Convert mixvoice.c to C++

1 files changed, 761 insertions, 0 deletions

diff --git a/Alc/mixvoice.cpp b/Alc/mixvoice.cpp
new file mode 100644
index 00000000..03d51dd6
--- /dev/null
+++ b/Alc/mixvoice.cpp
@@ -0,0 +1,761 @@
+/**
+ * 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 = static_cast<enum Resampler>(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, static_cast<const ST*>(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 = static_cast<const ALubyte*>(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 = static_cast<const ALubyte*>(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 = static_cast<const ALubyte*>(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 = static_cast<const ALubyte*>(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, static_cast<ALuint>(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;
+}