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, ALsizei *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 = ASYNC_EVENT(enumtype);
 201     evt.u.user.type = type;
 202     evt.u.user.id = objid;
 203     evt.u.user.param = param;
 204     strcpy(evt.u.user.msg, msg);
 205     if(ll_ringbuffer_write(context->AsyncEvents, (const char*)&evt, 1) == 1)
 206         alsem_post(&context->EventSem);
 207 }
 208
 209
 210 static inline ALfloat Sample_ALubyte(ALubyte val)
 211 { return (val-128) * (1.0f/128.0f); }
 212
 213 static inline ALfloat Sample_ALshort(ALshort val)
 214 { return val * (1.0f/32768.0f); }
 215
 216 static inline ALfloat Sample_ALfloat(ALfloat val)
 217 { return val; }
 218
 219 static inline ALfloat Sample_ALdouble(ALdouble val)
 220 { return (ALfloat)val; }
 221
 222 typedef ALubyte ALmulaw;
 223 static inline ALfloat Sample_ALmulaw(ALmulaw val)
 224 { return muLawDecompressionTable[val] * (1.0f/32768.0f); }
 225
 226 typedef ALubyte ALalaw;
 227 static inline ALfloat Sample_ALalaw(ALalaw val)
 228 { return aLawDecompressionTable[val] * (1.0f/32768.0f); }
 229
 230 #define DECL_TEMPLATE(T)                                                      \
 231 static inline void Load_##T(ALfloat *restrict dst, const T *restrict src,     \
 232                             ALint srcstep, ALsizei samples)                   \
 233 {                                                                             \
 234     ALsizei i;                                                                \
 235     for(i = 0;i < samples;i++)                                                \
 236         dst[i] += Sample_##T(src[i*srcstep]);                                 \
 237 }
 238
 239 DECL_TEMPLATE(ALubyte)
 240 DECL_TEMPLATE(ALshort)
 241 DECL_TEMPLATE(ALfloat)
 242 DECL_TEMPLATE(ALdouble)
 243 DECL_TEMPLATE(ALmulaw)
 244 DECL_TEMPLATE(ALalaw)
 245
 246 #undef DECL_TEMPLATE
 247
 248 static void LoadSamples(ALfloat *restrict dst, const ALvoid *restrict src, ALint srcstep,
 249                         enum FmtType srctype, ALsizei samples)
 250 {
 251 #define HANDLE_FMT(ET, ST) case ET: Load_##ST(dst, src, srcstep, samples); break
 252     switch(srctype)
 253     {
 254         HANDLE_FMT(FmtUByte, ALubyte);
 255         HANDLE_FMT(FmtShort, ALshort);
 256         HANDLE_FMT(FmtFloat, ALfloat);
 257         HANDLE_FMT(FmtDouble, ALdouble);
 258         HANDLE_FMT(FmtMulaw, ALmulaw);
 259         HANDLE_FMT(FmtAlaw, ALalaw);
 260     }
 261 #undef HANDLE_FMT
 262 }
 263
 264
 265 static const ALfloat *DoFilters(BiquadFilter *lpfilter, BiquadFilter *hpfilter,
 266                                 ALfloat *restrict dst, const ALfloat *restrict src,
 267                                 ALsizei numsamples, enum ActiveFilters type)
 268 {
 269     ALsizei i;
 270     switch(type)
 271     {
 272         case AF_None:
 273             BiquadFilter_passthru(lpfilter, numsamples);
 274             BiquadFilter_passthru(hpfilter, numsamples);
 275             break;
 276
 277         case AF_LowPass:
 278             BiquadFilter_process(lpfilter, dst, src, numsamples);
 279             BiquadFilter_passthru(hpfilter, numsamples);
 280             return dst;
 281         case AF_HighPass:
 282             BiquadFilter_passthru(lpfilter, numsamples);
 283             BiquadFilter_process(hpfilter, dst, src, numsamples);
 284             return dst;
 285
 286         case AF_BandPass:
 287             for(i = 0;i < numsamples;)
 288             {
 289                 ALfloat temp[256];
 290                 ALsizei todo = mini(256, numsamples-i);
 291
 292                 BiquadFilter_process(lpfilter, temp, src+i, todo);
 293                 BiquadFilter_process(hpfilter, dst+i, temp, todo);
 294                 i += todo;
 295             }
 296             return dst;
 297     }
 298     return src;
 299 }
 300
 301
 302 /* This function uses these device temp buffers. */
