(frame.o, keyboard.o, xdisp.o, xfaces.o): Depend on macgui.h.
[emacs.git] / src / sound.c
blob04a46b0550301b7c69590b24ed9a4fce998ac9e5
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, or (at your option)
10 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; see the file COPYING. If not, write to
19 the Free Software Foundation, Inc., 51 Franklin Street, Fifth Floor,
20 Boston, MA 02110-1301, USA. */
22 /* Written by Gerd Moellmann <gerd@gnu.org>. Tested with Luigi's
23 driver on FreeBSD 2.2.7 with a SoundBlaster 16. */
26 Modified by Ben Key <Bkey1@tampabay.rr.com> to add a partial
27 implementation of the play-sound specification for Windows.
29 Notes:
30 In the Windows implementation of play-sound-internal only the
31 :file and :volume keywords are supported. The :device keyword,
32 if present, is ignored. The :data keyword, if present, will
33 cause an error to be generated.
35 The Windows implementation of play-sound is implemented via the
36 Win32 API functions mciSendString, waveOutGetVolume, and
37 waveOutSetVolume which are exported by Winmm.dll.
40 #include <config.h>
42 #if defined HAVE_SOUND
44 /* BEGIN: Common Includes */
45 #include <fcntl.h>
46 #include <unistd.h>
47 #include <sys/types.h>
48 #include <errno.h>
49 #include "lisp.h"
50 #include "dispextern.h"
51 #include "atimer.h"
52 #include <signal.h>
53 #include "syssignal.h"
54 /* END: Common Includes */
57 /* BEGIN: Non Windows Includes */
58 #ifndef WINDOWSNT
60 #ifndef MSDOS
61 #include <sys/ioctl.h>
62 #endif
64 /* FreeBSD has machine/soundcard.h. Voxware sound driver docs mention
65 sys/soundcard.h. So, let's try whatever's there. */
67 #ifdef HAVE_MACHINE_SOUNDCARD_H
68 #include <machine/soundcard.h>
69 #endif
70 #ifdef HAVE_SYS_SOUNDCARD_H
71 #include <sys/soundcard.h>
72 #endif
73 #ifdef HAVE_SOUNDCARD_H
74 #include <soundcard.h>
75 #endif
76 #ifdef HAVE_ALSA
77 #ifdef ALSA_SUBDIR_INCLUDE
78 #include <alsa/asoundlib.h>
79 #else
80 #include <asoundlib.h>
81 #endif /* ALSA_SUBDIR_INCLUDE */
82 #endif /* HAVE_ALSA */
84 /* END: Non Windows Includes */
86 #else /* WINDOWSNT */
88 /* BEGIN: Windows Specific Includes */
89 #include <stdio.h>
90 #include <stdlib.h>
91 #include <string.h>
92 #include <limits.h>
93 #include <windows.h>
94 #include <mmsystem.h>
95 /* END: Windows Specific Includes */
97 #endif /* WINDOWSNT */
99 /* BEGIN: Common Definitions */
100 #define abs(X) ((X) < 0 ? -(X) : (X))
102 /* Symbols. */
104 extern Lisp_Object QCfile, QCdata;
105 Lisp_Object QCvolume, QCdevice;
106 Lisp_Object Qsound;
107 Lisp_Object Qplay_sound_functions;
109 /* Indices of attributes in a sound attributes vector. */
111 enum sound_attr
113 SOUND_FILE,
114 SOUND_DATA,
115 SOUND_DEVICE,
116 SOUND_VOLUME,
117 SOUND_ATTR_SENTINEL
120 static void alsa_sound_perror P_ ((char *, int)) NO_RETURN;
121 static void sound_perror P_ ((char *)) NO_RETURN;
122 static void sound_warning P_ ((char *));
123 static int parse_sound P_ ((Lisp_Object, Lisp_Object *));
125 /* END: Common Definitions */
127 /* BEGIN: Non Windows Definitions */
128 #ifndef WINDOWSNT
130 #ifndef DEFAULT_SOUND_DEVICE
131 #define DEFAULT_SOUND_DEVICE "/dev/dsp"
132 #endif
133 #ifndef DEFAULT_ALSA_SOUND_DEVICE
134 #define DEFAULT_ALSA_SOUND_DEVICE "default"
135 #endif
138 /* Structure forward declarations. */
140 struct sound;
141 struct sound_device;
143 /* The file header of RIFF-WAVE files (*.wav). Files are always in
144 little-endian byte-order. */
146 struct wav_header
148 u_int32_t magic;
149 u_int32_t length;
150 u_int32_t chunk_type;
151 u_int32_t chunk_format;
152 u_int32_t chunk_length;
153 u_int16_t format;
154 u_int16_t channels;
155 u_int32_t sample_rate;
156 u_int32_t bytes_per_second;
157 u_int16_t sample_size;
158 u_int16_t precision;
159 u_int32_t chunk_data;
160 u_int32_t data_length;
163 /* The file header of Sun adio files (*.au). Files are always in
164 big-endian byte-order. */
166 struct au_header
168 /* ASCII ".snd" */
169 u_int32_t magic_number;
171 /* Offset of data part from start of file. Minimum value is 24. */
172 u_int32_t data_offset;
174 /* Size of data part, 0xffffffff if unknown. */
175 u_int32_t data_size;
177 /* Data encoding format.
