diff --git a/logging/rtc_event_log/rtc_event_log_parser.cc b/logging/rtc_event_log/rtc_event_log_parser.cc
index 4016f84..a3982ba 100644
--- a/logging/rtc_event_log/rtc_event_log_parser.cc
+++ b/logging/rtc_event_log/rtc_event_log_parser.cc
@@ -1215,6 +1215,32 @@
   StoreFirstAndLastTimestamp(generic_packets_received_);
   StoreFirstAndLastTimestamp(generic_acks_received_);
 
+  // TODO(terelius): This should be cleaned up. We could also handle
+  // a "missing" end event, by inserting the last previous regular
+  // event rather than the next start event.
+  auto start_iter = start_log_events().begin();
+  auto stop_iter = stop_log_events().begin();
+  while (start_iter != start_log_events().end()) {
+    int64_t start_us = start_iter->log_time_us();
+    ++start_iter;
+    absl::optional<int64_t> next_start_us;
+    if (start_iter != start_log_events().end())
+      next_start_us.emplace(start_iter->log_time_us());
+    if (stop_iter != stop_log_events().end() &&
+        stop_iter->log_time_us() <=
+            next_start_us.value_or(std::numeric_limits<int64_t>::max())) {
+      int64_t stop_us = stop_iter->log_time_us();
+      RTC_PARSE_CHECK_OR_RETURN_LE(start_us, stop_us);
+      log_segments_.emplace_back(start_us, stop_us);
+      ++stop_iter;
+    } else {
+      // We're missing an end event. Assume that it occurred just before the
+      // next start.
+      log_segments_.emplace_back(start_us,
+                                 next_start_us.value_or(last_timestamp()));
+    }
+  }
+
   return status;
 }
 
diff --git a/logging/rtc_event_log/rtc_event_log_parser.h b/logging/rtc_event_log/rtc_event_log_parser.h
index 7a162af..be6b99a 100644
--- a/logging/rtc_event_log/rtc_event_log_parser.h
+++ b/logging/rtc_event_log/rtc_event_log_parser.h
@@ -330,6 +330,20 @@
     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 +611,8 @@
   int64_t first_timestamp() const { return first_timestamp_; }
   int64_t last_timestamp() const { return last_timestamp_; }
 
+  const std::vector<LogSegment>& log_segments() const { return log_segments_; }
+
   std::vector<LoggedPacketInfo> GetPacketInfos(PacketDirection direction) const;
   std::vector<LoggedPacketInfo> GetIncomingPacketInfos() const {
     return GetPacketInfos(kIncomingPacket);
@@ -850,6 +866,9 @@
   int64_t first_timestamp_;
   int64_t last_timestamp_;
 
+  // Stores the start and end timestamp for each log segments.
+  std::vector<LogSegment> log_segments_;
+
   // 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/logging/rtc_event_log/rtc_event_log_unittest.cc b/logging/rtc_event_log/rtc_event_log_unittest.cc
index 579c652..5989006 100644
--- a/logging/rtc_event_log/rtc_event_log_unittest.cc
+++ b/logging/rtc_event_log/rtc_event_log_unittest.cc
@@ -739,6 +739,11 @@
   EXPECT_EQ(first_timestamp_ms_, parsed_log.first_timestamp() / 1000);
   EXPECT_EQ(last_timestamp_ms_, parsed_log.last_timestamp() / 1000);
 