 303 #define SOURCE_DATA_BUF 0
 304 #define RESAMPLED_BUF 1
 305 #define FILTERED_BUF 2
 306 #define NFC_DATA_BUF 3
 307 ALboolean MixSource(ALvoice *voice, ALuint SourceID, ALCcontext *Context, ALsizei SamplesToDo)
 308 {
 309     ALCdevice *Device = Context->Device;
 310     ALbufferlistitem *BufferListItem;
 311     ALbufferlistitem *BufferLoopItem;
 312     ALsizei NumChannels, SampleSize;
 313     ALbitfieldSOFT enabledevt;
 314     ALsizei buffers_done = 0;
 315     ResamplerFunc Resample;
 316     ALsizei DataPosInt;
 317     ALsizei DataPosFrac;
 318     ALint64 DataSize64;
 319     ALint increment;
 320     ALsizei Counter;
 321     ALsizei OutPos;
 322     ALsizei IrSize;
 323     bool isplaying;
 324     bool firstpass;
 325     bool isstatic;
 326     ALsizei chan;
 327     ALsizei send;
 328
 329     /* Get source info */
 330     isplaying      = true; /* Will only be called while playing. */
 331     isstatic       = !!(voice->Flags&VOICE_IS_STATIC);
 332     DataPosInt     = ATOMIC_LOAD(&voice->position, almemory_order_acquire);
 333     DataPosFrac    = ATOMIC_LOAD(&voice->position_fraction, almemory_order_relaxed);
 334     BufferListItem = ATOMIC_LOAD(&voice->current_buffer, almemory_order_relaxed);
 335     BufferLoopItem = ATOMIC_LOAD(&voice->loop_buffer, almemory_order_relaxed);
 336     NumChannels    = voice->NumChannels;
 337     SampleSize     = voice->SampleSize;
 338     increment      = voice->Step;
 339
 340     IrSize = (Device->HrtfHandle ? Device->HrtfHandle->irSize : 0);
 341
 342     Resample = ((increment == FRACTIONONE && DataPosFrac == 0) ?
 343                 Resample_copy_C : voice->Resampler);
 344
 345     Counter = (voice->Flags&VOICE_IS_FADING) ? SamplesToDo : 0;
 346     firstpass = true;
 347     OutPos = 0;
 348
 349     do {
 350         ALsizei SrcBufferSize, DstBufferSize;
 351
 352         /* Figure out how many buffer samples will be needed */
 353         DataSize64  = SamplesToDo-OutPos;
 354         DataSize64 *= increment;
 355         DataSize64 += DataPosFrac+FRACTIONMASK;
 356         DataSize64 >>= FRACTIONBITS;
 357         DataSize64 += MAX_RESAMPLE_PADDING*2;
 358         SrcBufferSize = (ALsizei)mini64(DataSize64, BUFFERSIZE);
 359
 360         /* Figure out how many samples we can actually mix from this. */
 361         DataSize64  = SrcBufferSize;
 362         DataSize64 -= MAX_RESAMPLE_PADDING*2;
 363         DataSize64 <<= FRACTIONBITS;
 364         DataSize64 -= DataPosFrac;
 365         DstBufferSize = (ALsizei)mini64((DataSize64+(increment-1)) / increment,
 366                                         SamplesToDo - OutPos);
 367
 368         /* Some mixers like having a multiple of 4, so try to give that unless
 369          * this is the last update. */
 370         if(DstBufferSize < SamplesToDo-OutPos)
 371             DstBufferSize &= ~3;
 372
 373         /* It's impossible to have a buffer list item with no entries. */
 374         assert(BufferListItem->num_buffers > 0);
 375
 376         for(chan = 0;chan < NumChannels;chan++)
 377         {
 378             const ALfloat *ResampledData;
 379             ALfloat *SrcData = Device->TempBuffer[SOURCE_DATA_BUF];
 380             ALsizei FilledAmt;
 381
 382             /* Load the previous samples into the source data first, and clear the rest. */
 383             memcpy(SrcData, voice->PrevSamples[chan], MAX_RESAMPLE_PADDING*sizeof(ALfloat));
 384             memset(SrcData+MAX_RESAMPLE_PADDING, 0, (BUFFERSIZE-MAX_RESAMPLE_PADDING)*
 385                                                     sizeof(ALfloat));
 386             FilledAmt = MAX_RESAMPLE_PADDING;
 387
 388             if(isstatic)
 389             {
 390                 /* TODO: For static sources, loop points are taken from the
 391                  * first buffer (should be adjusted by any buffer offset, to
 392                  * possibly be added later).