178 1 8-bit ISDN u-law
179 2 8-bit linear PCM (REF-PCM)
180 3 16-bit linear PCM
181 4 24-bit linear PCM
182 5 32-bit linear PCM
183 6 32-bit IEEE floating-point
184 7 64-bit IEEE floating-point
185 23 8-bit u-law compressed using CCITT 0.721 ADPCM voice data
186 encoding scheme. */
187 u_int32_t encoding;
189 /* Number of samples per second. */
190 u_int32_t sample_rate;
192 /* Number of interleaved channels. */
193 u_int32_t channels;
196 /* Maximum of all sound file headers sizes. */
198 #define MAX_SOUND_HEADER_BYTES \
199 max (sizeof (struct wav_header), sizeof (struct au_header))
201 /* Interface structure for sound devices. */
203 struct sound_device
205 /* The name of the device or null meaning use a default device name. */
206 char *file;
208 /* File descriptor of the device. */
209 int fd;
211 /* Device-dependent format. */
212 int format;
214 /* Volume (0..100). Zero means unspecified. */
215 int volume;
217 /* Sample size. */
218 int sample_size;
220 /* Sample rate. */
221 int sample_rate;
223 /* Bytes per second. */
224 int bps;
226 /* 1 = mono, 2 = stereo, 0 = don't set. */
227 int channels;
229 /* Open device SD. */
230 void (* open) P_ ((struct sound_device *sd));
232 /* Close device SD. */
233 void (* close) P_ ((struct sound_device *sd));
235 /* Configure SD accoring to device-dependent parameters. */
236 void (* configure) P_ ((struct sound_device *device));
238 /* Choose a device-dependent format for outputting sound S. */
239 void (* choose_format) P_ ((struct sound_device *sd,
240 struct sound *s));
242 /* Return a preferred data size in bytes to be sent to write (below)
243 each time. 2048 is used if this is NULL. */
244 int (* period_size) P_ ((struct sound_device *sd));
246 /* Write NYBTES bytes from BUFFER to device SD. */
247 void (* write) P_ ((struct sound_device *sd, const char *buffer,
248 int nbytes));
250 /* A place for devices to store additional data. */
251 void *data;
254 /* An enumerator for each supported sound file type. */
256 enum sound_type
258 RIFF,
259 SUN_AUDIO
262 /* Interface structure for sound files. */
264 struct sound
266 /* The type of the file. */
267 enum sound_type type;
269 /* File descriptor of a sound file. */
270 int fd;
272 /* Pointer to sound file header. This contains header_size bytes
273 read from the start of a sound file. */
274 char *header;
276 /* Number of bytes raed from sound file. This is always <=
277 MAX_SOUND_HEADER_BYTES. */
278 int header_size;
280 /* Sound data, if a string. */
281 Lisp_Object data;
283 /* Play sound file S on device SD. */
284 void (* play) P_ ((struct sound *s, struct sound_device *sd));
287 /* These are set during `play-sound-internal' so that sound_cleanup has
288 access to them. */
290 struct sound_device *current_sound_device;
291 struct sound *current_sound;
293 /* Function prototypes. */
295 static void vox_open P_ ((struct sound_device *));
296 static void vox_configure P_ ((struct sound_device *));
297 static void vox_close P_ ((struct sound_device *sd));
298 static void vox_choose_format P_ ((struct sound_device *, struct sound *));
299 static int vox_init P_ ((struct sound_device *));
300 static void vox_write P_ ((struct sound_device *, const char *, int));
301 static void find_sound_type P_ ((struct sound *));
302 static u_int32_t le2hl P_ ((u_int32_t));
303 static u_int16_t le2hs P_ ((u_int16_t));
304 static u_int32_t be2hl P_ ((u_int32_t));
305 static int wav_init P_ ((struct sound *));
306 static void wav_play P_ ((struct sound *, struct sound_device *));
307 static int au_init P_ ((struct sound *));
308 static void au_play P_ ((struct sound *, struct sound_device *));
310 #if 0 /* Currently not used. */
311 static u_int16_t be2hs P_ ((u_int16_t));
312 #endif
314 /* END: Non Windows Definitions */
315 #else /* WINDOWSNT */
317 /* BEGIN: Windows Specific Definitions */
318 static int do_play_sound P_ ((const char *, unsigned long));
320 END: Windows Specific Definitions */
321 #endif /* WINDOWSNT */
324 /***********************************************************************
325 General
326 ***********************************************************************/
328 /* BEGIN: Common functions */
330 /* Like perror, but signals an error. */
332 static void
333 sound_perror (msg)
334 char *msg;
336 int saved_errno = errno;
338 turn_on_atimers (1);
339 #ifdef SIGIO
340 sigunblock (sigmask (SIGIO));
341 #endif
342 if (saved_errno != 0)
343 error ("%s: %s", msg, strerror (saved_errno));
344 else
345 error ("%s", msg);
349 /* Display a warning message. */
351 static void
352 sound_warning (msg)
353 char *msg;
355 message (msg);
359 /* Parse sound specification SOUND, and fill ATTRS with what is
360 found. Value is non-zero if SOUND Is a valid sound specification.
361 A valid sound specification is a list starting with the symbol
362 `sound'. The rest of the list is a property list which may
363 contain the following key/value pairs:
365 - `:file FILE'
367 FILE is the sound file to play. If it isn't an absolute name,
368 it's searched under `data-directory'.
370 - `:data DATA'
372 DATA is a string containing sound data. Either :file or :data
373 may be present, but not both.
375 - `:device DEVICE'
377 DEVICE is the name of the device to play on, e.g. "/dev/dsp2".
378 If not specified, a default device is used.
380 - `:volume VOL'
382 VOL must be an integer in the range [0, 100], or a float in the
383 range [0, 1]. */
385 static int
386 parse_sound (sound, attrs)
387 Lisp_Object sound;
388 Lisp_Object *attrs;
390 /* SOUND must be a list starting with the symbol `sound'. */
391 if (!CONSP (sound) || !EQ (XCAR (sound), Qsound))
392 return 0;
394 sound = XCDR (sound);
395 attrs[SOUND_FILE] = Fplist_get (sound, QCfile);
396 attrs[SOUND_DATA] = Fplist_get (sound, QCdata);
397 attrs[SOUND_DEVICE] = Fplist_get (sound, QCdevice);
398 attrs[SOUND_VOLUME] = Fplist_get (sound, QCvolume);
400 #ifndef WINDOWSNT
401 /* File name or data must be specified. */
402 if (!STRINGP (attrs[SOUND_FILE])
403 && !STRINGP (attrs[SOUND_DATA]))
404 return 0;
