Alc/mixvoice.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 "sample_cvt.h"
  37 #include "alu.h"
  38 #include "alconfig.h"
  39 #include "ringbuffer.h"
  40
  41 #include "cpu_caps.h"
  42 #include "mixer/defs.h"
  43
  44
  45 static_assert((INT_MAX>>FRACTIONBITS)/MAX_PITCH > BUFFERSIZE,
  46               "MAX_PITCH and/or BUFFERSIZE are too large for FRACTIONBITS!");
  47
  48 extern inline void InitiatePositionArrays(ALsizei frac, ALint increment, ALsizei *restrict frac_arr, ALint *restrict pos_arr, ALsizei size);
  49
  50
  51 /* BSinc24 requires up to 23 extra samples before the current position, and 24 after. */
  52 static_assert(MAX_RESAMPLE_PADDING >= 24, "MAX_RESAMPLE_PADDING must be at least 24!");
  53
  54
  55 enum Resampler ResamplerDefault = LinearResampler;
  56
  57 MixerFunc MixSamples = Mix_C;
  58 RowMixerFunc MixRowSamples = MixRow_C;
  59 static HrtfMixerFunc MixHrtfSamples = MixHrtf_C;
  60 static HrtfMixerBlendFunc MixHrtfBlendSamples = MixHrtfBlend_C;
  61
  62 static MixerFunc SelectMixer(void)
  63 {
  64 #ifdef HAVE_NEON
  65     if((CPUCapFlags&CPU_CAP_NEON))
  66         return Mix_Neon;
  67 #endif
  68 #ifdef HAVE_SSE
  69     if((CPUCapFlags&CPU_CAP_SSE))
  70         return Mix_SSE;
  71 #endif
  72     return Mix_C;
  73 }
  74
  75 static RowMixerFunc SelectRowMixer(void)
  76 {
  77 #ifdef HAVE_NEON
  78     if((CPUCapFlags&CPU_CAP_NEON))
  79         return MixRow_Neon;
  80 #endif
  81 #ifdef HAVE_SSE
  82     if((CPUCapFlags&CPU_CAP_SSE))
  83         return MixRow_SSE;
  84 #endif
  85     return MixRow_C;
  86 }
  87
  88 static inline HrtfMixerFunc SelectHrtfMixer(void)
  89 {
  90 #ifdef HAVE_NEON
  91     if((CPUCapFlags&CPU_CAP_NEON))
  92         return MixHrtf_Neon;
  93 #endif
  94 #ifdef HAVE_SSE
  95     if((CPUCapFlags&CPU_CAP_SSE))
  96         return MixHrtf_SSE;
  97 #endif
  98     return MixHrtf_C;
  99 }
 100
 101 static inline HrtfMixerBlendFunc SelectHrtfBlendMixer(void)
 102 {
 103 #ifdef HAVE_NEON
 104     if((CPUCapFlags&CPU_CAP_NEON))
 105         return MixHrtfBlend_Neon;
 106 #endif
 107 #ifdef HAVE_SSE
 108     if((CPUCapFlags&CPU_CAP_SSE))
 109         return MixHrtfBlend_SSE;
 110 #endif
 111     return MixHrtfBlend_C;
 112 }
 113
 114 ResamplerFunc SelectResampler(enum Resampler resampler)
 115 {
 116     switch(resampler)
 117     {
 118         case PointResampler:
 119             return Resample_point_C;
 120         case LinearResampler:
 121 #ifdef HAVE_NEON
 122             if((CPUCapFlags&CPU_CAP_NEON))
 123                 return Resample_lerp_Neon;
 124 #endif
 125 #ifdef HAVE_SSE4_1
 126             if((CPUCapFlags&CPU_CAP_SSE4_1))
 127                 return Resample_lerp_SSE41;
 128 #endif
 129 #ifdef HAVE_SSE2
 130             if((CPUCapFlags&CPU_CAP_SSE2))
 131                 return Resample_lerp_SSE2;
 132 #endif
 133             return Resample_lerp_C;
 134         case FIR4Resampler:
 135             return Resample_cubic_C;
 136         case BSinc12Resampler:
 137         case BSinc24Resampler:
 138 #ifdef HAVE_NEON
 139             if((CPUCapFlags&CPU_CAP_NEON))
 140                 return Resample_bsinc_Neon;
 141 #endif
 142 #ifdef HAVE_SSE
 143             if((CPUCapFlags&CPU_CAP_SSE))
 144                 return Resample_bsinc_SSE;
 145 #endif
 146             return Resample_bsinc_C;
 147     }
