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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2025-02-22 05:56:31 +00:00
Fixed several GIR warnings
This commit is contained in:
parent
533d237754
commit
e756324490
9 changed files with 19 additions and 14 deletions
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@ -82,12 +82,15 @@ GstRtspServer-@GST_API_VERSION@.gir: $(INTROSPECTION_SCANNER) libgstrtspserver-@
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--add-include-path=`$(PKG_CONFIG) --variable=girdir gstreamer-@GST_API_VERSION@` \
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--add-include-path=`$(PKG_CONFIG) --variable=girdir gstreamer-rtsp-@GST_API_VERSION@` \
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--add-include-path=`$(PKG_CONFIG) --variable=girdir gstreamer-sdp-@GST_API_VERSION@` \
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--add-include-path=`$(PKG_CONFIG) --variable=girdir gstreamer-net-@GST_API_VERSION@` \
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--library=libgstrtspserver-@GST_API_VERSION@.la \
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--include=Gst-@GST_API_VERSION@ \
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--include=GstRtsp-@GST_API_VERSION@ \
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--include=GstNet-@GST_API_VERSION@ \
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--libtool="$(top_builddir)/libtool" \
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--pkg gstreamer-@GST_API_VERSION@ \
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--pkg gstreamer-rtsp-@GST_API_VERSION@ \
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--pkg gstreamer-net-@GST_API_VERSION@ \
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--pkg-export gstreamer-rtsp-server-@GST_API_VERSION@ \
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--output $@ \
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$(gir_headers) \
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@ -110,6 +113,7 @@ typelibs_DATA = $(BUILT_GIRSOURCES:.gir=.typelib)
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--includedir=`$(PKG_CONFIG) --variable=girdir gstreamer-@GST_API_VERSION@` \
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--includedir=`$(PKG_CONFIG) --variable=girdir gstreamer-rtsp-@GST_API_VERSION@` \
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--includedir=`$(PKG_CONFIG) --variable=girdir gstreamer-sdp-@GST_API_VERSION@` \
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--includedir=`$(PKG_CONFIG) --variable=girdir gstreamer-net-@GST_API_VERSION@` \
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$(INTROSPECTION_COMPILER_OPTS) $< -o $(@F)
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CLEANFILES += $(BUILT_GIRSOURCES) $(typelibs_DATA)
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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: (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: (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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@ -29,7 +29,7 @@
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static GPrivate current_context;
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/**
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* gst_rtsp_context_get_current:
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* gst_rtsp_context_get_current: (skip)
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*
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* Get the current #GstRTSPContext. This object is retrieved from the
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* current thread that is handling the request for a client.
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@ -483,7 +483,7 @@ gst_rtsp_media_new (GstElement * element)
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*
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* Get the element that was used when constructing @media.
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*
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* Returns: a #GstElement. Unref after usage.
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* Returns: (transfer full): a #GstElement. Unref after usage.
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*/
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GstElement *
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gst_rtsp_media_get_element (GstRTSPMedia * media)
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@ -2000,7 +2000,7 @@ get_clock_unlocked (GstRTSPMedia * media)
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*
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* @media must be prepared before this method returns a valid clock object.
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*
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* Returns: the #GstClock used by @media. unref after usage.
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* Returns: (transfer full): the #GstClock used by @media. unref after usage.
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*/
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GstClock *
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gst_rtsp_media_get_clock (GstRTSPMedia * media)
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@ -2066,7 +2066,7 @@ not_prepared:
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* Get the #GstNetTimeProvider for the clock used by @media. The time provider
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* will listen on @address and @port for client time requests.
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*
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* Returns: the #GstNetTimeProvider of @media.
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* Returns: (transfer full): the #GstNetTimeProvider of @media.
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*/
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GstNetTimeProvider *
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gst_rtsp_media_get_time_provider (GstRTSPMedia * media, const gchar * address,
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@ -2131,7 +2131,8 @@ 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: a #GPtrArray of #GstRTSPStreamTransport pointers
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* @transports: (element-type GstRtspServer.RTSPStreamTransport): a #GPtrArray
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* 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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@ -221,7 +221,7 @@ has_prefix (DataItem * str, DataItem * prefix)
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* gst_rtsp_mount_points_match:
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* @mounts: a #GstRTSPMountPoints
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* @path: a mount point
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* @matched: the amount of @path matched
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* @matched: (out): the amount of @path matched
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*
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* Find the factory in @mounts that has the longest match with @path.
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*
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@ -119,7 +119,7 @@ guint gst_rtsp_server_attach (GstRTSPServer *serve
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/**
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* GstRTSPServerClientFilterFunc:
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* @server: a #GstRTSPServer object
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* @sess: a #GstRTSPClient in @server
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* @client: a #GstRTSPClient in @server
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* @user_data: user data that has been given to gst_rtsp_server_client_filter()
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*
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* This function will be called by the gst_rtsp_server_client_filter(). An
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@ -163,7 +163,7 @@ gst_rtsp_session_media_new (const gchar * path, GstRTSPMedia * media)
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* gst_rtsp_session_media_matches:
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* @media: a #GstRTSPSessionMedia
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* @path: a path
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* @matched: the amount of matched characters of @path
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* @matched: (out): the amount of matched characters of @path
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*
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* Check if the path of @media matches @path. It @path matches, the amount of
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* matched characters is returned in @matched.
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@ -264,7 +264,7 @@ gst_rtsp_session_release_media (GstRTSPSession * sess,
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* gst_rtsp_session_get_media:
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* @sess: a #GstRTSPSession
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* @path: the path for the media
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* @matched: the amount of matched characters
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* @matched: (out): the amount of matched characters
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*
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* Get the session media for @path. @matched will contain the number of matched
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* characters of @path.
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@ -312,7 +312,7 @@ gst_rtsp_stream_get_index (GstRTSPStream * stream)
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*
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* Get the srcpad associated with @stream.
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*
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* Return: the srcpad. Unref after usage.
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* Returns: (transfer full): the srcpad. Unref after usage.
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*/
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GstPad *
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gst_rtsp_stream_get_srcpad (GstRTSPStream * stream)
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@ -328,7 +328,7 @@ gst_rtsp_stream_get_srcpad (GstRTSPStream * stream)
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*
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* Get the control string to identify this stream.
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*
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* Return: the control string. free after usage.
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* Returns: (transfer full): the control string. free after usage.
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*/
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gchar *
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gst_rtsp_stream_get_control (GstRTSPStream * stream)
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@ -1076,7 +1076,7 @@ gst_rtsp_stream_get_server_port (GstRTSPStream * stream,
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*
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* Get the RTP session of this stream.
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*
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* Returns: The RTP session of this stream. Unref after usage.
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* Returns: (transfer full): The RTP session of this stream. Unref after usage.
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*/
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GObject *
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gst_rtsp_stream_get_rtpsession (GstRTSPStream * stream)
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