* sound.c (alsa_sound_perror, alsa_open, alsa_period_size)
(alsa_configure, alsa_close, alsa_choose_format, alsa_write) (snd_error_quiet, alsa_init): New functions. (vox_init): Return 0 if unable to open device. (Fplay_sound_internal): Test for alsa first and use vox (oss) as a fallback. (struct sound_device): Add period_size. (wav_play, au_play): Use period_size if set.
This commit is contained in:
Jan Djärv
parent
4900d996ca
commit
2d2643f653
1 changed files with 392 additions and 9 deletions
401
src/sound.c
401
src/sound.c
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@ -73,6 +73,10 @@ Boston, MA 02110-1301, USA. */
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#ifdef HAVE_SOUNDCARD_H
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#include <soundcard.h>
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#endif
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#ifdef HAVE_ALSA
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#include <asoundlib.h>
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#endif
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/* END: Non Windows Includes */
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#else /* WINDOWSNT */
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@ -121,6 +125,9 @@ static int parse_sound P_ ((Lisp_Object, Lisp_Object *));
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#ifndef DEFAULT_SOUND_DEVICE
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#define DEFAULT_SOUND_DEVICE "/dev/dsp"
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#endif
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#ifndef DEFAULT_ALSA_SOUND_DEVICE
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#define DEFAULT_ALSA_SOUND_DEVICE "default"
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#endif
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/* Structure forward declarations. */
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@ -227,6 +234,10 @@ struct sound_device
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void (* choose_format) P_ ((struct sound_device *sd,
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struct sound *s));
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/* Return a preferred data size in bytes to be sent to write (below)
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each time. 2048 is used if this is NULL. */
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int (* period_size) P_ ((struct sound_device *sd));
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/* Write NYBTES bytes from BUFFER to device SD. */
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void (* write) P_ ((struct sound_device *sd, const char *buffer,
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int nbytes));
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@ -280,7 +291,7 @@ static void vox_open P_ ((struct sound_device *));
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static void vox_configure P_ ((struct sound_device *));
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static void vox_close P_ ((struct sound_device *sd));
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static void vox_choose_format P_ ((struct sound_device *, struct sound *));
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static void vox_init P_ ((struct sound_device *));
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static int vox_init P_ ((struct sound_device *));
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static void vox_write P_ ((struct sound_device *, const char *, int));
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static void find_sound_type P_ ((struct sound *));
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static u_int32_t le2hl P_ ((u_int32_t));
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@ -604,7 +615,7 @@ wav_play (s, sd)
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{
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char *buffer;
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int nbytes;
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int blksize = 2048;
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int blksize = sd->period_size ? sd->period_size (sd) : 2048;
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buffer = (char *) alloca (blksize);
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lseek (s->fd, sizeof *header, SEEK_SET);
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@ -633,7 +644,8 @@ enum au_encoding
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AU_ENCODING_32,
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AU_ENCODING_IEEE32,
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AU_ENCODING_IEEE64,
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AU_COMPRESSED = 23
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AU_COMPRESSED = 23,
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AU_ENCODING_ALAW_8 = 27
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};
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@ -689,7 +701,7 @@ au_play (s, sd)
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SBYTES (s->data) - header->data_offset);
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else
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{
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int blksize = 2048;
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int blksize = sd->period_size ? sd->period_size (sd) : 2048;
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char *buffer;
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int nbytes;
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@ -868,16 +880,33 @@ vox_choose_format (sd, s)
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/* Initialize device SD. Set up the interface functions in the device
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structure. */
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static void
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static int
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vox_init (sd)
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struct sound_device *sd;
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{
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char *file;
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int fd;
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/* Open the sound device. Default is /dev/dsp. */
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if (sd->file)
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file = sd->file;
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else
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file = DEFAULT_SOUND_DEVICE;
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fd = emacs_open (file, O_WRONLY, 0);
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if (fd >= 0)
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emacs_close (fd);
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else
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return 0;
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sd->fd = -1;
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sd->open = vox_open;
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sd->close = vox_close;
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sd->configure = vox_configure;
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sd->choose_format = vox_choose_format;
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sd->write = vox_write;
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sd->period_size = NULL;
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return 1;
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}
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/* Write NBYTES bytes from BUFFER to device SD. */
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@ -893,6 +922,359 @@ vox_write (sd, buffer, nbytes)
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sound_perror ("Error writing to sound device");
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}
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#ifdef HAVE_ALSA
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/***********************************************************************
