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-rw-r--r--Alc/mixvoice.c762
1 files changed, 0 insertions, 762 deletions
diff --git a/Alc/mixvoice.c b/Alc/mixvoice.c
deleted file mode 100644
index d019b898..00000000
--- a/Alc/mixvoice.c
+++ /dev/null
@@ -1,762 +0,0 @@
-/**
- * OpenAL cross platform audio library
- * Copyright (C) 1999-2007 by authors.
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Library General Public
- * License as published by the Free Software Foundation; either
- * version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
- * Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- * License along with this library; if not, write to the
- * Free Software Foundation, Inc.,
- * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
- * Or go to http://www.gnu.org/copyleft/lgpl.html
- */
-
-#include "config.h"
-
-#include <math.h>
-#include <stdlib.h>
-#include <string.h>
-#include <ctype.h>
-#include <assert.h>
-
-#include "alMain.h"
-#include "AL/al.h"
-#include "AL/alc.h"
-#include "alSource.h"
-#include "alBuffer.h"
-#include "alListener.h"
-#include "alAuxEffectSlot.h"
-#include "sample_cvt.h"
-#include "alu.h"
-#include "alconfig.h"
-#include "ringbuffer.h"
-
-#include "cpu_caps.h"
-#include "mixer/defs.h"
-
-
-static_assert((INT_MAX>>FRACTIONBITS)/MAX_PITCH > BUFFERSIZE,
- "MAX_PITCH and/or BUFFERSIZE are too large for FRACTIONBITS!");
-
-extern inline void InitiatePositionArrays(ALsizei frac, ALint increment, ALsizei *restrict frac_arr, ALsizei *restrict pos_arr, ALsizei size);
-
-
-/* BSinc24 requires up to 23 extra samples before the current position, and 24 after. */
-static_assert(MAX_RESAMPLE_PADDING >= 24, "MAX_RESAMPLE_PADDING must be at least 24!");
-
-
-enum Resampler ResamplerDefault = LinearResampler;
-
-MixerFunc MixSamples = Mix_C;
-RowMixerFunc MixRowSamples = MixRow_C;
-static HrtfMixerFunc MixHrtfSamples = MixHrtf_C;
-static HrtfMixerBlendFunc MixHrtfBlendSamples = MixHrtfBlend_C;
-
-static MixerFunc SelectMixer(void)
-{
-#ifdef HAVE_NEON
- if((CPUCapFlags&CPU_CAP_NEON))
- return Mix_Neon;
-#endif
-#ifdef HAVE_SSE
- if((CPUCapFlags&CPU_CAP_SSE))
- return Mix_SSE;
-#endif
- return Mix_C;
-}
-
-static RowMixerFunc SelectRowMixer(void)
-{
-#ifdef HAVE_NEON
- if((CPUCapFlags&CPU_CAP_NEON))
- return MixRow_Neon;
-#endif
-#ifdef HAVE_SSE
- if((CPUCapFlags&CPU_CAP_SSE))
- return MixRow_SSE;
-#endif
- return MixRow_C;
-}
-
-static inline HrtfMixerFunc SelectHrtfMixer(void)
-{
-#ifdef HAVE_NEON
- if((CPUCapFlags&CPU_CAP_NEON))
- return MixHrtf_Neon;
-#endif
-#ifdef HAVE_SSE
- if((CPUCapFlags&CPU_CAP_SSE))
- return MixHrtf_SSE;
-#endif
- return MixHrtf_C;
-}
-
-static inline HrtfMixerBlendFunc SelectHrtfBlendMixer(void)
-{
-#ifdef HAVE_NEON
- if((CPUCapFlags&CPU_CAP_NEON))
- return MixHrtfBlend_Neon;
-#endif
-#ifdef HAVE_SSE
- if((CPUCapFlags&CPU_CAP_SSE))
- return MixHrtfBlend_SSE;
-#endif
- return MixHrtfBlend_C;
-}
-
-ResamplerFunc SelectResampler(enum Resampler resampler)
-{
- switch(resampler)
- {
- case PointResampler:
- return Resample_point_C;
