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

624 lines
17 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., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#include <sys/ioctl.h>
#include "rtsp-server.h"
#include "rtsp-client.h"
#define DEFAULT_BACKLOG 5
#define DEFAULT_PORT 8554
enum
{
PROP_0,
PROP_BACKLOG,
PROP_PORT,
PROP_SESSION_POOL,
PROP_MEDIA_MAPPING,
PROP_LAST
};
G_DEFINE_TYPE (GstRTSPServer, gst_rtsp_server, G_TYPE_OBJECT);
static void gst_rtsp_server_get_property (GObject *object, guint propid,
GValue *value, GParamSpec *pspec);
static void gst_rtsp_server_set_property (GObject *object, guint propid,
const GValue *value, GParamSpec *pspec);
static void gst_rtsp_server_finalize (GObject *object);
static GstRTSPClient * default_accept_client (GstRTSPServer *server,
GIOChannel *channel);
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::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::port
*
* The session port of the server. This is the port where the server will
* listen on.
*/
g_object_class_install_property (gobject_class, PROP_PORT,
g_param_spec_int ("port", "Port", "The port the server uses to listen on",
1, 65535, DEFAULT_PORT, 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));
klass->accept_client = default_accept_client;
}
static void
gst_rtsp_server_init (GstRTSPServer * server)
{
server->port = DEFAULT_PORT;
server->backlog = DEFAULT_BACKLOG;
server->session_pool = gst_rtsp_session_pool_new ();
server->media_mapping = gst_rtsp_media_mapping_new ();
}
static void
gst_rtsp_server_finalize (GObject *object)
{
GstRTSPServer *server = GST_RTSP_SERVER (object);
g_object_unref (server->session_pool);
g_object_unref (server->media_mapping);
}
/**
* 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_port:
* @server: a #GstRTSPServer
* @port: the port
*
* Configure @server to accept connections on the given port.
* @port should be a port number between 1 and 65535.
*
* This function must be called before the server is bound.
*/
void
gst_rtsp_server_set_port (GstRTSPServer *server, gint port)
{
g_return_if_fail (GST_IS_RTSP_SERVER (server));
g_return_if_fail (port >= 1 && port <= 65535);
server->port = port;
}
/**
* gst_rtsp_server_get_port:
* @server: a #GstRTSPServer
*
* Get the port number on which the server will accept connections.
*
* Returns: the server port.
*/
gint
gst_rtsp_server_get_port (GstRTSPServer *server)
{
g_return_val_if_fail (GST_IS_RTSP_SERVER (server), -1);
return server->port;
}
/**
* 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));
server->backlog = backlog;
}
/**
* 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)
{
g_return_val_if_fail (GST_IS_RTSP_SERVER (server), -1);
return server->backlog;
}
/**
* 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));
old = server->session_pool;
if (old != pool) {
if (pool)
g_object_ref (pool);
server->session_pool = pool;
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: 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);
if ((result = server->session_pool))
g_object_ref (result);
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));
old = server->media_mapping;
if (old != mapping) {
if (mapping)
g_object_ref (mapping);
server->media_mapping = mapping;
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: 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);
if ((result = server->media_mapping))
g_object_ref (result);
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_PORT:
g_value_set_int (value, gst_rtsp_server_get_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;
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_PORT:
gst_rtsp_server_set_port (server, g_value_get_int (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;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
}
}
/* Prepare a server socket for @server and make it listen on the configured port */
static gboolean
gst_rtsp_server_sink_init_send (GstRTSPServer * server)
{
int ret;
/* create server socket */
if ((server->server_sock.fd = socket (AF_INET, SOCK_STREAM, 0)) == -1)
goto no_socket;
GST_DEBUG_OBJECT (server, "opened sending server socket with fd %d",
server->server_sock.fd);
/* make address reusable */
ret = 1;
if (setsockopt (server->server_sock.fd, SOL_SOCKET, SO_REUSEADDR,
(void *) &ret, sizeof (ret)) < 0)
goto reuse_failed;
/* keep connection alive; avoids SIGPIPE during write */
ret = 1;
if (setsockopt (server->server_sock.fd, SOL_SOCKET, SO_KEEPALIVE,
(void *) &ret, sizeof (ret)) < 0)
goto keepalive_failed;
/* name the socket */
memset (&server->server_sin, 0, sizeof (server->server_sin));
server->server_sin.sin_family = AF_INET; /* network socket */
server->server_sin.sin_port = htons (server->port); /* on port */
server->server_sin.sin_addr.s_addr = htonl (INADDR_ANY); /* for hosts */
/* bind it */
GST_DEBUG_OBJECT (server, "binding server socket to address");
ret = bind (server->server_sock.fd, (struct sockaddr *) &server->server_sin,
sizeof (server->server_sin));
if (ret)
goto bind_failed;
