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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, 2009 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 err = snd_pcm_sw_params (p->handle, p->swparams);
1070 if (err < 0)
1071 alsa_sound_perror ("Unable to set sw params for playback\n", err);
1073 snd_pcm_hw_params_free (p->hwparams);
1074 p->hwparams = NULL;
1075 snd_pcm_sw_params_free (p->swparams);
1076 p->swparams = NULL;
1078 err = snd_pcm_prepare (p->handle);
1079 if (err < 0)
1080 alsa_sound_perror ("Could not prepare audio interface for use", err);
1082 if (sd->volume > 0)
1084 int chn;
1085 snd_mixer_t *handle;
1086 snd_mixer_elem_t *e;
1087 char *file = sd->file ? sd->file : DEFAULT_ALSA_SOUND_DEVICE;
1089 if (snd_mixer_open (&handle, 0) >= 0)
1091 if (snd_mixer_attach (handle, file) >= 0
1092 && snd_mixer_load (handle) >= 0
1093 && snd_mixer_selem_register (handle, NULL, NULL) >= 0)
1094 for (e = snd_mixer_first_elem (handle);
1096 e = snd_mixer_elem_next (e))
1098 if (snd_mixer_selem_has_playback_volume (e))
1100 long pmin, pmax, vol;
1101 snd_mixer_selem_get_playback_volume_range (e, &pmin, &pmax);
1102 vol = pmin + (sd->volume * (pmax - pmin)) / 100;
1104 for (chn = 0; chn <= SND_MIXER_SCHN_LAST; chn++)
1105 snd_mixer_selem_set_playback_volume (e, chn, vol);
1108 snd_mixer_close(handle);
1114 /* Close device SD if it is open. */
1116 static void
1117 alsa_close (sd)
1118 struct sound_device *sd;
1120 struct alsa_params *p = (struct alsa_params *) sd->data;
1121 if (p)
1123 if (p->hwparams)
1124 snd_pcm_hw_params_free (p->hwparams);
1125 if (p->swparams)
1126 snd_pcm_sw_params_free (p->swparams);
1127 if (p->handle)
1129 snd_pcm_drain (p->handle);
1130 snd_pcm_close (p->handle);
1132 free (p);
1136 /* Choose device-dependent format for device SD from sound file S. */
1138 static void
1139 alsa_choose_format (sd, s)
1140 struct sound_device *sd;
1141 struct sound *s;
1143 struct alsa_params *p = (struct alsa_params *) sd->data;
1144 if (s->type == RIFF)
1146 struct wav_header *h = (struct wav_header *) s->header;
1147 if (h->precision == 8)
1148 sd->format = SND_PCM_FORMAT_U8;
1149 else if (h->precision == 16)
1150 sd->format = SND_PCM_FORMAT_S16_LE;
1151 else
1152 error ("Unsupported WAV file format");
1154 else if (s->type == SUN_AUDIO)
1156 struct au_header *header = (struct au_header *) s->header;
1157 switch (header->encoding)
1159 case AU_ENCODING_ULAW_8:
1160 sd->format = SND_PCM_FORMAT_MU_LAW;
1161 break;
1162 case AU_ENCODING_ALAW_8:
1163 sd->format = SND_PCM_FORMAT_A_LAW;
1164 break;
1165 case AU_ENCODING_IEEE32:
1166 sd->format = SND_PCM_FORMAT_FLOAT_BE;
1167 break;
1168 case AU_ENCODING_IEEE64:
1169 sd->format = SND_PCM_FORMAT_FLOAT64_BE;
1170 break;
1171 case AU_ENCODING_8:
1172 sd->format = SND_PCM_FORMAT_S8;
1173 break;
1174 case AU_ENCODING_16:
1175 sd->format = SND_PCM_FORMAT_S16_BE;
1176 break;
1177 case AU_ENCODING_24:
1178 sd->format = SND_PCM_FORMAT_S24_BE;
1179 break;
1180 case AU_ENCODING_32:
1181 sd->format = SND_PCM_FORMAT_S32_BE;
1182 break;
1184 default:
1185 error ("Unsupported AU file format");
1188 else
1189 abort ();
1193 /* Write NBYTES bytes from BUFFER to device SD. */
1195 static void
1196 alsa_write (sd, buffer, nbytes)
1197 struct sound_device *sd;
1198 const char *buffer;
1199 int nbytes;
1201 struct alsa_params *p = (struct alsa_params *) sd->data;
