/**
* 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 "alu.h"
#include <algorithm>
#include <array>
#include <atomic>
#include <cassert>
#include <chrono>
#include <climits>
#include <cmath>
#include <cstdarg>
#include <cstdio>
#include <cstdlib>
#include <cstring>
#include <functional>
#include <iterator>
#include <limits>
#include <memory>
#include <new>
#include <numeric>
#include <utility>
#include "AL/al.h"
#include "AL/alc.h"
#include "AL/efx.h"
#include "alAuxEffectSlot.h"
#include "alBuffer.h"
#include "alcmain.h"
#include "alEffect.h"
#include "alListener.h"
#include "alcontext.h"
#include "almalloc.h"
#include "alnumeric.h"
#include "alspan.h"
#include "ambidefs.h"
#include "atomic.h"
#include "bformatdec.h"
#include "bs2b.h"
#include "cpu_caps.h"
#include "effects/base.h"
#include "filters/biquad.h"
#include "filters/nfc.h"
#include "filters/splitter.h"
#include "fpu_modes.h"
#include "hrtf.h"
#include "inprogext.h"
#include "mastering.h"
#include "math_defs.h"
#include "mixer/defs.h"
#include "opthelpers.h"
#include "ringbuffer.h"
#include "threads.h"
#include "uhjfilter.h"
#include "vecmat.h"
#include "vector.h"
#include "bsinc_inc.h"
namespace {
using namespace std::placeholders;
ALfloat InitConeScale()
{
ALfloat ret{1.0f};
const char *str{getenv("__ALSOFT_HALF_ANGLE_CONES")};
if(str && (strcasecmp(str, "true") == 0 || strtol(str, nullptr, 0) == 1))
ret *= 0.5f;
return ret;
}
ALfloat InitZScale()
{
ALfloat ret{1.0f};
const char *str{getenv("__ALSOFT_REVERSE_Z")};
if(str && (strcasecmp(str, "true") == 0 || strtol(str, nullptr, 0) == 1))
ret *= -1.0f;
return ret;
}
ALboolean InitReverbSOS()
{
ALboolean ret{AL_FALSE};
const char *str{getenv("__ALSOFT_REVERB_IGNORES_SOUND_SPEED")};
if(str && (strcasecmp(str, "true") == 0 || strtol(str, nullptr, 0) == 1))
ret = AL_TRUE;
return ret;
}
} // namespace
/* Cone scalar */
const ALfloat ConeScale{InitConeScale()};
/* Localized Z scalar for mono sources */
const ALfloat ZScale{InitZScale()};
/* Force default speed of sound for distance-related reverb decay. */
const ALboolean OverrideReverbSpeedOfSound{InitReverbSOS()};
namespace {
void ClearArray(ALfloat (&f)[MAX_OUTPUT_CHANNELS])
{
std::fill(std::begin(f), std::end(f), 0.0f);
}
struct ChanMap {
Channel channel;
ALfloat angle;
ALfloat elevation;
};
HrtfDirectMixerFunc MixDirectHrtf = MixDirectHrtf_<CTag>;
inline HrtfDirectMixerFunc SelectHrtfMixer(void)
{
#ifdef HAVE_NEON
if((CPUCapFlags&CPU_CAP_NEON))
return MixDirectHrtf_<NEONTag>;
#endif
#ifdef HAVE_SSE
if((CPUCapFlags&CPU_CAP_SSE))
return MixDirectHrtf_<SSETag>;
#endif
return MixDirectHrtf_<CTag>;
}
} // namespace
void aluInit(void)
{
MixDirectHrtf = SelectHrtfMixer();
}
void ProcessHrtf(ALCdevice *device, const ALsizei SamplesToDo)
{
/* HRTF is stereo output only. */
const int lidx{device->RealOut.ChannelIndex[FrontLeft]};
const int ridx{device->RealOut.ChannelIndex[FrontRight]};
ASSUME(lidx >= 0 && ridx >= 0);
DirectHrtfState *state{device->mHrtfState.get()};
MixDirectHrtf(device->RealOut.Buffer[lidx], device->RealOut.Buffer[ridx], device->Dry.Buffer,
device->HrtfAccumData, state, SamplesToDo);
}
void ProcessAmbiDec(ALCdevice *device, const ALsizei SamplesToDo)
{
BFormatDec *ambidec{device->AmbiDecoder.get()};
ambidec->process(device->RealOut.Buffer, device->Dry.Buffer.data(), SamplesToDo);
}
void ProcessUhj(ALCdevice *device, const ALsizei SamplesToDo)
{
/* UHJ is stereo output only. */
const int lidx{device->RealOut.ChannelIndex[FrontLeft]};
const int ridx{device->RealOut.ChannelIndex[FrontRight]};
ASSUME(lidx >= 0 && ridx >= 0);
/* Encode to stereo-compatible 2-channel UHJ output. */
Uhj2Encoder *uhj2enc{device->Uhj_Encoder.get()};
uhj2enc->encode(device->RealOut.Buffer[lidx], device->RealOut.Buffer[ridx],
device->Dry.Buffer.data(), SamplesToDo);
}
void ProcessBs2b(ALCdevice *device, const ALsizei SamplesToDo)
{
/* First, decode the ambisonic mix to the "real" output. */
BFormatDec *ambidec{device->AmbiDecoder.get()};
ambidec->process(device->RealOut.Buffer, device->Dry.Buffer.data(), SamplesToDo);
/* BS2B is stereo output only. */
const int lidx{device->RealOut.ChannelIndex[FrontLeft]};
const int ridx{device->RealOut.ChannelIndex[FrontRight]};
ASSUME(lidx >= 0 && ridx >= 0);
/* Now apply the BS2B binaural/crossfeed filter. */
bs2b_cross_feed(device->Bs2b.get(), device->RealOut.Buffer[lidx].data(),
device->RealOut.Buffer[ridx].data(), SamplesToDo);
}
/* Prepares the interpolator for a given rate (determined by increment).
*
* With a bit of work, and a trade of memory for CPU cost, this could be
* modified for use with an interpolated increment for buttery-smooth pitch
* changes.
*/
void BsincPrepare(const ALuint increment, BsincState *state, const BSincTable *table)
{
ALsizei si{BSINC_SCALE_COUNT - 1};
ALfloat sf{0.0f};
if(increment > FRACTIONONE)
{
sf = static_cast<ALfloat>FRACTIONONE / increment;
sf = maxf(0.0f, (BSINC_SCALE_COUNT-1) * (sf-table->scaleBase) * table->scaleRange);
si = float2int(sf);
/* The interpolation factor is fit to this diagonally-symmetric curve
* to reduce the transition ripple caused by interpolating different
* scales of the sinc function.
*/
sf = 1.0f - std::cos(std::asin(sf - si));
}
state->sf = sf;
state->m = table->m[si];
state->l = (state->m/2) - 1;
state->filter = table->Tab + table->filterOffset[si];
}
namespace {
/* This RNG method was created based on the math found in opusdec. It's quick,
* and starting with a seed value of 22222, is suitable for generating
* whitenoise.
