10a9753183
All previous versions of the library treated the source cone angles as half angles, which is contrary to the spec. Setting the __ALSOFT_HALF_ANGLE_CONES environment variable to "true" or "1" restores the buggy behavior for compatibility with applications that expect it. This is not a config file option because new apps should not be made to depend on the old behavior.
1031 lines
37 KiB
C
1031 lines
37 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 "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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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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ALbufferlistitem *BufferListItem;
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enum DevFmtChannels DevChans;
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enum FmtChannels Channels;
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ALfloat SrcMatrix[MAXCHANNELS][MAXCHANNELS];
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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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ALboolean DupStereo;
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ALfloat Pitch;
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ALfloat cw;
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ALint i;
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/* Get device properties */
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DevChans = ALContext->Device->FmtChans;
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DupStereo = ALContext->Device->DuplicateStereo;
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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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Pitch = ALSource->flPitch;
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/* Calculate the stepping value */
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Channels = FmtMono;
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BufferListItem = ALSource->queue;
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while(BufferListItem != NULL)
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{
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ALbuffer *ALBuffer;
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if((ALBuffer=BufferListItem->buffer) != NULL)
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{
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ALint maxstep = STACK_DATA_SIZE / FrameSizeFromFmt(ALBuffer->FmtChannels,
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ALBuffer->FmtType);
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maxstep -= ResamplerPadding[ALSource->Resampler] +
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ResamplerPrePadding[ALSource->Resampler] + 1;
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maxstep = min(maxstep, INT_MAX>>FRACTIONBITS);
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Pitch = Pitch * ALBuffer->Frequency / Frequency;
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if(Pitch > (ALfloat)maxstep)
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ALSource->Params.Step = maxstep<<FRACTIONBITS;
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else
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{
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ALSource->Params.Step = Pitch*FRACTIONONE;
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if(ALSource->Params.Step == 0)
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ALSource->Params.Step = 1;
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}
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Channels = ALBuffer->FmtChannels;
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break;
