Add histogram stats for average send delay of sent packets for a sent video stream. The delay is measured from a packet is sent to the transport until leaving the socket.
- "WebRTC.Video.SendDelayInMs"
Change so that PacketOption packet id is always set in RtpSender (if having a TransportSequenceNumberAllocator).
Add SendDelayStats class for computing delays.
Add SendPacketObserver to RtpRtcp config and register SendDelayStats as observer.
Wire up OnSentPacket to SendDelayStats.
BUG=webrtc:5215
Review-Url: https://codereview.webrtc.org/1478253002
Cr-Commit-Position: refs/heads/master@{#12600}
diff --git a/webrtc/video/send_delay_stats.cc b/webrtc/video/send_delay_stats.cc
new file mode 100644
index 0000000..8701066
--- /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
