examples/alplay.c

   1 /*
   2  * OpenAL Source Play Example
   3  *
   4  * Copyright (c) 2017 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 playing a sound buffer. */
  26
  27 #include <assert.h>
  28 #include <inttypes.h>
  29 #include <limits.h>
  30 #include <stdio.h>
  31 #include <stdlib.h>
  32
  33 #include "sndfile.h"
  34
  35 #include "AL/al.h"
  36 #include "AL/alext.h"
  37
  38 #include "common/alhelpers.h"
  39
  40 #include "win_main_utf8.h"
  41
  42
  43 enum FormatType {
  44     Int16,
  45     Float,
  46     IMA4,
  47     MSADPCM
  48 };
  49
  50 /* LoadBuffer loads the named audio file into an OpenAL buffer object, and
  51  * returns the new buffer ID.
  52  */
  53 static ALuint LoadSound(const char *filename)
  54 {
  55     enum FormatType sample_format = Int16;
  56     ALint byteblockalign = 0;
  57     ALint splblockalign = 0;
  58     sf_count_t num_frames;
  59     ALenum err, format;
  60     ALsizei num_bytes;
  61     SNDFILE *sndfile;
  62     SF_INFO sfinfo;
  63     ALuint buffer;
  64     void *membuf;
  65
  66     /* Open the audio file and check that it's usable. */
  67     sndfile = sf_open(filename, SFM_READ, &sfinfo);
  68     if(!sndfile)
  69     {
  70         fprintf(stderr, "Could not open audio in %s: %s\n", filename, sf_strerror(sndfile));
  71         return 0;
  72     }
  73     if(sfinfo.frames < 1)
  74     {
  75         fprintf(stderr, "Bad sample count in %s (%" PRId64 ")\n", filename, sfinfo.frames);
  76         sf_close(sndfile);
  77         return 0;
  78     }
  79
  80     /* Detect a suitable format to load. Formats like Vorbis and Opus use float
  81      * natively, so load as float to avoid clipping when possible. Formats
  82      * larger than 16-bit can also use float to preserve a bit more precision.
  83      */
  84     switch((sfinfo.format&SF_FORMAT_SUBMASK))
  85     {
  86     case SF_FORMAT_PCM_24:
  87     case SF_FORMAT_PCM_32:
  88     case SF_FORMAT_FLOAT:
  89     case SF_FORMAT_DOUBLE:
  90     case SF_FORMAT_VORBIS:
  91     case SF_FORMAT_OPUS:
  92     case SF_FORMAT_ALAC_20:
  93     case SF_FORMAT_ALAC_24:
  94     case SF_FORMAT_ALAC_32:
  95     case 0x0080/*SF_FORMAT_MPEG_LAYER_I*/:
  96     case 0x0081/*SF_FORMAT_MPEG_LAYER_II*/:
  97     case 0x0082/*SF_FORMAT_MPEG_LAYER_III*/:
  98         if(alIsExtensionPresent("AL_EXT_FLOAT32"))
  99             sample_format = Float;
 100         break;
 101     case SF_FORMAT_IMA_ADPCM:
 102         /* ADPCM formats require setting a block alignment as specified in the
 103          * file, which needs to be read from the wave 'fmt ' chunk manually
 104          * since libsndfile doesn't provide it in a format-agnostic way.
 105          */
 106         if(sfinfo.channels <= 2 && (sfinfo.format&SF_FORMAT_TYPEMASK) == SF_FORMAT_WAV
 107             && alIsExtensionPresent("AL_EXT_IMA4")
 108             && alIsExtensionPresent("AL_SOFT_block_alignment"))
 109             sample_format = IMA4;
 110         break;
 111     case SF_FORMAT_MS_ADPCM:
 112         if(sfinfo.channels <= 2 && (sfinfo.format&SF_FORMAT_TYPEMASK) == SF_FORMAT_WAV
 113             && alIsExtensionPresent("AL_SOFT_MSADPCM")
 114             && alIsExtensionPresent("AL_SOFT_block_alignment"))
 115             sample_format = MSADPCM;
 116         break;
 117     }
 118
 119     if(sample_format == IMA4 || sample_format == MSADPCM)
 120     {
 121         /* For ADPCM, lookup the wave file's "fmt " chunk, which is a
 122          * WAVEFORMATEX-based structure for the audio format.
