604 lines
22 KiB
C
604 lines
22 KiB
C
/*
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* OpenAL Multi-Zone Reverb Example
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*
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* Copyright (c) 2018 by Chris Robinson <chris.kcat@gmail.com>
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*
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* Permission is hereby granted, free of charge, to any person obtaining a copy
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* of this software and associated documentation files (the "Software"), to deal
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* in the Software without restriction, including without limitation the rights
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* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
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* copies of the Software, and to permit persons to whom the Software is
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* furnished to do so, subject to the following conditions:
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*
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* The above copyright notice and this permission notice shall be included in
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* all copies or substantial portions of the Software.
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*
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* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
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* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
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* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
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* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
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* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
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* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
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* THE SOFTWARE.
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*/
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/* This file contains an example for controlling multiple reverb zones to
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* smoothly transition between reverb environments. The general concept is to
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* extend single-reverb by also tracking the closest adjacent environment, and
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* utilize EAX Reverb's panning vectors to position them relative to the
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* listener.
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*/
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#include <stdio.h>
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#include <assert.h>
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#include <math.h>
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#include <SDL_sound.h>
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#include "AL/al.h"
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#include "AL/alc.h"
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#include "AL/alext.h"
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#include "AL/efx-presets.h"
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#include "common/alhelpers.h"
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#ifndef M_PI
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#define M_PI 3.14159265358979323846
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#endif
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/* Filter object functions */
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static LPALGENFILTERS alGenFilters;
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static LPALDELETEFILTERS alDeleteFilters;
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static LPALISFILTER alIsFilter;
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static LPALFILTERI alFilteri;
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static LPALFILTERIV alFilteriv;
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static LPALFILTERF alFilterf;
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static LPALFILTERFV alFilterfv;
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static LPALGETFILTERI alGetFilteri;
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static LPALGETFILTERIV alGetFilteriv;
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static LPALGETFILTERF alGetFilterf;
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static LPALGETFILTERFV alGetFilterfv;
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/* Effect object functions */
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static LPALGENEFFECTS alGenEffects;
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static LPALDELETEEFFECTS alDeleteEffects;
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static LPALISEFFECT alIsEffect;
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static LPALEFFECTI alEffecti;
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static LPALEFFECTIV alEffectiv;
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static LPALEFFECTF alEffectf;
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static LPALEFFECTFV alEffectfv;
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static LPALGETEFFECTI alGetEffecti;
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static LPALGETEFFECTIV alGetEffectiv;
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static LPALGETEFFECTF alGetEffectf;
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static LPALGETEFFECTFV alGetEffectfv;
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/* Auxiliary Effect Slot object functions */
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static LPALGENAUXILIARYEFFECTSLOTS alGenAuxiliaryEffectSlots;
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static LPALDELETEAUXILIARYEFFECTSLOTS alDeleteAuxiliaryEffectSlots;
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static LPALISAUXILIARYEFFECTSLOT alIsAuxiliaryEffectSlot;
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static LPALAUXILIARYEFFECTSLOTI alAuxiliaryEffectSloti;
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static LPALAUXILIARYEFFECTSLOTIV alAuxiliaryEffectSlotiv;
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static LPALAUXILIARYEFFECTSLOTF alAuxiliaryEffectSlotf;
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static LPALAUXILIARYEFFECTSLOTFV alAuxiliaryEffectSlotfv;
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static LPALGETAUXILIARYEFFECTSLOTI alGetAuxiliaryEffectSloti;
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static LPALGETAUXILIARYEFFECTSLOTIV alGetAuxiliaryEffectSlotiv;
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static LPALGETAUXILIARYEFFECTSLOTF alGetAuxiliaryEffectSlotf;
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static LPALGETAUXILIARYEFFECTSLOTFV alGetAuxiliaryEffectSlotfv;
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/* LoadEffect loads the given initial reverb properties into the given OpenAL
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* effect object, and returns non-zero on success.
