* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
+/**
+ * SECTION:rtsp-client
+ * @short_description: A client connection state
+ * @see_also: #GstRTSPServer, #GstRTSPThreadPool
+ *
+ * The client object handles the connection with a client for as long as a TCP
+ * connection is open.
+ *
+ * A #GstRTSPClient is created by #GstRTSPServer when a new connection is
+ * accepted and it inherits the #GstRTSPMountPoints, #GstRTSPSessionPool,
+ * #GstRTSPAuth and #GstRTSPThreadPool from the server.
+ *
+ * The client connection should be configured with the #GstRTSPConnection using
+ * gst_rtsp_client_set_connection() before it can be attached to a #GMainContext
+ * using gst_rtsp_client_attach(). From then on the client will handle requests
+ * on the connection.
+ *
+ * Use gst_rtsp_client_session_filter() to iterate or modify all the
+ * #GstRTSPSession objects managed by the client object.
+ *
+ * Last reviewed on 2013-07-11 (1.0.0)
+ */
#include <stdio.h>
#include <string.h>
#include "rtsp-sdp.h"
#include "rtsp-params.h"
+#define GST_RTSP_CLIENT_GET_PRIVATE(obj) \
+ (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTSP_CLIENT, GstRTSPClientPrivate))
+
+/* locking order:
+ * send_lock, lock, tunnels_lock
+ */
+
+struct _GstRTSPClientPrivate
+{
+ GMutex lock; /* protects everything else */
+ GMutex send_lock;
+ GstRTSPConnection *connection;
+ GstRTSPWatch *watch;
+ guint close_seq;
+ gchar *server_ip;
+ gboolean is_ipv6;
+
+ GstRTSPClientSendFunc send_func; /* protected by send_lock */
+ gpointer send_data; /* protected by send_lock */
+ GDestroyNotify send_notify; /* protected by send_lock */
+
+ GstRTSPSessionPool *session_pool;
+ GstRTSPMountPoints *mount_points;
+ GstRTSPAuth *auth;
+ GstRTSPThreadPool *thread_pool;
+
+ /* used to cache the media in the last requested DESCRIBE so that
+ * we can pick it up in the next SETUP immediately */
+ gchar *path;
+ GstRTSPMedia *media;
+
+ GList *transports;
+ GList *sessions;
+};
+
static GMutex tunnels_lock;
-static GHashTable *tunnels;
+static GHashTable *tunnels; /* protected by tunnels_lock */
#define DEFAULT_SESSION_POOL NULL
#define DEFAULT_MOUNT_POINTS NULL
-#define DEFAULT_USE_CLIENT_SETTINGS FALSE
enum
{
PROP_0,
PROP_SESSION_POOL,
PROP_MOUNT_POINTS,
- PROP_USE_CLIENT_SETTINGS,
PROP_LAST
};
SIGNAL_TEARDOWN_REQUEST,
SIGNAL_SET_PARAMETER_REQUEST,
SIGNAL_GET_PARAMETER_REQUEST,
+ SIGNAL_HANDLE_RESPONSE,
SIGNAL_LAST
};
static void client_session_finalized (GstRTSPClient * client,
GstRTSPSession * session);
static void unlink_session_transports (GstRTSPClient * client,
- GstRTSPSession * session, GstRTSPSessionMedia * media);
+ GstRTSPSession * session, GstRTSPSessionMedia * sessmedia);
+static gboolean default_configure_client_media (GstRTSPClient * client,
+ GstRTSPMedia * media, GstRTSPStream * stream, GstRTSPContext * ctx);
+static gboolean default_configure_client_transport (GstRTSPClient * client,
+ GstRTSPContext * ctx, GstRTSPTransport * ct);
+static GstRTSPResult default_params_set (GstRTSPClient * client,
+ GstRTSPContext * ctx);
+static GstRTSPResult default_params_get (GstRTSPClient * client,
+ GstRTSPContext * ctx);
+static gchar *default_make_path_from_uri (GstRTSPClient * client,
+ const GstRTSPUrl * uri);
G_DEFINE_TYPE (GstRTSPClient, gst_rtsp_client, G_TYPE_OBJECT);
{
GObjectClass *gobject_class;
+ g_type_class_add_private (klass, sizeof (GstRTSPClientPrivate));
+
gobject_class = G_OBJECT_CLASS (klass);
gobject_class->get_property = gst_rtsp_client_get_property;
gobject_class->finalize = gst_rtsp_client_finalize;
klass->create_sdp = create_sdp;
+ klass->configure_client_media = default_configure_client_media;
+ klass->configure_client_transport = default_configure_client_transport;
+ klass->params_set = default_params_set;
+ klass->params_get = default_params_get;
+ klass->make_path_from_uri = default_make_path_from_uri;
g_object_class_install_property (gobject_class, PROP_SESSION_POOL,
g_param_spec_object ("session-pool", "Session Pool",
GST_TYPE_RTSP_MOUNT_POINTS,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
- 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 and destination in multicast",
- DEFAULT_USE_CLIENT_SETTINGS,
- G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
-
gst_rtsp_client_signals[SIGNAL_CLOSED] =
g_signal_new ("closed", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
G_STRUCT_OFFSET (GstRTSPClientClass, closed), NULL, NULL,
get_parameter_request), NULL, NULL, g_cclosure_marshal_VOID__POINTER,
G_TYPE_NONE, 1, G_TYPE_POINTER);
+ gst_rtsp_client_signals[SIGNAL_HANDLE_RESPONSE] =
+ g_signal_new ("handle-response", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
+ handle_response), NULL, NULL, g_cclosure_marshal_VOID__POINTER,
+ G_TYPE_NONE, 1, G_TYPE_POINTER);
+
tunnels =
g_hash_table_new_full (g_str_hash, g_str_equal, g_free, g_object_unref);
g_mutex_init (&tunnels_lock);
static void
gst_rtsp_client_init (GstRTSPClient * client)
{
- g_mutex_init (&client->lock);
- client->use_client_settings = DEFAULT_USE_CLIENT_SETTINGS;
- client->close_response_seq = 0;
+ GstRTSPClientPrivate *priv = GST_RTSP_CLIENT_GET_PRIVATE (client);
+
+ client->priv = priv;
+
+ g_mutex_init (&priv->lock);
+ g_mutex_init (&priv->send_lock);
+ priv->close_seq = 0;
+}
+
+static GstRTSPFilterResult
+filter_session (GstRTSPSession * sess, GstRTSPSessionMedia * sessmedia,
+ gpointer user_data)
+{
+ GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
+
+ gst_rtsp_session_media_set_state (sessmedia, GST_STATE_NULL);
+ unlink_session_transports (client, sess, sessmedia);
+
+ /* unmanage the media in the session */
+ return GST_RTSP_FILTER_REMOVE;
}
static void
client_unlink_session (GstRTSPClient * client, GstRTSPSession * session)
{
/* unlink all media managed in this session */
- while (session->medias) {
- GstRTSPSessionMedia *media = session->medias->data;
+ gst_rtsp_session_filter (session, filter_session, client);
+}
+
+static void
+client_watch_session (GstRTSPClient * client, GstRTSPSession * session)
+{
+ GstRTSPClientPrivate *priv = client->priv;
+ GList *walk;
+
+ for (walk = priv->sessions; walk; walk = g_list_next (walk)) {
+ GstRTSPSession *msession = (GstRTSPSession *) walk->data;
- gst_rtsp_session_media_set_state (media, GST_STATE_NULL);
- unlink_session_transports (client, session, media);
- /* unmanage the media in the session. this will modify session->medias */
- gst_rtsp_session_release_media (session, media);
+ /* we already know about this session */
+ if (msession == session)
+ return;
}
+
+ GST_INFO ("watching session %p", session);
+
+ g_object_weak_ref (G_OBJECT (session), (GWeakNotify) client_session_finalized,
+ client);
+ priv->sessions = g_list_prepend (priv->sessions, session);
+}
+
+static void
+client_unwatch_session (GstRTSPClient * client, GstRTSPSession * session)
+{
+ GstRTSPClientPrivate *priv = client->priv;
+
+ GST_INFO ("unwatching session %p", session);
+
+ g_object_weak_unref (G_OBJECT (session),
+ (GWeakNotify) client_session_finalized, client);
+ priv->sessions = g_list_remove (priv->sessions, session);
+}
+
+static void
+client_cleanup_session (GstRTSPClient * client, GstRTSPSession * session)
+{
+ g_object_weak_unref (G_OBJECT (session),
+ (GWeakNotify) client_session_finalized, client);
+ client_unlink_session (client, session);
}
static void
client_cleanup_sessions (GstRTSPClient * client)
{
+ GstRTSPClientPrivate *priv = client->priv;
GList *sessions;
/* remove weak-ref from sessions */
- for (sessions = client->sessions; sessions; sessions = g_list_next (sessions)) {
- GstRTSPSession *session = (GstRTSPSession *) sessions->data;
- g_object_weak_unref (G_OBJECT (session),
- (GWeakNotify) client_session_finalized, client);
- client_unlink_session (client, session);
+ for (sessions = priv->sessions; sessions; sessions = g_list_next (sessions)) {
+ client_cleanup_session (client, (GstRTSPSession *) sessions->data);
}
- g_list_free (client->sessions);
- client->sessions = NULL;
+ g_list_free (priv->sessions);
+ priv->sessions = NULL;
}
/* A client is finalized when the connection is broken */
gst_rtsp_client_finalize (GObject * obj)
{
GstRTSPClient *client = GST_RTSP_CLIENT (obj);
+ GstRTSPClientPrivate *priv = client->priv;
GST_INFO ("finalize client %p", client);
- if (client->watch)
- g_source_destroy ((GSource *) client->watch);
+ gst_rtsp_client_set_send_func (client, NULL, NULL, NULL);
+
+ if (priv->watch)
+ g_source_destroy ((GSource *) priv->watch);
client_cleanup_sessions (client);
- gst_rtsp_connection_free (client->connection);
- if (client->session_pool)
- g_object_unref (client->session_pool);
- if (client->mount_points)
- g_object_unref (client->mount_points);
- if (client->auth)
- g_object_unref (client->auth);
+ if (priv->connection)
+ gst_rtsp_connection_free (priv->connection);
+ if (priv->session_pool)
+ g_object_unref (priv->session_pool);
+ if (priv->mount_points)
+ g_object_unref (priv->mount_points);
+ if (priv->auth)
+ g_object_unref (priv->auth);
+ if (priv->thread_pool)
+ g_object_unref (priv->thread_pool);
- if (client->uri)
- gst_rtsp_url_free (client->uri);
- if (client->media) {
- gst_rtsp_media_unprepare (client->media);
- g_object_unref (client->media);
+ if (priv->path)
+ g_free (priv->path);
+ if (priv->media) {
+ gst_rtsp_media_unprepare (priv->media);
+ g_object_unref (priv->media);
}
- g_free (client->server_ip);
- g_mutex_clear (&client->lock);
+ g_free (priv->server_ip);
+ g_mutex_clear (&priv->lock);
+ g_mutex_clear (&priv->send_lock);
G_OBJECT_CLASS (gst_rtsp_client_parent_class)->finalize (obj);
}
case PROP_MOUNT_POINTS:
g_value_take_object (value, gst_rtsp_client_get_mount_points (client));
break;
- case PROP_USE_CLIENT_SETTINGS:
- g_value_set_boolean (value,
- gst_rtsp_client_get_use_client_settings (client));
- break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
}
case PROP_MOUNT_POINTS:
gst_rtsp_client_set_mount_points (client, g_value_get_object (value));
break;
- case PROP_USE_CLIENT_SETTINGS:
- gst_rtsp_client_set_use_client_settings (client,
- g_value_get_boolean (value));
- break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
}
}
static void
-send_response (GstRTSPClient * client, GstRTSPSession * session,
- GstRTSPMessage * response, gboolean close)
+send_message (GstRTSPClient * client, GstRTSPSession * session,
+ GstRTSPMessage * message, gboolean close)
{
- gst_rtsp_message_add_header (response, GST_RTSP_HDR_SERVER,
+ GstRTSPClientPrivate *priv = client->priv;
+
+ gst_rtsp_message_add_header (message, GST_RTSP_HDR_SERVER,
"GStreamer RTSP server");
/* remove any previous header */
- gst_rtsp_message_remove_header (response, GST_RTSP_HDR_SESSION, -1);
+ gst_rtsp_message_remove_header (message, GST_RTSP_HDR_SESSION, -1);
/* add the new session header for new session ids */
if (session) {
- gst_rtsp_message_take_header (response, GST_RTSP_HDR_SESSION,
+ gst_rtsp_message_take_header (message, GST_RTSP_HDR_SESSION,
gst_rtsp_session_get_header (session));
}
if (gst_debug_category_get_threshold (rtsp_client_debug) >= GST_LEVEL_LOG) {
- gst_rtsp_message_dump (response);
+ gst_rtsp_message_dump (message);
}
- if (close) {
- gst_rtsp_message_add_header (response, GST_RTSP_HDR_CONNECTION, "close");
- }
- /* send the response and store the seq number so we can wait until it's
- * written to the client to close the connection */
- gst_rtsp_watch_send_message (client->watch, response, close ?
