examples/alconvolve.c

   1 /*
   2  * OpenAL Convolution Reverb Example
   3  *
   4  * Copyright (c) 2020 by Chris Robinson <chris.kcat@gmail.com>
   5  *
   6  * Permission is hereby granted, free of charge, to any person obtaining a copy
   7  * of this software and associated documentation files (the "Software"), to deal
   8  * in the Software without restriction, including without limitation the rights
   9  * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
  10  * copies of the Software, and to permit persons to whom the Software is
  11  * furnished to do so, subject to the following conditions:
  12  *
  13  * The above copyright notice and this permission notice shall be included in
  14  * all copies or substantial portions of the Software.
  15  *
  16  * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
  17  * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
  18  * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
  19  * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
  20  * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
  21  * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
  22  * THE SOFTWARE.
  23  */
  24
  25 /* This file contains an example for applying convolution reverb to a source. */
  26
  27 #include <assert.h>
  28 #include <inttypes.h>
  29 #include <limits.h>
  30 #include <stdio.h>
  31 #include <stdlib.h>
  32 #include <string.h>
  33
  34 #include "sndfile.h"
  35
  36 #include "AL/al.h"
  37 #include "AL/alext.h"
  38
  39 #include "common/alhelpers.h"
  40
  41
  42 #ifndef AL_SOFT_convolution_reverb
  43 #define AL_SOFT_convolution_reverb
  44 #define AL_EFFECT_CONVOLUTION_REVERB_SOFT        0xA000
  45 #endif
  46
  47
  48 /* Filter object functions */
  49 static LPALGENFILTERS alGenFilters;
  50 static LPALDELETEFILTERS alDeleteFilters;
  51 static LPALISFILTER alIsFilter;
  52 static LPALFILTERI alFilteri;
  53 static LPALFILTERIV alFilteriv;
  54 static LPALFILTERF alFilterf;
  55 static LPALFILTERFV alFilterfv;
  56 static LPALGETFILTERI alGetFilteri;
  57 static LPALGETFILTERIV alGetFilteriv;
  58 static LPALGETFILTERF alGetFilterf;
  59 static LPALGETFILTERFV alGetFilterfv;
  60
  61 /* Effect object functions */
  62 static LPALGENEFFECTS alGenEffects;
  63 static LPALDELETEEFFECTS alDeleteEffects;
  64 static LPALISEFFECT alIsEffect;
  65 static LPALEFFECTI alEffecti;
  66 static LPALEFFECTIV alEffectiv;
  67 static LPALEFFECTF alEffectf;
  68 static LPALEFFECTFV alEffectfv;
  69 static LPALGETEFFECTI alGetEffecti;
  70 static LPALGETEFFECTIV alGetEffectiv;
  71 static LPALGETEFFECTF alGetEffectf;
  72 static LPALGETEFFECTFV alGetEffectfv;
  73
  74 /* Auxiliary Effect Slot object functions */
  75 static LPALGENAUXILIARYEFFECTSLOTS alGenAuxiliaryEffectSlots;
  76 static LPALDELETEAUXILIARYEFFECTSLOTS alDeleteAuxiliaryEffectSlots;
  77 static LPALISAUXILIARYEFFECTSLOT alIsAuxiliaryEffectSlot;
  78 static LPALAUXILIARYEFFECTSLOTI alAuxiliaryEffectSloti;
  79 static LPALAUXILIARYEFFECTSLOTIV alAuxiliaryEffectSlotiv;
  80 static LPALAUXILIARYEFFECTSLOTF alAuxiliaryEffectSlotf;
  81 static LPALAUXILIARYEFFECTSLOTFV alAuxiliaryEffectSlotfv;
  82 static LPALGETAUXILIARYEFFECTSLOTI alGetAuxiliaryEffectSloti;
  83 static LPALGETAUXILIARYEFFECTSLOTIV alGetAuxiliaryEffectSlotiv;
  84 static LPALGETAUXILIARYEFFECTSLOTF alGetAuxiliaryEffectSlotf;
  85 static LPALGETAUXILIARYEFFECTSLOTFV alGetAuxiliaryEffectSlotfv;
  86
  87
  88 /* This stuff defines a simple streaming player object, the same as alstream.c.
  89  * Comments are removed for brevity, see alstream.c for more details.
