mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-09-29 23:32:33 +00:00
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
This commit is contained in:
Sebastian Rasmussen
Tim-Philipp Müller
parent
0b617dd5bd
commit
b1b5301577
18 changed files with 132 additions and 119 deletions
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@ -181,7 +181,7 @@ gst_rtsp_address_pool_finalize (GObject * obj)
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*
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* Make a new #GstRTSPAddressPool.
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*
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* Returns: a new #GstRTSPAddressPool
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* Returns: (transfer full): a new #GstRTSPAddressPool
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*/
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GstRTSPAddressPool *
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gst_rtsp_address_pool_new (void)
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@ -424,7 +424,7 @@ split_range (GstRTSPAddressPool * pool, AddrRange * range, guint skip_addr,
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* one can be found in @port.
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*
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* Returns: a #GstRTSPAddress that should be freed with gst_rtsp_address_free
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* after use or %NULL when no address could be acquired.
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* after use or %NULL when no address could be acquired.
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*/
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GstRTSPAddress *
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gst_rtsp_address_pool_acquire_address (GstRTSPAddressPool * pool,
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@ -161,7 +161,7 @@ gst_rtsp_auth_set_property (GObject * object, guint propid,
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*
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* Create a new #GstRTSPAuth instance.
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*
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* Returns: a new #GstRTSPAuth
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* Returns: (transfer full): a new #GstRTSPAuth
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*/
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GstRTSPAuth *
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gst_rtsp_auth_new (void)
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@ -176,7 +176,7 @@ gst_rtsp_auth_new (void)
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/**
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* gst_rtsp_auth_set_tls_certificate:
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* @auth: a #GstRTSPAuth
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* @cert: (allow-none): a #GTlsCertificate
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* @cert: (transfer none) (allow-none): a #GTlsCertificate
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*
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* Set the TLS certificate for the auth. Client connections will only
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* be accepted when TLS is negotiated.
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@ -233,7 +233,7 @@ gst_rtsp_auth_get_tls_certificate (GstRTSPAuth * auth)
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/**
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* gst_rtsp_auth_set_default_token:
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* @auth: a #GstRTSPAuth
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* @token: (allow-none): a #GstRTSPToken
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* @token: (transfer none) (allow-none): a #GstRTSPToken
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*
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* Set the default #GstRTSPToken to @token in @auth. The default token will
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* be used for unauthenticated users.
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@ -292,7 +292,7 @@ gst_rtsp_auth_get_default_token (GstRTSPAuth * auth)
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* gst_rtsp_auth_add_basic:
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* @auth: a #GstRTSPAuth
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* @basic: the basic token
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* @token: authorisation token
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* @token: (transfer none): authorisation token
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*
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* Add a basic token for the default authentication algorithm that
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* enables the client with privileges listed in @token.
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@ -616,8 +616,8 @@ no_context:
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*
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* Construct a Basic authorisation token from @user and @pass.
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*
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* Returns: the base64 encoding of the string @user:@pass. g_free()
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* after usage.
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* Returns: (transfer full): the base64 encoding of the string @user:@pass.
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* g_free() after usage.
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*/
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gchar *
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gst_rtsp_auth_make_basic (const gchar * user, const gchar * pass)
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@ -412,7 +412,7 @@ gst_rtsp_client_set_property (GObject * object, guint propid,
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*
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* Create a new #GstRTSPClient instance.
