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[audio-libsamplerate.git] / libsamplerate / examples / audio_out.c
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8529da43
MG		 1/*
2** Copyright (c) 1999-2016, Erik de Castro Lopo <erikd@mega-nerd.com>
3** All rights reserved.
4**
5** This code is released under 2-clause BSD license. Please see the
6** file at : https://github.com/erikd/libsamplerate/blob/master/COPYING
7*/
8
9#include <stdio.h>
10#include <stdlib.h>
11#include <string.h>
12#include <unistd.h>
13
14#include <config.h>
15
16#include "audio_out.h"
17
18#if HAVE_ALSA_ASOUNDLIB_H
19 #define ALSA_PCM_NEW_HW_PARAMS_API
20 #define ALSA_PCM_NEW_SW_PARAMS_API
21 #include <alsa/asoundlib.h>
22 #include <sys/time.h>
23#endif
24
25#if (HAVE_SNDFILE)
26
27#include <float_cast.h>
28
29#include <sndfile.h>
30
31#define BUFFER_LEN (2048)
32
33#define MAKE_MAGIC(a,b,c,d,e,f,g,h) \
34 ((a) + ((b) << 1) + ((c) << 2) + ((d) << 3) + ((e) << 4) + ((f) << 5) + ((g) << 6) + ((h) << 7))
35
36typedef struct AUDIO_OUT_s
37{ int magic ;
38} AUDIO_OUT ;
39
40
41/*------------------------------------------------------------------------------
42** Linux (ALSA and OSS) functions for playing a sound.
43*/
44
45#if defined (__linux__)
46
47#if HAVE_ALSA_ASOUNDLIB_H
48
49#define ALSA_MAGIC MAKE_MAGIC ('L', 'n', 'x', '-', 'A', 'L', 'S', 'A')
50
51typedef struct
52{ int magic ;
53 snd_pcm_t * dev ;
54 int channels ;
55} ALSA_AUDIO_OUT ;
56
57static int alsa_write_float (snd_pcm_t *alsa_dev, float *data, int frames, int channels) ;
58
59static AUDIO_OUT *
60alsa_open (int channels, unsigned samplerate)
61{ ALSA_AUDIO_OUT *alsa_out ;
62 const char * device = "default" ;
63 snd_pcm_hw_params_t *hw_params ;
64 snd_pcm_uframes_t buffer_size ;
65 snd_pcm_uframes_t alsa_period_size, alsa_buffer_frames ;
66 snd_pcm_sw_params_t *sw_params ;
67
68 int err ;
69
70 alsa_period_size = 1024 ;
71 alsa_buffer_frames = 4 * alsa_period_size ;
72
73 if ((alsa_out = calloc (1, sizeof (ALSA_AUDIO_OUT))) == NULL)
74 { perror ("alsa_open : malloc ") ;
75 exit (1) ;
76 } ;
77
78 alsa_out->magic = ALSA_MAGIC ;
79 alsa_out->channels = channels ;
80
81 if ((err = snd_pcm_open (&alsa_out->dev, device, SND_PCM_STREAM_PLAYBACK, 0)) < 0)
82 { fprintf (stderr, "cannot open audio device \"%s\" (%s)\n", device, snd_strerror (err)) ;
83 goto catch_error ;
84 } ;
85
86 snd_pcm_nonblock (alsa_out->dev, 0) ;
87
88 if ((err = snd_pcm_hw_params_malloc (&hw_params)) < 0)
89 { fprintf (stderr, "cannot allocate hardware parameter structure (%s)\n", snd_strerror (err)) ;
90 goto catch_error ;
91 } ;
92
93 if ((err = snd_pcm_hw_params_any (alsa_out->dev, hw_params)) < 0)
94 { fprintf (stderr, "cannot initialize hardware parameter structure (%s)\n", snd_strerror (err)) ;
95 goto catch_error ;
96 } ;
97
98 if ((err = snd_pcm_hw_params_set_access (alsa_out->dev, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0)
99 { fprintf (stderr, "cannot set access type (%s)\n", snd_strerror (err)) ;
100 goto catch_error ;
101 } ;
102
103 if ((err = snd_pcm_hw_params_set_format (alsa_out->dev, hw_params, SND_PCM_FORMAT_FLOAT)) < 0)
104 { fprintf (stderr, "cannot set sample format (%s)\n", snd_strerror (err)) ;
105 goto catch_error ;
106 } ;
107
108 if ((err = snd_pcm_hw_params_set_rate_near (alsa_out->dev, hw_params, &samplerate, 0)) < 0)
109 { fprintf (stderr, "cannot set sample rate (%s)\n", snd_strerror (err)) ;
110 goto catch_error ;
111 } ;
112
113 if ((err = snd_pcm_hw_params_set_channels (alsa_out->dev, hw_params, channels)) < 0)
114 { fprintf (stderr, "cannot set channel count (%s)\n", snd_strerror (err)) ;
115 goto catch_error ;
116 } ;
117
118 if ((err = snd_pcm_hw_params_set_buffer_size_near (alsa_out->dev, hw_params, &alsa_buffer_frames)) < 0)
119 { fprintf (stderr, "cannot set buffer size (%s)\n", snd_strerror (err)) ;
120 goto catch_error ;
121 } ;
122
123 if ((err = snd_pcm_hw_params_set_period_size_near (alsa_out->dev, hw_params, &alsa_period_size, 0)) < 0)
124 { fprintf (stderr, "cannot set period size (%s)\n", snd_strerror (err)) ;
125 goto catch_error ;
126 } ;
127
128 if ((err = snd_pcm_hw_params (alsa_out->dev, hw_params)) < 0)
129 { fprintf (stderr, "cannot set parameters (%s)\n", snd_strerror (err)) ;
130 goto catch_error ;
131 } ;
132
