path: root/src/sound.c

/* sound.c -- sound support.
   Copyright (C) 1998, 1999, 2001, 2002, 2003, 2004,
                 2005, 2006, 2007, 2008, 2009 Free Software Foundation, Inc.

This file is part of GNU Emacs.

GNU Emacs is free software: you can redistribute it and/or modify
it under the terms of the GNU General Public License as published by
the Free Software Foundation, either version 3 of the License, or
(at your option) any later version.

GNU Emacs is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
GNU General Public License for more details.

You should have received a copy of the GNU General Public License
along with GNU Emacs.  If not, see <http://www.gnu.org/licenses/>.  */

/* Written by Gerd Moellmann <gerd@gnu.org>.  Tested with Luigi's
   driver on FreeBSD 2.2.7 with a SoundBlaster 16.  */

/*
  Modified by Ben Key <Bkey1@tampabay.rr.com> to add a partial
  implementation of the play-sound specification for Windows.

  Notes:
  In the Windows implementation of play-sound-internal only the
  :file and :volume keywords are supported.  The :device keyword,
  if present, is ignored.  The :data keyword, if present, will
  cause an error to be generated.

  The Windows implementation of play-sound is implemented via the
  Win32 API functions mciSendString, waveOutGetVolume, and
  waveOutSetVolume which are exported by Winmm.dll.
*/

#include <config.h>

#if defined HAVE_SOUND

/* BEGIN: Common Includes */
#include <fcntl.h>
#include <unistd.h>
#include <sys/types.h>
#include <errno.h>
#include "lisp.h"
#include "dispextern.h"
#include "atimer.h"
#include <signal.h>
#include "syssignal.h"
/* END: Common Includes */


/* BEGIN: Non Windows Includes */
#ifndef WINDOWSNT

#ifndef MSDOS
#include <sys/ioctl.h>
#endif

/* FreeBSD has machine/soundcard.h.  Voxware sound driver docs mention
   sys/soundcard.h.  So, let's try whatever's there.  */

#ifdef HAVE_MACHINE_SOUNDCARD_H
#include <machine/soundcard.h>
#endif
#ifdef HAVE_SYS_SOUNDCARD_H
#include <sys/soundcard.h>
#endif
#ifdef HAVE_SOUNDCARD_H
#include <soundcard.h>
#endif
#ifdef HAVE_ALSA
#ifdef ALSA_SUBDIR_INCLUDE
#include <alsa/asoundlib.h>
#else
#include <asoundlib.h>
#endif /* ALSA_SUBDIR_INCLUDE */
#endif /* HAVE_ALSA */

/* END: Non Windows Includes */

#else /* WINDOWSNT */

/* BEGIN: Windows Specific Includes */
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <limits.h>
#include <windows.h>
#include <mmsystem.h>
/* END: Windows Specific Includes */

#endif /* WINDOWSNT */

/* BEGIN: Common Definitions */

/* Symbols.  */

extern Lisp_Object QCfile, QCdata;
Lisp_Object QCvolume, QCdevice;
Lisp_Object Qsound;
Lisp_Object Qplay_sound_functions;

/* Indices of attributes in a sound attributes vector.  */

enum sound_attr
{
  SOUND_FILE,
  SOUND_DATA,
  SOUND_DEVICE,
  SOUND_VOLUME,
  SOUND_ATTR_SENTINEL
};

static void alsa_sound_perror P_ ((char *, int)) NO_RETURN;
static void sound_perror P_ ((char *)) NO_RETURN;
static void sound_warning P_ ((char *));
static int parse_sound P_ ((Lisp_Object, Lisp_Object *));

/* END: Common Definitions */

/* BEGIN: Non Windows Definitions */
#ifndef WINDOWSNT

#ifndef DEFAULT_SOUND_DEVICE
#define DEFAULT_SOUND_DEVICE "/dev/dsp"
#endif
#ifndef DEFAULT_ALSA_SOUND_DEVICE
#define DEFAULT_ALSA_SOUND_DEVICE "default"
#endif


/* Structure forward declarations.  */

struct sound;
struct sound_device;

/* The file header of RIFF-WAVE files (*.wav).  Files are always in
   little-endian byte-order.  */

struct wav_header
{
  u_int32_t magic;
  u_int32_t length;
  u_int32_t chunk_type;
  u_int32_t chunk_format;
  u_int32_t chunk_length;
  u_int16_t format;
  u_int16_t channels;
  u_int32_t sample_rate;
  u_int32_t bytes_per_second;
  u_int16_t sample_size;
  u_int16_t precision;
  u_int32_t chunk_data;
  u_int32_t data_length;
};

/* The file header of Sun adio files (*.au).  Files are always in
   big-endian byte-order.  */

struct au_header
{
  /* ASCII ".snd" */
  u_int32_t magic_number;

  /* Offset of data part from start of file. Minimum value is 24.  */
  u_int32_t data_offset;

  /* Size of data part, 0xffffffff if unknown.  */
  u_int32_t data_size;

  /* Data encoding format.
     1	8-bit ISDN u-law
     2  8-bit linear PCM (REF-PCM)
     3  16-bit linear PCM
     4	24-bit linear PCM
     5	32-bit linear PCM
     6	32-bit IEEE floating-point
     7	64-bit IEEE floating-point
     23 8-bit u-law compressed using CCITT 0.721 ADPCM voice data
     encoding scheme.  */
  u_int32_t encoding;

  /* Number of samples per second.  */
  u_int32_t sample_rate;

  /* Number of interleaved channels.  */
  u_int32_t channels;
};

/* Maximum of all sound file headers sizes.  */

#define MAX_SOUND_HEADER_BYTES \
     max (sizeof (struct wav_header), sizeof (struct au_header))