 148
 149     return Resample_point_C;
 150 }
 151
 152
 153 void aluInitMixer(void)
 154 {
 155     const char *str;
 156
 157     if(ConfigValueStr(NULL, NULL, "resampler", &str))
 158     {
 159         if(strcasecmp(str, "point") == 0 || strcasecmp(str, "none") == 0)
 160             ResamplerDefault = PointResampler;
 161         else if(strcasecmp(str, "linear") == 0)
 162             ResamplerDefault = LinearResampler;
 163         else if(strcasecmp(str, "cubic") == 0)
 164             ResamplerDefault = FIR4Resampler;
 165         else if(strcasecmp(str, "bsinc12") == 0)
 166             ResamplerDefault = BSinc12Resampler;
 167         else if(strcasecmp(str, "bsinc24") == 0)
 168             ResamplerDefault = BSinc24Resampler;
 169         else if(strcasecmp(str, "bsinc") == 0)
 170         {
 171             WARN("Resampler option \"%s\" is deprecated, using bsinc12\n", str);
 172             ResamplerDefault = BSinc12Resampler;
 173         }
 174         else if(strcasecmp(str, "sinc4") == 0 || strcasecmp(str, "sinc8") == 0)
 175         {
 176             WARN("Resampler option \"%s\" is deprecated, using cubic\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     MixRowSamples = SelectRowMixer();
 194 }
 195
 196
 197 static void SendAsyncEvent(ALCcontext *context, ALuint enumtype, ALenum type,
 198                            ALuint objid, ALuint param, const char *msg)
 199 {
 200     AsyncEvent evt;
 201     evt.EnumType = enumtype;
 202     evt.Type = type;
 203     evt.ObjectId = objid;
 204     evt.Param = param;
 205     strcpy(evt.Message, msg);
 206     if(ll_ringbuffer_write(context->AsyncEvents, (const char*)&evt, 1) == 1)
 207         alsem_post(&context->EventSem);
 208 }
 209
 210
 211 static inline ALfloat Sample_ALubyte(ALubyte val)
 212 { return (val-128) * (1.0f/128.0f); }
 213
 214 static inline ALfloat Sample_ALshort(ALshort val)
 215 { return val * (1.0f/32768.0f); }
 216
 217 static inline ALfloat Sample_ALfloat(ALfloat val)
 218 { return val; }
 219
 220 static inline ALfloat Sample_ALdouble(ALdouble val)
 221 { return (ALfloat)val; }
 222
 223 typedef ALubyte ALmulaw;
 224 static inline ALfloat Sample_ALmulaw(ALmulaw val)
 225 { return muLawDecompressionTable[val] * (1.0f/32768.0f); }
 226
 227 typedef ALubyte ALalaw;
 228 static inline ALfloat Sample_ALalaw(ALalaw val)
 229 { return aLawDecompressionTable[val] * (1.0f/32768.0f); }
 230
 231 #define DECL_TEMPLATE(T)                                                      \
 232 static inline void Load_##T(ALfloat *restrict dst, const T *restrict src,     \
 233                             ALint srcstep, ALsizei samples)                   \
 234 {                                                                             \
 235     ALsizei i;                                                                \
 236     for(i = 0;i < samples;i++)                                                \
 237         dst[i] += Sample_##T(src[i*srcstep]);                                 \
 238 }
 239
 240 DECL_TEMPLATE(ALubyte)
 241 DECL_TEMPLATE(ALshort)
 242 DECL_TEMPLATE(ALfloat)
 243 DECL_TEMPLATE(ALdouble)
 244 DECL_TEMPLATE(ALmulaw)
 245 DECL_TEMPLATE(ALalaw)
 246
 247 #undef DECL_TEMPLATE
 248
 249 static void LoadSamples(ALfloat *restrict dst, const ALvoid *restrict src, ALint srcstep,
