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authorChris Robinson <[email protected]>2018-11-16 20:32:19 -0800
committerChris Robinson <[email protected]>2018-11-16 20:32:19 -0800
commit53373a43b8984aea6a7e2107b264d208c00a5f53 (patch)
tree0d9326fd52f7818adc007f76acd452e3e6a03246 /Alc/ALu.c
parent317acd6ae2f110c76fd1e019a3066c8c45b64921 (diff)
Convert ALu.c to C++
Required changes to bsincgen to generate C++-friendly structures.
Diffstat (limited to 'Alc/ALu.c')
-rw-r--r--Alc/ALu.c1879
1 files changed, 0 insertions, 1879 deletions
diff --git a/Alc/ALu.c b/Alc/ALu.c
deleted file mode 100644
index a9b5a009..00000000
--- a/Alc/ALu.c
+++ /dev/null
@@ -1,1879 +0,0 @@
-/**
- * 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 "alMain.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 "static_assert.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;
-
-const aluMatrixf IdentityMatrixf = {{
- { 1.0f, 0.0f, 0.0f, 0.0f },
- { 0.0f, 1.0f, 0.0f, 0.0f },
- { 0.0f, 0.0f, 1.0f, 0.0f },
- { 0.0f, 0.0f, 0.0f, 1.0f },
-}};
-
-
-static 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;
-};
-
-static HrtfDirectMixerFunc MixDirectHrtf = MixDirectHrtf_C;
-
-
-void DeinitVoice(ALvoice *voice)
-{
- al_free(ATOMIC_EXCHANGE_PTR_SEQ(&voice->Update, NULL));
-}
-
-
-static 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;
-}
-
-
-/* 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.
- */
-static inline ALuint dither_rng(ALuint *seed)
-{
- *seed = (*seed * 96314165) + 907633515;
- return *seed;
-}
-
-
-static 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];
-}
-
-static 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];
-}
-
-static 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;
-}
-
-static 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];
-}
-
-static 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 aluInit(void)
-{
- MixDirectHrtf = SelectHrtfMixer();
-}
-
-
-static void SendSourceStoppedEvent(ALCcontext *context, ALuint id)
-{
- AsyncEvent evt = ASYNC_EVENT(EventType_SourceStateChange);
- ALbitfieldSOFT enabledevt;
- size_t strpos;
- ALuint scale;
-
- enabledevt = ATOMIC_LOAD(&context->EnabledEvts, almemory_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, (const char*)&evt, 1) == 1)
- alsem_post(&context->EventSem);
-}
-
-
-static void ProcessHrtf(ALCdevice *device, ALsizei SamplesToDo)
-{
- DirectHrtfState *state;
- int lidx, ridx;
- ALsizei c;
-
- if(device->AmbiUp)
- ambiup_process(device->AmbiUp,
- device->Dry.Buffer, device->Dry.NumChannels, device->FOAOut.Buffer,
- SamplesToDo
- );
-
- lidx = GetChannelIdxByName(&device->RealOut, FrontLeft);
- ridx = GetChannelIdxByName(&device->RealOut, FrontRight);
- assert(lidx != -1 && ridx != -1);
-
- state = device->Hrtf;
- for(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;
-}
-
-static void ProcessAmbiDec(ALCdevice *device, ALsizei SamplesToDo)
-{
- if(device->Dry.Buffer != device->FOAOut.Buffer)
- bformatdec_upSample(device->AmbiDecoder,
- device->Dry.Buffer, device->FOAOut.Buffer, device->FOAOut.NumChannels,
- SamplesToDo
- );
- bformatdec_process(device->AmbiDecoder,
- device->RealOut.Buffer, device->RealOut.NumChannels, device->Dry.Buffer,
- SamplesToDo
- );
-}
-
-static void ProcessAmbiUp(ALCdevice *device, ALsizei SamplesToDo)
-{
- ambiup_process(device->AmbiUp,
- device->RealOut.Buffer, device->RealOut.NumChannels, device->FOAOut.Buffer,
- SamplesToDo
- );
-}
-
-static 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,
- device->RealOut.Buffer[lidx], device->RealOut.Buffer[ridx],
- device->Dry.Buffer, SamplesToDo
- );
-}
-
