1314 lines
52 KiB
C
1314 lines
52 KiB
C
/**
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* OpenAL cross platform audio library
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* Copyright (C) 1999-2007 by authors.
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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* Or go to http://www.gnu.org/copyleft/lgpl.html
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*/
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#include "config.h"
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#include <math.h>
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#include <stdlib.h>
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#include <string.h>
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#include <ctype.h>
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#include <assert.h>
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#include "alMain.h"
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#include "AL/al.h"
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#include "AL/alc.h"
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#include "alSource.h"
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#include "alBuffer.h"
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#include "alThunk.h"
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#include "alListener.h"
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#include "alAuxEffectSlot.h"
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#include "alu.h"
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#include "bs2b.h"
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#define FRACTIONBITS 14
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#define FRACTIONMASK ((1L<<FRACTIONBITS)-1)
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#define MAX_PITCH 65536
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/* Minimum ramp length in milliseconds. The value below was chosen to
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* adequately reduce clicks and pops from harsh gain changes. */
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#define MIN_RAMP_LENGTH 16
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static __inline ALfloat aluF2F(ALfloat Value)
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{
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return Value;
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}
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static __inline ALshort aluF2S(ALfloat Value)
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{
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ALint i;
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if(Value < 0.0f)
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{
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i = (ALint)(Value*32768.0f);
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i = max(-32768, i);
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}
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else
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{
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i = (ALint)(Value*32767.0f);
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i = min( 32767, i);
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}
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return ((ALshort)i);
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}
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static __inline ALubyte aluF2UB(ALfloat Value)
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{
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ALshort i = aluF2S(Value);
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return (i>>8)+128;
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}
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static __inline ALvoid aluCrossproduct(const ALfloat *inVector1, const ALfloat *inVector2, ALfloat *outVector)
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{
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outVector[0] = inVector1[1]*inVector2[2] - inVector1[2]*inVector2[1];
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outVector[1] = inVector1[2]*inVector2[0] - inVector1[0]*inVector2[2];
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outVector[2] = inVector1[0]*inVector2[1] - inVector1[1]*inVector2[0];
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}
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static __inline ALfloat aluDotproduct(const ALfloat *inVector1, const ALfloat *inVector2)
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{
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return inVector1[0]*inVector2[0] + inVector1[1]*inVector2[1] +
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inVector1[2]*inVector2[2];
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}
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static __inline ALvoid aluNormalize(ALfloat *inVector)
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{
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ALfloat length, inverse_length;
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length = aluSqrt(aluDotproduct(inVector, inVector));
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if(length != 0.0f)
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{
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inverse_length = 1.0f/length;
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inVector[0] *= inverse_length;
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inVector[1] *= inverse_length;
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inVector[2] *= inverse_length;
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}
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}
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static __inline ALvoid aluMatrixVector(ALfloat *vector,ALfloat w,ALfloat matrix[4][4])
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{
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ALfloat temp[4] = {
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vector[0], vector[1], vector[2], w
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};
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vector[0] = temp[0]*matrix[0][0] + temp[1]*matrix[1][0] + temp[2]*matrix[2][0] + temp[3]*matrix[3][0];
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vector[1] = temp[0]*matrix[0][1] + temp[1]*matrix[1][1] + temp[2]*matrix[2][1] + temp[3]*matrix[3][1];
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vector[2] = temp[0]*matrix[0][2] + temp[1]*matrix[1][2] + temp[2]*matrix[2][2] + temp[3]*matrix[3][2];
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}
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ALvoid CalcNonAttnSourceParams(ALsource *ALSource, const ALCcontext *ALContext)
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{
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ALfloat SourceVolume,ListenerGain,MinVolume,MaxVolume;
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ALfloat DryGain, DryGainHF;
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ALfloat WetGain[MAX_SENDS];
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ALfloat WetGainHF[MAX_SENDS];
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ALint NumSends, Frequency;
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ALfloat cw;
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ALint i;
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//Get context properties
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NumSends = ALContext->Device->NumAuxSends;
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Frequency = ALContext->Device->Frequency;
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//Get listener properties
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ListenerGain = ALContext->Listener.Gain;
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//Get source properties
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SourceVolume = ALSource->flGain;
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MinVolume = ALSource->flMinGain;
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MaxVolume = ALSource->flMaxGain;
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//1. Multi-channel buffers always play "normal"
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ALSource->Params.Pitch = ALSource->flPitch;
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DryGain = SourceVolume;
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DryGain = __min(DryGain,MaxVolume);
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DryGain = __max(DryGain,MinVolume);
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DryGainHF = 1.0f;
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switch(ALSource->DirectFilter.type)
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{
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case AL_FILTER_LOWPASS:
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DryGain *= ALSource->DirectFilter.Gain;
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DryGainHF *= ALSource->DirectFilter.GainHF;
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break;
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}
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for(i = 0;i < OUTPUTCHANNELS;i++)
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ALSource->Params.DryGains[i] = DryGain * ListenerGain;
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for(i = 0;i < NumSends;i++)
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{
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WetGain[i] = SourceVolume;
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WetGain[i] = __min(WetGain[i],MaxVolume);