+  ASSERT_EQ(parsed_log.log_segments().size(), 1u);
+  EXPECT_EQ(parsed_log.log_segments()[0].start_time_ms(),
+            start_time_us_ / 1000);
+  EXPECT_EQ(parsed_log.log_segments()[0].stop_time_ms(), stop_time_us_ / 1000);
+
   // Clean up temporary file - can be pretty slow.
   remove(temp_filename_.c_str());
 }
diff --git a/rtc_tools/BUILD.gn b/rtc_tools/BUILD.gn
index f293853..3d39845 100644
--- a/rtc_tools/BUILD.gn
+++ b/rtc_tools/BUILD.gn
@@ -319,8 +319,12 @@
     rtc_library("event_log_visualizer_utils") {
       visibility = [ "*" ]
       sources = [
+        "rtc_event_log_visualizer/alerts.cc",
+        "rtc_event_log_visualizer/alerts.h",
         "rtc_event_log_visualizer/analyzer.cc",
         "rtc_event_log_visualizer/analyzer.h",
+        "rtc_event_log_visualizer/analyzer_common.cc",
+        "rtc_event_log_visualizer/analyzer_common.h",
         "rtc_event_log_visualizer/log_simulation.cc",
         "rtc_event_log_visualizer/log_simulation.h",
         "rtc_event_log_visualizer/plot_base.cc",
@@ -329,7 +333,6 @@
         "rtc_event_log_visualizer/plot_protobuf.h",
         "rtc_event_log_visualizer/plot_python.cc",
         "rtc_event_log_visualizer/plot_python.h",
-        "rtc_event_log_visualizer/triage_notifications.h",
       ]
       deps = [
         ":chart_proto",
diff --git a/rtc_tools/rtc_event_log_visualizer/alerts.cc b/rtc_tools/rtc_event_log_visualizer/alerts.cc
new file mode 100644
index 0000000..da059cb
--- /dev/null
+++ b/rtc_tools/rtc_event_log_visualizer/alerts.cc
@@ -0,0 +1,228 @@
+/*
+ *  Copyright (c) 2020 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 "rtc_tools/rtc_event_log_visualizer/alerts.h"
+
+#include <stdio.h>
+
+#include <algorithm>
+#include <limits>
+#include <map>
+#include <string>
+
+#include "logging/rtc_event_log/rtc_event_processor.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/format_macros.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/sequence_number_util.h"
+#include "rtc_base/strings/string_builder.h"
+
+namespace webrtc {
+
+void TriageHelper::Print(FILE* file) {
+  fprintf(file, "========== TRIAGE NOTIFICATIONS ==========\n");
+  for (const auto& alert : triage_alerts_) {
+    fprintf(file, "%d %s. First occurence at %3.3lf\n", alert.second.count,
+            alert.second.explanation.c_str(), alert.second.first_occurence);
+  }
+  fprintf(file, "========== END TRIAGE NOTIFICATIONS ==========\n");
+}
+
+void TriageHelper::AnalyzeStreamGaps(const ParsedRtcEventLog& parsed_log,
+                                     PacketDirection direction) {
+  // With 100 packets/s (~800kbps), false positives would require 10 s without
+  // data.
+  constexpr int64_t kMaxSeqNumJump = 1000;
+  // With a 90 kHz clock, false positives would require 10 s without data.
+  constexpr int64_t kMaxCaptureTimeJump = 900000;
+
+  std::string seq_num_explanation =
+      direction == kIncomingPacket
+          ? "Incoming RTP sequence number jumps more than 1000. Counter may "
+            "have been reset or rewritten incorrectly in a group call."
+          : "Outgoing RTP sequence number jumps more than 1000. Counter may "
+            "have been reset.";
+  std::string capture_time_explanation =
+      direction == kIncomingPacket ? "Incoming capture time jumps more than "
+                                     "10s. Clock might have been reset."
+                                   : "Outgoing capture time jumps more than "
+                                     "10s. Clock might have been reset.";
+  TriageAlertType seq_num_alert = direction == kIncomingPacket
+                                      ? TriageAlertType::kIncomingSeqNumJump
+                                      : TriageAlertType::kOutgoingSeqNumJump;
+  TriageAlertType capture_time_alert =
+      direction == kIncomingPacket ? TriageAlertType::kIncomingCaptureTimeJump
+                                   : TriageAlertType::kOutgoingCaptureTimeJump;
+
+  const int64_t segment_end_us =
+      parsed_log.log_segments().empty()
+          ? std::numeric_limits<int64_t>::max()
+          : parsed_log.log_segments().front().stop_time_us();
+
+  // Check for gaps in sequence numbers and capture timestamps.
+  for (const auto& stream : parsed_log.rtp_packets_by_ssrc(direction)) {
+    if (IsRtxSsrc(parsed_log, direction, stream.ssrc)) {
+      continue;
+    }
+    SeqNumUnwrapper<uint16_t> seq_num_unwrapper;
+    absl::optional<int64_t> last_seq_num;
+    SeqNumUnwrapper<uint32_t> capture_time_unwrapper;
+    absl::optional<int64_t> last_capture_time;
+    for (const auto& packet : stream.packet_view) {
