Line data    Source code

       1             : /*
       2             :  *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
       3             :  *
       4             :  *  Use of this source code is governed by a BSD-style license
       5             :  *  that can be found in the LICENSE file in the root of the source
       6             :  *  tree. An additional intellectual property rights grant can be found
       7             :  *  in the file PATENTS.  All contributing project authors may
       8             :  *  be found in the AUTHORS file in the root of the source tree.
       9             :  */
      10             : 
      11             : #include "webrtc/video/send_delay_stats.h"
      12             : 
      13             : #include <utility>
      14             : 
      15             : #include "webrtc/base/logging.h"
      16             : #include "webrtc/system_wrappers/include/metrics.h"
      17             : 
      18             : namespace webrtc {
      19             : namespace {
      20             : // Packet with a larger delay are removed and excluded from the delay stats.
      21             : // Set to larger than max histogram delay which is 10000.
      22             : const int64_t kMaxSentPacketDelayMs = 11000;
      23             : const size_t kMaxPacketMapSize = 2000;
      24             : 
      25             : // Limit for the maximum number of streams to calculate stats for.
      26             : const size_t kMaxSsrcMapSize = 50;
      27             : const int kMinRequiredPeriodicSamples = 5;
      28             : }  // namespace
      29             : 
      30           0 : SendDelayStats::SendDelayStats(Clock* clock)
      31           0 :     : clock_(clock), num_old_packets_(0), num_skipped_packets_(0) {}
      32             : 
      33           0 : SendDelayStats::~SendDelayStats() {
      34           0 :   if (num_old_packets_ > 0 || num_skipped_packets_ > 0) {
      35           0 :     LOG(LS_WARNING) << "Delay stats: number of old packets " << num_old_packets_
      36           0 :                     << ", skipped packets " << num_skipped_packets_
      37           0 :                     << ". Number of streams " << send_delay_counters_.size();
      38             :   }
      39           0 :   UpdateHistograms();
      40           0 : }
      41             : 
      42           0 : void SendDelayStats::UpdateHistograms() {
      43           0 :   rtc::CritScope lock(&crit_);
      44           0 :   for (const auto& it : send_delay_counters_) {
      45           0 :     AggregatedStats stats = it.second->GetStats();
      46           0 :     if (stats.num_samples >= kMinRequiredPeriodicSamples) {
      47           0 :       RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.SendDelayInMs", stats.average);
      48           0 :       LOG(LS_INFO) << "WebRTC.Video.SendDelayInMs, " << stats.ToString();
      49             :     }
      50             :   }
      51           0 : }
      52             : 
      53           0 : void SendDelayStats::AddSsrcs(const VideoSendStream::Config& config) {
      54           0 :   rtc::CritScope lock(&crit_);
      55           0 :   if (ssrcs_.size() > kMaxSsrcMapSize)
      56           0 :     return;
      57           0 :   for (const auto& ssrc : config.rtp.ssrcs)
      58           0 :     ssrcs_.insert(ssrc);
      59             : }
      60             : 
      61           0 : AvgCounter* SendDelayStats::GetSendDelayCounter(uint32_t ssrc) {
      62           0 :   const auto& it = send_delay_counters_.find(ssrc);
      63           0 :   if (it != send_delay_counters_.end())
      64           0 :     return it->second.get();
      65             : 
      66           0 :   AvgCounter* counter = new AvgCounter(clock_, nullptr, false);
      67           0 :   send_delay_counters_[ssrc].reset(counter);
      68           0 :   return counter;
      69             : }
      70             : 
      71           0 : void SendDelayStats::OnSendPacket(uint16_t packet_id,
      72             :                                   int64_t capture_time_ms,
      73             :                                   uint32_t ssrc) {
      74             :   // Packet sent to transport.
      75           0 :   rtc::CritScope lock(&crit_);
      76           0 :   if (ssrcs_.find(ssrc) == ssrcs_.end())
      77           0 :     return;
      78             : 
      79           0 :   int64_t now = clock_->TimeInMilliseconds();
      80           0 :   RemoveOld(now, &packets_);
      81             : 
      82           0 :   if (packets_.size() > kMaxPacketMapSize) {
      83           0 :     ++num_skipped_packets_;
      84           0 :     return;
      85             :   }
      86             :   packets_.insert(
      87           0 :       std::make_pair(packet_id, Packet(ssrc, capture_time_ms, now)));
      88             : }
      89             : 
      90           0 : bool SendDelayStats::OnSentPacket(int packet_id, int64_t time_ms) {
      91             :   // Packet leaving socket.
      92           0 :   if (packet_id == -1)
      93           0 :     return false;
      94             : 
      95           0 :   rtc::CritScope lock(&crit_);
      96           0 :   auto it = packets_.find(packet_id);
      97           0 :   if (it == packets_.end())
      98           0 :     return false;
      99             : 
     100             :   // TODO(asapersson): Remove SendSideDelayUpdated(), use capture -> sent.
     101             :   // Elapsed time from send (to transport) -> sent (leaving socket).
     102           0 :   int diff_ms = time_ms - it->second.send_time_ms;
     103           0 :   GetSendDelayCounter(it->second.ssrc)->Add(diff_ms);
     104           0 :   packets_.erase(it);
     105           0 :   return true;
     106             : }
     107             : 
     108           0 : void SendDelayStats::RemoveOld(int64_t now, PacketMap* packets) {
     109           0 :   while (!packets->empty()) {
     110           0 :     auto it = packets->begin();
     111           0 :     if (now - it->second.capture_time_ms < kMaxSentPacketDelayMs)
     112           0 :       break;
     113             : 
     114           0 :     packets->erase(it);
     115           0 :     ++num_old_packets_;
     116             :   }
     117           0 : }
     118             : 
     119             : }  // namespace webrtc