/* Interface structure for sound devices.  */

struct sound_device
{
  /* The name of the device or null meaning use a default device name.  */
  char *file;

  /* File descriptor of the device.  */
  int fd;

  /* Device-dependent format.  */
  int format;

  /* Volume (0..100).  Zero means unspecified.  */
  int volume;

  /* Sample size.  */
  int sample_size;

  /* Sample rate.  */
  int sample_rate;

  /* Bytes per second.  */
  int bps;

  /* 1 = mono, 2 = stereo, 0 = don't set.  */
  int channels;

  /* Open device SD.  */
  void (* open) P_ ((struct sound_device *sd));

  /* Close device SD.  */
  void (* close) P_ ((struct sound_device *sd));

  /* Configure SD accoring to device-dependent parameters.  */
  void (* configure) P_ ((struct sound_device *device));

  /* Choose a device-dependent format for outputting sound S.  */
  void (* choose_format) P_ ((struct sound_device *sd,
			      struct sound *s));

  /* Return a preferred data size in bytes to be sent to write (below)
     each time.  2048 is used if this is NULL.  */
  int (* period_size) P_ ((struct sound_device *sd));

  /* Write NYBTES bytes from BUFFER to device SD.  */
  void (* write) P_ ((struct sound_device *sd, const char *buffer,
		      int nbytes));

  /* A place for devices to store additional data.  */
  void *data;
};

/* An enumerator for each supported sound file type.  */

enum sound_type
{
  RIFF,
  SUN_AUDIO
};

/* Interface structure for sound files.  */

struct sound
{
  /* The type of the file.  */
  enum sound_type type;

  /* File descriptor of a sound file.  */
  int fd;

  /* Pointer to sound file header.  This contains header_size bytes
     read from the start of a sound file.  */
  char *header;

  /* Number of bytes raed from sound file.  This is always <=
     MAX_SOUND_HEADER_BYTES.  */
  int header_size;

  /* Sound data, if a string.  */
  Lisp_Object data;

  /* Play sound file S on device SD.  */
  void (* play) P_ ((struct sound *s, struct sound_device *sd));
};

/* These are set during `play-sound-internal' so that sound_cleanup has
   access to them.  */

struct sound_device *current_sound_device;
struct sound *current_sound;

/* Function prototypes.  */

static void vox_open P_ ((struct sound_device *));
static void vox_configure P_ ((struct sound_device *));
static void vox_close P_ ((struct sound_device *sd));
static void vox_choose_format P_ ((struct sound_device *, struct sound *));
static int vox_init P_ ((struct sound_device *));
static void vox_write P_ ((struct sound_device *, const char *, int));
static void find_sound_type P_ ((struct sound *));
static u_int32_t le2hl P_ ((u_int32_t));
static u_int16_t le2hs P_ ((u_int16_t));
static u_int32_t be2hl P_ ((u_int32_t));
static int wav_init P_ ((struct sound *));
static void wav_play P_ ((struct sound *, struct sound_device *));
static int au_init P_ ((struct sound *));
static void au_play P_ ((struct sound *, struct sound_device *));

#if 0 /* Currently not used.  */
static u_int16_t be2hs P_ ((u_int16_t));
#endif

/* END: Non Windows Definitions */
#else /* WINDOWSNT */

/* BEGIN: Windows Specific Definitions */
static int do_play_sound P_ ((const char *, unsigned long));
/*
  END: Windows Specific Definitions */
#endif /* WINDOWSNT */


/***********************************************************************
			       General
 ***********************************************************************/

/* BEGIN: Common functions */

/* Like perror, but signals an error.  */

static void
sound_perror (msg)
     char *msg;
{
  int saved_errno = errno;

  turn_on_atimers (1);
#ifdef SIGIO
  sigunblock (sigmask (SIGIO));
#endif
  if (saved_errno != 0)
    error ("%s: %s", msg, strerror (saved_errno));
  else
    error ("%s", msg);
}


/* Display a warning message.  */

static void
sound_warning (msg)
     char *msg;
{
  message (msg);
}


/* Parse sound specification SOUND, and fill ATTRS with what is
   found.  Value is non-zero if SOUND Is a valid sound specification.
   A valid sound specification is a list starting with the symbol
   `sound'.  The rest of the list is a property list which may
   contain the following key/value pairs:

   - `:file FILE'

   FILE is the sound file to play.  If it isn't an absolute name,
   it's searched under `data-directory'.

   - `:data DATA'

   DATA is a string containing sound data.  Either :file or :data
   may be present, but not both.

   - `:device DEVICE'

   DEVICE is the name of the device to play on, e.g. "/dev/dsp2".
   If not specified, a default device is used.

   - `:volume VOL'

   VOL must be an integer in the range [0, 100], or a float in the
   range [0, 1].  */

static int
parse_sound (sound, attrs)
     Lisp_Object sound;
     Lisp_Object *attrs;
{
  /* SOUND must be a list starting with the symbol `sound'.  */
  if (!CONSP (sound) || !EQ (XCAR (sound), Qsound))
    return 0;

  sound = XCDR (sound);
  attrs[SOUND_FILE] = Fplist_get (sound, QCfile);
  attrs[SOUND_DATA] = Fplist_get (sound, QCdata);
  attrs[SOUND_DEVICE] = Fplist_get (sound, QCdevice);
  attrs[SOUND_VOLUME] = Fplist_get (sound, QCvolume);

#ifndef WINDOWSNT
  /* File name or data must be specified.  */
  if (!STRINGP (attrs[SOUND_FILE])
      && !STRINGP (attrs[SOUND_DATA]))
    return 0;
#else /* WINDOWSNT */
  /*
    Data is not supported in Windows.  Therefore a
    File name MUST be supplied.
  */
  if (!STRINGP (attrs[SOUND_FILE]))
    {
      return 0;
    }
#endif /* WINDOWSNT */

