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Correctly handle missing webrtc-audio-processing

This commit is contained in:
Marvin W 2021-05-01 16:00:37 +02:00
parent 23ffd37dde
commit d388525fc6
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GPG Key ID: 072E9235DB996F2A
3 changed files with 7 additions and 4 deletions
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2
.github/workflows/build.yml vendored
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@ -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

4
plugins/rtp/src/device.vala
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@ -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);

5
plugins/rtp/src/plugin.vala
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@ -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;