chan_alsa.c File Reference

ALSA sound card channel driver. More...

#include "asterisk.h"
#include <errno.h>
#include <fcntl.h>
#include <sys/ioctl.h>
#include <sys/time.h>
#include <alsa/asoundlib.h>
#include "asterisk/frame.h"
#include "asterisk/channel.h"
#include "asterisk/module.h"
#include "asterisk/pbx.h"
#include "asterisk/config.h"
#include "asterisk/cli.h"
#include "asterisk/utils.h"
#include "asterisk/causes.h"
#include "asterisk/endian.h"
#include "asterisk/stringfields.h"
#include "asterisk/abstract_jb.h"
#include "asterisk/musiconhold.h"
#include "asterisk/poll-compat.h"
#include "asterisk/stasis_channels.h"
#include "asterisk/format_cache.h"

Include dependency graph for chan_alsa.c:

Go to the source code of this file.

Data Structures
struct	chan_alsa_pvt

Macros
#define	ALSA_INDEV   "default"
#define	ALSA_OUTDEV   "default"
#define	ALSA_PCM_NEW_HW_PARAMS_API
#define	ALSA_PCM_NEW_SW_PARAMS_API
#define	BUFFER_FMT   ((buffersize * 10) << 16) | (0x0006);
#define	DEBUG   0
#define	DESIRED_RATE   8000
#define	ESTRPIPE   EPIPE
#define	FRAME_SIZE   160
#define	MAX_BUFFER_SIZE   100
#define	MIN_SWITCH_TIME   600
#define	PERIOD_FRAMES   80 /* 80 Frames, at 2 bytes each */

Functions
static void	__reg_module (void)
static void	__unreg_module (void)
static int	alsa_answer (struct ast_channel *c)
static int	alsa_call (struct ast_channel *c, const char *dest, int timeout)
static snd_pcm_t *	alsa_card_init (char *dev, snd_pcm_stream_t stream)
static int	alsa_digit (struct ast_channel *c, char digit, unsigned int duration)
static int	alsa_fixup (struct ast_channel *oldchan, struct ast_channel *newchan)
static int	alsa_hangup (struct ast_channel *c)
static int	alsa_indicate (struct ast_channel *chan, int cond, const void *data, size_t datalen)
static struct ast_channel *	alsa_new (struct chan_alsa_pvt *p, int state, const struct ast_assigned_ids *assignedids, const struct ast_channel *requestor)
static struct ast_frame *	alsa_read (struct ast_channel *chan)
static struct ast_channel *	alsa_request (const char *type, struct ast_format_cap *cap, const struct ast_assigned_ids *assignedids, const struct ast_channel *requestor, const char *data, int *cause)
static int	alsa_text (struct ast_channel *c, const char *text)
static int	alsa_write (struct ast_channel *chan, struct ast_frame *f)
struct ast_module *	AST_MODULE_SELF_SYM (void)
static char *	console_answer (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
static char *	console_autoanswer (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
static char *	console_dial (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
static char *	console_hangup (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
static char *	console_mute (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
static char *	console_sendtext (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
static void	grab_owner (void)
static int	load_module (void)
	Load the module. More...
static int	soundcard_init (void)
static int	unload_module (void)

Variables
static struct ast_module_info	__mod_info = { .name = AST_MODULE, .flags = AST_MODFLAG_LOAD_ORDER , .description = "ALSA Console Channel Driver" , .key = "This paragraph is copyright (c) 2006 by Digium, Inc. \In order for your module to load, it must return this \key via a function called \"key\". Any code which \includes this paragraph must be licensed under the GNU \General Public License version 2 or later (at your \option). In addition to Digium's general reservations \of rights, Digium expressly reserves the right to \allow other parties to license this paragraph under \different terms. Any use of Digium, Inc. trademarks or \logos (including \"Asterisk\" or \"Digium\") without \express written permission of Digium, Inc. is prohibited.\n" , .buildopt_sum = "30ef0c93b36035ec78c9cfd712d36d9b" , .support_level = AST_MODULE_SUPPORT_EXTENDED, .load = load_module, .unload = unload_module, .load_pri = AST_MODPRI_CHANNEL_DRIVER, }
static struct chan_alsa_pvt	alsa
static struct ast_channel_tech	alsa_tech
static ast_mutex_t	alsalock = { PTHREAD_RECURSIVE_MUTEX_INITIALIZER_NP , NULL, {1, 0} }
static const struct ast_module_info *	ast_module_info = &__mod_info
static int	autoanswer = 1
static struct ast_cli_entry	cli_alsa []
static const char	config [] = "alsa.conf"
static char	context [AST_MAX_CONTEXT] = "default"
static struct ast_jb_conf	default_jbconf
static char	exten [AST_MAX_EXTENSION] = "s"
static snd_pcm_format_t	format = SND_PCM_FORMAT_S16_BE
static struct ast_jb_conf	global_jbconf
static int	hookstate = 0
static char	indevname [50] = ALSA_INDEV
static char	language [MAX_LANGUAGE] = ""
static char	mohinterpret [MAX_MUSICCLASS]
static int	mute = 0
static int	noaudiocapture = 0
static char	outdevname [50] = ALSA_OUTDEV
static int	readdev = -1
static int	silencesuppression = 0
static int	silencethreshold = 1000
static const char	tdesc [] = "ALSA Console Channel Driver"
static int	writedev = -1

Detailed Description

ALSA sound card channel driver.

Author

Matthew Fredrickson cresl.nosp@m.in@d.nosp@m.igium.nosp@m..com

Definition in file chan_alsa.c.

Macro Definition Documentation

ALSA_INDEV

#define ALSA_INDEV   "default"

Definition at line 83 of file chan_alsa.c.

ALSA_OUTDEV

#define ALSA_OUTDEV   "default"

Definition at line 84 of file chan_alsa.c.

ALSA_PCM_NEW_HW_PARAMS_API

#define ALSA_PCM_NEW_HW_PARAMS_API

Definition at line 50 of file chan_alsa.c.

ALSA_PCM_NEW_SW_PARAMS_API

#define ALSA_PCM_NEW_SW_PARAMS_API

Definition at line 51 of file chan_alsa.c.

BUFFER_FMT

#define BUFFER_FMT   ((buffersize * 10) << 16) | (0x0006);

Definition at line 94 of file chan_alsa.c.

DEBUG

#define DEBUG   0

Definition at line 81 of file chan_alsa.c.

Referenced by __big_split(), and hash_seq().

DESIRED_RATE

#define DESIRED_RATE   8000

Definition at line 85 of file chan_alsa.c.

Referenced by alsa_card_init().

ESTRPIPE

#define ESTRPIPE   EPIPE

Definition at line 44 of file chan_alsa.c.

Referenced by alsa_read(), and alsa_write().

FRAME_SIZE

#define FRAME_SIZE   160

Definition at line 88 of file chan_alsa.c.

Referenced by alsa_read().

MAX_BUFFER_SIZE

#define MAX_BUFFER_SIZE   100

Definition at line 137 of file chan_alsa.c.

MIN_SWITCH_TIME

#define MIN_SWITCH_TIME   600

Definition at line 97 of file chan_alsa.c.

PERIOD_FRAMES

#define PERIOD_FRAMES   80 /* 80 Frames, at 2 bytes each */

Definition at line 89 of file chan_alsa.c.

Referenced by alsa_card_init().

Function Documentation

__reg_module()

static void __reg_module ( void  )

static

Definition at line 1043 of file chan_alsa.c.

__unreg_module()

static void __unreg_module ( void  )

static

Definition at line 1043 of file chan_alsa.c.

alsa_answer()

static int alsa_answer ( struct ast_channel *  c )

static

Definition at line 361 of file chan_alsa.c.

References alsa, alsalock, ast_mutex_lock, ast_mutex_unlock, ast_setstate(), AST_STATE_UP, ast_verbose(), chan_alsa_pvt::icard, and noaudiocapture.

