mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-24 18:51:11 +00:00
1495 lines
39 KiB
C
1495 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.
|
|
*/
|
|
|
|
#include <stdlib.h>
|
|
#include <string.h>
|
|
|
|
#include "rtsp-server.h"
|
|
#include "rtsp-client.h"
|
|
|
|
#define GST_RTSP_SERVER_GET_PRIVATE(obj) \
|
|
(G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTSP_SERVER, GstRTSPServerPrivate))
|
|
|
|
#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;
|
|
gint max_threads;
|
|
gboolean use_client_settings;
|
|
|
|
GSocket *socket;
|
|
|
|
/* sessions on this server */
|
|
GstRTSPSessionPool *session_pool;
|
|
|
|
/* mount points for this server */
|
|
GstRTSPMountPoints *mount_points;
|
|
|
|
/* authentication manager */
|
|
GstRTSPAuth *auth;
|
|
|
|
/* the TLS certificate */
|
|
GTlsCertificate *certificate;
|
|
|
|
/* the clients that are connected */
|
|
GList *clients;
|
|
GQueue loops; /* the main loops used in the threads */
|
|
};
|
|
|
|
#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 DEFAULT_MAX_THREADS 0
|
|
#define DEFAULT_USE_CLIENT_SETTINGS FALSE
|
|
|
|
/* 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_MAX_THREADS,
|
|
PROP_USE_CLIENT_SETTINGS,
|
|
PROP_LAST
|
|
};
|
|
|
|
enum
|
|
{
|
|
SIGNAL_CLIENT_CONNECTED,
|
|
SIGNAL_LAST
|
|
};
|
|
|
|
G_DEFINE_TYPE (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;
|
|
typedef struct _Loop Loop;
|
|
|
|
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 gpointer do_loop (Loop * loop);
|
|
static GstRTSPClient *default_create_client (GstRTSPServer * server);
|
|
static gboolean default_setup_connection (GstRTSPServer * server,
|
|
GstRTSPClient * client, GstRTSPConnection * conn);
|
|
|
|
static void
|
|
gst_rtsp_server_class_init (GstRTSPServerClass * klass)
|
|
{
|
|
GObjectClass *gobject_class;
|
|
|
|
g_type_class_add_private (klass, sizeof (GstRTSPServerPrivate));
|
|
|
|
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));
|
|
/**
|
|
* GstRTSPServer::max-threads:
|
|
*
|
|
* The maximum amount of threads to use for client connections. A value of
|
|
* 0 means to use only the mainloop, -1 means an unlimited amount of
|
|
* threads.
|
|
*/
|
|
g_object_class_install_property (gobject_class, PROP_MAX_THREADS,
|
|
g_param_spec_int ("max-threads", "Max Threads",
|
|
"The maximum amount of threads to use for client connections "
|
|
"(0 = only mainloop, -1 = unlimited)", -1, G_MAXINT,
|
|
DEFAULT_MAX_THREADS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
/**
|
|
* GstRTSPServer::use-client-settings:
|
|
*
|
|
* Use client transport settings (destination, port pair and ttl for
|
|
* multicast. FALSE means that the server settings will be used.
|
|
*/
|
|
g_object_class_install_property (gobject_class, PROP_USE_CLIENT_SETTINGS,
|
|
g_param_spec_boolean ("use-client-settings", "Use Client Settings",
|
|
"Use client settings for ttl, destination and port pair in multicast",
|
|
DEFAULT_USE_CLIENT_SETTINGS,
|
|
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_VOID__OBJECT, G_TYPE_NONE, 1,
|
|
gst_rtsp_client_get_type ());
|
|
|
|
klass->create_client = default_create_client;
|
|
klass->setup_connection = default_setup_connection;
|
|
|
|
klass->pool = g_thread_pool_new ((GFunc) do_loop, klass, -1, FALSE, NULL);
|
|
|
|
GST_DEBUG_CATEGORY_INIT (rtsp_server_debug, "rtspserver", 0, "GstRTSPServer");
|
|
}
|
|
|
|
static void
|
|
gst_rtsp_server_init (GstRTSPServer * server)
|
|
{
|
|
GstRTSPServerPrivate *priv = GST_RTSP_SERVER_GET_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->max_threads = DEFAULT_MAX_THREADS;
|
|
priv->use_client_settings = DEFAULT_USE_CLIENT_SETTINGS;
|
|
g_queue_init (&priv->loops);
|
|
}
|
|
|
|
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);
|
|
|
|
g_object_unref (priv->session_pool);
|
|
g_object_unref (priv->mount_points);
|
|
|
|
if (priv->auth)
|
|
g_object_unref (priv->auth);
|
|
|
|
if (priv->certificate)
|
|
g_object_unref (priv->certificate);
|
|
|
|
g_mutex_clear (&priv->lock);
|
|
|
|
G_OBJECT_CLASS (gst_rtsp_server_parent_class)->finalize (object);
|
|
}
|
|
|
|
/**
|
|
* 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_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: 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.