 123          */
 124         SF_CHUNK_INFO inf = { "fmt ", 4, 0, NULL };
 125         SF_CHUNK_ITERATOR *iter = sf_get_chunk_iterator(sndfile, &inf);
 126
 127         /* If there's an issue getting the chunk or block alignment, load as
 128          * 16-bit and have libsndfile do the conversion.
 129          */
 130         if(!iter || sf_get_chunk_size(iter, &inf) != SF_ERR_NO_ERROR || inf.datalen < 14)
 131             sample_format = Int16;
 132         else
 133         {
 134             ALubyte *fmtbuf = calloc(inf.datalen, 1);
 135             inf.data = fmtbuf;
 136             if(sf_get_chunk_data(iter, &inf) != SF_ERR_NO_ERROR)
 137                 sample_format = Int16;
 138             else
 139             {
 140                 /* Read the nBlockAlign field, and convert from bytes- to
 141                  * samples-per-block (verifying it's valid by converting back
 142                  * and comparing to the original value).
 143                  */
 144                 byteblockalign = fmtbuf[12] | (fmtbuf[13]<<8);
 145                 if(sample_format == IMA4)
 146                 {
 147                     splblockalign = (byteblockalign/sfinfo.channels - 4)/4*8 + 1;
 148                     if(splblockalign < 1
 149                         || ((splblockalign-1)/2 + 4)*sfinfo.channels != byteblockalign)
 150                         sample_format = Int16;
 151                 }
 152                 else
 153                 {
 154                     splblockalign = (byteblockalign/sfinfo.channels - 7)*2 + 2;
 155                     if(splblockalign < 2
 156                         || ((splblockalign-2)/2 + 7)*sfinfo.channels != byteblockalign)
 157                         sample_format = Int16;
 158                 }
 159             }
 160             free(fmtbuf);
 161         }
 162     }
 163
 164     if(sample_format == Int16)
 165     {
 166         splblockalign = 1;
 167         byteblockalign = sfinfo.channels * 2;
 168     }
 169     else if(sample_format == Float)
 170     {
 171         splblockalign = 1;
 172         byteblockalign = sfinfo.channels * 4;
 173     }
 174
 175     /* Figure out the OpenAL format from the file and desired sample type. */
 176     format = AL_NONE;
 177     if(sfinfo.channels == 1)
 178     {
 179         if(sample_format == Int16)
 180             format = AL_FORMAT_MONO16;
 181         else if(sample_format == Float)
 182             format = AL_FORMAT_MONO_FLOAT32;
 183         else if(sample_format == IMA4)
 184             format = AL_FORMAT_MONO_IMA4;
 185         else if(sample_format == MSADPCM)
 186             format = AL_FORMAT_MONO_MSADPCM_SOFT;
 187     }
 188     else if(sfinfo.channels == 2)
 189     {
 190         if(sample_format == Int16)
 191             format = AL_FORMAT_STEREO16;
 192         else if(sample_format == Float)
 193             format = AL_FORMAT_STEREO_FLOAT32;
 194         else if(sample_format == IMA4)
 195             format = AL_FORMAT_STEREO_IMA4;
 196         else if(sample_format == MSADPCM)
 197             format = AL_FORMAT_STEREO_MSADPCM_SOFT;
 198     }
 199     else if(sfinfo.channels == 3)
 200     {
 201         if(sf_command(sndfile, SFC_WAVEX_GET_AMBISONIC, NULL, 0) == SF_AMBISONIC_B_FORMAT)
 202         {
 203             if(sample_format == Int16)
 204                 format = AL_FORMAT_BFORMAT2D_16;
 205             else if(sample_format == Float)
 206                 format = AL_FORMAT_BFORMAT2D_FLOAT32;
 207         }
 208     }
 209     else if(sfinfo.channels == 4)
 210     {
 211         if(sf_command(sndfile, SFC_WAVEX_GET_AMBISONIC, NULL, 0) == SF_AMBISONIC_B_FORMAT)
 212         {
 213             if(sample_format == Int16)
 214                 format = AL_FORMAT_BFORMAT3D_16;
 215             else if(sample_format == Float)
