Merge "Merge upstream SHA 04cb763"android-cts_7.1_r1android-cts-7.1_r9android-cts-7.1_r8android-cts-7.1_r7android-cts-7.1_r6android-cts-7.1_r5android-cts-7.1_r4android-cts-7.1_r3android-cts-7.1_r29android-cts-7.1_r28android-cts-7.1_r27android-cts-7.1_r26android-cts-7.1_r25android-cts-7.1_r24android-cts-7.1_r23android-cts-7.1_r22android-cts-7.1_r21android-cts-7.1_r20android-cts-7.1_r2android-cts-7.1_r19android-cts-7.1_r18android-cts-7.1_r17android-cts-7.1_r16android-cts-7.1_r15android-cts-7.1_r14android-cts-7.1_r13android-cts-7.1_r12android-cts-7.1_r11android-cts-7.1_r10android-cts-7.1_r1android-cts-7.0_r9android-cts-7.0_r8android-cts-7.0_r7android-cts-7.0_r6android-cts-7.0_r5android-cts-7.0_r4android-cts-7.0_r33android-cts-7.0_r32android-cts-7.0_r31android-cts-7.0_r30android-cts-7.0_r3android-cts-7.0_r29android-cts-7.0_r28android-cts-7.0_r27android-cts-7.0_r26android-cts-7.0_r25android-cts-7.0_r24android-cts-7.0_r23android-cts-7.0_r22android-cts-7.0_r21android-cts-7.0_r20android-cts-7.0_r2android-cts-7.0_r19android-cts-7.0_r18android-cts-7.0_r17android-cts-7.0_r16android-cts-7.0_r15android-cts-7.0_r14android-cts-7.0_r13android-cts-7.0_r12android-cts-7.0_r11android-cts-7.0_r10android-cts-7.0_r1android-7.1.2_r9android-7.1.2_r8android-7.1.2_r6android-7.1.2_r5android-7.1.2_r4android-7.1.2_r39android-7.1.2_r38android-7.1.2_r37android-7.1.2_r36android-7.1.2_r33android-7.1.2_r32android-7.1.2_r30android-7.1.2_r3android-7.1.2_r29android-7.1.2_r28android-7.1.2_r27android-7.1.2_r25android-7.1.2_r24android-7.1.2_r23android-7.1.2_r2android-7.1.2_r19android-7.1.2_r18android-7.1.2_r17android-7.1.2_r16android-7.1.2_r15android-7.1.2_r14android-7.1.2_r13android-7.1.2_r12android-7.1.2_r11android-7.1.2_r10android-7.1.2_r1android-7.1.1_r9android-7.1.1_r8android-7.1.1_r7android-7.1.1_r61android-7.1.1_r60android-7.1.1_r6android-7.1.1_r59android-7.1.1_r58android-7.1.1_r57android-7.1.1_r56android-7.1.1_r55android-7.1.1_r54android-7.1.1_r53android-7.1.1_r52android-7.1.1_r51android-7.1.1_r50android-7.1.1_r49android-7.1.1_r48android-7.1.1_r47android-7.1.1_r46android-7.1.1_r45android-7.1.1_r44android-7.1.1_r43android-7.1.1_r42android-7.1.1_r41android-7.1.1_r40android-7.1.1_r4android-7.1.1_r39android-7.1.1_r38android-7.1.1_r35android-7.1.1_r33android-7.1.1_r32android-7.1.1_r31android-7.1.1_r3android-7.1.1_r28android-7.1.1_r27android-7.1.1_r26android-7.1.1_r25android-7.1.1_r24android-7.1.1_r23android-7.1.1_r22android-7.1.1_r21android-7.1.1_r20android-7.1.1_r2android-7.1.1_r17android-7.1.1_r16android-7.1.1_r15android-7.1.1_r14android-7.1.1_r13android-7.1.1_r12android-7.1.1_r11android-7.1.1_r10android-7.1.1_r1android-7.1.0_r7android-7.1.0_r6android-7.1.0_r5android-7.1.0_r4android-7.1.0_r3android-7.1.0_r2android-7.1.0_r1android-7.0.0_r9android-7.0.0_r8android-7.0.0_r7android-7.0.0_r6android-7.0.0_r5android-7.0.0_r4android-7.0.0_r36android-7.0.0_r35android-7.0.0_r34android-7.0.0_r33android-7.0.0_r32android-7.0.0_r31android-7.0.0_r30android-7.0.0_r3android-7.0.0_r29android-7.0.0_r28android-7.0.0_r27android-7.0.0_r24android-7.0.0_r21android-7.0.0_r19android-7.0.0_r17android-7.0.0_r15android-7.0.0_r14android-7.0.0_r13android-7.0.0_r12android-7.0.0_r11android-7.0.0_r10android-7.0.0_r1nougat-releasenougat-mr2.3-releasenougat-mr2.2-releasenougat-mr2.1-releasenougat-mr2-security-releasenougat-mr2-releasenougat-mr2-pixel-releasenougat-mr2-devnougat-mr1.8-releasenougat-mr1.7-releasenougat-mr1.6-releasenougat-mr1.5-releasenougat-mr1.4-releasenougat-mr1.3-releasenougat-mr1.2-releasenougat-mr1.1-releasenougat-mr1-volantis-releasenougat-mr1-security-releasenougat-mr1-releasenougat-mr1-flounder-releasenougat-mr1-devnougat-mr1-cts-releasenougat-mr0.5-releasenougat-dr1-releasenougat-devnougat-cts-releasenougat-bugfix-release

am: 9a337512d9

* commit '9a337512d97e37afc142dee4fd50a41b741a87d2': (797 commits)
  Add tests for verifying transport feedback for audio and video.
  Eliminate defines in talk/
  Revert of Update with new default boringssl no-aes cipher suites. Re-enable tests. (patchset #3 id:40001 of https://codereview.webrtc.org/1550773002/ )
  Remove assert which was incorrectly added to TcpPort::OnSentPacket.
  Reland Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket.
  Update with new default boringssl no-aes cipher suites. Re-enable tests.
  Revert of Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket. (patchset #3 id:40001 of https://codereview.webrtc.org/1577873003/ )
  Re-land: "Use an explicit identifier in Config"
  Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket.
  Revert of Delete remnants of non-square pixel support from cricket::VideoFrame. (patchset #1 id:1 of https://codereview.webrtc.org/1586613002/ )
  Remove libfuzzer trybot from default trybot set.
  Add ramp-up tests for transport sequence number with and w/o audio.
  Delete remnants of non-square pixel support from cricket::VideoFrame.
  Fix IPAddress::ToSensitiveString() to avoid dependency on inet_ntop().
  Revert of Storing raw audio sink for default audio track. (patchset #7 id:120001 of https://codereview.chromium.org/1551813002/ )
  Re-enable tests that failed under Linux_Msan.
  Revert of Use an explicit identifier in Config (patchset #4 id:60001 of https://codereview.webrtc.org/1538643004/ )
  Roll chromium_revision 346fea9..099be58 (369082:369139)
  Disable WebRtcVideoChannel2BaseTest.SendManyResizeOnce for TSan
  Add build_protobuf variable.
  ...

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