Fix the video buffer size should take rtt into consideration

webrtc/modules/video_coding/frame_buffer2.cc

Index: webrtc/modules/video_coding/frame_buffer2.cc
diff --git a/webrtc/modules/video_coding/frame_buffer2.cc b/webrtc/modules/video_coding/frame_buffer2.cc
index b8a80e1f681cc77ae634ef7faf879d3b14ad156d..2540ae594148795ce62280e9df31e525348f988b 100644
--- a/webrtc/modules/video_coding/frame_buffer2.cc
+++ b/webrtc/modules/video_coding/frame_buffer2.cc
@@ -142,6 +142,8 @@ FrameBuffer::ReturnReason FrameBuffer::NextFrame(
         float rtt_mult = protection_mode_ == kProtectionNackFEC ? 0.0 : 1.0;
         timing_->SetJitterDelay(jitter_estimator_->GetJitterEstimate(rtt_mult));
         timing_->UpdateCurrentDelay(frame->RenderTime(), now_ms);
+      } else {
+        jitter_estimator_->FrameNacked();
       }
 
       // Gracefully handle bad RTP timestamps and render time issues.
@@ -242,6 +244,11 @@ void FrameBuffer::Stop() {
   new_continuous_frame_event_.Set();
 }
 
+void FrameBuffer::UpdateRtt(int64_t rtt_ms) {
+  rtc::CritScope lock(&crit_);
+  jitter_estimator_->UpdateRtt(rtt_ms);
+}
+
 bool FrameBuffer::ValidReferences(const FrameObject& frame) const {
   for (size_t i = 0; i < frame.num_references; ++i) {
     if (AheadOrAt(frame.references[i], frame.picture_id))