- &client->close_response_seq : NULL);
- gst_rtsp_message_unset (response);
-}
+ if (close)
+ gst_rtsp_message_add_header (message, GST_RTSP_HDR_CONNECTION, "close");
-static void
-send_generic_response (GstRTSPClient * client, GstRTSPStatusCode code,
- GstRTSPClientState * state)
-{
- gst_rtsp_message_init_response (state->response, code,
- gst_rtsp_status_as_text (code), state->request);
+ g_mutex_lock (&priv->send_lock);
+ if (priv->send_func)
+ priv->send_func (client, message, close, priv->send_data);
+ g_mutex_unlock (&priv->send_lock);
- send_response (client, NULL, state->response, FALSE);
+ gst_rtsp_message_unset (message);
}
static void
-handle_unauthorized_request (GstRTSPClient * client, GstRTSPAuth * auth,
- GstRTSPClientState * state)
+send_generic_response (GstRTSPClient * client, GstRTSPStatusCode code,
+ GstRTSPContext * ctx)
{
- gst_rtsp_message_init_response (state->response, GST_RTSP_STS_UNAUTHORIZED,
- gst_rtsp_status_as_text (GST_RTSP_STS_UNAUTHORIZED), state->request);
-
- if (auth) {
- /* and let the authentication manager setup the auth tokens */
- gst_rtsp_auth_setup_auth (auth, client, 0, state);
- }
+ gst_rtsp_message_init_response (ctx->response, code,
+ gst_rtsp_status_as_text (code), ctx->request);
- send_response (client, state->session, state->response, FALSE);
+ send_message (client, NULL, ctx->response, FALSE);
}
-
static gboolean
-compare_uri (const GstRTSPUrl * uri1, const GstRTSPUrl * uri2)
+paths_are_equal (const gchar * path1, const gchar * path2, gint len2)
{
- if (uri1 == NULL || uri2 == NULL)
+ if (path1 == NULL || path2 == NULL)
return FALSE;
- if (strcmp (uri1->abspath, uri2->abspath))
+ if (strlen (path1) != len2)
+ return FALSE;
+
+ if (strncmp (path1, path2, len2))
return FALSE;
return TRUE;
* but is cached for when the same client (without breaking the connection) is
* doing a setup for the exact same url. */
static GstRTSPMedia *
-find_media (GstRTSPClient * client, GstRTSPClientState * state)
+find_media (GstRTSPClient * client, GstRTSPContext * ctx, gchar * path,
+ gint * matched)
{
+ GstRTSPClientPrivate *priv = client->priv;
GstRTSPMediaFactory *factory;
GstRTSPMedia *media;
- GstRTSPAuth *auth;
+ gint path_len;
- if (!compare_uri (client->uri, state->uri)) {
- /* remove any previously cached values before we try to construct a new
- * media for uri */
- if (client->uri)
- gst_rtsp_url_free (client->uri);
- client->uri = NULL;
- if (client->media) {
- gst_rtsp_media_unprepare (client->media);
- g_object_unref (client->media);
- }
- client->media = NULL;
+ /* find the longest matching factory for the uri first */
+ if (!(factory = gst_rtsp_mount_points_match (priv->mount_points,
+ path, matched)))
+ goto no_factory;
- if (!client->mount_points)
- goto no_mount_points;
+ ctx->factory = factory;
- /* find the factory for the uri first */
- if (!(factory =
- gst_rtsp_mount_points_find_factory (client->mount_points,
- state->uri)))
- goto no_factory;
+ if (!gst_rtsp_auth_check (GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_ACCESS))
+ goto no_factory_access;
- state->factory = factory;
+ if (!gst_rtsp_auth_check (GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_CONSTRUCT))
+ goto not_authorized;
+
+ if (matched)
+ path_len = *matched;
+ else
+ path_len = strlen (path);
- /* check if we have access to the factory */
- if ((auth = gst_rtsp_media_factory_get_auth (factory))) {
- if (!gst_rtsp_auth_check (auth, client, 0, state))
- goto not_allowed;
+ if (!paths_are_equal (priv->path, path, path_len)) {
+ GstRTSPThread *thread;
- g_object_unref (auth);
+ /* remove any previously cached values before we try to construct a new
+ * media for uri */
+ if (priv->path)
+ g_free (priv->path);
+ priv->path = NULL;
+ if (priv->media) {
+ gst_rtsp_media_unprepare (priv->media);
+ g_object_unref (priv->media);
}
+ priv->media = NULL;
/* prepare the media and add it to the pipeline */
- if (!(media = gst_rtsp_media_factory_construct (factory, state->uri)))
+ if (!(media = gst_rtsp_media_factory_construct (factory, ctx->uri)))
goto no_media;
- g_object_unref (factory);
- factory = NULL;
- state->factory = NULL;
+ ctx->media = media;
- /* set ipv6 on the media before preparing */
- media->is_ipv6 = client->is_ipv6;
- state->media = media;
+ thread = gst_rtsp_thread_pool_get_thread (priv->thread_pool,
+ GST_RTSP_THREAD_TYPE_MEDIA, ctx);
+ if (thread == NULL)
+ goto no_thread;
/* prepare the media */
- if (!(gst_rtsp_media_prepare (media)))
+ if (!(gst_rtsp_media_prepare (media, thread)))
goto no_prepare;
/* now keep track of the uri and the media */
- client->uri = gst_rtsp_url_copy (state->uri);
- client->media = media;
+ priv->path = g_strndup (path, path_len);
+ priv->media = media;
} else {
- /* we have seen this uri before, used cached media */
- media = client->media;
- state->media = media;
- GST_INFO ("reusing cached media %p", media);
+ /* we have seen this path before, used cached media */
+ media = priv->media;
+ ctx->media = media;
+ GST_INFO ("reusing cached media %p for path %s", media, priv->path);
}
+ g_object_unref (factory);
+ ctx->factory = NULL;
+
if (media)
g_object_ref (media);
return media;
/* ERRORS */
-no_mount_points:
+no_factory:
{
- GST_ERROR ("client %p: no mount points configured", client);
- send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
+ GST_ERROR ("client %p: no factory for path %s", client, path);
+ send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
return NULL;
}
-no_factory:
+no_factory_access:
{
- GST_ERROR ("client %p: no factory for uri", client);
- send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
+ GST_ERROR ("client %p: not authorized to see factory path %s", client,
+ path);
+ /* error reply is already sent */
return NULL;
}
-not_allowed:
+not_authorized:
{
- GST_ERROR ("client %p: unauthorized request", client);
- handle_unauthorized_request (client, auth, state);
- g_object_unref (factory);
- g_object_unref (auth);
+ GST_ERROR ("client %p: not authorized for factory path %s", client, path);
+ /* error reply is already sent */
return NULL;
}
no_media:
{
GST_ERROR ("client %p: can't create media", client);
- send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, state);
+ send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
+ g_object_unref (factory);
+ ctx->factory = NULL;
+ return NULL;
+ }
+no_thread:
+ {
+ GST_ERROR ("client %p: can't create thread", client);
+ send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
+ g_object_unref (media);
+ ctx->media = NULL;
g_object_unref (factory);
+ ctx->factory = NULL;
return NULL;
}
no_prepare:
{
GST_ERROR ("client %p: can't prepare media", client);
- send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, state);
+ send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
g_object_unref (media);
+ ctx->media = NULL;
+ g_object_unref (factory);
+ ctx->factory = NULL;
return NULL;
}
}
static gboolean
do_send_data (GstBuffer * buffer, guint8 channel, GstRTSPClient * client)
{
+ GstRTSPClientPrivate *priv = client->priv;
GstRTSPMessage message = { 0 };
GstMapInfo map_info;
guint8 *data;
gst_rtsp_message_take_body (&message, map_info.data, map_info.size);
- /* FIXME, client->watch could have been finalized here, we need to keep an
- * extra refcount to the watch. */
- gst_rtsp_watch_send_message (client->watch, &message, NULL);
+ g_mutex_lock (&priv->send_lock);
+ if (priv->send_func)
+ priv->send_func (client, &message, FALSE, priv->send_data);
+ g_mutex_unlock (&priv->send_lock);
gst_rtsp_message_steal_body (&message, &data, &usize);
gst_buffer_unmap (buffer, &map_info);
link_transport (GstRTSPClient * client, GstRTSPSession * session,
GstRTSPStreamTransport * trans)
{
+ GstRTSPClientPrivate *priv = client->priv;
+
GST_DEBUG ("client %p: linking transport %p", client, trans);
+
gst_rtsp_stream_transport_set_callbacks (trans,
(GstRTSPSendFunc) do_send_data,
(GstRTSPSendFunc) do_send_data, client, NULL);
- client->transports = g_list_prepend (client->transports, trans);
+ priv->transports = g_list_prepend (priv->transports, trans);
/* make sure our session can't expire */
gst_rtsp_session_prevent_expire (session);
unlink_transport (GstRTSPClient * client, GstRTSPSession * session,
GstRTSPStreamTransport * trans)
{
+ GstRTSPClientPrivate *priv = client->priv;
+
GST_DEBUG ("client %p: unlinking transport %p", client, trans);
+
gst_rtsp_stream_transport_set_callbacks (trans, NULL, NULL, NULL, NULL);
- client->transports = g_list_remove (client->transports, trans);
+ priv->transports = g_list_remove (priv->transports, trans);
/* our session can now expire */
gst_rtsp_session_allow_expire (session);
static void
unlink_session_transports (GstRTSPClient * client, GstRTSPSession * session,
- GstRTSPSessionMedia * media)
+ GstRTSPSessionMedia * sessmedia)
{
guint n_streams, i;
- n_streams = gst_rtsp_media_n_streams (media->media);
+ n_streams =
+ gst_rtsp_media_n_streams (gst_rtsp_session_media_get_media (sessmedia));
for (i = 0; i < n_streams; i++) {
GstRTSPStreamTransport *trans;
- GstRTSPTransport *tr;
+ const GstRTSPTransport *tr;
/* get the transport, if there is no transport configured, skip this stream */
- trans = gst_rtsp_session_media_get_transport (media, i);
+ trans = gst_rtsp_session_media_get_transport (sessmedia, i);
if (trans == NULL)
continue;
- tr = trans->transport;
+ tr = gst_rtsp_stream_transport_get_transport (trans);
if (tr->lower_transport == GST_RTSP_LOWER_TRANS_TCP) {
/* for TCP, unlink the stream from the TCP connection of the client */
static void
close_connection (GstRTSPClient * client)
{
+ GstRTSPClientPrivate *priv = client->priv;
const gchar *tunnelid;
GST_DEBUG ("client %p: closing connection", client);
- if ((tunnelid = gst_rtsp_connection_get_tunnelid (client->connection))) {
+ if ((tunnelid = gst_rtsp_connection_get_tunnelid (priv->connection))) {
g_mutex_lock (&tunnels_lock);
/* remove from tunnelids */
g_hash_table_remove (tunnels, tunnelid);
g_mutex_unlock (&tunnels_lock);
}
- gst_rtsp_connection_close (client->connection);
+ gst_rtsp_connection_close (priv->connection);
+}
+
+static gchar *
+default_make_path_from_uri (GstRTSPClient * client, const GstRTSPUrl * uri)
+{
+ gchar *path;
+
+ if (uri->query)
+ path = g_strconcat (uri->abspath, "?", uri->query, NULL);
+ else
+ path = g_strdup (uri->abspath);
+
+ return path;
}
static gboolean
-handle_teardown_request (GstRTSPClient * client, GstRTSPClientState * state)
+handle_teardown_request (GstRTSPClient * client, GstRTSPContext * ctx)
{
+ GstRTSPClientPrivate *priv = client->priv;
+ GstRTSPClientClass *klass;
GstRTSPSession *session;
- GstRTSPSessionMedia *media;
+ GstRTSPSessionMedia *sessmedia;
GstRTSPStatusCode code;
+ gchar *path;
+ gint matched;
- if (!state->session)
+ if (!ctx->session)
goto no_session;
- session = state->session;
+ session = ctx->session;
+
+ if (!ctx->uri)
+ goto no_uri;
+
+ klass = GST_RTSP_CLIENT_GET_CLASS (client);
+ path = klass->make_path_from_uri (client, ctx->uri);
/* get a handle to the configuration of the media in the session */
- media = gst_rtsp_session_get_media (session, state->uri);
- if (!media)
+ sessmedia = gst_rtsp_session_get_media (session, path, &matched);
+ if (!sessmedia)
goto not_found;
- state->sessmedia = media;
+ /* only aggregate control for now.. */
+ if (path[matched] != '\0')
+ goto no_aggregate;
+
+ g_free (path);
+
+ ctx->sessmedia = sessmedia;
+
+ /* we emit the signal before closing the connection */
+ g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_TEARDOWN_REQUEST],
+ 0, ctx);
/* unlink the all TCP callbacks */
- unlink_session_transports (client, session, media);
+ unlink_session_transports (client, session, sessmedia);
/* remove the session from the watched sessions */
- g_object_weak_unref (G_OBJECT (session),
- (GWeakNotify) client_session_finalized, client);
- client->sessions = g_list_remove (client->sessions, session);
+ client_unwatch_session (client, session);
- gst_rtsp_session_media_set_state (media, GST_STATE_NULL);
+ gst_rtsp_session_media_set_state (sessmedia, GST_STATE_NULL);
/* unmanage the media in the session, returns false if all media session
