Report timing frames info in GetStats.

webrtc/video/receive_statistics_proxy.cc

 /*
  *  Copyright (c) 2013 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 388 matching lines @@
   // us from ever correctly displaying frame rate of 0.
   int64_t now_ms = clock_->TimeInMilliseconds();
   UpdateFramerate(now_ms);
   stats_.render_frame_rate = renders_fps_estimator_.Rate(now_ms).value_or(0);
   stats_.decode_frame_rate = decode_fps_estimator_.Rate(now_ms).value_or(0);
   stats_.total_bitrate_bps =
       static_cast<int>(total_byte_tracker_.ComputeRate() * 8);
   return stats_;
 }
 
+rtc::Optional<TimingFrameInfo> ReceiveStatisticsProxy::GetTimingFrameInfo() {

[sprang_webrtc, 2017/07/06 09:13:08]

nit: Rename this method so it's clear we're mutating the state, for instance
GetAndResetTimingFrameInfo().

[ilnik, 2017/07/06 09:28:37]

Done.

+  rtc::CritScope lock(&crit_);
+  rtc::Optional<TimingFrameInfo> info = timing_frame_info_;
+  // Reset reported value to empty, so it will be always
+  // overwritten in |OnTimingFrameInfoUpdated|, thus allowing to store new
+  // value instead of reported one.
+  timing_frame_info_.reset();
+  return info;
+}
+
 void ReceiveStatisticsProxy::OnIncomingPayloadType(int payload_type) {
   rtc::CritScope lock(&crit_);
   stats_.current_payload_type = payload_type;
 }
 
 void ReceiveStatisticsProxy::OnDecoderImplementationName(
     const char* implementation_name) {
   rtc::CritScope lock(&crit_);
   stats_.decoder_implementation_name = implementation_name;
 }
@@ skipping 38 matching lines @@
                        "target_delay_ms", target_delay_ms,
                        "ssrc", stats_.ssrc);
   TRACE_EVENT_INSTANT2("webrtc_stats", "WebRTC.Video.JitterBufferDelayInMs",
                        "jitter_buffer_ms", jitter_buffer_ms,
                        "ssrc", stats_.ssrc);
   TRACE_EVENT_INSTANT2("webrtc_stats", "WebRTC.Video.RenderDelayInMs",
                        "render_delay_ms", render_delay_ms,
                        "ssrc", stats_.ssrc);
 }
 
+void ReceiveStatisticsProxy::OnTimingFrameInfoUpdated(
+    const TimingFrameInfo& info) {
+  rtc::CritScope lock(&crit_);
+  // Only the frame which was processed the longest since the last
+  // GetStats() call is reported. Therefore, only single 'longest' frame is
+  // stored.
+  if (!timing_frame_info_ || info.IsLongerThan(*timing_frame_info_)) {
+    timing_frame_info_.emplace(info);
+  }
+}
+
 void ReceiveStatisticsProxy::RtcpPacketTypesCounterUpdated(
     uint32_t ssrc,
     const RtcpPacketTypeCounter& packet_counter) {
   rtc::CritScope lock(&crit_);
   if (stats_.ssrc != ssrc)
     return;
   stats_.rtcp_packet_type_counts = packet_counter;
 }
 
 void ReceiveStatisticsProxy::StatisticsUpdated(
@@ skipping 201 matching lines @@
   sum = 0;
 }
 
 void ReceiveStatisticsProxy::OnRttUpdate(int64_t avg_rtt_ms,
                                          int64_t max_rtt_ms) {
   rtc::CritScope lock(&crit_);
   avg_rtt_ms_ = avg_rtt_ms;
 }
 
 }  // namespace webrtc