- case LinearResampler:
-#ifdef HAVE_NEON
- if((CPUCapFlags&CPU_CAP_NEON))
- return Resample_lerp_Neon;
-#endif
-#ifdef HAVE_SSE4_1
- if((CPUCapFlags&CPU_CAP_SSE4_1))
- return Resample_lerp_SSE41;
-#endif
-#ifdef HAVE_SSE2
- if((CPUCapFlags&CPU_CAP_SSE2))
- return Resample_lerp_SSE2;
-#endif
- return Resample_lerp_C;
- case FIR4Resampler:
- return Resample_cubic_C;
- case BSinc12Resampler:
- case BSinc24Resampler:
-#ifdef HAVE_NEON
- if((CPUCapFlags&CPU_CAP_NEON))
- return Resample_bsinc_Neon;
-#endif
-#ifdef HAVE_SSE
- if((CPUCapFlags&CPU_CAP_SSE))
- return Resample_bsinc_SSE;
-#endif
- return Resample_bsinc_C;
- }
-
- return Resample_point_C;
-}
-
-
-void aluInitMixer(void)
-{
- const char *str;
-
- if(ConfigValueStr(NULL, NULL, "resampler", &str))
- {
- if(strcasecmp(str, "point") == 0 || strcasecmp(str, "none") == 0)
- ResamplerDefault = PointResampler;
- else if(strcasecmp(str, "linear") == 0)
- ResamplerDefault = LinearResampler;
- else if(strcasecmp(str, "cubic") == 0)
- ResamplerDefault = FIR4Resampler;
- else if(strcasecmp(str, "bsinc12") == 0)
- ResamplerDefault = BSinc12Resampler;
- else if(strcasecmp(str, "bsinc24") == 0)
- ResamplerDefault = BSinc24Resampler;
- else if(strcasecmp(str, "bsinc") == 0)
- {
- WARN("Resampler option \"%s\" is deprecated, using bsinc12\n", str);
- ResamplerDefault = BSinc12Resampler;
- }
- else if(strcasecmp(str, "sinc4") == 0 || strcasecmp(str, "sinc8") == 0)
- {
- WARN("Resampler option \"%s\" is deprecated, using cubic\n", str);
- ResamplerDefault = FIR4Resampler;
- }
- else
- {
- char *end;
- long n = strtol(str, &end, 0);
- if(*end == '\0' && (n == PointResampler || n == LinearResampler || n == FIR4Resampler))
- ResamplerDefault = n;
- else
- WARN("Invalid resampler: %s\n", str);
- }
- }
-
- MixHrtfBlendSamples = SelectHrtfBlendMixer();
- MixHrtfSamples = SelectHrtfMixer();
- MixSamples = SelectMixer();
- MixRowSamples = SelectRowMixer();
-}
-
-
-static void SendAsyncEvent(ALCcontext *context, ALuint enumtype, ALenum type,
- ALuint objid, ALuint param, const char *msg)
-{
- AsyncEvent evt = ASYNC_EVENT(enumtype);
- evt.u.user.type = type;
- evt.u.user.id = objid;
- evt.u.user.param = param;
- strcpy(evt.u.user.msg, msg);
- if(ll_ringbuffer_write(context->AsyncEvents, (const char*)&evt, 1) == 1)
- alsem_post(&context->EventSem);
-}
-
-
-static inline ALfloat Sample_ALubyte(ALubyte val)
-{ return (val-128) * (1.0f/128.0f); }
-
-static inline ALfloat Sample_ALshort(ALshort val)
-{ return val * (1.0f/32768.0f); }
-
-static inline ALfloat Sample_ALfloat(ALfloat val)
-{ return val; }
-
-static inline ALfloat Sample_ALdouble(ALdouble val)
-{ return (ALfloat)val; }
-
-typedef ALubyte ALmulaw;
-static inline ALfloat Sample_ALmulaw(ALmulaw val)
-{ return muLawDecompressionTable[val] * (1.0f/32768.0f); }
-
-typedef ALubyte ALalaw;
-static inline ALfloat Sample_ALalaw(ALalaw val)
-{ return aLawDecompressionTable[val] * (1.0f/32768.0f); }
-
-#define DECL_TEMPLATE(T) \
-static inline void Load_##T(ALfloat *restrict dst, const T *restrict src, \
