gstreamer/subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-server.c

1521 lines
41 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., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
/**
* SECTION:rtsp-server
* @short_description: The main server object
* @see_also: #GstRTSPClient, #GstRTSPThreadPool
*
* The server object is the object listening for connections on a port and
* creating #GstRTSPClient objects to handle those connections.
*
* The server will listen on the address set with gst_rtsp_server_set_address()
* and the port or service configured with gst_rtsp_server_set_service().
* Use gst_rtsp_server_set_backlog() to configure the amount of pending requests
* that the server will keep. By default the server listens on the current
* network (0.0.0.0) and port 8554.
*
* The server will require an SSL connection when a TLS certificate has been
* set in the auth object with gst_rtsp_auth_set_tls_certificate().
*
* To start the server, use gst_rtsp_server_attach() to attach it to a
* #GMainContext. For more control, gst_rtsp_server_create_source() and
* gst_rtsp_server_create_socket() can be used to get a #GSource and #GSocket
* respectively.
*
* gst_rtsp_server_transfer_connection() can be used to transfer an existing
* socket to the RTSP server, for example from an HTTP server.
*
* Once the server socket is attached to a mainloop, it will start accepting
* connections. When a new connection is received, a new #GstRTSPClient object
* is created to handle the connection. The new client will be configured with
* the server #GstRTSPAuth, #GstRTSPMountPoints, #GstRTSPSessionPool and
* #GstRTSPThreadPool.
*
* The server uses the configured #GstRTSPThreadPool object to handle the
* remainder of the communication with this client.
*
* Last reviewed on 2013-07-11 (1.0.0)
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <stdlib.h>
#include <string.h>
#include "rtsp-context.h"
#include "rtsp-server-object.h"
#include "rtsp-client.h"
#define GST_RTSP_SERVER_GET_LOCK(server) (&(GST_RTSP_SERVER_CAST(server)->priv->lock))
#define GST_RTSP_SERVER_LOCK(server) (g_mutex_lock(GST_RTSP_SERVER_GET_LOCK(server)))
#define GST_RTSP_SERVER_UNLOCK(server) (g_mutex_unlock(GST_RTSP_SERVER_GET_LOCK(server)))
struct _GstRTSPServerPrivate
{
GMutex lock; /* protects everything in this struct */
/* server information */
gchar *address;
gchar *service;
gint backlog;
GSocket *socket;
/* sessions on this server */
GstRTSPSessionPool *session_pool;
/* mount points for this server */
GstRTSPMountPoints *mount_points;
/* request size limit */
guint content_length_limit;
/* authentication manager */
GstRTSPAuth *auth;
/* resource manager */
GstRTSPThreadPool *thread_pool;
/* the clients that are connected */
GList *clients;
guint clients_cookie;
};
#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 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_BOUND_PORT,
PROP_BACKLOG,
PROP_SESSION_POOL,
PROP_MOUNT_POINTS,
PROP_CONTENT_LENGTH_LIMIT,
PROP_LAST
};
enum
{
SIGNAL_CLIENT_CONNECTED,
SIGNAL_LAST
};
G_DEFINE_TYPE_WITH_PRIVATE (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_create_client (GstRTSPServer * server);
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::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
* 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::mount-points:
*
* The mount points to use for this server. By default the server has no
* mount points and thus cannot map urls to media streams.
*/
g_object_class_install_property (gobject_class, PROP_MOUNT_POINTS,
g_param_spec_object ("mount-points", "Mount Points",
"The mount points to use for client session",
GST_TYPE_RTSP_MOUNT_POINTS,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* RTSPServer::content-length-limit:
*
* Define an appropriate request size limit and reject requests exceeding the
* limit.