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}
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BufferListItem = BufferListItem->next;
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}
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/* Calculate gains */
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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 < MAXCHANNELS;i++)
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{
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ALuint i2;
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for(i2 = 0;i2 < MAXCHANNELS;i2++)
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SrcMatrix[i][i2] = 0.0f;
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}
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switch(Channels)
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{
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case FmtMono:
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SrcMatrix[0][FRONT_CENTER] = DryGain * ListenerGain;
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break;
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case FmtStereo:
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if(DupStereo == AL_FALSE)
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{
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SrcMatrix[0][FRONT_LEFT] = DryGain * ListenerGain;
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SrcMatrix[1][FRONT_RIGHT] = DryGain * ListenerGain;
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}
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else
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{
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switch(DevChans)
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{
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case DevFmtMono:
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case DevFmtStereo:
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SrcMatrix[0][FRONT_LEFT] = DryGain * ListenerGain;
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SrcMatrix[1][FRONT_RIGHT] = DryGain * ListenerGain;
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break;
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case DevFmtQuad:
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case DevFmtX51:
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DryGain *= aluSqrt(2.0f/4.0f);
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SrcMatrix[0][FRONT_LEFT] = DryGain * ListenerGain;
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SrcMatrix[1][FRONT_RIGHT] = DryGain * ListenerGain;
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SrcMatrix[0][BACK_LEFT] = DryGain * ListenerGain;
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SrcMatrix[1][BACK_RIGHT] = DryGain * ListenerGain;
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break;
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case DevFmtX61:
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DryGain *= aluSqrt(2.0f/4.0f);
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SrcMatrix[0][FRONT_LEFT] = DryGain * ListenerGain;
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SrcMatrix[1][FRONT_RIGHT] = DryGain * ListenerGain;
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SrcMatrix[0][SIDE_LEFT] = DryGain * ListenerGain;
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SrcMatrix[1][SIDE_RIGHT] = DryGain * ListenerGain;
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break;
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case DevFmtX71:
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DryGain *= aluSqrt(2.0f/6.0f);