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*/
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static int LoadEffect(ALuint effect, const EFXEAXREVERBPROPERTIES *reverb)
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{
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ALenum err;
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alGetError();
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/* Prepare the effect for EAX Reverb (standard reverb doesn't contain
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* the needed panning vectors).
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*/
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alEffecti(effect, AL_EFFECT_TYPE, AL_EFFECT_EAXREVERB);
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if((err=alGetError()) != AL_NO_ERROR)
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{
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fprintf(stderr, "Failed to set EAX Reverb: %s (0x%04x)\n", alGetString(err), err);
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return 0;
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}
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/* Load the reverb properties. */
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alEffectf(effect, AL_EAXREVERB_DENSITY, reverb->flDensity);
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alEffectf(effect, AL_EAXREVERB_DIFFUSION, reverb->flDiffusion);
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alEffectf(effect, AL_EAXREVERB_GAIN, reverb->flGain);
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alEffectf(effect, AL_EAXREVERB_GAINHF, reverb->flGainHF);
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alEffectf(effect, AL_EAXREVERB_GAINLF, reverb->flGainLF);
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alEffectf(effect, AL_EAXREVERB_DECAY_TIME, reverb->flDecayTime);
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alEffectf(effect, AL_EAXREVERB_DECAY_HFRATIO, reverb->flDecayHFRatio);
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alEffectf(effect, AL_EAXREVERB_DECAY_LFRATIO, reverb->flDecayLFRatio);
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alEffectf(effect, AL_EAXREVERB_REFLECTIONS_GAIN, reverb->flReflectionsGain);
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alEffectf(effect, AL_EAXREVERB_REFLECTIONS_DELAY, reverb->flReflectionsDelay);
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alEffectfv(effect, AL_EAXREVERB_REFLECTIONS_PAN, reverb->flReflectionsPan);
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alEffectf(effect, AL_EAXREVERB_LATE_REVERB_GAIN, reverb->flLateReverbGain);
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alEffectf(effect, AL_EAXREVERB_LATE_REVERB_DELAY, reverb->flLateReverbDelay);
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alEffectfv(effect, AL_EAXREVERB_LATE_REVERB_PAN, reverb->flLateReverbPan);
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alEffectf(effect, AL_EAXREVERB_ECHO_TIME, reverb->flEchoTime);
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alEffectf(effect, AL_EAXREVERB_ECHO_DEPTH, reverb->flEchoDepth);
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alEffectf(effect, AL_EAXREVERB_MODULATION_TIME, reverb->flModulationTime);
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alEffectf(effect, AL_EAXREVERB_MODULATION_DEPTH, reverb->flModulationDepth);
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alEffectf(effect, AL_EAXREVERB_AIR_ABSORPTION_GAINHF, reverb->flAirAbsorptionGainHF);
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alEffectf(effect, AL_EAXREVERB_HFREFERENCE, reverb->flHFReference);
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alEffectf(effect, AL_EAXREVERB_LFREFERENCE, reverb->flLFReference);
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alEffectf(effect, AL_EAXREVERB_ROOM_ROLLOFF_FACTOR, reverb->flRoomRolloffFactor);
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alEffecti(effect, AL_EAXREVERB_DECAY_HFLIMIT, reverb->iDecayHFLimit);
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/* Check if an error occured, and return failure if so. */
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if((err=alGetError()) != AL_NO_ERROR)
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{
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fprintf(stderr, "Error setting up reverb: %s\n", alGetString(err));
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return 0;
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}
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return 1;
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}
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/* LoadBuffer loads the named audio file into an OpenAL buffer object, and
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* returns the new buffer ID.