+      if (packet.log_time_us() > segment_end_us) {
+        // Only process the first (LOG_START, LOG_END) segment.
+        break;
+      }
+
+      int64_t seq_num = seq_num_unwrapper.Unwrap(packet.header.sequenceNumber);
+      if (last_seq_num.has_value() &&
+          std::abs(seq_num - last_seq_num.value()) > kMaxSeqNumJump) {
+        Alert(seq_num_alert, config_.GetCallTimeSec(packet.log_time_us()),
+              seq_num_explanation);
+      }
+      last_seq_num.emplace(seq_num);
+
+      int64_t capture_time =
+          capture_time_unwrapper.Unwrap(packet.header.timestamp);
+      if (last_capture_time.has_value() &&
+          std::abs(capture_time - last_capture_time.value()) >
+              kMaxCaptureTimeJump) {
+        Alert(capture_time_alert, config_.GetCallTimeSec(packet.log_time_us()),
+              capture_time_explanation);
+      }
+      last_capture_time.emplace(capture_time);
+    }
+  }
+}
+
+void TriageHelper::AnalyzeTransmissionGaps(const ParsedRtcEventLog& parsed_log,
+                                           PacketDirection direction) {
+  constexpr int64_t kMaxRtpTransmissionGap = 500000;
+  constexpr int64_t kMaxRtcpTransmissionGap = 2000000;
+  std::string rtp_explanation =
+      direction == kIncomingPacket
+          ? "No RTP packets received for more than 500ms. This indicates a "
+            "network problem. Temporary video freezes and choppy or robotic "
+            "audio is unavoidable. Unnecessary BWE drops is a known issue."
+          : "No RTP packets sent for more than 500 ms. This might be an issue "
+            "with the pacer.";
+  std::string rtcp_explanation =
+      direction == kIncomingPacket
+          ? "No RTCP packets received for more than 2 s. Could be a longer "
+            "connection outage"
+          : "No RTCP sent for more than 2 s. This is most likely a bug.";
+  TriageAlertType rtp_alert = direction == kIncomingPacket
+                                  ? TriageAlertType::kIncomingRtpGap
+                                  : TriageAlertType::kOutgoingRtpGap;
+  TriageAlertType rtcp_alert = direction == kIncomingPacket
+                                   ? TriageAlertType::kIncomingRtcpGap
+                                   : TriageAlertType::kOutgoingRtcpGap;
+
+  const int64_t segment_end_us =
+      parsed_log.log_segments().empty()
+          ? std::numeric_limits<int64_t>::max()
+          : parsed_log.log_segments().front().stop_time_us();
+
+  // TODO(terelius): The parser could provide a list of all packets, ordered
+  // by time, for each direction.
+  std::multimap<int64_t, const LoggedRtpPacket*> rtp_in_direction;
+  for (const auto& stream : parsed_log.rtp_packets_by_ssrc(direction)) {
+    for (const LoggedRtpPacket& rtp_packet : stream.packet_view)
+      rtp_in_direction.emplace(rtp_packet.log_time_us(), &rtp_packet);
+  }
+  absl::optional<int64_t> last_rtp_time;
+  for (const auto& kv : rtp_in_direction) {
+    int64_t timestamp = kv.first;
+    if (timestamp > segment_end_us) {
+      // Only process the first (LOG_START, LOG_END) segment.
+      break;
+    }
+    int64_t duration = timestamp - last_rtp_time.value_or(0);
+    if (last_rtp_time.has_value() && duration > kMaxRtpTransmissionGap) {
+      // No packet sent/received for more than 500 ms.
+      Alert(rtp_alert, config_.GetCallTimeSec(timestamp), rtp_explanation);
+    }
+    last_rtp_time.emplace(timestamp);
+  }
+
+  absl::optional<int64_t> last_rtcp_time;
+  if (direction == kIncomingPacket) {
+    for (const auto& rtcp : parsed_log.incoming_rtcp_packets()) {
+      if (rtcp.log_time_us() > segment_end_us) {
+        // Only process the first (LOG_START, LOG_END) segment.
+        break;
+      }
+      int64_t duration = rtcp.log_time_us() - last_rtcp_time.value_or(0);
+      if (last_rtcp_time.has_value() && duration > kMaxRtcpTransmissionGap) {
+        // No feedback sent/received for more than 2000 ms.
+        Alert(rtcp_alert, config_.GetCallTimeSec(rtcp.log_time_us()),
+              rtcp_explanation);
+      }
+      last_rtcp_time.emplace(rtcp.log_time_us());
+    }
+  } else {
+    for (const auto& rtcp : parsed_log.outgoing_rtcp_packets()) {
+      if (rtcp.log_time_us() > segment_end_us) {
+        // Only process the first (LOG_START, LOG_END) segment.
+        break;
+      }
+      int64_t duration = rtcp.log_time_us() - last_rtcp_time.value_or(0);
+      if (last_rtcp_time.has_value() && duration > kMaxRtcpTransmissionGap) {
+        // No feedback sent/received for more than 2000 ms.
+        Alert(rtcp_alert, config_.GetCallTimeSec(rtcp.log_time_us()),
+              rtcp_explanation);