*/
inline ALuint dither_rng(ALuint *seed) noexcept
{
*seed = (*seed * 96314165) + 907633515;
return *seed;
}
inline alu::Vector aluCrossproduct(const alu::Vector &in1, const alu::Vector &in2)
{
return alu::Vector{
in1[1]*in2[2] - in1[2]*in2[1],
in1[2]*in2[0] - in1[0]*in2[2],
in1[0]*in2[1] - in1[1]*in2[0],
0.0f
};
}
inline ALfloat aluDotproduct(const alu::Vector &vec1, const alu::Vector &vec2)
{
return vec1[0]*vec2[0] + vec1[1]*vec2[1] + vec1[2]*vec2[2];
}
alu::Vector operator*(const alu::Matrix &mtx, const alu::Vector &vec) noexcept
{
return alu::Vector{
vec[0]*mtx[0][0] + vec[1]*mtx[1][0] + vec[2]*mtx[2][0] + vec[3]*mtx[3][0],
vec[0]*mtx[0][1] + vec[1]*mtx[1][1] + vec[2]*mtx[2][1] + vec[3]*mtx[3][1],
vec[0]*mtx[0][2] + vec[1]*mtx[1][2] + vec[2]*mtx[2][2] + vec[3]*mtx[3][2],
vec[0]*mtx[0][3] + vec[1]*mtx[1][3] + vec[2]*mtx[2][3] + vec[3]*mtx[3][3]
};
}
bool CalcContextParams(ALCcontext *Context)
{
ALcontextProps *props{Context->Update.exchange(nullptr, std::memory_order_acq_rel)};
if(!props) return false;
ALlistener &Listener = Context->Listener;
Listener.Params.MetersPerUnit = props->MetersPerUnit;
Listener.Params.DopplerFactor = props->DopplerFactor;
Listener.Params.SpeedOfSound = props->SpeedOfSound * props->DopplerVelocity;
if(!OverrideReverbSpeedOfSound)
Listener.Params.ReverbSpeedOfSound = Listener.Params.SpeedOfSound *
Listener.Params.MetersPerUnit;
Listener.Params.SourceDistanceModel = props->SourceDistanceModel;
Listener.Params.mDistanceModel = props->mDistanceModel;
AtomicReplaceHead(Context->FreeContextProps, props);
return true;
}
bool CalcListenerParams(ALCcontext *Context)
{
ALlistener &Listener = Context->Listener;
ALlistenerProps *props{Listener.Update.exchange(nullptr, std::memory_order_acq_rel)};
if(!props) return false;
/* AT then UP */
alu::Vector N{props->OrientAt[0], props->OrientAt[1], props->OrientAt[2], 0.0f};
N.normalize();
alu::Vector V{props->OrientUp[0], props->OrientUp[1], props->OrientUp[2], 0.0f};
V.normalize();
/* Build and normalize right-vector */
alu::Vector U{aluCrossproduct(N, V)};
U.normalize();
Listener.Params.Matrix = alu::Matrix{
U[0], V[0], -N[0], 0.0f,
U[1], V[1], -N[1], 0.0f,
U[2], V[2], -N[2], 0.0f,
0.0f, 0.0f, 0.0f, 1.0f
};
const alu::Vector P{Listener.Params.Matrix *
alu::Vector{props->Position[0], props->Position[1], props->Position[2], 1.0f}};
Listener.Params.Matrix.setRow(3, -P[0], -P[1], -P[2], 1.0f);
const alu::Vector vel{props->Velocity[0], props->Velocity[1], props->Velocity[2], 0.0f};
Listener.Params.Velocity = Listener.Params.Matrix * vel;
Listener.Params.Gain = props->Gain * Context->GainBoost;
AtomicReplaceHead(Context->FreeListenerProps, props);
return true;
}
bool CalcEffectSlotParams(ALeffectslot *slot, ALCcontext *context, bool force)
{
ALeffectslotProps *props{slot->Update.exchange(nullptr, std::memory_order_acq_rel)};
if(!props && !force) return false;
EffectState *state;
if(!props)
state = slot->Params.mEffectState;
else
{
slot->Params.Gain = props->Gain;
slot->Params.AuxSendAuto = props->AuxSendAuto;
slot->Params.Target = props->Target;
slot->Params.EffectType = props->Type;
slot->Params.mEffectProps = props->Props;
if(IsReverbEffect(props->Type))
{
slot->Params.RoomRolloff = props->Props.Reverb.RoomRolloffFactor;
slot->Params.DecayTime = props->Props.Reverb.DecayTime;
slot->Params.DecayLFRatio = props->Props.Reverb.DecayLFRatio;
slot->Params.DecayHFRatio = props->Props.Reverb.DecayHFRatio;
slot->Params.DecayHFLimit = props->Props.Reverb.DecayHFLimit;
slot->Params.AirAbsorptionGainHF = props->Props.Reverb.AirAbsorptionGainHF;
}
else
{
slot->Params.RoomRolloff = 0.0f;
slot->Params.DecayTime = 0.0f;
slot->Params.DecayLFRatio = 0.0f;
slot->Params.DecayHFRatio = 0.0f;
slot->Params.DecayHFLimit = AL_FALSE;
slot->Params.AirAbsorptionGainHF = 1.0f;
}
state = props->State;
props->State = nullptr;
EffectState *oldstate{slot->Params.mEffectState};
slot->Params.mEffectState = state;
/* Manually decrement the old effect state's refcount if it's greater
* than 1. We need to be a bit clever here to avoid the refcount
* reaching 0 since it can't be deleted in the mixer.
*/
ALuint oldval{oldstate->mRef.load(std::memory_order_acquire)};
while(oldval > 1 && !oldstate->mRef.compare_exchange_weak(oldval, oldval-1,
std::memory_order_acq_rel, std::memory_order_acquire))
{
/* oldval was updated with the current value on failure, so just
* try again.
*/
}
if(oldval < 2)
{
/* Otherwise, if it would be deleted, send it off with a release
* event.
*/
RingBuffer *ring{context->AsyncEvents.get()};
auto evt_vec = ring->getWriteVector();
if(LIKELY(evt_vec.first.len > 0))
{
AsyncEvent *evt{new (evt_vec.first.buf) AsyncEvent{EventType_ReleaseEffectState}};
evt->u.mEffectState = oldstate;
ring->writeAdvance(1);
context->EventSem.post();
}
else
{
/* If writing the event failed, the queue was probably full.
* Store the old state in the property object where it can
* eventually be cleaned up sometime later (not ideal, but
* better than blocking or leaking).
*/
props->State = oldstate;
}
}
AtomicReplaceHead(context->FreeEffectslotProps, props);
}
EffectTarget output;
if(ALeffectslot *target{slot->Params.Target})
output = EffectTarget{&target->Wet, nullptr};
else
{
ALCdevice *device{context->Device};
output = EffectTarget{&device->Dry, &device->RealOut};
}
state->update(context, slot, &slot->Params.mEffectProps, output);
return true;
}
/* Scales the given azimuth toward the side (+/- pi/2 radians) for positions in
* front.
*/
inline float ScaleAzimuthFront(float azimuth, float scale)
{
const ALfloat abs_azi{std::fabs(azimuth)};
if(!(abs_azi > al::MathDefs<float>::Pi()*0.5f))
return minf(abs_azi*scale, al::MathDefs<float>::Pi()*0.5f) * std::copysign(1.0f, azimuth);
return azimuth;
}
void CalcPanningAndFilters(ALvoice *voice, const ALfloat xpos, const ALfloat ypos,
const ALfloat zpos, const ALfloat Distance, const ALfloat Spread, const ALfloat DryGain,
const ALfloat DryGainHF, const ALfloat DryGainLF, const ALfloat (&WetGain)[MAX_SENDS],
const ALfloat (&WetGainLF)[MAX_SENDS], const ALfloat (&WetGainHF)[MAX_SENDS],
ALeffectslot *(&SendSlots)[MAX_SENDS], const ALvoicePropsBase *props,
const ALlistener &Listener, const ALCdevice *Device)
{
static constexpr ChanMap MonoMap[1]{
{ FrontCenter, 0.0f, 0.0f }
}, RearMap[2]{
{ BackLeft, Deg2Rad(-150.0f), Deg2Rad(0.0f) },
{ BackRight, Deg2Rad( 150.0f), Deg2Rad(0.0f) }
}, QuadMap[4]{
{ FrontLeft, Deg2Rad( -45.0f), Deg2Rad(0.0f) },
{ FrontRight, Deg2Rad( 45.0f), Deg2Rad(0.0f) },
{ BackLeft, Deg2Rad(-135.0f), Deg2Rad(0.0f) },
{ BackRight, Deg2Rad( 135.0f), Deg2Rad(0.0f) }
}, X51Map[6]{
{ FrontLeft, Deg2Rad( -30.0f), Deg2Rad(0.0f) },
{ FrontRight, Deg2Rad( 30.0f), Deg2Rad(0.0f) },
{ FrontCenter, Deg2Rad( 0.0f), Deg2Rad(0.0f) },
{ LFE, 0.0f, 0.0f },
{ SideLeft, Deg2Rad(-110.0f), Deg2Rad(0.0f) },
{ SideRight, Deg2Rad( 110.0f), Deg2Rad(0.0f) }
}, X61Map[7]{
{ FrontLeft, Deg2Rad(-30.0f), Deg2Rad(0.0f) },
{ FrontRight, Deg2Rad( 30.0f), Deg2Rad(0.0f) },
{ FrontCenter, Deg2Rad( 0.0f), Deg2Rad(0.0f) },
{ LFE, 0.0f, 0.0f },
