BASELINE: Update Chromium to 85.0.4183.14085-based

Change-Id: Iaa42f4680837c57725b1344f108c0196741f6057
Reviewed-by: Allan Sandfeld Jensen <allan.jensen@qt.io>

7 files changed, 94 insertions, 19 deletions

diff --git a/chromium/third_party/webrtc/logging/BUILD.gn b/chromium/third_party/webrtc/logging/BUILD.gn
index 28176d2583a..c1edd69680c 100644
--- a/chromium/third_party/webrtc/logging/BUILD.gn
+++ b/chromium/third_party/webrtc/logging/BUILD.gn
@@ -53,8 +53,8 @@ rtc_library("rtc_event_pacing") {
   deps = [
     "../api:scoped_refptr",
     "../api/rtc_event_log",
-    "//third_party/abseil-cpp/absl/memory",
   ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/memory" ]
 }
 
 rtc_library("rtc_event_audio") {
@@ -75,8 +75,8 @@ rtc_library("rtc_event_audio") {
     "../api/rtc_event_log",
     "../modules/audio_coding:audio_network_adaptor_config",
     "../rtc_base:checks",
-    "//third_party/abseil-cpp/absl/memory",
   ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/memory" ]
 }
 
 rtc_library("rtc_event_bwe") {
@@ -101,6 +101,8 @@ rtc_library("rtc_event_bwe") {
     "../api/rtc_event_log",
     "../api/units:data_rate",
     "../modules/remote_bitrate_estimator",
+  ]
+  absl_deps = [
     "//third_party/abseil-cpp/absl/memory",
     "//third_party/abseil-cpp/absl/types:optional",
   ]
@@ -119,6 +121,8 @@ rtc_library("rtc_event_generic_packet_events") {
   deps = [
     "../api/rtc_event_log",
     "../rtc_base:timeutils",
+  ]
+  absl_deps = [
     "//third_party/abseil-cpp/absl/memory",
     "//third_party/abseil-cpp/absl/types:optional",
   ]
@@ -143,8 +147,8 @@ rtc_library("rtc_event_rtp_rtcp") {
     "../modules/rtp_rtcp:rtp_rtcp_format",
     "../rtc_base:checks",
     "../rtc_base:rtc_base_approved",
-    "//third_party/abseil-cpp/absl/memory",
   ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/memory" ]
 }
 
 rtc_library("rtc_event_video") {
@@ -160,8 +164,8 @@ rtc_library("rtc_event_video") {
     "../api:scoped_refptr",
     "../api/rtc_event_log",
     "../rtc_base:checks",
-    "//third_party/abseil-cpp/absl/memory",
   ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/memory" ]
 }
 
 # TODO(eladalon): Break down into (1) encoder and (2) decoder; we don't need
@@ -187,6 +191,8 @@ rtc_library("rtc_event_log_impl_encoder") {
     "../rtc_base:checks",
     "../rtc_base:ignore_wundef",
     "../rtc_base:rtc_base_approved",
+  ]
+  absl_deps = [
     "//third_party/abseil-cpp/absl/memory",
     "//third_party/abseil-cpp/absl/strings",
     "//third_party/abseil-cpp/absl/types:optional",
@@ -245,8 +251,8 @@ if (rtc_enable_protobuf) {
       "../rtc_base:rtc_task_queue",
       "../rtc_base:safe_minmax",
       "../rtc_base/synchronization:sequence_checker",
-      "//third_party/abseil-cpp/absl/types:optional",
     ]
+    absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
   }
 }
 