*
* Since: 1.18
*/
g_object_class_install_property (gobject_class, PROP_CONTENT_LENGTH_LIMIT,
g_param_spec_uint ("content-length-limit", "Limitation of Content-Length",
"Limitation of Content-Length",
0, G_MAXUINT, G_MAXUINT, 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, NULL, G_TYPE_NONE, 1, GST_TYPE_RTSP_CLIENT);
klass->create_client = default_create_client;
GST_DEBUG_CATEGORY_INIT (rtsp_server_debug, "rtspserver", 0, "GstRTSPServer");
}
static void
gst_rtsp_server_init (GstRTSPServer * server)
{
GstRTSPServerPrivate *priv = gst_rtsp_server_get_instance_private (server);
server->priv = priv;
g_mutex_init (&priv->lock);
priv->address = g_strdup (DEFAULT_ADDRESS);
priv->service = g_strdup (DEFAULT_SERVICE);
priv->socket = NULL;
priv->backlog = DEFAULT_BACKLOG;
priv->session_pool = gst_rtsp_session_pool_new ();
priv->mount_points = gst_rtsp_mount_points_new ();
priv->content_length_limit = G_MAXUINT;
priv->thread_pool = gst_rtsp_thread_pool_new ();
}
static void
gst_rtsp_server_finalize (GObject * object)
{
GstRTSPServer *server = GST_RTSP_SERVER (object);
GstRTSPServerPrivate *priv = server->priv;
GST_DEBUG_OBJECT (server, "finalize server");
g_free (priv->address);
g_free (priv->service);
if (priv->socket)
g_object_unref (priv->socket);
if (priv->session_pool)
g_object_unref (priv->session_pool);
if (priv->mount_points)
g_object_unref (priv->mount_points);
if (priv->thread_pool)
g_object_unref (priv->thread_pool);
if (priv->auth)
g_object_unref (priv->auth);
g_mutex_clear (&priv->lock);
G_OBJECT_CLASS (gst_rtsp_server_parent_class)->finalize (object);
}
/**
* gst_rtsp_server_new:
*
* Create a new #GstRTSPServer instance.
*
* Returns: (transfer full): a new #GstRTSPServer
*/
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)
{
GstRTSPServerPrivate *priv;
g_return_if_fail (GST_IS_RTSP_SERVER (server));
g_return_if_fail (address != NULL);
priv = server->priv;
GST_RTSP_SERVER_LOCK (server);
g_free (priv->address);
priv->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: (transfer full) (nullable): the server address. g_free() after usage.
*/
gchar *
gst_rtsp_server_get_address (GstRTSPServer * server)
{
GstRTSPServerPrivate *priv;
gchar *result;
g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
priv = server->priv;
GST_RTSP_SERVER_LOCK (server);
result = g_strdup (priv->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)
{
GstRTSPServerPrivate *priv;
GSocketAddress *address;
int result = -1;
g_return_val_if_fail (GST_IS_RTSP_SERVER (server), result);
priv = server->priv;
GST_RTSP_SERVER_LOCK (server);
if (priv->socket == NULL)
goto out;
address = g_socket_get_local_address (priv->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;
}
/**
* 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.
*
* When @service is set to "0", the server will listen on a random free
* port. The actual used port can be retrieved with
* gst_rtsp_server_get_bound_port().
*
* This function must be called before the server is bound.
*/
void
gst_rtsp_server_set_service (GstRTSPServer * server, const gchar * service)
{
GstRTSPServerPrivate *priv;
g_return_if_fail (GST_IS_RTSP_SERVER (server));
g_return_if_fail (service != NULL);
priv = server->priv;
GST_RTSP_SERVER_LOCK (server);
g_free (priv->service);
priv->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: (transfer full) (nullable): the service. use g_free() after usage.
