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[emacs.git] / src / sound.c
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1 /* sound.c -- sound support.
2 Copyright (C) 1998, 1999, 2001, 2002, 2003, 2004,
3 2005, 2006, 2007, 2008 Free Software Foundation, Inc.
5 This file is part of GNU Emacs.
7 GNU Emacs is free software: you can redistribute it and/or modify
8 it under the terms of the GNU General Public License as published by
9 the Free Software Foundation, either version 3 of the License, or
10 (at your option) any later version.
12 GNU Emacs is distributed in the hope that it will be useful,
13 but WITHOUT ANY WARRANTY; without even the implied warranty of
14 MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
15 GNU General Public License for more details.
17 You should have received a copy of the GNU General Public License
18 along with GNU Emacs. If not, see <http://www.gnu.org/licenses/>. */
20 /* Written by Gerd Moellmann <gerd@gnu.org>. Tested with Luigi's
21 driver on FreeBSD 2.2.7 with a SoundBlaster 16. */
24 Modified by Ben Key <Bkey1@tampabay.rr.com> to add a partial
25 implementation of the play-sound specification for Windows.
27 Notes:
28 In the Windows implementation of play-sound-internal only the
29 :file and :volume keywords are supported. The :device keyword,
30 if present, is ignored. The :data keyword, if present, will
31 cause an error to be generated.
33 The Windows implementation of play-sound is implemented via the
34 Win32 API functions mciSendString, waveOutGetVolume, and
35 waveOutSetVolume which are exported by Winmm.dll.
38 #include <config.h>
40 #if defined HAVE_SOUND
42 /* BEGIN: Common Includes */
43 #include <fcntl.h>
44 #include <unistd.h>
45 #include <sys/types.h>
46 #include <errno.h>
47 #include "lisp.h"
48 #include "dispextern.h"
49 #include "atimer.h"
50 #include <signal.h>
51 #include "syssignal.h"
52 /* END: Common Includes */
55 /* BEGIN: Non Windows Includes */
56 #ifndef WINDOWSNT
58 #ifndef MSDOS
59 #include <sys/ioctl.h>
60 #endif
62 /* FreeBSD has machine/soundcard.h. Voxware sound driver docs mention
63 sys/soundcard.h. So, let's try whatever's there. */
65 #ifdef HAVE_MACHINE_SOUNDCARD_H
66 #include <machine/soundcard.h>
67 #endif
68 #ifdef HAVE_SYS_SOUNDCARD_H
69 #include <sys/soundcard.h>
70 #endif
71 #ifdef HAVE_SOUNDCARD_H
72 #include <soundcard.h>
73 #endif
74 #ifdef HAVE_ALSA
75 #ifdef ALSA_SUBDIR_INCLUDE
76 #include <alsa/asoundlib.h>
77 #else
78 #include <asoundlib.h>
79 #endif /* ALSA_SUBDIR_INCLUDE */
80 #endif /* HAVE_ALSA */
82 /* END: Non Windows Includes */
84 #else /* WINDOWSNT */
86 /* BEGIN: Windows Specific Includes */
87 #include <stdio.h>
88 #include <stdlib.h>
89 #include <string.h>
90 #include <limits.h>
91 #include <windows.h>
92 #include <mmsystem.h>
93 /* END: Windows Specific Includes */
95 #endif /* WINDOWSNT */
97 /* BEGIN: Common Definitions */
99 /* Symbols. */
101 extern Lisp_Object QCfile, QCdata;
102 Lisp_Object QCvolume, QCdevice;
103 Lisp_Object Qsound;
104 Lisp_Object Qplay_sound_functions;
106 /* Indices of attributes in a sound attributes vector. */
108 enum sound_attr
110 SOUND_FILE,
111 SOUND_DATA,
112 SOUND_DEVICE,
113 SOUND_VOLUME,
114 SOUND_ATTR_SENTINEL
117 static void alsa_sound_perror P_ ((char *, int)) NO_RETURN;
118 static void sound_perror P_ ((char *)) NO_RETURN;
119 static void sound_warning P_ ((char *));
120 static int parse_sound P_ ((Lisp_Object, Lisp_Object *));
122 /* END: Common Definitions */
124 /* BEGIN: Non Windows Definitions */
125 #ifndef WINDOWSNT
127 #ifndef DEFAULT_SOUND_DEVICE
128 #define DEFAULT_SOUND_DEVICE "/dev/dsp"
129 #endif
130 #ifndef DEFAULT_ALSA_SOUND_DEVICE
131 #define DEFAULT_ALSA_SOUND_DEVICE "default"
132 #endif
135 /* Structure forward declarations. */
137 struct sound;
138 struct sound_device;
140 /* The file header of RIFF-WAVE files (*.wav). Files are always in
141 little-endian byte-order. */
143 struct wav_header
145 u_int32_t magic;
146 u_int32_t length;
147 u_int32_t chunk_type;
148 u_int32_t chunk_format;
149 u_int32_t chunk_length;
150 u_int16_t format;
151 u_int16_t channels;
152 u_int32_t sample_rate;
153 u_int32_t bytes_per_second;
154 u_int16_t sample_size;
155 u_int16_t precision;
156 u_int32_t chunk_data;
157 u_int32_t data_length;
160 /* The file header of Sun adio files (*.au). Files are always in
161 big-endian byte-order. */
163 struct au_header
165 /* ASCII ".snd" */
166 u_int32_t magic_number;
168 /* Offset of data part from start of file. Minimum value is 24. */
169 u_int32_t data_offset;
171 /* Size of data part, 0xffffffff if unknown. */
172 u_int32_t data_size;
174 /* Data encoding format.