{ BackCenter, Deg2Rad(180.0f), Deg2Rad(0.0f) },
{ SideLeft, Deg2Rad(-90.0f), Deg2Rad(0.0f) },
{ SideRight, Deg2Rad( 90.0f), Deg2Rad(0.0f) }
}, X71Map[8]{
{ FrontLeft, Deg2Rad( -30.0f), Deg2Rad(0.0f) },
{ FrontRight, Deg2Rad( 30.0f), Deg2Rad(0.0f) },
{ FrontCenter, Deg2Rad( 0.0f), Deg2Rad(0.0f) },
{ LFE, 0.0f, 0.0f },
{ BackLeft, Deg2Rad(-150.0f), Deg2Rad(0.0f) },
{ BackRight, Deg2Rad( 150.0f), Deg2Rad(0.0f) },
{ SideLeft, Deg2Rad( -90.0f), Deg2Rad(0.0f) },
{ SideRight, Deg2Rad( 90.0f), Deg2Rad(0.0f) }
};
ChanMap StereoMap[2]{
{ FrontLeft, Deg2Rad(-30.0f), Deg2Rad(0.0f) },
{ FrontRight, Deg2Rad( 30.0f), Deg2Rad(0.0f) }
};
const auto Frequency = static_cast<ALfloat>(Device->Frequency);
const ALsizei NumSends{Device->NumAuxSends};
ASSUME(NumSends >= 0);
bool DirectChannels{props->DirectChannels != AL_FALSE};
const ChanMap *chans{nullptr};
ALsizei num_channels{0};
bool isbformat{false};
ALfloat downmix_gain{1.0f};
switch(voice->mFmtChannels)
{
case FmtMono:
chans = MonoMap;
num_channels = 1;
/* Mono buffers are never played direct. */
DirectChannels = false;
break;
case FmtStereo:
/* Convert counter-clockwise to clockwise. */
StereoMap[0].angle = -props->StereoPan[0];
StereoMap[1].angle = -props->StereoPan[1];
chans = StereoMap;
num_channels = 2;
downmix_gain = 1.0f / 2.0f;
break;
case FmtRear:
chans = RearMap;
num_channels = 2;
downmix_gain = 1.0f / 2.0f;
break;
case FmtQuad:
chans = QuadMap;
num_channels = 4;
downmix_gain = 1.0f / 4.0f;
break;
case FmtX51:
chans = X51Map;
num_channels = 6;
/* NOTE: Excludes LFE. */
downmix_gain = 1.0f / 5.0f;
break;
case FmtX61:
chans = X61Map;
num_channels = 7;
/* NOTE: Excludes LFE. */
downmix_gain = 1.0f / 6.0f;
break;
case FmtX71:
chans = X71Map;
num_channels = 8;
/* NOTE: Excludes LFE. */
downmix_gain = 1.0f / 7.0f;
break;
case FmtBFormat2D:
num_channels = 3;
isbformat = true;
DirectChannels = false;
break;
case FmtBFormat3D:
num_channels = 4;
isbformat = true;
DirectChannels = false;
break;
}
ASSUME(num_channels > 0);
std::for_each(voice->mChans.begin(), voice->mChans.begin()+num_channels,
[NumSends](ALvoice::ChannelData &chandata) -> void
{
chandata.mDryParams.Hrtf.Target = HrtfFilter{};
ClearArray(chandata.mDryParams.Gains.Target);
std::for_each(chandata.mWetParams.begin(), chandata.mWetParams.begin()+NumSends,
[](SendParams ¶ms) -> void { ClearArray(params.Gains.Target); });
});
voice->mFlags &= ~(VOICE_HAS_HRTF | VOICE_HAS_NFC);
if(isbformat)
{
/* Special handling for B-Format sources. */
if(Distance > std::numeric_limits<float>::epsilon())
{
/* Panning a B-Format sound toward some direction is easy. Just pan
* the first (W) channel as a normal mono sound and silence the
* others.
*/
if(Device->AvgSpeakerDist > 0.0f)
{
/* Clamp the distance for really close sources, to prevent
* excessive bass.
*/
const ALfloat mdist{maxf(Distance, Device->AvgSpeakerDist/4.0f)};
const ALfloat w0{SPEEDOFSOUNDMETRESPERSEC / (mdist * Frequency)};
/* Only need to adjust the first channel of a B-Format source. */
voice->mChans[0].mDryParams.NFCtrlFilter.adjust(w0);
voice->mFlags |= VOICE_HAS_NFC;
}
ALfloat coeffs[MAX_AMBI_CHANNELS];
if(Device->mRenderMode != StereoPair)
CalcDirectionCoeffs({xpos, ypos, zpos}, Spread, coeffs);
else
{
/* Clamp Y, in case rounding errors caused it to end up outside
* of -1...+1.
*/
const ALfloat ev{std::asin(clampf(ypos, -1.0f, 1.0f))};
/* Negate Z for right-handed coords with -Z in front. */
const ALfloat az{std::atan2(xpos, -zpos)};
/* A scalar of 1.5 for plain stereo results in +/-60 degrees
* being moved to +/-90 degrees for direct right and left
* speaker responses.
*/
CalcAngleCoeffs(ScaleAzimuthFront(az, 1.5f), ev, Spread, coeffs);
}
/* NOTE: W needs to be scaled due to FuMa normalization. */
const ALfloat &scale0 = AmbiScale::FromFuMa[0];
ComputePanGains(&Device->Dry, coeffs, DryGain*scale0,
voice->mChans[0].mDryParams.Gains.Target);
for(ALsizei i{0};i < NumSends;i++)
{
if(const ALeffectslot *Slot{SendSlots[i]})
ComputePanGains(&Slot->Wet, coeffs, WetGain[i]*scale0,
voice->mChans[0].mWetParams[i].Gains.Target);
}
}
else
{
if(Device->AvgSpeakerDist > 0.0f)
{
/* NOTE: The NFCtrlFilters were created with a w0 of 0, which
* is what we want for FOA input. The first channel may have
* been previously re-adjusted if panned, so reset it.
*/
voice->mChans[0].mDryParams.NFCtrlFilter.adjust(0.0f);
voice->mFlags |= VOICE_HAS_NFC;
}
/* Local B-Format sources have their XYZ channels rotated according
* to the orientation.
*/
/* AT then UP */
alu::Vector N{props->OrientAt[0], props->OrientAt[1], props->OrientAt[2], 0.0f};
N.normalize();
alu::Vector V{props->OrientUp[0], props->OrientUp[1], props->OrientUp[2], 0.0f};
V.normalize();
if(!props->HeadRelative)
{
N = Listener.Params.Matrix * N;
V = Listener.Params.Matrix * V;
}
/* Build and normalize right-vector */
alu::Vector U{aluCrossproduct(N, V)};
U.normalize();
/* Build a rotate + conversion matrix (FuMa -> ACN+N3D). NOTE: This
* matrix is transposed, for the inputs to align on the rows and
* outputs on the columns.
*/
const ALfloat &wscale = AmbiScale::FromFuMa[0];
const ALfloat &yscale = AmbiScale::FromFuMa[1];
const ALfloat &zscale = AmbiScale::FromFuMa[2];
const ALfloat &xscale = AmbiScale::FromFuMa[3];
const ALfloat matrix[4][MAX_AMBI_CHANNELS]{
// ACN0 ACN1 ACN2 ACN3
{ wscale, 0.0f, 0.0f, 0.0f }, // FuMa W
{ 0.0f, -N[0]*xscale, N[1]*xscale, -N[2]*xscale }, // FuMa X
{ 0.0f, U[0]*yscale, -U[1]*yscale, U[2]*yscale }, // FuMa Y
{ 0.0f, -V[0]*zscale, V[1]*zscale, -V[2]*zscale } // FuMa Z
};
for(ALsizei c{0};c < num_channels;c++)
{
ComputePanGains(&Device->Dry, matrix[c], DryGain,
voice->mChans[c].mDryParams.Gains.Target);
for(ALsizei i{0};i < NumSends;i++)
{
if(const ALeffectslot *Slot{SendSlots[i]})
ComputePanGains(&Slot->Wet, matrix[c], WetGain[i],
voice->mChans[c].mWetParams[i].Gains.Target);
}
}
}
}
else if(DirectChannels)
{
/* Direct source channels always play local. Skip the virtual channels
* and write inputs to the matching real outputs.
*/
voice->mDirect.Buffer = Device->RealOut.Buffer;
for(ALsizei c{0};c < num_channels;c++)
{
int idx{GetChannelIdxByName(Device->RealOut, chans[c].channel)};
if(idx != -1) voice->mChans[c].mDryParams.Gains.Target[idx] = DryGain;
}
/* Auxiliary sends still use normal channel panning since they mix to
* B-Format, which can't channel-match.
*/
for(ALsizei c{0};c < num_channels;c++)
{
ALfloat coeffs[MAX_AMBI_CHANNELS];
CalcAngleCoeffs(chans[c].angle, chans[c].elevation, 0.0f, coeffs);
for(ALsizei i{0};i < NumSends;i++)
{
if(const ALeffectslot *Slot{SendSlots[i]})
ComputePanGains(&Slot->Wet, coeffs, WetGain[i],
voice->mChans[c].mWetParams[i].Gains.Target);
}
}
}
else if(Device->mRenderMode == HrtfRender)
{
/* Full HRTF rendering. Skip the virtual channels and render to the
* real outputs.
*/
voice->mDirect.Buffer = Device->RealOut.Buffer;
if(Distance > std::numeric_limits<float>::epsilon())
{
const ALfloat ev{std::asin(clampf(ypos, -1.0f, 1.0f))};
const ALfloat az{std::atan2(xpos, -zpos)};
/* Get the HRIR coefficients and delays just once, for the given
* source direction.