+      }
+      last_rtcp_time.emplace(rtcp.log_time_us());
+    }
+  }
+}
+
+// TODO(terelius): Notifications could possibly be generated by the same code
+// that produces the graphs. There is some code duplication that could be
+// avoided, but that might be solved anyway when we move functionality from the
+// analyzer to the parser.
+void TriageHelper::AnalyzeLog(const ParsedRtcEventLog& parsed_log) {
+  AnalyzeStreamGaps(parsed_log, kIncomingPacket);
+  AnalyzeStreamGaps(parsed_log, kOutgoingPacket);
+  AnalyzeTransmissionGaps(parsed_log, kIncomingPacket);
+  AnalyzeTransmissionGaps(parsed_log, kOutgoingPacket);
+
+  const int64_t segment_end_us =
+      parsed_log.log_segments().empty()
+          ? std::numeric_limits<int64_t>::max()
+          : parsed_log.log_segments().front().stop_time_us();
+
+  int64_t first_occurence = parsed_log.last_timestamp();
+  constexpr double kMaxLossFraction = 0.05;
+  // Loss feedback
+  int64_t total_lost_packets = 0;
+  int64_t total_expected_packets = 0;
+  for (auto& bwe_update : parsed_log.bwe_loss_updates()) {
+    if (bwe_update.log_time_us() > segment_end_us) {
+      // Only process the first (LOG_START, LOG_END) segment.
+      break;
+    }
+    int64_t lost_packets = static_cast<double>(bwe_update.fraction_lost) / 255 *
+                           bwe_update.expected_packets;
+    total_lost_packets += lost_packets;
+    total_expected_packets += bwe_update.expected_packets;
+    if (bwe_update.fraction_lost >= 255 * kMaxLossFraction) {
+      first_occurence = std::min(first_occurence, bwe_update.log_time_us());
+    }
+  }
+  double avg_outgoing_loss =
+      static_cast<double>(total_lost_packets) / total_expected_packets;
+  if (avg_outgoing_loss > kMaxLossFraction) {
+    Alert(TriageAlertType::kOutgoingHighLoss, first_occurence,
+          "More than 5% of outgoing packets lost.");
+  }
+}
+
+}  // namespace webrtc
diff --git a/rtc_tools/rtc_event_log_visualizer/alerts.h b/rtc_tools/rtc_event_log_visualizer/alerts.h
new file mode 100644
index 0000000..9e559e6
--- /dev/null
+++ b/rtc_tools/rtc_event_log_visualizer/alerts.h
@@ -0,0 +1,86 @@
+/*
+ *  Copyright (c) 2020 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.
+ */
+
+#ifndef RTC_TOOLS_RTC_EVENT_LOG_VISUALIZER_ALERTS_H_
+#define RTC_TOOLS_RTC_EVENT_LOG_VISUALIZER_ALERTS_H_
+
+#include <stdio.h>
+
+#include <map>
+#include <string>
+#include <utility>
+
+#include "absl/strings/string_view.h"
+#include "logging/rtc_event_log/rtc_event_log_parser.h"
+#include "rtc_base/constructor_magic.h"
+#include "rtc_tools/rtc_event_log_visualizer/analyzer_common.h"
+
+namespace webrtc {
+
+enum class TriageAlertType {
+  kUnknown = 0,
+  kIncomingRtpGap,
+  kOutgoingRtpGap,
+  kIncomingRtcpGap,
+  kOutgoingRtcpGap,
+  kIncomingSeqNumJump,
+  kOutgoingSeqNumJump,
+  kIncomingCaptureTimeJump,
+  kOutgoingCaptureTimeJump,
+  kOutgoingHighLoss,
+  kLast,
+};
+
+struct TriageAlert {
+  TriageAlertType type = TriageAlertType::kUnknown;
+  int count = 0;
+  float first_occurence = -1;
+  std::string explanation;
+};
+
+class TriageHelper {
+ public:
+  explicit TriageHelper(const AnalyzerConfig& config) : config_(config) {}
+
+  void AnalyzeLog(const ParsedRtcEventLog& parsed_log);
+
+  void AnalyzeStreamGaps(const ParsedRtcEventLog& parsed_log,
+                         PacketDirection direction);
+  void AnalyzeTransmissionGaps(const ParsedRtcEventLog& parsed_log,
+                               PacketDirection direction);
+  void Print(FILE* file);
+
+ private:
+  AnalyzerConfig config_;
+  std::map<TriageAlertType, TriageAlert> triage_alerts_;
+
+  void Alert(TriageAlertType type,
+             float time_seconds,
+             absl::string_view explanation) {
+    std::map<TriageAlertType, TriageAlert>::iterator it =
+        triage_alerts_.find(type);
+
+    if (it == triage_alerts_.end()) {
+      TriageAlert alert;
+      alert.type = type;
+      alert.first_occurence = time_seconds;
+      alert.count = 1;
+      alert.explanation = std::string(explanation);
+      triage_alerts_.insert(std::make_pair(type, alert));
+    } else {
+      it->second.count += 1;
+    }
+  }
+  RTC_DISALLOW_COPY_AND_ASSIGN(TriageHelper);
+};
+
+}  // namespace webrtc
+
+#endif  // RTC_TOOLS_RTC_EVENT_LOG_VISUALIZER_ALERTS_H_
diff --git a/rtc_tools/rtc_event_log_visualizer/analyzer.cc b/rtc_tools/rtc_event_log_visualizer/analyzer.cc
index 9fcb510..9a9a455 100644
--- a/rtc_tools/rtc_event_log_visualizer/analyzer.cc
+++ b/rtc_tools/rtc_event_log_visualizer/analyzer.cc
@@ -465,31 +465,14 @@
     config_.begin_time_ = config_.end_time_ = 0;
   }
 
