Commit b1b53015 authored by Sebastian Rasmussen's avatar Sebastian Rasmussen Committed by Tim-Philipp Müller

gobject-introspection: Add annotations to support language bindings

In addition a few cosmetic changes:

 * Adjust the order of arguments
 * Fix typo: occured -> occurred
 * Fix indentation after Return:-clauses

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726941
parent 0b617dd5
......@@ -181,7 +181,7 @@ gst_rtsp_address_pool_finalize (GObject * obj)
*
* Make a new #GstRTSPAddressPool.
*
* Returns: a new #GstRTSPAddressPool
* Returns: (transfer full): a new #GstRTSPAddressPool
*/
GstRTSPAddressPool *
gst_rtsp_address_pool_new (void)
......@@ -424,7 +424,7 @@ split_range (GstRTSPAddressPool * pool, AddrRange * range, guint skip_addr,
* one can be found in @port.
*
* Returns: a #GstRTSPAddress that should be freed with gst_rtsp_address_free
* after use or %NULL when no address could be acquired.
* after use or %NULL when no address could be acquired.
*/
GstRTSPAddress *
gst_rtsp_address_pool_acquire_address (GstRTSPAddressPool * pool,
......
......@@ -161,7 +161,7 @@ gst_rtsp_auth_set_property (GObject * object, guint propid,
*
* Create a new #GstRTSPAuth instance.
*
* Returns: a new #GstRTSPAuth
* Returns: (transfer full): a new #GstRTSPAuth
*/
GstRTSPAuth *
gst_rtsp_auth_new (void)
......@@ -176,7 +176,7 @@ gst_rtsp_auth_new (void)
/**
* gst_rtsp_auth_set_tls_certificate:
* @auth: a #GstRTSPAuth
* @cert: (allow-none): a #GTlsCertificate
* @cert: (transfer none) (allow-none): a #GTlsCertificate
*
* Set the TLS certificate for the auth. Client connections will only
* be accepted when TLS is negotiated.
......@@ -233,7 +233,7 @@ gst_rtsp_auth_get_tls_certificate (GstRTSPAuth * auth)
/**
* gst_rtsp_auth_set_default_token:
* @auth: a #GstRTSPAuth
* @token: (allow-none): a #GstRTSPToken
* @token: (transfer none) (allow-none): a #GstRTSPToken
*
* Set the default #GstRTSPToken to @token in @auth. The default token will
* be used for unauthenticated users.
......@@ -292,7 +292,7 @@ gst_rtsp_auth_get_default_token (GstRTSPAuth * auth)
* gst_rtsp_auth_add_basic:
* @auth: a #GstRTSPAuth
* @basic: the basic token
* @token: authorisation token
* @token: (transfer none): authorisation token
*
* Add a basic token for the default authentication algorithm that
* enables the client with privileges listed in @token.
......@@ -616,8 +616,8 @@ no_context:
*
* Construct a Basic authorisation token from @user and @pass.
*
* Returns: the base64 encoding of the string @user:@pass. g_free()
* after usage.
* Returns: (transfer full): the base64 encoding of the string @user:@pass.
* g_free() after usage.
*/
gchar *
gst_rtsp_auth_make_basic (const gchar * user, const gchar * pass)
......
......@@ -412,7 +412,7 @@ gst_rtsp_client_set_property (GObject * object, guint propid,
*
* Create a new #GstRTSPClient instance.
*
* Returns: a new #GstRTSPClient
* Returns: (transfer full): a new #GstRTSPClient
*/
GstRTSPClient *
gst_rtsp_client_new (void)
......@@ -2199,7 +2199,7 @@ handle_data (GstRTSPClient * client, GstRTSPMessage * message)
/**
* gst_rtsp_client_set_session_pool:
* @client: a #GstRTSPClient
* @pool: a #GstRTSPSessionPool
* @pool: (transfer none): a #GstRTSPSessionPool
*
* Set @pool as the sessionpool for @client which it will use to find
* or allocate sessions. the sessionpool is usually inherited from the server
......@@ -2257,7 +2257,7 @@ gst_rtsp_client_get_session_pool (GstRTSPClient * client)
/**
* gst_rtsp_client_set_mount_points:
* @client: a #GstRTSPClient
* @mounts: a #GstRTSPMountPoints
* @mounts: (transfer none): a #GstRTSPMountPoints
*
* Set @mounts as the mount points for @client which it will use to map urls
* to media streams. These mount points are usually inherited from the server that
......@@ -2315,7 +2315,7 @@ gst_rtsp_client_get_mount_points (GstRTSPClient * client)
/**
* gst_rtsp_client_set_auth:
* @client: a #GstRTSPClient
* @auth: a #GstRTSPAuth
* @auth: (transfer none): a #GstRTSPAuth
*
* configure @auth to be used as the authentication manager of @client.