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*/
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static ALuint LoadSound(const char *filename)
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{
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Sound_Sample *sample;
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ALenum err, format;
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ALuint buffer;
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Uint32 slen;
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/* Open the audio file */
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sample = Sound_NewSampleFromFile(filename, NULL, 65536);
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if(!sample)
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{
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fprintf(stderr, "Could not open audio in %s\n", filename);
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return 0;
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}
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/* Get the sound format, and figure out the OpenAL format */
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if(sample->actual.channels == 1)
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{
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if(sample->actual.format == AUDIO_U8)
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format = AL_FORMAT_MONO8;
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else if(sample->actual.format == AUDIO_S16SYS)
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format = AL_FORMAT_MONO16;
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else
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{
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fprintf(stderr, "Unsupported sample format: 0x%04x\n", sample->actual.format);
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Sound_FreeSample(sample);
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return 0;
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}
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}
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else if(sample->actual.channels == 2)
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{
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if(sample->actual.format == AUDIO_U8)
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format = AL_FORMAT_STEREO8;
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else if(sample->actual.format == AUDIO_S16SYS)
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format = AL_FORMAT_STEREO16;
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else
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{
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fprintf(stderr, "Unsupported sample format: 0x%04x\n", sample->actual.format);
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Sound_FreeSample(sample);
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return 0;
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}
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}
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else
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{
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fprintf(stderr, "Unsupported channel count: %d\n", sample->actual.channels);
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Sound_FreeSample(sample);
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return 0;
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}
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/* Decode the whole audio stream to a buffer. */
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slen = Sound_DecodeAll(sample);
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if(!sample->buffer || slen == 0)
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{
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fprintf(stderr, "Failed to read audio from %s\n", filename);
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Sound_FreeSample(sample);
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return 0;
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}
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/* Buffer the audio data into a new buffer object, then free the data and
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* close the file. */
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buffer = 0;
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alGenBuffers(1, &buffer);
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alBufferData(buffer, format, sample->buffer, slen, sample->actual.rate);
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Sound_FreeSample(sample);
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/* Check if an error occured, and clean up if so. */
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err = alGetError();
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if(err != AL_NO_ERROR)
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{
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fprintf(stderr, "OpenAL Error: %s\n", alGetString(err));
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if(buffer && alIsBuffer(buffer))
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alDeleteBuffers(1, &buffer);
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return 0;
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}
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return buffer;
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}
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/* Helper to calculate the dot-product of the two given vectors. */
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static ALfloat dot_product(const ALfloat vec0[3], const ALfloat vec1[3])
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{
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return vec0[0]*vec1[0] + vec0[1]*vec1[1] + vec0[2]*vec1[2];
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}
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int main(int argc, char **argv)
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{
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static const int MaxTransitions = 8;
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EFXEAXREVERBPROPERTIES reverb0 = EFX_REVERB_PRESET_CASTLE_LARGEROOM;
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EFXEAXREVERBPROPERTIES reverb1 = EFX_REVERB_PRESET_CASTLE_LONGPASSAGE;
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struct timespec basetime;
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ALCdevice *device = NULL;
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ALCcontext *context = NULL;
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ALuint effects[2] = { 0, 0 };
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ALuint slots[2] = { 0, 0 };
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ALuint direct_filter = 0;
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ALuint buffer = 0;
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ALuint source = 0;
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ALCint num_sends = 0;
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ALenum state = AL_INITIAL;
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ALfloat direct_gain = 1.0f;
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int loops = 0;
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/* Print out usage if no arguments were specified */
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if(argc < 2)
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{
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fprintf(stderr, "Usage: %s [-device <name>] [options] <filename>\n\n"
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"Options:\n"
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"\t-nodirect\tSilence direct path output (easier to hear reverb)\n\n",
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argv[0]);
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return 1;
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}
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/* Initialize OpenAL, and check for EFX support with at least 2 auxiliary
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* sends (if multiple sends are supported, 2 are provided by default; if
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* you want more, you have to request it through alcCreateContext).