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*
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* Returns: a new #GstRTSPClient
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* Returns: (transfer full): a new #GstRTSPClient
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*/
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GstRTSPClient *
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gst_rtsp_client_new (void)
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@ -2199,7 +2199,7 @@ handle_data (GstRTSPClient * client, GstRTSPMessage * message)
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/**
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* gst_rtsp_client_set_session_pool:
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* @client: a #GstRTSPClient
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* @pool: a #GstRTSPSessionPool
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* @pool: (transfer none): a #GstRTSPSessionPool
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*
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* Set @pool as the sessionpool for @client which it will use to find
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* or allocate sessions. the sessionpool is usually inherited from the server
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@ -2257,7 +2257,7 @@ gst_rtsp_client_get_session_pool (GstRTSPClient * client)
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/**
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* gst_rtsp_client_set_mount_points:
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* @client: a #GstRTSPClient
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* @mounts: a #GstRTSPMountPoints
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* @mounts: (transfer none): a #GstRTSPMountPoints
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*
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* Set @mounts as the mount points for @client which it will use to map urls
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* to media streams. These mount points are usually inherited from the server that
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@ -2315,7 +2315,7 @@ gst_rtsp_client_get_mount_points (GstRTSPClient * client)
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/**
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* gst_rtsp_client_set_auth:
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* @client: a #GstRTSPClient
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* @auth: a #GstRTSPAuth
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* @auth: (transfer none): a #GstRTSPAuth
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*
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* configure @auth to be used as the authentication manager of @client.
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*/
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@ -2372,7 +2372,7 @@ gst_rtsp_client_get_auth (GstRTSPClient * client)
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/**
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* gst_rtsp_client_set_thread_pool:
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* @client: a #GstRTSPClient
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* @pool: a #GstRTSPThreadPool
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* @pool: (transfer none): a #GstRTSPThreadPool
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*
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* configure @pool to be used as the thread pool of @client.
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*/
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@ -2511,9 +2511,9 @@ gst_rtsp_client_get_connection (GstRTSPClient * client)
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/**
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* gst_rtsp_client_set_send_func:
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* @client: a #GstRTSPClient
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* @func: a #GstRTSPClientSendFunc
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* @user_data: user data passed to @func
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* @notify: called when @user_data is no longer in use
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* @func: (scope notified): a #GstRTSPClientSendFunc
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* @user_data: (closure): user data passed to @func
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* @notify: (allow-none): called when @user_data is no longer in use
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*
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* Set @func as the callback that will be called when a new message needs to be
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* sent to the client. @user_data is passed to @func and @notify is called when
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@ -2549,7 +2549,7 @@ gst_rtsp_client_set_send_func (GstRTSPClient * client,
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/**
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* gst_rtsp_client_handle_message:
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* @client: a #GstRTSPClient
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* @message: an #GstRTSPMessage
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* @message: (transfer none): an #GstRTSPMessage
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*
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* Let the client handle @message.
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*
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@ -2581,8 +2581,8 @@ gst_rtsp_client_handle_message (GstRTSPClient * client,
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/**
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* gst_rtsp_client_send_message:
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* @client: a #GstRTSPClient
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* @session: a #GstRTSPSession to send the message to or %NULL
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* @message: The #GstRTSPMessage to send
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* @session: (transfer none): a #GstRTSPSession to send the message to or %NULL
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* @message: (transfer none): The #GstRTSPMessage to send
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*
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* Send a message message to the remote end. @message must be a
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* #GST_RTSP_MESSAGE_REQUEST or a #GST_RTSP_MESSAGE_RESPONSE.
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@ -276,7 +276,7 @@ gst_rtsp_media_factory_uri_set_property (GObject * object, guint propid,
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*
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* Create a new #GstRTSPMediaFactoryURI instance.
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*
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* Returns: a new #GstRTSPMediaFactoryURI object.
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* Returns: (transfer full): a new #GstRTSPMediaFactoryURI object.
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*/
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GstRTSPMediaFactoryURI *
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gst_rtsp_media_factory_uri_new (void)
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@ -318,7 +318,7 @@ gst_rtsp_media_factory_uri_set_uri (GstRTSPMediaFactoryURI * factory,
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*
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* Get the URI that will provide media for this factory.
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*
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* Returns: the configured URI. g_free() after usage.
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* Returns: (transfer full): the configured URI. g_free() after usage.
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*/
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gchar *
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gst_rtsp_media_factory_uri_get_uri (GstRTSPMediaFactoryURI * factory)
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@ -317,7 +317,7 @@ gst_rtsp_media_factory_set_property (GObject * object, guint propid,
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*
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* Create a new #GstRTSPMediaFactory instance.