*/
gchar *
gst_rtsp_server_get_service (GstRTSPServer * server)
{
GstRTSPServerPrivate *priv;
gchar *result;
g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
priv = server->priv;
GST_RTSP_SERVER_LOCK (server);
result = g_strdup (priv->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)
{
GstRTSPServerPrivate *priv;
g_return_if_fail (GST_IS_RTSP_SERVER (server));
priv = server->priv;
GST_RTSP_SERVER_LOCK (server);
priv->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)
{
GstRTSPServerPrivate *priv;
gint result;
g_return_val_if_fail (GST_IS_RTSP_SERVER (server), -1);
priv = server->priv;
GST_RTSP_SERVER_LOCK (server);
result = priv->backlog;
GST_RTSP_SERVER_UNLOCK (server);
return result;
}
/**
* gst_rtsp_server_set_session_pool:
* @server: a #GstRTSPServer
* @pool: (transfer none) (nullable): a #GstRTSPSessionPool
*
* configure @pool to be used as the session pool of @server.
*/
void
gst_rtsp_server_set_session_pool (GstRTSPServer * server,
GstRTSPSessionPool * pool)
{
GstRTSPServerPrivate *priv;
GstRTSPSessionPool *old;
g_return_if_fail (GST_IS_RTSP_SERVER (server));
priv = server->priv;
if (pool)
g_object_ref (pool);
GST_RTSP_SERVER_LOCK (server);
old = priv->session_pool;
priv->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: (transfer full) (nullable): the #GstRTSPSessionPool used for sessions. g_object_unref() after
* usage.
*/
GstRTSPSessionPool *
gst_rtsp_server_get_session_pool (GstRTSPServer * server)
{
GstRTSPServerPrivate *priv;
GstRTSPSessionPool *result;
g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
priv = server->priv;
GST_RTSP_SERVER_LOCK (server);
if ((result = priv->session_pool))
g_object_ref (result);
GST_RTSP_SERVER_UNLOCK (server);
return result;
}
/**
* gst_rtsp_server_set_mount_points:
* @server: a #GstRTSPServer
* @mounts: (transfer none) (nullable): a #GstRTSPMountPoints
*
* configure @mounts to be used as the mount points of @server.
*/
void
gst_rtsp_server_set_mount_points (GstRTSPServer * server,
GstRTSPMountPoints * mounts)
{
GstRTSPServerPrivate *priv;
GstRTSPMountPoints *old;
g_return_if_fail (GST_IS_RTSP_SERVER (server));
priv = server->priv;
if (mounts)
g_object_ref (mounts);
GST_RTSP_SERVER_LOCK (server);
old = priv->mount_points;
priv->mount_points = mounts;
GST_RTSP_SERVER_UNLOCK (server);
if (old)
g_object_unref (old);
}
/**
* gst_rtsp_server_get_mount_points:
* @server: a #GstRTSPServer
*
* Get the #GstRTSPMountPoints used as the mount points of @server.
*
* Returns: (transfer full) (nullable): the #GstRTSPMountPoints of @server. g_object_unref() after
* usage.
*/
GstRTSPMountPoints *
gst_rtsp_server_get_mount_points (GstRTSPServer * server)
{
GstRTSPServerPrivate *priv;
GstRTSPMountPoints *result;
g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
priv = server->priv;
GST_RTSP_SERVER_LOCK (server);
if ((result = priv->mount_points))
g_object_ref (result);
GST_RTSP_SERVER_UNLOCK (server);
return result;
}
/**
* gst_rtsp_server_set_content_length_limit
* @server: a #GstRTSPServer
* Configure @server to use the specified Content-Length limit.
*
* Define an appropriate request size limit and reject requests exceeding the
* limit.
*
* Since: 1.18
*/
void
gst_rtsp_server_set_content_length_limit (GstRTSPServer * server, guint limit)
{
GstRTSPServerPrivate *priv;
g_return_if_fail (GST_IS_RTSP_SERVER (server));
priv = server->priv;
GST_RTSP_SERVER_LOCK (server);
priv->content_length_limit = limit;
GST_RTSP_SERVER_UNLOCK (server);
}
/**
* gst_rtsp_server_get_content_length_limit:
* @server: a #GstRTSPServer
*
* Get the Content-Length limit of @server.