*/
......@@ -2372,7 +2372,7 @@ gst_rtsp_client_get_auth (GstRTSPClient * client)
/**
* gst_rtsp_client_set_thread_pool:
* @client: a #GstRTSPClient
* @pool: a #GstRTSPThreadPool
* @pool: (transfer none): a #GstRTSPThreadPool
*
* configure @pool to be used as the thread pool of @client.
*/
......@@ -2511,9 +2511,9 @@ gst_rtsp_client_get_connection (GstRTSPClient * client)
/**
* gst_rtsp_client_set_send_func:
* @client: a #GstRTSPClient
* @func: a #GstRTSPClientSendFunc
* @user_data: user data passed to @func
* @notify: called when @user_data is no longer in use
* @func: (scope notified): a #GstRTSPClientSendFunc
* @user_data: (closure): user data passed to @func
* @notify: (allow-none): called when @user_data is no longer in use
*
* Set @func as the callback that will be called when a new message needs to be
* sent to the client. @user_data is passed to @func and @notify is called when
......@@ -2549,7 +2549,7 @@ gst_rtsp_client_set_send_func (GstRTSPClient * client,
/**
* gst_rtsp_client_handle_message:
* @client: a #GstRTSPClient
* @message: an #GstRTSPMessage
* @message: (transfer none): an #GstRTSPMessage
*
* Let the client handle @message.
*
......@@ -2581,8 +2581,8 @@ gst_rtsp_client_handle_message (GstRTSPClient * client,
/**
* gst_rtsp_client_send_message:
* @client: a #GstRTSPClient
* @session: a #GstRTSPSession to send the message to or %NULL
* @message: The #GstRTSPMessage to send
* @session: (transfer none): a #GstRTSPSession to send the message to or %NULL
* @message: (transfer none): The #GstRTSPMessage to send
*
* Send a message message to the remote end. @message must be a
* #GST_RTSP_MESSAGE_REQUEST or a #GST_RTSP_MESSAGE_RESPONSE.
......
......@@ -276,7 +276,7 @@ gst_rtsp_media_factory_uri_set_property (GObject * object, guint propid,
*
* Create a new #GstRTSPMediaFactoryURI instance.
*
* Returns: a new #GstRTSPMediaFactoryURI object.
* Returns: (transfer full): a new #GstRTSPMediaFactoryURI object.
*/
GstRTSPMediaFactoryURI *
gst_rtsp_media_factory_uri_new (void)
......@@ -318,7 +318,7 @@ gst_rtsp_media_factory_uri_set_uri (GstRTSPMediaFactoryURI * factory,
*
* Get the URI that will provide media for this factory.
*
* Returns: the configured URI. g_free() after usage.
* Returns: (transfer full): the configured URI. g_free() after usage.
*/
gchar *
gst_rtsp_media_factory_uri_get_uri (GstRTSPMediaFactoryURI * factory)
......
......@@ -317,7 +317,7 @@ gst_rtsp_media_factory_set_property (GObject * object, guint propid,
*
* Create a new #GstRTSPMediaFactory instance.
*
* Returns: a new #GstRTSPMediaFactory object.
* Returns: (transfer full): a new #GstRTSPMediaFactory object.
*/
GstRTSPMediaFactory *
gst_rtsp_media_factory_new (void)
......@@ -332,7 +332,7 @@ gst_rtsp_media_factory_new (void)
/**
* gst_rtsp_media_factory_set_permissions:
* @factory: a #GstRTSPMediaFactory
* @permissions: a #GstRTSPPermissions
* @permissions: (transfer none): a #GstRTSPPermissions
*
* Set @permissions on @factory.
*/
......@@ -455,7 +455,8 @@ gst_rtsp_media_factory_set_launch (GstRTSPMediaFactory * factory,
* Get the gst_parse_launch() pipeline description that will be used in the
* default prepare vmethod.
*
* Returns: the configured launch description. g_free() after usage.
* Returns: (transfer full): the configured launch description. g_free() after
* usage.
