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(BiquadState *lpfilter, BiquadState *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             BiquadState_processPassthru(lpfilter, src, numsamples);
 275             BiquadState_processPassthru(hpfilter, src, numsamples);
 276             break;
 277
 278         case AF_LowPass:
 279             BiquadState_process(lpfilter, dst, src, numsamples);
 280             BiquadState_processPassthru(hpfilter, dst, numsamples);
 281             return dst;
 282         case AF_HighPass:
 283             BiquadState_processPassthru(lpfilter, src, numsamples);
 284             BiquadState_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                 BiquadState_process(lpfilter, temp, src+i, todo);
 294                 BiquadState_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 CompLen = 0;
 491                     ALsizei i;
 492
 493                     for(i = 0;i < tmpiter->num_buffers;i++)
 494                     {
 495                         const ALbuffer *ALBuffer = tmpiter->buffers[i];
 496                         ALsizei DataSize = ALBuffer ? ALBuffer->SampleLen : 0;
 497                         CompLen = maxi(CompLen, DataSize);
 498
 499                         if(DataSize > pos)
 500                         {
 501                             const ALubyte *Data = ALBuffer->data;
 502                             Data += (pos*NumChannels + chan)*SampleSize;
 503
 504                             DataSize = minu(SizeToDo, DataSize - pos);
 505                             LoadSamples(&SrcData[FilledAmt], Data, NumChannels,
 506                                         ALBuffer->FmtType, DataSize);
 507                         }
 508                     }
 509                     if(pos > CompLen)
 510                         pos -= CompLen;
 511                     else
 512                     {
 513                         FilledAmt += CompLen - pos;
 514                         pos = 0;
 515                     }
 516                     if(SrcBufferSize > FilledAmt)
 517                     {
 518                         tmpiter = ATOMIC_LOAD(&tmpiter->next, almemory_order_acquire);
 519                         if(!tmpiter) tmpiter = BufferLoopItem;
 520                     }
 521                 }
 522             }
 523
 524             /* Store the last source samples used for next time. */
 525             memcpy(voice->PrevSamples[chan],
 526                 &SrcData[(increment*DstBufferSize + DataPosFrac)>>FRACTIONBITS],
 527                 MAX_RESAMPLE_PADDING*sizeof(ALfloat)
 528             );
 529
 530             /* Now resample, then filter and mix to the appropriate outputs. */
 531             ResampledData = Resample(&voice->ResampleState,
 532                 &SrcData[MAX_RESAMPLE_PADDING], DataPosFrac, increment,
 533                 Device->TempBuffer[RESAMPLED_BUF], DstBufferSize
 534             );
 535             {
 536                 DirectParams *parms = &voice->Direct.Params[chan];
 537                 const ALfloat *samples;
 538
 539                 samples = DoFilters(
 540                     &parms->LowPass, &parms->HighPass, Device->TempBuffer[FILTERED_BUF],
 541                     ResampledData, DstBufferSize, voice->Direct.FilterType
 542                 );
 543                 if(!(voice->Flags&VOICE_HAS_HRTF))
 544                 {
 545                     if(!Counter)
 546                         memcpy(parms->Gains.Current, parms->Gains.Target,
 547                                sizeof(parms->Gains.Current));
 548                     if(!(voice->Flags&VOICE_HAS_NFC))
 549                         MixSamples(samples, voice->Direct.Channels, voice->Direct.Buffer,
 550                             parms->Gains.Current, parms->Gains.Target, Counter, OutPos,
 551                             DstBufferSize
 552                         );
 553                     else
 554                     {
 555                         ALfloat *nfcsamples = Device->TempBuffer[NFC_DATA_BUF];
 556                         ALsizei chanoffset = 0;
 557
 558                         MixSamples(samples,
 559                             voice->Direct.ChannelsPerOrder[0], voice->Direct.Buffer,
 560                             parms->Gains.Current, parms->Gains.Target, Counter, OutPos,
 561                             DstBufferSize
 562                         );
 563                         chanoffset += voice->Direct.ChannelsPerOrder[0];
 564 #define APPLY_NFC_MIX(order)                                                  \
 565     if(voice->Direct.ChannelsPerOrder[order] > 0)                             \
 566     {                                                                         \
 567         NfcFilterProcess##order(&parms->NFCtrlFilter, nfcsamples, samples,    \
 568                                DstBufferSize);                                \
 569         MixSamples(nfcsamples, voice->Direct.ChannelsPerOrder[order],         \
 570             voice->Direct.Buffer+chanoffset, parms->Gains.Current+chanoffset, \
 571             parms->Gains.Target+chanoffset, Counter, OutPos, DstBufferSize    \
 572         );                                                                    \
 573         chanoffset += voice->Direct.ChannelsPerOrder[order];                  \
 574     }
 575                         APPLY_NFC_MIX(1)
 576                         APPLY_NFC_MIX(2)
 577                         APPLY_NFC_MIX(3)
 578 #undef APPLY_NFC_MIX
 579                     }
 580                 }
 581                 else
 582                 {
 583                     MixHrtfParams hrtfparams;
 584                     ALsizei fademix = 0;
 585                     int lidx, ridx;
 586
 587                     lidx = GetChannelIdxByName(&Device->RealOut, FrontLeft);
 588                     ridx = GetChannelIdxByName(&Device->RealOut, FrontRight);
 589                     assert(lidx != -1 && ridx != -1);
 590
 591                     if(!Counter)
 592                     {
 593                         /* No fading, just overwrite the old HRTF params. */
 594                         parms->Hrtf.Old = parms->Hrtf.Target;
 595                     }
 596                     else if(!(parms->Hrtf.Old.Gain > GAIN_SILENCE_THRESHOLD))
 597                     {
 598                         /* The old HRTF params are silent, so overwrite the old
 599                          * coefficients with the new, and reset the old gain to
 600                          * 0. The future mix will then fade from silence.
