Update AvgCounter to have the ability to include last period metric for subsequent intervals withou…

webrtc/video/send_delay_stats.cc

 /*
  *  Copyright (c) 2016 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.
  */
 
 #include "webrtc/video/send_delay_stats.h"
 
+#include <utility>
+
 #include "webrtc/base/logging.h"
 #include "webrtc/system_wrappers/include/metrics.h"
 
 namespace webrtc {
 namespace {
 // Packet with a larger delay are removed and excluded from the delay stats.
 // Set to larger than max histogram delay which is 10000.
 const int64_t kMaxSentPacketDelayMs = 11000;
 const size_t kMaxPacketMapSize = 2000;
 
@@ skipping 31 matching lines @@
     return;
   for (const auto& ssrc : config.rtp.ssrcs)
     ssrcs_.insert(ssrc);
 }
 
 AvgCounter* SendDelayStats::GetSendDelayCounter(uint32_t ssrc) {
   const auto& it = send_delay_counters_.find(ssrc);
   if (it != send_delay_counters_.end())
     return it->second.get();
 
-  AvgCounter* counter = new AvgCounter(clock_, nullptr);
+  AvgCounter* counter = new AvgCounter(clock_, nullptr, false);
   send_delay_counters_[ssrc].reset(counter);
   return counter;
 }
 
 void SendDelayStats::OnSendPacket(uint16_t packet_id,
                                   int64_t capture_time_ms,
                                   uint32_t ssrc) {
   // Packet sent to transport.
   rtc::CritScope lock(&crit_);
   if (ssrcs_.find(ssrc) == ssrcs_.end())
@@ skipping 33 matching lines @@
     auto it = packets->begin();
     if (now - it->second.capture_time_ms < kMaxSentPacketDelayMs)
       break;
 
     packets->erase(it);
     ++num_old_packets_;
   }
 }
 
 }  // namespace webrtc