*
* Make a new #GstRTSPAddressPool.
*
- * Returns: a new #GstRTSPAddressPool
+ * Returns: (transfer full): a new #GstRTSPAddressPool
*/
GstRTSPAddressPool *
gst_rtsp_address_pool_new (void)
* 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,
*
* Create a new #GstRTSPAuth instance.
*
- * Returns: a new #GstRTSPAuth
+ * Returns: (transfer full): a new #GstRTSPAuth
*/
GstRTSPAuth *
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.
/**
* 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.
* 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.
*
* 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)
*
* Create a new #GstRTSPClient instance.
*
- * Returns: a new #GstRTSPClient
+ * Returns: (transfer full): a new #GstRTSPClient
*/
GstRTSPClient *
gst_rtsp_client_new (void)
/**
* 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
/**
* 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
/**
* 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.
*/
/**
* 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.
*/
/**
* 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
/**
* gst_rtsp_client_handle_message:
* @client: a #GstRTSPClient
- * @message: an #GstRTSPMessage
+ * @message: (transfer none): an #GstRTSPMessage
*
* Let the client handle @message.
*
/**
* 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.
*
* 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)
*
* 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)
*
* Create a new #GstRTSPMediaFactory instance.
*
- * Returns: a new #GstRTSPMediaFactory object.
+ * Returns: (transfer full): a new #GstRTSPMediaFactory object.
*/
GstRTSPMediaFactory *
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.
*/
* 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)
/**
* 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.
*/
*
* Ownership is taken of @element.
*
- * Returns: a new #GstRTSPMedia object.
+ * Returns: (transfer full): a new #GstRTSPMedia object.
*/
GstRTSPMedia *
gst_rtsp_media_new (GstElement * element)
/**
* gst_rtsp_media_set_permissions:
* @media: a #GstRTSPMedia
- * @permissions: a #GstRTSPPermissions
+ * @permissions: (transfer none): a #GstRTSPPermissions
*
* Set @permissions on @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.
*/
* @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,
* 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,
/**
* 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().
/**
* 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
/**
* 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.
* 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.
*
*
* Make a new mount points object.
*
- * Returns: a new #GstRTSPMountPoints
+ * Returns: (transfer full): a new #GstRTSPMountPoints
*/
GstRTSPMountPoints *
gst_rtsp_mount_points_new (void)
*
* 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,
* 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,
/**
* gst_rtsp_params_set:
* @client: a #GstRTSPClient
- * @ctx: a #GstRTSPContext
+ * @ctx: (transfer none): a #GstRTSPContext
*
* Set parameters (not implemented yet)
*
/**
* gst_rtsp_params_get:
* @client: a #GstRTSPClient
- * @ctx: a #GstRTSPContext
+ * @ctx: (transfer none): a #GstRTSPContext
*
* Get parameters (not implemented yet)
*
*
* 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,
/**
* 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.
* gst_rtsp_server_new:
*
* Create a new #GstRTSPServer instance.
+ *
+ * Returns: (transfer full): a new #GstRTSPServer
*/
GstRTSPServer *
gst_rtsp_server_new (void)
*
* 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)
*
* 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)
/**
* 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.
*/
/**
* 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.
*/
/**
* 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.
*/
/**
* 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.
*/
/**
* 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,
* 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,
* 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,
/**
* 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().
* 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,
/**
* 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)
* 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)
* 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)
* 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.
*
/**
* 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.
*
* 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)
/**
* 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.
*
* 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
* 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)
/**
* 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.
*
* 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
* @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)
*
* 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)
/**
* 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.
*
/**
* 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.
*
* 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)
* @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.
/**
* 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.
* 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)
* 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)
*
* 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,
/**
* 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.
*
/**
* 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.
*
* 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)
*
* 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)
/**
* 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
*
/**
* 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.
*/
*
* 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,
/**
* 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.
/**
* 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.
