[disorder] / lib / uaudio-alsa.c

/*
 * This file is part of DisOrder.
 * Copyright (C) 2009 Richard Kettlewell
 *
 * This program 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.
 *
 * This program 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 this program.  If not, see <http://www.gnu.org/licenses/>.
 */
/** @file lib/uaudio-alsa.c
 * @brief Support for ALSA backend */
#include "common.h"

#if HAVE_ALSA_ASOUNDLIB_H

#include <alsa/asoundlib.h>

#include "mem.h"
#include "log.h"
#include "uaudio.h"
#include "configuration.h"

/** @brief The current PCM handle */
static snd_pcm_t *alsa_pcm;

static const char *const alsa_options[] = {
  "device",
  "mixer-control",
  "mixer-channel",
  NULL
};

/** @brief Mixer handle */
snd_mixer_t *alsa_mixer_handle;

/** @brief Mixer control */
static snd_mixer_elem_t *alsa_mixer_elem;

/** @brief Left channel */
static snd_mixer_selem_channel_id_t alsa_mixer_left;

/** @brief Right channel */
static snd_mixer_selem_channel_id_t alsa_mixer_right;

/** @brief Minimum level */
static long alsa_mixer_min;

/** @brief Maximum level */
static long alsa_mixer_max;

/** @brief Actually play sound via ALSA */
static size_t alsa_play(void *buffer, size_t samples, unsigned flags) {
  /* If we're paused we just pretend.  We rely on snd_pcm_writei() blocking so
   * we have to fake up a sleep here.  However it doesn't have to be all that
   * accurate - in particular it's quite acceptable to greatly underestimate
   * the required wait time.  For 'lengthy' waits we do this by the blunt
   * instrument of halving it.  */
  if(flags & UAUDIO_PAUSED) {
    if(samples > 64)
      samples /= 2;
    const uint64_t ns = ((uint64_t)samples * 1000000000
                         / (uaudio_rate * uaudio_channels));
    struct timespec ts[1];
    ts->tv_sec = ns / 1000000000;
    ts->tv_nsec = ns % 1000000000;
    while(nanosleep(ts, ts) < 0 && errno == EINTR)
      ;
    return samples;
  }
  int err;
  /* ALSA wants 'frames', where frame = several concurrently played samples */
  const snd_pcm_uframes_t frames = samples / uaudio_channels;

  snd_pcm_sframes_t rc = snd_pcm_writei(alsa_pcm, buffer, frames);
  if(rc < 0) {
    switch(rc) {
    case -EPIPE:
      if((err = snd_pcm_prepare(alsa_pcm)))
	fatal(0, "error calling snd_pcm_prepare: %d", err);
      return 0;
    case -EAGAIN:
      return 0;
    default:
      fatal(0, "error calling snd_pcm_writei: %d", (int)rc);
    }
  }
  return rc * uaudio_channels;
}

/** @brief Open the ALSA sound device */
static void alsa_open(void) {
  const char *device = uaudio_get("device", "default");
  int err;

  if((err = snd_pcm_open(&alsa_pcm,
			 device,
			 SND_PCM_STREAM_PLAYBACK,
			 0)))
    fatal(0, "error from snd_pcm_open: %d", err);
  snd_pcm_hw_params_t *hwparams;
  snd_pcm_hw_params_alloca(&hwparams);
  if((err = snd_pcm_hw_params_any(alsa_pcm, hwparams)) < 0)
    fatal(0, "error from snd_pcm_hw_params_any: %d", err);
  if((err = snd_pcm_hw_params_set_access(alsa_pcm, hwparams,
                                         SND_PCM_ACCESS_RW_INTERLEAVED)) < 0)
    fatal(0, "error from snd_pcm_hw_params_set_access: %d", err);
  int sample_format;
  if(uaudio_bits == 16)
    sample_format = uaudio_signed ? SND_PCM_FORMAT_S16 : SND_PCM_FORMAT_U16;
  else
    sample_format = uaudio_signed ? SND_PCM_FORMAT_S8 : SND_PCM_FORMAT_U8;
  if((err = snd_pcm_hw_params_set_format(alsa_pcm, hwparams,
                                         sample_format)) < 0)
    fatal(0, "error from snd_pcm_hw_params_set_format (%d): %d",
          sample_format, err);
  unsigned rate = uaudio_rate;
  if((err = snd_pcm_hw_params_set_rate_near(alsa_pcm, hwparams, &rate, 0)) < 0)
    fatal(0, "error from snd_pcm_hw_params_set_rate_near (%d): %d",
          rate, err);
  if((err = snd_pcm_hw_params_set_channels(alsa_pcm, hwparams,
                                           uaudio_channels)) < 0)
    fatal(0, "error from snd_pcm_hw_params_set_channels (%d): %d",
          uaudio_channels, err);
  if((err = snd_pcm_hw_params(alsa_pcm, hwparams)) < 0)
    fatal(0, "error calling snd_pcm_hw_params: %d", err);
  