  90  */
  91 #define NUM_BUFFERS 4
  92 #define BUFFER_SAMPLES 8192
  93
  94 typedef struct StreamPlayer {
  95     ALuint buffers[NUM_BUFFERS];
  96     ALuint source;
  97
  98     SNDFILE *sndfile;
  99     SF_INFO sfinfo;
 100     float *membuf;
 101
 102     ALenum format;
 103 } StreamPlayer;
 104
 105 static StreamPlayer *NewPlayer(void)
 106 {
 107     StreamPlayer *player;
 108
 109     player = calloc(1, sizeof(*player));
 110     assert(player != NULL);
 111
 112     alGenBuffers(NUM_BUFFERS, player->buffers);
 113     assert(alGetError() == AL_NO_ERROR && "Could not create buffers");
 114
 115     alGenSources(1, &player->source);
 116     assert(alGetError() == AL_NO_ERROR && "Could not create source");
 117
 118     alSource3i(player->source, AL_POSITION, 0, 0, -1);
 119     alSourcei(player->source, AL_SOURCE_RELATIVE, AL_TRUE);
 120     alSourcei(player->source, AL_ROLLOFF_FACTOR, 0);
 121     assert(alGetError() == AL_NO_ERROR && "Could not set source parameters");
 122
 123     return player;
 124 }
 125
 126 static void ClosePlayerFile(StreamPlayer *player)
 127 {
 128     if(player->sndfile)
 129         sf_close(player->sndfile);
 130     player->sndfile = NULL;
 131
 132     free(player->membuf);
 133     player->membuf = NULL;
 134 }
 135
 136 static void DeletePlayer(StreamPlayer *player)
 137 {
 138     ClosePlayerFile(player);
 139
 140     alDeleteSources(1, &player->source);
 141     alDeleteBuffers(NUM_BUFFERS, player->buffers);
 142     if(alGetError() != AL_NO_ERROR)
 143         fprintf(stderr, "Failed to delete object IDs\n");
 144
 145     memset(player, 0, sizeof(*player));
 146     free(player);
 147 }
 148
 149 static int OpenPlayerFile(StreamPlayer *player, const char *filename)
 150 {
 151     size_t frame_size;
 152
 153     ClosePlayerFile(player);
 154
 155     player->sndfile = sf_open(filename, SFM_READ, &player->sfinfo);
 156     if(!player->sndfile)
 157     {
 158         fprintf(stderr, "Could not open audio in %s: %s\n", filename, sf_strerror(NULL));
 159         return 0;
 160     }
 161
 162     player->format = AL_NONE;
 163     if(player->sfinfo.channels == 1)
 164         player->format = AL_FORMAT_MONO_FLOAT32;
 165     else if(player->sfinfo.channels == 2)
 166         player->format = AL_FORMAT_STEREO_FLOAT32;
 167     else if(player->sfinfo.channels == 6)
 168         player->format = AL_FORMAT_51CHN32;
 169     else if(player->sfinfo.channels == 3)
 170     {
 171         if(sf_command(player->sndfile, SFC_WAVEX_GET_AMBISONIC, NULL, 0) == SF_AMBISONIC_B_FORMAT)
 172             player->format = AL_FORMAT_BFORMAT2D_FLOAT32;
 173     }
 174     else if(player->sfinfo.channels == 4)
 175     {
 176         if(sf_command(player->sndfile, SFC_WAVEX_GET_AMBISONIC, NULL, 0) == SF_AMBISONIC_B_FORMAT)
 177             player->format = AL_FORMAT_BFORMAT3D_FLOAT32;
 178     }
 179     if(!player->format)
 180     {
 181         fprintf(stderr, "Unsupported channel count: %d\n", player->sfinfo.channels);
 182         sf_close(player->sndfile);
 183         player->sndfile = NULL;
 184         return 0;
 185     }
 186
 187     frame_size = (size_t)(BUFFER_SAMPLES * player->sfinfo.channels) * sizeof(float);
 188     player->membuf = malloc(frame_size);
 189
 190     return 1;
 191 }
 192
 193 static int StartPlayer(StreamPlayer *player)
 194 {
 195     ALsizei i;
 196
 197     alSourceRewind(player->source);
 198     alSourcei(player->source, AL_BUFFER, 0);
 199
