Convert mixvoice.c to C++

1 files changed, 0 insertions, 759 deletions

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 <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!");
-
-/* 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;
-}