Correctly handle missing webrtc-audio-processing

1 files changed, 3 insertions, 2 deletions

diff --git a/plugins/rtp/src/plugin.vala b/plugins/rtp/src/plugin.vala
index e3d5ee41..f575a7d0 100644
--- a/plugins/rtp/src/plugin.vala
+++ b/plugins/rtp/src/plugin.vala
@@ -8,7 +8,7 @@ public class Dino.Plugins.Rtp.Plugin : RootInterface, VideoCallPlugin, Object {
     public Gst.DeviceMonitor device_monitor { get; private set; }
     public Gst.Pipeline pipe { get; private set; }
     public Gst.Bin rtpbin { get; private set; }
-    public EchoProbe echoprobe { get; private set; }
+    public Gst.Element echoprobe { get; private set; }
 
     private Gee.List<Stream> streams = new ArrayList<Stream>();
     private Gee.List<Device> devices = new ArrayList<Device>();
@@ -71,10 +71,11 @@ public class Dino.Plugins.Rtp.Plugin : RootInterface, VideoCallPlugin, Object {
         rtpbin.connect("signal::request-pt-map", request_pt_map, this);
         pipe.add(rtpbin);
 
+#if WITH_VOICE_PROCESSOR
         // Audio echo probe
-//        echoprobe = Gst.ElementFactory.make("webrtcechoprobe", "echo-probe");
         echoprobe = new EchoProbe();
         if (echoprobe != null) pipe.add(echoprobe);
+#endif
 
         // Pipeline
         pipe.auto_flush_bus = true;