dect
/
asterisk
Archived
13
0
Fork 0
Code Releases Activity
This repository has been archived on 2022-02-17. You can view files and clone it, but cannot push or open issues or pull requests.
asterisk/channels/chan_oss.c

849 lines
21 KiB
C
Executable File
Raw Blame History

/*
* Asterisk -- A telephony toolkit for Linux.
*
* Use /dev/dsp as a channel, and the console to command it :).
*
* The full-duplex "simulation" is pretty weak. This is generally a
* VERY BADLY WRITTEN DRIVER so please don't use it as a model for
* writing a driver.
*
* Copyright (C) 1999, Mark Spencer
*
* Mark Spencer <markster@linux-support.net>
*
* This program is free software, distributed under the terms of
* the GNU General Public License
*/
#include <asterisk/frame.h>
#include <asterisk/logger.h>
#include <asterisk/channel.h>
#include <asterisk/module.h>
#include <asterisk/channel_pvt.h>
#include <asterisk/options.h>
#include <asterisk/pbx.h>
#include <asterisk/config.h>
#include <asterisk/cli.h>
#include <unistd.h>
#include <fcntl.h>
#include <errno.h>
#include <sys/ioctl.h>
#include <sys/time.h>
#include <string.h>
#include <stdlib.h>
#include <stdio.h>
#include <linux/soundcard.h>
/* Which device to use */
#define DEV_DSP "/dev/dsp"
/* Lets use 160 sample frames, just like GSM. */
#define FRAME_SIZE 160
/* 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 * 5) << 16) | (0x0006);
/* Don't switch between read/write modes faster than every 300 ms */
#define MIN_SWITCH_TIME 600
static struct timeval lasttime;
static int usecnt;
static int needanswer = 0;
static int needhangup = 0;
static int silencesuppression = 0;
static int silencethreshold = 1000;
static char digits[80] = "";
static char text2send[80] = "";
static pthread_mutex_t usecnt_lock = PTHREAD_MUTEX_INITIALIZER;
static char *type = "Console";
static char *desc = "OSS Console Channel Driver";
static char *tdesc = "OSS Console Channel Driver";
static char *config = "oss.conf";
static char context[AST_MAX_EXTENSION] = "default";
static char language[MAX_LANGUAGE] = "";
static char exten[AST_MAX_EXTENSION] = "s";
/* Some pipes to prevent overflow */
static int funnel[2];
static pthread_mutex_t sound_lock = PTHREAD_MUTEX_INITIALIZER;
static pthread_t silly;
static struct chan_oss_pvt {
/* We only have one OSS 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_EXTENSION];
} oss;
static int time_has_passed()
{
struct timeval tv;
int ms;
gettimeofday(&tv, NULL);
ms = (tv.tv_sec - lasttime.tv_sec) * 1000 +
(tv.tv_usec - lasttime.tv_usec) / 1000;
if (ms > MIN_SWITCH_TIME)
return -1;
return 0;
}
/* 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
static int buffersize = 3;
static int full_duplex = 0;
/* Are we reading or writing (simulated full duplex) */
static int readmode = 1;
/* File descriptor for sound device */
static int sounddev = -1;
static int autoanswer = 1;
static int calc_loudness(short *frame)
{
int sum = 0;
int x;
for (x=0;x<FRAME_SIZE;x++) {
if (frame[x] < 0)
sum -= frame[x];
else
sum += frame[x];
}
sum = sum/FRAME_SIZE;
return sum;
}
static int silence_suppress(short *buf)
{
#define SILBUF 3
int loudness;
static int silentframes = 0;
static char silbuf[FRAME_SIZE * 2 * SILBUF];
static int silbufcnt=0;
if (!silencesuppression)
return 0;
loudness = calc_loudness((short *)(buf));