175 1 8-bit ISDN u-law
176 2 8-bit linear PCM (REF-PCM)
177 3 16-bit linear PCM
178 4 24-bit linear PCM
179 5 32-bit linear PCM
180 6 32-bit IEEE floating-point
181 7 64-bit IEEE floating-point
182 23 8-bit u-law compressed using CCITT 0.721 ADPCM voice data
183 encoding scheme. */
184 u_int32_t encoding;
186 /* Number of samples per second. */
187 u_int32_t sample_rate;
189 /* Number of interleaved channels. */
190 u_int32_t channels;
193 /* Maximum of all sound file headers sizes. */
195 #define MAX_SOUND_HEADER_BYTES \
196 max (sizeof (struct wav_header), sizeof (struct au_header))
198 /* Interface structure for sound devices. */
200 struct sound_device
202 /* The name of the device or null meaning use a default device name. */
203 char *file;
205 /* File descriptor of the device. */
206 int fd;
208 /* Device-dependent format. */
209 int format;
211 /* Volume (0..100). Zero means unspecified. */
212 int volume;
214 /* Sample size. */
215 int sample_size;
217 /* Sample rate. */
218 int sample_rate;
220 /* Bytes per second. */
221 int bps;
223 /* 1 = mono, 2 = stereo, 0 = don't set. */
224 int channels;
226 /* Open device SD. */
227 void (* open) P_ ((struct sound_device *sd));
229 /* Close device SD. */
230 void (* close) P_ ((struct sound_device *sd));
232 /* Configure SD accoring to device-dependent parameters. */
233 void (* configure) P_ ((struct sound_device *device));
235 /* Choose a device-dependent format for outputting sound S. */
236 void (* choose_format) P_ ((struct sound_device *sd,
237 struct sound *s));
239 /* Return a preferred data size in bytes to be sent to write (below)
240 each time. 2048 is used if this is NULL. */
241 int (* period_size) P_ ((struct sound_device *sd));
243 /* Write NYBTES bytes from BUFFER to device SD. */
244 void (* write) P_ ((struct sound_device *sd, const char *buffer,
245 int nbytes));
247 /* A place for devices to store additional data. */
248 void *data;
251 /* An enumerator for each supported sound file type. */
253 enum sound_type
255 RIFF,
256 SUN_AUDIO
259 /* Interface structure for sound files. */
261 struct sound
263 /* The type of the file. */
264 enum sound_type type;
266 /* File descriptor of a sound file. */
267 int fd;
269 /* Pointer to sound file header. This contains header_size bytes
270 read from the start of a sound file. */
271 char *header;
273 /* Number of bytes raed from sound file. This is always <=
274 MAX_SOUND_HEADER_BYTES. */
275 int header_size;
277 /* Sound data, if a string. */
278 Lisp_Object data;
280 /* Play sound file S on device SD. */
281 void (* play) P_ ((struct sound *s, struct sound_device *sd));
284 /* These are set during `play-sound-internal' so that sound_cleanup has
285 access to them. */
287 struct sound_device *current_sound_device;
288 struct sound *current_sound;
290 /* Function prototypes. */
292 static void vox_open P_ ((struct sound_device *));
293 static void vox_configure P_ ((struct sound_device *));
294 static void vox_close P_ ((struct sound_device *sd));
295 static void vox_choose_format P_ ((struct sound_device *, struct sound *));
296 static int vox_init P_ ((struct sound_device *));
297 static void vox_write P_ ((struct sound_device *, const char *, int));
298 static void find_sound_type P_ ((struct sound *));
299 static u_int32_t le2hl P_ ((u_int32_t));
300 static u_int16_t le2hs P_ ((u_int16_t));
301 static u_int32_t be2hl P_ ((u_int32_t));
302 static int wav_init P_ ((struct sound *));
303 static void wav_play P_ ((struct sound *, struct sound_device *));
304 static int au_init P_ ((struct sound *));
305 static void au_play P_ ((struct sound *, struct sound_device *));
307 #if 0 /* Currently not used. */
308 static u_int16_t be2hs P_ ((u_int16_t));
309 #endif
311 /* END: Non Windows Definitions */
312 #else /* WINDOWSNT */
314 /* BEGIN: Windows Specific Definitions */
315 static int do_play_sound P_ ((const char *, unsigned long));
317 END: Windows Specific Definitions */
318 #endif /* WINDOWSNT */
321 /***********************************************************************
322 General
323 ***********************************************************************/
325 /* BEGIN: Common functions */
327 /* Like perror, but signals an error. */
329 static void
330 sound_perror (msg)
331 char *msg;
333 int saved_errno = errno;
335 turn_on_atimers (1);
336 #ifdef SIGIO
337 sigunblock (sigmask (SIGIO));
338 #endif
339 if (saved_errno != 0)
340 error ("%s: %s", msg, strerror (saved_errno));
341 else
342 error ("%s", msg);
346 /* Display a warning message. */
348 static void
349 sound_warning (msg)
350 char *msg;
352 message (msg);
356 /* Parse sound specification SOUND, and fill ATTRS with what is
357 found. Value is non-zero if SOUND Is a valid sound specification.
358 A valid sound specification is a list starting with the symbol
359 `sound'. The rest of the list is a property list which may
360 contain the following key/value pairs:
362 - `:file FILE'
364 FILE is the sound file to play. If it isn't an absolute name,
365 it's searched under `data-directory'.
367 - `:data DATA'
369 DATA is a string containing sound data. Either :file or :data
370 may be present, but not both.
372 - `:device DEVICE'
374 DEVICE is the name of the device to play on, e.g. "/dev/dsp2".
375 If not specified, a default device is used.
377 - `:volume VOL'
379 VOL must be an integer in the range [0, 100], or a float in the
380 range [0, 1]. */
382 static int
383 parse_sound (sound, attrs)
384 Lisp_Object sound;
385 Lisp_Object *attrs;
387 /* SOUND must be a list starting with the symbol `sound'. */
388 if (!CONSP (sound) || !EQ (XCAR (sound), Qsound))
389 return 0;
391 sound = XCDR (sound);
392 attrs[SOUND_FILE] = Fplist_get (sound, QCfile);
393 attrs[SOUND_DATA] = Fplist_get (sound, QCdata);
394 attrs[SOUND_DEVICE] = Fplist_get (sound, QCdevice);
395 attrs[SOUND_VOLUME] = Fplist_get (sound, QCvolume);
397 #ifndef WINDOWSNT
398 /* File name or data must be specified. */
399 if (!STRINGP (attrs[SOUND_FILE])
400 && !STRINGP (attrs[SOUND_DATA]))
401 return 0;
