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+/*
+ * OpenAL Multi-Zone Reverb Example
+ *
+ * Copyright (c) 2018 by Chris Robinson <[email protected]>
+ *
+ * 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
+ };
+ struct timespec basetime;
+ 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 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. */
+ altimespec_get(&basetime, AL_TIME_UTC);
+ loops = 0;
+ printf("Transition %d of %d...\n", loops+1, MaxTransitions);
+
+ /* Play the sound for a while. */
+ alSourcePlay(source);
+ do {
+ struct timespec 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.
+ */
+ altimespec_get(&curtime, AL_TIME_UTC);
+ timediff = (ALfloat)(curtime.tv_sec - basetime.tv_sec);
+ timediff += (ALfloat)(curtime.tv_nsec - basetime.tv_nsec) / 1000000000.0f;
+
+ /* 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.tv_sec += 8;
+ }
+ }
+
+ /* 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;
+}