@@ -318,6 +324,8 @@ if (rtc_enable_protobuf) {
       "../rtc_base:protobuf_utils",
       "../rtc_base:rtc_base_approved",
       "../rtc_base:rtc_numerics",
+    ]
+    absl_deps = [
       "//third_party/abseil-cpp/absl/memory",
       "//third_party/abseil-cpp/absl/types:optional",
     ]
@@ -369,6 +377,8 @@ if (rtc_enable_protobuf) {
         "../test:fileutils",
         "../test:test_support",
         "//testing/gtest",
+      ]
+      absl_deps = [
         "//third_party/abseil-cpp/absl/memory",
         "//third_party/abseil-cpp/absl/types:optional",
       ]
@@ -417,8 +427,8 @@ rtc_library("ice_log") {
     "../api:libjingle_peerconnection_api",  # For api/dtls_transport_interface.h
     "../api/rtc_event_log",
     "../rtc_base:rtc_base_approved",
-    "//third_party/abseil-cpp/absl/memory",
   ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/memory" ]
 }
 
 if (rtc_include_tests) {
diff --git a/chromium/third_party/webrtc/logging/rtc_event_log/encoder/blob_encoding.h b/chromium/third_party/webrtc/logging/rtc_event_log/encoder/blob_encoding.h
index 4a38dc5d0b6..b5b589aaf6a 100644
--- a/chromium/third_party/webrtc/logging/rtc_event_log/encoder/blob_encoding.h
+++ b/chromium/third_party/webrtc/logging/rtc_event_log/encoder/blob_encoding.h
@@ -43,7 +43,7 @@ namespace webrtc {
 //
 // Note that the returned std::string might have been reserved for significantly
 // more memory than it ends up using. If the caller to EncodeBlobs() intends
-// to store the result long-term, he should consider shrink_to_fit()-ing it.
+// to store the result long-term, they should consider shrink_to_fit()-ing it.
 std::string EncodeBlobs(const std::vector<std::string>& blobs);
 std::vector<absl::string_view> DecodeBlobs(absl::string_view encoded_blobs,
                                            size_t num_of_blobs);
diff --git a/chromium/third_party/webrtc/logging/rtc_event_log/encoder/rtc_event_log_encoder_unittest.cc b/chromium/third_party/webrtc/logging/rtc_event_log/encoder/rtc_event_log_encoder_unittest.cc
index 30596216907..cf85775f527 100644
--- a/chromium/third_party/webrtc/logging/rtc_event_log/encoder/rtc_event_log_encoder_unittest.cc
+++ b/chromium/third_party/webrtc/logging/rtc_event_log/encoder/rtc_event_log_encoder_unittest.cc
@@ -675,13 +675,18 @@ TEST_P(RtcEventLogEncoderTest, RtcEventLoggingStarted) {
 }
 
 TEST_P(RtcEventLogEncoderTest, RtcEventLoggingStopped) {
-  const int64_t timestamp_us = rtc::TimeMicros();
-  std::string encoded = encoder_->EncodeLogEnd(timestamp_us);
+  const int64_t start_timestamp_us = rtc::TimeMicros();
+  const int64_t start_utc_time_us = rtc::TimeUTCMicros();
+  std::string encoded =
+      encoder_->EncodeLogStart(start_timestamp_us, start_utc_time_us);
+
+  const int64_t stop_timestamp_us = rtc::TimeMicros();
+  encoded += encoder_->EncodeLogEnd(stop_timestamp_us);
   ASSERT_TRUE(parsed_log_.ParseString(encoded).ok());
   const auto& stop_log_events = parsed_log_.stop_log_events();
 
   ASSERT_EQ(stop_log_events.size(), 1u);
-  verifier_.VerifyLoggedStopEvent(timestamp_us, stop_log_events[0]);
+  verifier_.VerifyLoggedStopEvent(stop_timestamp_us, stop_log_events[0]);
 }
 
 // TODO(eladalon/terelius): Test with multiple events in the batch.
diff --git a/chromium/third_party/webrtc/logging/rtc_event_log/mock/mock_rtc_event_log.h b/chromium/third_party/webrtc/logging/rtc_event_log/mock/mock_rtc_event_log.h
index 66a2065ed47..646831de272 100644
--- a/chromium/third_party/webrtc/logging/rtc_event_log/mock/mock_rtc_event_log.h
+++ b/chromium/third_party/webrtc/logging/rtc_event_log/mock/mock_rtc_event_log.h
@@ -21,20 +21,20 @@ namespace webrtc {
 class MockRtcEventLog : public RtcEventLog {
  public:
   MockRtcEventLog();
-  ~MockRtcEventLog();
+  ~MockRtcEventLog() override;
 
