Add delay metric (includes network delay (rtt/2) + jitter delay + decode time +

webrtc/video_engine/vie_channel.cc

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 97 matching lines @@
       paced_sender_(paced_sender),
       packet_router_(packet_router),
       bandwidth_observer_(bandwidth_observer),
       transport_feedback_observer_(transport_feedback_observer),
       nack_history_size_sender_(kSendSidePacketHistorySize),
       max_nack_reordering_threshold_(kMaxPacketAgeToNack),
       pre_render_callback_(NULL),
       report_block_stats_sender_(new ReportBlockStats()),
       time_of_first_rtt_ms_(-1),
       rtt_sum_ms_(0),
+      last_rtt_ms_(0),
       num_rtts_(0),
       rtp_rtcp_modules_(
           CreateRtpRtcpModules(!sender,
                                vie_receiver_.GetReceiveStatistics(),
                                transport,
                                sender ? intra_frame_observer_ : nullptr,
                                sender ? bandwidth_observer_.get() : nullptr,
                                transport_feedback_observer_,
                                rtt_stats_,
                                &rtcp_packet_type_counter_observer_,
@@ skipping 933 matching lines @@
                                  int current_delay_ms,
                                  int target_delay_ms,
                                  int jitter_buffer_ms,
                                  int min_playout_delay_ms,
                                  int render_delay_ms) {
   CriticalSectionScoped cs(crit_.get());
   if (!receive_stats_callback_)
     return;
   receive_stats_callback_->OnDecoderTiming(
       decode_ms, max_decode_ms, current_delay_ms, target_delay_ms,
-      jitter_buffer_ms, min_playout_delay_ms, render_delay_ms);
+      jitter_buffer_ms, min_playout_delay_ms, render_delay_ms, last_rtt_ms_);
 }
 
 int32_t ViEChannel::RequestKeyFrame() {
   return rtp_rtcp_modules_[0]->RequestKeyFrame();
 }
 
 int32_t ViEChannel::SliceLossIndicationRequest(
   const uint64_t picture_id) {
   return rtp_rtcp_modules_[0]->SendRTCPSliceLossIndication(
       static_cast<uint8_t>(picture_id));
@@ skipping 13 matching lines @@
   return true;
 }
 
 void ViEChannel::OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) {
   vcm_->SetReceiveChannelParameters(max_rtt_ms);
 
   CriticalSectionScoped cs(crit_.get());
   if (time_of_first_rtt_ms_ == -1)
     time_of_first_rtt_ms_ = Clock::GetRealTimeClock()->TimeInMilliseconds();
   rtt_sum_ms_ += avg_rtt_ms;
+  last_rtt_ms_ = avg_rtt_ms;
   ++num_rtts_;
 }
 
 int ViEChannel::ProtectionRequest(const FecProtectionParams* delta_fec_params,
                                   const FecProtectionParams* key_fec_params,
                                   uint32_t* video_rate_bps,
                                   uint32_t* nack_rate_bps,
                                   uint32_t* fec_rate_bps) {
   *video_rate_bps = 0;
   *nack_rate_bps = 0;
@@ skipping 138 matching lines @@
   CriticalSectionScoped cs(crit_.get());
   receive_stats_callback_ = receive_statistics_proxy;
 }
 
 void ViEChannel::SetIncomingVideoStream(
     IncomingVideoStream* incoming_video_stream) {
   CriticalSectionScoped cs(crit_.get());
   incoming_video_stream_ = incoming_video_stream;
 }
 }  // namespace webrtc