|
|
*
|
|
* 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: 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: 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): 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: 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): 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: 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): 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_max_threads:
|
|
* @server: a #GstRTSPServer
|
|
* @max_threads: maximum threads
|
|
*
|
|
* Set the maximum threads used by the server to handle client requests.
|
|
* A value of 0 will use the server mainloop, a value of -1 will use an
|
|
* unlimited number of threads.
|
|
*/
|
|
void
|
|
gst_rtsp_server_set_max_threads (GstRTSPServer * server, gint max_threads)
|
|
{
|
|
GstRTSPServerPrivate *priv;
|
|
|
|
g_return_if_fail (GST_IS_RTSP_SERVER (server));
|
|
|
|
priv = server->priv;
|
|
|
|
GST_RTSP_SERVER_LOCK (server);
|
|
priv->max_threads = max_threads;
|
|
GST_RTSP_SERVER_UNLOCK (server);
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_server_get_max_threads:
|
|
* @server: a #GstRTSPServer
|
|
*
|
|
* Get the maximum number of threads used for client connections.
|
|
* See gst_rtsp_server_set_max_threads().
|
|
*
|
|
* Returns: the maximum number of threads.
|
|
*/
|
|
gint
|
|
gst_rtsp_server_get_max_threads (GstRTSPServer * server)
|
|
{
|
|
GstRTSPServerPrivate *priv;
|
|
gint res;
|
|
|
|
g_return_val_if_fail (GST_IS_RTSP_SERVER (server), -1);
|
|
|
|
priv = server->priv;
|
|
|
|
GST_RTSP_SERVER_LOCK (server);
|
|
res = priv->max_threads;
|
|
GST_RTSP_SERVER_UNLOCK (server);
|
|
|
|
return res;
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_server_set_use_client_settings:
|
|
* @server: a #GstRTSPServer
|
|
* @use_client_settings: whether to use client settings for multicast
|
|
*
|
|
* Use client transport settings (destination, port pair and ttl) for
|
|
* multicast.
|
|
* When @use_client_settings is %FALSE, the server settings will be
|
|
* used.
|
|
*/
|
|
void
|
|
gst_rtsp_server_set_use_client_settings (GstRTSPServer * server,
|
|
gboolean use_client_settings)
|
|
{
|
|
GstRTSPServerPrivate *priv;
|
|
|
|
g_return_if_fail (GST_IS_RTSP_SERVER (server));
|
|
|
|
priv = server->priv;
|
|
|
|
GST_RTSP_SERVER_LOCK (server);
|
|
priv->use_client_settings = use_client_settings;
|
|
GST_RTSP_SERVER_UNLOCK (server);
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_server_get_use_client_settings:
|
|
* @server: a #GstRTSPServer
|
|
*
|
|
* Check if client transport settings (destination, port pair and ttl) for
|
|
* multicast will be used.
