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Diffstat (limited to 'Alc/ALu.c')
| -rw-r--r-- | Alc/ALu.c | 1923 |
1 files changed, 0 insertions, 1923 deletions
diff --git a/Alc/ALu.c b/Alc/ALu.c deleted file mode 100644 index 03abb116..00000000 --- a/Alc/ALu.c +++ /dev/null @@ -1,1923 +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" - -#include "backends/base.h" - - -extern inline ALfloat minf(ALfloat a, ALfloat b); -extern inline ALfloat maxf(ALfloat a, ALfloat b); -extern inline ALfloat clampf(ALfloat val, ALfloat min, ALfloat max); - -extern inline ALdouble mind(ALdouble a, ALdouble b); -extern inline ALdouble maxd(ALdouble a, ALdouble b); -extern inline ALdouble clampd(ALdouble val, ALdouble min, ALdouble max); - -extern inline ALuint minu(ALuint a, ALuint b); -extern inline ALuint maxu(ALuint a, ALuint b); -extern inline ALuint clampu(ALuint val, ALuint min, ALuint max); - -extern inline ALint mini(ALint a, ALint b); -extern inline ALint maxi(ALint a, ALint b); -extern inline ALint clampi(ALint val, ALint min, ALint max); - -extern inline ALint64 mini64(ALint64 a, ALint64 b); -extern inline ALint64 maxi64(ALint64 a, ALint64 b); -extern inline ALint64 clampi64(ALint64 val, ALint64 min, ALint64 max); - -extern inline ALuint64 minu64(ALuint64 a, ALuint64 b); -extern inline ALuint64 maxu64(ALuint64 a, ALuint64 b); -extern inline ALuint64 clampu64(ALuint64 val, ALuint64 min, ALuint64 max); - -extern inline size_t minz(size_t a, size_t b); -extern inline size_t maxz(size_t a, size_t b); -extern inline size_t clampz(size_t val, size_t min, size_t max); - -extern inline ALfloat lerp(ALfloat val1, ALfloat val2, ALfloat mu); -extern inline ALfloat cubic(ALfloat val1, ALfloat val2, ALfloat val3, ALfloat val4, ALfloat mu); - -extern inline void aluVectorSet(aluVector *restrict vector, ALfloat x, ALfloat y, ALfloat z, ALfloat w); - -extern inline void aluMatrixfSetRow(aluMatrixf *matrix, ALuint row, - ALfloat m0, ALfloat m1, ALfloat m2, ALfloat m3); -extern inline void aluMatrixfSet(aluMatrixf *matrix, - ALfloat m00, ALfloat m01, ALfloat m02, ALfloat m03, - ALfloat m10, ALfloat m11, ALfloat m12, ALfloat m13, - ALfloat m20, ALfloat m21, ALfloat m22, ALfloat m23, - ALfloat m30, ALfloat m31, ALfloat m32, ALfloat m33); - - -/* 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); - } -} |