*
* @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
* 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)
gst_object_unref (element);
} else {
ret = GST_FLOW_OK;
+ gst_buffer_unref (buffer);
}
return ret;
}
/**
* gst_rtsp_stream_add_transport:
* @stream: a #GstRTSPStream
- * @trans: a #GstRTSPStreamTransport
+ * @trans: (transfer none): a #GstRTSPStreamTransport
*
* Add the transport in @trans to @stream. The media of @stream will
* then also be send to the values configured in @trans.
/**
* gst_rtsp_stream_remove_transport:
* @stream: a #GstRTSPStream
- * @trans: a #GstRTSPStreamTransport
+ * @trans: (transfer none): a #GstRTSPStreamTransport
*
* Remove the transport in @trans from @stream. The media of @stream will
* not be sent to the values configured in @trans.
* @stream must be joined to a bin.
*
* Returns: (transfer full): the RTP socket or %NULL if no socket could be
- * allocated for @family. Unref after usage
+ * allocated for @family. Unref after usage
*/
GSocket *
gst_rtsp_stream_get_rtp_socket (GstRTSPStream * stream, GSocketFamily family)
* @stream must be joined to a bin.
*
* Returns: (transfer full): the RTCP socket or %NULL if no socket could be
- * allocated for @family. Unref after usage
+ * allocated for @family. Unref after usage
*/
GSocket *
gst_rtsp_stream_get_rtcp_socket (GstRTSPStream * stream, GSocketFamily family)
* gst_rtsp_stream_transport_filter:
* @stream: a #GstRTSPStream
* @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 transport managed by @stream. The result value of @func
* determines what happens to the transport. @func will be called with @stream
*
* Create a new thread object that can run a mainloop.
*
- * Returns: a #GstRTSPThread.
+ * Returns: (transfer full): a #GstRTSPThread.
*/
GstRTSPThread *
gst_rtsp_thread_new (GstRTSPThreadType type)
/**
* gst_rtsp_thread_reuse:
- * @thread: a #GstRTSPThread
+ * @thread: (transfer none): a #GstRTSPThread
*
* Reuse the mainloop of @thread
*
*
* Create a new #GstRTSPThreadPool instance.
*
- * Returns: a new #GstRTSPThreadPool
+ * Returns: (transfer full): a new #GstRTSPThreadPool
*/
GstRTSPThreadPool *
gst_rtsp_thread_pool_new (void)
* gst_rtsp_thread_pool_get_thread:
* @pool: a #GstRTSPThreadPool
* @type: the #GstRTSPThreadType
- * @ctx: a #GstRTSPContext
+ * @ctx: (transfer none): a #GstRTSPContext
*
* Get a new #GstRTSPThread for @type and @ctx.
*
- * Returns: a new #GstRTSPThread, gst_rtsp_thread_stop() after usage
+ * Returns: (transfer full): a new #GstRTSPThread, gst_rtsp_thread_stop() after usage
*/
GstRTSPThread *
gst_rtsp_thread_pool_get_thread (GstRTSPThreadPool * pool,
*
* Access the structure of the token.
*
- * Returns: The structure of the token. The structure is still
- * owned by the token, which means that you should not free it and
- * that the pointer becomes invalid when you free the token.
+ * Returns: (transfer none): The structure of the token. The structure is still
+ * owned by the token, which means that you should not free it and that the
+ * pointer becomes invalid when you free the token.
*
* MT safe.
*/
*
* Get a writable version of the structure.
*
- * Returns: The structure of the token. The structure is still
- * owned by the token, which means that you should not free it and
- * that the pointer becomes invalid when you free the token.
- * This function checks if @token is writable and will never return %NULL.
+ * Returns: (transfer none): The structure of the token. The structure is still
+ * owned by the token, which means that you should not free it and that the
+ * pointer becomes invalid when you free the token. This function checks if
+ * @token is writable and will never return %NULL.
*
* MT safe.
*/
*
* Get the string value of @field in @token.
*
- * Returns: the string value of @field in @token or %NULL when @field is not
- * defined in @token.
+ * Returns: (transfer none): the string value of @field in @token or %NULL when
+ * @field is not defined in @token. The string becomes invalid when you free
+ * @token.
*/
const gchar *
gst_rtsp_token_get_string (GstRTSPToken * token, const gchar * field)
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