|
|
*/
|
|
gboolean
|
|
gst_rtsp_server_get_use_client_settings (GstRTSPServer * server)
|
|
{
|
|
GstRTSPServerPrivate *priv;
|
|
gboolean res;
|
|
|
|
g_return_val_if_fail (GST_IS_RTSP_SERVER (server), FALSE);
|
|
|
|
priv = server->priv;
|
|
|
|
GST_RTSP_SERVER_LOCK (server);
|
|
res = priv->use_client_settings;
|
|
GST_RTSP_SERVER_UNLOCK (server);
|
|
|
|
return res;
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_server_set_tls_certificate:
|
|
* @server: a #GstRTSPServer
|
|
* @cert: (allow none): a #GTlsCertificate
|
|
*
|
|
* Set the TLS certificate for the server. Client connections will only
|
|
* be accepted when TLS is negotiated.
|
|
*/
|
|
void
|
|
gst_rtsp_server_set_tls_certificate (GstRTSPServer * server,
|
|
GTlsCertificate * cert)
|
|
{
|
|
GstRTSPServerPrivate *priv;
|
|
GTlsCertificate *old;
|
|
|
|
g_return_if_fail (GST_IS_RTSP_SERVER (server));
|
|
|
|
priv = server->priv;
|
|
|
|
if (cert)
|
|
g_object_ref (cert);
|
|
|
|
GST_RTSP_SERVER_LOCK (server);
|
|
old = priv->certificate;
|
|
priv->certificate = cert;
|
|
GST_RTSP_SERVER_UNLOCK (server);
|
|
|
|
if (old)
|
|
g_object_unref (old);
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_server_get_tls_certificate:
|
|
* @server: a #GstRTSPServer
|
|
*
|
|
* Get the #GTlsCertificate used for negotiating TLS @server.
|
|
*
|
|
* Returns: (transfer full): the #GTlsCertificate of @server. g_object_unref() after
|
|
* usage.
|
|
*/
|
|
GTlsCertificate *
|
|
gst_rtsp_server_get_tls_certificate (GstRTSPServer * server)
|
|
{
|
|
GstRTSPServerPrivate *priv;
|
|
GTlsCertificate *result;
|
|
|
|
g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
|
|
|
|
priv = server->priv;
|
|
|
|
GST_RTSP_SERVER_LOCK (server);
|
|
if ((result = priv->certificate))
|
|
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_MAX_THREADS:
|
|
g_value_set_int (value, gst_rtsp_server_get_max_threads (server));
|
|
break;
|
|
case PROP_USE_CLIENT_SETTINGS:
|
|
g_value_set_boolean (value,
|
|
gst_rtsp_server_get_use_client_settings (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_MAX_THREADS:
|
|
gst_rtsp_server_set_max_threads (server, g_value_get_int (value));
|
|
break;
|
|
case PROP_USE_CLIENT_SETTINGS:
|
|
gst_rtsp_server_set_use_client_settings (server,
|
|
g_value_get_boolean (value));
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
|
|
}
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_server_create_socket:
|
|
* @server: a #GstRTSPServer
|
|
* @cancellable: a #GCancellable
|
|
* @error: 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 occured.
|
|
*/
|
|
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)) {
|
|
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 _Loop
|
|
{
|
|
gint refcnt;
|
|
|
|
GstRTSPServer *server;
|
|
GMainLoop *mainloop;
|
|
GMainContext *mainctx;
|
|
};
|
|
|
|
/* must be called with the lock held */
|
|
static void
|
|
loop_unref (Loop * loop)
|
|
{
|
|
GstRTSPServer *server = loop->server;
|
|
GstRTSPServerPrivate *priv = server->priv;
|
|
|
|
loop->refcnt--;
|
|
|
|
if (loop->refcnt <= 0) {
|
|
g_queue_remove (&priv->loops, loop);
|
|
g_main_loop_quit (loop->mainloop);
|
|
}
|
|
}
|
|
|
|
struct _ClientContext
|
|
{
|
|
GstRTSPServer *server;
|
|
Loop *loop;
|
|
GstRTSPClient *client;
|
|
};
|
|
|
|
static gboolean
|
|
free_client_context (ClientContext * ctx)
|
|
{
|
|
GST_RTSP_SERVER_LOCK (ctx->server);
|
|
if (ctx->loop)
|
|
loop_unref (ctx->loop);
|
|
GST_RTSP_SERVER_UNLOCK (ctx->server);
|
|
|
|
g_object_unref (ctx->client);
|
|
g_slice_free (ClientContext, ctx);
|
|
|
|
return G_SOURCE_REMOVE;
|
|
}
|
|
|
|
static gpointer
|
|
do_loop (Loop * loop)
|
|
{
|
|
GST_INFO ("enter mainloop");
|
|
g_main_loop_run (loop->mainloop);
|
|
GST_INFO ("exit mainloop");
|
|
|
|