*/
GetHrtfCoeffs(Device->mHrtf, ev, az, Distance, Spread,
voice->mChans[0].mDryParams.Hrtf.Target.Coeffs,
voice->mChans[0].mDryParams.Hrtf.Target.Delay);
voice->mChans[0].mDryParams.Hrtf.Target.Gain = DryGain * downmix_gain;
/* Remaining channels use the same results as the first. */
for(ALsizei c{1};c < num_channels;c++)
{
/* Skip LFE */
if(chans[c].channel == LFE) continue;
voice->mChans[c].mDryParams.Hrtf.Target = voice->mChans[0].mDryParams.Hrtf.Target;
}
/* Calculate the directional coefficients once, which apply to all
* input channels of the source sends.
*/
ALfloat coeffs[MAX_AMBI_CHANNELS];
CalcDirectionCoeffs({xpos, ypos, zpos}, Spread, coeffs);
for(ALsizei c{0};c < num_channels;c++)
{
/* Skip LFE */
if(chans[c].channel == LFE)
continue;
for(ALsizei i{0};i < NumSends;i++)
{
if(const ALeffectslot *Slot{SendSlots[i]})
ComputePanGains(&Slot->Wet, coeffs, WetGain[i] * downmix_gain,
voice->mChans[c].mWetParams[i].Gains.Target);
}
}
}
else
{
/* Local sources on HRTF play with each channel panned to its
* relative location around the listener, providing "virtual
* speaker" responses.
*/
for(ALsizei c{0};c < num_channels;c++)
{
/* Skip LFE */
if(chans[c].channel == LFE)
continue;
/* Get the HRIR coefficients and delays for this channel
* position.
*/
GetHrtfCoeffs(Device->mHrtf, chans[c].elevation, chans[c].angle,
std::numeric_limits<float>::infinity(), Spread,
voice->mChans[c].mDryParams.Hrtf.Target.Coeffs,
voice->mChans[c].mDryParams.Hrtf.Target.Delay);
voice->mChans[c].mDryParams.Hrtf.Target.Gain = DryGain;
/* Normal panning for auxiliary sends. */
ALfloat coeffs[MAX_AMBI_CHANNELS];
CalcAngleCoeffs(chans[c].angle, chans[c].elevation, Spread, coeffs);
for(ALsizei i{0};i < NumSends;i++)
{
if(const ALeffectslot *Slot{SendSlots[i]})
ComputePanGains(&Slot->Wet, coeffs, WetGain[i],
voice->mChans[c].mWetParams[i].Gains.Target);
}
}
}
voice->mFlags |= VOICE_HAS_HRTF;
}
else
{
/* Non-HRTF rendering. Use normal panning to the output. */
if(Distance > std::numeric_limits<float>::epsilon())
{
/* Calculate NFC filter coefficient if needed. */
if(Device->AvgSpeakerDist > 0.0f)
{
/* Clamp the distance for really close sources, to prevent
* excessive bass.
*/
const ALfloat mdist{maxf(Distance, Device->AvgSpeakerDist/4.0f)};
const ALfloat w0{SPEEDOFSOUNDMETRESPERSEC / (mdist * Frequency)};
/* Adjust NFC filters. */
for(ALsizei c{0};c < num_channels;c++)
voice->mChans[c].mDryParams.NFCtrlFilter.adjust(w0);
voice->mFlags |= VOICE_HAS_NFC;
}
/* Calculate the directional coefficients once, which apply to all
* input channels.
*/
ALfloat coeffs[MAX_AMBI_CHANNELS];
if(Device->mRenderMode != StereoPair)
CalcDirectionCoeffs({xpos, ypos, zpos}, Spread, coeffs);
else
{
const ALfloat ev{std::asin(clampf(ypos, -1.0f, 1.0f))};
const ALfloat az{std::atan2(xpos, -zpos)};
CalcAngleCoeffs(ScaleAzimuthFront(az, 1.5f), ev, Spread, coeffs);
}
for(ALsizei c{0};c < num_channels;c++)
{
/* Special-case LFE */
if(chans[c].channel == LFE)
{
if(Device->Dry.Buffer.data() == Device->RealOut.Buffer.data())
{
int idx = GetChannelIdxByName(Device->RealOut, chans[c].channel);
if(idx != -1) voice->mChans[c].mDryParams.Gains.Target[idx] = DryGain;
}
continue;
}
ComputePanGains(&Device->Dry, coeffs, DryGain * downmix_gain,
voice->mChans[c].mDryParams.Gains.Target);
}
for(ALsizei c{0};c < num_channels;c++)
{
/* Skip LFE */
if(chans[c].channel == LFE)
continue;
for(ALsizei i{0};i < NumSends;i++)
{
if(const ALeffectslot *Slot{SendSlots[i]})
ComputePanGains(&Slot->Wet, coeffs, WetGain[i] * downmix_gain,
voice->mChans[c].mWetParams[i].Gains.Target);
}
}
}
else
{
if(Device->AvgSpeakerDist > 0.0f)
{
/* If the source distance is 0, set w0 to w1 to act as a pass-
* through. We still want to pass the signal through the
* filters so they keep an appropriate history, in case the
* source moves away from the listener.
*/
const ALfloat w0{SPEEDOFSOUNDMETRESPERSEC / (Device->AvgSpeakerDist * Frequency)};
for(ALsizei c{0};c < num_channels;c++)
voice->mChans[c].mDryParams.NFCtrlFilter.adjust(w0);
voice->mFlags |= VOICE_HAS_NFC;
}
for(ALsizei c{0};c < num_channels;c++)
{
/* Special-case LFE */
if(chans[c].channel == LFE)
{
if(Device->Dry.Buffer.data() == Device->RealOut.Buffer.data())
{
int idx = GetChannelIdxByName(Device->RealOut, chans[c].channel);
if(idx != -1) voice->mChans[c].mDryParams.Gains.Target[idx] = DryGain;
}
continue;
}
ALfloat coeffs[MAX_AMBI_CHANNELS];
CalcAngleCoeffs(
(Device->mRenderMode==StereoPair) ? ScaleAzimuthFront(chans[c].angle, 3.0f)
: chans[c].angle,
chans[c].elevation, Spread, coeffs
);
ComputePanGains(&Device->Dry, coeffs, DryGain,
voice->mChans[c].mDryParams.Gains.Target);
for(ALsizei i{0};i < NumSends;i++)
{
if(const ALeffectslot *Slot{SendSlots[i]})
ComputePanGains(&Slot->Wet, coeffs, WetGain[i],
voice->mChans[c].mWetParams[i].Gains.Target);
}
}
}
}
{
const ALfloat hfScale{props->Direct.HFReference / Frequency};
const ALfloat lfScale{props->Direct.LFReference / Frequency};
const ALfloat gainHF{maxf(DryGainHF, 0.001f)}; /* Limit -60dB */
const ALfloat gainLF{maxf(DryGainLF, 0.001f)};
voice->mDirect.FilterType = AF_None;
if(gainHF != 1.0f) voice->mDirect.FilterType |= AF_LowPass;
if(gainLF != 1.0f) voice->mDirect.FilterType |= AF_HighPass;
auto &lowpass = voice->mChans[0].mDryParams.LowPass;
auto &highpass = voice->mChans[0].mDryParams.HighPass;
lowpass.setParams(BiquadType::HighShelf, gainHF, hfScale,
lowpass.rcpQFromSlope(gainHF, 1.0f));
highpass.setParams(BiquadType::LowShelf, gainLF, lfScale,
highpass.rcpQFromSlope(gainLF, 1.0f));
for(ALsizei c{1};c < num_channels;c++)
{
voice->mChans[c].mDryParams.LowPass.copyParamsFrom(lowpass);
voice->mChans[c].mDryParams.HighPass.copyParamsFrom(highpass);
}
}
for(ALsizei i{0};i < NumSends;i++)
{
const ALfloat hfScale{props->Send[i].HFReference / Frequency};
const ALfloat lfScale{props->Send[i].LFReference / Frequency};
const ALfloat gainHF{maxf(WetGainHF[i], 0.001f)};
const ALfloat gainLF{maxf(WetGainLF[i], 0.001f)};
voice->mSend[i].FilterType = AF_None;
