gstreamer/gst/rtsp-server/rtsp-server.c
Wim Taymans edd2175695 Fix some leaks and change default port
Fix some memory leaks by setting the udpsrc elements to the unlocked state after
we finished the initial preroll. If we keep them locked, setting the pipeline to
NULL will not stop and clean up the sources correctly.

Change the default RTSP port to 8554 aka the official alternative RTSP port.
2009-01-30 12:18:01 +01:00

612 lines
16 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 <sys/ioctl.h>
#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;
}
}