* gst_rtsp_auth_setup_auth:
* @auth: a #GstRTSPAuth
* @client: the client
- * @uri: the requested uri
- * @session: the session
- * @request: the request
- * @response: the response
+ * @hint: TODO
+ * @state: TODO
*
* Add authentication tokens to @response.
*
*
* Get the #GstRTSPSessionPool object that @client uses to manage its sessions.
*
- * Returns: a #GstRTSPSessionPool, unref after usage.
+ * Returns: (transfer full): a #GstRTSPSessionPool, unref after usage.
*/
GstRTSPSessionPool *
gst_rtsp_client_get_session_pool (GstRTSPClient * client)
*
* Get the #GstRTSPServer object that @client was created from.
*
- * Returns: a #GstRTSPServer, unref after usage.
+ * Returns: (transfer full): a #GstRTSPServer, unref after usage.
*/
GstRTSPServer *
gst_rtsp_client_get_server (GstRTSPClient * client)
*
* Get the #GstRTSPMediaMapping object that @client uses to manage its sessions.
*
- * Returns: a #GstRTSPMediaMapping, unref after usage.
+ * Returns: (transfer full): a #GstRTSPMediaMapping, unref after usage.
*/
GstRTSPMediaMapping *
gst_rtsp_client_get_media_mapping (GstRTSPClient * client)
*
* Get the #GstRTSPAuth used as the authentication manager of @client.
*
- * Returns: the #GstRTSPAuth of @client. g_object_unref() after
+ * Returns: (transfer full): the #GstRTSPAuth of @client. g_object_unref() after
* usage.
*/
GstRTSPAuth *
gobject_class->finalize = gst_rtsp_media_factory_uri_finalize;
/**
- * GstRTSPMediaFactoryURI::uri
+ * GstRTSPMediaFactoryURI::uri:
*
* The uri of the resource that will be served by this factory.
*/
"The URI of the resource to stream", DEFAULT_URI,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
- * GstRTSPMediaFactoryURI::use-gstpay
+ * GstRTSPMediaFactoryURI::use-gstpay:
*
* Allow the usage of gstpay in order to avoid decoding of compressed formats
* without a payloader.
gobject_class->finalize = gst_rtsp_media_factory_finalize;
/**
- * GstRTSPMediaFactory::launch
+ * GstRTSPMediaFactory::launch:
*
* The gst_parse_launch() line to use for constructing the pipeline in the
* default prepare vmethod.
*
* Get the #GstRTSPAuth used as the authentication manager of @factory.
*
- * Returns: the #GstRTSPAuth of @factory. g_object_unref() after
+ * Returns: (transfer full): the #GstRTSPAuth of @factory. g_object_unref() after
* usage.
*/
GstRTSPAuth *
* the srcpad member set to a source pad that produces buffer of type
* application/x-rtp.
*
- * Returns: a new #GstRTSPMedia if the media could be prepared.
+ * Returns: (transfer full): a new #GstRTSPMedia if the media could be prepared.
*/
GstRTSPMedia *
gst_rtsp_media_factory_construct (GstRTSPMediaFactory * factory,
* Find the #GstRTSPMediaFactory for @url. The default implementation of this object
* will use the mappings added with gst_rtsp_media_mapping_add_factory ().
*
- * Returns: the #GstRTSPMediaFactory for @url. g_object_unref() after usage.
+ * Returns: (transfer full): the #GstRTSPMediaFactory for @url. g_object_unref() after usage.
*/
GstRTSPMediaFactory *
gst_rtsp_media_mapping_find_factory (GstRTSPMediaMapping * mapping,
*
* Get the #GstRTSPAuth used as the authentication manager of @media.
*
- * Returns: the #GstRTSPAuth of @media. g_object_unref() after
+ * Returns: (transfer full): the #GstRTSPAuth of @media. g_object_unref() after
* usage.
*/
GstRTSPAuth *
gobject_class->finalize = gst_rtsp_server_finalize;
/**
- * GstRTSPServer::address
+ * GstRTSPServer::address:
*
* The address of the server. This is the address where the server will
* listen on.
"The address the server uses to listen on", DEFAULT_ADDRESS,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
- * GstRTSPServer::service
+ * GstRTSPServer::service:
*
* The service of the server. This is either a string with the service name or
* a port number (as a string) the server will listen on.
