Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3194>
/**
* gst_rtsp_auth_set_realm:
+ * @realm: (nullable): The realm to set
*
* Set the @realm of @auth
*
/**
* gst_rtsp_auth_get_realm:
*
- * Returns: (transfer full): the @realm of @auth
+ * Returns: (transfer full) (nullable): the @realm of @auth
*
* Since: 1.16
*/
* Create a bin that encapsulates an @element and prevents it from affecting
* latency on the whole pipeline.
*
- * Returns: A newly created #GstRTSPLatencyBin element, or %NULL on failure
+ * Returns: (nullable): A newly created #GstRTSPLatencyBin element, or %NULL on failure
*/
GstElement *
gst_rtsp_latency_bin_new (GstElement * element)
* After the media is constructed, it can be configured and then prepared
* with gst_rtsp_media_prepare ().
*
- * Returns: (transfer full): a new #GstRTSPMedia if the media could be prepared.
+ * Returns: (transfer full) (nullable): a new #GstRTSPMedia if the media could be prepared.
*/
GstRTSPMedia *
gst_rtsp_media_factory_construct (GstRTSPMediaFactory * factory,
* Returns the clock that is going to be used by the pipelines
* of all medias created from this factory.
*
- * Returns: (transfer full): The GstClock
+ * Returns: (transfer full) (nullable): The GstClock
*
* Since: 1.8
*/
* implementation of this function returns the bin created from the
* launch parameter.
*
- * Returns: (transfer floating): a new #GstElement.
+ * Returns: (transfer floating) (nullable): a new #GstElement.
*/
GstElement *
gst_rtsp_media_factory_create_element (GstRTSPMediaFactory * factory,
* Get the #GstNetTimeProvider for the clock used by @media. The time provider
* will listen on @address and @port for client time requests.
*
- * Returns: (transfer full): the #GstNetTimeProvider of @media.
+ * Returns: (transfer full) (nullable): the #GstNetTimeProvider of @media.
*/
GstNetTimeProvider *
gst_rtsp_media_get_time_provider (GstRTSPMedia * media, const gchar * address,
/**
* gst_rtsp_onvif_media_factory_set_backchannel_launch:
* @factory: a #GstRTSPMediaFactory
- * @launch: the launch description
+ * @launch: (nullable): the launch description
*
* The gst_parse_launch() line to use for constructing the ONVIF backchannel
* pipeline in the default prepare vmethod if requested by the client.
* Get the gst_parse_launch() pipeline description that will be used in the
* default prepare vmethod for generating the ONVIF backchannel pipeline.
*
- * Returns: (transfer full): the configured backchannel launch description. g_free() after
+ * Returns: (transfer full) (nullable): the configured backchannel launch description. g_free() after
* usage.
*
* Since: 1.14
*
* Get the service on which the server will accept connections.
*
- * Returns: (transfer full) (nullable): the service. use g_free() after usage.
+ * Returns: (transfer full): the service. use g_free() after usage.
*/
gchar *
gst_rtsp_server_get_service (GstRTSPServer * server)
/**
* gst_rtsp_server_create_source:
* @server: a #GstRTSPServer
- * @cancellable: (allow-none): a #GCancellable or %NULL.
+ * @cancellable: (nullable): a #GCancellable or %NULL.
* @error: (out): a #GError
*
* Create a #GSource for @server. The new source will have a default
/**
* gst_rtsp_server_attach:
* @server: a #GstRTSPServer
- * @context: (allow-none): a #GMainContext
+ * @context: (nullable): 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
/**
* gst_rtsp_server_client_filter:
* @server: a #GstRTSPServer
- * @func: (scope call) (allow-none): a callback
+ * @func: (scope call) (nullable): a callback
* @user_data: user data passed to @func
*
* Call @func for each client managed by @server. The result value of @func
*
* Get the RTP session of this stream.
*
- * Returns: (transfer full): The RTP session of this stream. Unref after usage.
+ * Returns: (transfer full) (nullable): The RTP session of this stream. Unref after usage.
*/
GObject *
gst_rtsp_stream_get_rtpsession (GstRTSPStream * stream)
*
* Get the SRTP encoder for this stream.
*
- * Returns: (transfer full): The SRTP encoder for this stream. Unref after usage.
+ * Returns: (transfer full) (nullable): The SRTP encoder for this stream. Unref after usage.
*/
GstElement *
gst_rtsp_stream_get_srtp_encoder (GstRTSPStream * stream)