{
   ast_mutex_lock(&alsalock);
   ast_verbose(" << Console call has been answered >> \n");
   ast_setstate(c, AST_STATE_UP);
   if (!noaudiocapture) {
      snd_pcm_prepare(alsa.icard);
      snd_pcm_start(alsa.icard);
   }
   ast_mutex_unlock(&alsalock);

   return 0;
}

alsa_call()

static int alsa_call ( struct ast_channel *  c, const char *  dest, int  timeout  )

static

Definition at line 325 of file chan_alsa.c.

References alsa, alsalock, ast_channel_unlock, AST_CONTROL_ANSWER, AST_CONTROL_RINGING, AST_FRAME_CONTROL, ast_indicate(), ast_mutex_lock, ast_mutex_unlock, ast_queue_frame(), ast_verbose(), autoanswer, grab_owner(), chan_alsa_pvt::icard, ast_frame_subclass::integer, mute, noaudiocapture, chan_alsa_pvt::owner, and ast_frame::subclass.

{
   struct ast_frame f = { AST_FRAME_CONTROL };

   ast_mutex_lock(&alsalock);
   ast_verbose(" << Call placed to '%s' on console >> \n", dest);
   if (autoanswer) {
      ast_verbose(" << Auto-answered >> \n");
      if (mute) {
         ast_verbose( " << Muted >> \n" );
      }
      grab_owner();
      if (alsa.owner) {
         f.subclass.integer = AST_CONTROL_ANSWER;
         ast_queue_frame(alsa.owner, &f);
         ast_channel_unlock(alsa.owner);
      }
   } else {
      ast_verbose(" << Type 'answer' to answer, or use 'autoanswer' for future calls >> \n");
      grab_owner();
      if (alsa.owner) {
         f.subclass.integer = AST_CONTROL_RINGING;
         ast_queue_frame(alsa.owner, &f);
         ast_channel_unlock(alsa.owner);
         ast_indicate(alsa.owner, AST_CONTROL_RINGING);
      }
   }
   if (!noaudiocapture) {
      snd_pcm_prepare(alsa.icard);
      snd_pcm_start(alsa.icard);
   }
   ast_mutex_unlock(&alsalock);

   return 0;
}

alsa_card_init()

static snd_pcm_t* alsa_card_init ( char *  dev, snd_pcm_stream_t  stream  )

static

Definition at line 173 of file chan_alsa.c.

References ast_alloca, ast_debug, ast_log, DESIRED_RATE, format, LOG_ERROR, LOG_WARNING, NULL, PERIOD_FRAMES, readdev, and writedev.

Referenced by soundcard_init().

{
   int err;
   int direction;
   snd_pcm_t *handle = NULL;
   snd_pcm_hw_params_t *hwparams = NULL;
   snd_pcm_sw_params_t *swparams = NULL;
   struct pollfd pfd;
   snd_pcm_uframes_t period_size = PERIOD_FRAMES * 4;
   snd_pcm_uframes_t buffer_size = 0;
   unsigned int rate = DESIRED_RATE;
   snd_pcm_uframes_t start_threshold, stop_threshold;

   err = snd_pcm_open(&handle, dev, stream, SND_PCM_NONBLOCK);
   if (err < 0) {
      ast_log(LOG_ERROR, "snd_pcm_open failed: %s\n", snd_strerror(err));
      return NULL;
   } else {
      ast_debug(1, "Opening device %s in %s mode\n", dev, (stream == SND_PCM_STREAM_CAPTURE) ? "read" : "write");
   }

   hwparams = ast_alloca(snd_pcm_hw_params_sizeof());
   memset(hwparams, 0, snd_pcm_hw_params_sizeof());
   snd_pcm_hw_params_any(handle, hwparams);

   err = snd_pcm_hw_params_set_access(handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED);
   if (err < 0)
      ast_log(LOG_ERROR, "set_access failed: %s\n", snd_strerror(err));

   err = snd_pcm_hw_params_set_format(handle, hwparams, format);
   if (err < 0)
      ast_log(LOG_ERROR, "set_format failed: %s\n", snd_strerror(err));

   err = snd_pcm_hw_params_set_channels(handle, hwparams, 1);
   if (err < 0)
      ast_log(LOG_ERROR, "set_channels failed: %s\n", snd_strerror(err));

   direction = 0;
   err = snd_pcm_hw_params_set_rate_near(handle, hwparams, &rate, &direction);
   if (rate != DESIRED_RATE)
      ast_log(LOG_WARNING, "Rate not correct, requested %d, got %u\n", DESIRED_RATE, rate);

   direction = 0;
   err = snd_pcm_hw_params_set_period_size_near(handle, hwparams, &period_size, &direction);
   if (err < 0)
      ast_log(LOG_ERROR, "period_size(%lu frames) is bad: %s\n", period_size, snd_strerror(err));
   else {
      ast_debug(1, "Period size is %d\n", err);
   }

   buffer_size = 4096 * 2;    /* period_size * 16; */
   err = snd_pcm_hw_params_set_buffer_size_near(handle, hwparams, &buffer_size);
   if (err < 0)
      ast_log(LOG_WARNING, "Problem setting buffer size of %lu: %s\n", buffer_size, snd_strerror(err));
   else {
      ast_debug(1, "Buffer size is set to %d frames\n", err);
   }

   err = snd_pcm_hw_params(handle, hwparams);
   if (err < 0)
      ast_log(LOG_ERROR, "Couldn't set the new hw params: %s\n", snd_strerror(err));

   swparams = ast_alloca(snd_pcm_sw_params_sizeof());
   memset(swparams, 0, snd_pcm_sw_params_sizeof());
   snd_pcm_sw_params_current(handle, swparams);

   if (stream == SND_PCM_STREAM_PLAYBACK)
      start_threshold = period_size;
   else
      start_threshold = 1;

   err = snd_pcm_sw_params_set_start_threshold(handle, swparams, start_threshold);
   if (err < 0)
      ast_log(LOG_ERROR, "start threshold: %s\n", snd_strerror(err));

   if (stream == SND_PCM_STREAM_PLAYBACK)
      stop_threshold = buffer_size;
   else
      stop_threshold = buffer_size;

   err = snd_pcm_sw_params_set_stop_threshold(handle, swparams, stop_threshold);
   if (err < 0)
      ast_log(LOG_ERROR, "stop threshold: %s\n", snd_strerror(err));

   err = snd_pcm_sw_params(handle, swparams);
   if (err < 0)
      ast_log(LOG_ERROR, "sw_params: %s\n", snd_strerror(err));

   err = snd_pcm_poll_descriptors_count(handle);
   if (err <= 0)
      ast_log(LOG_ERROR, "Unable to get a poll descriptors count, error is %s\n", snd_strerror(err));
   if (err != 1) {
      ast_debug(1, "Can't handle more than one device\n");
   }

   snd_pcm_poll_descriptors(handle, &pfd, err);
   ast_debug(1, "Acquired fd %d from the poll descriptor\n", pfd.fd);

   if (stream == SND_PCM_STREAM_CAPTURE)
      readdev = pfd.fd;
   else
      writedev = pfd.fd;

   return handle;
}

alsa_digit()

static int alsa_digit ( struct ast_channel *  c, char  digit, unsigned int  duration  )

static

Definition at line 299 of file chan_alsa.c.

References alsalock, ast_mutex_lock, ast_mutex_unlock, and ast_verbose().

{
   ast_mutex_lock(&alsalock);
   ast_verbose(" << Console Received digit %c of duration %u ms >> \n",
      digit, duration);
   ast_mutex_unlock(&alsalock);

   return 0;
}

alsa_fixup()

static int alsa_fixup ( struct ast_channel *  oldchan, struct ast_channel *  newchan  )

static

Definition at line 531 of file chan_alsa.c.

