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/*
* 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 "video/send_delay_stats.h"
#include <utility>
#include "rtc_base/logging.h"
#include "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 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) {
RTC_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() {
MutexLock lock(&mutex_);
for (const auto& it : send_delay_counters_) {
AggregatedStats stats = it.second->GetStats();
if (stats.num_samples >= kMinRequiredPeriodicSamples) {
RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.SendDelayInMs", stats.average);
RTC_LOG(LS_INFO) << "WebRTC.Video.SendDelayInMs, " << stats.ToString();
}
}
}
void SendDelayStats::AddSsrcs(const VideoSendStream::Config& config) {
MutexLock lock(&mutex_);
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, false);
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.
MutexLock lock(&mutex_);
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;
MutexLock lock(&mutex_);
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;
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_;
}
}
} // namespace webrtc
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