g_main_context_unref (loop->mainctx);
|
|
g_main_loop_unref (loop->mainloop);
|
|
g_object_unref (loop->server);
|
|
g_slice_free (Loop, loop);
|
|
|
|
return NULL;
|
|
}
|
|
|
|
/* Must be called with lock held */
|
|
|
|
static Loop *
|
|
gst_rtsp_server_get_main_loop (GstRTSPServer * server)
|
|
{
|
|
GstRTSPServerPrivate *priv = server->priv;
|
|
Loop *loop;
|
|
|
|
if (priv->max_threads > 0 &&
|
|
g_queue_get_length (&priv->loops) >= priv->max_threads) {
|
|
loop = g_queue_pop_head (&priv->loops);
|
|
loop->refcnt++;
|
|
} else {
|
|
GstRTSPServerClass *klass = GST_RTSP_SERVER_GET_CLASS (server);
|
|
|
|
loop = g_slice_new0 (Loop);
|
|
loop->refcnt = 1;
|
|
loop->server = g_object_ref (server);
|
|
loop->mainctx = g_main_context_new ();
|
|
loop->mainloop = g_main_loop_new (loop->mainctx, FALSE);
|
|
|
|
g_thread_pool_push (klass->pool, loop, NULL);
|
|
}
|
|
|
|
g_queue_push_tail (&priv->loops, loop);
|
|
|
|
return loop;
|
|
}
|
|
|
|
static void
|
|
unmanage_client (GstRTSPClient * client, ClientContext * ctx)
|
|
{
|
|
GstRTSPServer *server = ctx->server;
|
|
GstRTSPServerPrivate *priv = server->priv;
|
|
|
|
GST_DEBUG_OBJECT (server, "unmanage client %p", client);
|
|
|
|
g_object_ref (server);
|
|
|
|
GST_RTSP_SERVER_LOCK (server);
|
|
priv->clients = g_list_remove (priv->clients, ctx);
|
|
GST_RTSP_SERVER_UNLOCK (server);
|
|
|
|
if (ctx->loop) {
|
|
GSource *src;
|
|
|
|
src = g_idle_source_new ();
|
|
g_source_set_callback (src, (GSourceFunc) free_client_context, ctx, NULL);
|
|
g_source_attach (src, ctx->loop->mainctx);
|
|
g_source_unref (src);
|
|
} else {
|
|
free_client_context (ctx);
|
|
}
|
|
|
|
g_object_unref (server);
|
|
}
|
|
|
|
/* add the client context to the active list of clients, takes ownership
|
|
* of client */
|
|
static void
|
|
manage_client (GstRTSPServer * server, GstRTSPClient * client)
|
|
{
|
|
ClientContext *ctx;
|
|
GstRTSPServerPrivate *priv = server->priv;
|
|
GMainContext *mainctx;
|
|
|
|
GST_DEBUG_OBJECT (server, "manage client %p", client);
|
|
|
|
ctx = g_slice_new0 (ClientContext);
|
|
ctx->server = server;
|
|
ctx->client = client;
|
|
|
|
GST_RTSP_SERVER_LOCK (server);
|
|
if (priv->max_threads == 0) {
|
|
GSource *source;
|
|
|
|
/* find the context to add the watch */
|
|
if ((source = g_main_current_source ()))
|
|
mainctx = g_source_get_context (source);
|
|
else
|
|
mainctx = NULL;
|
|
} else {
|
|
ctx->loop = gst_rtsp_server_get_main_loop (server);
|
|
mainctx = ctx->loop->mainctx;
|
|
}
|
|
|
|
g_signal_connect (client, "closed", (GCallback) unmanage_client, ctx);
|
|
priv->clients = g_list_prepend (priv->clients, ctx);
|
|
|
|
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);
|
|
/* check if client transport settings for multicast are allowed */
|
|
gst_rtsp_client_set_use_client_settings (client, priv->use_client_settings);
|
|
GST_RTSP_SERVER_UNLOCK (server);
|
|
|
|
return client;
|
|
}
|
|
|
|
static gboolean
|
|
default_setup_connection (GstRTSPServer * server, GstRTSPClient * client,
|
|
GstRTSPConnection * conn)
|
|
{
|
|
GstRTSPServerPrivate *priv = server->priv;
|
|
|
|
GST_RTSP_SERVER_LOCK (server);
|
|
if (priv->certificate) {
|
|
GTlsConnection *tls;
|
|
|
|
/* configure the connection */
|
|
tls = gst_rtsp_connection_get_tls (conn, NULL);
|
|
g_tls_connection_set_certificate (tls, priv->certificate);
|
|
}
|
|
GST_RTSP_SERVER_UNLOCK (server);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_server_transfer_connection:
|
|
* @server: a #GstRTSPServer
|
|
* @socket: a network socket
|
|
* @ip: the IP address of the remote client
|
|
* @port: the port used by the other end
|
|
* @initial_buffer: any initial data that was already read from the socket
|
|
*
|
|
* 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 occured.
