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.
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.
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
34 #include "alListener.h"
35 #include "alAuxEffectSlot.h"
38 #include "mixer_defs.h"
41 static_assert((INT_MAX>>FRACTIONBITS)/MAX_PITCH > BUFFERSIZE,
42 "MAX_PITCH and/or BUFFERSIZE are too large for FRACTIONBITS!");
44 extern inline void InitiatePositionArrays(ALuint frac, ALuint increment, ALuint *frac_arr, ALuint *pos_arr, ALuint size);
46 alignas(16) union ResamplerCoeffs ResampleCoeffs;
56 ResamplerDefault = LinearResampler
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!");
64 static HrtfMixerFunc MixHrtfSamples = MixHrtf_C;
65 static MixerFunc MixSamples = Mix_C;
66 static ResamplerFunc ResampleSamples = Resample_point32_C;
68 static inline HrtfMixerFunc SelectHrtfMixer(void)
71 if((CPUCapFlags&CPU_CAP_SSE))
75 if((CPUCapFlags&CPU_CAP_NEON))
82 static inline MixerFunc SelectMixer(void)
85 if((CPUCapFlags&CPU_CAP_SSE))
89 if((CPUCapFlags&CPU_CAP_NEON))
96 static inline ResamplerFunc SelectResampler(enum Resampler resampler)
101 return Resample_point32_C;
102 case LinearResampler:
104 if((CPUCapFlags&CPU_CAP_SSE4_1))
105 return Resample_lerp32_SSE41;
108 if((CPUCapFlags&CPU_CAP_SSE2))
109 return Resample_lerp32_SSE2;
111 return Resample_lerp32_C;
114 if((CPUCapFlags&CPU_CAP_SSE4_1))
115 return Resample_fir4_32_SSE41;
118 if((CPUCapFlags&CPU_CAP_SSE3))
119 return Resample_fir4_32_SSE3;
121 return Resample_fir4_32_C;
124 if((CPUCapFlags&CPU_CAP_SSE4_1))
125 return Resample_fir8_32_SSE41;
128 if((CPUCapFlags&CPU_CAP_SSE3))
129 return Resample_fir8_32_SSE3;
131 return Resample_fir8_32_C;
134 if((CPUCapFlags&CPU_CAP_SSE))
135 return Resample_bsinc32_SSE;
137 return Resample_bsinc32_C;
140 return Resample_point32_C;
144 /* The sinc resampler makes use of a Kaiser window to limit the needed sample
145 * points to 4 and 8, respectively.
149 #define M_PI (3.14159265358979323846)
151 static inline double Sinc(double x)
153 if(x == 0.0) return 1.0;
154 return sin(x*M_PI) / (x*M_PI);
157 /* The zero-order modified Bessel function of the first kind, used for the
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
163 static double BesselI_0(double x)
165 double term, sum, x2, y, last_sum;
168 /* Start at k=1 since k=0 is trivial. */
174 /* Let the integration converge until the term of the sum is no longer
183 } while(sum != last_sum);
187 /* Calculate a Kaiser window from the given beta value and a normalized k
190 * w(k) = { I_0(B sqrt(1 - k^2)) / I_0(B), -1 <= k <= 1
193 * Where k can be calculated as:
195 * k = i / l, where -l <= i <= l.
199 * k = 2 i / M - 1, where 0 <= i <= M.
