/**
* OpenAL cross platform audio library
* Copyright (C) 1999-2007 by authors.
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
* Or go to http://www.gnu.org/copyleft/lgpl.html
*/
#include "config.h"
#include <math.h>
#include <stdlib.h>
#include <string.h>
#include <ctype.h>
#include <assert.h>
#include <algorithm>
#include "alMain.h"
#include "alcontext.h"
#include "alSource.h"
#include "alBuffer.h"
#include "alListener.h"
#include "alAuxEffectSlot.h"
#include "alu.h"
#include "bs2b.h"
#include "hrtf.h"
#include "mastering.h"
#include "uhjfilter.h"
#include "bformatdec.h"
#include "ringbuffer.h"
#include "filters/splitter.h"
#include "mixer/defs.h"
#include "fpu_modes.h"
#include "cpu_caps.h"
#include "bsinc_inc.h"
/* Cone scalar */
ALfloat ConeScale = 1.0f;
/* Localized Z scalar for mono sources */
ALfloat ZScale = 1.0f;
/* Force default speed of sound for distance-related reverb decay. */
ALboolean OverrideReverbSpeedOfSound = AL_FALSE;
namespace {
void ClearArray(ALfloat f[MAX_OUTPUT_CHANNELS])
{
size_t i;
for(i = 0;i < MAX_OUTPUT_CHANNELS;i++)
f[i] = 0.0f;
}
struct ChanMap {
enum Channel channel;
ALfloat angle;
ALfloat elevation;
};
HrtfDirectMixerFunc MixDirectHrtf = MixDirectHrtf_C;
inline HrtfDirectMixerFunc SelectHrtfMixer(void)
{
#ifdef HAVE_NEON
if((CPUCapFlags&CPU_CAP_NEON))
return MixDirectHrtf_Neon;
#endif
#ifdef HAVE_SSE
if((CPUCapFlags&CPU_CAP_SSE))
return MixDirectHrtf_SSE;
#endif
return MixDirectHrtf_C;
}
} // namespace
void aluInit(void)
{
MixDirectHrtf = SelectHrtfMixer();
}
void DeinitVoice(ALvoice *voice)
{
al_free(voice->Update.exchange(nullptr));
}
namespace {
void ProcessHrtf(ALCdevice *device, ALsizei SamplesToDo)
{
if(device->AmbiUp)
ambiup_process(device->AmbiUp.get(),
device->Dry.Buffer, device->Dry.NumChannels, device->FOAOut.Buffer,
SamplesToDo
);
int lidx{GetChannelIdxByName(&device->RealOut, FrontLeft)};
int ridx{GetChannelIdxByName(&device->RealOut, FrontRight)};
assert(lidx != -1 && ridx != -1);
DirectHrtfState *state{device->mHrtfState.get()};
for(ALsizei c{0};c < device->Dry.NumChannels;c++)
{
MixDirectHrtf(device->RealOut.Buffer[lidx], device->RealOut.Buffer[ridx],
device->Dry.Buffer[c], state->Offset, state->IrSize,
state->Chan[c].Coeffs, state->Chan[c].Values, SamplesToDo
);
}
state->Offset += SamplesToDo;
}
void ProcessAmbiDec(ALCdevice *device, ALsizei SamplesToDo)
{
if(device->Dry.Buffer != device->FOAOut.Buffer)
bformatdec_upSample(device->AmbiDecoder.get(),
device->Dry.Buffer, device->FOAOut.Buffer, device->FOAOut.NumChannels,
SamplesToDo
);
bformatdec_process(device->AmbiDecoder.get(),
device->RealOut.Buffer, device->RealOut.NumChannels, device->Dry.Buffer,
SamplesToDo
);
}
void ProcessAmbiUp(ALCdevice *device, ALsizei SamplesToDo)
{
ambiup_process(device->AmbiUp.get(),
device->RealOut.Buffer, device->RealOut.NumChannels, device->FOAOut.Buffer,
SamplesToDo
);
}
void ProcessUhj(ALCdevice *device, ALsizei SamplesToDo)
{
int lidx = GetChannelIdxByName(&device->RealOut, FrontLeft);
int ridx = GetChannelIdxByName(&device->RealOut, FrontRight);
assert(lidx != -1 && ridx != -1);
/* Encode to stereo-compatible 2-channel UHJ output. */
EncodeUhj2(device->Uhj_Encoder.get(),
device->RealOut.Buffer[lidx], device->RealOut.Buffer[ridx],
device->Dry.Buffer, SamplesToDo
);
}
void ProcessBs2b(ALCdevice *device, ALsizei SamplesToDo)
{
int lidx = GetChannelIdxByName(&device->RealOut, FrontLeft);
int ridx = GetChannelIdxByName(&device->RealOut, FrontRight);
assert(lidx != -1 && ridx != -1);
/* Apply binaural/crossfeed filter */
bs2b_cross_feed(device->Bs2b.get(), device->RealOut.Buffer[lidx],
device->RealOut.Buffer[ridx], SamplesToDo);
}
} // namespace
void aluSelectPostProcess(ALCdevice *device)
{
if(device->HrtfHandle)
device->PostProcess = ProcessHrtf;
else if(device->AmbiDecoder)
device->PostProcess = ProcessAmbiDec;
else if(device->AmbiUp)
device->PostProcess = ProcessAmbiUp;
else if(device->Uhj_Encoder)
device->PostProcess = ProcessUhj;
else if(device->Bs2b)
device->PostProcess = ProcessBs2b;
else
device->PostProcess = NULL;
}
/* 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)
{
ALfloat sf = 0.0f;
ALsizei si = BSINC_SCALE_COUNT-1;
if(increment > FRACTIONONE)
{
sf = (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 - cosf(asinf(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 void aluCrossproduct(const ALfloat *inVector1, const ALfloat *inVector2, ALfloat *outVector)
{
outVector[0] = inVector1[1]*inVector2[2] - inVector1[2]*inVector2[1];