*
* Returns: the Content-Length limit.
*
* Since: 1.18
*/
guint
gst_rtsp_server_get_content_length_limit (GstRTSPServer * server)
{
GstRTSPServerPrivate *priv;
guint result;
g_return_val_if_fail (GST_IS_RTSP_SERVER (server), G_MAXUINT);
priv = server->priv;
GST_RTSP_SERVER_LOCK (server);
result = priv->content_length_limit;
GST_RTSP_SERVER_UNLOCK (server);
return result;
}
/**
* gst_rtsp_server_set_auth:
* @server: a #GstRTSPServer
* @auth: (transfer none) (nullable): a #GstRTSPAuth
*
* configure @auth to be used as the authentication manager of @server.
*/
void
gst_rtsp_server_set_auth (GstRTSPServer * server, GstRTSPAuth * auth)
{
GstRTSPServerPrivate *priv;
GstRTSPAuth *old;
g_return_if_fail (GST_IS_RTSP_SERVER (server));
priv = server->priv;
if (auth)
g_object_ref (auth);
GST_RTSP_SERVER_LOCK (server);
old = priv->auth;
priv->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: (transfer full) (nullable): the #GstRTSPAuth of @server. g_object_unref() after
* usage.
*/
GstRTSPAuth *
gst_rtsp_server_get_auth (GstRTSPServer * server)
{
GstRTSPServerPrivate *priv;
GstRTSPAuth *result;
g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
priv = server->priv;
GST_RTSP_SERVER_LOCK (server);
if ((result = priv->auth))
g_object_ref (result);
GST_RTSP_SERVER_UNLOCK (server);
return result;
}
/**
* gst_rtsp_server_set_thread_pool:
* @server: a #GstRTSPServer
* @pool: (transfer none) (nullable): a #GstRTSPThreadPool
*
* configure @pool to be used as the thread pool of @server.
*/
void
gst_rtsp_server_set_thread_pool (GstRTSPServer * server,
GstRTSPThreadPool * pool)
{
GstRTSPServerPrivate *priv;
GstRTSPThreadPool *old;
g_return_if_fail (GST_IS_RTSP_SERVER (server));
priv = server->priv;
if (pool)
g_object_ref (pool);
GST_RTSP_SERVER_LOCK (server);
old = priv->thread_pool;
priv->thread_pool = pool;
GST_RTSP_SERVER_UNLOCK (server);
if (old)
g_object_unref (old);
}
/**
* gst_rtsp_server_get_thread_pool:
* @server: a #GstRTSPServer
*
* Get the #GstRTSPThreadPool used as the thread pool of @server.
*
* Returns: (transfer full) (nullable): the #GstRTSPThreadPool of @server. g_object_unref() after
* usage.
*/
GstRTSPThreadPool *
gst_rtsp_server_get_thread_pool (GstRTSPServer * server)
{
GstRTSPServerPrivate *priv;
GstRTSPThreadPool *result;
g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
priv = server->priv;
GST_RTSP_SERVER_LOCK (server);
if ((result = priv->thread_pool))
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_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_SESSION_POOL:
g_value_take_object (value, gst_rtsp_server_get_session_pool (server));
break;
case PROP_MOUNT_POINTS:
g_value_take_object (value, gst_rtsp_server_get_mount_points (server));
break;
case PROP_CONTENT_LENGTH_LIMIT:
g_value_set_uint (value,
gst_rtsp_server_get_content_length_limit (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_MOUNT_POINTS:
gst_rtsp_server_set_mount_points (server, g_value_get_object (value));
break;
case PROP_CONTENT_LENGTH_LIMIT:
gst_rtsp_server_set_content_length_limit (server,
g_value_get_uint (value));
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
}
}
/**
* gst_rtsp_server_create_socket:
* @server: a #GstRTSPServer
* @cancellable: (allow-none): a #GCancellable
* @error: (out): 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
* occurred.
