Alc/mixer.c

   1 /**
   2  * OpenAL cross platform audio library
   3  * Copyright (C) 1999-2007 by authors.
   4  * This library is free software; you can redistribute it and/or
   5  *  modify it under the terms of the GNU Library General Public
   6  *  License as published by the Free Software Foundation; either
   7  *  version 2 of the License, or (at your option) any later version.
   8  *
   9  * This library is distributed in the hope that it will be useful,
  10  *  but WITHOUT ANY WARRANTY; without even the implied warranty of
  11  *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  12  *  Library General Public License for more details.
  13  *
  14  * You should have received a copy of the GNU Library General Public
  15  *  License along with this library; if not, write to the
  16  *  Free Software Foundation, Inc.,
  17  *  51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
  18  * Or go to http://www.gnu.org/copyleft/lgpl.html
  19  */
  20
  21 #include "config.h"
  22
  23 #include <math.h>
  24 #include <stdlib.h>
  25 #include <string.h>
  26 #include <ctype.h>
  27 #include <assert.h>
  28
  29 #include "alMain.h"
  30 #include "AL/al.h"
  31 #include "AL/alc.h"
  32 #include "alSource.h"
  33 #include "alBuffer.h"
  34 #include "alListener.h"
  35 #include "alAuxEffectSlot.h"
  36 #include "alu.h"
  37
  38 #include "mixer_defs.h"
  39
  40
  41 static_assert((INT_MAX>>FRACTIONBITS)/MAX_PITCH > BUFFERSIZE,
  42               "MAX_PITCH and/or BUFFERSIZE are too large for FRACTIONBITS!");
  43
  44 extern inline void InitiatePositionArrays(ALuint frac, ALuint increment, ALuint *frac_arr, ALuint *pos_arr, ALuint size);
  45
  46 alignas(16) union ResamplerCoeffs ResampleCoeffs;
  47
  48
  49 enum Resampler {
  50     PointResampler,
  51     LinearResampler,
  52     FIR4Resampler,
  53     FIR8Resampler,
  54     BSincResampler,
  55
  56     ResamplerDefault = LinearResampler
  57 };
  58
  59 /* FIR8 requires 3 extra samples before the current position, and 4 after. */
  60 static_assert(MAX_PRE_SAMPLES >= 3, "MAX_PRE_SAMPLES must be at least 3!");
  61 static_assert(MAX_POST_SAMPLES >= 4, "MAX_POST_SAMPLES must be at least 4!");
  62
  63
  64 static HrtfMixerFunc MixHrtfSamples = MixHrtf_C;
  65 static MixerFunc MixSamples = Mix_C;
  66 static ResamplerFunc ResampleSamples = Resample_point32_C;
  67
  68 static inline HrtfMixerFunc SelectHrtfMixer(void)
  69 {
  70 #ifdef HAVE_SSE
  71     if((CPUCapFlags&CPU_CAP_SSE))
  72         return MixHrtf_SSE;
  73 #endif
  74 #ifdef HAVE_NEON
  75     if((CPUCapFlags&CPU_CAP_NEON))
  76         return MixHrtf_Neon;
  77 #endif
  78
  79     return MixHrtf_C;
  80 }
  81
  82 static inline MixerFunc SelectMixer(void)
  83 {
  84 #ifdef HAVE_SSE
  85     if((CPUCapFlags&CPU_CAP_SSE))
  86         return Mix_SSE;
  87 #endif
  88 #ifdef HAVE_NEON
  89     if((CPUCapFlags&CPU_CAP_NEON))
  90         return Mix_Neon;
  91 #endif
  92
  93     return Mix_C;
  94 }
  95
  96 static inline ResamplerFunc SelectResampler(enum Resampler resampler)