}

static void alsa_start(uaudio_callback *callback,
                      void *userdata) {
  if(uaudio_channels != 1 && uaudio_channels != 2)
    fatal(0, "asked for %d channels but only support 1 or 2",
          uaudio_channels); 
  if(uaudio_bits != 8 && uaudio_bits != 16)
    fatal(0, "asked for %d bits/channel but only support 8 or 16",
          uaudio_bits); 
  alsa_open();
  uaudio_thread_start(callback, userdata, alsa_play,
                      32 / uaudio_sample_size,
                      4096 / uaudio_sample_size,
                      0);
}

static void alsa_stop(void) {
  uaudio_thread_stop();
  snd_pcm_close(alsa_pcm);
  alsa_pcm = 0;
}

/** @brief Convert a level to a percentage */
static int to_percent(long n) {
  return (n - alsa_mixer_min) * 100 / (alsa_mixer_max - alsa_mixer_min);
}

/** @brief Convert a percentage to a level */
static int from_percent(int n) {
  return alsa_mixer_min + n * (alsa_mixer_max - alsa_mixer_min) / 100;
}

static void alsa_open_mixer(void) {
  int err;
  snd_mixer_selem_id_t *id;
  const char *device = uaudio_get("device", "default");
  const char *mixer = uaudio_get("mixer-control", "0");
  const char *channel = uaudio_get("mixer-channel", "PCM");

  snd_mixer_selem_id_alloca(&id);
  if((err = snd_mixer_open(&alsa_mixer_handle, 0)))
    fatal(0, "snd_mixer_open: %s", snd_strerror(err));
  if((err = snd_mixer_attach(alsa_mixer_handle, device)))
    fatal(0, "snd_mixer_attach %s: %s", device, snd_strerror(err));
  if((err = snd_mixer_selem_register(alsa_mixer_handle,
                                     0/*options*/, 0/*classp*/)))
    fatal(0, "snd_mixer_selem_register %s: %s",
          device, snd_strerror(err));
  if((err = snd_mixer_load(alsa_mixer_handle)))
    fatal(0, "snd_mixer_load %s: %s", device, snd_strerror(err));
  snd_mixer_selem_id_set_name(id, channel);
  snd_mixer_selem_id_set_index(id, atoi(mixer));
  if(!(alsa_mixer_elem = snd_mixer_find_selem(alsa_mixer_handle, id)))
    fatal(0, "device '%s' mixer control '%s,%s' does not exist",
	  device, channel, mixer);
  if(!snd_mixer_selem_has_playback_volume(alsa_mixer_elem))
    fatal(0, "device '%s' mixer control '%s,%s' has no playback volume",
	  device, channel, mixer);
  if(snd_mixer_selem_is_playback_mono(alsa_mixer_elem)) {
    alsa_mixer_left = alsa_mixer_right = SND_MIXER_SCHN_MONO;
  } else {
    alsa_mixer_left = SND_MIXER_SCHN_FRONT_LEFT;
    alsa_mixer_right = SND_MIXER_SCHN_FRONT_RIGHT;
  }
  if(!snd_mixer_selem_has_playback_channel(alsa_mixer_elem,
                                           alsa_mixer_left)
     || !snd_mixer_selem_has_playback_channel(alsa_mixer_elem,
                                              alsa_mixer_right))
    fatal(0, "device '%s' mixer control '%s,%s' lacks required playback channels",
	  device, channel, mixer);
  snd_mixer_selem_get_playback_volume_range(alsa_mixer_elem,
                                            &alsa_mixer_min, &alsa_mixer_max);

