo);
 710
 711     voice->Flags |= VOICE_IS_FADING;
 712
 713     /* Update source info */
 714     ATOMIC_STORE(&voice->position,          DataPosInt, almemory_order_relaxed);
 715     ATOMIC_STORE(&voice->position_fraction, DataPosFrac, almemory_order_relaxed);
 716     ATOMIC_STORE(&voice->current_buffer,    BufferListItem, almemory_order_release);
 717     return isplaying;
 718 }