From: Sangchul Lee <sc11.lee@samsung.com>
Date: Fri, 31 May 2024 02:46:37 +0000 (+0900)
Subject: webrtcbin: Remove incomplete codes
X-Git-Tag: accepted/tizen/unified/20240611.123015~1
X-Git-Url: http://review.tizen.org/git/?a=commitdiff_plain;h=refs%2Fchanges%2F44%2F311944%2F3;p=platform%2Fupstream%2Fgstreamer.git

webrtcbin: Remove incomplete codes

It turns out to have misbehavior when making sdp description
in case of renegotiation. So it has been removed.

Change-Id: Ibc3b3052c404eae5bcd988fade20e14dd40771d3
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>
---

diff --git a/packaging/gstreamer.spec b/packaging/gstreamer.spec
index 7c9f713..70f3172 100644
--- a/packaging/gstreamer.spec
+++ b/packaging/gstreamer.spec
@@ -62,7 +62,7 @@
 
 Name:           %{_name}
 Version:        1.22.8
-Release:        8
+Release:        9
 Summary:        Streaming-Media Framework Runtime
 License:        LGPL-2.0+
 Group:          Multimedia/Framework
diff --git a/subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c b/subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c
index d50d997..8f22462 100644
--- a/subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c
+++ b/subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c
@@ -2473,16 +2473,6 @@ _create_webrtc_transceiver (GstWebRTCBin * webrtc,
   return trans;
 }
 
-#ifdef TIZEN_FEATURE_WEBRTC_MODIFICATION
-static void
-_remove_webrtc_transceiver (GstWebRTCBin * webrtc,
-    GstWebRTCRTPTransceiver * trans)
-{
-  g_ptr_array_remove (webrtc->priv->transceivers, trans);
-  gst_object_unref (trans);
-}
-#endif
-
 static TransportStream *
 _create_transport_channel (GstWebRTCBin * webrtc, guint session_id)
 {
@@ -8350,11 +8340,7 @@ gst_webrtc_bin_release_pad (GstElement * element, GstPad * pad)
    * a possibly dead transceiver */
   PC_LOCK (webrtc);
   if (webrtc_pad->trans)
-#ifdef TIZEN_FEATURE_WEBRTC_MODIFICATION
-    _remove_webrtc_transceiver (webrtc, webrtc_pad->trans);
-#else
     gst_object_unref (webrtc_pad->trans);
-#endif
   webrtc_pad->trans = NULL;
   gst_caps_replace (&webrtc_pad->received_caps, NULL);
   PC_UNLOCK (webrtc);