}

static void alsa_close_mixer(void) {
  /* TODO alsa_mixer_elem */
  if(alsa_mixer_handle)
    snd_mixer_close(alsa_mixer_handle);
}

static void alsa_get_volume(int *left, int *right) {
  long l, r;
  int err;
  
  if((err = snd_mixer_selem_get_playback_volume(alsa_mixer_elem,
                                                alsa_mixer_left, &l))
     || (err = snd_mixer_selem_get_playback_volume(alsa_mixer_elem,
                                                   alsa_mixer_right, &r)))
    fatal(0, "snd_mixer_selem_get_playback_volume: %s", snd_strerror(err));
  *left = to_percent(l);
  *right = to_percent(r);
}

static void alsa_set_volume(int *left, int *right) {
  long l, r;
  int err;
  
  /* Set the volume */
  if(alsa_mixer_left == alsa_mixer_right) {
    /* Mono output - just use the loudest */
    if((err = snd_mixer_selem_set_playback_volume
	(alsa_mixer_elem, alsa_mixer_left,
	 from_percent(*left > *right ? *left : *right))))
      fatal(0, "snd_mixer_selem_set_playback_volume: %s", snd_strerror(err));
  } else {
    /* Stereo output */
    if((err = snd_mixer_selem_set_playback_volume
	(alsa_mixer_elem, alsa_mixer_left, from_percent(*left)))
       || (err = snd_mixer_selem_set_playback_volume
	   (alsa_mixer_elem, alsa_mixer_right, from_percent(*right))))
      fatal(0, "snd_mixer_selem_set_playback_volume: %s", snd_strerror(err));
  }
  /* Read it back to see what we ended up at */
  if((err = snd_mixer_selem_get_playback_volume(alsa_mixer_elem,
                                                alsa_mixer_left, &l))
     || (err = snd_mixer_selem_get_playback_volume(alsa_mixer_elem,
                                                   alsa_mixer_right, &r)))
    fatal(0, "snd_mixer_selem_get_playback_volume: %s", snd_strerror(err));
  *left = to_percent(l);
  *right = to_percent(r);
}

static void alsa_configure(void) {
  uaudio_set("device", config->device);
  uaudio_set("mixer-control", config->mixer);
  uaudio_set("mixer-channel", config->channel);
}

const struct uaudio uaudio_alsa = {
  .name = "alsa",
  .options = alsa_options,
  .start = alsa_start,
  .stop = alsa_stop,
  .activate = uaudio_thread_activate,
  .deactivate = uaudio_thread_deactivate,
  .open_mixer = alsa_open_mixer,
  .close_mixer = alsa_close_mixer,
  .get_volume = alsa_get_volume,
  .set_volume = alsa_set_volume,
  .configure = alsa_configure
};