  /* Volume must be in the range 0..100 or unspecified.  */
  if (!NILP (attrs[SOUND_VOLUME]))
    {
      if (INTEGERP (attrs[SOUND_VOLUME]))
	{
	  if (XINT (attrs[SOUND_VOLUME]) < 0
	      || XINT (attrs[SOUND_VOLUME]) > 100)
	    return 0;
	}
      else if (FLOATP (attrs[SOUND_VOLUME]))
	{
	  if (XFLOAT_DATA (attrs[SOUND_VOLUME]) < 0
	      || XFLOAT_DATA (attrs[SOUND_VOLUME]) > 1)
	    return 0;
	}
      else
	return 0;
    }

#ifndef WINDOWSNT
  /* Device must be a string or unspecified.  */
  if (!NILP (attrs[SOUND_DEVICE])
      && !STRINGP (attrs[SOUND_DEVICE]))
    return 0;
#endif  /* WINDOWSNT */
  /*
    Since device is ignored in Windows, it does not matter
    what it is.
   */
  return 1;
}

/* END: Common functions */

/* BEGIN: Non Windows functions */
#ifndef WINDOWSNT

/* Find out the type of the sound file whose file descriptor is FD.
   S is the sound file structure to fill in.  */

static void
find_sound_type (s)
     struct sound *s;
{
  if (!wav_init (s) && !au_init (s))
    error ("Unknown sound format");
}


/* Function installed by play-sound-internal with record_unwind_protect.  */

static Lisp_Object
sound_cleanup (arg)
     Lisp_Object arg;
{
  if (current_sound_device->close)
    current_sound_device->close (current_sound_device);
  if (current_sound->fd > 0)
    emacs_close (current_sound->fd);
  free (current_sound_device);
  free (current_sound);

  return Qnil;
}

/***********************************************************************
			Byte-order Conversion
 ***********************************************************************/

/* Convert 32-bit value VALUE which is in little-endian byte-order
   to host byte-order.  */

static u_int32_t
le2hl (value)
     u_int32_t value;
{
#ifdef WORDS_BIG_ENDIAN
  unsigned char *p = (unsigned char *) &value;
  value = p[0] + (p[1] << 8) + (p[2] << 16) + (p[3] << 24);
#endif
  return value;
}


/* Convert 16-bit value VALUE which is in little-endian byte-order
   to host byte-order.  */

static u_int16_t
le2hs (value)
     u_int16_t value;
{
#ifdef WORDS_BIG_ENDIAN
  unsigned char *p = (unsigned char *) &value;
  value = p[0] + (p[1] << 8);
#endif
  return value;
}


/* Convert 32-bit value VALUE which is in big-endian byte-order
   to host byte-order.  */

static u_int32_t
be2hl (value)
     u_int32_t value;
{
#ifndef WORDS_BIG_ENDIAN
  unsigned char *p = (unsigned char *) &value;
  value = p[3] + (p[2] << 8) + (p[1] << 16) + (p[0] << 24);
#endif
  return value;
}


#if 0 /* Currently not used.  */

/* Convert 16-bit value VALUE which is in big-endian byte-order
   to host byte-order.  */

static u_int16_t
be2hs (value)
     u_int16_t value;
{
#ifndef WORDS_BIG_ENDIAN
  unsigned char *p = (unsigned char *) &value;
  value = p[1] + (p[0] << 8);
#endif
  return value;
}

#endif /* 0 */

/***********************************************************************
			  RIFF-WAVE (*.wav)
 ***********************************************************************/

/* Try to initialize sound file S from S->header.  S->header
   contains the first MAX_SOUND_HEADER_BYTES number of bytes from the
   sound file.  If the file is a WAV-format file, set up interface
   functions in S and convert header fields to host byte-order.
   Value is non-zero if the file is a WAV file.  */

static int
wav_init (s)
     struct sound *s;
{
  struct wav_header *header = (struct wav_header *) s->header;

  if (s->header_size < sizeof *header
      || bcmp (s->header, "RIFF", 4) != 0)
    return 0;

  /* WAV files are in little-endian order.  Convert the header
     if on a big-endian machine.  */
  header->magic = le2hl (header->magic);
  header->length = le2hl (header->length);
  header->chunk_type = le2hl (header->chunk_type);
  header->chunk_format = le2hl (header->chunk_format);
  header->chunk_length = le2hl (header->chunk_length);
  header->format = le2hs (header->format);
  header->channels = le2hs (header->channels);
  header->sample_rate = le2hl (header->sample_rate);
  header->bytes_per_second = le2hl (header->bytes_per_second);
  header->sample_size = le2hs (header->sample_size);
  header->precision = le2hs (header->precision);
  header->chunk_data = le2hl (header->chunk_data);
  header->data_length = le2hl (header->data_length);

  /* Set up the interface functions for WAV.  */
  s->type = RIFF;
  s->play = wav_play;

  return 1;
}


/* Play RIFF-WAVE audio file S on sound device SD.  */

static void
wav_play (s, sd)
     struct sound *s;
     struct sound_device *sd;
{
  struct wav_header *header = (struct wav_header *) s->header;

  /* Let the device choose a suitable device-dependent format
     for the file.  */
  sd->choose_format (sd, s);

  /* Configure the device.  */
  sd->sample_size = header->sample_size;
  sd->sample_rate = header->sample_rate;
  sd->bps = header->bytes_per_second;
  sd->channels = header->channels;
  sd->configure (sd);