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SrcMatrix[0][FRONT_LEFT] = DryGain * ListenerGain;
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SrcMatrix[1][FRONT_RIGHT] = DryGain * ListenerGain;
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SrcMatrix[0][BACK_LEFT] = DryGain * ListenerGain;
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SrcMatrix[1][BACK_RIGHT] = DryGain * ListenerGain;
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SrcMatrix[0][SIDE_LEFT] = DryGain * ListenerGain;
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SrcMatrix[1][SIDE_RIGHT] = DryGain * ListenerGain;
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break;
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}
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}
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break;
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case FmtRear:
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SrcMatrix[0][BACK_LEFT] = DryGain * ListenerGain;
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SrcMatrix[1][BACK_RIGHT] = DryGain * ListenerGain;
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break;
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case FmtQuad:
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SrcMatrix[0][FRONT_LEFT] = DryGain * ListenerGain;
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SrcMatrix[1][FRONT_RIGHT] = DryGain * ListenerGain;
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SrcMatrix[2][BACK_LEFT] = DryGain * ListenerGain;
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SrcMatrix[3][BACK_RIGHT] = DryGain * ListenerGain;
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break;
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case FmtX51:
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SrcMatrix[0][FRONT_LEFT] = DryGain * ListenerGain;
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SrcMatrix[1][FRONT_RIGHT] = DryGain * ListenerGain;
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SrcMatrix[2][FRONT_CENTER] = DryGain * ListenerGain;
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SrcMatrix[3][LFE] = DryGain * ListenerGain;
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SrcMatrix[4][BACK_LEFT] = DryGain * ListenerGain;
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SrcMatrix[5][BACK_RIGHT] = DryGain * ListenerGain;
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break;
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case FmtX61:
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SrcMatrix[0][FRONT_LEFT] = DryGain * ListenerGain;
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SrcMatrix[1][FRONT_RIGHT] = DryGain * ListenerGain;
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SrcMatrix[2][FRONT_CENTER] = DryGain * ListenerGain;
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SrcMatrix[3][LFE] = DryGain * ListenerGain;
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SrcMatrix[4][BACK_CENTER] = DryGain * ListenerGain;
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SrcMatrix[5][SIDE_LEFT] = DryGain * ListenerGain;
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SrcMatrix[6][SIDE_RIGHT] = DryGain * ListenerGain;
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break;
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case FmtX71:
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SrcMatrix[0][FRONT_LEFT] = DryGain * ListenerGain;
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SrcMatrix[1][FRONT_RIGHT] = DryGain * ListenerGain;
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SrcMatrix[2][FRONT_CENTER] = DryGain * ListenerGain;
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SrcMatrix[3][LFE] = DryGain * ListenerGain;