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WetGain[i] = __max(WetGain[i],MinVolume);
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WetGainHF[i] = 1.0f;
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switch(ALSource->Send[i].WetFilter.type)
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{
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case AL_FILTER_LOWPASS:
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WetGain[i] *= ALSource->Send[i].WetFilter.Gain;
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WetGainHF[i] *= ALSource->Send[i].WetFilter.GainHF;
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break;
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}
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ALSource->Params.WetGains[i] = WetGain[i] * ListenerGain;
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}
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for(i = NumSends;i < MAX_SENDS;i++)
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{
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ALSource->Params.WetGains[i] = 0.0f;
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WetGainHF[i] = 1.0f;
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}
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/* Update filter coefficients. Calculations based on the I3DL2
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* spec. */
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cw = cos(2.0*M_PI * LOWPASSFREQCUTOFF / Frequency);
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/* We use two chained one-pole filters, so we need to take the
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* square root of the squared gain, which is the same as the base
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* gain. */
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ALSource->Params.iirFilter.coeff = lpCoeffCalc(DryGainHF, cw);
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for(i = 0;i < NumSends;i++)
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{
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/* We use a one-pole filter, so we need to take the squared gain */
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ALfloat a = lpCoeffCalc(WetGainHF[i]*WetGainHF[i], cw);
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ALSource->Params.Send[i].iirFilter.coeff = a;
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}
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}
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ALvoid CalcSourceParams(ALsource *ALSource, const ALCcontext *ALContext)
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{
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const ALCdevice *Device = ALContext->Device;
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ALfloat InnerAngle,OuterAngle,Angle,Distance,DryMix,OrigDist;
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ALfloat Direction[3],Position[3],SourceToListener[3];
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ALfloat Velocity[3],ListenerVel[3];
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ALfloat MinVolume,MaxVolume,MinDist,MaxDist,Rolloff,OuterGainHF;
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ALfloat ConeVolume,ConeHF,SourceVolume,ListenerGain;
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ALfloat DopplerFactor, DopplerVelocity, flSpeedOfSound;
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ALfloat Matrix[4][4];
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ALfloat flAttenuation, effectiveDist;
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ALfloat RoomAttenuation[MAX_SENDS];
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ALfloat MetersPerUnit;
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ALfloat RoomRolloff[MAX_SENDS];
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ALfloat DryGainHF = 1.0f;
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ALfloat WetGain[MAX_SENDS];
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ALfloat WetGainHF[MAX_SENDS];
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ALfloat DirGain, AmbientGain;
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const ALfloat *SpeakerGain;
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ALfloat length;
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ALuint Frequency;
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ALint NumSends;
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ALint pos, s, i;
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ALfloat cw;
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for(i = 0;i < MAX_SENDS;i++)
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WetGainHF[i] = 1.0f;
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//Get context properties
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DopplerFactor = ALContext->DopplerFactor * ALSource->DopplerFactor;
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DopplerVelocity = ALContext->DopplerVelocity;
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flSpeedOfSound = ALContext->flSpeedOfSound;
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NumSends = Device->NumAuxSends;
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Frequency = Device->Frequency;
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//Get listener properties
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ListenerGain = ALContext->Listener.Gain;
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MetersPerUnit = ALContext->Listener.MetersPerUnit;
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memcpy(ListenerVel, ALContext->Listener.Velocity, sizeof(ALContext->Listener.Velocity));
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//Get source properties
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SourceVolume = ALSource->flGain;
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memcpy(Position, ALSource->vPosition, sizeof(ALSource->vPosition));
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memcpy(Direction, ALSource->vOrientation, sizeof(ALSource->vOrientation));
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memcpy(Velocity, ALSource->vVelocity, sizeof(ALSource->vVelocity));
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MinVolume = ALSource->flMinGain;
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MaxVolume = ALSource->flMaxGain;
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MinDist = ALSource->flRefDistance;
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MaxDist = ALSource->flMaxDistance;
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Rolloff = ALSource->flRollOffFactor;
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InnerAngle = ALSource->flInnerAngle;
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OuterAngle = ALSource->flOuterAngle;
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OuterGainHF = ALSource->OuterGainHF;
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//1. Translate Listener to origin (convert to head relative)
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if(ALSource->bHeadRelative==AL_FALSE)
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{
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ALfloat U[3],V[3],N[3];
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// Build transform matrix
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memcpy(N, ALContext->Listener.Forward, sizeof(N)); // At-vector
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aluNormalize(N); // Normalized At-vector
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memcpy(V, ALContext->Listener.Up, sizeof(V)); // Up-vector
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aluNormalize(V); // Normalized Up-vector
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aluCrossproduct(N, V, U); // Right-vector
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aluNormalize(U); // Normalized Right-vector
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Matrix[0][0] = U[0]; Matrix[0][1] = V[0]; Matrix[0][2] = -N[0]; Matrix[0][3] = 0.0f;
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Matrix[1][0] = U[1]; Matrix[1][1] = V[1]; Matrix[1][2] = -N[1]; Matrix[1][3] = 0.0f;
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Matrix[2][0] = U[2]; Matrix[2][1] = V[2]; Matrix[2][2] = -N[2]; Matrix[2][3] = 0.0f;
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Matrix[3][0] = 0.0f; Matrix[3][1] = 0.0f; Matrix[3][2] = 0.0f; Matrix[3][3] = 1.0f;
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// Translate position
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Position[0] -= ALContext->Listener.Position[0];
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Position[1] -= ALContext->Listener.Position[1];
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Position[2] -= ALContext->Listener.Position[2];
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// Transform source position and direction into listener space
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aluMatrixVector(Position, 1.0f, Matrix);
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aluMatrixVector(Direction, 0.0f, Matrix);
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// Transform source and listener velocity into listener space
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aluMatrixVector(Velocity, 0.0f, Matrix);
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aluMatrixVector(ListenerVel, 0.0f, Matrix);
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}
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else
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ListenerVel[0] = ListenerVel[1] = ListenerVel[2] = 0.0f;
