gstreamer/gst/rtsp-server/rtsp-server.c
2012-11-13 12:05:42 +01:00

1176 lines
31 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 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 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_MEDIA_MAPPING,
PROP_MAX_THREADS,
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;
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 (ClientContext * ctx);
static GstRTSPClient *default_create_client (GstRTSPServer * server);
static gboolean default_accept_client (GstRTSPServer * server,
GstRTSPClient * client, GSocket * socket, GError ** error);
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::media-mapping:
*
* The media mapping to use for this server. By default the server has no
* media mapping and thus cannot map urls to media streams.
*/
g_object_class_install_property (gobject_class, PROP_MEDIA_MAPPING,
g_param_spec_object ("media-mapping", "Media Mapping",
"The media mapping to use for client session",
GST_TYPE_RTSP_MEDIA_MAPPING,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* 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));
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->accept_client = default_accept_client;
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)
{
g_mutex_init (&server->lock);
server->address = g_strdup (DEFAULT_ADDRESS);
server->service = g_strdup (DEFAULT_SERVICE);
server->socket = NULL;
server->backlog = DEFAULT_BACKLOG;
server->session_pool = gst_rtsp_session_pool_new ();
server->media_mapping = gst_rtsp_media_mapping_new ();
server->max_threads = DEFAULT_MAX_THREADS;
}
static void
gst_rtsp_server_finalize (GObject * object)
{
GstRTSPServer *server = GST_RTSP_SERVER (object);
GST_DEBUG_OBJECT (server, "finalize server");
g_free (server->address);
g_free (server->service);
if (server->socket)
g_object_unref (server->socket);
g_object_unref (server->session_pool);
g_object_unref (server->media_mapping);
if (server->auth)
g_object_unref (server->auth);
g_mutex_clear (&server->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)
{
g_return_if_fail (GST_IS_RTSP_SERVER (server));
g_return_if_fail (address != NULL);
GST_RTSP_SERVER_LOCK (server);
g_free (server->address);
server->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)
{
gchar *result;
g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
GST_RTSP_SERVER_LOCK (server);
result = g_strdup (server->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)
{
GSocketAddress *address;
int result = -1;
g_return_val_if_fail (GST_IS_RTSP_SERVER (server), result);
GST_RTSP_SERVER_LOCK (server);
if (server->socket == NULL)
goto out;
address = g_socket_get_local_address (server->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)
{
g_return_if_fail (GST_IS_RTSP_SERVER (server));
g_return_if_fail (service != NULL);
GST_RTSP_SERVER_LOCK (server);
g_free (server->service);
server->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)
{
gchar *result;
g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
GST_RTSP_SERVER_LOCK (server);
result = g_strdup (server->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)
{
g_return_if_fail (GST_IS_RTSP_SERVER (server));
GST_RTSP_SERVER_LOCK (server);
server->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)
{
gint result;
g_return_val_if_fail (GST_IS_RTSP_SERVER (server), -1);
GST_RTSP_SERVER_LOCK (server);
result = server->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)
{
GstRTSPSessionPool *old;
g_return_if_fail (GST_IS_RTSP_SERVER (server));
if (pool)
g_object_ref (pool);
GST_RTSP_SERVER_LOCK (server);
old = server->session_pool;
server->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)
{
GstRTSPSessionPool *result;
g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
GST_RTSP_SERVER_LOCK (server);
if ((result = server->session_pool))
g_object_ref (result);
GST_RTSP_SERVER_UNLOCK (server);
return result;
}
/**
* gst_rtsp_server_set_media_mapping:
* @server: a #GstRTSPServer
* @mapping: a #GstRTSPMediaMapping
*
* configure @mapping to be used as the media mapping of @server.
