Add rtcAudioJitterBufferMinDelayMs WebRTC configuration option under origin trial

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

Bug: chromium:904764
Change-Id: Ifaa3df3dffaa9266f55a36840b9f26b323e50478
Reviewed-on: https://chromium-review.googlesource.com/c/1352752
Commit-Queue: Jakob Ivarsson <jakobi@google.com>
Reviewed-by: Guido Urdaneta <guidou@chromium.org>
Cr-Commit-Position: refs/heads/master@{#611713}

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

@@ -50,5 +50,6 @@ dictionary RTCConfiguration {
     [RuntimeEnabled=RTCUnifiedPlan] SdpSemantics sdpSemantics;
     [OriginTrialEnabled=RtcAudioJitterBufferMaxPackets] long rtcAudioJitterBufferMaxPackets;
     [OriginTrialEnabled=RtcAudioJitterBufferMaxPackets] boolean rtcAudioJitterBufferFastAccelerate;
+    [OriginTrialEnabled=RtcAudioJitterBufferMaxPackets] long rtcAudioJitterBufferMinDelayMs;
 };
 

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

@@ -426,6 +426,12 @@ webrtc::PeerConnectionInterface::RTCConfiguration ParseConfiguration(
         configuration->hasRtcAudioJitterBufferFastAccelerate();
   }
 
+  if (configuration->hasRtcAudioJitterBufferMinDelayMs()) {
+    UseCounter::Count(context, WebFeature::kRTCMaxAudioBufferSize);
+    web_configuration.audio_jitter_buffer_min_delay_ms =
+        static_cast<int>(configuration->rtcAudioJitterBufferMinDelayMs());
+  }
+
   return web_configuration;
 }
 
@@ -1394,6 +1400,10 @@ RTCConfiguration* RTCPeerConnection::getConfiguration(
         static_cast<int32_t>(audio_jitter_buffer_max_packets));
     result->setRtcAudioJitterBufferFastAccelerate(
         webrtc_configuration.audio_jitter_buffer_fast_accelerate);
+    int audio_jitter_buffer_min_delay_ms =
+        webrtc_configuration.audio_jitter_buffer_min_delay_ms;
+    result->setRtcAudioJitterBufferMinDelayMs(
+        static_cast<int32_t>(audio_jitter_buffer_min_delay_ms));
   }
 
   return result;

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

@@ -9,10 +9,10 @@ generate_token.py http://127.0.0.1:8000 RtcAudioJitterBufferMaxPackets --expire-
 <script src="/resources/testharnessreport.js"></script>
 <script>
 test(t => {
-  var peerconnection = new RTCPeerConnection({
+  let peerconnection = new RTCPeerConnection({
       rtcAudioJitterBufferMaxPackets: 79
   });
-  var configuration = peerconnection.getConfiguration();
+  let configuration = peerconnection.getConfiguration();
   assert_true(
       configuration.rtcAudioJitterBufferMaxPackets === 79,
       'rtcAudioJitterBufferMaxPackets equals passed values'
@@ -20,14 +20,25 @@ test(t => {
 }, 'rtcAudioJitterBufferMaxPackets property in Origin-Trial enabled document.');
 
 test(t => {
-  var peerconnection = new RTCPeerConnection({
+  let peerconnection = new RTCPeerConnection({
       rtcAudioJitterBufferFastAccelerate: true
   });
-  var configuration = peerconnection.getConfiguration();
+  let configuration = peerconnection.getConfiguration();
   assert_true(
       configuration.rtcAudioJitterBufferFastAccelerate,
       'rtcAudioJitterBufferFastAccelerate equals passed values'
   );
 }, 'rtcAudioJitterBufferFastAccelerate property in Origin-Trial enabled document.');
 
+test(t => {
+  let peerconnection = new RTCPeerConnection({
+      rtcAudioJitterBufferMinDelayMs: 20
+  });
+  let configuration = peerconnection.getConfiguration();
+  assert_true(
+      configuration.rtcAudioJitterBufferMinDelayMs === 20,
+      'rtcAudioJitterBufferMinDelayMs equals passed values'
+  );
+}, 'rtcAudioJitterBufferMinDelayMs property in Origin-Trial enabled document.');
+
 </script>