*
* 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.
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
*/
-#include <sys/ioctl.h>
+#include <stdlib.h>
+#include <string.h>
#include "rtsp-server.h"
#include "rtsp-client.h"
#define DEFAULT_ADDRESS "0.0.0.0"
+#define DEFAULT_BOUND_PORT -1
/* #define DEFAULT_ADDRESS "::0" */
#define DEFAULT_SERVICE "8554"
#define DEFAULT_BACKLOG 5
+#define DEFAULT_MAX_THREADS 0
/* 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
PROP_0,
PROP_ADDRESS,
PROP_SERVICE,
+ PROP_BOUND_PORT,
PROP_BACKLOG,
PROP_SESSION_POOL,
PROP_MEDIA_MAPPING,
+ PROP_MAX_THREADS,
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
+typedef struct _ClientContext ClientContext;
+
+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_accept_client (GstRTSPServer * server,
- GIOChannel * channel);
+static gpointer do_loop (ClientContext * ctx);
+static GstRTSPClient *default_create_client (GstRTSPServer * server);
+static gboolean default_accept_client (GstRTSPServer * server,
+ GstRTSPClient * client, GSocket * socket, GError ** error);
static void
gst_rtsp_server_class_init (GstRTSPServerClass * klass)
gobject_class->finalize = gst_rtsp_server_finalize;
/**
- * GstRTSPServer::address
+ * GstRTSPServer::address:
*
* The address of the server. This is the address where the server will
* listen on.
"The address the server uses to listen on", DEFAULT_ADDRESS,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
- * GstRTSPServer::service
+ * 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.
"The service or port number the server uses to listen on",
DEFAULT_SERVICE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
- * GstRTSPServer::backlog
+ * GstRTSPServer::bound-port:
+ *
+ * The actual port the server is listening on. Can be used to retrieve the
+ * port number when the server is started on port 0, which means bind to a
+ * random port. Set to -1 if the server has not been bound yet.
+ */
+ g_object_class_install_property (gobject_class, PROP_BOUND_PORT,
+ g_param_spec_int ("bound-port", "Bound port",
+ "The port number the server is listening on",
+ -1, G_MAXUINT16, DEFAULT_BOUND_PORT,
+ G_PARAM_READABLE | 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
"of pending connections may grow", 0, G_MAXINT, DEFAULT_BACKLOG,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
- * GstRTSPServer::session-pool
+ * 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
GST_TYPE_RTSP_SESSION_POOL,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
- * GstRTSPServer::media-mapping
+ * 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.
"The media mapping to use for client session",
GST_TYPE_RTSP_MEDIA_MAPPING,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
-
+ /**
+ * GstRTSPServer::max-threads:
+ *
+ * The maximum amount of threads to use for client connections. A value of
+ * 0 means to use only the mainloop, -1 means an unlimited amount of
+ * threads.
+ */
+ g_object_class_install_property (gobject_class, PROP_MAX_THREADS,
+ g_param_spec_int ("max-threads", "Max Threads",
+ "The maximum amount of threads to use for client connections "
+ "(0 = only mainloop, -1 = unlimited)", -1, G_MAXINT,
+ DEFAULT_MAX_THREADS, 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;
+ klass->pool = g_thread_pool_new ((GFunc) do_loop, klass, -1, FALSE, NULL);
+
GST_DEBUG_CATEGORY_INIT (rtsp_server_debug, "rtspserver", 0, "GstRTSPServer");
}
static void
gst_rtsp_server_init (GstRTSPServer * server)
{
+ g_mutex_init (&server->lock);
server->address = g_strdup (DEFAULT_ADDRESS);
server->service = g_strdup (DEFAULT_SERVICE);
+ server->socket = NULL;
server->backlog = DEFAULT_BACKLOG;
server->session_pool = gst_rtsp_session_pool_new ();
server->media_mapping = gst_rtsp_media_mapping_new ();
{
GstRTSPServer *server = GST_RTSP_SERVER (object);
+ GST_DEBUG_OBJECT (server, "finalize server");
+
g_free (server->address);
g_free (server->service);
+ if (server->socket)
+ g_object_unref (server->socket);
g_object_unref (server->session_pool);
g_object_unref (server->media_mapping);
+
+ if (server->auth)
+ g_object_unref (server->auth);
+
+ g_mutex_clear (&server->lock);
+
+ G_OBJECT_CLASS (gst_rtsp_server_parent_class)->finalize (object);
}
/**
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);
}
/**
gchar *
gst_rtsp_server_get_address (GstRTSPServer * server)
{
+ gchar *result;
g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
- return g_strdup (server->address);
+ GST_RTSP_SERVER_LOCK (server);
+ result = g_strdup (server->address);
+ GST_RTSP_SERVER_UNLOCK (server);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_server_get_bound_port:
+ * @server: a #GstRTSPServer
+ *
+ * Get the port number where the server was bound to.