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*
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* Returns: a new #GstRTSPMediaFactory object.
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* Returns: (transfer full): a new #GstRTSPMediaFactory object.
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*/
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GstRTSPMediaFactory *
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gst_rtsp_media_factory_new (void)
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@ -332,7 +332,7 @@ gst_rtsp_media_factory_new (void)
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/**
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* gst_rtsp_media_factory_set_permissions:
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* @factory: a #GstRTSPMediaFactory
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* @permissions: a #GstRTSPPermissions
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* @permissions: (transfer none): a #GstRTSPPermissions
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*
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* Set @permissions on @factory.
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*/
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@ -455,7 +455,8 @@ gst_rtsp_media_factory_set_launch (GstRTSPMediaFactory * factory,
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* Get the gst_parse_launch() pipeline description that will be used in the
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* default prepare vmethod.
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*
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* Returns: the configured launch description. g_free() after usage.
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* Returns: (transfer full): the configured launch description. g_free() after
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* usage.
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*/
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gchar *
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gst_rtsp_media_factory_get_launch (GstRTSPMediaFactory * factory)
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@ -668,7 +669,7 @@ gst_rtsp_media_factory_get_buffer_size (GstRTSPMediaFactory * factory)
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/**
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* gst_rtsp_media_factory_set_address_pool:
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* @factory: a #GstRTSPMediaFactory
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* @pool: a #GstRTSPAddressPool
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* @pool: (transfer none): a #GstRTSPAddressPool
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*
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* configure @pool to be used as the address pool of @factory.
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*/
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@ -563,7 +563,7 @@ collect_media_stats (GstRTSPMedia * media)
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*
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* Ownership is taken of @element.
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*
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* Returns: a new #GstRTSPMedia object.
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* Returns: (transfer full): a new #GstRTSPMedia object.
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*/
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GstRTSPMedia *
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gst_rtsp_media_new (GstElement * element)
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@ -632,7 +632,7 @@ gst_rtsp_media_take_pipeline (GstRTSPMedia * media, GstPipeline * pipeline)
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/**
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* gst_rtsp_media_set_permissions:
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* @media: a #GstRTSPMedia
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* @permissions: a #GstRTSPPermissions
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* @permissions: (transfer none): a #GstRTSPPermissions
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*
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* Set @permissions on @media.
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*/
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@ -1089,7 +1089,7 @@ gst_rtsp_media_is_time_provider (GstRTSPMedia * media)
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/**
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* gst_rtsp_media_set_address_pool:
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* @media: a #GstRTSPMedia
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* @pool: a #GstRTSPAddressPool
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* @pool: (transfer none): a #GstRTSPAddressPool
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*
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* configure @pool to be used as the address pool of @media.
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*/
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@ -1217,7 +1217,7 @@ gst_rtsp_media_collect_streams (GstRTSPMedia * media)
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* @srcpad should be a pad of an element inside @media->element.
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*
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* Returns: (transfer none): a new #GstRTSPStream that remains valid for as long
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* as @media exists.
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* as @media exists.
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*/
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GstRTSPStream *
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gst_rtsp_media_create_stream (GstRTSPMedia * media, GstElement * payloader,
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@ -1397,7 +1397,7 @@ default_convert_range (GstRTSPMedia * media, GstRTSPTimeRange * range,
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* Get the current range as a string. @media must be prepared with
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* gst_rtsp_media_prepare ().
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*
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* Returns: The range as a string, g_free() after usage.
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* Returns: (transfer full): The range as a string, g_free() after usage.
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*/
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gchar *
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gst_rtsp_media_get_range_string (GstRTSPMedia * media, gboolean play,
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@ -1474,7 +1474,7 @@ media_streams_set_blocked (GstRTSPMedia * media, gboolean blocked)
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/**
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* gst_rtsp_media_seek:
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* @media: a #GstRTSPMedia
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* @range: a #GstRTSPTimeRange
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* @range: (transfer none): a #GstRTSPTimeRange
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*
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* Seek the pipeline of @media to @range. @media must be prepared with
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* gst_rtsp_media_prepare().
