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Side by Side Diff: webrtc/video_engine/call_stats.cc

Issue 1506773002: Merge webrtc/video_engine/ into webrtc/video/ (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: rebase Created 5 years ago
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1 /*
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
3 *
4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree.
9 */
10
11 #include "webrtc/video_engine/call_stats.h"
12
13 #include <assert.h>
14
15 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
16 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
17 #include "webrtc/system_wrappers/include/tick_util.h"
18
19 namespace webrtc {
20 namespace {
21 // Time interval for updating the observers.
22 const int64_t kUpdateIntervalMs = 1000;
23 // Weight factor to apply to the average rtt.
24 const float kWeightFactor = 0.3f;
25
26 void RemoveOldReports(int64_t now, std::list<CallStats::RttTime>* reports) {
27 // A rtt report is considered valid for this long.
28 const int64_t kRttTimeoutMs = 1500;
29 while (!reports->empty() &&
30 (now - reports->front().time) > kRttTimeoutMs) {
31 reports->pop_front();
32 }
33 }
34
35 int64_t GetMaxRttMs(std::list<CallStats::RttTime>* reports) {
36 int64_t max_rtt_ms = 0;
37 for (std::list<CallStats::RttTime>::const_iterator it = reports->begin();
38 it != reports->end(); ++it) {
39 max_rtt_ms = std::max(it->rtt, max_rtt_ms);
40 }
41 return max_rtt_ms;
42 }
43
44 int64_t GetAvgRttMs(std::list<CallStats::RttTime>* reports) {
45 if (reports->empty()) {
46 return 0;
47 }
48 int64_t sum = 0;
49 for (std::list<CallStats::RttTime>::const_iterator it = reports->begin();
50 it != reports->end(); ++it) {
51 sum += it->rtt;
52 }
53 return sum / reports->size();
54 }
55
56 void UpdateAvgRttMs(std::list<CallStats::RttTime>* reports, int64_t* avg_rtt) {
57 uint32_t cur_rtt_ms = GetAvgRttMs(reports);
58 if (cur_rtt_ms == 0) {
59 // Reset.
60 *avg_rtt = 0;
61 return;
62 }
63 if (*avg_rtt == 0) {
64 // Initialize.
65 *avg_rtt = cur_rtt_ms;
66 return;
67 }
68 *avg_rtt = *avg_rtt * (1.0f - kWeightFactor) + cur_rtt_ms * kWeightFactor;
69 }
70 } // namespace
71
72 class RtcpObserver : public RtcpRttStats {
73 public:
74 explicit RtcpObserver(CallStats* owner) : owner_(owner) {}
75 virtual ~RtcpObserver() {}
76
77 virtual void OnRttUpdate(int64_t rtt) {
78 owner_->OnRttUpdate(rtt);
79 }
80
81 // Returns the average RTT.
82 virtual int64_t LastProcessedRtt() const {
83 return owner_->avg_rtt_ms();
84 }
85
86 private:
87 CallStats* owner_;
88
89 RTC_DISALLOW_COPY_AND_ASSIGN(RtcpObserver);
90 };
91
92 CallStats::CallStats()
93 : crit_(CriticalSectionWrapper::CreateCriticalSection()),
94 rtcp_rtt_stats_(new RtcpObserver(this)),
95 last_process_time_(TickTime::MillisecondTimestamp()),
96 max_rtt_ms_(0),
97 avg_rtt_ms_(0) {
98 }
99
100 CallStats::~CallStats() {
101 assert(observers_.empty());
102 }
103
104 int64_t CallStats::TimeUntilNextProcess() {
105 return last_process_time_ + kUpdateIntervalMs -
106 TickTime::MillisecondTimestamp();
107 }
108
109 int32_t CallStats::Process() {
110 CriticalSectionScoped cs(crit_.get());
111 int64_t now = TickTime::MillisecondTimestamp();
112 if (now < last_process_time_ + kUpdateIntervalMs)
113 return 0;
114
115 last_process_time_ = now;
116
117 RemoveOldReports(now, &reports_);
118 max_rtt_ms_ = GetMaxRttMs(&reports_);
119 UpdateAvgRttMs(&reports_, &avg_rtt_ms_);
120
121 // If there is a valid rtt, update all observers with the max rtt.
122 // TODO(asapersson): Consider changing this to report the average rtt.
123 if (max_rtt_ms_ > 0) {
124 for (std::list<CallStatsObserver*>::iterator it = observers_.begin();
125 it != observers_.end(); ++it) {
126 (*it)->OnRttUpdate(avg_rtt_ms_, max_rtt_ms_);
127 }
128 }
129 return 0;
130 }
131
132 int64_t CallStats::avg_rtt_ms() const {
133 CriticalSectionScoped cs(crit_.get());
134 return avg_rtt_ms_;
135 }
136
137 RtcpRttStats* CallStats::rtcp_rtt_stats() const {
138 return rtcp_rtt_stats_.get();
139 }
140
141 void CallStats::RegisterStatsObserver(CallStatsObserver* observer) {
142 CriticalSectionScoped cs(crit_.get());
143 for (std::list<CallStatsObserver*>::iterator it = observers_.begin();
144 it != observers_.end(); ++it) {
145 if (*it == observer)
146 return;
147 }
148 observers_.push_back(observer);
149 }
150
151 void CallStats::DeregisterStatsObserver(CallStatsObserver* observer) {
152 CriticalSectionScoped cs(crit_.get());
153 for (std::list<CallStatsObserver*>::iterator it = observers_.begin();
154 it != observers_.end(); ++it) {
155 if (*it == observer) {
156 observers_.erase(it);
157 return;
158 }
159 }
160 }
161
162 void CallStats::OnRttUpdate(int64_t rtt) {
163 CriticalSectionScoped cs(crit_.get());
164 reports_.push_back(RttTime(rtt, TickTime::MillisecondTimestamp()));
165 }
166
167 } // namespace webrtc
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