 250                         enum FmtType srctype, ALsizei samples)
 251 {
 252 #define HANDLE_FMT(ET, ST) case ET: Load_##ST(dst, src, srcstep, samples); break
 253     switch(srctype)
 254     {
 255         HANDLE_FMT(FmtUByte, ALubyte);
 256         HANDLE_FMT(FmtShort, ALshort);
 257         HANDLE_FMT(FmtFloat, ALfloat);
 258         HANDLE_FMT(FmtDouble, ALdouble);
 259         HANDLE_FMT(FmtMulaw, ALmulaw);
 260         HANDLE_FMT(FmtAlaw, ALalaw);
 261     }
 262 #undef HANDLE_FMT
 263 }
 264
 265
 266 static const ALfloat *DoFilters(BiquadFilter *lpfilter, BiquadFilter *hpfilter,
 267                                 ALfloat *restrict dst, const ALfloat *restrict src,
 268                                 ALsizei numsamples, enum ActiveFilters type)
 269 {
 270     ALsizei i;
 271     switch(type)
 272     {
 273         case AF_None:
 274             BiquadFilter_passthru(lpfilter, numsamples);
 275             BiquadFilter_passthru(hpfilter, numsamples);
 276             break;
 277
 278         case AF_LowPass:
 279             BiquadFilter_process(lpfilter, dst, src, numsamples);
 280             BiquadFilter_passthru(hpfilter, numsamples);
 281             return dst;
 282         case AF_HighPass:
 283             BiquadFilter_passthru(lpfilter, numsamples);
 284             BiquadFilter_process(hpfilter, dst, src, numsamples);
 285             return dst;
 286
 287         case AF_BandPass:
 288             for(i = 0;i < numsamples;)
 289             {
 290                 ALfloat temp[256];
 291                 ALsizei todo = mini(256, numsamples-i);
 292
 293                 BiquadFilter_process(lpfilter, temp, src+i, todo);
 294                 BiquadFilter_process(hpfilter, dst+i, temp, todo);
 295                 i += todo;
 296             }
 297             return dst;
 298     }
 299     return src;
 300 }
 301
 302
 303 /* This function uses these device temp buffers. */
 304 #define SOURCE_DATA_BUF 0
 305 #define RESAMPLED_BUF 1
 306 #define FILTERED_BUF 2
 307 #define NFC_DATA_BUF 3
 308 ALboolean MixSource(ALvoice *voice, ALuint SourceID, ALCcontext *Context, ALsizei SamplesToDo)
 309 {
 310     ALCdevice *Device = Context->Device;
 311     ALbufferlistitem *BufferListItem;
 312     ALbufferlistitem *BufferLoopItem;
 313     ALsizei NumChannels, SampleSize;
 314     ALbitfieldSOFT enabledevt;
 315     ALsizei buffers_done = 0;
 316     ResamplerFunc Resample;
 317     ALsizei DataPosInt;
 318     ALsizei DataPosFrac;
 319     ALint64 DataSize64;
 320     ALint increment;
 321     ALsizei Counter;
 322     ALsizei OutPos;
 323     ALsizei IrSize;
 324     bool isplaying;
 325     bool firstpass;
 326     bool isstatic;
 327     ALsizei chan;
 328     ALsizei send;
 329
 330     /* Get source info */
 331     isplaying      = true; /* Will only be called while playing. */
 332     isstatic       = !!(voice->Flags&VOICE_IS_STATIC);
 333     DataPosInt     = ATOMIC_LOAD(&voice->position, almemory_order_acquire);
 334     DataPosFrac    = ATOMIC_LOAD(&voice->position_fraction, almemory_order_relaxed);
 335     BufferListItem = ATOMIC_LOAD(&voice->current_buffer, almemory_order_relaxed);
 336     BufferLoopItem = ATOMIC_LOAD(&voice->loop_buffer, almemory_order_relaxed);
 337     NumChannels    = voice->NumChannels;
 338     SampleSize     = voice->SampleSize;
 339     increment      = voice->Step;