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*/
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argv++; argc--;
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if(InitAL(&argv, &argc) != 0)
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return 1;
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while(argc > 0)
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{
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if(strcmp(argv[0], "-nodirect") == 0)
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direct_gain = 0.0f;
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else
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break;
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argv++;
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argc--;
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}
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if(argc < 1)
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{
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fprintf(stderr, "No filename spacified.\n");
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CloseAL();
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return 1;
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}
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context = alcGetCurrentContext();
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device = alcGetContextsDevice(context);
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if(!alcIsExtensionPresent(device, "ALC_EXT_EFX"))
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{
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fprintf(stderr, "Error: EFX not supported\n");
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CloseAL();
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return 1;
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}
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num_sends = 0;
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alcGetIntegerv(device, ALC_MAX_AUXILIARY_SENDS, 1, &num_sends);
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if(alcGetError(device) != ALC_NO_ERROR || num_sends < 2)
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{
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fprintf(stderr, "Error: Device does not support multiple sends (got %d, need 2)\n",
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num_sends);
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CloseAL();
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return 1;
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}
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/* Define a macro to help load the function pointers. */
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#define LOAD_PROC(x) ((x) = alGetProcAddress(#x))
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LOAD_PROC(alGenFilters);
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LOAD_PROC(alDeleteFilters);
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LOAD_PROC(alIsFilter);
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LOAD_PROC(alFilteri);
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LOAD_PROC(alFilteriv);
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LOAD_PROC(alFilterf);
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LOAD_PROC(alFilterfv);
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LOAD_PROC(alGetFilteri);
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LOAD_PROC(alGetFilteriv);
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LOAD_PROC(alGetFilterf);
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LOAD_PROC(alGetFilterfv);
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LOAD_PROC(alGenEffects);
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LOAD_PROC(alDeleteEffects);
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LOAD_PROC(alIsEffect);
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LOAD_PROC(alEffecti);
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LOAD_PROC(alEffectiv);
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LOAD_PROC(alEffectf);
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LOAD_PROC(alEffectfv);
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LOAD_PROC(alGetEffecti);
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LOAD_PROC(alGetEffectiv);
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LOAD_PROC(alGetEffectf);
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LOAD_PROC(alGetEffectfv);
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LOAD_PROC(alGenAuxiliaryEffectSlots);
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LOAD_PROC(alDeleteAuxiliaryEffectSlots);
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LOAD_PROC(alIsAuxiliaryEffectSlot);
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LOAD_PROC(alAuxiliaryEffectSloti);
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LOAD_PROC(alAuxiliaryEffectSlotiv);
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LOAD_PROC(alAuxiliaryEffectSlotf);
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LOAD_PROC(alAuxiliaryEffectSlotfv);
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LOAD_PROC(alGetAuxiliaryEffectSloti);
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LOAD_PROC(alGetAuxiliaryEffectSlotiv);
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LOAD_PROC(alGetAuxiliaryEffectSlotf);
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LOAD_PROC(alGetAuxiliaryEffectSlotfv);
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#undef LOAD_PROC
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/* Initialize SDL_sound. */
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Sound_Init();
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/* Load the sound into a buffer. */
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buffer = LoadSound(argv[0]);
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if(!buffer)
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{
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CloseAL();
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Sound_Quit();
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return 1;
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}
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/* Generate two effects for two "zones", and load a reverb into each one.
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* Note that unlike single-zone reverb, where you can store one effect per
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* preset, for multi-zone reverb you should have one effect per environment
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* instance, or one per audible zone. This is because we'll be changing the
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* effects' properties in real-time based on the environment instance
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* relative to the listener.
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*/
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alGenEffects(2, effects);
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if(!LoadEffect(effects[0], &reverb0) || !LoadEffect(effects[1], &reverb1))
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{
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alDeleteEffects(2, effects);
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alDeleteBuffers(1, &buffer);
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Sound_Quit();
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CloseAL();
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return 1;
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}
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/* Create the effect slot objects, one for each "active" effect. */
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alGenAuxiliaryEffectSlots(2, slots);
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/* Tell the effect slots to use the loaded effect objects, with slot 0 for
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* Zone 0 and slot 1 for Zone 1. Note that this effectively copies the
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* effect properties. Modifying or deleting the effect object afterward
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* won't directly affect the effect slot until they're reapplied like this.
