/* GStreamer * Copyright (C) 2008 Wim Taymans * * 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 #include "rtsp-server.h" #include "rtsp-client.h" #define DEFAULT_BACKLOG 5 #define DEFAULT_PORT 8554 enum { PROP_0, PROP_BACKLOG, PROP_PORT, PROP_SESSION_POOL, PROP_MEDIA_MAPPING, PROP_LAST }; G_DEFINE_TYPE (GstRTSPServer, gst_rtsp_server, G_TYPE_OBJECT); 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 GstRTSPClient * gst_rtsp_server_accept_client (GstRTSPServer *server, 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; /** * 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::port * * The session port of the server. This is the port where the server will * listen on. */ g_object_class_install_property (gobject_class, PROP_PORT, g_param_spec_int ("port", "Port", "The port the server uses to listen on", 1, 65535, DEFAULT_PORT, 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)); klass->accept_client = gst_rtsp_server_accept_client; } static void gst_rtsp_server_init (GstRTSPServer * server) { server->server_port = DEFAULT_PORT; server->backlog = DEFAULT_BACKLOG; server->session_pool = gst_rtsp_session_pool_new (); server->media_mapping = gst_rtsp_media_mapping_new (); } /** * 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_port: * @server: a #GstRTSPServer * @port: the port * * Configure @server to accept connections on the given port. * @port should be a port number between 1 and 65535. * * This function must be called before the server is bound. */ void gst_rtsp_server_set_port (GstRTSPServer *server, gint port) { g_return_if_fail (GST_IS_RTSP_SERVER (server)); g_return_if_fail (port >= 1 && port <= 65535); server->server_port = port; } /** * gst_rtsp_server_get_port: * @server: a #GstRTSPServer * * Get the port number on which the server will accept connections. * * Returns: the server port. */ gint gst_rtsp_server_get_port (GstRTSPServer *server) { g_return_val_if_fail (GST_IS_RTSP_SERVER (server), -1); return server->server_port; } /** * 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)); server->backlog = backlog; } /** * 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) { g_return_val_if_fail (GST_IS_RTSP_SERVER (server), -1); return server->backlog; } /** * 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)); old = server->session_pool; if (old != pool) { if (pool) g_object_ref (pool); server->session_pool = pool; 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); if ((result = server->session_pool)) g_object_ref (result); 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)); old = server->media_mapping; if (old != mapping) { if (mapping) g_object_ref (mapping); server->media_mapping = mapping; 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); if ((result = server->media_mapping)) g_object_ref (result); 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_PORT: g_value_set_int (value, gst_rtsp_server_get_port (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_PORT: gst_rtsp_server_set_port (server, g_value_get_int (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); } } /* Prepare a server socket for @server and make it listen on the configured port */ static gboolean gst_rtsp_server_sink_init_send (GstRTSPServer * server) { int ret; /* create server socket */ if ((server->server_sock.fd = socket (AF_INET, SOCK_STREAM, 0)) == -1) goto no_socket; GST_DEBUG_OBJECT (server, "opened sending server socket with fd %d", server->server_sock.fd); /* make address reusable */ ret = 1; if (setsockopt (server->server_sock.fd, SOL_SOCKET, SO_REUSEADDR, (void *) &ret, sizeof (ret)) < 0) goto reuse_failed; /* 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; /* name the socket */ memset (&server->server_sin, 0, sizeof (server->server_sin)); server->server_sin.sin_family = AF_INET; /* network socket */ server->server_sin.sin_port = htons (server->server_port); /* on port */ server->server_sin.sin_addr.s_addr = htonl (INADDR_ANY); /* for hosts */ /* bind it */ GST_DEBUG_OBJECT (server, "binding server socket to address"); ret = bind (server->server_sock.fd, (struct sockaddr *) &server->server_sin, sizeof (server->server_sin)); if (ret) goto bind_failed; /* set the server socket to nonblocking */ fcntl (server->server_sock.fd, F_SETFL, O_NONBLOCK); 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) goto listen_failed; GST_DEBUG_OBJECT (server, "listened on server socket %d, returning from connection setup", server->server_sock.fd); return TRUE; /* ERRORS */ no_socket: { GST_ERROR_OBJECT (server, "failed to create socket: %s", g_strerror (errno)); return FALSE; } reuse_failed: { if (server->server_sock.fd >= 0) { close (server->server_sock.fd); server->server_sock.fd = -1; } GST_ERROR_OBJECT (server, "failed to reuse socket: %s", g_strerror (errno)); return FALSE; } keepalive_failed: { if (server->server_sock.fd >= 0) { close (server->server_sock.fd); server->server_sock.fd = -1; } GST_ERROR_OBJECT (server, "failed to configure keepalive socket: %s", g_strerror (errno)); return FALSE; } listen_failed: { if (server->server_sock.fd >= 0) { close (server->server_sock.fd); server->server_sock.fd = -1; } GST_ERROR_OBJECT (server, "failed to listen on socket: %s", g_strerror (errno)); return FALSE; } bind_failed: { if (server->server_sock.fd >= 0) { close (server->server_sock.fd); server->server_sock.fd = -1; } GST_ERROR_OBJECT (server, "failed to bind on socket: %s", g_strerror (errno)); return FALSE; } } /* default method for creating a new client object in the server to accept and * handle a client connection on this server */ static GstRTSPClient * gst_rtsp_server_accept_client (GstRTSPServer *server, GIOChannel *channel) { 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_client_set_session_pool (client, server->session_pool); /* set the session pool that this client should use */ gst_rtsp_client_set_media_mapping (client, server->media_mapping); /* 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 client; /* ERRORS */ accept_failed: { g_error ("Could not accept client on server socket %d: %s (%d)", server->server_sock.fd, g_strerror (errno), errno); gst_object_unref (client); return NULL; } } /** * 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) { GstRTSPClient *client = NULL; GstRTSPServerClass *klass; if (condition & G_IO_IN) { 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; /* can unref the client now, when the request is finished, it will be * unreffed async. */ gst_object_unref (client); } else { g_print ("received unknown event %08x", condition); } return TRUE; /* ERRORS */ client_failed: { GST_ERROR_OBJECT (server, "failed to create a client"); return FALSE; } } /** * gst_rtsp_server_get_io_channel: * @server: a #GstRTSPServer * * Create a #GIOChannel for @server. * * Returns: the GIOChannel for @server or NULL when an error occured. */ GIOChannel * gst_rtsp_server_get_io_channel (GstRTSPServer *server) { g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL); if (server->io_channel == NULL) { if (!gst_rtsp_server_sink_init_send (server)) goto init_failed; /* create IO channel for the socket */ server->io_channel = g_io_channel_unix_new (server->server_sock.fd); } return server->io_channel; init_failed: { return NULL; } } /** * 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) { g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL); if (server->io_watch == NULL) { GIOChannel *channel; channel = gst_rtsp_server_get_io_channel (server); if (channel == NULL) goto no_channel; /* 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); /* configure the callback */ g_source_set_callback (server->io_watch, (GSourceFunc) gst_rtsp_server_io_func, server, NULL); } return server->io_watch; no_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. * * 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); return res; /* ERRORS */ no_source: { return 0; } }