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@ -2116,7 +2116,7 @@ preroll_failed:
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/**
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* gst_rtsp_media_prepare:
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* @media: a #GstRTSPMedia
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* @thread: a #GstRTSPThread to run the bus handler or %NULL
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* @thread: (transfer full): a #GstRTSPThread to run the bus handler or %NULL
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*
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* Prepare @media for streaming. This function will create the objects
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* to manage the streaming. A pipeline must have been set on @media with
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@ -2541,9 +2541,9 @@ default_setup_sdp (GstRTSPMedia * media, GstSDPMessage * sdp, GstSDPInfo * info)
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/**
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* gst_rtsp_media_setup_sdp:
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* @sdp: a #GstSDPMessage
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* @info: info
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* @media: a #GstRTSPMedia
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* @sdp: (transfer none): a #GstSDPMessage
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* @info: (transfer none): a #GstSDPInfo
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*
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* Add @media specific info to @sdp. @info is used to configure the connection
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* information in the SDP.
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@ -2772,8 +2772,8 @@ gst_rtsp_media_set_pipeline_state (GstRTSPMedia * media, GstState state)
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* gst_rtsp_media_set_state:
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* @media: a #GstRTSPMedia
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* @state: the target state of the media
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* @transports: (element-type GstRtspServer.RTSPStreamTransport): a #GPtrArray
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* of #GstRTSPStreamTransport pointers
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* @transports: (transfer none) (element-type GstRtspServer.RTSPStreamTransport):
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* a #GPtrArray of #GstRTSPStreamTransport pointers
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*
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* Set the state of @media to @state and for the transports in @transports.
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*
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@ -156,7 +156,7 @@ gst_rtsp_mount_points_finalize (GObject * obj)
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*
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* Make a new mount points object.
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*
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* Returns: a new #GstRTSPMountPoints
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* Returns: (transfer full): a new #GstRTSPMountPoints
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*/
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GstRTSPMountPoints *
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gst_rtsp_mount_points_new (void)
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@ -181,7 +181,7 @@ default_make_path (GstRTSPMountPoints * mounts, const GstRTSPUrl * url)
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*
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* Make a path string from @url.
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*
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* Returns: a path string for @url, g_free() after usage.
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* Returns: (transfer full): a path string for @url, g_free() after usage.
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*/
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gchar *
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gst_rtsp_mount_points_make_path (GstRTSPMountPoints * mounts,
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@ -229,7 +229,7 @@ has_prefix (DataItem * str, DataItem * prefix)
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* the amount of characters that matched is returned in @matched.
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*
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* Returns: (transfer full): the #GstRTSPMediaFactory for @path.
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* g_object_unref() after usage.
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* g_object_unref() after usage.
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*/
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GstRTSPMediaFactory *
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gst_rtsp_mount_points_match (GstRTSPMountPoints * mounts,
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@ -31,7 +31,7 @@
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/**
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* gst_rtsp_params_set:
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* @client: a #GstRTSPClient
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* @ctx: a #GstRTSPContext
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* @ctx: (transfer none): a #GstRTSPContext
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*
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* Set parameters (not implemented yet)
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*
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@ -55,7 +55,7 @@ gst_rtsp_params_set (GstRTSPClient * client, GstRTSPContext * ctx)
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/**
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* gst_rtsp_params_get:
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* @client: a #GstRTSPClient
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* @ctx: a #GstRTSPContext
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* @ctx: (transfer none): a #GstRTSPContext
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*
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* Get parameters (not implemented yet)
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*
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@ -221,7 +221,8 @@ gst_rtsp_permissions_remove_role (GstRTSPPermissions * permissions,
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*
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* Get all permissions for @role in @permissions.