if (option_debug)
ast_log(LOG_DEBUG, "loudness is %d\n", loudness);
if (loudness < silencethreshold) {
silentframes++;
silbufcnt++;
/* Keep track of the last few bits of silence so we can play
them as lead-in when the time is right */
if (silbufcnt >= SILBUF) {
/* Make way for more buffer */
memmove(silbuf, silbuf + FRAME_SIZE * 2, FRAME_SIZE * 2 * (SILBUF - 1));
silbufcnt--;
}
memcpy(silbuf + FRAME_SIZE * 2 * silbufcnt, buf, FRAME_SIZE * 2);
if (silentframes > 10) {
/* We've had plenty of silence, so compress it now */
return 1;
}
} else {
silentframes=0;
/* Write any buffered silence we have, it may have something
important */
if (silbufcnt) {
write(funnel[1], silbuf, silbufcnt * FRAME_SIZE);
silbufcnt = 0;
}
}
return 0;
}
static void *silly_thread(void *ignore)
{
char buf[FRAME_SIZE * 2];
int pos=0;
int res=0;
/* Read from the sound device, and write to the pipe. */
for (;;) {
/* Give the writer a better shot at the lock */
#if 0
usleep(1000);
#endif
pthread_testcancel();
pthread_mutex_lock(&sound_lock);
res = read(sounddev, buf + pos, FRAME_SIZE * 2 - pos);
pthread_mutex_unlock(&sound_lock);
if (res > 0) {
pos += res;
if (pos == FRAME_SIZE * 2) {
if (needhangup || needanswer || strlen(digits) ||
!silence_suppress((short *)buf)) {
res = write(funnel[1], buf, sizeof(buf));
}
pos = 0;
}
} else {
close(funnel[1]);
break;
}
pthread_testcancel();
}
return NULL;
}
static int setformat(void)
{
int fmt, desired, res, fd = sounddev;
static int warnedalready = 0;
static int warnedalready2 = 0;
pthread_mutex_lock(&sound_lock);
fmt = AFMT_S16_LE;
res = ioctl(fd, SNDCTL_DSP_SETFMT, &fmt);
if (res < 0) {
ast_log(LOG_WARNING, "Unable to set format to 16-bit signed\n");
pthread_mutex_unlock(&sound_lock);
return -1;
}
res = ioctl(fd, SNDCTL_DSP_SETDUPLEX, 0);
if (res >= 0) {
if (option_verbose > 1)
ast_verbose(VERBOSE_PREFIX_2 "Console is full duplex\n");
full_duplex = -1;
}
fmt = 0;
res = ioctl(fd, SNDCTL_DSP_STEREO, &fmt);
if (res < 0) {
ast_log(LOG_WARNING, "Failed to set audio device to mono\n");
pthread_mutex_unlock(&sound_lock);
return -1;
}
/* 8000 Hz desired */
desired = 8000;
fmt = desired;
res = ioctl(fd, SNDCTL_DSP_SPEED, &fmt);
if (res < 0) {
ast_log(LOG_WARNING, "Failed to set audio device to mono\n");
pthread_mutex_unlock(&sound_lock);
return -1;
}
if (fmt != desired) {
if (!warnedalready++)
ast_log(LOG_WARNING, "Requested %d Hz, got %d Hz -- sound may be choppy\n", desired, fmt);
}
#if 1
fmt = BUFFER_FMT;
res = ioctl(fd, SNDCTL_DSP_SETFRAGMENT, &fmt);
if (res < 0) {
if (!warnedalready2++)
ast_log(LOG_WARNING, "Unable to set fragment size -- sound may be choppy\n");
}
#endif
pthread_mutex_unlock(&sound_lock);
return 0;
}
static int soundcard_setoutput(int force)
{
/* Make sure the soundcard is in output mode. */
int fd = sounddev;
if (full_duplex || (!readmode && !force))
return 0;
pthread_mutex_lock(&sound_lock);
readmode = 0;
if (force || time_has_passed()) {
ioctl(sounddev, SNDCTL_DSP_RESET);
/* Keep the same fd reserved by closing the sound device and copying stdin at the same
time. */
/* dup2(0, sound); */
close(sounddev);
fd = open(DEV_DSP, O_WRONLY);
if (fd < 0) {
ast_log(LOG_WARNING, "Unable to re-open DSP device: %s\n", strerror(errno));
pthread_mutex_unlock(&sound_lock);
return -1;
}
