Adding ViE CallStats to keep track of call statistics. As a start, only rtt is handled.

This CL will be followed by another CL connecting the dots.

BUG=769
TEST=New unittest added.

Review URL: https://webrtc-codereview.appspot.com/968006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3117 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
mflodman@webrtc.org 2012-11-16 13:57:26 +00:00
parent c289f9f209
commit b2f474e8bb
5 changed files with 392 additions and 0 deletions

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@ -245,6 +245,13 @@ class RtcpBandwidthObserver {
virtual ~RtcpBandwidthObserver() {}
};
class RtcpRttObserver {
public:
virtual void OnRttUpdate(uint32_t rtt) = 0;
virtual ~RtcpRttObserver() {};
};
// A clock interface that allows reading of absolute and relative
// timestamps in an RTP/RTCP module.
class RtpRtcpClock {

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@ -0,0 +1,119 @@
/*
* Copyright (c) 2012 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/video_engine/call_stats.h"
#include <cassert>
#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
#include "webrtc/system_wrappers/interface/tick_util.h"
#include "webrtc/system_wrappers/interface/trace.h"
namespace webrtc {
// A rtt report is considered valid for this long.
const int kRttTimeoutMs = 1500;
// Time interval for updating the observers.
const int kUpdateIntervalMs = 1000;
class RtcpObserver : public RtcpRttObserver {
public:
explicit RtcpObserver(CallStats* owner) : owner_(owner) {}
virtual ~RtcpObserver() {}
virtual void OnRttUpdate(uint32_t rtt) {
owner_->OnRttUpdate(rtt);
}
private:
CallStats* owner_;
DISALLOW_COPY_AND_ASSIGN(RtcpObserver);
};
CallStats::CallStats()
: crit_(CriticalSectionWrapper::CreateCriticalSection()),
rtcp_rtt_observer_(new RtcpObserver(this)),
last_process_time_(TickTime::MillisecondTimestamp()) {
}
CallStats::~CallStats() {
assert(observers_.empty());
}
int32_t CallStats::TimeUntilNextProcess() {
return last_process_time_ + kUpdateIntervalMs -
TickTime::MillisecondTimestamp();
}
int32_t CallStats::Process() {
CriticalSectionScoped cs(crit_.get());
if (TickTime::MillisecondTimestamp() < last_process_time_ + kUpdateIntervalMs)
return 0;
// Remove invalid, as in too old, rtt values.
int64_t time_now = TickTime::MillisecondTimestamp();
while (!reports_.empty() && reports_.front().time + kRttTimeoutMs <
time_now) {
reports_.pop_front();
}
// Find the max stored RTT.
uint32_t max_rtt = 0;
for (std::list<RttTime>::const_iterator it = reports_.begin();
it != reports_.end(); ++it) {
if (it->rtt > max_rtt)
max_rtt = it->rtt;
}
// If there is a valid rtt, update all observers.
if (max_rtt > 0) {
for (std::list<StatsObserver*>::iterator it = observers_.begin();
it != observers_.end(); ++it) {
(*it)->OnRttUpdate(max_rtt);
}
}
last_process_time_ = time_now;
return 0;
}
RtcpRttObserver* CallStats::rtcp_rtt_observer() const {
return rtcp_rtt_observer_.get();
}
void CallStats::RegisterStatsObserver(StatsObserver* observer) {
CriticalSectionScoped cs(crit_.get());
for (std::list<StatsObserver*>::iterator it = observers_.begin();
it != observers_.end(); ++it) {
if (*it == observer)
return;
}
observers_.push_back(observer);
}
void CallStats::DeregisterStatsObserver(StatsObserver* observer) {
CriticalSectionScoped cs(crit_.get());
for (std::list<StatsObserver*>::iterator it = observers_.begin();
it != observers_.end(); ++it) {
if (*it == observer) {
observers_.erase(it);
return;
}
}
}
void CallStats::OnRttUpdate(uint32_t rtt) {
CriticalSectionScoped cs(crit_.get());
int64_t time_now = TickTime::MillisecondTimestamp();
reports_.push_back(RttTime(rtt, time_now));
}
} // namespace webrtc

