Add histogram stats for send delay for a sent video stream.

webrtc/video/send_delay_stats.cc

Index: webrtc/video/send_delay_stats.cc
diff --git a/webrtc/video/send_delay_stats.cc b/webrtc/video/send_delay_stats.cc
new file mode 100644
index 0000000000000000000000000000000000000000..8701066519807bc1cb2634dc5e3a03ed9c2a9584
--- /dev/null
+++ b/webrtc/video/send_delay_stats.cc
@@ -0,0 +1,118 @@
+/*
+ *  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;
+}  // 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) {
+      RTC_LOGGED_HISTOGRAM_COUNTS_10000("WebRTC.Video.SendDelayInMs",
+                                        send_delay_ms);
+    }
+  }
+}
+
+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);
+}
+
+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_);
+
+  if (packets_.size() > kMaxPacketMapSize) {
+    ++num_skipped_packets_;
+    return;
+  }
+  packets_.insert(
+      std::make_pair(packet_id, Packet(ssrc, capture_time_ms, now)));
+}
+
+bool SendDelayStats::OnSentPacket(int packet_id, int64_t time_ms) {
+  // Packet leaving socket.
+  if (packet_id == -1)
+    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);
+  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