(Eval During Compile): Clarify putting macros in eval-when-compile.
[emacs.git] / src / sound.c
blob7fb79e64048d20a4948ff28f273afebb078134c7
1 /* sound.c -- sound support.
2 Copyright (C) 1998, 1999, 2001, 2002, 2003, 2004,
3 2005, 2006, 2007 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 2, 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;
625 buffer = (char *) alloca (blksize);
626 lseek (s->fd, sizeof *header, SEEK_SET);
628 while ((nbytes = emacs_read (s->fd, buffer, blksize)) > 0)
629 sd->write (sd, buffer, nbytes);
631 if (nbytes < 0)
632 sound_perror ("Error reading sound file");
637 /***********************************************************************
638 Sun Audio (*.au)
639 ***********************************************************************/
641 /* Sun audio file encodings. */
643 enum au_encoding
645 AU_ENCODING_ULAW_8 = 1,
646 AU_ENCODING_8,
647 AU_ENCODING_16,
648 AU_ENCODING_24,
649 AU_ENCODING_32,
650 AU_ENCODING_IEEE32,
651 AU_ENCODING_IEEE64,
652 AU_COMPRESSED = 23,
653 AU_ENCODING_ALAW_8 = 27
657 /* Try to initialize sound file S from S->header. S->header
658 contains the first MAX_SOUND_HEADER_BYTES number of bytes from the
659 sound file. If the file is a AU-format file, set up interface
660 functions in S and convert header fields to host byte-order.
661 Value is non-zero if the file is an AU file. */
663 static int
664 au_init (s)
665 struct sound *s;
667 struct au_header *header = (struct au_header *) s->header;
669 if (s->header_size < sizeof *header
670 || bcmp (s->header, ".snd", 4) != 0)
671 return 0;
673 header->magic_number = be2hl (header->magic_number);
674 header->data_offset = be2hl (header->data_offset);
675 header->data_size = be2hl (header->data_size);
676 header->encoding = be2hl (header->encoding);
677 header->sample_rate = be2hl (header->sample_rate);
678 header->channels = be2hl (header->channels);
680 /* Set up the interface functions for AU. */
681 s->type = SUN_AUDIO;
682 s->play = au_play;
684 return 1;
688 /* Play Sun audio file S on sound device SD. */
690 static void
691 au_play (s, sd)
692 struct sound *s;
693 struct sound_device *sd;
695 struct au_header *header = (struct au_header *) s->header;
697 sd->sample_size = 0;
698 sd->sample_rate = header->sample_rate;
699 sd->bps = 0;
700 sd->channels = header->channels;
701 sd->choose_format (sd, s);
702 sd->configure (sd);
704 if (STRINGP (s->data))
705 sd->write (sd, SDATA (s->data) + header->data_offset,
706 SBYTES (s->data) - header->data_offset);
707 else
709 int blksize = sd->period_size ? sd->period_size (sd) : 2048;
710 char *buffer;
711 int nbytes;
713 /* Seek */
714 lseek (s->fd, header->data_offset, SEEK_SET);
716 /* Copy sound data to the device. */
717 buffer = (char *) alloca (blksize);
718 while ((nbytes = emacs_read (s->fd, buffer, blksize)) > 0)
719 sd->write (sd, buffer, nbytes);
721 if (nbytes < 0)
722 sound_perror ("Error reading sound file");
727 /***********************************************************************
728 Voxware Driver Interface
729 ***********************************************************************/
731 /* This driver is available on GNU/Linux, and the free BSDs. FreeBSD
732 has a compatible own driver aka Luigi's driver. */
735 /* Open device SD. If SD->file is non-null, open that device,
736 otherwise use a default device name. */
738 static void
739 vox_open (sd)
740 struct sound_device *sd;
742 char *file;
744 /* Open the sound device. Default is /dev/dsp. */
745 if (sd->file)
746 file = sd->file;
747 else
748 file = DEFAULT_SOUND_DEVICE;
750 sd->fd = emacs_open (file, O_WRONLY, 0);
751 if (sd->fd < 0)
752 sound_perror (file);
756 /* Configure device SD from parameters in it. */
758 static void
759 vox_configure (sd)
760 struct sound_device *sd;
762 int val;
764 xassert (sd->fd >= 0);
766 /* On GNU/Linux, it seems that the device driver doesn't like to be
767 interrupted by a signal. Block the ones we know to cause
768 troubles. */
769 turn_on_atimers (0);
770 #ifdef SIGIO
771 sigblock (sigmask (SIGIO));
772 #endif
774 val = sd->format;
775 if (ioctl (sd->fd, SNDCTL_DSP_SETFMT, &sd->format) < 0
776 || val != sd->format)
777 sound_perror ("Could not set sound format");
779 val = sd->channels != 1;
780 if (ioctl (sd->fd, SNDCTL_DSP_STEREO, &val) < 0
781 || val != (sd->channels != 1))
782 sound_perror ("Could not set stereo/mono");
784 /* I think bps and sampling_rate are the same, but who knows.