 340
 341     IrSize = (Device->HrtfHandle ? Device->HrtfHandle->irSize : 0);
 342
 343     Resample = ((increment == FRACTIONONE && DataPosFrac == 0) ?
 344                 Resample_copy_C : voice->Resampler);
 345
 346     Counter = (voice->Flags&VOICE_IS_FADING) ? SamplesToDo : 0;
 347     firstpass = true;
 348     OutPos = 0;
 349
 350     do {
 351         ALsizei SrcBufferSize, DstBufferSize;
 352
 353         /* Figure out how many buffer samples will be needed */
 354         DataSize64  = SamplesToDo-OutPos;
 355         DataSize64 *= increment;
 356         DataSize64 += DataPosFrac+FRACTIONMASK;
 357         DataSize64 >>= FRACTIONBITS;
 358         DataSize64 += MAX_RESAMPLE_PADDING*2;
 359         SrcBufferSize = (ALsizei)mini64(DataSize64, BUFFERSIZE);
 360
 361         /* Figure out how many samples we can actually mix from this. */
 362         DataSize64  = SrcBufferSize;
 363         DataSize64 -= MAX_RESAMPLE_PADDING*2;
 364         DataSize64 <<= FRACTIONBITS;
 365         DataSize64 -= DataPosFrac;
 366         DstBufferSize = (ALsizei)mini64((DataSize64+(increment-1)) / increment,
 367                                         SamplesToDo - OutPos);
 368
 369         /* Some mixers like having a multiple of 4, so try to give that unless
 370          * this is the last update. */
 371         if(DstBufferSize < SamplesToDo-OutPos)
 372             DstBufferSize &= ~3;
 373
 374         /* It's impossible to have a buffer list item with no entries. */
 375         assert(BufferListItem->num_buffers > 0);
 376
 377         for(chan = 0;chan < NumChannels;chan++)
 378         {
 379             const ALfloat *ResampledData;
 380             ALfloat *SrcData = Device->TempBuffer[SOURCE_DATA_BUF];
 381             ALsizei FilledAmt;
 382
 383             /* Load the previous samples into the source data first, and clear the rest. */
 384             memcpy(SrcData, voice->PrevSamples[chan], MAX_RESAMPLE_PADDING*sizeof(ALfloat));
 385             memset(SrcData+MAX_RESAMPLE_PADDING, 0, (BUFFERSIZE-MAX_RESAMPLE_PADDING)*
 386                                                     sizeof(ALfloat));
 387             FilledAmt = MAX_RESAMPLE_PADDING;
 388
 389             if(isstatic)
 390             {
 391                 /* TODO: For static sources, loop points are taken from the
 392                  * first buffer (should be adjusted by any buffer offset, to
 393                  * possibly be added later).