#endif

/*
Local Variables:
c-basic-offset:2
comment-column:40
fill-column:79
indent-tabs-mode:nil
End:
*/
Commit	Line	Data
4fd38868 RK	1	/*
	2	* This file is part of DisOrder.
	3	* Copyright (C) 2009 Richard Kettlewell
	4	*
	5	* This program is free software: you can redistribute it and/or modify
	6	* it under the terms of the GNU General Public License as published by
	7	* the Free Software Foundation, either version 3 of the License, or
	8	* (at your option) any later version.
	9	*
	10	* This program is distributed in the hope that it will be useful,
	11	* but WITHOUT ANY WARRANTY; without even the implied warranty of
	12	* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
	13	* GNU General Public License for more details.
	14	*
	15	* You should have received a copy of the GNU General Public License
	16	* along with this program. If not, see <http://www.gnu.org/licenses/>.
	17	*/
	18	/** @file lib/uaudio-alsa.c
	19	* @brief Support for ALSA backend */
	20	#include "common.h"
	21
	22	#if HAVE_ALSA_ASOUNDLIB_H
	23
	24	#include <alsa/asoundlib.h>
	25
	26	#include "mem.h"
	27	#include "log.h"
	28	#include "uaudio.h"
ba70caca	29	#include "configuration.h"
4fd38868 RK	30
	31	/** @brief The current PCM handle */
	32	static snd_pcm_t *alsa_pcm;
	33
	34	static const char *const alsa_options[] = {
	35	"device",
b50cfb8a RK	36	"mixer-control",
b50cfb8a RK	37	"mixer-channel",
4fd38868 RK	38	NULL
	39	};
	40
b50cfb8a RK	41	/** @brief Mixer handle */
	42	snd_mixer_t *alsa_mixer_handle;
	43
	44	/** @brief Mixer control */
	45	static snd_mixer_elem_t *alsa_mixer_elem;
	46
	47	/** @brief Left channel */
	48	static snd_mixer_selem_channel_id_t alsa_mixer_left;
	49
	50	/** @brief Right channel */
	51	static snd_mixer_selem_channel_id_t alsa_mixer_right;
	52
	53	/** @brief Minimum level */
	54	static long alsa_mixer_min;
	55
	56	/** @brief Maximum level */
	57	static long alsa_mixer_max;
	58
4fd38868	59	/** @brief Actually play sound via ALSA */
b1f6ca8c RK	60	static size_t alsa_play(void *buffer, size_t samples, unsigned flags) {
	61	/* If we're paused we just pretend. We rely on snd_pcm_writei() blocking so
	62	* we have to fake up a sleep here. However it doesn't have to be all that
	63	* accurate - in particular it's quite acceptable to greatly underestimate
	64	* the required wait time. For 'lengthy' waits we do this by the blunt
	65	* instrument of halving it. */
	66	if(flags & UAUDIO_PAUSED) {
	67	if(samples > 64)
	68	samples /= 2;
	69	const uint64_t ns = ((uint64_t)samples * 1000000000
	70	/ (uaudio_rate * uaudio_channels));
	71	struct timespec ts[1];
	72	ts->tv_sec = ns / 1000000000;
	73	ts->tv_nsec = ns % 1000000000;
	74	while(nanosleep(ts, ts) < 0 && errno == EINTR)
	75	;
	76	return samples;
	77	}
4fd38868 RK	78	int err;
	79	/* ALSA wants 'frames', where frame = several concurrently played samples */
	80	const snd_pcm_uframes_t frames = samples / uaudio_channels;
	81
	82	snd_pcm_sframes_t rc = snd_pcm_writei(alsa_pcm, buffer, frames);
	83	if(rc < 0) {
	84	switch(rc) {
	85	case -EPIPE:
	86	if((err = snd_pcm_prepare(alsa_pcm)))
	87	fatal(0, "error calling snd_pcm_prepare: %d", err);
	88	return 0;
	89	case -EAGAIN:
	90	return 0;
	91	default:
	92	fatal(0, "error calling snd_pcm_writei: %d", (int)rc);
	93	}
	94	}
	95	return rc * uaudio_channels;
	96	}
	97
	98	/** @brief Open the ALSA sound device */
	99	static void alsa_open(void) {
b50cfb8a	100	const char *device = uaudio_get("device", "default");
4fd38868 RK	101	int err;
4fd38868 RK	102
4fd38868 RK	103	if((err = snd_pcm_open(&alsa_pcm,