+ *
+ * Returns: the port number
+ */
+int
+gst_rtsp_server_get_bound_port (GstRTSPServer * server)
+{
+ GSocketAddress *address;
+ int result = -1;
+
+ g_return_val_if_fail (GST_IS_RTSP_SERVER (server), result);
+
+ GST_RTSP_SERVER_LOCK (server);
+ if (server->socket == NULL)
+ goto out;
+
+ address = g_socket_get_local_address (server->socket, NULL);
+ result = g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (address));
+ g_object_unref (address);
+
+out:
+ GST_RTSP_SERVER_UNLOCK (server);
+
+ return result;
}
/**
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);
}
/**
gchar *
gst_rtsp_server_get_service (GstRTSPServer * server)
{
+ gchar *result;
+
g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
- return g_strdup (server->service);
+ GST_RTSP_SERVER_LOCK (server);
+ result = g_strdup (server->service);
+ GST_RTSP_SERVER_UNLOCK (server);
+
+ return result;
}
/**
{
g_return_if_fail (GST_IS_RTSP_SERVER (server));
+ GST_RTSP_SERVER_LOCK (server);
server->backlog = backlog;
+ GST_RTSP_SERVER_UNLOCK (server);
}
/**
gint
gst_rtsp_server_get_backlog (GstRTSPServer * server)
{
+ gint result;
+
g_return_val_if_fail (GST_IS_RTSP_SERVER (server), -1);
- return server->backlog;
+ GST_RTSP_SERVER_LOCK (server);
+ result = server->backlog;
+ GST_RTSP_SERVER_UNLOCK (server);
+
+ return result;
}
/**
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 != pool) {
- if (pool)
- g_object_ref (pool);
- server->session_pool = pool;
- if (old)
- g_object_unref (old);
- }
+ if (old)
+ g_object_unref (old);
}
/**
*
* Get the #GstRTSPSessionPool used as the session pool of @server.
*
- * Returns: the #GstRTSPSessionPool used for sessions. g_object_unref() after
+ * Returns: (transfer full): the #GstRTSPSessionPool used for sessions. g_object_unref() after
* usage.
*/
GstRTSPSessionPool *
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;
}
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 != mapping) {
- if (mapping)
- g_object_ref (mapping);
- server->media_mapping = mapping;
- if (old)
- g_object_unref (old);
- }
+ if (old)
+ g_object_unref (old);
}
*
* Get the #GstRTSPMediaMapping used as the media mapping of @server.
*
- * Returns: the #GstRTSPMediaMapping of @server. g_object_unref() after
+ * Returns: (transfer full): the #GstRTSPMediaMapping of @server. g_object_unref() after
* usage.
*/
GstRTSPMediaMapping *
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;
}
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 != auth) {
- if (auth)
- g_object_ref (auth);
- server->auth = auth;
- if (old)
- g_object_unref (old);
- }
+ if (old)
+ g_object_unref (old);
}
*
* Get the #GstRTSPAuth used as the authentication manager of @server.
*
- * Returns: the #GstRTSPAuth of @server. g_object_unref() after
+ * Returns: (transfer full): the #GstRTSPAuth of @server. g_object_unref() after
* usage.
*/
GstRTSPAuth *
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;
}
+/**
+ * gst_rtsp_server_set_max_threads:
+ * @server: a #GstRTSPServer
+ * @max_threads: maximum threads
+ *
+ * Set the maximum threads used by the server to handle client requests.
