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 /* C doesn't allow casting between function and non-function pointer types, so
  88  * with C99 we need to use a union to reinterpret the pointer type. Pre-C99
  89  * still needs to use a normal cast and live with the warning (C++ is fine with
  90  * a regular reinterpret_cast).
  91  */
  92 #if __STDC_VERSION__ >= 199901L
  93 #define FUNCTION_CAST(T, ptr) (union{void *p; T f;}){ptr}.f
  94 #else
  95 #define FUNCTION_CAST(T, ptr) (T)(ptr)
  96 #endif
  97
  98
  99 /* This stuff defines a simple streaming player object, the same as alstream.c.
 100  * Comments are removed for brevity, see alstream.c for more details.
 101  */
 102 #define NUM_BUFFERS 4
 103 #define BUFFER_SAMPLES 8192
 104
 105 typedef struct StreamPlayer {
 106     ALuint buffers[NUM_BUFFERS];
 107     ALuint source;
 108
 109     SNDFILE *sndfile;
 110     SF_INFO sfinfo;
 111     float *membuf;
 112
 113     ALenum format;
 114 } StreamPlayer;
 115
 116 static StreamPlayer *NewPlayer(void)
 117 {
 118     StreamPlayer *player;
 119
 120     player = calloc(1, sizeof(*player));
 121     assert(player != NULL);
 122
 123     alGenBuffers(NUM_BUFFERS, player->buffers);
 124     assert(alGetError() == AL_NO_ERROR && "Could not create buffers");
 125
 126     alGenSources(1, &player->source);
 127     assert(alGetError() == AL_NO_ERROR && "Could not create source");
 128
 129     alSource3i(player->source, AL_POSITION, 0, 0, -1);
 130     alSourcei(player->source, AL_SOURCE_RELATIVE, AL_TRUE);
 131     alSourcei(player->source, AL_ROLLOFF_FACTOR, 0);
 132     assert(alGetError() == AL_NO_ERROR && "Could not set source parameters");
 133
 134     return player;
 135 }
 136
 137 static void ClosePlayerFile(StreamPlayer *player)
 138 {
 139     if(player->sndfile)
 140         sf_close(player->sndfile);
 141     player->sndfile = NULL;
 142
 143     free(player->membuf);
 144     player->membuf = NULL;
 145 }
 146
 147 static void DeletePlayer(StreamPlayer *player)
 148 {
 149     ClosePlayerFile(player);
 150
 151     alDeleteSources(1, &player->source);
 152     alDeleteBuffers(NUM_BUFFERS, player->buffers);
 153     if(alGetError() != AL_NO_ERROR)
 154         fprintf(stderr, "Failed to delete object IDs\n");
 155
 156     memset(player, 0, sizeof(*player));
 157     free(player);
 158 }
 159
 160 static int OpenPlayerFile(StreamPlayer *player, const char *filename)
 161 {
 162     size_t frame_size;
 163
 164     ClosePlayerFile(player);
 165
 166     player->sndfile = sf_open(filename, SFM_READ, &player->sfinfo);
 167     if(!player->sndfile)
 168     {
 169         fprintf(stderr, "Could not open audio in %s: %s\n", filename, sf_strerror(NULL));
 170         return 0;
 171     }
 172
 173     player->format = AL_NONE;
 174     if(player->sfinfo.channels == 1)
 175         player->format = AL_FORMAT_MONO_FLOAT32;
 176     else if(player->sfinfo.channels == 2)
 177         player->format = AL_FORMAT_STEREO_FLOAT32;
 178     else if(player->sfinfo.channels == 6)
 179         player->format = AL_FORMAT_51CHN32;
 180     else if(player->sfinfo.channels == 3)
 181     {
 182         if(sf_command(player->sndfile, SFC_WAVEX_GET_AMBISONIC, NULL, 0) == SF_AMBISONIC_B_FORMAT)
 183             player->format = AL_FORMAT_BFORMAT2D_FLOAT32;
 184     }
 185     else if(player->sfinfo.channels == 4)
 186     {
 187         if(sf_command(player->sndfile, SFC_WAVEX_GET_AMBISONIC, NULL, 0) == SF_AMBISONIC_B_FORMAT)
 188             player->format = AL_FORMAT_BFORMAT3D_FLOAT32;
 189     }
 190     if(!player->format)
 191     {
 192         fprintf(stderr, "Unsupported channel count: %d\n", player->sfinfo.channels);
 193         sf_close(player->sndfile);
 194         player->sndfile = NULL;
 195         return 0;
 196     }
 197