* are torn down. */
- if (!gst_rtsp_session_release_media (session, media)) {
+ if (!gst_rtsp_session_release_media (session, sessmedia)) {
/* remove the session */
- gst_rtsp_session_pool_remove (client->session_pool, session);
+ gst_rtsp_session_pool_remove (priv->session_pool, session);
}
/* construct the response now */
code = GST_RTSP_STS_OK;
- gst_rtsp_message_init_response (state->response, code,
- gst_rtsp_status_as_text (code), state->request);
+ gst_rtsp_message_init_response (ctx->response, code,
+ gst_rtsp_status_as_text (code), ctx->request);
- send_response (client, session, state->response, TRUE);
-
- /* we emit the signal before closing the connection */
- g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_TEARDOWN_REQUEST],
- 0, state);
+ send_message (client, session, ctx->response, TRUE);
return TRUE;
no_session:
{
GST_ERROR ("client %p: no session", client);
- send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, state);
+ send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
+ return FALSE;
+ }
+no_uri:
+ {
+ GST_ERROR ("client %p: no uri supplied", client);
+ send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
return FALSE;
}
not_found:
{
GST_ERROR ("client %p: no media for uri", client);
- send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
+ send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
+ g_free (path);
+ return FALSE;
+ }
+no_aggregate:
+ {
+ GST_ERROR ("client %p: no aggregate path %s", client, path);
+ send_generic_response (client,
+ GST_RTSP_STS_ONLY_AGGREGATE_OPERATION_ALLOWED, ctx);
+ g_free (path);
return FALSE;
}
}
+static GstRTSPResult
+default_params_set (GstRTSPClient * client, GstRTSPContext * ctx)
+{
+ GstRTSPResult res;
+
+ res = gst_rtsp_params_set (client, ctx);
+
+ return res;
+}
+
+static GstRTSPResult
+default_params_get (GstRTSPClient * client, GstRTSPContext * ctx)
+{
+ GstRTSPResult res;
+
+ res = gst_rtsp_params_get (client, ctx);
+
+ return res;
+}
+
static gboolean
-handle_get_param_request (GstRTSPClient * client, GstRTSPClientState * state)
+handle_get_param_request (GstRTSPClient * client, GstRTSPContext * ctx)
{
GstRTSPResult res;
guint8 *data;
guint size;
- res = gst_rtsp_message_get_body (state->request, &data, &size);
+ res = gst_rtsp_message_get_body (ctx->request, &data, &size);
if (res != GST_RTSP_OK)
goto bad_request;
if (size == 0) {
/* no body, keep-alive request */
- send_generic_response (client, GST_RTSP_STS_OK, state);
+ send_generic_response (client, GST_RTSP_STS_OK, ctx);
} else {
/* there is a body, handle the params */
- res = gst_rtsp_params_get (client, state);
+ res = GST_RTSP_CLIENT_GET_CLASS (client)->params_get (client, ctx);
if (res != GST_RTSP_OK)
goto bad_request;
- send_response (client, state->session, state->response, FALSE);
+ send_message (client, ctx->session, ctx->response, FALSE);
}
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_GET_PARAMETER_REQUEST],
- 0, state);
+ 0, ctx);
return TRUE;
bad_request:
{
GST_ERROR ("client %p: bad request", client);
- send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, state);
+ send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
return FALSE;
}
}
static gboolean
-handle_set_param_request (GstRTSPClient * client, GstRTSPClientState * state)
+handle_set_param_request (GstRTSPClient * client, GstRTSPContext * ctx)
{
GstRTSPResult res;
guint8 *data;
guint size;
- res = gst_rtsp_message_get_body (state->request, &data, &size);
+ res = gst_rtsp_message_get_body (ctx->request, &data, &size);
if (res != GST_RTSP_OK)
goto bad_request;
if (size == 0) {
/* no body, keep-alive request */
- send_generic_response (client, GST_RTSP_STS_OK, state);
+ send_generic_response (client, GST_RTSP_STS_OK, ctx);
} else {
/* there is a body, handle the params */
- res = gst_rtsp_params_set (client, state);
+ res = GST_RTSP_CLIENT_GET_CLASS (client)->params_set (client, ctx);
if (res != GST_RTSP_OK)
goto bad_request;
- send_response (client, state->session, state->response, FALSE);
+ send_message (client, ctx->session, ctx->response, FALSE);
}
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_SET_PARAMETER_REQUEST],
- 0, state);
+ 0, ctx);
return TRUE;
bad_request:
{
GST_ERROR ("client %p: bad request", client);
- send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, state);
+ send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
return FALSE;
}
}
static gboolean
-handle_pause_request (GstRTSPClient * client, GstRTSPClientState * state)
+handle_pause_request (GstRTSPClient * client, GstRTSPContext * ctx)
{
GstRTSPSession *session;
- GstRTSPSessionMedia *media;
+ GstRTSPClientClass *klass;
+ GstRTSPSessionMedia *sessmedia;
GstRTSPStatusCode code;
+ GstRTSPState rtspstate;
+ gchar *path;
+ gint matched;
- if (!(session = state->session))
+ if (!(session = ctx->session))
goto no_session;
+ if (!ctx->uri)
+ goto no_uri;
+
+ klass = GST_RTSP_CLIENT_GET_CLASS (client);
+ path = klass->make_path_from_uri (client, ctx->uri);
+
/* get a handle to the configuration of the media in the session */
- media = gst_rtsp_session_get_media (session, state->uri);
- if (!media)
+ sessmedia = gst_rtsp_session_get_media (session, path, &matched);
+ if (!sessmedia)
goto not_found;
- state->sessmedia = media;
+ if (path[matched] != '\0')
+ goto no_aggregate;
+
+ g_free (path);
+
+ ctx->sessmedia = sessmedia;
+ rtspstate = gst_rtsp_session_media_get_rtsp_state (sessmedia);
/* the session state must be playing or recording */
- if (media->state != GST_RTSP_STATE_PLAYING &&
- media->state != GST_RTSP_STATE_RECORDING)
+ if (rtspstate != GST_RTSP_STATE_PLAYING &&
+ rtspstate != GST_RTSP_STATE_RECORDING)
goto invalid_state;
/* unlink the all TCP callbacks */
- unlink_session_transports (client, session, media);
+ unlink_session_transports (client, session, sessmedia);
/* then pause sending */
- gst_rtsp_session_media_set_state (media, GST_STATE_PAUSED);
+ gst_rtsp_session_media_set_state (sessmedia, GST_STATE_PAUSED);
/* construct the response now */
code = GST_RTSP_STS_OK;
- gst_rtsp_message_init_response (state->response, code,
- gst_rtsp_status_as_text (code), state->request);
+ gst_rtsp_message_init_response (ctx->response, code,
+ gst_rtsp_status_as_text (code), ctx->request);
- send_response (client, session, state->response, FALSE);
+ send_message (client, session, ctx->response, FALSE);
/* the state is now READY */
- media->state = GST_RTSP_STATE_READY;
+ gst_rtsp_session_media_set_rtsp_state (sessmedia, GST_RTSP_STATE_READY);
- g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PAUSE_REQUEST],
- 0, state);
+ g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PAUSE_REQUEST], 0, ctx);
return TRUE;
no_session:
{
GST_ERROR ("client %p: no seesion", client);
- send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, state);
+ send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
+ return FALSE;
+ }
+no_uri:
+ {
+ GST_ERROR ("client %p: no uri supplied", client);
+ send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
return FALSE;
}
not_found:
{
GST_ERROR ("client %p: no media for uri", client);
- send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
+ send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
+ g_free (path);
+ return FALSE;
+ }
+no_aggregate:
+ {
+ GST_ERROR ("client %p: no aggregate path %s", client, path);
+ send_generic_response (client,
+ GST_RTSP_STS_ONLY_AGGREGATE_OPERATION_ALLOWED, ctx);
+ g_free (path);
return FALSE;
}
invalid_state:
{
GST_ERROR ("client %p: not PLAYING or RECORDING", client);
send_generic_response (client, GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE,
- state);
+ ctx);
return FALSE;
}
}
+/* convert @url and @path to a URL used as a content base for the factory
+ * located at @path */
+static gchar *
+make_base_url (GstRTSPClient * client, GstRTSPUrl * url, gchar * path)
+{
+ GstRTSPUrl tmp;
+ gchar *result, *trail;
+
+ /* check for trailing '/' and append one */
+ trail = (path[strlen (path) - 1] != '/' ? "/" : "");
+
+ tmp = *url;
+ tmp.user = NULL;
+ tmp.passwd = NULL;
+ tmp.abspath = g_strdup_printf ("%s%s", path, trail);
+ tmp.query = NULL;
+ result = gst_rtsp_url_get_request_uri (&tmp);
+ g_free (tmp.abspath);
+
+ return result;
+}
+
static gboolean
-handle_play_request (GstRTSPClient * client, GstRTSPClientState * state)
+handle_play_request (GstRTSPClient * client, GstRTSPContext * ctx)
{
GstRTSPSession *session;
- GstRTSPSessionMedia *media;
+ GstRTSPClientClass *klass;
+ GstRTSPSessionMedia *sessmedia;
+ GstRTSPMedia *media;
GstRTSPStatusCode code;
+ GstRTSPUrl *uri;
GString *rtpinfo;
guint n_streams, i, infocount;
gchar *str;
GstRTSPTimeRange *range;
GstRTSPResult res;
+ GstRTSPState rtspstate;
+ GstRTSPRangeUnit unit = GST_RTSP_RANGE_NPT;
+ gchar *path;
+ gint matched;
- if (!(session = state->session))
+ if (!(session = ctx->session))
goto no_session;
+ if (!(uri = ctx->uri))
+ goto no_uri;
+
+ klass = GST_RTSP_CLIENT_GET_CLASS (client);
+ path = klass->make_path_from_uri (client, uri);
+
/* get a handle to the configuration of the media in the session */
- media = gst_rtsp_session_get_media (session, state->uri);
- if (!media)
+ sessmedia = gst_rtsp_session_get_media (session, path, &matched);
+ if (!sessmedia)
goto not_found;
- state->sessmedia = media;
+ if (path[matched] != '\0')
+ goto no_aggregate;
+
+ ctx->sessmedia = sessmedia;
+ ctx->media = media = gst_rtsp_session_media_get_media (sessmedia);
/* the session state must be playing or ready */
- if (media->state != GST_RTSP_STATE_PLAYING &&
- media->state != GST_RTSP_STATE_READY)
+ rtspstate = gst_rtsp_session_media_get_rtsp_state (sessmedia);
+ if (rtspstate != GST_RTSP_STATE_PLAYING && rtspstate != GST_RTSP_STATE_READY)
goto invalid_state;
+ /* in play we first unsuspend, media could be suspended from SDP or PAUSED */
+ if (!gst_rtsp_media_unsuspend (media))
+ goto unsuspend_failed;
+
/* parse the range header if we have one */
- res =
- gst_rtsp_message_get_header (state->request, GST_RTSP_HDR_RANGE, &str, 0);
+ res = gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_RANGE, &str, 0);
if (res == GST_RTSP_OK) {
if (gst_rtsp_range_parse (str, &range) == GST_RTSP_OK) {
/* we have a range, seek to the position */
- gst_rtsp_media_seek (media->media, range);
+ unit = range->unit;
+ gst_rtsp_media_seek (media, range);
gst_rtsp_range_free (range);
}
}
/* grab RTPInfo from the payloaders now */
rtpinfo = g_string_new ("");
- n_streams = gst_rtsp_media_n_streams (media->media);
+ n_streams = gst_rtsp_media_n_streams (media);
for (i = 0, infocount = 0; i < n_streams; i++) {
GstRTSPStreamTransport *trans;
- GstRTSPTransport *tr;
- gchar *uristr;
+ GstRTSPStream *stream;
+ const GstRTSPTransport *tr;
guint rtptime, seq;
/* get the transport, if there is no transport configured, skip this stream */
- trans = gst_rtsp_session_media_get_transport (media, i);
+ trans = gst_rtsp_session_media_get_transport (sessmedia, i);
if (trans == NULL) {
GST_INFO ("stream %d is not configured", i);
continue;
}
- tr = trans->transport;
+ tr = gst_rtsp_stream_transport_get_transport (trans);
if (tr->lower_transport == GST_RTSP_LOWER_TRANS_TCP) {
/* for TCP, link the stream to the TCP connection of the client */
link_transport (client, session, trans);
}
- if (gst_rtsp_stream_get_rtpinfo (trans->stream, &rtptime, &seq)) {
+ stream = gst_rtsp_stream_transport_get_stream (trans);
+ if (gst_rtsp_stream_get_rtpinfo (stream, &rtptime, &seq)) {
+ const GstRTSPUrl *url;
+ gchar *url_str;
+
if (infocount > 0)
g_string_append (rtpinfo, ", ");
- uristr = gst_rtsp_url_get_request_uri (state->uri);
- g_string_append_printf (rtpinfo, "url=%s/stream=%d;seq=%u;rtptime=%u",
- uristr, i, seq, rtptime);
- g_free (uristr);
+ url = gst_rtsp_stream_transport_get_url (trans);
+ url_str = gst_rtsp_url_get_request_uri (url);
+ g_string_append_printf (rtpinfo, "url=%s;seq=%u;rtptime=%u",
+ url_str, seq, rtptime);
+ g_free (url_str);
infocount++;
} else {
GST_WARNING ("RTP-Info cannot be determined for stream %d", i);
}
}
+ g_free (path);
/* construct the response now */
code = GST_RTSP_STS_OK;
- gst_rtsp_message_init_response (state->response, code,