 393                  */
 394                 const ALbuffer *Buffer0 = BufferListItem->buffers[0];
 395                 const ALsizei LoopStart = Buffer0->LoopStart;
 396                 const ALsizei LoopEnd   = Buffer0->LoopEnd;
 397                 const ALsizei LoopSize  = LoopEnd - LoopStart;
 398
 399                 /* If current pos is beyond the loop range, do not loop */
 400                 if(!BufferLoopItem || DataPosInt >= LoopEnd)
 401                 {
 402                     ALsizei SizeToDo = SrcBufferSize - FilledAmt;
 403                     ALsizei CompLen = 0;
 404                     ALsizei i;
 405
 406                     BufferLoopItem = NULL;
 407
 408                     for(i = 0;i < BufferListItem->num_buffers;i++)
 409                     {
 410                         const ALbuffer *buffer = BufferListItem->buffers[i];
 411                         const ALubyte *Data = buffer->data;
 412                         ALsizei DataSize;
 413
 414                         if(DataPosInt >= buffer->SampleLen)
 415                             continue;
 416
 417                         /* Load what's left to play from the buffer */
 418                         DataSize = mini(SizeToDo, buffer->SampleLen - DataPosInt);
 419                         CompLen = maxi(CompLen, DataSize);
 420
 421                         LoadSamples(&SrcData[FilledAmt],
 422                             &Data[(DataPosInt*NumChannels + chan)*SampleSize],
 423                             NumChannels, buffer->FmtType, DataSize
 424                         );
 425                     }
 426                     FilledAmt += CompLen;
 427                 }
 428                 else
 429                 {
 430                     ALsizei SizeToDo = mini(SrcBufferSize - FilledAmt, LoopEnd - DataPosInt);
 431                     ALsizei CompLen = 0;
 432                     ALsizei i;
 433
 434                     for(i = 0;i < BufferListItem->num_buffers;i++)
 435                     {
 436                         const ALbuffer *buffer = BufferListItem->buffers[i];
 437                         const ALubyte *Data = buffer->data;
 438                         ALsizei DataSize;
 439
 440                         if(DataPosInt >= buffer->SampleLen)
 441                             continue;
 442
 443                         /* Load what's left of this loop iteration */
 444                         DataSize = mini(SizeToDo, buffer->SampleLen - DataPosInt);
 445                         CompLen = maxi(CompLen, DataSize);
 446
 447                         LoadSamples(&SrcData[FilledAmt],
 448                             &Data[(DataPosInt*NumChannels + chan)*SampleSize],
 449                             NumChannels, buffer->FmtType, DataSize
 450                         );
 451                     }
 452                     FilledAmt += CompLen;
 453
 454                     while(SrcBufferSize > FilledAmt)
 455                     {
 456                         const ALsizei SizeToDo = mini(SrcBufferSize - FilledAmt, LoopSize);
 457
 458                         CompLen = 0;
 459                         for(i = 0;i < BufferListItem->num_buffers;i++)
 460                         {
 461                             const ALbuffer *buffer = BufferListItem->buffers[i];
 462                             const ALubyte *Data = buffer->data;
 463                             ALsizei DataSize;
 464
 465                             if(LoopStart >= buffer->SampleLen)
 466                                 continue;
 467
 468                             DataSize = mini(SizeToDo, buffer->SampleLen - LoopStart);
 469                             CompLen = maxi(CompLen, DataSize);
 470
 471                             LoadSamples(&SrcData[FilledAmt],
 472                                 &Data[(LoopStart*NumChannels + chan)*SampleSize],
 473                                 NumChannels, buffer->FmtType, DataSize
 474                             );
 475                         }
 476                         FilledAmt += CompLen;
 477                     }
 478                 }
 479             }
 480             else
 481             {
 482                 /* Crawl the buffer queue to fill in the temp buffer */
 483                 ALbufferlistitem *tmpiter = BufferListItem;
 484                 ALsizei pos = DataPosInt;
 485
 486                 while(tmpiter && SrcBufferSize > FilledAmt)
 487                 {
 488                     ALsizei SizeToDo = SrcBufferSize - FilledAmt;
 489                     ALsizei CompLen = 0;
 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 = mini(SizeToDo, DataSize - pos);
 503                             CompLen = maxi(CompLen, DataSize);
 504
 505                             LoadSamples(&SrcData[FilledAmt], Data, NumChannels,
 506                                         ALBuffer->FmtType, DataSize);