References alsalock, ast_channel_tech_pvt(), ast_mutex_lock, ast_mutex_unlock, and chan_alsa_pvt::owner.

{
   struct chan_alsa_pvt *p = ast_channel_tech_pvt(newchan);

   ast_mutex_lock(&alsalock);
   p->owner = newchan;
   ast_mutex_unlock(&alsalock);

   return 0;
}

alsa_hangup()

static int alsa_hangup ( struct ast_channel *  c )

static

Definition at line 375 of file chan_alsa.c.

References alsa, alsalock, ast_channel_tech_pvt_set(), ast_module_unref, ast_mutex_lock, ast_mutex_unlock, ast_verbose(), hookstate, chan_alsa_pvt::icard, noaudiocapture, NULL, chan_alsa_pvt::owner, and ast_module_info::self.

{
   ast_mutex_lock(&alsalock);
   ast_channel_tech_pvt_set(c, NULL);
   alsa.owner = NULL;
   ast_verbose(" << Hangup on console >> \n");
   ast_module_unref(ast_module_info->self);
   hookstate = 0;
   if (!noaudiocapture) {
      snd_pcm_drop(alsa.icard);
   }
   ast_mutex_unlock(&alsalock);

   return 0;
}

alsa_indicate()

static int alsa_indicate ( struct ast_channel *  chan, int  cond, const void *  data, size_t  datalen  )

static

Definition at line 542 of file chan_alsa.c.

References alsalock, ast_channel_name(), AST_CONTROL_BUSY, AST_CONTROL_CONGESTION, AST_CONTROL_HOLD, AST_CONTROL_INCOMPLETE, AST_CONTROL_PROCEEDING, AST_CONTROL_PROGRESS, AST_CONTROL_PVT_CAUSE_CODE, AST_CONTROL_RINGING, AST_CONTROL_SRCUPDATE, AST_CONTROL_UNHOLD, AST_CONTROL_VIDUPDATE, ast_log, ast_moh_start(), ast_moh_stop(), ast_mutex_lock, ast_mutex_unlock, ast_verbose(), LOG_WARNING, and mohinterpret.

{
   int res = 0;

   ast_mutex_lock(&alsalock);

   switch (cond) {
   case AST_CONTROL_BUSY:
   case AST_CONTROL_CONGESTION:
   case AST_CONTROL_RINGING:
   case AST_CONTROL_INCOMPLETE:
   case AST_CONTROL_PVT_CAUSE_CODE:
   case -1:
      res = -1;  /* Ask for inband indications */
      break;
   case AST_CONTROL_PROGRESS:
   case AST_CONTROL_PROCEEDING:
   case AST_CONTROL_VIDUPDATE:
   case AST_CONTROL_SRCUPDATE:
      break;
   case AST_CONTROL_HOLD:
      ast_verbose(" << Console Has Been Placed on Hold >> \n");
      ast_moh_start(chan, data, mohinterpret);
      break;
   case AST_CONTROL_UNHOLD:
      ast_verbose(" << Console Has Been Retrieved from Hold >> \n");
      ast_moh_stop(chan);
      break;
   default:
      ast_log(LOG_WARNING, "Don't know how to display condition %d on %s\n", cond, ast_channel_name(chan));
      res = -1;
   }

   ast_mutex_unlock(&alsalock);

   return res;
}

alsa_new()

static struct ast_channel* alsa_new ( struct chan_alsa_pvt *  p, int  state, const struct ast_assigned_ids *  assignedids, const struct ast_channel *  requestor  )

static

Definition at line 580 of file chan_alsa.c.

References ast_channel_alloc, ast_channel_context_set(), ast_channel_exten_set(), ast_channel_name(), ast_channel_nativeformats_set(), ast_channel_set_fd(), ast_channel_set_readformat(), ast_channel_set_writeformat(), ast_channel_stage_snapshot(), ast_channel_stage_snapshot_done(), ast_channel_tech_pvt_set(), ast_channel_tech_set(), ast_channel_unlock, ast_format_slin, ast_hangup(), ast_jb_configure(), ast_log, ast_module_ref, ast_pbx_start(), AST_STATE_DOWN, ast_strlen_zero, ast_channel_tech::capabilities, chan_alsa_pvt::context, chan_alsa_pvt::exten, global_jbconf, indevname, language, LOG_WARNING, NULL, chan_alsa_pvt::owner, readdev, ast_module_info::self, and tmp().

Referenced by alsa_request(), and console_dial().

{
   struct ast_channel *tmp = NULL;

   if (!(tmp = ast_channel_alloc(1, state, 0, 0, "", p->exten, p->context, assignedids, requestor, 0, "ALSA/%s", indevname)))
      return NULL;

   ast_channel_stage_snapshot(tmp);

   ast_channel_tech_set(tmp, &alsa_tech);
   ast_channel_set_fd(tmp, 0, readdev);
   ast_channel_set_readformat(tmp, ast_format_slin);
   ast_channel_set_writeformat(tmp, ast_format_slin);
   ast_channel_nativeformats_set(tmp, alsa_tech.capabilities);

   ast_channel_tech_pvt_set(tmp, p);
   if (!ast_strlen_zero(p->context))
      ast_channel_context_set(tmp, p->context);
   if (!ast_strlen_zero(p->exten))
      ast_channel_exten_set(tmp, p->exten);
   if (!ast_strlen_zero(language))
      ast_channel_language_set(tmp, language);
   p->owner = tmp;
   ast_module_ref(ast_module_info->self);
   ast_jb_configure(tmp, &global_jbconf);

   ast_channel_stage_snapshot_done(tmp);
   ast_channel_unlock(tmp);

   if (state != AST_STATE_DOWN) {
      if (ast_pbx_start(tmp)) {
         ast_log(LOG_WARNING, "Unable to start PBX on %s\n", ast_channel_name(tmp));
         ast_hangup(tmp);
         tmp = NULL;
      }
   }

   return tmp;
}

alsa_read()

static struct ast_frame * alsa_read ( struct ast_channel *  chan )

static

Definition at line 443 of file chan_alsa.c.

References alsa, alsalock, ast_format_slin, AST_FRAME_NULL, AST_FRAME_VOICE, AST_FRIENDLY_OFFSET, ast_log, ast_mutex_lock, ast_mutex_unlock, AST_STATE_UP, buf, ast_frame::data, ast_frame::datalen, ast_frame::delivery, ESTRPIPE, ast_frame_subclass::format, FRAME_SIZE, ast_frame::frametype, chan_alsa_pvt::icard, ast_frame_subclass::integer, LOG_ERROR, ast_frame::mallocd, mute, noaudiocapture, NULL, ast_frame::offset, ast_frame::ptr, ast_frame::samples, ast_frame::src, state, and ast_frame::subclass.