 200     for(i = 0;i < NUM_BUFFERS;i++)
 201     {
 202         sf_count_t slen = sf_readf_float(player->sndfile, player->membuf, BUFFER_SAMPLES);
 203         if(slen < 1) break;
 204
 205         slen *= player->sfinfo.channels * (sf_count_t)sizeof(float);
 206         alBufferData(player->buffers[i], player->format, player->membuf, (ALsizei)slen,
 207             player->sfinfo.samplerate);
 208     }
 209     if(alGetError() != AL_NO_ERROR)
 210     {
 211         fprintf(stderr, "Error buffering for playback\n");
 212         return 0;
 213     }
 214
 215     alSourceQueueBuffers(player->source, i, player->buffers);
 216     alSourcePlay(player->source);
 217     if(alGetError() != AL_NO_ERROR)
 218     {
 219         fprintf(stderr, "Error starting playback\n");
 220         return 0;
 221     }
 222
 223     return 1;
 224 }
 225
 226 static int UpdatePlayer(StreamPlayer *player)
 227 {
 228     ALint processed, state;
 229
 230     alGetSourcei(player->source, AL_SOURCE_STATE, &state);
 231     alGetSourcei(player->source, AL_BUFFERS_PROCESSED, &processed);
 232     if(alGetError() != AL_NO_ERROR)
 233     {
 234         fprintf(stderr, "Error checking source state\n");
 235         return 0;
 236     }
 237
 238     while(processed > 0)
 239     {
 240         ALuint bufid;
 241         sf_count_t slen;
 242
 243         alSourceUnqueueBuffers(player->source, 1, &bufid);
 244         processed--;
 245
 246         slen = sf_readf_float(player->sndfile, player->membuf, BUFFER_SAMPLES);
 247         if(slen > 0)
 248         {
 249             slen *= player->sfinfo.channels * (sf_count_t)sizeof(float);
 250             alBufferData(bufid, player->format, player->membuf, (ALsizei)slen,
 251                 player->sfinfo.samplerate);
 252             alSourceQueueBuffers(player->source, 1, &bufid);
 253         }
 254         if(alGetError() != AL_NO_ERROR)
 255         {
 256             fprintf(stderr, "Error buffering data\n");
 257             return 0;
 258         }
 259     }
 260
 261     if(state != AL_PLAYING && state != AL_PAUSED)
 262     {
 263         ALint queued;
 264
 265         alGetSourcei(player->source, AL_BUFFERS_QUEUED, &queued);
 266         if(queued == 0)
 267             return 0;
 268
 269         alSourcePlay(player->source);
 270         if(alGetError() != AL_NO_ERROR)
 271         {
 272             fprintf(stderr, "Error restarting playback\n");
 273             return 0;
 274         }
 275     }
 276
 277     return 1;
 278 }
 279
 280
 281 /* CreateEffect creates a new OpenAL effect object with a convolution reverb
 282  * type, and returns the new effect ID.
 283  */
 284 static ALuint CreateEffect(void)
 285 {
 286     ALuint effect = 0;
 287     ALenum err;
 288
 289     printf("Using Convolution Reverb\n");
 290
 291     /* Create the effect object and set the convolution reverb effect type. */
 292     alGenEffects(1, &effect);
 293     alEffecti(effect, AL_EFFECT_TYPE, AL_EFFECT_CONVOLUTION_REVERB_SOFT);
 294
 295     /* Check if an error occured, and clean up if so. */
 296     err = alGetError();
 297     if(err != AL_NO_ERROR)
 298     {
 299         fprintf(stderr, "OpenAL error: %s\n", alGetString(err));
 300         if(alIsEffect(effect))
 301             alDeleteEffects(1, &effect);
 302         return 0;
 303     }
 304
 305     return effect;
 306 }
 307
 308 /* LoadBuffer loads the named audio file into an OpenAL buffer object, and
 309  * returns the new buffer ID.