	104	device,
	105	SND_PCM_STREAM_PLAYBACK,
	106	0)))
	107	fatal(0, "error from snd_pcm_open: %d", err);
	108	snd_pcm_hw_params_t *hwparams;
	109	snd_pcm_hw_params_alloca(&hwparams);
	110	if((err = snd_pcm_hw_params_any(alsa_pcm, hwparams)) < 0)
	111	fatal(0, "error from snd_pcm_hw_params_any: %d", err);
	112	if((err = snd_pcm_hw_params_set_access(alsa_pcm, hwparams,
	113	SND_PCM_ACCESS_RW_INTERLEAVED)) < 0)
	114	fatal(0, "error from snd_pcm_hw_params_set_access: %d", err);
	115	int sample_format;
	116	if(uaudio_bits == 16)
	117	sample_format = uaudio_signed ? SND_PCM_FORMAT_S16 : SND_PCM_FORMAT_U16;
	118	else
	119	sample_format = uaudio_signed ? SND_PCM_FORMAT_S8 : SND_PCM_FORMAT_U8;
	120	if((err = snd_pcm_hw_params_set_format(alsa_pcm, hwparams,
	121	sample_format)) < 0)
	122	fatal(0, "error from snd_pcm_hw_params_set_format (%d): %d",
	123	sample_format, err);
	124	unsigned rate = uaudio_rate;
	125	if((err = snd_pcm_hw_params_set_rate_near(alsa_pcm, hwparams, &rate, 0)) < 0)
	126	fatal(0, "error from snd_pcm_hw_params_set_rate_near (%d): %d",
	127	rate, err);
	128	if((err = snd_pcm_hw_params_set_channels(alsa_pcm, hwparams,
	129	uaudio_channels)) < 0)
	130	fatal(0, "error from snd_pcm_hw_params_set_channels (%d): %d",
	131	uaudio_channels, err);
	132	if((err = snd_pcm_hw_params(alsa_pcm, hwparams)) < 0)
	133	fatal(0, "error calling snd_pcm_hw_params: %d", err);
	134
	135	}
	136
4fd38868 RK	137	static void alsa_start(uaudio_callback *callback,
	138	void *userdata) {
	139	if(uaudio_channels != 1 && uaudio_channels != 2)
	140	fatal(0, "asked for %d channels but only support 1 or 2",
	141	uaudio_channels);
	142	if(uaudio_bits != 8 && uaudio_bits != 16)
	143	fatal(0, "asked for %d bits/channel but only support 8 or 16",
	144	uaudio_bits);
	145	alsa_open();
	146	uaudio_thread_start(callback, userdata, alsa_play,
	147	32 / uaudio_sample_size,
63761c19 RK	148	4096 / uaudio_sample_size,
63761c19 RK	149	0);
4fd38868 RK	150	}
	151
	152	static void alsa_stop(void) {
	153	uaudio_thread_stop();
	154	snd_pcm_close(alsa_pcm);
	155	alsa_pcm = 0;
	156	}
	157
b50cfb8a RK	158	/** @brief Convert a level to a percentage */
	159	static int to_percent(long n) {
	160	return (n - alsa_mixer_min) * 100 / (alsa_mixer_max - alsa_mixer_min);
	161	}
	162
42f738c2 RK	163	/** @brief Convert a percentage to a level */
	164	static int from_percent(int n) {
	165	return alsa_mixer_min + n * (alsa_mixer_max - alsa_mixer_min) / 100;
	166	}
	167
b50cfb8a RK	168	static void alsa_open_mixer(void) {
	169	int err;
	170	snd_mixer_selem_id_t *id;
	171	const char *device = uaudio_get("device", "default");
	172	const char *mixer = uaudio_get("mixer-control", "0");
	173	const char *channel = uaudio_get("mixer-channel", "PCM");
	174
	175	snd_mixer_selem_id_alloca(&id);
	176	if((err = snd_mixer_open(&alsa_mixer_handle, 0)))
	177	fatal(0, "snd_mixer_open: %s", snd_strerror(err));
42f738c2 RK	178	if((err = snd_mixer_attach(alsa_mixer_handle, device)))
42f738c2 RK	179	fatal(0, "snd_mixer_attach %s: %s", device, snd_strerror(err));
b50cfb8a RK	180	if((err = snd_mixer_selem_register(alsa_mixer_handle,
	181	0/options/, 0/classp/)))
	182	fatal(0, "snd_mixer_selem_register %s: %s",
	183	device, snd_strerror(err));
	184	if((err = snd_mixer_load(alsa_mixer_handle)))
	185	fatal(0, "snd_mixer_load %s: %s", device, snd_strerror(err));
	186	snd_mixer_selem_id_set_name(id, channel);
	187	snd_mixer_selem_id_set_index(id, atoi(mixer));
	188	if(!(alsa_mixer_elem = snd_mixer_find_selem(alsa_mixer_handle, id)))
