Lothario
/
axmol
mirror of https://github.com/axmolengine/axmol.git
1
0
Fork 0
Code Issues Packages Projects Releases Wiki Activity
axmol/thirdparty/openal/core/voice.cpp

946 lines
34 KiB
C++
Raw Blame History

#include "config.h"
#include "voice.h"
#include <algorithm>
#include <array>
#include <atomic>
#include <cassert>
#include <cstdint>
#include <iterator>
#include <memory>
#include <new>
#include <stdlib.h>
#include <utility>
#include <vector>
#include "albyte.h"
#include "alnumeric.h"
#include "aloptional.h"
#include "alspan.h"
#include "alstring.h"
#include "ambidefs.h"
#include "async_event.h"
#include "buffer_storage.h"
#include "context.h"
#include "cpu_caps.h"
#include "devformat.h"
#include "device.h"
#include "filters/biquad.h"
#include "filters/nfc.h"
#include "filters/splitter.h"
#include "fmt_traits.h"
#include "logging.h"
#include "mixer.h"
#include "mixer/defs.h"
#include "mixer/hrtfdefs.h"
#include "opthelpers.h"
#include "resampler_limits.h"
#include "ringbuffer.h"
#include "vector.h"
#include "voice_change.h"
struct CTag;
#ifdef HAVE_SSE
struct SSETag;
#endif
#ifdef HAVE_NEON
struct NEONTag;
#endif
struct CopyTag;
static_assert(!(sizeof(DeviceBase::MixerBufferLine)&15),
"DeviceBase::MixerBufferLine must be a multiple of 16 bytes");
static_assert(!(MaxResamplerEdge&3), "MaxResamplerEdge is not a multiple of 4");
Resampler ResamplerDefault{Resampler::Linear};
namespace {
using uint = unsigned int;
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 Voice::InitMixer(al::optional<std::string> resampler)
{
if(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{resampler->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(ContextBase *context, uint id)
{
RingBuffer *ring{context->mAsyncEvents.get()};
auto evt_vec = ring->getWriteVector();
if(evt_vec.first.len < 1) return;
AsyncEvent *evt{al::construct_at(reinterpret_cast<AsyncEvent*>(evt_vec.first.buf),
AsyncEvent::SourceStateChange)};
evt->u.srcstate.id = id;
evt->u.srcstate.state = AsyncEvent::SrcState::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();
}
template<FmtType Type>
inline void LoadSamples(const al::span<float*> dstSamples, const size_t dstOffset,
const al::byte *src, const size_t srcOffset, const FmtChannels srcChans, const size_t srcStep,
const size_t samples) noexcept
{
constexpr size_t sampleSize{sizeof(typename al::FmtTypeTraits<Type>::Type)};
auto s = src + srcOffset*srcStep*sampleSize;
if(srcChans == FmtUHJ2 || srcChans == FmtSuperStereo)
{
al::LoadSampleArray<Type>(dstSamples[0]+dstOffset, s, srcStep, samples);
al::LoadSampleArray<Type>(dstSamples[1]+dstOffset, s+sampleSize, srcStep, samples);
std::fill_n(dstSamples[2]+dstOffset, samples, 0.0f);
}
else
{
for(auto *dst : dstSamples)
{
al::LoadSampleArray<Type>(dst+dstOffset, s, srcStep, samples);
s += sampleSize;
}
}
}
void LoadSamples(const al::span<float*> dstSamples, const size_t dstOffset, const al::byte *src,
const size_t srcOffset, const FmtType srcType, const FmtChannels srcChans,
const size_t srcStep, const size_t samples) noexcept
{
#define HANDLE_FMT(T) case T: \
LoadSamples<T>(dstSamples, dstOffset, src, srcOffset, srcChans, 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
}
void LoadBufferStatic(VoiceBufferItem *buffer, VoiceBufferItem *bufferLoopItem,
const size_t dataPosInt, const FmtType sampleType, const FmtChannels sampleChannels,
const size_t srcStep, const size_t samplesToLoad, const al::span<float*> voiceSamples)
{
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)
{
/* Load what's left to play from the buffer */
const size_t remaining{minz(samplesToLoad, buffer->mSampleLen-dataPosInt)};
LoadSamples(voiceSamples, 0, buffer->mSamples, dataPosInt, sampleType, sampleChannels,
srcStep, remaining);
