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,
);