rtsp-server: Add/fix various annotations

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3194>

subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-auth.c

@@ -1231,6 +1231,7 @@ gst_rtsp_auth_make_basic (const gchar * user, const gchar * pass)
 
 /**
  * gst_rtsp_auth_set_realm:
+ * @realm: (nullable): The realm to set
  *
  * Set the @realm of @auth
  *
@@ -1251,7 +1252,7 @@ gst_rtsp_auth_set_realm (GstRTSPAuth * auth, const gchar * realm)
 /**
  * gst_rtsp_auth_get_realm:
  *
- * Returns: (transfer full): the @realm of @auth
+ * Returns: (transfer full) (nullable): the @realm of @auth
  *
  * Since: 1.16
  */

subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-latency-bin.c

@@ -336,7 +336,7 @@ gst_rtsp_latency_bin_change_state (GstElement * element, GstStateChange
  * 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)

subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-media-factory.c

@@ -1375,7 +1375,7 @@ weak_ref_free (GWeakRef * ref)
  * 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,
@@ -1534,7 +1534,7 @@ gst_rtsp_media_factory_set_clock (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
  */
@@ -1987,7 +1987,7 @@ default_configure (GstRTSPMediaFactory * factory, GstRTSPMedia * media)
  * 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,

subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-media.c

@@ -4316,7 +4316,7 @@ not_prepared:
  * 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,

subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-onvif-media-factory.c

@@ -360,7 +360,7 @@ gst_rtsp_onvif_media_factory_init (GstRTSPOnvifMediaFactory * factory)
 /**
  * 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.
@@ -400,7 +400,7 @@ gst_rtsp_onvif_media_factory_set_backchannel_launch (GstRTSPOnvifMediaFactory *
  * 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

subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-server.c

@@ -434,7 +434,7 @@ gst_rtsp_server_set_service (GstRTSPServer * server, const gchar * service)
  *
  * 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)
@@ -1318,7 +1318,7 @@ watch_destroyed (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
@@ -1380,7 +1380,7 @@ no_socket:
 /**
  * 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
@@ -1427,7 +1427,7 @@ no_source:
 /**
  * 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

subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-stream.c

@@ -1993,7 +1993,7 @@ gst_rtsp_stream_get_server_port (GstRTSPStream * stream,
  *
  * 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)
@@ -2019,7 +2019,7 @@ 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)