gstreamer/gst/rtsp-server/rtsp-server.c
Fabian Deutsch 6ef7c966ae Add a signal for newly connected clients.
Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
2011-05-17 09:44:14 +02:00

916 lines
24 KiB
C

/* GStreamer
* Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#include <stdio.h>
#include <stdlib.h>
#include <unistd.h>
#include <errno.h>
#include <string.h>
#include <sys/time.h>
#include <sys/types.h>
#include <netinet/in.h>
#include <netdb.h>
#include <sys/socket.h>
#include <sys/wait.h>
#include <fcntl.h>
#include <arpa/inet.h>
#include <sys/ioctl.h>
#include "rtsp-server.h"
#include "rtsp-client.h"
#define DEFAULT_ADDRESS "0.0.0.0"
/* #define DEFAULT_ADDRESS "::0" */
#define DEFAULT_SERVICE "8554"
#define DEFAULT_BACKLOG 5
/* Define to use the SO_LINGER option so that the server sockets can be resused
* sooner. Disabled for now because it is not very well implemented by various
* OSes and it causes clients to fail to read the TEARDOWN response. */
#undef USE_SOLINGER
enum
{
PROP_0,
PROP_ADDRESS,
PROP_SERVICE,
PROP_BACKLOG,
PROP_SESSION_POOL,
PROP_MEDIA_MAPPING,
PROP_LAST
};
enum
{
SIGNAL_CLIENT_CONNECTED,
SIGNAL_LAST
};
G_DEFINE_TYPE (GstRTSPServer, gst_rtsp_server, G_TYPE_OBJECT);
GST_DEBUG_CATEGORY_STATIC (rtsp_server_debug);
#define GST_CAT_DEFAULT rtsp_server_debug
static guint gst_rtsp_server_signals[SIGNAL_LAST] = { 0 };
static void gst_rtsp_server_get_property (GObject * object, guint propid,
GValue * value, GParamSpec * pspec);
static void gst_rtsp_server_set_property (GObject * object, guint propid,
const GValue * value, GParamSpec * pspec);
static void gst_rtsp_server_finalize (GObject * object);
static GstRTSPClient *default_create_client (GstRTSPServer * server);
static gboolean default_accept_client (GstRTSPServer * server,
GstRTSPClient * client, GIOChannel * channel);
static void
gst_rtsp_server_class_init (GstRTSPServerClass * klass)
{
GObjectClass *gobject_class;
gobject_class = G_OBJECT_CLASS (klass);
gobject_class->get_property = gst_rtsp_server_get_property;
gobject_class->set_property = gst_rtsp_server_set_property;
gobject_class->finalize = gst_rtsp_server_finalize;
/**
* GstRTSPServer::address
*
* The address of the server. This is the address where the server will
* listen on.
*/
g_object_class_install_property (gobject_class, PROP_ADDRESS,
g_param_spec_string ("address", "Address",
"The address the server uses to listen on", DEFAULT_ADDRESS,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstRTSPServer::service
*
* The service of the server. This is either a string with the service name or
* a port number (as a string) the server will listen on.
*/
g_object_class_install_property (gobject_class, PROP_SERVICE,
g_param_spec_string ("service", "Service",
"The service or port number the server uses to listen on",
DEFAULT_SERVICE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstRTSPServer::backlog
*
* The backlog argument defines the maximum length to which the queue of
* pending connections for the server may grow. If a connection request arrives
* when the queue is full, the client may receive an error with an indication of
* ECONNREFUSED or, if the underlying protocol supports retransmission, the
* request may be ignored so that a later reattempt at connection succeeds.
*/
g_object_class_install_property (gobject_class, PROP_BACKLOG,
g_param_spec_int ("backlog", "Backlog",
"The maximum length to which the queue "
"of pending connections may grow", 0, G_MAXINT, DEFAULT_BACKLOG,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstRTSPServer::session-pool
*
* The session pool of the server. By default each server has a separate
* session pool but sessions can be shared between servers by setting the same
* session pool on multiple servers.
*/
g_object_class_install_property (gobject_class, PROP_SESSION_POOL,
g_param_spec_object ("session-pool", "Session Pool",
"The session pool to use for client session",
GST_TYPE_RTSP_SESSION_POOL,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstRTSPServer::media-mapping
*
* The media mapping to use for this server. By default the server has no
* media mapping and thus cannot map urls to media streams.