133 /* extra check: if we have only one period, this code won't work */
134 snd_pcm_hw_params_get_period_size (hw_params, &alsa_period_size, 0) ;
135 snd_pcm_hw_params_get_buffer_size (hw_params, &buffer_size) ;
136 if (alsa_period_size == buffer_size)
137 { fprintf (stderr, "Can't use period equal to buffer size (%lu == %lu)", alsa_period_size, buffer_size) ;
138 goto catch_error ;
139 } ;
140
141 snd_pcm_hw_params_free (hw_params) ;
142
143 if ((err = snd_pcm_sw_params_malloc (&sw_params)) != 0)
144 { fprintf (stderr, "%s: snd_pcm_sw_params_malloc: %s", __func__, snd_strerror (err)) ;
145 goto catch_error ;
146 } ;
147
148 if ((err = snd_pcm_sw_params_current (alsa_out->dev, sw_params)) != 0)
149 { fprintf (stderr, "%s: snd_pcm_sw_params_current: %s", __func__, snd_strerror (err)) ;
150 goto catch_error ;
151 } ;
152
153 /* note: set start threshold to delay start until the ring buffer is full */
154 snd_pcm_sw_params_current (alsa_out->dev, sw_params) ;
155
156 if ((err = snd_pcm_sw_params_set_start_threshold (alsa_out->dev, sw_params, buffer_size)) < 0)
157 { fprintf (stderr, "cannot set start threshold (%s)\n", snd_strerror (err)) ;
158 goto catch_error ;
159 } ;
160
161 if ((err = snd_pcm_sw_params (alsa_out->dev, sw_params)) != 0)
162 { fprintf (stderr, "%s: snd_pcm_sw_params: %s", __func__, snd_strerror (err)) ;
163 goto catch_error ;
164 } ;
165
166 snd_pcm_sw_params_free (sw_params) ;
167
168 snd_pcm_reset (alsa_out->dev) ;
169
170catch_error :
171
172 if (err < 0 && alsa_out->dev != NULL)
173 { snd_pcm_close (alsa_out->dev) ;
174 return NULL ;
175 } ;
176
177 return (AUDIO_OUT *) alsa_out ;
178} /* alsa_open */
179
180static void
181alsa_play (get_audio_callback_t callback, AUDIO_OUT *audio_out, void *callback_data)
182{ static float buffer [BUFFER_LEN] ;
183 ALSA_AUDIO_OUT *alsa_out ;
184 int read_frames ;
185
186 if ((alsa_out = (ALSA_AUDIO_OUT*) audio_out) == NULL)
187 { printf ("alsa_close : AUDIO_OUT is NULL.\n") ;
188 return ;
189 } ;
190
191 if (alsa_out->magic != ALSA_MAGIC)
192 { printf ("alsa_close : Bad magic number.\n") ;
193 return ;
194 } ;
195
196 while ((read_frames = callback (callback_data, buffer, BUFFER_LEN / alsa_out->channels)))
197 alsa_write_float (alsa_out->dev, buffer, read_frames, alsa_out->channels) ;
198
199 return ;
200} /* alsa_play */
201
202static int
203alsa_write_float (snd_pcm_t *alsa_dev, float *data, int frames, int channels)
204{ static int epipe_count = 0 ;
205
206 int total = 0 ;
207 int retval ;
208
209 if (epipe_count > 0)
210 epipe_count -- ;
211
212 while (total < frames)
213 { retval = snd_pcm_writei (alsa_dev, data + total * channels, frames - total) ;
214
215 if (retval >= 0)
216 { total += retval ;
217 if (total == frames)
218 return total ;
219
220 continue ;
221 } ;
222
223 switch (retval)
224 { case -EAGAIN :
225 puts ("alsa_write_float: EAGAIN") ;
226 continue ;
227 break ;
228
229 case -EPIPE :
230 if (epipe_count > 0)
231 { printf ("alsa_write_float: EPIPE %d\n", epipe_count) ;
232 if (epipe_count > 140)
233 return retval ;
234 } ;
235 epipe_count += 100 ;
236
237#if 0
238 if (0)
239 { snd_pcm_status_t *status ;
240
241 snd_pcm_status_alloca (&status) ;
242 if ((retval = snd_pcm_status (alsa_dev, status)) < 0)
243 fprintf (stderr, "alsa_out: xrun. can't determine length\n") ;
244 else if (snd_pcm_status_get_state (status) == SND_PCM_STATE_XRUN)
245 { struct timeval now, diff, tstamp ;
246
247 gettimeofday (&now, 0) ;
248 snd_pcm_status_get_trigger_tstamp (status, &tstamp) ;
249 timersub (&now, &tstamp, &diff) ;
250
251 fprintf (stderr, "alsa_write_float xrun: of at least %.3f msecs. resetting stream\n",
252 diff.tv_sec * 1000 + diff.tv_usec / 1000.0) ;
253 }
254 else
255 fprintf (stderr, "alsa_write_float: xrun. can't determine length\n") ;
256 } ;
257#endif
258
259 snd_pcm_prepare (alsa_dev) ;
260 break ;
261
262 case -EBADFD :
263 fprintf (stderr, "alsa_write_float: Bad PCM state.n") ;
264 return 0 ;
265 break ;
266
267 case -ESTRPIPE :
268 fprintf (stderr, "alsa_write_float: Suspend event.n") ;
269 return 0 ;
270 break ;
271
272 case -EIO :
273 puts ("alsa_write_float: EIO") ;
274 return 0 ;
275
276 default :
277 fprintf (stderr, "alsa_write_float: retval = %d\n", retval) ;
278 return 0 ;
279 break ;
280 } ; /* switch */
281 } ; /* while */
282