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ALSA Driver Interface
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***********************************************************************/
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/* This driver is available on GNU/Linux. */
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static void
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alsa_sound_perror (msg, err)
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char *msg;
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int err;
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{
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error ("%s: %s", msg, snd_strerror (err));
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}
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struct alsa_params
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{
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snd_pcm_t *handle;
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snd_pcm_hw_params_t *hwparams;
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snd_pcm_sw_params_t *swparams;
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snd_pcm_uframes_t period_size;
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};
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/* Open device SD. If SD->file is non-null, open that device,
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otherwise use a default device name. */
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static void
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alsa_open (sd)
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struct sound_device *sd;
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{
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char *file;
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struct alsa_params *p;
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int err;
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/* Open the sound device. Default is "default". */
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if (sd->file)
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file = sd->file;
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else
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file = DEFAULT_ALSA_SOUND_DEVICE;
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p = xmalloc (sizeof (*p));
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p->handle = NULL;
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p->hwparams = NULL;
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p->swparams = NULL;
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sd->fd = -1;
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sd->data = p;
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if ((err = snd_pcm_open (&p->handle, file, SND_PCM_STREAM_PLAYBACK, 0)) < 0)
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alsa_sound_perror (file, err);
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}
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static int
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alsa_period_size (sd)
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struct sound_device *sd;
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{
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struct alsa_params *p = (struct alsa_params *) sd->data;
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return p->period_size;
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}
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static void
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alsa_configure (sd)
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struct sound_device *sd;
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{
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int val, err, dir;
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struct alsa_params *p = (struct alsa_params *) sd->data;
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snd_pcm_uframes_t buffer_size;
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xassert (p->handle != 0);
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if ((err = snd_pcm_hw_params_malloc (&p->hwparams)) < 0)
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alsa_sound_perror ("Could not allocate hardware parameter structure", err);
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if ((err = snd_pcm_sw_params_malloc (&p->swparams)) < 0)
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alsa_sound_perror ("Could not allocate software parameter structure", err);
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if ((err = snd_pcm_hw_params_any (p->handle, p->hwparams)) < 0)
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alsa_sound_perror ("Could not initialize hardware parameter structure", err);
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if ((err = snd_pcm_hw_params_set_access (p->handle, p->hwparams,
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SND_PCM_ACCESS_RW_INTERLEAVED)) < 0)
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alsa_sound_perror ("Could not set access type", err);
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val = sd->format;
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if ((err = snd_pcm_hw_params_set_format (p->handle, p->hwparams, val)) < 0)
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alsa_sound_perror ("Could not set sound format", err);
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val = sd->sample_rate;
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if ((err = snd_pcm_hw_params_set_rate_near (p->handle, p->hwparams, &val, 0))
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< 0)
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alsa_sound_perror ("Could not set sample rate", err);
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val = sd->channels;
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if ((err = snd_pcm_hw_params_set_channels (p->handle, p->hwparams, val)) < 0)
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alsa_sound_perror ("Could not set channel count", err);
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err = snd_pcm_hw_params_get_period_size (p->hwparams, &p->period_size, &dir);
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if (err < 0)
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alsa_sound_perror ("Unable to get period size for playback", err);
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err = snd_pcm_hw_params_get_buffer_size (p->hwparams, &buffer_size);
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if (err < 0)
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alsa_sound_perror("Unable to get buffer size for playback", err);
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if ((err = snd_pcm_hw_params (p->handle, p->hwparams)) < 0)
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alsa_sound_perror ("Could not set parameters", err);
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err = snd_pcm_sw_params_current (p->handle, p->swparams);
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if (err < 0)
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alsa_sound_perror ("Unable to determine current swparams for playback",
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err);
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/* Start the transfer when the buffer is almost full */
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err = snd_pcm_sw_params_set_start_threshold (p->handle, p->swparams,
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(buffer_size / p->period_size)
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* p->period_size);
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if (err < 0)
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alsa_sound_perror ("Unable to set start threshold mode for playback", err);
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/* Allow the transfer when at least period_size samples can be processed */
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err = snd_pcm_sw_params_set_avail_min (p->handle, p->swparams, p->period_size);
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if (err < 0)
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alsa_sound_perror ("Unable to set avail min for playback", err);