-  const auto& log_start_events = parsed_log_.start_log_events();
-  const auto& log_end_events = parsed_log_.stop_log_events();
-  auto start_iter = log_start_events.begin();
-  auto end_iter = log_end_events.begin();
-  while (start_iter != log_start_events.end()) {
-    int64_t start = start_iter->log_time_us();
-    ++start_iter;
-    absl::optional<int64_t> next_start;
-    if (start_iter != log_start_events.end())
-      next_start.emplace(start_iter->log_time_us());
-    if (end_iter != log_end_events.end() &&
-        end_iter->log_time_us() <=
-            next_start.value_or(std::numeric_limits<int64_t>::max())) {
-      int64_t end = end_iter->log_time_us();
-      RTC_DCHECK_LE(start, end);
-      log_segments_.push_back(std::make_pair(start, end));
-      ++end_iter;
-    } else {
-      // we're missing an end event. Assume that it occurred just before the
-      // next start.
-      log_segments_.push_back(
-          std::make_pair(start, next_start.value_or(config_.end_time_)));
-    }
-  }
-  RTC_LOG(LS_INFO) << "Found " << log_segments_.size()
+  RTC_LOG(LS_INFO) << "Found " << parsed_log_.log_segments().size()
+                   << " (LOG_START, LOG_END) segments in log.";
+}
+
+EventLogAnalyzer::EventLogAnalyzer(const ParsedRtcEventLog& log,
+                                   const AnalyzerConfig& config)
+    : parsed_log_(log), config_(config) {
+  RTC_LOG(LS_INFO) << "Found " << parsed_log_.log_segments().size()
                    << " (LOG_START, LOG_END) segments in log.";
 }
 