/* dup2 will close the original and make fd be sound */
if (dup2(fd, sounddev) < 0) {
ast_log(LOG_WARNING, "dup2() failed: %s\n", strerror(errno));
pthread_mutex_unlock(&sound_lock);
return -1;
}
if (setformat()) {
pthread_mutex_unlock(&sound_lock);
return -1;
}
pthread_mutex_unlock(&sound_lock);
return 0;
}
pthread_mutex_unlock(&sound_lock);
return 1;
}
static int soundcard_setinput(int force)
{
int fd = sounddev;
if (full_duplex || (readmode && !force))
return 0;
pthread_mutex_lock(&sound_lock);
readmode = -1;
if (force || time_has_passed()) {
ioctl(sounddev, SNDCTL_DSP_RESET);
close(sounddev);
/* dup2(0, sound); */
fd = open(DEV_DSP, O_RDONLY);
if (fd < 0) {
ast_log(LOG_WARNING, "Unable to re-open DSP device: %s\n", strerror(errno));
pthread_mutex_unlock(&sound_lock);
return -1;
}
/* dup2 will close the original and make fd be sound */
if (dup2(fd, sounddev) < 0) {
ast_log(LOG_WARNING, "dup2() failed: %s\n", strerror(errno));
pthread_mutex_unlock(&sound_lock);
return -1;
}
if (setformat()) {
pthread_mutex_unlock(&sound_lock);
return -1;
}
pthread_mutex_unlock(&sound_lock);
return 0;
}
pthread_mutex_unlock(&sound_lock);
return 1;
}
static int soundcard_init()
{
/* Assume it's full duplex for starters */
int fd = open(DEV_DSP, O_RDWR);
if (fd < 0) {
ast_log(LOG_ERROR, "Unable to open %s: %s\n", DEV_DSP, strerror(errno));
return fd;
}
gettimeofday(&lasttime, NULL);
sounddev = fd;
setformat();
if (!full_duplex)
soundcard_setinput(1);
return sounddev;
}
static int oss_digit(struct ast_channel *c, char digit)
{
ast_verbose( " << Console Received digit %c >> \n", digit);
return 0;
}
static int oss_text(struct ast_channel *c, char *text)
{
ast_verbose( " << Console Received text %s >> \n", text);
return 0;
}
static int oss_call(struct ast_channel *c, char *dest, int timeout)
{
ast_verbose( " << Call placed to '%s' on console >> \n", dest);
if (autoanswer) {
ast_verbose( " << Auto-answered >> \n" );
needanswer = 1;
} else {
ast_verbose( " << Type 'answer' to answer, or use 'autoanswer' for future calls >> \n");
}
return 0;
}
static int oss_answer(struct ast_channel *c)
{
ast_verbose( " << Console call has been answered >> \n");
c->state = AST_STATE_UP;
return 0;
}
static int oss_hangup(struct ast_channel *c)
{
c->pvt->pvt = NULL;
oss.owner = NULL;
ast_verbose( " << Hangup on console >> \n");
pthread_mutex_lock(&usecnt_lock);
usecnt--;
pthread_mutex_unlock(&usecnt_lock);
needhangup = 0;
needanswer = 0;
return 0;
}
static int soundcard_writeframe(short *data)
{
/* Write an exactly FRAME_SIZE sized of frame */
static int bufcnt = 0;
static short buffer[FRAME_SIZE * MAX_BUFFER_SIZE * 5];
struct audio_buf_info info;
int res;
int fd = sounddev;
static int warned=0;
pthread_mutex_lock(&sound_lock);
if (ioctl(fd, SNDCTL_DSP_GETOSPACE, &info)) {
if (!warned)
ast_log(LOG_WARNING, "Error reading output space\n");
bufcnt = buffersize;
warned++;
}
if ((info.fragments >= buffersize * 5) && (bufcnt == buffersize)) {
/* We've run out of stuff, buffer again */
bufcnt = 0;
}
if (bufcnt == buffersize) {
/* Write sample immediately */
res = write(fd, ((void *)data), FRAME_SIZE * 2);
} else {
/* Copy the data into our buffer */
res = FRAME_SIZE * 2;
memcpy(buffer + (bufcnt * FRAME_SIZE), data, FRAME_SIZE * 2);
bufcnt++;
if (bufcnt == buffersize) {
