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/*
* OpenAL Multi-Zone Reverb Example
*
* Copyright (c) 2018 by Chris Robinson <chris.kcat@gmail.com>
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/* This file contains an example for controlling multiple reverb zones to
* smoothly transition between reverb environments. The general concept is to
* extend single-reverb by also tracking the closest adjacent environment, and
* utilize EAX Reverb's panning vectors to position them relative to the
* listener.
*/
#include <stdio.h>
#include <assert.h>
#include <math.h>
#include <SDL_sound.h>
#include "AL/al.h"
#include "AL/alc.h"
#include "AL/alext.h"
#include "AL/efx-presets.h"
#include "common/alhelpers.h"
#ifndef M_PI
#define M_PI 3.14159265358979323846
#endif
/* Filter object functions */
static LPALGENFILTERS alGenFilters;
static LPALDELETEFILTERS alDeleteFilters;
static LPALISFILTER alIsFilter;
static LPALFILTERI alFilteri;
static LPALFILTERIV alFilteriv;
static LPALFILTERF alFilterf;
static LPALFILTERFV alFilterfv;
static LPALGETFILTERI alGetFilteri;
static LPALGETFILTERIV alGetFilteriv;
static LPALGETFILTERF alGetFilterf;
static LPALGETFILTERFV alGetFilterfv;
/* Effect object functions */
static LPALGENEFFECTS alGenEffects;
static LPALDELETEEFFECTS alDeleteEffects;
static LPALISEFFECT alIsEffect;
static LPALEFFECTI alEffecti;
static LPALEFFECTIV alEffectiv;
static LPALEFFECTF alEffectf;
static LPALEFFECTFV alEffectfv;
static LPALGETEFFECTI alGetEffecti;
static LPALGETEFFECTIV alGetEffectiv;
static LPALGETEFFECTF alGetEffectf;
static LPALGETEFFECTFV alGetEffectfv;
/* Auxiliary Effect Slot object functions */
static LPALGENAUXILIARYEFFECTSLOTS alGenAuxiliaryEffectSlots;
static LPALDELETEAUXILIARYEFFECTSLOTS alDeleteAuxiliaryEffectSlots;
static LPALISAUXILIARYEFFECTSLOT alIsAuxiliaryEffectSlot;
static LPALAUXILIARYEFFECTSLOTI alAuxiliaryEffectSloti;
static LPALAUXILIARYEFFECTSLOTIV alAuxiliaryEffectSlotiv;
static LPALAUXILIARYEFFECTSLOTF alAuxiliaryEffectSlotf;
static LPALAUXILIARYEFFECTSLOTFV alAuxiliaryEffectSlotfv;
static LPALGETAUXILIARYEFFECTSLOTI alGetAuxiliaryEffectSloti;
static LPALGETAUXILIARYEFFECTSLOTIV alGetAuxiliaryEffectSlotiv;
static LPALGETAUXILIARYEFFECTSLOTF alGetAuxiliaryEffectSlotf;
static LPALGETAUXILIARYEFFECTSLOTFV alGetAuxiliaryEffectSlotfv;
/* LoadEffect loads the given initial reverb properties into the given OpenAL
* effect object, and returns non-zero on success.
*/
static int LoadEffect(ALuint effect, const EFXEAXREVERBPROPERTIES *reverb)
{
ALenum err;
alGetError();
/* Prepare the effect for EAX Reverb (standard reverb doesn't contain
* the needed panning vectors).