outVector[1] = inVector1[2]*inVector2[0] - inVector1[0]*inVector2[2];
outVector[2] = inVector1[0]*inVector2[1] - inVector1[1]*inVector2[0];
}
inline ALfloat aluDotproduct(const aluVector *vec1, const aluVector *vec2)
{
return vec1->v[0]*vec2->v[0] + vec1->v[1]*vec2->v[1] + vec1->v[2]*vec2->v[2];
}
ALfloat aluNormalize(ALfloat *vec)
{
ALfloat length = sqrtf(vec[0]*vec[0] + vec[1]*vec[1] + vec[2]*vec[2]);
if(length > FLT_EPSILON)
{
ALfloat inv_length = 1.0f/length;
vec[0] *= inv_length;
vec[1] *= inv_length;
vec[2] *= inv_length;
return length;
}
vec[0] = vec[1] = vec[2] = 0.0f;
return 0.0f;
}
void aluMatrixfFloat3(ALfloat *vec, ALfloat w, const aluMatrixf *mtx)
{
ALfloat v[4] = { vec[0], vec[1], vec[2], w };
vec[0] = v[0]*mtx->m[0][0] + v[1]*mtx->m[1][0] + v[2]*mtx->m[2][0] + v[3]*mtx->m[3][0];
vec[1] = v[0]*mtx->m[0][1] + v[1]*mtx->m[1][1] + v[2]*mtx->m[2][1] + v[3]*mtx->m[3][1];
vec[2] = v[0]*mtx->m[0][2] + v[1]*mtx->m[1][2] + v[2]*mtx->m[2][2] + v[3]*mtx->m[3][2];
}
aluVector aluMatrixfVector(const aluMatrixf *mtx, const aluVector *vec)
{
aluVector v;
v.v[0] = vec->v[0]*mtx->m[0][0] + vec->v[1]*mtx->m[1][0] + vec->v[2]*mtx->m[2][0] + vec->v[3]*mtx->m[3][0];
v.v[1] = vec->v[0]*mtx->m[0][1] + vec->v[1]*mtx->m[1][1] + vec->v[2]*mtx->m[2][1] + vec->v[3]*mtx->m[3][1];
v.v[2] = vec->v[0]*mtx->m[0][2] + vec->v[1]*mtx->m[1][2] + vec->v[2]*mtx->m[2][2] + vec->v[3]*mtx->m[3][2];
v.v[3] = vec->v[0]*mtx->m[0][3] + vec->v[1]*mtx->m[1][3] + vec->v[2]*mtx->m[2][3] + vec->v[3]*mtx->m[3][3];
return v;
}
void SendSourceStoppedEvent(ALCcontext *context, ALuint id)
{
AsyncEvent evt = ASYNC_EVENT(EventType_SourceStateChange);
ALbitfieldSOFT enabledevt;
size_t strpos;
ALuint scale;
enabledevt = context->EnabledEvts.load(std::memory_order_acquire);
if(!(enabledevt&EventType_SourceStateChange)) return;
evt.u.user.type = AL_EVENT_TYPE_SOURCE_STATE_CHANGED_SOFT;
evt.u.user.id = id;
evt.u.user.param = AL_STOPPED;
/* Normally snprintf would be used, but this is called from the mixer and
* that function's not real-time safe, so we have to construct it manually.
*/
strcpy(evt.u.user.msg, "Source ID "); strpos = 10;
scale = 1000000000;
while(scale > 0 && scale > id)
scale /= 10;
while(scale > 0)
{
evt.u.user.msg[strpos++] = '0' + ((id/scale)%10);
scale /= 10;
}
strcpy(evt.u.user.msg+strpos, " state changed to AL_STOPPED");
if(ll_ringbuffer_write(context->AsyncEvents, &evt, 1) == 1)
alsem_post(&context->EventSem);
}
bool CalcContextParams(ALCcontext *Context)
{
ALlistener &Listener = Context->Listener;
struct ALcontextProps *props;
props = Context->Update.exchange(nullptr, std::memory_order_acq_rel);
if(!props) return false;
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;
ALfloat N[3], V[3], U[3], P[3];
struct ALlistenerProps *props;
aluVector vel;
props = Listener.Update.exchange(nullptr, std::memory_order_acq_rel);
if(!props) return false;
/* AT then UP */
N[0] = props->Forward[0];
N[1] = props->Forward[1];
N[2] = props->Forward[2];
aluNormalize(N);
V[0] = props->Up[0];
V[1] = props->Up[1];
V[2] = props->Up[2];
aluNormalize(V);
/* Build and normalize right-vector */
aluCrossproduct(N, V, U);
aluNormalize(U);
aluMatrixfSet(&Listener.Params.Matrix,
U[0], V[0], -N[0], 0.0,
U[1], V[1], -N[1], 0.0,
U[2], V[2], -N[2], 0.0,
0.0, 0.0, 0.0, 1.0
);
P[0] = props->Position[0];
P[1] = props->Position[1];
P[2] = props->Position[2];
aluMatrixfFloat3(P, 1.0, &Listener.Params.Matrix);
aluMatrixfSetRow(&Listener.Params.Matrix, 3, -P[0], -P[1], -P[2], 1.0f);
aluVectorSet(&vel, props->Velocity[0], props->Velocity[1], props->Velocity[2], 0.0f);
Listener.Params.Velocity = aluMatrixfVector(&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)
{
struct ALeffectslotProps *props;
EffectState *state;
props = slot->Update.exchange(nullptr, std::memory_order_acq_rel);
if(!props && !force) return false;
if(props)
{
slot->Params.Gain = props->Gain;
slot->Params.AuxSendAuto = props->AuxSendAuto;
slot->Params.EffectType = props->Type;
slot->Params.EffectProps = 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;
if(state == slot->Params.mEffectState)
{
/* If the effect state is the same as current, we can decrement its
* count safely to remove it from the update object (it can't reach
* 0 refs since the current params also hold a reference).
*/
DecrementRef(&state->mRef);
props->State = nullptr;
}
else
{
/* Otherwise, replace it and send off the old one with a release
* event.