 394                  */
 395                 const ALbuffer *Buffer0 = BufferListItem->buffers[0];
 396                 const ALsizei LoopStart = Buffer0->LoopStart;
 397                 const ALsizei LoopEnd   = Buffer0->LoopEnd;
 398                 const ALsizei LoopSize  = LoopEnd - LoopStart;
 399
 400                 /* If current pos is beyond the loop range, do not loop */
 401                 if(!BufferLoopItem || DataPosInt >= LoopEnd)
 402                 {
 403                     ALsizei SizeToDo = SrcBufferSize - FilledAmt;
 404                     ALsizei CompLen = 0;
 405                     ALsizei i;
 406
 407                     BufferLoopItem = NULL;
 408
 409                     for(i = 0;i < BufferListItem->num_buffers;i++)
 410                     {
 411                         const ALbuffer *buffer = BufferListItem->buffers[i];
 412                         const ALubyte *Data = buffer->data;
 413                         ALsizei DataSize;
 414
 415                         if(DataPosInt >= buffer->SampleLen)
 416                             continue;
 417
 418                         /* Load what's left to play from the buffer */
 419                         DataSize = mini(SizeToDo, buffer->SampleLen - DataPosInt);
 420                         CompLen = maxi(CompLen, DataSize);
 421
 422                         LoadSamples(&SrcData[FilledAmt],
 423                             &Data[(DataPosInt*NumChannels + chan)*SampleSize],
 424                             NumChannels, buffer->FmtType, DataSize
 425                         );
 426                     }
 427                     FilledAmt += CompLen;
 428                 }
 429                 else
 430                 {
 431                     ALsizei SizeToDo = mini(SrcBufferSize - FilledAmt, LoopEnd - DataPosInt);
 432                     ALsizei CompLen = 0;
 433                     ALsizei i;
 434
 435                     for(i = 0;i < BufferListItem->num_buffers;i++)
 436                     {
 437                         const ALbuffer *buffer = BufferListItem->buffers[i];
 438                         const ALubyte *Data = buffer->data;
 439                         ALsizei DataSize;
 440
 441                         if(DataPosInt >= buffer->SampleLen)
 442                             continue;
 443
 444                         /* Load what's left of this loop iteration */
 445                         DataSize = mini(SizeToDo, buffer->SampleLen - DataPosInt);
 446                         CompLen = maxi(CompLen, DataSize);
 447
 448                         LoadSamples(&SrcData[FilledAmt],
 449                             &Data[(DataPosInt*NumChannels + chan)*SampleSize],
 450                             NumChannels, buffer->FmtType, DataSize
 451                         );
 452                     }
 453                     FilledAmt += CompLen;
 454
 455                     while(SrcBufferSize > FilledAmt)
 456                     {
 457                         const ALsizei SizeToDo = mini(SrcBufferSize - FilledAmt, LoopSize);
 458
 459                         CompLen = 0;
 460                         for(i = 0;i < BufferListItem->num_buffers;i++)
 461                         {
 462                             const ALbuffer *buffer = BufferListItem->buffers[i];
 463                             const ALubyte *Data = buffer->data;
 464                             ALsizei DataSize;
 465
 466                             if(LoopStart >= buffer->SampleLen)
 467                                 continue;
 468
 469                             DataSize = mini(SizeToDo, buffer->SampleLen - LoopStart);
 470                             CompLen = maxi(CompLen, DataSize);
 471
 472                             LoadSamples(&SrcData[FilledAmt],
 473                                 &Data[(LoopStart*NumChannels + chan)*SampleSize],
 474                                 NumChannels, buffer->FmtType, DataSize
 475                             );
 476                         }
 477                         FilledAmt += CompLen;
 478                     }
 479                 }
 480             }
 481             else
 482             {
 483                 /* Crawl the buffer queue to fill in the temp buffer */
 484                 ALbufferlistitem *tmpiter = BufferListItem;
 485                 ALsizei pos = DataPosInt;
 486
 487                 while(tmpiter && SrcBufferSize > FilledAmt)
 488                 {
 489                     ALsizei SizeToDo = SrcBufferSize - FilledAmt;
 490                     ALsizei i;
 491
 492                     for(i = 0;i < tmpiter->num_buffers;i++)
 493                     {
 494                         const ALbuffer *ALBuffer = tmpiter->buffers[i];
 495                         ALsizei DataSize = ALBuffer ? ALBuffer->SampleLen : 0;
 496
 497                         if(DataSize > pos)
 498                         {
 499                             const ALubyte *Data = ALBuffer->data;
 500                             Data += (pos*NumChannels + chan)*SampleSize;
 501
 502                             DataSize = minu(SizeToDo, DataSize - pos);
