Correctly handle missing webrtc-audio-processing

.github/workflows/build.yml

@@ -6,7 +6,7 @@ jobs:
     steps:
       - uses: actions/checkout@v2
       - run: sudo apt-get update
-      - run: sudo apt-get install -y build-essential gettext cmake valac libgee-0.8-dev libsqlite3-dev libgtk-3-dev libnotify-dev libgpgme-dev libsoup2.4-dev libgcrypt20-dev libqrencode-dev libgspell-1-dev libnice-dev libgstreamer1.0-dev libgstreamer-plugins-base1.0-dev libsrtp2-dev
+      - run: sudo apt-get install -y build-essential gettext cmake valac libgee-0.8-dev libsqlite3-dev libgtk-3-dev libnotify-dev libgpgme-dev libsoup2.4-dev libgcrypt20-dev libqrencode-dev libgspell-1-dev libnice-dev libgstreamer1.0-dev libgstreamer-plugins-base1.0-dev libsrtp2-dev libwebrtc-audio-processing-dev
       - run: ./configure --with-tests --with-libsignal-in-tree
       - run: make
       - run: build/xmpp-vala-test

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);

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;