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SourceToListener[0] = -Position[0];
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SourceToListener[1] = -Position[1];
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SourceToListener[2] = -Position[2];
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aluNormalize(SourceToListener);
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aluNormalize(Direction);
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//2. Calculate distance attenuation
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Distance = aluSqrt(aluDotproduct(Position, Position));
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OrigDist = Distance;
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flAttenuation = 1.0f;
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for(i = 0;i < NumSends;i++)
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{
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RoomAttenuation[i] = 1.0f;
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RoomRolloff[i] = ALSource->RoomRolloffFactor;
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if(ALSource->Send[i].Slot &&
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(ALSource->Send[i].Slot->effect.type == AL_EFFECT_REVERB ||
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ALSource->Send[i].Slot->effect.type == AL_EFFECT_EAXREVERB))
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RoomRolloff[i] += ALSource->Send[i].Slot->effect.Reverb.RoomRolloffFactor;
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}
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switch(ALContext->SourceDistanceModel ? ALSource->DistanceModel :
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ALContext->DistanceModel)
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{
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case AL_INVERSE_DISTANCE_CLAMPED:
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Distance=__max(Distance,MinDist);
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Distance=__min(Distance,MaxDist);
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if(MaxDist < MinDist)
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break;
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//fall-through
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case AL_INVERSE_DISTANCE:
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if(MinDist > 0.0f)
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{
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if((MinDist + (Rolloff * (Distance - MinDist))) > 0.0f)
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flAttenuation = MinDist / (MinDist + (Rolloff * (Distance - MinDist)));
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for(i = 0;i < NumSends;i++)
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{
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if((MinDist + (RoomRolloff[i] * (Distance - MinDist))) > 0.0f)
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RoomAttenuation[i] = MinDist / (MinDist + (RoomRolloff[i] * (Distance - MinDist)));
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}
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}
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break;
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case AL_LINEAR_DISTANCE_CLAMPED:
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Distance=__max(Distance,MinDist);
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Distance=__min(Distance,MaxDist);
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if(MaxDist < MinDist)
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break;
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//fall-through
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case AL_LINEAR_DISTANCE:
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Distance=__min(Distance,MaxDist);
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if(MaxDist != MinDist)
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{
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flAttenuation = 1.0f - (Rolloff*(Distance-MinDist)/(MaxDist - MinDist));
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for(i = 0;i < NumSends;i++)
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RoomAttenuation[i] = 1.0f - (RoomRolloff[i]*(Distance-MinDist)/(MaxDist - MinDist));
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}
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break;
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case AL_EXPONENT_DISTANCE_CLAMPED:
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Distance=__max(Distance,MinDist);
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Distance=__min(Distance,MaxDist);
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if(MaxDist < MinDist)
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break;
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//fall-through
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case AL_EXPONENT_DISTANCE:
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if(Distance > 0.0f && MinDist > 0.0f)
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{
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flAttenuation = aluPow(Distance/MinDist, -Rolloff);
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for(i = 0;i < NumSends;i++)
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RoomAttenuation[i] = aluPow(Distance/MinDist, -RoomRolloff[i]);
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}
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break;
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case AL_NONE:
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break;
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}
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// Source Gain + Attenuation
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DryMix = SourceVolume * flAttenuation;
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for(i = 0;i < NumSends;i++)
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WetGain[i] = SourceVolume * RoomAttenuation[i];
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effectiveDist = 0.0f;
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if(MinDist > 0.0f)
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effectiveDist = (MinDist/flAttenuation - MinDist)*MetersPerUnit;
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// Distance-based air absorption
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if(ALSource->AirAbsorptionFactor > 0.0f && effectiveDist > 0.0f)
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{
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ALfloat absorb;
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// Absorption calculation is done in dB
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absorb = (ALSource->AirAbsorptionFactor*AIRABSORBGAINDBHF) *
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effectiveDist;
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// Convert dB to linear gain before applying
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absorb = aluPow(10.0f, absorb/20.0f);
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DryGainHF *= absorb;
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}
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//3. Apply directional soundcones
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Angle = aluAcos(aluDotproduct(Direction,SourceToListener)) * 180.0f/M_PI;
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if(Angle >= InnerAngle && Angle <= OuterAngle)
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{
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ALfloat scale = (Angle-InnerAngle) / (OuterAngle-InnerAngle);
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ConeVolume = (1.0f+(ALSource->flOuterGain-1.0f)*scale);
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ConeHF = (1.0f+(OuterGainHF-1.0f)*scale);
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}
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else if(Angle > OuterAngle)
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{
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ConeVolume = (1.0f+(ALSource->flOuterGain-1.0f));
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ConeHF = (1.0f+(OuterGainHF-1.0f));
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}
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else
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{
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ConeVolume = 1.0f;
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ConeHF = 1.0f;
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}
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// Apply some high-frequency attenuation for sources behind the listener
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// NOTE: This should be aluDotproduct({0,0,-1}, ListenerToSource), however
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// that is equivalent to aluDotproduct({0,0,1}, SourceToListener), which is
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// the same as SourceToListener[2]
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Angle = aluAcos(SourceToListener[2]) * 180.0f/M_PI;
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// Sources within the minimum distance attenuate less
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if(OrigDist < MinDist)
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Angle *= OrigDist/MinDist;
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if(Angle > 90.0f)
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{
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ALfloat scale = (Angle-90.0f) / (180.1f-90.0f); // .1 to account for fp errors
