Merge webrtc/video_engine/ into webrtc/video/

webrtc/video_engine/call_stats.cc

Index: webrtc/video_engine/call_stats.cc
diff --git a/webrtc/video_engine/call_stats.cc b/webrtc/video_engine/call_stats.cc
deleted file mode 100644
index 4d5338c5fc162301796aba889ce8442c63cd4c9d..0000000000000000000000000000000000000000
--- a/webrtc/video_engine/call_stats.cc
+++ /dev/null
@@ -1,167 +0,0 @@
-/*
- *  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/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()
-    : 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(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, TickTime::MillisecondTimestamp()));
-}
-
-}  // namespace webrtc