  /* Copy sound data to the device.  The WAV file specification is
     actually more complex.  This simple scheme worked with all WAV
     files I found so far.  If someone feels inclined to implement the
     whole RIFF-WAVE spec, please do.  */
  if (STRINGP (s->data))
    sd->write (sd, SDATA (s->data) + sizeof *header,
	       SBYTES (s->data) - sizeof *header);
  else
    {
      char *buffer;
      int nbytes;
      int blksize = sd->period_size ? sd->period_size (sd) : 2048;
      int data_left = header->data_length;

      buffer = (char *) alloca (blksize);
      lseek (s->fd, sizeof *header, SEEK_SET);
      while (data_left > 0
             && (nbytes = emacs_read (s->fd, buffer, blksize)) > 0)
        {
          /* Don't play possible garbage at the end of file */
          if (data_left < nbytes) nbytes = data_left;
          data_left -= nbytes;
          sd->write (sd, buffer, nbytes);
        }

      if (nbytes < 0)
	sound_perror ("Error reading sound file");
    }
}


/***********************************************************************
			   Sun Audio (*.au)
 ***********************************************************************/

/* Sun audio file encodings.  */

enum au_encoding
{
  AU_ENCODING_ULAW_8 = 1,
  AU_ENCODING_8,
  AU_ENCODING_16,
  AU_ENCODING_24,
  AU_ENCODING_32,
  AU_ENCODING_IEEE32,
  AU_ENCODING_IEEE64,
  AU_COMPRESSED = 23,
  AU_ENCODING_ALAW_8 = 27
};


/* Try to initialize sound file S from S->header.  S->header
   contains the first MAX_SOUND_HEADER_BYTES number of bytes from the
   sound file.  If the file is a AU-format file, set up interface
   functions in S and convert header fields to host byte-order.
   Value is non-zero if the file is an AU file.  */

static int
au_init (s)
     struct sound *s;
{
  struct au_header *header = (struct au_header *) s->header;

  if (s->header_size < sizeof *header
      || bcmp (s->header, ".snd", 4) != 0)
    return 0;

  header->magic_number = be2hl (header->magic_number);
  header->data_offset = be2hl (header->data_offset);
  header->data_size = be2hl (header->data_size);
  header->encoding = be2hl (header->encoding);
  header->sample_rate = be2hl (header->sample_rate);
  header->channels = be2hl (header->channels);

  /* Set up the interface functions for AU.  */
  s->type = SUN_AUDIO;
  s->play = au_play;

  return 1;
}


/* Play Sun audio file S on sound device SD.  */

static void
au_play (s, sd)
     struct sound *s;
     struct sound_device *sd;
{
  struct au_header *header = (struct au_header *) s->header;

  sd->sample_size = 0;
  sd->sample_rate = header->sample_rate;
  sd->bps = 0;
  sd->channels = header->channels;
  sd->choose_format (sd, s);
  sd->configure (sd);

  if (STRINGP (s->data))
    sd->write (sd, SDATA (s->data) + header->data_offset,
	       SBYTES (s->data) - header->data_offset);
  else
    {
      int blksize = sd->period_size ? sd->period_size (sd) : 2048;
      char *buffer;
      int nbytes;

      /* Seek */
      lseek (s->fd, header->data_offset, SEEK_SET);

      /* Copy sound data to the device.  */
      buffer = (char *) alloca (blksize);
      while ((nbytes = emacs_read (s->fd, buffer, blksize)) > 0)
	sd->write (sd, buffer, nbytes);

      if (nbytes < 0)
	sound_perror ("Error reading sound file");
    }
}


/***********************************************************************
		       Voxware Driver Interface
 ***********************************************************************/

/* This driver is available on GNU/Linux, and the free BSDs.  FreeBSD
   has a compatible own driver aka Luigi's driver.  */


/* Open device SD.  If SD->file is non-null, open that device,
   otherwise use a default device name.  */

static void
vox_open (sd)
     struct sound_device *sd;
{
  char *file;

  /* Open the sound device.  Default is /dev/dsp.  */
  if (sd->file)
    file = sd->file;
  else
    file = DEFAULT_SOUND_DEVICE;

  sd->fd = emacs_open (file, O_WRONLY, 0);
  if (sd->fd < 0)
    sound_perror (file);
}


/* Configure device SD from parameters in it.  */

static void
vox_configure (sd)
     struct sound_device *sd;
{
  int val;

  xassert (sd->fd >= 0);

  /* On GNU/Linux, it seems that the device driver doesn't like to be
     interrupted by a signal.  Block the ones we know to cause
     troubles.  */
  turn_on_atimers (0);
#ifdef SIGIO
  sigblock (sigmask (SIGIO));
#endif

  val = sd->format;
  if (ioctl (sd->fd, SNDCTL_DSP_SETFMT, &sd->format) < 0
      || val != sd->format)
    sound_perror ("Could not set sound format");

  val = sd->channels != 1;
  if (ioctl (sd->fd, SNDCTL_DSP_STEREO, &val) < 0
      || val != (sd->channels != 1))
    sound_perror ("Could not set stereo/mono");

  /* I think bps and sampling_rate are the same, but who knows.
     Check this. and use SND_DSP_SPEED for both.  */
  if (sd->sample_rate > 0)
    {
      val = sd->sample_rate;
      if (ioctl (sd->fd, SNDCTL_DSP_SPEED, &sd->sample_rate) < 0)
	sound_perror ("Could not set sound speed");
      else if (val != sd->sample_rate)
	sound_warning ("Could not set sample rate");
    }

  if (sd->volume > 0)
    {
      int volume = sd->volume & 0xff;
      volume |= volume << 8;
      /* This may fail if there is no mixer.  Ignore the failure.  */
      ioctl (sd->fd, SOUND_MIXER_WRITE_PCM, &volume);
    }

  turn_on_atimers (1);
#ifdef SIGIO
  sigunblock (sigmask (SIGIO));
#endif
}


/* Close device SD if it is open.  */

static void
vox_close (sd)
     struct sound_device *sd;
{
  if (sd->fd >= 0)
    {
      /* On GNU/Linux, it seems that the device driver doesn't like to
	 be interrupted by a signal.  Block the ones we know to cause
	 troubles.  */
#ifdef SIGIO
      sigblock (sigmask (SIGIO));
#endif
      turn_on_atimers (0);