 310  */
 311 static ALuint LoadSound(const char *filename)
 312 {
 313     const char *namepart;
 314     ALenum err, format;
 315     ALuint buffer;
 316     SNDFILE *sndfile;
 317     SF_INFO sfinfo;
 318     float *membuf;
 319     sf_count_t num_frames;
 320     ALsizei num_bytes;
 321
 322     /* Open the audio file and check that it's usable. */
 323     sndfile = sf_open(filename, SFM_READ, &sfinfo);
 324     if(!sndfile)
 325     {
 326         fprintf(stderr, "Could not open audio in %s: %s\n", filename, sf_strerror(sndfile));
 327         return 0;
 328     }
 329     if(sfinfo.frames < 1 || sfinfo.frames > (sf_count_t)(INT_MAX/sizeof(float))/sfinfo.channels)
 330     {
 331         fprintf(stderr, "Bad sample count in %s (%" PRId64 ")\n", filename, sfinfo.frames);
 332         sf_close(sndfile);
 333         return 0;
 334     }
 335
 336     /* Get the sound format, and figure out the OpenAL format. Use floats since
 337      * impulse responses will usually have more than 16-bit precision.
 338      */
 339     format = AL_NONE;
 340     if(sfinfo.channels == 1)
 341         format = AL_FORMAT_MONO_FLOAT32;
 342     else if(sfinfo.channels == 2)
 343         format = AL_FORMAT_STEREO_FLOAT32;
 344     else if(sfinfo.channels == 3)
 345     {
 346         if(sf_command(sndfile, SFC_WAVEX_GET_AMBISONIC, NULL, 0) == SF_AMBISONIC_B_FORMAT)
 347             format = AL_FORMAT_BFORMAT2D_FLOAT32;
 348     }
 349     else if(sfinfo.channels == 4)
 350     {
 351         if(sf_command(sndfile, SFC_WAVEX_GET_AMBISONIC, NULL, 0) == SF_AMBISONIC_B_FORMAT)
 352             format = AL_FORMAT_BFORMAT3D_FLOAT32;
 353     }
 354     if(!format)
 355     {
 356         fprintf(stderr, "Unsupported channel count: %d\n", sfinfo.channels);
 357         sf_close(sndfile);
 358         return 0;
 359     }
 360
 361     namepart = strrchr(filename, '/');
 362     if(namepart || (namepart=strrchr(filename, '\\')))
 363         namepart++;
 364     else
 365         namepart = filename;
 366     printf("Loading: %s (%s, %dhz, %" PRId64 " samples / %.2f seconds)\n", namepart,
 367         FormatName(format), sfinfo.samplerate, sfinfo.frames,
 368         (double)sfinfo.frames / sfinfo.samplerate);
 369     fflush(stdout);
 370
 371     /* Decode the whole audio file to a buffer. */
 372     membuf = malloc((size_t)(sfinfo.frames * sfinfo.channels) * sizeof(float));
 373
 374     num_frames = sf_readf_float(sndfile, membuf, sfinfo.frames);
 375     if(num_frames < 1)
 376     {
 377         free(membuf);
 378         sf_close(sndfile);
 379         fprintf(stderr, "Failed to read samples in %s (%" PRId64 ")\n", filename, num_frames);
 380         return 0;
 381     }
 382     num_bytes = (ALsizei)(num_frames * sfinfo.channels) * (ALsizei)sizeof(float);
 383
 384     /* Buffer the audio data into a new buffer object, then free the data and
 385      * close the file.