283 return total ;
284} /* alsa_write_float */
285
286static void
287alsa_close (AUDIO_OUT *audio_out)
288{ ALSA_AUDIO_OUT *alsa_out ;
289
290 if ((alsa_out = (ALSA_AUDIO_OUT*) audio_out) == NULL)
291 { printf ("alsa_close : AUDIO_OUT is NULL.\n") ;
292 return ;
293 } ;
294
295 if (alsa_out->magic != ALSA_MAGIC)
296 { printf ("alsa_close : Bad magic number.\n") ;
297 return ;
298 } ;
299
300 memset (alsa_out, 0, sizeof (ALSA_AUDIO_OUT)) ;
301
302 free (alsa_out) ;
303
304 return ;
305} /* alsa_close */
306
307#endif /* HAVE_ALSA_ASOUNDLIB_H */
308
309#include <fcntl.h>
310#include <sys/ioctl.h>
311#include <sys/soundcard.h>
312
313#define OSS_MAGIC MAKE_MAGIC ('L', 'i', 'n', 'u', 'x', 'O', 'S', 'S')
314
315typedef struct
316{ int magic ;
317 int fd ;
318 int channels ;
319} OSS_AUDIO_OUT ;
320
321static AUDIO_OUT *opensoundsys_open (int channels, int samplerate) ;
322static void opensoundsys_play (get_audio_callback_t callback, AUDIO_OUT *audio_out, void *callback_data) ;
323static void opensoundsys_close (AUDIO_OUT *audio_out) ;
324
325
326static AUDIO_OUT *
327opensoundsys_open (int channels, int samplerate)
328{ OSS_AUDIO_OUT *opensoundsys_out ;
329 int stereo, fmt, error ;
330
331 if ((opensoundsys_out = calloc (1, sizeof (OSS_AUDIO_OUT))) == NULL)
332 { perror ("opensoundsys_open : malloc ") ;
333 exit (1) ;
334 } ;
335
336 opensoundsys_out->magic = OSS_MAGIC ;
337 opensoundsys_out->channels = channels ;
338
339 if ((opensoundsys_out->fd = open ("/dev/dsp", O_WRONLY, 0)) == -1)
340 { perror ("opensoundsys_open : open ") ;
341 exit (1) ;
342 } ;
343
344 stereo = 0 ;
345 if (ioctl (opensoundsys_out->fd, SNDCTL_DSP_STEREO, &stereo) == -1)
346 { /* Fatal error */
347 perror ("opensoundsys_open : stereo ") ;
348 exit (1) ;
349 } ;
350
351 if (ioctl (opensoundsys_out->fd, SNDCTL_DSP_RESET, 0))
352 { perror ("opensoundsys_open : reset ") ;
353 exit (1) ;
354 } ;
355
356 fmt = CPU_IS_BIG_ENDIAN ? AFMT_S16_BE : AFMT_S16_LE ;
357 if (ioctl (opensoundsys_out->fd, SNDCTL_DSP_SETFMT, &fmt) != 0)
358 { perror ("opensoundsys_open_dsp_device : set format ") ;
359 exit (1) ;
360 } ;
361
362 if ((error = ioctl (opensoundsys_out->fd, SNDCTL_DSP_CHANNELS, &channels)) != 0)
363 { perror ("opensoundsys_open : channels ") ;
364 exit (1) ;
365 } ;
366
367 if ((error = ioctl (opensoundsys_out->fd, SNDCTL_DSP_SPEED, &samplerate)) != 0)
368 { perror ("opensoundsys_open : sample rate ") ;
369 exit (1) ;
370 } ;
371
372 if ((error = ioctl (opensoundsys_out->fd, SNDCTL_DSP_SYNC, 0)) != 0)
373 { perror ("opensoundsys_open : sync ") ;
374 exit (1) ;
375 } ;
376
377 return (AUDIO_OUT*) opensoundsys_out ;
378} /* opensoundsys_open */
379
380static void
381opensoundsys_play (get_audio_callback_t callback, AUDIO_OUT *audio_out, void *callback_data)
382{ OSS_AUDIO_OUT *opensoundsys_out ;
383 static float float_buffer [BUFFER_LEN] ;
384 static short buffer [BUFFER_LEN] ;
385 int k, read_frames ;
386
387 if ((opensoundsys_out = (OSS_AUDIO_OUT*) audio_out) == NULL)
388 { printf ("opensoundsys_play : AUDIO_OUT is NULL.\n") ;
389 return ;
390 } ;
391
392 if (opensoundsys_out->magic != OSS_MAGIC)
393 { printf ("opensoundsys_play : Bad magic number.\n") ;
394 return ;
395 } ;
396
397 while ((read_frames = callback (callback_data, float_buffer, BUFFER_LEN / opensoundsys_out->channels)))
398 { for (k = 0 ; k < read_frames * opensoundsys_out->channels ; k++)
399 buffer [k] = lrint (32767.0 * float_buffer [k]) ;
400 (void) write (opensoundsys_out->fd, buffer, read_frames * opensoundsys_out->channels * sizeof (short)) ;
401 } ;
402
403 return ;
404} /* opensoundsys_play */
405
406static void
407opensoundsys_close (AUDIO_OUT *audio_out)
408{ OSS_AUDIO_OUT *opensoundsys_out ;
409
410 if ((opensoundsys_out = (OSS_AUDIO_OUT*) audio_out) == NULL)
411 { printf ("opensoundsys_close : AUDIO_OUT is NULL.\n") ;
412 return ;
413 } ;
414
415 if (opensoundsys_out->magic != OSS_MAGIC)
416 { printf ("opensoundsys_close : Bad magic number.\n") ;
417 return ;
418 } ;
419
420 memset (opensoundsys_out, 0, sizeof (OSS_AUDIO_OUT)) ;
421
422 free (opensoundsys_out) ;
423
424 return ;
425} /* opensoundsys_close */
426
427#endif /* __linux__ */
428
429/*------------------------------------------------------------------------------
430** Mac OS X functions for playing a sound.