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*/
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alAuxiliaryEffectSloti(slots[0], AL_EFFECTSLOT_EFFECT, effects[0]);
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alAuxiliaryEffectSloti(slots[1], AL_EFFECTSLOT_EFFECT, effects[1]);
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assert(alGetError()==AL_NO_ERROR && "Failed to set effect slot");
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/* For the purposes of this example, prepare a filter that optionally
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* silences the direct path which allows us to hear just the reverberation.
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* A filter like this is normally used for obstruction, where the path
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* directly between the listener and source is blocked (the exact
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* properties depending on the type and thickness of the obstructing
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* material).
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*/
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alGenFilters(1, &direct_filter);
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alFilteri(direct_filter, AL_FILTER_TYPE, AL_FILTER_LOWPASS);
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alFilterf(direct_filter, AL_LOWPASS_GAIN, direct_gain);
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assert(alGetError()==AL_NO_ERROR && "Failed to set direct filter");
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/* Create the source to play the sound with. */
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source = 0;
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alGenSources(1, &source);
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alSourcei(source, AL_LOOPING, AL_TRUE);
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alSourcei(source, AL_DIRECT_FILTER, direct_filter);
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alSourcei(source, AL_BUFFER, buffer);
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/* Connect the source to the effect slots. Here, we connect source send 0
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* to Zone 0's slot, and send 1 to Zone 1's slot. Filters can be specified
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* to occlude the source from each zone by varying amounts; for example, a
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* source within a particular zone would be unfiltered, while a source that
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* can only see a zone through a window may be attenuated for that zone.
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*/
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alSource3i(source, AL_AUXILIARY_SEND_FILTER, slots[0], 0, AL_FILTER_NULL);
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alSource3i(source, AL_AUXILIARY_SEND_FILTER, slots[1], 1, AL_FILTER_NULL);
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assert(alGetError()==AL_NO_ERROR && "Failed to setup sound source");
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/* Get the current time as the base for timing in the main loop. */
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altimespec_get(&basetime, AL_TIME_UTC);
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loops = 0;
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printf("Transition %d of %d...\n", loops+1, MaxTransitions);
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/* Play the sound for a while. */
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alSourcePlay(source);
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do {
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/* Individual reverb zones are connected via "portals". Each portal has
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* a position (center point of the connecting area), a normal (facing
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* direction), and a radius (approximate size of the connecting area).
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* For this example it also has movement velocity, although normally it
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* would be the listener that moves relative to the portal instead of
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* the portal itself.
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*/
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const ALfloat portal_pos[3] = { -10.0f, 0.0f, 0.0f };
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const ALfloat portal_norm[3] = { 1.0f, 0.0f, 0.0f };
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const ALfloat portal_vel[3] = { 5.0f, 0.0f, 0.0f };
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const ALfloat portal_radius = 2.5f;
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ALfloat other_dir[3], this_dir[3];
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ALfloat local_norm[3];
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ALfloat local_dir[3];
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ALfloat local_radius;
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ALfloat dist, timediff;
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struct timespec curtime;
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/* Start a batch update, to ensure all changes apply simultaneously. */
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alcSuspendContext(context);
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/* Get the current time to track the amount of time that passed.
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* Convert the difference to seconds.