*/
g_object_class_install_property (gobject_class, PROP_MEDIA_MAPPING,
g_param_spec_object ("media-mapping", "Media Mapping",
"The media mapping to use for client session",
GST_TYPE_RTSP_MEDIA_MAPPING,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
gst_rtsp_server_signals[SIGNAL_CLIENT_CONNECTED] =
g_signal_new ("client-connected", G_TYPE_FROM_CLASS (gobject_class),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPServerClass, client_connected),
NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
gst_rtsp_client_get_type ());
klass->create_client = default_create_client;
klass->accept_client = default_accept_client;
GST_DEBUG_CATEGORY_INIT (rtsp_server_debug, "rtspserver", 0, "GstRTSPServer");
}
static void
gst_rtsp_server_init (GstRTSPServer * server)
{
server->lock = g_mutex_new ();
server->address = g_strdup (DEFAULT_ADDRESS);
server->service = g_strdup (DEFAULT_SERVICE);
server->backlog = DEFAULT_BACKLOG;
server->session_pool = gst_rtsp_session_pool_new ();
server->media_mapping = gst_rtsp_media_mapping_new ();
}
static void
gst_rtsp_server_finalize (GObject * object)
{
GstRTSPServer *server = GST_RTSP_SERVER (object);
GST_DEBUG_OBJECT (server, "finalize server");
g_free (server->address);
g_free (server->service);
g_object_unref (server->session_pool);
g_object_unref (server->media_mapping);
if (server->auth)
g_object_unref (server->auth);
g_mutex_free (server->lock);
G_OBJECT_CLASS (gst_rtsp_server_parent_class)->finalize (object);
}
/**
* gst_rtsp_server_new:
*
* Create a new #GstRTSPServer instance.
*/
GstRTSPServer *
gst_rtsp_server_new (void)
{
GstRTSPServer *result;
result = g_object_new (GST_TYPE_RTSP_SERVER, NULL);
return result;
}
/**
* gst_rtsp_server_set_address:
* @server: a #GstRTSPServer
* @address: the address
*
* Configure @server to accept connections on the given address.
*
* This function must be called before the server is bound.
*/
void
gst_rtsp_server_set_address (GstRTSPServer * server, const gchar * address)
{
g_return_if_fail (GST_IS_RTSP_SERVER (server));
g_return_if_fail (address != NULL);
GST_RTSP_SERVER_LOCK (server);
g_free (server->address);
server->address = g_strdup (address);
GST_RTSP_SERVER_UNLOCK (server);
}
/**
* gst_rtsp_server_get_address:
* @server: a #GstRTSPServer
*
* Get the address on which the server will accept connections.
*
* Returns: the server address. g_free() after usage.
*/
gchar *
gst_rtsp_server_get_address (GstRTSPServer * server)
{
gchar *result;
g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
GST_RTSP_SERVER_LOCK (server);
result = g_strdup (server->address);
GST_RTSP_SERVER_UNLOCK (server);
return result;
}
/**
* gst_rtsp_server_set_service:
* @server: a #GstRTSPServer
* @service: the service
*
* Configure @server to accept connections on the given service.
* @service should be a string containing the service name (see services(5)) or
* a string containing a port number between 1 and 65535.
*
* This function must be called before the server is bound.
*/
void
gst_rtsp_server_set_service (GstRTSPServer * server, const gchar * service)
{
g_return_if_fail (GST_IS_RTSP_SERVER (server));
g_return_if_fail (service != NULL);
GST_RTSP_SERVER_LOCK (server);
g_free (server->service);
server->service = g_strdup (service);
GST_RTSP_SERVER_UNLOCK (server);
}
/**
* gst_rtsp_server_get_service:
* @server: a #GstRTSPServer
*
* Get the service on which the server will accept connections.
*
* Returns: the service. use g_free() after usage.
*/
gchar *
gst_rtsp_server_get_service (GstRTSPServer * server)
{
gchar *result;
g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
GST_RTSP_SERVER_LOCK (server);
result = g_strdup (server->service);
GST_RTSP_SERVER_UNLOCK (server);
return result;
}
/**
* gst_rtsp_server_set_backlog:
* @server: a #GstRTSPServer
* @backlog: the backlog
*
* configure the maximum amount of requests that may be queued for the
* server.