*/
void
gst_rtsp_server_set_media_mapping (GstRTSPServer * server,
GstRTSPMediaMapping * mapping)
{
GstRTSPMediaMapping *old;
g_return_if_fail (GST_IS_RTSP_SERVER (server));
if (mapping)
g_object_ref (mapping);
GST_RTSP_SERVER_LOCK (server);
old = server->media_mapping;
server->media_mapping = mapping;
GST_RTSP_SERVER_UNLOCK (server);
if (old)
g_object_unref (old);
}
/**
* gst_rtsp_server_get_media_mapping:
* @server: a #GstRTSPServer
*
* Get the #GstRTSPMediaMapping used as the media mapping of @server.
*
* Returns: (transfer full): the #GstRTSPMediaMapping of @server. g_object_unref() after
* usage.
*/
GstRTSPMediaMapping *
gst_rtsp_server_get_media_mapping (GstRTSPServer * server)
{
GstRTSPMediaMapping *result;
g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
GST_RTSP_SERVER_LOCK (server);
if ((result = server->media_mapping))
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)
{
GstRTSPAuth *old;
g_return_if_fail (GST_IS_RTSP_SERVER (server));
if (auth)
g_object_ref (auth);
GST_RTSP_SERVER_LOCK (server);
old = server->auth;
server->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)
{
GstRTSPAuth *result;
g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
GST_RTSP_SERVER_LOCK (server);
if ((result = server->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)
{
g_return_if_fail (GST_IS_RTSP_SERVER (server));
GST_RTSP_SERVER_LOCK (server);
server->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)
{
gint res;
g_return_val_if_fail (GST_IS_RTSP_SERVER (server), -1);
GST_RTSP_SERVER_LOCK (server);
res = server->max_threads;
GST_RTSP_SERVER_UNLOCK (server);
return res;
}
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_MEDIA_MAPPING:
g_value_take_object (value, gst_rtsp_server_get_media_mapping (server));
break;
case PROP_MAX_THREADS:
g_value_set_int (value, gst_rtsp_server_get_max_threads (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_MEDIA_MAPPING:
gst_rtsp_server_set_media_mapping (server, g_value_get_object (value));
break;
case PROP_MAX_THREADS:
gst_rtsp_server_set_max_threads (server, g_value_get_int (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)
{
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);
GST_RTSP_SERVER_LOCK (server);
GST_DEBUG_OBJECT (server, "getting address info of %s/%s", server->address,
server->service);
/* resolve the server IP address */
port = atoi (server->service);
if (port != 0 || !strcmp (server->service, "0"))
conn = g_network_address_new (server->address, port);
else
conn = g_network_service_new (server->service, "tcp", server->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);
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, server->backlog);
if (!g_socket_listen (socket, error))
goto listen_failed;
GST_DEBUG_OBJECT (server, "listening on server socket %p with queue of %d",
socket, server->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;
GMainLoop *loop;
GMainContext *context;
GstRTSPClient *client;
};
static void
free_client_context (ClientContext * ctx)
{
g_main_context_unref (ctx->context);
if (ctx->loop)
g_main_loop_unref (ctx->loop);
g_object_unref (ctx->client);
g_slice_free (ClientContext, ctx);
}
static gpointer
do_loop (ClientContext * ctx)
{
GST_INFO ("enter mainloop");
g_main_loop_run (ctx->loop);
GST_INFO ("exit mainloop");
free_client_context (ctx);
return NULL;
}
static void
unmanage_client (GstRTSPClient * client, ClientContext * ctx)
{
GstRTSPServer *server = ctx->server;
GST_DEBUG_OBJECT (server, "unmanage client %p", client);
g_object_ref (server);
gst_rtsp_client_set_server (client, NULL);
GST_RTSP_SERVER_LOCK (server);
server->clients = g_list_remove (server->clients, ctx);
GST_RTSP_SERVER_UNLOCK (server);
if (ctx->loop)
g_main_loop_quit (ctx->loop);
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;
GST_DEBUG_OBJECT (server, "manage client %p", client);
gst_rtsp_client_set_server (client, server);
ctx = g_slice_new0 (ClientContext);