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*/
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altimespec_get(&curtime, AL_TIME_UTC);
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timediff = (ALfloat)(curtime.tv_sec - basetime.tv_sec);
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timediff += (ALfloat)(curtime.tv_nsec - basetime.tv_nsec) / 1000000000.0f;
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/* Avoid negative time deltas, in case of non-monotonic clocks. */
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if(timediff < 0.0f)
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timediff = 0.0f;
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else while(timediff >= 4.0f)
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{
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/* For this example, each transition occurs over 4 seconds.
|
|
* Decrease the delta and increase the base time to start a new
|
|
* transition.
|
|
*/
|
|
timediff -= 4.0f;
|
|
basetime.tv_sec += 4;
|
|
if(++loops < MaxTransitions)
|
|
printf("Transition %d of %d...\n", loops+1, MaxTransitions);
|
|
}
|
|
|
|
/* Move the portal according to the amount of time passed. local_dir
|
|
* represents the listener-relative point to the adjacent zone.
|
|
*/
|
|
local_dir[0] = portal_pos[0] + portal_vel[0]*timediff;
|
|
local_dir[1] = portal_pos[1] + portal_vel[1]*timediff;
|
|
local_dir[2] = portal_pos[2] + portal_vel[2]*timediff;
|
|
/* A normal application would also rotate the portal's normal given the
|
|
* listener orientation, to get the listener-relative normal.
|
|
*
|
|
* For this example, the portal is always head-on but every other
|
|
* transition negates the normal. This effectively simulates a
|
|
* different portal moving in closer than the last one that faces the
|
|
* other way, switching the old adjacent zone to a new one.
|
|
*/
|
|
local_norm[0] = portal_norm[0] * ((loops&1) ? -1.0f : 1.0f);
|
|
local_norm[1] = portal_norm[1] * ((loops&1) ? -1.0f : 1.0f);
|
|
local_norm[2] = portal_norm[2] * ((loops&1) ? -1.0f : 1.0f);
|
|
|
|
/* Calculate the distance from the listener to the portal. */
|
|
dist = sqrtf(dot_product(local_dir, local_dir));
|
|
if(!(dist > 0.00001f))
|
|
{
|
|
/* We're practically in the center of the portal. Give the panning
|
|
* vectors a 50/50 split, with Zone 0 covering the half in front of
|
|
* the normal, and Zone 1 covering the half behind.
|
|
*/
|
|
this_dir[0] = local_norm[0] / 2.0f;
|
|
this_dir[1] = local_norm[1] / 2.0f;
|
|
this_dir[2] = local_norm[2] / 2.0f;
|
|
|
|
other_dir[0] = local_norm[0] / -2.0f;
|
|
other_dir[1] = local_norm[1] / -2.0f;
|
|
other_dir[2] = local_norm[2] / -2.0f;
|
|
|
|
alEffectf(effects[0], AL_EAXREVERB_GAIN, reverb0.flGain);
|
|
alEffectfv(effects[0], AL_EAXREVERB_REFLECTIONS_PAN, this_dir);
|
|
alEffectfv(effects[0], AL_EAXREVERB_LATE_REVERB_PAN, this_dir);
|
|
|
|
alEffectf(effects[1], AL_EAXREVERB_GAIN, reverb1.flGain);
|
|
alEffectfv(effects[1], AL_EAXREVERB_REFLECTIONS_PAN, other_dir);
|
|
alEffectfv(effects[1], AL_EAXREVERB_LATE_REVERB_PAN, other_dir);
|
|
}
|
|
else
|
|
{
|
|
const EFXEAXREVERBPROPERTIES *other_reverb;
|
|
const EFXEAXREVERBPROPERTIES *this_reverb;
|
|
ALuint other_effect, this_effect;
|
|
ALfloat spread, attn;
|
|
|
|
/* Normalize the direction to the portal. */
|
|
local_dir[0] /= dist;
|
|
local_dir[1] /= dist;
|
|
local_dir[2] /= dist;
|
|
|
|
/* Scale the radius according to its local angle. The visibility to
|
|
* the other zone reduces as the portal becomes perpendicular.