*
* This function must be called before the server is bound.
*/
void
gst_rtsp_server_set_backlog (GstRTSPServer * server, gint backlog)
{
g_return_if_fail (GST_IS_RTSP_SERVER (server));
GST_RTSP_SERVER_LOCK (server);
server->backlog = backlog;
GST_RTSP_SERVER_UNLOCK (server);
}
/**
* gst_rtsp_server_get_backlog:
* @server: a #GstRTSPServer
*
* The maximum amount of queued requests for the server.
*
* Returns: the server backlog.
*/
gint
gst_rtsp_server_get_backlog (GstRTSPServer * server)
{
gint result;
g_return_val_if_fail (GST_IS_RTSP_SERVER (server), -1);
GST_RTSP_SERVER_LOCK (server);
result = server->backlog;
GST_RTSP_SERVER_UNLOCK (server);
return result;
}
/**
* gst_rtsp_server_set_session_pool:
* @server: a #GstRTSPServer
* @pool: a #GstRTSPSessionPool
*
* configure @pool to be used as the session pool of @server.
*/
void
gst_rtsp_server_set_session_pool (GstRTSPServer * server,
GstRTSPSessionPool * pool)
{
GstRTSPSessionPool *old;
g_return_if_fail (GST_IS_RTSP_SERVER (server));
if (pool)
g_object_ref (pool);
GST_RTSP_SERVER_LOCK (server);
old = server->session_pool;
server->session_pool = pool;
GST_RTSP_SERVER_UNLOCK (server);
if (old)
g_object_unref (old);
}
/**
* gst_rtsp_server_get_session_pool:
* @server: a #GstRTSPServer
*
* Get the #GstRTSPSessionPool used as the session pool of @server.
*
* Returns: the #GstRTSPSessionPool used for sessions. g_object_unref() after
* usage.
*/
GstRTSPSessionPool *
gst_rtsp_server_get_session_pool (GstRTSPServer * server)
{
GstRTSPSessionPool *result;
g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
GST_RTSP_SERVER_LOCK (server);
if ((result = server->session_pool))
g_object_ref (result);
GST_RTSP_SERVER_UNLOCK (server);
return result;
}
/**
* gst_rtsp_server_set_media_mapping:
* @server: a #GstRTSPServer
* @mapping: a #GstRTSPMediaMapping
*
* configure @mapping to be used as the media mapping of @server.
*/
void
gst_rtsp_server_set_media_mapping (GstRTSPServer * server,
GstRTSPMediaMapping * mapping)
{
GstRTSPMediaMapping *old;
g_return_if_fail (GST_IS_RTSP_SERVER (server));
if (mapping)
g_object_ref (mapping);
GST_RTSP_SERVER_LOCK (server);
old = server->media_mapping;
server->media_mapping = mapping;
GST_RTSP_SERVER_UNLOCK (server);
if (old)
g_object_unref (old);
}
/**
* gst_rtsp_server_get_media_mapping:
* @server: a #GstRTSPServer
*
* Get the #GstRTSPMediaMapping used as the media mapping of @server.
*
* Returns: the #GstRTSPMediaMapping of @server. g_object_unref() after
* usage.
*/
GstRTSPMediaMapping *
gst_rtsp_server_get_media_mapping (GstRTSPServer * server)
{
GstRTSPMediaMapping *result;
g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
GST_RTSP_SERVER_LOCK (server);
if ((result = server->media_mapping))
g_object_ref (result);
GST_RTSP_SERVER_UNLOCK (server);
return result;
}
/**
* gst_rtsp_server_set_auth:
* @server: a #GstRTSPServer
* @auth: a #GstRTSPAuth
*
* configure @auth to be used as the authentication manager of @server.
*/
void
gst_rtsp_server_set_auth (GstRTSPServer * server, GstRTSPAuth * auth)
{
GstRTSPAuth *old;
g_return_if_fail (GST_IS_RTSP_SERVER (server));
if (auth)
g_object_ref (auth);
GST_RTSP_SERVER_LOCK (server);
old = server->auth;
server->auth = auth;
GST_RTSP_SERVER_UNLOCK (server);
if (old)
g_object_unref (old);
}
/**
* gst_rtsp_server_get_auth:
* @server: a #GstRTSPServer
*
* Get the #GstRTSPAuth used as the authentication manager of @server.