{
   static struct ast_frame f;
   static short __buf[FRAME_SIZE + AST_FRIENDLY_OFFSET / 2];
   short *buf;
   static int readpos = 0;
   static int left = FRAME_SIZE;
   snd_pcm_state_t state;
   int r = 0;

   ast_mutex_lock(&alsalock);
   f.frametype = AST_FRAME_NULL;
   f.subclass.integer = 0;
   f.samples = 0;
   f.datalen = 0;
   f.data.ptr = NULL;
   f.offset = 0;
   f.src = "Console";
   f.mallocd = 0;
   f.delivery.tv_sec = 0;
   f.delivery.tv_usec = 0;

   if (noaudiocapture) {
      /* Return null frame to asterisk*/
      ast_mutex_unlock(&alsalock);
      return &f;
   }

   state = snd_pcm_state(alsa.icard);
   if ((state != SND_PCM_STATE_PREPARED) && (state != SND_PCM_STATE_RUNNING)) {
      snd_pcm_prepare(alsa.icard);
   }

   buf = __buf + AST_FRIENDLY_OFFSET / 2;

   r = snd_pcm_readi(alsa.icard, buf + readpos, left);
   if (r == -EPIPE) {
#if DEBUG
      ast_log(LOG_ERROR, "XRUN read\n");
#endif
      snd_pcm_prepare(alsa.icard);
   } else if (r == -ESTRPIPE) {
      ast_log(LOG_ERROR, "-ESTRPIPE\n");
      snd_pcm_prepare(alsa.icard);
   } else if (r < 0) {
      ast_log(LOG_ERROR, "Read error: %s\n", snd_strerror(r));
   }

   /* Return NULL frame on error */
   if (r < 0) {
      ast_mutex_unlock(&alsalock);
      return &f;
   }

   /* Update positions */
   readpos += r;
   left -= r;

   if (readpos >= FRAME_SIZE) {
      /* A real frame */
      readpos = 0;
      left = FRAME_SIZE;
      if (ast_channel_state(chan) != AST_STATE_UP) {
         /* Don't transmit unless it's up */
         ast_mutex_unlock(&alsalock);
         return &f;
      }
      if (mute) {
         /* Don't transmit if muted */
         ast_mutex_unlock(&alsalock);
         return &f;
      }

      f.frametype = AST_FRAME_VOICE;
      f.subclass.format = ast_format_slin;
      f.samples = FRAME_SIZE;
      f.datalen = FRAME_SIZE * 2;
      f.data.ptr = buf;
      f.offset = AST_FRIENDLY_OFFSET;
      f.src = "Console";
      f.mallocd = 0;

   }
   ast_mutex_unlock(&alsalock);

   return &f;
}

alsa_request()

static struct ast_channel * alsa_request ( const char *  type, struct ast_format_cap *  cap, const struct ast_assigned_ids *  assignedids, const struct ast_channel *  requestor, const char *  data, int *  cause  )

static

Definition at line 620 of file chan_alsa.c.

References alsa, alsa_new(), alsalock, AST_CAUSE_BUSY, ast_format_cap_get_names(), ast_format_cap_iscompatible_format(), AST_FORMAT_CAP_NAMES_LEN, AST_FORMAT_CMP_NOT_EQUAL, ast_format_slin, ast_log, ast_mutex_lock, ast_mutex_unlock, AST_STATE_DOWN, ast_str_alloca, LOG_NOTICE, LOG_WARNING, NULL, chan_alsa_pvt::owner, and tmp().

{
   struct ast_channel *tmp = NULL;

   if (ast_format_cap_iscompatible_format(cap, ast_format_slin) == AST_FORMAT_CMP_NOT_EQUAL) {
      struct ast_str *codec_buf = ast_str_alloca(AST_FORMAT_CAP_NAMES_LEN);
      ast_log(LOG_NOTICE, "Asked to get a channel of format '%s'\n", ast_format_cap_get_names(cap, &codec_buf));
      return NULL;
   }

   ast_mutex_lock(&alsalock);

   if (alsa.owner) {
      ast_log(LOG_NOTICE, "Already have a call on the ALSA channel\n");
      *cause = AST_CAUSE_BUSY;
   } else if (!(tmp = alsa_new(&alsa, AST_STATE_DOWN, assignedids, requestor))) {
      ast_log(LOG_WARNING, "Unable to create new ALSA channel\n");
   }

   ast_mutex_unlock(&alsalock);

   return tmp;
}

alsa_text()

static int alsa_text ( struct ast_channel *  c, const char *  text  )

static

Definition at line 309 of file chan_alsa.c.

References alsalock, ast_mutex_lock, ast_mutex_unlock, and ast_verbose().

{
   ast_mutex_lock(&alsalock);
   ast_verbose(" << Console Received text %s >> \n", text);
   ast_mutex_unlock(&alsalock);

   return 0;
}

alsa_write()

static int alsa_write ( struct ast_channel *  chan, struct ast_frame *  f  )

static

Definition at line 391 of file chan_alsa.c.

References alsa, alsalock, ast_debug, ast_log, ast_mutex_lock, ast_mutex_unlock, ast_frame::data, ast_frame::datalen, ESTRPIPE, len(), LOG_ERROR, LOG_NOTICE, LOG_WARNING, chan_alsa_pvt::ocard, ast_frame::ptr, and state.

{
   static char sizbuf[8000];
   static int sizpos = 0;
   int len = sizpos;
   int res = 0;
   /* size_t frames = 0; */
   snd_pcm_state_t state;

   ast_mutex_lock(&alsalock);

   /* We have to digest the frame in 160-byte portions */
   if (f->datalen > sizeof(sizbuf) - sizpos) {
      ast_log(LOG_WARNING, "Frame too large\n");
      res = -1;
   } else {
      memcpy(sizbuf + sizpos, f->data.ptr, f->datalen);
      len += f->datalen;
      state = snd_pcm_state(alsa.ocard);
      if (state == SND_PCM_STATE_XRUN)
         snd_pcm_prepare(alsa.ocard);
      while ((res = snd_pcm_writei(alsa.ocard, sizbuf, len / 2)) == -EAGAIN) {
         usleep(1);
      }
      if (res == -EPIPE) {
#if DEBUG
         ast_debug(1, "XRUN write\n");
#endif
         snd_pcm_prepare(alsa.ocard);
         while ((res = snd_pcm_writei(alsa.ocard, sizbuf, len / 2)) == -EAGAIN) {
            usleep(1);
         }
         if (res != len / 2) {
            ast_log(LOG_ERROR, "Write error: %s\n", snd_strerror(res));
            res = -1;
         } else if (res < 0) {
            ast_log(LOG_ERROR, "Write error %s\n", snd_strerror(res));
            res = -1;
         }
      } else {
         if (res == -ESTRPIPE)
            ast_log(LOG_ERROR, "You've got some big problems\n");
         else if (res < 0)
            ast_log(LOG_NOTICE, "Error %d on write\n", res);
      }
   }
   ast_mutex_unlock(&alsalock);

   return res >= 0 ? 0 : res;
}

AST_MODULE_SELF_SYM()

struct ast_module* AST_MODULE_SELF_SYM

(

void

)

Definition at line 1043 of file chan_alsa.c.

console_answer()

static char* console_answer ( struct ast_cli_entry *  e, int  cmd, struct ast_cli_args *  a  )

static

Definition at line 680 of file chan_alsa.c.

References alsa, alsalock, ast_cli_args::argc, ast_channel_unlock, ast_cli(), AST_CONTROL_ANSWER, ast_mutex_lock, ast_mutex_unlock, ast_queue_control(), ast_verbose(), CLI_FAILURE, CLI_GENERATE, CLI_INIT, CLI_SHOWUSAGE, CLI_SUCCESS, ast_cli_entry::command, ast_cli_args::fd, grab_owner(), hookstate, chan_alsa_pvt::icard, mute, noaudiocapture, NULL, chan_alsa_pvt::owner, and ast_cli_entry::usage.

{
   char *res = CLI_SUCCESS;

   switch (cmd) {
   case CLI_INIT:
      e->command = "console answer";
      e->usage =
         "Usage: console answer\n"
         "       Answers an incoming call on the console (ALSA) channel.\n";

      return NULL;
   case CLI_GENERATE:
      return NULL;
   }

   if (a->argc != 2)
      return CLI_SHOWUSAGE;

   ast_mutex_lock(&alsalock);

   if (!alsa.owner) {
      ast_cli(a->fd, "No one is calling us\n");
      res = CLI_FAILURE;
   } else {
      if (mute) {
         ast_verbose( " << Muted >> \n" );
      }
      hookstate = 1;
      grab_owner();
      if (alsa.owner) {
         ast_queue_control(alsa.owner, AST_CONTROL_ANSWER);
         ast_channel_unlock(alsa.owner);
      }
   }

   if (!noaudiocapture) {
      snd_pcm_prepare(alsa.icard);
      snd_pcm_start(alsa.icard);
   }

   ast_mutex_unlock(&alsalock);

   return res;
}

console_autoanswer()

static char* console_autoanswer ( struct ast_cli_entry *  e, int  cmd, struct ast_cli_args *  a  )

static

Definition at line 644 of file chan_alsa.c.

