Start integrating StatsCounter class.

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 "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;
 
 // Limit for the maximum number of streams to calculate stats for.
 const size_t kMaxSsrcMapSize = 50;
-const int kMinRequiredSamples = 200;
+const int kMinRequiredPeriodicSamples = 5;
 }  // namespace
 
 SendDelayStats::SendDelayStats(Clock* clock)
     : clock_(clock), num_old_packets_(0), num_skipped_packets_(0) {}
 
 SendDelayStats::~SendDelayStats() {
   if (num_old_packets_ > 0 || num_skipped_packets_ > 0) {
     LOG(LS_WARNING) << "Delay stats: number of old packets " << num_old_packets_
                     << ", skipped packets " << num_skipped_packets_
                     << ". Number of streams " << send_delay_counters_.size();
   }
   UpdateHistograms();
 }
 
 void SendDelayStats::UpdateHistograms() {
   rtc::CritScope lock(&crit_);
   for (const auto& it : send_delay_counters_) {
-    int send_delay_ms = it.second.Avg(kMinRequiredSamples);
-    if (send_delay_ms != -1) {
+    AggregatedStats stats = it.second->GetStats();
+    if (stats.num_samples >= kMinRequiredPeriodicSamples) {
       RTC_LOGGED_HISTOGRAM_COUNTS_10000("WebRTC.Video.SendDelayInMs",
-                                        send_delay_ms);
+                                        stats.average);
     }
   }
 }
 
 void SendDelayStats::AddSsrcs(const VideoSendStream::Config& config) {
   rtc::CritScope lock(&crit_);
   if (ssrcs_.size() > kMaxSsrcMapSize)
     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);
+  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())
     return;
 
   int64_t now = clock_->TimeInMilliseconds();
   RemoveOld(now, &packets_);
@@ skipping 12 matching lines @@
     return false;
 
   rtc::CritScope lock(&crit_);
   auto it = packets_.find(packet_id);
   if (it == packets_.end())
     return false;
 
   // TODO(asapersson): Remove SendSideDelayUpdated(), use capture -> sent.
   // Elapsed time from send (to transport) -> sent (leaving socket).
   int diff_ms = time_ms - it->second.send_time_ms;
-  send_delay_counters_[it->second.ssrc].Add(diff_ms);
+  GetSendDelayCounter(it->second.ssrc)->Add(diff_ms);
   packets_.erase(it);
   return true;
 }
 
 void SendDelayStats::RemoveOld(int64_t now, PacketMap* packets) {
   while (!packets->empty()) {
     auto it = packets->begin();
     if (now - it->second.capture_time_ms < kMaxSentPacketDelayMs)
       break;
 
     packets->erase(it);
     ++num_old_packets_;
   }
 }
 
-void SendDelayStats::SampleCounter::Add(int sample) {
-  sum += sample;
-  ++num_samples;
-}
-
-int SendDelayStats::SampleCounter::Avg(int min_required_samples) const {
-  if (num_samples < min_required_samples || num_samples == 0)
-    return -1;
-  return (sum + (num_samples / 2)) / num_samples;
-}
-
 }  // namespace webrtc