*
* Returns: the #GstRTSPAuth of @server. g_object_unref() after
* usage.
*/
GstRTSPAuth *
gst_rtsp_server_get_auth (GstRTSPServer * server)
{
GstRTSPAuth *result;
g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
GST_RTSP_SERVER_LOCK (server);
if ((result = server->auth))
g_object_ref (result);
GST_RTSP_SERVER_UNLOCK (server);
return result;
}
static void
gst_rtsp_server_get_property (GObject * object, guint propid,
GValue * value, GParamSpec * pspec)
{
GstRTSPServer *server = GST_RTSP_SERVER (object);
switch (propid) {
case PROP_ADDRESS:
g_value_take_string (value, gst_rtsp_server_get_address (server));
break;
case PROP_SERVICE:
g_value_take_string (value, gst_rtsp_server_get_service (server));
break;
case PROP_BACKLOG:
g_value_set_int (value, gst_rtsp_server_get_backlog (server));
break;
case PROP_SESSION_POOL:
g_value_take_object (value, gst_rtsp_server_get_session_pool (server));
break;
case PROP_MEDIA_MAPPING:
g_value_take_object (value, gst_rtsp_server_get_media_mapping (server));
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
}
}
static void
gst_rtsp_server_set_property (GObject * object, guint propid,
const GValue * value, GParamSpec * pspec)
{
GstRTSPServer *server = GST_RTSP_SERVER (object);
switch (propid) {
case PROP_ADDRESS:
gst_rtsp_server_set_address (server, g_value_get_string (value));
break;
case PROP_SERVICE:
gst_rtsp_server_set_service (server, g_value_get_string (value));
break;
case PROP_BACKLOG:
gst_rtsp_server_set_backlog (server, g_value_get_int (value));
break;
case PROP_SESSION_POOL:
gst_rtsp_server_set_session_pool (server, g_value_get_object (value));
break;
case PROP_MEDIA_MAPPING:
gst_rtsp_server_set_media_mapping (server, g_value_get_object (value));
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
}
}
/**
* gst_rtsp_server_get_io_channel:
* @server: a #GstRTSPServer
*
* Create a #GIOChannel for @server. The io channel will listen on the
* configured service.
*
* Returns: the GIOChannel for @server or NULL when an error occured.
*/
GIOChannel *
gst_rtsp_server_get_io_channel (GstRTSPServer * server)
{
GIOChannel *channel;
int ret, sockfd = -1;
struct addrinfo hints;
struct addrinfo *result, *rp;
#ifdef USE_SOLINGER
struct linger linger;
#endif
g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
memset (&hints, 0, sizeof (struct addrinfo));
hints.ai_family = AF_UNSPEC; /* Allow IPv4 or IPv6 */
hints.ai_socktype = SOCK_STREAM; /* stream socket */
hints.ai_flags = AI_PASSIVE | AI_CANONNAME; /* For wildcard IP address */
hints.ai_protocol = 0; /* Any protocol */
hints.ai_canonname = NULL;
hints.ai_addr = NULL;
hints.ai_next = NULL;
GST_DEBUG_OBJECT (server, "getting address info of %s/%s", server->address,
server->service);
GST_RTSP_SERVER_LOCK (server);
/* resolve the server IP address */
if ((ret =
getaddrinfo (server->address, server->service, &hints, &result)) != 0)
goto no_address;
/* create server socket, we loop through all the addresses until we manage to
* create a socket and bind. */
for (rp = result; rp; rp = rp->ai_next) {
sockfd = socket (rp->ai_family, rp->ai_socktype, rp->ai_protocol);
if (sockfd == -1) {
GST_DEBUG_OBJECT (server, "failed to make socket (%s), try next",
g_strerror (errno));
continue;
}
/* make address reusable */
ret = 1;
if (setsockopt (sockfd, SOL_SOCKET, SO_REUSEADDR,
(void *) &ret, sizeof (ret)) < 0) {
/* warn but try to bind anyway */
GST_WARNING_OBJECT (server, "failed to reuse socker (%s)",
g_strerror (errno));
}
if (bind (sockfd, rp->ai_addr, rp->ai_addrlen) == 0) {
GST_DEBUG_OBJECT (server, "bind on %s", rp->ai_canonname);
break;
}
GST_DEBUG_OBJECT (server, "failed to bind socket (%s), try next",
g_strerror (errno));
close (sockfd);
sockfd = -1;
}
freeaddrinfo (result);
if (sockfd == -1)
goto no_socket;
GST_DEBUG_OBJECT (server, "opened sending server socket with fd %d", sockfd);
/* keep connection alive; avoids SIGPIPE during write */
ret = 1;
if (setsockopt (sockfd, SOL_SOCKET, SO_KEEPALIVE,