*/
gchar *
gst_rtsp_media_factory_get_launch (GstRTSPMediaFactory * factory)
......@@ -668,7 +669,7 @@ gst_rtsp_media_factory_get_buffer_size (GstRTSPMediaFactory * factory)
/**
* gst_rtsp_media_factory_set_address_pool:
* @factory: a #GstRTSPMediaFactory
* @pool: a #GstRTSPAddressPool
* @pool: (transfer none): a #GstRTSPAddressPool
*
* configure @pool to be used as the address pool of @factory.
*/
......
......@@ -563,7 +563,7 @@ collect_media_stats (GstRTSPMedia * media)
*
* Ownership is taken of @element.
*
* Returns: a new #GstRTSPMedia object.
* Returns: (transfer full): a new #GstRTSPMedia object.
*/
GstRTSPMedia *
gst_rtsp_media_new (GstElement * element)
......@@ -632,7 +632,7 @@ gst_rtsp_media_take_pipeline (GstRTSPMedia * media, GstPipeline * pipeline)
/**
* gst_rtsp_media_set_permissions:
* @media: a #GstRTSPMedia
* @permissions: a #GstRTSPPermissions
* @permissions: (transfer none): a #GstRTSPPermissions
*
* Set @permissions on @media.
*/
......@@ -1089,7 +1089,7 @@ gst_rtsp_media_is_time_provider (GstRTSPMedia * media)
/**
* gst_rtsp_media_set_address_pool:
* @media: a #GstRTSPMedia
* @pool: a #GstRTSPAddressPool
* @pool: (transfer none): a #GstRTSPAddressPool
*
* configure @pool to be used as the address pool of @media.
*/
......@@ -1217,7 +1217,7 @@ gst_rtsp_media_collect_streams (GstRTSPMedia * media)
* @srcpad should be a pad of an element inside @media->element.
*
* Returns: (transfer none): a new #GstRTSPStream that remains valid for as long
* as @media exists.
* as @media exists.
*/
GstRTSPStream *
gst_rtsp_media_create_stream (GstRTSPMedia * media, GstElement * payloader,
......@@ -1397,7 +1397,7 @@ default_convert_range (GstRTSPMedia * media, GstRTSPTimeRange * range,
* Get the current range as a string. @media must be prepared with
* gst_rtsp_media_prepare ().
*
* Returns: The range as a string, g_free() after usage.
* Returns: (transfer full): The range as a string, g_free() after usage.
*/
gchar *
gst_rtsp_media_get_range_string (GstRTSPMedia * media, gboolean play,
......@@ -1474,7 +1474,7 @@ media_streams_set_blocked (GstRTSPMedia * media, gboolean blocked)
/**
* gst_rtsp_media_seek:
* @media: a #GstRTSPMedia
* @range: a #GstRTSPTimeRange
* @range: (transfer none): a #GstRTSPTimeRange
*
* Seek the pipeline of @media to @range. @media must be prepared with
* gst_rtsp_media_prepare().
......@@ -2116,7 +2116,7 @@ preroll_failed:
/**
* gst_rtsp_media_prepare:
* @media: a #GstRTSPMedia
* @thread: a #GstRTSPThread to run the bus handler or %NULL
* @thread: (transfer full): a #GstRTSPThread to run the bus handler or %NULL
*
* Prepare @media for streaming. This function will create the objects
* to manage the streaming. A pipeline must have been set on @media with
......@@ -2541,9 +2541,9 @@ default_setup_sdp (GstRTSPMedia * media, GstSDPMessage * sdp, GstSDPInfo * info)
/**
* gst_rtsp_media_setup_sdp:
* @sdp: a #GstSDPMessage
* @info: info
* @media: a #GstRTSPMedia
* @sdp: (transfer none): a #GstSDPMessage
* @info: (transfer none): a #GstSDPInfo
*
* Add @media specific info to @sdp. @info is used to configure the connection
* information in the SDP.
......@@ -2772,8 +2772,8 @@ gst_rtsp_media_set_pipeline_state (GstRTSPMedia * media, GstState state)
* gst_rtsp_media_set_state:
* @media: a #GstRTSPMedia
* @state: the target state of the media
* @transports: (element-type GstRtspServer.RTSPStreamTransport): a #GPtrArray
* of #GstRTSPStreamTransport pointers
* @transports: (transfer none) (element-type GstRtspServer.RTSPStreamTransport):
* a #GPtrArray of #GstRTSPStreamTransport pointers
*
* Set the state of @media to @state and for the transports in @transports.