785 Check this. and use SND_DSP_SPEED for both. */
786 if (sd->sample_rate > 0)
788 val = sd->sample_rate;
789 if (ioctl (sd->fd, SNDCTL_DSP_SPEED, &sd->sample_rate) < 0)
790 sound_perror ("Could not set sound speed");
791 else if (val != sd->sample_rate)
792 sound_warning ("Could not set sample rate");
795 if (sd->volume > 0)
797 int volume = sd->volume & 0xff;
798 volume |= volume << 8;
799 /* This may fail if there is no mixer. Ignore the failure. */
800 ioctl (sd->fd, SOUND_MIXER_WRITE_PCM, &volume);
803 turn_on_atimers (1);
804 #ifdef SIGIO
805 sigunblock (sigmask (SIGIO));
806 #endif
810 /* Close device SD if it is open. */
812 static void
813 vox_close (sd)
814 struct sound_device *sd;
816 if (sd->fd >= 0)
818 /* On GNU/Linux, it seems that the device driver doesn't like to
819 be interrupted by a signal. Block the ones we know to cause
820 troubles. */
821 #ifdef SIGIO
822 sigblock (sigmask (SIGIO));
823 #endif
824 turn_on_atimers (0);
826 /* Flush sound data, and reset the device. */
827 ioctl (sd->fd, SNDCTL_DSP_SYNC, NULL);
829 turn_on_atimers (1);
830 #ifdef SIGIO
831 sigunblock (sigmask (SIGIO));
832 #endif
834 /* Close the device. */
835 emacs_close (sd->fd);
836 sd->fd = -1;
841 /* Choose device-dependent format for device SD from sound file S. */
843 static void
844 vox_choose_format (sd, s)
845 struct sound_device *sd;
846 struct sound *s;
848 if (s->type == RIFF)
850 struct wav_header *h = (struct wav_header *) s->header;
851 if (h->precision == 8)
852 sd->format = AFMT_U8;
853 else if (h->precision == 16)
854 sd->format = AFMT_S16_LE;
855 else
856 error ("Unsupported WAV file format");
858 else if (s->type == SUN_AUDIO)
860 struct au_header *header = (struct au_header *) s->header;
861 switch (header->encoding)
863 case AU_ENCODING_ULAW_8:
864 case AU_ENCODING_IEEE32:
865 case AU_ENCODING_IEEE64:
866 sd->format = AFMT_MU_LAW;
867 break;
869 case AU_ENCODING_8:
870 case AU_ENCODING_16:
871 case AU_ENCODING_24:
872 case AU_ENCODING_32:
873 sd->format = AFMT_S16_LE;
874 break;
876 default:
877 error ("Unsupported AU file format");
880 else
881 abort ();
885 /* Initialize device SD. Set up the interface functions in the device
886 structure. */
888 static int
889 vox_init (sd)
890 struct sound_device *sd;
892 char *file;
893 int fd;
895 /* Open the sound device. Default is /dev/dsp. */
896 if (sd->file)
897 file = sd->file;
898 else
899 file = DEFAULT_SOUND_DEVICE;
900 fd = emacs_open (file, O_WRONLY, 0);
901 if (fd >= 0)
902 emacs_close (fd);
903 else
904 return 0;
906 sd->fd = -1;
907 sd->open = vox_open;
908 sd->close = vox_close;
909 sd->configure = vox_configure;
910 sd->choose_format = vox_choose_format;
911 sd->write = vox_write;
912 sd->period_size = NULL;
914 return 1;
917 /* Write NBYTES bytes from BUFFER to device SD. */
919 static void
920 vox_write (sd, buffer, nbytes)
921 struct sound_device *sd;
922 const char *buffer;
923 int nbytes;
925 int nwritten = emacs_write (sd->fd, buffer, nbytes);
926 if (nwritten < 0)
927 sound_perror ("Error writing to sound device");
930 #ifdef HAVE_ALSA
931 /***********************************************************************
932 ALSA Driver Interface
933 ***********************************************************************/
935 /* This driver is available on GNU/Linux. */
937 static void