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*
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* Returns: the structure with permissions for @role.
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* Returns: (transfer none): the structure with permissions for @role. It
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* remains valid for as long as @permissions is valid.
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*/
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const GstStructure *
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gst_rtsp_permissions_get_role (GstRTSPPermissions * permissions,
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@ -245,8 +245,8 @@ no_multicast:
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/**
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* gst_rtsp_sdp_from_media:
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* @sdp: a #GstSDPMessage
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* @info: info
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* @media: a #GstRTSPMedia
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* @info: (transfer none): a #GstSDPInfo
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* @media: (transfer none): a #GstRTSPMedia
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*
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* Add @media specific info to @sdp. @info is used to configure the connection
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* information in the SDP.
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@ -283,6 +283,8 @@ gst_rtsp_server_finalize (GObject * object)
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* gst_rtsp_server_new:
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*
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* Create a new #GstRTSPServer instance.
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*
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* Returns: (transfer full): a new #GstRTSPServer
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*/
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GstRTSPServer *
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gst_rtsp_server_new (void)
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@ -325,7 +327,7 @@ gst_rtsp_server_set_address (GstRTSPServer * server, const gchar * address)
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*
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* Get the address on which the server will accept connections.
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*
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* Returns: the server address. g_free() after usage.
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* 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);
|
||||
} else {
|
||||
ret = GST_FLOW_OK;
|
||||
gst_buffer_unref (buffer);
|
||||
}
|
||||
return ret;
|
||||
}
|
||||
|
|
@ -2143,7 +2144,7 @@ unknown_transport:
|
|||
/**
|
||||
* 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.
|
||||
|
|
@ -2176,7 +2177,7 @@ gst_rtsp_stream_add_transport (GstRTSPStream * stream,
|
|||
/**
|
||||
* 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.
|
||||
|
|
@ -2216,7 +2217,7 @@ gst_rtsp_stream_remove_transport (GstRTSPStream * stream,
|
|||
* @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)
|
||||
|
|
@ -2250,7 +2251,7 @@ 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)
|
||||
|
|
@ -2278,7 +2279,7 @@ 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
|
||||
|
|
|
|||
|
|
@ -106,7 +106,7 @@ gst_rtsp_thread_init (GstRTSPThreadImpl * impl)
|
|||
*
|
||||
* 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)
|
||||
|
|
@ -125,7 +125,7 @@ gst_rtsp_thread_new (GstRTSPThreadType type)
|
|||
|
||||
/**
|
||||
* gst_rtsp_thread_reuse:
|
||||
* @thread: a #GstRTSPThread
|
||||
* @thread: (transfer none): a #GstRTSPThread
|
||||
*
|
||||
* Reuse the mainloop of @thread
|
||||
*
|
||||
|
|
@ -346,7 +346,7 @@ do_loop (GstRTSPThread * thread)
|
|||
*
|
||||
* Create a new #GstRTSPThreadPool instance.
|
||||
*
|
||||
* Returns: a new #GstRTSPThreadPool
|
||||
* Returns: (transfer full): a new #GstRTSPThreadPool
|
||||
*/
|
||||
GstRTSPThreadPool *
|
||||
gst_rtsp_thread_pool_new (void)
|
||||
|
|
@ -506,11 +506,11 @@ thread_error:
|
|||
* 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,
|
||||
|
|
|
|||
|
|
@ -172,9 +172,9 @@ gst_rtsp_token_new_valist (const gchar * firstfield, va_list var_args)
|
|||
*
|
||||
* 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.
|
||||
*/
|
||||
|
|
@ -192,10 +192,10 @@ gst_rtsp_token_get_structure (GstRTSPToken * token)
|
|||
*
|
||||
* 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.
|
||||
*/
|
||||
|
|
@ -216,8 +216,9 @@ gst_rtsp_token_writable_structure (GstRTSPToken * token)
|
|||
*
|
||||
* 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)
|
||||
|
|
|
|||
Loading…
Reference in a new issue