webrtc: Also remove rtcp_transport from the structure

Change-Id: Id63ee4d28b7ebaf72004af554d16f711d6d72ee9
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1765>
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

diff --git a/gst-libs/gst/webrtc/rtpreceiver.c b/gst-libs/gst/webrtc/rtpreceiver.c
index d20239a..fb8f6e6 100644 (file)
--- a/gst-libs/gst/webrtc/rtpreceiver.c
+++ b/gst-libs/gst/webrtc/rtpreceiver.c
@@ -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);
 }
 

diff --git a/gst-libs/gst/webrtc/rtpreceiver.h b/gst-libs/gst/webrtc/rtpreceiver.h
index c2fa210..a02bad1 100644 (file)
--- a/gst-libs/gst/webrtc/rtpreceiver.h
+++ b/gst-libs/gst/webrtc/rtpreceiver.h
@@ -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];
 };

diff --git a/gst-libs/gst/webrtc/rtpsender.c b/gst-libs/gst/webrtc/rtpsender.c
index 40d871c..388c48d 100644 (file)
--- a/gst-libs/gst/webrtc/rtpsender.c
+++ b/gst-libs/gst/webrtc/rtpsender.c
@@ -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);
 }
 

diff --git a/gst-libs/gst/webrtc/rtpsender.h b/gst-libs/gst/webrtc/rtpsender.h
index 5fa9fe8..44eee81 100644 (file)
--- a/gst-libs/gst/webrtc/rtpsender.h
+++ b/gst-libs/gst/webrtc/rtpsender.h
@@ -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;