if(gainHF != 1.0f) voice->mSend[i].FilterType |= AF_LowPass;
if(gainLF != 1.0f) voice->mSend[i].FilterType |= AF_HighPass;
auto &lowpass = voice->mChans[0].mWetParams[i].LowPass;
auto &highpass = voice->mChans[0].mWetParams[i].HighPass;
lowpass.setParams(BiquadType::HighShelf, gainHF, hfScale,
lowpass.rcpQFromSlope(gainHF, 1.0f));
highpass.setParams(BiquadType::LowShelf, gainLF, lfScale,
highpass.rcpQFromSlope(gainLF, 1.0f));
for(ALsizei c{1};c < num_channels;c++)
{
voice->mChans[c].mWetParams[i].LowPass.copyParamsFrom(lowpass);
voice->mChans[c].mWetParams[i].HighPass.copyParamsFrom(highpass);
}
}
}
void CalcNonAttnSourceParams(ALvoice *voice, const ALvoicePropsBase *props, const ALCcontext *ALContext)
{
const ALCdevice *Device{ALContext->Device};
ALeffectslot *SendSlots[MAX_SENDS];
voice->mDirect.Buffer = Device->Dry.Buffer;
for(ALsizei i{0};i < Device->NumAuxSends;i++)
{
SendSlots[i] = props->Send[i].Slot;
if(!SendSlots[i] && i == 0)
SendSlots[i] = ALContext->DefaultSlot.get();
if(!SendSlots[i] || SendSlots[i]->Params.EffectType == AL_EFFECT_NULL)
{
SendSlots[i] = nullptr;
voice->mSend[i].Buffer = {};
}
else
voice->mSend[i].Buffer = SendSlots[i]->Wet.Buffer;
}
/* Calculate the stepping value */
const auto Pitch = static_cast<ALfloat>(voice->mFrequency) /
static_cast<ALfloat>(Device->Frequency) * props->Pitch;
if(Pitch > static_cast<ALfloat>(MAX_PITCH))
voice->mStep = MAX_PITCH<<FRACTIONBITS;
else
voice->mStep = maxi(fastf2i(Pitch * FRACTIONONE), 1);
if(props->mResampler == BSinc24Resampler)
BsincPrepare(voice->mStep, &voice->mResampleState.bsinc, &bsinc24);
else if(props->mResampler == BSinc12Resampler)
BsincPrepare(voice->mStep, &voice->mResampleState.bsinc, &bsinc12);
voice->mResampler = SelectResampler(props->mResampler);
/* Calculate gains */
const ALlistener &Listener = ALContext->Listener;
ALfloat DryGain{clampf(props->Gain, props->MinGain, props->MaxGain)};
DryGain *= props->Direct.Gain * Listener.Params.Gain;
DryGain = minf(DryGain, GAIN_MIX_MAX);
ALfloat DryGainHF{props->Direct.GainHF};
ALfloat DryGainLF{props->Direct.GainLF};
ALfloat WetGain[MAX_SENDS], WetGainHF[MAX_SENDS], WetGainLF[MAX_SENDS];
for(ALsizei i{0};i < Device->NumAuxSends;i++)
{
WetGain[i] = clampf(props->Gain, props->MinGain, props->MaxGain);
WetGain[i] *= props->Send[i].Gain * Listener.Params.Gain;
WetGain[i] = minf(WetGain[i], GAIN_MIX_MAX);
WetGainHF[i] = props->Send[i].GainHF;
WetGainLF[i] = props->Send[i].GainLF;
}
CalcPanningAndFilters(voice, 0.0f, 0.0f, -1.0f, 0.0f, 0.0f, DryGain, DryGainHF, DryGainLF,
WetGain, WetGainLF, WetGainHF, SendSlots, props, Listener, Device);
}
void CalcAttnSourceParams(ALvoice *voice, const ALvoicePropsBase *props, const ALCcontext *ALContext)
{
const ALCdevice *Device{ALContext->Device};
const ALsizei NumSends{Device->NumAuxSends};
const ALlistener &Listener = ALContext->Listener;
/* Set mixing buffers and get send parameters. */
voice->mDirect.Buffer = Device->Dry.Buffer;
ALeffectslot *SendSlots[MAX_SENDS];
ALfloat RoomRolloff[MAX_SENDS];
ALfloat DecayDistance[MAX_SENDS];
ALfloat DecayLFDistance[MAX_SENDS];
ALfloat DecayHFDistance[MAX_SENDS];
for(ALsizei i{0};i < NumSends;i++)
{
SendSlots[i] = props->Send[i].Slot;
if(!SendSlots[i] && i == 0)
SendSlots[i] = ALContext->DefaultSlot.get();
if(!SendSlots[i] || SendSlots[i]->Params.EffectType == AL_EFFECT_NULL)
{
SendSlots[i] = nullptr;
RoomRolloff[i] = 0.0f;
DecayDistance[i] = 0.0f;
DecayLFDistance[i] = 0.0f;
DecayHFDistance[i] = 0.0f;
}
else if(SendSlots[i]->Params.AuxSendAuto)
{
RoomRolloff[i] = SendSlots[i]->Params.RoomRolloff + props->RoomRolloffFactor;
/* Calculate the distances to where this effect's decay reaches
* -60dB.
*/
DecayDistance[i] = SendSlots[i]->Params.DecayTime *
Listener.Params.ReverbSpeedOfSound;
DecayLFDistance[i] = DecayDistance[i] * SendSlots[i]->Params.DecayLFRatio;
DecayHFDistance[i] = DecayDistance[i] * SendSlots[i]->Params.DecayHFRatio;
if(SendSlots[i]->Params.DecayHFLimit)
{
ALfloat airAbsorption{SendSlots[i]->Params.AirAbsorptionGainHF};
if(airAbsorption < 1.0f)
{
/* Calculate the distance to where this effect's air
* absorption reaches -60dB, and limit the effect's HF
* decay distance (so it doesn't take any longer to decay
* than the air would allow).
*/
ALfloat absorb_dist{std::log10(REVERB_DECAY_GAIN) / std::log10(airAbsorption)};
DecayHFDistance[i] = minf(absorb_dist, DecayHFDistance[i]);
}
}
}
else
{
/* If the slot's auxiliary send auto is off, the data sent to the
* effect slot is the same as the dry path, sans filter effects */
RoomRolloff[i] = props->RolloffFactor;
DecayDistance[i] = 0.0f;
DecayLFDistance[i] = 0.0f;
DecayHFDistance[i] = 0.0f;
}
if(!SendSlots[i])
voice->mSend[i].Buffer = {};
else
voice->mSend[i].Buffer = SendSlots[i]->Wet.Buffer;
}
/* Transform source to listener space (convert to head relative) */
alu::Vector Position{props->Position[0], props->Position[1], props->Position[2], 1.0f};
alu::Vector Velocity{props->Velocity[0], props->Velocity[1], props->Velocity[2], 0.0f};
alu::Vector Direction{props->Direction[0], props->Direction[1], props->Direction[2], 0.0f};
if(props->HeadRelative == AL_FALSE)
{
/* Transform source vectors */
Position = Listener.Params.Matrix * Position;
Velocity = Listener.Params.Matrix * Velocity;
Direction = Listener.Params.Matrix * Direction;
}
else
{
/* Offset the source velocity to be relative of the listener velocity */
Velocity += Listener.Params.Velocity;
}
const bool directional{Direction.normalize() > 0.0f};
alu::Vector ToSource{Position[0], Position[1], Position[2], 0.0f};
const ALfloat Distance{ToSource.normalize()};
/* Initial source gain */
ALfloat DryGain{props->Gain};
ALfloat DryGainHF{1.0f};
ALfloat DryGainLF{1.0f};
ALfloat WetGain[MAX_SENDS], WetGainHF[MAX_SENDS], WetGainLF[MAX_SENDS];
for(ALsizei i{0};i < NumSends;i++)
{
WetGain[i] = props->Gain;
WetGainHF[i] = 1.0f;
WetGainLF[i] = 1.0f;
}
/* Calculate distance attenuation */
ALfloat ClampedDist{Distance};
switch(Listener.Params.SourceDistanceModel ?
