chan_alsa.c

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/*
 * Asterisk -- An open source telephony toolkit.
 *
 * Copyright (C) 1999 - 2005, Digium, Inc.
 *
 * By Matthew Fredrickson <[email protected]>
 *
 * See http://www.asterisk.org for more information about
 * the Asterisk project. Please do not directly contact
 * any of the maintainers of this project for assistance;
 * the project provides a web site, mailing lists and IRC
 * channels for your use.
 *
 * This program is free software, distributed under the terms of
 * the GNU General Public License Version 2. See the LICENSE file
 * at the top of the source tree.
 */

/*! \file
 * \brief ALSA sound card channel driver
 *
 * \author Matthew Fredrickson <[email protected]>
 *
 * \ingroup channel_drivers
 */

/*! \li \ref chan_alsa.c uses the configuration file \ref alsa.conf
 * \addtogroup configuration_file
 */

/*! \page alsa.conf alsa.conf
 * \verbinclude alsa.conf.sample
 */

/*** MODULEINFO
 <depend>alsa</depend>
 <support_level>extended</support_level>
 ***/

#include "asterisk.h"

#include <errno.h>
#ifndef ESTRPIPE
#define ESTRPIPE EPIPE
#endif
#include <fcntl.h>
#include <sys/ioctl.h>
#include <sys/time.h>

#define ALSA_PCM_NEW_HW_PARAMS_API
#define ALSA_PCM_NEW_SW_PARAMS_API
#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"

/*! Global jitterbuffer configuration - by default, jb is disabled
 * \note Values shown here match the defaults shown in alsa.conf.sample */
static struct ast_jb_conf default_jbconf = {
 .flags = 0,
 .max_size = 200,
 .resync_threshold = 1000,
 .impl = "fixed",
 .target_extra = 40,
};
static struct ast_jb_conf global_jbconf;

#define DEBUG 0
/* Which device to use */
#define ALSA_INDEV "default"
#define ALSA_OUTDEV "default"
#define DESIRED_RATE 8000

/* Lets use 160 sample frames, just like GSM. */
#define FRAME_SIZE 160
#define PERIOD_FRAMES 80 /* 80 Frames, at 2 bytes each */

/* When you set the frame size, you have to come up with
 the right buffer format as well. */
/* 5 64-byte frames = one frame */
#define BUFFER_FMT ((buffersize * 10) << 16) | (0x0006);

/* Don't switch between read/write modes faster than every 300 ms */
#define MIN_SWITCH_TIME 600

#if __BYTE_ORDER == __LITTLE_ENDIAN
static snd_pcm_format_t format = SND_PCM_FORMAT_S16_LE;
#else
static snd_pcm_format_t format = SND_PCM_FORMAT_S16_BE;
#endif

static char indevname[50] = ALSA_INDEV;
static char outdevname[50] = ALSA_OUTDEV;

static int silencesuppression = 0;
static int silencethreshold = 1000;

AST_MUTEX_DEFINE_STATIC(alsalock);

static const char tdesc[] = "ALSA Console Channel Driver";
static const char config[] = "alsa.conf";

static char context[AST_MAX_CONTEXT] = "default";
static char language[MAX_LANGUAGE] = "";
static char exten[AST_MAX_EXTENSION] = "s";
static char mohinterpret[MAX_MUSICCLASS];

static int hookstate = 0;

static struct chan_alsa_pvt {
 /* We only have one ALSA structure -- near sighted perhaps, but it
 keeps this driver as simple as possible -- as it should be. */
 struct ast_channel *owner;
 char exten[AST_MAX_EXTENSION];
 char context[AST_MAX_CONTEXT];
 snd_pcm_t *icard, *ocard;

} alsa;

/* Number of buffers... Each is FRAMESIZE/8 ms long. For example
 with 160 sample frames, and a buffer size of 3, we have a 60ms buffer,
 usually plenty. */

#define MAX_BUFFER_SIZE 100

/* File descriptors for sound device */
static int readdev = -1;
static int writedev = -1;

static int autoanswer = 1;
static int mute = 0;
static int noaudiocapture = 0;

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_digit(struct ast_channel *c, char digit, unsigned int duration);
static int alsa_text(struct ast_channel *c, const char *text);
static int alsa_hangup(struct ast_channel *c);
static int alsa_answer(struct ast_channel *c);
static struct ast_frame *alsa_read(struct ast_channel *chan);
static int alsa_call(struct ast_channel *c, const char *dest, int timeout);
static int alsa_write(struct ast_channel *chan, struct ast_frame *f);
static int alsa_indicate(struct ast_channel *chan, int cond, const void *data, size_t datalen);
static int alsa_fixup(struct ast_channel *oldchan, struct ast_channel *newchan);

static struct ast_channel_tech alsa_tech = {
 .type = "Console",
 .description = tdesc,
 .requester = alsa_request,
 .send_digit_end = alsa_digit,
 .send_text = alsa_text,
 .hangup = alsa_hangup,
 .answer = alsa_answer,
 .read = alsa_read,
 .call = alsa_call,
 .write = alsa_write,
 .indicate = alsa_indicate,
 .fixup = alsa_fixup,
};

static snd_pcm_t *alsa_card_init(char *dev, snd_pcm_stream_t stream)
{
 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;
}

static int soundcard_init(void)
{
 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;
}

static int alsa_digit(struct ast_channel *c, char digit, unsigned int duration)
{
 ast_mutex_lock(&alsalock);
 ast_verbose(" << Console Received digit %c of duration %u ms >> \n",
 digit, duration);
 ast_mutex_unlock(&alsalock);

 return 0;
}

static int alsa_text(struct ast_channel *c, const char *text)
{
 ast_mutex_lock(&alsalock);
 ast_verbose(" << Console Received text %s >> \n", text);
 ast_mutex_unlock(&alsalock);

 return 0;
}

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

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

static int alsa_answer(struct ast_channel *c)
{
 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;
}

static int alsa_hangup(struct ast_channel *c)
{
 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;
}

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


static struct ast_frame *alsa_read(struct ast_channel *chan)
{
 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;
}

static int alsa_fixup(struct ast_channel *oldchan, struct ast_channel *newchan)
{
 struct chan_alsa_pvt *p = ast_channel_tech_pvt(newchan);

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

 return 0;
}

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

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

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)
{
 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;
}

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

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

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

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

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

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

static struct ast_cli_entry cli_alsa[] = {
 AST_CLI_DEFINE(console_answer, "Answer an incoming console call"),
 AST_CLI_DEFINE(console_hangup, "Hangup a call on the console"),
 AST_CLI_DEFINE(console_dial, "Dial an extension on the console"),
 AST_CLI_DEFINE(console_sendtext, "Send text to the remote device"),
 AST_CLI_DEFINE(console_autoanswer, "Sets/displays autoanswer"),
 AST_CLI_DEFINE(console_mute, "Disable/Enable mic input"),
};

static int unload_module(void)
{
 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;
}

/*!
 * \brief 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.
 */
static int load_module(void)
{
 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;
}

AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "ALSA Console Channel Driver",
 .support_level = AST_MODULE_SUPPORT_EXTENDED,
 .load = load_module,
 .unload = unload_module,
 .load_pri = AST_MODPRI_CHANNEL_DRIVER,
);