- gst_rtsp_status_as_text (code), state->request);
+ gst_rtsp_message_init_response (ctx->response, code,
+ gst_rtsp_status_as_text (code), ctx->request);
/* add the RTP-Info header */
if (infocount > 0) {
str = g_string_free (rtpinfo, FALSE);
- gst_rtsp_message_take_header (state->response, GST_RTSP_HDR_RTP_INFO, str);
+ gst_rtsp_message_take_header (ctx->response, GST_RTSP_HDR_RTP_INFO, str);
} else {
g_string_free (rtpinfo, TRUE);
}
/* add the range */
- str = gst_rtsp_media_get_range_string (media->media, TRUE);
- gst_rtsp_message_take_header (state->response, GST_RTSP_HDR_RANGE, str);
+ str = gst_rtsp_media_get_range_string (media, TRUE, unit);
+ if (str)
+ gst_rtsp_message_take_header (ctx->response, GST_RTSP_HDR_RANGE, str);
- send_response (client, session, state->response, FALSE);
+ send_message (client, session, ctx->response, FALSE);
/* start playing after sending the request */
- gst_rtsp_session_media_set_state (media, GST_STATE_PLAYING);
+ gst_rtsp_session_media_set_state (sessmedia, GST_STATE_PLAYING);
- media->state = GST_RTSP_STATE_PLAYING;
+ gst_rtsp_session_media_set_rtsp_state (sessmedia, GST_RTSP_STATE_PLAYING);
- g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PLAY_REQUEST],
- 0, state);
+ g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PLAY_REQUEST], 0, ctx);
return TRUE;
no_session:
{
GST_ERROR ("client %p: no session", client);
- send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, state);
+ send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
+ return FALSE;
+ }
+no_uri:
+ {
+ GST_ERROR ("client %p: no uri supplied", client);
+ send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
return FALSE;
}
not_found:
{
GST_ERROR ("client %p: media not found", client);
- send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
+ send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
+ return FALSE;
+ }
+no_aggregate:
+ {
+ GST_ERROR ("client %p: no aggregate path %s", client, path);
+ send_generic_response (client,
+ GST_RTSP_STS_ONLY_AGGREGATE_OPERATION_ALLOWED, ctx);
+ g_free (path);
return FALSE;
}
invalid_state:
{
GST_ERROR ("client %p: not PLAYING or READY", client);
send_generic_response (client, GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE,
- state);
+ ctx);
+ g_free (path);
+ return FALSE;
+ }
+unsuspend_failed:
+ {
+ GST_ERROR ("client %p: unsuspend failed", client);
+ send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
+ g_free (path);
return FALSE;
}
}
}
static gboolean
-handle_blocksize (GstRTSPMedia * media, GstRTSPStream * stream,
- GstRTSPMessage * request)
+default_configure_client_media (GstRTSPClient * client, GstRTSPMedia * media,
+ GstRTSPStream * stream, GstRTSPContext * ctx)
{
+ GstRTSPMessage *request = ctx->request;
gchar *blocksize_str;
- gboolean ret = TRUE;
if (gst_rtsp_message_get_header (request, GST_RTSP_HDR_BLOCKSIZE,
&blocksize_str, 0) == GST_RTSP_OK) {
gchar *end;
blocksize = g_ascii_strtoull (blocksize_str, &end, 10);
- if (end == blocksize_str) {
- GST_ERROR ("failed to parse blocksize");
- ret = FALSE;
- } else {
- /* we don't want to change the mtu when this media
- * can be shared because it impacts other clients */
- if (gst_rtsp_media_is_shared (media))
- return TRUE;
-
- if (blocksize > G_MAXUINT)
- blocksize = G_MAXUINT;
- gst_rtsp_stream_set_mtu (stream, blocksize);
- }
+ if (end == blocksize_str)
+ goto parse_failed;
+
+ /* we don't want to change the mtu when this media
+ * can be shared because it impacts other clients */
+ if (gst_rtsp_media_is_shared (media))
+ goto done;
+
+ if (blocksize > G_MAXUINT)
+ blocksize = G_MAXUINT;
+
+ gst_rtsp_stream_set_mtu (stream, blocksize);
+ }
+done:
+ return TRUE;
+
+ /* ERRORS */
+parse_failed:
+ {
+ GST_ERROR_OBJECT (client, "failed to parse blocksize");
+ send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
+ return FALSE;
}
- return ret;
}
static gboolean
-configure_client_transport (GstRTSPClient * client, GstRTSPClientState * state,
- GstRTSPTransport * ct)
+default_configure_client_transport (GstRTSPClient * client,
+ GstRTSPContext * ctx, GstRTSPTransport * ct)
{
+ GstRTSPClientPrivate *priv = client->priv;
+
/* we have a valid transport now, set the destination of the client. */
if (ct->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
- if (ct->destination == NULL || !client->use_client_settings) {
+ gboolean use_client_settings;
+
+ use_client_settings =
+ gst_rtsp_auth_check (GST_RTSP_AUTH_CHECK_TRANSPORT_CLIENT_SETTINGS);
+
+ if (ct->destination && use_client_settings) {
+ GstRTSPAddress *addr;
+
+ addr = gst_rtsp_stream_reserve_address (ctx->stream, ct->destination,
+ ct->port.min, ct->port.max - ct->port.min + 1, ct->ttl);
+
+ if (addr == NULL)
+ goto no_address;
+
+ gst_rtsp_address_free (addr);
+ } else {
GstRTSPAddress *addr;
+ GSocketFamily family;
+
+ family = priv->is_ipv6 ? G_SOCKET_FAMILY_IPV6 : G_SOCKET_FAMILY_IPV4;
- addr = gst_rtsp_stream_get_address (state->stream);
+ addr = gst_rtsp_stream_get_multicast_address (ctx->stream, family);
if (addr == NULL)
goto no_address;
ct->port.min = addr->port;
ct->port.max = addr->port + addr->n_ports - 1;
ct->ttl = addr->ttl;
+
+ gst_rtsp_address_free (addr);
}
} else {
GstRTSPUrl *url;
- url = gst_rtsp_connection_get_url (client->connection);
+ url = gst_rtsp_connection_get_url (priv->connection);
g_free (ct->destination);
ct->destination = g_strdup (url->host);
if (ct->lower_transport & GST_RTSP_LOWER_TRANS_TCP) {
/* check if the client selected channels for TCP */
if (ct->interleaved.min == -1 || ct->interleaved.max == -1) {
- gst_rtsp_session_media_alloc_channels (state->sessmedia,
+ gst_rtsp_session_media_alloc_channels (ctx->sessmedia,
&ct->interleaved);
}
}
}
static GstRTSPTransport *
-make_server_transport (GstRTSPClient * client, GstRTSPClientState * state,
+make_server_transport (GstRTSPClient * client, GstRTSPContext * ctx,
GstRTSPTransport * ct)
{
GstRTSPTransport *st;
+ GInetAddress *addr;
+ GSocketFamily family;
/* prepare the server transport */
gst_rtsp_transport_new (&st);
st->profile = ct->profile;
st->lower_transport = ct->lower_transport;
+ addr = g_inet_address_new_from_string (ct->destination);
+
+ if (!addr) {
+ GST_ERROR ("failed to get inet addr from client destination");
+ family = G_SOCKET_FAMILY_IPV4;
+ } else {
+ family = g_inet_address_get_family (addr);
+ g_object_unref (addr);
+ addr = NULL;
+ }
+
switch (st->lower_transport) {
case GST_RTSP_LOWER_TRANS_UDP:
st->client_port = ct->client_port;
- st->server_port = state->stream->server_port;
+ gst_rtsp_stream_get_server_port (ctx->stream, &st->server_port, family);
break;
case GST_RTSP_LOWER_TRANS_UDP_MCAST:
st->port = ct->port;
break;
}
- if (state->stream->session)
- g_object_get (state->stream->session, "internal-ssrc", &st->ssrc, NULL);
+ gst_rtsp_stream_get_ssrc (ctx->stream, &st->ssrc);
return st;
}
static gboolean
-handle_setup_request (GstRTSPClient * client, GstRTSPClientState * state)
+handle_setup_request (GstRTSPClient * client, GstRTSPContext * ctx)
{
+ GstRTSPClientPrivate *priv = client->priv;
GstRTSPResult res;
GstRTSPUrl *uri;
gchar *transport;
GstRTSPStatusCode code;
GstRTSPSession *session;
GstRTSPStreamTransport *trans;
- gchar *trans_str, *pos;
- guint streamid;
+ gchar *trans_str;
GstRTSPSessionMedia *sessmedia;
GstRTSPMedia *media;
GstRTSPStream *stream;
+ GstRTSPState rtspstate;
+ GstRTSPClientClass *klass;
+ gchar *path, *control;
+ gint matched;
- uri = state->uri;
-
- /* the uri contains the stream number we added in the SDP config, which is
- * always /stream=%d so we need to strip that off
- * parse the stream we need to configure, look for the stream in the abspath
- * first and then in the query. */
- if (uri->abspath == NULL || !(pos = strstr (uri->abspath, "/stream="))) {
- if (uri->query == NULL || !(pos = strstr (uri->query, "/stream=")))
- goto bad_request;
- }
-
- /* we can mofify the parsed uri in place */
- *pos = '\0';
+ if (!ctx->uri)
+ goto no_uri;
- pos += strlen ("/stream=");
- if (sscanf (pos, "%u", &streamid) != 1)
- goto bad_request;
+ uri = ctx->uri;
+ klass = GST_RTSP_CLIENT_GET_CLASS (client);
+ path = klass->make_path_from_uri (client, uri);
/* parse the transport */
res =
- gst_rtsp_message_get_header (state->request, GST_RTSP_HDR_TRANSPORT,
+ gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_TRANSPORT,
&transport, 0);
if (res != GST_RTSP_OK)
goto no_transport;
- gst_rtsp_transport_new (&ct);
-
- /* our supported transports */
- supported = GST_RTSP_LOWER_TRANS_UDP |
- GST_RTSP_LOWER_TRANS_UDP_MCAST | GST_RTSP_LOWER_TRANS_TCP;
-
- /* parse and find a usable supported transport */
- if (!parse_transport (transport, supported, ct))
- goto unsupported_transports;
-
/* we create the session after parsing stuff so that we don't make
* a session for malformed requests */
- if (client->session_pool == NULL)
+ if (priv->session_pool == NULL)
goto no_pool;
- session = state->session;
+ session = ctx->session;
if (session) {
g_object_ref (session);
/* get a handle to the configuration of the media in the session, this can
* return NULL if this is a new url to manage in this session. */
- sessmedia = gst_rtsp_session_get_media (session, uri);
+ sessmedia = gst_rtsp_session_get_media (session, path, &matched);
} else {
- /* create a session if this fails we probably reached our session limit or
- * something. */
- if (!(session = gst_rtsp_session_pool_create (client->session_pool)))
- goto service_unavailable;
-
- state->session = session;
-
/* we need a new media configuration in this session */
sessmedia = NULL;
}
- /* we have no media, find one and manage it */
+ /* we have no session media, find one and manage it */
if (sessmedia == NULL) {
/* get a handle to the configuration of the media in the session */
- if ((media = find_media (client, state))) {
- /* manage the media in our session now */
- sessmedia = gst_rtsp_session_manage_media (session, uri, media);
- }
+ media = find_media (client, ctx, path, &matched);
+ } else {
+ if ((media = gst_rtsp_session_media_get_media (sessmedia)))
+ g_object_ref (media);
+ else
+ goto media_not_found;
}
+ /* no media, not found then */
+ if (media == NULL)
+ goto media_not_found_no_reply;
- /* if we stil have no media, error */
- if (sessmedia == NULL)
- goto not_found;
+ if (path[matched] == '\0')
+ goto control_not_found;
- state->sessmedia = sessmedia;
- state->media = media = sessmedia->media;
+ /* path is what matched. */
+ path[matched] = '\0';
+ /* control is remainder */
+ control = &path[matched + 1];
- /* now get the stream */
- stream = gst_rtsp_media_get_stream (media, streamid);
+ /* find the stream now using the control part */
+ stream = gst_rtsp_media_find_stream (media, control);
if (stream == NULL)
- goto not_found;
+ goto stream_not_found;
- state->stream = stream;
+ /* now we have a uri identifying a valid media and stream */
+ ctx->stream = stream;
+ ctx->media = media;
- /* set blocksize on this stream */
- if (!handle_blocksize (media, stream, state->request))
- goto invalid_blocksize;
+ if (session == NULL) {
+ /* create a session if this fails we probably reached our session limit or
+ * something. */
+ if (!(session = gst_rtsp_session_pool_create (priv->session_pool)))
+ goto service_unavailable;
- /* update the client transport */
- if (!configure_client_transport (client, state, ct))
- goto unsupported_client_transport;
+ /* make sure this client is closed when the session is closed */
+ client_watch_session (client, session);
- /* set in the session media transport */
- trans = gst_rtsp_session_media_set_transport (sessmedia, stream, ct);
+ /* signal new session */
+ g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_NEW_SESSION], 0,
+ session);
- /* configure keepalive for this transport */
+ ctx->session = session;
+ }
+
+ if (sessmedia == NULL) {
+ /* manage the media in our session now, if not done already */
+ sessmedia = gst_rtsp_session_manage_media (session, path, media);
+ /* if we stil have no media, error */
+ if (sessmedia == NULL)
+ goto sessmedia_unavailable;
+ } else {
+ g_object_unref (media);
+ }
+
+ ctx->sessmedia = sessmedia;
+