*/
GSocket *
gst_rtsp_server_create_socket (GstRTSPServer * server,
GCancellable * cancellable, GError ** error)
{
GstRTSPServerPrivate *priv;
GSocketConnectable *conn;
GSocketAddressEnumerator *enumerator;
GSocket *socket = NULL;
#ifdef USE_SOLINGER
struct linger linger;
#endif
GError *sock_error = NULL;
GError *bind_error = NULL;
guint16 port;
g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
priv = server->priv;
GST_RTSP_SERVER_LOCK (server);
GST_DEBUG_OBJECT (server, "getting address info of %s/%s", priv->address,
priv->service);
/* resolve the server IP address */
port = atoi (priv->service);
if (port != 0 || !strcmp (priv->service, "0"))
conn = g_network_address_new (priv->address, port);
else
conn = g_network_service_new (priv->service, "tcp", priv->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. */
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;
}
/* 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);
g_object_unref (sockaddr);
continue;
}
if (g_socket_bind (socket, sockaddr, TRUE, bind_error ? NULL : &bind_error)) {
/* ask what port the socket has been bound to */
if (port == 0 || !strcmp (priv->service, "0")) {
GError *addr_error = NULL;
g_object_unref (sockaddr);
sockaddr = g_socket_get_local_address (socket, &addr_error);
if (addr_error != NULL) {
GST_DEBUG_OBJECT (server,
"failed to get the local address of a bound socket %s",
addr_error->message);
g_clear_error (&addr_error);
break;
}
port =
g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (sockaddr));
if (port != 0) {
g_free (priv->service);
priv->service = g_strdup_printf ("%d", port);
} else {
GST_DEBUG_OBJECT (server, "failed to get the port of a bound socket");
}
}
g_object_unref (sockaddr);
break;
}
GST_DEBUG_OBJECT (server, "failed to bind socket (%s), try next",
bind_error->message);
g_object_unref (sockaddr);
g_object_unref (socket);
socket = NULL;
}
g_object_unref (enumerator);
if (socket == NULL)
goto no_socket;
g_clear_error (&sock_error);
g_clear_error (&bind_error);
GST_DEBUG_OBJECT (server, "opened sending server socket");
/* keep connection alive; avoids SIGPIPE during write */
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 (sockfd, SOL_SOCKET, SO_LINGER,
(void *) &linger, sizeof (linger)) < 0)
goto linger_failed;
#endif
#endif
/* set the server socket to nonblocking */
g_socket_set_blocking (socket, FALSE);
/* set listen backlog */
g_socket_set_listen_backlog (socket, priv->backlog);
if (!g_socket_listen (socket, error))
goto listen_failed;
GST_DEBUG_OBJECT (server, "listening on server socket %p with queue of %d",
socket, priv->backlog);
GST_RTSP_SERVER_UNLOCK (server);
return socket;
/* ERRORS */
no_socket:
{
GST_ERROR_OBJECT (server, "failed to create socket");
goto close_error;
}
#if 0
#ifdef USE_SOLINGER
linger_failed:
{
GST_ERROR_OBJECT (server, "failed to no linger socket: %s",
g_strerror (errno));
goto close_error;
}
#endif
#endif
listen_failed:
{
GST_ERROR_OBJECT (server, "failed to listen on socket: %s",
(*error)->message);
goto close_error;
}
close_error:
{
if (socket)
g_object_unref (socket);
if (sock_error) {
if (error == NULL)
g_propagate_error (error, sock_error);
else
g_error_free (sock_error);
}
