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axmol/external/openal/alc/voice.cpp

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/**
* 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 "voice.h"
#include <algorithm>
#include <array>
#include <atomic>
#include <cassert>
#include <climits>
#include <cstddef>
#include <cstdint>
#include <iterator>
#include <memory>
#include <new>
#include <utility>
#include "alcmain.h"
#include "albyte.h"
#include "alconfig.h"
#include "alcontext.h"
#include "alnumeric.h"
#include "aloptional.h"
#include "alspan.h"
#include "alstring.h"
#include "alu.h"
#include "async_event.h"
#include "buffer_storage.h"
#include "core/cpu_caps.h"
#include "core/devformat.h"
#include "core/filters/biquad.h"
#include "core/filters/nfc.h"
#include "core/filters/splitter.h"
#include "core/fmt_traits.h"
#include "core/logging.h"
#include "core/mixer/defs.h"
#include "core/mixer/hrtfdefs.h"
#include "hrtf.h"
#include "inprogext.h"
#include "opthelpers.h"
#include "ringbuffer.h"
#include "threads.h"
#include "vector.h"
#include "voice_change.h"
struct CTag;
#ifdef HAVE_SSE
struct SSETag;
#endif
#ifdef HAVE_NEON
struct NEONTag;
#endif
struct CopyTag;
Resampler ResamplerDefault{Resampler::Linear};
MixerFunc MixSamples{Mix_<CTag>};
namespace {
using HrtfMixerFunc = void(*)(const float *InSamples, float2 *AccumSamples, const uint IrSize,
const MixHrtfFilter *hrtfparams, const size_t BufferSize);
using HrtfMixerBlendFunc = void(*)(const float *InSamples, float2 *AccumSamples,
const uint IrSize, const HrtfFilter *oldparams, const MixHrtfFilter *newparams,
const size_t BufferSize);
HrtfMixerFunc MixHrtfSamples{MixHrtf_<CTag>};
HrtfMixerBlendFunc MixHrtfBlendSamples{MixHrtfBlend_<CTag>};
inline 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>;
}
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>;
}
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>;
}
} // namespace
void aluInitMixer()
{
if(auto resopt = ConfigValueStr(nullptr, nullptr, "resampler"))
{
struct ResamplerEntry {
const char name[16];
const Resampler resampler;
};
constexpr ResamplerEntry ResamplerList[]{
{ "none", Resampler::Point },
{ "point", Resampler::Point },
{ "linear", Resampler::Linear },
{ "cubic", Resampler::Cubic },
{ "bsinc12", Resampler::BSinc12 },
{ "fast_bsinc12", Resampler::FastBSinc12 },
{ "bsinc24", Resampler::BSinc24 },
{ "fast_bsinc24", Resampler::FastBSinc24 },
};
const char *str{resopt->c_str()};
if(al::strcasecmp(str, "bsinc") == 0)
{
WARN("Resampler option \"%s\" is deprecated, using bsinc12\n", str);
str = "bsinc12";
}
else if(al::strcasecmp(str, "sinc4") == 0 || al::strcasecmp(str, "sinc8") == 0)
{
WARN("Resampler option \"%s\" is deprecated, using cubic\n", str);
str = "cubic";
}
auto iter = std::find_if(std::begin(ResamplerList), std::end(ResamplerList),
[str](const ResamplerEntry &entry) -> bool
{ return al::strcasecmp(str, entry.name) == 0; });
if(iter == std::end(ResamplerList))
ERR("Invalid resampler: %s\n", str);
else
ResamplerDefault = iter->resampler;
}
MixSamples = SelectMixer();
MixHrtfBlendSamples = SelectHrtfBlendMixer();
MixHrtfSamples = SelectHrtfMixer();
}
namespace {
void SendSourceStoppedEvent(ALCcontext *context, uint id)
{
RingBuffer *ring{context->mAsyncEvents.get()};
auto evt_vec = ring->getWriteVector();
if(evt_vec.first.len < 1) return;
AsyncEvent *evt{::new(evt_vec.first.buf) AsyncEvent{EventType_SourceStateChange}};
evt->u.srcstate.id = id;
evt->u.srcstate.state = VChangeState::Stop;
ring->writeAdvance(1);
}
const float *DoFilters(BiquadFilter &lpfilter, BiquadFilter &hpfilter, float *dst,
const al::span<const float> src, int type)
{
switch(type)
{
case AF_None:
lpfilter.clear();
hpfilter.clear();
break;
case AF_LowPass:
lpfilter.process(src, dst);