+ * A value of 0 will use the server mainloop, a value of -1 will use an
+ * unlimited number of threads.
+ */
+void
+gst_rtsp_server_set_max_threads (GstRTSPServer * server, gint max_threads)
+{
+ g_return_if_fail (GST_IS_RTSP_SERVER (server));
+
+ GST_RTSP_SERVER_LOCK (server);
+ server->max_threads = max_threads;
+ GST_RTSP_SERVER_UNLOCK (server);
+}
+
+/**
+ * gst_rtsp_server_get_max_threads:
+ * @server: a #GstRTSPServer
+ *
+ * Get the maximum number of threads used for client connections.
+ * See gst_rtsp_server_set_max_threads().
+ *
+ * Returns: the maximum number of threads.
+ */
+gint
+gst_rtsp_server_get_max_threads (GstRTSPServer * server)
+{
+ gint res;
+
+ g_return_val_if_fail (GST_IS_RTSP_SERVER (server), -1);
+
+ GST_RTSP_SERVER_LOCK (server);
+ res = server->max_threads;
+ GST_RTSP_SERVER_UNLOCK (server);
+
+ return res;
+}
+
+
static void
gst_rtsp_server_get_property (GObject * object, guint propid,
GValue * value, GParamSpec * pspec)
case PROP_SERVICE:
g_value_take_string (value, gst_rtsp_server_get_service (server));
break;
+ case PROP_BOUND_PORT:
+ g_value_set_int (value, gst_rtsp_server_get_bound_port (server));
+ break;
case PROP_BACKLOG:
g_value_set_int (value, gst_rtsp_server_get_backlog (server));
break;
case PROP_MEDIA_MAPPING:
g_value_take_object (value, gst_rtsp_server_get_media_mapping (server));
break;
+ case PROP_MAX_THREADS:
+ g_value_set_int (value, gst_rtsp_server_get_max_threads (server));
+ break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
}
case PROP_MEDIA_MAPPING:
gst_rtsp_server_set_media_mapping (server, g_value_get_object (value));
break;
+ case PROP_MAX_THREADS:
+ gst_rtsp_server_set_max_threads (server, g_value_get_int (value));
+ break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
}
}
-/* Prepare a server socket for @server and make it listen on the configured port */
-static gboolean
-gst_rtsp_server_sink_init_send (GstRTSPServer * server)
+/**
+ * gst_rtsp_server_create_socket:
+ * @server: a #GstRTSPServer
+ * @cancellable: a #GCancellable
+ * @error: a #GError
+ *
+ * Create a #GSocket for @server. The socket will listen on the
+ * configured service.
+ *
+ * Returns: (transfer full): the #GSocket for @server or NULL when an error occured.
+ */
+GSocket *
+gst_rtsp_server_create_socket (GstRTSPServer * server,
+ GCancellable * cancellable, GError ** error)
{
- int ret, sockfd;
- struct addrinfo hints;
- struct addrinfo *result, *rp;
+ GSocketConnectable *conn;
+ GSocketAddressEnumerator *enumerator;
+ GSocket *socket = NULL;
#ifdef USE_SOLINGER
struct linger linger;
#endif
+ GError *sock_error = NULL;
+ GError *bind_error = NULL;
+ guint16 port;
- 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;
+ g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
+ GST_RTSP_SERVER_LOCK (server);
GST_DEBUG_OBJECT (server, "getting address info of %s/%s", server->address,
server->service);
/* resolve the server IP address */
- if ((ret =
- getaddrinfo (server->address, server->service, &hints, &result)) != 0)
- goto no_address;
+ port = atoi (server->service);
+ if (port != 0 || !strcmp (server->service, "0"))
+ conn = g_network_address_new (server->address, port);
+ else
+ conn = g_network_service_new (server->service, "tcp", server->address);
+
+ enumerator = g_socket_connectable_enumerate (conn);
+ g_object_unref (conn);
/* 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;
+ while (TRUE) {
+ GSocketAddress *sockaddr;
+
+ sockaddr =
+ g_socket_address_enumerator_next (enumerator, cancellable, error);
+ if (!sockaddr) {
+ if (!*error)
+ GST_DEBUG_OBJECT (server, "no more addresses %s",
+ *error ? (*error)->message : "");