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SrcMatrix[4][BACK_LEFT] = DryGain * ListenerGain;
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SrcMatrix[5][BACK_RIGHT] = DryGain * ListenerGain;
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SrcMatrix[6][SIDE_LEFT] = DryGain * ListenerGain;
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SrcMatrix[7][SIDE_RIGHT] = DryGain * ListenerGain;
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break;
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}
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for(i = 0;i < MAXCHANNELS;i++)
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{
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ALuint j, k;
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for(j = 0;j < MAXCHANNELS;j++)
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{
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ALfloat (*DevMatrix)[MAXCHANNELS] = ALContext->Device->ChannelMatrix;
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ALSource->Params.DryGains[i][j] = 0.0f;
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for(k = 0;k < MAXCHANNELS;k++)
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{
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/* Matrix mult: O[i][j] += A[i][k] * B[k][j]
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* However, our device matrix is transposed, so we do:
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* O[i][j] += A[i][k] * B[j][k]
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*/
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ALSource->Params.DryGains[i][j] += SrcMatrix[i][k] * DevMatrix[j][k];
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}
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}
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}
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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.Send[i].WetGain = WetGain[i] * ListenerGain;
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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,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, SpeedOfSound;
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ALfloat AirAbsorptionFactor;
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ALbufferlistitem *BufferListItem;
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ALfloat Attenuation, 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 DryGain;
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ALfloat DryGainHF;
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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 Pitch;
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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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DryGainHF = 1.0f;
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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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SpeedOfSound = 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 * ConeScale;
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OuterAngle = ALSource->flOuterAngle * ConeScale;
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OuterGainHF = ALSource->OuterGainHF;
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AirAbsorptionFactor = ALSource->AirAbsorptionFactor;
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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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ALfloat Matrix[4][4];
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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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Attenuation = 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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Attenuation = 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;