*/
alEffecti(effect, AL_EFFECT_TYPE, AL_EFFECT_EAXREVERB);
if((err=alGetError()) != AL_NO_ERROR)
{
fprintf(stderr, "Failed to set EAX Reverb: %s (0x%04x)\n", alGetString(err), err);
return 0;
}
/* Load the reverb properties. */
alEffectf(effect, AL_EAXREVERB_DENSITY, reverb->flDensity);
alEffectf(effect, AL_EAXREVERB_DIFFUSION, reverb->flDiffusion);
alEffectf(effect, AL_EAXREVERB_GAIN, reverb->flGain);
alEffectf(effect, AL_EAXREVERB_GAINHF, reverb->flGainHF);
alEffectf(effect, AL_EAXREVERB_GAINLF, reverb->flGainLF);
alEffectf(effect, AL_EAXREVERB_DECAY_TIME, reverb->flDecayTime);
alEffectf(effect, AL_EAXREVERB_DECAY_HFRATIO, reverb->flDecayHFRatio);
alEffectf(effect, AL_EAXREVERB_DECAY_LFRATIO, reverb->flDecayLFRatio);
alEffectf(effect, AL_EAXREVERB_REFLECTIONS_GAIN, reverb->flReflectionsGain);
alEffectf(effect, AL_EAXREVERB_REFLECTIONS_DELAY, reverb->flReflectionsDelay);
alEffectfv(effect, AL_EAXREVERB_REFLECTIONS_PAN, reverb->flReflectionsPan);
alEffectf(effect, AL_EAXREVERB_LATE_REVERB_GAIN, reverb->flLateReverbGain);
alEffectf(effect, AL_EAXREVERB_LATE_REVERB_DELAY, reverb->flLateReverbDelay);
alEffectfv(effect, AL_EAXREVERB_LATE_REVERB_PAN, reverb->flLateReverbPan);
alEffectf(effect, AL_EAXREVERB_ECHO_TIME, reverb->flEchoTime);
alEffectf(effect, AL_EAXREVERB_ECHO_DEPTH, reverb->flEchoDepth);
alEffectf(effect, AL_EAXREVERB_MODULATION_TIME, reverb->flModulationTime);
alEffectf(effect, AL_EAXREVERB_MODULATION_DEPTH, reverb->flModulationDepth);
alEffectf(effect, AL_EAXREVERB_AIR_ABSORPTION_GAINHF, reverb->flAirAbsorptionGainHF);
alEffectf(effect, AL_EAXREVERB_HFREFERENCE, reverb->flHFReference);
alEffectf(effect, AL_EAXREVERB_LFREFERENCE, reverb->flLFReference);
alEffectf(effect, AL_EAXREVERB_ROOM_ROLLOFF_FACTOR, reverb->flRoomRolloffFactor);
alEffecti(effect, AL_EAXREVERB_DECAY_HFLIMIT, reverb->iDecayHFLimit);
/* Check if an error occured, and return failure if so. */
if((err=alGetError()) != AL_NO_ERROR)
{
fprintf(stderr, "Error setting up reverb: %s\n", alGetString(err));
return 0;
}
return 1;
}
/* LoadBuffer loads the named audio file into an OpenAL buffer object, and
* returns the new buffer ID.
*/
static ALuint LoadSound(const char *filename)
{
Sound_Sample *sample;
ALenum err, format;
ALuint buffer;
Uint32 slen;
/* Open the audio file */
sample = Sound_NewSampleFromFile(filename, NULL, 65536);
if(!sample)
{
fprintf(stderr, "Could not open audio in %s\n", filename);
return 0;
}
/* Get the sound format, and figure out the OpenAL format */
if(sample->actual.channels == 1)
{
if(sample->actual.format == AUDIO_U8)
format = AL_FORMAT_MONO8;
else if(sample->actual.format == AUDIO_S16SYS)
format = AL_FORMAT_MONO16;
else
{
fprintf(stderr, "Unsupported sample format: 0x%04x\n", sample->actual.format);
Sound_FreeSample(sample);
return 0;
}
}
else if(sample->actual.channels == 2)
{
if(sample->actual.format == AUDIO_U8)
format = AL_FORMAT_STEREO8;
else if(sample->actual.format == AUDIO_S16SYS)
format = AL_FORMAT_STEREO16;
else
{
fprintf(stderr, "Unsupported sample format: 0x%04x\n", sample->actual.format);
Sound_FreeSample(sample);
return 0;
}
}
else
{
fprintf(stderr, "Unsupported channel count: %d\n", sample->actual.channels);
Sound_FreeSample(sample);
return 0;
}
/* Decode the whole audio stream to a buffer. */
slen = Sound_DecodeAll(sample);
if(!sample->buffer || slen == 0)
{
fprintf(stderr, "Failed to read audio from %s\n", filename);
Sound_FreeSample(sample);
return 0;
}
/* Buffer the audio data into a new buffer object, then free the data and
* close the file. */
buffer = 0;
alGenBuffers(1, &buffer);
alBufferData(buffer, format, sample->buffer, slen, sample->actual.rate);
Sound_FreeSample(sample);
/* Check if an error occured, and clean up if so. */
err = alGetError();
if(err != AL_NO_ERROR)
{
fprintf(stderr, "OpenAL Error: %s\n", alGetString(err));
if(buffer && alIsBuffer(buffer))
alDeleteBuffers(1, &buffer);
return 0;
}
return buffer;
}
/* Helper to calculate the dot-product of the two given vectors. */
static ALfloat dot_product(const ALfloat vec0[3], const ALfloat vec1[3])
{
return vec0[0]*vec1[0] + vec0[1]*vec1[1] + vec0[2]*vec1[2];
}
/* Helper to normalize a given vector. */
static void normalize(ALfloat vec[3])
{
ALfloat mag = sqrtf(dot_product(vec, vec));
if(mag > 0.00001f)
{
vec[0] /= mag;
vec[1] /= mag;
vec[2] /= mag;
}
else
{
vec[0] = 0.0f;
vec[1] = 0.0f;
vec[2] = 0.0f;
}
}
/* The main update function to update the listener and environment effects. */
static void UpdateListenerAndEffects(float timediff, const ALuint slots[2], const ALuint effects[2], const EFXEAXREVERBPROPERTIES reverbs[2])
{
static const ALfloat listener_move_scale = 10.0f;
/* Individual reverb zones are connected via "portals". Each portal has a
* position (center point of the connecting area), a normal (facing
* direction), and a radius (approximate size of the connecting area).