- ALint srcstep, ALsizei samples) \
-{ \
- ALsizei i; \
- for(i = 0;i < samples;i++) \
- dst[i] += Sample_##T(src[i*srcstep]); \
-}
-
-DECL_TEMPLATE(ALubyte)
-DECL_TEMPLATE(ALshort)
-DECL_TEMPLATE(ALfloat)
-DECL_TEMPLATE(ALdouble)
-DECL_TEMPLATE(ALmulaw)
-DECL_TEMPLATE(ALalaw)
-
-#undef DECL_TEMPLATE
-
-static void LoadSamples(ALfloat *restrict dst, const ALvoid *restrict src, ALint srcstep,
- enum FmtType srctype, ALsizei samples)
-{
-#define HANDLE_FMT(ET, ST) case ET: Load_##ST(dst, src, srcstep, samples); break
- switch(srctype)
- {
- HANDLE_FMT(FmtUByte, ALubyte);
- HANDLE_FMT(FmtShort, ALshort);
- HANDLE_FMT(FmtFloat, ALfloat);
- HANDLE_FMT(FmtDouble, ALdouble);
- HANDLE_FMT(FmtMulaw, ALmulaw);
- HANDLE_FMT(FmtAlaw, ALalaw);
- }
-#undef HANDLE_FMT
-}
-
-
-static const ALfloat *DoFilters(BiquadFilter *lpfilter, BiquadFilter *hpfilter,
- ALfloat *restrict dst, const ALfloat *restrict src,
- ALsizei numsamples, enum ActiveFilters type)
-{
- ALsizei i;
- switch(type)
- {
- case AF_None:
- BiquadFilter_passthru(lpfilter, numsamples);
- BiquadFilter_passthru(hpfilter, numsamples);
- break;
-
- case AF_LowPass:
- BiquadFilter_process(lpfilter, dst, src, numsamples);
- BiquadFilter_passthru(hpfilter, numsamples);
- return dst;
- case AF_HighPass:
- BiquadFilter_passthru(lpfilter, numsamples);
- BiquadFilter_process(hpfilter, dst, src, numsamples);
- return dst;
-
- case AF_BandPass:
- for(i = 0;i < numsamples;)
- {
- ALfloat temp[256];
- ALsizei todo = mini(256, numsamples-i);
-
- BiquadFilter_process(lpfilter, temp, src+i, todo);
- BiquadFilter_process(hpfilter, dst+i, temp, todo);
- i += todo;
- }
- return dst;
- }
- return src;
-}
-
-
-/* This function uses these device temp buffers. */
-#define SOURCE_DATA_BUF 0
-#define RESAMPLED_BUF 1
-#define FILTERED_BUF 2
-#define NFC_DATA_BUF 3
-ALboolean MixSource(ALvoice *voice, ALuint SourceID, ALCcontext *Context, ALsizei SamplesToDo)
-{
- ALCdevice *Device = Context->Device;
- ALbufferlistitem *BufferListItem;
- ALbufferlistitem *BufferLoopItem;
- ALsizei NumChannels, SampleSize;
- ALbitfieldSOFT enabledevt;
- ALsizei buffers_done = 0;
- ResamplerFunc Resample;
- ALsizei DataPosInt;
- ALsizei DataPosFrac;
- ALint64 DataSize64;
- ALint increment;
- ALsizei Counter;
- ALsizei OutPos;
- ALsizei IrSize;
- bool isplaying;
- bool firstpass;
- bool isstatic;
- ALsizei chan;
- ALsizei send;
-
- /* Get source info */
- isplaying = true; /* Will only be called while playing. */
- isstatic = !!(voice->Flags&VOICE_IS_STATIC);
- DataPosInt = ATOMIC_LOAD(&voice->position, almemory_order_acquire);
- DataPosFrac = ATOMIC_LOAD(&voice->position_fraction, almemory_order_relaxed);
- BufferListItem = ATOMIC_LOAD(&voice->current_buffer, almemory_order_relaxed);
- BufferLoopItem = ATOMIC_LOAD(&voice->loop_buffer, almemory_order_relaxed);
- NumChannels = voice->NumChannels;
- SampleSize = voice->SampleSize;
- increment = voice->Step;
-
- IrSize = (Device->HrtfHandle ? Device->HrtfHandle->irSize : 0);
-
- Resample = ((increment == FRACTIONONE && DataPosFrac == 0) ?