hpfilter.clear();
return dst;
case AF_HighPass:
lpfilter.clear();
hpfilter.process(src, dst);
return dst;
case AF_BandPass:
DualBiquad{lpfilter, hpfilter}.process(src, dst);
return dst;
}
return src.data();
}
void LoadSamples(float *RESTRICT dst, const al::byte *src, const size_t srcstep, FmtType srctype,
const size_t samples) noexcept
{
#define HANDLE_FMT(T) case T: al::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
}
float *LoadBufferStatic(VoiceBufferItem *buffer, VoiceBufferItem *&bufferLoopItem,
const size_t numChannels, const FmtType sampleType, const size_t sampleSize, const size_t chan,
size_t dataPosInt, al::span<float> srcBuffer)
{
const uint LoopStart{buffer->mLoopStart};
const uint LoopEnd{buffer->mLoopEnd};
ASSUME(LoopEnd > LoopStart);
/* If current pos is beyond the loop range, do not loop */
if(!bufferLoopItem || dataPosInt >= LoopEnd)
{
bufferLoopItem = nullptr;
/* Load what's left to play from the buffer */
const size_t DataRem{minz(srcBuffer.size(), buffer->mSampleLen-dataPosInt)};
const al::byte *Data{buffer->mSamples + (dataPosInt*numChannels + chan)*sampleSize};
LoadSamples(srcBuffer.data(), Data, numChannels, sampleType, DataRem);
srcBuffer = srcBuffer.subspan(DataRem);
}
else
{
/* Load what's left of this loop iteration */
const size_t DataRem{minz(srcBuffer.size(), LoopEnd-dataPosInt)};
const al::byte *Data{buffer->mSamples + (dataPosInt*numChannels + chan)*sampleSize};
LoadSamples(srcBuffer.data(), Data, numChannels, sampleType, DataRem);
srcBuffer = srcBuffer.subspan(DataRem);
/* Load any repeats of the loop we can to fill the buffer. */
const auto LoopSize = static_cast<size_t>(LoopEnd - LoopStart);
while(!srcBuffer.empty())
{
const size_t DataSize{minz(srcBuffer.size(), LoopSize)};
Data = buffer->mSamples + (LoopStart*numChannels + chan)*sampleSize;
LoadSamples(srcBuffer.data(), Data, numChannels, sampleType, DataSize);
srcBuffer = srcBuffer.subspan(DataSize);
}
}
return srcBuffer.begin();
}
float *LoadBufferCallback(VoiceBufferItem *buffer, const size_t numChannels,
const FmtType sampleType, const size_t sampleSize, const size_t chan,
size_t numCallbackSamples, al::span<float> srcBuffer)
{
/* Load what's left to play from the buffer */
const size_t DataRem{minz(srcBuffer.size(), numCallbackSamples)};
const al::byte *Data{buffer->mSamples + chan*sampleSize};
LoadSamples(srcBuffer.data(), Data, numChannels, sampleType, DataRem);
srcBuffer = srcBuffer.subspan(DataRem);
return srcBuffer.begin();
}
float *LoadBufferQueue(VoiceBufferItem *buffer, VoiceBufferItem *bufferLoopItem,
const size_t numChannels, const FmtType sampleType, const size_t sampleSize, const size_t chan,
size_t dataPosInt, al::span<float> srcBuffer)
{
/* Crawl the buffer queue to fill in the temp buffer */
while(buffer && !srcBuffer.empty())
{
if(dataPosInt >= buffer->mSampleLen)
{
dataPosInt -= buffer->mSampleLen;
buffer = buffer->mNext.load(std::memory_order_acquire);
if(!buffer) buffer = bufferLoopItem;
continue;
}
const size_t DataSize{minz(srcBuffer.size(), buffer->mSampleLen-dataPosInt)};
const al::byte *Data{buffer->mSamples + (dataPosInt*numChannels + chan)*sampleSize};
LoadSamples(srcBuffer.data(), Data, numChannels, sampleType, DataSize);
srcBuffer = srcBuffer.subspan(DataSize);
if(srcBuffer.empty()) break;
dataPosInt = 0;
buffer = buffer->mNext.load(std::memory_order_acquire);
if(!buffer) buffer = bufferLoopItem;
}
return srcBuffer.begin();
}
void DoHrtfMix(const float *samples, const uint DstBufferSize, DirectParams &parms,
const float TargetGain, const uint Counter, uint OutPos, const uint IrSize,
ALCdevice *Device)
{
auto &HrtfSamples = Device->HrtfSourceData;
/* Source HRTF mixing needs to include the direct delay so it remains
* aligned with the direct mix's HRTF filtering.