  97 {
  98     switch(resampler)
  99     {
 100         case PointResampler:
 101             return Resample_point32_C;
 102         case LinearResampler:
 103 #ifdef HAVE_SSE4_1
 104             if((CPUCapFlags&CPU_CAP_SSE4_1))
 105                 return Resample_lerp32_SSE41;
 106 #endif
 107 #ifdef HAVE_SSE2
 108             if((CPUCapFlags&CPU_CAP_SSE2))
 109                 return Resample_lerp32_SSE2;
 110 #endif
 111             return Resample_lerp32_C;
 112         case FIR4Resampler:
 113 #ifdef HAVE_SSE4_1
 114             if((CPUCapFlags&CPU_CAP_SSE4_1))
 115                 return Resample_fir4_32_SSE41;
 116 #endif
 117 #ifdef HAVE_SSE3
 118             if((CPUCapFlags&CPU_CAP_SSE3))
 119                 return Resample_fir4_32_SSE3;
 120 #endif
 121             return Resample_fir4_32_C;
 122         case FIR8Resampler:
 123 #ifdef HAVE_SSE4_1
 124             if((CPUCapFlags&CPU_CAP_SSE4_1))
 125                 return Resample_fir8_32_SSE41;
 126 #endif
 127 #ifdef HAVE_SSE3
 128             if((CPUCapFlags&CPU_CAP_SSE3))
 129                 return Resample_fir8_32_SSE3;
 130 #endif
 131             return Resample_fir8_32_C;
 132         case BSincResampler:
 133 #ifdef HAVE_SSE
 134             if((CPUCapFlags&CPU_CAP_SSE))
 135                 return Resample_bsinc32_SSE;
 136 #endif
 137             return Resample_bsinc32_C;
 138     }
 139
 140     return Resample_point32_C;
 141 }
 142
 143
 144 /* The sinc resampler makes use of a Kaiser window to limit the needed sample
 145  * points to 4 and 8, respectively.
 146  */
 147
 148 #ifndef M_PI
 149 #define M_PI                         (3.14159265358979323846)
 150 #endif
 151 static inline double Sinc(double x)
 152 {
 153     if(x == 0.0) return 1.0;
 154     return sin(x*M_PI) / (x*M_PI);
 155 }
 156
 157 /* The zero-order modified Bessel function of the first kind, used for the
 158  * Kaiser window.
 159  *
 160  *   I_0(x) = sum_{k=0}^inf (1 / k!)^2 (x / 2)^(2 k)
 161  *          = sum_{k=0}^inf ((x / 2)^k / k!)^2
 162  */
 163 static double BesselI_0(double x)
 164 {
 165     double term, sum, x2, y, last_sum;
 166     int k;
 167
 168     /* Start at k=1 since k=0 is trivial. */
 169     term = 1.0;
 170     sum = 1.0;
 171     x2 = x / 2.0;
 172     k = 1;
 173
 174     /* Let the integration converge until the term of the sum is no longer
 175      * significant.
 176      */
 177     do {
 178         y = x2 / k;
 179         k ++;
 180         last_sum = sum;
 181         term *= y * y;
 182         sum += term;
 183     } while(sum != last_sum);
 184     return sum;
 185 }
 186
 187 /* Calculate a Kaiser window from the given beta value and a normalized k
 188  * [-1, 1].
 189  *
 190  *   w(k) = { I_0(B sqrt(1 - k^2)) / I_0(B),  -1 <= k <= 1
 191  *          { 0,                              elsewhere.
 192  *
 193  * Where k can be calculated as:
 194  *
 195  *   k = i / l,         where -l <= i <= l.
 196  *
 197  * or:
 198  *
 199  *   k = 2 i / M - 1,   where 0 <= i <= M.