*/
AsyncEvent evt = ASYNC_EVENT(EventType_ReleaseEffectState);
evt.u.mEffectState = slot->Params.mEffectState;
slot->Params.mEffectState = state;
props->State = NULL;
if(LIKELY(ll_ringbuffer_write(context->AsyncEvents, &evt, 1) != 0))
alsem_post(&context->EventSem);
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 = evt.u.mEffectState;
}
}
AtomicReplaceHead(context->FreeEffectslotProps, props);
}
else
state = slot->Params.mEffectState;
state->update(context, slot, &slot->Params.EffectProps);
return true;
}
constexpr struct 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) }
};
void CalcPanningAndFilters(ALvoice *voice, const ALfloat Azi, const ALfloat Elev,
const ALfloat Distance, const ALfloat Spread,
const ALfloat DryGain, const ALfloat DryGainHF,
const ALfloat DryGainLF, const ALfloat *WetGain,
const ALfloat *WetGainLF, const ALfloat *WetGainHF,
ALeffectslot **SendSlots, const ALbuffer *Buffer,
const struct ALvoiceProps *props, const ALlistener &Listener,
const ALCdevice *Device)
{
struct ChanMap StereoMap[2] = {
{ FrontLeft, DEG2RAD(-30.0f), DEG2RAD(0.0f) },
{ FrontRight, DEG2RAD( 30.0f), DEG2RAD(0.0f) }
};
bool DirectChannels = props->DirectChannels;
const ALsizei NumSends = Device->NumAuxSends;
const ALuint Frequency = Device->Frequency;
const struct ChanMap *chans = NULL;
ALsizei num_channels = 0;
bool isbformat = false;
ALfloat downmix_gain = 1.0f;
ALsizei c, i;
switch(Buffer->FmtChannels)
{
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;
}
for(c = 0;c < num_channels;c++)
{
memset(&voice->Direct.Params[c].Hrtf.Target, 0,
sizeof(voice->Direct.Params[c].Hrtf.Target));
ClearArray(voice->Direct.Params[c].Gains.Target);
}
for(i = 0;i < NumSends;i++)
{
for(c = 0;c < num_channels;c++)
ClearArray(voice->Send[i].Params[c].Gains.Target);
}
voice->Flags &= ~(VOICE_HAS_HRTF | VOICE_HAS_NFC);
if(isbformat)
{
/* Special handling for B-Format sources. */
if(Distance > FLT_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.
*/
ALfloat coeffs[MAX_AMBI_COEFFS];
if(Device->AvgSpeakerDist > 0.0f)
{
ALfloat mdist = Distance * Listener.Params.MetersPerUnit;
ALfloat w0 = SPEEDOFSOUNDMETRESPERSEC /
(mdist * (ALfloat)Device->Frequency);
ALfloat w1 = SPEEDOFSOUNDMETRESPERSEC /
(Device->AvgSpeakerDist * (ALfloat)Device->Frequency);
/* Clamp w0 for really close distances, to prevent excessive
* bass.
*/
w0 = minf(w0, w1*4.0f);
/* Only need to adjust the first channel of a B-Format source. */
NfcFilterAdjust(&voice->Direct.Params[0].NFCtrlFilter, w0);
for(i = 0;i < MAX_AMBI_ORDER+1;i++)
voice->Direct.ChannelsPerOrder[i] = Device->NumChannelsPerOrder[i];
voice->Flags |= VOICE_HAS_NFC;
}
/* 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((Device->Render_Mode==StereoPair) ? ScaleAzimuthFront(Azi, 1.5f) : Azi,
Elev, Spread, coeffs);
/* NOTE: W needs to be scaled by sqrt(2) due to FuMa normalization. */
ComputePanGains(&Device->Dry, coeffs, DryGain*SQRTF_2,
voice->Direct.Params[0].Gains.Target);
for(i = 0;i < NumSends;i++)
{
const ALeffectslot *Slot = SendSlots[i];
if(Slot)
ComputePanningGainsBF(Slot->ChanMap, Slot->NumChannels, coeffs,
WetGain[i]*SQRTF_2, voice->Send[i].Params[0].Gains.Target
);
}
}
else
{
/* Local B-Format sources have their XYZ channels rotated according
* to the orientation.
*/
ALfloat N[3], V[3], U[3];
aluMatrixf matrix;
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.
*/
NfcFilterAdjust(&voice->Direct.Params[0].NFCtrlFilter, 0.0f);
voice->Direct.ChannelsPerOrder[0] = 1;
voice->Direct.ChannelsPerOrder[1] = mini(voice->Direct.Channels-1, 3);
for(i = 2;i < MAX_AMBI_ORDER+1;i++)
voice->Direct.ChannelsPerOrder[i] = 0;
voice->Flags |= VOICE_HAS_NFC;
}
/* AT then UP */
N[0] = props->Orientation[0][0];
N[1] = props->Orientation[0][1];
N[2] = props->Orientation[0][2];
aluNormalize(N);
V[0] = props->Orientation[1][0];
V[1] = props->Orientation[1][1];
V[2] = props->Orientation[1][2];
aluNormalize(V);
if(!props->HeadRelative)
{
const aluMatrixf *lmatrix = &Listener.Params.Matrix;
aluMatrixfFloat3(N, 0.0f, lmatrix);
aluMatrixfFloat3(V, 0.0f, lmatrix);
}
/* Build and normalize right-vector */
aluCrossproduct(N, V, U);
aluNormalize(U);
/* 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.
*/
aluMatrixfSet(&matrix,
// ACN0 ACN1 ACN2 ACN3
SQRTF_2, 0.0f, 0.0f, 0.0f, // Ambi W
0.0f, -N[0]*SQRTF_3, N[1]*SQRTF_3, -N[2]*SQRTF_3, // Ambi X
0.0f, U[0]*SQRTF_3, -U[1]*SQRTF_3, U[2]*SQRTF_3, // Ambi Y
0.0f, -V[0]*SQRTF_3, V[1]*SQRTF_3, -V[2]*SQRTF_3 // Ambi Z
);
voice->Direct.Buffer = Device->FOAOut.Buffer;
voice->Direct.Channels = Device->FOAOut.NumChannels;
for(c = 0;c < num_channels;c++)
ComputePanGains(&Device->FOAOut, matrix.m[c], DryGain,
voice->Direct.Params[c].Gains.Target);
for(i = 0;i < NumSends;i++)
{
const ALeffectslot *Slot = SendSlots[i];
if(Slot)
{
for(c = 0;c < num_channels;c++)
ComputePanningGainsBF(Slot->ChanMap, Slot->NumChannels,
matrix.m[c], WetGain[i], voice->Send[i].Params[c].Gains.Target
);
}
}
}
}
else if(DirectChannels)
{
/* Direct source channels always play local. Skip the virtual channels
* and write inputs to the matching real outputs.