*/
const ALfloat portal_pos[3] = { 0.0f, 0.0f, 0.0f };
const ALfloat portal_norm[3] = { sqrtf(0.5f), 0.0f, -sqrtf(0.5f) };
const ALfloat portal_radius = 2.5f;
ALfloat other_dir[3], this_dir[3];
ALfloat listener_pos[3];
ALfloat local_norm[3];
ALfloat local_dir[3];
ALfloat near_edge[3];
ALfloat far_edge[3];
ALfloat dist, edist;
/* Update the listener position for the amount of time passed. This uses a
* simple triangular LFO to offset the position (moves along the X axis
* between -listener_move_scale and +listener_move_scale for each
* transition).
*/
listener_pos[0] = (fabsf(2.0f - timediff/2.0f) - 1.0f) * listener_move_scale;
listener_pos[1] = 0.0f;
listener_pos[2] = 0.0f;
alListenerfv(AL_POSITION, listener_pos);
/* Calculate local_dir, which represents the listener-relative point to the
* adjacent zone (should also include orientation). Because EAX Reverb uses
* left-handed coordinates instead of right-handed like the rest of OpenAL,
* negate Z for the local values.
*/
local_dir[0] = portal_pos[0] - listener_pos[0];
local_dir[1] = portal_pos[1] - listener_pos[1];
local_dir[2] = -(portal_pos[2] - listener_pos[2]);
/* A normal application would also rotate the portal's normal given the
* listener orientation, to get the listener-relative normal.
*/
local_norm[0] = portal_norm[0];
local_norm[1] = portal_norm[1];
local_norm[2] = -portal_norm[2];
/* Calculate the distance from the listener to the portal, and ensure it's
* far enough away to not suffer severe floating-point precision issues.
*/
dist = sqrtf(dot_product(local_dir, local_dir));
if(dist > 0.00001f)
{
const EFXEAXREVERBPROPERTIES *other_reverb, *this_reverb;
ALuint other_effect, this_effect;
ALfloat magnitude, dir_dot_norm;
/* Normalize the direction to the portal. */
local_dir[0] /= dist;
local_dir[1] /= dist;
local_dir[2] /= dist;
/* Calculate the dot product of the portal's local direction and local
* normal, which is used for angular and side checks later on.
*/
dir_dot_norm = dot_product(local_dir, local_norm);
/* Figure out which zone we're in. */
if(dir_dot_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 = &reverbs[0];
other_reverb = &reverbs[1];
}
else
{
/* We're behind the portal, so we're in Zone 1. */
this_effect = effects[1];
other_effect = effects[0];
this_reverb = &reverbs[1];
other_reverb = &reverbs[0];
}
/* Calculate the listener-relative extents of the portal. */
/* First, project the listener-to-portal vector onto the portal's plane
* to get the portal-relative direction along the plane that goes away
* from the listener (toward the farthest edge of the portal).
*/
far_edge[0] = local_dir[0] - local_norm[0]*dir_dot_norm;
far_edge[1] = local_dir[1] - local_norm[1]*dir_dot_norm;
far_edge[2] = local_dir[2] - local_norm[2]*dir_dot_norm;
edist = sqrtf(dot_product(far_edge, far_edge));
if(edist > 0.0001f)
{
/* Rescale the portal-relative vector to be at the radius edge. */
ALfloat mag = portal_radius / edist;
far_edge[0] *= mag;
far_edge[1] *= mag;
far_edge[2] *= mag;
/* Calculate the closest edge of the portal by negating the
* farthest, and add an offset to make them both relative to the
* listener.