-static 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, device->RealOut.Buffer[lidx],
- device->RealOut.Buffer[ridx], SamplesToDo);
-}
-
-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];
-}
-
-
-static bool CalcContextParams(ALCcontext *Context)
-{
- ALlistener *Listener = Context->Listener;
- struct ALcontextProps *props;
-
- props = ATOMIC_EXCHANGE_PTR(&Context->Update, NULL, almemory_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.DistanceModel = props->DistanceModel;
-
- ATOMIC_REPLACE_HEAD(struct ALcontextProps*, &Context->FreeContextProps, props);
- return true;
-}
-
-static bool CalcListenerParams(ALCcontext *Context)
-{
- ALlistener *Listener = Context->Listener;
- ALfloat N[3], V[3], U[3], P[3];
- struct ALlistenerProps *props;
- aluVector vel;
-
- props = ATOMIC_EXCHANGE_PTR(&Listener->Update, NULL, almemory_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;
-
- ATOMIC_REPLACE_HEAD(struct ALlistenerProps*, &Context->FreeListenerProps, props);
- return true;
-}
-
-static bool CalcEffectSlotParams(ALeffectslot *slot, ALCcontext *context, bool force)
-{
- struct ALeffectslotProps *props;
- ALeffectState *state;
-
- props = ATOMIC_EXCHANGE_PTR(&slot->Update, NULL, almemory_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.EffectState)
- {
- /* 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->Ref);
- props->State = NULL;
- }
- else
- {
- /* Otherwise, replace it and send off the old one with a release
- * event.
- */
- AsyncEvent evt = ASYNC_EVENT(EventType_ReleaseEffectState);
- evt.u.EffectState = slot->Params.EffectState;
-
- slot->Params.EffectState = state;
- props->State = NULL;
-
- if(LIKELY(ll_ringbuffer_write(context->AsyncEvents, (const char*)&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.EffectState;
- }
- }
-
- ATOMIC_REPLACE_HEAD(struct ALeffectslotProps*, &context->FreeEffectslotProps, props);
- }
- else
- state = slot->Params.EffectState;
-
- V(state,update)(context, slot, &slot->Params.EffectProps);
- return true;
-}
-
-
-static const 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) }
-};
-
-static 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);
- }
- }
-}
-
-static 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;
- 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);
-}
-
-static 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;
- 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->DistanceModel : Listener->Params.DistanceModel)
- {
- case InverseDistanceClamped:
- ClampedDist = clampf(ClampedDist, props->RefDistance, props->MaxDistance);
- if(props->MaxDistance < props->RefDistance)
- break;
- /*fall-through*/
- case InverseDistance:
- 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 LinearDistanceClamped:
- ClampedDist = clampf(ClampedDist, props->RefDistance, props->MaxDistance);
- if(props->MaxDistance < props->RefDistance)
- break;
- /*fall-through*/
- case LinearDistance:
- 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 ExponentDistanceClamped:
- ClampedDist = clampf(ClampedDist, props->RefDistance, props->MaxDistance);
- if(props->MaxDistance < props->RefDistance)
- break;
- /*fall-through*/
- case ExponentDistance:
- 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 DisableDistance:
- 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);
-}
-
-static void CalcSourceParams(ALvoice *voice, ALCcontext *context, bool force)
-{
- ALbufferlistitem *BufferListItem;
- struct ALvoiceProps *props;
-
- props = ATOMIC_EXCHANGE_PTR(&voice->Update, NULL, almemory_order_acq_rel);
- if(!props && !force) return;
-
- if(props)
- {