(void *) &ret, sizeof (ret)) < 0)
goto keepalive_failed;
#ifdef USE_SOLINGER
/* make sure socket is reset 5 seconds after close. This ensure that we can
* reuse the socket quickly while still having a chance to send data to the
* client. */
linger.l_onoff = 1;
linger.l_linger = 5;
if (setsockopt (sockfd, SOL_SOCKET, SO_LINGER,
(void *) &linger, sizeof (linger)) < 0)
goto linger_failed;
#endif
/* set the server socket to nonblocking */
fcntl (sockfd, F_SETFL, O_NONBLOCK);
GST_DEBUG_OBJECT (server, "listening on server socket %d with queue of %d",
sockfd, server->backlog);
if (listen (sockfd, server->backlog) == -1)
goto listen_failed;
GST_DEBUG_OBJECT (server,
"listened on server socket %d, returning from connection setup", sockfd);
/* create IO channel for the socket */
channel = g_io_channel_unix_new (sockfd);
g_io_channel_set_close_on_unref (channel, TRUE);
GST_INFO_OBJECT (server, "listening on service %s", server->service);
GST_RTSP_SERVER_UNLOCK (server);
return channel;
/* ERRORS */
no_address:
{
GST_ERROR_OBJECT (server, "failed to resolve address: %s",
gai_strerror (ret));
goto close_error;
}
no_socket:
{
GST_ERROR_OBJECT (server, "failed to create socket: %s",
g_strerror (errno));
goto close_error;
}
keepalive_failed:
{
GST_ERROR_OBJECT (server, "failed to configure keepalive socket: %s",
g_strerror (errno));
goto close_error;
}
#ifdef USE_SOLINGER
linger_failed:
{
GST_ERROR_OBJECT (server, "failed to no linger socket: %s",
g_strerror (errno));
goto close_error;
}
#endif
listen_failed:
{
GST_ERROR_OBJECT (server, "failed to listen on socket: %s",
g_strerror (errno));
goto close_error;
}
close_error:
{
if (sockfd >= 0) {
close (sockfd);
}
GST_RTSP_SERVER_UNLOCK (server);
return NULL;
}
}
static void
unmanage_client (GstRTSPClient * client, GstRTSPServer * server)
{
GST_DEBUG_OBJECT (server, "unmanage client %p", client);
gst_rtsp_client_set_server (client, NULL);
GST_RTSP_SERVER_LOCK (server);
server->clients = g_list_remove (server->clients, client);
GST_RTSP_SERVER_UNLOCK (server);
g_object_unref (client);
}
/* add the client to the active list of clients, takes ownership of
* the client */
static void
manage_client (GstRTSPServer * server, GstRTSPClient * client)
{
GST_DEBUG_OBJECT (server, "manage client %p", client);
gst_rtsp_client_set_server (client, server);
GST_RTSP_SERVER_LOCK (server);
g_signal_connect (client, "closed", (GCallback) unmanage_client, server);
server->clients = g_list_prepend (server->clients, client);
GST_RTSP_SERVER_UNLOCK (server);
}
static GstRTSPClient *
default_create_client (GstRTSPServer * server)
{
GstRTSPClient *client;
/* a new client connected, create a session to handle the client. */
client = gst_rtsp_client_new ();
/* set the session pool that this client should use */
GST_RTSP_SERVER_LOCK (server);
gst_rtsp_client_set_session_pool (client, server->session_pool);
/* set the media mapping that this client should use */
gst_rtsp_client_set_media_mapping (client, server->media_mapping);
/* set authentication manager */
gst_rtsp_client_set_auth (client, server->auth);
GST_RTSP_SERVER_UNLOCK (server);
return client;
}
/* default method for creating a new client object in the server to accept and
* handle a client connection on this server */
static gboolean
default_accept_client (GstRTSPServer * server, GstRTSPClient * client,
GIOChannel * channel)
{
/* accept connections for that client, this function returns after accepting
* the connection and will run the remainder of the communication with the
* client asyncronously. */
if (!gst_rtsp_client_accept (client, channel))
goto accept_failed;
return TRUE;
/* ERRORS */
accept_failed:
{
GST_ERROR_OBJECT (server,
"Could not accept client on server : %s (%d)", g_strerror (errno),
errno);
return FALSE;
}
}
/**
* gst_rtsp_server_io_func:
* @channel: a #GIOChannel
* @condition: the condition on @source
*
* A default #GIOFunc that creates a new #GstRTSPClient to accept and handle a
* new connection on @channel or @server.
