2 * OpenAL Source Play Example
4 * Copyright (c) 2017 by Chris Robinson <chris.kcat@gmail.com>
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
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13 * The above copyright notice and this permission notice shall be included in
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25 /* This file contains an example for playing a sound buffer. */
38 #include "common/alhelpers.h"
48 /* LoadBuffer loads the named audio file into an OpenAL buffer object, and
49 * returns the new buffer ID.
51 static ALuint LoadSound(const char *filename)
53 enum FormatType sample_format = Int16;
54 ALint byteblockalign = 0;
55 ALint splblockalign = 0;
56 sf_count_t num_frames;
64 /* Open the audio file and check that it's usable. */
65 sndfile = sf_open(filename, SFM_READ, &sfinfo);
68 fprintf(stderr, "Could not open audio in %s: %s\n", filename, sf_strerror(sndfile));
73 fprintf(stderr, "Bad sample count in %s (%" PRId64 ")\n", filename, sfinfo.frames);
78 /* Detect a suitable format to load. Formats like Vorbis and Opus use float
79 * natively, so load as float to avoid clipping when possible. Formats
80 * larger than 16-bit can also use float to preserve a bit more precision.
82 switch((sfinfo.format&SF_FORMAT_SUBMASK))
84 case SF_FORMAT_PCM_24:
85 case SF_FORMAT_PCM_32:
87 case SF_FORMAT_DOUBLE:
88 case SF_FORMAT_VORBIS:
90 case SF_FORMAT_ALAC_20:
91 case SF_FORMAT_ALAC_24:
92 case SF_FORMAT_ALAC_32:
93 case 0x0080/*SF_FORMAT_MPEG_LAYER_I*/:
94 case 0x0081/*SF_FORMAT_MPEG_LAYER_II*/:
95 case 0x0082/*SF_FORMAT_MPEG_LAYER_III*/:
96 if(alIsExtensionPresent("AL_EXT_FLOAT32"))
97 sample_format = Float;
99 case SF_FORMAT_IMA_ADPCM:
100 /* ADPCM formats require setting a block alignment as specified in the
101 * file, which needs to be read from the wave 'fmt ' chunk manually
102 * since libsndfile doesn't provide it in a format-agnostic way.
104 if(sfinfo.channels <= 2 && (sfinfo.format&SF_FORMAT_TYPEMASK) == SF_FORMAT_WAV
105 && alIsExtensionPresent("AL_EXT_IMA4")
106 && alIsExtensionPresent("AL_SOFT_block_alignment"))
107 sample_format = IMA4;
109 case SF_FORMAT_MS_ADPCM:
110 if(sfinfo.channels <= 2 && (sfinfo.format&SF_FORMAT_TYPEMASK) == SF_FORMAT_WAV
111 && alIsExtensionPresent("AL_SOFT_MSADPCM")
112 && alIsExtensionPresent("AL_SOFT_block_alignment"))
113 sample_format = MSADPCM;
117 if(sample_format == IMA4 || sample_format == MSADPCM)
119 /* For ADPCM, lookup the wave file's "fmt " chunk, which is a
120 * WAVEFORMATEX-based structure for the audio format.
122 SF_CHUNK_INFO inf = { "fmt ", 4, 0, NULL };
123 SF_CHUNK_ITERATOR *iter = sf_get_chunk_iterator(sndfile, &inf);
125 /* If there's an issue getting the chunk or block alignment, load as
126 * 16-bit and have libsndfile do the conversion.
128 if(!iter || sf_get_chunk_size(iter, &inf) != SF_ERR_NO_ERROR || inf.datalen < 14)
129 sample_format = Int16;
132 ALubyte *fmtbuf = calloc(inf.datalen, 1);
134 if(sf_get_chunk_data(iter, &inf) != SF_ERR_NO_ERROR)
135 sample_format = Int16;
138 /* Read the nBlockAlign field, and convert from bytes- to
139 * samples-per-block (verifying it's valid by converting back
140 * and comparing to the original value).