 507                         }
 508                     }
 509                     if(UNLIKELY(!CompLen))
 510                         pos -= tmpiter->max_samples;
 511                     else
 512                     {
 513                         FilledAmt += CompLen;
 514                         if(SrcBufferSize <= FilledAmt)
 515                             break;
 516                         pos = 0;
 517                     }
 518                     tmpiter = ATOMIC_LOAD(&tmpiter->next, almemory_order_acquire);
 519                     if(!tmpiter) tmpiter = BufferLoopItem;
 520                 }
 521             }
 522
 523             /* Store the last source samples used for next time. */
 524             memcpy(voice->PrevSamples[chan],
 525                 &SrcData[(increment*DstBufferSize + DataPosFrac)>>FRACTIONBITS],
 526                 MAX_RESAMPLE_PADDING*sizeof(ALfloat)
 527             );
 528
 529             /* Now resample, then filter and mix to the appropriate outputs. */
 530             ResampledData = Resample(&voice->ResampleState,
 531                 &SrcData[MAX_RESAMPLE_PADDING], DataPosFrac, increment,
 532                 Device->TempBuffer[RESAMPLED_BUF], DstBufferSize
 533             );
 534             {
 535                 DirectParams *parms = &voice->Direct.Params[chan];
 536                 const ALfloat *samples;
 537
 538                 samples = DoFilters(
 539                     &parms->LowPass, &parms->HighPass, Device->TempBuffer[FILTERED_BUF],
 540                     ResampledData, DstBufferSize, voice->Direct.FilterType
 541                 );
 542                 if(!(voice->Flags&VOICE_HAS_HRTF))
 543                 {
 544                     if(!Counter)
 545                         memcpy(parms->Gains.Current, parms->Gains.Target,
 546                                sizeof(parms->Gains.Current));
 547                     if(!(voice->Flags&VOICE_HAS_NFC))
 548                         MixSamples(samples, voice->Direct.Channels, voice->Direct.Buffer,
 549                             parms->Gains.Current, parms->Gains.Target, Counter, OutPos,
 550                             DstBufferSize
 551                         );
 552                     else
 553                     {
 554                         ALfloat *nfcsamples = Device->TempBuffer[NFC_DATA_BUF];
 555                         ALsizei chanoffset = 0;
 556
 557                         MixSamples(samples,
 558                             voice->Direct.ChannelsPerOrder[0], voice->Direct.Buffer,
 559                             parms->Gains.Current, parms->Gains.Target, Counter, OutPos,
 560                             DstBufferSize
 561                         );
 562                         chanoffset += voice->Direct.ChannelsPerOrder[0];
 563 #define APPLY_NFC_MIX(order)                                                  \
 564     if(voice->Direct.ChannelsPerOrder[order] > 0)                             \
 565     {                                                                         \
 566         NfcFilterProcess##order(&parms->NFCtrlFilter, nfcsamples, samples,    \
 567                                DstBufferSize);                                \
 568         MixSamples(nfcsamples, voice->Direct.ChannelsPerOrder[order],         \
 569             voice->Direct.Buffer+chanoffset, parms->Gains.Current+chanoffset, \
 570             parms->Gains.Target+chanoffset, Counter, OutPos, DstBufferSize    \
 571         );                                                                    \
 572         chanoffset += voice->Direct.ChannelsPerOrder[order];                  \
 573     }
 574                         APPLY_NFC_MIX(1)
 575                         APPLY_NFC_MIX(2)
 576                         APPLY_NFC_MIX(3)
 577 #undef APPLY_NFC_MIX
 578                     }
 579                 }
 580                 else
 581                 {
 582                     MixHrtfParams hrtfparams;
 583                     ALsizei fademix = 0;
 584                     int lidx, ridx;
 585
 586                     lidx = GetChannelIdxByName(&Device->RealOut, FrontLeft);
 587                     ridx = GetChannelIdxByName(&Device->RealOut, FrontRight);
 588                     assert(lidx != -1 && ridx != -1);
 589
 590                     if(!Counter)
 591                     {
 592                         /* No fading, just overwrite the old HRTF params. */
 593                         parms->Hrtf.Old = parms->Hrtf.Target;
 594                     }
 595                     else if(!(parms->Hrtf.Old.Gain > GAIN_SILENCE_THRESHOLD))
 596                     {
 597                         /* The old HRTF params are silent, so overwrite the old
 598                          * coefficients with the new, and reset the old gain to
 599                          * 0. The future mix will then fade from silence.