*
......
......@@ -156,7 +156,7 @@ gst_rtsp_mount_points_finalize (GObject * obj)
*
* Make a new mount points object.
*
* Returns: a new #GstRTSPMountPoints
* Returns: (transfer full): a new #GstRTSPMountPoints
*/
GstRTSPMountPoints *
gst_rtsp_mount_points_new (void)
......@@ -181,7 +181,7 @@ default_make_path (GstRTSPMountPoints * mounts, const GstRTSPUrl * url)
*
* Make a path string from @url.
*
* Returns: a path string for @url, g_free() after usage.
* Returns: (transfer full): a path string for @url, g_free() after usage.
*/
gchar *
gst_rtsp_mount_points_make_path (GstRTSPMountPoints * mounts,
......@@ -229,7 +229,7 @@ has_prefix (DataItem * str, DataItem * prefix)
* the amount of characters that matched is returned in @matched.
*
* Returns: (transfer full): the #GstRTSPMediaFactory for @path.
* g_object_unref() after usage.
* g_object_unref() after usage.
*/
GstRTSPMediaFactory *
gst_rtsp_mount_points_match (GstRTSPMountPoints * mounts,
......
......@@ -31,7 +31,7 @@
/**
* gst_rtsp_params_set:
* @client: a #GstRTSPClient
* @ctx: a #GstRTSPContext
* @ctx: (transfer none): a #GstRTSPContext
*
* Set parameters (not implemented yet)
*
......@@ -55,7 +55,7 @@ gst_rtsp_params_set (GstRTSPClient * client, GstRTSPContext * ctx)
/**
* gst_rtsp_params_get:
* @client: a #GstRTSPClient
* @ctx: a #GstRTSPContext
* @ctx: (transfer none): a #GstRTSPContext
*
* Get parameters (not implemented yet)
*
......
......@@ -221,7 +221,8 @@ gst_rtsp_permissions_remove_role (GstRTSPPermissions * permissions,
*
* Get all permissions for @role in @permissions.
*
* Returns: the structure with permissions for @role.
* Returns: (transfer none): the structure with permissions for @role. It
* remains valid for as long as @permissions is valid.
*/
const GstStructure *
gst_rtsp_permissions_get_role (GstRTSPPermissions * permissions,
......
......@@ -245,8 +245,8 @@ no_multicast:
/**
* gst_rtsp_sdp_from_media:
* @sdp: a #GstSDPMessage
* @info: info
* @media: a #GstRTSPMedia
* @info: (transfer none): a #GstSDPInfo
* @media: (transfer none): a #GstRTSPMedia
*
* Add @media specific info to @sdp. @info is used to configure the connection
* information in the SDP.
......
......@@ -283,6 +283,8 @@ gst_rtsp_server_finalize (GObject * object)
* gst_rtsp_server_new:
*
* Create a new #GstRTSPServer instance.
*
* Returns: (transfer full): a new #GstRTSPServer
*/
GstRTSPServer *
gst_rtsp_server_new (void)
......@@ -325,7 +327,7 @@ gst_rtsp_server_set_address (GstRTSPServer * server, const gchar * address)
*
* Get the address on which the server will accept connections.
*
* Returns: the server address. g_free() after usage.
* Returns: (transfer full): the server address. g_free() after usage.
*/
gchar *
gst_rtsp_server_get_address (GstRTSPServer * server)
......@@ -414,7 +416,7 @@ gst_rtsp_server_set_service (GstRTSPServer * server, const gchar * service)
*
* Get the service on which the server will accept connections.
*
* Returns: 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)
......@@ -485,7 +487,7 @@ gst_rtsp_server_get_backlog (GstRTSPServer * server)
/**
* gst_rtsp_server_set_session_pool:
* @server: a #GstRTSPServer
* @pool: a #GstRTSPSessionPool
* @pool: (transfer none): a #GstRTSPSessionPool
*
* configure @pool to be used as the session pool of @server.
*/
......@@ -542,7 +544,7 @@ gst_rtsp_server_get_session_pool (GstRTSPServer * server)
/**
* gst_rtsp_server_set_mount_points:
* @server: a #GstRTSPServer
* @mounts: a #GstRTSPMountPoints
* @mounts: (transfer none): a #GstRTSPMountPoints
*
* configure @mounts to be used as the mount points of @server.