201 static inline double Kaiser(double b, double k)
203 if(k <= -1.0 || k >= 1.0) return 0.0;
204 return BesselI_0(b * sqrt(1.0 - (k*k))) / BesselI_0(b);
207 static inline double CalcKaiserBeta(double rejection)
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));
217 static float SincKaiser(double r, double x)
219 /* Limit rippling to -60dB. */
220 return (float)(Kaiser(CalcKaiserBeta(60.0), x / r) * Sinc(x));
224 void aluInitMixer(void)
226 enum Resampler resampler = ResamplerDefault;
230 if(ConfigValueStr(NULL, NULL, "resampler", &str))
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)
244 WARN("Resampler option \"cubic\" is deprecated, using sinc4\n");
245 resampler = FIR4Resampler;
250 long n = strtol(str, &end, 0);
251 if(*end == '\0' && (n == PointResampler || n == LinearResampler || n == FIR4Resampler))
254 WARN("Invalid resampler: %s\n", str);
258 if(resampler == FIR8Resampler)
259 for(i = 0;i < FRACTIONONE;i++)
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);
271 else if(resampler == FIR4Resampler)
272 for(i = 0;i < FRACTIONONE;i++)
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);
281 MixHrtfSamples = SelectHrtfMixer();
282 MixSamples = SelectMixer();
283 ResampleSamples = SelectResampler(resampler);
287 static inline ALfloat Sample_ALbyte(ALbyte val)
288 { return val * (1.0f/127.0f); }
290 static inline ALfloat Sample_ALshort(ALshort val)
291 { return val * (1.0f/32767.0f); }
293 static inline ALfloat Sample_ALfloat(ALfloat val)
296 #define DECL_TEMPLATE(T) \
297 static inline void Load_##T(ALfloat *dst, const T *src, ALuint srcstep, ALuint samples)\
300 for(i = 0;i < samples;i++) \
301 dst[i] = Sample_##T(src[i*srcstep]); \
304 DECL_TEMPLATE(ALbyte)
305 DECL_TEMPLATE(ALshort)
306 DECL_TEMPLATE(ALfloat)
310 static void LoadSamples(ALfloat *dst, const ALvoid *src, ALuint srcstep, enum FmtType srctype, ALuint samples)
315 Load_ALbyte(dst, src, srcstep, samples);
318 Load_ALshort(dst, src, srcstep, samples);
321 Load_ALfloat(dst, src, srcstep, samples);
326 static inline void SilenceSamples(ALfloat *dst, ALuint samples)
329 for(i = 0;i < samples;i++)
334 static const ALfloat *DoFilters(ALfilterState *lpfilter, ALfilterState *hpfilter,
335 ALfloat *restrict dst, const ALfloat *restrict src,
336 ALuint numsamples, enum ActiveFilters type)
342 ALfilterState_processPassthru(lpfilter, src, numsamples);
343 ALfilterState_processPassthru(hpfilter, src, numsamples);
347 ALfilterState_process(lpfilter, dst, src, numsamples);
348 ALfilterState_processPassthru(hpfilter, dst, numsamples);
351 ALfilterState_processPassthru(lpfilter, src, numsamples);
352 ALfilterState_process(hpfilter, dst, src, numsamples);
356 for(i = 0;i < numsamples;)
359 ALuint todo = minu(256, numsamples-i);
361 ALfilterState_process(lpfilter, temp, src+i, todo);
362 ALfilterState_process(hpfilter, dst+i, temp, todo);
371 ALvoid MixSource(ALvoice *voice, ALsource *Source, ALCdevice *Device, ALuint SamplesToDo)
373 ResamplerFunc Resample;
374 ALbufferlistitem *BufferListItem;
375 ALuint DataPosInt, DataPosFrac;
386 /* Get source info */
387 State = Source->state;
388 BufferListItem = ATOMIC_LOAD(&Source->current_buffer);
389 DataPosInt = Source->position;
390 DataPosFrac = Source->position_fraction;
391 Looping = Source->Looping;
392 NumChannels = Source->NumChannels;
393 SampleSize = Source->SampleSize;
394 increment = voice->Step;
