/* sound.c -- sound support. Copyright (C) 1998-1999, 2001-2012 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 . */ /* Written by Gerd Moellmann . Tested with Luigi's driver on FreeBSD 2.2.7 with a SoundBlaster 16. */ /* Modified by Ben Key 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 #if defined HAVE_SOUND /* BEGIN: Common Includes */ #include #include #include #include #include #include "lisp.h" #include "dispextern.h" #include "atimer.h" #include #include "syssignal.h" /* END: Common Includes */ /* BEGIN: Non Windows Includes */ #ifndef WINDOWSNT #include /* 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 #endif #ifdef HAVE_SYS_SOUNDCARD_H #include #endif #ifdef HAVE_SOUNDCARD_H #include #endif #ifdef HAVE_ALSA #ifdef ALSA_SUBDIR_INCLUDE #include #else #include #endif /* ALSA_SUBDIR_INCLUDE */ #endif /* HAVE_ALSA */ /* END: Non Windows Includes */ #else /* WINDOWSNT */ /* BEGIN: Windows Specific Includes */ #include #include #include #include /* END: Windows Specific Includes */ #endif /* WINDOWSNT */ /* BEGIN: Common Definitions */ /* Symbols. */ static Lisp_Object QCvolume, QCdevice; static Lisp_Object Qsound; static 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 }; #ifdef HAVE_ALSA static void alsa_sound_perror (const char *, int) NO_RETURN; #endif static void sound_perror (const char *) NO_RETURN; static void sound_warning (const char *); static int parse_sound (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) (struct sound_device *sd); /* Close device SD. */ void (* close) (struct sound_device *sd); /* Configure SD according to device-dependent parameters. */ void (* configure) (struct sound_device *device); /* Choose a device-dependent format for outputting sound S. */ void (* choose_format) (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. */ EMACS_INT (* period_size) (struct sound_device *sd); /* Write NYBTES bytes from BUFFER to device SD. */ void (* write) (struct sound_device *sd, const char *buffer, EMACS_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 read 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) (struct sound *s, struct sound_device *sd); }; /* These are set during `play-sound-internal' so that sound_cleanup has access to them. */ static struct sound_device *current_sound_device; static struct sound *current_sound; /* Function prototypes. */ static void vox_open (struct sound_device *); static void vox_configure (struct sound_device *); static void vox_close (struct sound_device *sd); static void vox_choose_format (struct sound_device *, struct sound *); static int vox_init (struct sound_device *); static void vox_write (struct sound_device *, const char *, EMACS_INT); static void find_sound_type (struct sound *); static u_int32_t le2hl (u_int32_t); static u_int16_t le2hs (u_int16_t); static u_int32_t be2hl (u_int32_t); static int wav_init (struct sound *); static void wav_play (struct sound *, struct sound_device *); static int au_init (struct sound *); static void au_play (struct sound *, struct sound_device *); #if 0 /* Currently not used. */ static u_int16_t be2hs (u_int16_t); #endif /* END: Non Windows Definitions */ #else /* WINDOWSNT */ /* BEGIN: Windows Specific Definitions */ static int do_play_sound (const char *, unsigned long); /* END: Windows Specific Definitions */ #endif /* WINDOWSNT */ /*********************************************************************** General ***********************************************************************/ /* BEGIN: Common functions */ /* Like perror, but signals an error. */ static void sound_perror (const 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 (const char *msg) { message ("%s", 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 (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 (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 (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); xfree (current_sound_device); xfree (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 (u_int32_t value) { #ifdef WORDS_BIGENDIAN 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 (u_int16_t value) { #ifdef WORDS_BIGENDIAN 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 (u_int32_t value) { #ifndef WORDS_BIGENDIAN 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 (u_int16_t value) { #ifndef WORDS_BIGENDIAN 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 (struct sound *s) { struct wav_header *header = (struct wav_header *) s->header; if (s->header_size < sizeof *header || memcmp (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 (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, SSDATA (s->data) + sizeof *header, SBYTES (s->data) - sizeof *header); else { char *buffer; EMACS_INT nbytes = 0; EMACS_INT blksize = sd->period_size ? sd->period_size (sd) : 2048; EMACS_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 (struct sound *s) { struct au_header *header = (struct au_header *) s->header; if (s->header_size < sizeof *header || memcmp (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 (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, SSDATA (s->data) + header->data_offset, SBYTES (s->data) - header->data_offset); else { EMACS_INT blksize = sd->period_size ? sd->period_size (sd) : 2048; char *buffer; EMACS_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 (struct sound_device *sd) { const 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 (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 (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 (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 (struct sound_device *sd) { const 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 (struct sound_device *sd, const char *buffer, EMACS_INT nbytes) { if (emacs_write (sd->fd, buffer, nbytes) != nbytes) 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 (const 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 (struct sound_device *sd) { const 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 EMACS_INT alsa_period_size (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 (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); 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; const 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 (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); } xfree (p); } } /* Choose device-dependent format for device SD from sound file S. */ static void alsa_choose_format (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 = 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 (struct sound_device *sd, const char *buffer, EMACS_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; EMACS_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 (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 (struct sound_device *sd) { const 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 */ #define SOUND_WARNING(fun, error, text) \ { \ char buf[1024]; \ char err_string[MAXERRORLENGTH]; \ fun (error, err_string, sizeof (err_string)); \ _snprintf (buf, sizeof (buf), "%s\nError: %s", \ text, err_string); \ sound_warning (buf); \ } static int do_play_sound (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, sizeof (sz_ret_buf), NULL); if (mci_error != 0) { SOUND_WARNING (mciGetErrorString, mci_error, "The open mciSendString command failed to open " "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 (waveOutGetErrorText, mm_result, "waveOutSetVolume failed to set the volume level " "of the WAVE_MAPPER device.\n" "As a result, the user selected volume level will " "not be used."); } } else { SOUND_WARNING (waveOutGetErrorText, mm_result, "waveOutGetVolume failed to obtain the original " "volume level of the WAVE_MAPPER device.\n" "As a result, the user selected volume level will " "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, sizeof (sz_ret_buf), NULL); if (mci_error != 0) { SOUND_WARNING (mciGetErrorString, mci_error, "The play mciSendString command failed to play the " "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, sizeof (sz_ret_buf), NULL); if (b_reset_volume == TRUE) { mm_result = waveOutSetVolume ((HWAVEOUT) WAVE_MAPPER, ui_volume_org); if (mm_result != MMSYSERR_NOERROR) { SOUND_WARNING (waveOutGetErrorText, mm_result, "waveOutSetVolume failed to reset the original volume " "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. */) (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)); memset (current_sound_device, 0, sizeof (struct sound_device)); current_sound = (struct sound *) xmalloc (sizeof (struct sound)); memset (current_sound, 0, 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)); memcpy (current_sound->header, SDATA (current_sound->data), 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, SSDATA (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 = 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 (void) { DEFSYM (QCdevice, ":device"); DEFSYM (QCvolume, ":volume"); DEFSYM (Qsound, "sound"); DEFSYM (Qplay_sound_functions, "play-sound-functions"); defsubr (&Splay_sound_internal); } void init_sound (void) { } #endif /* HAVE_SOUND */