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webrtc: Also remove rtcp_transport from the structure

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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1765>
This commit is contained in:
Olivier Crête 2020-11-04 17:06:02 -05:00 committed by GStreamer Merge Bot
parent c5d76d944e
commit 52c676546d
4 changed files with 2 additions and 14 deletions
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4
gst-libs/gst/webrtc/rtpreceiver.c
View file

@ -84,10 +84,6 @@ gst_webrtc_rtp_receiver_finalize (GObject * object)
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);
}

4
gst-libs/gst/webrtc/rtpreceiver.h
View file

@ -38,7 +38,6 @@ GType gst_webrtc_rtp_receiver_get_type(void);
/**
* 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
*
@ -50,9 +49,8 @@ struct _GstWebRTCRTPReceiver
{
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];
};

4
gst-libs/gst/webrtc/rtpsender.c
View file

@ -121,10 +121,6 @@ gst_webrtc_rtp_sender_finalize (GObject * object)
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);
}

4
gst-libs/gst/webrtc/rtpsender.h
View file

@ -38,7 +38,6 @@ GType gst_webrtc_rtp_sender_get_type(void);
/**
* 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)
*
@ -59,9 +58,8 @@ struct _GstWebRTCRTPSender
{
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;