props->mDistanceModel : Listener.Params.mDistanceModel)
{
case DistanceModel::InverseClamped:
ClampedDist = clampf(ClampedDist, props->RefDistance, props->MaxDistance);
if(props->MaxDistance < props->RefDistance) break;
/*fall-through*/
case DistanceModel::Inverse:
if(!(props->RefDistance > 0.0f))
ClampedDist = props->RefDistance;
else
{
ALfloat dist = lerp(props->RefDistance, ClampedDist, props->RolloffFactor);
if(dist > 0.0f) DryGain *= props->RefDistance / dist;
for(ALsizei i{0};i < NumSends;i++)
{
dist = lerp(props->RefDistance, ClampedDist, RoomRolloff[i]);
if(dist > 0.0f) WetGain[i] *= props->RefDistance / dist;
}
}
break;
case DistanceModel::LinearClamped:
ClampedDist = clampf(ClampedDist, props->RefDistance, props->MaxDistance);
if(props->MaxDistance < props->RefDistance) break;
/*fall-through*/
case DistanceModel::Linear:
if(!(props->MaxDistance != props->RefDistance))
ClampedDist = props->RefDistance;
else
{
ALfloat attn = props->RolloffFactor * (ClampedDist-props->RefDistance) /
(props->MaxDistance-props->RefDistance);
DryGain *= maxf(1.0f - attn, 0.0f);
for(ALsizei i{0};i < NumSends;i++)
{
attn = RoomRolloff[i] * (ClampedDist-props->RefDistance) /
(props->MaxDistance-props->RefDistance);
WetGain[i] *= maxf(1.0f - attn, 0.0f);
}
}
break;
case DistanceModel::ExponentClamped:
ClampedDist = clampf(ClampedDist, props->RefDistance, props->MaxDistance);
if(props->MaxDistance < props->RefDistance) break;
/*fall-through*/
case DistanceModel::Exponent:
if(!(ClampedDist > 0.0f && props->RefDistance > 0.0f))
ClampedDist = props->RefDistance;
else
{
DryGain *= std::pow(ClampedDist/props->RefDistance, -props->RolloffFactor);
for(ALsizei i{0};i < NumSends;i++)
WetGain[i] *= std::pow(ClampedDist/props->RefDistance, -RoomRolloff[i]);
}
break;
case DistanceModel::Disable:
ClampedDist = props->RefDistance;
break;
}
/* Calculate directional soundcones */
if(directional && props->InnerAngle < 360.0f)
{
const ALfloat Angle{Rad2Deg(std::acos(-aluDotproduct(Direction, ToSource)) *
ConeScale * 2.0f)};
ALfloat ConeVolume, ConeHF;
if(!(Angle > props->InnerAngle))
{
ConeVolume = 1.0f;
ConeHF = 1.0f;
}
else if(Angle < props->OuterAngle)
{
ALfloat scale = ( Angle-props->InnerAngle) /
(props->OuterAngle-props->InnerAngle);
ConeVolume = lerp(1.0f, props->OuterGain, scale);
ConeHF = lerp(1.0f, props->OuterGainHF, scale);
}
else
{
ConeVolume = props->OuterGain;
ConeHF = props->OuterGainHF;
}
DryGain *= ConeVolume;
if(props->DryGainHFAuto)
DryGainHF *= ConeHF;
if(props->WetGainAuto)
std::transform(std::begin(WetGain), std::begin(WetGain)+NumSends, std::begin(WetGain),
[ConeVolume](ALfloat gain) noexcept -> ALfloat { return gain * ConeVolume; }
);
if(props->WetGainHFAuto)
std::transform(std::begin(WetGainHF), std::begin(WetGainHF)+NumSends,
std::begin(WetGainHF),
[ConeHF](ALfloat gain) noexcept -> ALfloat { return gain * ConeHF; }
);
}
/* Apply gain and frequency filters */
DryGain = clampf(DryGain, props->MinGain, props->MaxGain);
DryGain = minf(DryGain*props->Direct.Gain*Listener.Params.Gain, GAIN_MIX_MAX);
DryGainHF *= props->Direct.GainHF;
DryGainLF *= props->Direct.GainLF;
for(ALsizei i{0};i < NumSends;i++)
{
WetGain[i] = clampf(WetGain[i], props->MinGain, props->MaxGain);
WetGain[i] = minf(WetGain[i]*props->Send[i].Gain*Listener.Params.Gain, GAIN_MIX_MAX);
WetGainHF[i] *= props->Send[i].GainHF;
WetGainLF[i] *= props->Send[i].GainLF;
}
/* Distance-based air absorption and initial send decay. */
if(ClampedDist > props->RefDistance && props->RolloffFactor > 0.0f)
{
ALfloat meters_base{(ClampedDist-props->RefDistance) * props->RolloffFactor *
Listener.Params.MetersPerUnit};
if(props->AirAbsorptionFactor > 0.0f)
{
ALfloat hfattn{std::pow(AIRABSORBGAINHF, meters_base * props->AirAbsorptionFactor)};
DryGainHF *= hfattn;
std::transform(std::begin(WetGainHF), std::begin(WetGainHF)+NumSends,
std::begin(WetGainHF),
[hfattn](ALfloat gain) noexcept -> ALfloat { return gain * hfattn; }
);
}
if(props->WetGainAuto)
{
/* Apply a decay-time transformation to the wet path, based on the
* source distance in meters. The initial decay of the reverb
* effect is calculated and applied to the wet path.
*/
for(ALsizei i{0};i < NumSends;i++)
{
if(!(DecayDistance[i] > 0.0f))
continue;
const ALfloat gain{std::pow(REVERB_DECAY_GAIN, meters_base/DecayDistance[i])};
WetGain[i] *= gain;
/* Yes, the wet path's air absorption is applied with
* WetGainAuto on, rather than WetGainHFAuto.
*/
if(gain > 0.0f)
{
ALfloat gainhf{std::pow(REVERB_DECAY_GAIN, meters_base/DecayHFDistance[i])};
WetGainHF[i] *= minf(gainhf / gain, 1.0f);
ALfloat gainlf{std::pow(REVERB_DECAY_GAIN, meters_base/DecayLFDistance[i])};
WetGainLF[i] *= minf(gainlf / gain, 1.0f);
}
}
}
}
/* Initial source pitch */
ALfloat Pitch{props->Pitch};
/* Calculate velocity-based doppler effect */
ALfloat DopplerFactor{props->DopplerFactor * Listener.Params.DopplerFactor};
if(DopplerFactor > 0.0f)
{
const alu::Vector &lvelocity = Listener.Params.Velocity;
ALfloat vss{aluDotproduct(Velocity, ToSource) * -DopplerFactor};
ALfloat vls{aluDotproduct(lvelocity, ToSource) * -DopplerFactor};
const ALfloat SpeedOfSound{Listener.Params.SpeedOfSound};
if(!(vls < SpeedOfSound))
{
/* Listener moving away from the source at the speed of sound.
* Sound waves can't catch it.
*/
Pitch = 0.0f;
}
else if(!(vss < SpeedOfSound))
{
/* Source moving toward the listener at the speed of sound. Sound
* waves bunch up to extreme frequencies.
*/
Pitch = std::numeric_limits<float>::infinity();
}
else
{
/* Source and listener movement is nominal. Calculate the proper
* doppler shift.
*/
Pitch *= (SpeedOfSound-vls) / (SpeedOfSound-vss);
}
}
/* Adjust pitch based on the buffer and output frequencies, and calculate
* fixed-point stepping value.
*/
Pitch *= static_cast<ALfloat>(voice->mFrequency)/static_cast<ALfloat>(Device->Frequency);
if(Pitch > static_cast<ALfloat>(MAX_PITCH))
voice->mStep = MAX_PITCH<<FRACTIONBITS;
else
voice->mStep = maxi(fastf2i(Pitch * FRACTIONONE), 1);
if(props->mResampler == BSinc24Resampler)
BsincPrepare(voice->mStep, &voice->mResampleState.bsinc, &bsinc24);
else if(props->mResampler == BSinc12Resampler)
BsincPrepare(voice->mStep, &voice->mResampleState.bsinc, &bsinc12);
voice->mResampler = SelectResampler(props->mResampler);
ALfloat spread{0.0f};
if(props->Radius > Distance)
spread = al::MathDefs<float>::Tau() - Distance/props->Radius*al::MathDefs<float>::Pi();
else if(Distance > 0.0f)
spread = std::asin(props->Radius/Distance) * 2.0f;
CalcPanningAndFilters(voice, ToSource[0], ToSource[1], ToSource[2]*ZScale,
Distance*Listener.Params.MetersPerUnit, spread, DryGain, DryGainHF, DryGainLF, WetGain,
WetGainLF, WetGainHF, SendSlots, props, Listener, Device);
}
void CalcSourceParams(ALvoice *voice, ALCcontext *context, bool force)
{
ALvoiceProps *props{voice->mUpdate.exchange(nullptr, std::memory_order_acq_rel)};
if(!props && !force) return;
if(props)
{
voice->mProps = *props;
AtomicReplaceHead(context->FreeVoiceProps, props);
}
if((voice->mProps.mSpatializeMode == SpatializeAuto && voice->mFmtChannels == FmtMono) ||
voice->mProps.mSpatializeMode == SpatializeOn)
CalcAttnSourceParams(voice, &voice->mProps, context);
else
CalcNonAttnSourceParams(voice, &voice->mProps, context);
}
void ProcessParamUpdates(ALCcontext *ctx, const ALeffectslotArray *slots)
{
IncrementRef(&ctx->UpdateCount);
if(LIKELY(!ctx->HoldUpdates.load(std::memory_order_acquire)))
{
bool cforce{CalcContextParams(ctx)};
bool force{CalcListenerParams(ctx) || cforce};
force = std::accumulate(slots->begin(), slots->end(), force,
[ctx,cforce](bool force, ALeffectslot *slot) -> bool
{ return CalcEffectSlotParams(slot, ctx, cforce) | force; }
);
std::for_each(ctx->Voices->begin(),
ctx->Voices->begin() + ctx->VoiceCount.load(std::memory_order_acquire),
[ctx,force](ALvoice &voice) -> void
{
ALuint sid{voice.mSourceID.load(std::memory_order_acquire)};
if(sid) CalcSourceParams(&voice, ctx, force);
}
);
}
IncrementRef(&ctx->UpdateCount);
}
void ProcessContext(ALCcontext *ctx, const ALsizei SamplesToDo)
{
ASSUME(SamplesToDo > 0);
const ALeffectslotArray *auxslots{ctx->ActiveAuxSlots.load(std::memory_order_acquire)};
/* Process pending propery updates for objects on the context. */
ProcessParamUpdates(ctx, auxslots);
/* Clear auxiliary effect slot mixing buffers. */
std::for_each(auxslots->begin(), auxslots->end(),
[SamplesToDo](ALeffectslot *slot) -> void
{
for(auto &buffer : slot->MixBuffer)
std::fill_n(buffer.begin(), SamplesToDo, 0.0f);
}
);
/* Process voices that have a playing source. */
std::for_each(ctx->Voices->begin(),
ctx->Voices->begin() + ctx->VoiceCount.load(std::memory_order_acquire),
[SamplesToDo,ctx](ALvoice &voice) -> void
{
const ALvoice::State vstate{voice.mPlayState.load(std::memory_order_acquire)};
if(vstate == ALvoice::Stopped) return;
const ALuint sid{voice.mSourceID.load(std::memory_order_relaxed)};
if(voice.mStep < 1) return;
MixVoice(&voice, vstate, sid, ctx, SamplesToDo);
}
);
/* Process effects. */
if(auxslots->size() < 1) return;
auto slots = auxslots->data();
auto slots_end = slots + auxslots->size();
/* First sort the slots into scratch storage, so that effects come before
* their effect target (or their targets' target).
