openal-soft/Alc/mixvoice.cpp

/**
 * 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 <cmath>
#include <cstdlib>
#include <cstring>
#include <cctype>
#include <cassert>

#include <numeric>
#include <algorithm>

#include "AL/al.h"
#include "AL/alc.h"

#include "alMain.h"
#include "alcontext.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!");


Resampler ResamplerDefault = LinearResampler;

MixerFunc MixSamples = Mix_<CTag>;
RowMixerFunc MixRowSamples = MixRow_<CTag>;
static HrtfMixerFunc MixHrtfSamples = MixHrtf_<CTag>;
static HrtfMixerBlendFunc MixHrtfBlendSamples = MixHrtfBlend_<CTag>;

static MixerFunc SelectMixer()
{
#ifdef HAVE_NEON
    if((CPUCapFlags&CPU_CAP_NEON))
        return Mix_<NEONTag>;
#endif
#ifdef HAVE_SSE
    if((CPUCapFlags&CPU_CAP_SSE))
        return Mix_<SSETag>;
#endif
    return Mix_<CTag>;
}

static RowMixerFunc SelectRowMixer()
{
#ifdef HAVE_NEON
    if((CPUCapFlags&CPU_CAP_NEON))
        return MixRow_<NEONTag>;
#endif
#ifdef HAVE_SSE
    if((CPUCapFlags&CPU_CAP_SSE))
        return MixRow_<SSETag>;
#endif
    return MixRow_<CTag>;
}

static inline HrtfMixerFunc SelectHrtfMixer()
{
#ifdef HAVE_NEON
    if((CPUCapFlags&CPU_CAP_NEON))
        return MixHrtf_<NEONTag>;
#endif
#ifdef HAVE_SSE
    if((CPUCapFlags&CPU_CAP_SSE))
        return MixHrtf_<SSETag>;
#endif
    return MixHrtf_<CTag>;
}

static inline HrtfMixerBlendFunc SelectHrtfBlendMixer()
{
#ifdef HAVE_NEON
    if((CPUCapFlags&CPU_CAP_NEON))
        return MixHrtfBlend_<NEONTag>;
#endif
#ifdef HAVE_SSE
    if((CPUCapFlags&CPU_CAP_SSE))
        return MixHrtfBlend_<SSETag>;
#endif
    return MixHrtfBlend_<CTag>;
}

ResamplerFunc SelectResampler(Resampler resampler)
{
    switch(resampler)
    {
        case PointResampler:
            return Resample_<PointTag,CTag>;
        case LinearResampler:
#ifdef HAVE_NEON
            if((CPUCapFlags&CPU_CAP_NEON))
                return Resample_<LerpTag,NEONTag>;
#endif
#ifdef HAVE_SSE4_1
            if((CPUCapFlags&CPU_CAP_SSE4_1))
                return Resample_<LerpTag,SSE4Tag>;
#endif
#ifdef HAVE_SSE2
            if((CPUCapFlags&CPU_CAP_SSE2))
                return Resample_<LerpTag,SSE2Tag>;
#endif
            return Resample_<LerpTag,CTag>;
        case FIR4Resampler:
            return Resample_<CubicTag,CTag>;
        case BSinc12Resampler:
        case BSinc24Resampler:
#ifdef HAVE_NEON
            if((CPUCapFlags&CPU_CAP_NEON))
                return Resample_<BSincTag,NEONTag>;
#endif
#ifdef HAVE_SSE
            if((CPUCapFlags&CPU_CAP_SSE))
                return Resample_<BSincTag,SSETag>;
#endif
            return Resample_<BSincTag,CTag>;
    }

    return Resample_<PointTag,CTag>;
}


void aluInitMixer()
{
    const char *str;

    if(ConfigValueStr(nullptr, nullptr, "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<Resampler>(n);
            else
                WARN("Invalid resampler: %s\n", str);
        }
    }