405 #else /* WINDOWSNT */
407 Data is not supported in Windows. Therefore a
408 File name MUST be supplied.
410 if (!STRINGP (attrs[SOUND_FILE]))
412 return 0;
414 #endif /* WINDOWSNT */
416 /* Volume must be in the range 0..100 or unspecified. */
417 if (!NILP (attrs[SOUND_VOLUME]))
419 if (INTEGERP (attrs[SOUND_VOLUME]))
421 if (XINT (attrs[SOUND_VOLUME]) < 0
422 || XINT (attrs[SOUND_VOLUME]) > 100)
423 return 0;
425 else if (FLOATP (attrs[SOUND_VOLUME]))
427 if (XFLOAT_DATA (attrs[SOUND_VOLUME]) < 0
428 || XFLOAT_DATA (attrs[SOUND_VOLUME]) > 1)
429 return 0;
431 else
432 return 0;
435 #ifndef WINDOWSNT
436 /* Device must be a string or unspecified. */
437 if (!NILP (attrs[SOUND_DEVICE])
438 && !STRINGP (attrs[SOUND_DEVICE]))
439 return 0;
440 #endif /* WINDOWSNT */
442 Since device is ignored in Windows, it does not matter
443 what it is.
445 return 1;
448 /* END: Common functions */
450 /* BEGIN: Non Windows functions */
451 #ifndef WINDOWSNT
453 /* Find out the type of the sound file whose file descriptor is FD.
454 S is the sound file structure to fill in. */
456 static void
457 find_sound_type (s)
458 struct sound *s;
460 if (!wav_init (s) && !au_init (s))
461 error ("Unknown sound format");
465 /* Function installed by play-sound-internal with record_unwind_protect. */
467 static Lisp_Object
468 sound_cleanup (arg)
469 Lisp_Object arg;
471 if (current_sound_device->close)
472 current_sound_device->close (current_sound_device);
473 if (current_sound->fd > 0)
474 emacs_close (current_sound->fd);
475 free (current_sound_device);
476 free (current_sound);
478 return Qnil;
481 /***********************************************************************
482 Byte-order Conversion
483 ***********************************************************************/
485 /* Convert 32-bit value VALUE which is in little-endian byte-order
486 to host byte-order. */
488 static u_int32_t
489 le2hl (value)
490 u_int32_t value;
492 #ifdef WORDS_BIG_ENDIAN
493 unsigned char *p = (unsigned char *) &value;
494 value = p[0] + (p[1] << 8) + (p[2] << 16) + (p[3] << 24);
495 #endif
496 return value;
500 /* Convert 16-bit value VALUE which is in little-endian byte-order
501 to host byte-order. */
503 static u_int16_t
504 le2hs (value)
505 u_int16_t value;
507 #ifdef WORDS_BIG_ENDIAN
508 unsigned char *p = (unsigned char *) &value;
509 value = p[0] + (p[1] << 8);
510 #endif
511 return value;
515 /* Convert 32-bit value VALUE which is in big-endian byte-order
516 to host byte-order. */
518 static u_int32_t
519 be2hl (value)
520 u_int32_t value;
522 #ifndef WORDS_BIG_ENDIAN
523 unsigned char *p = (unsigned char *) &value;
524 value = p[3] + (p[2] << 8) + (p[1] << 16) + (p[0] << 24);
525 #endif
526 return value;
530 #if 0 /* Currently not used. */
532 /* Convert 16-bit value VALUE which is in big-endian byte-order
533 to host byte-order. */
535 static u_int16_t
536 be2hs (value)
537 u_int16_t value;
539 #ifndef WORDS_BIG_ENDIAN
540 unsigned char *p = (unsigned char *) &value;
541 value = p[1] + (p[0] << 8);
542 #endif
543 return value;
546 #endif /* 0 */
548 /***********************************************************************
549 RIFF-WAVE (*.wav)
550 ***********************************************************************/
552 /* Try to initialize sound file S from S->header. S->header
553 contains the first MAX_SOUND_HEADER_BYTES number of bytes from the
554 sound file. If the file is a WAV-format file, set up interface
555 functions in S and convert header fields to host byte-order.
556 Value is non-zero if the file is a WAV file. */
558 static int
559 wav_init (s)
560 struct sound *s;
562 struct wav_header *header = (struct wav_header *) s->header;
564 if (s->header_size < sizeof *header
565 || bcmp (s->header, "RIFF", 4) != 0)
566 return 0;
568 /* WAV files are in little-endian order. Convert the header
569 if on a big-endian machine. */
570 header->magic = le2hl (header->magic);
571 header->length = le2hl (header->length);
572 header->chunk_type = le2hl (header->chunk_type);
573 header->chunk_format = le2hl (header->chunk_format);
574 header->chunk_length = le2hl (header->chunk_length);
575 header->format = le2hs (header->format);
576 header->channels = le2hs (header->channels);
577 header->sample_rate = le2hl (header->sample_rate);
578 header->bytes_per_second = le2hl (header->bytes_per_second);
579 header->sample_size = le2hs (header->sample_size);
580 header->precision = le2hs (header->precision);
581 header->chunk_data = le2hl (header->chunk_data);
582 header->data_length = le2hl (header->data_length);
584 /* Set up the interface functions for WAV. */
585 s->type = RIFF;
586 s->play = wav_play;
588 return 1;
592 /* Play RIFF-WAVE audio file S on sound device SD. */
594 static void
595 wav_play (s, sd)
596 struct sound *s;
597 struct sound_device *sd;
599 struct wav_header *header = (struct wav_header *) s->header;
601 /* Let the device choose a suitable device-dependent format
602 for the file. */
603 sd->choose_format (sd, s);
605 /* Configure the device. */
606 sd->sample_size = header->sample_size;
607 sd->sample_rate = header->sample_rate;
608 sd->bps = header->bytes_per_second;
609 sd->channels = header->channels;
610 sd->configure (sd);
612 /* Copy sound data to the device. The WAV file specification is
613 actually more complex. This simple scheme worked with all WAV
614 files I found so far. If someone feels inclined to implement the
615 whole RIFF-WAVE spec, please do. */
616 if (STRINGP (s->data))
617 sd->write (sd, SDATA (s->data) + sizeof *header,
618 SBYTES (s->data) - sizeof *header);
619 else
621 char *buffer;
622 int nbytes;
623 int blksize = sd->period_size ? sd->period_size (sd) : 2048;
624 int data_left = header->data_length;
626 buffer = (char *) alloca (blksize);
627 lseek (s->fd, sizeof *header, SEEK_SET);
628 while (data_left > 0
629 && (nbytes = emacs_read (s->fd, buffer, blksize)) > 0)
631 /* Don't play possible garbage at the end of file */
632 if (data_left < nbytes) nbytes = data_left;
633 data_left -= nbytes;
634 sd->write (sd, buffer, nbytes);
637 if (nbytes < 0)
638 sound_perror ("Error reading sound file");
643 /***********************************************************************
644 Sun Audio (*.au)
645 ***********************************************************************/
647 /* Sun audio file encodings. */
649 enum au_encoding
651 AU_ENCODING_ULAW_8 = 1,
652 AU_ENCODING_8,
653 AU_ENCODING_16,
654 AU_ENCODING_24,
655 AU_ENCODING_32,
656 AU_ENCODING_IEEE32,
657 AU_ENCODING_IEEE64,
658 AU_COMPRESSED = 23,
659 AU_ENCODING_ALAW_8 = 27
663 /* Try to initialize sound file S from S->header. S->header
664 contains the first MAX_SOUND_HEADER_BYTES number of bytes from the
665 sound file. If the file is a AU-format file, set up interface
666 functions in S and convert header fields to host byte-order.