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ConeHF *= 1.0f - (Device->HeadDampen*scale);
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}
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DryMix *= ConeVolume;
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if(ALSource->DryGainHFAuto)
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DryGainHF *= ConeHF;
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// Clamp to Min/Max Gain
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DryMix = __min(DryMix,MaxVolume);
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DryMix = __max(DryMix,MinVolume);
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for(i = 0;i < NumSends;i++)
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{
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ALeffectslot *Slot = ALSource->Send[i].Slot;
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if(!Slot || Slot->effect.type == AL_EFFECT_NULL)
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{
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ALSource->Params.WetGains[i] = 0.0f;
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WetGainHF[i] = 1.0f;
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continue;
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}
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if(Slot->AuxSendAuto)
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{
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if(ALSource->WetGainAuto)
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WetGain[i] *= ConeVolume;
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if(ALSource->WetGainHFAuto)
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WetGainHF[i] *= ConeHF;
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// Clamp to Min/Max Gain
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WetGain[i] = __min(WetGain[i],MaxVolume);
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WetGain[i] = __max(WetGain[i],MinVolume);
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if(Slot->effect.type == AL_EFFECT_REVERB ||
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Slot->effect.type == AL_EFFECT_EAXREVERB)
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{
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/* Apply a decay-time transformation to the wet path, based on
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* the attenuation of the dry path.
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*
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* Using the approximate (effective) source to listener
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* distance, the initial decay of the reverb effect is
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* calculated and applied to the wet path.
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*/
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WetGain[i] *= aluPow(10.0f, effectiveDist /
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(SPEEDOFSOUNDMETRESPERSEC *
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Slot->effect.Reverb.DecayTime) *
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-60.0 / 20.0);
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WetGainHF[i] *= aluPow(10.0f,
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log10(Slot->effect.Reverb.AirAbsorptionGainHF) *
|
|
ALSource->AirAbsorptionFactor * effectiveDist);
|
|
}
|
|
}
|
|
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 */
|
|
WetGain[i] = DryMix;
|
|
WetGainHF[i] = DryGainHF;
|
|
}
|
|
|
|
switch(ALSource->Send[i].WetFilter.type)
|
|
{
|
|
case AL_FILTER_LOWPASS:
|
|
WetGain[i] *= ALSource->Send[i].WetFilter.Gain;
|
|
WetGainHF[i] *= ALSource->Send[i].WetFilter.GainHF;
|
|
break;
|
|
}
|
|
ALSource->Params.WetGains[i] = WetGain[i] * ListenerGain;
|
|
}
|
|
for(i = NumSends;i < MAX_SENDS;i++)
|
|
{
|
|
ALSource->Params.WetGains[i] = 0.0f;
|
|
WetGainHF[i] = 1.0f;
|
|
}
|
|
|
|
// Apply filter gains and filters
|
|
switch(ALSource->DirectFilter.type)
|
|
{
|
|
case AL_FILTER_LOWPASS:
|
|
DryMix *= ALSource->DirectFilter.Gain;
|
|
DryGainHF *= ALSource->DirectFilter.GainHF;
|
|
break;
|
|
}
|
|
DryMix *= ListenerGain;
|
|
|
|
// Calculate Velocity
|
|
if(DopplerFactor != 0.0f)
|
|
{
|
|
ALfloat flVSS, flVLS;
|
|
ALfloat flMaxVelocity = (DopplerVelocity * flSpeedOfSound) /
|
|
DopplerFactor;
|
|
|
|
flVSS = aluDotproduct(Velocity, SourceToListener);
|
|
if(flVSS >= flMaxVelocity)
|
|
flVSS = (flMaxVelocity - 1.0f);
|
|
else if(flVSS <= -flMaxVelocity)
|
|
flVSS = -flMaxVelocity + 1.0f;
|
|
|
|
flVLS = aluDotproduct(ListenerVel, SourceToListener);
|
|
if(flVLS >= flMaxVelocity)
|
|
flVLS = (flMaxVelocity - 1.0f);
|
|
else if(flVLS <= -flMaxVelocity)
|
|
flVLS = -flMaxVelocity + 1.0f;
|
|
|
|
ALSource->Params.Pitch = ALSource->flPitch *
|
|
((flSpeedOfSound * DopplerVelocity) - (DopplerFactor * flVLS)) /
|
|
((flSpeedOfSound * DopplerVelocity) - (DopplerFactor * flVSS));
|
|
}
|
|
else
|
|
ALSource->Params.Pitch = ALSource->flPitch;
|
|
|
|
// Use energy-preserving panning algorithm for multi-speaker playback
|
|
length = __max(OrigDist, MinDist);
|
|
if(length > 0.0f)
|
|
{
|
|
ALfloat invlen = 1.0f/length;
|
|
Position[0] *= invlen;
|
|
Position[1] *= invlen;
|
|
Position[2] *= invlen;
|
|
}
|
|
|
|
pos = aluCart2LUTpos(-Position[2], Position[0]);
|
|
SpeakerGain = &Device->PanningLUT[OUTPUTCHANNELS * pos];
|
|
|
|
DirGain = aluSqrt(Position[0]*Position[0] + Position[2]*Position[2]);
|
|
// elevation adjustment for directional gain. this sucks, but
|
|
// has low complexity
|
|
AmbientGain = 1.0/aluSqrt(Device->NumChan) * (1.0-DirGain);
|
|
for(s = 0;s < OUTPUTCHANNELS;s++)
|
|
ALSource->Params.DryGains[s] = 0.0f;
|
|
for(s = 0;s < (ALsizei)Device->NumChan;s++)
|
|
{
|
|
Channel chan = Device->Speaker2Chan[s];
|
|
ALfloat gain = SpeakerGain[chan]*DirGain + AmbientGain;
|
|
ALSource->Params.DryGains[chan] = DryMix * gain;
|
|
}
|
|
|
|
/* Update filter coefficients. */
|
|
cw = cos(2.0*M_PI * LOWPASSFREQCUTOFF / Frequency);
|
|
|
|
/* Spatialized sources use four chained one-pole filters, so we need to
|
|
* take the fourth root of the squared gain, which is the same as the
|
|
* square root of the base gain. */
|
|
ALSource->Params.iirFilter.coeff = lpCoeffCalc(aluSqrt(DryGainHF), cw);
|
|
|
|
for(i = 0;i < NumSends;i++)
|
|
{
|
|
/* The wet path uses two chained one-pole filters, so take the
|
|
* base gain (square root of the squared gain) */
|
|
ALSource->Params.Send[i].iirFilter.coeff = lpCoeffCalc(WetGainHF[i], cw);
|
|
}
|
|
}
|
|
|
|
static __inline ALfloat point(ALfloat val1, ALfloat val2, ALint frac)
|
|
{
|
|
return val1;
|
|
(void)val2;
|
|
(void)frac;
|
|
}
|
|
static __inline ALfloat lerp(ALfloat val1, ALfloat val2, ALint frac)
|
|
{
|
|
return val1 + ((val2-val1)*(frac * (1.0f/(1<<FRACTIONBITS))));
|
|
}
|
|
static __inline ALfloat cos_lerp(ALfloat val1, ALfloat val2, ALint frac)
|
|
{
|
|
ALfloat mult = (1.0f-cos(frac * (1.0f/(1<<FRACTIONBITS)) * M_PI)) * 0.5f;
|
|
return val1 + ((val2-val1)*mult);
|
|
}
|
|
|
|
static void MixSource(ALsource *ALSource, ALCcontext *ALContext,
|
|
float (*DryBuffer)[OUTPUTCHANNELS], ALuint SamplesToDo)
|
|
{
|
|
static float DummyBuffer[BUFFERSIZE];
|
|
ALfloat *WetBuffer[MAX_SENDS];
|
|
ALfloat DrySend[OUTPUTCHANNELS];
|
|
ALfloat dryGainStep[OUTPUTCHANNELS];
|
|
ALfloat wetGainStep[MAX_SENDS];
|
|
ALuint i, j, out;
|
|
ALfloat value, outsamp;
|
|
ALbufferlistitem *BufferListItem;
|
|
ALint64 DataSize64,DataPos64;
|
|
FILTER *DryFilter, *WetFilter[MAX_SENDS];
|
|
ALfloat WetSend[MAX_SENDS];
|
|
ALuint rampLength;
|
|
ALboolean DuplicateStereo;
|
|
ALuint DeviceFreq;
|
|
ALint increment;
|
|
ALuint DataPosInt, DataPosFrac;
|
|
ALuint Channels, Bytes;
|
|
ALuint Frequency;
|
|
resampler_t Resampler;
|
|
ALuint BuffersPlayed;
|
|
ALboolean Looping;
|
|
ALfloat Pitch;
|
|
ALenum State;
|
|
|
|
DuplicateStereo = ALContext->Device->DuplicateStereo;
|
|
DeviceFreq = ALContext->Device->Frequency;
|
|
|
|
rampLength = DeviceFreq * MIN_RAMP_LENGTH / 1000;
|
|
rampLength = max(rampLength, SamplesToDo);
|
|
|
|
/* Find buffer format */
|
|
Frequency = 0;
|
|
Channels = 0;
|
|
Bytes = 0;
|
|
BufferListItem = ALSource->queue;
|
|
while(BufferListItem != NULL)
|
|
{
|
|
ALbuffer *ALBuffer;
|
|
if((ALBuffer=BufferListItem->buffer) != NULL)
|
|
{
|
|
Channels = aluChannelsFromFormat(ALBuffer->format);
|
|
Bytes = aluBytesFromFormat(ALBuffer->format);
|
|
Frequency = ALBuffer->frequency;
|
|
break;
|
|
}
|
|
BufferListItem = BufferListItem->next;
|
|
}
|
|
|
|
if(ALSource->NeedsUpdate)
|
|
{
|
|
ALsource_Update(ALSource, ALContext);
|
|
ALSource->NeedsUpdate = AL_FALSE;
|
|
}
|
|
|
|
/* Get source info */
|
|
Resampler = ALSource->Resampler;
|
|
State = ALSource->state;
|
|
BuffersPlayed = ALSource->BuffersPlayed;
|
|
DataPosInt = ALSource->position;
|
|
DataPosFrac = ALSource->position_fraction;
|
|
Looping = ALSource->bLooping;
|
|
|
|
/* Compute 18.14 fixed point step */
|
|
Pitch = (ALSource->Params.Pitch*Frequency) / DeviceFreq;
|
|
if(Pitch > (float)MAX_PITCH) Pitch = (float)MAX_PITCH;
|
|
increment = (ALint)(Pitch*(ALfloat)(1L<<FRACTIONBITS));
|
|
if(increment <= 0) increment = (1<<FRACTIONBITS);
|
|
|
|
if(ALSource->FirstStart)
|
|
{
|
|
for(i = 0;i < OUTPUTCHANNELS;i++)
|
|
DrySend[i] = ALSource->Params.DryGains[i];
|
|
for(i = 0;i < MAX_SENDS;i++)
|
|
WetSend[i] = ALSource->Params.WetGains[i];
|
|
}
|
|
else
|
|
{
|
|
for(i = 0;i < OUTPUTCHANNELS;i++)
|
|
DrySend[i] = ALSource->DryGains[i];
|
|
for(i = 0;i < MAX_SENDS;i++)
|
|
WetSend[i] = ALSource->WetGains[i];
|
|
}
|
|
|
|
DryFilter = &ALSource->Params.iirFilter;
|
|
for(i = 0;i < MAX_SENDS;i++)
|
|
{
|
|
WetFilter[i] = &ALSource->Params.Send[i].iirFilter;
|
|
WetBuffer[i] = (ALSource->Send[i].Slot ?
