From 502eddfc36ca6f0d6d507520f0b9e8f8322f6633 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Sebastian=20Dr=C3=B6ge?= Date: Sat, 15 Oct 2022 12:40:04 +0300 Subject: [PATCH] rtsp-server: Add/fix various annotations Part-of: --- subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-auth.c | 3 ++- .../gst-rtsp-server/gst/rtsp-server/rtsp-latency-bin.c | 2 +- .../gst-rtsp-server/gst/rtsp-server/rtsp-media-factory.c | 6 +++--- subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-media.c | 2 +- .../gst/rtsp-server/rtsp-onvif-media-factory.c | 4 ++-- subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-server.c | 8 ++++---- subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-stream.c | 4 ++-- 7 files changed, 15 insertions(+), 14 deletions(-) diff --git a/subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-auth.c b/subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-auth.c index b6286e173c..c16f7a6732 100644 --- a/subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-auth.c +++ b/subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-auth.c @@ -1231,6 +1231,7 @@ gst_rtsp_auth_make_basic (const gchar * user, const gchar * pass) /** * gst_rtsp_auth_set_realm: + * @realm: (nullable): The realm to set * * Set the @realm of @auth * @@ -1251,7 +1252,7 @@ gst_rtsp_auth_set_realm (GstRTSPAuth * auth, const gchar * realm) /** * gst_rtsp_auth_get_realm: * - * Returns: (transfer full): the @realm of @auth + * Returns: (transfer full) (nullable): the @realm of @auth * * Since: 1.16 */ diff --git a/subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-latency-bin.c b/subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-latency-bin.c index c297ab63ee..13dd57ec72 100644 --- a/subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-latency-bin.c +++ b/subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-latency-bin.c @@ -336,7 +336,7 @@ gst_rtsp_latency_bin_change_state (GstElement * element, GstStateChange * Create a bin that encapsulates an @element and prevents it from affecting * latency on the whole pipeline. * - * Returns: A newly created #GstRTSPLatencyBin element, or %NULL on failure + * Returns: (nullable): A newly created #GstRTSPLatencyBin element, or %NULL on failure */ GstElement * gst_rtsp_latency_bin_new (GstElement * element) diff --git a/subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-media-factory.c b/subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-media-factory.c index 5dd9dc0a1b..a044da0cfb 100644 --- a/subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-media-factory.c +++ b/subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-media-factory.c @@ -1375,7 +1375,7 @@ weak_ref_free (GWeakRef * ref) * After the media is constructed, it can be configured and then prepared * with gst_rtsp_media_prepare (). * - * Returns: (transfer full): a new #GstRTSPMedia if the media could be prepared. + * Returns: (transfer full) (nullable): a new #GstRTSPMedia if the media could be prepared. */ GstRTSPMedia * gst_rtsp_media_factory_construct (GstRTSPMediaFactory * factory, @@ -1534,7 +1534,7 @@ gst_rtsp_media_factory_set_clock (GstRTSPMediaFactory * factory, * Returns the clock that is going to be used by the pipelines * of all medias created from this factory. * - * Returns: (transfer full): The GstClock + * Returns: (transfer full) (nullable): The GstClock * * Since: 1.8 */ @@ -1987,7 +1987,7 @@ default_configure (GstRTSPMediaFactory * factory, GstRTSPMedia * media) * implementation of this function returns the bin created from the * launch parameter. * - * Returns: (transfer floating): a new #GstElement. + * Returns: (transfer floating) (nullable): a new #GstElement. */ GstElement * gst_rtsp_media_factory_create_element (GstRTSPMediaFactory * factory, diff --git a/subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-media.c b/subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-media.c index 57db50556a..b5c300d8a7 100644 --- a/subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-media.c +++ b/subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-media.c @@ -4316,7 +4316,7 @@ not_prepared: * Get the #GstNetTimeProvider for the clock used by @media. The time provider * will listen on @address and @port for client time requests. * - * Returns: (transfer full): the #GstNetTimeProvider of @media. + * Returns: (transfer full) (nullable): the #GstNetTimeProvider of @media. */ GstNetTimeProvider * gst_rtsp_media_get_time_provider (GstRTSPMedia * media, const gchar * address, diff --git a/subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-onvif-media-factory.c b/subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-onvif-media-factory.c index dbd461a5ab..420ca052f4 100644 --- a/subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-onvif-media-factory.c +++ b/subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-onvif-media-factory.c @@ -360,7 +360,7 @@ gst_rtsp_onvif_media_factory_init (GstRTSPOnvifMediaFactory * factory) /** * gst_rtsp_onvif_media_factory_set_backchannel_launch: * @factory: a #GstRTSPMediaFactory - * @launch: the launch description + * @launch: (nullable): the launch description * * The gst_parse_launch() line to use for constructing the ONVIF backchannel * pipeline in the default prepare vmethod if requested by the client. @@ -400,7 +400,7 @@ gst_rtsp_onvif_media_factory_set_backchannel_launch (GstRTSPOnvifMediaFactory * * Get the gst_parse_launch() pipeline description that will be used in the * default prepare vmethod for generating the ONVIF backchannel pipeline. * - * Returns: (transfer full): the configured backchannel launch description. g_free() after + * Returns: (transfer full) (nullable): the configured backchannel launch description. g_free() after * usage. * * Since: 1.14 diff --git a/subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-server.c b/subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-server.c index ddde369a7a..8c810901a0 100644 --- a/subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-server.c +++ b/subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-server.c @@ -434,7 +434,7 @@ gst_rtsp_server_set_service (GstRTSPServer * server, const gchar * service) * * Get the service on which the server will accept connections. * - * Returns: (transfer full) (nullable): 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) @@ -1318,7 +1318,7 @@ watch_destroyed (GstRTSPServer * server) /** * gst_rtsp_server_create_source: * @server: a #GstRTSPServer - * @cancellable: (allow-none): a #GCancellable or %NULL. + * @cancellable: (nullable): a #GCancellable or %NULL. * @error: (out): a #GError * * Create a #GSource for @server. The new source will have a default @@ -1380,7 +1380,7 @@ no_socket: /** * gst_rtsp_server_attach: * @server: a #GstRTSPServer - * @context: (allow-none): a #GMainContext + * @context: (nullable): a #GMainContext * * Attaches @server to @context. When the mainloop for @context is run, the * server will be dispatched. When @context is %NULL, the default context will be @@ -1427,7 +1427,7 @@ no_source: /** * gst_rtsp_server_client_filter: * @server: a #GstRTSPServer - * @func: (scope call) (allow-none): a callback + * @func: (scope call) (nullable): a callback * @user_data: user data passed to @func * * Call @func for each client managed by @server. The result value of @func diff --git a/subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-stream.c b/subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-stream.c index 99d87e9d97..24e8e63e0b 100644 --- a/subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-stream.c +++ b/subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-stream.c @@ -1993,7 +1993,7 @@ gst_rtsp_stream_get_server_port (GstRTSPStream * stream, * * Get the RTP session of this stream. * - * Returns: (transfer full): The RTP session of this stream. Unref after usage. + * Returns: (transfer full) (nullable): The RTP session of this stream. Unref after usage. */ GObject * gst_rtsp_stream_get_rtpsession (GstRTSPStream * stream) @@ -2019,7 +2019,7 @@ gst_rtsp_stream_get_rtpsession (GstRTSPStream * stream) * * Get the SRTP encoder for this stream. * - * Returns: (transfer full): The SRTP encoder for this stream. Unref after usage. + * Returns: (transfer full) (nullable): The SRTP encoder for this stream. Unref after usage. */ GstElement * gst_rtsp_stream_get_srtp_encoder (GstRTSPStream * stream)