webrtc: Remove unused parameter from rtpsender constructor

[platform/upstream/gstreamer.git] / gst-libs / gst / webrtc / rtpsender.c

diff --git a/gst-libs/gst/webrtc/rtpsender.c b/gst-libs/gst/webrtc/rtpsender.c
index 4dc6dec..da743f3 100644 (file)
--- a/gst-libs/gst/webrtc/rtpsender.c
+++ b/gst-libs/gst/webrtc/rtpsender.c
@@ -139,7 +139,7 @@ gst_webrtc_rtp_sender_init (GstWebRTCRTPSender * webrtc)
 }
 
 GstWebRTCRTPSender *
-gst_webrtc_rtp_sender_new (GArray * send_encodings /* FIXME */ )
+gst_webrtc_rtp_sender_new (void)
 {
   return g_object_new (GST_TYPE_WEBRTC_RTP_SENDER, NULL);
 }