asterisk/res/res_audiosocket.c

345 lines
7.6 KiB
C

/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 2019, CyCore Systems, Inc
*
* Seán C McCord <scm@cycoresys.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 AudioSocket support for Asterisk
*
* \author Seán C McCord <scm@cycoresys.com>
*
*/
/*** MODULEINFO
<support_level>extended</support_level>
***/
#include "asterisk.h"
#include "errno.h"
#include <uuid/uuid.h>
#include "asterisk/file.h"
#include "asterisk/res_audiosocket.h"
#include "asterisk/channel.h"
#include "asterisk/module.h"
#include "asterisk/uuid.h"
#include "asterisk/format_cache.h"
#define MODULE_DESCRIPTION "AudioSocket support functions for Asterisk"
#define MAX_CONNECT_TIMEOUT_MSEC 2000
/*!
* \internal
* \brief Attempt to complete the audiosocket connection.
*
* \param server Url that we are trying to connect to.
* \param addr Address that host was resolved to.
* \param netsockfd File descriptor of socket.
*
* \retval 0 when connection is succesful.
* \retval 1 when there is an error.
*/
static int handle_audiosocket_connection(const char *server,
const struct ast_sockaddr addr, const int netsockfd)
{
struct pollfd pfds[1];
int res, conresult;
socklen_t reslen;
reslen = sizeof(conresult);
pfds[0].fd = netsockfd;
pfds[0].events = POLLOUT;
while ((res = ast_poll(pfds, 1, MAX_CONNECT_TIMEOUT_MSEC)) != 1) {
if (errno != EINTR) {
if (!res) {
ast_log(LOG_WARNING, "AudioSocket connection to '%s' timed"
"out after MAX_CONNECT_TIMEOUT_MSEC (%d) milliseconds.\n",
server, MAX_CONNECT_TIMEOUT_MSEC);
} else {
ast_log(LOG_WARNING, "Connect to '%s' failed: %s\n", server,
strerror(errno));
}
return -1;
}
}
if (getsockopt(pfds[0].fd, SOL_SOCKET, SO_ERROR, &conresult, &reslen) < 0) {
ast_log(LOG_WARNING, "Connection to %s failed with error: %s\n",
ast_sockaddr_stringify(&addr), strerror(errno));
return -1;
}
if (conresult) {
ast_log(LOG_WARNING, "Connecting to '%s' failed for url '%s': %s\n",
ast_sockaddr_stringify(&addr), server, strerror(conresult));
return -1;
}
return 0;
}
const int ast_audiosocket_connect(const char *server, struct ast_channel *chan)
{
int s = -1;
struct ast_sockaddr *addrs = NULL;
int num_addrs = 0, i = 0;
if (chan && ast_autoservice_start(chan) < 0) {
ast_log(LOG_WARNING, "Failed to start autoservice for channel "
"%s\n", ast_channel_name(chan));
goto end;
}
if (ast_strlen_zero(server)) {
ast_log(LOG_ERROR, "No AudioSocket server provided\n");
goto end;
}
if (!(num_addrs = ast_sockaddr_resolve(&addrs, server, PARSE_PORT_REQUIRE,
AST_AF_UNSPEC))) {
ast_log(LOG_ERROR, "Failed to resolve AudioSocket service using %s - "
"requires a valid hostname and port\n", server);
goto end;
}
/* Connect to AudioSocket service */
for (i = 0; i < num_addrs; i++) {
if (!ast_sockaddr_port(&addrs[i])) {
/* If there's no port, other addresses should have the
* same problem. Stop here.