 200  */
 201 static inline double Kaiser(double b, double k)
 202 {
 203     if(k <= -1.0 || k >= 1.0) return 0.0;
 204     return BesselI_0(b * sqrt(1.0 - (k*k))) / BesselI_0(b);
 205 }
 206
 207 static inline double CalcKaiserBeta(double rejection)
 208 {
 209     if(rejection > 50.0)
 210         return 0.1102 * (rejection - 8.7);
 211     if(rejection >= 21.0)
 212         return (0.5842 * pow(rejection - 21.0, 0.4)) +
 213                (0.07886 * (rejection - 21.0));
 214     return 0.0;
 215 }
 216
 217 static float SincKaiser(double r, double x)
 218 {
 219     /* Limit rippling to -60dB. */
 220     return (float)(Kaiser(CalcKaiserBeta(60.0), x / r) * Sinc(x));
 221 }
 222
 223
 224 void aluInitMixer(void)
 225 {
 226     enum Resampler resampler = ResamplerDefault;
 227     const char *str;
 228     ALuint i;
 229
 230     if(ConfigValueStr(NULL, NULL, "resampler", &str))
 231     {
 232         if(strcasecmp(str, "point") == 0 || strcasecmp(str, "none") == 0)
 233             resampler = PointResampler;
 234         else if(strcasecmp(str, "linear") == 0)
 235             resampler = LinearResampler;
 236         else if(strcasecmp(str, "sinc4") == 0)
 237             resampler = FIR4Resampler;
 238         else if(strcasecmp(str, "sinc8") == 0)
 239             resampler = FIR8Resampler;
 240         else if(strcasecmp(str, "bsinc") == 0)
 241             resampler = BSincResampler;
 242         else if(strcasecmp(str, "cubic") == 0)
 243         {
 244             WARN("Resampler option \"cubic\" is deprecated, using sinc4\n");
 245             resampler = FIR4Resampler;
 246         }
 247         else
 248         {
 249             char *end;
 250             long n = strtol(str, &end, 0);
 251             if(*end == '\0' && (n == PointResampler || n == LinearResampler || n == FIR4Resampler))
 252                 resampler = n;
 253             else
 254                 WARN("Invalid resampler: %s\n", str);
 255         }
 256     }
 257
 258     if(resampler == FIR8Resampler)
 259         for(i = 0;i < FRACTIONONE;i++)
 260         {
 261             ALdouble mu = (ALdouble)i / FRACTIONONE;
 262             ResampleCoeffs.FIR8[i][0] = SincKaiser(4.0, mu - -3.0);
 263             ResampleCoeffs.FIR8[i][1] = SincKaiser(4.0, mu - -2.0);
 264             ResampleCoeffs.FIR8[i][2] = SincKaiser(4.0, mu - -1.0);
 265             ResampleCoeffs.FIR8[i][3] = SincKaiser(4.0, mu -  0.0);
 266             ResampleCoeffs.FIR8[i][4] = SincKaiser(4.0, mu -  1.0);
 267             ResampleCoeffs.FIR8[i][5] = SincKaiser(4.0, mu -  2.0);
 268             ResampleCoeffs.FIR8[i][6] = SincKaiser(4.0, mu -  3.0);
 269             ResampleCoeffs.FIR8[i][7] = SincKaiser(4.0, mu -  4.0);
 270         }
 271     else if(resampler == FIR4Resampler)
 272         for(i = 0;i < FRACTIONONE;i++)
 273         {
 274             ALdouble mu = (ALdouble)i / FRACTIONONE;
 275             ResampleCoeffs.FIR4[i][0] = SincKaiser(2.0, mu - -1.0);
 276             ResampleCoeffs.FIR4[i][1] = SincKaiser(2.0, mu -  0.0);
 277             ResampleCoeffs.FIR4[i][2] = SincKaiser(2.0, mu -  1.0);
 278             ResampleCoeffs.FIR4[i][3] = SincKaiser(2.0, mu -  2.0);
 279         }
 280
 281     MixHrtfSamples = SelectHrtfMixer();
 282     MixSamples = SelectMixer();
 283     ResampleSamples = SelectResampler(resampler);
 284 }
 285
 286
 287 static inline ALfloat Sample_ALbyte(ALbyte val)
 288 { return val * (1.0f/127.0f); }
 289
 290 static inline ALfloat Sample_ALshort(ALshort val)
 291 { return val * (1.0f/32767.0f); }
 292
 293 static inline ALfloat Sample_ALfloat(ALfloat val)
 294 { return val; }
 295
 296 #define DECL_TEMPLATE(T)                                                      \
