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rtsp-server: fixed comments and GIR annotations
https://bugzilla.gnome.org/show_bug.cgi?id=680777
This commit is contained in:
Sebastian Pölsterl
Tim-Philipp Müller
parent
bc474a5b26
commit
e11e855ac8
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@ -152,10 +152,8 @@ default_setup_auth (GstRTSPAuth * auth, GstRTSPClient * client,
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* gst_rtsp_auth_setup_auth:
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* @auth: a #GstRTSPAuth
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* @client: the client
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* @uri: the requested uri
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* @session: the session
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* @request: the request
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* @response: the response
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* @hint: TODO
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* @state: TODO
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*
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* Add authentication tokens to @response.
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*
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@ -1629,7 +1629,7 @@ gst_rtsp_client_set_session_pool (GstRTSPClient * client,
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*
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* Get the #GstRTSPSessionPool object that @client uses to manage its sessions.
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*
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* Returns: a #GstRTSPSessionPool, unref after usage.
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* Returns: (transfer full): a #GstRTSPSessionPool, unref after usage.
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*/
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GstRTSPSessionPool *
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gst_rtsp_client_get_session_pool (GstRTSPClient * client)
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@ -1670,7 +1670,7 @@ gst_rtsp_client_set_server (GstRTSPClient * client, GstRTSPServer * server)
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*
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* Get the #GstRTSPServer object that @client was created from.
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*
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* Returns: a #GstRTSPServer, unref after usage.
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* Returns: (transfer full): a #GstRTSPServer, unref after usage.
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*/
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GstRTSPServer *
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gst_rtsp_client_get_server (GstRTSPClient * client)
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@ -1715,7 +1715,7 @@ gst_rtsp_client_set_media_mapping (GstRTSPClient * client,
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*
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* Get the #GstRTSPMediaMapping object that @client uses to manage its sessions.
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*
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* Returns: a #GstRTSPMediaMapping, unref after usage.
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* Returns: (transfer full): a #GstRTSPMediaMapping, unref after usage.
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*/
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GstRTSPMediaMapping *
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gst_rtsp_client_get_media_mapping (GstRTSPClient * client)
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@ -1789,7 +1789,7 @@ gst_rtsp_client_set_auth (GstRTSPClient * client, GstRTSPAuth * auth)
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*
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* Get the #GstRTSPAuth used as the authentication manager of @client.
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*
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* Returns: the #GstRTSPAuth of @client. g_object_unref() after
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* Returns: (transfer full): the #GstRTSPAuth of @client. g_object_unref() after
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* usage.
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*/
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GstRTSPAuth *
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@ -86,7 +86,7 @@ gst_rtsp_media_factory_uri_class_init (GstRTSPMediaFactoryURIClass * klass)
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gobject_class->finalize = gst_rtsp_media_factory_uri_finalize;
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/**
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* GstRTSPMediaFactoryURI::uri
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* GstRTSPMediaFactoryURI::uri:
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*
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* The uri of the resource that will be served by this factory.
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*/
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@ -95,7 +95,7 @@ gst_rtsp_media_factory_uri_class_init (GstRTSPMediaFactoryURIClass * klass)
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"The URI of the resource to stream", DEFAULT_URI,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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/**
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* GstRTSPMediaFactoryURI::use-gstpay
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* GstRTSPMediaFactoryURI::use-gstpay:
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*
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* Allow the usage of gstpay in order to avoid decoding of compressed formats
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* without a payloader.
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@ -81,7 +81,7 @@ gst_rtsp_media_factory_class_init (GstRTSPMediaFactoryClass * klass)
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gobject_class->finalize = gst_rtsp_media_factory_finalize;
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/**
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* GstRTSPMediaFactory::launch
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* GstRTSPMediaFactory::launch:
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*
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* The gst_parse_launch() line to use for constructing the pipeline in the
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* default prepare vmethod.
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@ -510,7 +510,7 @@ gst_rtsp_media_factory_set_auth (GstRTSPMediaFactory * factory,
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*
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* Get the #GstRTSPAuth used as the authentication manager of @factory.
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*
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* Returns: the #GstRTSPAuth of @factory. g_object_unref() after
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* Returns: (transfer full): the #GstRTSPAuth of @factory. g_object_unref() after
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* usage.
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*/
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GstRTSPAuth *
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@ -586,7 +586,7 @@ media_unprepared (GstRTSPMedia * media, GstRTSPMediaFactory * factory)
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* the srcpad member set to a source pad that produces buffer of type
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* application/x-rtp.