1203 /* The the third parameter to snd_pcm_writei is frames, not bytes. */
1204 int fact = snd_pcm_format_size (sd->format, 1) * sd->channels;
1205 int nwritten = 0;
1206 int err;
1208 while (nwritten < nbytes)
1210 snd_pcm_uframes_t frames = (nbytes - nwritten)/fact;
1211 if (frames == 0) break;
1213 err = snd_pcm_writei (p->handle, buffer + nwritten, frames);
1214 if (err < 0)
1216 if (err == -EPIPE)
1217 { /* under-run */
1218 err = snd_pcm_prepare (p->handle);
1219 if (err < 0)
1220 alsa_sound_perror ("Can't recover from underrun, prepare failed",
1221 err);
1223 else if (err == -ESTRPIPE)
1225 while ((err = snd_pcm_resume (p->handle)) == -EAGAIN)
1226 sleep(1); /* wait until the suspend flag is released */
1227 if (err < 0)
1229 err = snd_pcm_prepare (p->handle);
1230 if (err < 0)
1231 alsa_sound_perror ("Can't recover from suspend, "
1232 "prepare failed",
1233 err);
1236 else
1237 alsa_sound_perror ("Error writing to sound device", err);
1240 else
1241 nwritten += err * fact;
1245 static void
1246 snd_error_quiet (file, line, function, err, fmt)
1247 const char *file;
1248 int line;
1249 const char *function;
1250 int err;
1251 const char *fmt;
1255 /* Initialize device SD. Set up the interface functions in the device
1256 structure. */
1258 static int
1259 alsa_init (sd)
1260 struct sound_device *sd;
1262 char *file;
1263 snd_pcm_t *handle;
1264 int err;
1266 /* Open the sound device. Default is "default". */
1267 if (sd->file)
1268 file = sd->file;
1269 else
1270 file = DEFAULT_ALSA_SOUND_DEVICE;
1272 snd_lib_error_set_handler ((snd_lib_error_handler_t) snd_error_quiet);
1273 err = snd_pcm_open (&handle, file, SND_PCM_STREAM_PLAYBACK, 0);
1274 snd_lib_error_set_handler (NULL);
1275 if (err < 0)
1276 return 0;
1277 snd_pcm_close (handle);
1279 sd->fd = -1;
1280 sd->open = alsa_open;
1281 sd->close = alsa_close;
1282 sd->configure = alsa_configure;
1283 sd->choose_format = alsa_choose_format;
1284 sd->write = alsa_write;
1285 sd->period_size = alsa_period_size;
1287 return 1;
1290 #endif /* HAVE_ALSA */
1293 /* END: Non Windows functions */
1294 #else /* WINDOWSNT */
1296 /* BEGIN: Windows specific functions */
1298 #define SOUND_WARNING(fun, error, text) \
1300 char buf[1024]; \
1301 char err_string[MAXERRORLENGTH]; \
1302 fun (error, err_string, sizeof (err_string)); \
1303 _snprintf (buf, sizeof (buf), "%s\nError: %s", \
1304 text, err_string); \
1305 sound_warning (buf); \
1308 static int
1309 do_play_sound (psz_file, ui_volume)
1310 const char *psz_file;
1311 unsigned long ui_volume;
1313 int i_result = 0;
1314 MCIERROR mci_error = 0;
1315 char sz_cmd_buf[520] = {0};
1316 char sz_ret_buf[520] = {0};
1317 MMRESULT mm_result = MMSYSERR_NOERROR;
1318 unsigned long ui_volume_org = 0;
1319 BOOL b_reset_volume = FALSE;
1321 memset (sz_cmd_buf, 0, sizeof (sz_cmd_buf));
1322 memset (sz_ret_buf, 0, sizeof (sz_ret_buf));
1323 sprintf (sz_cmd_buf,
1324 "open \"%s\" alias GNUEmacs_PlaySound_Device wait",
1325 psz_file);
1326 mci_error = mciSendString (sz_cmd_buf, sz_ret_buf, sizeof (sz_ret_buf), NULL);
1327 if (mci_error != 0)
1329 SOUND_WARNING (mciGetErrorString, mci_error,
1330 "The open mciSendString command failed to open "
1331 "the specified sound file.");
1332 i_result = (int) mci_error;
1333 return i_result;
1335 if ((ui_volume > 0) && (ui_volume != UINT_MAX))