402 #else /* WINDOWSNT */
404 Data is not supported in Windows. Therefore a
405 File name MUST be supplied.
407 if (!STRINGP (attrs[SOUND_FILE]))
409 return 0;
411 #endif /* WINDOWSNT */
413 /* Volume must be in the range 0..100 or unspecified. */
414 if (!NILP (attrs[SOUND_VOLUME]))
416 if (INTEGERP (attrs[SOUND_VOLUME]))
418 if (XINT (attrs[SOUND_VOLUME]) < 0
419 || XINT (attrs[SOUND_VOLUME]) > 100)
420 return 0;
422 else if (FLOATP (attrs[SOUND_VOLUME]))
424 if (XFLOAT_DATA (attrs[SOUND_VOLUME]) < 0
425 || XFLOAT_DATA (attrs[SOUND_VOLUME]) > 1)
426 return 0;
428 else
429 return 0;
432 #ifndef WINDOWSNT
433 /* Device must be a string or unspecified. */
434 if (!NILP (attrs[SOUND_DEVICE])
435 && !STRINGP (attrs[SOUND_DEVICE]))
436 return 0;
437 #endif /* WINDOWSNT */
439 Since device is ignored in Windows, it does not matter
440 what it is.
442 return 1;
445 /* END: Common functions */
447 /* BEGIN: Non Windows functions */
448 #ifndef WINDOWSNT
450 /* Find out the type of the sound file whose file descriptor is FD.
451 S is the sound file structure to fill in. */
453 static void
454 find_sound_type (s)
455 struct sound *s;
457 if (!wav_init (s) && !au_init (s))
458 error ("Unknown sound format");
462 /* Function installed by play-sound-internal with record_unwind_protect. */
464 static Lisp_Object
465 sound_cleanup (arg)
466 Lisp_Object arg;
468 if (current_sound_device->close)
469 current_sound_device->close (current_sound_device);
470 if (current_sound->fd > 0)
471 emacs_close (current_sound->fd);
472 free (current_sound_device);
473 free (current_sound);
475 return Qnil;
478 /***********************************************************************
479 Byte-order Conversion
480 ***********************************************************************/
482 /* Convert 32-bit value VALUE which is in little-endian byte-order
483 to host byte-order. */
485 static u_int32_t
486 le2hl (value)
487 u_int32_t value;
489 #ifdef WORDS_BIG_ENDIAN
490 unsigned char *p = (unsigned char *) &value;
491 value = p[0] + (p[1] << 8) + (p[2] << 16) + (p[3] << 24);
492 #endif
493 return value;
497 /* Convert 16-bit value VALUE which is in little-endian byte-order
498 to host byte-order. */
500 static u_int16_t
501 le2hs (value)
502 u_int16_t value;
504 #ifdef WORDS_BIG_ENDIAN
505 unsigned char *p = (unsigned char *) &value;
506 value = p[0] + (p[1] << 8);
507 #endif
508 return value;
512 /* Convert 32-bit value VALUE which is in big-endian byte-order
513 to host byte-order. */
515 static u_int32_t
516 be2hl (value)
517 u_int32_t value;
519 #ifndef WORDS_BIG_ENDIAN
520 unsigned char *p = (unsigned char *) &value;
521 value = p[3] + (p[2] << 8) + (p[1] << 16) + (p[0] << 24);
522 #endif
523 return value;
527 #if 0 /* Currently not used. */
529 /* Convert 16-bit value VALUE which is in big-endian byte-order
530 to host byte-order. */
532 static u_int16_t
533 be2hs (value)
534 u_int16_t value;
536 #ifndef WORDS_BIG_ENDIAN
537 unsigned char *p = (unsigned char *) &value;
538 value = p[1] + (p[0] << 8);
539 #endif
540 return value;
543 #endif /* 0 */
545 /***********************************************************************
546 RIFF-WAVE (*.wav)
547 ***********************************************************************/
549 /* Try to initialize sound file S from S->header. S->header
550 contains the first MAX_SOUND_HEADER_BYTES number of bytes from the
551 sound file. If the file is a WAV-format file, set up interface
552 functions in S and convert header fields to host byte-order.
553 Value is non-zero if the file is a WAV file. */
555 static int
556 wav_init (s)
557 struct sound *s;
559 struct wav_header *header = (struct wav_header *) s->header;
561 if (s->header_size < sizeof *header
562 || bcmp (s->header, "RIFF", 4) != 0)
563 return 0;
565 /* WAV files are in little-endian order. Convert the header
566 if on a big-endian machine. */
567 header->magic = le2hl (header->magic);
568 header->length = le2hl (header->length);
569 header->chunk_type = le2hl (header->chunk_type);
570 header->chunk_format = le2hl (header->chunk_format);
571 header->chunk_length = le2hl (header->chunk_length);
572 header->format = le2hs (header->format);
573 header->channels = le2hs (header->channels);
574 header->sample_rate = le2hl (header->sample_rate);
575 header->bytes_per_second = le2hl (header->bytes_per_second);
576 header->sample_size = le2hs (header->sample_size);
577 header->precision = le2hs (header->precision);
578 header->chunk_data = le2hl (header->chunk_data);
579 header->data_length = le2hl (header->data_length);
581 /* Set up the interface functions for WAV. */
582 s->type = RIFF;
583 s->play = wav_play;
585 return 1;
589 /* Play RIFF-WAVE audio file S on sound device SD. */
591 static void
592 wav_play (s, sd)
593 struct sound *s;
594 struct sound_device *sd;
596 struct wav_header *header = (struct wav_header *) s->header;
598 /* Let the device choose a suitable device-dependent format
599 for the file. */
600 sd->choose_format (sd, s);
602 /* Configure the device. */
603 sd->sample_size = header->sample_size;
604 sd->sample_rate = header->sample_rate;
605 sd->bps = header->bytes_per_second;
606 sd->channels = header->channels;
607 sd->configure (sd);
609 /* Copy sound data to the device. The WAV file specification is
610 actually more complex. This simple scheme worked with all WAV
611 files I found so far. If someone feels inclined to implement the
612 whole RIFF-WAVE spec, please do. */
613 if (STRINGP (s->data))
614 sd->write (sd, SDATA (s->data) + sizeof *header,
615 SBYTES (s->data) - sizeof *header);
616 else
618 char *buffer;
619 int nbytes;
620 int blksize = sd->period_size ? sd->period_size (sd) : 2048;
621 int data_left = header->data_length;
623 buffer = (char *) alloca (blksize);
624 lseek (s->fd, sizeof *header, SEEK_SET);
625 while (data_left > 0
626 && (nbytes = emacs_read (s->fd, buffer, blksize)) > 0)
628 /* Don't play possible garbage at the end of file */
629 if (data_left < nbytes) nbytes = data_left;
630 data_left -= nbytes;
631 sd->write (sd, buffer, nbytes);
634 if (nbytes < 0)
635 sound_perror ("Error reading sound file");
640 /***********************************************************************
641 Sun Audio (*.au)
642 ***********************************************************************/
644 /* Sun audio file encodings. */
646 enum au_encoding
648 AU_ENCODING_ULAW_8 = 1,
649 AU_ENCODING_8,
650 AU_ENCODING_16,
651 AU_ENCODING_24,
652 AU_ENCODING_32,
653 AU_ENCODING_IEEE32,
654 AU_ENCODING_IEEE64,
655 AU_COMPRESSED = 23,
656 AU_ENCODING_ALAW_8 = 27
660 /* Try to initialize sound file S from S->header. S->header
661 contains the first MAX_SOUND_HEADER_BYTES number of bytes from the
662 sound file. If the file is a AU-format file, set up interface
663 functions in S and convert header fields to host byte-order.