431*/
432
433#if (defined (__MACH__) && defined (__APPLE__)) /* MacOSX */
434
435#include <Carbon.h>
436#include <CoreAudio/AudioHardware.h>
437
438#define MACOSX_MAGIC MAKE_MAGIC ('M', 'a', 'c', ' ', 'O', 'S', ' ', 'X')
439
440typedef struct
441{ int magic ;
442 AudioStreamBasicDescription format ;
443
444 UInt32 buf_size ;
445 AudioDeviceID device ;
446
447 int channels ;
448 int samplerate ;
449 int buffer_size ;
450 int done_playing ;
451
452 get_audio_callback_t callback ;
453
454 void *callback_data ;
455} MACOSX_AUDIO_OUT ;
456
457static AUDIO_OUT *macosx_open (int channels, int samplerate) ;
458static void macosx_play (get_audio_callback_t callback, AUDIO_OUT *audio_out, void *callback_data) ;
459static void macosx_close (AUDIO_OUT *audio_out) ;
460
461static OSStatus
462macosx_audio_out_callback (AudioDeviceID device, const AudioTimeStamp* current_time,
463 const AudioBufferList* data_in, const AudioTimeStamp* time_in,
464 AudioBufferList* data_out, const AudioTimeStamp* time_out, void* client_data) ;
465
466
467static AUDIO_OUT *
468macosx_open (int channels, int samplerate)
469{ MACOSX_AUDIO_OUT *macosx_out ;
470 OSStatus err ;
471 size_t count ;
472
473 if ((macosx_out = calloc (1, sizeof (MACOSX_AUDIO_OUT))) == NULL)
474 { perror ("macosx_open : malloc ") ;
475 exit (1) ;
476 } ;
477
478 macosx_out->magic = MACOSX_MAGIC ;
479 macosx_out->channels = channels ;
480 macosx_out->samplerate = samplerate ;
481
482 macosx_out->device = kAudioDeviceUnknown ;
483
484 /* get the default output device for the HAL */
485 count = sizeof (AudioDeviceID) ;
486 if ((err = AudioHardwareGetProperty (kAudioHardwarePropertyDefaultOutputDevice,
487 &count, (void *) &(macosx_out->device))) != noErr)
488 { printf ("AudioHardwareGetProperty failed.\n") ;
489 free (macosx_out) ;
490 return NULL ;
491 } ;
492
493 /* get the buffersize that the default device uses for IO */
494 count = sizeof (UInt32) ;
495 if ((err = AudioDeviceGetProperty (macosx_out->device, 0, false, kAudioDevicePropertyBufferSize,
496 &count, &(macosx_out->buffer_size))) != noErr)
497 { printf ("AudioDeviceGetProperty (AudioDeviceGetProperty) failed.\n") ;
498 free (macosx_out) ;
499 return NULL ;
500 } ;
501
502 /* get a description of the data format used by the default device */
503 count = sizeof (AudioStreamBasicDescription) ;
504 if ((err = AudioDeviceGetProperty (macosx_out->device, 0, false, kAudioDevicePropertyStreamFormat,
505 &count, &(macosx_out->format))) != noErr)
506 { printf ("AudioDeviceGetProperty (kAudioDevicePropertyStreamFormat) failed.\n") ;
507 free (macosx_out) ;
508 return NULL ;
509 } ;
510
511 macosx_out->format.mSampleRate = samplerate ;
512 macosx_out->format.mChannelsPerFrame = channels ;
513
514 if ((err = AudioDeviceSetProperty (macosx_out->device, NULL, 0, false, kAudioDevicePropertyStreamFormat,
515 sizeof (AudioStreamBasicDescription), &(macosx_out->format))) != noErr)
516 { printf ("AudioDeviceSetProperty (kAudioDevicePropertyStreamFormat) failed.\n") ;
517 free (macosx_out) ;
518 return NULL ;
519 } ;
520
521 /* we want linear pcm */
522 if (macosx_out->format.mFormatID != kAudioFormatLinearPCM)
523 { free (macosx_out) ;
524 return NULL ;
525 } ;
526
527 macosx_out->done_playing = 0 ;
528
529 /* Fire off the device. */
530 if ((err = AudioDeviceAddIOProc (macosx_out->device, macosx_audio_out_callback,