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;
GstRTSPThread *thread;
GstRTSPClient *client;
};
static gboolean
free_client_context (ClientContext * ctx)
{
GST_DEBUG ("free context %p", ctx);
GST_RTSP_SERVER_LOCK (ctx->server);
if (ctx->thread)
gst_rtsp_thread_stop (ctx->thread);
GST_RTSP_SERVER_UNLOCK (ctx->server);
g_object_unref (ctx->client);
g_object_unref (ctx->server);
g_slice_free (ClientContext, ctx);
return G_SOURCE_REMOVE;
}
static void
unmanage_client (GstRTSPClient * client, ClientContext * ctx)
{
GstRTSPServer *server = ctx->server;
GstRTSPServerPrivate *priv = server->priv;
GST_DEBUG_OBJECT (server, "unmanage client %p", client);
GST_RTSP_SERVER_LOCK (server);
priv->clients = g_list_remove (priv->clients, ctx);
priv->clients_cookie++;
GST_RTSP_SERVER_UNLOCK (server);
if (ctx->thread) {
GSource *src;
src = g_idle_source_new ();
g_source_set_callback (src, (GSourceFunc) free_client_context, ctx, NULL);
g_source_attach (src, ctx->thread->context);
g_source_unref (src);
} else {
free_client_context (ctx);
}
}
/* add the client context to the active list of clients, takes ownership
* of client */
static void
manage_client (GstRTSPServer * server, GstRTSPClient * client)
{
ClientContext *cctx;
GstRTSPServerPrivate *priv = server->priv;
GMainContext *mainctx = NULL;
GstRTSPContext ctx = { NULL };
GST_DEBUG_OBJECT (server, "manage client %p", client);
g_signal_emit (server, gst_rtsp_server_signals[SIGNAL_CLIENT_CONNECTED], 0,
client);
cctx = g_slice_new0 (ClientContext);
cctx->server = g_object_ref (server);
cctx->client = client;
GST_RTSP_SERVER_LOCK (server);
ctx.server = server;
ctx.client = client;
cctx->thread = gst_rtsp_thread_pool_get_thread (priv->thread_pool,
GST_RTSP_THREAD_TYPE_CLIENT, &ctx);
if (cctx->thread)
mainctx = cctx->thread->context;
else {
GSource *source;
/* find the context to add the watch */
if ((source = g_main_current_source ()))
mainctx = g_source_get_context (source);
}
g_signal_connect (client, "closed", (GCallback) unmanage_client, cctx);
priv->clients = g_list_prepend (priv->clients, cctx);
priv->clients_cookie++;
gst_rtsp_client_attach (client, mainctx);
GST_RTSP_SERVER_UNLOCK (server);
}
static GstRTSPClient *
default_create_client (GstRTSPServer * server)
{
GstRTSPClient *client;
GstRTSPServerPrivate *priv = server->priv;
/* 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, priv->session_pool);
/* set the mount points that this client should use */
gst_rtsp_client_set_mount_points (client, priv->mount_points);
/* Set content-length limit */
gst_rtsp_client_set_content_length_limit (GST_RTSP_CLIENT (client),
priv->content_length_limit);
/* set authentication manager */
gst_rtsp_client_set_auth (client, priv->auth);
/* set threadpool */
gst_rtsp_client_set_thread_pool (client, priv->thread_pool);
GST_RTSP_SERVER_UNLOCK (server);
return client;
}
/**
* gst_rtsp_server_transfer_connection:
* @server: a #GstRTSPServer
* @socket: (transfer full): a network socket
* @ip: the IP address of the remote client
* @port: the port used by the other end
* @initial_buffer: (nullable): any initial data that was already read from the socket
*
* 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 all was ok, FALSE if an error occurred.