664 Value is non-zero if the file is an AU file. */
666 static int
667 au_init (s)
668 struct sound *s;
670 struct au_header *header = (struct au_header *) s->header;
672 if (s->header_size < sizeof *header
673 || bcmp (s->header, ".snd", 4) != 0)
674 return 0;
676 header->magic_number = be2hl (header->magic_number);
677 header->data_offset = be2hl (header->data_offset);
678 header->data_size = be2hl (header->data_size);
679 header->encoding = be2hl (header->encoding);
680 header->sample_rate = be2hl (header->sample_rate);
681 header->channels = be2hl (header->channels);
683 /* Set up the interface functions for AU. */
684 s->type = SUN_AUDIO;
685 s->play = au_play;
687 return 1;
691 /* Play Sun audio file S on sound device SD. */
693 static void
694 au_play (s, sd)
695 struct sound *s;
696 struct sound_device *sd;
698 struct au_header *header = (struct au_header *) s->header;
700 sd->sample_size = 0;
701 sd->sample_rate = header->sample_rate;
702 sd->bps = 0;
703 sd->channels = header->channels;
704 sd->choose_format (sd, s);
705 sd->configure (sd);
707 if (STRINGP (s->data))
708 sd->write (sd, SDATA (s->data) + header->data_offset,
709 SBYTES (s->data) - header->data_offset);
710 else
712 int blksize = sd->period_size ? sd->period_size (sd) : 2048;
713 char *buffer;
714 int nbytes;
716 /* Seek */
717 lseek (s->fd, header->data_offset, SEEK_SET);
719 /* Copy sound data to the device. */
720 buffer = (char *) alloca (blksize);
721 while ((nbytes = emacs_read (s->fd, buffer, blksize)) > 0)
722 sd->write (sd, buffer, nbytes);
724 if (nbytes < 0)
725 sound_perror ("Error reading sound file");
730 /***********************************************************************
731 Voxware Driver Interface
732 ***********************************************************************/
734 /* This driver is available on GNU/Linux, and the free BSDs. FreeBSD
735 has a compatible own driver aka Luigi's driver. */
738 /* Open device SD. If SD->file is non-null, open that device,
739 otherwise use a default device name. */
741 static void
742 vox_open (sd)
743 struct sound_device *sd;
745 char *file;
747 /* Open the sound device. Default is /dev/dsp. */
748 if (sd->file)
749 file = sd->file;
750 else
751 file = DEFAULT_SOUND_DEVICE;
753 sd->fd = emacs_open (file, O_WRONLY, 0);
754 if (sd->fd < 0)
755 sound_perror (file);
759 /* Configure device SD from parameters in it. */
761 static void
762 vox_configure (sd)
763 struct sound_device *sd;
765 int val;
767 xassert (sd->fd >= 0);
769 /* On GNU/Linux, it seems that the device driver doesn't like to be
770 interrupted by a signal. Block the ones we know to cause
771 troubles. */
772 turn_on_atimers (0);
773 #ifdef SIGIO
774 sigblock (sigmask (SIGIO));
775 #endif
777 val = sd->format;
778 if (ioctl (sd->fd, SNDCTL_DSP_SETFMT, &sd->format) < 0
779 || val != sd->format)
780 sound_perror ("Could not set sound format");
782 val = sd->channels != 1;
783 if (ioctl (sd->fd, SNDCTL_DSP_STEREO, &val) < 0
784 || val != (sd->channels != 1))
785 sound_perror ("Could not set stereo/mono");
787 /* I think bps and sampling_rate are the same, but who knows.
