mirror of
https://github.com/DrBeef/Quake2Quest.git
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ece44600d1
removes the audio lag
203 lines
6.4 KiB
C
203 lines
6.4 KiB
C
/**
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* OpenAL cross platform audio library
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* Copyright (C) 2009 by Chris Robinson.
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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 "alMain.h"
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#include "alFilter.h"
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#include "alAuxEffectSlot.h"
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#include "alError.h"
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#include "alu.h"
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typedef struct ALmodulatorState {
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// Must be first in all effects!
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ALeffectState state;
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enum {
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SINUSOID,
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SAWTOOTH,
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SQUARE
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} Waveform;
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ALuint index;
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ALuint step;
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ALfloat Gain[MAXCHANNELS];
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FILTER iirFilter;
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ALfloat history[1];
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} ALmodulatorState;
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#define WAVEFORM_FRACBITS 16
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#define WAVEFORM_FRACONE (1<<WAVEFORM_FRACBITS)
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#define WAVEFORM_FRACMASK (WAVEFORM_FRACONE-1)
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static __inline ALfloat Sin(ALuint index)
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{
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return aluSin(index * (F_PI*2.0f / WAVEFORM_FRACONE));
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}
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static __inline ALfloat Saw(ALuint index)
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{
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return index*(2.0f/WAVEFORM_FRACONE) - 1.0f;
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}
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static __inline ALfloat Square(ALuint index)
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{
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return ((index>>(WAVEFORM_FRACBITS-1))&1)*2.0f - 1.0f;
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}
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static __inline ALfloat hpFilter1P(FILTER *iir, ALuint offset, ALfloat input)
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{
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ALfloat *history = &iir->history[offset];
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ALfloat a = iir->coeff;
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ALfloat output = input;
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output = output + (history[0]-output)*a;
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history[0] = output;
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return input - output;
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}
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#define DECL_TEMPLATE(func) \
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static void Process##func(ALmodulatorState *state, ALuint SamplesToDo, \
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const ALfloat *SamplesIn, ALfloat (*SamplesOut)[MAXCHANNELS]) \
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{ \
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const ALuint step = state->step; \
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ALuint index = state->index; \
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ALfloat samp; \
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ALuint i, k; \
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\
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for(i = 0;i < SamplesToDo;i++) \
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{ \
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samp = SamplesIn[i]; \
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\
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index += step; \
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index &= WAVEFORM_FRACMASK; \
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samp *= func(index); \
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\
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samp = hpFilter1P(&state->iirFilter, 0, samp); \
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\
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for(k = 0;k < MAXCHANNELS;k++) \
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SamplesOut[i][k] += state->Gain[k] * samp; \
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} \
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state->index = index; \
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}
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DECL_TEMPLATE(Sin)
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DECL_TEMPLATE(Saw)
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DECL_TEMPLATE(Square)
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#undef DECL_TEMPLATE
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static ALvoid ModulatorDestroy(ALeffectState *effect)
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{
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ALmodulatorState *state = (ALmodulatorState*)effect;
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free(state);
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}
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static ALboolean ModulatorDeviceUpdate(ALeffectState *effect, ALCdevice *Device)
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{
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return AL_TRUE;
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(void)effect;
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(void)Device;
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}
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static ALvoid ModulatorUpdate(ALeffectState *effect, ALCdevice *Device, const ALeffectslot *Slot)
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{
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ALmodulatorState *state = (ALmodulatorState*)effect;
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ALfloat gain, cw, a = 0.0f;
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ALuint index;
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if(Slot->effect.Modulator.Waveform == AL_RING_MODULATOR_SINUSOID)
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state->Waveform = SINUSOID;
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else if(Slot->effect.Modulator.Waveform == AL_RING_MODULATOR_SAWTOOTH)
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state->Waveform = SAWTOOTH;
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else if(Slot->effect.Modulator.Waveform == AL_RING_MODULATOR_SQUARE)
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state->Waveform = SQUARE;
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state->step = fastf2u(Slot->effect.Modulator.Frequency*WAVEFORM_FRACONE /
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Device->Frequency);
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if(state->step == 0) state->step = 1;
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cw = aluCos(F_PI*2.0f * Slot->effect.Modulator.HighPassCutoff /
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Device->Frequency);
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a = (2.0f-cw) - aluSqrt(aluPow(2.0f-cw, 2.0f) - 1.0f);
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state->iirFilter.coeff = a;
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gain = aluSqrt(1.0f/Device->NumChan);
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gain *= Slot->Gain;
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for(index = 0;index < MAXCHANNELS;index++)
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state->Gain[index] = 0.0f;
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for(index = 0;index < Device->NumChan;index++)
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{
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enum Channel chan = Device->Speaker2Chan[index];
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state->Gain[chan] = gain;
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}
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}
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static ALvoid ModulatorProcess(ALeffectState *effect, ALuint SamplesToDo, const ALfloat *SamplesIn, ALfloat (*SamplesOut)[MAXCHANNELS])
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{
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ALmodulatorState *state = (ALmodulatorState*)effect;
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switch(state->Waveform)
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{
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case SINUSOID:
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ProcessSin(state, SamplesToDo, SamplesIn, SamplesOut);
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break;
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case SAWTOOTH:
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ProcessSaw(state, SamplesToDo, SamplesIn, SamplesOut);
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break;
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case SQUARE:
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ProcessSquare(state, SamplesToDo, SamplesIn, SamplesOut);
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break;
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}
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}
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ALeffectState *ModulatorCreate(void)
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{
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ALmodulatorState *state;
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state = malloc(sizeof(*state));
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if(!state)
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return NULL;
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state->state.Destroy = ModulatorDestroy;
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state->state.DeviceUpdate = ModulatorDeviceUpdate;
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state->state.Update = ModulatorUpdate;
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state->state.Process = ModulatorProcess;
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state->index = 0;
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state->step = 1;
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state->iirFilter.coeff = 0.0f;
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state->iirFilter.history[0] = 0.0f;
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return &state->state;
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}
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