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@ -0,0 +1,84 @@
/*
* Copyright (c) 2012 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.
*/
#ifndef WEBRTC_VIDEO_ENGINE_CALL_STATS_H_
#define WEBRTC_VIDEO_ENGINE_CALL_STATS_H_
#include <list>
#include "webrtc/modules/interface/module.h"
#include "webrtc/system_wrappers/interface/constructor_magic.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
namespace webrtc {
class CriticalSectionWrapper;
class RtcpRttObserver;
// Interface used to distribute call statistics. Callbacks will be triggered as
// soon as the class has been registered using RegisterStatsObserver.
class StatsObserver {
public:
virtual void OnRttUpdate(uint32_t rtt) = 0;
virtual ~StatsObserver() {}
};
// CallStats keeps track of statistics for a call.
class CallStats : public Module {
public:
friend class RtcpObserver;
CallStats();
~CallStats();
// Implements Module, to use the process thread.
virtual int32_t TimeUntilNextProcess();
virtual int32_t Process();
// Returns a RtcpRttObserver to register at a statistics provider. The object
// has the same lifetime as the CallStats instance.
RtcpRttObserver* rtcp_rtt_observer() const;
// Registers/deregisters a new observer to receive statistics updates.
void RegisterStatsObserver(StatsObserver* observer);
void DeregisterStatsObserver(StatsObserver* observer);
protected:
void OnRttUpdate(uint32_t rtt);
private:
// Helper struct keeping track of the time a rtt value is reported.
struct RttTime {
RttTime(uint32_t new_rtt, int64_t rtt_time)
: rtt(new_rtt), time(rtt_time) {}
const uint32_t rtt;
const int64_t time;
};
// Protecting all members.
scoped_ptr<CriticalSectionWrapper> crit_;
// Observer receiving statistics updates.
scoped_ptr<RtcpRttObserver> rtcp_rtt_observer_;
// The last time 'Process' resulted in statistic update.
int64_t last_process_time_;
// All Rtt reports within valid time interval, oldest first.
std::list<RttTime> reports_;
// Observers getting stats reports.
std::list<StatsObserver*> observers_;
DISALLOW_COPY_AND_ASSIGN(CallStats);
};
} // namespace webrtc
#endif // WEBRTC_VIDEO_ENGINE_CALL_STATS_H_