788 Check this. and use SND_DSP_SPEED for both. */
789 if (sd->sample_rate > 0)
791 val = sd->sample_rate;
792 if (ioctl (sd->fd, SNDCTL_DSP_SPEED, &sd->sample_rate) < 0)
793 sound_perror ("Could not set sound speed");
794 else if (val != sd->sample_rate)
795 sound_warning ("Could not set sample rate");
798 if (sd->volume > 0)
800 int volume = sd->volume & 0xff;
801 volume |= volume << 8;
802 /* This may fail if there is no mixer. Ignore the failure. */
803 ioctl (sd->fd, SOUND_MIXER_WRITE_PCM, &volume);
806 turn_on_atimers (1);
807 #ifdef SIGIO
808 sigunblock (sigmask (SIGIO));
809 #endif
813 /* Close device SD if it is open. */
815 static void
816 vox_close (sd)
817 struct sound_device *sd;
819 if (sd->fd >= 0)
821 /* On GNU/Linux, it seems that the device driver doesn't like to
822 be interrupted by a signal. Block the ones we know to cause
823 troubles. */
824 #ifdef SIGIO
825 sigblock (sigmask (SIGIO));
826 #endif
827 turn_on_atimers (0);
829 /* Flush sound data, and reset the device. */
830 ioctl (sd->fd, SNDCTL_DSP_SYNC, NULL);
832 turn_on_atimers (1);
833 #ifdef SIGIO
834 sigunblock (sigmask (SIGIO));
835 #endif
837 /* Close the device. */
838 emacs_close (sd->fd);
839 sd->fd = -1;
844 /* Choose device-dependent format for device SD from sound file S. */
846 static void
847 vox_choose_format (sd, s)
848 struct sound_device *sd;
849 struct sound *s;
851 if (s->type == RIFF)
853 struct wav_header *h = (struct wav_header *) s->header;
854 if (h->precision == 8)
855 sd->format = AFMT_U8;
856 else if (h->precision == 16)
857 sd->format = AFMT_S16_LE;
858 else
859 error ("Unsupported WAV file format");
861 else if (s->type == SUN_AUDIO)
863 struct au_header *header = (struct au_header *) s->header;
864 switch (header->encoding)
866 case AU_ENCODING_ULAW_8:
867 case AU_ENCODING_IEEE32:
868 case AU_ENCODING_IEEE64:
869 sd->format = AFMT_MU_LAW;
870 break;
872 case AU_ENCODING_8:
873 case AU_ENCODING_16:
874 case AU_ENCODING_24:
875 case AU_ENCODING_32:
876 sd->format = AFMT_S16_LE;
877 break;
879 default:
880 error ("Unsupported AU file format");
883 else
884 abort ();
888 /* Initialize device SD. Set up the interface functions in the device
889 structure. */
891 static int
892 vox_init (sd)
893 struct sound_device *sd;
895 char *file;
896 int fd;
898 /* Open the sound device. Default is /dev/dsp. */
899 if (sd->file)
900 file = sd->file;
901 else
902 file = DEFAULT_SOUND_DEVICE;
903 fd = emacs_open (file, O_WRONLY, 0);
904 if (fd >= 0)
905 emacs_close (fd);
906 else
907 return 0;
909 sd->fd = -1;
910 sd->open = vox_open;
911 sd->close = vox_close;
912 sd->configure = vox_configure;
913 sd->choose_format = vox_choose_format;
914 sd->write = vox_write;
915 sd->period_size = NULL;
917 return 1;
920 /* Write NBYTES bytes from BUFFER to device SD. */
922 static void
923 vox_write (sd, buffer, nbytes)
924 struct sound_device *sd;
925 const char *buffer;
926 int nbytes;
928 int nwritten = emacs_write (sd->fd, buffer, nbytes);
929 if (nwritten < 0)
930 sound_perror ("Error writing to sound device");
933 #ifdef HAVE_ALSA
934 /***********************************************************************
935 ALSA Driver Interface
936 ***********************************************************************/
938 /* This driver is available on GNU/Linux. */
940 static void
941 alsa_sound_perror (msg, err)
942 char *msg;
943 int err;
945 error ("%s: %s", msg, snd_strerror (err));
948 struct alsa_params
950 snd_pcm_t *handle;
951 snd_pcm_hw_params_t *hwparams;
952 snd_pcm_sw_params_t *swparams;
953 snd_pcm_uframes_t period_size;
956 /* Open device SD. If SD->file is non-null, open that device,
957 otherwise use a default device name. */
959 static void
960 alsa_open (sd)
961 struct sound_device *sd;
963 char *file;
964 struct alsa_params *p;
965 int err;
967 /* Open the sound device. Default is "default". */
968 if (sd->file)
969 file = sd->file;
970 else
971 file = DEFAULT_ALSA_SOUND_DEVICE;
973 p = xmalloc (sizeof (*p));
974 p->handle = NULL;
975 p->hwparams = NULL;
976 p->swparams = NULL;
978 sd->fd = -1;
979 sd->data = p;
982 err = snd_pcm_open (&p->handle, file, SND_PCM_STREAM_PLAYBACK, 0);
983 if (err < 0)
984 alsa_sound_perror (file, err);
987 static int
988 alsa_period_size (sd)
989 struct sound_device *sd;
991 struct alsa_params *p = (struct alsa_params *) sd->data;
992 int fact = snd_pcm_format_size (sd->format, 1) * sd->channels;
993 return p->period_size * (fact > 0 ? fact : 1);
996 static void
997 alsa_configure (sd)
998 struct sound_device *sd;
1000 int val, err, dir;
1001 unsigned uval;
1002 struct alsa_params *p = (struct alsa_params *) sd->data;
1003 snd_pcm_uframes_t buffer_size;
1005 xassert (p->handle != 0);
1007 err = snd_pcm_hw_params_malloc (&p->hwparams);
1008 if (err < 0)
1009 alsa_sound_perror ("Could not allocate hardware parameter structure", err);
1011 err = snd_pcm_sw_params_malloc (&p->swparams);
1012 if (err < 0)
1013 alsa_sound_perror ("Could not allocate software parameter structure", err);
1015 err = snd_pcm_hw_params_any (p->handle, p->hwparams);
1016 if (err < 0)
