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 /* Effect object functions */
  49 static LPALGENEFFECTS alGenEffects;
  50 static LPALDELETEEFFECTS alDeleteEffects;
  51 static LPALISEFFECT alIsEffect;
  52 static LPALEFFECTI alEffecti;
  53 static LPALEFFECTIV alEffectiv;
  54 static LPALEFFECTF alEffectf;
  55 static LPALEFFECTFV alEffectfv;
  56 static LPALGETEFFECTI alGetEffecti;
  57 static LPALGETEFFECTIV alGetEffectiv;
  58 static LPALGETEFFECTF alGetEffectf;
  59 static LPALGETEFFECTFV alGetEffectfv;
  60
  61 /* Auxiliary Effect Slot object functions */
  62 static LPALGENAUXILIARYEFFECTSLOTS alGenAuxiliaryEffectSlots;
  63 static LPALDELETEAUXILIARYEFFECTSLOTS alDeleteAuxiliaryEffectSlots;
  64 static LPALISAUXILIARYEFFECTSLOT alIsAuxiliaryEffectSlot;
  65 static LPALAUXILIARYEFFECTSLOTI alAuxiliaryEffectSloti;
  66 static LPALAUXILIARYEFFECTSLOTIV alAuxiliaryEffectSlotiv;
  67 static LPALAUXILIARYEFFECTSLOTF alAuxiliaryEffectSlotf;
  68 static LPALAUXILIARYEFFECTSLOTFV alAuxiliaryEffectSlotfv;
  69 static LPALGETAUXILIARYEFFECTSLOTI alGetAuxiliaryEffectSloti;
  70 static LPALGETAUXILIARYEFFECTSLOTIV alGetAuxiliaryEffectSlotiv;
  71 static LPALGETAUXILIARYEFFECTSLOTF alGetAuxiliaryEffectSlotf;
  72 static LPALGETAUXILIARYEFFECTSLOTFV alGetAuxiliaryEffectSlotfv;
  73
  74
  75 /* This stuff defines a simple streaming player object, the same as alstream.c.
  76  * Comments are removed for brevity, see alstream.c for more details.
  77  */
  78 #define NUM_BUFFERS 4
  79 #define BUFFER_SAMPLES 8192
  80
  81 typedef struct StreamPlayer {
  82     ALuint buffers[NUM_BUFFERS];
  83     ALuint source;
  84
  85     SNDFILE *sndfile;
  86     SF_INFO sfinfo;
  87     float *membuf;
  88
  89     ALenum format;
  90 } StreamPlayer;
  91
  92 static StreamPlayer *NewPlayer(void)
  93 {
  94     StreamPlayer *player;
  95
  96     player = calloc(1, sizeof(*player));
  97     assert(player != NULL);
  98
  99     alGenBuffers(NUM_BUFFERS, player->buffers);
 100     assert(alGetError() == AL_NO_ERROR && "Could not create buffers");
 101
 102     alGenSources(1, &player->source);
 103     assert(alGetError() == AL_NO_ERROR && "Could not create source");
 104
 105     alSource3i(player->source, AL_POSITION, 0, 0, -1);
 106     alSourcei(player->source, AL_SOURCE_RELATIVE, AL_TRUE);
 107     alSourcei(player->source, AL_ROLLOFF_FACTOR, 0);
 108     assert(alGetError() == AL_NO_ERROR && "Could not set source parameters");
 109
 110     return player;
 111 }
 112
 113 static void ClosePlayerFile(StreamPlayer *player)
 114 {
 115     if(player->sndfile)
 116         sf_close(player->sndfile);
 117     player->sndfile = NULL;
 118
 119     free(player->membuf);
 120     player->membuf = NULL;
 121 }
 122
 123 static void DeletePlayer(StreamPlayer *player)
 124 {
 125     ClosePlayerFile(player);
 126
 127     alDeleteSources(1, &player->source);
 128     alDeleteBuffers(NUM_BUFFERS, player->buffers);
 129     if(alGetError() != AL_NO_ERROR)
 130         fprintf(stderr, "Failed to delete object IDs\n");
 131
 132     memset(player, 0, sizeof(*player));
 133     free(player);
 134 }
 135
 136 static int OpenPlayerFile(StreamPlayer *player, const char *filename)
 137 {
 138     size_t frame_size;
 139
 140     ClosePlayerFile(player);
 141
 142     player->sndfile = sf_open(filename, SFM_READ, &player->sfinfo);
 143     if(!player->sndfile)
 144     {
 145         fprintf(stderr, "Could not open audio in %s: %s\n", filename, sf_strerror(NULL));
 146         return 0;
 147     }