*/
auto sorted_slots = const_cast<ALeffectslot**>(slots_end);
auto sorted_slots_end = sorted_slots;
auto in_chain = [](const ALeffectslot *slot1, const ALeffectslot *slot2) noexcept -> bool
{
while((slot1=slot1->Params.Target) != nullptr) {
if(slot1 == slot2) return true;
}
return false;
};
*sorted_slots_end = *slots;
++sorted_slots_end;
while(++slots != slots_end)
{
/* If this effect slot targets an effect slot already in the list (i.e.
* slots outputs to something in sorted_slots), directly or indirectly,
* insert it prior to that element.
*/
auto checker = sorted_slots;
do {
if(in_chain(*slots, *checker)) break;
} while(++checker != sorted_slots_end);
checker = std::move_backward(checker, sorted_slots_end, sorted_slots_end+1);
*--checker = *slots;
++sorted_slots_end;
}
std::for_each(sorted_slots, sorted_slots_end,
[SamplesToDo](const ALeffectslot *slot) -> void
{
EffectState *state{slot->Params.mEffectState};
state->process(SamplesToDo, slot->Wet.Buffer.data(),
static_cast<ALsizei>(slot->Wet.Buffer.size()), state->mOutTarget);
}
);
}
void ApplyStablizer(FrontStablizer *Stablizer, const al::span<FloatBufferLine> Buffer,
const ALuint lidx, const ALuint ridx, const ALuint cidx, const ALsizei SamplesToDo)
{
ASSUME(SamplesToDo > 0);
/* Apply a delay to all channels, except the front-left and front-right, so
* they maintain correct timing.
*/
const size_t NumChannels{Buffer.size()};
for(size_t i{0u};i < NumChannels;i++)
{
if(i == lidx || i == ridx)
continue;
auto &DelayBuf = Stablizer->DelayBuf[i];
auto buffer_end = Buffer[i].begin() + SamplesToDo;
if(LIKELY(SamplesToDo >= ALsizei{FrontStablizer::DelayLength}))
{
auto delay_end = std::rotate(Buffer[i].begin(),
buffer_end - FrontStablizer::DelayLength, buffer_end);
std::swap_ranges(Buffer[i].begin(), delay_end, std::begin(DelayBuf));
}
else
{
auto delay_start = std::swap_ranges(Buffer[i].begin(), buffer_end,
std::begin(DelayBuf));
std::rotate(std::begin(DelayBuf), delay_start, std::end(DelayBuf));
}
}
ALfloat (&lsplit)[2][BUFFERSIZE] = Stablizer->LSplit;
ALfloat (&rsplit)[2][BUFFERSIZE] = Stablizer->RSplit;
auto &tmpbuf = Stablizer->TempBuf;
/* This applies the band-splitter, preserving phase at the cost of some
* delay. The shorter the delay, the more error seeps into the result.
*/
auto apply_splitter = [&tmpbuf,SamplesToDo](const FloatBufferLine &Buffer,
ALfloat (&DelayBuf)[FrontStablizer::DelayLength], BandSplitter &Filter,
ALfloat (&splitbuf)[2][BUFFERSIZE]) -> void
{
/* Combine the delayed samples and the input samples into the temp
* buffer, in reverse. Then copy the final samples back into the delay
* buffer for next time. Note that the delay buffer's samples are
* stored backwards here.
*/
auto tmpbuf_end = std::begin(tmpbuf) + SamplesToDo;
std::copy_n(std::begin(DelayBuf), FrontStablizer::DelayLength, tmpbuf_end);
std::reverse_copy(Buffer.begin(), Buffer.begin()+SamplesToDo, std::begin(tmpbuf));
std::copy_n(std::begin(tmpbuf), FrontStablizer::DelayLength, std::begin(DelayBuf));
/* Apply an all-pass on the reversed signal, then reverse the samples
* to get the forward signal with a reversed phase shift.
*/
Filter.applyAllpass(tmpbuf, SamplesToDo+FrontStablizer::DelayLength);
std::reverse(std::begin(tmpbuf), tmpbuf_end+FrontStablizer::DelayLength);
/* Now apply the band-splitter, combining its phase shift with the
* reversed phase shift, restoring the original phase on the split
* signal.
*/
Filter.process(splitbuf[1], splitbuf[0], tmpbuf, SamplesToDo);
};
apply_splitter(Buffer[lidx], Stablizer->DelayBuf[lidx], Stablizer->LFilter, lsplit);
apply_splitter(Buffer[ridx], Stablizer->DelayBuf[ridx], Stablizer->RFilter, rsplit);
for(ALsizei i{0};i < SamplesToDo;i++)
{
ALfloat lfsum{lsplit[0][i] + rsplit[0][i]};
ALfloat hfsum{lsplit[1][i] + rsplit[1][i]};
ALfloat s{lsplit[0][i] + lsplit[1][i] - rsplit[0][i] - rsplit[1][i]};
/* This pans the separate low- and high-frequency sums between being on
* the center channel and the left/right channels. The low-frequency
* sum is 1/3rd toward center (2/3rds on left/right) and the high-
* frequency sum is 1/4th toward center (3/4ths on left/right). These
* values can be tweaked.
*/
ALfloat m{lfsum*std::cos(1.0f/3.0f * (al::MathDefs<float>::Pi()*0.5f)) +
hfsum*std::cos(1.0f/4.0f * (al::MathDefs<float>::Pi()*0.5f))};
ALfloat c{lfsum*std::sin(1.0f/3.0f * (al::MathDefs<float>::Pi()*0.5f)) +
hfsum*std::sin(1.0f/4.0f * (al::MathDefs<float>::Pi()*0.5f))};
/* The generated center channel signal adds to the existing signal,
* while the modified left and right channels replace.
*/
Buffer[lidx][i] = (m + s) * 0.5f;
Buffer[ridx][i] = (m - s) * 0.5f;
Buffer[cidx][i] += c * 0.5f;
}
}
void ApplyDistanceComp(const al::span<FloatBufferLine> Samples, const ALsizei SamplesToDo,
const DistanceComp::DistData *distcomp)
{
ASSUME(SamplesToDo > 0);
for(auto &chanbuffer : Samples)
{
const ALfloat gain{distcomp->Gain};
const ALsizei base{distcomp->Length};
ALfloat *distbuf{al::assume_aligned<16>(distcomp->Buffer)};
++distcomp;
if(base < 1)
continue;
ALfloat *inout{al::assume_aligned<16>(chanbuffer.data())};
auto inout_end = inout + SamplesToDo;
if(LIKELY(SamplesToDo >= base))
{
auto delay_end = std::rotate(inout, inout_end - base, inout_end);
std::swap_ranges(inout, delay_end, distbuf);
}
else
{
auto delay_start = std::swap_ranges(inout, inout_end, distbuf);
std::rotate(distbuf, delay_start, distbuf + base);
}
std::transform(inout, inout_end, inout, std::bind(std::multiplies<float>{}, _1, gain));
}
}
void ApplyDither(const al::span<FloatBufferLine> Samples, ALuint *dither_seed,
const ALfloat quant_scale, const ALsizei SamplesToDo)
{
/* Dithering. Generate whitenoise (uniform distribution of random values
* between -1 and +1) and add it to the sample values, after scaling up to
* the desired quantization depth amd before rounding.