@@ -527,7 +510,7 @@
       continue;
     }
 
-    TimeSeries time_series(GetStreamName(direction, stream.ssrc),
+    TimeSeries time_series(GetStreamName(parsed_log_, direction, stream.ssrc),
                            LineStyle::kBar);
     auto GetPacketSize = [](const LoggedRtpPacket& packet) {
       return absl::optional<float>(packet.total_length);
@@ -597,8 +580,8 @@
   for (const auto& stream : parsed_log_.rtp_packets_by_ssrc(direction)) {
     if (!MatchingSsrc(stream.ssrc, desired_ssrc_))
       continue;
-    std::string label =
-        std::string("RTP ") + GetStreamName(direction, stream.ssrc);
+    std::string label = std::string("RTP ") +
+                        GetStreamName(parsed_log_, direction, stream.ssrc);
     CreateAccumulatedPacketsTimeSeries(plot, stream.packet_view, label);
   }
   std::string label =
@@ -627,7 +610,8 @@
       continue;
     }
     TimeSeries time_series(
-        std::string("RTP ") + GetStreamName(direction, stream.ssrc),
+        std::string("RTP ") +
+            GetStreamName(parsed_log_, direction, stream.ssrc),
         LineStyle::kLine);
     MovingAverage<LoggedRtpPacket, double>(CountPackets, stream.packet_view,
                                            config_, &time_series);
@@ -736,9 +720,9 @@
 void EventLogAnalyzer::CreateAudioLevelGraph(PacketDirection direction,
                                              Plot* plot) {
   for (const auto& stream : parsed_log_.rtp_packets_by_ssrc(direction)) {
-    if (!IsAudioSsrc(direction, stream.ssrc))
+    if (!IsAudioSsrc(parsed_log_, direction, stream.ssrc))
       continue;
-    TimeSeries time_series(GetStreamName(direction, stream.ssrc),
+    TimeSeries time_series(GetStreamName(parsed_log_, direction, stream.ssrc),
                            LineStyle::kLine);
     for (auto& packet : stream.packet_view) {
       if (packet.header.extension.hasAudioLevel) {
@@ -767,8 +751,9 @@
       continue;
     }
 
-    TimeSeries time_series(GetStreamName(kIncomingPacket, stream.ssrc),
-                           LineStyle::kBar);
+    TimeSeries time_series(
+        GetStreamName(parsed_log_, kIncomingPacket, stream.ssrc),
+        LineStyle::kBar);
     auto GetSequenceNumberDiff = [](const LoggedRtpPacketIncoming& old_packet,
                                     const LoggedRtpPacketIncoming& new_packet) {
       int64_t diff =
@@ -801,8 +786,9 @@
       continue;
     }
 