- Resample_copy_C : voice->Resampler);
-
- Counter = (voice->Flags&VOICE_IS_FADING) ? SamplesToDo : 0;
- firstpass = true;
- OutPos = 0;
-
- do {
- ALsizei SrcBufferSize, DstBufferSize;
-
- /* Figure out how many buffer samples will be needed */
- DataSize64 = SamplesToDo-OutPos;
- DataSize64 *= increment;
- DataSize64 += DataPosFrac+FRACTIONMASK;
- DataSize64 >>= FRACTIONBITS;
- DataSize64 += MAX_RESAMPLE_PADDING*2;
- SrcBufferSize = (ALsizei)mini64(DataSize64, BUFFERSIZE);
-
- /* Figure out how many samples we can actually mix from this. */
- DataSize64 = SrcBufferSize;
- DataSize64 -= MAX_RESAMPLE_PADDING*2;
- DataSize64 <<= FRACTIONBITS;
- DataSize64 -= DataPosFrac;
- DstBufferSize = (ALsizei)mini64((DataSize64+(increment-1)) / increment,
- SamplesToDo - OutPos);
-
- /* Some mixers like having a multiple of 4, so try to give that unless
- * this is the last update. */
- if(DstBufferSize < SamplesToDo-OutPos)
- DstBufferSize &= ~3;
-
- /* It's impossible to have a buffer list item with no entries. */
- assert(BufferListItem->num_buffers > 0);
-
- for(chan = 0;chan < NumChannels;chan++)
- {
- const ALfloat *ResampledData;
- ALfloat *SrcData = Device->TempBuffer[SOURCE_DATA_BUF];
- ALsizei FilledAmt;
-
- /* Load the previous samples into the source data first, and clear the rest. */
- memcpy(SrcData, voice->PrevSamples[chan], MAX_RESAMPLE_PADDING*sizeof(ALfloat));
- memset(SrcData+MAX_RESAMPLE_PADDING, 0, (BUFFERSIZE-MAX_RESAMPLE_PADDING)*
- sizeof(ALfloat));
- FilledAmt = MAX_RESAMPLE_PADDING;
-
- if(isstatic)
- {
- /* TODO: For static sources, loop points are taken from the
- * first buffer (should be adjusted by any buffer offset, to
- * possibly be added later).
- */
- const ALbuffer *Buffer0 = BufferListItem->buffers[0];
- const ALsizei LoopStart = Buffer0->LoopStart;
- const ALsizei LoopEnd = Buffer0->LoopEnd;
- const ALsizei LoopSize = LoopEnd - LoopStart;
-
- /* If current pos is beyond the loop range, do not loop */
- if(!BufferLoopItem || DataPosInt >= LoopEnd)
- {
- ALsizei SizeToDo = SrcBufferSize - FilledAmt;
- ALsizei CompLen = 0;
- ALsizei i;
-
- BufferLoopItem = NULL;
-
- for(i = 0;i < BufferListItem->num_buffers;i++)
- {
- const ALbuffer *buffer = BufferListItem->buffers[i];
- const ALubyte *Data = buffer->data;
- ALsizei DataSize;
-
- if(DataPosInt >= buffer->SampleLen)
- continue;
-
- /* Load what's left to play from the buffer */
- DataSize = mini(SizeToDo, buffer->SampleLen - DataPosInt);
- CompLen = maxi(CompLen, DataSize);
-
- LoadSamples(&SrcData[FilledAmt],
- &Data[(DataPosInt*NumChannels + chan)*SampleSize],
- NumChannels, buffer->FmtType, DataSize
- );
- }
- FilledAmt += CompLen;
- }
- else
- {
- ALsizei SizeToDo = mini(SrcBufferSize - FilledAmt, LoopEnd - DataPosInt);
- ALsizei CompLen = 0;
- ALsizei i;
-
- for(i = 0;i < BufferListItem->num_buffers;i++)
- {
- const ALbuffer *buffer = BufferListItem->buffers[i];
- const ALubyte *Data = buffer->data;
- ALsizei DataSize;
-
- if(DataPosInt >= buffer->SampleLen)
- continue;
-
- /* Load what's left of this loop iteration */
- DataSize = mini(SizeToDo, buffer->SampleLen - DataPosInt);