 601                          */
 602                         parms->Hrtf.Old = parms->Hrtf.Target;
 603                         parms->Hrtf.Old.Gain = 0.0f;
 604                     }
 605                     else if(firstpass)
 606                     {
 607                         ALfloat gain;
 608
 609                         /* Fade between the coefficients over 128 samples. */
 610                         fademix = mini(DstBufferSize, 128);
 611
 612                         /* The new coefficients need to fade in completely
 613                          * since they're replacing the old ones. To keep the
 614                          * gain fading consistent, interpolate between the old
 615                          * and new target gains given how much of the fade time
 616                          * this mix handles.
 617                          */
 618                         gain = lerp(parms->Hrtf.Old.Gain, parms->Hrtf.Target.Gain,
 619                                     minf(1.0f, (ALfloat)fademix/Counter));
 620                         hrtfparams.Coeffs = parms->Hrtf.Target.Coeffs;
 621                         hrtfparams.Delay[0] = parms->Hrtf.Target.Delay[0];
 622                         hrtfparams.Delay[1] = parms->Hrtf.Target.Delay[1];
 623                         hrtfparams.Gain = 0.0f;
 624                         hrtfparams.GainStep = gain / (ALfloat)fademix;
 625
 626                         MixHrtfBlendSamples(
 627                             voice->Direct.Buffer[lidx], voice->Direct.Buffer[ridx],
 628                             samples, voice->Offset, OutPos, IrSize, &parms->Hrtf.Old,
 629                             &hrtfparams, &parms->Hrtf.State, fademix
 630                         );
 631                         /* Update the old parameters with the result. */
 632                         parms->Hrtf.Old = parms->Hrtf.Target;
 633                         if(fademix < Counter)
 634                             parms->Hrtf.Old.Gain = hrtfparams.Gain;
 635                     }
 636
 637                     if(fademix < DstBufferSize)
 638                     {
 639                         ALsizei todo = DstBufferSize - fademix;
 640                         ALfloat gain = parms->Hrtf.Target.Gain;
 641
 642                         /* Interpolate the target gain if the gain fading lasts
 643                          * longer than this mix.
 644                          */
 645                         if(Counter > DstBufferSize)
 646                             gain = lerp(parms->Hrtf.Old.Gain, gain,
 647                                         (ALfloat)todo/(Counter-fademix));
 648
 649                         hrtfparams.Coeffs = parms->Hrtf.Target.Coeffs;
 650                         hrtfparams.Delay[0] = parms->Hrtf.Target.Delay[0];
 651                         hrtfparams.Delay[1] = parms->Hrtf.Target.Delay[1];
 652                         hrtfparams.Gain = parms->Hrtf.Old.Gain;
 653                         hrtfparams.GainStep = (gain - parms->Hrtf.Old.Gain) / (ALfloat)todo;
 654                         MixHrtfSamples(
 655                             voice->Direct.Buffer[lidx], voice->Direct.Buffer[ridx],
 656                             samples+fademix, voice->Offset+fademix, OutPos+fademix, IrSize,
 657                             &hrtfparams, &parms->Hrtf.State, todo
 658                         );
 659                         /* Store the interpolated gain or the final target gain
 660                          * depending if the fade is done.
