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

Issue 1507903005: Revert of Merge webrtc/video_engine/ into webrtc/video/ (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Resolved merge conflict 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/call_stats.h"
12
13 #include <assert.h>
14
15 #include <algorithm>
16
17 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
18 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
19 #include "webrtc/system_wrappers/include/tick_util.h"
20
21 namespace webrtc {
22 namespace {
23 // Time interval for updating the observers.
24 const int64_t kUpdateIntervalMs = 1000;
25 // Weight factor to apply to the average rtt.
26 const float kWeightFactor = 0.3f;
27
28 void RemoveOldReports(int64_t now, std::list<CallStats::RttTime>* reports) {
29 // A rtt report is considered valid for this long.
30 const int64_t kRttTimeoutMs = 1500;
31 while (!reports->empty() &&
32 (now - reports->front().time) > kRttTimeoutMs) {
33 reports->pop_front();
34 }
35 }
36
37 int64_t GetMaxRttMs(std::list<CallStats::RttTime>* reports) {
38 int64_t max_rtt_ms = 0;
39 for (std::list<CallStats::RttTime>::const_iterator it = reports->begin();
40 it != reports->end(); ++it) {
41 max_rtt_ms = std::max(it->rtt, max_rtt_ms);
42 }
43 return max_rtt_ms;
44 }
45
46 int64_t GetAvgRttMs(std::list<CallStats::RttTime>* reports) {
47 if (reports->empty()) {
48 return 0;
49 }
50 int64_t sum = 0;
51 for (std::list<CallStats::RttTime>::const_iterator it = reports->begin();
52 it != reports->end(); ++it) {
53 sum += it->rtt;
54 }
55 return sum / reports->size();
56 }
57
58 void UpdateAvgRttMs(std::list<CallStats::RttTime>* reports, int64_t* avg_rtt) {
59 uint32_t cur_rtt_ms = GetAvgRttMs(reports);
60 if (cur_rtt_ms == 0) {
61 // Reset.
62 *avg_rtt = 0;
63 return;
64 }
65 if (*avg_rtt == 0) {
66 // Initialize.
67 *avg_rtt = cur_rtt_ms;
68 return;
69 }
70 *avg_rtt = *avg_rtt * (1.0f - kWeightFactor) + cur_rtt_ms * kWeightFactor;
71 }
72 } // namespace
73
74 class RtcpObserver : public RtcpRttStats {
75 public:
76 explicit RtcpObserver(CallStats* owner) : owner_(owner) {}
77 virtual ~RtcpObserver() {}
78
79 virtual void OnRttUpdate(int64_t rtt) {
80 owner_->OnRttUpdate(rtt);
81 }
82
83 // Returns the average RTT.
84 virtual int64_t LastProcessedRtt() const {
85 return owner_->avg_rtt_ms();
86 }
87
88 private:
89 CallStats* owner_;
90
91 RTC_DISALLOW_COPY_AND_ASSIGN(RtcpObserver);
92 };
93
94 CallStats::CallStats()
95 : crit_(CriticalSectionWrapper::CreateCriticalSection()),
96 rtcp_rtt_stats_(new RtcpObserver(this)),
97 last_process_time_(TickTime::MillisecondTimestamp()),
98 max_rtt_ms_(0),
99 avg_rtt_ms_(0) {
100 }
101
102 CallStats::~CallStats() {
103 assert(observers_.empty());
104 }
105
106 int64_t CallStats::TimeUntilNextProcess() {
107 return last_process_time_ + kUpdateIntervalMs -
108 TickTime::MillisecondTimestamp();
109 }
110
111 int32_t CallStats::Process() {
112 CriticalSectionScoped cs(crit_.get());
113 int64_t now = TickTime::MillisecondTimestamp();
114 if (now < last_process_time_ + kUpdateIntervalMs)
115 return 0;
116
117 last_process_time_ = now;
118
119 RemoveOldReports(now, &reports_);
120 max_rtt_ms_ = GetMaxRttMs(&reports_);
121 UpdateAvgRttMs(&reports_, &avg_rtt_ms_);
122
123 // If there is a valid rtt, update all observers with the max rtt.
124 // TODO(asapersson): Consider changing this to report the average rtt.
125 if (max_rtt_ms_ > 0) {
126 for (std::list<CallStatsObserver*>::iterator it = observers_.begin();
127 it != observers_.end(); ++it) {
128 (*it)->OnRttUpdate(avg_rtt_ms_, max_rtt_ms_);
129 }
130 }
131 return 0;
132 }
133
134 int64_t CallStats::avg_rtt_ms() const {
135 CriticalSectionScoped cs(crit_.get());
136 return avg_rtt_ms_;
137 }
138
139 RtcpRttStats* CallStats::rtcp_rtt_stats() const {
140 return rtcp_rtt_stats_.get();
141 }
142
143 void CallStats::RegisterStatsObserver(CallStatsObserver* observer) {
144 CriticalSectionScoped cs(crit_.get());
145 for (std::list<CallStatsObserver*>::iterator it = observers_.begin();
146 it != observers_.end(); ++it) {
147 if (*it == observer)
148 return;
149 }
150 observers_.push_back(observer);
151 }
152
153 void CallStats::DeregisterStatsObserver(CallStatsObserver* observer) {
154 CriticalSectionScoped cs(crit_.get());
155 for (std::list<CallStatsObserver*>::iterator it = observers_.begin();
156 it != observers_.end(); ++it) {
157 if (*it == observer) {
158 observers_.erase(it);
159 return;
160 }
161 }
162 }
163
164 void CallStats::OnRttUpdate(int64_t rtt) {
165 CriticalSectionScoped cs(crit_.get());
166 reports_.push_back(RttTime(rtt, TickTime::MillisecondTimestamp()));
167 }
168
169 } // namespace webrtc
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