      /* Flush sound data, and reset the device.  */
      ioctl (sd->fd, SNDCTL_DSP_SYNC, NULL);

      turn_on_atimers (1);
#ifdef SIGIO
      sigunblock (sigmask (SIGIO));
#endif

      /* Close the device.  */
      emacs_close (sd->fd);
      sd->fd = -1;
    }
}


/* Choose device-dependent format for device SD from sound file S.  */

static void
vox_choose_format (sd, s)
     struct sound_device *sd;
     struct sound *s;
{
  if (s->type == RIFF)
    {
      struct wav_header *h = (struct wav_header *) s->header;
      if (h->precision == 8)
	sd->format = AFMT_U8;
      else if (h->precision == 16)
	sd->format = AFMT_S16_LE;
      else
	error ("Unsupported WAV file format");
    }
  else if (s->type == SUN_AUDIO)
    {
      struct au_header *header = (struct au_header *) s->header;
      switch (header->encoding)
	{
	case AU_ENCODING_ULAW_8:
	case AU_ENCODING_IEEE32:
	case AU_ENCODING_IEEE64:
	  sd->format = AFMT_MU_LAW;
	  break;

	case AU_ENCODING_8:
	case AU_ENCODING_16:
	case AU_ENCODING_24:
	case AU_ENCODING_32:
	  sd->format = AFMT_S16_LE;
	  break;

	default:
	  error ("Unsupported AU file format");
	}
    }
  else
    abort ();
}


/* Initialize device SD.  Set up the interface functions in the device
   structure.  */

static int
vox_init (sd)
     struct sound_device *sd;
{
  char *file;
  int fd;

  /* Open the sound device.  Default is /dev/dsp.  */
  if (sd->file)
    file = sd->file;
  else
    file = DEFAULT_SOUND_DEVICE;
  fd = emacs_open (file, O_WRONLY, 0);
  if (fd >= 0)
    emacs_close (fd);
  else
    return 0;

  sd->fd = -1;
  sd->open = vox_open;
  sd->close = vox_close;
  sd->configure = vox_configure;
  sd->choose_format = vox_choose_format;
  sd->write = vox_write;
  sd->period_size = NULL;

  return 1;
}

/* Write NBYTES bytes from BUFFER to device SD.  */

static void
vox_write (sd, buffer, nbytes)
     struct sound_device *sd;
     const char *buffer;
     int nbytes;
{
  int nwritten = emacs_write (sd->fd, buffer, nbytes);
  if (nwritten < 0)
    sound_perror ("Error writing to sound device");
}

#ifdef HAVE_ALSA
/***********************************************************************
		       ALSA Driver Interface
 ***********************************************************************/

/* This driver is available on GNU/Linux. */

static void
alsa_sound_perror (msg, err)
     char *msg;
     int err;
{
  error ("%s: %s", msg, snd_strerror (err));
}

struct alsa_params
{
  snd_pcm_t *handle;
  snd_pcm_hw_params_t *hwparams;
  snd_pcm_sw_params_t *swparams;
  snd_pcm_uframes_t period_size;
};

/* Open device SD.  If SD->file is non-null, open that device,
   otherwise use a default device name.  */

static void
alsa_open (sd)
     struct sound_device *sd;
{
  char *file;
  struct alsa_params *p;
  int err;

  /* Open the sound device.  Default is "default".  */
  if (sd->file)
    file = sd->file;
  else
    file = DEFAULT_ALSA_SOUND_DEVICE;

  p = xmalloc (sizeof (*p));
  p->handle = NULL;
  p->hwparams = NULL;
  p->swparams = NULL;

  sd->fd = -1;
  sd->data = p;


  err = snd_pcm_open (&p->handle, file, SND_PCM_STREAM_PLAYBACK, 0);
  if (err < 0)
    alsa_sound_perror (file, err);
}

static int
alsa_period_size (sd)
       struct sound_device *sd;
{
  struct alsa_params *p = (struct alsa_params *) sd->data;
  int fact = snd_pcm_format_size (sd->format, 1) * sd->channels;
  return p->period_size * (fact > 0 ? fact : 1);
}

static void
alsa_configure (sd)
     struct sound_device *sd;
{
  int val, err, dir;
  unsigned uval;
  struct alsa_params *p = (struct alsa_params *) sd->data;
  snd_pcm_uframes_t buffer_size;

  xassert (p->handle != 0);

  err = snd_pcm_hw_params_malloc (&p->hwparams);
  if (err < 0)
    alsa_sound_perror ("Could not allocate hardware parameter structure", err);

  err = snd_pcm_sw_params_malloc (&p->swparams);
  if (err < 0)
    alsa_sound_perror ("Could not allocate software parameter structure", err);

  err = snd_pcm_hw_params_any (p->handle, p->hwparams);
  if (err < 0)
    alsa_sound_perror ("Could not initialize hardware parameter structure", err);

  err = snd_pcm_hw_params_set_access (p->handle, p->hwparams,
                                      SND_PCM_ACCESS_RW_INTERLEAVED);
  if (err < 0)
    alsa_sound_perror ("Could not set access type", err);

  val = sd->format;
  err = snd_pcm_hw_params_set_format (p->handle, p->hwparams, val);
  if (err < 0)
    alsa_sound_perror ("Could not set sound format", err);

  uval = sd->sample_rate;
  err = snd_pcm_hw_params_set_rate_near (p->handle, p->hwparams, &uval, 0);
  if (err < 0)
    alsa_sound_perror ("Could not set sample rate", err);

  val = sd->channels;
  err = snd_pcm_hw_params_set_channels (p->handle, p->hwparams, val);
  if (err < 0)
    alsa_sound_perror ("Could not set channel count", err);

  err = snd_pcm_hw_params (p->handle, p->hwparams);
  if (err < 0)
    alsa_sound_perror ("Could not set parameters", err);


  err = snd_pcm_hw_params_get_period_size (p->hwparams, &p->period_size, &dir);
  if (err < 0)
    alsa_sound_perror ("Unable to get period size for playback", err);

  err = snd_pcm_hw_params_get_buffer_size (p->hwparams, &buffer_size);
  if (err < 0)
    alsa_sound_perror("Unable to get buffer size for playback", err);

  err = snd_pcm_sw_params_current (p->handle, p->swparams);
  if (err < 0)
    alsa_sound_perror ("Unable to determine current swparams for playback",
                       err);