+ else
+ GST_DEBUG_OBJECT (server, "failed to retrieve next address %s",
+ (*error)->message);
+ break;
}
- /* 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));
+ /* only keep the first error */
+ socket = g_socket_new (g_socket_address_get_family (sockaddr),
+ G_SOCKET_TYPE_STREAM, G_SOCKET_PROTOCOL_TCP,
+ sock_error ? NULL : &sock_error);
+
+ if (socket == NULL) {
+ GST_DEBUG_OBJECT (server, "failed to make socket (%s), try next",
+ sock_error->message);
+ continue;
}
- if (bind (sockfd, rp->ai_addr, rp->ai_addrlen) == 0) {
- GST_DEBUG_OBJECT (server, "bind on %s", rp->ai_canonname);
+ if (g_socket_bind (socket, sockaddr, TRUE, bind_error ? NULL : &bind_error)) {
+ g_object_unref (sockaddr);
break;
}
GST_DEBUG_OBJECT (server, "failed to bind socket (%s), try next",
- g_strerror (errno));
- close (sockfd);
+ bind_error->message);
+ g_object_unref (sockaddr);
+ g_object_unref (socket);
+ socket = NULL;
}
- freeaddrinfo (result);
+ g_object_unref (enumerator);
- if (rp == NULL)
+ if (socket == NULL)
goto no_socket;
- server->server_sock.fd = sockfd;
+ g_clear_error (&sock_error);
+ g_clear_error (&bind_error);
- GST_DEBUG_OBJECT (server, "opened sending server socket with fd %d",
- server->server_sock.fd);
+ GST_DEBUG_OBJECT (server, "opened sending server socket");
/* keep connection alive; avoids SIGPIPE during write */
- ret = 1;
- if (setsockopt (server->server_sock.fd, SOL_SOCKET, SO_KEEPALIVE,
- (void *) &ret, sizeof (ret)) < 0)
- goto keepalive_failed;
+ g_socket_set_keepalive (socket, TRUE);
+#if 0
#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 (server->server_sock.fd, SOL_SOCKET, SO_LINGER,
+ if (setsockopt (sockfd, SOL_SOCKET, SO_LINGER,
(void *) &linger, sizeof (linger)) < 0)
goto linger_failed;
#endif
+#endif
/* set the server socket to nonblocking */
- fcntl (server->server_sock.fd, F_SETFL, O_NONBLOCK);
+ g_socket_set_blocking (socket, FALSE);
+
+ /* set listen backlog */
+ g_socket_set_listen_backlog (socket, server->backlog);
- GST_DEBUG_OBJECT (server, "listening on server socket %d with queue of %d",
- server->server_sock.fd, server->backlog);
- if (listen (server->server_sock.fd, server->backlog) == -1)
+ if (!g_socket_listen (socket, error))
goto listen_failed;
- GST_DEBUG_OBJECT (server,
- "listened on server socket %d, returning from connection setup",
- server->server_sock.fd);
+ GST_DEBUG_OBJECT (server, "listening on server socket %p with queue of %d",
+ socket, server->backlog);
- GST_INFO_OBJECT (server, "listening on service %s", server->service);
+ GST_RTSP_SERVER_UNLOCK (server);
- return TRUE;
+ return socket;
/* ERRORS */
-no_address:
- {
- GST_ERROR_OBJECT (server, "failed to resolve address: %s",
- gai_strerror (ret));
- return FALSE;
- }
no_socket:
{
- GST_ERROR_OBJECT (server, "failed to create socket: %s",
- g_strerror (errno));
- return FALSE;
- }
-keepalive_failed:
- {
- GST_ERROR_OBJECT (server, "failed to configure keepalive socket: %s",
- g_strerror (errno));
+ GST_ERROR_OBJECT (server, "failed to create socket");
goto close_error;
}
+#if 0
#ifdef USE_SOLINGER
linger_failed:
{
goto close_error;
}
#endif
+#endif
listen_failed:
{
GST_ERROR_OBJECT (server, "failed to listen on socket: %s",
- g_strerror (errno));
+ (*error)->message);
goto close_error;
}
close_error:
{
- if (server->server_sock.fd >= 0) {
- close (server->server_sock.fd);
- server->server_sock.fd = -1;
+ if (socket)
+ g_object_unref (socket);
+
+ if (sock_error) {
+ if (error == NULL)
+ g_propagate_error (error, sock_error);
+ else
+ g_error_free (sock_error);
}
- return FALSE;
+ if (bind_error) {
+ if ((error == NULL) || (*error == NULL))