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*
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* Returns: a new #GstRTSPMedia if the media could be prepared.
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* Returns: (transfer full): a new #GstRTSPMedia if the media could be prepared.
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*/
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GstRTSPMedia *
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gst_rtsp_media_factory_construct (GstRTSPMediaFactory * factory,
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@ -100,7 +100,7 @@ find_media (GstRTSPMediaMapping * mapping, const GstRTSPUrl * url)
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* Find the #GstRTSPMediaFactory for @url. The default implementation of this object
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* will use the mappings added with gst_rtsp_media_mapping_add_factory ().
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*
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* Returns: the #GstRTSPMediaFactory for @url. g_object_unref() after usage.
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* Returns: (transfer full): the #GstRTSPMediaFactory for @url. g_object_unref() after usage.
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*/
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GstRTSPMediaFactory *
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gst_rtsp_media_mapping_find_factory (GstRTSPMediaMapping * mapping,
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@ -608,7 +608,7 @@ gst_rtsp_media_set_auth (GstRTSPMedia * media, GstRTSPAuth * auth)
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*
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* Get the #GstRTSPAuth used as the authentication manager of @media.
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*
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* Returns: the #GstRTSPAuth of @media. g_object_unref() after
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* Returns: (transfer full): the #GstRTSPAuth of @media. g_object_unref() after
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* usage.
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*/
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GstRTSPAuth *
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@ -82,7 +82,7 @@ gst_rtsp_server_class_init (GstRTSPServerClass * klass)
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gobject_class->finalize = gst_rtsp_server_finalize;
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/**
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* GstRTSPServer::address
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* GstRTSPServer::address:
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*
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* The address of the server. This is the address where the server will
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* listen on.
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@ -92,7 +92,7 @@ gst_rtsp_server_class_init (GstRTSPServerClass * klass)
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"The address the server uses to listen on", DEFAULT_ADDRESS,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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/**
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* GstRTSPServer::service
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* GstRTSPServer::service:
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*
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* The service of the server. This is either a string with the service name or
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* a port number (as a string) the server will listen on.
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@ -102,7 +102,7 @@ gst_rtsp_server_class_init (GstRTSPServerClass * klass)
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"The service or port number the server uses to listen on",
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DEFAULT_SERVICE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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/**
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* GstRTSPServer::bound-port
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* GstRTSPServer::bound-port:
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*
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* The actual port the server is listening on. Can be used to retrieve the
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* port number when the server is started on port 0, which means bind to a
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@ -114,7 +114,7 @@ gst_rtsp_server_class_init (GstRTSPServerClass * klass)
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-1, G_MAXUINT16, DEFAULT_BOUND_PORT,
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G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
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/**
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* GstRTSPServer::backlog
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* GstRTSPServer::backlog:
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*
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* The backlog argument defines the maximum length to which the queue of
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* pending connections for the server may grow. If a connection request arrives
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@ -128,7 +128,7 @@ gst_rtsp_server_class_init (GstRTSPServerClass * klass)
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"of pending connections may grow", 0, G_MAXINT, DEFAULT_BACKLOG,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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/**
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* GstRTSPServer::session-pool
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* GstRTSPServer::session-pool:
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*
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* The session pool of the server. By default each server has a separate
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* session pool but sessions can be shared between servers by setting the same
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@ -140,7 +140,7 @@ gst_rtsp_server_class_init (GstRTSPServerClass * klass)
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GST_TYPE_RTSP_SESSION_POOL,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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/**
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* GstRTSPServer::media-mapping
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* GstRTSPServer::media-mapping:
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*
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* The media mapping to use for this server. By default the server has no
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* media mapping and thus cannot map urls to media streams.
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@ -397,7 +397,7 @@ gst_rtsp_server_set_session_pool (GstRTSPServer * server,
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*
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* Get the #GstRTSPSessionPool used as the session pool of @server.
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*
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* Returns: the #GstRTSPSessionPool used for sessions. g_object_unref() after
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* Returns: (transfer full): the #GstRTSPSessionPool used for sessions. g_object_unref() after
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* usage.
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*/
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GstRTSPSessionPool *
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@ -449,7 +449,7 @@ gst_rtsp_server_set_media_mapping (GstRTSPServer * server,
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*
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* Get the #GstRTSPMediaMapping used as the media mapping of @server.
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*
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* Returns: the #GstRTSPMediaMapping of @server. g_object_unref() after
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* Returns: (transfer full): the #GstRTSPMediaMapping of @server. g_object_unref() after
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* usage.