*/
voice->Direct.Buffer = Device->RealOut.Buffer;
voice->Direct.Channels = Device->RealOut.NumChannels;
for(c = 0;c < num_channels;c++)
{
int idx = GetChannelIdxByName(&Device->RealOut, chans[c].channel);
if(idx != -1) voice->Direct.Params[c].Gains.Target[idx] = DryGain;
}
/* Auxiliary sends still use normal channel panning since they mix to
* B-Format, which can't channel-match.
*/
for(c = 0;c < num_channels;c++)
{
ALfloat coeffs[MAX_AMBI_COEFFS];
CalcAngleCoeffs(chans[c].angle, chans[c].elevation, 0.0f, coeffs);
for(i = 0;i < NumSends;i++)
{
const ALeffectslot *Slot = SendSlots[i];
if(Slot)
ComputePanningGainsBF(Slot->ChanMap, Slot->NumChannels,
coeffs, WetGain[i], voice->Send[i].Params[c].Gains.Target
);
}
}
}
else if(Device->Render_Mode == HrtfRender)
{
/* Full HRTF rendering. Skip the virtual channels and render to the
* real outputs.
*/
voice->Direct.Buffer = Device->RealOut.Buffer;
voice->Direct.Channels = Device->RealOut.NumChannels;
if(Distance > FLT_EPSILON)
{
ALfloat coeffs[MAX_AMBI_COEFFS];
/* Get the HRIR coefficients and delays just once, for the given
* source direction.
*/
GetHrtfCoeffs(Device->HrtfHandle, Elev, Azi, Spread,
voice->Direct.Params[0].Hrtf.Target.Coeffs,
voice->Direct.Params[0].Hrtf.Target.Delay);
voice->Direct.Params[0].Hrtf.Target.Gain = DryGain * downmix_gain;
/* Remaining channels use the same results as the first. */
for(c = 1;c < num_channels;c++)
{
/* Skip LFE */
if(chans[c].channel != LFE)
voice->Direct.Params[c].Hrtf.Target = voice->Direct.Params[0].Hrtf.Target;
}
/* Calculate the directional coefficients once, which apply to all
* input channels of the source sends.
*/
CalcAngleCoeffs(Azi, Elev, Spread, coeffs);
for(i = 0;i < NumSends;i++)
{
const ALeffectslot *Slot = SendSlots[i];
if(Slot)
for(c = 0;c < num_channels;c++)
{
/* Skip LFE */
if(chans[c].channel != LFE)
ComputePanningGainsBF(Slot->ChanMap,
Slot->NumChannels, coeffs, WetGain[i] * downmix_gain,
voice->Send[i].Params[c].Gains.Target
);
}
}
}
else
{
/* Local sources on HRTF play with each channel panned to its
* relative location around the listener, providing "virtual
* speaker" responses.
*/
for(c = 0;c < num_channels;c++)
{
ALfloat coeffs[MAX_AMBI_COEFFS];
if(chans[c].channel == LFE)
{
/* Skip LFE */
continue;
}
/* Get the HRIR coefficients and delays for this channel
* position.
*/
GetHrtfCoeffs(Device->HrtfHandle,
chans[c].elevation, chans[c].angle, Spread,
voice->Direct.Params[c].Hrtf.Target.Coeffs,
voice->Direct.Params[c].Hrtf.Target.Delay
);
voice->Direct.Params[c].Hrtf.Target.Gain = DryGain;
/* Normal panning for auxiliary sends. */
CalcAngleCoeffs(chans[c].angle, chans[c].elevation, Spread, coeffs);
for(i = 0;i < NumSends;i++)
{
const ALeffectslot *Slot = SendSlots[i];
if(Slot)
ComputePanningGainsBF(Slot->ChanMap, Slot->NumChannels,
coeffs, WetGain[i], voice->Send[i].Params[c].Gains.Target
);
}
}
}
voice->Flags |= VOICE_HAS_HRTF;
}
else
{
/* Non-HRTF rendering. Use normal panning to the output. */
if(Distance > FLT_EPSILON)
{
ALfloat coeffs[MAX_AMBI_COEFFS];
ALfloat w0 = 0.0f;
/* Calculate NFC filter coefficient if needed. */
if(Device->AvgSpeakerDist > 0.0f)
{
ALfloat mdist = Distance * Listener.Params.MetersPerUnit;
ALfloat w1 = SPEEDOFSOUNDMETRESPERSEC /
(Device->AvgSpeakerDist * (ALfloat)Device->Frequency);
w0 = SPEEDOFSOUNDMETRESPERSEC /
(mdist * (ALfloat)Device->Frequency);
/* Clamp w0 for really close distances, to prevent excessive
* bass.
*/
w0 = minf(w0, w1*4.0f);
/* Adjust NFC filters. */
for(c = 0;c < num_channels;c++)
NfcFilterAdjust(&voice->Direct.Params[c].NFCtrlFilter, w0);
for(i = 0;i < MAX_AMBI_ORDER+1;i++)
voice->Direct.ChannelsPerOrder[i] = Device->NumChannelsPerOrder[i];
voice->Flags |= VOICE_HAS_NFC;
}
/* Calculate the directional coefficients once, which apply to all
* input channels.
*/
CalcAngleCoeffs((Device->Render_Mode==StereoPair) ? ScaleAzimuthFront(Azi, 1.5f) : Azi,
Elev, Spread, coeffs);
for(c = 0;c < num_channels;c++)
{
/* Special-case LFE */
if(chans[c].channel == LFE)
{
if(Device->Dry.Buffer == Device->RealOut.Buffer)
{
int idx = GetChannelIdxByName(&Device->RealOut, chans[c].channel);
if(idx != -1) voice->Direct.Params[c].Gains.Target[idx] = DryGain;
}
continue;
}
ComputePanGains(&Device->Dry, coeffs, DryGain * downmix_gain,
voice->Direct.Params[c].Gains.Target);
}
for(i = 0;i < NumSends;i++)
{
const ALeffectslot *Slot = SendSlots[i];
if(Slot)
for(c = 0;c < num_channels;c++)
{
/* Skip LFE */
if(chans[c].channel != LFE)
ComputePanningGainsBF(Slot->ChanMap,
Slot->NumChannels, coeffs, WetGain[i] * downmix_gain,
voice->Send[i].Params[c].Gains.Target
);
}
}
}
else
{
ALfloat w0 = 0.0f;
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.
