Merge changes I7bbf776e,I1b827825

am: fe8b4a6579

* commit 'fe8b4a657979b49e1701bd92f6d5814a99e0b2be': (7237 commits)
  WIP: Changes after merge commit 'cb3f9bd'
  Make the nonlinear beamformer steerable
  Utilize bitrate above codec max to protect video.
  Enable VP9 internal resize by default.
  Filter overlapping RTP header extensions.
  Make VCMEncodedFrameCallback const.
  MediaCodecVideoEncoder: Add number of quality resolution downscales to Encoded callback.
  Remove redudant encoder rate calls.
  Create isolate files for nonparallel tests.
  Register header extensions in RtpRtcpObserver to avoid log spam.
  Make an enum class out of NetEqDecoder, and hide the neteq_decoders_ table
  ACM: Move NACK functionality inside NetEq
  Fix chromium-style warnings in webrtc/sound/.
  Create a 'webrtc_nonparallel_tests' target.
  Update scalability structure data according to updates in the RTP payload profile.
  audio_coding: rename interface -> include
  Rewrote perform_action_on_all_files to be parallell.
  Update reference indices according to updates in the RTP payload profile.
  Disable P2PTransport...TestFailoverControlledSide on Memcheck
  pass clangcl compile options to ignore warnings in gflags.cc
  ...

1 files changed, 167 insertions, 0 deletions

diff --git a/webrtc/video_engine/call_stats.cc b/webrtc/video_engine/call_stats.cc
new file mode 100644
index 0000000000..0b71cc346c
--- /dev/null
+++ b/webrtc/video_engine/call_stats.cc
@@ -0,0 +1,167 @@
+/*
+ *  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/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