938 alsa_sound_perror (msg, err)
939 char *msg;
940 int err;
942 error ("%s: %s", msg, snd_strerror (err));
945 struct alsa_params
947 snd_pcm_t *handle;
948 snd_pcm_hw_params_t *hwparams;
949 snd_pcm_sw_params_t *swparams;
950 snd_pcm_uframes_t period_size;
953 /* Open device SD. If SD->file is non-null, open that device,
954 otherwise use a default device name. */
956 static void
957 alsa_open (sd)
958 struct sound_device *sd;
960 char *file;
961 struct alsa_params *p;
962 int err;
964 /* Open the sound device. Default is "default". */
965 if (sd->file)
966 file = sd->file;
967 else
968 file = DEFAULT_ALSA_SOUND_DEVICE;
970 p = xmalloc (sizeof (*p));
971 p->handle = NULL;
972 p->hwparams = NULL;
973 p->swparams = NULL;
975 sd->fd = -1;
976 sd->data = p;
979 err = snd_pcm_open (&p->handle, file, SND_PCM_STREAM_PLAYBACK, 0);
980 if (err < 0)
981 alsa_sound_perror (file, err);
984 static int
985 alsa_period_size (sd)
986 struct sound_device *sd;
988 struct alsa_params *p = (struct alsa_params *) sd->data;
989 return p->period_size;
992 static void
993 alsa_configure (sd)
994 struct sound_device *sd;
996 int val, err, dir;
997 unsigned uval;
998 struct alsa_params *p = (struct alsa_params *) sd->data;
999 snd_pcm_uframes_t buffer_size;
1001 xassert (p->handle != 0);
1003 err = snd_pcm_hw_params_malloc (&p->hwparams);
1004 if (err < 0)
1005 alsa_sound_perror ("Could not allocate hardware parameter structure", err);
1007 err = snd_pcm_sw_params_malloc (&p->swparams);
1008 if (err < 0)
1009 alsa_sound_perror ("Could not allocate software parameter structure", err);
1011 err = snd_pcm_hw_params_any (p->handle, p->hwparams);
1012 if (err < 0)
1013 alsa_sound_perror ("Could not initialize hardware parameter structure", err);
1015 err = snd_pcm_hw_params_set_access (p->handle, p->hwparams,
1016 SND_PCM_ACCESS_RW_INTERLEAVED);
1017 if (err < 0)
1018 alsa_sound_perror ("Could not set access type", err);
1020 val = sd->format;
1021 err = snd_pcm_hw_params_set_format (p->handle, p->hwparams, val);
1022 if (err < 0)
1023 alsa_sound_perror ("Could not set sound format", err);
1025 uval = sd->sample_rate;
1026 err = snd_pcm_hw_params_set_rate_near (p->handle, p->hwparams, &uval, 0);
1027 if (err < 0)
1028 alsa_sound_perror ("Could not set sample rate", err);
1030 val = sd->channels;
1031 err = snd_pcm_hw_params_set_channels (p->handle, p->hwparams, val);
1032 if (err < 0)
1033 alsa_sound_perror ("Could not set channel count", err);
1035 err = snd_pcm_hw_params (p->handle, p->hwparams);
1036 if (err < 0)
1037 alsa_sound_perror ("Could not set parameters", err);
1040 err = snd_pcm_hw_params_get_period_size (p->hwparams, &p->period_size, &dir);
1041 if (err < 0)
1042 alsa_sound_perror ("Unable to get period size for playback", err);
1044 err = snd_pcm_hw_params_get_buffer_size (p->hwparams, &buffer_size);
1045 if (err < 0)
1046 alsa_sound_perror("Unable to get buffer size for playback", err);
1048 err = snd_pcm_sw_params_current (p->handle, p->swparams);
1049 if (err < 0)
1050 alsa_sound_perror ("Unable to determine current swparams for playback",
1051 err);
1053 /* Start the transfer when the buffer is almost full */
1054 err = snd_pcm_sw_params_set_start_threshold (p->handle, p->swparams,
1055 (buffer_size / p->period_size)
1056 * p->period_size);
1057 if (err < 0)