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@ -0,0 +1,179 @@
/*
* Copyright (c) 2012 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 <gmock/gmock.h>
#include <gtest/gtest.h>
#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
#include "webrtc/system_wrappers/interface/tick_util.h"
#include "webrtc/video_engine/call_stats.h"
using ::testing::_;
using ::testing::AnyNumber;
using ::testing::Return;
namespace webrtc {
class MockStatsObserver : public StatsObserver {
public:
MockStatsObserver() {}
virtual ~MockStatsObserver() {}
MOCK_METHOD1(OnRttUpdate, void(uint32_t));
};
class CallStatsTest : public ::testing::Test {
protected:
virtual void SetUp() {
TickTime::UseFakeClock(12345);
call_stats_.reset(new CallStats());
}
scoped_ptr<CallStats> call_stats_;
};
TEST_F(CallStatsTest, AddAndTriggerCallback) {
MockStatsObserver stats_observer;
RtcpRttObserver* rtcp_observer = call_stats_->rtcp_rtt_observer();
call_stats_->RegisterStatsObserver(&stats_observer);
TickTime::AdvanceFakeClock(1000);
uint32_t rtt = 25;
rtcp_observer->OnRttUpdate(rtt);
EXPECT_CALL(stats_observer, OnRttUpdate(rtt))
.Times(1);
call_stats_->Process();
call_stats_->DeregisterStatsObserver(&stats_observer);
}
TEST_F(CallStatsTest, ProcessTime) {
MockStatsObserver stats_observer;
call_stats_->RegisterStatsObserver(&stats_observer);
RtcpRttObserver* rtcp_observer = call_stats_->rtcp_rtt_observer();
rtcp_observer->OnRttUpdate(100);
// Time isn't updated yet.
EXPECT_CALL(stats_observer, OnRttUpdate(_))
.Times(0);
call_stats_->Process();
// Advance clock and verify we get an update.
TickTime::AdvanceFakeClock(1000);
EXPECT_CALL(stats_observer, OnRttUpdate(_))
.Times(1);
call_stats_->Process();
// Advance clock just too little to get an update.
TickTime::AdvanceFakeClock(999);
rtcp_observer->OnRttUpdate(100);
EXPECT_CALL(stats_observer, OnRttUpdate(_))
.Times(0);
call_stats_->Process();
// Advance enough to trigger a new update.
TickTime::AdvanceFakeClock(1);
EXPECT_CALL(stats_observer, OnRttUpdate(_))
.Times(1);
call_stats_->Process();
call_stats_->DeregisterStatsObserver(&stats_observer);
}
// Verify all observers get correct estimates and observers can be added and
// removed.
TEST_F(CallStatsTest, MultipleObservers) {
MockStatsObserver stats_observer_1;
call_stats_->RegisterStatsObserver(&stats_observer_1);
// Add the secondobserver twice, there should still be only one report to the
// observer.
MockStatsObserver stats_observer_2;
call_stats_->RegisterStatsObserver(&stats_observer_2);
call_stats_->RegisterStatsObserver(&stats_observer_2);
RtcpRttObserver* rtcp_observer = call_stats_->rtcp_rtt_observer();
uint32_t rtt = 100;
rtcp_observer->OnRttUpdate(rtt);
// Verify both observers are updated.
TickTime::AdvanceFakeClock(1000);
EXPECT_CALL(stats_observer_1, OnRttUpdate(rtt))
.Times(1);
EXPECT_CALL(stats_observer_2, OnRttUpdate(rtt))
.Times(1);
call_stats_->Process();
// Deregister the second observer and verify update is only sent to the first
// observer.
call_stats_->DeregisterStatsObserver(&stats_observer_2);
rtcp_observer->OnRttUpdate(rtt);
TickTime::AdvanceFakeClock(1000);
EXPECT_CALL(stats_observer_1, OnRttUpdate(rtt))
.Times(1);
EXPECT_CALL(stats_observer_2, OnRttUpdate(rtt))
.Times(0);
call_stats_->Process();
// Deregister the first observer.
call_stats_->DeregisterStatsObserver(&stats_observer_1);
rtcp_observer->OnRttUpdate(rtt);
TickTime::AdvanceFakeClock(1000);
EXPECT_CALL(stats_observer_1, OnRttUpdate(rtt))
.Times(0);
EXPECT_CALL(stats_observer_2, OnRttUpdate(rtt))
.Times(0);
call_stats_->Process();
}
// Verify increasing and decreasing rtt triggers callbacks with correct values.
TEST_F(CallStatsTest, ChangeRtt) {
MockStatsObserver stats_observer;
call_stats_->RegisterStatsObserver(&stats_observer);
RtcpRttObserver* rtcp_observer = call_stats_->rtcp_rtt_observer();
// Advance clock to be ready for an update.
TickTime::AdvanceFakeClock(1000);
// Set a first value and verify the callback is triggered.
const uint32_t first_rtt = 100;
rtcp_observer->OnRttUpdate(first_rtt);
EXPECT_CALL(stats_observer, OnRttUpdate(first_rtt))
.Times(1);
call_stats_->Process();
// Increase rtt and verify the new value is reported.
TickTime::AdvanceFakeClock(1000);
const uint32_t high_rtt = first_rtt + 20;
rtcp_observer->OnRttUpdate(high_rtt);
EXPECT_CALL(stats_observer, OnRttUpdate(high_rtt))
.Times(1);
call_stats_->Process();
// Increase time enough for a new update, but not too much to make the
// rtt invalid. Report a lower rtt and verify the old/high value still is sent
// in the callback.
TickTime::AdvanceFakeClock(1000);
const uint32_t low_rtt = first_rtt - 20;
rtcp_observer->OnRttUpdate(low_rtt);
EXPECT_CALL(stats_observer, OnRttUpdate(high_rtt))
.Times(1);
call_stats_->Process();
// Advance time to make the high report invalid, the lower rtt should no be in
// the callback.
TickTime::AdvanceFakeClock(1000);
EXPECT_CALL(stats_observer, OnRttUpdate(low_rtt))
.Times(1);
call_stats_->Process();
call_stats_->DeregisterStatsObserver(&stats_observer);
}
} // namespace webrtc

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@ -67,6 +67,7 @@
'include/vie_rtp_rtcp.h',
# headers
'call_stats.h',
'encoder_state_feedback.h',
'stream_synchronization.h',
'vie_base_impl.h',
@ -102,6 +103,7 @@
'vie_sync_module.h',
# ViE
'call_stats.cc',
'encoder_state_feedback.cc',
'stream_synchronization.cc',
'vie_base_impl.cc',
@ -155,6 +157,7 @@
'../modules/rtp_rtcp/interface',
],
'sources': [
'call_stats_unittest.cc',
'encoder_state_feedback_unittest.cc',
'stream_synchronization_unittest.cc',
'vie_remb_unittest.cc',