*/
float2 *AccumSamples{Device->HrtfAccumData + HrtfDirectDelay};
/* Copy the HRTF history and new input samples into a temp buffer. */
auto src_iter = std::copy(parms.Hrtf.History.begin(), parms.Hrtf.History.end(),
std::begin(HrtfSamples));
std::copy_n(samples, DstBufferSize, src_iter);
/* Copy the last used samples back into the history buffer for later. */
std::copy_n(std::begin(HrtfSamples) + DstBufferSize, parms.Hrtf.History.size(),
parms.Hrtf.History.begin());
/* If fading and this is the first mixing pass, fade between the IRs. */
uint fademix{0u};
if(Counter && OutPos == 0)
{
fademix = minu(DstBufferSize, Counter);
float gain{TargetGain};
/* 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.
*/
if(Counter > fademix)
{
const float a{static_cast<float>(fademix) / static_cast<float>(Counter)};
gain = lerp(parms.Hrtf.Old.Gain, TargetGain, a);
}
MixHrtfFilter hrtfparams;
hrtfparams.Coeffs = &parms.Hrtf.Target.Coeffs;
hrtfparams.Delay = parms.Hrtf.Target.Delay;
hrtfparams.Gain = 0.0f;
hrtfparams.GainStep = gain / static_cast<float>(fademix);
MixHrtfBlendSamples(HrtfSamples, AccumSamples+OutPos, IrSize, &parms.Hrtf.Old, &hrtfparams,
fademix);
/* Update the old parameters with the result. */
parms.Hrtf.Old = parms.Hrtf.Target;
parms.Hrtf.Old.Gain = gain;
OutPos += fademix;
}
if(fademix < DstBufferSize)
{
const uint todo{DstBufferSize - fademix};
float gain{TargetGain};
/* Interpolate the target gain if the gain fading lasts longer than
* this mix.
*/
if(Counter > DstBufferSize)
{
const float a{static_cast<float>(todo) / static_cast<float>(Counter-fademix)};
gain = lerp(parms.Hrtf.Old.Gain, TargetGain, a);
}
MixHrtfFilter hrtfparams;
hrtfparams.Coeffs = &parms.Hrtf.Target.Coeffs;
hrtfparams.Delay = parms.Hrtf.Target.Delay;
hrtfparams.Gain = parms.Hrtf.Old.Gain;
hrtfparams.GainStep = (gain - parms.Hrtf.Old.Gain) / static_cast<float>(todo);
MixHrtfSamples(HrtfSamples+fademix, AccumSamples+OutPos, IrSize, &hrtfparams, todo);
/* Store the now-current gain for next time. */
parms.Hrtf.Old.Gain = gain;
}
}
void DoNfcMix(const al::span<const float> samples, FloatBufferLine *OutBuffer, DirectParams &parms,
const float *TargetGains, const uint Counter, const uint OutPos, ALCdevice *Device)
{
using FilterProc = void (NfcFilter::*)(const al::span<const float>, float*);
static constexpr FilterProc NfcProcess[MaxAmbiOrder+1]{
nullptr, &NfcFilter::process1, &NfcFilter::process2, &NfcFilter::process3};
float *CurrentGains{parms.Gains.Current.data()};
MixSamples(samples, {OutBuffer, 1u}, CurrentGains, TargetGains, Counter, OutPos);
++OutBuffer;
++CurrentGains;
++TargetGains;
const al::span<float> nfcsamples{Device->NfcSampleData, samples.size()};
size_t order{1};
while(const size_t chancount{Device->NumChannelsPerOrder[order]})
{
(parms.NFCtrlFilter.*NfcProcess[order])(samples, nfcsamples.data());
MixSamples(nfcsamples, {OutBuffer, chancount}, CurrentGains, TargetGains, Counter, OutPos);
OutBuffer += chancount;
CurrentGains += chancount;
TargetGains += chancount;
if(++order == MaxAmbiOrder+1)
break;
}
}
} // namespace
void Voice::mix(const State vstate, ALCcontext *Context, const uint SamplesToDo)
{
static constexpr std::array<float,MAX_OUTPUT_CHANNELS> SilentTarget{};
ASSUME(SamplesToDo > 0);
/* Get voice info */
uint DataPosInt{mPosition.load(std::memory_order_relaxed)};
uint DataPosFrac{mPositionFrac.load(std::memory_order_relaxed)};
VoiceBufferItem *BufferListItem{mCurrentBuffer.load(std::memory_order_relaxed)};
VoiceBufferItem *BufferLoopItem{mLoopBuffer.load(std::memory_order_relaxed)};
const FmtType SampleType{mFmtType};
const uint SampleSize{mSampleSize};
const uint increment{mStep};
if UNLIKELY(increment < 1)
{
/* If the voice is supposed to be stopping but can't be mixed, just
* stop it before bailing.