|
|
|
|
case AL_LINEAR_DISTANCE_CLAMPED:
|
|
Distance=__max(Distance,MinDist);
|
|
Distance=__min(Distance,MaxDist);
|
|
if(MaxDist < MinDist)
|
|
break;
|
|
//fall-through
|
|
case AL_LINEAR_DISTANCE:
|
|
if(MaxDist != MinDist)
|
|
{
|
|
Attenuation = 1.0f - (Rolloff*(Distance-MinDist)/(MaxDist - MinDist));
|
|
Attenuation = __max(Attenuation, 0.0f);
|
|
for(i = 0;i < NumSends;i++)
|
|
{
|
|
RoomAttenuation[i] = 1.0f - (RoomRolloff[i]*(Distance-MinDist)/(MaxDist - MinDist));
|
|
RoomAttenuation[i] = __max(RoomAttenuation[i], 0.0f);
|
|
}
|
|
}
|
|
break;
|
|
|
|
case AL_EXPONENT_DISTANCE_CLAMPED:
|
|
Distance=__max(Distance,MinDist);
|
|
Distance=__min(Distance,MaxDist);
|
|
if(MaxDist < MinDist)
|
|
break;
|
|
//fall-through
|
|
case AL_EXPONENT_DISTANCE:
|
|
if(Distance > 0.0f && MinDist > 0.0f)
|
|
{
|
|
Attenuation = aluPow(Distance/MinDist, -Rolloff);
|
|
for(i = 0;i < NumSends;i++)
|
|
RoomAttenuation[i] = aluPow(Distance/MinDist, -RoomRolloff[i]);
|
|
}
|
|
break;
|
|
|
|
case AL_NONE:
|
|
break;
|
|
}
|
|
|
|
// Source Gain + Attenuation
|
|
DryGain = SourceVolume * Attenuation;
|
|
for(i = 0;i < NumSends;i++)
|
|
WetGain[i] = SourceVolume * RoomAttenuation[i];
|
|
|
|
EffectiveDist = 0.0f;
|
|
if(MinDist > 0.0f && Attenuation < 1.0f)
|
|
EffectiveDist = (MinDist/Attenuation - MinDist)*MetersPerUnit;
|
|
|
|
// Distance-based air absorption
|
|
if(AirAbsorptionFactor > 0.0f && EffectiveDist > 0.0f)
|
|
{
|
|
ALfloat absorb;
|
|
|
|
// Absorption calculation is done in dB
|
|
absorb = (AirAbsorptionFactor*AIRABSORBGAINDBHF) *
|
|
EffectiveDist;
|
|
// Convert dB to linear gain before applying
|
|
absorb = aluPow(10.0f, absorb/20.0f);
|
|
|
|
DryGainHF *= absorb;
|
|
}
|
|
|
|
//3. Apply directional soundcones
|
|
Angle = aluAcos(aluDotproduct(Direction,SourceToListener)) * 180.0/M_PI;
|
|
if(Angle >= InnerAngle && Angle <= OuterAngle)
|
|
{
|
|
ALfloat scale = (Angle-InnerAngle) / (OuterAngle-InnerAngle);
|
|
ConeVolume = (1.0f+(ALSource->flOuterGain-1.0f)*scale);
|
|
ConeHF = (1.0f+(OuterGainHF-1.0f)*scale);
|
|
}
|
|
else if(Angle > OuterAngle)
|
|
{
|
|
ConeVolume = (1.0f+(ALSource->flOuterGain-1.0f));
|
|
ConeHF = (1.0f+(OuterGainHF-1.0f));
|
|
}
|
|
else
|
|
{
|
|
ConeVolume = 1.0f;
|
|
ConeHF = 1.0f;
|
|
}
|
|
|
|
// Apply some high-frequency attenuation for sources behind the listener
|
|
// NOTE: This should be aluDotproduct({0,0,-1}, ListenerToSource), however
|
|
// that is equivalent to aluDotproduct({0,0,1}, SourceToListener), which is
|
|
// the same as SourceToListener[2]
|
|
Angle = aluAcos(SourceToListener[2]) * 180.0f/M_PI;
|
|
// Sources within the minimum distance attenuate less
|
|
if(OrigDist < MinDist)
|
|
Angle *= OrigDist/MinDist;
|
|
if(Angle > 90.0f)
|
|
{
|
|
ALfloat scale = (Angle-90.0f) / (180.1f-90.0f); // .1 to account for fp errors
|
|
ConeHF *= 1.0f - (Device->HeadDampen*scale);
|
|
}
|
|
|
|
DryGain *= ConeVolume;
|
|
if(ALSource->DryGainHFAuto)
|
|
DryGainHF *= ConeHF;
|
|
|
|
// Clamp to Min/Max Gain
|
|
DryGain = __min(DryGain,MaxVolume);
|
|
DryGain = __max(DryGain,MinVolume);
|
|
|
|
for(i = 0;i < NumSends;i++)
|
|
{
|
|
ALeffectslot *Slot = ALSource->Send[i].Slot;
|
|
|
|
if(!Slot || Slot->effect.type == AL_EFFECT_NULL)
|
|
{
|
|
ALSource->Params.Send[i].WetGain = 0.0f;
|
|
WetGainHF[i] = 1.0f;
|
|
continue;
|
|
}
|
|
|
|
if(Slot->AuxSendAuto)
|
|
{
|
|
if(ALSource->WetGainAuto)
|
|
WetGain[i] *= ConeVolume;
|
|
if(ALSource->WetGainHFAuto)
|
|
WetGainHF[i] *= ConeHF;
|
|
|
|
// Clamp to Min/Max Gain
|
|
WetGain[i] = __min(WetGain[i],MaxVolume);
|
|
WetGain[i] = __max(WetGain[i],MinVolume);
|
|
|