References alsalock, ast_cli_args::argc, ast_cli_args::argv, ast_cli(), ast_mutex_lock, ast_mutex_unlock, autoanswer, CLI_GENERATE, CLI_INIT, CLI_SHOWUSAGE, CLI_SUCCESS, ast_cli_entry::command, ast_cli_args::fd, NULL, and ast_cli_entry::usage.

{
   char *res = CLI_SUCCESS;

   switch (cmd) {
   case CLI_INIT:
      e->command = "console autoanswer [on|off]";
      e->usage =
         "Usage: console autoanswer [on|off]\n"
         "       Enables or disables autoanswer feature.  If used without\n"
         "       argument, displays the current on/off status of autoanswer.\n"
         "       The default value of autoanswer is in 'alsa.conf'.\n";
      return NULL;
   case CLI_GENERATE:
      return NULL;
   }

   if ((a->argc != 2) && (a->argc != 3))
      return CLI_SHOWUSAGE;

   ast_mutex_lock(&alsalock);
   if (a->argc == 2) {
      ast_cli(a->fd, "Auto answer is %s.\n", autoanswer ? "on" : "off");
   } else {
      if (!strcasecmp(a->argv[2], "on"))
         autoanswer = -1;
      else if (!strcasecmp(a->argv[2], "off"))
         autoanswer = 0;
      else
         res = CLI_SHOWUSAGE;
   }
   ast_mutex_unlock(&alsalock);

   return res;
}

console_dial()

static char* console_dial ( struct ast_cli_entry *  e, int  cmd, struct ast_cli_args *  a  )

static

Definition at line 813 of file chan_alsa.c.

References alsa, alsa_new(), alsalock, ast_cli_args::argc, ast_cli_args::argv, ast_cli(), ast_copy_string(), ast_exists_extension(), AST_FRAME_DTMF, ast_mutex_lock, ast_mutex_unlock, ast_queue_frame(), AST_STATE_RINGING, ast_strlen_zero, CLI_FAILURE, CLI_GENERATE, CLI_INIT, CLI_SHOWUSAGE, CLI_SUCCESS, ast_cli_entry::command, chan_alsa_pvt::context, d, chan_alsa_pvt::exten, ast_cli_args::fd, ast_frame::frametype, hookstate, NULL, chan_alsa_pvt::owner, strsep(), tmp(), and ast_cli_entry::usage.

{
   char tmp[256], *tmp2;
   char *mye, *myc;
   const char *d;
   char *res = CLI_SUCCESS;

   switch (cmd) {
   case CLI_INIT:
      e->command = "console dial";
      e->usage =
         "Usage: console dial [extension[@context]]\n"
         "       Dials a given extension (and context if specified)\n";
      return NULL;
   case CLI_GENERATE:
      return NULL;
   }

   if ((a->argc != 2) && (a->argc != 3))
      return CLI_SHOWUSAGE;

   ast_mutex_lock(&alsalock);

   if (alsa.owner) {
      if (a->argc == 3) {
         if (alsa.owner) {
            for (d = a->argv[2]; *d; d++) {
               struct ast_frame f = { .frametype = AST_FRAME_DTMF, .subclass.integer = *d };

               ast_queue_frame(alsa.owner, &f);
            }
         }
      } else {
         ast_cli(a->fd, "You're already in a call.  You can use this only to dial digits until you hangup\n");
         res = CLI_FAILURE;
      }
   } else {
      mye = exten;
      myc = context;
      if (a->argc == 3) {
         char *stringp = NULL;

         ast_copy_string(tmp, a->argv[2], sizeof(tmp));
         stringp = tmp;
         strsep(&stringp, "@");
         tmp2 = strsep(&stringp, "@");
         if (!ast_strlen_zero(tmp))
            mye = tmp;
         if (!ast_strlen_zero(tmp2))
            myc = tmp2;
      }
      if (ast_exists_extension(NULL, myc, mye, 1, NULL)) {
         ast_copy_string(alsa.exten, mye, sizeof(alsa.exten));
         ast_copy_string(alsa.context, myc, sizeof(alsa.context));
         hookstate = 1;
         alsa_new(&alsa, AST_STATE_RINGING, NULL, NULL);
      } else
         ast_cli(a->fd, "No such extension '%s' in context '%s'\n", mye, myc);
   }

   ast_mutex_unlock(&alsalock);

   return res;
}

console_hangup()

static char* console_hangup ( struct ast_cli_entry *  e, int  cmd, struct ast_cli_args *  a  )

static

Definition at line 775 of file chan_alsa.c.

References alsa, alsalock, ast_cli_args::argc, AST_CAUSE_NORMAL_CLEARING, ast_channel_unlock, ast_cli(), ast_mutex_lock, ast_mutex_unlock, ast_queue_hangup_with_cause(), CLI_FAILURE, CLI_GENERATE, CLI_INIT, CLI_SHOWUSAGE, CLI_SUCCESS, ast_cli_entry::command, ast_cli_args::fd, grab_owner(), hookstate, NULL, chan_alsa_pvt::owner, and ast_cli_entry::usage.

{
   char *res = CLI_SUCCESS;

   switch (cmd) {
   case CLI_INIT:
      e->command = "console hangup";
      e->usage =
         "Usage: console hangup\n"
         "       Hangs up any call currently placed on the console.\n";
      return NULL;
   case CLI_GENERATE:
      return NULL;
   }


   if (a->argc != 2)
      return CLI_SHOWUSAGE;

   ast_mutex_lock(&alsalock);

   if (!alsa.owner && !hookstate) {
      ast_cli(a->fd, "No call to hangup\n");
      res = CLI_FAILURE;
   } else {
      hookstate = 0;
      grab_owner();
      if (alsa.owner) {
         ast_queue_hangup_with_cause(alsa.owner, AST_CAUSE_NORMAL_CLEARING);
         ast_channel_unlock(alsa.owner);
      }
   }

   ast_mutex_unlock(&alsalock);

   return res;
}

console_mute()

static char* console_mute ( struct ast_cli_entry *  e, int  cmd, struct ast_cli_args *  a  )

static

Definition at line 878 of file chan_alsa.c.

References ast_cli_args::argc, ast_cli_args::argv, ast_cli(), CLI_GENERATE, CLI_INIT, CLI_SHOWUSAGE, CLI_SUCCESS, ast_cli_entry::command, ast_cli_args::fd, mute, NULL, and ast_cli_entry::usage.

{
   int toggle = 0;
   char *res = CLI_SUCCESS;

   switch (cmd) {
   case CLI_INIT:
      e->command = "console {mute|unmute} [toggle]";
      e->usage =
         "Usage: console {mute|unmute} [toggle]\n"
         "       Mute/unmute the microphone.\n";
      return NULL;
   case CLI_GENERATE:
      return NULL;
   }


   if (a->argc > 3) {
      return CLI_SHOWUSAGE;
   }

   if (a->argc == 3) {
      if (strcasecmp(a->argv[2], "toggle"))
         return CLI_SHOWUSAGE;
      toggle = 1;
   }

   if (a->argc < 2) {
      return CLI_SHOWUSAGE;
   }

   if (!strcasecmp(a->argv[1], "mute")) {
      mute = toggle ? !mute : 1;
   } else if (!strcasecmp(a->argv[1], "unmute")) {
      mute = toggle ? !mute : 0;
   } else {
      return CLI_SHOWUSAGE;
   }

   ast_cli(a->fd, "Console mic is %s\n", mute ? "off" : "on");

   return res;
}

console_sendtext()

static char* console_sendtext ( struct ast_cli_entry *  e, int  cmd, struct ast_cli_args *  a  )

static

Definition at line 726 of file chan_alsa.c.