ctx->server = server;
ctx->client = client;
if (server->max_threads == 0) {
GSource *source;
/* find the context to add the watch */
if ((source = g_main_current_source ()))
ctx->context = g_main_context_ref (g_source_get_context (source));
else
ctx->context = NULL;
} else {
ctx->context = g_main_context_new ();
ctx->loop = g_main_loop_new (ctx->context, TRUE);
}
gst_rtsp_client_attach (client, ctx->context);
GST_RTSP_SERVER_LOCK (server);
g_signal_connect (client, "closed", (GCallback) unmanage_client, ctx);
server->clients = g_list_prepend (server->clients, ctx);
GST_RTSP_SERVER_UNLOCK (server);
if (ctx->loop) {
GstRTSPServerClass *klass = GST_RTSP_SERVER_GET_CLASS (server);
g_thread_pool_push (klass->pool, ctx, NULL);
}
}
static GstRTSPClient *
default_create_client (GstRTSPServer * server)
{
GstRTSPClient *client;
/* a new client connected, create a session to handle the client. */
client = gst_rtsp_client_new ();
/* set the session pool that this client should use */
GST_RTSP_SERVER_LOCK (server);
gst_rtsp_client_set_session_pool (client, server->session_pool);
/* set the media mapping that this client should use */
gst_rtsp_client_set_media_mapping (client, server->media_mapping);
/* set authentication manager */
gst_rtsp_client_set_auth (client, server->auth);
GST_RTSP_SERVER_UNLOCK (server);
return client;
}
/* default method for creating a new client object in the server to accept and
* handle a client connection on this server */
static gboolean
default_accept_client (GstRTSPServer * server, GstRTSPClient * client,
GSocket * socket, GError ** error)
{
/* accept connections for that client, this function returns after accepting
* the connection and will run the remainder of the communication with the
* client asyncronously. */
if (!gst_rtsp_client_accept (client, socket, NULL, error))
goto accept_failed;
return TRUE;
/* ERRORS */
accept_failed:
{
GST_ERROR_OBJECT (server,
"Could not accept client on server : %s", (*error)->message);
return FALSE;
}
}
/**
* 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;
GError *error = NULL;
klass = GST_RTSP_SERVER_GET_CLASS (server);
if (klass->create_client)
client = klass->create_client (server);
if (client == NULL)
goto client_failed;
/* a new client connected, create a client object to handle the client. */
if (!gst_rtsp_client_use_socket (client, socket, ip,
port, initial_buffer, &error)) {
goto transfer_failed;
}
/* 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;
}
transfer_failed:
{
GST_ERROR_OBJECT (server, "failed to accept client: %s", error->message);
g_error_free (error);
g_object_unref (client);
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)
{
gboolean result = TRUE;
GstRTSPClient *client = NULL;
GstRTSPServerClass *klass;
GError *error = NULL;
if (condition & G_IO_IN) {
klass = GST_RTSP_SERVER_GET_CLASS (server);
if (klass->create_client)
client = klass->create_client (server);
if (client == NULL)
goto client_failed;
/* a new client connected, create a client object to handle the client. */
if (klass->accept_client)
result = klass->accept_client (server, client, socket, &error);
if (!result)
goto accept_failed;
/* 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 TRUE;
/* ERRORS */
client_failed:
{
GST_ERROR_OBJECT (server, "failed to create a client");
return FALSE;
}
accept_failed:
{
GST_ERROR_OBJECT (server, "failed to accept client: %s", error->message);
g_error_free (error);
g_object_unref (client);
return FALSE;
}
}
static void
watch_destroyed (GstRTSPServer * server)
{
GST_DEBUG_OBJECT (server, "source destroyed");
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)
{
GSocket *socket, *old;
GSource *source;
g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
socket = gst_rtsp_server_create_socket (server, NULL, error);
if (socket == NULL)
goto no_socket;
GST_RTSP_SERVER_LOCK (server);
old = server->socket;
server->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;
}
}