1337 mm_result = waveOutGetVolume ((HWAVEOUT) WAVE_MAPPER, &ui_volume_org);
1338 if (mm_result == MMSYSERR_NOERROR)
1340 b_reset_volume = TRUE;
1341 mm_result = waveOutSetVolume ((HWAVEOUT) WAVE_MAPPER, ui_volume);
1342 if (mm_result != MMSYSERR_NOERROR)
1344 SOUND_WARNING (waveOutGetErrorText, mm_result,
1345 "waveOutSetVolume failed to set the volume level "
1346 "of the WAVE_MAPPER device.\n"
1347 "As a result, the user selected volume level will "
1348 "not be used.");
1351 else
1353 SOUND_WARNING (waveOutGetErrorText, mm_result,
1354 "waveOutGetVolume failed to obtain the original "
1355 "volume level of the WAVE_MAPPER device.\n"
1356 "As a result, the user selected volume level will "
1357 "not be used.");
1360 memset (sz_cmd_buf, 0, sizeof (sz_cmd_buf));
1361 memset (sz_ret_buf, 0, sizeof (sz_ret_buf));
1362 strcpy (sz_cmd_buf, "play GNUEmacs_PlaySound_Device wait");
1363 mci_error = mciSendString (sz_cmd_buf, sz_ret_buf, sizeof (sz_ret_buf), NULL);
1364 if (mci_error != 0)
1366 SOUND_WARNING (mciGetErrorString, mci_error,
1367 "The play mciSendString command failed to play the "
1368 "opened sound file.");
1369 i_result = (int) mci_error;
1371 memset (sz_cmd_buf, 0, sizeof (sz_cmd_buf));
1372 memset (sz_ret_buf, 0, sizeof (sz_ret_buf));
1373 strcpy (sz_cmd_buf, "close GNUEmacs_PlaySound_Device wait");
1374 mci_error = mciSendString (sz_cmd_buf, sz_ret_buf, sizeof (sz_ret_buf), NULL);
1375 if (b_reset_volume == TRUE)
1377 mm_result = waveOutSetVolume ((HWAVEOUT) WAVE_MAPPER, ui_volume_org);
1378 if (mm_result != MMSYSERR_NOERROR)
1380 SOUND_WARNING (waveOutGetErrorText, mm_result,
1381 "waveOutSetVolume failed to reset the original volume "
1382 "level of the WAVE_MAPPER device.");
1385 return i_result;
1388 /* END: Windows specific functions */
1390 #endif /* WINDOWSNT */
1392 DEFUN ("play-sound-internal", Fplay_sound_internal, Splay_sound_internal, 1, 1, 0,
1393 doc: /* Play sound SOUND.
1395 Internal use only, use `play-sound' instead. */)
1396 (sound)
1397 Lisp_Object sound;
1399 Lisp_Object attrs[SOUND_ATTR_SENTINEL];
1400 int count = SPECPDL_INDEX ();
1402 #ifndef WINDOWSNT
1403 Lisp_Object file;
1404 struct gcpro gcpro1, gcpro2;
1405 Lisp_Object args[2];
1406 #else /* WINDOWSNT */
1407 int len = 0;
1408 Lisp_Object lo_file = {0};
1409 char * psz_file = NULL;
1410 unsigned long ui_volume_tmp = UINT_MAX;
1411 unsigned long ui_volume = UINT_MAX;
1412 int i_result = 0;
1413 #endif /* WINDOWSNT */
1415 /* Parse the sound specification. Give up if it is invalid. */
1416 if (!parse_sound (sound, attrs))
1417 error ("Invalid sound specification");
1419 #ifndef WINDOWSNT
1420 file = Qnil;
1421 GCPRO2 (sound, file);
1422 current_sound_device = (struct sound_device *) xmalloc (sizeof (struct sound_device));
1423 bzero (current_sound_device, sizeof (struct sound_device));
1424 current_sound = (struct sound *) xmalloc (sizeof (struct sound));
1425 bzero (current_sound, sizeof (struct sound));
1426 record_unwind_protect (sound_cleanup, Qnil);
1427 current_sound->header = (char *) alloca (MAX_SOUND_HEADER_BYTES);
1429 if (STRINGP (attrs[SOUND_FILE]))
1431 /* Open the sound file. */
1432 current_sound->fd = openp (Fcons (Vdata_directory, Qnil),
1433 attrs[SOUND_FILE], Qnil, &file, Qnil);
1434 if (current_sound->fd < 0)
1435 sound_perror ("Could not open sound file");
1437 /* Read the first bytes from the file. */
