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rtsp-server: fixed comments and GIR annotations

Browse source 
https://bugzilla.gnome.org/show_bug.cgi?id=680777
This commit is contained in:
Sebastian Pölsterl 2012-10-01 19:46:15 +02:00 committed by Tim-Philipp Müller
parent bc474a5b26
commit e11e855ac8
9 changed files with 33 additions and 38 deletions
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6
gst/rtsp-server/rtsp-auth.c
View file

@ -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
* @session: the session
* @request: the request
* @response: the response
* @hint: TODO
* @state: TODO
*
* Add authentication tokens to @response.
*

8
gst/rtsp-server/rtsp-client.c
View file

@ -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 *

4
gst/rtsp-server/rtsp-media-factory-uri.c
View file

@ -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.

6
gst/rtsp-server/rtsp-media-factory.c
View file

@ -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,

2
gst/rtsp-server/rtsp-media-mapping.c
View file

@ -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,

2
gst/rtsp-server/rtsp-media.c
View file

@ -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 *

23
gst/rtsp-server/rtsp-server.c
View file

@ -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
* @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

12
gst/rtsp-server/rtsp-session-pool.c
View file

@ -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
* #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,

8
gst/rtsp-server/rtsp-session.c
View file

@ -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_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.
*