396 IrSize = (Device->Hrtf ? GetHrtfIrSize(Device->Hrtf) : 0);
398 Resample = ((increment == FRACTIONONE && DataPosFrac == 0) ?
399 Resample_copy32_C : ResampleSamples);
403 ALuint SrcBufferSize, DstBufferSize;
405 /* Figure out how many buffer samples will be needed */
406 DataSize64 = SamplesToDo-OutPos;
407 DataSize64 *= increment;
408 DataSize64 += DataPosFrac+FRACTIONMASK;
409 DataSize64 >>= FRACTIONBITS;
410 DataSize64 += MAX_POST_SAMPLES+MAX_PRE_SAMPLES;
412 SrcBufferSize = (ALuint)mini64(DataSize64, BUFFERSIZE);
414 /* Figure out how many samples we can actually mix from this. */
415 DataSize64 = SrcBufferSize;
416 DataSize64 -= MAX_POST_SAMPLES+MAX_PRE_SAMPLES;
417 DataSize64 <<= FRACTIONBITS;
418 DataSize64 -= DataPosFrac;
420 DstBufferSize = (ALuint)((DataSize64+(increment-1)) / increment);
421 DstBufferSize = minu(DstBufferSize, (SamplesToDo-OutPos));
423 /* Some mixers like having a multiple of 4, so try to give that unless
424 * this is the last update. */
425 if(OutPos+DstBufferSize < SamplesToDo)
428 for(chan = 0;chan < NumChannels;chan++)
430 const ALfloat *ResampledData;
431 ALfloat *SrcData = Device->SourceData;
434 /* Load the previous samples into the source data first. */
435 memcpy(SrcData, voice->PrevSamples[chan], MAX_PRE_SAMPLES*sizeof(ALfloat));
436 SrcDataSize = MAX_PRE_SAMPLES;
438 if(Source->SourceType == AL_STATIC)
440 const ALbuffer *ALBuffer = BufferListItem->buffer;
441 const ALubyte *Data = ALBuffer->data;
445 /* Offset buffer data to current channel */
446 Data += chan*SampleSize;
448 /* If current pos is beyond the loop range, do not loop */
449 if(Looping == AL_FALSE || DataPosInt >= (ALuint)ALBuffer->LoopEnd)
453 /* Load what's left to play from the source buffer, and
454 * clear the rest of the temp buffer */
456 DataSize = minu(SrcBufferSize - SrcDataSize, ALBuffer->SampleLen - pos);
458 LoadSamples(&SrcData[SrcDataSize], &Data[pos * NumChannels*SampleSize],
459 NumChannels, ALBuffer->FmtType, DataSize);
460 SrcDataSize += DataSize;
462 SilenceSamples(&SrcData[SrcDataSize], SrcBufferSize - SrcDataSize);
463 SrcDataSize += SrcBufferSize - SrcDataSize;
467 ALuint LoopStart = ALBuffer->LoopStart;
468 ALuint LoopEnd = ALBuffer->LoopEnd;
470 /* Load what's left of this loop iteration, then load
471 * repeats of the loop section */
473 DataSize = LoopEnd - pos;
474 DataSize = minu(SrcBufferSize - SrcDataSize, DataSize);
476 LoadSamples(&SrcData[SrcDataSize], &Data[pos * NumChannels*SampleSize],
477 NumChannels, ALBuffer->FmtType, DataSize);
478 SrcDataSize += DataSize;
480 DataSize = LoopEnd-LoopStart;
481 while(SrcBufferSize > SrcDataSize)
483 DataSize = minu(SrcBufferSize - SrcDataSize, DataSize);
485 LoadSamples(&SrcData[SrcDataSize], &Data[LoopStart * NumChannels*SampleSize],
486 NumChannels, ALBuffer->FmtType, DataSize);
487 SrcDataSize += DataSize;
493 /* Crawl the buffer queue to fill in the temp buffer */
494 ALbufferlistitem *tmpiter = BufferListItem;
495 ALuint pos = DataPosInt;
497 while(tmpiter && SrcBufferSize > SrcDataSize)
499 const ALbuffer *ALBuffer;
500 if((ALBuffer=tmpiter->buffer) != NULL)
502 const ALubyte *Data = ALBuffer->data;