  /* Start the transfer when the buffer is almost full */
  err = snd_pcm_sw_params_set_start_threshold (p->handle, p->swparams,
                                               (buffer_size / p->period_size)
                                               * p->period_size);
  if (err < 0)
    alsa_sound_perror ("Unable to set start threshold mode for playback", err);

  /* Allow the transfer when at least period_size samples can be processed */
  err = snd_pcm_sw_params_set_avail_min (p->handle, p->swparams, p->period_size);
  if (err < 0)
    alsa_sound_perror ("Unable to set avail min for playback", err);

  /* Align all transfers to 1 period */
  err = snd_pcm_sw_params_set_xfer_align (p->handle, p->swparams,
                                          p->period_size);
  if (err < 0)
    alsa_sound_perror ("Unable to set transfer align for playback", err);

  err = snd_pcm_sw_params (p->handle, p->swparams);
  if (err < 0)
    alsa_sound_perror ("Unable to set sw params for playback\n", err);

  snd_pcm_hw_params_free (p->hwparams);
  p->hwparams = NULL;
  snd_pcm_sw_params_free (p->swparams);
  p->swparams = NULL;

  err = snd_pcm_prepare (p->handle);
  if (err < 0)
    alsa_sound_perror ("Could not prepare audio interface for use", err);

  if (sd->volume > 0)
    {
      int chn;
      snd_mixer_t *handle;
      snd_mixer_elem_t *e;
      char *file = sd->file ? sd->file : DEFAULT_ALSA_SOUND_DEVICE;

      if (snd_mixer_open (&handle, 0) >= 0)
        {
          if (snd_mixer_attach (handle, file) >= 0
              && snd_mixer_load (handle) >= 0
              && snd_mixer_selem_register (handle, NULL, NULL) >= 0)
            for (e = snd_mixer_first_elem (handle);
                 e;
                 e = snd_mixer_elem_next (e))
              {
                if (snd_mixer_selem_has_playback_volume (e))
                  {
                    long pmin, pmax, vol;
                    snd_mixer_selem_get_playback_volume_range (e, &pmin, &pmax);
                    vol = pmin + (sd->volume * (pmax - pmin)) / 100;

                    for (chn = 0; chn <= SND_MIXER_SCHN_LAST; chn++)
                      snd_mixer_selem_set_playback_volume (e, chn, vol);
                  }
              }
          snd_mixer_close(handle);
        }
    }
}


/* Close device SD if it is open.  */

static void
alsa_close (sd)
     struct sound_device *sd;
{
  struct alsa_params *p = (struct alsa_params *) sd->data;
  if (p)
    {
      if (p->hwparams)
        snd_pcm_hw_params_free (p->hwparams);
      if (p->swparams)
        snd_pcm_sw_params_free (p->swparams);
      if (p->handle)
        {
          snd_pcm_drain (p->handle);
          snd_pcm_close (p->handle);
        }
      free (p);
    }
}

/* Choose device-dependent format for device SD from sound file S.  */

static void
alsa_choose_format (sd, s)
     struct sound_device *sd;
     struct sound *s;
{
  struct alsa_params *p = (struct alsa_params *) sd->data;
  if (s->type == RIFF)
    {
      struct wav_header *h = (struct wav_header *) s->header;
      if (h->precision == 8)
	sd->format = SND_PCM_FORMAT_U8;
      else if (h->precision == 16)
          sd->format = SND_PCM_FORMAT_S16_LE;
      else
	error ("Unsupported WAV file format");
    }
  else if (s->type == SUN_AUDIO)
    {
      struct au_header *header = (struct au_header *) s->header;
      switch (header->encoding)
	{
	case AU_ENCODING_ULAW_8:
	  sd->format = SND_PCM_FORMAT_MU_LAW;
          break;
	case AU_ENCODING_ALAW_8:
	  sd->format = SND_PCM_FORMAT_A_LAW;
          break;
	case AU_ENCODING_IEEE32:
          sd->format = SND_PCM_FORMAT_FLOAT_BE;
          break;
	case AU_ENCODING_IEEE64:
	  sd->format = SND_PCM_FORMAT_FLOAT64_BE;
	  break;
	case AU_ENCODING_8:
	  sd->format = SND_PCM_FORMAT_S8;
	  break;
	case AU_ENCODING_16:
	  sd->format = SND_PCM_FORMAT_S16_BE;
	  break;
	case AU_ENCODING_24:
	  sd->format = SND_PCM_FORMAT_S24_BE;
	  break;
	case AU_ENCODING_32:
	  sd->format = SND_PCM_FORMAT_S32_BE;
	  break;

	default:
	  error ("Unsupported AU file format");
	}
    }
  else
    abort ();
}


/* Write NBYTES bytes from BUFFER to device SD.  */

static void
alsa_write (sd, buffer, nbytes)
     struct sound_device *sd;
     const char *buffer;
     int nbytes;
{
  struct alsa_params *p = (struct alsa_params *) sd->data;