*/
w0 = SPEEDOFSOUNDMETRESPERSEC /
(Device->AvgSpeakerDist * (ALfloat)Device->Frequency);
for(c = 0;c < num_channels;c++)
NfcFilterAdjust(&voice->Direct.Params[c].NFCtrlFilter, w0);
for(i = 0;i < MAX_AMBI_ORDER+1;i++)
voice->Direct.ChannelsPerOrder[i] = Device->NumChannelsPerOrder[i];
voice->Flags |= VOICE_HAS_NFC;
}
for(c = 0;c < num_channels;c++)
{
ALfloat coeffs[MAX_AMBI_COEFFS];
/* Special-case LFE */
if(chans[c].channel == LFE)
{
if(Device->Dry.Buffer == Device->RealOut.Buffer)
{
int idx = GetChannelIdxByName(&Device->RealOut, chans[c].channel);
if(idx != -1) voice->Direct.Params[c].Gains.Target[idx] = DryGain;
}
continue;
}
CalcAngleCoeffs(
(Device->Render_Mode==StereoPair) ? ScaleAzimuthFront(chans[c].angle, 3.0f)
: chans[c].angle,
chans[c].elevation, Spread, coeffs
);
ComputePanGains(&Device->Dry, coeffs, DryGain,
voice->Direct.Params[c].Gains.Target);
for(i = 0;i < NumSends;i++)
{
const ALeffectslot *Slot = SendSlots[i];
if(Slot)
ComputePanningGainsBF(Slot->ChanMap, Slot->NumChannels,
coeffs, WetGain[i], voice->Send[i].Params[c].Gains.Target
);
}
}
}
}
{
ALfloat hfScale = props->Direct.HFReference / Frequency;
ALfloat lfScale = props->Direct.LFReference / Frequency;
ALfloat gainHF = maxf(DryGainHF, 0.001f); /* Limit -60dB */
ALfloat gainLF = maxf(DryGainLF, 0.001f);
voice->Direct.FilterType = AF_None;
if(gainHF != 1.0f) voice->Direct.FilterType |= AF_LowPass;
if(gainLF != 1.0f) voice->Direct.FilterType |= AF_HighPass;
BiquadFilter_setParams(
&voice->Direct.Params[0].LowPass, BiquadType::HighShelf,
gainHF, hfScale, calc_rcpQ_from_slope(gainHF, 1.0f)
);
BiquadFilter_setParams(
&voice->Direct.Params[0].HighPass, BiquadType::LowShelf,
gainLF, lfScale, calc_rcpQ_from_slope(gainLF, 1.0f)
);
for(c = 1;c < num_channels;c++)
{
BiquadFilter_copyParams(&voice->Direct.Params[c].LowPass,
&voice->Direct.Params[0].LowPass);
BiquadFilter_copyParams(&voice->Direct.Params[c].HighPass,
&voice->Direct.Params[0].HighPass);
}
}
for(i = 0;i < NumSends;i++)
{
ALfloat hfScale = props->Send[i].HFReference / Frequency;
ALfloat lfScale = props->Send[i].LFReference / Frequency;
ALfloat gainHF = maxf(WetGainHF[i], 0.001f);
ALfloat gainLF = maxf(WetGainLF[i], 0.001f);
voice->Send[i].FilterType = AF_None;
if(gainHF != 1.0f) voice->Send[i].FilterType |= AF_LowPass;
if(gainLF != 1.0f) voice->Send[i].FilterType |= AF_HighPass;
BiquadFilter_setParams(
&voice->Send[i].Params[0].LowPass, BiquadType::HighShelf,
gainHF, hfScale, calc_rcpQ_from_slope(gainHF, 1.0f)
);
BiquadFilter_setParams(
&voice->Send[i].Params[0].HighPass, BiquadType::LowShelf,
gainLF, lfScale, calc_rcpQ_from_slope(gainLF, 1.0f)
);
for(c = 1;c < num_channels;c++)
{
BiquadFilter_copyParams(&voice->Send[i].Params[c].LowPass,
&voice->Send[i].Params[0].LowPass);
BiquadFilter_copyParams(&voice->Send[i].Params[c].HighPass,
&voice->Send[i].Params[0].HighPass);
}
}
}
void CalcNonAttnSourceParams(ALvoice *voice, const struct ALvoiceProps *props, const ALbuffer *ALBuffer, const ALCcontext *ALContext)
{
const ALCdevice *Device = ALContext->Device;
const ALlistener &Listener = ALContext->Listener;
ALfloat DryGain, DryGainHF, DryGainLF;
ALfloat WetGain[MAX_SENDS];
ALfloat WetGainHF[MAX_SENDS];
ALfloat WetGainLF[MAX_SENDS];
ALeffectslot *SendSlots[MAX_SENDS];
ALfloat Pitch;
ALsizei i;
voice->Direct.Buffer = Device->Dry.Buffer;
voice->Direct.Channels = Device->Dry.NumChannels;
for(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] = NULL;
voice->Send[i].Buffer = NULL;
voice->Send[i].Channels = 0;
}
else
{
voice->Send[i].Buffer = SendSlots[i]->WetBuffer;
voice->Send[i].Channels = SendSlots[i]->NumChannels;
}
}
/* Calculate the stepping value */
Pitch = (ALfloat)ALBuffer->Frequency/(ALfloat)Device->Frequency * props->Pitch;
if(Pitch > (ALfloat)MAX_PITCH)
voice->Step = MAX_PITCH<<FRACTIONBITS;
else
voice->Step = maxi(fastf2i(Pitch * FRACTIONONE), 1);
if(props->Resampler == BSinc24Resampler)
BsincPrepare(voice->Step, &voice->ResampleState.bsinc, &bsinc24);
else if(props->Resampler == BSinc12Resampler)
BsincPrepare(voice->Step, &voice->ResampleState.bsinc, &bsinc12);
voice->Resampler = SelectResampler(props->Resampler);
/* Calculate gains */
DryGain = clampf(props->Gain, props->MinGain, props->MaxGain);
DryGain *= props->Direct.Gain * Listener.Params.Gain;
DryGain = minf(DryGain, GAIN_MIX_MAX);
DryGainHF = props->Direct.GainHF;
DryGainLF = props->Direct.GainLF;
for(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, 0.0f, 0.0f, DryGain, DryGainHF, DryGainLF, WetGain,
WetGainLF, WetGainHF, SendSlots, ALBuffer, props, Listener, Device);
}
void CalcAttnSourceParams(ALvoice *voice, const struct ALvoiceProps *props, const ALbuffer *ALBuffer, const ALCcontext *ALContext)
{
const ALCdevice *Device = ALContext->Device;
const ALlistener &Listener = ALContext->Listener;
const ALsizei NumSends = Device->NumAuxSends;
aluVector Position, Velocity, Direction, SourceToListener;