1058 alsa_sound_perror ("Unable to set start threshold mode for playback", err);
1060 /* Allow the transfer when at least period_size samples can be processed */
1061 err = snd_pcm_sw_params_set_avail_min (p->handle, p->swparams, p->period_size);
1062 if (err < 0)
1063 alsa_sound_perror ("Unable to set avail min for playback", err);
1065 /* Align all transfers to 1 period */
1066 err = snd_pcm_sw_params_set_xfer_align (p->handle, p->swparams,
1067 p->period_size);
1068 if (err < 0)
1069 alsa_sound_perror ("Unable to set transfer align for playback", err);
1071 err = snd_pcm_sw_params (p->handle, p->swparams);
1072 if (err < 0)
1073 alsa_sound_perror ("Unable to set sw params for playback\n", err);
1075 snd_pcm_hw_params_free (p->hwparams);
1076 p->hwparams = NULL;
1077 snd_pcm_sw_params_free (p->swparams);
1078 p->swparams = NULL;
1080 err = snd_pcm_prepare (p->handle);
1081 if (err < 0)
1082 alsa_sound_perror ("Could not prepare audio interface for use", err);
1084 if (sd->volume > 0)
1086 int chn;
1087 snd_mixer_t *handle;
1088 snd_mixer_elem_t *e;
1089 char *file = sd->file ? sd->file : DEFAULT_ALSA_SOUND_DEVICE;
1091 if (snd_mixer_open (&handle, 0) >= 0)
1093 if (snd_mixer_attach (handle, file) >= 0
1094 && snd_mixer_load (handle) >= 0
1095 && snd_mixer_selem_register (handle, NULL, NULL) >= 0)
1096 for (e = snd_mixer_first_elem (handle);
1098 e = snd_mixer_elem_next (e))
1100 if (snd_mixer_selem_has_playback_volume (e))
1102 long pmin, pmax;
1103 snd_mixer_selem_get_playback_volume_range (e, &pmin, &pmax);
1104 long vol = pmin + (sd->volume * (pmax - pmin)) / 100;
1106 for (chn = 0; chn <= SND_MIXER_SCHN_LAST; chn++)
1107 snd_mixer_selem_set_playback_volume (e, chn, vol);
1110 snd_mixer_close(handle);
1116 /* Close device SD if it is open. */
1118 static void
1119 alsa_close (sd)
1120 struct sound_device *sd;
1122 struct alsa_params *p = (struct alsa_params *) sd->data;
1123 if (p)
1125 if (p->hwparams)
1126 snd_pcm_hw_params_free (p->hwparams);
1127 if (p->swparams)
1128 snd_pcm_sw_params_free (p->swparams);
1129 if (p->handle)
1131 snd_pcm_drain (p->handle);
1132 snd_pcm_close (p->handle);
1134 free (p);
1138 /* Choose device-dependent format for device SD from sound file S. */
1140 static void
1141 alsa_choose_format (sd, s)
1142 struct sound_device *sd;
1143 struct sound *s;
1145 struct alsa_params *p = (struct alsa_params *) sd->data;
1146 if (s->type == RIFF)
1148 struct wav_header *h = (struct wav_header *) s->header;
1149 if (h->precision == 8)
1150 sd->format = SND_PCM_FORMAT_U8;
1151 else if (h->precision == 16)
1152 sd->format = SND_PCM_FORMAT_S16_LE;
1153 else
1154 error ("Unsupported WAV file format");
1156 else if (s->type == SUN_AUDIO)
1158 struct au_header *header = (struct au_header *) s->header;
1159 switch (header->encoding)
1161 case AU_ENCODING_ULAW_8:
1162 sd->format = SND_PCM_FORMAT_MU_LAW;
1163 break;
1164 case AU_ENCODING_ALAW_8:
1165 sd->format = SND_PCM_FORMAT_A_LAW;
1166 break;
1167 case AU_ENCODING_IEEE32:
1168 sd->format = SND_PCM_FORMAT_FLOAT_BE;
1169 break;
1170 case AU_ENCODING_IEEE64:
1171 sd->format = SND_PCM_FORMAT_FLOAT64_BE;
1172 break;
1173 case AU_ENCODING_8:
1174 sd->format = SND_PCM_FORMAT_S8;
1175 break;
1176 case AU_ENCODING_16:
1177 sd->format = SND_PCM_FORMAT_S16_BE;
1178 break;
1179 case AU_ENCODING_24:
1180 sd->format = SND_PCM_FORMAT_S24_BE;
1181 break;
1182 case AU_ENCODING_32:
1183 sd->format = SND_PCM_FORMAT_S32_BE;
1184 break;
1186 default:
1187 error ("Unsupported AU file format");
1190 else
1191 abort ();
1195 /* Write NBYTES bytes from BUFFER to device SD. */
1197 static void
1198 alsa_write (sd, buffer, nbytes)
1199 struct sound_device *sd;
1200 const char *buffer;
1201 int nbytes;
1203 struct alsa_params *p = (struct alsa_params *) sd->data;
1205 /* The the third parameter to snd_pcm_writei is frames, not bytes. */
1206 int fact = snd_pcm_format_size (sd->format, 1) * sd->channels;
1207 int nwritten = 0;
1208 int err;
1210 while (nwritten < nbytes)
1212 err = snd_pcm_writei (p->handle,
1213 buffer + nwritten,
1214 (nbytes - nwritten)/fact);
1215 if (err < 0)
1217 if (err == -EPIPE)
1218 { /* under-run */
1219 err = snd_pcm_prepare (p->handle);
1220 if (err < 0)
1221 alsa_sound_perror ("Can't recover from underrun, prepare failed",
1222 err);
1224 else if (err == -ESTRPIPE)
1226 while ((err = snd_pcm_resume (p->handle)) == -EAGAIN)
1227 sleep(1); /* wait until the suspend flag is released */
1228 if (err < 0)
1230 err = snd_pcm_prepare (p->handle);
1231 if (err < 0)
1232 alsa_sound_perror ("Can't recover from suspend, "
1233 "prepare failed",
1234 err);
1237 else
1238 alsa_sound_perror ("Error writing to sound device", err);
1241 else
1242 nwritten += err * fact;
1246 static void
1247 snd_error_quiet (file, line, function, err, fmt)
1248 const char *file;
1249 int line;
1250 const char *function;
1251 int err;
1252 const char *fmt;
1256 /* Initialize device SD. Set up the interface functions in the device
1257 structure. */
1259 static int
1260 alsa_init (sd)
1261 struct sound_device *sd;
1263 char *file;
1264 snd_pcm_t *handle;
1265 int err;
1267 /* Open the sound device. Default is "default". */
1268 if (sd->file)
1269 file = sd->file;
1270 else
1271 file = DEFAULT_ALSA_SOUND_DEVICE;
1273 snd_lib_error_set_handler ((snd_lib_error_handler_t) snd_error_quiet);
1274 err = snd_pcm_open (&handle, file, SND_PCM_STREAM_PLAYBACK, 0);
1275 snd_lib_error_set_handler (NULL);
1276 if (err < 0)
1277 return 0;
1278 snd_pcm_close (handle);
1280 sd->fd = -1;
1281 sd->open = alsa_open;
1282 sd->close = alsa_close;
1283 sd->configure = alsa_configure;
1284 sd->choose_format = alsa_choose_format;
1285 sd->write = alsa_write;
1286 sd->period_size = alsa_period_size;
1288 return 1;
1291 #endif /* HAVE_ALSA */
1294 /* END: Non Windows functions */
1295 #else /* WINDOWSNT */
1297 /* BEGIN: Windows specific functions */
1299 static int
1300 do_play_sound (psz_file, ui_volume)
1301 const char *psz_file;
1302 unsigned long ui_volume;
1304 int i_result = 0;
1305 MCIERROR mci_error = 0;
1306 char sz_cmd_buf[520] = {0};
1307 char sz_ret_buf[520] = {0};
1308 MMRESULT mm_result = MMSYSERR_NOERROR;
1309 unsigned long ui_volume_org = 0;
1310 BOOL b_reset_volume = FALSE;
1312 memset (sz_cmd_buf, 0, sizeof(sz_cmd_buf));
1313 memset (sz_ret_buf, 0, sizeof(sz_ret_buf));
1314 sprintf (sz_cmd_buf,
1315 "open \"%s\" alias GNUEmacs_PlaySound_Device wait",
1316 psz_file);
1317 mci_error = mciSendString (sz_cmd_buf, sz_ret_buf, 520, NULL);
1318 if (mci_error != 0)
1320 sound_warning ("The open mciSendString command failed to open\n"
1321 "the specified sound file");
1322 i_result = (int) mci_error;
