/* * OpenAL Multi-Zone Reverb Example * * Copyright (c) 2018 by Chris Robinson * * 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 #include #include #include #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 * right-handed coordinates instead of left-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 ] [options] \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; }