Add usage counter for WebRTC jitter buffer origin trial.

Bug: chromium:904764
Change-Id: I3ad24d066a2c60328921867b559fdd5e1a2dad00
Reviewed-on: https://chromium-review.googlesource.com/c/1346092Reviewed-by: Guido Urdaneta <guidou@chromium.org>
Commit-Queue: Jonas Olsson <jonasolsson@chromium.org>
Cr-Commit-Position: refs/heads/master@{#610017}

third_party/blink/renderer/modules/peerconnection/rtc_peer_connection.cc

@@ -415,6 +415,7 @@ webrtc::PeerConnectionInterface::RTCConfiguration ParseConfiguration(
       configuration->iceCandidatePoolSize();
 
   if (configuration->hasRtcAudioJitterBufferMaxPackets()) {
+    UseCounter::Count(context, WebFeature::kRTCMaxAudioBufferSize);
     web_configuration.audio_jitter_buffer_max_packets =
         static_cast<int>(configuration->rtcAudioJitterBufferMaxPackets());
   }