 297 static inline void Load_##T(ALfloat *dst, const T *src, ALuint srcstep, ALuint samples)\
 298 {                                                                             \
 299     ALuint i;                                                                 \
 300     for(i = 0;i < samples;i++)                                                \
 301         dst[i] = Sample_##T(src[i*srcstep]);                                  \
 302 }
 303
 304 DECL_TEMPLATE(ALbyte)
 305 DECL_TEMPLATE(ALshort)
 306 DECL_TEMPLATE(ALfloat)
 307
 308 #undef DECL_TEMPLATE
 309
 310 static void LoadSamples(ALfloat *dst, const ALvoid *src, ALuint srcstep, enum FmtType srctype, ALuint samples)
 311 {
 312     switch(srctype)
 313     {
 314         case FmtByte:
 315             Load_ALbyte(dst, src, srcstep, samples);
 316             break;
 317         case FmtShort:
 318             Load_ALshort(dst, src, srcstep, samples);
 319             break;
 320         case FmtFloat:
 321             Load_ALfloat(dst, src, srcstep, samples);
 322             break;
 323     }
 324 }
 325
 326 static inline void SilenceSamples(ALfloat *dst, ALuint samples)
 327 {
 328     ALuint i;
 329     for(i = 0;i < samples;i++)
 330         dst[i] = 0.0f;
 331 }
 332
 333
 334 static const ALfloat *DoFilters(ALfilterState *lpfilter, ALfilterState *hpfilter,
 335                                 ALfloat *restrict dst, const ALfloat *restrict src,
 336                                 ALuint numsamples, enum ActiveFilters type)
 337 {
 338     ALuint i;
 339     switch(type)
 340     {
 341         case AF_None:
 342             ALfilterState_processPassthru(lpfilter, src, numsamples);
 343             ALfilterState_processPassthru(hpfilter, src, numsamples);
 344             break;
 345
 346         case AF_LowPass:
 347             ALfilterState_process(lpfilter, dst, src, numsamples);
 348             ALfilterState_processPassthru(hpfilter, dst, numsamples);
 349             return dst;
 350         case AF_HighPass:
 351             ALfilterState_processPassthru(lpfilter, src, numsamples);
 352             ALfilterState_process(hpfilter, dst, src, numsamples);
 353             return dst;
 354
 355         case AF_BandPass:
 356             for(i = 0;i < numsamples;)
 357             {
 358                 ALfloat temp[256];
 359                 ALuint todo = minu(256, numsamples-i);
 360
 361                 ALfilterState_process(lpfilter, temp, src+i, todo);
 362                 ALfilterState_process(hpfilter, dst+i, temp, todo);
 363                 i += todo;
 364             }
 365             return dst;
 366     }
 367     return src;
 368 }
 369
 370
 371 ALvoid MixSource(ALvoice *voice, ALsource *Source, ALCdevice *Device, ALuint SamplesToDo)
 372 {
 373     ResamplerFunc Resample;
 374     ALbufferlistitem *BufferListItem;
 375     ALuint DataPosInt, DataPosFrac;
 376     ALboolean Looping;
 377     ALuint increment;
 378     ALenum State;
 379     ALuint OutPos;
 380     ALuint NumChannels;
 381     ALuint SampleSize;
 382     ALint64 DataSize64;
 383     ALuint IrSize;
 384     ALuint chan, send, j;
 385
 386     /* Get source info */
 387     State          = AL_PLAYING; /* Only called while playing. */
 388     BufferListItem = ATOMIC_LOAD(&Source->current_buffer);
 389     DataPosInt     = ATOMIC_LOAD(&Source->position, almemory_order_relaxed);
 390     DataPosFrac    = ATOMIC_LOAD(&Source->position_fraction, almemory_order_relaxed);
 391     Looping        = ATOMIC_LOAD(&Source->looping, almemory_order_relaxed);
 392     NumChannels    = Source->NumChannels;
 393     SampleSize     = Source->SampleSize;
 394     increment      = voice->Step;
