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, 204 insertions, 0 deletions

diff --git a/webrtc/video/call_stats_unittest.cc b/webrtc/video/call_stats_unittest.cc
new file mode 100644
index 0000000000..6226a5bf6e
--- /dev/null
+++ b/webrtc/video/call_stats_unittest.cc
@@ -0,0 +1,204 @@
+/*
+ *  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 "testing/gmock/include/gmock/gmock.h"
+#include "testing/gtest/include/gtest/gtest.h"
+
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "webrtc/system_wrappers/include/tick_util.h"
+#include "webrtc/video/call_stats.h"
+
+using ::testing::_;
+using ::testing::AnyNumber;
+using ::testing::Return;
+
+namespace webrtc {
+
+class MockStatsObserver : public CallStatsObserver {
+ public:
+  MockStatsObserver() {}
+  virtual ~MockStatsObserver() {}
+
+  MOCK_METHOD2(OnRttUpdate, void(int64_t, int64_t));
+};
+
+class CallStatsTest : public ::testing::Test {
+ public:
+  CallStatsTest() : fake_clock_(12345) {}
+
+ protected:
+  virtual void SetUp() { call_stats_.reset(new CallStats(&fake_clock_)); }
+  SimulatedClock fake_clock_;
+  rtc::scoped_ptr<CallStats> call_stats_;
+};
+
+TEST_F(CallStatsTest, AddAndTriggerCallback) {
+  MockStatsObserver stats_observer;
+  RtcpRttStats* rtcp_rtt_stats = call_stats_->rtcp_rtt_stats();
+  call_stats_->RegisterStatsObserver(&stats_observer);
+  fake_clock_.AdvanceTimeMilliseconds(1000);
+  EXPECT_EQ(0, rtcp_rtt_stats->LastProcessedRtt());
+
+  const int64_t kRtt = 25;
+  rtcp_rtt_stats->OnRttUpdate(kRtt);
+  EXPECT_CALL(stats_observer, OnRttUpdate(kRtt, kRtt)).Times(1);
+  call_stats_->Process();
+  EXPECT_EQ(kRtt, rtcp_rtt_stats->LastProcessedRtt());
+
+  const int64_t kRttTimeOutMs = 1500 + 10;
+  fake_clock_.AdvanceTimeMilliseconds(kRttTimeOutMs);
+  EXPECT_CALL(stats_observer, OnRttUpdate(_, _)).Times(0);
+  call_stats_->Process();
+  EXPECT_EQ(0, rtcp_rtt_stats->LastProcessedRtt());
+
+  call_stats_->DeregisterStatsObserver(&stats_observer);
+}
+
+TEST_F(CallStatsTest, ProcessTime) {
+  MockStatsObserver stats_observer;
+  call_stats_->RegisterStatsObserver(&stats_observer);
+  RtcpRttStats* rtcp_rtt_stats = call_stats_->rtcp_rtt_stats();
+  rtcp_rtt_stats->OnRttUpdate(100);
+
+  // Time isn't updated yet.
+  EXPECT_CALL(stats_observer, OnRttUpdate(_, _)).Times(0);
+  call_stats_->Process();
+
+  // Advance clock and verify we get an update.
+  fake_clock_.AdvanceTimeMilliseconds(1000);
+  EXPECT_CALL(stats_observer, OnRttUpdate(_, _)).Times(1);
+  call_stats_->Process();
+
+  // Advance clock just too little to get an update.
+  fake_clock_.AdvanceTimeMilliseconds(999);
+  rtcp_rtt_stats->OnRttUpdate(100);
+  EXPECT_CALL(stats_observer, OnRttUpdate(_, _)).Times(0);
+  call_stats_->Process();
+
+  // Advance enough to trigger a new update.
+  fake_clock_.AdvanceTimeMilliseconds(1);
+  EXPECT_CALL(stats_observer, OnRttUpdate(_, _)).Times(1);
+  call_stats_->Process();
+
+  call_stats_->DeregisterStatsObserver(&stats_observer);
+}
+
+// Verify all observers get correct estimates and observers can be added and
+// removed.
+TEST_F(CallStatsTest, MultipleObservers) {
+  MockStatsObserver stats_observer_1;
+  call_stats_->RegisterStatsObserver(&stats_observer_1);
+  // Add the second observer twice, there should still be only one report to the
+  // observer.
+  MockStatsObserver stats_observer_2;
+  call_stats_->RegisterStatsObserver(&stats_observer_2);
+  call_stats_->RegisterStatsObserver(&stats_observer_2);
+
+  RtcpRttStats* rtcp_rtt_stats = call_stats_->rtcp_rtt_stats();
+  const int64_t kRtt = 100;
+  rtcp_rtt_stats->OnRttUpdate(kRtt);
+
+  // Verify both observers are updated.
+  fake_clock_.AdvanceTimeMilliseconds(1000);
+  EXPECT_CALL(stats_observer_1, OnRttUpdate(kRtt, kRtt)).Times(1);
+  EXPECT_CALL(stats_observer_2, OnRttUpdate(kRtt, kRtt)).Times(1);
+  call_stats_->Process();
+
+  // Deregister the second observer and verify update is only sent to the first
+  // observer.
