Fix the video buffer size should take rtt into consideration

BUG=webrtc:8010

Review-Url: https://codereview.webrtc.org/2980413002
Cr-Commit-Position: refs/heads/master@{#19285}

AUTHORS

@@ -61,6 +61,7 @@ Agora IO <*@agora.io>
 ARM Holdings <*@arm.com>
 BroadSoft Inc. <*@broadsoft.com>
 Google Inc. <*@google.com>
+Life On Air Inc. <*@lifeonair.com>
 Intel Corporation <*@intel.com>
 MIPS Technologies <*@mips.com>
 Mozilla Foundation <*@mozilla.com>

webrtc/modules/video_coding/frame_buffer2.cc

@@ -147,6 +147,8 @@ 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.
@@ -247,6 +249,11 @@ 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))

webrtc/modules/video_coding/frame_buffer2.h

@@ -74,6 +74,9 @@ class FrameBuffer {
   // return immediately.
   void Stop();
 
+  // Updates the RTT for jitter buffer estimation.
+  void UpdateRtt(int64_t rtt_ms);
+
  private:
   struct FrameKey {
     FrameKey() : picture_id(0), spatial_layer(0) {}

webrtc/video/video_receive_stream.cc

@@ -267,6 +267,7 @@ void VideoReceiveStream::Start() {
 
   frame_buffer_->Start();
   call_stats_->RegisterStatsObserver(&rtp_video_stream_receiver_);
+  call_stats_->RegisterStatsObserver(this);
 
   if (rtp_video_stream_receiver_.IsRetransmissionsEnabled() &&
       protected_by_fec) {
@@ -316,6 +317,7 @@ void VideoReceiveStream::Stop() {
   rtp_video_stream_receiver_.StopReceive();
 
   frame_buffer_->Stop();
+  call_stats_->DeregisterStatsObserver(this);
   call_stats_->DeregisterStatsObserver(&rtp_video_stream_receiver_);
   process_thread_->DeRegisterModule(&video_receiver_);
 
@@ -443,6 +445,10 @@ void VideoReceiveStream::OnCompleteFrame(
     rtp_video_stream_receiver_.FrameContinuous(last_continuous_pid);
 }
 
+void VideoReceiveStream::OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) {
+  frame_buffer_->UpdateRtt(max_rtt_ms);
+}
+
 int VideoReceiveStream::id() const {
   RTC_DCHECK_RUN_ON(&worker_thread_checker_);
   return config_.rtp.remote_ssrc;

webrtc/video/video_receive_stream.h

@@ -49,7 +49,8 @@ class VideoReceiveStream : public webrtc::VideoReceiveStream,
                            public NackSender,
                            public KeyFrameRequestSender,
                            public video_coding::OnCompleteFrameCallback,
-                           public Syncable {
+                           public Syncable,
+                           public CallStatsObserver {
  public:
   VideoReceiveStream(RtpStreamReceiverControllerInterface* receiver_controller,
                      int num_cpu_cores,
@@ -104,6 +105,9 @@ class VideoReceiveStream : public webrtc::VideoReceiveStream,
   void OnCompleteFrame(
       std::unique_ptr<video_coding::FrameObject> frame) override;
 
+  // Implements CallStatsObserver::OnRttUpdate
+  void OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) override;
+
   // Implements Syncable.
   int id() const override;
   rtc::Optional<Syncable::Info> GetInfo() const override;