35151f35ec
- "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}
119 lines
3.6 KiB
C++
119 lines
3.6 KiB
C++
/*
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* Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "webrtc/video/send_delay_stats.h"
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#include "webrtc/base/logging.h"
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#include "webrtc/system_wrappers/include/metrics.h"
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namespace webrtc {
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namespace {
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// Packet with a larger delay are removed and excluded from the delay stats.
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// Set to larger than max histogram delay which is 10000.
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const int64_t kMaxSentPacketDelayMs = 11000;
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const size_t kMaxPacketMapSize = 2000;
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// Limit for the maximum number of streams to calculate stats for.
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const size_t kMaxSsrcMapSize = 50;
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const int kMinRequiredSamples = 200;
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} // namespace
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SendDelayStats::SendDelayStats(Clock* clock)
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: clock_(clock), num_old_packets_(0), num_skipped_packets_(0) {}
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SendDelayStats::~SendDelayStats() {
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if (num_old_packets_ > 0 || num_skipped_packets_ > 0) {
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LOG(LS_WARNING) << "Delay stats: number of old packets " << num_old_packets_
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<< ", skipped packets " << num_skipped_packets_
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<< ". Number of streams " << send_delay_counters_.size();
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}
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UpdateHistograms();
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}
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void SendDelayStats::UpdateHistograms() {
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rtc::CritScope lock(&crit_);
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for (const auto& it : send_delay_counters_) {
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int send_delay_ms = it.second.Avg(kMinRequiredSamples);
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if (send_delay_ms != -1) {
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RTC_LOGGED_HISTOGRAM_COUNTS_10000("WebRTC.Video.SendDelayInMs",
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send_delay_ms);
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}
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}
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}
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void SendDelayStats::AddSsrcs(const VideoSendStream::Config& config) {
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rtc::CritScope lock(&crit_);
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if (ssrcs_.size() > kMaxSsrcMapSize)
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return;
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for (const auto& ssrc : config.rtp.ssrcs)
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ssrcs_.insert(ssrc);
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}
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void SendDelayStats::OnSendPacket(uint16_t packet_id,
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int64_t capture_time_ms,
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uint32_t ssrc) {
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// Packet sent to transport.
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rtc::CritScope lock(&crit_);
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if (ssrcs_.find(ssrc) == ssrcs_.end())
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return;
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int64_t now = clock_->TimeInMilliseconds();
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RemoveOld(now, &packets_);
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if (packets_.size() > kMaxPacketMapSize) {
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++num_skipped_packets_;
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return;
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}
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packets_.insert(
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std::make_pair(packet_id, Packet(ssrc, capture_time_ms, now)));
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}
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bool SendDelayStats::OnSentPacket(int packet_id, int64_t time_ms) {
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// Packet leaving socket.
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if (packet_id == -1)
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return false;
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rtc::CritScope lock(&crit_);
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auto it = packets_.find(packet_id);
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if (it == packets_.end())
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return false;
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// TODO(asapersson): Remove SendSideDelayUpdated(), use capture -> sent.
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// Elapsed time from send (to transport) -> sent (leaving socket).
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int diff_ms = time_ms - it->second.send_time_ms;
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send_delay_counters_[it->second.ssrc].Add(diff_ms);
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packets_.erase(it);
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return true;
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}
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void SendDelayStats::RemoveOld(int64_t now, PacketMap* packets) {
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while (!packets->empty()) {
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auto it = packets->begin();
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if (now - it->second.capture_time_ms < kMaxSentPacketDelayMs)
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break;
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packets->erase(it);
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++num_old_packets_;
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}
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}
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void SendDelayStats::SampleCounter::Add(int sample) {
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sum += sample;
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++num_samples;
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}
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int SendDelayStats::SampleCounter::Avg(int min_required_samples) const {
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if (num_samples < min_required_samples || num_samples == 0)
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return -1;
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return (sum + (num_samples / 2)) / num_samples;
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}
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} // namespace webrtc
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