blob: d2938a4c29d4c59d4022e968c091679df7b13c59 [file] [log] [blame]
/*
* Copyright 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/api/rtcstatscollector.h"
#include <memory>
#include <utility>
#include <vector>
#include "webrtc/api/peerconnection.h"
#include "webrtc/base/checks.h"
namespace webrtc {
rtc::scoped_refptr<RTCStatsCollector> RTCStatsCollector::Create(
PeerConnection* pc, int64_t cache_lifetime_us) {
return rtc::scoped_refptr<RTCStatsCollector>(
new rtc::RefCountedObject<RTCStatsCollector>(pc, cache_lifetime_us));
}
RTCStatsCollector::RTCStatsCollector(PeerConnection* pc,
int64_t cache_lifetime_us)
: pc_(pc),
signaling_thread_(pc->session()->signaling_thread()),
worker_thread_(pc->session()->worker_thread()),
network_thread_(pc->session()->network_thread()),
num_pending_partial_reports_(0),
partial_report_timestamp_us_(0),
cache_timestamp_us_(0),
cache_lifetime_us_(cache_lifetime_us) {
RTC_DCHECK(pc_);
RTC_DCHECK(signaling_thread_);
RTC_DCHECK(worker_thread_);
RTC_DCHECK(network_thread_);
RTC_DCHECK_GE(cache_lifetime_us_, 0);
}
void RTCStatsCollector::GetStatsReport(
rtc::scoped_refptr<RTCStatsCollectorCallback> callback) {
RTC_DCHECK(signaling_thread_->IsCurrent());
RTC_DCHECK(callback);
callbacks_.push_back(callback);
// "Now" using a monotonically increasing timer.
int64_t cache_now_us = rtc::TimeMicros();
if (cached_report_ &&
cache_now_us - cache_timestamp_us_ <= cache_lifetime_us_) {
// We have a fresh cached report to deliver.
DeliverCachedReport();
} else if (!num_pending_partial_reports_) {
// Only start gathering stats if we're not already gathering stats. In the
// case of already gathering stats, |callback_| will be invoked when there
// are no more pending partial reports.
// "Now" using a system clock, relative to the UNIX epoch (Jan 1, 1970,
// UTC), in microseconds. The system clock could be modified and is not
// necessarily monotonically increasing.
int64_t timestamp_us = rtc::TimeUTCMicros();
num_pending_partial_reports_ = 3;
partial_report_timestamp_us_ = cache_now_us;
invoker_.AsyncInvoke<void>(RTC_FROM_HERE, signaling_thread_,
rtc::Bind(&RTCStatsCollector::ProducePartialResultsOnSignalingThread,
rtc::scoped_refptr<RTCStatsCollector>(this), timestamp_us));
invoker_.AsyncInvoke<void>(RTC_FROM_HERE, worker_thread_,
rtc::Bind(&RTCStatsCollector::ProducePartialResultsOnWorkerThread,
rtc::scoped_refptr<RTCStatsCollector>(this), timestamp_us));
invoker_.AsyncInvoke<void>(RTC_FROM_HERE, network_thread_,
rtc::Bind(&RTCStatsCollector::ProducePartialResultsOnNetworkThread,
rtc::scoped_refptr<RTCStatsCollector>(this), timestamp_us));
}
}
void RTCStatsCollector::ClearCachedStatsReport() {
RTC_DCHECK(signaling_thread_->IsCurrent());
cached_report_ = nullptr;
}
void RTCStatsCollector::ProducePartialResultsOnSignalingThread(
int64_t timestamp_us) {
RTC_DCHECK(signaling_thread_->IsCurrent());
rtc::scoped_refptr<RTCStatsReport> report = RTCStatsReport::Create();
report->AddStats(ProducePeerConnectionStats_s(timestamp_us));
AddPartialResults(report);
}
void RTCStatsCollector::ProducePartialResultsOnWorkerThread(
int64_t timestamp_us) {
RTC_DCHECK(worker_thread_->IsCurrent());
rtc::scoped_refptr<RTCStatsReport> report = RTCStatsReport::Create();
// TODO(hbos): Gather stats on worker thread.
AddPartialResults(report);
}
void RTCStatsCollector::ProducePartialResultsOnNetworkThread(
int64_t timestamp_us) {
RTC_DCHECK(network_thread_->IsCurrent());
rtc::scoped_refptr<RTCStatsReport> report = RTCStatsReport::Create();
// TODO(hbos): Gather stats on network thread.
AddPartialResults(report);
}
void RTCStatsCollector::AddPartialResults(
const rtc::scoped_refptr<RTCStatsReport>& partial_report) {
if (!signaling_thread_->IsCurrent()) {
invoker_.AsyncInvoke<void>(RTC_FROM_HERE, signaling_thread_,
rtc::Bind(&RTCStatsCollector::AddPartialResults_s,
rtc::scoped_refptr<RTCStatsCollector>(this),
partial_report));
return;
}
AddPartialResults_s(partial_report);
}
void RTCStatsCollector::AddPartialResults_s(
rtc::scoped_refptr<RTCStatsReport> partial_report) {
RTC_DCHECK(signaling_thread_->IsCurrent());
RTC_DCHECK_GT(num_pending_partial_reports_, 0);
if (!partial_report_)
partial_report_ = partial_report;
else
partial_report_->TakeMembersFrom(partial_report);
--num_pending_partial_reports_;
if (!num_pending_partial_reports_) {
cache_timestamp_us_ = partial_report_timestamp_us_;
cached_report_ = partial_report_;
partial_report_ = nullptr;
DeliverCachedReport();
}
}
void RTCStatsCollector::DeliverCachedReport() {
RTC_DCHECK(signaling_thread_->IsCurrent());
RTC_DCHECK(!callbacks_.empty());
RTC_DCHECK(cached_report_);
for (const rtc::scoped_refptr<RTCStatsCollectorCallback>& callback :
callbacks_) {
callback->OnStatsDelivered(cached_report_);
}
callbacks_.clear();
}
std::unique_ptr<RTCPeerConnectionStats>
RTCStatsCollector::ProducePeerConnectionStats_s(int64_t timestamp_us) const {
RTC_DCHECK(signaling_thread_->IsCurrent());
// TODO(hbos): If data channels are removed from the peer connection this will
// yield incorrect counts. Address before closing crbug.com/636818. See
// https://w3c.github.io/webrtc-stats/webrtc-stats.html#pcstats-dict*.
uint32_t data_channels_opened = 0;
const std::vector<rtc::scoped_refptr<DataChannel>>& data_channels =
pc_->sctp_data_channels();
for (const rtc::scoped_refptr<DataChannel>& data_channel : data_channels) {
if (data_channel->state() == DataChannelInterface::kOpen)
++data_channels_opened;
}
// There is always just one |RTCPeerConnectionStats| so its |id| can be a
// constant.
std::unique_ptr<RTCPeerConnectionStats> stats(
new RTCPeerConnectionStats("RTCPeerConnection", timestamp_us));
stats->data_channels_opened = data_channels_opened;
stats->data_channels_closed = static_cast<uint32_t>(data_channels.size()) -
data_channels_opened;
return stats;
}
} // namespace webrtc