 503                             LoadSamples(&SrcData[FilledAmt], Data, NumChannels,
 504                                         ALBuffer->FmtType, DataSize);
 505                         }
 506                     }
 507                     if(pos > tmpiter->max_samples)
 508                         pos -= tmpiter->max_samples;
 509                     else
 510                     {
 511                         FilledAmt += tmpiter->max_samples - pos;
 512                         pos = 0;
 513                     }
 514                     if(SrcBufferSize > FilledAmt)
 515                     {
 516                         tmpiter = ATOMIC_LOAD(&tmpiter->next, almemory_order_acquire);
 517                         if(!tmpiter) tmpiter = BufferLoopItem;
 518                     }
 519                 }
 520             }
 521
 522             /* Store the last source samples used for next time. */
 523             memcpy(voice->PrevSamples[chan],
 524                 &SrcData[(increment*DstBufferSize + DataPosFrac)>>FRACTIONBITS],
 525                 MAX_RESAMPLE_PADDING*sizeof(ALfloat)
 526             );
 527
 528             /* Now resample, then filter and mix to the appropriate outputs. */
 529             ResampledData = Resample(&voice->ResampleState,
 530                 &SrcData[MAX_RESAMPLE_PADDING], DataPosFrac, increment,
 531                 Device->TempBuffer[RESAMPLED_BUF], DstBufferSize
 532             );
 533             {
 534                 DirectParams *parms = &voice->Direct.Params[chan];
 535                 const ALfloat *samples;
 536
 537                 samples = DoFilters(
 538                     &parms->LowPass, &parms->HighPass, Device->TempBuffer[FILTERED_BUF],
 539                     ResampledData, DstBufferSize, voice->Direct.FilterType
 540                 );
 541                 if(!(voice->Flags&VOICE_HAS_HRTF))
 542                 {
 543                     if(!Counter)
 544                         memcpy(parms->Gains.Current, parms->Gains.Target,
 545                                sizeof(parms->Gains.Current));
 546                     if(!(voice->Flags&VOICE_HAS_NFC))
 547                         MixSamples(samples, voice->Direct.Channels, voice->Direct.Buffer,
 548                             parms->Gains.Current, parms->Gains.Target, Counter, OutPos,
 549                             DstBufferSize
 550                         );
 551                     else
 552                     {
 553                         ALfloat *nfcsamples = Device->TempBuffer[NFC_DATA_BUF];
 554                         ALsizei chanoffset = 0;
 555
 556                         MixSamples(samples,
 557                             voice->Direct.ChannelsPerOrder[0], voice->Direct.Buffer,
 558                             parms->Gains.Current, parms->Gains.Target, Counter, OutPos,
 559                             DstBufferSize
 560                         );
 561                         chanoffset += voice->Direct.ChannelsPerOrder[0];
 562 #define APPLY_NFC_MIX(order)                                                  \
 563     if(voice->Direct.ChannelsPerOrder[order] > 0)                             \
 564     {                                                                         \
 565         NfcFilterProcess##order(&parms->NFCtrlFilter, nfcsamples, samples,    \
 566                                DstBufferSize);                                \
 567         MixSamples(nfcsamples, voice->Direct.ChannelsPerOrder[order],         \
 568             voice->Direct.Buffer+chanoffset, parms->Gains.Current+chanoffset, \
 569             parms->Gains.Target+chanoffset, Counter, OutPos, DstBufferSize    \
 570         );                                                                    \
 571         chanoffset += voice->Direct.ChannelsPerOrder[order];                  \
 572     }
 573                         APPLY_NFC_MIX(1)
 574                         APPLY_NFC_MIX(2)
 575                         APPLY_NFC_MIX(3)
 576 #undef APPLY_NFC_MIX
 577                     }
 578                 }
 579                 else
 580                 {
 581                     MixHrtfParams hrtfparams;
 582                     ALsizei fademix = 0;
 583                     int lidx, ridx;
 584
 585                     lidx = GetChannelIdxByName(&Device->RealOut, FrontLeft);
 586                     ridx = GetChannelIdxByName(&Device->RealOut, FrontRight);
 587                     assert(lidx != -1 && ridx != -1);
 588
 589                     if(!Counter)
 590                     {
 591                         /* No fading, just overwrite the old HRTF params. */
 592                         parms->Hrtf.Old = parms->Hrtf.Target;
 593                     }
 594                     else if(!(parms->Hrtf.Old.Gain > GAIN_SILENCE_THRESHOLD))
 595                     {
 596                         /* The old HRTF params are silent, so overwrite the old
 597                          * coefficients with the new, and reset the old gain to
 598                          * 0. The future mix will then fade from silence.