*/
......@@ -600,7 +602,7 @@ gst_rtsp_server_get_mount_points (GstRTSPServer * server)
/**
* gst_rtsp_server_set_auth:
* @server: a #GstRTSPServer
* @auth: a #GstRTSPAuth
* @auth: (transfer none): a #GstRTSPAuth
*
* configure @auth to be used as the authentication manager of @server.
*/
......@@ -657,7 +659,7 @@ gst_rtsp_server_get_auth (GstRTSPServer * server)
/**
* gst_rtsp_server_set_thread_pool:
* @server: a #GstRTSPServer
* @pool: a #GstRTSPThreadPool
* @pool: (transfer none): a #GstRTSPThreadPool
*
* configure @pool to be used as the thread pool of @server.
*/
......@@ -771,13 +773,14 @@ gst_rtsp_server_set_property (GObject * object, guint propid,
/**
* gst_rtsp_server_create_socket:
* @server: a #GstRTSPServer
* @cancellable: a #GCancellable
* @error: a #GError
* @cancellable: (allow-none): a #GCancellable
* @error: (out): a #GError
*
* Create a #GSocket for @server. The socket will listen on the
* configured service.
*
* Returns: (transfer full): the #GSocket for @server or %NULL when an error occured.
* Returns: (transfer full): the #GSocket for @server or %NULL when an error
* occurred.
*/
GSocket *
gst_rtsp_server_create_socket (GstRTSPServer * server,
......@@ -1089,7 +1092,7 @@ default_create_client (GstRTSPServer * server)
* as an RTSP over HTTP tunnel. The @initial_buffer contains any remaining data
* that the HTTP server read from the socket while parsing the HTTP header.
*
* Returns: TRUE if all was ok, FALSE if an error occured.
* Returns: TRUE if all was ok, FALSE if an error occurred.
*/
gboolean
gst_rtsp_server_transfer_connection (GstRTSPServer * server, GSocket * socket,
......@@ -1140,12 +1143,12 @@ no_connection:
* gst_rtsp_server_io_func:
* @socket: a #GSocket
* @condition: the condition on @source
* @server: a #GstRTSPServer
* @server: (transfer none): a #GstRTSPServer
*
* A default #GSocketSourceFunc that creates a new #GstRTSPClient to accept and handle a
* new connection on @socket or @server.
*
* Returns: TRUE if the source could be connected, FALSE if an error occured.
* Returns: TRUE if the source could be connected, FALSE if an error occurred.
*/
gboolean
gst_rtsp_server_io_func (GSocket * socket, GIOCondition condition,
......@@ -1229,8 +1232,8 @@ watch_destroyed (GstRTSPServer * server)
/**
* gst_rtsp_server_create_source:
* @server: a #GstRTSPServer
* @cancellable: a #GCancellable or %NULL.
* @error: a #GError
* @cancellable: (allow-none): a #GCancellable or %NULL.
* @error: (out): a #GError
*
* Create a #GSource for @server. The new source will have a default
* #GSocketSourceFunc of gst_rtsp_server_io_func().
......@@ -1240,8 +1243,8 @@ watch_destroyed (GstRTSPServer * server)
* unless cancellation happened at the same time as a condition change). You can
* check for this in the callback using g_cancellable_is_cancelled().
*
* Returns: the #GSource for @server or %NULL when an error occured. Free with
* g_source_unref ()
* Returns: (transfer full): the #GSource for @server or %NULL when an error
* occurred. Free with g_source_unref ()
*/
GSource *
gst_rtsp_server_create_source (GstRTSPServer * server,
......
......@@ -123,14 +123,14 @@ free_session_media (gpointer data)
/**
* gst_rtsp_session_media_new:
* @path: the path
* @media: the #GstRTSPMedia
* @media: (transfer full): the #GstRTSPMedia
*
* Create a new #GstRTSPSessionMedia that manages the streams
* in @media for @path. @media should be prepared.
*
* Ownership is taken of @media.
*
* Returns: a new #GstRTSPSessionMedia.
* Returns: (transfer full): a new #GstRTSPSessionMedia.
*/
GstRTSPSessionMedia *
gst_rtsp_session_media_new (const gchar * path, GstRTSPMedia * media)
......@@ -206,7 +206,7 @@ gst_rtsp_session_media_matches (GstRTSPSessionMedia * media,
* Get the #GstRTSPMedia that was used when constructing @media
*
* Returns: (transfer none): the #GstRTSPMedia of @media. Remains valid as long
* as @media is valid.