  /* The the third parameter to snd_pcm_writei is frames, not bytes. */
  int fact = snd_pcm_format_size (sd->format, 1) * sd->channels;
  int nwritten = 0;
  int err;

  while (nwritten < nbytes)
    {
      snd_pcm_uframes_t frames = (nbytes - nwritten)/fact;
      if (frames == 0) break;
      
      err = snd_pcm_writei (p->handle, buffer + nwritten, frames);
      if (err < 0)
        {
          if (err == -EPIPE)
            {	/* under-run */
              err = snd_pcm_prepare (p->handle);
              if (err < 0)
                alsa_sound_perror ("Can't recover from underrun, prepare failed",
                                   err);
            }
          else if (err == -ESTRPIPE)
            {
              while ((err = snd_pcm_resume (p->handle)) == -EAGAIN)
                sleep(1);	/* wait until the suspend flag is released */
              if (err < 0)
                {
                  err = snd_pcm_prepare (p->handle);
                  if (err < 0)
                    alsa_sound_perror ("Can't recover from suspend, "
                                       "prepare failed",
                                       err);
                }
            }
          else
            alsa_sound_perror ("Error writing to sound device", err);

        }
      else
        nwritten += err * fact;
    }
}

static void
snd_error_quiet (file, line, function, err, fmt)
     const char *file;
     int line;
     const char *function;
     int err;
     const char *fmt;
{
}

/* Initialize device SD.  Set up the interface functions in the device
   structure.  */

static int
alsa_init (sd)
     struct sound_device *sd;
{
  char *file;
  snd_pcm_t *handle;
  int err;

  /* Open the sound device.  Default is "default".  */
  if (sd->file)
    file = sd->file;
  else
    file = DEFAULT_ALSA_SOUND_DEVICE;

  snd_lib_error_set_handler ((snd_lib_error_handler_t) snd_error_quiet);
  err = snd_pcm_open (&handle, file, SND_PCM_STREAM_PLAYBACK, 0);
  snd_lib_error_set_handler (NULL);
  if (err < 0)
      return 0;
  snd_pcm_close (handle);

  sd->fd = -1;
  sd->open = alsa_open;
  sd->close = alsa_close;
  sd->configure = alsa_configure;
  sd->choose_format = alsa_choose_format;
  sd->write = alsa_write;
  sd->period_size = alsa_period_size;

  return 1;
}

#endif /* HAVE_ALSA */


/* END: Non Windows functions */
#else /* WINDOWSNT */

/* BEGIN: Windows specific functions */

static int
do_play_sound (psz_file, ui_volume)
     const char *psz_file;
     unsigned long ui_volume;
{
  int i_result = 0;
  MCIERROR mci_error = 0;
  char sz_cmd_buf[520] = {0};
  char sz_ret_buf[520] = {0};
  MMRESULT mm_result = MMSYSERR_NOERROR;
  unsigned long ui_volume_org = 0;
  BOOL b_reset_volume = FALSE;

  memset (sz_cmd_buf, 0, sizeof(sz_cmd_buf));
  memset (sz_ret_buf, 0, sizeof(sz_ret_buf));
  sprintf (sz_cmd_buf,
           "open \"%s\" alias GNUEmacs_PlaySound_Device wait",
           psz_file);
  mci_error = mciSendString (sz_cmd_buf, sz_ret_buf, 520, NULL);
  if (mci_error != 0)
    {
      sound_warning ("The open mciSendString command failed to open\n"
                     "the specified sound file");
      i_result = (int) mci_error;
      return i_result;
    }
  if ((ui_volume > 0) && (ui_volume != UINT_MAX))
    {
      mm_result = waveOutGetVolume ((HWAVEOUT) WAVE_MAPPER, &ui_volume_org);
      if (mm_result == MMSYSERR_NOERROR)
        {
          b_reset_volume = TRUE;
          mm_result = waveOutSetVolume ((HWAVEOUT) WAVE_MAPPER, ui_volume);
          if ( mm_result != MMSYSERR_NOERROR)
            {
              sound_warning ("waveOutSetVolume failed to set the volume level\n"
                             "of the WAVE_MAPPER device.\n"
                             "As a result, the user selected volume level will\n"
                             "not be used.");
            }
        }
      else
        {
          sound_warning ("waveOutGetVolume failed to obtain the original\n"
                         "volume level of the WAVE_MAPPER device.\n"
                         "As a result, the user selected volume level will\n"
                         "not be used.");
        }
    }
  memset (sz_cmd_buf, 0, sizeof(sz_cmd_buf));
  memset (sz_ret_buf, 0, sizeof(sz_ret_buf));
  strcpy (sz_cmd_buf, "play GNUEmacs_PlaySound_Device wait");
  mci_error = mciSendString (sz_cmd_buf, sz_ret_buf, 520, NULL);
  if (mci_error != 0)
    {
      sound_warning ("The play mciSendString command failed to play the\n"
                     "opened sound file.");
      i_result = (int) mci_error;
    }
  memset (sz_cmd_buf, 0, sizeof(sz_cmd_buf));
  memset (sz_ret_buf, 0, sizeof(sz_ret_buf));
  strcpy (sz_cmd_buf, "close GNUEmacs_PlaySound_Device wait");
  mci_error = mciSendString (sz_cmd_buf, sz_ret_buf, 520, NULL);
  if (b_reset_volume == TRUE)
    {
      mm_result=waveOutSetVolume ((HWAVEOUT) WAVE_MAPPER, ui_volume_org);
      if (mm_result != MMSYSERR_NOERROR)
        {
          sound_warning ("waveOutSetVolume failed to reset the original volume\n"
                         "level of the WAVE_MAPPER device.");
        }
    }
  return i_result;
}

/* END: Windows specific functions */

#endif /* WINDOWSNT */

DEFUN ("play-sound-internal", Fplay_sound_internal, Splay_sound_internal, 1, 1, 0,
       doc: /* Play sound SOUND.