"The service or port number the server uses to listen on",
DEFAULT_SERVICE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
- * GstRTSPServer::bound-port
+ * GstRTSPServer::bound-port:
*
* The actual port the server is listening on. Can be used to retrieve the
* port number when the server is started on port 0, which means bind to a
-1, G_MAXUINT16, DEFAULT_BOUND_PORT,
G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
/**
- * GstRTSPServer::backlog
+ * GstRTSPServer::backlog:
*
* The backlog argument defines the maximum length to which the queue of
* pending connections for the server may grow. If a connection request arrives
"of pending connections may grow", 0, G_MAXINT, DEFAULT_BACKLOG,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
- * GstRTSPServer::session-pool
+ * GstRTSPServer::session-pool:
*
* The session pool of the server. By default each server has a separate
* session pool but sessions can be shared between servers by setting the same
GST_TYPE_RTSP_SESSION_POOL,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
- * GstRTSPServer::media-mapping
+ * GstRTSPServer::media-mapping:
*
* The media mapping to use for this server. By default the server has no
* media mapping and thus cannot map urls to media streams.
*
* Get the #GstRTSPSessionPool used as the session pool of @server.
*
- * Returns: the #GstRTSPSessionPool used for sessions. g_object_unref() after
+ * Returns: (transfer full): the #GstRTSPSessionPool used for sessions. g_object_unref() after
* usage.
*/
GstRTSPSessionPool *
*
* Get the #GstRTSPMediaMapping used as the media mapping of @server.
*
- * Returns: the #GstRTSPMediaMapping of @server. g_object_unref() after
+ * Returns: (transfer full): the #GstRTSPMediaMapping of @server. g_object_unref() after
* usage.
*/
GstRTSPMediaMapping *
*
* Get the #GstRTSPAuth used as the authentication manager of @server.
*
- * Returns: the #GstRTSPAuth of @server. g_object_unref() after
+ * Returns: (transfer full): the #GstRTSPAuth of @server. g_object_unref() after
* usage.
*/
GstRTSPAuth *
* Create a #GSocket for @server. The socket will listen on the
* configured service.
*
- * Returns: the #GSocket for @server or NULL when an error occured.
+ * Returns: (transfer full): the #GSocket for @server or NULL when an error occured.
*/
GSocket *
gst_rtsp_server_create_socket (GstRTSPServer * server,
/**
* gst_rtsp_server_attach:
* @server: a #GstRTSPServer
- * @context: a #GMainContext
- * @error: a #GError
+ * @context: (allow-none): a #GMainContext
*
* Attaches @server to @context. When the mainloop for @context is run, the
* server will be dispatched. When @context is NULL, the default context will be
* Find the session with @sessionid in @pool. The access time of the session
* will be updated with gst_rtsp_session_touch().
*
- * Returns: the #GstRTSPSession with @sessionid or %NULL when the session did
+ * Returns: (transfer full): the #GstRTSPSession with @sessionid or %NULL when the session did
* not exist. g_object_unref() after usage.
*/
GstRTSPSession *
*
* Create a new #GstRTSPSession object in @pool.
*
- * Returns: a new #GstRTSPSession.
+ * Returns: (transfer none): a new #GstRTSPSession.
*/
GstRTSPSession *
gst_rtsp_session_pool_create (GstRTSPSessionPool * pool)
/**
* gst_rtsp_session_pool_filter:
* @pool: a #GstRTSPSessionPool
- * @func: a callback
+ * @func: (scope call): a callback
* @user_data: user data passed to @func
*
* Call @func for each session in @pool. The result value of @func determines
* will also be added with an additional ref to the result GList of this
* function..
*
- * Returns: 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.
+ * 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.
*/
GList *
gst_rtsp_session_pool_filter (GstRTSPSessionPool * pool,
* gst_rtsp_session_manage_media:
* @sess: a #GstRTSPSession
* @uri: the uri for the media
- * @media: a #GstRTSPMedia
+ * @media: (transfer full): a #GstRTSPMedia
*
* Manage the media object @obj in @sess. @uri will be used to retrieve this
* media from the session with gst_rtsp_session_get_media().
/**
* gst_rtsp_session_stream_set_callbacks:
* @stream: a #GstRTSPSessionStream
- * @send_rtp: a callback called when RTP should be sent
- * @send_rtcp: a callback called when RTCP should be sent
- * @send_rtp_list: a callback called when RTP should be sent
- * @send_rtcp_list: a callback called when RTCP should be sent
+ * @send_rtp: (scope notified): a callback called when RTP should be sent
+ * @send_rtcp: (scope notified): a callback called when RTCP should be sent
* @user_data: user data passed to callbacks
* @notify: called with the user_data when no longer needed.
*
projects / platform / upstream / gstreamer.git / commitdiff