 386      */
 387     buffer = 0;
 388     alGenBuffers(1, &buffer);
 389     alBufferData(buffer, format, membuf, num_bytes, sfinfo.samplerate);
 390
 391     free(membuf);
 392     sf_close(sndfile);
 393
 394     /* Check if an error occured, and clean up if so. */
 395     err = alGetError();
 396     if(err != AL_NO_ERROR)
 397     {
 398         fprintf(stderr, "OpenAL Error: %s\n", alGetString(err));
 399         if(buffer && alIsBuffer(buffer))
 400             alDeleteBuffers(1, &buffer);
 401         return 0;
 402     }
 403
 404     return buffer;
 405 }
 406
 407
 408 int main(int argc, char **argv)
 409 {
 410     ALuint ir_buffer, filter, effect, slot;
 411     StreamPlayer *player;
 412     int i;
 413
 414     /* Print out usage if no arguments were specified */
 415     if(argc < 2)
 416     {
 417         fprintf(stderr, "Usage: %s [-device <name>] <impulse response file> "
 418             "<[-dry | -nodry] filename>...\n", argv[0]);
 419         return 1;
 420     }
 421
 422     argv++; argc--;
 423     if(InitAL(&argv, &argc) != 0)
 424         return 1;
 425
 426     if(!alIsExtensionPresent("AL_SOFTX_convolution_reverb"))
 427     {
 428         CloseAL();
 429         fprintf(stderr, "Error: Convolution revern not supported\n");
 430         return 1;
 431     }
 432
 433     if(argc < 2)
 434     {
 435         CloseAL();
 436         fprintf(stderr, "Error: Missing impulse response or sound files\n");
 437         return 1;
 438     }
 439
 440     /* Define a macro to help load the function pointers. */
 441 #define LOAD_PROC(T, x)  ((x) = FUNCTION_CAST(T, alGetProcAddress(#x)))
 442     LOAD_PROC(LPALGENFILTERS, alGenFilters);
 443     LOAD_PROC(LPALDELETEFILTERS, alDeleteFilters);
 444     LOAD_PROC(LPALISFILTER, alIsFilter);
 445     LOAD_PROC(LPALFILTERI, alFilteri);
 446     LOAD_PROC(LPALFILTERIV, alFilteriv);
 447     LOAD_PROC(LPALFILTERF, alFilterf);
 448     LOAD_PROC(LPALFILTERFV, alFilterfv);
 449     LOAD_PROC(LPALGETFILTERI, alGetFilteri);
 450     LOAD_PROC(LPALGETFILTERIV, alGetFilteriv);
 451     LOAD_PROC(LPALGETFILTERF, alGetFilterf);
 452     LOAD_PROC(LPALGETFILTERFV, alGetFilterfv);
 453
 454     LOAD_PROC(LPALGENEFFECTS, alGenEffects);
 455     LOAD_PROC(LPALDELETEEFFECTS, alDeleteEffects);
 456     LOAD_PROC(LPALISEFFECT, alIsEffect);
 457     LOAD_PROC(LPALEFFECTI, alEffecti);
 458     LOAD_PROC(LPALEFFECTIV, alEffectiv);
 459     LOAD_PROC(LPALEFFECTF, alEffectf);
 460     LOAD_PROC(LPALEFFECTFV, alEffectfv);
 461     LOAD_PROC(LPALGETEFFECTI, alGetEffecti);
 462     LOAD_PROC(LPALGETEFFECTIV, alGetEffectiv);
 463     LOAD_PROC(LPALGETEFFECTF, alGetEffectf);
 464     LOAD_PROC(LPALGETEFFECTFV, alGetEffectfv);
 465
 466     LOAD_PROC(LPALGENAUXILIARYEFFECTSLOTS, alGenAuxiliaryEffectSlots);
 467     LOAD_PROC(LPALDELETEAUXILIARYEFFECTSLOTS, alDeleteAuxiliaryEffectSlots);
 468     LOAD_PROC(LPALISAUXILIARYEFFECTSLOT, alIsAuxiliaryEffectSlot);
 469     LOAD_PROC(LPALAUXILIARYEFFECTSLOTI, alAuxiliaryEffectSloti);
 470     LOAD_PROC(LPALAUXILIARYEFFECTSLOTIV, alAuxiliaryEffectSlotiv);
 471     LOAD_PROC(LPALAUXILIARYEFFECTSLOTF, alAuxiliaryEffectSlotf);
 472     LOAD_PROC(LPALAUXILIARYEFFECTSLOTFV, alAuxiliaryEffectSlotfv);
 473     LOAD_PROC(LPALGETAUXILIARYEFFECTSLOTI, alGetAuxiliaryEffectSloti);
 474     LOAD_PROC(LPALGETAUXILIARYEFFECTSLOTIV, alGetAuxiliaryEffectSlotiv);
 475     LOAD_PROC(LPALGETAUXILIARYEFFECTSLOTF, alGetAuxiliaryEffectSlotf);
 476     LOAD_PROC(LPALGETAUXILIARYEFFECTSLOTFV, alGetAuxiliaryEffectSlotfv);
 477 #undef LOAD_PROC
 478
 479     /* Load the reverb into an effect. */
 480     effect = CreateEffect();
 481     if(!effect)
 482     {
 483         CloseAL();
 484         return 1;
 485     }
 486
 487     /* Load the impulse response sound into a buffer. */
 488     ir_buffer = LoadSound(argv[0]);
 489     if(!ir_buffer)
 490     {
 491         alDeleteEffects(1, &effect);
 492         CloseAL();
 493         return 1;
 494     }
 495
 496     /* Create the effect slot object. This is what "plays" an effect on sources
 497      * that connect to it.