 198     frame_size = (size_t)(BUFFER_SAMPLES * player->sfinfo.channels) * sizeof(float);
 199     player->membuf = malloc(frame_size);
 200
 201     return 1;
 202 }
 203
 204 static int StartPlayer(StreamPlayer *player)
 205 {
 206     ALsizei i;
 207
 208     alSourceRewind(player->source);
 209     alSourcei(player->source, AL_BUFFER, 0);
 210
 211     for(i = 0;i < NUM_BUFFERS;i++)
 212     {
 213         sf_count_t slen = sf_readf_float(player->sndfile, player->membuf, BUFFER_SAMPLES);
 214         if(slen < 1) break;
 215
 216         slen *= player->sfinfo.channels * (sf_count_t)sizeof(float);
 217         alBufferData(player->buffers[i], player->format, player->membuf, (ALsizei)slen,
 218             player->sfinfo.samplerate);
 219     }
 220     if(alGetError() != AL_NO_ERROR)
 221     {
 222         fprintf(stderr, "Error buffering for playback\n");
 223         return 0;
 224     }
 225
 226     alSourceQueueBuffers(player->source, i, player->buffers);
 227     alSourcePlay(player->source);
 228     if(alGetError() != AL_NO_ERROR)
 229     {
 230         fprintf(stderr, "Error starting playback\n");
 231         return 0;
 232     }
 233
 234     return 1;
 235 }
 236
 237 static int UpdatePlayer(StreamPlayer *player)
 238 {
 239     ALint processed, state;
 240
 241     alGetSourcei(player->source, AL_SOURCE_STATE, &state);
 242     alGetSourcei(player->source, AL_BUFFERS_PROCESSED, &processed);
 243     if(alGetError() != AL_NO_ERROR)
 244     {
 245         fprintf(stderr, "Error checking source state\n");
 246         return 0;
 247     }
 248
 249     while(processed > 0)
 250     {
 251         ALuint bufid;
 252         sf_count_t slen;
 253
 254         alSourceUnqueueBuffers(player->source, 1, &bufid);
 255         processed--;
 256
 257         slen = sf_readf_float(player->sndfile, player->membuf, BUFFER_SAMPLES);
 258         if(slen > 0)
 259         {
 260             slen *= player->sfinfo.channels * (sf_count_t)sizeof(float);
 261             alBufferData(bufid, player->format, player->membuf, (ALsizei)slen,
 262                 player->sfinfo.samplerate);
 263             alSourceQueueBuffers(player->source, 1, &bufid);
 264         }
 265         if(alGetError() != AL_NO_ERROR)
 266         {
 267             fprintf(stderr, "Error buffering data\n");
 268             return 0;
 269         }
 270     }
 271
 272     if(state != AL_PLAYING && state != AL_PAUSED)
 273     {
 274         ALint queued;
 275
 276         alGetSourcei(player->source, AL_BUFFERS_QUEUED, &queued);
 277         if(queued == 0)
 278             return 0;
 279
 280         alSourcePlay(player->source);
 281         if(alGetError() != AL_NO_ERROR)
 282         {
 283             fprintf(stderr, "Error restarting playback\n");
 284             return 0;
 285         }
 286     }
 287
 288     return 1;
 289 }
 290
 291
 292 /* CreateEffect creates a new OpenAL effect object with a convolution reverb
 293  * type, and returns the new effect ID.
 294  */
 295 static ALuint CreateEffect(void)
 296 {
 297     ALuint effect = 0;
 298     ALenum err;
 299
 300     printf("Using Convolution Reverb\n");
 301
 302     /* Create the effect object and set the convolution reverb effect type. */
 303     alGenEffects(1, &effect);
 304     alEffecti(effect, AL_EFFECT_TYPE, AL_EFFECT_CONVOLUTION_REVERB_SOFT);
 305
 306     /* Check if an error occured, and clean up if so. */
 307     err = alGetError();
 308     if(err != AL_NO_ERROR)
 309     {
 310         fprintf(stderr, "OpenAL error: %s\n", alGetString(err));
 311         if(alIsEffect(effect))
 312             alDeleteEffects(1, &effect);
 313         return 0;
 314     }
 315
 316     return effect;
 317 }
 318
 319 /* LoadBuffer loads the named audio file into an OpenAL buffer object, and
 320  * returns the new buffer ID.