|
|
*/
|
|
local_radius = portal_radius * fabsf(dot_product(local_dir, local_norm));
|
|
|
|
/* Calculate distance attenuation for the other zone, using the
|
|
* standard inverse distance model with the radius as a reference.
|
|
*/
|
|
attn = local_radius / dist;
|
|
if(attn > 1.0f) attn = 1.0f;
|
|
|
|
/* Calculate the 'spread' of the portal, which is the amount of
|
|
* coverage the other zone has around the listener.
|
|
*/
|
|
spread = atan2f(local_radius, dist) / (ALfloat)M_PI;
|
|
|
|
/* Figure out which zone we're in, given the direction to the
|
|
* portal and its normal.
|
|
*/
|
|
if(dot_product(local_dir, local_norm) <= 0.0f)
|
|
{
|
|
/* We're in front of the portal, so we're in Zone 0. */
|
|
this_effect = effects[0];
|
|
other_effect = effects[1];
|
|
this_reverb = &reverb0;
|
|
other_reverb = &reverb1;
|
|
}
|
|
else
|
|
{
|
|
/* We're behind the portal, so we're in Zone 1. */
|
|
this_effect = effects[1];
|
|
other_effect = effects[0];
|
|
this_reverb = &reverb1;
|
|
other_reverb = &reverb0;
|
|
}
|
|
|
|
/* Scale the other zone's panning vector down as the portal's
|
|
* spread increases, so that it envelops the listener more.
|
|
*/
|
|
other_dir[0] = local_dir[0] * (1.0f-spread);
|
|
other_dir[1] = local_dir[1] * (1.0f-spread);
|
|
other_dir[2] = local_dir[2] * (1.0f-spread);
|
|
/* Pan the current zone to the opposite direction of the portal,
|
|
* and take the remaining percentage of the portal's spread.
|
|
*/
|
|
this_dir[0] = local_dir[0] * -spread;
|
|
this_dir[1] = local_dir[1] * -spread;
|
|
this_dir[2] = local_dir[2] * -spread;
|
|
|
|
/* Now set the effects' panning vectors and distance attenuation. */
|
|
alEffectf(this_effect, AL_EAXREVERB_GAIN, this_reverb->flGain);
|
|
alEffectfv(this_effect, AL_EAXREVERB_REFLECTIONS_PAN, this_dir);
|
|
alEffectfv(this_effect, AL_EAXREVERB_LATE_REVERB_PAN, this_dir);
|
|
|
|
alEffectf(other_effect, AL_EAXREVERB_GAIN, other_reverb->flGain * attn);
|
|
alEffectfv(other_effect, AL_EAXREVERB_REFLECTIONS_PAN, other_dir);
|
|
alEffectfv(other_effect, AL_EAXREVERB_LATE_REVERB_PAN, other_dir);
|
|
}
|
|
|
|
/* Finally, update the effect slots with the updated effect parameters,
|
|
* and finish the update batch.
|
|
*/
|
|
alAuxiliaryEffectSloti(slots[0], AL_EFFECTSLOT_EFFECT, effects[0]);
|
|
alAuxiliaryEffectSloti(slots[1], AL_EFFECTSLOT_EFFECT, effects[1]);
|
|
alcProcessContext(context);
|
|
|
|
al_nssleep(10000000);
|
|
|
|
alGetSourcei(source, AL_SOURCE_STATE, &state);
|
|
} while(alGetError() == AL_NO_ERROR && state == AL_PLAYING && loops < MaxTransitions);
|
|
|
|
/* All done. Delete resources, and close down SDL_sound and OpenAL. */
|
|
alDeleteSources(1, &source);
|
|
alDeleteAuxiliaryEffectSlots(2, slots);
|
|
alDeleteEffects(2, effects);
|
|
alDeleteFilters(1, &direct_filter);
|
|
alDeleteBuffers(1, &buffer);
|
|
|
|
Sound_Quit();
|
|
CloseAL();
|
|
|
|
return 0;
|
|
}
|