|
|
ALSource->Send[i].Slot->WetBuffer :
|
|
DummyBuffer);
|
|
}
|
|
|
|
/* Get current buffer queue item */
|
|
BufferListItem = ALSource->queue;
|
|
for(i = 0;i < BuffersPlayed && BufferListItem;i++)
|
|
BufferListItem = BufferListItem->next;
|
|
|
|
j = 0;
|
|
do {
|
|
ALfloat *Data = NULL;
|
|
ALuint LoopStart = 0;
|
|
ALuint LoopEnd = 0;
|
|
ALuint DataSize = 0;
|
|
ALbuffer *ALBuffer;
|
|
ALuint BufferSize;
|
|
|
|
/* Get buffer info */
|
|
if((ALBuffer=BufferListItem->buffer) != NULL)
|
|
{
|
|
Data = ALBuffer->data;
|
|
DataSize = ALBuffer->size;
|
|
DataSize /= Channels * Bytes;
|
|
LoopStart = ALBuffer->LoopStart;
|
|
LoopEnd = ALBuffer->LoopEnd;
|
|
}
|
|
|
|
if(Looping && ALSource->lSourceType == AL_STATIC)
|
|
{
|
|
/* If current offset is beyond the loop range, do not loop */
|
|
if(DataPosInt >= LoopEnd)
|
|
Looping = AL_FALSE;
|
|
}
|
|
if(!Looping || ALSource->lSourceType != AL_STATIC)
|
|
{
|
|
/* Non-looping and non-static sources ignore loop points */
|
|
LoopStart = 0;
|
|
LoopEnd = DataSize;
|
|
}
|
|
|
|
if(DataPosInt >= DataSize)
|
|
goto skipmix;
|
|
|
|
if(BufferListItem->next)
|
|
{
|
|
ALbuffer *NextBuf = BufferListItem->next->buffer;
|
|
if(NextBuf && NextBuf->size)
|
|
{
|
|
ALint ulExtraSamples = BUFFER_PADDING*Channels*Bytes;
|
|
ulExtraSamples = min(NextBuf->size, ulExtraSamples);
|
|
memcpy(&Data[DataSize*Channels], NextBuf->data, ulExtraSamples);
|
|
}
|
|
}
|
|
else if(Looping)
|
|
{
|
|
ALbuffer *NextBuf = ALSource->queue->buffer;
|
|
if(NextBuf && NextBuf->size)
|
|
{
|
|
ALint ulExtraSamples = BUFFER_PADDING*Channels*Bytes;
|
|
ulExtraSamples = min(NextBuf->size, ulExtraSamples);
|
|
memcpy(&Data[DataSize*Channels], &NextBuf->data[LoopStart*Channels], ulExtraSamples);
|
|
}
|
|
}
|
|
else
|
|
memset(&Data[DataSize*Channels], 0, (BUFFER_PADDING*Channels*Bytes));
|
|
|
|
/* Compute the gain steps for each output channel */
|
|
for(i = 0;i < OUTPUTCHANNELS;i++)
|
|
dryGainStep[i] = (ALSource->Params.DryGains[i]-DrySend[i]) /
|
|
rampLength;
|
|
for(i = 0;i < MAX_SENDS;i++)
|
|
wetGainStep[i] = (ALSource->Params.WetGains[i]-WetSend[i]) /
|
|
rampLength;
|
|
|
|
/* Figure out how many samples we can mix. */
|
|
DataSize64 = LoopEnd;
|
|
DataSize64 <<= FRACTIONBITS;
|
|
DataPos64 = DataPosInt;
|
|
DataPos64 <<= FRACTIONBITS;
|
|
DataPos64 += DataPosFrac;
|
|
BufferSize = (ALuint)((DataSize64-DataPos64+(increment-1)) / increment);
|
|
|
|
BufferSize = min(BufferSize, (SamplesToDo-j));
|
|
|
|
/* Actual sample mixing loops */
|
|
if(Channels == 1) /* Mono */
|
|
{
|
|
#define DO_MIX(resampler) do { \
|
|
while(BufferSize--) \
|
|
{ \
|
|
for(i = 0;i < OUTPUTCHANNELS;i++) \
|
|
DrySend[i] += dryGainStep[i]; \
|
|
for(i = 0;i < MAX_SENDS;i++) \
|
|
WetSend[i] += wetGainStep[i]; \
|
|
\
|
|
/* First order interpolator */ \
|
|
value = (resampler)(Data[DataPosInt], Data[DataPosInt+1], \
|
|
DataPosFrac); \
|
|
\
|
|
/* Direct path final mix buffer and panning */ \
|
|
outsamp = lpFilter4P(DryFilter, 0, value); \
|
|
DryBuffer[j][FRONT_LEFT] += outsamp*DrySend[FRONT_LEFT]; \
|
|
DryBuffer[j][FRONT_RIGHT] += outsamp*DrySend[FRONT_RIGHT]; \
|
|
DryBuffer[j][SIDE_LEFT] += outsamp*DrySend[SIDE_LEFT]; \
|
|
DryBuffer[j][SIDE_RIGHT] += outsamp*DrySend[SIDE_RIGHT]; \
|
|
DryBuffer[j][BACK_LEFT] += outsamp*DrySend[BACK_LEFT]; \
|
|
DryBuffer[j][BACK_RIGHT] += outsamp*DrySend[BACK_RIGHT]; \
|
|
DryBuffer[j][FRONT_CENTER] += outsamp*DrySend[FRONT_CENTER]; \
|
|
DryBuffer[j][BACK_CENTER] += outsamp*DrySend[BACK_CENTER]; \
|
|
\
|
|
/* Room path final mix buffer and panning */ \
|
|
for(i = 0;i < MAX_SENDS;i++) \
|
|
{ \
|
|
outsamp = lpFilter2P(WetFilter[i], 0, value); \
|
|
WetBuffer[i][j] += outsamp*WetSend[i]; \
|
|
} \
|
|
\
|
|
DataPosFrac += increment; \
|
|
DataPosInt += DataPosFrac>>FRACTIONBITS; \
|
|
DataPosFrac &= FRACTIONMASK; \
|
|
j++; \
|
|
} \
|
|
} while(0)
|
|
|
|
switch(Resampler)
|
|
{
|
|
case POINT_RESAMPLER:
|
|
DO_MIX(point); break;
|
|
case LINEAR_RESAMPLER:
|
|
DO_MIX(lerp); break;
|
|
case COSINE_RESAMPLER:
|
|
DO_MIX(cos_lerp); break;
|
|
case RESAMPLER_MIN:
|
|
case RESAMPLER_MAX:
|
|
break;
|
|
}
|
|
#undef DO_MIX
|
|
}
|
|
else if(Channels == 2 && DuplicateStereo) /* Stereo */