 600                          */
 601                         parms->Hrtf.Old = parms->Hrtf.Target;
 602                         parms->Hrtf.Old.Gain = 0.0f;
 603                     }
 604                     else if(firstpass)
 605                     {
 606                         ALfloat gain;
 607
 608                         /* Fade between the coefficients over 128 samples. */
 609                         fademix = mini(DstBufferSize, 128);
 610
 611                         /* The new coefficients need to fade in completely
 612                          * since they're replacing the old ones. To keep the
 613                          * gain fading consistent, interpolate between the old
 614                          * and new target gains given how much of the fade time
 615                          * this mix handles.
 616                          */
 617                         gain = lerp(parms->Hrtf.Old.Gain, parms->Hrtf.Target.Gain,
 618                                     minf(1.0f, (ALfloat)fademix/Counter));
 619                         hrtfparams.Coeffs = parms->Hrtf.Target.Coeffs;
 620                         hrtfparams.Delay[0] = parms->Hrtf.Target.Delay[0];
 621                         hrtfparams.Delay[1] = parms->Hrtf.Target.Delay[1];
 622                         hrtfparams.Gain = 0.0f;
 623                         hrtfparams.GainStep = gain / (ALfloat)fademix;
 624
 625                         MixHrtfBlendSamples(
 626                             voice->Direct.Buffer[lidx], voice->Direct.Buffer[ridx],
 627                             samples, voice->Offset, OutPos, IrSize, &parms->Hrtf.Old,
 628                             &hrtfparams, &parms->Hrtf.State, fademix
 629                         );
 630                         /* Update the old parameters with the result. */
 631                         parms->Hrtf.Old = parms->Hrtf.Target;
 632                         if(fademix < Counter)
 633                             parms->Hrtf.Old.Gain = hrtfparams.Gain;
 634                     }
 635
 636                     if(fademix < DstBufferSize)
 637                     {
 638                         ALsizei todo = DstBufferSize - fademix;
 639                         ALfloat gain = parms->Hrtf.Target.Gain;
 640
 641                         /* Interpolate the target gain if the gain fading lasts
 642                          * longer than this mix.
 643                          */
 644                         if(Counter > DstBufferSize)
 645                             gain = lerp(parms->Hrtf.Old.Gain, gain,
 646                                         (ALfloat)todo/(Counter-fademix));
 647
 648                         hrtfparams.Coeffs = parms->Hrtf.Target.Coeffs;
 649                         hrtfparams.Delay[0] = parms->Hrtf.Target.Delay[0];
 650                         hrtfparams.Delay[1] = parms->Hrtf.Target.Delay[1];
 651                         hrtfparams.Gain = parms->Hrtf.Old.Gain;
 652                         hrtfparams.GainStep = (gain - parms->Hrtf.Old.Gain) / (ALfloat)todo;
 653                         MixHrtfSamples(
 654                             voice->Direct.Buffer[lidx], voice->Direct.Buffer[ridx],
 655                             samples+fademix, voice->Offset+fademix, OutPos+fademix, IrSize,
 656                             &hrtfparams, &parms->Hrtf.State, todo
 657                         );
 658                         /* Store the interpolated gain or the final target gain
 659                          * depending if the fade is done.