ALfloat Distance, ClampedDist, DopplerFactor;
ALeffectslot *SendSlots[MAX_SENDS];
ALfloat RoomRolloff[MAX_SENDS];
ALfloat DecayDistance[MAX_SENDS];
ALfloat DecayLFDistance[MAX_SENDS];
ALfloat DecayHFDistance[MAX_SENDS];
ALfloat DryGain, DryGainHF, DryGainLF;
ALfloat WetGain[MAX_SENDS];
ALfloat WetGainHF[MAX_SENDS];
ALfloat WetGainLF[MAX_SENDS];
bool directional;
ALfloat ev, az;
ALfloat spread;
ALfloat Pitch;
ALint i;
/* Set mixing buffers and get send parameters. */
voice->Direct.Buffer = Device->Dry.Buffer;
voice->Direct.Channels = Device->Dry.NumChannels;
for(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] = NULL;
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 = log10f(REVERB_DECAY_GAIN) / log10f(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->Send[i].Buffer = NULL;
voice->Send[i].Channels = 0;
}
else
{
voice->Send[i].Buffer = SendSlots[i]->WetBuffer;
voice->Send[i].Channels = SendSlots[i]->NumChannels;
}
}
/* Transform source to listener space (convert to head relative) */
aluVectorSet(&Position, props->Position[0], props->Position[1], props->Position[2], 1.0f);
aluVectorSet(&Direction, props->Direction[0], props->Direction[1], props->Direction[2], 0.0f);
aluVectorSet(&Velocity, props->Velocity[0], props->Velocity[1], props->Velocity[2], 0.0f);
if(props->HeadRelative == AL_FALSE)
{
const aluMatrixf *Matrix = &Listener.Params.Matrix;
/* Transform source vectors */
Position = aluMatrixfVector(Matrix, &Position);
Velocity = aluMatrixfVector(Matrix, &Velocity);
Direction = aluMatrixfVector(Matrix, &Direction);
}
else
{
const aluVector *lvelocity = &Listener.Params.Velocity;
/* Offset the source velocity to be relative of the listener velocity */
Velocity.v[0] += lvelocity->v[0];
Velocity.v[1] += lvelocity->v[1];
Velocity.v[2] += lvelocity->v[2];
}
directional = aluNormalize(Direction.v) > 0.0f;
SourceToListener.v[0] = -Position.v[0];
SourceToListener.v[1] = -Position.v[1];
SourceToListener.v[2] = -Position.v[2];
SourceToListener.v[3] = 0.0f;
Distance = aluNormalize(SourceToListener.v);
/* Initial source gain */
DryGain = props->Gain;
DryGainHF = 1.0f;
DryGainLF = 1.0f;
for(i = 0;i < NumSends;i++)
{
WetGain[i] = props->Gain;
WetGainHF[i] = 1.0f;
WetGainLF[i] = 1.0f;
}
/* Calculate distance attenuation */
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(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(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 *= powf(ClampedDist/props->RefDistance, -props->RolloffFactor);
for(i = 0;i < NumSends;i++)
WetGain[i] *= powf(ClampedDist/props->RefDistance, -RoomRolloff[i]);
}
break;
case DistanceModel::Disable:
ClampedDist = props->RefDistance;
break;
}
/* Calculate directional soundcones */
if(directional && props->InnerAngle < 360.0f)
{
ALfloat ConeVolume;
ALfloat ConeHF;
ALfloat Angle;
Angle = acosf(aluDotproduct(&Direction, &SourceToListener));
Angle = RAD2DEG(Angle * ConeScale * 2.0f);
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)
{
for(i = 0;i < NumSends;i++)
WetGain[i] *= ConeVolume;
}
if(props->WetGainHFAuto)
{
for(i = 0;i < NumSends;i++)
WetGainHF[i] *= 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(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 = powf(AIRABSORBGAINHF, meters_base * props->AirAbsorptionFactor);
DryGainHF *= hfattn;
for(i = 0;i < NumSends;i++)
WetGainHF[i] *= 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(i = 0;i < NumSends;i++)
{
ALfloat gain, gainhf, gainlf;
if(!(DecayDistance[i] > 0.0f))
continue;
gain = powf(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)
{
gainhf = powf(REVERB_DECAY_GAIN, meters_base/DecayHFDistance[i]);
WetGainHF[i] *= minf(gainhf / gain, 1.0f);
gainlf = powf(REVERB_DECAY_GAIN, meters_base/DecayLFDistance[i]);
WetGainLF[i] *= minf(gainlf / gain, 1.0f);
}
}
}
}
/* Initial source pitch */
Pitch = props->Pitch;
/* Calculate velocity-based doppler effect */
DopplerFactor = props->DopplerFactor * Listener.Params.DopplerFactor;
if(DopplerFactor > 0.0f)
{
const aluVector *lvelocity = &Listener.Params.Velocity;
const ALfloat SpeedOfSound = Listener.Params.SpeedOfSound;
ALfloat vss, vls;
vss = aluDotproduct(&Velocity, &SourceToListener) * DopplerFactor;
vls = aluDotproduct(lvelocity, &SourceToListener) * DopplerFactor;
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 = HUGE_VALF;
}
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 *= (ALfloat)ALBuffer->Frequency/(ALfloat)Device->Frequency;
if(Pitch > (ALfloat)MAX_PITCH)
voice->Step = MAX_PITCH<<FRACTIONBITS;
else
voice->Step = maxi(fastf2i(Pitch * FRACTIONONE), 1);
if(props->Resampler == BSinc24Resampler)
BsincPrepare(voice->Step, &voice->ResampleState.bsinc, &bsinc24);
else if(props->Resampler == BSinc12Resampler)
BsincPrepare(voice->Step, &voice->ResampleState.bsinc, &bsinc12);
voice->Resampler = SelectResampler(props->Resampler);
if(Distance > 0.0f)
{
/* Clamp Y, in case rounding errors caused it to end up outside of
* -1...+1.