res = write(fd, ((void *)buffer), FRAME_SIZE * 2 * buffersize);
}
}
pthread_mutex_unlock(&sound_lock);
return res;
}
static int oss_write(struct ast_channel *chan, struct ast_frame *f)
{
int res;
static char sizbuf[8000];
static int sizpos = 0;
int len = sizpos;
int pos;
if (!full_duplex && (strlen(digits) || needhangup || needanswer)) {
/* If we're half duplex, we have to switch to read mode
to honor immediate needs if necessary */
res = soundcard_setinput(1);
if (res < 0) {
ast_log(LOG_WARNING, "Unable to set device to input mode\n");
return -1;
}
return 0;
}
res = soundcard_setoutput(0);
if (res < 0) {
ast_log(LOG_WARNING, "Unable to set output device\n");
return -1;
} else if (res > 0) {
/* The device is still in read mode, and it's too soon to change it,
so just pretend we wrote it */
return 0;
}
/* We have to digest the frame in 160-byte portions */
if (f->datalen > sizeof(sizbuf) - sizpos) {
ast_log(LOG_WARNING, "Frame too large\n");
return -1;
}
memcpy(sizbuf + sizpos, f->data, f->datalen);
len += f->datalen;
pos = 0;
while(len - pos > FRAME_SIZE * 2) {
soundcard_writeframe((short *)(sizbuf + pos));
pos += FRAME_SIZE * 2;
}
if (len - pos)
memmove(sizbuf, sizbuf + pos, len - pos);
sizpos = len - pos;
return 0;
}
static struct ast_frame *oss_read(struct ast_channel *chan)
{
static struct ast_frame f;
static char buf[FRAME_SIZE * 2 + AST_FRIENDLY_OFFSET];
static int readpos = 0;
int res;
#if 0
ast_log(LOG_DEBUG, "oss_read()\n");
#endif
f.frametype = AST_FRAME_NULL;
f.subclass = 0;
f.timelen = 0;
f.datalen = 0;
f.data = NULL;
f.offset = 0;
f.src = type;
f.mallocd = 0;
if (needhangup) {
return NULL;
}
if (strlen(text2send)) {
f.frametype = AST_FRAME_TEXT;
f.subclass = 0;
f.data = text2send;
f.datalen = strlen(text2send);
strcpy(text2send,"");
return &f;
}
if (strlen(digits)) {
f.frametype = AST_FRAME_DTMF;
f.subclass = digits[0];
for (res=0;res<strlen(digits);res++)
digits[res] = digits[res + 1];
return &f;
}
if (needanswer) {
needanswer = 0;
f.frametype = AST_FRAME_CONTROL;
f.subclass = AST_CONTROL_ANSWER;
chan->state = AST_STATE_UP;
return &f;
}
res = soundcard_setinput(0);
if (res < 0) {
ast_log(LOG_WARNING, "Unable to set input mode\n");
return NULL;
}
if (res > 0) {
/* Theoretically shouldn't happen, but anyway, return a NULL frame */
return &f;
}
res = read(funnel[0], buf + AST_FRIENDLY_OFFSET + readpos, FRAME_SIZE * 2 - readpos);
if (res < 0) {
ast_log(LOG_WARNING, "Error reading from sound device: %s\n", strerror(errno));
return NULL;
}
readpos += res;
if (readpos == FRAME_SIZE * 2) {
/* A real frame */
readpos = 0;
f.frametype = AST_FRAME_VOICE;
f.subclass = AST_FORMAT_SLINEAR;
f.timelen = FRAME_SIZE / 8;
f.datalen = FRAME_SIZE * 2;
f.data = buf + AST_FRIENDLY_OFFSET;
f.offset = AST_FRIENDLY_OFFSET;
f.src = type;
f.mallocd = 0;
}
return &f;
}
static struct ast_channel *oss_new(struct chan_oss_pvt *p, int state)
{
struct ast_channel *tmp;
tmp = ast_channel_alloc();
if (tmp) {
snprintf(tmp->name, sizeof(tmp->name), "OSS/%s", DEV_DSP + 5);
tmp->type = type;
tmp->fd = funnel[0];
tmp->nativeformats = AST_FORMAT_SLINEAR;
tmp->pvt->pvt = p;
tmp->pvt->send_digit = oss_digit;
tmp->pvt->send_text = oss_text;
tmp->pvt->hangup = oss_hangup;
tmp->pvt->answer = oss_answer;
tmp->pvt->read = oss_read;
tmp->pvt->call = oss_call;
tmp->pvt->write = oss_write;