References alsa, alsalock, ast_cli_args::argc, ast_cli_args::argv, ast_channel_unlock, ast_cli(), AST_CONTROL_ANSWER, AST_FRAME_TEXT, ast_mutex_lock, ast_mutex_unlock, ast_queue_control(), ast_queue_frame(), CLI_FAILURE, CLI_GENERATE, CLI_INIT, CLI_SHOWUSAGE, CLI_SUCCESS, ast_cli_entry::command, ast_frame::data, ast_frame::datalen, ast_cli_args::fd, grab_owner(), NULL, chan_alsa_pvt::owner, ast_frame::ptr, and ast_cli_entry::usage.

{
   int tmparg = 3;
   char *res = CLI_SUCCESS;

   switch (cmd) {
   case CLI_INIT:
      e->command = "console send text";
      e->usage =
         "Usage: console send text <message>\n"
         "       Sends a text message for display on the remote terminal.\n";
      return NULL;
   case CLI_GENERATE:
      return NULL;
   }

   if (a->argc < 3)
      return CLI_SHOWUSAGE;

   ast_mutex_lock(&alsalock);

   if (!alsa.owner) {
      ast_cli(a->fd, "No channel active\n");
      res = CLI_FAILURE;
   } else {
      struct ast_frame f = { AST_FRAME_TEXT };
      char text2send[256] = "";

      while (tmparg < a->argc) {
         strncat(text2send, a->argv[tmparg++], sizeof(text2send) - strlen(text2send) - 1);
         strncat(text2send, " ", sizeof(text2send) - strlen(text2send) - 1);
      }

      text2send[strlen(text2send) - 1] = '\n';
      f.data.ptr = text2send;
      f.datalen = strlen(text2send) + 1;
      grab_owner();
      if (alsa.owner) {
         ast_queue_frame(alsa.owner, &f);
         ast_queue_control(alsa.owner, AST_CONTROL_ANSWER);
         ast_channel_unlock(alsa.owner);
      }
   }

   ast_mutex_unlock(&alsalock);

   return res;
}

grab_owner()

static void grab_owner ( void  )

static

Definition at line 318 of file chan_alsa.c.

References alsa, alsalock, ast_channel_trylock, DEADLOCK_AVOIDANCE, and chan_alsa_pvt::owner.

Referenced by alsa_call(), console_answer(), console_hangup(), and console_sendtext().

{
   while (alsa.owner && ast_channel_trylock(alsa.owner)) {
      DEADLOCK_AVOIDANCE(&alsalock);
   }
}

load_module()

static int load_module ( void  )

static

Load the module.

Module loading including tests for configuration or dependencies. This function can return AST_MODULE_LOAD_FAILURE, AST_MODULE_LOAD_DECLINE, or AST_MODULE_LOAD_SUCCESS. If a dependency or environment variable fails tests return AST_MODULE_LOAD_FAILURE. If the module can not load the configuration file or other non-critical problem return AST_MODULE_LOAD_DECLINE. On success return AST_MODULE_LOAD_SUCCESS.

Definition at line 961 of file chan_alsa.c.

References ARRAY_LEN, ast_channel_register(), ast_cli_register_multiple, ast_config_destroy(), ast_config_load, ast_copy_string(), ast_format_cap_alloc, ast_format_cap_append, AST_FORMAT_CAP_FLAG_DEFAULT, ast_format_slin, ast_jb_read_conf(), ast_log, AST_MODFLAG_LOAD_ORDER, AST_MODPRI_CHANNEL_DRIVER, AST_MODULE_INFO(), AST_MODULE_LOAD_DECLINE, AST_MODULE_LOAD_SUCCESS, AST_MODULE_SUPPORT_EXTENDED, ast_true(), ast_variable_browse(), ast_verb, ASTERISK_GPL_KEY, autoanswer, ast_channel_tech::capabilities, config, CONFIG_STATUS_FILEINVALID, chan_alsa_pvt::context, chan_alsa_pvt::exten, global_jbconf, indevname, language, LOG_ERROR, mohinterpret, mute, ast_variable::name, ast_variable::next, noaudiocapture, outdevname, silencesuppression, silencethreshold, soundcard_init(), unload_module(), and ast_variable::value.

{
   struct ast_config *cfg;
   struct ast_variable *v;
   struct ast_flags config_flags = { 0 };

   if (!(alsa_tech.capabilities = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT))) {
      return AST_MODULE_LOAD_DECLINE;
   }
   ast_format_cap_append(alsa_tech.capabilities, ast_format_slin, 0);

   /* Copy the default jb config over global_jbconf */
   memcpy(&global_jbconf, &default_jbconf, sizeof(struct ast_jb_conf));

   strcpy(mohinterpret, "default");

   if (!(cfg = ast_config_load(config, config_flags))) {
      ast_log(LOG_ERROR, "Unable to read ALSA configuration file %s.  Aborting.\n", config);
      return AST_MODULE_LOAD_DECLINE;
   } else if (cfg == CONFIG_STATUS_FILEINVALID) {
      ast_log(LOG_ERROR, "%s is in an invalid format.  Aborting.\n", config);
      return AST_MODULE_LOAD_DECLINE;
   }

   v = ast_variable_browse(cfg, "general");
   for (; v; v = v->next) {
      /* handle jb conf */
      if (!ast_jb_read_conf(&global_jbconf, v->name, v->value)) {
         continue;
      }

      if (!strcasecmp(v->name, "autoanswer")) {
         autoanswer = ast_true(v->value);
      } else if (!strcasecmp(v->name, "mute")) {
         mute = ast_true(v->value);
      } else if (!strcasecmp(v->name, "noaudiocapture")) {
         noaudiocapture = ast_true(v->value);
      } else if (!strcasecmp(v->name, "silencesuppression")) {
         silencesuppression = ast_true(v->value);
      } else if (!strcasecmp(v->name, "silencethreshold")) {
         silencethreshold = atoi(v->value);
      } else if (!strcasecmp(v->name, "context")) {
         ast_copy_string(context, v->value, sizeof(context));
      } else if (!strcasecmp(v->name, "language")) {
         ast_copy_string(language, v->value, sizeof(language));
      } else if (!strcasecmp(v->name, "extension")) {
         ast_copy_string(exten, v->value, sizeof(exten));
      } else if (!strcasecmp(v->name, "input_device")) {
         ast_copy_string(indevname, v->value, sizeof(indevname));
      } else if (!strcasecmp(v->name, "output_device")) {
         ast_copy_string(outdevname, v->value, sizeof(outdevname));
      } else if (!strcasecmp(v->name, "mohinterpret")) {
         ast_copy_string(mohinterpret, v->value, sizeof(mohinterpret));
      }
   }
   ast_config_destroy(cfg);

   if (soundcard_init() < 0) {
      ast_verb(2, "No sound card detected -- console channel will be unavailable\n");
      ast_verb(2, "Turn off ALSA support by adding 'noload=chan_alsa.so' in /etc/asterisk/modules.conf\n");
      unload_module();

      return AST_MODULE_LOAD_DECLINE;
   }

   if (ast_channel_register(&alsa_tech)) {
      ast_log(LOG_ERROR, "Unable to register channel class 'Console'\n");
      unload_module();

      return AST_MODULE_LOAD_DECLINE;
   }

   ast_cli_register_multiple(cli_alsa, ARRAY_LEN(cli_alsa));

   return AST_MODULE_LOAD_SUCCESS;
}

soundcard_init()

static int soundcard_init ( void  )

static

Definition at line 279 of file chan_alsa.c.

References alsa, alsa_card_init(), ast_log, chan_alsa_pvt::icard, indevname, LOG_ERROR, noaudiocapture, chan_alsa_pvt::ocard, outdevname, and writedev.

Referenced by load_module().