    MixHrtfBlendSamples = SelectHrtfBlendMixer();
    MixHrtfSamples = SelectHrtfMixer();
    MixSamples = SelectMixer();
    MixRowSamples = SelectRowMixer();
}


namespace {

/* Base template left undefined. Should be marked =delete, but Clang 3.8.1
 * chokes on that given the inline specializations.
 */
template<FmtType T>
inline ALfloat LoadSample(typename FmtTypeTraits<T>::Type val);

template<> inline ALfloat LoadSample<FmtUByte>(FmtTypeTraits<FmtUByte>::Type val)
{ return (val-128) * (1.0f/128.0f); }
template<> inline ALfloat LoadSample<FmtShort>(FmtTypeTraits<FmtShort>::Type val)
{ return val * (1.0f/32768.0f); }
template<> inline ALfloat LoadSample<FmtFloat>(FmtTypeTraits<FmtFloat>::Type val)
{ return val; }
template<> inline ALfloat LoadSample<FmtDouble>(FmtTypeTraits<FmtDouble>::Type val)
{ return static_cast<ALfloat>(val); }
template<> inline ALfloat LoadSample<FmtMulaw>(FmtTypeTraits<FmtMulaw>::Type val)
{ return muLawDecompressionTable[val] * (1.0f/32768.0f); }
template<> inline ALfloat LoadSample<FmtAlaw>(FmtTypeTraits<FmtAlaw>::Type val)
{ return aLawDecompressionTable[val] * (1.0f/32768.0f); }

template<FmtType T>
inline void LoadSampleArray(ALfloat *RESTRICT dst, const void *src, ALint srcstep, ALsizei samples)
{
    using SampleType = typename FmtTypeTraits<T>::Type;

    const SampleType *ssrc = static_cast<const SampleType*>(src);
    for(ALsizei i{0};i < samples;i++)
        dst[i] += LoadSample<T>(ssrc[i*srcstep]);
}

void LoadSamples(ALfloat *RESTRICT dst, const ALvoid *RESTRICT src, ALint srcstep, FmtType srctype,
                 ALsizei samples)
{
#define HANDLE_FMT(T)                                                         \
    case T: LoadSampleArray<T>(dst, src, srcstep, samples); break
    switch(srctype)
    {
        HANDLE_FMT(FmtUByte);
        HANDLE_FMT(FmtShort);
        HANDLE_FMT(FmtFloat);
        HANDLE_FMT(FmtDouble);
        HANDLE_FMT(FmtMulaw);
        HANDLE_FMT(FmtAlaw);
    }
#undef HANDLE_FMT
}


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:
            lpfilter->passthru(numsamples);
            hpfilter->passthru(numsamples);
            break;

        case AF_LowPass:
            lpfilter->process(dst, src, numsamples);
            hpfilter->passthru(numsamples);
            return dst;
        case AF_HighPass:
            lpfilter->passthru(numsamples);
            hpfilter->process(dst, src, numsamples);
            return dst;

        case AF_BandPass:
            for(i = 0;i < numsamples;)
            {
                ALfloat temp[256];
                ALsizei todo = mini(256, numsamples-i);

                lpfilter->process(temp, src+i, todo);
                hpfilter->process(dst+i, temp, todo);
                i += todo;
            }
            return dst;
    }
    return src;
}

} // namespace

/* This function uses these device temp buffers. */
#define RESAMPLED_BUF 0
#define FILTERED_BUF 1
#define NFC_DATA_BUF 2
ALboolean MixSource(ALvoice *voice, const ALuint SourceID, ALCcontext *Context, const ALsizei SamplesToDo)
{
    ASSUME(SamplesToDo > 0);

    /* Get source info */
    bool isplaying{true}; /* Will only be called while playing. */
    bool isstatic{(voice->Flags&VOICE_IS_STATIC) != 0};
    ALsizei DataPosInt{static_cast<ALsizei>(voice->position.load(std::memory_order_acquire))};
    ALsizei DataPosFrac{voice->position_fraction.load(std::memory_order_relaxed)};
    ALbufferlistitem *BufferListItem{voice->current_buffer.load(std::memory_order_relaxed)};
    ALbufferlistitem *BufferLoopItem{voice->loop_buffer.load(std::memory_order_relaxed)};
    ALsizei NumChannels{voice->NumChannels};
    ALsizei SampleSize{voice->SampleSize};
    ALint increment{voice->Step};