667 Value is non-zero if the file is an AU file. */
669 static int
670 au_init (s)
671 struct sound *s;
673 struct au_header *header = (struct au_header *) s->header;
675 if (s->header_size < sizeof *header
676 || bcmp (s->header, ".snd", 4) != 0)
677 return 0;
679 header->magic_number = be2hl (header->magic_number);
680 header->data_offset = be2hl (header->data_offset);
681 header->data_size = be2hl (header->data_size);
682 header->encoding = be2hl (header->encoding);
683 header->sample_rate = be2hl (header->sample_rate);
684 header->channels = be2hl (header->channels);
686 /* Set up the interface functions for AU. */
687 s->type = SUN_AUDIO;
688 s->play = au_play;
690 return 1;
694 /* Play Sun audio file S on sound device SD. */
696 static void
697 au_play (s, sd)
698 struct sound *s;
699 struct sound_device *sd;
701 struct au_header *header = (struct au_header *) s->header;
703 sd->sample_size = 0;
704 sd->sample_rate = header->sample_rate;
705 sd->bps = 0;
706 sd->channels = header->channels;
707 sd->choose_format (sd, s);
708 sd->configure (sd);
710 if (STRINGP (s->data))
711 sd->write (sd, SDATA (s->data) + header->data_offset,
712 SBYTES (s->data) - header->data_offset);
713 else
715 int blksize = sd->period_size ? sd->period_size (sd) : 2048;
716 char *buffer;
717 int nbytes;
719 /* Seek */
720 lseek (s->fd, header->data_offset, SEEK_SET);
722 /* Copy sound data to the device. */
723 buffer = (char *) alloca (blksize);
724 while ((nbytes = emacs_read (s->fd, buffer, blksize)) > 0)
725 sd->write (sd, buffer, nbytes);
727 if (nbytes < 0)
728 sound_perror ("Error reading sound file");
733 /***********************************************************************
734 Voxware Driver Interface
735 ***********************************************************************/
737 /* This driver is available on GNU/Linux, and the free BSDs. FreeBSD
738 has a compatible own driver aka Luigi's driver. */
741 /* Open device SD. If SD->file is non-null, open that device,
742 otherwise use a default device name. */
744 static void
745 vox_open (sd)
746 struct sound_device *sd;
748 char *file;
750 /* Open the sound device. Default is /dev/dsp. */
751 if (sd->file)
752 file = sd->file;
753 else
754 file = DEFAULT_SOUND_DEVICE;
756 sd->fd = emacs_open (file, O_WRONLY, 0);
757 if (sd->fd < 0)
758 sound_perror (file);
762 /* Configure device SD from parameters in it. */
764 static void
765 vox_configure (sd)
766 struct sound_device *sd;
768 int val;
770 xassert (sd->fd >= 0);
772 /* On GNU/Linux, it seems that the device driver doesn't like to be
773 interrupted by a signal. Block the ones we know to cause
774 troubles. */
775 turn_on_atimers (0);
776 #ifdef SIGIO
777 sigblock (sigmask (SIGIO));
778 #endif
780 val = sd->format;
781 if (ioctl (sd->fd, SNDCTL_DSP_SETFMT, &sd->format) < 0
782 || val != sd->format)
783 sound_perror ("Could not set sound format");
785 val = sd->channels != 1;
786 if (ioctl (sd->fd, SNDCTL_DSP_STEREO, &val) < 0
787 || val != (sd->channels != 1))
788 sound_perror ("Could not set stereo/mono");
790 /* I think bps and sampling_rate are the same, but who knows.