1017 alsa_sound_perror ("Could not initialize hardware parameter structure", err);
1019 err = snd_pcm_hw_params_set_access (p->handle, p->hwparams,
1020 SND_PCM_ACCESS_RW_INTERLEAVED);
1021 if (err < 0)
1022 alsa_sound_perror ("Could not set access type", err);
1024 val = sd->format;
1025 err = snd_pcm_hw_params_set_format (p->handle, p->hwparams, val);
1026 if (err < 0)
1027 alsa_sound_perror ("Could not set sound format", err);
1029 uval = sd->sample_rate;
1030 err = snd_pcm_hw_params_set_rate_near (p->handle, p->hwparams, &uval, 0);
1031 if (err < 0)
1032 alsa_sound_perror ("Could not set sample rate", err);
1034 val = sd->channels;
1035 err = snd_pcm_hw_params_set_channels (p->handle, p->hwparams, val);
1036 if (err < 0)
1037 alsa_sound_perror ("Could not set channel count", err);
1039 err = snd_pcm_hw_params (p->handle, p->hwparams);
1040 if (err < 0)
1041 alsa_sound_perror ("Could not set parameters", err);
1044 err = snd_pcm_hw_params_get_period_size (p->hwparams, &p->period_size, &dir);
1045 if (err < 0)
1046 alsa_sound_perror ("Unable to get period size for playback", err);
1048 err = snd_pcm_hw_params_get_buffer_size (p->hwparams, &buffer_size);
1049 if (err < 0)
1050 alsa_sound_perror("Unable to get buffer size for playback", err);
1052 err = snd_pcm_sw_params_current (p->handle, p->swparams);
1053 if (err < 0)
1054 alsa_sound_perror ("Unable to determine current swparams for playback",
1055 err);
1057 /* Start the transfer when the buffer is almost full */
1058 err = snd_pcm_sw_params_set_start_threshold (p->handle, p->swparams,
1059 (buffer_size / p->period_size)
1060 * p->period_size);
1061 if (err < 0)
1062 alsa_sound_perror ("Unable to set start threshold mode for playback", err);
1064 /* Allow the transfer when at least period_size samples can be processed */
1065 err = snd_pcm_sw_params_set_avail_min (p->handle, p->swparams, p->period_size);
1066 if (err < 0)
1067 alsa_sound_perror ("Unable to set avail min for playback", err);
1069 /* Align all transfers to 1 period */
1070 err = snd_pcm_sw_params_set_xfer_align (p->handle, p->swparams,
1071 p->period_size);
1072 if (err < 0)
1073 alsa_sound_perror ("Unable to set transfer align for playback", err);
1075 err = snd_pcm_sw_params (p->handle, p->swparams);
1076 if (err < 0)
1077 alsa_sound_perror ("Unable to set sw params for playback\n", err);
1079 snd_pcm_hw_params_free (p->hwparams);
1080 p->hwparams = NULL;
1081 snd_pcm_sw_params_free (p->swparams);
1082 p->swparams = NULL;
1084 err = snd_pcm_prepare (p->handle);
1085 if (err < 0)
1086 alsa_sound_perror ("Could not prepare audio interface for use", err);
1088 if (sd->volume > 0)
1090 int chn;
1091 snd_mixer_t *handle;
1092 snd_mixer_elem_t *e;
1093 char *file = sd->file ? sd->file : DEFAULT_ALSA_SOUND_DEVICE;
1095 if (snd_mixer_open (&handle, 0) >= 0)
1097 if (snd_mixer_attach (handle, file) >= 0
1098 && snd_mixer_load (handle) >= 0
1099 && snd_mixer_selem_register (handle, NULL, NULL) >= 0)
1100 for (e = snd_mixer_first_elem (handle);
1102 e = snd_mixer_elem_next (e))
1104 if (snd_mixer_selem_has_playback_volume (e))
1106 long pmin, pmax, vol;
1107 snd_mixer_selem_get_playback_volume_range (e, &pmin, &pmax);
1108 vol = pmin + (sd->volume * (pmax - pmin)) / 100;
1110 for (chn = 0; chn <= SND_MIXER_SCHN_LAST; chn++)
1111 snd_mixer_selem_set_playback_volume (e, chn, vol);
1114 snd_mixer_close(handle);
1120 /* Close device SD if it is open. */
1122 static void
1123 alsa_close (sd)
1124 struct sound_device *sd;
1126 struct alsa_params *p = (struct alsa_params *) sd->data;
1127 if (p)
1129 if (p->hwparams)
1130 snd_pcm_hw_params_free (p->hwparams);
1131 if (p->swparams)
1132 snd_pcm_sw_params_free (p->swparams);
1133 if (p->handle)
1135 snd_pcm_drain (p->handle);
1136 snd_pcm_close (p->handle);
1138 free (p);
1142 /* Choose device-dependent format for device SD from sound file S. */
1144 static void
1145 alsa_choose_format (sd, s)
1146 struct sound_device *sd;
1147 struct sound *s;
1149 struct alsa_params *p = (struct alsa_params *) sd->data;
1150 if (s->type == RIFF)
1152 struct wav_header *h = (struct wav_header *) s->header;
1153 if (h->precision == 8)
1154 sd->format = SND_PCM_FORMAT_U8;
1155 else if (h->precision == 16)
1156 sd->format = SND_PCM_FORMAT_S16_LE;
1157 else
1158 error ("Unsupported WAV file format");
1160 else if (s->type == SUN_AUDIO)
1162 struct au_header *header = (struct au_header *) s->header;
1163 switch (header->encoding)
1165 case AU_ENCODING_ULAW_8:
1166 sd->format = SND_PCM_FORMAT_MU_LAW;
1167 break;
1168 case AU_ENCODING_ALAW_8:
1169 sd->format = SND_PCM_FORMAT_A_LAW;
1170 break;
1171 case AU_ENCODING_IEEE32:
1172 sd->format = SND_PCM_FORMAT_FLOAT_BE;
1173 break;
1174 case AU_ENCODING_IEEE64:
1175 sd->format = SND_PCM_FORMAT_FLOAT64_BE;
1176 break;
1177 case AU_ENCODING_8:
1178 sd->format = SND_PCM_FORMAT_S8;
1179 break;
1180 case AU_ENCODING_16:
1181 sd->format = SND_PCM_FORMAT_S16_BE;
1182 break;
1183 case AU_ENCODING_24:
1184 sd->format = SND_PCM_FORMAT_S24_BE;
1185 break;
1186 case AU_ENCODING_32:
1187 sd->format = SND_PCM_FORMAT_S32_BE;
1188 break;