-    TimeSeries time_series(GetStreamName(kIncomingPacket, stream.ssrc),
-                           LineStyle::kLine, PointStyle::kHighlight);
+    TimeSeries time_series(
+        GetStreamName(parsed_log_, kIncomingPacket, stream.ssrc),
+        LineStyle::kLine, PointStyle::kHighlight);
     // TODO(terelius): Should the window and step size be read from the class
     // instead?
     const int64_t kWindowUs = 1000000;
@@ -855,7 +841,7 @@
   for (const auto& stream : parsed_log_.incoming_rtp_packets_by_ssrc()) {
     // Filter on SSRC.
     if (!MatchingSsrc(stream.ssrc, desired_ssrc_) ||
-        IsRtxSsrc(kIncomingPacket, stream.ssrc)) {
+        IsRtxSsrc(parsed_log_, kIncomingPacket, stream.ssrc)) {
       continue;
     }
 
@@ -866,15 +852,17 @@
                           << packets.size() << " packets in the stream.";
       continue;
     }
-    int64_t end_time_us = log_segments_.empty()
-                              ? std::numeric_limits<int64_t>::max()
-                              : log_segments_.front().second;
+    int64_t segment_end_us =
+        parsed_log_.log_segments().empty()
+            ? std::numeric_limits<int64_t>::max()
+            : parsed_log_.log_segments().front().stop_time_us();
     absl::optional<uint32_t> estimated_frequency =
-        EstimateRtpClockFrequency(packets, end_time_us);
+        EstimateRtpClockFrequency(packets, segment_end_us);
     if (!estimated_frequency)
       continue;
     const double frequency_hz = *estimated_frequency;
-    if (IsVideoSsrc(kIncomingPacket, stream.ssrc) && frequency_hz != 90000) {
+    if (IsVideoSsrc(parsed_log_, kIncomingPacket, stream.ssrc) &&
+        frequency_hz != 90000) {
       RTC_LOG(LS_WARNING)
           << "Video stream should use a 90 kHz clock but appears to use "
           << frequency_hz / 1000 << ". Discarding.";
@@ -891,14 +879,16 @@
     };
 
     TimeSeries capture_time_data(
-        GetStreamName(kIncomingPacket, stream.ssrc) + " capture-time",
+        GetStreamName(parsed_log_, kIncomingPacket, stream.ssrc) +
+            " capture-time",
         LineStyle::kLine);
     AccumulatePairs<LoggedRtpPacketIncoming, double>(
         ToCallTime, ToNetworkDelay, packets, &capture_time_data);
     plot->AppendTimeSeries(std::move(capture_time_data));
 
     TimeSeries send_time_data(
-        GetStreamName(kIncomingPacket, stream.ssrc) + " abs-send-time",
+        GetStreamName(parsed_log_, kIncomingPacket, stream.ssrc) +
+            " abs-send-time",
         LineStyle::kLine);
     AccumulatePairs<LoggedRtpPacketIncoming, double>(
         ToCallTime, NetworkDelayDiff_AbsSendTime, packets, &send_time_data);
@@ -1191,7 +1181,7 @@
       continue;
     }
 
-    TimeSeries time_series(GetStreamName(direction, stream.ssrc),
+    TimeSeries time_series(GetStreamName(parsed_log_, direction, stream.ssrc),
                            LineStyle::kLine);
     auto GetPacketSizeKilobits = [](const LoggedRtpPacket& packet) {
       return packet.total_length * 8.0 / 1000.0;
@@ -1483,7 +1473,7 @@
   std::multimap<int64_t, const RtpPacketType*> incoming_rtp;
 
   for (const auto& stream : parsed_log_.incoming_rtp_packets_by_ssrc()) {
-    if (IsVideoSsrc(kIncomingPacket, stream.ssrc)) {
+    if (IsVideoSsrc(parsed_log_, kIncomingPacket, stream.ssrc)) {
       for (const auto& rtp_packet : stream.incoming_packets)
         incoming_rtp.insert(
             std::make_pair(rtp_packet.rtp.log_time_us(), &rtp_packet));
@@ -1586,7 +1576,7 @@
     const std::vector<LoggedRtpPacketOutgoing>& packets =
         stream.outgoing_packets;
 