*/
if(vstate == Stopping)
mPlayState.store(Stopped, std::memory_order_release);
return;
}
ASSUME(SampleSize > 0);
const size_t FrameSize{mChans.size() * SampleSize};
ASSUME(FrameSize > 0);
ALCdevice *Device{Context->mDevice.get()};
const uint NumSends{Device->NumAuxSends};
const uint IrSize{Device->mIrSize};
ResamplerFunc Resample{(increment == MixerFracOne && DataPosFrac == 0) ?
Resample_<CopyTag,CTag> : mResampler};
uint Counter{(mFlags&VoiceIsFading) ? SamplesToDo : 0};
if(!Counter)
{
/* No fading, just overwrite the old/current params. */
for(auto &chandata : mChans)
{
{
DirectParams &parms = chandata.mDryParams;
if(!(mFlags&VoiceHasHrtf))
parms.Gains.Current = parms.Gains.Target;
else
parms.Hrtf.Old = parms.Hrtf.Target;
}
for(uint send{0};send < NumSends;++send)
{
if(mSend[send].Buffer.empty())
continue;
SendParams &parms = chandata.mWetParams[send];
parms.Gains.Current = parms.Gains.Target;
}
}
}
else if UNLIKELY(!BufferListItem)
Counter = std::min(Counter, 64u);
uint buffers_done{0u};
uint OutPos{0u};
do {
/* Figure out how many buffer samples will be needed */
uint DstBufferSize{SamplesToDo - OutPos};
uint SrcBufferSize;
if(increment <= MixerFracOne)
{
/* Calculate the last written dst sample pos. */
uint64_t DataSize64{DstBufferSize - 1};
/* Calculate the last read src sample pos. */
DataSize64 = (DataSize64*increment + DataPosFrac) >> MixerFracBits;
/* +1 to get the src sample count, include padding. */
DataSize64 += 1 + MaxResamplerPadding;
/* Result is guaranteed to be <= BufferLineSize+MaxResamplerPadding
* since we won't use more src samples than dst samples+padding.
*/
SrcBufferSize = static_cast<uint>(DataSize64);
}
else
{
uint64_t DataSize64{DstBufferSize};
/* Calculate the end src sample pos, include padding. */
DataSize64 = (DataSize64*increment + DataPosFrac) >> MixerFracBits;
DataSize64 += MaxResamplerPadding;
if(DataSize64 <= BufferLineSize + MaxResamplerPadding)
SrcBufferSize = static_cast<uint>(DataSize64);
else
{
/* If the source size got saturated, we can't fill the desired
* dst size. Figure out how many samples we can actually mix.
*/
SrcBufferSize = BufferLineSize + MaxResamplerPadding;
DataSize64 = SrcBufferSize - MaxResamplerPadding;
DataSize64 = ((DataSize64<<MixerFracBits) - DataPosFrac) / increment;
if(DataSize64 < DstBufferSize)
{
/* Some mixers require being 16-byte aligned, so also limit
* to a multiple of 4 samples to maintain alignment.