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/* Align all transfers to 1 period */
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err = snd_pcm_sw_params_set_xfer_align (p->handle, p->swparams,
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p->period_size);
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if (err < 0)
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alsa_sound_perror ("Unable to set transfer align for playback", err);
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err = snd_pcm_sw_params (p->handle, p->swparams);
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if (err < 0)
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alsa_sound_perror ("Unable to set sw params for playback\n", err);
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snd_pcm_hw_params_free (p->hwparams);
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p->hwparams = NULL;
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snd_pcm_sw_params_free (p->swparams);
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p->swparams = NULL;
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if ((err = snd_pcm_prepare (p->handle)) < 0)
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alsa_sound_perror ("Could not prepare audio interface for use", err);
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if (sd->volume > 0)
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{
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int chn;
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snd_mixer_t *handle;
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snd_mixer_elem_t *e;
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char *file = sd->file ? sd->file : DEFAULT_ALSA_SOUND_DEVICE;
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if (snd_mixer_open (&handle, 0) >= 0)
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{
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if (snd_mixer_attach (handle, file) >= 0
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&& snd_mixer_load (handle) >= 0
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&& snd_mixer_selem_register (handle, NULL, NULL) >= 0)
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for (e = snd_mixer_first_elem (handle);
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e;
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e = snd_mixer_elem_next (e))
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{
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if (snd_mixer_selem_has_playback_volume (e))
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{
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long pmin, pmax;
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snd_mixer_selem_get_playback_volume_range (e, &pmin, &pmax);
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long vol = pmin + (sd->volume * (pmax - pmin)) / 100;
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for (chn = 0; chn <= SND_MIXER_SCHN_LAST; chn++)
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snd_mixer_selem_set_playback_volume (e, chn, vol);
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}
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}
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snd_mixer_close(handle);
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}
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}
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}
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/* Close device SD if it is open. */
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static void
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alsa_close (sd)
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struct sound_device *sd;
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{
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struct alsa_params *p = (struct alsa_params *) sd->data;
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if (p)
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{
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if (p->hwparams)
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snd_pcm_hw_params_free (p->hwparams);
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if (p->swparams)
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snd_pcm_sw_params_free (p->swparams);
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if (p->handle)
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{
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snd_pcm_drain(p->handle);
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snd_pcm_close (p->handle);
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}
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free (p);
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}
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}
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/* Choose device-dependent format for device SD from sound file S. */
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static void
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alsa_choose_format (sd, s)
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struct sound_device *sd;
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struct sound *s;
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{
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struct alsa_params *p = (struct alsa_params *) sd->data;
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if (s->type == RIFF)
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{
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struct wav_header *h = (struct wav_header *) s->header;
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if (h->precision == 8)
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sd->format = SND_PCM_FORMAT_U8;
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else if (h->precision == 16)
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sd->format = SND_PCM_FORMAT_S16_LE;
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else
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error ("Unsupported WAV file format");
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}
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else if (s->type == SUN_AUDIO)
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{
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struct au_header *header = (struct au_header *) s->header;
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switch (header->encoding)
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{
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case AU_ENCODING_ULAW_8:
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sd->format = SND_PCM_FORMAT_MU_LAW;
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break;
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case AU_ENCODING_ALAW_8:
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sd->format = SND_PCM_FORMAT_A_LAW;
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break;
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case AU_ENCODING_IEEE32:
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sd->format = SND_PCM_FORMAT_FLOAT_BE;
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break;
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case AU_ENCODING_IEEE64:
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sd->format = SND_PCM_FORMAT_FLOAT64_BE;
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break;
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case AU_ENCODING_8:
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sd->format = SND_PCM_FORMAT_S8;
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break;
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case AU_ENCODING_16:
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sd->format = SND_PCM_FORMAT_S16_BE;
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break;
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case AU_ENCODING_24:
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sd->format = SND_PCM_FORMAT_S24_BE;
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break;
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case AU_ENCODING_32:
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sd->format = SND_PCM_FORMAT_S32_BE;