 216                 format = AL_FORMAT_BFORMAT3D_FLOAT32;
 217         }
 218     }
 219     if(!format)
 220     {
 221         fprintf(stderr, "Unsupported channel count: %d\n", sfinfo.channels);
 222         sf_close(sndfile);
 223         return 0;
 224     }
 225
 226     if(sfinfo.frames/splblockalign > (sf_count_t)(INT_MAX/byteblockalign))
 227     {
 228         fprintf(stderr, "Too many samples in %s (%" PRId64 ")\n", filename, sfinfo.frames);
 229         sf_close(sndfile);
 230         return 0;
 231     }
 232
 233     /* Decode the whole audio file to a buffer. */
 234     membuf = malloc((size_t)(sfinfo.frames / splblockalign * byteblockalign));
 235
 236     if(sample_format == Int16)
 237         num_frames = sf_readf_short(sndfile, membuf, sfinfo.frames);
 238     else if(sample_format == Float)
 239         num_frames = sf_readf_float(sndfile, membuf, sfinfo.frames);
 240     else
 241     {
 242         sf_count_t count = sfinfo.frames / splblockalign * byteblockalign;
 243         num_frames = sf_read_raw(sndfile, membuf, count);
 244         if(num_frames > 0)
 245             num_frames = num_frames / byteblockalign * splblockalign;
 246     }
 247     if(num_frames < 1)
 248     {
 249         free(membuf);
 250         sf_close(sndfile);
 251         fprintf(stderr, "Failed to read samples in %s (%" PRId64 ")\n", filename, num_frames);
 252         return 0;
 253     }
 254     num_bytes = (ALsizei)(num_frames / splblockalign * byteblockalign);
 255
 256     printf("Loading: %s (%s, %dhz)\n", filename, FormatName(format), sfinfo.samplerate);
 257     fflush(stdout);
 258
 259     /* Buffer the audio data into a new buffer object, then free the data and
 260      * close the file.
 261      */
 262     buffer = 0;
 263     alGenBuffers(1, &buffer);
 264     if(splblockalign > 1)
 265         alBufferi(buffer, AL_UNPACK_BLOCK_ALIGNMENT_SOFT, splblockalign);
 266     alBufferData(buffer, format, membuf, num_bytes, sfinfo.samplerate);
 267
 268     free(membuf);
 269     sf_close(sndfile);
 270
 271     /* Check if an error occurred, 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(buffer && alIsBuffer(buffer))
 277             alDeleteBuffers(1, &buffer);
 278         return 0;
 279     }
 280
 281     return buffer;
 282 }
 283
 284
 285 int main(int argc, char **argv)
 286 {
 287     ALuint source, buffer;
 288     ALfloat offset;
 289     ALenum state;
 290
 291     /* Print out usage if no arguments were specified */
 292     if(argc < 2)
 293     {
 294         fprintf(stderr, "Usage: %s [-device <name>] <filename>\n", argv[0]);
 295         return 1;
 296     }
 297
 298     /* Initialize OpenAL. */
 299     argv++; argc--;
 300     if(InitAL(&argv, &argc) != 0)
 301         return 1;
 302
 303     /* Load the sound into a buffer. */
 304     buffer = LoadSound(argv[0]);
 305     if(!buffer)
 306     {
 307         CloseAL();
 308         return 1;
 309     }
 310
 311     /* Create the source to play the sound with. */
 312     source = 0;
 313     alGenSources(1, &source);
 314     alSourcei(source, AL_BUFFER, (ALint)buffer);
 315     assert(alGetError()==AL_NO_ERROR && "Failed to setup sound source");
 316
 317     /* Play the sound until it finishes. */
 318     alSourcePlay(source);
 319     do {
 320         al_nssleep(10000000);
 321         alGetSourcei(source, AL_SOURCE_STATE, &state);
 322
 323         /* Get the source offset. */
 324         alGetSourcef(source, AL_SEC_OFFSET, &offset);
 325         printf("\rOffset: %f  ", offset);
 326         fflush(stdout);
 327     } while(alGetError() == AL_NO_ERROR && state == AL_PLAYING);
 328     printf("\n");
 329
 330     /* All done. Delete resources, and close down OpenAL. */
 331     alDeleteSources(1, &source);
 332     alDeleteBuffers(1, &buffer);
 333
 334     CloseAL();
 335
 336     return 0;
 337 }