*/
ev = asinf(clampf(-SourceToListener.v[1], -1.0f, 1.0f));
/* Double negation on Z cancels out; negate once for changing source-
* to-listener to listener-to-source, and again for right-handed coords
* with -Z in front.
*/
az = atan2f(-SourceToListener.v[0], SourceToListener.v[2]*ZScale);
}
else
ev = az = 0.0f;
if(props->Radius > Distance)
spread = F_TAU - Distance/props->Radius*F_PI;
else if(Distance > 0.0f)
spread = asinf(props->Radius / Distance) * 2.0f;
else
spread = 0.0f;
CalcPanningAndFilters(voice, az, ev, Distance, spread, DryGain, DryGainHF, DryGainLF, WetGain,
WetGainLF, WetGainHF, SendSlots, ALBuffer, props, Listener, Device);
}
void CalcSourceParams(ALvoice *voice, ALCcontext *context, bool force)
{
ALvoiceProps *props{voice->Update.exchange(nullptr, std::memory_order_acq_rel)};
if(!props && !force) return;
if(props)
{
memcpy(voice->Props, props,
FAM_SIZE(struct ALvoiceProps, Send, context->Device->NumAuxSends)
);
AtomicReplaceHead(context->FreeVoiceProps, props);
}
props = voice->Props;
ALbufferlistitem *BufferListItem{voice->current_buffer.load(std::memory_order_relaxed)};
while(BufferListItem)
{
auto buffers_end = BufferListItem->buffers+BufferListItem->num_buffers;
auto buffer = std::find_if(BufferListItem->buffers, buffers_end,
[](const ALbuffer *buffer) noexcept -> bool
{ return buffer != nullptr; }
);
if(LIKELY(buffer != buffers_end))
{
if(props->SpatializeMode == SpatializeOn ||
(props->SpatializeMode == SpatializeAuto && (*buffer)->FmtChannels == FmtMono))
CalcAttnSourceParams(voice, props, *buffer, context);
else
CalcNonAttnSourceParams(voice, props, *buffer, context);
break;
}
BufferListItem = BufferListItem->next.load(std::memory_order_acquire);
}
}
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;
std::for_each(slots->slot, slots->slot+slots->count,
[ctx,cforce,&force](ALeffectslot *slot) -> void
{ force |= CalcEffectSlotParams(slot, ctx, cforce); }
);
std::for_each(ctx->Voices, ctx->Voices+ctx->VoiceCount.load(std::memory_order_acquire),
[ctx,force](ALvoice *voice) -> void
{
ALsource *source{voice->Source.load(std::memory_order_acquire)};
if(source) CalcSourceParams(voice, ctx, force);
}
);
}
IncrementRef(&ctx->UpdateCount);
}
void ProcessContext(ALCcontext *ctx, ALsizei SamplesToDo)
{
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->slot, auxslots->slot+auxslots->count,
[SamplesToDo](ALeffectslot *slot) -> void
{
std::for_each(slot->WetBuffer, slot->WetBuffer+slot->NumChannels,
[SamplesToDo](ALfloat *buffer) -> void
{ std::fill_n(buffer, SamplesToDo, 0.0f); }
);
}
);
/* Process voices that have a playing source. */
std::for_each(ctx->Voices, ctx->Voices+ctx->VoiceCount.load(std::memory_order_acquire),
[SamplesToDo,ctx](ALvoice *voice) -> void
{
ALsource *source{voice->Source.load(std::memory_order_acquire)};
if(!source) return;
if(!voice->Playing.load(std::memory_order_relaxed) || voice->Step < 1)
return;
if(!MixSource(voice, source->id, ctx, SamplesToDo))
{
voice->Source.store(nullptr, std::memory_order_relaxed);
voice->Playing.store(false, std::memory_order_release);
SendSourceStoppedEvent(ctx, source->id);
}
}
);
/* Process effects. */
std::for_each(auxslots->slot, auxslots->slot+auxslots->count,
[SamplesToDo](const ALeffectslot *slot) -> void
{
EffectState *state{slot->Params.mEffectState};
state->process(SamplesToDo, slot->WetBuffer, state->mOutBuffer,
state->mOutChannels);
}
);
}
void ApplyStablizer(FrontStablizer *Stablizer, ALfloat (*RESTRICT Buffer)[BUFFERSIZE],
int lidx, int ridx, int cidx, ALsizei SamplesToDo, ALsizei NumChannels)
{
ALfloat (*RESTRICT lsplit)[BUFFERSIZE] = Stablizer->LSplit;
ALfloat (*RESTRICT rsplit)[BUFFERSIZE] = Stablizer->RSplit;
ALsizei i;
/* Apply an all-pass to all channels, except the front-left and front-
* right, so they maintain the same relative phase.
*/
for(i = 0;i < NumChannels;i++)
{
if(i == lidx || i == ridx)
continue;
splitterap_process(&Stablizer->APFilter[i], Buffer[i], SamplesToDo);
}
bandsplit_process(&Stablizer->LFilter, lsplit[1], lsplit[0], Buffer[lidx], SamplesToDo);
bandsplit_process(&Stablizer->RFilter, rsplit[1], rsplit[0], Buffer[ridx], SamplesToDo);
for(i = 0;i < SamplesToDo;i++)
{
ALfloat lfsum, hfsum;
ALfloat m, s, c;
lfsum = lsplit[0][i] + rsplit[0][i];
hfsum = lsplit[1][i] + rsplit[1][i];
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.