 148
 149     if(player->sfinfo.channels == 1)
 150         player->format = AL_FORMAT_MONO_FLOAT32;
 151     else if(player->sfinfo.channels == 2)
 152         player->format = AL_FORMAT_STEREO_FLOAT32;
 153     else if(player->sfinfo.channels == 6)
 154         player->format = AL_FORMAT_51CHN32;
 155     else
 156     {
 157         fprintf(stderr, "Unsupported channel count: %d\n", player->sfinfo.channels);
 158         sf_close(player->sndfile);
 159         player->sndfile = NULL;
 160         return 0;
 161     }
 162
 163     frame_size = (size_t)(BUFFER_SAMPLES * player->sfinfo.channels) * sizeof(float);
 164     player->membuf = malloc(frame_size);
 165
 166     return 1;
 167 }
 168
 169 static int StartPlayer(StreamPlayer *player)
 170 {
 171     ALsizei i;
 172
 173     alSourceRewind(player->source);
 174     alSourcei(player->source, AL_BUFFER, 0);
 175
 176     for(i = 0;i < NUM_BUFFERS;i++)
 177     {
 178         sf_count_t slen = sf_readf_float(player->sndfile, player->membuf, BUFFER_SAMPLES);
 179         if(slen < 1) break;
 180
 181         slen *= player->sfinfo.channels * (sf_count_t)sizeof(float);
 182         alBufferData(player->buffers[i], player->format, player->membuf, (ALsizei)slen,
 183             player->sfinfo.samplerate);
 184     }
 185     if(alGetError() != AL_NO_ERROR)
 186     {
 187         fprintf(stderr, "Error buffering for playback\n");
 188         return 0;
 189     }
 190
 191     alSourceQueueBuffers(player->source, i, player->buffers);
 192     alSourcePlay(player->source);
 193     if(alGetError() != AL_NO_ERROR)
 194     {
 195         fprintf(stderr, "Error starting playback\n");
 196         return 0;
 197     }
 198
 199     return 1;
 200 }
 201
 202 static int UpdatePlayer(StreamPlayer *player)
 203 {
 204     ALint processed, state;
 205
 206     alGetSourcei(player->source, AL_SOURCE_STATE, &state);
 207     alGetSourcei(player->source, AL_BUFFERS_PROCESSED, &processed);
 208     if(alGetError() != AL_NO_ERROR)
 209     {
 210         fprintf(stderr, "Error checking source state\n");
 211         return 0;
 212     }
 213
 214     while(processed > 0)
 215     {
 216         ALuint bufid;
 217         sf_count_t slen;
 218
 219         alSourceUnqueueBuffers(player->source, 1, &bufid);
 220         processed--;
 221
 222         slen = sf_readf_float(player->sndfile, player->membuf, BUFFER_SAMPLES);
 223         if(slen > 0)
 224         {
 225             slen *= player->sfinfo.channels * (sf_count_t)sizeof(float);
 226             alBufferData(bufid, player->format, player->membuf, (ALsizei)slen,
 227                 player->sfinfo.samplerate);
 228             alSourceQueueBuffers(player->source, 1, &bufid);
 229         }
 230         if(alGetError() != AL_NO_ERROR)
 231         {
 232             fprintf(stderr, "Error buffering data\n");
 233             return 0;
 234         }
 235     }
 236
 237     if(state != AL_PLAYING && state != AL_PAUSED)
 238     {
 239         ALint queued;
 240
 241         alGetSourcei(player->source, AL_BUFFERS_QUEUED, &queued);
 242         if(queued == 0)
 243             return 0;
 244
 245         alSourcePlay(player->source);
 246         if(alGetError() != AL_NO_ERROR)
 247         {
 248             fprintf(stderr, "Error restarting playback\n");
 249             return 0;
 250         }
 251     }
 252
 253     return 1;
 254 }
 255
 256
 257 /* CreateEffect creates a new OpenAL effect object with a convolution reverb
 258  * type, and returns the new effect ID.
