diff options
Diffstat (limited to 'webrtc/video/call_stats.cc')
-rw-r--r-- | webrtc/video/call_stats.cc | 168 |
1 files changed, 168 insertions, 0 deletions
diff --git a/webrtc/video/call_stats.cc b/webrtc/video/call_stats.cc new file mode 100644 index 0000000000..69ea1a3d78 --- /dev/null +++ b/webrtc/video/call_stats.cc @@ -0,0 +1,168 @@ +/* + * 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/call_stats.h" + +#include <assert.h> + +#include <algorithm> + +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" +#include "webrtc/system_wrappers/include/critical_section_wrapper.h" +#include "webrtc/system_wrappers/include/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(); + } +} + +int64_t GetMaxRttMs(std::list<CallStats::RttTime>* reports) { + int64_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; +} + +int64_t GetAvgRttMs(std::list<CallStats::RttTime>* reports) { + if (reports->empty()) { + return 0; + } + int64_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, int64_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(int64_t rtt) { + owner_->OnRttUpdate(rtt); + } + + // Returns the average RTT. + virtual int64_t LastProcessedRtt() const { + return owner_->avg_rtt_ms(); + } + + private: + CallStats* owner_; + + RTC_DISALLOW_COPY_AND_ASSIGN(RtcpObserver); +}; + +CallStats::CallStats(Clock* clock) + : clock_(clock), + crit_(CriticalSectionWrapper::CreateCriticalSection()), + rtcp_rtt_stats_(new RtcpObserver(this)), + last_process_time_(clock_->TimeInMilliseconds()), + max_rtt_ms_(0), + avg_rtt_ms_(0) {} + +CallStats::~CallStats() { + assert(observers_.empty()); +} + +int64_t CallStats::TimeUntilNextProcess() { + return last_process_time_ + kUpdateIntervalMs - clock_->TimeInMilliseconds(); +} + +int32_t CallStats::Process() { + CriticalSectionScoped cs(crit_.get()); + int64_t now = clock_->TimeInMilliseconds(); + 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(avg_rtt_ms_, max_rtt_ms_); + } + } + return 0; +} + +int64_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(int64_t rtt) { + CriticalSectionScoped cs(crit_.get()); + reports_.push_back(RttTime(rtt, clock_->TimeInMilliseconds())); +} + +} // namespace webrtc |