webrtc: Remove APIs to set transport on sender/receiver

They're not not used ever.

Change-Id: Ia90b7edfd32571bc018cb0cb2e5c1a8132da79e7
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 dd8c5a9..d20239a 100644 (file)
--- a/gst-libs/gst/webrtc/rtpreceiver.c
+++ b/gst-libs/gst/webrtc/rtpreceiver.c
@@ -53,21 +53,6 @@ enum
 
 //static guint gst_webrtc_rtp_receiver_signals[LAST_SIGNAL] = { 0 };
 
-void
-gst_webrtc_rtp_receiver_set_transport (GstWebRTCRTPReceiver * receiver,
-    GstWebRTCDTLSTransport * transport)
-{
-  g_return_if_fail (GST_IS_WEBRTC_RTP_RECEIVER (receiver));
-  g_return_if_fail (GST_IS_WEBRTC_DTLS_TRANSPORT (transport));
-
-  GST_OBJECT_LOCK (receiver);
-  gst_object_replace ((GstObject **) & receiver->transport,
-      GST_OBJECT (transport));
-  gst_object_replace ((GstObject **) & receiver->rtcp_transport,
-      GST_OBJECT (transport));
-  GST_OBJECT_UNLOCK (receiver);
-}
-
 static void
 gst_webrtc_rtp_receiver_set_property (GObject * object, guint prop_id,
     const GValue * value, GParamSpec * pspec)

diff --git a/gst-libs/gst/webrtc/rtpreceiver.h b/gst-libs/gst/webrtc/rtpreceiver.h
index c4260f7..c2fa210 100644 (file)
--- a/gst-libs/gst/webrtc/rtpreceiver.h
+++ b/gst-libs/gst/webrtc/rtpreceiver.h
@@ -66,9 +66,6 @@ struct _GstWebRTCRTPReceiverClass
 
 GST_WEBRTC_API
 GstWebRTCRTPReceiver *      gst_webrtc_rtp_receiver_new                 (void);
-GST_WEBRTC_API
-void                        gst_webrtc_rtp_receiver_set_transport       (GstWebRTCRTPReceiver * receiver,
-                                                                         GstWebRTCDTLSTransport * transport);
 
 G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstWebRTCRTPReceiver, gst_object_unref)
 

diff --git a/gst-libs/gst/webrtc/rtpsender.c b/gst-libs/gst/webrtc/rtpsender.c
index 3c533bf..40d871c 100644 (file)
--- a/gst-libs/gst/webrtc/rtpsender.c
+++ b/gst-libs/gst/webrtc/rtpsender.c
@@ -56,21 +56,6 @@ enum
 
 //static guint gst_webrtc_rtp_sender_signals[LAST_SIGNAL] = { 0 };
 
-void
-gst_webrtc_rtp_sender_set_transport (GstWebRTCRTPSender * sender,
-    GstWebRTCDTLSTransport * transport)
-{
-  g_return_if_fail (GST_IS_WEBRTC_RTP_SENDER (sender));
-  g_return_if_fail (GST_IS_WEBRTC_DTLS_TRANSPORT (transport));
-
-  GST_OBJECT_LOCK (sender);
-  gst_object_replace ((GstObject **) & sender->transport,
-      GST_OBJECT (transport));
-  gst_object_replace ((GstObject **) & sender->rtcp_transport,
-      GST_OBJECT (transport));
-  GST_OBJECT_UNLOCK (sender);
-}
-
 /**
  * gst_webrtc_rtp_sender_set_priority:
  * @sender: a #GstWebRTCRTPSender

diff --git a/gst-libs/gst/webrtc/rtpsender.h b/gst-libs/gst/webrtc/rtpsender.h
index 6c42f2d..5fa9fe8 100644 (file)
--- a/gst-libs/gst/webrtc/rtpsender.h
+++ b/gst-libs/gst/webrtc/rtpsender.h
@@ -80,12 +80,6 @@ GST_WEBRTC_API
 GstWebRTCRTPSender *        gst_webrtc_rtp_sender_new                   (void);
 
 GST_WEBRTC_API
-void                        gst_webrtc_rtp_sender_set_transport         (GstWebRTCRTPSender * sender,
-                                                                         GstWebRTCDTLSTransport * transport);
-GST_WEBRTC_API
-void                        gst_webrtc_rtp_sender_set_rtcp_transport    (GstWebRTCRTPSender * sender,
-                                                                         GstWebRTCDTLSTransport * transport);
-GST_WEBRTC_API
 void                        gst_webrtc_rtp_sender_set_priority          (GstWebRTCRTPSender *sender,
                                                                          GstWebRTCPriorityType priority);