+ if (!klass->configure_client_media (client, media, stream, ctx))
+ goto configure_media_failed_no_reply;
+
+ gst_rtsp_transport_new (&ct);
+
+ /* our supported transports */
+ supported = gst_rtsp_stream_get_protocols (stream);
+
+ /* parse and find a usable supported transport */
+ if (!parse_transport (transport, supported, ct))
+ goto unsupported_transports;
+
+ /* update the client transport */
+ if (!klass->configure_client_transport (client, ctx, ct))
+ goto unsupported_client_transport;
+
+ /* set in the session media transport */
+ trans = gst_rtsp_session_media_set_transport (sessmedia, stream, ct);
+
+ /* configure the url used to set this transport, this we will use when
+ * generating the response for the PLAY request */
+ gst_rtsp_stream_transport_set_url (trans, uri);
+
+ /* configure keepalive for this transport */
gst_rtsp_stream_transport_set_keepalive (trans,
(GstRTSPKeepAliveFunc) do_keepalive, session, NULL);
/* create and serialize the server transport */
- st = make_server_transport (client, state, ct);
+ st = make_server_transport (client, ctx, ct);
trans_str = gst_rtsp_transport_as_text (st);
gst_rtsp_transport_free (st);
/* construct the response now */
code = GST_RTSP_STS_OK;
- gst_rtsp_message_init_response (state->response, code,
- gst_rtsp_status_as_text (code), state->request);
+ gst_rtsp_message_init_response (ctx->response, code,
+ gst_rtsp_status_as_text (code), ctx->request);
- gst_rtsp_message_add_header (state->response, GST_RTSP_HDR_TRANSPORT,
+ gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_TRANSPORT,
trans_str);
g_free (trans_str);
- send_response (client, session, state->response, FALSE);
+ send_message (client, session, ctx->response, FALSE);
/* update the state */
- switch (sessmedia->state) {
+ rtspstate = gst_rtsp_session_media_get_rtsp_state (sessmedia);
+ switch (rtspstate) {
case GST_RTSP_STATE_PLAYING:
case GST_RTSP_STATE_RECORDING:
case GST_RTSP_STATE_READY:
/* no state change */
break;
default:
- sessmedia->state = GST_RTSP_STATE_READY;
+ gst_rtsp_session_media_set_rtsp_state (sessmedia, GST_RTSP_STATE_READY);
break;
}
g_object_unref (session);
+ g_free (path);
- g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_SETUP_REQUEST],
- 0, state);
+ g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_SETUP_REQUEST], 0, ctx);
return TRUE;
/* ERRORS */
-bad_request:
+no_uri:
{
- GST_ERROR ("client %p: bad request", client);
- send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, state);
+ GST_ERROR ("client %p: no uri", client);
+ send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
return FALSE;
}
-not_found:
+no_transport:
{
- GST_ERROR ("client %p: media not found", client);
- send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
- g_object_unref (session);
- gst_rtsp_transport_free (ct);
+ GST_ERROR ("client %p: no transport", client);
+ send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, ctx);
+ g_free (path);
return FALSE;
}
-invalid_blocksize:
+no_pool:
{
- GST_ERROR ("client %p: invalid blocksize", client);
- send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, state);
- g_object_unref (session);
- gst_rtsp_transport_free (ct);
+ GST_ERROR ("client %p: no session pool configured", client);
+ send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
+ g_free (path);
return FALSE;
}
-unsupported_client_transport:
+media_not_found_no_reply:
{
- GST_ERROR ("client %p: unsupported client transport", client);
- send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, state);
- g_object_unref (session);
- gst_rtsp_transport_free (ct);
+ GST_ERROR ("client %p: media '%s' not found", client, path);
+ g_free (path);
+ /* error reply is already sent */
return FALSE;
}
-no_transport:
+media_not_found:
{
- GST_ERROR ("client %p: no transport", client);
- send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, state);
+ GST_ERROR ("client %p: media '%s' not found", client, path);
+ send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
+ g_free (path);
return FALSE;
}
-unsupported_transports:
+control_not_found:
{
- GST_ERROR ("client %p: unsupported transports", client);
- send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, state);
- gst_rtsp_transport_free (ct);
+ GST_ERROR ("client %p: no control in path '%s'", client, path);
+ send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
+ g_object_unref (media);
+ g_free (path);
return FALSE;
}
-no_pool:
+stream_not_found:
{
- GST_ERROR ("client %p: no session pool configured", client);
- send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, state);
- gst_rtsp_transport_free (ct);
+ GST_ERROR ("client %p: stream '%s' not found", client, control);
+ send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
+ g_object_unref (media);
+ g_free (path);
return FALSE;
}
service_unavailable:
{
GST_ERROR ("client %p: can't create session", client);
- send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, state);
+ send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
+ g_object_unref (media);
+ g_free (path);
+ return FALSE;
+ }
+sessmedia_unavailable:
+ {
+ GST_ERROR ("client %p: can't create session media", client);
+ send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
+ g_object_unref (media);
+ g_object_unref (session);
+ g_free (path);
+ return FALSE;
+ }
+configure_media_failed_no_reply:
+ {
+ GST_ERROR ("client %p: configure_media failed", client);
+ g_object_unref (session);
+ g_free (path);
+ /* error reply is already sent */
+ return FALSE;
+ }
+unsupported_transports:
+ {
+ GST_ERROR ("client %p: unsupported transports", client);
+ send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, ctx);
gst_rtsp_transport_free (ct);
+ g_object_unref (session);
+ g_free (path);
+ return FALSE;
+ }
+unsupported_client_transport:
+ {
+ GST_ERROR ("client %p: unsupported client transport", client);
+ send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, ctx);
+ gst_rtsp_transport_free (ct);
+ g_object_unref (session);
+ g_free (path);
return FALSE;
}
}
static GstSDPMessage *
create_sdp (GstRTSPClient * client, GstRTSPMedia * media)
{
+ GstRTSPClientPrivate *priv = client->priv;
GstSDPMessage *sdp;
GstSDPInfo info;
const gchar *proto;
/* some standard things first */
gst_sdp_message_set_version (sdp, "0");
- if (client->is_ipv6)
+ if (priv->is_ipv6)
proto = "IP6";
else
proto = "IP4";
gst_sdp_message_set_origin (sdp, "-", "1188340656180883", "1", "IN", proto,
- client->server_ip);
+ priv->server_ip);
gst_sdp_message_set_session_name (sdp, "Session streamed with GStreamer");
gst_sdp_message_set_information (sdp, "rtsp-server");
gst_sdp_message_add_attribute (sdp, "type", "broadcast");
gst_sdp_message_add_attribute (sdp, "control", "*");
- info.server_proto = proto;
- info.server_ip = g_strdup (client->server_ip);
+ info.is_ipv6 = priv->is_ipv6;
+ info.server_ip = priv->server_ip;
/* create an SDP for the media object */
if (!gst_rtsp_sdp_from_media (sdp, &info, media))
goto no_sdp;
- g_free (info.server_ip);
-
return sdp;
/* ERRORS */
no_sdp:
{
GST_ERROR ("client %p: could not create SDP", client);
- g_free (info.server_ip);
gst_sdp_message_free (sdp);
return NULL;
}
/* for the describe we must generate an SDP */
static gboolean
-handle_describe_request (GstRTSPClient * client, GstRTSPClientState * state)
+handle_describe_request (GstRTSPClient * client, GstRTSPContext * ctx)
{
+ GstRTSPClientPrivate *priv = client->priv;
GstRTSPResult res;
GstSDPMessage *sdp;
- guint i, str_len;
- gchar *str, *content_base;
+ guint i;
+ gchar *path, *str;
GstRTSPMedia *media;
GstRTSPClientClass *klass;
klass = GST_RTSP_CLIENT_GET_CLASS (client);
+ if (!ctx->uri)
+ goto no_uri;
+
/* check what kind of format is accepted, we don't really do anything with it
* and always return SDP for now. */
for (i = 0; i++;) {
gchar *accept;
res =
- gst_rtsp_message_get_header (state->request, GST_RTSP_HDR_ACCEPT,
+ gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_ACCEPT,
&accept, i);
if (res == GST_RTSP_ENOTIMPL)
break;
break;
}
+ if (!priv->mount_points)
+ goto no_mount_points;
+
+ if (!(path = gst_rtsp_mount_points_make_path (priv->mount_points, ctx->uri)))
+ goto no_path;
+
/* find the media object for the uri */
- if (!(media = find_media (client, state)))
+ if (!(media = find_media (client, ctx, path, NULL)))
goto no_media;
/* create an SDP for the media object on this client */
if (!(sdp = klass->create_sdp (client, media)))
goto no_sdp;
+ /* we suspend after the describe */
+ gst_rtsp_media_suspend (media);
g_object_unref (media);
- gst_rtsp_message_init_response (state->response, GST_RTSP_STS_OK,
- gst_rtsp_status_as_text (GST_RTSP_STS_OK), state->request);
+ gst_rtsp_message_init_response (ctx->response, GST_RTSP_STS_OK,
+ gst_rtsp_status_as_text (GST_RTSP_STS_OK), ctx->request);
- gst_rtsp_message_add_header (state->response, GST_RTSP_HDR_CONTENT_TYPE,
+ gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_CONTENT_TYPE,
"application/sdp");
/* content base for some clients that might screw up creating the setup uri */
- str = gst_rtsp_url_get_request_uri (state->uri);
- str_len = strlen (str);
+ str = make_base_url (client, ctx->uri, path);
+ g_free (path);
- /* check for trailing '/' and append one */
- if (str[str_len - 1] != '/') {
- content_base = g_malloc (str_len + 2);
- memcpy (content_base, str, str_len);
- content_base[str_len] = '/';
- content_base[str_len + 1] = '\0';
- g_free (str);
- } else {
- content_base = str;
- }
-
- GST_INFO ("adding content-base: %s", content_base);
-
- gst_rtsp_message_add_header (state->response, GST_RTSP_HDR_CONTENT_BASE,
- content_base);
- g_free (content_base);
+ GST_INFO ("adding content-base: %s", str);
+ gst_rtsp_message_take_header (ctx->response, GST_RTSP_HDR_CONTENT_BASE, str);
/* add SDP to the response body */
str = gst_sdp_message_as_text (sdp);
- gst_rtsp_message_take_body (state->response, (guint8 *) str, strlen (str));
+ gst_rtsp_message_take_body (ctx->response, (guint8 *) str, strlen (str));
gst_sdp_message_free (sdp);
- send_response (client, state->session, state->response, FALSE);
+ send_message (client, ctx->session, ctx->response, FALSE);
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_DESCRIBE_REQUEST],
- 0, state);
+ 0, ctx);
return TRUE;
/* ERRORS */
+no_uri:
+ {
+ GST_ERROR ("client %p: no uri", client);
+ send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
+ return FALSE;
+ }
+no_mount_points:
+ {
+ GST_ERROR ("client %p: no mount points configured", client);
+ send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
+ return FALSE;
+ }
+no_path:
+ {
+ GST_ERROR ("client %p: can't find path for url", client);
+ send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
+ return FALSE;
+ }
no_media:
{
GST_ERROR ("client %p: no media", client);
+ g_free (path);
/* error reply is already sent */
return FALSE;
}
no_sdp:
{
GST_ERROR ("client %p: can't create SDP", client);
- send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, state);
+ send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
+ g_free (path);
g_object_unref (media);
return FALSE;
}
}
static gboolean
-handle_options_request (GstRTSPClient * client, GstRTSPClientState * state)
+handle_options_request (GstRTSPClient * client, GstRTSPContext * ctx)
{
GstRTSPMethod options;
gchar *str;
str = gst_rtsp_options_as_text (options);
- gst_rtsp_message_init_response (state->response, GST_RTSP_STS_OK,
- gst_rtsp_status_as_text (GST_RTSP_STS_OK), state->request);
+ gst_rtsp_message_init_response (ctx->response, GST_RTSP_STS_OK,
+ gst_rtsp_status_as_text (GST_RTSP_STS_OK), ctx->request);
- gst_rtsp_message_add_header (state->response, GST_RTSP_HDR_PUBLIC, str);
+ gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_PUBLIC, str);
g_free (str);
- send_response (client, state->session, state->response, FALSE);
+ send_message (client, ctx->session, ctx->response, FALSE);
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_OPTIONS_REQUEST],
- 0, state);
+ 0, ctx);
return TRUE;
}
static void
client_session_finalized (GstRTSPClient * client, GstRTSPSession * session)
{
+ GstRTSPClientPrivate *priv = client->priv;
+