|
if(Slot->effect.type == AL_EFFECT_REVERB ||
|
|
Slot->effect.type == AL_EFFECT_EAXREVERB)
|
|
{
|
|
/* Apply a decay-time transformation to the wet path, based on
|
|
* the attenuation of the dry path.
|
|
*
|
|
* Using the approximate (effective) source to listener
|
|
* distance, the initial decay of the reverb effect is
|
|
* calculated and applied to the wet path.
|
|
*/
|
|
WetGain[i] *= aluPow(10.0f, EffectiveDist /
|
|
(SPEEDOFSOUNDMETRESPERSEC *
|
|
Slot->effect.Reverb.DecayTime) *
|
|
-60.0 / 20.0);
|
|
|
|
WetGainHF[i] *= aluPow(Slot->effect.Reverb.AirAbsorptionGainHF,
|
|
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] = DryGain;
|
|
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.Send[i].WetGain = WetGain[i] * ListenerGain;
|
|
}
|
|
|
|
// Apply filter gains and filters
|
|
switch(ALSource->DirectFilter.type)
|
|
{
|
|
case AL_FILTER_LOWPASS:
|
|
DryGain *= ALSource->DirectFilter.Gain;
|
|
DryGainHF *= ALSource->DirectFilter.GainHF;
|
|
break;
|
|
}
|
|
DryGain *= ListenerGain;
|
|
|
|
// Calculate Velocity
|
|
Pitch = ALSource->flPitch;
|
|
if(DopplerFactor != 0.0f)
|
|
{
|
|
ALfloat VSS, VLS;
|
|
ALfloat MaxVelocity = (SpeedOfSound*DopplerVelocity) /
|
|
DopplerFactor;
|
|
|
|
VSS = aluDotproduct(Velocity, SourceToListener);
|
|
if(VSS >= MaxVelocity)
|
|
VSS = (MaxVelocity - 1.0f);
|
|
else if(VSS <= -MaxVelocity)
|
|
VSS = -MaxVelocity + 1.0f;
|
|
|
|
VLS = aluDotproduct(ListenerVel, SourceToListener);
|
|
if(VLS >= MaxVelocity)
|
|
VLS = (MaxVelocity - 1.0f);
|
|
else if(VLS <= -MaxVelocity)
|
|
VLS = -MaxVelocity + 1.0f;
|
|
|
|
Pitch *= ((SpeedOfSound*DopplerVelocity) - (DopplerFactor*VLS)) /
|
|
((SpeedOfSound*DopplerVelocity) - (DopplerFactor*VSS));
|
|
}
|
|
|
|
BufferListItem = ALSource->queue;
|
|
while(BufferListItem != NULL)
|
|
{
|
|
ALbuffer *ALBuffer;
|
|
if((ALBuffer=BufferListItem->buffer) != NULL)
|
|
{
|
|
ALint maxstep = STACK_DATA_SIZE / FrameSizeFromFmt(ALBuffer->FmtChannels,
|
|
ALBuffer->FmtType);
|
|
maxstep -= ResamplerPadding[ALSource->Resampler] +
|
|
ResamplerPrePadding[ALSource->Resampler] + 1;
|
|
maxstep = min(maxstep, INT_MAX>>FRACTIONBITS);
|
|
|
|
Pitch = Pitch * ALBuffer->Frequency / Frequency;
|
|
if(Pitch > (ALfloat)maxstep)
|
|
ALSource->Params.Step = maxstep<<FRACTIONBITS;
|
|
else
|
|
{
|
|
ALSource->Params.Step = Pitch*FRACTIONONE;
|
|
if(ALSource->Params.Step == 0)
|
|
ALSource->Params.Step = 1;
|
|
}
|
|
break;
|
|
}
|
|
BufferListItem = BufferListItem->next;
|
|
}
|
|
|
|
// 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[MAXCHANNELS * pos];
|
|
|
|
DirGain = aluSqrt(Position[0]*Position[0] + Position[2]*Position[2]);
|
|
// elevation adjustment for directional gain. this sucks, but
|
|
// has low complexity
|
|
AmbientGain = aluSqrt(1.0/Device->NumChan);
|
|
for(s = 0;s < MAXCHANNELS;s++)
|
|
{
|
|
ALuint s2;
|
|
for(s2 = 0;s2 < MAXCHANNELS;s2++)
|
|
ALSource->Params.DryGains[s][s2] = 0.0f;
|
|
}
|
|
for(s = 0;s < (ALsizei)Device->NumChan;s++)
|
|
{
|
|
Channel chan = Device->Speaker2Chan[s];
|
|
ALfloat gain = AmbientGain + (SpeakerGain[chan]-AmbientGain)*DirGain;
|
|
ALSource->Params.DryGains[0][chan] = DryGain * 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 aluF2F(ALfloat val)
|
|
{
|
|
return val;
|
|
}
|
|
static __inline ALushort aluF2US(ALfloat val)
|
|
{
|
|
if(val > 1.0f) return 65535;
|
|
if(val < -1.0f) return 0;
|
|
return (ALint)(val*32767.0f) + 32768;
|
|
}
|
|
static __inline ALshort aluF2S(ALfloat val)
|
|
{
|
|
if(val > 1.0f) return 32767;
|
|
if(val < -1.0f) return -32768;
|
|
return (ALint)(val*32767.0f);
|
|
}
|
|
static __inline ALubyte aluF2UB(ALfloat val)
|
|
{
|
|
ALushort i = aluF2US(val);
|
|
return i>>8;
|
|
}
|
|
static __inline ALbyte aluF2B(ALfloat val)
|
|
{
|
|
ALshort i = aluF2S(val);