    ASSUME(DataPosInt >= 0);
    ASSUME(DataPosFrac >= 0);
    ASSUME(NumChannels > 0);
    ASSUME(SampleSize > 0);
    ASSUME(increment > 0);

    ALCdevice *Device{Context->Device};
    const ALsizei IrSize{Device->mHrtf ? Device->mHrtf->irSize : 0};
    const int OutLIdx{GetChannelIdxByName(Device->RealOut, FrontLeft)};
    const int OutRIdx{GetChannelIdxByName(Device->RealOut, FrontRight)};

    ASSUME(IrSize >= 0);

    ResamplerFunc Resample{(increment == FRACTIONONE && DataPosFrac == 0) ?
                           Resample_<CopyTag,CTag> : voice->Resampler};

    ALsizei Counter{(voice->Flags&VOICE_IS_FADING) ? SamplesToDo : 0};
    if(!Counter)
    {
        /* No fading, just overwrite the old/current params. */
        for(ALsizei chan{0};chan < NumChannels;chan++)
        {
            DirectParams &parms = voice->Direct.Params[chan];
            if(!(voice->Flags&VOICE_HAS_HRTF))
                std::copy(std::begin(parms.Gains.Target), std::end(parms.Gains.Target),
                    std::begin(parms.Gains.Current));
            else
                parms.Hrtf.Old = parms.Hrtf.Target;
            auto set_current = [chan](ALvoice::SendData &send) -> void
            {
                if(!send.Buffer)
                    return;

                SendParams &parms = send.Params[chan];
                std::copy(std::begin(parms.Gains.Target), std::end(parms.Gains.Target),
                    std::begin(parms.Gains.Current));
            };
            std::for_each(voice->Send.begin(), voice->Send.end(), set_current);
        }
    }
    else if((voice->Flags&VOICE_HAS_HRTF))
    {
        for(ALsizei chan{0};chan < NumChannels;chan++)
        {
            DirectParams &parms = voice->Direct.Params[chan];
            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;
            }
        }
    }

    ALsizei buffers_done{0};
    ALsizei OutPos{0};
    do {
        /* Figure out how many buffer samples will be needed */
        ALsizei DstBufferSize{SamplesToDo - OutPos};

        /* Calculate the last written dst sample pos. */
        int64_t DataSize64{DstBufferSize - 1};
        /* Calculate the last read src sample pos. */
        DataSize64 = (DataSize64*increment + DataPosFrac) >> FRACTIONBITS;
        /* +1 to get the src sample count, include padding. */
        DataSize64 += 1 + MAX_RESAMPLE_PADDING*2;

        auto SrcBufferSize = static_cast<ALsizei>(
            mini64(DataSize64, BUFFERSIZE + MAX_RESAMPLE_PADDING*2 + 1));
        if(SrcBufferSize > BUFFERSIZE + MAX_RESAMPLE_PADDING*2)
        {
            SrcBufferSize = BUFFERSIZE + MAX_RESAMPLE_PADDING*2;
            /* If the source buffer got saturated, we can't fill the desired
             * dst size. Figure out how many samples we can actually mix from
             * this.
             */
            DataSize64 = SrcBufferSize - MAX_RESAMPLE_PADDING*2;
            DataSize64 = ((DataSize64<<FRACTIONBITS) - DataPosFrac + increment-1) / increment;
            DstBufferSize = static_cast<ALsizei>(mini64(DataSize64, DstBufferSize));

            /* 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(ALsizei chan{0};chan < NumChannels;chan++)
        {
            auto &SrcData = Device->SourceData;

            /* Load the previous samples into the source data first, and clear the rest. */
            auto srciter = std::copy(std::begin(voice->PrevSamples[chan]),
                std::end(voice->PrevSamples[chan]), std::begin(SrcData));
            std::fill(srciter, std::end(SrcData), 0.0f);

            auto FilledAmt = static_cast<ALsizei>(voice->PrevSamples[chan].size());
            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};
                ASSUME(LoopStart >= 0);
                ASSUME(LoopEnd > LoopStart);