{
   if (!noaudiocapture) {
      alsa.icard = alsa_card_init(indevname, SND_PCM_STREAM_CAPTURE);
      if (!alsa.icard) {
         ast_log(LOG_ERROR, "Problem opening alsa capture device\n");
         return -1;
      }
   }

   alsa.ocard = alsa_card_init(outdevname, SND_PCM_STREAM_PLAYBACK);

   if (!alsa.ocard) {
      ast_log(LOG_ERROR, "Problem opening ALSA playback device\n");
      return -1;
   }

   return writedev;
}

unload_module()

static int unload_module ( void  )

static

Definition at line 931 of file chan_alsa.c.

References alsa, ao2_cleanup, ARRAY_LEN, ast_channel_unregister(), ast_cli_unregister_multiple(), ast_softhangup(), AST_SOFTHANGUP_APPUNLOAD, ast_channel_tech::capabilities, chan_alsa_pvt::icard, NULL, chan_alsa_pvt::ocard, and chan_alsa_pvt::owner.

Referenced by load_module().

{
   ast_channel_unregister(&alsa_tech);
   ast_cli_unregister_multiple(cli_alsa, ARRAY_LEN(cli_alsa));

   if (alsa.icard)
      snd_pcm_close(alsa.icard);
   if (alsa.ocard)
      snd_pcm_close(alsa.ocard);
   if (alsa.owner)
      ast_softhangup(alsa.owner, AST_SOFTHANGUP_APPUNLOAD);
   if (alsa.owner)
      return -1;

   ao2_cleanup(alsa_tech.capabilities);
   alsa_tech.capabilities = NULL;

   return 0;
}

Variable Documentation

__mod_info

struct ast_module_info __mod_info = { .name = AST_MODULE, .flags = AST_MODFLAG_LOAD_ORDER , .description = "ALSA Console Channel Driver" , .key = "This paragraph is copyright (c) 2006 by Digium, Inc. \In order for your module to load, it must return this \key via a function called \"key\". Any code which \includes this paragraph must be licensed under the GNU \General Public License version 2 or later (at your \option). In addition to Digium's general reservations \of rights, Digium expressly reserves the right to \allow other parties to license this paragraph under \different terms. Any use of Digium, Inc. trademarks or \logos (including \"Asterisk\" or \"Digium\") without \express written permission of Digium, Inc. is prohibited.\n" , .buildopt_sum = "30ef0c93b36035ec78c9cfd712d36d9b" , .support_level = AST_MODULE_SUPPORT_EXTENDED, .load = load_module, .unload = unload_module, .load_pri = AST_MODPRI_CHANNEL_DRIVER, }

static

Definition at line 1043 of file chan_alsa.c.

alsa

struct chan_alsa_pvt alsa

static

Referenced by alsa_answer(), alsa_call(), alsa_hangup(), alsa_read(), alsa_request(), alsa_write(), console_answer(), console_dial(), console_hangup(), console_sendtext(), grab_owner(), soundcard_init(), and unload_module().

alsa_tech

struct ast_channel_tech alsa_tech

static

Definition at line 158 of file chan_alsa.c.

alsalock

ast_mutex_t alsalock = { PTHREAD_RECURSIVE_MUTEX_INITIALIZER_NP , NULL, {1, 0} }

static

Definition at line 111 of file chan_alsa.c.

Referenced by alsa_answer(), alsa_call(), alsa_digit(), alsa_fixup(), alsa_hangup(), alsa_indicate(), alsa_read(), alsa_request(), alsa_text(), alsa_write(), console_answer(), console_autoanswer(), console_dial(), console_hangup(), console_sendtext(), and grab_owner().

ast_module_info

const struct ast_module_info* ast_module_info = &__mod_info

static

Definition at line 1043 of file chan_alsa.c.

autoanswer

int autoanswer = 1

static

Definition at line 143 of file chan_alsa.c.

Referenced by alsa_call(), console_autoanswer(), and load_module().

cli_alsa

struct ast_cli_entry cli_alsa[]

static

Definition at line 922 of file chan_alsa.c.

config

const char config[] = "alsa.conf"

static

Definition at line 114 of file chan_alsa.c.

Referenced by load_module().

context

char context[AST_MAX_CONTEXT] = "default"

static

Examples:

/usr/src/asterisk-18.5.0/main/app.c.

Definition at line 116 of file chan_alsa.c.

Referenced by __ast_channel_alloc_ap(), __ast_goto_if_exists(), __sip_alloc(), acf_isexten_exec(), acf_vm_info(), action_atxfer(), action_blind_transfer(), action_dialplan_exec(), action_extensionstate(), action_originate(), action_redirect(), add_action_to_menu_entry(), advanced_options(), append_mailbox_mapping(), ast_ari_channels_continue_in_dialplan(), ast_bridge_setup_after_goto(), ast_cdr_fork(), ast_channel_by_exten_cb(), ast_channel_yank(), ast_compile_ael2(), ast_get_enum(), ast_log(), ast_msg_alloc(), ast_msg_set_context(), ast_res_pjsip_initialize_configuration(), ast_sip_cli_print_sorcery_objectset(), ast_walk_context_includes(), ast_xml_query(), attended_transfer_exec(), base_process_party_a(), blind_transfer_exec(), bridge_exec(), build_peer(), cdr_object_create_and_append(), channel_snapshot_dialplan_create(), check_access(), check_peer_ok(), cli_aor_print_body(), cli_aor_print_header(), cli_channel_print_body(), cli_channel_print_header(), cli_channelstats_print_body(), cli_channelstats_print_header(), cli_contact_print_body(), cli_contact_print_header(), cli_endpoint_print_body(), cli_endpoint_print_header(), cli_print_body(), cli_print_header(), cli_unid_print_body(), cli_unid_print_header(), complete_dialplan_add_include(), complete_dialplan_remove_extension(), complete_dialplan_remove_include(), conf_run(), config_parse_variables(), create_addr_from_peer(), create_queue_member(), delete_existing_cb(), disa_exec(), dns_system_resolver_add_record(), dns_system_resolver_set_response(), dundi_do_lookup(), dundi_do_precache(), dundi_do_query(), dundi_exec(), dundi_helper(), dundi_show_cache(), dundi_show_hints(), ebl_callback(), enum_callback(), eval_exten_read(), extenspy_exec(), extstate_read(), find_label_in_current_db(), free_zone(), get_also_info(), get_cid_name(), get_destination(), get_transfer_context(), goto_parser(), handle_blind_transfer(), handle_cli_dialplan_remove_extension(), handle_request_bye(), handle_request_invite(), handle_request_options(), handle_request_refer(), has_voicemail(), hint_read(), iax2_call(), iax2_transfer(), inboxcount2(), init_queue(), inprocess_count(), ip_identify_match_srv_lookup(), isexten_function_read(), launch_ha_netscript(), leave_voicemail(), load_module(), local_alloc(), local_devicestate(), log_exec(), lua_pbx_exec(), lua_register_hints(), lua_register_switches(), lua_sort_extensions(), manager_dialplan_extension_add(), manager_dialplan_extension_remove(), manager_show_dialplan(), mb_poll_thread(), messagecount(), mwi_to_event(), orig_exten(), pbx_builtin_background(), pbx_find_extension(), pbx_parseable_goto(), pickup_exec(), play_message(), print_uri_debug(), process_ast_dsp(), publish_mwi_to_stasis(), pubsub_on_rx_mwi_notify_request(), pvalGotoSetTarget(), queue_mwi_event(), queue_set_param(), readexten_exec(), receive_message(), refer_incoming_attended_request(), refer_incoming_blind_request(), register_exten(), register_peer_exten(), reload(), reload_config(), retrydial_exec(), rx_data_to_ast_msg(), set_peer_defaults(), setup_peer_after_bridge_goto(), socket_process_helper(), str2tech(), try_redirect(), txt_callback(), unregister_exten(), vm_allocate_dh(), vm_sayname(), vmsayname_exec(), and wait_for_answer().

default_jbconf

struct ast_jb_conf default_jbconf

static

Global jitterbuffer configuration - by default, jb is disabled

Note

Values shown here match the defaults shown in alsa.conf.sample

Definition at line 72 of file chan_alsa.c.

exten

char exten[AST_MAX_EXTENSION] = "s"

static

Examples:

/usr/src/asterisk-18.5.0/main/app.c.