|
|
*/
|
|
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);
|
|
|
|
/* set connection on the client now */
|
|
gst_rtsp_client_set_connection (client, conn);
|
|
|
|
/* manage the client connection */
|
|
manage_client (server, client);
|
|
|
|
g_signal_emit (server, gst_rtsp_server_signals[SIGNAL_CLIENT_CONNECTED], 0,
|
|
client);
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
client_failed:
|
|
{
|
|
GST_ERROR_OBJECT (server, "failed to create a client");
|
|
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);
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_server_io_func:
|
|
* @socket: a #GSocket
|
|
* @condition: the condition on @source
|
|
* @server: 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 occured.
|
|
*/
|
|
gboolean
|
|
gst_rtsp_server_io_func (GSocket * socket, GIOCondition condition,
|
|
GstRTSPServer * server)
|
|
{
|
|
GstRTSPClient *client = NULL;
|
|
GstRTSPServerClass *klass;
|
|
GstRTSPResult res;
|
|
GstRTSPConnection *conn = NULL;
|
|
|
|
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->create_client)
|
|
client = klass->create_client (server);
|
|
if (client == NULL)
|
|
goto client_failed;
|
|
|
|
/* a new client connected. */
|
|
GST_RTSP_CHECK (gst_rtsp_connection_accept (socket, &conn, NULL),
|
|
accept_failed);
|
|
|
|
if (klass->setup_connection)
|
|
if (!klass->setup_connection (server, client, conn))
|
|
goto setup_failed;
|
|
|
|
/* set connection on the client now */
|
|
gst_rtsp_client_set_connection (client, conn);
|
|
|
|
/* manage the client connection */
|
|
manage_client (server, client);
|
|
|
|
g_signal_emit (server, gst_rtsp_server_signals[SIGNAL_CLIENT_CONNECTED], 0,
|
|
client);
|
|
} else {
|
|
GST_WARNING_OBJECT (server, "received unknown event %08x", condition);
|
|
}
|
|
return G_SOURCE_CONTINUE;
|
|
|
|
/* ERRORS */
|
|
client_failed:
|
|
{
|
|
GST_ERROR_OBJECT (server, "failed to create a client");
|
|
return G_SOURCE_CONTINUE;
|
|
}
|
|
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);
|
|
g_object_unref (client);
|
|
return G_SOURCE_CONTINUE;
|
|
}
|
|
setup_failed:
|
|
{
|
|
GST_ERROR_OBJECT (server, "failed to setup client connection");
|
|
gst_rtsp_connection_free (conn);
|
|
g_object_unref (client);
|
|
return G_SOURCE_CONTINUE;
|
|
}
|
|
}
|
|
|
|
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: a #GCancellable or %NULL.
|
|
* @error: 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().
|
|
*
|
|
* Returns: the #GSource for @server or NULL when an error occured. 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.
|
|
*
|
|
* 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;
|
|
}
|
|
}
|