*/
near_edge[0] = local_dir[0]*dist - far_edge[0];
near_edge[1] = local_dir[1]*dist - far_edge[1];
near_edge[2] = local_dir[2]*dist - far_edge[2];
far_edge[0] += local_dir[0]*dist;
far_edge[1] += local_dir[1]*dist;
far_edge[2] += local_dir[2]*dist;
/* Normalize the listener-relative extents of the portal, then
* calculate the panning magnitude for the other zone given the
* apparent size of the opening. The panning magnitude affects the
* envelopment of the environment, with 1 being a point, 0.5 being
* half coverage around the listener, and 0 being full coverage.
*/
normalize(far_edge);
normalize(near_edge);
magnitude = 1.0f - acosf(dot_product(far_edge, near_edge))/(float)(M_PI*2.0);
/* Recalculate the panning direction, to be directly between the
* direction of the two extents.
*/
local_dir[0] = far_edge[0] + near_edge[0];
local_dir[1] = far_edge[1] + near_edge[1];
local_dir[2] = far_edge[2] + near_edge[2];
normalize(local_dir);
}
else
{
/* If we get here, the listener is directly in front of or behind
* the center of the portal, making all aperture edges effectively
* equidistant. Calculating the panning magnitude is simplified,
* using the arctangent of the radius and distance.
*/
magnitude = 1.0f - (atan2f(portal_radius, dist) / (float)M_PI);
}
/* Scale the other zone's panning vector. */
other_dir[0] = local_dir[0] * magnitude;
other_dir[1] = local_dir[1] * magnitude;
other_dir[2] = local_dir[2] * magnitude;
/* Pan the current zone to the opposite direction of the portal, and
* take the remaining percentage of the portal's magnitude.
*/
this_dir[0] = local_dir[0] * (magnitude-1.0f);
this_dir[1] = local_dir[1] * (magnitude-1.0f);
this_dir[2] = local_dir[2] * (magnitude-1.0f);
/* Now set the effects' panning vectors and gain. Energy is shared
* between environments, so attenuate according to each zone's
* contribution (note: gain^2 = energy).
*/
alEffectf(this_effect, AL_EAXREVERB_REFLECTIONS_GAIN, this_reverb->flReflectionsGain * sqrtf(magnitude));
alEffectf(this_effect, AL_EAXREVERB_LATE_REVERB_GAIN, this_reverb->flLateReverbGain * sqrtf(magnitude));
alEffectfv(this_effect, AL_EAXREVERB_REFLECTIONS_PAN, this_dir);
alEffectfv(this_effect, AL_EAXREVERB_LATE_REVERB_PAN, this_dir);
alEffectf(other_effect, AL_EAXREVERB_REFLECTIONS_GAIN, other_reverb->flReflectionsGain * sqrtf(1.0f-magnitude));
alEffectf(other_effect, AL_EAXREVERB_LATE_REVERB_GAIN, other_reverb->flLateReverbGain * sqrtf(1.0f-magnitude));
alEffectfv(other_effect, AL_EAXREVERB_REFLECTIONS_PAN, other_dir);
alEffectfv(other_effect, AL_EAXREVERB_LATE_REVERB_PAN, other_dir);
}
else
{
/* 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_REFLECTIONS_GAIN, reverbs[0].flReflectionsGain * sqrtf(0.5f));
alEffectf(effects[0], AL_EAXREVERB_LATE_REVERB_GAIN, reverbs[0].flLateReverbGain * sqrtf(0.5f));
alEffectfv(effects[0], AL_EAXREVERB_REFLECTIONS_PAN, this_dir);
alEffectfv(effects[0], AL_EAXREVERB_LATE_REVERB_PAN, this_dir);
alEffectf(effects[1], AL_EAXREVERB_REFLECTIONS_GAIN, reverbs[1].flReflectionsGain * sqrtf(0.5f));
alEffectf(effects[1], AL_EAXREVERB_LATE_REVERB_GAIN, reverbs[1].flLateReverbGain * sqrtf(0.5f));
alEffectfv(effects[1], AL_EAXREVERB_REFLECTIONS_PAN, other_dir);