GST_INFO ("client %p: session %p finished", client, session);
/* unlink all media managed in this session */
client_unlink_session (client, session);
/* remove the session */
- if (!(client->sessions = g_list_remove (client->sessions, session))) {
+ if (!(priv->sessions = g_list_remove (priv->sessions, session))) {
GST_INFO ("client %p: all sessions finalized, close the connection",
client);
close_connection (client);
}
static void
-client_watch_session (GstRTSPClient * client, GstRTSPSession * session)
-{
- GList *walk;
-
- for (walk = client->sessions; walk; walk = g_list_next (walk)) {
- GstRTSPSession *msession = (GstRTSPSession *) walk->data;
-
- /* we already know about this session */
- if (msession == session)
- return;
- }
-
- GST_INFO ("watching session %p", session);
-
- g_object_weak_ref (G_OBJECT (session), (GWeakNotify) client_session_finalized,
- client);
- client->sessions = g_list_prepend (client->sessions, session);
-
- g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_NEW_SESSION], 0,
- session);
-}
-
-static void
handle_request (GstRTSPClient * client, GstRTSPMessage * request)
{
+ GstRTSPClientPrivate *priv = client->priv;
GstRTSPMethod method;
const gchar *uristr;
- GstRTSPUrl *uri;
+ GstRTSPUrl *uri = NULL;
GstRTSPVersion version;
GstRTSPResult res;
- GstRTSPSession *session;
- GstRTSPClientState state = { NULL };
+ GstRTSPSession *session = NULL;
+ GstRTSPContext sctx = { NULL }, *ctx;
GstRTSPMessage response = { 0 };
gchar *sessid;
- state.request = request;
- state.response = &response;
+ if (!(ctx = gst_rtsp_context_get_current ())) {
+ ctx = &sctx;
+ ctx->auth = priv->auth;
+ gst_rtsp_context_push_current (ctx);
+ }
+
+ ctx->conn = priv->connection;
+ ctx->client = client;
+ ctx->request = request;
+ ctx->response = &response;
if (gst_debug_category_get_threshold (rtsp_client_debug) >= GST_LEVEL_LOG) {
gst_rtsp_message_dump (request);
}
- GST_INFO ("client %p: received a request", client);
-
gst_rtsp_message_parse_request (request, &method, &uristr, &version);
- if (version != GST_RTSP_VERSION_1_0) {
- /* we can only handle 1.0 requests */
- send_generic_response (client, GST_RTSP_STS_RTSP_VERSION_NOT_SUPPORTED,
- &state);
- return;
- }
- state.method = method;
+ GST_INFO ("client %p: received a request %s %s %s", client,
+ gst_rtsp_method_as_text (method), uristr,
+ gst_rtsp_version_as_text (version));
+
+ /* we can only handle 1.0 requests */
+ if (version != GST_RTSP_VERSION_1_0)
+ goto not_supported;
+
+ ctx->method = method;
/* we always try to parse the url first */
- if (gst_rtsp_url_parse (uristr, &uri) != GST_RTSP_OK) {
- send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, &state);
- return;
- }
+ if (strcmp (uristr, "*") == 0) {
+ /* special case where we have * as uri, keep uri = NULL */
+ } else if (gst_rtsp_url_parse (uristr, &uri) != GST_RTSP_OK) {
+ /* check if the uristr is an absolute path <=> scheme and host information
+ * is missing */
+ gchar *scheme;
+
+ scheme = g_uri_parse_scheme (uristr);
+ if (scheme == NULL && g_str_has_prefix (uristr, "/")) {
+ gchar *absolute_uristr = NULL;
+
+ GST_WARNING_OBJECT (client, "request doesn't contain absolute url");
+ if (priv->server_ip == NULL) {
+ GST_WARNING_OBJECT (client, "host information missing");
+ goto bad_request;
+ }
- /* sanitize the uri */
- sanitize_uri (uri);
- state.uri = uri;
+ absolute_uristr =
+ g_strdup_printf ("rtsp://%s%s", priv->server_ip, uristr);
+
+ GST_DEBUG_OBJECT (client, "absolute url: %s", absolute_uristr);
+ if (gst_rtsp_url_parse (absolute_uristr, &uri) != GST_RTSP_OK) {
+ g_free (absolute_uristr);
+ goto bad_request;
+ }
+ g_free (absolute_uristr);
+ } else {
+ g_free (scheme);
+ goto bad_request;
+ }
+ }
/* get the session if there is any */
res = gst_rtsp_message_get_header (request, GST_RTSP_HDR_SESSION, &sessid, 0);
if (res == GST_RTSP_OK) {
- if (client->session_pool == NULL)
+ if (priv->session_pool == NULL)
goto no_pool;
/* we had a session in the request, find it again */
- if (!(session = gst_rtsp_session_pool_find (client->session_pool, sessid)))
+ if (!(session = gst_rtsp_session_pool_find (priv->session_pool, sessid)))
goto session_not_found;
/* we add the session to the client list of watched sessions. When a session
* disappears because it times out, we will be notified. If all sessions are
* gone, we will close the connection */
client_watch_session (client, session);
- } else
- session = NULL;
+ }
- state.session = session;
+ /* sanitize the uri */
+ if (uri)
+ sanitize_uri (uri);
+ ctx->uri = uri;
+ ctx->session = session;
- if (client->auth) {
- if (!gst_rtsp_auth_check (client->auth, client, 0, &state))
- goto not_authorized;
- }
+ if (!gst_rtsp_auth_check (GST_RTSP_AUTH_CHECK_URL))
+ goto not_authorized;
/* now see what is asked and dispatch to a dedicated handler */
switch (method) {
case GST_RTSP_OPTIONS:
- handle_options_request (client, &state);
+ handle_options_request (client, ctx);
break;
case GST_RTSP_DESCRIBE:
- handle_describe_request (client, &state);
+ handle_describe_request (client, ctx);
break;
case GST_RTSP_SETUP:
- handle_setup_request (client, &state);
+ handle_setup_request (client, ctx);
break;
case GST_RTSP_PLAY:
- handle_play_request (client, &state);
+ handle_play_request (client, ctx);
break;
case GST_RTSP_PAUSE:
- handle_pause_request (client, &state);
+ handle_pause_request (client, ctx);
break;
case GST_RTSP_TEARDOWN:
- handle_teardown_request (client, &state);
+ handle_teardown_request (client, ctx);
break;
case GST_RTSP_SET_PARAMETER:
- handle_set_param_request (client, &state);
+ handle_set_param_request (client, ctx);
break;
case GST_RTSP_GET_PARAMETER:
- handle_get_param_request (client, &state);
+ handle_get_param_request (client, ctx);
break;
case GST_RTSP_ANNOUNCE:
case GST_RTSP_RECORD:
case GST_RTSP_REDIRECT:
- send_generic_response (client, GST_RTSP_STS_NOT_IMPLEMENTED, &state);
- break;
+ goto not_implemented;
case GST_RTSP_INVALID:
default:
- send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, &state);
- break;
+ goto bad_request;
}
+
+done:
+ if (ctx == &sctx)
+ gst_rtsp_context_pop_current (ctx);
if (session)
g_object_unref (session);
-
- gst_rtsp_url_free (uri);
+ if (uri)
+ gst_rtsp_url_free (uri);
return;
/* ERRORS */
+not_supported:
+ {
+ GST_ERROR ("client %p: version %d not supported", client, version);
+ send_generic_response (client, GST_RTSP_STS_RTSP_VERSION_NOT_SUPPORTED,
+ ctx);
+ goto done;
+ }
+bad_request:
+ {
+ GST_ERROR ("client %p: bad request", client);
+ send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
+ goto done;
+ }
no_pool:
{
GST_ERROR ("client %p: no pool configured", client);
- send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, &state);
- return;
+ send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
+ goto done;
}
session_not_found:
{
GST_ERROR ("client %p: session not found", client);
- send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, &state);
- return;
+ send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
+ goto done;
}
not_authorized:
{
GST_ERROR ("client %p: not allowed", client);
- handle_unauthorized_request (client, client->auth, &state);
- return;
+ /* error reply is already sent */
+ goto done;
+ }
+not_implemented:
+ {
+ GST_ERROR ("client %p: method %d not implemented", client, method);
+ send_generic_response (client, GST_RTSP_STS_NOT_IMPLEMENTED, ctx);
+ goto done;
+ }
+}
+
+
+static void
+handle_response (GstRTSPClient * client, GstRTSPMessage * response)
+{
+ GstRTSPClientPrivate *priv = client->priv;
+ GstRTSPResult res;
+ GstRTSPSession *session = NULL;
+ GstRTSPContext sctx = { NULL }, *ctx;
+ gchar *sessid;
+
+ if (!(ctx = gst_rtsp_context_get_current ())) {
+ ctx = &sctx;
+ ctx->auth = priv->auth;
+ gst_rtsp_context_push_current (ctx);
+ }
+
+ ctx->conn = priv->connection;
+ ctx->client = client;
+ ctx->request = NULL;
+ ctx->uri = NULL;
+ ctx->method = GST_RTSP_INVALID;
+ ctx->response = response;
+
+ if (gst_debug_category_get_threshold (rtsp_client_debug) >= GST_LEVEL_LOG) {
+ gst_rtsp_message_dump (response);
+ }
+
+ GST_INFO ("client %p: received a response", client);
+
+ /* get the session if there is any */
+ res =
+ gst_rtsp_message_get_header (response, GST_RTSP_HDR_SESSION, &sessid, 0);
+ if (res == GST_RTSP_OK) {
+ if (priv->session_pool == NULL)
+ goto no_pool;
+
+ /* we had a session in the request, find it again */
+ if (!(session = gst_rtsp_session_pool_find (priv->session_pool, sessid)))
+ goto session_not_found;
+
+ /* we add the session to the client list of watched sessions. When a session
+ * disappears because it times out, we will be notified. If all sessions are
+ * gone, we will close the connection */
+ client_watch_session (client, session);
+ }
+
+ ctx->session = session;
+
+ g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_HANDLE_RESPONSE],
+ 0, ctx);
+
+done:
+ if (ctx == &sctx)
+ gst_rtsp_context_pop_current (ctx);
+ if (session)
+ g_object_unref (session);
+ return;
+
+no_pool:
+ {
+ GST_ERROR ("client %p: no pool configured", client);
+ goto done;
+ }
+session_not_found:
+ {
+ GST_ERROR ("client %p: session not found", client);
+ goto done;
}
}
static void
handle_data (GstRTSPClient * client, GstRTSPMessage * message)
{
+ GstRTSPClientPrivate *priv = client->priv;
GstRTSPResult res;
guint8 channel;
GList *walk;
buffer = gst_buffer_new_wrapped (data, size);
handled = FALSE;
- for (walk = client->transports; walk; walk = g_list_next (walk)) {
+ for (walk = priv->transports; walk; walk = g_list_next (walk)) {
GstRTSPStreamTransport *trans;
GstRTSPStream *stream;
- GstRTSPTransport *tr;
+ const GstRTSPTransport *tr;
trans = walk->data;
- /* we only add clients with a transport to the list */
- tr = trans->transport;
- stream = trans->stream;
+ tr = gst_rtsp_stream_transport_get_transport (trans);
+ stream = gst_rtsp_stream_transport_get_stream (trans);
/* check for TCP transport */
if (tr->lower_transport == GST_RTSP_LOWER_TRANS_TCP) {
GstRTSPSessionPool * pool)
{
GstRTSPSessionPool *old;
+ GstRTSPClientPrivate *priv;
g_return_if_fail (GST_IS_RTSP_CLIENT (client));
+ priv = client->priv;
+
if (pool)
g_object_ref (pool);
- g_mutex_lock (&client->lock);
- old = client->session_pool;
- client->session_pool = pool;
- g_mutex_unlock (&client->lock);
+ g_mutex_lock (&priv->lock);
+ old = priv->session_pool;
+ priv->session_pool = pool;
+ g_mutex_unlock (&priv->lock);
if (old)
g_object_unref (old);
GstRTSPSessionPool *
gst_rtsp_client_get_session_pool (GstRTSPClient * client)
{
+ GstRTSPClientPrivate *priv;
GstRTSPSessionPool *result;
g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
- g_mutex_lock (&client->lock);
- if ((result = client->session_pool))
+ priv = client->priv;
+
+ g_mutex_lock (&priv->lock);
+ if ((result = priv->session_pool))
g_object_ref (result);
- g_mutex_unlock (&client->lock);
+ g_mutex_unlock (&priv->lock);
return result;
}
gst_rtsp_client_set_mount_points (GstRTSPClient * client,
GstRTSPMountPoints * mounts)
{
+ GstRTSPClientPrivate *priv;
GstRTSPMountPoints *old;
g_return_if_fail (GST_IS_RTSP_CLIENT (client));
+ priv = client->priv;
+
if (mounts)
g_object_ref (mounts);
- g_mutex_lock (&client->lock);
- old = client->mount_points;
- client->mount_points = mounts;
- g_mutex_unlock (&client->lock);
+ g_mutex_lock (&priv->lock);
+ old = priv->mount_points;
+ priv->mount_points = mounts;
+ g_mutex_unlock (&priv->lock);
if (old)
g_object_unref (old);
GstRTSPMountPoints *
gst_rtsp_client_get_mount_points (GstRTSPClient * client)
{
+ GstRTSPClientPrivate *priv;
GstRTSPMountPoints *result;
g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
- g_mutex_lock (&client->lock);
- if ((result = client->mount_points))
+ priv = client->priv;
+
+ g_mutex_lock (&priv->lock);
+ if ((result = priv->mount_points))
g_object_ref (result);
- g_mutex_unlock (&client->lock);
+ g_mutex_unlock (&priv->lock);
return result;
}
/**
- * gst_rtsp_client_set_use_client_settings:
+ * gst_rtsp_client_set_auth:
* @client: a #GstRTSPClient
- * @use_client_settings: whether to use client settings for multicast
+ * @auth: a #GstRTSPAuth
*
- * Use client transport settings (destination and ttl) for multicast.