+  call_stats_->DeregisterStatsObserver(&stats_observer_2);
+  rtcp_rtt_stats->OnRttUpdate(kRtt);
+  fake_clock_.AdvanceTimeMilliseconds(1000);
+  EXPECT_CALL(stats_observer_1, OnRttUpdate(kRtt, kRtt)).Times(1);
+  EXPECT_CALL(stats_observer_2, OnRttUpdate(kRtt, kRtt)).Times(0);
+  call_stats_->Process();
+
+  // Deregister the first observer.
+  call_stats_->DeregisterStatsObserver(&stats_observer_1);
+  rtcp_rtt_stats->OnRttUpdate(kRtt);
+  fake_clock_.AdvanceTimeMilliseconds(1000);
+  EXPECT_CALL(stats_observer_1, OnRttUpdate(kRtt, kRtt)).Times(0);
+  EXPECT_CALL(stats_observer_2, OnRttUpdate(kRtt, kRtt)).Times(0);
+  call_stats_->Process();
+}
+
+// Verify increasing and decreasing rtt triggers callbacks with correct values.
+TEST_F(CallStatsTest, ChangeRtt) {
+  MockStatsObserver stats_observer;
+  call_stats_->RegisterStatsObserver(&stats_observer);
+  RtcpRttStats* rtcp_rtt_stats = call_stats_->rtcp_rtt_stats();
+
+  // Advance clock to be ready for an update.
+  fake_clock_.AdvanceTimeMilliseconds(1000);
+
+  // Set a first value and verify the callback is triggered.
+  const int64_t kFirstRtt = 100;
+  rtcp_rtt_stats->OnRttUpdate(kFirstRtt);
+  EXPECT_CALL(stats_observer, OnRttUpdate(kFirstRtt, kFirstRtt)).Times(1);
+  call_stats_->Process();
+
+  // Increase rtt and verify the new value is reported.
+  fake_clock_.AdvanceTimeMilliseconds(1000);
+  const int64_t kHighRtt = kFirstRtt + 20;
+  const int64_t kAvgRtt1 = 103;
+  rtcp_rtt_stats->OnRttUpdate(kHighRtt);
+  EXPECT_CALL(stats_observer, OnRttUpdate(kAvgRtt1, kHighRtt)).Times(1);
+  call_stats_->Process();
+
+  // Increase time enough for a new update, but not too much to make the
+  // rtt invalid. Report a lower rtt and verify the old/high value still is sent
+  // in the callback.
+  fake_clock_.AdvanceTimeMilliseconds(1000);
+  const int64_t kLowRtt = kFirstRtt - 20;
+  const int64_t kAvgRtt2 = 102;
+  rtcp_rtt_stats->OnRttUpdate(kLowRtt);
+  EXPECT_CALL(stats_observer, OnRttUpdate(kAvgRtt2, kHighRtt)).Times(1);
+  call_stats_->Process();
+
+  // Advance time to make the high report invalid, the lower rtt should now be
+  // in the callback.
+  fake_clock_.AdvanceTimeMilliseconds(1000);
+  const int64_t kAvgRtt3 = 95;
+  EXPECT_CALL(stats_observer, OnRttUpdate(kAvgRtt3, kLowRtt)).Times(1);
+  call_stats_->Process();
+
+  call_stats_->DeregisterStatsObserver(&stats_observer);
+}
+
+TEST_F(CallStatsTest, LastProcessedRtt) {
+  MockStatsObserver stats_observer;
+  call_stats_->RegisterStatsObserver(&stats_observer);
+  RtcpRttStats* rtcp_rtt_stats = call_stats_->rtcp_rtt_stats();
+  fake_clock_.AdvanceTimeMilliseconds(1000);
+
+  // Set a first values and verify that LastProcessedRtt initially returns the
+  // average rtt.
+  const int64_t kRttLow = 10;
+  const int64_t kRttHigh = 30;
+  const int64_t kAvgRtt = 20;
+  rtcp_rtt_stats->OnRttUpdate(kRttLow);
+  rtcp_rtt_stats->OnRttUpdate(kRttHigh);
+  EXPECT_CALL(stats_observer, OnRttUpdate(kAvgRtt, kRttHigh)).Times(1);
+  call_stats_->Process();
+  EXPECT_EQ(kAvgRtt, rtcp_rtt_stats->LastProcessedRtt());
+
+  // Update values and verify LastProcessedRtt.
+  fake_clock_.AdvanceTimeMilliseconds(1000);
+  rtcp_rtt_stats->OnRttUpdate(kRttLow);
+  rtcp_rtt_stats->OnRttUpdate(kRttHigh);
+  EXPECT_CALL(stats_observer, OnRttUpdate(kAvgRtt, kRttHigh)).Times(1);
+  call_stats_->Process();
+  EXPECT_EQ(kAvgRtt, rtcp_rtt_stats->LastProcessedRtt());
+
+  call_stats_->DeregisterStatsObserver(&stats_observer);
+}
+
+}  // namespace webrtc