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break;
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default:
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error ("Unsupported AU file format");
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}
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}
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else
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abort ();
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}
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/* Write NBYTES bytes from BUFFER to device SD. */
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static void
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alsa_write (sd, buffer, nbytes)
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struct sound_device *sd;
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const char *buffer;
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int nbytes;
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{
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struct alsa_params *p = (struct alsa_params *) sd->data;
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/* The the third parameter to snd_pcm_writei is frames, not bytes. */
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int fact = snd_pcm_format_size (sd->format, 1) * sd->channels;
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int nwritten = 0;
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int err;
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while (nwritten < nbytes)
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{
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if ((err = snd_pcm_writei (p->handle,
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buffer + nwritten,
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(nbytes - nwritten)/fact)) < 0)
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{
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fprintf(stderr, "Err %d/%s\n", err, snd_strerror(err));
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if (err == -EPIPE)
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{ /* under-run */
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err = snd_pcm_prepare (p->handle);
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if (err < 0)
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alsa_sound_perror ("Can't recover from underrun, prepare failed",
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err);
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}
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else if (err == -ESTRPIPE)
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{
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while ((err = snd_pcm_resume (p->handle)) == -EAGAIN)
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sleep(1); /* wait until the suspend flag is released */
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if (err < 0)
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{
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err = snd_pcm_prepare (p->handle);
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if (err < 0)
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alsa_sound_perror ("Can't recover from suspend, "
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"prepare failed",
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err);
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}
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}
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else
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alsa_sound_perror ("Error writing to sound device", err);
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}
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else
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nwritten += err * fact;
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}
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}
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static void
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snd_error_quiet (file, line, function, err, fmt)
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const char *file;
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int line;
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const char *function;
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int err;
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const char *fmt;
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{
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}
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/* Initialize device SD. Set up the interface functions in the device
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structure. */
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static int
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alsa_init (sd)
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struct sound_device *sd;
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{
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char *file;
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snd_pcm_t *handle;
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int err;
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/* Open the sound device. Default is "default". */
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if (sd->file)
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file = sd->file;
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else
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file = DEFAULT_ALSA_SOUND_DEVICE;
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snd_lib_error_set_handler ((snd_lib_error_handler_t) snd_error_quiet);
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err = snd_pcm_open (&handle, file, SND_PCM_STREAM_PLAYBACK, 0);
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snd_lib_error_set_handler (NULL);
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if (err < 0)
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return 0;
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sd->fd = -1;
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sd->open = alsa_open;
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sd->close = alsa_close;
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sd->configure = alsa_configure;
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sd->choose_format = alsa_choose_format;
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sd->write = alsa_write;
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sd->period_size = alsa_period_size;
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return 1;
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}
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#endif /* HAVE_ALSA */
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/* END: Non Windows functions */
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#else /* WINDOWSNT */
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|
|
@ -1056,10 +1438,11 @@ Internal use only, use `play-sound' instead.\n */)
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args[1] = sound;
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Frun_hook_with_args (2, args);
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/* There is only one type of device we currently support, the VOX
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sound driver. Set up the device interface functions for that
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device. */
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vox_init (current_sound_device);
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#ifdef HAVE_ALSA
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if (!alsa_init (current_sound_device))
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#endif
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if (!vox_init (current_sound_device))
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error ("No usable sound device driver found");
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/* Open the device. */
|
||||
current_sound_device->open (current_sound_device);
|
||||
|
|
|
|||
Loading…
Add table
Reference in a new issue