- * When @use_client_settings is %FALSE, the server settings will be
- * used.
+ * configure @auth to be used as the authentication manager of @client.
*/
void
-gst_rtsp_client_set_use_client_settings (GstRTSPClient * client,
- gboolean use_client_settings)
+gst_rtsp_client_set_auth (GstRTSPClient * client, GstRTSPAuth * auth)
{
+ GstRTSPClientPrivate *priv;
+ GstRTSPAuth *old;
+
g_return_if_fail (GST_IS_RTSP_CLIENT (client));
- g_mutex_lock (&client->lock);
- client->use_client_settings = use_client_settings;
- g_mutex_unlock (&client->lock);
+ priv = client->priv;
+
+ if (auth)
+ g_object_ref (auth);
+
+ g_mutex_lock (&priv->lock);
+ old = priv->auth;
+ priv->auth = auth;
+ g_mutex_unlock (&priv->lock);
+
+ if (old)
+ g_object_unref (old);
}
+
/**
- * gst_rtsp_client_get_use_client_settings:
+ * gst_rtsp_client_get_auth:
* @client: a #GstRTSPClient
*
- * Check if client transport settings (destination and ttl) for multicast
- * will be used.
+ * Get the #GstRTSPAuth used as the authentication manager of @client.
+ *
+ * Returns: (transfer full): the #GstRTSPAuth of @client. g_object_unref() after
+ * usage.
*/
-gboolean
-gst_rtsp_client_get_use_client_settings (GstRTSPClient * client)
+GstRTSPAuth *
+gst_rtsp_client_get_auth (GstRTSPClient * client)
{
- gboolean res;
+ GstRTSPClientPrivate *priv;
+ GstRTSPAuth *result;
- g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), FALSE);
+ g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
- g_mutex_lock (&client->lock);
- res = client->use_client_settings;
- g_mutex_unlock (&client->lock);
+ priv = client->priv;
- return res;
+ g_mutex_lock (&priv->lock);
+ if ((result = priv->auth))
+ g_object_ref (result);
+ g_mutex_unlock (&priv->lock);
+
+ return result;
}
/**
- * gst_rtsp_client_set_auth:
+ * gst_rtsp_client_set_thread_pool:
* @client: a #GstRTSPClient
- * @auth: a #GstRTSPAuth
+ * @pool: a #GstRTSPThreadPool
*
- * configure @auth to be used as the authentication manager of @client.
+ * configure @pool to be used as the thread pool of @client.
*/
void
-gst_rtsp_client_set_auth (GstRTSPClient * client, GstRTSPAuth * auth)
+gst_rtsp_client_set_thread_pool (GstRTSPClient * client,
+ GstRTSPThreadPool * pool)
{
- GstRTSPAuth *old;
+ GstRTSPClientPrivate *priv;
+ GstRTSPThreadPool *old;
g_return_if_fail (GST_IS_RTSP_CLIENT (client));
- if (auth)
- g_object_ref (auth);
+ priv = client->priv;
- g_mutex_lock (&client->lock);
- old = client->auth;
- client->auth = auth;
- g_mutex_unlock (&client->lock);
+ if (pool)
+ g_object_ref (pool);
+
+ g_mutex_lock (&priv->lock);
+ old = priv->thread_pool;
+ priv->thread_pool = pool;
+ g_mutex_unlock (&priv->lock);
if (old)
g_object_unref (old);
}
-
/**
- * gst_rtsp_client_get_auth:
+ * gst_rtsp_client_get_thread_pool:
* @client: a #GstRTSPClient
*
- * Get the #GstRTSPAuth used as the authentication manager of @client.
+ * Get the #GstRTSPThreadPool used as the thread pool of @client.
*
- * Returns: (transfer full): the #GstRTSPAuth of @client. g_object_unref() after
+ * Returns: (transfer full): the #GstRTSPThreadPool of @client. g_object_unref() after
* usage.
*/
-GstRTSPAuth *
-gst_rtsp_client_get_auth (GstRTSPClient * client)
+GstRTSPThreadPool *
+gst_rtsp_client_get_thread_pool (GstRTSPClient * client)
{
- GstRTSPAuth *result;
+ GstRTSPClientPrivate *priv;
+ GstRTSPThreadPool *result;
g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
- g_mutex_lock (&client->lock);
- if ((result = client->auth))
+ priv = client->priv;
+
+ g_mutex_lock (&priv->lock);
+ if ((result = priv->thread_pool))
g_object_ref (result);
- g_mutex_unlock (&client->lock);
+ g_mutex_unlock (&priv->lock);
return result;
}
-static GstRTSPResult
-message_received (GstRTSPWatch * watch, GstRTSPMessage * message,
- gpointer user_data)
+/**
+ * gst_rtsp_client_set_connection:
+ * @client: a #GstRTSPClient
+ * @conn: (transfer full): a #GstRTSPConnection
+ *
+ * Set the #GstRTSPConnection of @client. This function takes ownership of
+ * @conn.
+ *
+ * Returns: %TRUE on success.
+ */
+gboolean
+gst_rtsp_client_set_connection (GstRTSPClient * client,
+ GstRTSPConnection * conn)
{
- GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
+ GstRTSPClientPrivate *priv;
+ GSocket *read_socket;
+ GSocketAddress *address;
+ GstRTSPUrl *url;
+ GError *error = NULL;
+
+ g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), FALSE);
+ g_return_val_if_fail (conn != NULL, FALSE);
+
+ priv = client->priv;
+
+ read_socket = gst_rtsp_connection_get_read_socket (conn);
+
+ if (!(address = g_socket_get_local_address (read_socket, &error)))
+ goto no_address;
+
+ g_free (priv->server_ip);
+ /* keep the original ip that the client connected to */
+ if (G_IS_INET_SOCKET_ADDRESS (address)) {
+ GInetAddress *iaddr;
+
+ iaddr = g_inet_socket_address_get_address (G_INET_SOCKET_ADDRESS (address));
+
+ /* socket might be ipv6 but adress still ipv4 */
+ priv->is_ipv6 = g_inet_address_get_family (iaddr) == G_SOCKET_FAMILY_IPV6;
+ priv->server_ip = g_inet_address_to_string (iaddr);
+ g_object_unref (address);
+ } else {
+ priv->is_ipv6 = g_socket_get_family (read_socket) == G_SOCKET_FAMILY_IPV6;
+ priv->server_ip = g_strdup ("unknown");
+ }
+
+ GST_INFO ("client %p connected to server ip %s, ipv6 = %d", client,
+ priv->server_ip, priv->is_ipv6);
+
+ url = gst_rtsp_connection_get_url (conn);
+ GST_INFO ("added new client %p ip %s:%d", client, url->host, url->port);
+
+ priv->connection = conn;
+
+ return TRUE;
+
+ /* ERRORS */
+no_address:
+ {
+ GST_ERROR ("could not get local address %s", error->message);
+ g_error_free (error);
+ return FALSE;
+ }
+}
+
+/**
+ * gst_rtsp_client_get_connection:
+ * @client: a #GstRTSPClient
+ *
+ * Get the #GstRTSPConnection of @client.
+ *
+ * Returns: (transfer none): the #GstRTSPConnection of @client.
+ * The connection object returned remains valid until the client is freed.
+ */
+GstRTSPConnection *
+gst_rtsp_client_get_connection (GstRTSPClient * client)
+{
+ g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
+
+ return client->priv->connection;
+}
+
+/**
+ * gst_rtsp_client_set_send_func:
+ * @client: a #GstRTSPClient
+ * @func: a #GstRTSPClientSendFunc
+ * @user_data: user data passed to @func
+ * @notify: called when @user_data is no longer in use
+ *
+ * Set @func as the callback that will be called when a new message needs to be
+ * sent to the client. @user_data is passed to @func and @notify is called when
+ * @user_data is no longer in use.
+ *
+ * By default, the client will send the messages on the #GstRTSPConnection that
+ * was configured with gst_rtsp_client_attach() was called.
+ */
+void
+gst_rtsp_client_set_send_func (GstRTSPClient * client,
+ GstRTSPClientSendFunc func, gpointer user_data, GDestroyNotify notify)
+{
+ GstRTSPClientPrivate *priv;
+ GDestroyNotify old_notify;
+ gpointer old_data;
+
+ g_return_if_fail (GST_IS_RTSP_CLIENT (client));
+
+ priv = client->priv;
+
+ g_mutex_lock (&priv->send_lock);
+ priv->send_func = func;
+ old_notify = priv->send_notify;
+ old_data = priv->send_data;
+ priv->send_notify = notify;
+ priv->send_data = user_data;
+ g_mutex_unlock (&priv->send_lock);
+
+ if (old_notify)
+ old_notify (old_data);
+}
+
+/**
+ * gst_rtsp_client_handle_message:
+ * @client: a #GstRTSPClient
+ * @message: an #GstRTSPMessage
+ *
+ * Let the client handle @message.
+ *
+ * Returns: a #GstRTSPResult.
+ */
+GstRTSPResult
+gst_rtsp_client_handle_message (GstRTSPClient * client,
+ GstRTSPMessage * message)
+{
+ g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), GST_RTSP_EINVAL);
+ g_return_val_if_fail (message != NULL, GST_RTSP_EINVAL);
switch (message->type) {
case GST_RTSP_MESSAGE_REQUEST:
handle_request (client, message);
break;
case GST_RTSP_MESSAGE_RESPONSE:
+ handle_response (client, message);
break;
case GST_RTSP_MESSAGE_DATA:
handle_data (client, message);
return GST_RTSP_OK;
}
+/**
+ * gst_rtsp_client_send_message:
+ * @client: a #GstRTSPClient
+ * @session: a #GstRTSPSession to send the message to or %NULL
+ * @message: The #GstRTSPMessage to send
+ *
+ * Send a message message to the remote end. @message must be a
+ * #GST_RTSP_MESSAGE_REQUEST or a #GST_RTSP_MESSAGE_RESPONSE.
+ */
+GstRTSPResult
+gst_rtsp_client_send_message (GstRTSPClient * client, GstRTSPSession * session,
+ GstRTSPMessage * message)
+{
+ g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), GST_RTSP_EINVAL);
+ g_return_val_if_fail (message != NULL, GST_RTSP_EINVAL);
+ g_return_val_if_fail (message->type == GST_RTSP_MESSAGE_REQUEST ||
+ message->type == GST_RTSP_MESSAGE_RESPONSE, GST_RTSP_EINVAL);
+
+ send_message (client, session, message, FALSE);
+
+ return GST_RTSP_OK;
+}
+
+static GstRTSPResult
+do_send_message (GstRTSPClient * client, GstRTSPMessage * message,
+ gboolean close, gpointer user_data)
+{
+ GstRTSPClientPrivate *priv = client->priv;
+
+ /* send the response and store the seq number so we can wait until it's
+ * written to the client to close the connection */
+ return gst_rtsp_watch_send_message (priv->watch, message, close ?