|
|
{
|
|
const int chans[] = {
|
|
FRONT_LEFT, FRONT_RIGHT
|
|
};
|
|
const int chans2[] = {
|
|
BACK_LEFT, SIDE_LEFT, BACK_RIGHT, SIDE_RIGHT
|
|
};
|
|
const ALfloat dupscaler = aluSqrt(1.0f/3.0f);
|
|
|
|
#define DO_MIX(resampler) do { \
|
|
const ALfloat scaler = 1.0f/Channels; \
|
|
while(BufferSize--) \
|
|
{ \
|
|
for(i = 0;i < OUTPUTCHANNELS;i++) \
|
|
DrySend[i] += dryGainStep[i]; \
|
|
for(i = 0;i < MAX_SENDS;i++) \
|
|
WetSend[i] += wetGainStep[i]; \
|
|
\
|
|
for(i = 0;i < Channels;i++) \
|
|
{ \
|
|
value = (resampler)(Data[DataPosInt*Channels + i], \
|
|
Data[(DataPosInt+1)*Channels + i], \
|
|
DataPosFrac); \
|
|
\
|
|
outsamp = lpFilter2P(DryFilter, chans[i]*2, value) * dupscaler; \
|
|
DryBuffer[j][chans[i]] += outsamp*DrySend[chans[i]]; \
|
|
DryBuffer[j][chans2[i*2+0]] += outsamp*DrySend[chans2[i*2+0]]; \
|
|
DryBuffer[j][chans2[i*2+1]] += outsamp*DrySend[chans2[i*2+1]]; \
|
|
\
|
|
for(out = 0;out < MAX_SENDS;out++) \
|
|
{ \
|
|
outsamp = lpFilter1P(WetFilter[out], chans[i], value); \
|
|
WetBuffer[out][j] += outsamp*WetSend[out]*scaler; \
|
|
} \
|
|
} \
|
|
\
|
|
DataPosFrac += increment; \
|
|
DataPosInt += DataPosFrac>>FRACTIONBITS; \
|
|
DataPosFrac &= FRACTIONMASK; \
|
|
j++; \
|
|
} \
|
|
} while(0)
|
|
|
|
switch(Resampler)
|
|
{
|
|
case POINT_RESAMPLER:
|
|
DO_MIX(point); break;
|
|
case LINEAR_RESAMPLER:
|
|
DO_MIX(lerp); break;
|
|
case COSINE_RESAMPLER:
|
|
DO_MIX(cos_lerp); break;
|
|
case RESAMPLER_MIN:
|
|
case RESAMPLER_MAX:
|
|
break;
|
|
}
|
|
#undef DO_MIX
|
|
}
|
|
else if(Channels == 2) /* Stereo */
|
|
{
|
|
const int chans[] = {
|
|
FRONT_LEFT, FRONT_RIGHT
|
|
};
|
|
|
|
#define DO_MIX(resampler) do { \
|
|
const ALfloat scaler = 1.0f/Channels; \
|
|
while(BufferSize--) \
|
|
{ \
|
|
for(i = 0;i < OUTPUTCHANNELS;i++) \
|
|
DrySend[i] += dryGainStep[i]; \
|
|
for(i = 0;i < MAX_SENDS;i++) \
|
|
WetSend[i] += wetGainStep[i]; \
|
|
\
|
|
for(i = 0;i < Channels;i++) \
|
|
{ \
|
|
value = (resampler)(Data[DataPosInt*Channels + i], \
|
|
Data[(DataPosInt+1)*Channels + i], \
|
|
DataPosFrac); \
|
|
\
|
|
outsamp = lpFilter2P(DryFilter, chans[i]*2, value); \
|
|
DryBuffer[j][chans[i]] += outsamp*DrySend[chans[i]]; \
|
|
\
|
|
for(out = 0;out < MAX_SENDS;out++) \
|
|
{ \
|
|
outsamp = lpFilter1P(WetFilter[out], chans[i], value); \
|
|
WetBuffer[out][j] += outsamp*WetSend[out]*scaler; \
|
|
} \
|
|
} \
|
|
\
|
|
DataPosFrac += increment; \
|
|
DataPosInt += DataPosFrac>>FRACTIONBITS; \
|
|
DataPosFrac &= FRACTIONMASK; \
|
|
j++; \
|
|
} \
|
|
} while(0)
|
|
|
|
switch(Resampler)
|
|
{
|
|
case POINT_RESAMPLER:
|
|
DO_MIX(point); break;
|
|
case LINEAR_RESAMPLER:
|
|
DO_MIX(lerp); break;
|
|
case COSINE_RESAMPLER:
|
|
DO_MIX(cos_lerp); break;
|
|
case RESAMPLER_MIN:
|
|
case RESAMPLER_MAX:
|
|
break;
|
|
}
|
|
}
|
|
else if(Channels == 4) /* Quad */
|
|
{
|
|
const int chans[] = {
|
|
FRONT_LEFT, FRONT_RIGHT,
|
|
BACK_LEFT, BACK_RIGHT
|
|
};
|
|
|
|
switch(Resampler)
|
|
{
|
|
case POINT_RESAMPLER:
|
|
DO_MIX(point); break;
|
|
case LINEAR_RESAMPLER:
|
|
DO_MIX(lerp); break;
|
|
case COSINE_RESAMPLER:
|
|
DO_MIX(cos_lerp); break;
|
|
case RESAMPLER_MIN:
|
|
case RESAMPLER_MAX:
|
|
break;
|
|
}
|
|
}
|
|
else if(Channels == 6) /* 5.1 */
|
|
{
|
|
const int chans[] = {
|
|
FRONT_LEFT, FRONT_RIGHT,
|
|
FRONT_CENTER, LFE,
|
|
BACK_LEFT, BACK_RIGHT
|
|
};
|
|
|
|
switch(Resampler)
|
|
{
|
|
case POINT_RESAMPLER:
|
|
DO_MIX(point); break;
|
|
case LINEAR_RESAMPLER:
|
|
DO_MIX(lerp); break;
|
|
case COSINE_RESAMPLER:
|
|
DO_MIX(cos_lerp); break;
|
|
case RESAMPLER_MIN:
|
|
case RESAMPLER_MAX:
|
|
break;
|
|
}
|
|
}
|
|
else if(Channels == 7) /* 6.1 */
|
|
{
|
|
const int chans[] = {
|
|
FRONT_LEFT, FRONT_RIGHT,
|
|
FRONT_CENTER, LFE,
|
|
BACK_CENTER,
|
|
SIDE_LEFT, SIDE_RIGHT
|
|
};
|
|
|
|
switch(Resampler)
|
|
{
|
|
case POINT_RESAMPLER:
|
|
DO_MIX(point); break;
|
|
case LINEAR_RESAMPLER:
|
|
DO_MIX(lerp); break;
|
|
case COSINE_RESAMPLER:
|
|
DO_MIX(cos_lerp); break;
|
|
case RESAMPLER_MIN:
|
|
case RESAMPLER_MAX:
|
|
break;
|
|
}
|
|
}
|
|
else if(Channels == 8) /* 7.1 */
|
|
{
|
|
const int chans[] = {
|
|
FRONT_LEFT, FRONT_RIGHT,
|
|