- CompLen = maxi(CompLen, DataSize);
-
- LoadSamples(&SrcData[FilledAmt],
- &Data[(DataPosInt*NumChannels + chan)*SampleSize],
- NumChannels, buffer->FmtType, DataSize
- );
- }
- FilledAmt += CompLen;
-
- while(SrcBufferSize > FilledAmt)
- {
- const ALsizei SizeToDo = mini(SrcBufferSize - FilledAmt, LoopSize);
-
- CompLen = 0;
- for(i = 0;i < BufferListItem->num_buffers;i++)
- {
- const ALbuffer *buffer = BufferListItem->buffers[i];
- const ALubyte *Data = buffer->data;
- ALsizei DataSize;
-
- if(LoopStart >= buffer->SampleLen)
- continue;
-
- DataSize = mini(SizeToDo, buffer->SampleLen - LoopStart);
- CompLen = maxi(CompLen, DataSize);
-
- LoadSamples(&SrcData[FilledAmt],
- &Data[(LoopStart*NumChannels + chan)*SampleSize],
- NumChannels, buffer->FmtType, DataSize
- );
- }
- FilledAmt += CompLen;
- }
- }
- }
- else
- {
- /* Crawl the buffer queue to fill in the temp buffer */
- ALbufferlistitem *tmpiter = BufferListItem;
- ALsizei pos = DataPosInt;
-
- while(tmpiter && SrcBufferSize > FilledAmt)
- {
- ALsizei SizeToDo = SrcBufferSize - FilledAmt;
- ALsizei CompLen = 0;
- ALsizei i;
-
- for(i = 0;i < tmpiter->num_buffers;i++)
- {
- const ALbuffer *ALBuffer = tmpiter->buffers[i];
- ALsizei DataSize = ALBuffer ? ALBuffer->SampleLen : 0;
-
- if(DataSize > pos)
- {
- const ALubyte *Data = ALBuffer->data;
- Data += (pos*NumChannels + chan)*SampleSize;
-
- DataSize = mini(SizeToDo, DataSize - pos);
- CompLen = maxi(CompLen, DataSize);
-
- LoadSamples(&SrcData[FilledAmt], Data, NumChannels,
- ALBuffer->FmtType, DataSize);
- }
- }
- if(UNLIKELY(!CompLen))
- pos -= tmpiter->max_samples;
- else
- {
- FilledAmt += CompLen;
- if(SrcBufferSize <= FilledAmt)
- break;
- pos = 0;
- }
- tmpiter = ATOMIC_LOAD(&tmpiter->next, almemory_order_acquire);
- if(!tmpiter) tmpiter = BufferLoopItem;
- }
- }
-
- /* Store the last source samples used for next time. */
- memcpy(voice->PrevSamples[chan],
- &SrcData[(increment*DstBufferSize + DataPosFrac)>>FRACTIONBITS],
- MAX_RESAMPLE_PADDING*sizeof(ALfloat)
- );
-
- /* Now resample, then filter and mix to the appropriate outputs. */
- ResampledData = Resample(&voice->ResampleState,
- &SrcData[MAX_RESAMPLE_PADDING], DataPosFrac, increment,
- Device->TempBuffer[RESAMPLED_BUF], DstBufferSize
- );
- {
- DirectParams *parms = &voice->Direct.Params[chan];
- const ALfloat *samples;
-
- samples = DoFilters(
- &parms->LowPass, &parms->HighPass, Device->TempBuffer[FILTERED_BUF],
- ResampledData, DstBufferSize, voice->Direct.FilterType
- );
- if(!(voice->Flags&VOICE_HAS_HRTF))
- {
- if(!Counter)
- memcpy(parms->Gains.Current, parms->Gains.Target,
- sizeof(parms->Gains.Current));
- if(!(voice->Flags&VOICE_HAS_NFC))
- MixSamples(samples, voice->Direct.Channels, voice->Direct.Buffer,
- parms->Gains.Current, parms->Gains.Target, Counter, OutPos,
- DstBufferSize
- );
- else
- {
- ALfloat *nfcsamples = Device->TempBuffer[NFC_DATA_BUF];
- ALsizei chanoffset = 0;
-
- MixSamples(samples,
- voice->Direct.ChannelsPerOrder[0], voice->Direct.Buffer,
- parms->Gains.Current, parms->Gains.Target, Counter, OutPos,
- DstBufferSize
- );
- chanoffset += voice->Direct.ChannelsPerOrder[0];