|
|
return i>>8;
|
|
}
|
|
|
|
static const Channel MonoChans[] = { FRONT_CENTER };
|
|
static const Channel StereoChans[] = { FRONT_LEFT, FRONT_RIGHT };
|
|
static const Channel QuadChans[] = { FRONT_LEFT, FRONT_RIGHT,
|
|
BACK_LEFT, BACK_RIGHT };
|
|
static const Channel X51Chans[] = { FRONT_LEFT, FRONT_RIGHT,
|
|
FRONT_CENTER, LFE,
|
|
BACK_LEFT, BACK_RIGHT };
|
|
static const Channel X61Chans[] = { FRONT_LEFT, FRONT_LEFT,
|
|
FRONT_CENTER, LFE, BACK_CENTER,
|
|
SIDE_LEFT, SIDE_RIGHT };
|
|
static const Channel X71Chans[] = { FRONT_LEFT, FRONT_RIGHT,
|
|
FRONT_CENTER, LFE,
|
|
BACK_LEFT, BACK_RIGHT,
|
|
SIDE_LEFT, SIDE_RIGHT };
|
|
|
|
#define DECL_TEMPLATE(T, chans,N, func) \
|
|
static void Write_##T##_##chans(ALCdevice *device, T *buffer, ALuint SamplesToDo)\
|
|
{ \
|
|
ALfloat (*DryBuffer)[MAXCHANNELS] = device->DryBuffer; \
|
|
const ALuint *ChanMap = device->DevChannels; \
|
|
ALuint i, j; \
|
|
\
|
|
for(i = 0;i < SamplesToDo;i++) \
|
|
{ \
|
|
for(j = 0;j < N;j++) \
|
|
buffer[ChanMap[chans[j]]] = func(DryBuffer[i][chans[j]]); \
|
|
buffer += N; \
|
|
} \
|
|
}
|
|
|
|
DECL_TEMPLATE(ALfloat, MonoChans,1, aluF2F)
|
|
DECL_TEMPLATE(ALfloat, QuadChans,4, aluF2F)
|
|
DECL_TEMPLATE(ALfloat, X51Chans,6, aluF2F)
|
|
DECL_TEMPLATE(ALfloat, X61Chans,7, aluF2F)
|
|
DECL_TEMPLATE(ALfloat, X71Chans,8, aluF2F)
|
|
|
|
DECL_TEMPLATE(ALushort, MonoChans,1, aluF2US)
|
|
DECL_TEMPLATE(ALushort, QuadChans,4, aluF2US)
|
|
DECL_TEMPLATE(ALushort, X51Chans,6, aluF2US)
|
|
DECL_TEMPLATE(ALushort, X61Chans,7, aluF2US)
|
|
DECL_TEMPLATE(ALushort, X71Chans,8, aluF2US)
|
|
|
|
DECL_TEMPLATE(ALshort, MonoChans,1, aluF2S)
|
|
DECL_TEMPLATE(ALshort, QuadChans,4, aluF2S)
|
|
DECL_TEMPLATE(ALshort, X51Chans,6, aluF2S)
|
|
DECL_TEMPLATE(ALshort, X61Chans,7, aluF2S)
|
|
DECL_TEMPLATE(ALshort, X71Chans,8, aluF2S)
|
|
|
|
DECL_TEMPLATE(ALubyte, MonoChans,1, aluF2UB)
|
|
DECL_TEMPLATE(ALubyte, QuadChans,4, aluF2UB)
|
|
DECL_TEMPLATE(ALubyte, X51Chans,6, aluF2UB)
|
|
DECL_TEMPLATE(ALubyte, X61Chans,7, aluF2UB)
|
|
DECL_TEMPLATE(ALubyte, X71Chans,8, aluF2UB)
|
|
|
|
DECL_TEMPLATE(ALbyte, MonoChans,1, aluF2B)
|
|
DECL_TEMPLATE(ALbyte, QuadChans,4, aluF2B)
|
|
DECL_TEMPLATE(ALbyte, X51Chans,6, aluF2B)
|
|
DECL_TEMPLATE(ALbyte, X61Chans,7, aluF2B)
|
|
DECL_TEMPLATE(ALbyte, X71Chans,8, aluF2B)
|
|
|
|
#undef DECL_TEMPLATE
|
|
|
|
#define DECL_TEMPLATE(T, chans,N, func) \
|
|
static void Write_##T##_##chans(ALCdevice *device, T *buffer, ALuint SamplesToDo)\
|
|
{ \
|
|
ALfloat (*DryBuffer)[MAXCHANNELS] = device->DryBuffer; \
|
|
const ALuint *ChanMap = device->DevChannels; \
|
|
ALuint i, j; \
|
|
\
|
|
if(device->Bs2b) \
|
|
{ \
|
|
for(i = 0;i < SamplesToDo;i++) \
|
|
{ \
|
|
float samples[2]; \
|
|
samples[0] = DryBuffer[i][chans[0]]; \
|
|
samples[1] = DryBuffer[i][chans[1]]; \
|
|
bs2b_cross_feed(device->Bs2b, samples); \
|
|
buffer[ChanMap[chans[0]]] = func(samples[0]); \
|
|
buffer[ChanMap[chans[1]]] = func(samples[1]); \
|
|
buffer += 2; \
|
|
} \
|
|
} \
|
|
else \
|
|
{ \
|
|
for(i = 0;i < SamplesToDo;i++) \
|
|
{ \
|
|
for(j = 0;j < N;j++) \
|
|
buffer[ChanMap[chans[j]]] = func(DryBuffer[i][chans[j]]); \
|
|
buffer += N; \
|
|
} \
|
|
} \
|
|
}
|
|
|
|
DECL_TEMPLATE(ALfloat, StereoChans,2, aluF2F)
|
|
DECL_TEMPLATE(ALushort, StereoChans,2, aluF2US)
|
|
DECL_TEMPLATE(ALshort, StereoChans,2, aluF2S)
|
|
DECL_TEMPLATE(ALubyte, StereoChans,2, aluF2UB)
|
|
DECL_TEMPLATE(ALbyte, StereoChans,2, aluF2B)
|
|
|
|
#undef DECL_TEMPLATE
|
|
|
|
#define DECL_TEMPLATE(T, func) \
|
|
static void Write_##T(ALCdevice *device, T *buffer, ALuint SamplesToDo) \
|
|
{ \
|
|
switch(device->FmtChans) \
|
|
{ \
|
|
case DevFmtMono: \
|
|
Write_##T##_MonoChans(device, buffer, SamplesToDo); \
|
|
break; \
|
|
case DevFmtStereo: \
|