- memcpy(voice->Props, props,
- FAM_SIZE(struct ALvoiceProps, Send, context->Device->NumAuxSends)
- );
-
- ATOMIC_REPLACE_HEAD(struct ALvoiceProps*, &context->FreeVoiceProps, props);
- }
- props = voice->Props;
-
- BufferListItem = ATOMIC_LOAD(&voice->current_buffer, almemory_order_relaxed);
- while(BufferListItem != NULL)
- {
- const ALbuffer *buffer = NULL;
- ALsizei i = 0;
- while(!buffer && i < BufferListItem->num_buffers)
- buffer = BufferListItem->buffers[i];
- if(LIKELY(buffer))
- {
- if(props->SpatializeMode == SpatializeOn ||
- (props->SpatializeMode == SpatializeAuto && buffer->FmtChannels == FmtMono))
- CalcAttnSourceParams(voice, props, buffer, context);
- else
- CalcNonAttnSourceParams(voice, props, buffer, context);
- break;
- }
- BufferListItem = ATOMIC_LOAD(&BufferListItem->next, almemory_order_acquire);
- }
-}
-
-
-static void ProcessParamUpdates(ALCcontext *ctx, const struct ALeffectslotArray *slots)
-{
- ALvoice **voice, **voice_end;
- ALsource *source;
- ALsizei i;
-
- IncrementRef(&ctx->UpdateCount);
- if(!ATOMIC_LOAD(&ctx->HoldUpdates, almemory_order_acquire))
- {
- bool cforce = CalcContextParams(ctx);
- bool force = CalcListenerParams(ctx) | cforce;
- for(i = 0;i < slots->count;i++)
- force |= CalcEffectSlotParams(slots->slot[i], ctx, cforce);
-
- voice = ctx->Voices;
- voice_end = voice + ctx->VoiceCount;
- for(;voice != voice_end;++voice)
- {
- source = ATOMIC_LOAD(&(*voice)->Source, almemory_order_acquire);
- if(source) CalcSourceParams(*voice, ctx, force);
- }
- }
- IncrementRef(&ctx->UpdateCount);
-}
-
-
-static void ApplyStablizer(FrontStablizer *Stablizer, ALfloat (*RESTRICT Buffer)[BUFFERSIZE],
- int lidx, int ridx, int cidx, ALsizei SamplesToDo,
- ALsizei NumChannels)
-{
- ALfloat (*RESTRICT lsplit)[BUFFERSIZE] = ASSUME_ALIGNED(Stablizer->LSplit, 16);
- ALfloat (*RESTRICT rsplit)[BUFFERSIZE] = ASSUME_ALIGNED(Stablizer->RSplit, 16);
- 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;
- }
-}
-
-static void ApplyDistanceComp(ALfloat (*RESTRICT Samples)[BUFFERSIZE], DistanceComp *distcomp,
- ALfloat *RESTRICT Values, ALsizei SamplesToDo, ALsizei numchans)
-{
- ALsizei i, c;
-
- Values = ASSUME_ALIGNED(Values, 16);
- for(c = 0;c < numchans;c++)
- {
- ALfloat *RESTRICT inout = ASSUME_ALIGNED(Samples[c], 16);
- const ALfloat gain = distcomp[c].Gain;
- const ALsizei base = distcomp[c].Length;
- ALfloat *RESTRICT distbuf = ASSUME_ALIGNED(distcomp[c].Buffer, 16);
-
- if(base == 0)
- {
- if(gain < 1.0f)
- {
- for(i = 0;i < SamplesToDo;i++)
- inout[i] *= gain;
- }
- continue;
- }
-
- if(LIKELY(SamplesToDo >= base))
- {
- for(i = 0;i < base;i++)
- Values[i] = distbuf[i];
- for(;i < SamplesToDo;i++)
- Values[i] = inout[i-base];
- memcpy(distbuf, &inout[SamplesToDo-base], base*sizeof(ALfloat));
- }
- else
- {
- for(i = 0;i < SamplesToDo;i++)
- Values[i] = distbuf[i];
- memmove(distbuf, distbuf+SamplesToDo, (base-SamplesToDo)*sizeof(ALfloat));
- memcpy(distbuf+base-SamplesToDo, inout, SamplesToDo*sizeof(ALfloat));
- }
- for(i = 0;i < SamplesToDo;i++)
- inout[i] = Values[i]*gain;
- }
-}
-
-static void ApplyDither(ALfloat (*RESTRICT Samples)[BUFFERSIZE], ALuint *dither_seed,
- const ALfloat quant_scale, const ALsizei SamplesToDo,
- const ALsizei numchans)
-{
- const ALfloat invscale = 1.0f / quant_scale;
- ALuint seed = *dither_seed;
- ALsizei c, i;
-
- ASSUME(numchans > 0);
- ASSUME(SamplesToDo > 0);
-
- /* Dithering. Step 1, generate whitenoise (uniform distribution of random
- * values between -1 and +1). Step 2 is to add the noise to the samples,
- * before rounding and after scaling up to the desired quantization depth.