*/
gboolean
gst_rtsp_server_transfer_connection (GstRTSPServer * server, GSocket * socket,
const gchar * ip, gint port, const gchar * initial_buffer)
{
GstRTSPClient *client = NULL;
GstRTSPServerClass *klass;
GstRTSPConnection *conn;
GstRTSPResult res;
klass = GST_RTSP_SERVER_GET_CLASS (server);
if (klass->create_client)
client = klass->create_client (server);
if (client == NULL)
goto client_failed;
GST_RTSP_CHECK (gst_rtsp_connection_create_from_socket (socket, ip, port,
initial_buffer, &conn), no_connection);
g_object_unref (socket);
/* set connection on the client now */
gst_rtsp_client_set_connection (client, conn);
/* manage the client connection */
manage_client (server, client);
return TRUE;
/* ERRORS */
client_failed:
{
GST_ERROR_OBJECT (server, "failed to create a client");
g_object_unref (socket);
return FALSE;
}
no_connection:
{
gchar *str = gst_rtsp_strresult (res);
GST_ERROR ("could not create connection from socket %p: %s", socket, str);
g_free (str);
g_object_unref (socket);
g_object_unref (client);
return FALSE;
}
}
/**
* gst_rtsp_server_io_func:
* @socket: a #GSocket
* @condition: the condition on @source
* @server: (transfer none): a #GstRTSPServer
*
* A default #GSocketSourceFunc that creates a new #GstRTSPClient to accept and handle a
* new connection on @socket or @server.
*
* Returns: TRUE if the source could be connected, FALSE if an error occurred.
*/
gboolean
gst_rtsp_server_io_func (GSocket * socket, GIOCondition condition,
GstRTSPServer * server)
{
GstRTSPServerPrivate *priv = server->priv;
GstRTSPClient *client = NULL;
GstRTSPServerClass *klass;
GstRTSPResult res;
GstRTSPConnection *conn = NULL;
GstRTSPContext ctx = { NULL };
if (condition & G_IO_IN) {
/* a new client connected. */
GST_RTSP_CHECK (gst_rtsp_connection_accept (socket, &conn, NULL),
accept_failed);
ctx.server = server;
ctx.conn = conn;
ctx.auth = priv->auth;
gst_rtsp_context_push_current (&ctx);
if (!gst_rtsp_auth_check (GST_RTSP_AUTH_CHECK_CONNECT))
goto connection_refused;
klass = GST_RTSP_SERVER_GET_CLASS (server);
/* a new client connected, create a client object to handle the client. */
if (klass->create_client)
client = klass->create_client (server);
if (client == NULL)
goto client_failed;
/* set connection on the client now */
gst_rtsp_client_set_connection (client, conn);
/* manage the client connection */
manage_client (server, client);
} else {
GST_WARNING_OBJECT (server, "received unknown event %08x", condition);
goto exit_no_ctx;
}
exit:
gst_rtsp_context_pop_current (&ctx);
exit_no_ctx:
return G_SOURCE_CONTINUE;
/* ERRORS */
accept_failed:
{
gchar *str = gst_rtsp_strresult (res);
GST_ERROR_OBJECT (server, "Could not accept client on socket %p: %s",
socket, str);
g_free (str);
/* We haven't pushed the context yet, so just return */
goto exit_no_ctx;
}
connection_refused:
{
GST_ERROR_OBJECT (server, "connection refused");
gst_rtsp_connection_free (conn);
goto exit;
}
client_failed:
{
GST_ERROR_OBJECT (server, "failed to create a client");
gst_rtsp_connection_free (conn);
goto exit;
}
}
static void
watch_destroyed (GstRTSPServer * server)
{
GstRTSPServerPrivate *priv = server->priv;
GST_DEBUG_OBJECT (server, "source destroyed");
g_object_unref (priv->socket);
priv->socket = NULL;
g_object_unref (server);
}
/**
* gst_rtsp_server_create_source:
* @server: a #GstRTSPServer
* @cancellable: (allow-none): a #GCancellable or %NULL.
* @error: (out): a #GError
*
* Create a #GSource for @server. The new source will have a default
* #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().
*
* This takes a reference on @server until @source is destroyed.