791 Check this. and use SND_DSP_SPEED for both. */
792 if (sd->sample_rate > 0)
794 val = sd->sample_rate;
795 if (ioctl (sd->fd, SNDCTL_DSP_SPEED, &sd->sample_rate) < 0)
796 sound_perror ("Could not set sound speed");
797 else if (val != sd->sample_rate)
798 sound_warning ("Could not set sample rate");
801 if (sd->volume > 0)
803 int volume = sd->volume & 0xff;
804 volume |= volume << 8;
805 /* This may fail if there is no mixer. Ignore the failure. */
806 ioctl (sd->fd, SOUND_MIXER_WRITE_PCM, &volume);
809 turn_on_atimers (1);
810 #ifdef SIGIO
811 sigunblock (sigmask (SIGIO));
812 #endif
816 /* Close device SD if it is open. */
818 static void
819 vox_close (sd)
820 struct sound_device *sd;
822 if (sd->fd >= 0)
824 /* On GNU/Linux, it seems that the device driver doesn't like to
825 be interrupted by a signal. Block the ones we know to cause
826 troubles. */
827 #ifdef SIGIO
828 sigblock (sigmask (SIGIO));
829 #endif
830 turn_on_atimers (0);
832 /* Flush sound data, and reset the device. */
833 ioctl (sd->fd, SNDCTL_DSP_SYNC, NULL);
835 turn_on_atimers (1);
836 #ifdef SIGIO
837 sigunblock (sigmask (SIGIO));
838 #endif
840 /* Close the device. */
841 emacs_close (sd->fd);
842 sd->fd = -1;
847 /* Choose device-dependent format for device SD from sound file S. */
849 static void
850 vox_choose_format (sd, s)
851 struct sound_device *sd;
852 struct sound *s;
854 if (s->type == RIFF)
856 struct wav_header *h = (struct wav_header *) s->header;
857 if (h->precision == 8)
858 sd->format = AFMT_U8;
859 else if (h->precision == 16)
860 sd->format = AFMT_S16_LE;
861 else
862 error ("Unsupported WAV file format");
864 else if (s->type == SUN_AUDIO)
866 struct au_header *header = (struct au_header *) s->header;
867 switch (header->encoding)
869 case AU_ENCODING_ULAW_8:
870 case AU_ENCODING_IEEE32:
871 case AU_ENCODING_IEEE64:
872 sd->format = AFMT_MU_LAW;
873 break;
875 case AU_ENCODING_8:
876 case AU_ENCODING_16:
877 case AU_ENCODING_24:
878 case AU_ENCODING_32:
879 sd->format = AFMT_S16_LE;
880 break;
882 default:
883 error ("Unsupported AU file format");
886 else
887 abort ();
891 /* Initialize device SD. Set up the interface functions in the device
892 structure. */
894 static int
895 vox_init (sd)
896 struct sound_device *sd;
898 char *file;
899 int fd;
901 /* Open the sound device. Default is /dev/dsp. */
902 if (sd->file)
903 file = sd->file;
904 else
905 file = DEFAULT_SOUND_DEVICE;
906 fd = emacs_open (file, O_WRONLY, 0);
907 if (fd >= 0)
908 emacs_close (fd);
909 else
910 return 0;
912 sd->fd = -1;
913 sd->open = vox_open;
914 sd->close = vox_close;
915 sd->configure = vox_configure;
916 sd->choose_format = vox_choose_format;
917 sd->write = vox_write;
918 sd->period_size = NULL;
920 return 1;
923 /* Write NBYTES bytes from BUFFER to device SD. */
925 static void
926 vox_write (sd, buffer, nbytes)
927 struct sound_device *sd;
928 const char *buffer;
929 int nbytes;
931 int nwritten = emacs_write (sd->fd, buffer, nbytes);
932 if (nwritten < 0)
933 sound_perror ("Error writing to sound device");
936 #ifdef HAVE_ALSA
937 /***********************************************************************
938 ALSA Driver Interface
939 ***********************************************************************/
941 /* This driver is available on GNU/Linux. */
943 static void
944 alsa_sound_perror (msg, err)
945 char *msg;
946 int err;
948 error ("%s: %s", msg, snd_strerror (err));
951 struct alsa_params
953 snd_pcm_t *handle;
954 snd_pcm_hw_params_t *hwparams;
955 snd_pcm_sw_params_t *swparams;
956 snd_pcm_uframes_t period_size;
959 /* Open device SD. If SD->file is non-null, open that device,
960 otherwise use a default device name. */
962 static void
963 alsa_open (sd)
964 struct sound_device *sd;
966 char *file;
967 struct alsa_params *p;
968 int err;
970 /* Open the sound device. Default is "default". */
971 if (sd->file)
972 file = sd->file;
973 else
974 file = DEFAULT_ALSA_SOUND_DEVICE;
976 p = xmalloc (sizeof (*p));
977 p->handle = NULL;
978 p->hwparams = NULL;
979 p->swparams = NULL;
981 sd->fd = -1;
982 sd->data = p;
985 err = snd_pcm_open (&p->handle, file, SND_PCM_STREAM_PLAYBACK, 0);
986 if (err < 0)
987 alsa_sound_perror (file, err);
990 static int
991 alsa_period_size (sd)
992 struct sound_device *sd;
994 struct alsa_params *p = (struct alsa_params *) sd->data;
995 int fact = snd_pcm_format_size (sd->format, 1) * sd->channels;
996 return p->period_size * (fact > 0 ? fact : 1);
999 static void
1000 alsa_configure (sd)
1001 struct sound_device *sd;
1003 int val, err, dir;
1004 unsigned uval;
1005 struct alsa_params *p = (struct alsa_params *) sd->data;
1006 snd_pcm_uframes_t buffer_size;
1008 xassert (p->handle != 0);
1010 err = snd_pcm_hw_params_malloc (&p->hwparams);
1011 if (err < 0)
1012 alsa_sound_perror ("Could not allocate hardware parameter structure", err);
1014 err = snd_pcm_sw_params_malloc (&p->swparams);
1015 if (err < 0)
1016 alsa_sound_perror ("Could not allocate software parameter structure", err);
1018 err = snd_pcm_hw_params_any (p->handle, p->hwparams);
1019 if (err < 0)
