Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1765>
gst_object_unref (webrtc->transport);
webrtc->transport = NULL;
- if (webrtc->rtcp_transport)
- gst_object_unref (webrtc->rtcp_transport);
- webrtc->rtcp_transport = NULL;
-
G_OBJECT_CLASS (parent_class)->finalize (object);
}
/**
* GstWebRTCRTPReceiver:
* @transport: The transport for RTP packets
- * @rtcp_transport: The transport for RTCP packets without rtcp-mux
*
* An object to track the receiving aspect of the stream
*
{
GstObject parent;
- /* The MediStreamTrack is represented by the stream and is output into @transport/@rtcp_transport as necessary */
+ /* The MediStreamTrack is represented by the stream and is output into @transport as necessary */
GstWebRTCDTLSTransport *transport;
- GstWebRTCDTLSTransport *rtcp_transport;
gpointer _padding[GST_PADDING];
};
gst_object_unref (sender->transport);
sender->transport = NULL;
- if (sender->rtcp_transport)
- gst_object_unref (sender->rtcp_transport);
- sender->rtcp_transport = NULL;
-
G_OBJECT_CLASS (parent_class)->finalize (object);
}
/**
* GstWebRTCRTPSender:
* @transport: The transport for RTP packets
- * @rtcp_transport: The transport for RTCP packets without rtcp-mux
* @send_encodings: Unused
* @priority: The priority of the stream (Since: 1.20)
*
{
GstObject parent;
- /* The MediStreamTrack is represented by the stream and is output into @transport/@rtcp_transport as necessary */
+ /* The MediStreamTrack is represented by the stream and is output into @transport as necessary */
GstWebRTCDTLSTransport *transport;
- GstWebRTCDTLSTransport *rtcp_transport;
GArray *send_encodings;
GstWebRTCPriorityType priority;