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authorMarvin W <git@larma.de>2021-05-01 16:00:37 +0200
committerMarvin W <git@larma.de>2021-05-01 16:00:37 +0200
commitd388525fc69ab688a90f19d2d2499e0f6f10f573 (patch)
tree74db651fe723e16a12fce3dd25ae8d7d833b247d /plugins
parent23ffd37dded3bf872e42d7a00727ab3c4d105a97 (diff)
downloaddino-d388525fc69ab688a90f19d2d2499e0f6f10f573.tar.gz
dino-d388525fc69ab688a90f19d2d2499e0f6f10f573.zip
Correctly handle missing webrtc-audio-processing
Diffstat (limited to 'plugins')
-rw-r--r--plugins/rtp/src/device.vala4
-rw-r--r--plugins/rtp/src/plugin.vala5
2 files changed, 6 insertions, 3 deletions
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;