531 (void *) macosx_out)) != noErr)
532 { printf ("AudioDeviceAddIOProc failed.\n") ;
533 free (macosx_out) ;
534 return NULL ;
535 } ;
536
537 return (MACOSX_AUDIO_OUT *) macosx_out ;
538} /* macosx_open */
539
540static void
541macosx_play (get_audio_callback_t callback, AUDIO_OUT *audio_out, void *callback_data)
542{ MACOSX_AUDIO_OUT *macosx_out ;
543 OSStatus err ;
544
545 if ((macosx_out = (MACOSX_AUDIO_OUT*) audio_out) == NULL)
546 { printf ("macosx_play : AUDIO_OUT is NULL.\n") ;
547 return ;
548 } ;
549
550 if (macosx_out->magic != MACOSX_MAGIC)
551 { printf ("macosx_play : Bad magic number.\n") ;
552 return ;
553 } ;
554
555 /* Set the callback function and callback data. */
556 macosx_out->callback = callback ;
557 macosx_out->callback_data = callback_data ;
558
559 err = AudioDeviceStart (macosx_out->device, macosx_audio_out_callback) ;
560 if (err != noErr)
561 printf ("AudioDeviceStart failed.\n") ;
562
563 while (macosx_out->done_playing == SF_FALSE)
564 usleep (10 * 1000) ; /* 10 000 milliseconds. */
565
566 return ;
567} /* macosx_play */
568
569static void
570macosx_close (AUDIO_OUT *audio_out)
571{ MACOSX_AUDIO_OUT *macosx_out ;
572 OSStatus err ;
573
574 if ((macosx_out = (MACOSX_AUDIO_OUT*) audio_out) == NULL)
575 { printf ("macosx_close : AUDIO_OUT is NULL.\n") ;
576 return ;
577 } ;
578
579 if (macosx_out->magic != MACOSX_MAGIC)
580 { printf ("macosx_close : Bad magic number.\n") ;
581 return ;
582 } ;
583
584
585 if ((err = AudioDeviceStop (macosx_out->device, macosx_audio_out_callback)) != noErr)
586 { printf ("AudioDeviceStop failed.\n") ;
587 return ;
588 } ;
589
590 err = AudioDeviceRemoveIOProc (macosx_out->device, macosx_audio_out_callback) ;
591 if (err != noErr)
592 { printf ("AudioDeviceRemoveIOProc failed.\n") ;
593 return ;
594 } ;
595
596} /* macosx_close */
597
598static OSStatus
599macosx_audio_out_callback (AudioDeviceID device, const AudioTimeStamp* current_time,
600 const AudioBufferList* data_in, const AudioTimeStamp* time_in,
601 AudioBufferList* data_out, const AudioTimeStamp* time_out, void* client_data)
602{ MACOSX_AUDIO_OUT *macosx_out ;
603 int k, size, frame_count, read_count ;
604 float *buffer ;
605
606 if ((macosx_out = (MACOSX_AUDIO_OUT*) client_data) == NULL)
607 { printf ("macosx_play : AUDIO_OUT is NULL.\n") ;
608 return 42 ;
609 } ;
610
611 if (macosx_out->magic != MACOSX_MAGIC)
612 { printf ("macosx_play : Bad magic number.\n") ;
613 return 42 ;
614 } ;
615
616 size = data_out->mBuffers [0].mDataByteSize ;
617 frame_count = size / sizeof (float) / macosx_out->channels ;
618
619 buffer = (float*) data_out->mBuffers [0].mData ;
620
621 read_count = macosx_out->callback (macosx_out->callback_data, buffer, frame_count) ;
622
623 if (read_count < frame_count)
624 { memset (&(buffer [read_count]), 0, (frame_count - read_count) * sizeof (float)) ;
625 macosx_out->done_playing = 1 ;
626 } ;
627
628 return noErr ;
629} /* macosx_audio_out_callback */
630
631#endif /* MacOSX */
632
633
634/*------------------------------------------------------------------------------
635** Win32 functions for playing a sound.
636**
637** This API sucks. Its needlessly complicated and is *WAY* too loose with
638** passing pointers arounf in integers and and using char* pointers to
639** point to data instead of short*. It plain sucks!
