rtsp-server: fixed comments and GIR annotations

https://bugzilla.gnome.org/show_bug.cgi?id=680777

gst/rtsp-server/rtsp-auth.c

@@ -152,10 +152,8 @@ default_setup_auth (GstRTSPAuth * auth, GstRTSPClient * client,
  * gst_rtsp_auth_setup_auth:
  * @auth: a #GstRTSPAuth
  * @client: the client
- * @uri: the requested uri
+ * @hint: TODO
- * @session: the session
+ * @state: TODO
- * @request: the request
- * @response: the response
  *
  * Add authentication tokens to @response.
  *

gst/rtsp-server/rtsp-client.c

@@ -1629,7 +1629,7 @@ gst_rtsp_client_set_session_pool (GstRTSPClient * client,
  *
  * 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)
@@ -1670,7 +1670,7 @@ gst_rtsp_client_set_server (GstRTSPClient * client, GstRTSPServer * server)
  *
  * 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)
@@ -1715,7 +1715,7 @@ gst_rtsp_client_set_media_mapping (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)
@@ -1789,7 +1789,7 @@ gst_rtsp_client_set_auth (GstRTSPClient * client, GstRTSPAuth * auth)
  *
  * 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 *

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

@@ -86,7 +86,7 @@ gst_rtsp_media_factory_uri_class_init (GstRTSPMediaFactoryURIClass * klass)
   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.
    */
@@ -95,7 +95,7 @@ gst_rtsp_media_factory_uri_class_init (GstRTSPMediaFactoryURIClass * klass)
           "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.

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

@@ -81,7 +81,7 @@ gst_rtsp_media_factory_class_init (GstRTSPMediaFactoryClass * klass)
   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.
@@ -510,7 +510,7 @@ gst_rtsp_media_factory_set_auth (GstRTSPMediaFactory * factory,
  *
  * 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 *
@@ -586,7 +586,7 @@ media_unprepared (GstRTSPMedia * media, GstRTSPMediaFactory * factory)
  * 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,

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

@@ -100,7 +100,7 @@ find_media (GstRTSPMediaMapping * mapping, const GstRTSPUrl * url)
  * 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,

gst/rtsp-server/rtsp-media.c

@@ -608,7 +608,7 @@ gst_rtsp_media_set_auth (GstRTSPMedia * media, GstRTSPAuth * auth)
  *
  * 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 *

gst/rtsp-server/rtsp-server.c

@@ -82,7 +82,7 @@ gst_rtsp_server_class_init (GstRTSPServerClass * klass)
   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.
@@ -92,7 +92,7 @@ gst_rtsp_server_class_init (GstRTSPServerClass * klass)
           "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.
@@ -102,7 +102,7 @@ gst_rtsp_server_class_init (GstRTSPServerClass * klass)
           "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
@@ -114,7 +114,7 @@ gst_rtsp_server_class_init (GstRTSPServerClass * klass)
           -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
@@ -128,7 +128,7 @@ gst_rtsp_server_class_init (GstRTSPServerClass * klass)
           "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
@@ -140,7 +140,7 @@ gst_rtsp_server_class_init (GstRTSPServerClass * klass)
           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.
@@ -397,7 +397,7 @@ gst_rtsp_server_set_session_pool (GstRTSPServer * server,
  *
  * 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 *
@@ -449,7 +449,7 @@ gst_rtsp_server_set_media_mapping (GstRTSPServer * server,
  *
  * 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 *
@@ -500,7 +500,7 @@ gst_rtsp_server_set_auth (GstRTSPServer * server, GstRTSPAuth * auth)
  *
  * 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 *
@@ -584,7 +584,7 @@ gst_rtsp_server_set_property (GObject * object, guint propid,
  * 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,
@@ -991,8 +991,7 @@ no_socket:
 /**
  * gst_rtsp_server_attach:
  * @server: a #GstRTSPServer
- * @context: a #GMainContext
+ * @context: (allow-none): a #GMainContext
- * @error: a #GError
  *
  * 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/rtsp-session-pool.c

@@ -214,7 +214,7 @@ gst_rtsp_session_pool_get_n_sessions (GstRTSPSessionPool * pool)
  * 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 *
@@ -257,7 +257,7 @@ create_session_id (GstRTSPSessionPool * pool)
  *
  * 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)
@@ -423,7 +423,7 @@ filter_func (gchar * sessionid, GstRTSPSession * sess, FilterData * data)
 /**
  * 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
@@ -439,9 +439,9 @@ filter_func (gchar * sessionid, GstRTSPSession * sess, FilterData * data)
  * 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
+ * Returns: (element-type GstRTSPSession) (transfer full): a GList with all
- * #GST_RTSP_FILTER_REF. After usage, each element in the GList should be unreffed
+ * sessions for which @func returned #GST_RTSP_FILTER_REF. After usage, each
- * before the list is freed.
+ * element in the GList should be unreffed before the list is freed.
  */
 GList *
 gst_rtsp_session_pool_filter (GstRTSPSessionPool * pool,

gst/rtsp-server/rtsp-session.c

@@ -182,7 +182,7 @@ gst_rtsp_session_set_property (GObject * object, guint propid,
  * 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().
@@ -551,10 +551,8 @@ gst_rtsp_session_stream_set_transport (GstRTSPSessionStream * stream,
 /**
  * gst_rtsp_session_stream_set_callbacks:
  * @stream: a #GstRTSPSessionStream
- * @send_rtp: a callback called when RTP should be sent
+ * @send_rtp: (scope notified): a callback called when RTP should be sent
- * @send_rtcp: a callback called when RTCP should be sent
+ * @send_rtcp: (scope notified): 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
  * @user_data: user data passed to callbacks
  * @notify: called with the user_data when no longer needed.
  *