-  virtual bool StartLogging(std::unique_ptr<RtcEventLogOutput> output,
-                            int64_t output_period_ms) {
-    return StartLoggingProxy(output.get(), output_period_ms);
-  }
-  MOCK_METHOD2(StartLoggingProxy, bool(RtcEventLogOutput*, int64_t));
+  MOCK_METHOD(bool,
+              StartLogging,
+              (std::unique_ptr<RtcEventLogOutput> output,
+               int64_t output_period_ms),
+              (override));
 
-  MOCK_METHOD0(StopLogging, void());
+  MOCK_METHOD(void, StopLogging, (), (override));
 
-  virtual void Log(std::unique_ptr<RtcEvent> event) {
+  void Log(std::unique_ptr<RtcEvent> event) override {
     return LogProxy(event.get());
   }
-  MOCK_METHOD1(LogProxy, void(RtcEvent*));
+  MOCK_METHOD(void, LogProxy, (RtcEvent*));
 };
 
 }  // namespace webrtc
diff --git a/chromium/third_party/webrtc/logging/rtc_event_log/rtc_event_log_parser.cc b/chromium/third_party/webrtc/logging/rtc_event_log/rtc_event_log_parser.cc
index 4016f84339a..c88207607cb 100644
--- a/chromium/third_party/webrtc/logging/rtc_event_log/rtc_event_log_parser.cc
+++ b/chromium/third_party/webrtc/logging/rtc_event_log/rtc_event_log_parser.cc
@@ -1076,6 +1076,7 @@ void ParsedRtcEventLog::Clear() {
 
   first_timestamp_ = std::numeric_limits<int64_t>::max();
   last_timestamp_ = std::numeric_limits<int64_t>::min();
+  first_log_segment_ = LogSegment(0, std::numeric_limits<int64_t>::max());
 
   incoming_rtp_extensions_maps_.clear();
   outgoing_rtp_extensions_maps_.clear();
@@ -1214,6 +1215,38 @@ ParsedRtcEventLog::ParseStatus ParsedRtcEventLog::ParseStream(
   StoreFirstAndLastTimestamp(generic_packets_sent_);
   StoreFirstAndLastTimestamp(generic_packets_received_);
   StoreFirstAndLastTimestamp(generic_acks_received_);
+  StoreFirstAndLastTimestamp(remote_estimate_events_);
+
+  // Stop events could be missing due to file size limits. If so, use the
+  // last event, or the next start timestamp if available.
+  // TODO(terelius): This could be improved. Instead of using the next start
+  // event, we could use the timestamp of the the last previous regular event.
+  auto start_iter = start_log_events().begin();
+  auto stop_iter = stop_log_events().begin();
+  int64_t start_us = first_timestamp();
+  int64_t next_start_us = std::numeric_limits<int64_t>::max();
+  int64_t stop_us = std::numeric_limits<int64_t>::max();
+  if (start_iter != start_log_events().end()) {
+    start_us = std::min(start_us, start_iter->log_time_us());
+    ++start_iter;
+    if (start_iter != start_log_events().end())
+      next_start_us = start_iter->log_time_us();
+  }
+  if (stop_iter != stop_log_events().end()) {
+    stop_us = stop_iter->log_time_us();
+  }
+  stop_us = std::min(stop_us, next_start_us);
+  if (stop_us == std::numeric_limits<int64_t>::max() &&
+      last_timestamp() != std::numeric_limits<int64_t>::min()) {
+    stop_us = last_timestamp();
+  }
+  RTC_PARSE_CHECK_OR_RETURN_LE(start_us, stop_us);
+  first_log_segment_ = LogSegment(start_us, stop_us);
+
+  if (first_timestamp_ == std::numeric_limits<int64_t>::max() &&
+      last_timestamp_ == std::numeric_limits<int64_t>::min()) {
+    first_timestamp_ = last_timestamp_ = 0;
+  }
 
   return status;
 }
diff --git a/chromium/third_party/webrtc/logging/rtc_event_log/rtc_event_log_parser.h b/chromium/third_party/webrtc/logging/rtc_event_log/rtc_event_log_parser.h
index 7a162af897e..8d3351e815f 100644
--- a/chromium/third_party/webrtc/logging/rtc_event_log/rtc_event_log_parser.h
+++ b/chromium/third_party/webrtc/logging/rtc_event_log/rtc_event_log_parser.h
@@ -11,6 +11,7 @@
 #define LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_PARSER_H_
 