 395
 396     IrSize = (Device->Hrtf ? Device->Hrtf->irSize : 0);
 397
 398     Resample = ((increment == FRACTIONONE && DataPosFrac == 0) ?
 399                 Resample_copy32_C : ResampleSamples);
 400
 401     OutPos = 0;
 402     do {
 403         ALuint SrcBufferSize, DstBufferSize;
 404         ALuint Counter;
 405         ALfloat Delta;
 406
 407         if(!voice->Moving)
 408         {
 409             Counter = 0;
 410             Delta = 0.0f;
 411         }
 412         else
 413         {
 414             Counter = SamplesToDo - OutPos;
 415             Delta = 1.0f / (ALfloat)Counter;
 416         }
 417
 418         /* Figure out how many buffer samples will be needed */
 419         DataSize64  = SamplesToDo-OutPos;
 420         DataSize64 *= increment;
 421         DataSize64 += DataPosFrac+FRACTIONMASK;
 422         DataSize64 >>= FRACTIONBITS;
 423         DataSize64 += MAX_POST_SAMPLES+MAX_PRE_SAMPLES;
 424
 425         SrcBufferSize = (ALuint)mini64(DataSize64, BUFFERSIZE);
 426
 427         /* Figure out how many samples we can actually mix from this. */
 428         DataSize64  = SrcBufferSize;
 429         DataSize64 -= MAX_POST_SAMPLES+MAX_PRE_SAMPLES;
 430         DataSize64 <<= FRACTIONBITS;
 431         DataSize64 -= DataPosFrac;
 432
 433         DstBufferSize = (ALuint)((DataSize64+(increment-1)) / increment);
 434         DstBufferSize = minu(DstBufferSize, (SamplesToDo-OutPos));
 435
 436         /* Some mixers like having a multiple of 4, so try to give that unless
 437          * this is the last update. */
 438         if(OutPos+DstBufferSize < SamplesToDo)
 439             DstBufferSize &= ~3;
 440
 441         for(chan = 0;chan < NumChannels;chan++)
 442         {
 443             const ALfloat *ResampledData;
 444             ALfloat *SrcData = Device->SourceData;
 445             ALuint SrcDataSize;
 446
 447             /* Load the previous samples into the source data first. */
 448             memcpy(SrcData, voice->PrevSamples[chan], MAX_PRE_SAMPLES*sizeof(ALfloat));
 449             SrcDataSize = MAX_PRE_SAMPLES;
 450
 451             if(Source->SourceType == AL_STATIC)
 452             {
 453                 const ALbuffer *ALBuffer = BufferListItem->buffer;
 454                 const ALubyte *Data = ALBuffer->data;
 455                 ALuint DataSize;
 456                 ALuint pos;
 457
 458                 /* Offset buffer data to current channel */
 459                 Data += chan*SampleSize;
 460
 461                 /* If current pos is beyond the loop range, do not loop */
 462                 if(Looping == AL_FALSE || DataPosInt >= (ALuint)ALBuffer->LoopEnd)
 463                 {
 464                     Looping = AL_FALSE;
 465
 466                     /* Load what's left to play from the source buffer, and
 467                      * clear the rest of the temp buffer */
 468                     pos = DataPosInt;
 469                     DataSize = minu(SrcBufferSize - SrcDataSize, ALBuffer->SampleLen - pos);
 470
 471                     LoadSamples(&SrcData[SrcDataSize], &Data[pos * NumChannels*SampleSize],
 472                                 NumChannels, ALBuffer->FmtType, DataSize);
 473                     SrcDataSize += DataSize;
 474
 475                     SilenceSamples(&SrcData[SrcDataSize], SrcBufferSize - SrcDataSize);
 476                     SrcDataSize += SrcBufferSize - SrcDataSize;
 477                 }
 478                 else
 479                 {
 480                     ALuint LoopStart = ALBuffer->LoopStart;
 481                     ALuint LoopEnd   = ALBuffer->LoopEnd;
 482
 483                     /* Load what's left of this loop iteration, then load
 484                      * repeats of the loop section */
 485                     pos = DataPosInt;
 486                     DataSize = LoopEnd - pos;
 487                     DataSize = minu(SrcBufferSize - SrcDataSize, DataSize);
 488
 489                     LoadSamples(&SrcData[SrcDataSize], &Data[pos * NumChannels*SampleSize],