*/
ast_log(LOG_ERROR, "No port provided for %s\n",
ast_sockaddr_stringify(&addrs[i]));
s = -1;
goto end;
}
if ((s = ast_socket_nonblock(addrs[i].ss.ss_family, SOCK_STREAM,
IPPROTO_TCP)) < 0) {
ast_log(LOG_WARNING, "Unable to create socket: %s\n", strerror(errno));
continue;
}
if (ast_connect(s, &addrs[i]) && errno == EINPROGRESS) {
if (handle_audiosocket_connection(server, addrs[i], s)) {
close(s);
continue;
}
} else {
ast_log(LOG_ERROR, "Connection to %s failed with unexpected error: %s\n",
ast_sockaddr_stringify(&addrs[i]), strerror(errno));
close(s);
s = -1;
}
break;
}
end:
if (addrs) {
ast_free(addrs);
}
if (chan && ast_autoservice_stop(chan) < 0) {
ast_log(LOG_WARNING, "Failed to stop autoservice for channel %s\n",
ast_channel_name(chan));
close(s);
return -1;
}
if (i == num_addrs) {
ast_log(LOG_ERROR, "Failed to connect to AudioSocket service\n");
close(s);
return -1;
}
return s;
}
const int ast_audiosocket_init(const int svc, const char *id)
{
uuid_t uu;
int ret = 0;
uint8_t buf[3 + 16];
if (ast_strlen_zero(id)) {
ast_log(LOG_ERROR, "No UUID for AudioSocket\n");
return -1;
}
if (uuid_parse(id, uu)) {
ast_log(LOG_ERROR, "Failed to parse UUID '%s'\n", id);
return -1;
}
buf[0] = 0x01;
buf[1] = 0x00;
buf[2] = 0x10;
memcpy(buf + 3, uu, 16);
if (write(svc, buf, 3 + 16) != 3 + 16) {
ast_log(LOG_WARNING, "Failed to write data to AudioSocket\n");
ret = -1;
}
return ret;
}
const int ast_audiosocket_send_frame(const int svc, const struct ast_frame *f)
{
int ret = 0;
uint8_t kind = 0x10; /* always 16-bit, 8kHz signed linear mono, for now */
uint8_t *p;
uint8_t buf[3 + f->datalen];
p = buf;
*(p++) = kind;
*(p++) = f->datalen >> 8;
*(p++) = f->datalen & 0xff;
memcpy(p, f->data.ptr, f->datalen);
if (write(svc, buf, 3 + f->datalen) != 3 + f->datalen) {
ast_log(LOG_WARNING, "Failed to write data to AudioSocket\n");
ret = -1;
}
return ret;
}
struct ast_frame *ast_audiosocket_receive_frame(const int svc)
{
int i = 0, n = 0, ret = 0, not_audio = 0;
struct ast_frame f = {
.frametype = AST_FRAME_VOICE,
.subclass.format = ast_format_slin,
.src = "AudioSocket",
.mallocd = AST_MALLOCD_DATA,
};
uint8_t kind;
uint8_t len_high;
uint8_t len_low;
uint16_t len = 0;
uint8_t *data;
n = read(svc, &kind, 1);
if (n < 0 && errno == EAGAIN) {
return &ast_null_frame;
}
if (n == 0) {
return &ast_null_frame;
}
if (n != 1) {
ast_log(LOG_WARNING, "Failed to read type header from AudioSocket\n");
return NULL;
}
if (kind == 0x00) {
/* AudioSocket ended by remote */
return NULL;
}
if (kind != 0x10) {
/* read but ignore non-audio message */
ast_log(LOG_WARNING, "Received non-audio AudioSocket message\n");
not_audio = 1;
}
n = read(svc, &len_high, 1);
if (n != 1) {
ast_log(LOG_WARNING, "Failed to read data length from AudioSocket\n");
return NULL;
}
len += len_high * 256;
n = read(svc, &len_low, 1);
if (n != 1) {
ast_log(LOG_WARNING, "Failed to read data length from AudioSocket\n");
return NULL;
}
len += len_low;
if (len < 1) {
return &ast_null_frame;
}
data = ast_malloc(len);
if (!data) {
ast_log(LOG_ERROR, "Failed to allocate for data from AudioSocket\n");
return NULL;
}
ret = 0;
n = 0;
i = 0;
while (i < len) {
n = read(svc, data + i, len - i);
if (n < 0) {
ast_log(LOG_ERROR, "Failed to read data from AudioSocket\n");
ret = n;
break;
}
if (n == 0) {
ast_log(LOG_ERROR, "Insufficient data read from AudioSocket\n");
ret = -1;
break;
}
i += n;
}
if (ret != 0) {
ast_free(data);
return NULL;
}
if (not_audio) {
ast_free(data);
return &ast_null_frame;
}
f.data.ptr = data;
f.datalen = len;
f.samples = len / 2;
/* The frame steals data, so it doesn't need to be freed here */
return ast_frisolate(&f);
}
static int load_module(void)
{
ast_verb(1, "Loading AudioSocket Support module\n");
return AST_MODULE_LOAD_SUCCESS;
}
static int unload_module(void)
{
ast_verb(1, "Unloading AudioSocket Support module\n");
return AST_MODULE_LOAD_SUCCESS;
}
AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_GLOBAL_SYMBOLS | AST_MODFLAG_LOAD_ORDER, "AudioSocket support",
.support_level = AST_MODULE_SUPPORT_EXTENDED,
.load = load_module,
.unload = unload_module,
.load_pri = AST_MODPRI_CHANNEL_DEPEND,
);