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asterisk/channels/chan_alsa.c

1026 lines
26 KiB
C

/*
* 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>
*
* \par See also
* \arg Config_alsa
*
* \ingroup channel_drivers
*/
/*** MODULEINFO
<depend>alsa</depend>
<support_level>extended</support_level>
***/
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#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"
/*! 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_channel *requestor, void *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, 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 = 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 %d\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(%ld 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 %ld: %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 = 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, 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);
c->tech_pvt = 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;
int off = 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));
} else if (r >= 0) {
off -= r;
}
/* Update positions */
readpos += r;
left -= r;
if (readpos >= FRAME_SIZE) {
/* A real frame */
readpos = 0;
left = FRAME_SIZE;
if (chan->_state != 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;
ast_format_set(&f.subclass.format, AST_FORMAT_SLINEAR, 0);
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 = newchan->tech_pvt;
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 -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, chan->name);
res = -1;
}
ast_mutex_unlock(&alsalock);
return res;
}
static struct ast_channel *alsa_new(struct chan_alsa_pvt *p, int state, const char *linkedid)
{
struct ast_channel *tmp = NULL;
if (!(tmp = ast_channel_alloc(1, state, 0, 0, "", p->exten, p->context, linkedid, 0, "ALSA/%s", indevname)))
return NULL;
tmp->tech = &alsa_tech;
ast_channel_set_fd(tmp, 0, readdev);
ast_format_set(&tmp->readformat, AST_FORMAT_SLINEAR, 0);
ast_format_set(&tmp->writeformat, AST_FORMAT_SLINEAR, 0);
ast_format_cap_add(tmp->nativeformats, &tmp->writeformat);
tmp->tech_pvt = p;
if (!ast_strlen_zero(p->context))
ast_copy_string(tmp->context, p->context, sizeof(tmp->context));
if (!ast_strlen_zero(p->exten))
ast_copy_string(tmp->exten, p->exten, sizeof(tmp->exten));
if (!ast_strlen_zero(language))
ast_string_field_set(tmp, language, language);
p->owner = tmp;
ast_module_ref(ast_module_info->self);
ast_jb_configure(tmp, &global_jbconf);
if (state != AST_STATE_DOWN) {
if (ast_pbx_start(tmp)) {
ast_log(LOG_WARNING, "Unable to start PBX on %s\n", tmp->name);
ast_hangup(tmp);
tmp = NULL;
}
}
return tmp;
}
static struct ast_channel *alsa_request(const char *type, struct ast_format_cap *cap, const struct ast_channel *requestor, void *data, int *cause)
{
struct ast_format tmpfmt;
char buf[256];
struct ast_channel *tmp = NULL;
ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR, 0);
if (!(ast_format_cap_iscompatible(cap, &tmpfmt))) {
ast_log(LOG_NOTICE, "Asked to get a channel of format '%s'\n", ast_getformatname_multiple(buf, sizeof(buf), cap));
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, requestor ? requestor->linkedid : NULL))) {
ast_log(LOG_WARNING, "Unable to create new ALSA channel\n");
}
ast_mutex_unlock(&alsalock);
return tmp;
}
static char *autoanswer_complete(const char *line, const char *word, int pos, int state)
{
switch (state) {
case 0:
if (!ast_strlen_zero(word) && !strncasecmp(word, "on", MIN(strlen(word), 2)))
return ast_strdup("on");
case 1:
if (!ast_strlen_zero(word) && !strncasecmp(word, "off", MIN(strlen(word), 3)))
return ast_strdup("off");
default:
return NULL;
}
return NULL;
}
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";
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 autoanswer_complete(a->line, a->word, a->pos, a->n);
}
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);
} 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 load_module(void)
{
struct ast_config *cfg;
struct ast_variable *v;
struct ast_flags config_flags = { 0 };
struct ast_format tmpfmt;
if (!(alsa_tech.capabilities = ast_format_cap_alloc())) {
return AST_MODULE_LOAD_DECLINE;
}
ast_format_cap_add(alsa_tech.capabilities, ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR, 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");
return AST_MODULE_LOAD_DECLINE;
}
if (ast_channel_register(&alsa_tech)) {
ast_log(LOG_ERROR, "Unable to register channel class 'Console'\n");
return AST_MODULE_LOAD_FAILURE;
}
ast_cli_register_multiple(cli_alsa, ARRAY_LEN(cli_alsa));
return AST_MODULE_LOAD_SUCCESS;
}
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;
alsa_tech.capabilities = ast_format_cap_destroy(alsa_tech.capabilities);
return 0;
}
AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "ALSA Console Channel Driver",
.load = load_module,
.unload = unload_module,
.load_pri = AST_MODPRI_CHANNEL_DRIVER,
);