 490                                 NumChannels, ALBuffer->FmtType, DataSize);
 491                     SrcDataSize += DataSize;
 492
 493                     DataSize = LoopEnd-LoopStart;
 494                     while(SrcBufferSize > SrcDataSize)
 495                     {
 496                         DataSize = minu(SrcBufferSize - SrcDataSize, DataSize);
 497
 498                         LoadSamples(&SrcData[SrcDataSize], &Data[LoopStart * NumChannels*SampleSize],
 499                                     NumChannels, ALBuffer->FmtType, DataSize);
 500                         SrcDataSize += DataSize;
 501                     }
 502                 }
 503             }
 504             else
 505             {
 506                 /* Crawl the buffer queue to fill in the temp buffer */
 507                 ALbufferlistitem *tmpiter = BufferListItem;
 508                 ALuint pos = DataPosInt;
 509
 510                 while(tmpiter && SrcBufferSize > SrcDataSize)
 511                 {
 512                     const ALbuffer *ALBuffer;
 513                     if((ALBuffer=tmpiter->buffer) != NULL)
 514                     {
 515                         const ALubyte *Data = ALBuffer->data;
 516                         ALuint DataSize = ALBuffer->SampleLen;
 517
 518                         /* Skip the data already played */
 519                         if(DataSize <= pos)
 520                             pos -= DataSize;
 521                         else
 522                         {
 523                             Data += (pos*NumChannels + chan)*SampleSize;
 524                             DataSize -= pos;
 525                             pos -= pos;
 526
 527                             DataSize = minu(SrcBufferSize - SrcDataSize, DataSize);
 528                             LoadSamples(&SrcData[SrcDataSize], Data, NumChannels,
 529                                         ALBuffer->FmtType, DataSize);
 530                             SrcDataSize += DataSize;
 531                         }
 532                     }
 533                     tmpiter = tmpiter->next;
 534                     if(!tmpiter && Looping)
 535                         tmpiter = ATOMIC_LOAD(&Source->queue);
 536                     else if(!tmpiter)
 537                     {
 538                         SilenceSamples(&SrcData[SrcDataSize], SrcBufferSize - SrcDataSize);
 539                         SrcDataSize += SrcBufferSize - SrcDataSize;
 540                     }
 541                 }
 542             }
 543
 544             /* Store the last source samples used for next time. */
 545             memcpy(voice->PrevSamples[chan],
 546                 &SrcData[(increment*DstBufferSize + DataPosFrac)>>FRACTIONBITS],
 547                 MAX_PRE_SAMPLES*sizeof(ALfloat)
 548             );
 549
 550             /* Now resample, then filter and mix to the appropriate outputs. */
 551             ResampledData = Resample(&voice->SincState,
 552                 &SrcData[MAX_PRE_SAMPLES], DataPosFrac, increment,
 553                 Device->ResampledData, DstBufferSize
 554             );
 555             {
 556                 DirectParams *parms = &voice->Chan[chan].Direct;
 557                 const ALfloat *samples;
 558
 559                 samples = DoFilters(
 560                     &parms->LowPass, &parms->HighPass, Device->FilteredData,
 561                     ResampledData, DstBufferSize, parms->FilterType
 562                 );
 563                 if(!voice->IsHrtf)
 564                 {
 565                     ALfloat *restrict currents = parms->Gains.Current;
 566                     const ALfloat *targets = parms->Gains.Target;
 567                     MixGains gains[MAX_OUTPUT_CHANNELS];
 568
 569                     if(!Counter)
 570                     {
 571                         for(j = 0;j < voice->DirectOut.Channels;j++)
 572                         {
 573                             gains[j].Target = targets[j];
 574                             gains[j].Current = gains[j].Target;
 575                             gains[j].Step = 0.0f;