1020 alsa_sound_perror ("Could not initialize hardware parameter structure", err);
1022 err = snd_pcm_hw_params_set_access (p->handle, p->hwparams,
1023 SND_PCM_ACCESS_RW_INTERLEAVED);
1024 if (err < 0)
1025 alsa_sound_perror ("Could not set access type", err);
1027 val = sd->format;
1028 err = snd_pcm_hw_params_set_format (p->handle, p->hwparams, val);
1029 if (err < 0)
1030 alsa_sound_perror ("Could not set sound format", err);
1032 uval = sd->sample_rate;
1033 err = snd_pcm_hw_params_set_rate_near (p->handle, p->hwparams, &uval, 0);
1034 if (err < 0)
1035 alsa_sound_perror ("Could not set sample rate", err);
1037 val = sd->channels;
1038 err = snd_pcm_hw_params_set_channels (p->handle, p->hwparams, val);
1039 if (err < 0)
1040 alsa_sound_perror ("Could not set channel count", err);
1042 err = snd_pcm_hw_params (p->handle, p->hwparams);
1043 if (err < 0)
1044 alsa_sound_perror ("Could not set parameters", err);
1047 err = snd_pcm_hw_params_get_period_size (p->hwparams, &p->period_size, &dir);
1048 if (err < 0)
1049 alsa_sound_perror ("Unable to get period size for playback", err);
1051 err = snd_pcm_hw_params_get_buffer_size (p->hwparams, &buffer_size);
1052 if (err < 0)
1053 alsa_sound_perror("Unable to get buffer size for playback", err);
1055 err = snd_pcm_sw_params_current (p->handle, p->swparams);
1056 if (err < 0)
1057 alsa_sound_perror ("Unable to determine current swparams for playback",
1058 err);
1060 /* Start the transfer when the buffer is almost full */
1061 err = snd_pcm_sw_params_set_start_threshold (p->handle, p->swparams,
1062 (buffer_size / p->period_size)
1063 * p->period_size);
1064 if (err < 0)
1065 alsa_sound_perror ("Unable to set start threshold mode for playback", err);
1067 /* Allow the transfer when at least period_size samples can be processed */
1068 err = snd_pcm_sw_params_set_avail_min (p->handle, p->swparams, p->period_size);
1069 if (err < 0)
1070 alsa_sound_perror ("Unable to set avail min for playback", err);
1072 /* Align all transfers to 1 period */
1073 err = snd_pcm_sw_params_set_xfer_align (p->handle, p->swparams,
1074 p->period_size);
1075 if (err < 0)
1076 alsa_sound_perror ("Unable to set transfer align for playback", err);
1078 err = snd_pcm_sw_params (p->handle, p->swparams);
1079 if (err < 0)
1080 alsa_sound_perror ("Unable to set sw params for playback\n", err);
1082 snd_pcm_hw_params_free (p->hwparams);
1083 p->hwparams = NULL;
1084 snd_pcm_sw_params_free (p->swparams);
1085 p->swparams = NULL;
1087 err = snd_pcm_prepare (p->handle);
1088 if (err < 0)
1089 alsa_sound_perror ("Could not prepare audio interface for use", err);
1091 if (sd->volume > 0)
1093 int chn;
1094 snd_mixer_t *handle;
1095 snd_mixer_elem_t *e;
1096 char *file = sd->file ? sd->file : DEFAULT_ALSA_SOUND_DEVICE;
1098 if (snd_mixer_open (&handle, 0) >= 0)
1100 if (snd_mixer_attach (handle, file) >= 0
1101 && snd_mixer_load (handle) >= 0
1102 && snd_mixer_selem_register (handle, NULL, NULL) >= 0)
1103 for (e = snd_mixer_first_elem (handle);
1105 e = snd_mixer_elem_next (e))
1107 if (snd_mixer_selem_has_playback_volume (e))
1109 long pmin, pmax;
1110 snd_mixer_selem_get_playback_volume_range (e, &pmin, &pmax);
1111 long vol = pmin + (sd->volume * (pmax - pmin)) / 100;
1113 for (chn = 0; chn <= SND_MIXER_SCHN_LAST; chn++)
1114 snd_mixer_selem_set_playback_volume (e, chn, vol);
1117 snd_mixer_close(handle);
1123 /* Close device SD if it is open. */
1125 static void
1126 alsa_close (sd)
1127 struct sound_device *sd;
1129 struct alsa_params *p = (struct alsa_params *) sd->data;
1130 if (p)
1132 if (p->hwparams)
1133 snd_pcm_hw_params_free (p->hwparams);
1134 if (p->swparams)
1135 snd_pcm_sw_params_free (p->swparams);
1136 if (p->handle)
1138 snd_pcm_drain (p->handle);
1139 snd_pcm_close (p->handle);
1141 free (p);
1145 /* Choose device-dependent format for device SD from sound file S. */
1147 static void
1148 alsa_choose_format (sd, s)
1149 struct sound_device *sd;
1150 struct sound *s;
1152 struct alsa_params *p = (struct alsa_params *) sd->data;
1153 if (s->type == RIFF)
1155 struct wav_header *h = (struct wav_header *) s->header;
1156 if (h->precision == 8)
1157 sd->format = SND_PCM_FORMAT_U8;
1158 else if (h->precision == 16)
1159 sd->format = SND_PCM_FORMAT_S16_LE;
1160 else
1161 error ("Unsupported WAV file format");
1163 else if (s->type == SUN_AUDIO)
1165 struct au_header *header = (struct au_header *) s->header;
1166 switch (header->encoding)
1168 case AU_ENCODING_ULAW_8:
1169 sd->format = SND_PCM_FORMAT_MU_LAW;
1170 break;
1171 case AU_ENCODING_ALAW_8:
1172 sd->format = SND_PCM_FORMAT_A_LAW;
1173 break;
1174 case AU_ENCODING_IEEE32:
1175 sd->format = SND_PCM_FORMAT_FLOAT_BE;
1176 break;
1177 case AU_ENCODING_IEEE64:
1178 sd->format = SND_PCM_FORMAT_FLOAT64_BE;
1179 break;
1180 case AU_ENCODING_8:
1181 sd->format = SND_PCM_FORMAT_S8;
1182 break;
1183 case AU_ENCODING_16:
1184 sd->format = SND_PCM_FORMAT_S16_BE;
1185 break;
1186 case AU_ENCODING_24:
1187 sd->format = SND_PCM_FORMAT_S24_BE;
1188 break;
1189 case AU_ENCODING_32:
1190 sd->format = SND_PCM_FORMAT_S32_BE;
1191 break;