+ g_propagate_error (error, bind_error);
+ else
+ g_error_free (bind_error);
+ }
+ GST_RTSP_SERVER_UNLOCK (server);
+ return NULL;
+ }
+}
+
+struct _ClientContext
+{
+ GstRTSPServer *server;
+ GMainLoop *loop;
+ GMainContext *context;
+ GstRTSPClient *client;
+};
+
+static void
+free_client_context (ClientContext * ctx)
+{
+ g_main_context_unref (ctx->context);
+ if (ctx->loop)
+ g_main_loop_unref (ctx->loop);
+ g_object_unref (ctx->client);
+ g_slice_free (ClientContext, ctx);
+}
+
+static gpointer
+do_loop (ClientContext * ctx)
+{
+ GST_INFO ("enter mainloop");
+ g_main_loop_run (ctx->loop);
+ GST_INFO ("exit mainloop");
+
+ free_client_context (ctx);
+
+ return NULL;
+}
+
+static void
+unmanage_client (GstRTSPClient * client, ClientContext * ctx)
+{
+ GstRTSPServer *server = ctx->server;
+
+ GST_DEBUG_OBJECT (server, "unmanage client %p", client);
+
+ g_object_ref (server);
+ gst_rtsp_client_set_server (client, NULL);
+
+ GST_RTSP_SERVER_LOCK (server);
+ server->clients = g_list_remove (server->clients, ctx);
+ GST_RTSP_SERVER_UNLOCK (server);
+
+ if (ctx->loop)
+ g_main_loop_quit (ctx->loop);
+ else
+ free_client_context (ctx);
+
+ g_object_unref (server);
+}
+
+/* add the client context to the active list of clients, takes ownership
+ * of client */
+static void
+manage_client (GstRTSPServer * server, GstRTSPClient * client)
+{
+ ClientContext *ctx;
+
+ GST_DEBUG_OBJECT (server, "manage client %p", client);
+ gst_rtsp_client_set_server (client, server);
+
+ ctx = g_slice_new0 (ClientContext);
+ ctx->server = server;
+ ctx->client = client;
+ if (server->max_threads == 0) {
+ GSource *source;
+
+ /* find the context to add the watch */
+ if ((source = g_main_current_source ()))
+ ctx->context = g_main_context_ref (g_source_get_context (source));
+ else
+ ctx->context = NULL;
+ } else {
+ ctx->context = g_main_context_new ();
+ ctx->loop = g_main_loop_new (ctx->context, TRUE);
+ }
+ gst_rtsp_client_attach (client, ctx->context);
+
+ GST_RTSP_SERVER_LOCK (server);
+ g_signal_connect (client, "closed", (GCallback) unmanage_client, ctx);
+ server->clients = g_list_prepend (server->clients, ctx);
+ GST_RTSP_SERVER_UNLOCK (server);
+
+ if (ctx->loop) {
+ GstRTSPServerClass *klass = GST_RTSP_SERVER_GET_CLASS (server);
+
+ g_thread_pool_push (klass->pool, ctx, NULL);
}
}
-/* default method for creating a new client object in the server to accept and
- * handle a client connection on this server */
static GstRTSPClient *
-default_accept_client (GstRTSPServer * server, GIOChannel * channel)
+default_create_client (GstRTSPServer * server)
{
GstRTSPClient *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,
+ GSocket * socket, GError ** error)
+{
/* 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))
+ if (!gst_rtsp_client_accept (client, socket, NULL, error))
goto accept_failed;
- return client;
+ return TRUE;
/* ERRORS */
accept_failed:
{
GST_ERROR_OBJECT (server,
- "Could not accept client on server socket %d: %s (%d)",
- server->server_sock.fd, g_strerror (errno), errno);
- gst_object_unref (client);
- return NULL;
+ "Could not accept client on server : %s", (*error)->message);
+ return FALSE;
}
}
/**
- * gst_rtsp_server_io_func:
- * @channel: a #GIOChannel
- * @condition: the condition on @source
+ * gst_rtsp_server_transfer_connection:
+ * @server: a #GstRTSPServer
+ * @socket: a network socket
+ * @ip: the IP address of the remote client
+ * @port: the port used by the other end
+ * @initial_buffer: any initial data that was already read from the socket
*
- * A default #GIOFunc that creates a new #GstRTSPClient to accept and handle a
- * new connection on @channel or @server.