                /* If current pos is beyond the loop range, do not loop */
                if(!BufferLoopItem || DataPosInt >= LoopEnd)
                {
                    const ALsizei SizeToDo{SrcBufferSize - FilledAmt};

                    BufferLoopItem = nullptr;

                    auto load_buffer = [DataPosInt,&SrcData,NumChannels,SampleSize,chan,FilledAmt,SizeToDo](ALsizei CompLen, const ALbuffer *buffer) -> ALsizei
                    {
                        if(DataPosInt >= buffer->SampleLen)
                            return CompLen;

                        /* Load what's left to play from the buffer */
                        const ALsizei DataSize{mini(SizeToDo, buffer->SampleLen - DataPosInt)};
                        CompLen = maxi(CompLen, DataSize);

                        const ALbyte *Data{buffer->mData.data()};
                        LoadSamples(&SrcData[FilledAmt],
                            &Data[(DataPosInt*NumChannels + chan)*SampleSize],
                            NumChannels, buffer->mFmtType, DataSize
                        );
                        return CompLen;
                    };
                    auto buffers_end = BufferListItem->buffers + BufferListItem->num_buffers;
                    FilledAmt += std::accumulate(BufferListItem->buffers, buffers_end, ALsizei{0},
                        load_buffer);
                }
                else
                {
                    const ALsizei SizeToDo{mini(SrcBufferSize - FilledAmt, LoopEnd - DataPosInt)};

                    auto load_buffer = [DataPosInt,&SrcData,NumChannels,SampleSize,chan,FilledAmt,SizeToDo](ALsizei CompLen, const ALbuffer *buffer) -> ALsizei
                    {
                        if(DataPosInt >= buffer->SampleLen)
                            return CompLen;

                        /* Load what's left of this loop iteration */
                        const ALsizei DataSize{mini(SizeToDo, buffer->SampleLen - DataPosInt)};
                        CompLen = maxi(CompLen, DataSize);

                        const ALbyte *Data{buffer->mData.data()};
                        LoadSamples(&SrcData[FilledAmt],
                            &Data[(DataPosInt*NumChannels + chan)*SampleSize],
                            NumChannels, buffer->mFmtType, DataSize
                        );
                        return CompLen;
                    };
                    auto buffers_end = BufferListItem->buffers + BufferListItem->num_buffers;
                    FilledAmt = std::accumulate(BufferListItem->buffers, buffers_end, ALsizei{0}, load_buffer);

                    const ALsizei LoopSize{LoopEnd - LoopStart};
                    while(SrcBufferSize > FilledAmt)
                    {
                        const ALsizei SizeToDo{mini(SrcBufferSize - FilledAmt, LoopSize)};

                        auto load_buffer_loop = [LoopStart,&SrcData,NumChannels,SampleSize,chan,FilledAmt,SizeToDo](ALsizei CompLen, const ALbuffer *buffer) -> ALsizei
                        {
                            if(LoopStart >= buffer->SampleLen)
                                return CompLen;

                            const ALsizei DataSize{mini(SizeToDo, buffer->SampleLen - LoopStart)};
                            CompLen = maxi(CompLen, DataSize);

                            const ALbyte *Data{buffer->mData.data()};
                            LoadSamples(&SrcData[FilledAmt],
                                &Data[(LoopStart*NumChannels + chan)*SampleSize],
                                NumChannels, buffer->mFmtType, DataSize
                            );
                            return CompLen;
                        };
                        FilledAmt += std::accumulate(BufferListItem->buffers, buffers_end,
                            ALsizei{0}, load_buffer_loop);
                    }
                }
            }
            else
            {
                /* Crawl the buffer queue to fill in the temp buffer */
                ALbufferlistitem *tmpiter{BufferListItem};
                ALsizei pos{DataPosInt};