1190 default:
1191 error ("Unsupported AU file format");
1194 else
1195 abort ();
1199 /* Write NBYTES bytes from BUFFER to device SD. */
1201 static void
1202 alsa_write (sd, buffer, nbytes)
1203 struct sound_device *sd;
1204 const char *buffer;
1205 int nbytes;
1207 struct alsa_params *p = (struct alsa_params *) sd->data;
1209 /* The the third parameter to snd_pcm_writei is frames, not bytes. */
1210 int fact = snd_pcm_format_size (sd->format, 1) * sd->channels;
1211 int nwritten = 0;
1212 int err;
1214 while (nwritten < nbytes)
1216 snd_pcm_uframes_t frames = (nbytes - nwritten)/fact;
1217 if (frames == 0) break;
1219 err = snd_pcm_writei (p->handle, buffer + nwritten, frames);
1220 if (err < 0)
1222 if (err == -EPIPE)
1223 { /* under-run */
1224 err = snd_pcm_prepare (p->handle);
1225 if (err < 0)
1226 alsa_sound_perror ("Can't recover from underrun, prepare failed",
1227 err);
1229 else if (err == -ESTRPIPE)
1231 while ((err = snd_pcm_resume (p->handle)) == -EAGAIN)
1232 sleep(1); /* wait until the suspend flag is released */
1233 if (err < 0)
1235 err = snd_pcm_prepare (p->handle);
1236 if (err < 0)
1237 alsa_sound_perror ("Can't recover from suspend, "
1238 "prepare failed",
1239 err);
1242 else
1243 alsa_sound_perror ("Error writing to sound device", err);
1246 else
1247 nwritten += err * fact;
1251 static void
1252 snd_error_quiet (file, line, function, err, fmt)
1253 const char *file;
1254 int line;
1255 const char *function;
1256 int err;
1257 const char *fmt;
1261 /* Initialize device SD. Set up the interface functions in the device
1262 structure. */
1264 static int
1265 alsa_init (sd)
1266 struct sound_device *sd;
1268 char *file;
1269 snd_pcm_t *handle;
1270 int err;
1272 /* Open the sound device. Default is "default". */
1273 if (sd->file)
1274 file = sd->file;
1275 else
1276 file = DEFAULT_ALSA_SOUND_DEVICE;
1278 snd_lib_error_set_handler ((snd_lib_error_handler_t) snd_error_quiet);
1279 err = snd_pcm_open (&handle, file, SND_PCM_STREAM_PLAYBACK, 0);
1280 snd_lib_error_set_handler (NULL);
1281 if (err < 0)
1282 return 0;
1283 snd_pcm_close (handle);
1285 sd->fd = -1;
1286 sd->open = alsa_open;
1287 sd->close = alsa_close;
1288 sd->configure = alsa_configure;
1289 sd->choose_format = alsa_choose_format;
1290 sd->write = alsa_write;
1291 sd->period_size = alsa_period_size;
1293 return 1;
1296 #endif /* HAVE_ALSA */
1299 /* END: Non Windows functions */
1300 #else /* WINDOWSNT */
1302 /* BEGIN: Windows specific functions */
1304 static int
1305 do_play_sound (psz_file, ui_volume)
1306 const char *psz_file;
1307 unsigned long ui_volume;
1309 int i_result = 0;
1310 MCIERROR mci_error = 0;
1311 char sz_cmd_buf[520] = {0};
1312 char sz_ret_buf[520] = {0};
1313 MMRESULT mm_result = MMSYSERR_NOERROR;
1314 unsigned long ui_volume_org = 0;
1315 BOOL b_reset_volume = FALSE;
1317 memset (sz_cmd_buf, 0, sizeof(sz_cmd_buf));
1318 memset (sz_ret_buf, 0, sizeof(sz_ret_buf));
1319 sprintf (sz_cmd_buf,
1320 "open \"%s\" alias GNUEmacs_PlaySound_Device wait",
1321 psz_file);
1322 mci_error = mciSendString (sz_cmd_buf, sz_ret_buf, 520, NULL);
1323 if (mci_error != 0)
1325 sound_warning ("The open mciSendString command failed to open\n"
1326 "the specified sound file");
1327 i_result = (int) mci_error;
1328 return i_result;
1330 if ((ui_volume > 0) && (ui_volume != UINT_MAX))
1332 mm_result = waveOutGetVolume ((HWAVEOUT) WAVE_MAPPER, &ui_volume_org);
1333 if (mm_result == MMSYSERR_NOERROR)
1335 b_reset_volume = TRUE;
1336 mm_result = waveOutSetVolume ((HWAVEOUT) WAVE_MAPPER, ui_volume);
1337 if ( mm_result != MMSYSERR_NOERROR)
1339 sound_warning ("waveOutSetVolume failed to set the volume level\n"
1340 "of the WAVE_MAPPER device.\n"
1341 "As a result, the user selected volume level will\n"
1342 "not be used.");
1345 else
1347 sound_warning ("waveOutGetVolume failed to obtain the original\n"
1348 "volume level of the WAVE_MAPPER device.\n"
1349 "As a result, the user selected volume level will\n"
1350 "not be used.");
1353 memset (sz_cmd_buf, 0, sizeof(sz_cmd_buf));
1354 memset (sz_ret_buf, 0, sizeof(sz_ret_buf));
1355 strcpy (sz_cmd_buf, "play GNUEmacs_PlaySound_Device wait");
1356 mci_error = mciSendString (sz_cmd_buf, sz_ret_buf, 520, NULL);
1357 if (mci_error != 0)
1359 sound_warning ("The play mciSendString command failed to play the\n"
1360 "opened sound file.");
1361 i_result = (int) mci_error;
1363 memset (sz_cmd_buf, 0, sizeof(sz_cmd_buf));
1364 memset (sz_ret_buf, 0, sizeof(sz_ret_buf));
1365 strcpy (sz_cmd_buf, "close GNUEmacs_PlaySound_Device wait");
1366 mci_error = mciSendString (sz_cmd_buf, sz_ret_buf, 520, NULL);
1367 if (b_reset_volume == TRUE)
1369 mm_result=waveOutSetVolume ((HWAVEOUT) WAVE_MAPPER, ui_volume_org);
1370 if (mm_result != MMSYSERR_NOERROR)
1372 sound_warning ("waveOutSetVolume failed to reset the original volume\n"
1373 "level of the WAVE_MAPPER device.");
1376 return i_result;
1379 /* END: Windows specific functions */
1381 #endif /* WINDOWSNT */
1383 DEFUN ("play-sound-internal", Fplay_sound_internal, Splay_sound_internal, 1, 1, 0,
1384 doc: /* Play sound SOUND.