if(const size_t toFill{samplesToLoad - remaining})
{
for(auto *chanbuffer : voiceSamples)
{
auto srcsamples = chanbuffer + remaining - 1;
std::fill_n(srcsamples + 1, toFill, *srcsamples);
}
}
}
else
{
/* Load what's left of this loop iteration */
const size_t remaining{minz(samplesToLoad, loopEnd-dataPosInt)};
LoadSamples(voiceSamples, 0, buffer->mSamples, dataPosInt, sampleType, sampleChannels,
srcStep, remaining);
/* Load repeats of the loop to fill the buffer. */
const auto loopSize = static_cast<size_t>(loopEnd - loopStart);
size_t samplesLoaded{remaining};
while(const size_t toFill{minz(samplesToLoad - samplesLoaded, loopSize)})
{
LoadSamples(voiceSamples, samplesLoaded, buffer->mSamples, loopStart, sampleType,
sampleChannels, srcStep, toFill);
samplesLoaded += toFill;
}
}
}
void LoadBufferCallback(VoiceBufferItem *buffer, const size_t numCallbackSamples,
const FmtType sampleType, const FmtChannels sampleChannels, const size_t srcStep,
const size_t samplesToLoad, const al::span<float*> voiceSamples)
{
/* Load what's left to play from the buffer */
const size_t remaining{minz(samplesToLoad, numCallbackSamples)};
LoadSamples(voiceSamples, 0, buffer->mSamples, 0, sampleType, sampleChannels, srcStep,
remaining);
if(const size_t toFill{samplesToLoad - remaining})
{
for(auto *chanbuffer : voiceSamples)
{
auto srcsamples = chanbuffer + remaining - 1;
std::fill_n(srcsamples + 1, toFill, *srcsamples);
}
}
}
void LoadBufferQueue(VoiceBufferItem *buffer, VoiceBufferItem *bufferLoopItem,
size_t dataPosInt, const FmtType sampleType, const FmtChannels sampleChannels,
const size_t srcStep, const size_t samplesToLoad, const al::span<float*> voiceSamples)
{
/* Crawl the buffer queue to fill in the temp buffer */
size_t samplesLoaded{0};
while(buffer && samplesLoaded != samplesToLoad)
{
if(dataPosInt >= buffer->mSampleLen)
{
dataPosInt -= buffer->mSampleLen;
buffer = buffer->mNext.load(std::memory_order_acquire);
if(!buffer) buffer = bufferLoopItem;
continue;
}
const size_t remaining{minz(samplesToLoad-samplesLoaded, buffer->mSampleLen-dataPosInt)};
LoadSamples(voiceSamples, samplesLoaded, buffer->mSamples, dataPosInt, sampleType,
sampleChannels, srcStep, remaining);
samplesLoaded += remaining;
if(samplesLoaded == samplesToLoad)
break;
dataPosInt = 0;
buffer = buffer->mNext.load(std::memory_order_acquire);
if(!buffer) buffer = bufferLoopItem;
}
if(const size_t toFill{samplesToLoad - samplesLoaded})
{
size_t chanidx{0};
for(auto *chanbuffer : voiceSamples)
{
auto srcsamples = chanbuffer + samplesLoaded - 1;
std::fill_n(srcsamples + 1, toFill, *srcsamples);
++chanidx;
}
}
}
void DoHrtfMix(const float *samples, const uint DstBufferSize, DirectParams &parms,
const float TargetGain, const uint Counter, uint OutPos, const bool IsPlaying,
DeviceBase *Device)
{
const uint IrSize{Device->mIrSize};
auto &HrtfSamples = Device->HrtfSourceData;
auto &AccumSamples = Device->HrtfAccumData;
/* 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. */
if(likely(IsPlaying))
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 = lerpf(parms.Hrtf.Old.Gain, TargetGain, a);
}
MixHrtfFilter hrtfparams{
parms.Hrtf.Target.Coeffs,
parms.Hrtf.Target.Delay,
0.0f, 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 = lerpf(parms.Hrtf.Old.Gain, TargetGain, a);
}
MixHrtfFilter hrtfparams{
parms.Hrtf.Target.Coeffs,
parms.Hrtf.Target.Delay,
parms.Hrtf.Old.Gain,
(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, DeviceBase *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, ContextBase *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 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;
}
DeviceBase *Device{Context->mDevice};
const uint NumSends{Device->NumAuxSends};
ResamplerFunc Resample{(increment == MixerFracOne && DataPosFrac == 0) ?
