chan_alsa.c

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/*
 * Asterisk -- An open source telephony toolkit.
 *
 * Copyright (C) 1999 - 2005, Digium, Inc.
 *
 * By Matthew Fredrickson <creslin@digium.com>
 *
 * 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 <creslin@digium.com>
 *
 * \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>
   <defaultenabled>no</defaultenabled>
   <support_level>deprecated</support_level>
   <replacement>chan_console</replacement>
   <deprecated_in>19</deprecated_in>
   <removed_in>21</removed_in>
 ***/

#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_DEPRECATED,
   .load = load_module,
   .unload = unload_module,
   .load_pri = AST_MODPRI_CHANNEL_DRIVER,
);