/* set the server socket to nonblocking */
fcntl (server->server_sock.fd, F_SETFL, O_NONBLOCK);
GST_DEBUG_OBJECT (server, "listening on server socket %d with queue of %d",
server->server_sock.fd, server->backlog);
if (listen (server->server_sock.fd, server->backlog) == -1)
goto listen_failed;
GST_DEBUG_OBJECT (server,
"listened on server socket %d, returning from connection setup",
server->server_sock.fd);
g_message ("listening on port %d", server->port);
return TRUE;
/* ERRORS */
no_socket:
{
GST_ERROR_OBJECT (server, "failed to create socket: %s", g_strerror (errno));
return FALSE;
}
reuse_failed:
{
if (server->server_sock.fd >= 0) {
close (server->server_sock.fd);
server->server_sock.fd = -1;
}
GST_ERROR_OBJECT (server, "failed to reuse socket: %s", g_strerror (errno));
return FALSE;
}
keepalive_failed:
{
if (server->server_sock.fd >= 0) {
close (server->server_sock.fd);
server->server_sock.fd = -1;
}
GST_ERROR_OBJECT (server, "failed to configure keepalive socket: %s", g_strerror (errno));
return FALSE;
}
listen_failed:
{
if (server->server_sock.fd >= 0) {
close (server->server_sock.fd);
server->server_sock.fd = -1;
}
GST_ERROR_OBJECT (server, "failed to listen on socket: %s", g_strerror (errno));
return FALSE;
}
bind_failed:
{
if (server->server_sock.fd >= 0) {
close (server->server_sock.fd);
server->server_sock.fd = -1;
}
GST_ERROR_OBJECT (server, "failed to bind on socket: %s", g_strerror (errno));
return FALSE;
}
}
/* default method for creating a new client object in the server to accept and
* handle a client connection on this server */
static GstRTSPClient *
default_accept_client (GstRTSPServer *server, GIOChannel *channel)
{
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_client_set_session_pool (client, server->session_pool);
/* set the session pool that this client should use */
gst_rtsp_client_set_media_mapping (client, server->media_mapping);
/* 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, channel))
goto accept_failed;
return client;
/* ERRORS */
accept_failed:
{
g_error ("Could not accept client on server socket %d: %s (%d)",
server->server_sock.fd, g_strerror (errno), errno);
gst_object_unref (client);
return NULL;
}
}
/**
* gst_rtsp_server_io_func:
* @channel: a #GIOChannel
* @condition: the condition on @source
*
* A default #GIOFunc that creates a new #GstRTSPClient to accept and handle a
* new connection on @channel or @server.
*
* Returns: TRUE if the source could be connected, FALSE if an error occured.
*/
gboolean
gst_rtsp_server_io_func (GIOChannel *channel, GIOCondition condition, GstRTSPServer *server)
{
GstRTSPClient *client = NULL;
GstRTSPServerClass *klass;
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->accept_client)
client = klass->accept_client (server, channel);
if (client == NULL)
goto client_failed;
/* can unref the client now, when the request is finished, it will be
* unreffed async. */
gst_object_unref (client);
}
else {
g_print ("received unknown event %08x", condition);
}
return TRUE;
/* ERRORS */
client_failed:
{
GST_ERROR_OBJECT (server, "failed to create a client");
return FALSE;
}
}
/**
* gst_rtsp_server_get_io_channel:
* @server: a #GstRTSPServer
*
* Create a #GIOChannel for @server.
*
* Returns: the GIOChannel for @server or NULL when an error occured.
*/
GIOChannel *
gst_rtsp_server_get_io_channel (GstRTSPServer *server)
{
g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
if (server->io_channel == NULL) {
if (!gst_rtsp_server_sink_init_send (server))
goto init_failed;
/* create IO channel for the socket */
server->io_channel = g_io_channel_unix_new (server->server_sock.fd);
}
return server->io_channel;
init_failed:
{
return NULL;
}
}
/**
* gst_rtsp_server_create_watch:
* @server: a #GstRTSPServer
*
* Create a #GSource for @server. The new source will have a default
* #GIOFunc of gst_rtsp_server_io_func().
*
* Returns: the #GSource for @server or NULL when an error occured.
*/
GSource *
gst_rtsp_server_create_watch (GstRTSPServer *server)
{
g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
if (server->io_watch == NULL) {
GIOChannel *channel;
channel = gst_rtsp_server_get_io_channel (server);
if (channel == NULL)
goto no_channel;
/* create a watch for reads (new connections) and possible errors */
server->io_watch = g_io_create_watch (channel, G_IO_IN |
G_IO_ERR | G_IO_HUP | G_IO_NVAL);
/* configure the callback */
g_source_set_callback (server->io_watch, (GSourceFunc) gst_rtsp_server_io_func, server, NULL);
}
return server->io_watch;
no_channel:
{
return NULL;
}
}
/**
* gst_rtsp_server_attach:
* @server: a #GstRTSPServer
* @context: a #GMainContext
*
* Attaches @server to @context. When the mainloop for @context is run, the
* server will be dispatched.
*
* 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;
g_return_val_if_fail (GST_IS_RTSP_SERVER (server), 0);
source = gst_rtsp_server_create_watch (server);
if (source == NULL)
goto no_source;
res = g_source_attach (source, context);
return res;
/* ERRORS */
no_source:
{
return 0;
}
}