*
* Returns: TRUE if the source could be connected, FALSE if an error occured.
*/
gboolean
gst_rtsp_server_io_func (GIOChannel * channel, GIOCondition condition,
GstRTSPServer * server)
{
gboolean result;
GstRTSPClient *client = NULL;
GstRTSPServerClass *klass;
if (condition & G_IO_IN) {
klass = GST_RTSP_SERVER_GET_CLASS (server);
if (klass->create_client)
client = klass->create_client (server);
if (client == NULL)
goto client_failed;
/* a new client connected, create a client object to handle the client. */
if (klass->accept_client)
result = klass->accept_client (server, client, channel);
if (!result)
goto accept_failed;
/* manage the client connection */
manage_client (server, client);
g_signal_emit (server, gst_rtsp_server_signals[SIGNAL_CLIENT_CONNECTED], 0,
client);
} else {
GST_WARNING_OBJECT (server, "received unknown event %08x", condition);
}
return TRUE;
/* ERRORS */
client_failed:
{
GST_ERROR_OBJECT (server, "failed to create a client");
return FALSE;
}
accept_failed:
{
GST_ERROR_OBJECT (server, "failed to accept client");
gst_object_unref (client);
return FALSE;
}
}
static void
watch_destroyed (GstRTSPServer * server)
{
GST_DEBUG_OBJECT (server, "source destroyed");
g_object_unref (server);
}
/**
* gst_rtsp_server_create_watch:
* @server: a #GstRTSPServer
*
* Create a #GSource for @server. The new source will have a default
* #GIOFunc of gst_rtsp_server_io_func().
*
* Returns: the #GSource for @server or NULL when an error occured.
*/
GSource *
gst_rtsp_server_create_watch (GstRTSPServer * server)
{
GIOChannel *channel;
GSource *source;
g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
channel = gst_rtsp_server_get_io_channel (server);
if (channel == NULL)
goto no_channel;
/* create a watch for reads (new connections) and possible errors */
source = g_io_create_watch (channel, G_IO_IN |
G_IO_ERR | G_IO_HUP | G_IO_NVAL);
g_io_channel_unref (channel);
/* configure the callback */
g_source_set_callback (source,
(GSourceFunc) gst_rtsp_server_io_func, g_object_ref (server),
(GDestroyNotify) watch_destroyed);
return source;
no_channel:
{
GST_ERROR_OBJECT (server, "failed to create IO channel");
return NULL;
}
}
/**
* gst_rtsp_server_attach:
* @server: a #GstRTSPServer
* @context: a #GMainContext
*
* Attaches @server to @context. When the mainloop for @context is run, the
* server will be dispatched. When @context is NULL, the default context will be
* used).
*
* This function should be called when the server properties and urls are fully
* configured and the server is ready to start.
*
* Returns: the ID (greater than 0) for the source within the GMainContext.
*/
guint
gst_rtsp_server_attach (GstRTSPServer * server, GMainContext * context)
{
guint res;
GSource *source;
g_return_val_if_fail (GST_IS_RTSP_SERVER (server), 0);
source = gst_rtsp_server_create_watch (server);
if (source == NULL)
goto no_source;
res = g_source_attach (source, context);
g_source_unref (source);
return res;
/* ERRORS */
no_source:
{
GST_ERROR_OBJECT (server, "failed to create watch");
return 0;
}
}