1386 Internal use only, use `play-sound' instead. */)
1387 (sound)
1388 Lisp_Object sound;
1390 Lisp_Object attrs[SOUND_ATTR_SENTINEL];
1391 int count = SPECPDL_INDEX ();
1393 #ifndef WINDOWSNT
1394 Lisp_Object file;
1395 struct gcpro gcpro1, gcpro2;
1396 Lisp_Object args[2];
1397 #else /* WINDOWSNT */
1398 int len = 0;
1399 Lisp_Object lo_file = {0};
1400 char * psz_file = NULL;
1401 unsigned long ui_volume_tmp = UINT_MAX;
1402 unsigned long ui_volume = UINT_MAX;
1403 int i_result = 0;
1404 #endif /* WINDOWSNT */
1406 /* Parse the sound specification. Give up if it is invalid. */
1407 if (!parse_sound (sound, attrs))
1408 error ("Invalid sound specification");
1410 #ifndef WINDOWSNT
1411 file = Qnil;
1412 GCPRO2 (sound, file);
1413 current_sound_device = (struct sound_device *) xmalloc (sizeof (struct sound_device));
1414 bzero (current_sound_device, sizeof (struct sound_device));
1415 current_sound = (struct sound *) xmalloc (sizeof (struct sound));
1416 bzero (current_sound, sizeof (struct sound));
1417 record_unwind_protect (sound_cleanup, Qnil);
1418 current_sound->header = (char *) alloca (MAX_SOUND_HEADER_BYTES);
1420 if (STRINGP (attrs[SOUND_FILE]))
1422 /* Open the sound file. */
1423 current_sound->fd = openp (Fcons (Vdata_directory, Qnil),
1424 attrs[SOUND_FILE], Qnil, &file, Qnil);
1425 if (current_sound->fd < 0)
1426 sound_perror ("Could not open sound file");
1428 /* Read the first bytes from the file. */
1429 current_sound->header_size
1430 = emacs_read (current_sound->fd, current_sound->header,
1431 MAX_SOUND_HEADER_BYTES);
1432 if (current_sound->header_size < 0)
1433 sound_perror ("Invalid sound file header");
1435 else
1437 current_sound->data = attrs[SOUND_DATA];
1438 current_sound->header_size = min (MAX_SOUND_HEADER_BYTES, SBYTES (current_sound->data));
1439 bcopy (SDATA (current_sound->data), current_sound->header, current_sound->header_size);
1442 /* Find out the type of sound. Give up if we can't tell. */
1443 find_sound_type (current_sound);
1445 /* Set up a device. */
1446 if (STRINGP (attrs[SOUND_DEVICE]))
1448 int len = SCHARS (attrs[SOUND_DEVICE]);
1449 current_sound_device->file = (char *) alloca (len + 1);
1450 strcpy (current_sound_device->file, SDATA (attrs[SOUND_DEVICE]));
1453 if (INTEGERP (attrs[SOUND_VOLUME]))
1454 current_sound_device->volume = XFASTINT (attrs[SOUND_VOLUME]);
1455 else if (FLOATP (attrs[SOUND_VOLUME]))
1456 current_sound_device->volume = XFLOAT_DATA (attrs[SOUND_VOLUME]) * 100;
1458 args[0] = Qplay_sound_functions;
1459 args[1] = sound;
1460 Frun_hook_with_args (2, args);
1462 #ifdef HAVE_ALSA
1463 if (!alsa_init (current_sound_device))
1464 #endif
1465 if (!vox_init (current_sound_device))
1466 error ("No usable sound device driver found");
1468 /* Open the device. */
1469 current_sound_device->open (current_sound_device);
1471 /* Play the sound. */
1472 current_sound->play (current_sound, current_sound_device);
1474 /* Clean up. */
1475 UNGCPRO;
1477 #else /* WINDOWSNT */
1479 lo_file = Fexpand_file_name (attrs[SOUND_FILE], Qnil);
1480 len = XSTRING (lo_file)->size;
1481 psz_file = (char *) alloca (len + 1);
1482 strcpy (psz_file, XSTRING (lo_file)->data);
1483 if (INTEGERP (attrs[SOUND_VOLUME]))
1485 ui_volume_tmp = XFASTINT (attrs[SOUND_VOLUME]);
1487 else if (FLOATP (attrs[SOUND_VOLUME]))
1489 ui_volume_tmp = (unsigned long) XFLOAT_DATA (attrs[SOUND_VOLUME]) * 100;
1492 Based on some experiments I have conducted, a value of 100 or less
1493 for the sound volume is much too low. You cannot even hear it.
1494 A value of UINT_MAX indicates that you wish for the sound to played
1495 at the maximum possible volume. A value of UINT_MAX/2 plays the
1496 sound at 50% maximum volume. Therefore the value passed to do_play_sound
1497 (and thus to waveOutSetVolume) must be some fraction of UINT_MAX.
1498 The following code adjusts the user specified volume level appropriately.
1500 if ((ui_volume_tmp > 0) && (ui_volume_tmp <= 100))
1502 ui_volume = ui_volume_tmp * (UINT_MAX / 100);
1504 i_result = do_play_sound (psz_file, ui_volume);
1506 #endif /* WINDOWSNT */
1508 unbind_to (count, Qnil);
1509 return Qnil;
1512 /***********************************************************************
1513 Initialization
1514 ***********************************************************************/
1516 void
1517 syms_of_sound ()
1519 QCdevice = intern (":device");
1520 staticpro (&QCdevice);
1521 QCvolume = intern (":volume");
1522 staticpro (&QCvolume);
1523 Qsound = intern ("sound");
1524 staticpro (&Qsound);
1525 Qplay_sound_functions = intern ("play-sound-functions");
1526 staticpro (&Qplay_sound_functions);
1528 defsubr (&Splay_sound_internal);
1532 void
1533 init_sound ()
1537 #endif /* HAVE_SOUND */
1539 /* arch-tag: dd850ad8-0433-4e2c-9cba-b7aeeccc0dbd
1540 (do not change this comment) */