*
* Returns: (transfer full): the #GSource for @server or %NULL when an error
* occurred. Free with g_source_unref ()
*/
GSource *
gst_rtsp_server_create_source (GstRTSPServer * server,
GCancellable * cancellable, GError ** error)
{
GstRTSPServerPrivate *priv;
GSocket *socket, *old;
GSource *source;
g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
priv = server->priv;
socket = gst_rtsp_server_create_socket (server, NULL, error);
if (socket == NULL)
goto no_socket;
GST_RTSP_SERVER_LOCK (server);
old = priv->socket;
priv->socket = g_object_ref (socket);
GST_RTSP_SERVER_UNLOCK (server);
if (old)
g_object_unref (old);
/* 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);
/* 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 socket");
return NULL;
}
}
/**
* gst_rtsp_server_attach:
* @server: a #GstRTSPServer
* @context: (allow-none): 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.
*
* This takes a reference on @server until the source is destroyed. Note that
* if @context is not the default main context as returned by
* g_main_context_default() (or %NULL), g_source_remove() cannot be used to
* destroy the source. In that case it is recommended to use
* gst_rtsp_server_create_source() and attach it to @context manually.
*
* 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;
GError *error = NULL;
g_return_val_if_fail (GST_IS_RTSP_SERVER (server), 0);
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: %s", error->message);
g_error_free (error);
return 0;
}
}
/**
* gst_rtsp_server_client_filter:
* @server: a #GstRTSPServer
* @func: (scope call) (allow-none): a callback
* @user_data: user data passed to @func
*
* Call @func for each client managed by @server. The result value of @func
* determines what happens to the client. @func will be called with @server
* locked so no further actions on @server can be performed from @func.
*
* If @func returns #GST_RTSP_FILTER_REMOVE, the client will be removed from
* @server.
*
* If @func returns #GST_RTSP_FILTER_KEEP, the client will remain in @server.
*
* If @func returns #GST_RTSP_FILTER_REF, the client will remain in @server but
* will also be added with an additional ref to the result #GList of this
* function..
*
* When @func is %NULL, #GST_RTSP_FILTER_REF will be assumed for each client.
*
* Returns: (element-type GstRTSPClient) (transfer full): a #GList with all
* clients for which @func returned #GST_RTSP_FILTER_REF. After usage, each
* element in the #GList should be unreffed before the list is freed.
*/
GList *
gst_rtsp_server_client_filter (GstRTSPServer * server,
GstRTSPServerClientFilterFunc func, gpointer user_data)
{
GstRTSPServerPrivate *priv;
GList *result, *walk, *next;
GHashTable *visited;
guint cookie;
g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
priv = server->priv;
result = NULL;
if (func)
visited = g_hash_table_new_full (NULL, NULL, g_object_unref, NULL);
GST_RTSP_SERVER_LOCK (server);
restart:
cookie = priv->clients_cookie;
for (walk = priv->clients; walk; walk = next) {
ClientContext *cctx = walk->data;
GstRTSPClient *client = cctx->client;
GstRTSPFilterResult res;
gboolean changed;
next = g_list_next (walk);
if (func) {
/* only visit each media once */
if (g_hash_table_contains (visited, client))
continue;
g_hash_table_add (visited, g_object_ref (client));
GST_RTSP_SERVER_UNLOCK (server);
res = func (server, client, user_data);
GST_RTSP_SERVER_LOCK (server);
} else
res = GST_RTSP_FILTER_REF;
changed = (cookie != priv->clients_cookie);
switch (res) {
case GST_RTSP_FILTER_REMOVE:
GST_RTSP_SERVER_UNLOCK (server);
gst_rtsp_client_close (client);
GST_RTSP_SERVER_LOCK (server);
changed |= (cookie != priv->clients_cookie);
break;
case GST_RTSP_FILTER_REF:
result = g_list_prepend (result, g_object_ref (client));
break;
case GST_RTSP_FILTER_KEEP:
default:
break;
}
if (changed)
goto restart;
}
GST_RTSP_SERVER_UNLOCK (server);
if (func)
g_hash_table_unref (visited);
return result;
}