*/
DstBufferSize = static_cast<uint>(DataSize64) & ~3u;
}
}
}
if((mFlags&(VoiceIsCallback|VoiceCallbackStopped)) == VoiceIsCallback && BufferListItem)
{
/* Exclude resampler pre-padding from the needed size. */
const uint toLoad{SrcBufferSize - (MaxResamplerPadding>>1)};
if(toLoad > mNumCallbackSamples)
{
const size_t byteOffset{mNumCallbackSamples*FrameSize};
const size_t needBytes{toLoad*FrameSize - byteOffset};
const int gotBytes{BufferListItem->mCallback(BufferListItem->mUserData,
&BufferListItem->mSamples[byteOffset], static_cast<int>(needBytes))};
if(gotBytes < 1)
mFlags |= VoiceCallbackStopped;
else if(static_cast<uint>(gotBytes) < needBytes)
{
mFlags |= VoiceCallbackStopped;
mNumCallbackSamples += static_cast<uint>(static_cast<uint>(gotBytes) /
FrameSize);
}
else
mNumCallbackSamples = toLoad;
}
}
const size_t num_chans{mChans.size()};
size_t chan_idx{0};
ASSUME(DstBufferSize > 0);
for(auto &chandata : mChans)
{
const al::span<float> SrcData{Device->SourceData, SrcBufferSize};
/* Load the previous samples into the source data first, then load
* what we can from the buffer queue.
*/
auto srciter = std::copy_n(chandata.mPrevSamples.begin(), MaxResamplerPadding>>1,
SrcData.begin());
if UNLIKELY(!BufferListItem)
{
/* When loading from a voice that ended prematurely, only take
* the samples that get closest to 0 amplitude. This helps
* certain sounds fade out better.
*/
auto abs_lt = [](const float lhs, const float rhs) noexcept -> bool
{ return std::abs(lhs) < std::abs(rhs); };
auto input = chandata.mPrevSamples.begin() + (MaxResamplerPadding>>1);
auto in_end = std::min_element(input, chandata.mPrevSamples.end(), abs_lt);
srciter = std::copy(input, in_end, srciter);
}
else if((mFlags&VoiceIsStatic))
srciter = LoadBufferStatic(BufferListItem, BufferLoopItem, num_chans, SampleType,
SampleSize, chan_idx, DataPosInt, {srciter, SrcData.end()});
else if((mFlags&VoiceIsCallback))
srciter = LoadBufferCallback(BufferListItem, num_chans, SampleType, SampleSize,
chan_idx, mNumCallbackSamples, {srciter, SrcData.end()});
else
srciter = LoadBufferQueue(BufferListItem, BufferLoopItem, num_chans, SampleType,
SampleSize, chan_idx, DataPosInt, {srciter, SrcData.end()});
if UNLIKELY(srciter != SrcData.end())
{
/* If the source buffer wasn't filled, copy the last sample for
* the remaining buffer. Ideally it should have ended with
* silence, but if not the gain fading should help avoid clicks
* from sudden amplitude changes.
*/
const float sample{*(srciter-1)};
std::fill(srciter, SrcData.end(), sample);
}
/* Store the last source samples used for next time. */
std::copy_n(&SrcData[(increment*DstBufferSize + DataPosFrac)>>MixerFracBits],
chandata.mPrevSamples.size(), chandata.mPrevSamples.begin());
/* Resample, then apply ambisonic upsampling as needed. */
float *ResampledData{Resample(&mResampleState, &SrcData[MaxResamplerPadding>>1],
DataPosFrac, increment, {Device->ResampledData, DstBufferSize})};
if((mFlags&VoiceIsAmbisonic))
chandata.mAmbiSplitter.processHfScale({ResampledData, DstBufferSize},
chandata.mAmbiScale);
/* Now filter and mix to the appropriate outputs. */
float (&FilterBuf)[BufferLineSize] = Device->FilteredData;
{
DirectParams &parms = chandata.mDryParams;
const float *samples{DoFilters(parms.LowPass, parms.HighPass, FilterBuf,
{ResampledData, DstBufferSize}, mDirect.FilterType)};
if((mFlags&VoiceHasHrtf))
{
const float TargetGain{UNLIKELY(vstate == Stopping) ? 0.0f :
parms.Hrtf.Target.Gain};
DoHrtfMix(samples, DstBufferSize, parms, TargetGain, Counter, OutPos, IrSize,
Device);
}
else if((mFlags&VoiceHasNfc))
{
const float *TargetGains{UNLIKELY(vstate == Stopping) ? SilentTarget.data()
: parms.Gains.Target.data()};