 661                          */
 662                         if(DstBufferSize < Counter)
 663                             parms->Hrtf.Old.Gain = gain;
 664                         else
 665                             parms->Hrtf.Old.Gain = parms->Hrtf.Target.Gain;
 666                     }
 667                 }
 668             }
 669
 670             for(send = 0;send < Device->NumAuxSends;send++)
 671             {
 672                 SendParams *parms = &voice->Send[send].Params[chan];
 673                 const ALfloat *samples;
 674
 675                 if(!voice->Send[send].Buffer)
 676                     continue;
 677
 678                 samples = DoFilters(
 679                     &parms->LowPass, &parms->HighPass, Device->TempBuffer[FILTERED_BUF],
 680                     ResampledData, DstBufferSize, voice->Send[send].FilterType
 681                 );
 682
 683                 if(!Counter)
 684                     memcpy(parms->Gains.Current, parms->Gains.Target,
 685                            sizeof(parms->Gains.Current));
 686                 MixSamples(samples, voice->Send[send].Channels, voice->Send[send].Buffer,
 687                     parms->Gains.Current, parms->Gains.Target, Counter, OutPos, DstBufferSize
 688                 );
 689             }
 690         }
 691         /* Update positions */
 692         DataPosFrac += increment*DstBufferSize;
 693         DataPosInt  += DataPosFrac>>FRACTIONBITS;
 694         DataPosFrac &= FRACTIONMASK;
 695
 696         OutPos += DstBufferSize;
 697         voice->Offset += DstBufferSize;
 698         Counter = maxi(DstBufferSize, Counter) - DstBufferSize;
 699         firstpass = false;
 700
 701         if(isstatic)
 702         {
 703             if(BufferLoopItem)
 704             {
 705                 /* Handle looping static source */
 706                 const ALbuffer *Buffer = BufferListItem->buffers[0];
 707                 ALsizei LoopStart = Buffer->LoopStart;
 708                 ALsizei LoopEnd = Buffer->LoopEnd;
 709                 if(DataPosInt >= LoopEnd)
 710                 {
 711                     assert(LoopEnd > LoopStart);
 712                     DataPosInt = ((DataPosInt-LoopStart)%(LoopEnd-LoopStart)) + LoopStart;
 713                 }
 714             }
 715             else
 716             {
 717                 /* Handle non-looping static source */
 718                 ALsizei CompLen = 0;
 719                 ALsizei i;
 720
 721                 for(i = 0;i < BufferListItem->num_buffers;i++)
 722                 {
 723                     const ALbuffer *buffer = BufferListItem->buffers[i];
 724                     if(buffer) CompLen = maxi(CompLen, buffer->SampleLen);
 725                 }
 726
 727                 if(DataPosInt >= CompLen)
 728                 {
 729                     isplaying = false;
 730                     BufferListItem = NULL;
 731                     DataPosInt = 0;
 732                     DataPosFrac = 0;
 733                     break;
 734                 }
 735             }
 736         }
 737         else while(1)
 738         {
 739             /* Handle streaming source */
 740             ALsizei CompLen = 0;
 741             ALsizei i;
 742
 743             for(i = 0;i < BufferListItem->num_buffers;i++)
 744             {
 745                 const ALbuffer *buffer = BufferListItem->buffers[i];
 746                 if(buffer) CompLen = maxi(CompLen, buffer->SampleLen);
 747             }
 748
 749             if(CompLen > DataPosInt)
 750                 break;
 751
 752             buffers_done += BufferListItem->num_buffers;
 753             BufferListItem = ATOMIC_LOAD(&BufferListItem->next, almemory_order_acquire);
 754             if(!BufferListItem && !(BufferListItem=BufferLoopItem))
 755             {
 756                 isplaying = false;
 757                 DataPosInt = 0;
 758                 DataPosFrac = 0;
 759                 break;
 760             }
 761
 762             DataPosInt -= CompLen;
 763         }
 764     } while(isplaying && OutPos < SamplesToDo);
 765
 766     voice->Flags |= VOICE_IS_FADING;
 767
 768     /* Update source info */
 769     ATOMIC_STORE(&voice->position,          DataPosInt, almemory_order_relaxed);
 770     ATOMIC_STORE(&voice->position_fraction, DataPosFrac, almemory_order_relaxed);
 771     ATOMIC_STORE(&voice->current_buffer,    BufferListItem, almemory_order_release);
 772
 773     /* Send any events now, after the position/buffer info was updated. */
 774     enabledevt = ATOMIC_LOAD(&Context->EnabledEvts, almemory_order_acquire);
 775     if(buffers_done > 0 && (enabledevt&EventType_BufferCompleted))
 776         SendAsyncEvent(Context, EventType_BufferCompleted,
 777             AL_EVENT_TYPE_BUFFER_COMPLETED_SOFT, SourceID, buffers_done, "Buffer completed"
 778         );
 779
 780     return isplaying;
 781 }