*/
m = lfsum*cosf(1.0f/3.0f * F_PI_2) + hfsum*cosf(1.0f/4.0f * F_PI_2);
c = lfsum*sinf(1.0f/3.0f * F_PI_2) + hfsum*sinf(1.0f/4.0f * F_PI_2);
/* 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(ALfloat (*RESTRICT Samples)[BUFFERSIZE], const DistanceComp &distcomp,
ALfloat *RESTRICT Values, ALsizei SamplesToDo, ALsizei numchans)
{
for(ALsizei c{0};c < numchans;c++)
{
ALfloat *RESTRICT inout = Samples[c];
const ALfloat gain = distcomp[c].Gain;
const ALsizei base = distcomp[c].Length;
ALfloat *RESTRICT distbuf = distcomp[c].Buffer;
if(base == 0)
{
if(gain < 1.0f)
std::for_each(inout, inout+SamplesToDo,
[gain](ALfloat &in) noexcept -> void
{ in *= gain; }
);
continue;
}
if(LIKELY(SamplesToDo >= base))
{
auto out = std::copy_n(distbuf, base, Values);
std::copy_n(inout, SamplesToDo-base, out);
std::copy_n(inout+SamplesToDo-base, base, distbuf);
}
else
{
std::copy_n(distbuf, SamplesToDo, Values);
auto out = std::copy(distbuf+SamplesToDo, distbuf+base, distbuf);
std::copy_n(inout, SamplesToDo, out);
}
std::transform<ALfloat*RESTRICT>(Values, Values+SamplesToDo, inout,
[gain](ALfloat in) noexcept -> ALfloat
{ return in * gain; }
);
}
}
void ApplyDither(ALfloat (*RESTRICT Samples)[BUFFERSIZE], ALuint *dither_seed,
const ALfloat quant_scale, const ALsizei SamplesToDo, const ALsizei numchans)
{
ASSUME(numchans > 0);
/* 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](ALfloat *buffer) -> void
{
ASSUME(SamplesToDo > 0);
std::transform(buffer, buffer+SamplesToDo, buffer,
[&seed,invscale,quant_scale](ALfloat sample) noexcept -> ALfloat
{
ALfloat val = sample * quant_scale;
ALuint rng0 = dither_rng(&seed);
ALuint rng1 = dither_rng(&seed);
val += (ALfloat)(rng0*(1.0/UINT_MAX) - rng1*(1.0/UINT_MAX));
return fast_roundf(val) * invscale;
}
);
};
std::for_each(Samples, Samples+numchans, 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. There is an implied 1 bit in the mantissa
* along with the sign bit, giving 25 bits total, so [-16777216, +16777216]
* is the max value a normalized float can be scaled to before losing
* precision.
*/
return fastf2i(clampf(val*16777216.0f, -16777216.0f, 16777215.0f))<<7;
}
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 ALfloat (*RESTRICT InBuffer)[BUFFERSIZE], ALvoid *OutBuffer,
ALsizei Offset, ALsizei SamplesToDo, ALsizei numchans)
{
using SampleType = typename DevFmtTypeTraits<T>::Type;
ASSUME(numchans > 0);
SampleType *outbase = static_cast<SampleType*>(OutBuffer) + Offset*numchans;
auto conv_channel = [&outbase,SamplesToDo,numchans](const ALfloat *inbuf) -> void
{
ASSUME(SamplesToDo > 0);
SampleType *out{outbase++};
std::for_each<const ALfloat*RESTRICT>(inbuf, inbuf+SamplesToDo,
[numchans,&out](const ALfloat s) noexcept -> void
{
*out = SampleConv<SampleType>(s);
out += numchans;
}
);
};
std::for_each(InBuffer, InBuffer+numchans, 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.
*/
ALCcontext *ctx{device->ContextList.load(std::memory_order_acquire)};
while(ctx)
{
ProcessContext(ctx, SamplesToDo);
ctx = ctx->next.load(std::memory_order_relaxed);
}
/* 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);
/* Apply front image stablization for surround sound, if applicable. */
if(device->Stablizer)
{
int lidx = GetChannelIdxByName(&device->RealOut, FrontLeft);
int ridx = GetChannelIdxByName(&device->RealOut, FrontRight);
int cidx = GetChannelIdxByName(&device->RealOut, FrontCenter);
assert(lidx >= 0 && ridx >= 0 && cidx >= 0);
ApplyStablizer(device->Stablizer.get(), device->RealOut.Buffer, lidx, ridx, cidx,
SamplesToDo, device->RealOut.NumChannels);
}
/* Apply delays and attenuation for mismatched speaker distances. */
ApplyDistanceComp(device->RealOut.Buffer, device->ChannelDelay, device->TempBuffer[0],
SamplesToDo, device->RealOut.NumChannels);
/* Apply compression, limiting final sample amplitude, if desired. */
if(device->Limiter)
ApplyCompression(device->Limiter.get(), SamplesToDo, device->RealOut.Buffer);
/* Apply dithering. The compressor should have left enough headroom for
* the dither noise to not saturate.
*/
if(device->DitherDepth > 0.0f)
ApplyDither(device->RealOut.Buffer, &device->DitherSeed, device->DitherDepth,
SamplesToDo, device->RealOut.NumChannels);
if(LIKELY(OutBuffer))
{
ALfloat (*Buffer)[BUFFERSIZE] = device->RealOut.Buffer;
ALsizei Channels = device->RealOut.NumChannels;
/* Finally, interleave and convert samples, writing to the device's
* output buffer.
*/
switch(device->FmtType)
{
#define HANDLE_WRITE(T) case T: \
Write<T>(Buffer, OutBuffer, SamplesDone, SamplesToDo, Channels); 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(AL_FALSE, std::memory_order_acq_rel))
return;
AsyncEvent evt = ASYNC_EVENT(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 || (size_t)msglen >= sizeof(evt.u.user.msg))
evt.u.user.msg[sizeof(evt.u.user.msg)-1] = 0;
ALCcontext *ctx{device->ContextList.load()};
while(ctx)
{
ALbitfieldSOFT enabledevt = ctx->EnabledEvts.load(std::memory_order_acquire);
if((enabledevt&EventType_Disconnected) &&
ll_ringbuffer_write(ctx->AsyncEvents, &evt, 1) == 1)
alsem_post(&ctx->EventSem);
std::for_each(ctx->Voices, ctx->Voices+ctx->VoiceCount.load(std::memory_order_acquire),
[ctx](ALvoice *voice) -> void
{
ALsource *source{voice->Source.exchange(nullptr, std::memory_order_relaxed)};
if(source && voice->Playing.load(std::memory_order_relaxed))
{
/* If the source's voice was playing, it's now effectively
* stopped (the source state will be updated the next time
* it's checked).
*/
SendSourceStoppedEvent(ctx, source->id);
}
voice->Playing.store(false, std::memory_order_release);
}
);
ctx = ctx->next.load(std::memory_order_relaxed);
}
}