From a68d0b68d74a9f3fa65096fdfddc5a04fa118dfa Mon Sep 17 00:00:00 2001 From: Chris Robinson Date: Fri, 16 Nov 2018 20:46:50 -0800 Subject: Convert mixvoice.c to C++ --- Alc/mixvoice.c | 759 --------------------------------------------------------- 1 file changed, 759 deletions(-) delete mode 100644 Alc/mixvoice.c (limited to 'Alc/mixvoice.c') diff --git a/Alc/mixvoice.c b/Alc/mixvoice.c deleted file mode 100644 index 9a774786..00000000 --- a/Alc/mixvoice.c +++ /dev/null @@ -1,759 +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 -#include -#include -#include -#include - -#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!"); - -/* 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, int 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; -} -- cgit v1.2.3