 259  */
 260 static ALuint CreateEffect(void)
 261 {
 262     ALuint effect = 0;
 263     ALenum err;
 264
 265     printf("Using Convolution Reverb\n");
 266
 267     /* Create the effect object and set the convolution reverb effect type. */
 268     alGenEffects(1, &effect);
 269     alEffecti(effect, AL_EFFECT_TYPE, AL_EFFECT_CONVOLUTION_REVERB_SOFT);
 270
 271     /* Check if an error occured, and clean up if so. */
 272     err = alGetError();
 273     if(err != AL_NO_ERROR)
 274     {
 275         fprintf(stderr, "OpenAL error: %s\n", alGetString(err));
 276         if(alIsEffect(effect))
 277             alDeleteEffects(1, &effect);
 278         return 0;
 279     }
 280
 281     return effect;
 282 }
 283
 284 /* LoadBuffer loads the named audio file into an OpenAL buffer object, and
 285  * returns the new buffer ID.
 286  */
 287 static ALuint LoadSound(const char *filename)
 288 {
 289     const char *namepart;
 290     ALenum err, format;
 291     ALuint buffer;
 292     SNDFILE *sndfile;
 293     SF_INFO sfinfo;
 294     float *membuf;
 295     sf_count_t num_frames;
 296     ALsizei num_bytes;
 297
 298     /* Open the audio file and check that it's usable. */
 299     sndfile = sf_open(filename, SFM_READ, &sfinfo);
 300     if(!sndfile)
 301     {
 302         fprintf(stderr, "Could not open audio in %s: %s\n", filename, sf_strerror(sndfile));
 303         return 0;
 304     }
 305     if(sfinfo.frames < 1 || sfinfo.frames > (sf_count_t)(INT_MAX/sizeof(float))/sfinfo.channels)
 306     {
 307         fprintf(stderr, "Bad sample count in %s (%" PRId64 ")\n", filename, sfinfo.frames);
 308         sf_close(sndfile);
 309         return 0;
 310     }
 311
 312     /* Get the sound format, and figure out the OpenAL format. Use floats since
 313      * impulse responses will usually have more than 16-bit precision.
 314      */
 315     if(sfinfo.channels == 1)
 316         format = AL_FORMAT_MONO_FLOAT32;
 317     else if(sfinfo.channels == 2)
 318         format = AL_FORMAT_STEREO_FLOAT32;
 319     else
 320     {
 321         fprintf(stderr, "Unsupported channel count: %d\n", sfinfo.channels);
 322         sf_close(sndfile);
 323         return 0;
 324     }
 325
 326     namepart = strrchr(filename, '/');
 327     if(namepart || (namepart=strrchr(filename, '\\')))
 328         namepart++;
 329     else
 330         namepart = filename;
 331     printf("Loading: %s (%s, %dhz, %" PRId64 " samples / %.2f seconds)\n", namepart,
 332         FormatName(format), sfinfo.samplerate, sfinfo.frames,
 333         (double)sfinfo.frames / sfinfo.samplerate);
 334     fflush(stdout);
 335
 336     /* Decode the whole audio file to a buffer. */
 337     membuf = malloc((size_t)(sfinfo.frames * sfinfo.channels) * sizeof(float));
 338
 339     num_frames = sf_readf_float(sndfile, membuf, sfinfo.frames);
 340     if(num_frames < 1)
 341     {
 342         free(membuf);
 343         sf_close(sndfile);
 344         fprintf(stderr, "Failed to read samples in %s (%" PRId64 ")\n", filename, num_frames);
 345         return 0;
 346     }
 347     num_bytes = (ALsizei)(num_frames * sfinfo.channels) * (ALsizei)sizeof(float);
 348
 349     /* Buffer the audio data into a new buffer object, then free the data and
 350      * close the file.