142 byteblockalign = fmtbuf[12] | (fmtbuf[13]<<8);
143 if(sample_format == IMA4)
145 splblockalign = (byteblockalign/sfinfo.channels - 4)/4*8 + 1;
147 || ((splblockalign-1)/2 + 4)*sfinfo.channels != byteblockalign)
148 sample_format = Int16;
152 splblockalign = (byteblockalign/sfinfo.channels - 7)*2 + 2;
154 || ((splblockalign-2)/2 + 7)*sfinfo.channels != byteblockalign)
155 sample_format = Int16;
162 if(sample_format == Int16)
165 byteblockalign = sfinfo.channels * 2;
167 else if(sample_format == Float)
170 byteblockalign = sfinfo.channels * 4;
173 /* Figure out the OpenAL format from the file and desired sample type. */
175 if(sfinfo.channels == 1)
177 if(sample_format == Int16)
178 format = AL_FORMAT_MONO16;
179 else if(sample_format == Float)
180 format = AL_FORMAT_MONO_FLOAT32;
181 else if(sample_format == IMA4)
182 format = AL_FORMAT_MONO_IMA4;
183 else if(sample_format == MSADPCM)
184 format = AL_FORMAT_MONO_MSADPCM_SOFT;
186 else if(sfinfo.channels == 2)
188 if(sample_format == Int16)
189 format = AL_FORMAT_STEREO16;
190 else if(sample_format == Float)
191 format = AL_FORMAT_STEREO_FLOAT32;
192 else if(sample_format == IMA4)
193 format = AL_FORMAT_STEREO_IMA4;
194 else if(sample_format == MSADPCM)
195 format = AL_FORMAT_STEREO_MSADPCM_SOFT;
197 else if(sfinfo.channels == 3)
199 if(sf_command(sndfile, SFC_WAVEX_GET_AMBISONIC, NULL, 0) == SF_AMBISONIC_B_FORMAT)
201 if(sample_format == Int16)
202 format = AL_FORMAT_BFORMAT2D_16;
203 else if(sample_format == Float)
204 format = AL_FORMAT_BFORMAT2D_FLOAT32;
207 else if(sfinfo.channels == 4)
209 if(sf_command(sndfile, SFC_WAVEX_GET_AMBISONIC, NULL, 0) == SF_AMBISONIC_B_FORMAT)
211 if(sample_format == Int16)
212 format = AL_FORMAT_BFORMAT3D_16;
213 else if(sample_format == Float)
214 format = AL_FORMAT_BFORMAT3D_FLOAT32;
219 fprintf(stderr, "Unsupported channel count: %d\n", sfinfo.channels);
224 if(sfinfo.frames/splblockalign > (sf_count_t)(INT_MAX/byteblockalign))
226 fprintf(stderr, "Too many samples in %s (%" PRId64 ")\n", filename, sfinfo.frames);
231 /* Decode the whole audio file to a buffer. */
232 membuf = malloc((size_t)(sfinfo.frames / splblockalign * byteblockalign));
234 if(sample_format == Int16)
235 num_frames = sf_readf_short(sndfile, membuf, sfinfo.frames);
236 else if(sample_format == Float)
237 num_frames = sf_readf_float(sndfile, membuf, sfinfo.frames);
240 sf_count_t count = sfinfo.frames / splblockalign * byteblockalign;
241 num_frames = sf_read_raw(sndfile, membuf, count);
243 num_frames = num_frames / byteblockalign * splblockalign;
249 fprintf(stderr, "Failed to read samples in %s (%" PRId64 ")\n", filename, num_frames);
252 num_bytes = (ALsizei)(num_frames / splblockalign * byteblockalign);
254 printf("Loading: %s (%s, %dhz)\n", filename, FormatName(format), sfinfo.samplerate);
257 /* Buffer the audio data into a new buffer object, then free the data and
261 alGenBuffers(1, &buffer);
262 if(splblockalign > 1)
263 alBufferi(buffer, AL_UNPACK_BLOCK_ALIGNMENT_SOFT, splblockalign);
264 alBufferData(buffer, format, membuf, num_bytes, sfinfo.samplerate);
269 /* Check if an error occured, and clean up if so. */
271 if(err != AL_NO_ERROR)
273 fprintf(stderr, "OpenAL Error: %s\n", alGetString(err));
274 if(buffer && alIsBuffer(buffer))
275 alDeleteBuffers(1, &buffer);
283 int main(int argc, char **argv)
285 ALuint source, buffer;
289 /* Print out usage if no arguments were specified */
292 fprintf(stderr, "Usage: %s [-device <name>] <filename>\n", argv[0]);
296 /* Initialize OpenAL. */
298 if(InitAL(&argv, &argc) != 0)
301 /* Load the sound into a buffer. */
302 buffer = LoadSound(argv[0]);
309 /* Create the source to play the sound with. */
311 alGenSources(1, &source);
312 alSourcei(source, AL_BUFFER, (ALint)buffer);
313 assert(alGetError()==AL_NO_ERROR && "Failed to setup sound source");
315 /* Play the sound until it finishes. */
316 alSourcePlay(source);
318 al_nssleep(10000000);
319 alGetSourcei(source, AL_SOURCE_STATE, &state);
321 /* Get the source offset. */
322 alGetSourcef(source, AL_SEC_OFFSET, &offset);
323 printf("\rOffset: %f ", offset);
325 } while(alGetError() == AL_NO_ERROR && state == AL_PLAYING);
328 /* All done. Delete resources, and close down OpenAL. */
329 alDeleteSources(1, &source);
330 alDeleteBuffers(1, &buffer);