1193 default:
1194 error ("Unsupported AU file format");
1197 else
1198 abort ();
1202 /* Write NBYTES bytes from BUFFER to device SD. */
1204 static void
1205 alsa_write (sd, buffer, nbytes)
1206 struct sound_device *sd;
1207 const char *buffer;
1208 int nbytes;
1210 struct alsa_params *p = (struct alsa_params *) sd->data;
1212 /* The the third parameter to snd_pcm_writei is frames, not bytes. */
1213 int fact = snd_pcm_format_size (sd->format, 1) * sd->channels;
1214 int nwritten = 0;
1215 int err;
1217 while (nwritten < nbytes)
1219 snd_pcm_uframes_t frames = (nbytes - nwritten)/fact;
1220 if (frames == 0) break;
1222 err = snd_pcm_writei (p->handle, buffer + nwritten, frames);
1223 if (err < 0)
1225 if (err == -EPIPE)
1226 { /* under-run */
1227 err = snd_pcm_prepare (p->handle);
1228 if (err < 0)
1229 alsa_sound_perror ("Can't recover from underrun, prepare failed",
1230 err);
1232 else if (err == -ESTRPIPE)
1234 while ((err = snd_pcm_resume (p->handle)) == -EAGAIN)
1235 sleep(1); /* wait until the suspend flag is released */
1236 if (err < 0)
1238 err = snd_pcm_prepare (p->handle);
1239 if (err < 0)
1240 alsa_sound_perror ("Can't recover from suspend, "
1241 "prepare failed",
1242 err);
1245 else
1246 alsa_sound_perror ("Error writing to sound device", err);
1249 else
1250 nwritten += err * fact;
1254 static void
1255 snd_error_quiet (file, line, function, err, fmt)
1256 const char *file;
1257 int line;
1258 const char *function;
1259 int err;
1260 const char *fmt;
1264 /* Initialize device SD. Set up the interface functions in the device
1265 structure. */
1267 static int
1268 alsa_init (sd)
1269 struct sound_device *sd;
1271 char *file;
1272 snd_pcm_t *handle;
1273 int err;
1275 /* Open the sound device. Default is "default". */
1276 if (sd->file)
1277 file = sd->file;
1278 else
1279 file = DEFAULT_ALSA_SOUND_DEVICE;
1281 snd_lib_error_set_handler ((snd_lib_error_handler_t) snd_error_quiet);
1282 err = snd_pcm_open (&handle, file, SND_PCM_STREAM_PLAYBACK, 0);
1283 snd_lib_error_set_handler (NULL);
1284 if (err < 0)
1285 return 0;
1286 snd_pcm_close (handle);
1288 sd->fd = -1;
1289 sd->open = alsa_open;
1290 sd->close = alsa_close;
1291 sd->configure = alsa_configure;
1292 sd->choose_format = alsa_choose_format;
1293 sd->write = alsa_write;
1294 sd->period_size = alsa_period_size;
1296 return 1;
1299 #endif /* HAVE_ALSA */
1302 /* END: Non Windows functions */
1303 #else /* WINDOWSNT */
1305 /* BEGIN: Windows specific functions */
1307 static int
1308 do_play_sound (psz_file, ui_volume)
1309 const char *psz_file;
1310 unsigned long ui_volume;
1312 int i_result = 0;
1313 MCIERROR mci_error = 0;
1314 char sz_cmd_buf[520] = {0};
1315 char sz_ret_buf[520] = {0};
1316 MMRESULT mm_result = MMSYSERR_NOERROR;
1317 unsigned long ui_volume_org = 0;
1318 BOOL b_reset_volume = FALSE;
1320 memset (sz_cmd_buf, 0, sizeof(sz_cmd_buf));
1321 memset (sz_ret_buf, 0, sizeof(sz_ret_buf));
1322 sprintf (sz_cmd_buf,
1323 "open \"%s\" alias GNUEmacs_PlaySound_Device wait",
1324 psz_file);
1325 mci_error = mciSendString (sz_cmd_buf, sz_ret_buf, 520, NULL);
1326 if (mci_error != 0)
1328 sound_warning ("The open mciSendString command failed to open\n"
1329 "the specified sound file");
1330 i_result = (int) mci_error;
1331 return i_result;
1333 if ((ui_volume > 0) && (ui_volume != UINT_MAX))
1335 mm_result = waveOutGetVolume ((HWAVEOUT) WAVE_MAPPER, &ui_volume_org);
1336 if (mm_result == MMSYSERR_NOERROR)
1338 b_reset_volume = TRUE;
1339 mm_result = waveOutSetVolume ((HWAVEOUT) WAVE_MAPPER, ui_volume);
1340 if ( mm_result != MMSYSERR_NOERROR)
1342 sound_warning ("waveOutSetVolume failed to set the volume level\n"
1343 "of the WAVE_MAPPER device.\n"
1344 "As a result, the user selected volume level will\n"
1345 "not be used.");
1348 else
1350 sound_warning ("waveOutGetVolume failed to obtain the original\n"
1351 "volume level of the WAVE_MAPPER device.\n"
1352 "As a result, the user selected volume level will\n"
1353 "not be used.");
1356 memset (sz_cmd_buf, 0, sizeof(sz_cmd_buf));
1357 memset (sz_ret_buf, 0, sizeof(sz_ret_buf));
1358 strcpy (sz_cmd_buf, "play GNUEmacs_PlaySound_Device wait");
1359 mci_error = mciSendString (sz_cmd_buf, sz_ret_buf, 520, NULL);
1360 if (mci_error != 0)
1362 sound_warning ("The play mciSendString command failed to play the\n"
1363 "opened sound file.");
1364 i_result = (int) mci_error;
1366 memset (sz_cmd_buf, 0, sizeof(sz_cmd_buf));
1367 memset (sz_ret_buf, 0, sizeof(sz_ret_buf));
1368 strcpy (sz_cmd_buf, "close GNUEmacs_PlaySound_Device wait");
1369 mci_error = mciSendString (sz_cmd_buf, sz_ret_buf, 520, NULL);
1370 if (b_reset_volume == TRUE)
1372 mm_result=waveOutSetVolume ((HWAVEOUT) WAVE_MAPPER, ui_volume_org);
1373 if (mm_result != MMSYSERR_NOERROR)
1375 sound_warning ("waveOutSetVolume failed to reset the original volume\n"
1376 "level of the WAVE_MAPPER device.");
1379 return i_result;
1382 /* END: Windows specific functions */
1384 #endif /* WINDOWSNT */
1386 DEFUN ("play-sound-internal", Fplay_sound_internal, Splay_sound_internal, 1, 1, 0,
1387 doc: /* Play sound SOUND.