-#define APPLY_NFC_MIX(order) \
- if(voice->Direct.ChannelsPerOrder[order] > 0) \
- { \
- NfcFilterProcess##order(&parms->NFCtrlFilter, nfcsamples, samples, \
- DstBufferSize); \
- MixSamples(nfcsamples, voice->Direct.ChannelsPerOrder[order], \
- voice->Direct.Buffer+chanoffset, parms->Gains.Current+chanoffset, \
- parms->Gains.Target+chanoffset, Counter, OutPos, DstBufferSize \
- ); \
- chanoffset += voice->Direct.ChannelsPerOrder[order]; \
- }
- APPLY_NFC_MIX(1)
- APPLY_NFC_MIX(2)
- APPLY_NFC_MIX(3)
-#undef APPLY_NFC_MIX
- }
- }
- else
- {
- MixHrtfParams hrtfparams;
- ALsizei fademix = 0;
- int lidx, ridx;
-
- lidx = GetChannelIdxByName(&Device->RealOut, FrontLeft);
- ridx = GetChannelIdxByName(&Device->RealOut, FrontRight);
- assert(lidx != -1 && ridx != -1);
-
- if(!Counter)
- {
- /* No fading, just overwrite the old HRTF params. */
- parms->Hrtf.Old = parms->Hrtf.Target;
- }
- else if(!(parms->Hrtf.Old.Gain > GAIN_SILENCE_THRESHOLD))
- {
- /* The old HRTF params are silent, so overwrite the old
- * coefficients with the new, and reset the old gain to
- * 0. The future mix will then fade from silence.
- */
- parms->Hrtf.Old = parms->Hrtf.Target;
- parms->Hrtf.Old.Gain = 0.0f;
- }
- else if(firstpass)
- {
- ALfloat gain;
-
- /* Fade between the coefficients over 128 samples. */
- fademix = mini(DstBufferSize, 128);
-
- /* The new coefficients need to fade in completely
- * since they're replacing the old ones. To keep the
- * gain fading consistent, interpolate between the old
- * and new target gains given how much of the fade time
- * this mix handles.
- */
- gain = lerp(parms->Hrtf.Old.Gain, parms->Hrtf.Target.Gain,
- minf(1.0f, (ALfloat)fademix/Counter));
- hrtfparams.Coeffs = parms->Hrtf.Target.Coeffs;
- hrtfparams.Delay[0] = parms->Hrtf.Target.Delay[0];
- hrtfparams.Delay[1] = parms->Hrtf.Target.Delay[1];
- hrtfparams.Gain = 0.0f;
- hrtfparams.GainStep = gain / (ALfloat)fademix;
-
- MixHrtfBlendSamples(
- voice->Direct.Buffer[lidx], voice->Direct.Buffer[ridx],
- samples, voice->Offset, OutPos, IrSize, &parms->Hrtf.Old,
- &hrtfparams, &parms->Hrtf.State, fademix
- );
- /* Update the old parameters with the result. */
- parms->Hrtf.Old = parms->Hrtf.Target;
- if(fademix < Counter)
- parms->Hrtf.Old.Gain = hrtfparams.Gain;
- }
-
- if(fademix < DstBufferSize)
- {
- ALsizei todo = DstBufferSize - fademix;
- ALfloat gain = parms->Hrtf.Target.Gain;
-
- /* Interpolate the target gain if the gain fading lasts
- * longer than this mix.
- */
- if(Counter > DstBufferSize)
- gain = lerp(parms->Hrtf.Old.Gain, gain,
- (ALfloat)todo/(Counter-fademix));
-
- hrtfparams.Coeffs = parms->Hrtf.Target.Coeffs;
- hrtfparams.Delay[0] = parms->Hrtf.Target.Delay[0];
- hrtfparams.Delay[1] = parms->Hrtf.Target.Delay[1];
- hrtfparams.Gain = parms->Hrtf.Old.Gain;
- hrtfparams.GainStep = (gain - parms->Hrtf.Old.Gain) / (ALfloat)todo;
- MixHrtfSamples(
- voice->Direct.Buffer[lidx], voice->Direct.Buffer[ridx],
- samples+fademix, voice->Offset+fademix, OutPos+fademix, IrSize,
- &hrtfparams, &parms->Hrtf.State, todo
- );
- /* Store the interpolated gain or the final target gain
- * depending if the fade is done.