 351      */
 352     buffer = 0;
 353     alGenBuffers(1, &buffer);
 354     alBufferData(buffer, format, membuf, num_bytes, sfinfo.samplerate);
 355
 356     free(membuf);
 357     sf_close(sndfile);
 358
 359     /* Check if an error occured, and clean up if so. */
 360     err = alGetError();
 361     if(err != AL_NO_ERROR)
 362     {
 363         fprintf(stderr, "OpenAL Error: %s\n", alGetString(err));
 364         if(buffer && alIsBuffer(buffer))
 365             alDeleteBuffers(1, &buffer);
 366         return 0;
 367     }
 368
 369     return buffer;
 370 }
 371
 372
 373 int main(int argc, char **argv)
 374 {
 375     ALuint ir_buffer, effect, slot;
 376     StreamPlayer *player;
 377     int i;
 378
 379     /* Print out usage if no arguments were specified */
 380     if(argc < 2)
 381     {
 382         fprintf(stderr, "Usage: %s [-device <name>] <impulse response file> <filenames...>\n",
 383             argv[0]);
 384         return 1;
 385     }
 386
 387     argv++; argc--;
 388     if(InitAL(&argv, &argc) != 0)
 389         return 1;
 390
 391     if(!alIsExtensionPresent("AL_SOFTX_convolution_reverb"))
 392     {
 393         CloseAL();
 394         fprintf(stderr, "Error: Convolution revern not supported\n");
 395         return 1;
 396     }
 397
 398     if(argc < 2)
 399     {
 400         CloseAL();
 401         fprintf(stderr, "Error: Missing impulse response or sound files\n");
 402         return 1;
 403     }
 404
 405     /* Define a macro to help load the function pointers. */
 406 #define LOAD_PROC(T, x)  ((x) = (T)alGetProcAddress(#x))
 407     LOAD_PROC(LPALGENEFFECTS, alGenEffects);
 408     LOAD_PROC(LPALDELETEEFFECTS, alDeleteEffects);
 409     LOAD_PROC(LPALISEFFECT, alIsEffect);
 410     LOAD_PROC(LPALEFFECTI, alEffecti);
 411     LOAD_PROC(LPALEFFECTIV, alEffectiv);
 412     LOAD_PROC(LPALEFFECTF, alEffectf);
 413     LOAD_PROC(LPALEFFECTFV, alEffectfv);
 414     LOAD_PROC(LPALGETEFFECTI, alGetEffecti);
 415     LOAD_PROC(LPALGETEFFECTIV, alGetEffectiv);
 416     LOAD_PROC(LPALGETEFFECTF, alGetEffectf);
 417     LOAD_PROC(LPALGETEFFECTFV, alGetEffectfv);
 418
 419     LOAD_PROC(LPALGENAUXILIARYEFFECTSLOTS, alGenAuxiliaryEffectSlots);
 420     LOAD_PROC(LPALDELETEAUXILIARYEFFECTSLOTS, alDeleteAuxiliaryEffectSlots);
 421     LOAD_PROC(LPALISAUXILIARYEFFECTSLOT, alIsAuxiliaryEffectSlot);
 422     LOAD_PROC(LPALAUXILIARYEFFECTSLOTI, alAuxiliaryEffectSloti);
 423     LOAD_PROC(LPALAUXILIARYEFFECTSLOTIV, alAuxiliaryEffectSlotiv);
 424     LOAD_PROC(LPALAUXILIARYEFFECTSLOTF, alAuxiliaryEffectSlotf);
 425     LOAD_PROC(LPALAUXILIARYEFFECTSLOTFV, alAuxiliaryEffectSlotfv);
 426     LOAD_PROC(LPALGETAUXILIARYEFFECTSLOTI, alGetAuxiliaryEffectSloti);
 427     LOAD_PROC(LPALGETAUXILIARYEFFECTSLOTIV, alGetAuxiliaryEffectSlotiv);
 428     LOAD_PROC(LPALGETAUXILIARYEFFECTSLOTF, alGetAuxiliaryEffectSlotf);
 429     LOAD_PROC(LPALGETAUXILIARYEFFECTSLOTFV, alGetAuxiliaryEffectSlotfv);
 430 #undef LOAD_PROC
 431
 432     /* Load the reverb into an effect. */
 433     effect = CreateEffect();
 434     if(!effect)
 435     {
 436         CloseAL();
 437         return 1;
 438     }
 439
 440     /* Load the impulse response sound into a buffer. */
 441     ir_buffer = LoadSound(argv[0]);
 442     if(!ir_buffer)
 443     {
 444         alDeleteEffects(1, &effect);
 445         CloseAL();
 446         return 1;
 447     }
 448
 449     /* Create the effect slot object. This is what "plays" an effect on sources
 450      * that connect to it.