* as @media is valid.
*/
GstRTSPMedia *
gst_rtsp_session_media_get_media (GstRTSPSessionMedia * media)
......@@ -240,7 +240,7 @@ gst_rtsp_session_media_get_base_time (GstRTSPSessionMedia * media)
* with configured transports.
*
* Returns: (transfer full): The RTP-Info as a string or %NULL when
* no RTP-Info could be generated, g_free() after usage.
* no RTP-Info could be generated, g_free() after usage.
*/
gchar *
gst_rtsp_session_media_get_rtpinfo (GstRTSPSessionMedia * media)
......@@ -340,7 +340,7 @@ not_prepared:
* gst_rtsp_session_media_set_transport:
* @media: a #GstRTSPSessionMedia
* @stream: a #GstRTSPStream
* @tr: a #GstRTSPTransport
* @tr: (transfer full): a #GstRTSPTransport
*
* Configure the transport for @stream to @tr in @media.
*
......@@ -405,7 +405,7 @@ gst_rtsp_session_media_get_transport (GstRTSPSessionMedia * media, guint idx)
/**
* gst_rtsp_session_media_alloc_channels:
* @media: a #GstRTSPSessionMedia
* @range: a #GstRTSPRange
* @range: (out): a #GstRTSPRange
*
* Fill @range with the next available min and max channels for
* interleaved transport.
......
......@@ -172,7 +172,8 @@ gst_rtsp_session_pool_set_property (GObject * object, guint propid,
*
* Create a new #GstRTSPSessionPool instance.
*
* Returns: A new #GstRTSPSessionPool. g_object_unref() after usage.
* Returns: (transfer full): A new #GstRTSPSessionPool. g_object_unref() after
* usage.
*/
GstRTSPSessionPool *
gst_rtsp_session_pool_new (void)
......@@ -409,7 +410,7 @@ too_many_sessions:
/**
* gst_rtsp_session_pool_remove:
* @pool: a #GstRTSPSessionPool
* @sess: a #GstRTSPSession
* @sess: (transfer none): a #GstRTSPSession
*
* Remove @sess from @pool, releasing the ref that the pool has on @sess.
*
......@@ -507,7 +508,7 @@ filter_func (gchar * sessionid, GstRTSPSession * sess, FilterData * data)
* gst_rtsp_session_pool_filter:
* @pool: a #GstRTSPSessionPool
* @func: (scope call) (allow-none): a callback
* @user_data: user data passed to @func
* @user_data: (closure): user data passed to @func
*
* Call @func for each session in @pool. The result value of @func determines
* what happens to the session. @func will be called with the session pool
......@@ -645,7 +646,10 @@ static GSourceFuncs gst_pool_source_funcs = {
* gst_rtsp_session_pool_create_watch:
* @pool: a #GstRTSPSessionPool
*
* A GSource that will be dispatched when the session should be cleaned up.
* Create a #GSource that will be dispatched when the session should be cleaned
* up.
*
* Returns: (transfer full): a #GSource
*/
GSource *
gst_rtsp_session_pool_create_watch (GstRTSPSessionPool * pool)
......
......@@ -230,7 +230,7 @@ gst_rtsp_session_manage_media (GstRTSPSession * sess, const gchar * path,
/**
* gst_rtsp_session_release_media:
* @sess: a #GstRTSPSession
* @media: a #GstRTSPMedia
* @media: (transfer none): a #GstRTSPMedia
*
* Release the managed @media in @sess, freeing the memory allocated by it.
*
......@@ -314,7 +314,7 @@ gst_rtsp_session_get_media (GstRTSPSession * sess, const gchar * path,
* gst_rtsp_session_filter:
* @sess: a #GstRTSPSession
* @func: (scope call) (allow-none): a callback
* @user_data: user data passed to @func
* @user_data: (closure): user data passed to @func
*
* Call @func for each media in @sess. The result value of @func determines
* what happens to the media. @func will be called with @sess
......@@ -383,6 +383,8 @@ gst_rtsp_session_filter (GstRTSPSession * sess,
* @sessionid: a session id
*
* Create a new #GstRTSPSession instance with @sessionid.
*
* Returns: (transfer full): a new #GstRTSPSession
*/
GstRTSPSession *
gst_rtsp_session_new (const gchar * sessionid)
......@@ -402,8 +404,8 @@ gst_rtsp_session_new (const gchar * sessionid)
*
* Get the sessionid of @session.