	189	fatal(0, "device '%s' mixer control '%s,%s' does not exist",
	190	device, channel, mixer);
	191	if(!snd_mixer_selem_has_playback_volume(alsa_mixer_elem))
	192	fatal(0, "device '%s' mixer control '%s,%s' has no playback volume",
	193	device, channel, mixer);
	194	if(snd_mixer_selem_is_playback_mono(alsa_mixer_elem)) {
	195	alsa_mixer_left = alsa_mixer_right = SND_MIXER_SCHN_MONO;
	196	} else {
	197	alsa_mixer_left = SND_MIXER_SCHN_FRONT_LEFT;
	198	alsa_mixer_right = SND_MIXER_SCHN_FRONT_RIGHT;
	199	}
	200	if(!snd_mixer_selem_has_playback_channel(alsa_mixer_elem,
	201	alsa_mixer_left)
	202	\|\| !snd_mixer_selem_has_playback_channel(alsa_mixer_elem,
	203	alsa_mixer_right))
	204	fatal(0, "device '%s' mixer control '%s,%s' lacks required playback channels",
	205	device, channel, mixer);
	206	snd_mixer_selem_get_playback_volume_range(alsa_mixer_elem,
	207	&alsa_mixer_min, &alsa_mixer_max);
	208
	209	}
	210
	211	static void alsa_close_mixer(void) {
	212	/* TODO alsa_mixer_elem */
	213	if(alsa_mixer_handle)
	214	snd_mixer_close(alsa_mixer_handle);
	215	}
	216
	217	static void alsa_get_volume(int left, int right) {
	218	long l, r;
	219	int err;
	220
	221	if((err = snd_mixer_selem_get_playback_volume(alsa_mixer_elem,
	222	alsa_mixer_left, &l))
	223	\|\| (err = snd_mixer_selem_get_playback_volume(alsa_mixer_elem,
	224	alsa_mixer_right, &r)))
	225	fatal(0, "snd_mixer_selem_get_playback_volume: %s", snd_strerror(err));
	226	*left = to_percent(l);
	227	*right = to_percent(r);
	228	}
	229
	230	static void alsa_set_volume(int left, int right) {
	231	long l, r;
	232	int err;
	233
	234	/* Set the volume */
	235	if(alsa_mixer_left == alsa_mixer_right) {
	236	/* Mono output - just use the loudest */
	237	if((err = snd_mixer_selem_set_playback_volume
	238	(alsa_mixer_elem, alsa_mixer_left,
	239	from_percent(left > right ? left : right))))
	240	fatal(0, "snd_mixer_selem_set_playback_volume: %s", snd_strerror(err));
	241	} else {
	242	/* Stereo output */
	243	if((err = snd_mixer_selem_set_playback_volume
244	(alsa_mixer_elem, alsa_mixer_left, from_percent(*left)))
245	\|\| (err = snd_mixer_selem_set_playback_volume
246	(alsa_mixer_elem, alsa_mixer_right, from_percent(*right))))
247	fatal(0, "snd_mixer_selem_set_playback_volume: %s", snd_strerror(err));
248	}
249	/* Read it back to see what we ended up at */
250	if((err = snd_mixer_selem_get_playback_volume(alsa_mixer_elem,
251	alsa_mixer_left, &l))
252	\|\| (err = snd_mixer_selem_get_playback_volume(alsa_mixer_elem,
253	alsa_mixer_right, &r)))
254	fatal(0, "snd_mixer_selem_get_playback_volume: %s", snd_strerror(err));
255	*left = to_percent(l);
256	*right = to_percent(r);
257	}
258
ba70caca RK	259	static void alsa_configure(void) {
	260	uaudio_set("device", config->device);
	261	uaudio_set("mixer-control", config->mixer);
	262	uaudio_set("mixer-channel", config->channel);
	263	}
	264
4fd38868 RK	265	const struct uaudio uaudio_alsa = {
	266	.name = "alsa",
	267	.options = alsa_options,
	268	.start = alsa_start,
	269	.stop = alsa_stop,
b1f6ca8c RK	270	.activate = uaudio_thread_activate,
b1f6ca8c RK	271	.deactivate = uaudio_thread_deactivate,
b50cfb8a RK	272	.open_mixer = alsa_open_mixer,
	273	.close_mixer = alsa_close_mixer,
	274	.get_volume = alsa_get_volume,
	275	.set_volume = alsa_set_volume,
ba70caca	276	.configure = alsa_configure
4fd38868 RK	277	};
	278
	279	#endif
	280
	281	/*
	282	Local Variables:
	283	c-basic-offset:2
	284	comment-column:40
	285	fill-column:79
	286	indent-tabs-mode:nil
	287	End:
	288	*/