*/
const ALfloat invscale{1.0f / quant_scale};
ALuint seed{*dither_seed};
auto dither_channel = [&seed,invscale,quant_scale,SamplesToDo](FloatBufferLine &input) -> void
{
ASSUME(SamplesToDo > 0);
auto dither_sample = [&seed,invscale,quant_scale](const ALfloat sample) noexcept -> ALfloat
{
ALfloat val{sample * quant_scale};
ALuint rng0{dither_rng(&seed)};
ALuint rng1{dither_rng(&seed)};
val += static_cast<ALfloat>(rng0*(1.0/UINT_MAX) - rng1*(1.0/UINT_MAX));
return fast_roundf(val) * invscale;
};
std::transform(input.begin(), input.begin()+SamplesToDo, input.begin(), dither_sample);
};
std::for_each(Samples.begin(), Samples.end(), dither_channel);
*dither_seed = seed;
}
/* Base template left undefined. Should be marked =delete, but Clang 3.8.1
* chokes on that given the inline specializations.
*/
template<typename T>
inline T SampleConv(ALfloat) noexcept;
template<> inline ALfloat SampleConv(ALfloat val) noexcept
{ return val; }
template<> inline ALint SampleConv(ALfloat val) noexcept
{
/* Floats have a 23-bit mantissa, plus an implied 1 bit and a sign bit.
* This means a normalized float has at most 25 bits of signed precision.
* When scaling and clamping for a signed 32-bit integer, these following
* values are the best a float can give.
*/
return fastf2i(clampf(val*2147483648.0f, -2147483648.0f, 2147483520.0f));
}
template<> inline ALshort SampleConv(ALfloat val) noexcept
{ return fastf2i(clampf(val*32768.0f, -32768.0f, 32767.0f)); }
template<> inline ALbyte SampleConv(ALfloat val) noexcept
{ return fastf2i(clampf(val*128.0f, -128.0f, 127.0f)); }
/* Define unsigned output variations. */
template<> inline ALuint SampleConv(ALfloat val) noexcept
{ return SampleConv<ALint>(val) + 2147483648u; }
template<> inline ALushort SampleConv(ALfloat val) noexcept
{ return SampleConv<ALshort>(val) + 32768; }
template<> inline ALubyte SampleConv(ALfloat val) noexcept
{ return SampleConv<ALbyte>(val) + 128; }
template<DevFmtType T>
void Write(const al::span<const FloatBufferLine> InBuffer, ALvoid *OutBuffer, const size_t Offset,
const ALsizei SamplesToDo)
{
using SampleType = typename DevFmtTypeTraits<T>::Type;
const size_t numchans{InBuffer.size()};
ASSUME(numchans > 0);
SampleType *outbase = static_cast<SampleType*>(OutBuffer) + Offset*numchans;
auto conv_channel = [&outbase,SamplesToDo,numchans](const FloatBufferLine &inbuf) -> void
{
ASSUME(SamplesToDo > 0);
SampleType *out{outbase++};
auto conv_sample = [numchans,&out](const ALfloat s) noexcept -> void
{
*out = SampleConv<SampleType>(s);
out += numchans;
};
std::for_each(inbuf.begin(), inbuf.begin()+SamplesToDo, conv_sample);
};
std::for_each(InBuffer.cbegin(), InBuffer.cend(), conv_channel);
}
} // namespace
void aluMixData(ALCdevice *device, ALvoid *OutBuffer, ALsizei NumSamples)
{
FPUCtl mixer_mode{};
for(ALsizei SamplesDone{0};SamplesDone < NumSamples;)
{
const ALsizei SamplesToDo{mini(NumSamples-SamplesDone, BUFFERSIZE)};
/* Clear main mixing buffers. */
std::for_each(device->MixBuffer.begin(), device->MixBuffer.end(),
[SamplesToDo](std::array<ALfloat,BUFFERSIZE> &buffer) -> void
{ std::fill_n(buffer.begin(), SamplesToDo, 0.0f); }
);
/* Increment the mix count at the start (lsb should now be 1). */
IncrementRef(&device->MixCount);
/* For each context on this device, process and mix its sources and
* effects.
*/
for(ALCcontext *ctx : *device->mContexts.load(std::memory_order_acquire))
ProcessContext(ctx, SamplesToDo);
/* Increment the clock time. Every second's worth of samples is
* converted and added to clock base so that large sample counts don't
* overflow during conversion. This also guarantees a stable
* conversion.
*/
device->SamplesDone += SamplesToDo;
device->ClockBase += std::chrono::seconds{device->SamplesDone / device->Frequency};
device->SamplesDone %= device->Frequency;
/* Increment the mix count at the end (lsb should now be 0). */
IncrementRef(&device->MixCount);
/* Apply any needed post-process for finalizing the Dry mix to the
* RealOut (Ambisonic decode, UHJ encode, etc).
*/
if(LIKELY(device->PostProcess))
device->PostProcess(device, SamplesToDo);
const al::span<FloatBufferLine> RealOut{device->RealOut.Buffer};
/* Apply front image stablization for surround sound, if applicable. */
if(device->Stablizer)
{
const int lidx{GetChannelIdxByName(device->RealOut, FrontLeft)};
const int ridx{GetChannelIdxByName(device->RealOut, FrontRight)};
const int cidx{GetChannelIdxByName(device->RealOut, FrontCenter)};
assert(lidx >= 0 && ridx >= 0 && cidx >= 0);
ApplyStablizer(device->Stablizer.get(), RealOut, lidx, ridx, cidx, SamplesToDo);
}
/* Apply compression, limiting sample amplitude if needed or desired. */
if(Compressor *comp{device->Limiter.get()})
comp->process(SamplesToDo, RealOut.data());
/* Apply delays and attenuation for mismatched speaker distances. */
ApplyDistanceComp(RealOut, SamplesToDo, device->ChannelDelay.as_span().cbegin());
/* Apply dithering. The compressor should have left enough headroom for
* the dither noise to not saturate.
*/
if(device->DitherDepth > 0.0f)
ApplyDither(RealOut, &device->DitherSeed, device->DitherDepth, SamplesToDo);
if(LIKELY(OutBuffer))
{
/* Finally, interleave and convert samples, writing to the device's
* output buffer.
*/
switch(device->FmtType)
{
#define HANDLE_WRITE(T) case T: \
Write<T>(RealOut, OutBuffer, SamplesDone, SamplesToDo); break;
HANDLE_WRITE(DevFmtByte)
HANDLE_WRITE(DevFmtUByte)
HANDLE_WRITE(DevFmtShort)
HANDLE_WRITE(DevFmtUShort)
HANDLE_WRITE(DevFmtInt)
HANDLE_WRITE(DevFmtUInt)
HANDLE_WRITE(DevFmtFloat)
#undef HANDLE_WRITE
}
}
SamplesDone += SamplesToDo;
}
}
void aluHandleDisconnect(ALCdevice *device, const char *msg, ...)
{
if(!device->Connected.exchange(false, std::memory_order_acq_rel))
return;
AsyncEvent evt{EventType_Disconnected};
evt.u.user.type = AL_EVENT_TYPE_DISCONNECTED_SOFT;
evt.u.user.id = 0;
evt.u.user.param = 0;
va_list args;
va_start(args, msg);
int msglen{vsnprintf(evt.u.user.msg, sizeof(evt.u.user.msg), msg, args)};
va_end(args);
if(msglen < 0 || static_cast<size_t>(msglen) >= sizeof(evt.u.user.msg))
evt.u.user.msg[sizeof(evt.u.user.msg)-1] = 0;
for(ALCcontext *ctx : *device->mContexts.load())
{
const ALbitfieldSOFT enabledevt{ctx->EnabledEvts.load(std::memory_order_acquire)};
if((enabledevt&EventType_Disconnected))
{
RingBuffer *ring{ctx->AsyncEvents.get()};
auto evt_data = ring->getWriteVector().first;
if(evt_data.len > 0)
{
new (evt_data.buf) AsyncEvent{evt};
ring->writeAdvance(1);
ctx->EventSem.post();
}
}
auto stop_voice = [](ALvoice &voice) -> void
{
voice.mCurrentBuffer.store(nullptr, std::memory_order_relaxed);
voice.mLoopBuffer.store(nullptr, std::memory_order_relaxed);
voice.mSourceID.store(0u, std::memory_order_relaxed);
voice.mPlayState.store(ALvoice::Stopped, std::memory_order_release);
};
std::for_each(ctx->Voices->begin(),
ctx->Voices->begin() + ctx->VoiceCount.load(std::memory_order_acquire),
stop_voice);
}
}