 576                         }
 577                     }
 578                     else
 579                     {
 580                         for(j = 0;j < voice->DirectOut.Channels;j++)
 581                         {
 582                             ALfloat diff;
 583                             gains[j].Target = targets[j];
 584                             gains[j].Current = currents[j];
 585                             diff = gains[j].Target - gains[j].Current;
 586                             if(fabsf(diff) >= GAIN_SILENCE_THRESHOLD)
 587                                 gains[j].Step = diff * Delta;
 588                             else
 589                             {
 590                                 gains[j].Current = gains[j].Target;
 591                                 gains[j].Step = 0.0f;
 592                             }
 593                         }
 594                     }
 595
 596                     MixSamples(samples, voice->DirectOut.Channels, voice->DirectOut.Buffer,
 597                                gains, Counter, OutPos, DstBufferSize);
 598
 599                     for(j = 0;j < voice->DirectOut.Channels;j++)
 600                         currents[j] = gains[j].Current;
 601                 }
 602                 else
 603                 {
 604                     MixHrtfParams hrtfparams;
 605                     int lidx, ridx;
 606
 607                     if(!Counter)
 608                     {
 609                         parms->Hrtf.Current = parms->Hrtf.Target;
 610                         for(j = 0;j < HRIR_LENGTH;j++)
 611                         {
 612                             hrtfparams.Steps.Coeffs[j][0] = 0.0f;
 613                             hrtfparams.Steps.Coeffs[j][1] = 0.0f;
 614                         }
 615                         hrtfparams.Steps.Delay[0] = 0;
 616                         hrtfparams.Steps.Delay[1] = 0;
 617                     }
 618                     else
 619                     {
 620                         ALfloat coeffdiff;
 621                         ALint delaydiff;
 622                         for(j = 0;j < IrSize;j++)
 623                         {
 624                             coeffdiff = parms->Hrtf.Target.Coeffs[j][0] - parms->Hrtf.Current.Coeffs[j][0];
 625                             hrtfparams.Steps.Coeffs[j][0] = coeffdiff * Delta;
 626                             coeffdiff = parms->Hrtf.Target.Coeffs[j][1] - parms->Hrtf.Current.Coeffs[j][1];
 627                             hrtfparams.Steps.Coeffs[j][1] = coeffdiff * Delta;
 628                         }
 629                         delaydiff = (ALint)(parms->Hrtf.Target.Delay[0] - parms->Hrtf.Current.Delay[0]);
 630                         hrtfparams.Steps.Delay[0] = fastf2i((ALfloat)delaydiff * Delta);
 631                         delaydiff = (ALint)(parms->Hrtf.Target.Delay[1] - parms->Hrtf.Current.Delay[1]);
 632                         hrtfparams.Steps.Delay[1] = fastf2i((ALfloat)delaydiff * Delta);
 633                     }
 634                     hrtfparams.Target = &parms->Hrtf.Target;
 635                     hrtfparams.Current = &parms->Hrtf.Current;
 636
 637                     lidx = GetChannelIdxByName(Device->RealOut, FrontLeft);
 638                     ridx = GetChannelIdxByName(Device->RealOut, FrontRight);
 639                     assert(lidx != -1 && ridx != -1);
 640
 641                     MixHrtfSamples(voice->DirectOut.Buffer, lidx, ridx, samples, Counter,
 642                                    voice->Offset, OutPos, IrSize, &hrtfparams,
 643                                    &parms->Hrtf.State, DstBufferSize);
 644                 }
 645             }
 646
 647             for(send = 0;send < Device->NumAuxSends;send++)
 648             {
 649                 SendParams *parms = &voice->Chan[chan].Send[send];
 650                 ALfloat *restrict currents = parms->Gains.Current;
 651                 const ALfloat *targets = parms->Gains.Target;
 652                 MixGains gains[MAX_OUTPUT_CHANNELS];
 653                 const ALfloat *samples;