640*/
641
642#if (defined (_WIN32) || defined (WIN32))
643
644#include <windows.h>
645#include <mmsystem.h>
646
647#define WIN32_BUFFER_LEN (1<<15)
648#define WIN32_MAGIC MAKE_MAGIC ('W', 'i', 'n', '3', '2', 's', 'u', 'x')
649
650typedef struct
651{ int magic ;
652
653 HWAVEOUT hwave ;
654 WAVEHDR whdr [2] ;
655
656 HANDLE Event ;
657
658 short short_buffer [WIN32_BUFFER_LEN / sizeof (short)] ;
659 float float_buffer [WIN32_BUFFER_LEN / sizeof (short) / 2] ;
660
661 int bufferlen, current ;
662
663 int channels ;
664
665 get_audio_callback_t callback ;
666
667 void *callback_data ;
668} WIN32_AUDIO_OUT ;
669
670static AUDIO_OUT *win32_open (int channels, int samplerate) ;
671static void win32_play (get_audio_callback_t callback, AUDIO_OUT *audio_out, void *callback_data) ;
672static void win32_close (AUDIO_OUT *audio_out) ;
673
674static DWORD CALLBACK
675 win32_audio_out_callback (HWAVEOUT hwave, UINT msg, DWORD data, DWORD param1, DWORD param2) ;
676
677static AUDIO_OUT*
678win32_open (int channels, int samplerate)
679{ WIN32_AUDIO_OUT *win32_out ;
680
681 WAVEFORMATEX wf ;
682 int error ;
683
684 if ((win32_out = calloc (1, sizeof (WIN32_AUDIO_OUT))) == NULL)
685 { perror ("win32_open : malloc ") ;
686 exit (1) ;
687 } ;
688
689 win32_out->magic = WIN32_MAGIC ;
690 win32_out->channels = channels ;
691
692 win32_out->current = 0 ;
693
694 win32_out->Event = CreateEvent (0, FALSE, FALSE, 0) ;
695
696 wf.nChannels = channels ;
697 wf.nSamplesPerSec = samplerate ;
698 wf.nBlockAlign = channels * sizeof (short) ;
699
700 wf.wFormatTag = WAVE_FORMAT_PCM ;
701 wf.cbSize = 0 ;
702 wf.wBitsPerSample = 16 ;
703 wf.nAvgBytesPerSec = wf.nBlockAlign * wf.nSamplesPerSec ;
704
705 error = waveOutOpen (&(win32_out->hwave), WAVE_MAPPER, &wf, (DWORD) win32_audio_out_callback,
706 (DWORD) win32_out, CALLBACK_FUNCTION) ;
707 if (error)
708 { puts ("waveOutOpen failed.") ;
709 free (win32_out) ;
710 return NULL ;
711 } ;
712
713 waveOutPause (win32_out->hwave) ;
714
715 return (WIN32_AUDIO_OUT *) win32_out ;
716} /* win32_open */
717
718static void
719win32_play (get_audio_callback_t callback, AUDIO_OUT *audio_out, void *callback_data)
720{ WIN32_AUDIO_OUT *win32_out ;
721 int error ;
722
723 if ((win32_out = (WIN32_AUDIO_OUT*) audio_out) == NULL)
724 { printf ("win32_play : AUDIO_OUT is NULL.\n") ;
725 return ;
726 } ;
727
728 if (win32_out->magic != WIN32_MAGIC)
729 { printf ("win32_play : Bad magic number (%d %d).\n", win32_out->magic, WIN32_MAGIC) ;
730 return ;
731 } ;
732
733 /* Set the callback function and callback data. */
734 win32_out->callback = callback ;
735 win32_out->callback_data = callback_data ;
736
737 win32_out->whdr [0].lpData = (char*) win32_out->short_buffer ;
738 win32_out->whdr [1].lpData = ((char*) win32_out->short_buffer) + sizeof (win32_out->short_buffer) / 2 ;
739
740 win32_out->whdr [0].dwBufferLength = sizeof (win32_out->short_buffer) / 2 ;
741 win32_out->whdr [1].dwBufferLength = sizeof (win32_out->short_buffer) / 2 ;
742
743 win32_out->bufferlen = sizeof (win32_out->short_buffer) / 2 / sizeof (short) ;
744
745 /* Prepare the WAVEHDRs */
746 if ((error = waveOutPrepareHeader (win32_out->hwave, &(win32_out->whdr [0]), sizeof (WAVEHDR))))
747 { printf ("waveOutPrepareHeader [0] failed : %08X\n", error) ;
748 waveOutClose (win32_out->hwave) ;
749 return ;
750 } ;
751
752 if ((error = waveOutPrepareHeader (win32_out->hwave, &(win32_out->whdr [1]), sizeof (WAVEHDR))))
753 { printf ("waveOutPrepareHeader [1] failed : %08X\n", error) ;
754 waveOutUnprepareHeader (win32_out->hwave, &(win32_out->whdr [0]), sizeof (WAVEHDR)) ;
755 waveOutClose (win32_out->hwave) ;
756 return ;
757 } ;
758
759 waveOutRestart (win32_out->hwave) ;
760
761 /* Fake 2 calls to the callback function to queue up enough audio. */
762 win32_audio_out_callback (0, MM_WOM_DONE, (DWORD) win32_out, 0, 0) ;
763 win32_audio_out_callback (0, MM_WOM_DONE, (DWORD) win32_out, 0, 0) ;
764
765 /* Wait for playback to finish. The callback notifies us when all
766 ** wave data has been played.
767 */
768 WaitForSingleObject (win32_out->Event, INFINITE) ;
769
770 waveOutPause (win32_out->hwave) ;
771 waveOutReset (win32_out->hwave) ;
772
773 waveOutUnprepareHeader (win32_out->hwave, &(win32_out->whdr [0]), sizeof (WAVEHDR)) ;
774 waveOutUnprepareHeader (win32_out->hwave, &(win32_out->whdr [1]), sizeof (WAVEHDR)) ;
775
776 waveOutClose (win32_out->hwave) ;
777 win32_out->hwave = 0 ;
778
779 return ;
780} /* win32_play */
781
782static void
783win32_close (AUDIO_OUT *audio_out)
784{ WIN32_AUDIO_OUT *win32_out ;
785
786 if ((win32_out = (WIN32_AUDIO_OUT*) audio_out) == NULL)
787 { printf ("win32_close : AUDIO_OUT is NULL.\n") ;
788 return ;
789 } ;
790
791 if (win32_out->magic != WIN32_MAGIC)
792 { printf ("win32_close : Bad magic number.\n") ;
793 return ;
794 } ;
795
796 memset (win32_out, 0, sizeof (WIN32_AUDIO_OUT)) ;
797
798 free (win32_out) ;
799} /* win32_close */
800
801static DWORD CALLBACK
802win32_audio_out_callback (HWAVEOUT hwave, UINT msg, DWORD data, DWORD param1, DWORD param2)
803{ WIN32_AUDIO_OUT *win32_out ;
804 int read_count, frame_count, k ;
805 short *sptr ;
806
807 /*
808 ** I consider this technique of passing a pointer via an integer as
809 ** fundamentally broken but thats the way microsoft has defined the
810 ** interface.