Definition at line 118 of file chan_alsa.c.

Referenced by __analog_ss_thread(), __ast_channel_alloc_ap(), action_atxfer(), action_blind_transfer(), action_dialplan_exec(), action_extensionstate(), action_originate(), action_redirect(), add_action_to_menu_entry(), analog_ss_thread(), ast_ari_channels_continue_in_dialplan(), ast_bridge_setup_after_goto(), ast_cdr_fork(), ast_channel_by_exten_cb(), ast_channel_yank(), ast_compile_ael2(), ast_ivr_menu_run_internal(), ast_log(), ast_msg_set_exten(), AST_TEST_DEFINE(), ast_walk_extension_priorities(), attended_transfer_exec(), base_process_party_a(), blind_transfer_exec(), cdr_object_create_and_append(), channel_snapshot_dialplan_create(), check_access(), check_user_full(), complete_dialplan_remove_extension(), complete_dpreply(), config_parse_variables(), context_used(), copy_plain_file(), create_queue_member(), delete_existing_cb(), dialandactivatesub(), disa_exec(), eval_exten_read(), expand_gosub_args(), extenspy_exec(), extstate_read(), feature_attended_transfer(), find_label_in_current_context(), find_label_in_current_db(), gen_prios(), get_cid_name(), get_destination(), get_rdnis(), goto_parser(), handle_blind_transfer(), handle_cli_dialplan_add_extension(), handle_cli_dialplan_remove_extension(), handle_request_invite(), hint_read(), initreqprep(), isexten_function_read(), leave_voicemail(), linkprio(), local_devicestate(), lua_pbx_exec(), lua_sort_extensions(), manager_dialplan_extension_add(), manager_dialplan_extension_remove(), mgcp_ss(), my_new_analog_ast_channel(), my_on_hook(), new_iax(), onedigit_goto(), options_on_rx_request(), orig_exten(), originate_exec(), osplookup_exec(), pbx_builtin_background(), pickup_exec(), pp_each_extension_helper(), readexten_exec(), refer_incoming_blind_request(), register_verify(), reload(), rx_data_to_ast_msg(), session_inv_on_redirected(), sip_new(), sip_request_call(), sla_build_station(), sla_station_destructor(), socket_process_helper(), test_vm_api_destroy_mailbox_voicemails(), test_vm_api_test_setup(), test_vm_api_update_test_snapshots(), transmit_notify_with_mwi(), transmit_register(), user_destructor(), vm_msg_snapshot_create(), and waitstream_core().

format

snd_pcm_format_t format = SND_PCM_FORMAT_S16_BE

static

Examples:

/usr/src/asterisk-18.5.0/include/asterisk/logger.h, and /usr/src/asterisk-18.5.0/include/asterisk/strings.h.

Definition at line 102 of file chan_alsa.c.

Referenced by __stub__ast_agi_unregister_multiple(), acf_sprintf(), acf_strftime(), acf_strptime(), FaxPcap::add(), add_format_information_cb(), add_sdp(), alsa_card_init(), ao2_container_unregister(), apply_cap_to_bundled(), ast_add_profile(), ast_ari_recordings_get_stored_file(), ast_format_attribute_set(), ast_format_cache_get_by_codec(), ast_format_create_named(), ast_get_number_str(), ast_monitor_stop(), ast_openvstream(), ast_rtp_codecs_get_payload_format(), ast_rtp_codecs_payload_replace_format(), ast_rtp_write(), ast_strftime_locale(), AST_TEST_DEFINE(), build_peer(), build_profile(), build_user(), check_header(), check_header_fmt(), configure_local_rtp(), create_addr(), create_outgoing_sdp_stream(), dialog_unlink_all(), does_id_conflict(), file_count_line(), file_read(), file_write(), format_destroy(), format_hash_cb(), get_codecs(), get_soxmix_format(), handle_clear_alarms(), handle_saydatetime(), iax2_codec_choose(), iax2_codec_pref_best_bitfield2cap(), iax2_codec_pref_from_bitfield(), iax2_format_compatibility_bitfield2cap(), iax2_format_compatibility_cap2bitfield(), iax2_request(), jingle_add_payloads_to_description(), spandspflow2pcap::main(), minivm_notify_exec(), multicast_rtp_write(), ooh323c_set_capability(), ooh323c_set_capability_for_call(), process_sdp_a_audio(), process_sdp_a_video(), reload_config(), sayunixtime_exec(), sendmail(), set_config(), set_format(), show_sound_info_cb(), socket_process_helper(), start_monitor_action(), start_monitor_exec(), AllowableRange::to_wiki(), AllowableList::to_wiki(), transmit_connect_with_sdp(), transmit_modify_request(), transmit_modify_with_sdp(), fe6592859b85_fix_mwi_subscribe_replaces_::upgrade(), and vm_allocate_dh().

global_jbconf

struct ast_jb_conf global_jbconf

static

Definition at line 79 of file chan_alsa.c.

Referenced by alsa_new(), and load_module().

hookstate

int hookstate = 0

static

Definition at line 121 of file chan_alsa.c.

Referenced by alsa_hangup(), console_answer(), console_dial(), and console_hangup().

indevname

char indevname[50] = ALSA_INDEV

static

Definition at line 105 of file chan_alsa.c.

Referenced by alsa_new(), load_module(), and soundcard_init().

language

char language[MAX_LANGUAGE] = ""

static

Definition at line 117 of file chan_alsa.c.

Referenced by add_format_information_cb(), alsa_new(), ari_bridges_play_helper(), ari_channels_handle_play(), ast_get_number_str(), ast_res_pjsip_initialize_configuration(), build_peer(), channel_snapshot_base_create(), check_peer_ok(), conf_load_config(), create_addr_from_peer(), func_channel_write_real(), load_module(), set_peer_defaults(), and show_sound_info_cb().

mohinterpret

char mohinterpret[MAX_MUSICCLASS]

static

Definition at line 119 of file chan_alsa.c.

Referenced by __sip_alloc(), alsa_indicate(), build_peer(), check_peer_ok(), create_addr_from_peer(), load_module(), and set_peer_defaults().

mute

int mute = 0

static

Definition at line 144 of file chan_alsa.c.

Referenced by action_toggle_mute(), action_toggle_mute_participants(), alsa_call(), alsa_read(), console_answer(), console_mute(), dahdi_read(), dtmf_detect(), load_module(), mf_detect(), send_mute(), send_select_output(), stasis_app_control_mute_in_bridge(), and tone_detect().

noaudiocapture

int noaudiocapture = 0

static

Definition at line 145 of file chan_alsa.c.

Referenced by alsa_answer(), alsa_call(), alsa_hangup(), alsa_read(), console_answer(), load_module(), and soundcard_init().

outdevname

char outdevname[50] = ALSA_OUTDEV

static

Definition at line 106 of file chan_alsa.c.

Referenced by load_module(), and soundcard_init().

readdev

int readdev = -1

static

Definition at line 140 of file chan_alsa.c.

Referenced by alsa_card_init(), and alsa_new().

silencesuppression

int silencesuppression = 0

static

Definition at line 108 of file chan_alsa.c.

Referenced by load_module().

silencethreshold

int silencethreshold = 1000

static

Definition at line 109 of file chan_alsa.c.

Referenced by load_module().

tdesc

const char tdesc[] = "ALSA Console Channel Driver"

static

Definition at line 113 of file chan_alsa.c.

writedev

int writedev = -1

static

Definition at line 141 of file chan_alsa.c.

Referenced by alsa_card_init(), and soundcard_init().