*
* Returns: the sessionid of @session. The value remains valid as long as
* @session is alive.
* Returns: (transfer none): the sessionid of @session. The value remains valid
* as long as @session is alive.
*/
const gchar *
gst_rtsp_session_get_sessionid (GstRTSPSession * session)
......@@ -538,7 +540,7 @@ gst_rtsp_session_allow_expire (GstRTSPSession * session)
/**
* gst_rtsp_session_next_timeout:
* @session: a #GstRTSPSession
* @now: the current system time
* @now: (transfer none): the current system time
*
* Get the amount of milliseconds till the session will expire.
*
......@@ -580,7 +582,7 @@ gst_rtsp_session_next_timeout (GstRTSPSession * session, GTimeVal * now)
/**
* gst_rtsp_session_is_expired:
* @session: a #GstRTSPSession
* @now: the current system time
* @now: (transfer none): the current system time
*
* Check if @session timeout out.
*
......
......@@ -135,7 +135,7 @@ gst_rtsp_stream_transport_finalize (GObject * obj)
* Create a new #GstRTSPStreamTransport that can be used to manage
* @stream with transport @tr.
*
* Returns: a new #GstRTSPStreamTransport
* Returns: (transfer full): a new #GstRTSPStreamTransport
*/
GstRTSPStreamTransport *
gst_rtsp_stream_transport_new (GstRTSPStream * stream, GstRTSPTransport * tr)
......@@ -175,8 +175,8 @@ gst_rtsp_stream_transport_get_stream (GstRTSPStreamTransport * trans)
* @trans: a #GstRTSPStreamTransport
* @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.
* @user_data: (closure): user data passed to callbacks
* @notify: (allow-none): called with the user_data when no longer needed.
*
* Install callbacks that will be called when data for a stream should be sent
* to a client. This is usually used when sending RTP/RTCP over TCP.
......@@ -203,9 +203,9 @@ gst_rtsp_stream_transport_set_callbacks (GstRTSPStreamTransport * trans,
/**
* gst_rtsp_stream_transport_set_keepalive:
* @trans: a #GstRTSPStreamTransport
* @keep_alive: a callback called when the receiver is active
* @user_data: user data passed to callback
* @notify: called with the user_data when no longer needed.
* @keep_alive: (scope notified): a callback called when the receiver is active
* @user_data: (closure): user data passed to callback
* @notify: (allow-none): called with the user_data when no longer needed.
*
* Install callbacks that will be called when RTCP packets are received from the
* receiver of @trans.
......@@ -260,7 +260,7 @@ gst_rtsp_stream_transport_set_transport (GstRTSPStreamTransport * trans,
* Get the transport configured in @trans.
*
* Returns: (transfer none): the transport configured in @trans. It remains
* valid for as long as @trans is valid.
* valid for as long as @trans is valid.
*/
const GstRTSPTransport *
gst_rtsp_stream_transport_get_transport (GstRTSPStreamTransport * trans)
......@@ -300,7 +300,7 @@ gst_rtsp_stream_transport_set_url (GstRTSPStreamTransport * trans,
* Get the url configured in @trans.
*
* Returns: (transfer none): the url configured in @trans. It remains
* valid for as long as @trans is valid.
* valid for as long as @trans is valid.
*/
const GstRTSPUrl *
gst_rtsp_stream_transport_get_url (GstRTSPStreamTransport * trans)
......@@ -317,8 +317,8 @@ gst_rtsp_stream_transport_get_url (GstRTSPStreamTransport * trans)
*
* Get the RTP-Info string for @trans and @start_time.
*
* Returns: the RTPInfo string for @trans and @start_time or %NULL when
* the RTP-Info could not be determined. g_free() after usage.
* Returns: (transfer full): the RTPInfo string for @trans and @start_time or
* %NULL when the RTP-Info could not be determined. g_free() after usage.
*/
gchar *
gst_rtsp_stream_transport_get_rtpinfo (GstRTSPStreamTransport * trans,
......@@ -435,7 +435,7 @@ gst_rtsp_stream_transport_is_timed_out (GstRTSPStreamTransport * trans)
/**
* gst_rtsp_stream_transport_send_rtp:
* @trans: a #GstRTSPStreamTransport
* @buffer: a #GstBuffer
* @buffer: (transfer none): a #GstBuffer
*
* Send @buffer to the installed RTP callback for @trans.