Resample_<CopyTag,CTag> : mResampler};
uint Counter{mFlags.test(VoiceIsFading) ? SamplesToDo : 0};
if(!Counter)
{
/* No fading, just overwrite the old/current params. */
for(auto &chandata : mChans)
{
{
DirectParams &parms = chandata.mDryParams;
if(!mFlags.test(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);
std::array<float*,DeviceBase::MixerChannelsMax> SamplePointers;
const al::span<float*> MixingSamples{SamplePointers.data(), mChans.size()};
auto offset_bufferline = [](DeviceBase::MixerBufferLine &bufline) noexcept -> float*
{ return bufline.data() + MaxResamplerEdge; };
std::transform(Device->mSampleData.end() - mChans.size(), Device->mSampleData.end(),
MixingSamples.begin(), offset_bufferline);
const uint PostPadding{MaxResamplerEdge +
(mDecoder ? uint{UhjDecoder::sFilterDelay} : 0u)};
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 + PostPadding;
/* Result is guaranteed to be <= BufferLineSize+PostPadding 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 += PostPadding;
if(DataSize64 <= DeviceBase::MixerLineSize - MaxResamplerEdge)
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 = DeviceBase::MixerLineSize - MaxResamplerEdge;
DataSize64 = SrcBufferSize - PostPadding;
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 the voice is stopping, only one mixing iteration will
* be done, so ensure it fades out completely this mix.
*/
if(unlikely(vstate == Stopping))
Counter = std::min(Counter, DstBufferSize);
}
ASSUME(DstBufferSize > 0);
}
}
if(unlikely(!BufferListItem))
{
const size_t srcOffset{(increment*DstBufferSize + DataPosFrac)>>MixerFracBits};
auto prevSamples = mPrevSamples.data();
SrcBufferSize = SrcBufferSize - PostPadding + MaxResamplerEdge;
for(auto *chanbuffer : MixingSamples)
{
auto srcend = std::copy_n(prevSamples->data(), MaxResamplerPadding,
chanbuffer-MaxResamplerEdge);
/* 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 srciter = std::min_element(chanbuffer, srcend, abs_lt);
std::fill(srciter+1, chanbuffer + SrcBufferSize, *srciter);
std::copy_n(chanbuffer-MaxResamplerEdge+srcOffset, prevSamples->size(),
prevSamples->data());
++prevSamples;
}
}
else
{
auto prevSamples = mPrevSamples.data();
for(auto *chanbuffer : MixingSamples)
{
std::copy_n(prevSamples->data(), MaxResamplerEdge, chanbuffer-MaxResamplerEdge);
++prevSamples;
}
if(mFlags.test(VoiceIsStatic))
LoadBufferStatic(BufferListItem, BufferLoopItem, DataPosInt, mFmtType,
mFmtChannels, mFrameStep, SrcBufferSize, MixingSamples);
else if(mFlags.test(VoiceIsCallback))
{
if(!mFlags.test(VoiceCallbackStopped) && SrcBufferSize > mNumCallbackSamples)
{
const size_t byteOffset{mNumCallbackSamples*mFrameSize};
const size_t needBytes{SrcBufferSize*mFrameSize - byteOffset};
const int gotBytes{BufferListItem->mCallback(BufferListItem->mUserData,
&BufferListItem->mSamples[byteOffset], static_cast<int>(needBytes))};
if(gotBytes < 0)
mFlags.set(VoiceCallbackStopped);
else if(static_cast<uint>(gotBytes) < needBytes)
{
mFlags.set(VoiceCallbackStopped);
mNumCallbackSamples += static_cast<uint>(gotBytes) / mFrameSize;
}
else
mNumCallbackSamples = SrcBufferSize;
}
LoadBufferCallback(BufferListItem, mNumCallbackSamples, mFmtType, mFmtChannels,
mFrameStep, SrcBufferSize, MixingSamples);
}
else
LoadBufferQueue(BufferListItem, BufferLoopItem, DataPosInt, mFmtType, mFmtChannels,