alEffectfv(effects[1], AL_EAXREVERB_LATE_REVERB_PAN, other_dir);
}
/* Finally, update the effect slots with the updated effect parameters. */
alAuxiliaryEffectSloti(slots[0], AL_EFFECTSLOT_EFFECT, effects[0]);
alAuxiliaryEffectSloti(slots[1], AL_EFFECTSLOT_EFFECT, effects[1]);
}
int main(int argc, char **argv)
{
static const int MaxTransitions = 8;
EFXEAXREVERBPROPERTIES reverbs[2] = {
EFX_REVERB_PRESET_CARPETEDHALLWAY,
EFX_REVERB_PRESET_BATHROOM
};
ALCdevice *device = NULL;
ALCcontext *context = NULL;
ALuint effects[2] = { 0, 0 };
ALuint slots[2] = { 0, 0 };
ALuint direct_filter = 0;
ALuint buffer = 0;
ALuint source = 0;
ALCint num_sends = 0;
ALenum state = AL_INITIAL;
ALfloat direct_gain = 1.0f;
int basetime = 0;
int loops = 0;
/* Print out usage if no arguments were specified */
if(argc < 2)
{
fprintf(stderr, "Usage: %s [-device <name>] [options] <filename>\n\n"
"Options:\n"
"\t-nodirect\tSilence direct path output (easier to hear reverb)\n\n",
argv[0]);
return 1;
}
/* Initialize OpenAL, and check for EFX support with at least 2 auxiliary
* sends (if multiple sends are supported, 2 are provided by default; if
* you want more, you have to request it through alcCreateContext).
*/
argv++; argc--;
if(InitAL(&argv, &argc) != 0)
return 1;
while(argc > 0)
{
if(strcmp(argv[0], "-nodirect") == 0)
direct_gain = 0.0f;
else
break;
argv++;
argc--;
}
if(argc < 1)
{
fprintf(stderr, "No filename spacified.\n");
CloseAL();
return 1;
}
context = alcGetCurrentContext();
device = alcGetContextsDevice(context);
if(!alcIsExtensionPresent(device, "ALC_EXT_EFX"))
{
fprintf(stderr, "Error: EFX not supported\n");
CloseAL();
return 1;
}
num_sends = 0;
alcGetIntegerv(device, ALC_MAX_AUXILIARY_SENDS, 1, &num_sends);
if(alcGetError(device) != ALC_NO_ERROR || num_sends < 2)
{
fprintf(stderr, "Error: Device does not support multiple sends (got %d, need 2)\n",
num_sends);
CloseAL();
return 1;
}
/* Define a macro to help load the function pointers. */
#define LOAD_PROC(x) ((x) = alGetProcAddress(#x))
LOAD_PROC(alGenFilters);
LOAD_PROC(alDeleteFilters);
LOAD_PROC(alIsFilter);
LOAD_PROC(alFilteri);
LOAD_PROC(alFilteriv);
LOAD_PROC(alFilterf);
LOAD_PROC(alFilterfv);
LOAD_PROC(alGetFilteri);
LOAD_PROC(alGetFilteriv);
LOAD_PROC(alGetFilterf);
LOAD_PROC(alGetFilterfv);
LOAD_PROC(alGenEffects);
LOAD_PROC(alDeleteEffects);
LOAD_PROC(alIsEffect);
LOAD_PROC(alEffecti);
LOAD_PROC(alEffectiv);
LOAD_PROC(alEffectf);
LOAD_PROC(alEffectfv);
LOAD_PROC(alGetEffecti);
LOAD_PROC(alGetEffectiv);
LOAD_PROC(alGetEffectf);
LOAD_PROC(alGetEffectfv);
LOAD_PROC(alGenAuxiliaryEffectSlots);
LOAD_PROC(alDeleteAuxiliaryEffectSlots);
LOAD_PROC(alIsAuxiliaryEffectSlot);
LOAD_PROC(alAuxiliaryEffectSloti);
LOAD_PROC(alAuxiliaryEffectSlotiv);
LOAD_PROC(alAuxiliaryEffectSlotf);
LOAD_PROC(alAuxiliaryEffectSlotfv);
LOAD_PROC(alGetAuxiliaryEffectSloti);
LOAD_PROC(alGetAuxiliaryEffectSlotiv);
LOAD_PROC(alGetAuxiliaryEffectSlotf);
LOAD_PROC(alGetAuxiliaryEffectSlotfv);
#undef LOAD_PROC
/* Initialize SDL_sound. */
Sound_Init();
/* Load the sound into a buffer. */
buffer = LoadSound(argv[0]);
if(!buffer)
{
CloseAL();
Sound_Quit();
return 1;
}
/* Generate two effects for two "zones", and load a reverb into each one.