- */
- if(DstBufferSize < Counter)
- parms->Hrtf.Old.Gain = gain;
- else
- parms->Hrtf.Old.Gain = parms->Hrtf.Target.Gain;
- }
- }
- }
-
- for(send = 0;send < Device->NumAuxSends;send++)
- {
- SendParams *parms = &voice->Send[send].Params[chan];
- const ALfloat *samples;
-
- if(!voice->Send[send].Buffer)
- continue;
-
- samples = DoFilters(
- &parms->LowPass, &parms->HighPass, Device->TempBuffer[FILTERED_BUF],
- ResampledData, DstBufferSize, voice->Send[send].FilterType
- );
-
- if(!Counter)
- memcpy(parms->Gains.Current, parms->Gains.Target,
- sizeof(parms->Gains.Current));
- MixSamples(samples, voice->Send[send].Channels, voice->Send[send].Buffer,
- parms->Gains.Current, parms->Gains.Target, Counter, OutPos, DstBufferSize
- );
- }
- }
- /* Update positions */
- DataPosFrac += increment*DstBufferSize;
- DataPosInt += DataPosFrac>>FRACTIONBITS;
- DataPosFrac &= FRACTIONMASK;
-
- OutPos += DstBufferSize;
- voice->Offset += DstBufferSize;
- Counter = maxi(DstBufferSize, Counter) - DstBufferSize;
- firstpass = false;
-
- if(isstatic)
- {
- if(BufferLoopItem)
- {
- /* Handle looping static source */
- const ALbuffer *Buffer = BufferListItem->buffers[0];
- ALsizei LoopStart = Buffer->LoopStart;
- ALsizei LoopEnd = Buffer->LoopEnd;
- if(DataPosInt >= LoopEnd)
- {
- assert(LoopEnd > LoopStart);
- DataPosInt = ((DataPosInt-LoopStart)%(LoopEnd-LoopStart)) + LoopStart;
- }
- }
- else
- {
- /* Handle non-looping static source */
- if(DataPosInt >= BufferListItem->max_samples)
- {
- isplaying = false;
- BufferListItem = NULL;
- DataPosInt = 0;
- DataPosFrac = 0;
- break;
- }
- }
- }
- else while(1)
- {
- /* Handle streaming source */
- if(BufferListItem->max_samples > DataPosInt)
- break;
-
- DataPosInt -= BufferListItem->max_samples;
-
- buffers_done += BufferListItem->num_buffers;
- BufferListItem = ATOMIC_LOAD(&BufferListItem->next, almemory_order_relaxed);
- if(!BufferListItem && !(BufferListItem=BufferLoopItem))
- {
- isplaying = false;
- DataPosInt = 0;
- DataPosFrac = 0;
- break;
- }
- }
- } while(isplaying && OutPos < SamplesToDo);
-
- voice->Flags |= VOICE_IS_FADING;
-
- /* Update source info */
- ATOMIC_STORE(&voice->position, DataPosInt, almemory_order_relaxed);
- ATOMIC_STORE(&voice->position_fraction, DataPosFrac, almemory_order_relaxed);
- ATOMIC_STORE(&voice->current_buffer, BufferListItem, almemory_order_release);
-
- /* Send any events now, after the position/buffer info was updated. */
- enabledevt = ATOMIC_LOAD(&Context->EnabledEvts, almemory_order_acquire);
- if(buffers_done > 0 && (enabledevt&EventType_BufferCompleted))
- SendAsyncEvent(Context, EventType_BufferCompleted,
- AL_EVENT_TYPE_BUFFER_COMPLETED_SOFT, SourceID, buffers_done, "Buffer completed"
- );
-
- return isplaying;
-}
generated by cgit v1.2.3 (git 2.39.1) at 2024-11-28 15:44:32 +0000