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, 146 insertions, 0 deletions

diff --git a/webrtc/video/report_block_stats_unittest.cc b/webrtc/video/report_block_stats_unittest.cc
new file mode 100644
index 0000000000..5cde9004b1
--- /dev/null
+++ b/webrtc/video/report_block_stats_unittest.cc
@@ -0,0 +1,146 @@
+/*
+ *  Copyright (c) 2014 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/gtest/include/gtest/gtest.h"
+
+#include "webrtc/video/report_block_stats.h"
+
+namespace webrtc {
+
+class ReportBlockStatsTest : public ::testing::Test {
+ protected:
+  ReportBlockStatsTest() : kSsrc1(0x12345), kSsrc2(0x23456) {}
+
+  void SetUp() override {
+    // kSsrc1: block 1-3.
+    block1_1_.cumulativeLost = 10;
+    block1_1_.fractionLost = 123;
+    block1_1_.extendedHighSeqNum = 24000;
+    block1_1_.jitter = 777;
+    block1_1_.sourceSSRC = kSsrc1;
+    block1_2_.cumulativeLost = 15;
+    block1_2_.fractionLost = 0;
+    block1_2_.extendedHighSeqNum = 24100;
+    block1_2_.jitter = 222;
+    block1_2_.sourceSSRC = kSsrc1;
+    block1_3_.cumulativeLost = 50;
+    block1_3_.fractionLost = 0;
+    block1_3_.extendedHighSeqNum = 24200;
+    block1_3_.jitter = 333;
+    block1_3_.sourceSSRC = kSsrc1;
+    // kSsrc2: block 1,2.
+    block2_1_.cumulativeLost = 111;
+    block2_1_.fractionLost = 222;
+    block2_1_.extendedHighSeqNum = 8500;
+    block2_1_.jitter = 555;
+    block2_1_.sourceSSRC = kSsrc2;
+    block2_2_.cumulativeLost = 136;
+    block2_2_.fractionLost = 0;
+    block2_2_.extendedHighSeqNum = 8800;
+    block2_2_.jitter = 888;
+    block2_2_.sourceSSRC = kSsrc2;
+
+    ssrc1block1_.push_back(block1_1_);
+    ssrc1block2_.push_back(block1_2_);
+    ssrc12block1_.push_back(block1_1_);
+    ssrc12block1_.push_back(block2_1_);
+    ssrc12block2_.push_back(block1_2_);
+    ssrc12block2_.push_back(block2_2_);
+  }
+
+  RtcpStatistics RtcpReportBlockToRtcpStatistics(
+      const RTCPReportBlock& stats) {
+    RtcpStatistics block;
+    block.cumulative_lost = stats.cumulativeLost;
+    block.fraction_lost = stats.fractionLost;
+    block.extended_max_sequence_number = stats.extendedHighSeqNum;
+    block.jitter = stats.jitter;
+    return block;
+  }
+
+  const uint32_t kSsrc1;
+  const uint32_t kSsrc2;
+  RTCPReportBlock block1_1_;
+  RTCPReportBlock block1_2_;
+  RTCPReportBlock block1_3_;
+  RTCPReportBlock block2_1_;
+  RTCPReportBlock block2_2_;
+  std::vector<RTCPReportBlock> ssrc1block1_;
+  std::vector<RTCPReportBlock> ssrc1block2_;
+  std::vector<RTCPReportBlock> ssrc12block1_;
+  std::vector<RTCPReportBlock> ssrc12block2_;
+};
+
+TEST_F(ReportBlockStatsTest, AggregateAndStore_NoSsrc) {
+  ReportBlockStats stats;
+  std::vector<RTCPReportBlock> empty;
+  RTCPReportBlock aggregated = stats.AggregateAndStore(empty);
+  EXPECT_EQ(0U, aggregated.fractionLost);
+  EXPECT_EQ(0U, aggregated.cumulativeLost);
+  EXPECT_EQ(0U, aggregated.jitter);
+  EXPECT_EQ(0U, aggregated.extendedHighSeqNum);
+}
+
+TEST_F(ReportBlockStatsTest, AggregateAndStore_OneSsrc) {
+  ReportBlockStats stats;
+  RTCPReportBlock aggregated = stats.AggregateAndStore(ssrc1block1_);
+  // One ssrc, no aggregation done.
+  EXPECT_EQ(123U, aggregated.fractionLost);
+  EXPECT_EQ(10U, aggregated.cumulativeLost);
+  EXPECT_EQ(777U, aggregated.jitter);
+  EXPECT_EQ(24000U, aggregated.extendedHighSeqNum);
+
+  aggregated = stats.AggregateAndStore(ssrc1block2_);
+  EXPECT_EQ(0U, aggregated.fractionLost);
+  EXPECT_EQ(15U, aggregated.cumulativeLost);
+  EXPECT_EQ(222U, aggregated.jitter);
+  EXPECT_EQ(24100U, aggregated.extendedHighSeqNum);
+
+  // fl: 100 * (15-10) / (24100-24000) = 5%
+  EXPECT_EQ(5, stats.FractionLostInPercent());
+}
+
+TEST_F(ReportBlockStatsTest, AggregateAndStore_TwoSsrcs) {
+  ReportBlockStats stats;
+  RTCPReportBlock aggregated = stats.AggregateAndStore(ssrc12block1_);
+  EXPECT_EQ(0U, aggregated.fractionLost);
+  EXPECT_EQ(10U + 111U, aggregated.cumulativeLost);
+  EXPECT_EQ((777U + 555U) / 2, aggregated.jitter);
+  EXPECT_EQ(0U, aggregated.extendedHighSeqNum);
+
+  aggregated = stats.AggregateAndStore(ssrc12block2_);
+  // fl: 255 * ((15-10) + (136-111)) / ((24100-24000) + (8800-8500)) = 19
+  EXPECT_EQ(19U, aggregated.fractionLost);
+  EXPECT_EQ(15U + 136U, aggregated.cumulativeLost);
+  EXPECT_EQ((222U + 888U) / 2, aggregated.jitter);
+  EXPECT_EQ(0U, aggregated.extendedHighSeqNum);
+
+  // fl: 100 * ((15-10) + (136-111)) / ((24100-24000) + (8800-8500)) = 7%
+  EXPECT_EQ(7, stats.FractionLostInPercent());
+}
+
+TEST_F(ReportBlockStatsTest, StoreAndGetFractionLost) {
+  const uint32_t kRemoteSsrc = 1;
+  ReportBlockStats stats;
+  EXPECT_EQ(-1, stats.FractionLostInPercent());
+
+  // First block.
+  stats.Store(RtcpReportBlockToRtcpStatistics(block1_1_), kRemoteSsrc, kSsrc1);
+  EXPECT_EQ(-1, stats.FractionLostInPercent());
+  // fl: 100 * (15-10) / (24100-24000) = 5%
+  stats.Store(RtcpReportBlockToRtcpStatistics(block1_2_), kRemoteSsrc, kSsrc1);
+  EXPECT_EQ(5, stats.FractionLostInPercent());
+  // fl: 100 * (50-10) / (24200-24000) = 20%
+  stats.Store(RtcpReportBlockToRtcpStatistics(block1_3_), kRemoteSsrc, kSsrc1);
+  EXPECT_EQ(20, stats.FractionLostInPercent());
+}
+
+}  // namespace webrtc
+