mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-12-11 19:06:33 +00:00
62d4c0b179
Export rtsp-server library API in headers when we're building the library itself, otherwise import the API from the headers. This fixes linker warnings on Windows when building with MSVC. Fix up some missing config.h includes when building the lib which is needed to get the export api define from config.h https://bugzilla.gnome.org/show_bug.cgi?id=797185
1440 lines
39 KiB
C
1440 lines
39 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-server.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;
|
|
|
|
/* 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_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));
|
|
|
|
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_generic, 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->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_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;
|
|
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;
|
|
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 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);
|
|
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;
|
|
}
|