FRONT_CENTER, LFE,
|
|
BACK_LEFT, BACK_RIGHT,
|
|
SIDE_LEFT, SIDE_RIGHT
|
|
};
|
|
|
|
switch(Resampler)
|
|
{
|
|
case POINT_RESAMPLER:
|
|
DO_MIX(point); break;
|
|
case LINEAR_RESAMPLER:
|
|
DO_MIX(lerp); break;
|
|
case COSINE_RESAMPLER:
|
|
DO_MIX(cos_lerp); break;
|
|
case RESAMPLER_MIN:
|
|
case RESAMPLER_MAX:
|
|
break;
|
|
}
|
|
#undef DO_MIX
|
|
}
|
|
else /* Unknown? */
|
|
{
|
|
for(i = 0;i < OUTPUTCHANNELS;i++)
|
|
DrySend[i] += dryGainStep[i]*BufferSize;
|
|
for(i = 0;i < MAX_SENDS;i++)
|
|
WetSend[i] += wetGainStep[i]*BufferSize;
|
|
while(BufferSize--)
|
|
{
|
|
DataPosFrac += increment;
|
|
DataPosInt += DataPosFrac>>FRACTIONBITS;
|
|
DataPosFrac &= FRACTIONMASK;
|
|
j++;
|
|
}
|
|
}
|
|
|
|
skipmix:
|
|
/* Handle looping sources */
|
|
if(DataPosInt >= LoopEnd)
|
|
{
|
|
if(BuffersPlayed < (ALSource->BuffersInQueue-1))
|
|
{
|
|
BufferListItem = BufferListItem->next;
|
|
BuffersPlayed++;
|
|
DataPosInt -= DataSize;
|
|
}
|
|
else if(Looping)
|
|
{
|
|
BufferListItem = ALSource->queue;
|
|
BuffersPlayed = 0;
|
|
if(ALSource->lSourceType == AL_STATIC)
|
|
DataPosInt = ((DataPosInt-LoopStart)%(LoopEnd-LoopStart)) + LoopStart;
|
|
else
|
|
DataPosInt -= DataSize;
|
|
}
|
|
else
|
|
{
|
|
State = AL_STOPPED;
|
|
BufferListItem = ALSource->queue;
|
|
BuffersPlayed = ALSource->BuffersInQueue;
|
|
DataPosInt = 0;
|
|
DataPosFrac = 0;
|
|
}
|
|
}
|
|
} while(State == AL_PLAYING && j < SamplesToDo);
|
|
|
|
/* Update source info */
|
|
ALSource->state = State;
|
|
ALSource->BuffersPlayed = BuffersPlayed;
|
|
ALSource->position = DataPosInt;
|
|
ALSource->position_fraction = DataPosFrac;
|
|
ALSource->Buffer = BufferListItem->buffer;
|
|
|
|
for(i = 0;i < OUTPUTCHANNELS;i++)
|
|
ALSource->DryGains[i] = DrySend[i];
|
|
for(i = 0;i < MAX_SENDS;i++)
|
|
ALSource->WetGains[i] = WetSend[i];
|
|
|
|
ALSource->FirstStart = AL_FALSE;
|
|
}
|
|
|
|
ALvoid aluMixData(ALCdevice *device, ALvoid *buffer, ALsizei size)
|
|
{
|
|
float (*DryBuffer)[OUTPUTCHANNELS];
|
|
ALfloat (*Matrix)[OUTPUTCHANNELS];
|
|
const ALuint *ChanMap;
|
|
ALuint SamplesToDo;
|
|
ALeffectslot *ALEffectSlot;
|
|
ALCcontext *ALContext;
|
|
ALfloat samp;
|
|
int fpuState;
|
|
ALuint i, j, c;
|
|
ALsizei e, s;
|
|
|
|
#if defined(HAVE_FESETROUND)
|
|
fpuState = fegetround();
|
|
fesetround(FE_TOWARDZERO);
|
|
#elif defined(HAVE__CONTROLFP)
|
|
fpuState = _controlfp(0, 0);
|
|
_controlfp(_RC_CHOP, _MCW_RC);
|
|
#else
|
|
(void)fpuState;
|
|
#endif
|
|
|
|
DryBuffer = device->DryBuffer;
|
|
while(size > 0)
|
|
{
|
|
/* Setup variables */
|
|
SamplesToDo = min(size, BUFFERSIZE);
|
|
|
|
/* Clear mixing buffer */
|
|
memset(DryBuffer, 0, SamplesToDo*OUTPUTCHANNELS*sizeof(ALfloat));
|
|
|
|
SuspendContext(NULL);
|
|
for(c = 0;c < device->NumContexts;c++)
|
|
{
|
|
ALContext = device->Contexts[c];
|
|
SuspendContext(ALContext);
|
|
|
|
s = 0;
|
|
while(s < ALContext->ActiveSourceCount)
|
|
{
|
|
ALsource *Source = ALContext->ActiveSources[s];
|
|
if(Source->state != AL_PLAYING)
|
|
{
|
|
ALsizei end = --(ALContext->ActiveSourceCount);
|
|
ALContext->ActiveSources[s] = ALContext->ActiveSources[end];
|
|
continue;
|
|
}
|
|
MixSource(Source, ALContext, DryBuffer, SamplesToDo);
|
|
s++;
|
|
}
|
|
|
|
/* effect slot processing */
|
|
for(e = 0;e < ALContext->EffectSlotMap.size;e++)
|
|
{
|
|
ALEffectSlot = ALContext->EffectSlotMap.array[e].value;
|
|
if(ALEffectSlot->EffectState)
|
|
ALEffect_Process(ALEffectSlot->EffectState, ALEffectSlot, SamplesToDo, ALEffectSlot->WetBuffer, DryBuffer);
|
|
|
|
for(i = 0;i < SamplesToDo;i++)
|
|
ALEffectSlot->WetBuffer[i] = 0.0f;
|
|
}
|
|
ProcessContext(ALContext);
|
|
}
|
|
device->SamplesPlayed += SamplesToDo;
|
|
ProcessContext(NULL);
|
|
|
|
//Post processing loop
|
|
ChanMap = device->DevChannels;
|
|
Matrix = device->ChannelMatrix;
|
|
switch(device->Format)
|
|
{
|
|
#define CHECK_WRITE_FORMAT(bits, type, func) \
|
|
case AL_FORMAT_MONO##bits: \
|
|
for(i = 0;i < SamplesToDo;i++) \
|
|
{ \
|
|
samp = 0.0f; \
|
|
for(c = 0;c < OUTPUTCHANNELS;c++) \
|
|
samp += DryBuffer[i][c] * Matrix[c][FRONT_CENTER]; \
|
|
((type*)buffer)[ChanMap[FRONT_CENTER]] = (func)(samp); \
|
|
buffer = ((type*)buffer) + 1; \