811 */
812 if ((win32_out = (WIN32_AUDIO_OUT*) data) == NULL)
813 { printf ("win32_audio_out_callback : AUDIO_OUT is NULL.\n") ;
814 return 1 ;
815 } ;
816
817 if (win32_out->magic != WIN32_MAGIC)
818 { printf ("win32_audio_out_callback : Bad magic number (%d %d).\n", win32_out->magic, WIN32_MAGIC) ;
819 return 1 ;
820 } ;
821
822 if (msg != MM_WOM_DONE)
823 return 0 ;
824
825 /* Do the actual audio. */
826 sample_count = win32_out->bufferlen ;
827 frame_count = sample_count / win32_out->channels ;
828
829 read_count = win32_out->callback (win32_out->callback_data, win32_out->float_buffer, frame_count) ;
830
831 sptr = (short*) win32_out->whdr [win32_out->current].lpData ;
832
833 for (k = 0 ; k < read_count ; k++)
834 sptr [k] = lrint (32767.0 * win32_out->float_buffer [k]) ;
835
836 if (read_count > 0)
837 { /* Fix buffer length is only a partial block. */
838 if (read_count * sizeof (short) < win32_out->bufferlen)
839 win32_out->whdr [win32_out->current].dwBufferLength = read_count * sizeof (short) ;
840
841 /* Queue the WAVEHDR */
842 waveOutWrite (win32_out->hwave, (LPWAVEHDR) &(win32_out->whdr [win32_out->current]), sizeof (WAVEHDR)) ;
843 }
844 else
845 { /* Stop playback */
846 waveOutPause (win32_out->hwave) ;
847
848 SetEvent (win32_out->Event) ;
849 } ;
850
851 win32_out->current = (win32_out->current + 1) % 2 ;
852
853 return 0 ;
854} /* win32_audio_out_callback */
855
856#endif /* Win32 */
857
858/*------------------------------------------------------------------------------
859** Solaris.
860*/
861
862#if (defined (sun) && defined (unix)) /* ie Solaris */
863
864#include <fcntl.h>
865#include <sys/ioctl.h>
866#include <sys/audioio.h>
867
868#define SOLARIS_MAGIC MAKE_MAGIC ('S', 'o', 'l', 'a', 'r', 'i', 's', ' ')
869
870typedef struct
871{ int magic ;
872 int fd ;
873 int channels ;
874 int samplerate ;
875} SOLARIS_AUDIO_OUT ;
876
877static AUDIO_OUT *solaris_open (int channels, int samplerate) ;
878static void solaris_play (get_audio_callback_t callback, AUDIO_OUT *audio_out, void *callback_data) ;
879static void solaris_close (AUDIO_OUT *audio_out) ;
880
881static AUDIO_OUT *
882solaris_open (int channels, int samplerate)
883{ SOLARIS_AUDIO_OUT *solaris_out ;
884 audio_info_t audio_info ;
885 int error ;
886
887 if ((solaris_out = calloc (1, sizeof (SOLARIS_AUDIO_OUT))) == NULL)
888 { perror ("solaris_open : malloc ") ;
889 exit (1) ;
890 } ;
891
892 solaris_out->magic = SOLARIS_MAGIC ;
893 solaris_out->channels = channels ;
894 solaris_out->samplerate = channels ;
895
896 /* open the audio device - write only, non-blocking */
897 if ((solaris_out->fd = open ("/dev/audio", O_WRONLY | O_NONBLOCK)) < 0)
898 { perror ("open (/dev/audio) failed") ;
899 exit (1) ;
900 } ;
901
902 /* Retrive standard values. */
903 AUDIO_INITINFO (&audio_info) ;
904
905 audio_info.play.sample_rate = samplerate ;
906 audio_info.play.channels = channels ;
907 audio_info.play.precision = 16 ;
908 audio_info.play.encoding = AUDIO_ENCODING_LINEAR ;
909 audio_info.play.gain = AUDIO_MAX_GAIN ;
910 audio_info.play.balance = AUDIO_MID_BALANCE ;
911
912 if ((error = ioctl (solaris_out->fd, AUDIO_SETINFO, &audio_info)))
913 { perror ("ioctl (AUDIO_SETINFO) failed") ;
914 exit (1) ;
915 } ;
916
917 return (AUDIO_OUT*) solaris_out ;
918} /* solaris_open */
919
920static void
921solaris_play (get_audio_callback_t callback, AUDIO_OUT *audio_out, void *callback_data)
922{ SOLARIS_AUDIO_OUT *solaris_out ;
923 static float float_buffer [BUFFER_LEN] ;
924 static short buffer [BUFFER_LEN] ;
925 int k, read_frames ;
926
927 if ((solaris_out = (SOLARIS_AUDIO_OUT*) audio_out) == NULL)
928 { printf ("solaris_play : AUDIO_OUT is NULL.\n") ;
929 return ;
930 } ;
931
932 if (solaris_out->magic != SOLARIS_MAGIC)
933 { printf ("solaris_play : Bad magic number.\n") ;
934 return ;
935 } ;
936
937 while ((read_frames = callback (callback_data, float_buffer, BUFFER_LEN / solaris_out->channels)))
938 { for (k = 0 ; k < read_frames * solaris_out->channels ; k++)
939 buffer [k] = lrint (32767.0 * float_buffer [k]) ;
940 write (solaris_out->fd, buffer, read_frames * solaris_out->channels * sizeof (short)) ;
941 } ;
942
943 return ;
944} /* solaris_play */
945
946static void
947solaris_close (AUDIO_OUT *audio_out)
948{ SOLARIS_AUDIO_OUT *solaris_out ;
949
950 if ((solaris_out = (SOLARIS_AUDIO_OUT*) audio_out) == NULL)
951 { printf ("solaris_close : AUDIO_OUT is NULL.\n") ;
952 return ;
953 } ;
954
955 if (solaris_out->magic != SOLARIS_MAGIC)
956 { printf ("solaris_close : Bad magic number.\n") ;
957 return ;
958 } ;
959
960 memset (solaris_out, 0, sizeof (SOLARIS_AUDIO_OUT)) ;
961
962 free (solaris_out) ;
963
964 return ;
965} /* solaris_close */
966
967#endif /* Solaris */
968
969/*==============================================================================
970** Main function.