+ * Take an existing network socket and use it for an RTSP connection. This
+ * is used when transferring a socket from an HTTP server which should be used
+ * as an RTSP over HTTP tunnel. The @initial_buffer contains any remaining data
+ * that the HTTP server read from the socket while parsing the HTTP header.
*
- * Returns: TRUE if the source could be connected, FALSE if an error occured.
+ * Returns: TRUE if all was ok, FALSE if an error occured.
*/
gboolean
-gst_rtsp_server_io_func (GIOChannel * channel, GIOCondition condition,
- GstRTSPServer * server)
+gst_rtsp_server_transfer_connection (GstRTSPServer * server, GSocket * socket,
+ const gchar * ip, gint port, const gchar * initial_buffer)
{
GstRTSPClient *client = NULL;
GstRTSPServerClass *klass;
+ GError *error = NULL;
- if (condition & G_IO_IN) {
- klass = GST_RTSP_SERVER_GET_CLASS (server);
+ klass = GST_RTSP_SERVER_GET_CLASS (server);
- /* a new client connected, create a client object to handle the client. */
- if (klass->accept_client)
- client = klass->accept_client (server, channel);
- if (client == NULL)
- goto client_failed;
+ if (klass->create_client)
+ client = klass->create_client (server);
+ if (client == NULL)
+ goto client_failed;
- /* can unref the client now, when the request is finished, it will be
- * unreffed async. */
- gst_object_unref (client);
- } else {
- GST_WARNING_OBJECT (server, "received unknown event %08x", condition);
+ /* a new client connected, create a client object to handle the client. */
+ if (!gst_rtsp_client_use_socket (client, socket, ip,
+ port, initial_buffer, &error)) {
+ goto transfer_failed;
}
+
+ /* manage the client connection */
+ manage_client (server, client);
+
+ g_signal_emit (server, gst_rtsp_server_signals[SIGNAL_CLIENT_CONNECTED], 0,
+ client);
+
return TRUE;
/* ERRORS */
GST_ERROR_OBJECT (server, "failed to create a client");
return FALSE;
}
+transfer_failed:
+ {
+ GST_ERROR_OBJECT (server, "failed to accept client: %s", error->message);
+ g_error_free (error);
+ g_object_unref (client);
+ return FALSE;
+ }
}
/**
- * gst_rtsp_server_get_io_channel:
+ * gst_rtsp_server_io_func:
+ * @socket: a #GSocket
+ * @condition: the condition on @source
* @server: a #GstRTSPServer
*
- * Create a #GIOChannel for @server.
+ * A default #GSocketSourceFunc that creates a new #GstRTSPClient to accept and handle a
+ * new connection on @socket or @server.
*
- * Returns: the GIOChannel for @server or NULL when an error occured.
+ * Returns: TRUE if the source could be connected, FALSE if an error occured.