 498      */
 499     slot = 0;
 500     alGenAuxiliaryEffectSlots(1, &slot);
 501
 502     /* Set the impulse response sound buffer on the effect slot. This allows
 503      * effects to access it as needed. In this case, convolution reverb uses it
 504      * as the filter source. NOTE: Unlike the effect object, the buffer *is*
 505      * kept referenced and may not be changed or deleted as long as it's set,
 506      * just like with a source. When another buffer is set, or the effect slot
 507      * is deleted, the buffer reference is released.
 508      *
 509      * The effect slot's gain is reduced because the impulse responses I've
 510      * tested with result in excessively loud reverb. Is that normal? Even with
 511      * this, it seems a bit on the loud side.
 512      *
 513      * Also note: unlike standard or EAX reverb, there is no automatic
 514      * attenuation of a source's reverb response with distance, so the reverb
 515      * will remain full volume regardless of a given sound's distance from the
 516      * listener. You can use a send filter to alter a given source's
 517      * contribution to reverb.
 518      */
 519     alAuxiliaryEffectSloti(slot, AL_BUFFER, (ALint)ir_buffer);
 520     alAuxiliaryEffectSlotf(slot, AL_EFFECTSLOT_GAIN, 1.0f / 16.0f);
 521     alAuxiliaryEffectSloti(slot, AL_EFFECTSLOT_EFFECT, (ALint)effect);
 522     assert(alGetError()==AL_NO_ERROR && "Failed to set effect slot");
 523
 524     /* Create a filter that can silence the dry path. */
 525     filter = 0;
 526     alGenFilters(1, &filter);
 527     alFilteri(filter, AL_FILTER_TYPE, AL_FILTER_LOWPASS);
 528     alFilterf(filter, AL_LOWPASS_GAIN, 0.0f);
 529
 530     player = NewPlayer();
 531     /* Connect the player's source to the effect slot. */
 532     alSource3i(player->source, AL_AUXILIARY_SEND_FILTER, (ALint)slot, 0, AL_FILTER_NULL);
 533     assert(alGetError()==AL_NO_ERROR && "Failed to setup sound source");
 534
 535     /* Play each file listed on the command line */
 536     for(i = 1;i < argc;i++)
 537     {
 538         const char *namepart;
 539
 540         if(argc-i > 1)
 541         {
 542             if(strcasecmp(argv[i], "-nodry") == 0)
 543             {
 544                 alSourcei(player->source, AL_DIRECT_FILTER, (ALint)filter);
 545                 ++i;
 546             }
 547             else if(strcasecmp(argv[i], "-dry") == 0)
 548             {
 549                 alSourcei(player->source, AL_DIRECT_FILTER, AL_FILTER_NULL);
 550                 ++i;
 551             }
 552         }
 553
 554         if(!OpenPlayerFile(player, argv[i]))
 555             continue;
 556
 557         namepart = strrchr(argv[i], '/');
 558         if(namepart || (namepart=strrchr(argv[i], '\\')))
 559             namepart++;
 560         else
 561             namepart = argv[i];
 562
 563         printf("Playing: %s (%s, %dhz)\n", namepart, FormatName(player->format),
 564             player->sfinfo.samplerate);
 565         fflush(stdout);
 566
 567         if(!StartPlayer(player))
 568         {
 569             ClosePlayerFile(player);
 570             continue;
 571         }
 572
 573         while(UpdatePlayer(player))
 574             al_nssleep(10000000);
 575
 576         ClosePlayerFile(player);
 577     }
 578     printf("Done.\n");
 579
 580     /* All files done. Delete the player and effect resources, and close down
 581      * OpenAL.
 582      */
 583     DeletePlayer(player);
 584     player = NULL;
 585
 586     alDeleteAuxiliaryEffectSlots(1, &slot);
 587     alDeleteEffects(1, &effect);
 588     alDeleteFilters(1, &filter);
 589     alDeleteBuffers(1, &ir_buffer);
 590
 591     CloseAL();
 592
 593     return 0;
 594 }