+ &priv->close_seq : NULL);
+}
+
+static GstRTSPResult
+message_received (GstRTSPWatch * watch, GstRTSPMessage * message,
+ gpointer user_data)
+{
+ return gst_rtsp_client_handle_message (GST_RTSP_CLIENT (user_data), message);
+}
+
static GstRTSPResult
message_sent (GstRTSPWatch * watch, guint cseq, gpointer user_data)
{
- GstRTSPClient *client;
+ GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
+ GstRTSPClientPrivate *priv = client->priv;
- client = GST_RTSP_CLIENT (user_data);
- if (client->close_response_seq && client->close_response_seq == cseq) {
- client->close_response_seq = 0;
+ if (priv->close_seq && priv->close_seq == cseq) {
+ priv->close_seq = 0;
close_connection (client);
}
closed (GstRTSPWatch * watch, gpointer user_data)
{
GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
+ GstRTSPClientPrivate *priv = client->priv;
const gchar *tunnelid;
GST_INFO ("client %p: connection closed", client);
- if ((tunnelid = gst_rtsp_connection_get_tunnelid (client->connection))) {
+ if ((tunnelid = gst_rtsp_connection_get_tunnelid (priv->connection))) {
g_mutex_lock (&tunnels_lock);
/* remove from tunnelids */
g_hash_table_remove (tunnels, tunnelid);
g_mutex_unlock (&tunnels_lock);
}
+ gst_rtsp_client_set_send_func (client, NULL, NULL, NULL);
+
return GST_RTSP_OK;
}
str = gst_rtsp_strresult (result);
GST_INFO
- ("client %p: received an error %s when handling message %p with id %d",
- client, str, message, id);
+ ("client %p: error when handling message %p with id %d: %s",
+ client, message, id, str);
g_free (str);
return GST_RTSP_OK;
static gboolean
remember_tunnel (GstRTSPClient * client)
{
+ GstRTSPClientPrivate *priv = client->priv;
const gchar *tunnelid;
/* store client in the pending tunnels */
- tunnelid = gst_rtsp_connection_get_tunnelid (client->connection);
+ tunnelid = gst_rtsp_connection_get_tunnelid (priv->connection);
if (tunnelid == NULL)
goto no_tunnelid;
static GstRTSPStatusCode
tunnel_start (GstRTSPWatch * watch, gpointer user_data)
{
- GstRTSPClient *client;
-
- client = GST_RTSP_CLIENT (user_data);
+ GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
+ GstRTSPClientPrivate *priv = client->priv;
GST_INFO ("client %p: tunnel start (connection %p)", client,
- client->connection);
+ priv->connection);
if (!remember_tunnel (client))
goto tunnel_error;
static GstRTSPResult
tunnel_lost (GstRTSPWatch * watch, gpointer user_data)
{
- GstRTSPClient *client;
-
- client = GST_RTSP_CLIENT (user_data);
+ GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
+ GstRTSPClientPrivate *priv = client->priv;
GST_WARNING ("client %p: tunnel lost (connection %p)", client,
- client->connection);
+ priv->connection);
/* ignore error, it'll only be a problem when the client does a POST again */
remember_tunnel (client);
{
const gchar *tunnelid;
GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
+ GstRTSPClientPrivate *priv = client->priv;
GstRTSPClient *oclient;
+ GstRTSPClientPrivate *opriv;
GST_INFO ("client %p: tunnel complete", client);
/* find previous tunnel */
- tunnelid = gst_rtsp_connection_get_tunnelid (client->connection);
+ tunnelid = gst_rtsp_connection_get_tunnelid (priv->connection);
if (tunnelid == NULL)
goto no_tunnelid;
g_object_ref (oclient);
g_hash_table_remove (tunnels, tunnelid);
- if (oclient->watch == NULL)
+ opriv = oclient->priv;
+
+ if (opriv->watch == NULL)
goto tunnel_closed;
g_mutex_unlock (&tunnels_lock);
GST_INFO ("client %p: found tunnel %p (old %p, new %p)", client, oclient,
- oclient->connection, client->connection);
+ opriv->connection, priv->connection);
/* merge the tunnels into the first client */
- gst_rtsp_connection_do_tunnel (oclient->connection, client->connection);
- gst_rtsp_watch_reset (oclient->watch);
+ gst_rtsp_connection_do_tunnel (opriv->connection, priv->connection);
+ gst_rtsp_watch_reset (opriv->watch);
g_object_unref (oclient);
return GST_RTSP_OK;
static void
client_watch_notify (GstRTSPClient * client)
{
+ GstRTSPClientPrivate *priv = client->priv;
+
GST_INFO ("client %p: watch destroyed", client);
- client->watch = NULL;
+ priv->watch = NULL;
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_CLOSED], 0, NULL);
g_object_unref (client);
}
-static gboolean
-setup_client (GstRTSPClient * client, GSocket * socket,
- GstRTSPConnection * conn, GError ** error)
-{
- GSocket *read_socket;
- GSocketAddress *address;
- GstRTSPUrl *url;
-
- read_socket = gst_rtsp_connection_get_read_socket (conn);
- client->is_ipv6 = g_socket_get_family (socket) == G_SOCKET_FAMILY_IPV6;
-
- if (!(address = g_socket_get_remote_address (read_socket, error)))
- goto no_address;
-
- g_free (client->server_ip);
- /* keep the original ip that the client connected to */
- if (G_IS_INET_SOCKET_ADDRESS (address)) {
- GInetAddress *iaddr;
-
- iaddr = g_inet_socket_address_get_address (G_INET_SOCKET_ADDRESS (address));
-
- client->server_ip = g_inet_address_to_string (iaddr);
- g_object_unref (address);
- } else {
- client->server_ip = g_strdup ("unknown");
- }
-
- GST_INFO ("client %p connected to server ip %s, ipv6 = %d", client,
- client->server_ip, client->is_ipv6);
-
- url = gst_rtsp_connection_get_url (conn);
- GST_INFO ("added new client %p ip %s:%d", client, url->host, url->port);
-
- client->connection = conn;
-
- return TRUE;
-
- /* ERRORS */
-no_address:
- {
- GST_ERROR ("could not get remote address %s", (*error)->message);
- return FALSE;
- }
-}
-
/**
- * gst_rtsp_client_use_socket:
+ * gst_rtsp_client_attach:
* @client: a #GstRTSPClient
- * @socket: a #GSocket
- * @ip: the IP address of the remote client
- * @port: the port used by the other end
- * @initial_buffer: any zero terminated initial data that was already read from
- * the socket
- * @error: a #GError
+ * @context: (allow-none): a #GMainContext
+ *
+ * Attaches @client to @context. When the mainloop for @context is run, the
+ * client will be dispatched. When @context is %NULL, the default context will be
+ * used).
*
- * Take an existing network socket and use it for an RTSP connection.
+ * This function should be called when the client properties and urls are fully
+ * configured and the client is ready to start.
*
- * Returns: %TRUE on success.
+ * Returns: the ID (greater than 0) for the source within the GMainContext.
*/
-gboolean
-gst_rtsp_client_use_socket (GstRTSPClient * client, GSocket * socket,
- const gchar * ip, gint port, const gchar * initial_buffer, GError ** error)
+guint
+gst_rtsp_client_attach (GstRTSPClient * client, GMainContext * context)
{
- GstRTSPConnection *conn;
- GstRTSPResult res;
+ GstRTSPClientPrivate *priv;
+ guint res;
- g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), FALSE);
- g_return_val_if_fail (G_IS_SOCKET (socket), FALSE);
+ g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), 0);
+ priv = client->priv;
+ g_return_val_if_fail (priv->connection != NULL, 0);
+ g_return_val_if_fail (priv->watch == NULL, 0);
- GST_RTSP_CHECK (gst_rtsp_connection_create_from_socket (socket, ip, port,
- initial_buffer, &conn), no_connection);
+ /* create watch for the connection and attach */
+ priv->watch = gst_rtsp_watch_new (priv->connection, &watch_funcs,
+ g_object_ref (client), (GDestroyNotify) client_watch_notify);
+ gst_rtsp_client_set_send_func (client, do_send_message, priv->watch,
+ (GDestroyNotify) gst_rtsp_watch_unref);
- return setup_client (client, socket, conn, error);
+ /* FIXME make this configurable. We don't want to do this yet because it will
+ * be superceeded by a cache object later */
+ gst_rtsp_watch_set_send_backlog (priv->watch, 0, 100);
- /* ERRORS */
-no_connection:
- {
- gchar *str = gst_rtsp_strresult (res);
+ GST_INFO ("attaching to context %p", context);
+ res = gst_rtsp_watch_attach (priv->watch, context);
- GST_ERROR ("could not create connection from socket %p: %s", socket, str);
- g_free (str);
- return FALSE;
- }
+ return res;
}
/**
- * gst_rtsp_client_accept:
+ * gst_rtsp_client_session_filter:
* @client: a #GstRTSPClient
- * @socket: a #GSocket
- * @context: the context to run in
- * @cancellable: a #GCancellable
- * @error: a #GError
+ * @func: (scope call) (allow-none): a callback
+ * @user_data: user data passed to @func
+ *
+ * Call @func for each session managed by @client. The result value of @func
+ * determines what happens to the session. @func will be called with @client
+ * locked so no further actions on @client can be performed from @func.
*
- * Accept a new connection for @client on @socket.
+ * If @func returns #GST_RTSP_FILTER_REMOVE, the session will be removed from
+ * @client.
*
- * Returns: %TRUE if the client could be accepted.
+ * If @func returns #GST_RTSP_FILTER_KEEP, the session will remain in @client.
+ *
+ * If @func returns #GST_RTSP_FILTER_REF, the session will remain in @client but
+ * will also be added with an additional ref to the result #GList of this
+ * function..
+ *
+ * When @func is %NULL, #GST_RTSP_FILTER_REF will be assumed for each session.
+ *
+ * Returns: (element-type GstRTSPSession) (transfer full): a #GList with all
+ * sessions for which @func returned #GST_RTSP_FILTER_REF. After usage, each
+ * element in the #GList should be unreffed before the list is freed.
*/
-gboolean
-gst_rtsp_client_accept (GstRTSPClient * client, GSocket * socket,
- GCancellable * cancellable, GError ** error)
+GList *
+gst_rtsp_client_session_filter (GstRTSPClient * client,
+ GstRTSPClientSessionFilterFunc func, gpointer user_data)
{
- GstRTSPConnection *conn;
- GstRTSPResult res;
+ GstRTSPClientPrivate *priv;
+ GList *result, *walk, *next;
- g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), FALSE);
- g_return_val_if_fail (G_IS_SOCKET (socket), FALSE);
+ g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
- /* a new client connected. */
- GST_RTSP_CHECK (gst_rtsp_connection_accept (socket, &conn, cancellable),
- accept_failed);
+ priv = client->priv;
- return setup_client (client, socket, conn, error);
+ result = NULL;
- /* ERRORS */
-accept_failed:
- {
- gchar *str = gst_rtsp_strresult (res);
+ g_mutex_lock (&priv->lock);
+ for (walk = priv->sessions; walk; walk = next) {
+ GstRTSPSession *sess = walk->data;
+ GstRTSPFilterResult res;
- GST_ERROR ("Could not accept client on server socket %p: %s", socket, str);
- g_free (str);
- return FALSE;
- }
-}
+ next = g_list_next (walk);
-/**
- * gst_rtsp_client_attach:
- * @client: a #GstRTSPClient
- * @context: (allow-none): a #GMainContext
- *
- * Attaches @client to @context. When the mainloop for @context is run, the
- * client will be dispatched. When @context is NULL, the default context will be
- * used).
- *
- * This function should be called when the client properties and urls are fully
- * configured and the client is ready to start.
- *
- * Returns: the ID (greater than 0) for the source within the GMainContext.
- */
-guint
-gst_rtsp_client_attach (GstRTSPClient * client, GMainContext * context)
-{
- guint res;
-
- g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), 0);
- g_return_val_if_fail (client->watch == NULL, 0);
+ if (func)
+ res = func (client, sess, user_data);
+ else
+ res = GST_RTSP_FILTER_REF;
- /* create watch for the connection and attach */
- client->watch = gst_rtsp_watch_new (client->connection, &watch_funcs,
- g_object_ref (client), (GDestroyNotify) client_watch_notify);
-
- GST_INFO ("attaching to context %p", context);
- res = gst_rtsp_watch_attach (client->watch, context);
- gst_rtsp_watch_unref (client->watch);
+ switch (res) {
+ case GST_RTSP_FILTER_REMOVE:
+ /* stop watching the session and pretent it went away */
+ client_cleanup_session (client, sess);
+ break;
+ case GST_RTSP_FILTER_REF:
+ result = g_list_prepend (result, g_object_ref (sess));
+ break;
+ case GST_RTSP_FILTER_KEEP:
+ default:
+ break;
+ }
+ }
+ g_mutex_unlock (&priv->lock);
- return res;
+ return result;
}
projects / platform / upstream / gstreamer.git / blobdiff