GST_INFO ("finalize client %p", client);
- if (client->watchid)
+ if (client->watch)
g_source_destroy ((GSource *) client->watch);
client_cleanup_sessions (client);
client_watch_notify (GstRTSPClient * client)
{
GST_INFO ("client %p: watch destroyed", client);
- client->watchid = 0;
client->watch = NULL;
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_CLOSED], 0, NULL);
g_object_unref (client);
}
static gboolean
-attach_client (GstRTSPClient * client, GSocket * socket,
+setup_client (GstRTSPClient * client, GSocket * socket,
GstRTSPConnection * conn, GError ** error)
{
GSocket *read_socket;
GSocketAddress *address;
- GSource *source;
- GMainContext *context;
GstRTSPUrl *url;
read_socket = gst_rtsp_connection_get_read_socket (conn);
client->connection = conn;
- /* create watch for the connection and attach */
- client->watch = gst_rtsp_watch_new (client->connection, &watch_funcs,
- g_object_ref (client), (GDestroyNotify) client_watch_notify);
-
- /* find the context to add the watch */
- if ((source = g_main_current_source ()))
- context = g_source_get_context (source);
- else
- context = NULL;
-
- GST_INFO ("attaching to context %p", context);
-
- client->watchid = gst_rtsp_watch_attach (client->watch, context);
- gst_rtsp_watch_unref (client->watch);
-
return TRUE;
/* ERRORS */
* @socket: a #GSocket
* @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
+ * @initial_buffer: any zero terminated initial data that was already read from
+ * the socket
* @error: a #GError
*
* Take an existing network socket and use it for an RTSP connection.
* Returns: %TRUE on success.
*/
gboolean
-gst_rtsp_client_create_from_socket (GstRTSPClient * client, GSocket * socket,
+gst_rtsp_client_use_socket (GstRTSPClient * client, GSocket * socket,
const gchar * ip, gint port, const gchar * initial_buffer, GError ** error)
{
GstRTSPConnection *conn;
GST_RTSP_CHECK (gst_rtsp_connection_create_from_socket (socket, ip, port,
initial_buffer, &conn), no_connection);
- return attach_client (client, socket, conn, error);
+ return setup_client (client, socket, conn, error);
/* ERRORS */
no_connection:
* gst_rtsp_client_accept:
* @client: a #GstRTSPClient
* @socket: a #GSocket
+ * @context: the context to run in
* @cancellable: a #GCancellable
* @error: a #GError
*
* Accept a new connection for @client on @socket.
*
- * This function should be called when the client properties and urls are fully
- * configured and the client is ready to start.
- *
* Returns: %TRUE if the client could be accepted.
*/
gboolean
GST_RTSP_CHECK (gst_rtsp_connection_accept (socket, &conn, cancellable),
accept_failed);
- return attach_client (client, socket, conn, error);
+ return setup_client (client, socket, conn, error);
/* ERRORS */
accept_failed:
return FALSE;
}
}
+
+/**
+ * gst_rtsp_client_attach:
+ * @client: a #GstRTSPClient
+ * @context: (allow-none): a #GMainContext
+ *
+ * Attaches @client to @context. When the mainloop for @context is run, the
+ * client will be dispatched. When @context is NULL, the default context will be
+ * used).
+ *
+ * This function should be called when the client properties and urls are fully
+ * configured and the client is ready to start.
+ *
+ * Returns: the ID (greater than 0) for the source within the GMainContext.
+ */
+guint
+gst_rtsp_client_attach (GstRTSPClient * client, GMainContext * context)
+{
+ guint res;
+
+ g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), 0);
+ g_return_val_if_fail (client->watch == NULL, 0);
+
+ /* create watch for the connection and attach */
+ client->watch = gst_rtsp_watch_new (client->connection, &watch_funcs,
+ g_object_ref (client), (GDestroyNotify) client_watch_notify);
+
+ GST_INFO ("attaching to context %p", context);
+ res = gst_rtsp_watch_attach (client->watch, context);
+ gst_rtsp_watch_unref (client->watch);
+
+ return res;
+}
}
}
+typedef struct
+{
+ GstRTSPServer *server;
+ GMainLoop *loop;
+ GMainContext *context;
+ GstRTSPClient *client;
+} ClientContext;
+
static void
-unmanage_client (GstRTSPClient * client, GstRTSPServer * server)
+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, client);
+ server->clients = g_list_remove (server->clients, ctx);
GST_RTSP_SERVER_UNLOCK (server);
- g_object_unref (server);
- g_object_unref (client);
+ if (ctx->loop)
+ g_main_loop_quit (ctx->loop);
+ else
+ free_client_context (ctx);
+
+ g_object_unref (server);
}
-/* add the client to the active list of clients, takes ownership of
- * the client */
+/* 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 1
+ {
+ 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);
+ ctx->dothread = TRUE;
+#endif
+ gst_rtsp_client_attach (client, ctx->context);
+
GST_RTSP_SERVER_LOCK (server);
- g_signal_connect (client, "closed", (GCallback) unmanage_client, server);
- server->clients = g_list_prepend (server->clients, client);
+ 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) {
+ GThread *thread;
+
+ thread = g_thread_new ("MainLoop Thread", (GThreadFunc) do_loop, ctx);
+ g_thread_unref (thread);
+ }
}
static GstRTSPClient *
goto client_failed;
/* a new client connected, create a client object to handle the client. */
- if (!gst_rtsp_client_create_from_socket (client, socket, ip, port,
- initial_buffer, &error)) {
+ if (!gst_rtsp_client_use_socket (client, socket, ip,
+ port, initial_buffer, &error)) {
goto transfer_failed;
}
{
GST_ERROR_OBJECT (server, "failed to accept client: %s", error->message);
g_error_free (error);
- gst_object_unref (client);
+ g_object_unref (client);
return FALSE;
}
}
{
GST_ERROR_OBJECT (server, "failed to accept client: %s", error->message);
g_error_free (error);
- gst_object_unref (client);
+ g_object_unref (client);
return FALSE;
}
}
projects / platform / upstream / gstreamer.git / commitdiff