 451      */
 452     slot = 0;
 453     alGenAuxiliaryEffectSlots(1, &slot);
 454
 455     /* Set the impulse response sound buffer on the effect slot. This allows
 456      * effects to access it as needed. In this case, convolution reverb uses it
 457      * as the filter source. NOTE: Unlike the effect object, the buffer *is*
 458      * kept referenced and may not be changed or deleted as long as it's set,
 459      * just like with a source. When another buffer is set, or the effect slot
 460      * is deleted, the buffer reference is released.
 461      *
 462      * The effect slot's gain is reduced because the impulse responses I've
 463      * tested with result in excessively loud reverb. Is that normal? Even with
 464      * this, it seems a bit on the loud side.
 465      *
 466      * Also note: unlike standard or EAX reverb, there is no automatic
 467      * attenuation of a source's reverb response with distance, so the reverb
 468      * will remain full volume regardless of a given sound's distance from the
 469      * listener. You can use a send filter to alter a given source's
 470      * contribution to reverb.
 471      */
 472     alAuxiliaryEffectSloti(slot, AL_BUFFER, (ALint)ir_buffer);
 473     alAuxiliaryEffectSlotf(slot, AL_EFFECTSLOT_GAIN, 1.0f / 16.0f);
 474     alAuxiliaryEffectSloti(slot, AL_EFFECTSLOT_EFFECT, (ALint)effect);
 475     assert(alGetError()==AL_NO_ERROR && "Failed to set effect slot");
 476
 477     player = NewPlayer();
 478     /* Connect the player's source to the effect slot. */
 479     alSource3i(player->source, AL_AUXILIARY_SEND_FILTER, (ALint)slot, 0, AL_FILTER_NULL);
 480     assert(alGetError()==AL_NO_ERROR && "Failed to setup sound source");
 481
 482     /* Play each file listed on the command line */
 483     for(i = 1;i < argc;i++)
 484     {
 485         const char *namepart;
 486
 487         if(!OpenPlayerFile(player, argv[i]))
 488             continue;
 489
 490         namepart = strrchr(argv[i], '/');
 491         if(namepart || (namepart=strrchr(argv[i], '\\')))
 492             namepart++;
 493         else
 494             namepart = argv[i];
 495
 496         printf("Playing: %s (%s, %dhz)\n", namepart, FormatName(player->format),
 497             player->sfinfo.samplerate);
 498         fflush(stdout);
 499
 500         if(!StartPlayer(player))
 501         {
 502             ClosePlayerFile(player);
 503             continue;
 504         }
 505
 506         while(UpdatePlayer(player))
 507             al_nssleep(10000000);
 508
 509         ClosePlayerFile(player);
 510     }
 511     printf("Done.\n");
 512
 513     /* All files done. Delete the player and effect resources, and close down
 514      * OpenAL.
 515      */
 516     DeletePlayer(player);
 517     player = NULL;
 518
 519     alDeleteAuxiliaryEffectSlots(1, &slot);
 520     alDeleteEffects(1, &effect);
 521     alDeleteBuffers(1, &ir_buffer);
 522
 523     CloseAL();
 524
 525     return 0;
 526 }