 654
 655                 if(!voice->SendOut[send].Buffer)
 656                     continue;
 657
 658                 samples = DoFilters(
 659                     &parms->LowPass, &parms->HighPass, Device->FilteredData,
 660                     ResampledData, DstBufferSize, parms->FilterType
 661                 );
 662
 663                 if(!Counter)
 664                 {
 665                     for(j = 0;j < voice->SendOut[send].Channels;j++)
 666                     {
 667                         gains[j].Target = targets[j];
 668                         gains[j].Current = gains[j].Target;
 669                         gains[j].Step = 0.0f;
 670                     }
 671                 }
 672                 else
 673                 {
 674                     for(j = 0;j < voice->SendOut[send].Channels;j++)
 675                     {
 676                         ALfloat diff;
 677                         gains[j].Target = targets[j];
 678                         gains[j].Current = currents[j];
 679                         diff = gains[j].Target - gains[j].Current;
 680                         if(fabsf(diff) >= GAIN_SILENCE_THRESHOLD)
 681                             gains[j].Step = diff * Delta;
 682                         else
 683                         {
 684                             gains[j].Current = gains[j].Target;
 685                             gains[j].Step = 0.0f;
 686                         }
 687                     }
 688                 }
 689
 690                 MixSamples(samples,
 691                     voice->SendOut[send].Channels, voice->SendOut[send].Buffer,
 692                     gains, Counter, OutPos, DstBufferSize
 693                 );
 694
 695                 for(j = 0;j < voice->SendOut[send].Channels;j++)
 696                     currents[j] = gains[j].Current;
 697             }
 698         }
 699         /* Update positions */
 700         DataPosFrac += increment*DstBufferSize;
 701         DataPosInt  += DataPosFrac>>FRACTIONBITS;
 702         DataPosFrac &= FRACTIONMASK;
 703
 704         OutPos += DstBufferSize;
 705         voice->Offset += DstBufferSize;
 706
 707         /* Handle looping sources */
 708         while(1)
 709         {
 710             const ALbuffer *ALBuffer;
 711             ALuint DataSize = 0;
 712             ALuint LoopStart = 0;
 713             ALuint LoopEnd = 0;
 714
 715             if((ALBuffer=BufferListItem->buffer) != NULL)
 716             {
 717                 DataSize = ALBuffer->SampleLen;
 718                 LoopStart = ALBuffer->LoopStart;
 719                 LoopEnd = ALBuffer->LoopEnd;
 720                 if(LoopEnd > DataPosInt)
 721                     break;
 722             }
 723
 724             if(Looping && Source->SourceType == AL_STATIC)
 725             {
 726                 assert(LoopEnd > LoopStart);
 727                 DataPosInt = ((DataPosInt-LoopStart)%(LoopEnd-LoopStart)) + LoopStart;
 728                 break;
 729             }
 730
 731             if(DataSize > DataPosInt)
 732                 break;
 733
 734             if(!(BufferListItem=BufferListItem->next))
 735             {
 736                 if(Looping)
 737                     BufferListItem = ATOMIC_LOAD(&Source->queue);
 738                 else
 739                 {
 740                     State = AL_STOPPED;
 741                     BufferListItem = NULL;
 742                     DataPosInt = 0;
 743                     DataPosFrac = 0;
 744                     break;
 745                 }
 746             }
 747
 748             DataPosInt -= DataSize;
 749         }
 750     } while(State == AL_PLAYING && OutPos < SamplesToDo);
 751
 752     voice->Moving = AL_TRUE;
 753
 754     /* Update source info */
 755     Source->state = State;
 756     ATOMIC_STORE(&Source->current_buffer,    BufferListItem, almemory_order_relaxed);
 757     ATOMIC_STORE(&Source->position,          DataPosInt, almemory_order_relaxed);
 758     ATOMIC_STORE(&Source->position_fraction, DataPosFrac);
 759 }