examples/webrtc: add some missing unrefs

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2958>

diff --git a/subprojects/gst-examples/webrtc/multiparty-sendrecv/gst/mp-webrtc-sendrecv.c b/subprojects/gst-examples/webrtc/multiparty-sendrecv/gst/mp-webrtc-sendrecv.c
index 5047510..80e8c0e 100644 (file)
--- a/subprojects/gst-examples/webrtc/multiparty-sendrecv/gst/mp-webrtc-sendrecv.c
+++ b/subprojects/gst-examples/webrtc/multiparty-sendrecv/gst/mp-webrtc-sendrecv.c
@@ -282,6 +282,7 @@ on_offer_created (GstPromise * promise, const gchar * peer_id)
   /* Send offer to peer */
   send_room_peer_sdp (offer, peer_id);
   gst_webrtc_session_description_free (offer);
+  gst_object_unref (webrtc);
 }
 
 static void
@@ -388,6 +389,9 @@ add_peer_to_pipeline (const gchar * peer_id, gboolean offer)
   g_assert_true (ret);
   ret = gst_element_sync_state_with_parent (webrtc);
   g_assert_true (ret);
+
+  gst_object_unref (q);
+  gst_object_unref (webrtc)
 }
 
 static void
@@ -606,6 +610,7 @@ on_answer_created (GstPromise * promise, const gchar * peer_id)
   /* Send offer to peer */
   send_room_peer_sdp (answer, peer_id);
   gst_webrtc_session_description_free (answer);
+  gst_object_unref (webrtc);
 
   app_state = ROOM_CALL_STARTED;
 }