add jitter-buffer-fast-accelerate for webrtc configuration under origin trial

Experiment discussion:
https://groups.google.com/a/chromium.org/forum/#!topic/blink-dev/hE2B1iItPDk

Bug: 908272
Change-Id: I97a5d26996469d33b75afa4babf40e54eb830e6a
Reviewed-on: https://chromium-review.googlesource.com/c/1350174
Commit-Queue: Ruslan Burakov <kuddai@google.com>
Reviewed-by: Guido Urdaneta <guidou@chromium.org>
Cr-Commit-Position: refs/heads/master@{#611043}

third_party/blink/renderer/modules/peerconnection/rtc_configuration.idl

@@ -49,5 +49,6 @@ dictionary RTCConfiguration {
     // Nonstandard, added for Unified Plan migration
     [RuntimeEnabled=RTCUnifiedPlan] SdpSemantics sdpSemantics;
     [OriginTrialEnabled=RtcAudioJitterBufferMaxPackets] long rtcAudioJitterBufferMaxPackets;
+    [OriginTrialEnabled=RtcAudioJitterBufferMaxPackets] boolean rtcAudioJitterBufferFastAccelerate;
 };
 

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

@@ -420,6 +420,12 @@ webrtc::PeerConnectionInterface::RTCConfiguration ParseConfiguration(
         static_cast<int>(configuration->rtcAudioJitterBufferMaxPackets());
   }
 
+  if (configuration->hasRtcAudioJitterBufferFastAccelerate()) {
+    UseCounter::Count(context, WebFeature::kRTCMaxAudioBufferSize);
+    web_configuration.audio_jitter_buffer_fast_accelerate =
+        configuration->hasRtcAudioJitterBufferFastAccelerate();
+  }
+
   return web_configuration;
 }
 
@@ -1255,6 +1261,8 @@ RTCConfiguration* RTCPeerConnection::getConfiguration(
         webrtc_configuration.audio_jitter_buffer_max_packets;
     result->setRtcAudioJitterBufferMaxPackets(
         static_cast<int32_t>(audio_jitter_buffer_max_packets));
+    result->setRtcAudioJitterBufferFastAccelerate(
+        webrtc_configuration.audio_jitter_buffer_fast_accelerate);
   }
 
   return result;

third_party/blink/web_tests/http/tests/origin_trials/webexposed/rtcaudiojitterbuffermaxpackets-origin-trial-enabled.html

@@ -19,4 +19,15 @@ test(t => {
   );
 }, 'rtcAudioJitterBufferMaxPackets property in Origin-Trial enabled document.');
 
+test(t => {
+  var peerconnection = new RTCPeerConnection({
+      rtcAudioJitterBufferFastAccelerate: true
+  });
+  var configuration = peerconnection.getConfiguration();
+  assert_true(
+      configuration.rtcAudioJitterBufferFastAccelerate,
+      'rtcAudioJitterBufferFastAccelerate equals passed values'
+  );
+}, 'rtcAudioJitterBufferFastAccelerate property in Origin-Trial enabled document.');
+
 </script>