                while(tmpiter && SrcBufferSize > FilledAmt)
                {
                    if(pos >= tmpiter->max_samples)
                    {
                        pos -= tmpiter->max_samples;
                        tmpiter = tmpiter->next.load(std::memory_order_acquire);
                        if(!tmpiter) tmpiter = BufferLoopItem;
                        continue;
                    }

                    const ALsizei SizeToDo{SrcBufferSize - FilledAmt};
                    auto load_buffer = [pos,&SrcData,NumChannels,SampleSize,chan,FilledAmt,SizeToDo](ALsizei CompLen, const ALbuffer *buffer) -> ALsizei
                    {
                        if(!buffer) return CompLen;
                        ALsizei DataSize{buffer->SampleLen};
                        if(pos >= DataSize) return CompLen;

                        DataSize = mini(SizeToDo, DataSize - pos);
                        CompLen = maxi(CompLen, DataSize);

                        const ALbyte *Data{buffer->mData.data()};
                        Data += (pos*NumChannels + chan)*SampleSize;

                        LoadSamples(&SrcData[FilledAmt], Data, NumChannels,
                                    buffer->mFmtType, DataSize);
                        return CompLen;
                    };
                    auto buffers_end = tmpiter->buffers + tmpiter->num_buffers;
                    FilledAmt += std::accumulate(tmpiter->buffers, buffers_end, ALsizei{0},
                        load_buffer);

                    if(SrcBufferSize <= FilledAmt)
                        break;
                    pos = 0;
                    tmpiter = tmpiter->next.load(std::memory_order_acquire);
                    if(!tmpiter) tmpiter = BufferLoopItem;
                }
            }

            /* Store the last source samples used for next time. */
            std::copy_n(&SrcData[(increment*DstBufferSize + DataPosFrac)>>FRACTIONBITS],
                voice->PrevSamples[chan].size(), std::begin(voice->PrevSamples[chan]));

            /* Resample, then apply ambisonic upsampling as needed. */
            const ALfloat *ResampledData{Resample(&voice->ResampleState,
                &SrcData[MAX_RESAMPLE_PADDING], DataPosFrac, increment,
                Device->TempBuffer[RESAMPLED_BUF], DstBufferSize
            )};
            if((voice->Flags&VOICE_IS_AMBISONIC))
            {
                /* TODO: Does not properly handle HOA sources. Currently only
                 * first-order sources are possible, but in the future it would
                 * be desirable.
                 */
                const ALfloat hfscale{(chan==0) ? voice->AmbiScales[0] : voice->AmbiScales[1]};
                ALfloat (&hfbuf)[BUFFERSIZE] = Device->TempBuffer[FILTERED_BUF];
                ALfloat (&lfbuf)[BUFFERSIZE] = Device->TempBuffer[RESAMPLED_BUF];

                voice->AmbiSplitter[chan].process(hfbuf, lfbuf, ResampledData, DstBufferSize);
                MixRowSamples(lfbuf, &hfscale, &hfbuf, 1, 0, DstBufferSize);

                ResampledData = lfbuf;
            }

            /* Now filter and mix to the appropriate outputs. */
            {
                DirectParams &parms = voice->Direct.Params[chan];
                const ALfloat *samples{DoFilters(&parms.LowPass, &parms.HighPass,
                    Device->TempBuffer[FILTERED_BUF], ResampledData, DstBufferSize,
                    voice->Direct.FilterType
                )};

                if(!(voice->Flags&VOICE_HAS_HRTF))
                {
                    if(!(voice->Flags&VOICE_HAS_NFC))
                        MixSamples(samples, voice->Direct.Channels, voice->Direct.Buffer,
                            parms.Gains.Current, parms.Gains.Target, Counter, OutPos,
                            DstBufferSize);
                    else
                    {
                        MixSamples(samples,
                            voice->Direct.ChannelsPerOrder[0], voice->Direct.Buffer,
                            parms.Gains.Current, parms.Gains.Target, Counter, OutPos,
                            DstBufferSize);