DoNfcMix({samples, DstBufferSize}, mDirect.Buffer.data(), parms, TargetGains,
Counter, OutPos, Device);
}
else
{
const float *TargetGains{UNLIKELY(vstate == Stopping) ? SilentTarget.data()
: parms.Gains.Target.data()};
MixSamples({samples, DstBufferSize}, mDirect.Buffer,
parms.Gains.Current.data(), TargetGains, Counter, OutPos);
}
}
for(uint send{0};send < NumSends;++send)
{
if(mSend[send].Buffer.empty())
continue;
SendParams &parms = chandata.mWetParams[send];
const float *samples{DoFilters(parms.LowPass, parms.HighPass, FilterBuf,
{ResampledData, DstBufferSize}, mSend[send].FilterType)};
const float *TargetGains{UNLIKELY(vstate == Stopping) ? SilentTarget.data()
: parms.Gains.Target.data()};
MixSamples({samples, DstBufferSize}, mSend[send].Buffer,
parms.Gains.Current.data(), TargetGains, Counter, OutPos);
}
++chan_idx;
}
/* Update positions */
DataPosFrac += increment*DstBufferSize;
const uint SrcSamplesDone{DataPosFrac>>MixerFracBits};
DataPosInt += SrcSamplesDone;
DataPosFrac &= MixerFracMask;
OutPos += DstBufferSize;
Counter = maxu(DstBufferSize, Counter) - DstBufferSize;
if UNLIKELY(!BufferListItem)
{
/* Do nothing extra when there's no buffers. */
}
else if((mFlags&VoiceIsStatic))
{
if(BufferLoopItem)
{
/* Handle looping static source */
const uint LoopStart{BufferListItem->mLoopStart};
const uint LoopEnd{BufferListItem->mLoopEnd};
if(DataPosInt >= LoopEnd)
{
assert(LoopEnd > LoopStart);
DataPosInt = ((DataPosInt-LoopStart)%(LoopEnd-LoopStart)) + LoopStart;
}
}
else
{
/* Handle non-looping static source */
if(DataPosInt >= BufferListItem->mSampleLen)
{
BufferListItem = nullptr;
break;
}
}
}
else if((mFlags&VoiceIsCallback))
{
if(SrcSamplesDone < mNumCallbackSamples)
{
const size_t byteOffset{SrcSamplesDone*FrameSize};
const size_t byteEnd{mNumCallbackSamples*FrameSize};
al::byte *data{BufferListItem->mSamples};
std::copy(data+byteOffset, data+byteEnd, data);
mNumCallbackSamples -= SrcSamplesDone;
}
else
{
BufferListItem = nullptr;
mNumCallbackSamples = 0;
}
}
else
{
/* Handle streaming source */
do {
if(BufferListItem->mSampleLen > DataPosInt)
break;
DataPosInt -= BufferListItem->mSampleLen;
++buffers_done;
BufferListItem = BufferListItem->mNext.load(std::memory_order_relaxed);
if(!BufferListItem) BufferListItem = BufferLoopItem;
} while(BufferListItem);
}
} while(OutPos < SamplesToDo);
mFlags |= VoiceIsFading;
/* Don't update positions and buffers if we were stopping. */
if UNLIKELY(vstate == Stopping)
{
mPlayState.store(Stopped, std::memory_order_release);
return;
}
/* Capture the source ID in case it's reset for stopping. */
const uint SourceID{mSourceID.load(std::memory_order_relaxed)};
/* Update voice info */
mPosition.store(DataPosInt, std::memory_order_relaxed);
mPositionFrac.store(DataPosFrac, std::memory_order_relaxed);
mCurrentBuffer.store(BufferListItem, std::memory_order_relaxed);
if(!BufferListItem)
{
mLoopBuffer.store(nullptr, std::memory_order_relaxed);
mSourceID.store(0u, std::memory_order_relaxed);
}
std::atomic_thread_fence(std::memory_order_release);
/* Send any events now, after the position/buffer info was updated. */
const uint enabledevt{Context->mEnabledEvts.load(std::memory_order_acquire)};
if(buffers_done > 0 && (enabledevt&EventType_BufferCompleted))
{
RingBuffer *ring{Context->mAsyncEvents.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);
}
}
if(!BufferListItem)
{
/* If the voice just ended, set it to Stopping so the next render
* ensures any residual noise fades to 0 amplitude.
*/
mPlayState.store(Stopping, std::memory_order_release);
if((enabledevt&EventType_SourceStateChange))
SendSourceStoppedEvent(Context, SourceID);
}
}
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