Internal use only, use `play-sound' instead.  */)
     (sound)
     Lisp_Object sound;
{
  Lisp_Object attrs[SOUND_ATTR_SENTINEL];
  int count = SPECPDL_INDEX ();

#ifndef WINDOWSNT
  Lisp_Object file;
  struct gcpro gcpro1, gcpro2;
  Lisp_Object args[2];
#else /* WINDOWSNT */
  int len = 0;
  Lisp_Object lo_file = {0};
  char * psz_file = NULL;
  unsigned long ui_volume_tmp = UINT_MAX;
  unsigned long ui_volume = UINT_MAX;
  int i_result = 0;
#endif /* WINDOWSNT */

  /* Parse the sound specification.  Give up if it is invalid.  */
  if (!parse_sound (sound, attrs))
    error ("Invalid sound specification");

#ifndef WINDOWSNT
  file = Qnil;
  GCPRO2 (sound, file);
  current_sound_device = (struct sound_device *) xmalloc (sizeof (struct sound_device));
  bzero (current_sound_device, sizeof (struct sound_device));
  current_sound = (struct sound *) xmalloc (sizeof (struct sound));
  bzero (current_sound, sizeof (struct sound));
  record_unwind_protect (sound_cleanup, Qnil);
  current_sound->header = (char *) alloca (MAX_SOUND_HEADER_BYTES);

  if (STRINGP (attrs[SOUND_FILE]))
    {
      /* Open the sound file.  */
      current_sound->fd = openp (Fcons (Vdata_directory, Qnil),
				 attrs[SOUND_FILE], Qnil, &file, Qnil);
      if (current_sound->fd < 0)
	sound_perror ("Could not open sound file");

      /* Read the first bytes from the file.  */
      current_sound->header_size
	= emacs_read (current_sound->fd, current_sound->header,
		      MAX_SOUND_HEADER_BYTES);
      if (current_sound->header_size < 0)
	sound_perror ("Invalid sound file header");
    }
  else
    {
      current_sound->data = attrs[SOUND_DATA];
      current_sound->header_size = min (MAX_SOUND_HEADER_BYTES, SBYTES (current_sound->data));
      bcopy (SDATA (current_sound->data), current_sound->header, current_sound->header_size);
    }

  /* Find out the type of sound.  Give up if we can't tell.  */
  find_sound_type (current_sound);

  /* Set up a device.  */
  if (STRINGP (attrs[SOUND_DEVICE]))
    {
      int len = SCHARS (attrs[SOUND_DEVICE]);
      current_sound_device->file = (char *) alloca (len + 1);
      strcpy (current_sound_device->file, SDATA (attrs[SOUND_DEVICE]));
    }

  if (INTEGERP (attrs[SOUND_VOLUME]))
    current_sound_device->volume = XFASTINT (attrs[SOUND_VOLUME]);
  else if (FLOATP (attrs[SOUND_VOLUME]))
    current_sound_device->volume = XFLOAT_DATA (attrs[SOUND_VOLUME]) * 100;

  args[0] = Qplay_sound_functions;
  args[1] = sound;
  Frun_hook_with_args (2, args);

#ifdef HAVE_ALSA
  if (!alsa_init (current_sound_device))
#endif
    if (!vox_init (current_sound_device))
      error ("No usable sound device driver found");

  /* Open the device.  */
  current_sound_device->open (current_sound_device);

  /* Play the sound.  */
  current_sound->play (current_sound, current_sound_device);

  /* Clean up.  */
  UNGCPRO;

#else /* WINDOWSNT */

  lo_file = Fexpand_file_name (attrs[SOUND_FILE], Qnil);
  len = XSTRING (lo_file)->size;
  psz_file = (char *) alloca (len + 1);
  strcpy (psz_file, XSTRING (lo_file)->data);
  if (INTEGERP (attrs[SOUND_VOLUME]))
    {
      ui_volume_tmp = XFASTINT (attrs[SOUND_VOLUME]);
    }
  else if (FLOATP (attrs[SOUND_VOLUME]))
    {
      ui_volume_tmp = (unsigned long) XFLOAT_DATA (attrs[SOUND_VOLUME]) * 100;
    }
  /*
    Based on some experiments I have conducted, a value of 100 or less
    for the sound volume is much too low.  You cannot even hear it.
    A value of UINT_MAX indicates that you wish for the sound to played
    at the maximum possible volume.  A value of UINT_MAX/2 plays the
    sound at 50% maximum volume.  Therefore the value passed to do_play_sound
    (and thus to waveOutSetVolume) must be some fraction of UINT_MAX.
    The following code adjusts the user specified volume level appropriately.
  */
  if ((ui_volume_tmp > 0) && (ui_volume_tmp <= 100))
    {
      ui_volume = ui_volume_tmp * (UINT_MAX / 100);
    }
  i_result = do_play_sound (psz_file, ui_volume);

#endif /* WINDOWSNT */

  unbind_to (count, Qnil);
  return Qnil;
}

/***********************************************************************
			    Initialization
 ***********************************************************************/

void
syms_of_sound ()
{
  QCdevice = intern (":device");
  staticpro (&QCdevice);
  QCvolume = intern (":volume");
  staticpro (&QCvolume);
  Qsound = intern ("sound");
  staticpro (&Qsound);
  Qplay_sound_functions = intern ("play-sound-functions");
  staticpro (&Qplay_sound_functions);

  defsubr (&Splay_sound_internal);
}


void
init_sound ()
{
}

#endif /* HAVE_SOUND */

/* arch-tag: dd850ad8-0433-4e2c-9cba-b7aeeccc0dbd
   (do not change this comment) */