1323 return i_result;
1325 if ((ui_volume > 0) && (ui_volume != UINT_MAX))
1327 mm_result = waveOutGetVolume ((HWAVEOUT) WAVE_MAPPER, &ui_volume_org);
1328 if (mm_result == MMSYSERR_NOERROR)
1330 b_reset_volume = TRUE;
1331 mm_result = waveOutSetVolume ((HWAVEOUT) WAVE_MAPPER, ui_volume);
1332 if ( mm_result != MMSYSERR_NOERROR)
1334 sound_warning ("waveOutSetVolume failed to set the volume level\n"
1335 "of the WAVE_MAPPER device.\n"
1336 "As a result, the user selected volume level will\n"
1337 "not be used.");
1340 else
1342 sound_warning ("waveOutGetVolume failed to obtain the original\n"
1343 "volume level of the WAVE_MAPPER device.\n"
1344 "As a result, the user selected volume level will\n"
1345 "not be used.");
1348 memset (sz_cmd_buf, 0, sizeof(sz_cmd_buf));
1349 memset (sz_ret_buf, 0, sizeof(sz_ret_buf));
1350 strcpy (sz_cmd_buf, "play GNUEmacs_PlaySound_Device wait");
1351 mci_error = mciSendString (sz_cmd_buf, sz_ret_buf, 520, NULL);
1352 if (mci_error != 0)
1354 sound_warning ("The play mciSendString command failed to play the\n"
1355 "opened sound file.");
1356 i_result = (int) mci_error;
1358 memset (sz_cmd_buf, 0, sizeof(sz_cmd_buf));
1359 memset (sz_ret_buf, 0, sizeof(sz_ret_buf));
1360 strcpy (sz_cmd_buf, "close GNUEmacs_PlaySound_Device wait");
1361 mci_error = mciSendString (sz_cmd_buf, sz_ret_buf, 520, NULL);
1362 if (b_reset_volume == TRUE)
1364 mm_result=waveOutSetVolume ((HWAVEOUT) WAVE_MAPPER, ui_volume_org);
1365 if (mm_result != MMSYSERR_NOERROR)
1367 sound_warning ("waveOutSetVolume failed to reset the original volume\n"
1368 "level of the WAVE_MAPPER device.");
1371 return i_result;
1374 /* END: Windows specific functions */
1376 #endif /* WINDOWSNT */
1378 DEFUN ("play-sound-internal", Fplay_sound_internal, Splay_sound_internal, 1, 1, 0,
1379 doc: /* Play sound SOUND.
1381 Internal use only, use `play-sound' instead. */)
1382 (sound)
1383 Lisp_Object sound;
1385 Lisp_Object attrs[SOUND_ATTR_SENTINEL];
1386 int count = SPECPDL_INDEX ();
1388 #ifndef WINDOWSNT
1389 Lisp_Object file;
1390 struct gcpro gcpro1, gcpro2;
1391 Lisp_Object args[2];
1392 #else /* WINDOWSNT */
1393 int len = 0;
1394 Lisp_Object lo_file = {0};
1395 char * psz_file = NULL;
1396 unsigned long ui_volume_tmp = UINT_MAX;
1397 unsigned long ui_volume = UINT_MAX;
1398 int i_result = 0;
1399 #endif /* WINDOWSNT */
1401 /* Parse the sound specification. Give up if it is invalid. */
1402 if (!parse_sound (sound, attrs))
1403 error ("Invalid sound specification");
1405 #ifndef WINDOWSNT
1406 file = Qnil;
1407 GCPRO2 (sound, file);
1408 current_sound_device = (struct sound_device *) xmalloc (sizeof (struct sound_device));
1409 bzero (current_sound_device, sizeof (struct sound_device));
1410 current_sound = (struct sound *) xmalloc (sizeof (struct sound));
1411 bzero (current_sound, sizeof (struct sound));
1412 record_unwind_protect (sound_cleanup, Qnil);
1413 current_sound->header = (char *) alloca (MAX_SOUND_HEADER_BYTES);
1415 if (STRINGP (attrs[SOUND_FILE]))
1417 /* Open the sound file. */
1418 current_sound->fd = openp (Fcons (Vdata_directory, Qnil),
1419 attrs[SOUND_FILE], Qnil, &file, Qnil);
1420 if (current_sound->fd < 0)
1421 sound_perror ("Could not open sound file");
1423 /* Read the first bytes from the file. */
1424 current_sound->header_size