 599                          */
 600                         parms->Hrtf.Old = parms->Hrtf.Target;
 601                         parms->Hrtf.Old.Gain = 0.0f;
 602                     }
 603                     else if(firstpass)
 604                     {
 605                         ALfloat gain;
 606
 607                         /* Fade between the coefficients over 128 samples. */
 608                         fademix = mini(DstBufferSize, 128);
 609
 610                         /* The new coefficients need to fade in completely
 611                          * since they're replacing the old ones. To keep the
 612                          * gain fading consistent, interpolate between the old
 613                          * and new target gains given how much of the fade time
 614                          * this mix handles.
 615                          */
 616                         gain = lerp(parms->Hrtf.Old.Gain, parms->Hrtf.Target.Gain,
 617                                     minf(1.0f, (ALfloat)fademix/Counter));
 618                         hrtfparams.Coeffs = parms->Hrtf.Target.Coeffs;
 619                         hrtfparams.Delay[0] = parms->Hrtf.Target.Delay[0];
 620                         hrtfparams.Delay[1] = parms->Hrtf.Target.Delay[1];
 621                         hrtfparams.Gain = 0.0f;
 622                         hrtfparams.GainStep = gain / (ALfloat)fademix;
 623
 624                         MixHrtfBlendSamples(
 625                             voice->Direct.Buffer[lidx], voice->Direct.Buffer[ridx],
 626                             samples, voice->Offset, OutPos, IrSize, &parms->Hrtf.Old,
 627                             &hrtfparams, &parms->Hrtf.State, fademix
 628                         );
 629                         /* Update the old parameters with the result. */
 630                         parms->Hrtf.Old = parms->Hrtf.Target;
 631                         if(fademix < Counter)
 632                             parms->Hrtf.Old.Gain = hrtfparams.Gain;
 633                     }
 634
 635                     if(fademix < DstBufferSize)
 636                     {
 637                         ALsizei todo = DstBufferSize - fademix;
 638                         ALfloat gain = parms->Hrtf.Target.Gain;
 639
 640                         /* Interpolate the target gain if the gain fading lasts
 641                          * longer than this mix.
 642                          */
 643                         if(Counter > DstBufferSize)
 644                             gain = lerp(parms->Hrtf.Old.Gain, gain,
 645                                         (ALfloat)todo/(Counter-fademix));
 646
 647                         hrtfparams.Coeffs = parms->Hrtf.Target.Coeffs;
 648                         hrtfparams.Delay[0] = parms->Hrtf.Target.Delay[0];
 649                         hrtfparams.Delay[1] = parms->Hrtf.Target.Delay[1];
 650                         hrtfparams.Gain = parms->Hrtf.Old.Gain;
 651                         hrtfparams.GainStep = (gain - parms->Hrtf.Old.Gain) / (ALfloat)todo;
 652                         MixHrtfSamples(
 653                             voice->Direct.Buffer[lidx], voice->Direct.Buffer[ridx],
 654                             samples+fademix, voice->Offset+fademix, OutPos+fademix, IrSize,
 655                             &hrtfparams, &parms->Hrtf.State, todo
 656                         );
 657                         /* Store the interpolated gain or the final target gain
 658                          * depending if the fade is done.