1389 Internal use only, use `play-sound' instead. */)
1390 (sound)
1391 Lisp_Object sound;
1393 Lisp_Object attrs[SOUND_ATTR_SENTINEL];
1394 int count = SPECPDL_INDEX ();
1396 #ifndef WINDOWSNT
1397 Lisp_Object file;
1398 struct gcpro gcpro1, gcpro2;
1399 Lisp_Object args[2];
1400 #else /* WINDOWSNT */
1401 int len = 0;
1402 Lisp_Object lo_file = {0};
1403 char * psz_file = NULL;
1404 unsigned long ui_volume_tmp = UINT_MAX;
1405 unsigned long ui_volume = UINT_MAX;
1406 int i_result = 0;
1407 #endif /* WINDOWSNT */
1409 /* Parse the sound specification. Give up if it is invalid. */
1410 if (!parse_sound (sound, attrs))
1411 error ("Invalid sound specification");
1413 #ifndef WINDOWSNT
1414 file = Qnil;
1415 GCPRO2 (sound, file);
1416 current_sound_device = (struct sound_device *) xmalloc (sizeof (struct sound_device));
1417 bzero (current_sound_device, sizeof (struct sound_device));
1418 current_sound = (struct sound *) xmalloc (sizeof (struct sound));
1419 bzero (current_sound, sizeof (struct sound));
1420 record_unwind_protect (sound_cleanup, Qnil);
1421 current_sound->header = (char *) alloca (MAX_SOUND_HEADER_BYTES);
1423 if (STRINGP (attrs[SOUND_FILE]))
1425 /* Open the sound file. */
1426 current_sound->fd = openp (Fcons (Vdata_directory, Qnil),
1427 attrs[SOUND_FILE], Qnil, &file, Qnil);
1428 if (current_sound->fd < 0)
1429 sound_perror ("Could not open sound file");
1431 /* Read the first bytes from the file. */
1432 current_sound->header_size
1433 = emacs_read (current_sound->fd, current_sound->header,
1434 MAX_SOUND_HEADER_BYTES);
1435 if (current_sound->header_size < 0)
1436 sound_perror ("Invalid sound file header");
1438 else
1440 current_sound->data = attrs[SOUND_DATA];
1441 current_sound->header_size = min (MAX_SOUND_HEADER_BYTES, SBYTES (current_sound->data));
1442 bcopy (SDATA (current_sound->data), current_sound->header, current_sound->header_size);
1445 /* Find out the type of sound. Give up if we can't tell. */
1446 find_sound_type (current_sound);
1448 /* Set up a device. */
1449 if (STRINGP (attrs[SOUND_DEVICE]))
1451 int len = SCHARS (attrs[SOUND_DEVICE]);
1452 current_sound_device->file = (char *) alloca (len + 1);
1453 strcpy (current_sound_device->file, SDATA (attrs[SOUND_DEVICE]));
1456 if (INTEGERP (attrs[SOUND_VOLUME]))
1457 current_sound_device->volume = XFASTINT (attrs[SOUND_VOLUME]);
1458 else if (FLOATP (attrs[SOUND_VOLUME]))
1459 current_sound_device->volume = XFLOAT_DATA (attrs[SOUND_VOLUME]) * 100;
1461 args[0] = Qplay_sound_functions;
1462 args[1] = sound;
1463 Frun_hook_with_args (2, args);
1465 #ifdef HAVE_ALSA
1466 if (!alsa_init (current_sound_device))
1467 #endif
1468 if (!vox_init (current_sound_device))
1469 error ("No usable sound device driver found");
1471 /* Open the device. */
1472 current_sound_device->open (current_sound_device);
1474 /* Play the sound. */
1475 current_sound->play (current_sound, current_sound_device);
1477 /* Clean up. */
1478 UNGCPRO;
1480 #else /* WINDOWSNT */
1482 lo_file = Fexpand_file_name (attrs[SOUND_FILE], Qnil);
1483 len = XSTRING (lo_file)->size;
1484 psz_file = (char *) alloca (len + 1);
1485 strcpy (psz_file, XSTRING (lo_file)->data);
1486 if (INTEGERP (attrs[SOUND_VOLUME]))
1488 ui_volume_tmp = XFASTINT (attrs[SOUND_VOLUME]);
1490 else if (FLOATP (attrs[SOUND_VOLUME]))
1492 ui_volume_tmp = (unsigned long) XFLOAT_DATA (attrs[SOUND_VOLUME]) * 100;
1495 Based on some experiments I have conducted, a value of 100 or less
1496 for the sound volume is much too low. You cannot even hear it.
1497 A value of UINT_MAX indicates that you wish for the sound to played
1498 at the maximum possible volume. A value of UINT_MAX/2 plays the
1499 sound at 50% maximum volume. Therefore the value passed to do_play_sound
1500 (and thus to waveOutSetVolume) must be some fraction of UINT_MAX.
1501 The following code adjusts the user specified volume level appropriately.
1503 if ((ui_volume_tmp > 0) && (ui_volume_tmp <= 100))
1505 ui_volume = ui_volume_tmp * (UINT_MAX / 100);
1507 i_result = do_play_sound (psz_file, ui_volume);
1509 #endif /* WINDOWSNT */
1511 unbind_to (count, Qnil);
1512 return Qnil;
1515 /***********************************************************************
1516 Initialization
1517 ***********************************************************************/
1519 void
1520 syms_of_sound ()
1522 QCdevice = intern (":device");
1523 staticpro (&QCdevice);
1524 QCvolume = intern (":volume");
1525 staticpro (&QCvolume);
1526 Qsound = intern ("sound");
1527 staticpro (&Qsound);
1528 Qplay_sound_functions = intern ("play-sound-functions");
1529 staticpro (&Qplay_sound_functions);
1531 defsubr (&Splay_sound_internal);
1535 void
1536 init_sound ()
1540 #endif /* HAVE_SOUND */
1542 /* arch-tag: dd850ad8-0433-4e2c-9cba-b7aeeccc0dbd
1543 (do not change this comment) */