1425 = emacs_read (current_sound->fd, current_sound->header,
1426 MAX_SOUND_HEADER_BYTES);
1427 if (current_sound->header_size < 0)
1428 sound_perror ("Invalid sound file header");
1430 else
1432 current_sound->data = attrs[SOUND_DATA];
1433 current_sound->header_size = min (MAX_SOUND_HEADER_BYTES, SBYTES (current_sound->data));
1434 bcopy (SDATA (current_sound->data), current_sound->header, current_sound->header_size);
1437 /* Find out the type of sound. Give up if we can't tell. */
1438 find_sound_type (current_sound);
1440 /* Set up a device. */
1441 if (STRINGP (attrs[SOUND_DEVICE]))
1443 int len = SCHARS (attrs[SOUND_DEVICE]);
1444 current_sound_device->file = (char *) alloca (len + 1);
1445 strcpy (current_sound_device->file, SDATA (attrs[SOUND_DEVICE]));
1448 if (INTEGERP (attrs[SOUND_VOLUME]))
1449 current_sound_device->volume = XFASTINT (attrs[SOUND_VOLUME]);
1450 else if (FLOATP (attrs[SOUND_VOLUME]))
1451 current_sound_device->volume = XFLOAT_DATA (attrs[SOUND_VOLUME]) * 100;
1453 args[0] = Qplay_sound_functions;
1454 args[1] = sound;
1455 Frun_hook_with_args (2, args);
1457 #ifdef HAVE_ALSA
1458 if (!alsa_init (current_sound_device))
1459 #endif
1460 if (!vox_init (current_sound_device))
1461 error ("No usable sound device driver found");
1463 /* Open the device. */
1464 current_sound_device->open (current_sound_device);
1466 /* Play the sound. */
1467 current_sound->play (current_sound, current_sound_device);
1469 /* Clean up. */
1470 UNGCPRO;
1472 #else /* WINDOWSNT */
1474 lo_file = Fexpand_file_name (attrs[SOUND_FILE], Qnil);
1475 len = XSTRING (lo_file)->size;
1476 psz_file = (char *) alloca (len + 1);
1477 strcpy (psz_file, XSTRING (lo_file)->data);
1478 if (INTEGERP (attrs[SOUND_VOLUME]))
1480 ui_volume_tmp = XFASTINT (attrs[SOUND_VOLUME]);
1482 else if (FLOATP (attrs[SOUND_VOLUME]))
1484 ui_volume_tmp = (unsigned long) XFLOAT_DATA (attrs[SOUND_VOLUME]) * 100;
1487 Based on some experiments I have conducted, a value of 100 or less
1488 for the sound volume is much too low. You cannot even hear it.
1489 A value of UINT_MAX indicates that you wish for the sound to played
1490 at the maximum possible volume. A value of UINT_MAX/2 plays the
1491 sound at 50% maximum volume. Therefore the value passed to do_play_sound
1492 (and thus to waveOutSetVolume) must be some fraction of UINT_MAX.
1493 The following code adjusts the user specified volume level appropriately.
1495 if ((ui_volume_tmp > 0) && (ui_volume_tmp <= 100))
1497 ui_volume = ui_volume_tmp * (UINT_MAX / 100);
1499 i_result = do_play_sound (psz_file, ui_volume);
1501 #endif /* WINDOWSNT */
1503 unbind_to (count, Qnil);
1504 return Qnil;
1507 /***********************************************************************
1508 Initialization
1509 ***********************************************************************/
1511 void
1512 syms_of_sound ()
1514 QCdevice = intern (":device");
1515 staticpro (&QCdevice);
1516 QCvolume = intern (":volume");
1517 staticpro (&QCvolume);
1518 Qsound = intern ("sound");
1519 staticpro (&Qsound);
1520 Qplay_sound_functions = intern ("play-sound-functions");
1521 staticpro (&Qplay_sound_functions);
1523 defsubr (&Splay_sound_internal);
1527 void
1528 init_sound ()
1532 #endif /* HAVE_SOUND */
1534 /* arch-tag: dd850ad8-0433-4e2c-9cba-b7aeeccc0dbd
1535 (do not change this comment) */