 321  */
 322 static ALuint LoadSound(const char *filename)
 323 {
 324     const char *namepart;
 325     ALenum err, format;
 326     ALuint buffer;
 327     SNDFILE *sndfile;
 328     SF_INFO sfinfo;
 329     float *membuf;
 330     sf_count_t num_frames;
 331     ALsizei num_bytes;
 332
 333     /* Open the audio file and check that it's usable. */
 334     sndfile = sf_open(filename, SFM_READ, &sfinfo);
 335     if(!sndfile)
 336     {
 337         fprintf(stderr, "Could not open audio in %s: %s\n", filename, sf_strerror(sndfile));
 338         return 0;
 339     }
 340     if(sfinfo.frames < 1 || sfinfo.frames > (sf_count_t)(INT_MAX/sizeof(float))/sfinfo.channels)
 341     {
 342         fprintf(stderr, "Bad sample count in %s (%" PRId64 ")\n", filename, sfinfo.frames);
 343         sf_close(sndfile);
 344         return 0;
 345     }
 346
 347     /* Get the sound format, and figure out the OpenAL format. Use floats since
 348      * impulse responses will usually have more than 16-bit precision.
 349      */
 350     format = AL_NONE;
 351     if(sfinfo.channels == 1)
 352         format = AL_FORMAT_MONO_FLOAT32;
 353     else if(sfinfo.channels == 2)
 354         format = AL_FORMAT_STEREO_FLOAT32;
 355     else if(sfinfo.channels == 3)
 356     {
 357         if(sf_command(sndfile, SFC_WAVEX_GET_AMBISONIC, NULL, 0) == SF_AMBISONIC_B_FORMAT)
 358             format = AL_FORMAT_BFORMAT2D_FLOAT32;
 359     }
 360     else if(sfinfo.channels == 4)
 361     {
 362         if(sf_command(sndfile, SFC_WAVEX_GET_AMBISONIC, NULL, 0) == SF_AMBISONIC_B_FORMAT)
 363             format = AL_FORMAT_BFORMAT3D_FLOAT32;
 364     }
 365     if(!format)
 366     {
 367         fprintf(stderr, "Unsupported channel count: %d\n", sfinfo.channels);
 368         sf_close(sndfile);
 369         return 0;
 370     }
 371
 372     namepart = strrchr(filename, '/');
 373     if(namepart || (namepart=strrchr(filename, '\\')))
 374         namepart++;
 375     else
 376         namepart = filename;
 377     printf("Loading: %s (%s, %dhz, %" PRId64 " samples / %.2f seconds)\n", namepart,
 378         FormatName(format), sfinfo.samplerate, sfinfo.frames,
 379         (double)sfinfo.frames / sfinfo.samplerate);
 380     fflush(stdout);
 381
 382     /* Decode the whole audio file to a buffer. */
 383     membuf = malloc((size_t)(sfinfo.frames * sfinfo.channels) * sizeof(float));
 384
 385     num_frames = sf_readf_float(sndfile, membuf, sfinfo.frames);
 386     if(num_frames < 1)
 387     {
 388         free(membuf);
 389         sf_close(sndfile);
 390         fprintf(stderr, "Failed to read samples in %s (%" PRId64 ")\n", filename, num_frames);
 391         return 0;
 392     }
 393     num_bytes = (ALsizei)(num_frames * sfinfo.channels) * (ALsizei)sizeof(float);
 394
 395     /* Buffer the audio data into a new buffer object, then free the data and
 396      * close the file.