*
......@@ -461,7 +461,7 @@ gst_rtsp_stream_transport_send_rtp (GstRTSPStreamTransport * trans,
/**
* gst_rtsp_stream_transport_send_rtcp:
* @trans: a #GstRTSPStreamTransport
* @buffer: a #GstBuffer
* @buffer: (transfer none): a #GstBuffer
*
* Send @buffer to the installed RTCP callback for @trans.
*
......
......@@ -290,7 +290,7 @@ gst_rtsp_stream_set_property (GObject * object, guint propid,
* Create a new media stream with index @idx that handles RTP data on
* @srcpad and has a payloader element @payloader.
*
* Returns: a new #GstRTSPStream
* Returns: (transfer full): a new #GstRTSPStream
*/
GstRTSPStream *
gst_rtsp_stream_new (guint idx, GstElement * payloader, GstPad * srcpad)
......@@ -372,7 +372,7 @@ gst_rtsp_stream_get_srcpad (GstRTSPStream * stream)
*
* Get the control string to identify this stream.
*
* Returns: (transfer full): the control string. free after usage.
* Returns: (transfer full): the control string. g_free() after usage.
*/
gchar *
gst_rtsp_stream_get_control (GstRTSPStream * stream)
......@@ -565,7 +565,7 @@ gst_rtsp_stream_get_dscp_qos (GstRTSPStream * stream)
/**
* gst_rtsp_stream_is_transport_supported:
* @stream: a #GstRTSPStream
* @transport: a #GstRTSPTransport
* @transport: (transfer none): a #GstRTSPTransport
*
* Check if @transport can be handled by stream
*
......@@ -713,7 +713,7 @@ gst_rtsp_stream_get_protocols (GstRTSPStream * stream)
/**
* gst_rtsp_stream_set_address_pool:
* @stream: a #GstRTSPStream
* @pool: a #GstRTSPAddressPool
* @pool: (transfer none): a #GstRTSPAddressPool
*
* configure @pool to be used as the address pool of @stream.
*/
......@@ -775,8 +775,8 @@ gst_rtsp_stream_get_address_pool (GstRTSPStream * stream)
*
* Get the multicast address of @stream for @family.
*
* Returns: the #GstRTSPAddress of @stream or %NULL when no address could be
* allocated. gst_rtsp_address_free() after usage.
* Returns: (transfer full): the #GstRTSPAddress of @stream or %NULL when no
* address could be allocated. gst_rtsp_address_free() after usage.
*/
GstRTSPAddress *
gst_rtsp_stream_get_multicast_address (GstRTSPStream * stream,
......@@ -1506,8 +1506,8 @@ static GstAppSinkCallbacks sink_cb = {
/**
* gst_rtsp_stream_join_bin:
* @stream: a #GstRTSPStream
* @bin: a #GstBin to join
* @rtpbin: a rtpbin element in @bin
* @bin: (transfer none): a #GstBin to join
* @rtpbin: (transfer none): a rtpbin element in @bin
* @state: the target state of the new elements
*
* Join the #GstBin @bin that contains the element @rtpbin.
......@@ -1771,8 +1771,8 @@ link_failed:
/**
* gst_rtsp_stream_leave_bin:
* @stream: a #GstRTSPStream
* @bin: a #GstBin
* @rtpbin: a rtpbin #GstElement
* @bin: (transfer none): a #GstBin
* @rtpbin: (transfer none): a rtpbin #GstElement
*
* Remove the elements of @stream from @bin.
*
......@@ -1890,7 +1890,7 @@ was_not_joined:
* @stream: a #GstRTSPStream
* @rtptime: (allow-none): result RTP timestamp
* @seq: (allow-none): result RTP seqnum
* @clock_rate: the clock rate
* @clock_rate: (allow-none): the clock rate
* @running_time: (allow-none): result running-time
*
* Retrieve the current rtptime, seq and running-time. This is used to
......@@ -1969,7 +1969,7 @@ no_stats:
* Retrieve the current caps of @stream.
*
* Returns: (transfer full): the #GstCaps of @stream. use gst_caps_unref()
* after usage.
* after usage.
*/
GstCaps *
gst_rtsp_stream_get_caps (GstRTSPStream * stream)
......@@ -2065,6 +2065,7 @@ gst_rtsp_stream_recv_rtcp (GstRTSPStream * stream, GstBuffer * buffer)
gst_object_unref (element);