971*/
972
973AUDIO_OUT *
974audio_open (int channels, int samplerate)
975{
976#if defined (__linux__)
977 #if HAVE_ALSA_ASOUNDLIB_H
978 if (access ("/proc/asound/cards", R_OK) == 0)
979 return alsa_open (channels, samplerate) ;
980 #endif
981 return opensoundsys_open (channels, samplerate) ;
982#elif (defined (__MACH__) && defined (__APPLE__))
983 return macosx_open (channels, samplerate) ;
984#elif (defined (sun) && defined (unix))
985 return solaris_open (channels, samplerate) ;
986#elif (defined (_WIN32) || defined (WIN32))
987 return win32_open (channels, samplerate) ;
988#else
989 #warning "*** Playing sound not yet supported on this platform."
990 #warning "*** Please feel free to submit a patch."
991 printf ("Error : Playing sound not yet supported on this platform.\n") ;
992 return NULL ;
993#endif
994
995
996 return NULL ;
997} /* audio_open */
998
999void
1000audio_play (get_audio_callback_t callback, AUDIO_OUT *audio_out, void *callback_data)
1001{
1002
1003 if (callback == NULL)
1004 { printf ("Error : bad callback pointer.\n") ;
1005 return ;
1006 } ;
1007
1008 if (audio_out == NULL)
1009 { printf ("Error : bad audio_out pointer.\n") ;
1010 return ;
1011 } ;
1012
1013 if (callback_data == NULL)
1014 { printf ("Error : bad callback_data pointer.\n") ;
1015 return ;
1016 } ;
1017
1018#if defined (__linux__)
1019 #if HAVE_ALSA_ASOUNDLIB_H
1020 if (audio_out->magic == ALSA_MAGIC)
1021 alsa_play (callback, audio_out, callback_data) ;
1022 #endif
1023 opensoundsys_play (callback, audio_out, callback_data) ;
1024#elif (defined (__MACH__) && defined (__APPLE__))
1025 macosx_play (callback, audio_out, callback_data) ;
1026#elif (defined (sun) && defined (unix))
1027 solaris_play (callback, audio_out, callback_data) ;
1028#elif (defined (_WIN32) || defined (WIN32))
1029 win32_play (callback, audio_out, callback_data) ;
1030#else
1031 #warning "*** Playing sound not yet supported on this platform."
1032 #warning "*** Please feel free to submit a patch."
1033 printf ("Error : Playing sound not yet supported on this platform.\n") ;
1034 return ;
1035#endif
1036
1037 return ;
1038} /* audio_play */
1039
1040void
1041audio_close (AUDIO_OUT *audio_out)
1042{
1043#if defined (__linux__)
1044 #if HAVE_ALSA_ASOUNDLIB_H
1045 if (audio_out->magic == ALSA_MAGIC)
1046 alsa_close (audio_out) ;
1047 #endif
1048 opensoundsys_close (audio_out) ;
1049#elif (defined (__MACH__) && defined (__APPLE__))
1050 macosx_close (audio_out) ;
1051#elif (defined (sun) && defined (unix))
1052 solaris_close (audio_out) ;
1053#elif (defined (_WIN32) || defined (WIN32))
1054 win32_close (audio_out) ;
1055#else
1056 #warning "*** Playing sound not yet supported on this platform."
1057 #warning "*** Please feel free to submit a patch."
1058 printf ("Error : Playing sound not yet supported on this platform.\n") ;
1059 return ;
1060#endif
1061
1062 return ;
1063} /* audio_close */
1064
1065#else /* (HAVE_SNDFILE == 0) */
1066
1067/* Do not have libsndfile installed so just return. */
1068
1069AUDIO_OUT *
1070audio_open (int channels, int samplerate)
1071{
1072 (void) channels ;
1073 (void) samplerate ;
1074
1075 return NULL ;
1076} /* audio_open */
1077
1078void
1079audio_play (get_audio_callback_t callback, AUDIO_OUT *audio_out, void *callback_data)
1080{
1081 (void) callback ;
1082 (void) audio_out ;
1083 (void) callback_data ;
1084
1085 return ;
1086} /* audio_play */
1087
1088void
1089audio_close (AUDIO_OUT *audio_out)
1090{
1091 audio_out = audio_out ;
1092
1093 return ;
1094} /* audio_close */
1095
1096#endif /* HAVE_SNDFILE */
1097
interface to Secret Rabbit Code audio sample rate converter
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