From d388525fc69ab688a90f19d2d2499e0f6f10f573 Mon Sep 17 00:00:00 2001
From: Marvin W <git@larma.de>
Date: Sat, 1 May 2021 16:00:37 +0200
Subject: Correctly handle missing webrtc-audio-processing

---
 plugins/rtp/src/device.vala | 4 +++-
 plugins/rtp/src/plugin.vala | 5 +++--
 2 files changed, 6 insertions(+), 3 deletions(-)

(limited to 'plugins/rtp')

diff --git a/plugins/rtp/src/device.vala b/plugins/rtp/src/device.vala
index f8894502..3c650ad6 100644
--- a/plugins/rtp/src/device.vala
+++ b/plugins/rtp/src/device.vala
@@ -132,12 +132,14 @@ public class Dino.Plugins.Rtp.Device : MediaDevice, Object {
             filter.@set("caps", get_best_caps());
             pipe.add(filter);
             element.link(filter);
+#if WITH_VOICE_PROCESSOR
             if (media == "audio" && plugin.echoprobe != null) {
-                dsp = new VoiceProcessor(plugin.echoprobe, element as Gst.Audio.StreamVolume);
+                dsp = new VoiceProcessor(plugin.echoprobe as EchoProbe, element as Gst.Audio.StreamVolume);
                 dsp.name = @"dsp_$id";
                 pipe.add(dsp);
                 filter.link(dsp);
             }
+#endif
             tee = Gst.ElementFactory.make("tee", @"tee_$id");
             tee.@set("allow-not-linked", true);
             pipe.add(tee);
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;
-- 
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