 #include <iterator>
+#include <limits>
 #include <map>
 #include <set>
 #include <sstream>  // no-presubmit-check TODO(webrtc:8982)
@@ -193,6 +194,8 @@ class PacketView {
 
   size_t size() const { return num_elements_; }
 
+  bool empty() const { return num_elements_ == 0; }
+
   T& operator[](size_t i) {
     auto elem_ptr = data_ + i * element_size_;
     return *reinterpret_cast<T*>(elem_ptr);
@@ -330,6 +333,20 @@ class ParsedRtcEventLog {
     PacketView<const LoggedRtpPacket> packet_view;
   };
 
+  class LogSegment {
+   public:
+    LogSegment(int64_t start_time_us, int64_t stop_time_us)
+        : start_time_us_(start_time_us), stop_time_us_(stop_time_us) {}
+    int64_t start_time_ms() const { return start_time_us_ / 1000; }
+    int64_t start_time_us() const { return start_time_us_; }
+    int64_t stop_time_ms() const { return stop_time_us_ / 1000; }
+    int64_t stop_time_us() const { return stop_time_us_; }
+
+   private:
+    int64_t start_time_us_;
+    int64_t stop_time_us_;
+  };
+
   static webrtc::RtpHeaderExtensionMap GetDefaultHeaderExtensionMap();
 
   explicit ParsedRtcEventLog(
@@ -597,6 +614,8 @@ class ParsedRtcEventLog {
   int64_t first_timestamp() const { return first_timestamp_; }
   int64_t last_timestamp() const { return last_timestamp_; }
 
+  const LogSegment& first_log_segment() const { return first_log_segment_; }
+
   std::vector<LoggedPacketInfo> GetPacketInfos(PacketDirection direction) const;
   std::vector<LoggedPacketInfo> GetIncomingPacketInfos() const {
     return GetPacketInfos(kIncomingPacket);
@@ -850,6 +869,9 @@ class ParsedRtcEventLog {
   int64_t first_timestamp_;
   int64_t last_timestamp_;
 
+  LogSegment first_log_segment_ =
+      LogSegment(0, std::numeric_limits<int64_t>::max());
+
   // The extension maps are mutable to allow us to insert the default
   // configuration when parsing an RTP header for an unconfigured stream.
   // TODO(terelius): This is only used for the legacy format. Remove once we've
diff --git a/chromium/third_party/webrtc/logging/rtc_event_log/rtc_event_log_unittest.cc b/chromium/third_party/webrtc/logging/rtc_event_log/rtc_event_log_unittest.cc
index 579c6528804..e785d6160a0 100644
--- a/chromium/third_party/webrtc/logging/rtc_event_log/rtc_event_log_unittest.cc
+++ b/chromium/third_party/webrtc/logging/rtc_event_log/rtc_event_log_unittest.cc
@@ -739,6 +739,11 @@ void RtcEventLogSession::ReadAndVerifyLog() {
   EXPECT_EQ(first_timestamp_ms_, parsed_log.first_timestamp() / 1000);
   EXPECT_EQ(last_timestamp_ms_, parsed_log.last_timestamp() / 1000);
 
+  EXPECT_EQ(parsed_log.first_log_segment().start_time_ms(),
+            std::min(start_time_us_ / 1000, first_timestamp_ms_));
+  EXPECT_EQ(parsed_log.first_log_segment().stop_time_ms(),
+            stop_time_us_ / 1000);
+
   // Clean up temporary file - can be pretty slow.
   remove(temp_filename_.c_str());
 }