*/
-GIOChannel *
-gst_rtsp_server_get_io_channel (GstRTSPServer * server)
+gboolean
+gst_rtsp_server_io_func (GSocket * socket, GIOCondition condition,
+ GstRTSPServer * server)
{
- g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
+ gboolean result = TRUE;
+ GstRTSPClient *client = NULL;
+ GstRTSPServerClass *klass;
+ GError *error = NULL;
+
+ 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, socket, &error);
+ if (!result)
+ goto accept_failed;
- if (server->io_channel == NULL) {
- if (!gst_rtsp_server_sink_init_send (server))
- goto init_failed;
+ /* manage the client connection */
+ manage_client (server, client);
- /* create IO channel for the socket */
- server->io_channel = g_io_channel_unix_new (server->server_sock.fd);
+ g_signal_emit (server, gst_rtsp_server_signals[SIGNAL_CLIENT_CONNECTED], 0,
+ client);
+ } else {
+ GST_WARNING_OBJECT (server, "received unknown event %08x", condition);
}
- return server->io_channel;
+ return TRUE;
-init_failed:
+ /* ERRORS */
+client_failed:
{
- GST_ERROR_OBJECT (server, "failed to initialize server");
- return NULL;
+ GST_ERROR_OBJECT (server, "failed to create a client");
+ return FALSE;
+ }
+accept_failed:
+ {
+ GST_ERROR_OBJECT (server, "failed to accept client: %s", error->message);
+ g_error_free (error);
+ g_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:
+ * gst_rtsp_server_create_source:
* @server: a #GstRTSPServer
+ * @cancellable: a #GCancellable or %NULL.
+ * @error: a #GError
*
* Create a #GSource for @server. The new source will have a default
- * #GIOFunc of gst_rtsp_server_io_func().
+ * #GSocketSourceFunc of gst_rtsp_server_io_func().
+ *
+ * @cancellable if not NULL can be used to cancel the source, which will cause
+ * the source to trigger, reporting the current condition (which is likely 0
+ * unless cancellation happened at the same time as a condition change). You can
+ * check for this in the callback using g_cancellable_is_cancelled().
*
- * Returns: the #GSource for @server or NULL when an error occured.
+ * Returns: the #GSource for @server or NULL when an error occured. Free with
+ * g_source_unref ()
*/
GSource *
-gst_rtsp_server_create_watch (GstRTSPServer * server)
+gst_rtsp_server_create_source (GstRTSPServer * server,
+ GCancellable * cancellable, GError ** error)
{
+ GSocket *socket, *old;
+ GSource *source;
+
g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
- if (server->io_watch == NULL) {
- GIOChannel *channel;
+ socket = gst_rtsp_server_create_socket (server, NULL, error);
+ if (socket == NULL)
+ goto no_socket;
- channel = gst_rtsp_server_get_io_channel (server);
- if (channel == NULL)
- goto no_channel;
+ GST_RTSP_SERVER_LOCK (server);
+ old = server->socket;
+ server->socket = g_object_ref (socket);
+ GST_RTSP_SERVER_UNLOCK (server);
- /* create a watch for reads (new connections) and possible errors */
- server->io_watch = g_io_create_watch (channel, G_IO_IN |
- G_IO_ERR | G_IO_HUP | G_IO_NVAL);
+ if (old)
+ g_object_unref (old);
- /* configure the callback */
- g_source_set_callback (server->io_watch,
- (GSourceFunc) gst_rtsp_server_io_func, server, NULL);
- }
- return server->io_watch;
+ /* create a watch for reads (new connections) and possible errors */
+ source = g_socket_create_source (socket, G_IO_IN |
+ G_IO_ERR | G_IO_HUP | G_IO_NVAL, cancellable);
+ g_object_unref (socket);
-no_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_socket:
{
- GST_ERROR_OBJECT (server, "failed to create IO channel");
+ GST_ERROR_OBJECT (server, "failed to create socket");
return NULL;
}
}
/**
* gst_rtsp_server_attach:
* @server: a #GstRTSPServer
- * @context: a #GMainContext
+ * @context: (allow-none): a #GMainContext
*
* Attaches @server to @context. When the mainloop for @context is run, the
- * server will be dispatched.
+ * 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.
{
guint res;
GSource *source;
+ GError *error = NULL;
g_return_val_if_fail (GST_IS_RTSP_SERVER (server), 0);
- source = gst_rtsp_server_create_watch (server);
+ source = gst_rtsp_server_create_source (server, NULL, &error);
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");
+ GST_ERROR_OBJECT (server, "failed to create watch: %s", error->message);
+ g_error_free (error);
return 0;
}
}
projects / platform / upstream / gstreamer.git / blobdiff