1438 current_sound->header_size
1439 = emacs_read (current_sound->fd, current_sound->header,
1440 MAX_SOUND_HEADER_BYTES);
1441 if (current_sound->header_size < 0)
1442 sound_perror ("Invalid sound file header");
1444 else
1446 current_sound->data = attrs[SOUND_DATA];
1447 current_sound->header_size = min (MAX_SOUND_HEADER_BYTES, SBYTES (current_sound->data));
1448 bcopy (SDATA (current_sound->data), current_sound->header, current_sound->header_size);
1451 /* Find out the type of sound. Give up if we can't tell. */
1452 find_sound_type (current_sound);
1454 /* Set up a device. */
1455 if (STRINGP (attrs[SOUND_DEVICE]))
1457 int len = SCHARS (attrs[SOUND_DEVICE]);
1458 current_sound_device->file = (char *) alloca (len + 1);
1459 strcpy (current_sound_device->file, SDATA (attrs[SOUND_DEVICE]));
1462 if (INTEGERP (attrs[SOUND_VOLUME]))
1463 current_sound_device->volume = XFASTINT (attrs[SOUND_VOLUME]);
1464 else if (FLOATP (attrs[SOUND_VOLUME]))
1465 current_sound_device->volume = XFLOAT_DATA (attrs[SOUND_VOLUME]) * 100;
1467 args[0] = Qplay_sound_functions;
1468 args[1] = sound;
1469 Frun_hook_with_args (2, args);
1471 #ifdef HAVE_ALSA
1472 if (!alsa_init (current_sound_device))
1473 #endif
1474 if (!vox_init (current_sound_device))
1475 error ("No usable sound device driver found");
1477 /* Open the device. */
1478 current_sound_device->open (current_sound_device);
1480 /* Play the sound. */
1481 current_sound->play (current_sound, current_sound_device);
1483 /* Clean up. */
1484 UNGCPRO;
1486 #else /* WINDOWSNT */
1488 lo_file = Fexpand_file_name (attrs[SOUND_FILE], Qnil);
1489 len = XSTRING (lo_file)->size;
1490 psz_file = (char *) alloca (len + 1);
1491 strcpy (psz_file, XSTRING (lo_file)->data);
1492 if (INTEGERP (attrs[SOUND_VOLUME]))
1494 ui_volume_tmp = XFASTINT (attrs[SOUND_VOLUME]);
1496 else if (FLOATP (attrs[SOUND_VOLUME]))
1498 ui_volume_tmp = (unsigned long) XFLOAT_DATA (attrs[SOUND_VOLUME]) * 100;
1501 Based on some experiments I have conducted, a value of 100 or less
1502 for the sound volume is much too low. You cannot even hear it.
1503 A value of UINT_MAX indicates that you wish for the sound to played
1504 at the maximum possible volume. A value of UINT_MAX/2 plays the
1505 sound at 50% maximum volume. Therefore the value passed to do_play_sound
1506 (and thus to waveOutSetVolume) must be some fraction of UINT_MAX.
1507 The following code adjusts the user specified volume level appropriately.
1509 if ((ui_volume_tmp > 0) && (ui_volume_tmp <= 100))
1511 ui_volume = ui_volume_tmp * (UINT_MAX / 100);
1513 i_result = do_play_sound (psz_file, ui_volume);
1515 #endif /* WINDOWSNT */
1517 unbind_to (count, Qnil);
1518 return Qnil;
1521 /***********************************************************************
1522 Initialization
1523 ***********************************************************************/
1525 void
1526 syms_of_sound ()
1528 QCdevice = intern (":device");
1529 staticpro (&QCdevice);
1530 QCvolume = intern (":volume");
1531 staticpro (&QCvolume);
1532 Qsound = intern ("sound");
1533 staticpro (&Qsound);
1534 Qplay_sound_functions = intern ("play-sound-functions");
1535 staticpro (&Qplay_sound_functions);
1537 defsubr (&Splay_sound_internal);
1541 void
1542 init_sound ()
1546 #endif /* HAVE_SOUND */
1548 /* arch-tag: dd850ad8-0433-4e2c-9cba-b7aeeccc0dbd
1549 (do not change this comment) */