if (strlen(p->context))
strncpy(tmp->context, p->context, sizeof(tmp->context));
if (strlen(p->exten))
strncpy(tmp->exten, p->exten, sizeof(tmp->exten));
if (strlen(language))
strncpy(tmp->language, language, sizeof(tmp->language));
p->owner = tmp;
tmp->state = state;
pthread_mutex_lock(&usecnt_lock);
usecnt++;
pthread_mutex_unlock(&usecnt_lock);
ast_update_use_count();
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 *oss_request(char *type, int format, void *data)
{
int oldformat = format;
struct ast_channel *tmp;
format &= AST_FORMAT_SLINEAR;
if (!format) {
ast_log(LOG_NOTICE, "Asked to get a channel of format '%d'\n", oldformat);
return NULL;
}
if (oss.owner) {
ast_log(LOG_NOTICE, "Already have a call on the OSS channel\n");
return NULL;
}
tmp= oss_new(&oss, AST_STATE_DOWN);
if (!tmp) {
ast_log(LOG_WARNING, "Unable to create new OSS channel\n");
}
return tmp;
}
static int console_autoanswer(int fd, int argc, char *argv[])
{
if ((argc != 1) && (argc != 2))
return RESULT_SHOWUSAGE;
if (argc == 1) {
ast_cli(fd, "Auto answer is %s.\n", autoanswer ? "on" : "off");
return RESULT_SUCCESS;
} else {
if (!strcasecmp(argv[1], "on"))
autoanswer = -1;
else if (!strcasecmp(argv[1], "off"))
autoanswer = 0;
else
return RESULT_SHOWUSAGE;
}
return RESULT_SUCCESS;
}
static char *autoanswer_complete(char *line, char *word, int pos, int state)
{
#ifndef MIN
#define MIN(a,b) ((a) < (b) ? (a) : (b))
#endif
switch(state) {
case 0:
if (strlen(word) && !strncasecmp(word, "on", MIN(strlen(word), 2)))
return strdup("on");
case 1:
if (strlen(word) && !strncasecmp(word, "off", MIN(strlen(word), 3)))
return strdup("off");
default:
return NULL;
}
return NULL;
}
static char autoanswer_usage[] =
"Usage: 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 'oss.conf'.\n";
static int console_answer(int fd, int argc, char *argv[])
{
if (argc != 1)
return RESULT_SHOWUSAGE;
if (!oss.owner) {
ast_cli(fd, "No one is calling us\n");
return RESULT_FAILURE;
}
needanswer++;
return RESULT_SUCCESS;
}
static char sendtext_usage[] =
"Usage: send text <message>\n"
" Sends a text message for display on the remote terminal.\n";
static int console_sendtext(int fd, int argc, char *argv[])
{
int tmparg = 1;
if (argc < 1)
return RESULT_SHOWUSAGE;
if (!oss.owner) {
ast_cli(fd, "No one is calling us\n");
return RESULT_FAILURE;
}
if (strlen(text2send))
ast_cli(fd, "Warning: message already waiting to be sent, overwriting\n");
strcpy(text2send, "");
while(tmparg <= argc) {
strncat(text2send, argv[tmparg++], sizeof(text2send) - strlen(text2send));
strncat(text2send, " ", sizeof(text2send) - strlen(text2send));
}
needanswer++;
return RESULT_SUCCESS;
}
static char answer_usage[] =
"Usage: answer\n"
" Answers an incoming call on the console (OSS) channel.\n";
static int console_hangup(int fd, int argc, char *argv[])
{
if (argc != 1)
return RESULT_SHOWUSAGE;
if (!oss.owner) {
ast_cli(fd, "No call to hangup up\n");
return RESULT_FAILURE;
}
needhangup++;
return RESULT_SUCCESS;
}
static char hangup_usage[] =
"Usage: hangup\n"
" Hangs up any call currently placed on the console.\n";
static int console_dial(int fd, int argc, char *argv[])
{
char tmp[256], *tmp2;
char *mye, *myc;
if ((argc != 1) && (argc != 2))
return RESULT_SHOWUSAGE;
if (oss.owner) {
if (argc == 2)