 659                          */
 660                         if(DstBufferSize < Counter)
 661                             parms->Hrtf.Old.Gain = gain;
 662                         else
 663                             parms->Hrtf.Old.Gain = parms->Hrtf.Target.Gain;
 664                     }
 665                 }
 666             }
 667
 668             for(send = 0;send < Device->NumAuxSends;send++)
 669             {
 670                 SendParams *parms = &voice->Send[send].Params[chan];
 671                 const ALfloat *samples;
 672
 673                 if(!voice->Send[send].Buffer)
 674                     continue;
 675
 676                 samples = DoFilters(
 677                     &parms->LowPass, &parms->HighPass, Device->TempBuffer[FILTERED_BUF],
 678                     ResampledData, DstBufferSize, voice->Send[send].FilterType
 679                 );
 680
 681                 if(!Counter)
 682                     memcpy(parms->Gains.Current, parms->Gains.Target,
 683                            sizeof(parms->Gains.Current));
 684                 MixSamples(samples, voice->Send[send].Channels, voice->Send[send].Buffer,
 685                     parms->Gains.Current, parms->Gains.Target, Counter, OutPos, DstBufferSize
 686                 );
 687             }
 688         }
 689         /* Update positions */
 690         DataPosFrac += increment*DstBufferSize;
 691         DataPosInt  += DataPosFrac>>FRACTIONBITS;
 692         DataPosFrac &= FRACTIONMASK;
 693
 694         OutPos += DstBufferSize;
 695         voice->Offset += DstBufferSize;
 696         Counter = maxi(DstBufferSize, Counter) - DstBufferSize;
 697         firstpass = false;
 698
 699         if(isstatic)
 700         {
 701             if(BufferLoopItem)
 702             {
 703                 /* Handle looping static source */
 704                 const ALbuffer *Buffer = BufferListItem->buffers[0];
 705                 ALsizei LoopStart = Buffer->LoopStart;
 706                 ALsizei LoopEnd = Buffer->LoopEnd;
 707                 if(DataPosInt >= LoopEnd)
 708                 {
 709                     assert(LoopEnd > LoopStart);
 710                     DataPosInt = ((DataPosInt-LoopStart)%(LoopEnd-LoopStart)) + LoopStart;
 711                 }
 712             }
 713             else
 714             {
 715                 /* Handle non-looping static source */
 716                 if(DataPosInt >= BufferListItem->max_samples)
 717                 {
 718                     isplaying = false;
 719                     BufferListItem = NULL;
 720                     DataPosInt = 0;
 721                     DataPosFrac = 0;
 722                     break;
 723                 }
 724             }
 725         }
 726         else while(1)
 727         {
 728             /* Handle streaming source */
 729             if(BufferListItem->max_samples > DataPosInt)
 730                 break;
 731
 732             buffers_done += BufferListItem->num_buffers;
 733             BufferListItem = ATOMIC_LOAD(&BufferListItem->next, almemory_order_acquire);
 734             if(!BufferListItem && !(BufferListItem=BufferLoopItem))
 735             {
 736                 isplaying = false;
 737                 DataPosInt = 0;
 738                 DataPosFrac = 0;
 739                 break;
 740             }
 741
 742             DataPosInt -= BufferListItem->max_samples;
 743         }
 744     } while(isplaying && OutPos < SamplesToDo);
 745
 746     voice->Flags |= VOICE_IS_FADING;
 747
 748     /* Update source info */
 749     ATOMIC_STORE(&voice->position,          DataPosInt, almemory_order_relaxed);
 750     ATOMIC_STORE(&voice->position_fraction, DataPosFrac, almemory_order_relaxed);
 751     ATOMIC_STORE(&voice->current_buffer,    BufferListItem, almemory_order_release);
 752
 753     /* Send any events now, after the position/buffer info was updated. */
 754     enabledevt = ATOMIC_LOAD(&Context->EnabledEvts, almemory_order_acquire);
 755     if(buffers_done > 0 && (enabledevt&EventType_BufferCompleted))
 756         SendAsyncEvent(Context, EventType_BufferCompleted,
 757             AL_EVENT_TYPE_BUFFER_COMPLETED_SOFT, SourceID, buffers_done, "Buffer completed"
 758         );
 759
 760     return isplaying;
 761 }