Merge branch 'v1.19' of git://repo.or.cz/openal-soft into v1.19v1.19

1 files changed, 696 insertions, 0 deletions

diff --git a/examples/almultireverb.c b/examples/almultireverb.c
new file mode 100644
index 00000000..a2587585
--- /dev/null
+++ b/examples/almultireverb.c
@@ -0,0 +1,696 @@
+/*
+ * 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;
+}