strncat(digits, argv[1], sizeof(digits) - strlen(digits));
else {
ast_cli(fd, "You're already in a call. You can use this only to dial digits until you hangup\n");
return RESULT_FAILURE;
}
return RESULT_SUCCESS;
}
mye = exten;
myc = context;
if (argc == 2) {
strncpy(tmp, argv[1], sizeof(tmp));
strtok(tmp, "@");
tmp2 = strtok(NULL, "@");
if (strlen(tmp))
mye = tmp;
if (tmp2 && strlen(tmp2))
myc = tmp2;
}
if (ast_exists_extension(NULL, myc, mye, 1)) {
strncpy(oss.exten, mye, sizeof(oss.exten));
strncpy(oss.context, myc, sizeof(oss.context));
oss_new(&oss, AST_STATE_UP);
} else
ast_cli(fd, "No such extension '%s' in context '%s'\n", mye, myc);
return RESULT_SUCCESS;
}
static char dial_usage[] =
"Usage: dial [extension[@context]]\n"
" Dials a given extensison (";
static struct ast_cli_entry myclis[] = {
{ { "answer", NULL }, console_answer, "Answer an incoming console call", answer_usage },
{ { "hangup", NULL }, console_hangup, "Hangup a call on the console", hangup_usage },
{ { "dial", NULL }, console_dial, "Dial an extension on the console", dial_usage },
{ { "send text", NULL }, console_sendtext, "Send text to the remote device", sendtext_usage },
{ { "autoanswer", NULL }, console_autoanswer, "Sets/displays autoanswer", autoanswer_usage, autoanswer_complete }
};
int load_module()
{
int res;
int x;
int flags;
struct ast_config *cfg = ast_load(config);
struct ast_variable *v;
res = pipe(funnel);
if (res) {
ast_log(LOG_ERROR, "Unable to create pipe\n");
return -1;
}
/* We make the funnel so that writes to the funnel don't block...
Our "silly" thread can read to its heart content, preventing
recording overruns */
flags = fcntl(funnel[1], F_GETFL);
#if 0
fcntl(funnel[0], F_SETFL, flags | O_NONBLOCK);
#endif
fcntl(funnel[1], F_SETFL, flags | O_NONBLOCK);
res = soundcard_init();
if (res < 0) {
close(funnel[1]);
close(funnel[0]);
return -1;
}
if (!full_duplex)
ast_log(LOG_WARNING, "XXX I don't work right with non-full duplex sound cards XXX\n");
pthread_create(&silly, NULL, silly_thread, NULL);
res = ast_channel_register(type, tdesc, AST_FORMAT_SLINEAR, oss_request);
if (res < 0) {
ast_log(LOG_ERROR, "Unable to register channel class '%s'\n", type);
return -1;
}
for (x=0;x<sizeof(myclis)/sizeof(struct ast_cli_entry); x++)
ast_cli_register(myclis + x);
if (cfg) {
v = ast_variable_browse(cfg, "general");
while(v) {
if (!strcasecmp(v->name, "autoanswer"))
autoanswer = 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"))
strncpy(context, v->value, sizeof(context));
else if (!strcasecmp(v->name, "language"))
strncpy(language, v->value, sizeof(language));
else if (!strcasecmp(v->name, "extension"))
strncpy(exten, v->value, sizeof(exten));
v=v->next;
}
ast_destroy(cfg);
}
return 0;
}
int unload_module()
{
int x;
for (x=0;x<sizeof(myclis)/sizeof(struct ast_cli_entry); x++)
ast_cli_unregister(myclis + x);
close(sounddev);
if (funnel[0] > 0) {
close(funnel[0]);
close(funnel[1]);
}
if (silly) {
pthread_cancel(silly);
pthread_join(silly, NULL);
}
if (oss.owner)
ast_softhangup(oss.owner);
if (oss.owner)
return -1;
return 0;
}
char *description()
{
return desc;
}
int usecount()
{
int res;
pthread_mutex_lock(&usecnt_lock);
res = usecnt;
pthread_mutex_unlock(&usecnt_lock);
return res;
}
char *key()
{
return ASTERISK_GPL_KEY;
}
View Git Blame Copy Permalink