 397      */
 398     buffer = 0;
 399     alGenBuffers(1, &buffer);
 400     alBufferData(buffer, format, membuf, num_bytes, sfinfo.samplerate);
 401
 402     free(membuf);
 403     sf_close(sndfile);
 404
 405     /* Check if an error occured, and clean up if so. */
 406     err = alGetError();
 407     if(err != AL_NO_ERROR)
 408     {
 409         fprintf(stderr, "OpenAL Error: %s\n", alGetString(err));
 410         if(buffer && alIsBuffer(buffer))
 411             alDeleteBuffers(1, &buffer);
 412         return 0;
 413     }
 414
 415     return buffer;
 416 }
 417
 418
 419 int main(int argc, char **argv)
 420 {
 421     ALuint ir_buffer, filter, effect, slot;
 422     StreamPlayer *player;
 423     int i;
 424
 425     /* Print out usage if no arguments were specified */
 426     if(argc < 2)
 427     {
 428         fprintf(stderr, "Usage: %s [-device <name>] <impulse response file> "
 429             "<[-dry | -nodry] filename>...\n", argv[0]);
 430         return 1;
 431     }
 432
 433     argv++; argc--;
 434     if(InitAL(&argv, &argc) != 0)
 435         return 1;
 436
 437     if(!alIsExtensionPresent("AL_SOFTX_convolution_reverb"))
 438     {
 439         CloseAL();
 440         fprintf(stderr, "Error: Convolution revern not supported\n");
 441         return 1;
 442     }
 443
 444     if(argc < 2)
 445     {
 446         CloseAL();
 447         fprintf(stderr, "Error: Missing impulse response or sound files\n");
 448         return 1;
 449     }
 450
 451     /* Define a macro to help load the function pointers. */
 452 #define LOAD_PROC(T, x)  ((x) = FUNCTION_CAST(T, alGetProcAddress(#x)))
 453     LOAD_PROC(LPALGENFILTERS, alGenFilters);
 454     LOAD_PROC(LPALDELETEFILTERS, alDeleteFilters);
 455     LOAD_PROC(LPALISFILTER, alIsFilter);
 456     LOAD_PROC(LPALFILTERI, alFilteri);
 457     LOAD_PROC(LPALFILTERIV, alFilteriv);
 458     LOAD_PROC(LPALFILTERF, alFilterf);
 459     LOAD_PROC(LPALFILTERFV, alFilterfv);
 460     LOAD_PROC(LPALGETFILTERI, alGetFilteri);
 461     LOAD_PROC(LPALGETFILTERIV, alGetFilteriv);
 462     LOAD_PROC(LPALGETFILTERF, alGetFilterf);
 463     LOAD_PROC(LPALGETFILTERFV, alGetFilterfv);
 464
 465     LOAD_PROC(LPALGENEFFECTS, alGenEffects);
 466     LOAD_PROC(LPALDELETEEFFECTS, alDeleteEffects);
 467     LOAD_PROC(LPALISEFFECT, alIsEffect);
 468     LOAD_PROC(LPALEFFECTI, alEffecti);
 469     LOAD_PROC(LPALEFFECTIV, alEffectiv);
 470     LOAD_PROC(LPALEFFECTF, alEffectf);
 471     LOAD_PROC(LPALEFFECTFV, alEffectfv);
 472     LOAD_PROC(LPALGETEFFECTI, alGetEffecti);
 473     LOAD_PROC(LPALGETEFFECTIV, alGetEffectiv);
 474     LOAD_PROC(LPALGETEFFECTF, alGetEffectf);
 475     LOAD_PROC(LPALGETEFFECTFV, alGetEffectfv);
 476
 477     LOAD_PROC(LPALGENAUXILIARYEFFECTSLOTS, alGenAuxiliaryEffectSlots);
 478     LOAD_PROC(LPALDELETEAUXILIARYEFFECTSLOTS, alDeleteAuxiliaryEffectSlots);
 479     LOAD_PROC(LPALISAUXILIARYEFFECTSLOT, alIsAuxiliaryEffectSlot);
 480     LOAD_PROC(LPALAUXILIARYEFFECTSLOTI, alAuxiliaryEffectSloti);
 481     LOAD_PROC(LPALAUXILIARYEFFECTSLOTIV, alAuxiliaryEffectSlotiv);
 482     LOAD_PROC(LPALAUXILIARYEFFECTSLOTF, alAuxiliaryEffectSlotf);
 483     LOAD_PROC(LPALAUXILIARYEFFECTSLOTFV, alAuxiliaryEffectSlotfv);
 484     LOAD_PROC(LPALGETAUXILIARYEFFECTSLOTI, alGetAuxiliaryEffectSloti);
 485     LOAD_PROC(LPALGETAUXILIARYEFFECTSLOTIV, alGetAuxiliaryEffectSlotiv);
 486     LOAD_PROC(LPALGETAUXILIARYEFFECTSLOTF, alGetAuxiliaryEffectSlotf);
 487     LOAD_PROC(LPALGETAUXILIARYEFFECTSLOTFV, alGetAuxiliaryEffectSlotfv);
 488 #undef LOAD_PROC
 489
 490     /* Load the reverb into an effect. */
 491     effect = CreateEffect();
 492     if(!effect)
 493     {
 494         CloseAL();
 495         return 1;
 496     }
 497
 498     /* Load the impulse response sound into a buffer. */
 499     ir_buffer = LoadSound(argv[0]);
 500     if(!ir_buffer)
 501     {
 502         alDeleteEffects(1, &effect);
 503         CloseAL();
 504         return 1;
 505     }
 506
 507     /* Create the effect slot object. This is what "plays" an effect on sources
 508      * that connect to it.
