pkasting@chromium.org 0b1534c52e Use int64_t for milliseconds more often, primarily for TimeUntilNextProcess.
This fixes a variety of MSVC warnings about value truncations when implicitly
storing the 64-bit values we get back from e.g. TimeTicks in 32-bit objects, and
removes the need for a number of explicit casts.

This also moves a number of constants so they're declared right where they're used, which is easier to read and maintain, and makes some of them of integral type rather than using the "enum hack".

BUG=chromium:81439
TEST=none
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7905 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 22:09:40 +00:00

168 lines
4.5 KiB
C++

/*
* 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 <assert.h>
#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"
namespace webrtc {
namespace {
// Time interval for updating the observers.
const int64_t kUpdateIntervalMs = 1000;
// Weight factor to apply to the average rtt.
const float kWeightFactor = 0.3f;
void RemoveOldReports(int64_t now, std::list<CallStats::RttTime>* reports) {
// A rtt report is considered valid for this long.
const int64_t kRttTimeoutMs = 1500;
while (!reports->empty() &&
(now - reports->front().time) > kRttTimeoutMs) {
reports->pop_front();
}
}
uint32_t GetMaxRttMs(std::list<CallStats::RttTime>* reports) {
uint32_t max_rtt_ms = 0;
for (std::list<CallStats::RttTime>::const_iterator it = reports->begin();
it != reports->end(); ++it) {
max_rtt_ms = std::max(it->rtt, max_rtt_ms);
}
return max_rtt_ms;
}
uint32_t GetAvgRttMs(std::list<CallStats::RttTime>* reports) {
if (reports->empty()) {
return 0;
}
uint32_t sum = 0;
for (std::list<CallStats::RttTime>::const_iterator it = reports->begin();
it != reports->end(); ++it) {
sum += it->rtt;
}
return sum / reports->size();
}
void UpdateAvgRttMs(std::list<CallStats::RttTime>* reports, uint32_t* avg_rtt) {
uint32_t cur_rtt_ms = GetAvgRttMs(reports);
if (cur_rtt_ms == 0) {
// Reset.
*avg_rtt = 0;
return;
}
if (*avg_rtt == 0) {
// Initialize.
*avg_rtt = cur_rtt_ms;
return;
}
*avg_rtt = *avg_rtt * (1.0f - kWeightFactor) + cur_rtt_ms * kWeightFactor;
}
} // namespace
class RtcpObserver : public RtcpRttStats {
public:
explicit RtcpObserver(CallStats* owner) : owner_(owner) {}
virtual ~RtcpObserver() {}
virtual void OnRttUpdate(uint32_t rtt) {
owner_->OnRttUpdate(rtt);
}
// Returns the average RTT.
virtual uint32_t LastProcessedRtt() const {
return owner_->avg_rtt_ms();
}
private:
CallStats* owner_;
DISALLOW_COPY_AND_ASSIGN(RtcpObserver);
};
CallStats::CallStats()
: crit_(CriticalSectionWrapper::CreateCriticalSection()),
rtcp_rtt_stats_(new RtcpObserver(this)),
last_process_time_(TickTime::MillisecondTimestamp()),
max_rtt_ms_(0),
avg_rtt_ms_(0) {
}
CallStats::~CallStats() {
assert(observers_.empty());
}
int64_t CallStats::TimeUntilNextProcess() {
return last_process_time_ + kUpdateIntervalMs -
TickTime::MillisecondTimestamp();
}
int32_t CallStats::Process() {
CriticalSectionScoped cs(crit_.get());
int64_t now = TickTime::MillisecondTimestamp();
if (now < last_process_time_ + kUpdateIntervalMs)
return 0;
last_process_time_ = now;
RemoveOldReports(now, &reports_);
max_rtt_ms_ = GetMaxRttMs(&reports_);
UpdateAvgRttMs(&reports_, &avg_rtt_ms_);
// If there is a valid rtt, update all observers with the max rtt.
// TODO(asapersson): Consider changing this to report the average rtt.
if (max_rtt_ms_ > 0) {
for (std::list<CallStatsObserver*>::iterator it = observers_.begin();
it != observers_.end(); ++it) {
(*it)->OnRttUpdate(max_rtt_ms_);
}
}
return 0;
}
uint32_t CallStats::avg_rtt_ms() const {
CriticalSectionScoped cs(crit_.get());
return avg_rtt_ms_;
}
RtcpRttStats* CallStats::rtcp_rtt_stats() const {
return rtcp_rtt_stats_.get();
}
void CallStats::RegisterStatsObserver(CallStatsObserver* observer) {
CriticalSectionScoped cs(crit_.get());
for (std::list<CallStatsObserver*>::iterator it = observers_.begin();
it != observers_.end(); ++it) {
if (*it == observer)
return;
}
observers_.push_back(observer);
}
void CallStats::DeregisterStatsObserver(CallStatsObserver* observer) {
CriticalSectionScoped cs(crit_.get());
for (std::list<CallStatsObserver*>::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());
reports_.push_back(RttTime(rtt, TickTime::MillisecondTimestamp()));
}
} // namespace webrtc