- */
- for(c = 0;c < numchans;c++)
- {
- ALfloat *RESTRICT samples = Samples[c];
- for(i = 0;i < SamplesToDo;i++)
- {
- ALfloat val = samples[i] * quant_scale;
- ALuint rng0 = dither_rng(&seed);
- ALuint rng1 = dither_rng(&seed);
- val += (ALfloat)(rng0*(1.0/UINT_MAX) - rng1*(1.0/UINT_MAX));
- samples[i] = fast_roundf(val) * invscale;
- }
- }
- *dither_seed = seed;
-}
-
-
-static inline ALfloat Conv_ALfloat(ALfloat val)
-{ return val; }
-static inline ALint Conv_ALint(ALfloat val)
-{
- /* 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;
-}
-static inline ALshort Conv_ALshort(ALfloat val)
-{ return fastf2i(clampf(val*32768.0f, -32768.0f, 32767.0f)); }
-static inline ALbyte Conv_ALbyte(ALfloat val)
-{ return fastf2i(clampf(val*128.0f, -128.0f, 127.0f)); }
-
-/* Define unsigned output variations. */
-#define DECL_TEMPLATE(T, func, O) \
-static inline T Conv_##T(ALfloat val) { return func(val)+O; }
-
-DECL_TEMPLATE(ALubyte, Conv_ALbyte, 128)
-DECL_TEMPLATE(ALushort, Conv_ALshort, 32768)
-DECL_TEMPLATE(ALuint, Conv_ALint, 2147483648u)
-
-#undef DECL_TEMPLATE
-
-#define DECL_TEMPLATE(T, A) \
-static void Write##A(const ALfloat (*RESTRICT InBuffer)[BUFFERSIZE], \
- ALvoid *OutBuffer, ALsizei Offset, ALsizei SamplesToDo, \
- ALsizei numchans) \
-{ \
- ALsizei i, j; \
- \
- ASSUME(numchans > 0); \
- ASSUME(SamplesToDo > 0); \
- \
- for(j = 0;j < numchans;j++) \
- { \
- const ALfloat *RESTRICT in = ASSUME_ALIGNED(InBuffer[j], 16); \
- T *RESTRICT out = (T*)OutBuffer + Offset*numchans + j; \
- \
- for(i = 0;i < SamplesToDo;i++) \
- out[i*numchans] = Conv_##T(in[i]); \
- } \
-}
-
-DECL_TEMPLATE(ALfloat, F32)
-DECL_TEMPLATE(ALuint, UI32)
-DECL_TEMPLATE(ALint, I32)
-DECL_TEMPLATE(ALushort, UI16)
-DECL_TEMPLATE(ALshort, I16)
-DECL_TEMPLATE(ALubyte, UI8)
-DECL_TEMPLATE(ALbyte, I8)
-
-#undef DECL_TEMPLATE
-
-
-void aluMixData(ALCdevice *device, ALvoid *OutBuffer, ALsizei NumSamples)
-{
- ALsizei SamplesToDo;
- ALsizei SamplesDone;
- ALCcontext *ctx;
- ALsizei i, c;
-
- START_MIXER_MODE();
- for(SamplesDone = 0;SamplesDone < NumSamples;)
- {
- SamplesToDo = mini(NumSamples-SamplesDone, BUFFERSIZE);
- for(c = 0;c < device->Dry.NumChannels;c++)
- memset(device->Dry.Buffer[c], 0, SamplesToDo*sizeof(ALfloat));
- if(device->Dry.Buffer != device->FOAOut.Buffer)
- for(c = 0;c < device->FOAOut.NumChannels;c++)
- memset(device->FOAOut.Buffer[c], 0, SamplesToDo*sizeof(ALfloat));
- if(device->Dry.Buffer != device->RealOut.Buffer)
- for(c = 0;c < device->RealOut.NumChannels;c++)
- memset(device->RealOut.Buffer[c], 0, SamplesToDo*sizeof(ALfloat));
-
- IncrementRef(&device->MixCount);
-
- ctx = ATOMIC_LOAD(&device->ContextList, almemory_order_acquire);
- while(ctx)
- {
- const struct ALeffectslotArray *auxslots;
-
- auxslots = ATOMIC_LOAD(&ctx->ActiveAuxSlots, almemory_order_acquire);
- ProcessParamUpdates(ctx, auxslots);
-
- for(i = 0;i < auxslots->count;i++)
- {
- ALeffectslot *slot = auxslots->slot[i];
- for(c = 0;c < slot->NumChannels;c++)
- memset(slot->WetBuffer[c], 0, SamplesToDo*sizeof(ALfloat));
- }
-
- /* source processing */
- for(i = 0;i < ctx->VoiceCount;i++)
- {
- ALvoice *voice = ctx->Voices[i];
- ALsource *source = ATOMIC_LOAD(&voice->Source, almemory_order_acquire);