mFrameStep, SrcBufferSize, MixingSamples);
const size_t srcOffset{(increment*DstBufferSize + DataPosFrac)>>MixerFracBits};
if(mDecoder)
{
SrcBufferSize = SrcBufferSize - PostPadding + MaxResamplerEdge;
((*mDecoder).*mDecoderFunc)(MixingSamples, SrcBufferSize,
srcOffset * likely(vstate == Playing));
}
/* Store the last source samples used for next time. */
if(likely(vstate == Playing))
{
prevSamples = mPrevSamples.data();
for(auto *chanbuffer : MixingSamples)
{
/* Store the last source samples used for next time. */
std::copy_n(chanbuffer-MaxResamplerEdge+srcOffset, prevSamples->size(),
prevSamples->data());
++prevSamples;
}
}
}
auto voiceSamples = MixingSamples.begin();
for(auto &chandata : mChans)
{
/* Resample, then apply ambisonic upsampling as needed. */
float *ResampledData{Resample(&mResampleState, *voiceSamples, DataPosFrac, increment,
{Device->ResampledData, DstBufferSize})};
++voiceSamples;
if(mFlags.test(VoiceIsAmbisonic))
chandata.mAmbiSplitter.processScale({ResampledData, DstBufferSize},
chandata.mAmbiHFScale, chandata.mAmbiLFScale);
/* Now filter and mix to the appropriate outputs. */
const al::span<float,BufferLineSize> FilterBuf{Device->FilteredData};
{
DirectParams &parms = chandata.mDryParams;
const float *samples{DoFilters(parms.LowPass, parms.HighPass, FilterBuf.data(),
{ResampledData, DstBufferSize}, mDirect.FilterType)};
if(mFlags.test(VoiceHasHrtf))
{
const float TargetGain{parms.Hrtf.Target.Gain * likely(vstate == Playing)};
DoHrtfMix(samples, DstBufferSize, parms, TargetGain, Counter, OutPos,
(vstate == Playing), Device);
}
else
{
const float *TargetGains{likely(vstate == Playing) ? parms.Gains.Target.data()
: SilentTarget.data()};
if(mFlags.test(VoiceHasNfc))
DoNfcMix({samples, DstBufferSize}, mDirect.Buffer.data(), parms,
TargetGains, Counter, OutPos, Device);
else
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.data(),
{ResampledData, DstBufferSize}, mSend[send].FilterType)};
const float *TargetGains{likely(vstate == Playing) ? parms.Gains.Target.data()
: SilentTarget.data()};
MixSamples({samples, DstBufferSize}, mSend[send].Buffer,
parms.Gains.Current.data(), TargetGains, Counter, OutPos);
}
}
/* If the voice is stopping, we're now done. */
if(unlikely(vstate == Stopping))
break;
/* 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.test(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.test(VoiceIsCallback))
{
/* Handle callback buffer source */
if(SrcSamplesDone < mNumCallbackSamples)
{
const size_t byteOffset{SrcSamplesDone*mFrameSize};
const size_t byteEnd{mNumCallbackSamples*mFrameSize};
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.set(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&AsyncEvent::BufferCompleted))
{
RingBuffer *ring{Context->mAsyncEvents.get()};
auto evt_vec = ring->getWriteVector();
if(evt_vec.first.len > 0)
{
AsyncEvent *evt{al::construct_at(reinterpret_cast<AsyncEvent*>(evt_vec.first.buf),
AsyncEvent::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&AsyncEvent::SourceStateChange))
SendSourceStoppedEvent(Context, SourceID);
}
}
void Voice::prepare(DeviceBase *device)
{
/* Even if storing really high order ambisonics, we only mix channels for
* orders up to the device order. The rest are simply dropped.