|
|
} \
|
|
break; \
|
|
case AL_FORMAT_STEREO##bits: \
|
|
if(device->Bs2b) \
|
|
{ \
|
|
for(i = 0;i < SamplesToDo;i++) \
|
|
{ \
|
|
float samples[2] = { 0.0f, 0.0f }; \
|
|
for(c = 0;c < OUTPUTCHANNELS;c++) \
|
|
{ \
|
|
samples[0] += DryBuffer[i][c]*Matrix[c][FRONT_LEFT]; \
|
|
samples[1] += DryBuffer[i][c]*Matrix[c][FRONT_RIGHT]; \
|
|
} \
|
|
bs2b_cross_feed(device->Bs2b, samples); \
|
|
((type*)buffer)[ChanMap[FRONT_LEFT]] = (func)(samples[0]);\
|
|
((type*)buffer)[ChanMap[FRONT_RIGHT]]= (func)(samples[1]);\
|
|
buffer = ((type*)buffer) + 2; \
|
|
} \
|
|
} \
|
|
else \
|
|
{ \
|
|
for(i = 0;i < SamplesToDo;i++) \
|
|
{ \
|
|
static const Channel chans[] = { \
|
|
FRONT_LEFT, FRONT_RIGHT \
|
|
}; \
|
|
for(j = 0;j < 2;j++) \
|
|
{ \
|
|
samp = 0.0f; \
|
|
for(c = 0;c < OUTPUTCHANNELS;c++) \
|
|
samp += DryBuffer[i][c] * Matrix[c][chans[j]]; \
|
|
((type*)buffer)[ChanMap[chans[j]]] = (func)(samp); \
|
|
} \
|
|
buffer = ((type*)buffer) + 2; \
|
|
} \
|
|
} \
|
|
break; \
|
|
case AL_FORMAT_QUAD##bits: \
|
|
for(i = 0;i < SamplesToDo;i++) \
|
|
{ \
|
|
static const Channel chans[] = { \
|
|
FRONT_LEFT, FRONT_RIGHT, \
|
|
BACK_LEFT, BACK_RIGHT, \
|
|
}; \
|
|
for(j = 0;j < 4;j++) \
|
|
{ \
|
|
samp = 0.0f; \
|
|
for(c = 0;c < OUTPUTCHANNELS;c++) \
|
|
samp += DryBuffer[i][c] * Matrix[c][chans[j]]; \
|
|
((type*)buffer)[ChanMap[chans[j]]] = (func)(samp); \
|
|
} \
|
|
buffer = ((type*)buffer) + 4; \
|
|
} \
|
|
break; \
|
|
case AL_FORMAT_51CHN##bits: \
|
|
for(i = 0;i < SamplesToDo;i++) \
|
|
{ \
|
|
static const Channel chans[] = { \
|
|
FRONT_LEFT, FRONT_RIGHT, \
|
|
FRONT_CENTER, LFE, \
|
|
BACK_LEFT, BACK_RIGHT, \
|
|
}; \
|
|
for(j = 0;j < 6;j++) \
|
|
{ \
|
|
samp = 0.0f; \
|
|
for(c = 0;c < OUTPUTCHANNELS;c++) \
|
|
samp += DryBuffer[i][c] * Matrix[c][chans[j]]; \
|
|
((type*)buffer)[ChanMap[chans[j]]] = (func)(samp); \
|
|
} \
|
|
buffer = ((type*)buffer) + 6; \
|
|
} \
|
|
break; \
|
|
case AL_FORMAT_61CHN##bits: \
|
|
for(i = 0;i < SamplesToDo;i++) \
|
|
{ \
|
|
static const Channel chans[] = { \
|
|
FRONT_LEFT, FRONT_RIGHT, \
|
|
FRONT_CENTER, LFE, BACK_CENTER, \
|
|
SIDE_LEFT, SIDE_RIGHT, \
|
|
}; \
|
|
for(j = 0;j < 7;j++) \
|
|
{ \
|
|
samp = 0.0f; \
|
|
for(c = 0;c < OUTPUTCHANNELS;c++) \
|
|
samp += DryBuffer[i][c] * Matrix[c][chans[j]]; \
|
|
((type*)buffer)[ChanMap[chans[j]]] = (func)(samp); \
|
|
} \
|
|
buffer = ((type*)buffer) + 7; \
|
|
} \
|
|
break; \
|
|
case AL_FORMAT_71CHN##bits: \
|
|
for(i = 0;i < SamplesToDo;i++) \
|
|
{ \
|
|
static const Channel chans[] = { \
|
|
FRONT_LEFT, FRONT_RIGHT, \
|
|
FRONT_CENTER, LFE, \
|
|
BACK_LEFT, BACK_RIGHT, \
|
|
SIDE_LEFT, SIDE_RIGHT \
|
|
}; \
|
|
for(j = 0;j < 8;j++) \
|
|
{ \
|
|
samp = 0.0f; \
|
|
for(c = 0;c < OUTPUTCHANNELS;c++) \
|
|
samp += DryBuffer[i][c] * Matrix[c][chans[j]]; \
|
|
((type*)buffer)[ChanMap[chans[j]]] = (func)(samp); \
|
|
} \
|
|
buffer = ((type*)buffer) + 8; \
|
|
} \
|
|
break;
|
|
|
|
#define AL_FORMAT_MONO32 AL_FORMAT_MONO_FLOAT32
|
|
#define AL_FORMAT_STEREO32 AL_FORMAT_STEREO_FLOAT32
|
|
CHECK_WRITE_FORMAT(8, ALubyte, aluF2UB)
|
|
CHECK_WRITE_FORMAT(16, ALshort, aluF2S)
|
|
CHECK_WRITE_FORMAT(32, ALfloat, aluF2F)
|
|
#undef AL_FORMAT_STEREO32
|
|
#undef AL_FORMAT_MONO32
|
|
#undef CHECK_WRITE_FORMAT
|
|
|
|
default:
|
|
break;
|
|
}
|
|
|
|
size -= SamplesToDo;
|
|
}
|
|
|
|
#if defined(HAVE_FESETROUND)
|
|
fesetround(fpuState);
|
|
#elif defined(HAVE__CONTROLFP)
|
|
_controlfp(fpuState, 0xfffff);
|
|
#endif
|
|
}
|
|
|
|
ALvoid aluHandleDisconnect(ALCdevice *device)
|
|
{
|
|
ALuint i;
|
|
|
|
SuspendContext(NULL);
|
|
for(i = 0;i < device->NumContexts;i++)
|
|
{
|
|
ALCcontext *Context = device->Contexts[i];
|
|
ALsource *source;
|
|
ALsizei pos;
|
|
|
|
SuspendContext(Context);
|
|
|
|
for(pos = 0;pos < Context->SourceMap.size;pos++)
|
|
{
|
|
source = Context->SourceMap.array[pos].value;
|
|
if(source->state == AL_PLAYING)
|
|
{
|
|
source->state = AL_STOPPED;
|
|
source->BuffersPlayed = source->BuffersInQueue;
|
|
source->position = 0;
|
|
source->position_fraction = 0;
|
|
}
|
|
}
|
|
ProcessContext(Context);
|
|
}
|
|
|
|
device->Connected = ALC_FALSE;
|
|
ProcessContext(NULL);
|
|
}
|