- if(source && ATOMIC_LOAD(&voice->Playing, almemory_order_relaxed) &&
- voice->Step > 0)
- {
- if(!MixSource(voice, source->id, ctx, SamplesToDo))
- {
- ATOMIC_STORE(&voice->Source, NULL, almemory_order_relaxed);
- ATOMIC_STORE(&voice->Playing, false, almemory_order_release);
- SendSourceStoppedEvent(ctx, source->id);
- }
- }
- }
-
- /* effect slot processing */
- for(i = 0;i < auxslots->count;i++)
- {
- const ALeffectslot *slot = auxslots->slot[i];
- ALeffectState *state = slot->Params.EffectState;
- V(state,process)(SamplesToDo, slot->WetBuffer, state->OutBuffer,
- state->OutChannels);
- }
-
- ctx = ATOMIC_LOAD(&ctx->next, almemory_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 an exact, stable
- * conversion. */
- device->SamplesDone += SamplesToDo;
- device->ClockBase += (device->SamplesDone/device->Frequency) * DEVICE_CLOCK_RES;
- device->SamplesDone %= device->Frequency;
- IncrementRef(&device->MixCount);
-
- /* Apply post-process for finalizing the Dry mix to the RealOut
- * (Ambisonic decode, UHJ encode, etc).
- */
- if(LIKELY(device->PostProcess))
- device->PostProcess(device, SamplesToDo);
-
- 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, device->RealOut.Buffer, lidx, ridx, cidx,
- SamplesToDo, device->RealOut.NumChannels);
- }
-
- ApplyDistanceComp(device->RealOut.Buffer, device->ChannelDelay, device->TempBuffer[0],
- SamplesToDo, device->RealOut.NumChannels);
-
- if(device->Limiter)
- ApplyCompression(device->Limiter, SamplesToDo, device->RealOut.Buffer);
-
- 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;
-
- switch(device->FmtType)
- {
-#define HANDLE_WRITE(T, S) case T: \
- Write##S(Buffer, OutBuffer, SamplesDone, SamplesToDo, Channels); break;
- HANDLE_WRITE(DevFmtByte, I8)
- HANDLE_WRITE(DevFmtUByte, UI8)
- HANDLE_WRITE(DevFmtShort, I16)
- HANDLE_WRITE(DevFmtUShort, UI16)
- HANDLE_WRITE(DevFmtInt, I32)
- HANDLE_WRITE(DevFmtUInt, UI32)
- HANDLE_WRITE(DevFmtFloat, F32)
-#undef HANDLE_WRITE
- }
- }
-
- SamplesDone += SamplesToDo;
- }
- END_MIXER_MODE();
-}
-
-
-void aluHandleDisconnect(ALCdevice *device, const char *msg, ...)
-{
- AsyncEvent evt = ASYNC_EVENT(EventType_Disconnected);
- ALCcontext *ctx;
- va_list args;
- int msglen;
-
- if(!ATOMIC_EXCHANGE(&device->Connected, AL_FALSE, almemory_order_acq_rel))
- return;
-
- evt.u.user.type = AL_EVENT_TYPE_DISCONNECTED_SOFT;
- evt.u.user.id = 0;
- evt.u.user.param = 0;
-
- va_start(args, msg);
- 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;
-
- ctx = ATOMIC_LOAD_SEQ(&device->ContextList);
- while(ctx)
- {
- ALbitfieldSOFT enabledevt = ATOMIC_LOAD(&ctx->EnabledEvts, almemory_order_acquire);
- ALsizei i;
-
- if((enabledevt&EventType_Disconnected) &&
- ll_ringbuffer_write(ctx->AsyncEvents, (const char*)&evt, 1) == 1)
- alsem_post(&ctx->EventSem);
-
- for(i = 0;i < ctx->VoiceCount;i++)
- {
- ALvoice *voice = ctx->Voices[i];
- ALsource *source;
-
- source = ATOMIC_EXCHANGE_PTR(&voice->Source, NULL, almemory_order_relaxed);
- if(source && ATOMIC_LOAD(&voice->Playing, almemory_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);
- }
- ATOMIC_STORE(&voice->Playing, false, almemory_order_release);
- }
-
- ctx = ATOMIC_LOAD(&ctx->next, almemory_order_relaxed);
- }
-}