 509      */
 510     slot = 0;
 511     alGenAuxiliaryEffectSlots(1, &slot);
 512
 513     /* Set the impulse response sound buffer on the effect slot. This allows
 514      * effects to access it as needed. In this case, convolution reverb uses it
 515      * as the filter source. NOTE: Unlike the effect object, the buffer *is*
 516      * kept referenced and may not be changed or deleted as long as it's set,
 517      * just like with a source. When another buffer is set, or the effect slot
 518      * is deleted, the buffer reference is released.
 519      *
 520      * The effect slot's gain is reduced because the impulse responses I've
 521      * tested with result in excessively loud reverb. Is that normal? Even with
 522      * this, it seems a bit on the loud side.
 523      *
 524      * Also note: unlike standard or EAX reverb, there is no automatic
 525      * attenuation of a source's reverb response with distance, so the reverb
 526      * will remain full volume regardless of a given sound's distance from the
 527      * listener. You can use a send filter to alter a given source's
 528      * contribution to reverb.
 529      */
 530     alAuxiliaryEffectSloti(slot, AL_BUFFER, (ALint)ir_buffer);
 531     alAuxiliaryEffectSlotf(slot, AL_EFFECTSLOT_GAIN, 1.0f / 16.0f);
 532     alAuxiliaryEffectSloti(slot, AL_EFFECTSLOT_EFFECT, (ALint)effect);
 533     assert(alGetError()==AL_NO_ERROR && "Failed to set effect slot");
 534
 535     /* Create a filter that can silence the dry path. */
 536     filter = 0;
 537     alGenFilters(1, &filter);
 538     alFilteri(filter, AL_FILTER_TYPE, AL_FILTER_LOWPASS);
 539     alFilterf(filter, AL_LOWPASS_GAIN, 0.0f);
 540
 541     player = NewPlayer();
 542     /* Connect the player's source to the effect slot. */
 543     alSource3i(player->source, AL_AUXILIARY_SEND_FILTER, (ALint)slot, 0, AL_FILTER_NULL);
 544     assert(alGetError()==AL_NO_ERROR && "Failed to setup sound source");
 545
 546     /* Play each file listed on the command line */
 547     for(i = 1;i < argc;i++)
 548     {
 549         const char *namepart;
 550
 551         if(argc-i > 1)
 552         {
 553             if(strcasecmp(argv[i], "-nodry") == 0)
 554             {
 555                 alSourcei(player->source, AL_DIRECT_FILTER, (ALint)filter);
 556                 ++i;
 557             }
 558             else if(strcasecmp(argv[i], "-dry") == 0)
 559             {
 560                 alSourcei(player->source, AL_DIRECT_FILTER, AL_FILTER_NULL);
 561                 ++i;
 562             }
 563         }
 564
 565         if(!OpenPlayerFile(player, argv[i]))
 566             continue;
 567
 568         namepart = strrchr(argv[i], '/');
 569         if(namepart || (namepart=strrchr(argv[i], '\\')))
 570             namepart++;
 571         else
 572             namepart = argv[i];
 573
 574         printf("Playing: %s (%s, %dhz)\n", namepart, FormatName(player->format),
 575             player->sfinfo.samplerate);
 576         fflush(stdout);
 577
 578         if(!StartPlayer(player))
 579         {
 580             ClosePlayerFile(player);
 581             continue;
 582         }
 583
 584         while(UpdatePlayer(player))
 585             al_nssleep(10000000);
 586
 587         ClosePlayerFile(player);
 588     }
 589     printf("Done.\n");
 590
 591     /* All files done. Delete the player and effect resources, and close down
 592      * OpenAL.
 593      */
 594     DeletePlayer(player);
 595     player = NULL;
 596
 597     alDeleteAuxiliaryEffectSlots(1, &slot);
 598     alDeleteEffects(1, &effect);
 599     alDeleteFilters(1, &filter);
 600     alDeleteBuffers(1, &ir_buffer);
 601
 602     CloseAL();
 603
 604     return 0;
 605 }