 660                          */
 661                         if(DstBufferSize < Counter)
 662                             parms->Hrtf.Old.Gain = gain;
 663                         else
 664                             parms->Hrtf.Old.Gain = parms->Hrtf.Target.Gain;
 665                     }
 666                 }
 667             }
 668
 669             for(send = 0;send < Device->NumAuxSends;send++)
 670             {
 671                 SendParams *parms = &voice->Send[send].Params[chan];
 672                 const ALfloat *samples;
 673
 674                 if(!voice->Send[send].Buffer)
 675                     continue;
 676
 677                 samples = DoFilters(
 678                     &parms->LowPass, &parms->HighPass, Device->TempBuffer[FILTERED_BUF],
 679                     ResampledData, DstBufferSize, voice->Send[send].FilterType
 680                 );
 681
 682                 if(!Counter)
 683                     memcpy(parms->Gains.Current, parms->Gains.Target,
 684                            sizeof(parms->Gains.Current));
 685                 MixSamples(samples, voice->Send[send].Channels, voice->Send[send].Buffer,
 686                     parms->Gains.Current, parms->Gains.Target, Counter, OutPos, DstBufferSize
 687                 );
 688             }
 689         }
 690         /* Update positions */
 691         DataPosFrac += increment*DstBufferSize;
 692         DataPosInt  += DataPosFrac>>FRACTIONBITS;
 693         DataPosFrac &= FRACTIONMASK;
 694
 695         OutPos += DstBufferSize;
 696         voice->Offset += DstBufferSize;
 697         Counter = maxi(DstBufferSize, Counter) - DstBufferSize;
 698         firstpass = false;
 699
 700         if(isstatic)
 701         {
 702             if(BufferLoopItem)
 703             {
 704                 /* Handle looping static source */
 705                 const ALbuffer *Buffer = BufferListItem->buffers[0];
 706                 ALsizei LoopStart = Buffer->LoopStart;
 707                 ALsizei LoopEnd = Buffer->LoopEnd;
 708                 if(DataPosInt >= LoopEnd)
 709                 {
 710                     assert(LoopEnd > LoopStart);
 711                     DataPosInt = ((DataPosInt-LoopStart)%(LoopEnd-LoopStart)) + LoopStart;
 712                 }
 713             }
 714             else
 715             {
 716                 /* Handle non-looping static source */
 717                 if(DataPosInt >= BufferListItem->max_samples)
 718                 {
 719                     isplaying = false;
 720                     BufferListItem = NULL;
 721                     DataPosInt = 0;
 722                     DataPosFrac = 0;
 723                     break;
 724                 }
 725             }
 726         }
 727         else while(1)
 728         {
 729             /* Handle streaming source */
 730             if(BufferListItem->max_samples > DataPosInt)
 731                 break;
 732
 733             DataPosInt -= BufferListItem->max_samples;
 734
 735             buffers_done += BufferListItem->num_buffers;
 736             BufferListItem = ATOMIC_LOAD(&BufferListItem->next, almemory_order_relaxed);
 737             if(!BufferListItem && !(BufferListItem=BufferLoopItem))
 738             {
 739                 isplaying = false;
 740                 DataPosInt = 0;
 741                 DataPosFrac = 0;
 742                 break;
 743             }
 744         }
 745     } while(isplaying && OutPos < SamplesToDo);
 746
 747     voice->Flags |= VOICE_IS_FADING;
 748
 749     /* Update source info */
 750     ATOMIC_STORE(&voice->position,          DataPosInt, almemory_order_relaxed);
 751     ATOMIC_STORE(&voice->position_fraction, DataPosFrac, almemory_order_relaxed);
 752     ATOMIC_STORE(&voice->current_buffer,    BufferListItem, almemory_order_release);
 753
 754     /* Send any events now, after the position/buffer info was updated. */
 755     enabledevt = ATOMIC_LOAD(&Context->EnabledEvts, almemory_order_acquire);
 756     if(buffers_done > 0 && (enabledevt&EventType_BufferCompleted))
 757         SendAsyncEvent(Context, EventType_BufferCompleted,
 758             AL_EVENT_TYPE_BUFFER_COMPLETED_SOFT, SourceID, buffers_done, "Buffer completed"
 759         );
 760
 761     return isplaying;
 762 }