|
Write_##T##_StereoChans(device, buffer, SamplesToDo); \
|
|
break; \
|
|
case DevFmtQuad: \
|
|
Write_##T##_QuadChans(device, buffer, SamplesToDo); \
|
|
break; \
|
|
case DevFmtX51: \
|
|
Write_##T##_X51Chans(device, buffer, SamplesToDo); \
|
|
break; \
|
|
case DevFmtX61: \
|
|
Write_##T##_X61Chans(device, buffer, SamplesToDo); \
|
|
break; \
|
|
case DevFmtX71: \
|
|
Write_##T##_X71Chans(device, buffer, SamplesToDo); \
|
|
break; \
|
|
} \
|
|
}
|
|
|
|
DECL_TEMPLATE(ALfloat, aluF2F)
|
|
DECL_TEMPLATE(ALushort, aluF2US)
|
|
DECL_TEMPLATE(ALshort, aluF2S)
|
|
DECL_TEMPLATE(ALubyte, aluF2UB)
|
|
DECL_TEMPLATE(ALbyte, aluF2B)
|
|
|
|
#undef DECL_TEMPLATE
|
|
|
|
ALvoid aluMixData(ALCdevice *device, ALvoid *buffer, ALsizei size)
|
|
{
|
|
ALuint SamplesToDo;
|
|
ALeffectslot *ALEffectSlot;
|
|
ALCcontext **ctx, **ctx_end;
|
|
ALsource **src, **src_end;
|
|
int fpuState;
|
|
ALuint i, c;
|
|
ALsizei e;
|
|
|
|
#if defined(HAVE_FESETROUND)
|
|
fpuState = fegetround();
|
|
fesetround(FE_TOWARDZERO);
|
|
#elif defined(HAVE__CONTROLFP)
|
|
fpuState = _controlfp(_RC_CHOP, _MCW_RC);
|
|
#else
|
|
(void)fpuState;
|
|
#endif
|
|
|
|
while(size > 0)
|
|
{
|
|
/* Setup variables */
|
|
SamplesToDo = min(size, BUFFERSIZE);
|
|
|
|
/* Clear mixing buffer */
|
|
memset(device->DryBuffer, 0, SamplesToDo*MAXCHANNELS*sizeof(ALfloat));
|
|
|
|
SuspendContext(NULL);
|
|
ctx = device->Contexts;
|
|
ctx_end = ctx + device->NumContexts;
|
|
while(ctx != ctx_end)
|
|
{
|
|
SuspendContext(*ctx);
|
|
|
|
src = (*ctx)->ActiveSources;
|
|
src_end = src + (*ctx)->ActiveSourceCount;
|
|
while(src != src_end)
|
|
{
|
|
if((*src)->state != AL_PLAYING)
|
|
{
|
|
--((*ctx)->ActiveSourceCount);
|
|
*src = *(--src_end);
|
|
continue;
|
|
}
|
|
|
|
if((*src)->NeedsUpdate)
|
|
{
|
|
ALsource_Update(*src, *ctx);
|
|
(*src)->NeedsUpdate = AL_FALSE;
|
|
}
|
|
|
|
MixSource(*src, device, SamplesToDo);
|
|
src++;
|
|
}
|
|
|
|
/* effect slot processing */
|
|
for(e = 0;e < (*ctx)->EffectSlotMap.size;e++)
|
|
{
|
|
ALEffectSlot = (*ctx)->EffectSlotMap.array[e].value;
|
|
|
|
for(i = 0;i < SamplesToDo;i++)
|
|
{
|
|
ALEffectSlot->ClickRemoval[0] -= ALEffectSlot->ClickRemoval[0] / 256.0f;
|
|
ALEffectSlot->WetBuffer[i] += ALEffectSlot->ClickRemoval[0];
|
|
}
|
|
for(i = 0;i < 1;i++)
|
|
{
|
|
ALEffectSlot->ClickRemoval[i] += ALEffectSlot->PendingClicks[i];
|
|
ALEffectSlot->PendingClicks[i] = 0.0f;
|
|
}
|
|
|
|
ALEffect_Process(ALEffectSlot->EffectState, ALEffectSlot,
|
|
SamplesToDo, ALEffectSlot->WetBuffer,
|
|
device->DryBuffer);
|
|
|
|
for(i = 0;i < SamplesToDo;i++)
|
|
ALEffectSlot->WetBuffer[i] = 0.0f;
|
|
}
|
|
|
|
ProcessContext(*ctx);
|
|
ctx++;
|
|
}
|
|
ProcessContext(NULL);
|
|
|
|
//Post processing loop
|
|
for(i = 0;i < SamplesToDo;i++)
|
|
{
|
|
for(c = 0;c < MAXCHANNELS;c++)
|
|
{
|
|
device->ClickRemoval[c] -= device->ClickRemoval[c] / 256.0f;
|
|
device->DryBuffer[i][c] += device->ClickRemoval[c];
|
|
}
|
|
}
|
|
for(i = 0;i < MAXCHANNELS;i++)
|
|
{
|
|
device->ClickRemoval[i] += device->PendingClicks[i];
|
|
device->PendingClicks[i] = 0.0f;
|
|
}
|
|
|
|
switch(device->FmtType)
|
|
{
|
|
case DevFmtByte:
|
|
Write_ALbyte(device, buffer, SamplesToDo);
|
|
break;
|
|
case DevFmtUByte:
|
|
Write_ALubyte(device, buffer, SamplesToDo);
|
|
break;
|
|
case DevFmtShort:
|
|
Write_ALshort(device, buffer, SamplesToDo);
|
|
break;
|
|
case DevFmtUShort:
|
|
Write_ALushort(device, buffer, SamplesToDo);
|
|
break;
|
|
case DevFmtFloat:
|
|
Write_ALfloat(device, buffer, SamplesToDo);
|
|
break;
|
|
}
|
|
|
|
size -= SamplesToDo;
|
|
}
|
|
|
|
#if defined(HAVE_FESETROUND)
|
|
fesetround(fpuState);
|
|
#elif defined(HAVE__CONTROLFP)
|
|
_controlfp(fpuState, _MCW_RC);
|
|
#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);
|
|
}
|