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*/
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GstRTSPMediaMapping *
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@ -500,7 +500,7 @@ gst_rtsp_server_set_auth (GstRTSPServer * server, GstRTSPAuth * auth)
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*
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* Get the #GstRTSPAuth used as the authentication manager of @server.
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*
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* Returns: the #GstRTSPAuth of @server. g_object_unref() after
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* Returns: (transfer full): the #GstRTSPAuth of @server. g_object_unref() after
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* usage.
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*/
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GstRTSPAuth *
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@ -584,7 +584,7 @@ gst_rtsp_server_set_property (GObject * object, guint propid,
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* Create a #GSocket for @server. The socket will listen on the
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* configured service.
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*
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* Returns: the #GSocket for @server or NULL when an error occured.
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* Returns: (transfer full): the #GSocket for @server or NULL when an error occured.
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*/
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GSocket *
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gst_rtsp_server_create_socket (GstRTSPServer * server,
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@ -991,8 +991,7 @@ no_socket:
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/**
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* gst_rtsp_server_attach:
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* @server: a #GstRTSPServer
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* @context: a #GMainContext
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* @error: a #GError
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* @context: (allow-none): a #GMainContext
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*
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* Attaches @server to @context. When the mainloop for @context is run, the
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* server will be dispatched. When @context is NULL, the default context will be
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@ -214,7 +214,7 @@ gst_rtsp_session_pool_get_n_sessions (GstRTSPSessionPool * pool)
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* Find the session with @sessionid in @pool. The access time of the session
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* will be updated with gst_rtsp_session_touch().
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*
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* Returns: the #GstRTSPSession with @sessionid or %NULL when the session did
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* Returns: (transfer full): the #GstRTSPSession with @sessionid or %NULL when the session did
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* not exist. g_object_unref() after usage.
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*/
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GstRTSPSession *
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@ -257,7 +257,7 @@ create_session_id (GstRTSPSessionPool * pool)
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*
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* Create a new #GstRTSPSession object in @pool.
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*
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* Returns: a new #GstRTSPSession.
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* Returns: (transfer none): a new #GstRTSPSession.
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*/
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GstRTSPSession *
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gst_rtsp_session_pool_create (GstRTSPSessionPool * pool)
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@ -423,7 +423,7 @@ filter_func (gchar * sessionid, GstRTSPSession * sess, FilterData * data)
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/**
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* gst_rtsp_session_pool_filter:
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* @pool: a #GstRTSPSessionPool
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* @func: a callback
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* @func: (scope call): a callback
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* @user_data: user data passed to @func
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*
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* Call @func for each session in @pool. The result value of @func determines
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@ -439,9 +439,9 @@ filter_func (gchar * sessionid, GstRTSPSession * sess, FilterData * data)
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* will also be added with an additional ref to the result GList of this
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* function..
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*
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* Returns: a GList with all sessions for which @func returned
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* #GST_RTSP_FILTER_REF. After usage, each element in the GList should be unreffed
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* before the list is freed.
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* Returns: (element-type GstRTSPSession) (transfer full): a GList with all
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* sessions for which @func returned #GST_RTSP_FILTER_REF. After usage, each
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* element in the GList should be unreffed before the list is freed.
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*/
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GList *
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gst_rtsp_session_pool_filter (GstRTSPSessionPool * pool,
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@ -182,7 +182,7 @@ gst_rtsp_session_set_property (GObject * object, guint propid,
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* gst_rtsp_session_manage_media:
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* @sess: a #GstRTSPSession
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* @uri: the uri for the media
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* @media: a #GstRTSPMedia
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* @media: (transfer full): a #GstRTSPMedia
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*
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* Manage the media object @obj in @sess. @uri will be used to retrieve this
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* media from the session with gst_rtsp_session_get_media().
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@ -551,10 +551,8 @@ gst_rtsp_session_stream_set_transport (GstRTSPSessionStream * stream,
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/**
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* gst_rtsp_session_stream_set_callbacks:
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* @stream: a #GstRTSPSessionStream
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* @send_rtp: a callback called when RTP should be sent
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* @send_rtcp: a callback called when RTCP should be sent
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* @send_rtp_list: a callback called when RTP should be sent
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* @send_rtcp_list: a callback called when RTCP should be sent
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* @send_rtp: (scope notified): a callback called when RTP should be sent
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* @send_rtcp: (scope notified): a callback called when RTCP should be sent
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* @user_data: user data passed to callbacks
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* @notify: called with the user_data when no longer needed.
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*
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