                        ALfloat *nfcsamples{Device->TempBuffer[NFC_DATA_BUF]};
                        ALsizei chanoffset{voice->Direct.ChannelsPerOrder[0]};
                        using FilterProc = void (NfcFilter::*)(float*,const float*,int);
                        auto apply_nfc = [voice,&parms,samples,DstBufferSize,Counter,OutPos,&chanoffset,nfcsamples](FilterProc process, ALsizei order) -> void
                        {
                            if(voice->Direct.ChannelsPerOrder[order] < 1)
                                return;
                            (parms.NFCtrlFilter.*process)(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(&NfcFilter::process1, 1);
                        apply_nfc(&NfcFilter::process2, 2);
                        apply_nfc(&NfcFilter::process3, 3);
                    }
                }
                else
                {
                    ALsizei fademix{0};
                    /* If fading, the old gain is not silence, and this is the
                     * first mixing pass, fade between the IRs.
                     */
                    if(Counter && (parms.Hrtf.Old.Gain > GAIN_SILENCE_THRESHOLD) && OutPos == 0)
                    {
                        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.
                         */
                        ALfloat gain{lerp(parms.Hrtf.Old.Gain, parms.Hrtf.Target.Gain,
                                          minf(1.0f, static_cast<ALfloat>(fademix))/Counter)};
                        MixHrtfParams hrtfparams;
                        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 / static_cast<ALfloat>(fademix);

                        MixHrtfBlendSamples(
                            voice->Direct.Buffer[OutLIdx], voice->Direct.Buffer[OutRIdx],
                            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)
                    {
                        const 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,
                                        static_cast<ALfloat>(todo)/(Counter-fademix));

                        MixHrtfParams hrtfparams;
                        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) / static_cast<ALfloat>(todo);
                        MixHrtfSamples(
                            voice->Direct.Buffer[OutLIdx], voice->Direct.Buffer[OutRIdx],
                            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;
                    }
                }
            }

            ALfloat (&FilterBuf)[BUFFERSIZE] = Device->TempBuffer[FILTERED_BUF];
            auto mix_send = [Counter,OutPos,DstBufferSize,chan,ResampledData,&FilterBuf](ALvoice::SendData &send) -> void
            {
                if(!send.Buffer)
                    return;

                SendParams &parms = send.Params[chan];
                const ALfloat *samples{DoFilters(&parms.LowPass, &parms.HighPass,
                    FilterBuf, ResampledData, DstBufferSize, send.FilterType)};

                MixSamples(samples, send.Channels, send.Buffer, parms.Gains.Current,
                    parms.Gains.Target, Counter, OutPos, DstBufferSize);
            };
            std::for_each(voice->Send.begin(), voice->Send.end(), mix_send);
        }
        /* Update positions */
        DataPosFrac += increment*DstBufferSize;
        DataPosInt  += DataPosFrac>>FRACTIONBITS;
        DataPosFrac &= FRACTIONMASK;

        OutPos += DstBufferSize;
        voice->Offset += DstBufferSize;
        Counter = maxi(DstBufferSize, Counter) - DstBufferSize;

        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 = nullptr;
                    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 = BufferListItem->next.load(std::memory_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 */
    voice->position.store(DataPosInt, std::memory_order_relaxed);
    voice->position_fraction.store(DataPosFrac, std::memory_order_relaxed);
    voice->current_buffer.store(BufferListItem, std::memory_order_release);

    /* Send any events now, after the position/buffer info was updated. */
    ALbitfieldSOFT enabledevt{Context->EnabledEvts.load(std::memory_order_acquire)};
    if(buffers_done > 0 && (enabledevt&EventType_BufferCompleted))
    {
        RingBuffer *ring{Context->AsyncEvents.get()};
        auto evt_vec = ring->getWriteVector();
        if(evt_vec.first.len > 0)
        {
            AsyncEvent *evt{new (evt_vec.first.buf) AsyncEvent{EventType_BufferCompleted}};
            evt->u.bufcomp.id = SourceID;
            evt->u.bufcomp.count = buffers_done;
            ring->writeAdvance(1);
            Context->EventSem.post();
        }
    }

    return isplaying;
}