gstreamer/gst/rtsp-server/rtsp-server.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;
}
}