503 ALuint DataSize = ALBuffer->SampleLen;
505 /* Skip the data already played */
510 Data += (pos*NumChannels + chan)*SampleSize;
514 DataSize = minu(SrcBufferSize - SrcDataSize, DataSize);
515 LoadSamples(&SrcData[SrcDataSize], Data, NumChannels,
516 ALBuffer->FmtType, DataSize);
517 SrcDataSize += DataSize;
520 tmpiter = tmpiter->next;
521 if(!tmpiter && Looping)
522 tmpiter = ATOMIC_LOAD(&Source->queue);
525 SilenceSamples(&SrcData[SrcDataSize], SrcBufferSize - SrcDataSize);
526 SrcDataSize += SrcBufferSize - SrcDataSize;
531 /* Store the last source samples used for next time. */
532 memcpy(voice->PrevSamples[chan],
533 &SrcData[(increment*DstBufferSize + DataPosFrac)>>FRACTIONBITS],
534 MAX_PRE_SAMPLES*sizeof(ALfloat)
537 /* Now resample, then filter and mix to the appropriate outputs. */
538 ResampledData = Resample(&voice->SincState,
539 &SrcData[MAX_PRE_SAMPLES], DataPosFrac, increment,
540 Device->ResampledData, DstBufferSize
543 DirectParams *parms = &voice->Direct;
544 const ALfloat *samples;
547 &parms->Filters[chan].LowPass, &parms->Filters[chan].HighPass,
548 Device->FilteredData, ResampledData, DstBufferSize,
549 parms->Filters[chan].ActiveType
552 MixSamples(samples, parms->OutChannels, parms->OutBuffer, parms->Gains[chan],
553 parms->Counter, OutPos, DstBufferSize);
555 MixHrtfSamples(parms->OutBuffer, samples, parms->Counter, voice->Offset,
556 OutPos, IrSize, &parms->Hrtf[chan].Params,
557 &parms->Hrtf[chan].State, DstBufferSize);
560 for(j = 0;j < Device->NumAuxSends;j++)
562 SendParams *parms = &voice->Send[j];
563 const ALfloat *samples;
565 if(!parms->OutBuffer)
569 &parms->Filters[chan].LowPass, &parms->Filters[chan].HighPass,
570 Device->FilteredData, ResampledData, DstBufferSize,
571 parms->Filters[chan].ActiveType
573 MixSamples(samples, 1, parms->OutBuffer, &parms->Gains[chan],
574 parms->Counter, OutPos, DstBufferSize);
577 /* Update positions */
578 DataPosFrac += increment*DstBufferSize;
579 DataPosInt += DataPosFrac>>FRACTIONBITS;
580 DataPosFrac &= FRACTIONMASK;
582 OutPos += DstBufferSize;
583 voice->Offset += DstBufferSize;
584 voice->Direct.Counter = maxu(voice->Direct.Counter, DstBufferSize) - DstBufferSize;
585 for(j = 0;j < Device->NumAuxSends;j++)
586 voice->Send[j].Counter = maxu(voice->Send[j].Counter, DstBufferSize) - DstBufferSize;
588 /* Handle looping sources */
591 const ALbuffer *ALBuffer;
593 ALuint LoopStart = 0;
596 if((ALBuffer=BufferListItem->buffer) != NULL)
598 DataSize = ALBuffer->SampleLen;
599 LoopStart = ALBuffer->LoopStart;
600 LoopEnd = ALBuffer->LoopEnd;
601 if(LoopEnd > DataPosInt)
605 if(Looping && Source->SourceType == AL_STATIC)
607 assert(LoopEnd > LoopStart);
608 DataPosInt = ((DataPosInt-LoopStart)%(LoopEnd-LoopStart)) + LoopStart;
612 if(DataSize > DataPosInt)
615 if(!(BufferListItem=BufferListItem->next))
618 BufferListItem = ATOMIC_LOAD(&Source->queue);
622 BufferListItem = NULL;
629 DataPosInt -= DataSize;
631 } while(State == AL_PLAYING && OutPos < SamplesToDo);
633 /* Update source info */
634 Source->state = State;
635 ATOMIC_STORE(&Source->current_buffer, BufferListItem);
636 Source->position = DataPosInt;
637 Source->position_fraction = DataPosFrac;