* Note that unlike single-zone reverb, where you can store one effect per
* preset, for multi-zone reverb you should have one effect per environment
* instance, or one per audible zone. This is because we'll be changing the
* effects' properties in real-time based on the environment instance
* relative to the listener.
*/
alGenEffects(2, effects);
if(!LoadEffect(effects[0], &reverbs[0]) || !LoadEffect(effects[1], &reverbs[1]))
{
alDeleteEffects(2, effects);
alDeleteBuffers(1, &buffer);
Sound_Quit();
CloseAL();
return 1;
}
/* Create the effect slot objects, one for each "active" effect. */
alGenAuxiliaryEffectSlots(2, slots);
/* Tell the effect slots to use the loaded effect objects, with slot 0 for
* Zone 0 and slot 1 for Zone 1. Note that this effectively copies the
* effect properties. Modifying or deleting the effect object afterward
* won't directly affect the effect slot until they're reapplied like this.
*/
alAuxiliaryEffectSloti(slots[0], AL_EFFECTSLOT_EFFECT, effects[0]);
alAuxiliaryEffectSloti(slots[1], AL_EFFECTSLOT_EFFECT, effects[1]);
assert(alGetError()==AL_NO_ERROR && "Failed to set effect slot");
/* For the purposes of this example, prepare a filter that optionally
* silences the direct path which allows us to hear just the reverberation.
* A filter like this is normally used for obstruction, where the path
* directly between the listener and source is blocked (the exact
* properties depending on the type and thickness of the obstructing
* material).
*/
alGenFilters(1, &direct_filter);
alFilteri(direct_filter, AL_FILTER_TYPE, AL_FILTER_LOWPASS);
alFilterf(direct_filter, AL_LOWPASS_GAIN, direct_gain);
assert(alGetError()==AL_NO_ERROR && "Failed to set direct filter");
/* Create the source to play the sound with, place it in front of the
* listener's path in the left zone.
*/
source = 0;
alGenSources(1, &source);
alSourcei(source, AL_LOOPING, AL_TRUE);
alSource3f(source, AL_POSITION, -5.0f, 0.0f, -2.0f);
alSourcei(source, AL_DIRECT_FILTER, direct_filter);
alSourcei(source, AL_BUFFER, buffer);
/* Connect the source to the effect slots. Here, we connect source send 0
* to Zone 0's slot, and send 1 to Zone 1's slot. Filters can be specified
* to occlude the source from each zone by varying amounts; for example, a
* source within a particular zone would be unfiltered, while a source that
* can only see a zone through a window or thin wall may be attenuated for
* that zone.
*/
alSource3i(source, AL_AUXILIARY_SEND_FILTER, slots[0], 0, AL_FILTER_NULL);
alSource3i(source, AL_AUXILIARY_SEND_FILTER, slots[1], 1, AL_FILTER_NULL);
assert(alGetError()==AL_NO_ERROR && "Failed to setup sound source");
/* Get the current time as the base for timing in the main loop. */
basetime = altime_get();
loops = 0;
printf("Transition %d of %d...\n", loops+1, MaxTransitions);
/* Play the sound for a while. */
alSourcePlay(source);
do {
int curtime;
ALfloat timediff;
/* Start a batch update, to ensure all changes apply simultaneously. */
alcSuspendContext(context);
/* Get the current time to track the amount of time that passed.
* Convert the difference to seconds.
*/
curtime = altime_get();
timediff = (ALfloat)(curtime - basetime) / 1000.0f;
/* Avoid negative time deltas, in case of non-monotonic clocks. */
if(timediff < 0.0f)
timediff = 0.0f;
else while(timediff >= 4.0f*((loops&1)+1))
{
/* For this example, each transition occurs over 4 seconds, and
* there's 2 transitions per cycle.
*/
if(++loops < MaxTransitions)
printf("Transition %d of %d...\n", loops+1, MaxTransitions);
if(!(loops&1))
{
/* Cycle completed. Decrease the delta and increase the base
* time to start a new cycle.
*/
timediff -= 8.0f;
basetime += 8000;
}
}
/* Update the listener and effects, and finish the batch. */
UpdateListenerAndEffects(timediff, slots, effects, reverbs);
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;
}