*/
uint num_channels{(mFmtChannels == FmtUHJ2 || mFmtChannels == FmtSuperStereo) ? 3 :
ChannelsFromFmt(mFmtChannels, minu(mAmbiOrder, device->mAmbiOrder))};
if(unlikely(num_channels > device->mSampleData.size()))
{
ERR("Unexpected channel count: %u (limit: %zu, %d:%d)\n", num_channels,
device->mSampleData.size(), mFmtChannels, mAmbiOrder);
num_channels = static_cast<uint>(device->mSampleData.size());
}
if(mChans.capacity() > 2 && num_channels < mChans.capacity())
{
decltype(mChans){}.swap(mChans);
decltype(mPrevSamples){}.swap(mPrevSamples);
}
mChans.reserve(maxu(2, num_channels));
mChans.resize(num_channels);
mPrevSamples.reserve(maxu(2, num_channels));
mPrevSamples.resize(num_channels);
if(IsUHJ(mFmtChannels))
{
mDecoder = std::make_unique<UhjDecoder>();
mDecoderFunc = (mFmtChannels == FmtSuperStereo) ? &UhjDecoder::decodeStereo
: &UhjDecoder::decode;
}
else
{
mDecoder = nullptr;
mDecoderFunc = nullptr;
}
/* Clear the stepping value explicitly so the mixer knows not to mix this
* until the update gets applied.
*/
mStep = 0;
/* Make sure the sample history is cleared. */
std::fill(mPrevSamples.begin(), mPrevSamples.end(), HistoryLine{});
/* Don't need to set the VoiceIsAmbisonic flag if the device is not higher
* order than the voice. No HF scaling is necessary to mix it.
*/
if(mAmbiOrder && device->mAmbiOrder > mAmbiOrder)
{
const uint8_t *OrderFromChan{Is2DAmbisonic(mFmtChannels) ?
AmbiIndex::OrderFrom2DChannel().data() : AmbiIndex::OrderFromChannel().data()};
const auto scales = AmbiScale::GetHFOrderScales(mAmbiOrder, device->mAmbiOrder);
const BandSplitter splitter{device->mXOverFreq / static_cast<float>(device->Frequency)};
for(auto &chandata : mChans)
{
chandata.mAmbiHFScale = scales[*(OrderFromChan++)];
chandata.mAmbiLFScale = 1.0f;
chandata.mAmbiSplitter = splitter;
chandata.mDryParams = DirectParams{};
chandata.mDryParams.NFCtrlFilter = device->mNFCtrlFilter;
std::fill_n(chandata.mWetParams.begin(), device->NumAuxSends, SendParams{});
}
/* 2-channel UHJ needs different shelf filters. However, we can't just
* use different shelf filters after mixing it and with any old speaker
* setup the user has. To make this work, we apply the expected shelf
* filters for decoding UHJ2 to quad (only needs LF scaling), and act
* as if those 4 quad channels are encoded right back onto first-order
* B-Format, which then upsamples to higher order as normal (only needs
* HF scaling).
*
* This isn't perfect, but without an entirely separate and limited
* UHJ2 path, it's better than nothing.
*/
if(mFmtChannels == FmtUHJ2)
{
mChans[0].mAmbiLFScale = 0.661f;
mChans[1].mAmbiLFScale = 1.293f;
mChans[2].mAmbiLFScale = 1.293f;
}
mFlags.set(VoiceIsAmbisonic);
}
else if(mFmtChannels == FmtUHJ2 && !device->mUhjEncoder)
{
/* 2-channel UHJ with first-order output also needs the shelf filter
* correction applied, except with UHJ output (UHJ2->B-Format->UHJ2 is
* identity, so don't mess with it).
*/
const BandSplitter splitter{device->mXOverFreq / static_cast<float>(device->Frequency)};
for(auto &chandata : mChans)
{
chandata.mAmbiHFScale = 1.0f;
chandata.mAmbiLFScale = 1.0f;
chandata.mAmbiSplitter = splitter;
chandata.mDryParams = DirectParams{};
chandata.mDryParams.NFCtrlFilter = device->mNFCtrlFilter;
std::fill_n(chandata.mWetParams.begin(), device->NumAuxSends, SendParams{});
}
mChans[0].mAmbiLFScale = 0.661f;
mChans[1].mAmbiLFScale = 1.293f;
mChans[2].mAmbiLFScale = 1.293f;
mFlags.set(VoiceIsAmbisonic);
}
else
{
for(auto &chandata : mChans)
{
chandata.mDryParams = DirectParams{};
chandata.mDryParams.NFCtrlFilter = device->mNFCtrlFilter;
std::fill_n(chandata.mWetParams.begin(), device->NumAuxSends, SendParams{});
}
mFlags.reset(VoiceIsAmbisonic);
}
}
Reference in New Issue View Git Blame Copy Permalink