-    if (IsRtxSsrc(kOutgoingPacket, stream.ssrc)) {
+    if (IsRtxSsrc(parsed_log_, kOutgoingPacket, stream.ssrc)) {
       continue;
     }
 
@@ -1596,14 +1586,15 @@
              "pacer delay with less than 2 packets in the stream";
       continue;
     }
-    int64_t end_time_us = log_segments_.empty()
-                              ? std::numeric_limits<int64_t>::max()
-                              : log_segments_.front().second;
+    int64_t segment_end_us =
+        parsed_log_.log_segments().empty()
+            ? std::numeric_limits<int64_t>::max()
+            : parsed_log_.log_segments().front().stop_time_us();
     absl::optional<uint32_t> estimated_frequency =
-        EstimateRtpClockFrequency(packets, end_time_us);
+        EstimateRtpClockFrequency(packets, segment_end_us);
     if (!estimated_frequency)
       continue;
-    if (IsVideoSsrc(kOutgoingPacket, stream.ssrc) &&
+    if (IsVideoSsrc(parsed_log_, kOutgoingPacket, stream.ssrc) &&
         *estimated_frequency != 90000) {
       RTC_LOG(LS_WARNING)
           << "Video stream should use a 90 kHz clock but appears to use "
@@ -1612,7 +1603,7 @@
     }
 
     TimeSeries pacer_delay_series(
-        GetStreamName(kOutgoingPacket, stream.ssrc) + "(" +
+        GetStreamName(parsed_log_, kOutgoingPacket, stream.ssrc) + "(" +
             std::to_string(*estimated_frequency / 1000) + " kHz)",
         LineStyle::kLine, PointStyle::kHighlight);
     SeqNumUnwrapper<uint32_t> timestamp_unwrapper;
@@ -1645,7 +1636,7 @@
                                             Plot* plot) {
   for (const auto& stream : parsed_log_.rtp_packets_by_ssrc(direction)) {
     TimeSeries rtp_timestamps(
-        GetStreamName(direction, stream.ssrc) + " capture-time",
+        GetStreamName(parsed_log_, direction, stream.ssrc) + " capture-time",
         LineStyle::kLine, PointStyle::kHighlight);
     for (const auto& packet : stream.packet_view) {
       float x = config_.GetCallTimeSec(packet.log_time_us());
@@ -1655,7 +1646,8 @@
     plot->AppendTimeSeries(std::move(rtp_timestamps));
 
     TimeSeries rtcp_timestamps(
-        GetStreamName(direction, stream.ssrc) + " rtcp capture-time",
+        GetStreamName(parsed_log_, direction, stream.ssrc) +
+            " rtcp capture-time",
         LineStyle::kLine, PointStyle::kHighlight);
     // TODO(terelius): Why only sender reports?
     const auto& sender_reports = parsed_log_.sender_reports(direction);
@@ -1692,7 +1684,8 @@
       bool inserted;
       if (sr_report_it == sr_reports_by_ssrc.end()) {
         std::tie(sr_report_it, inserted) = sr_reports_by_ssrc.emplace(
-            ssrc, TimeSeries(GetStreamName(direction, ssrc) + " Sender Reports",
+            ssrc, TimeSeries(GetStreamName(parsed_log_, direction, ssrc) +
+                                 " Sender Reports",
                              LineStyle::kLine, PointStyle::kHighlight));
       }
       sr_report_it->second.points.emplace_back(x, y);
@@ -1713,9 +1706,9 @@
       bool inserted;
       if (rr_report_it == rr_reports_by_ssrc.end()) {
         std::tie(rr_report_it, inserted) = rr_reports_by_ssrc.emplace(
-            ssrc,
-            TimeSeries(GetStreamName(direction, ssrc) + " Receiver Reports",
-                       LineStyle::kLine, PointStyle::kHighlight));
+            ssrc, TimeSeries(GetStreamName(parsed_log_, direction, ssrc) +
+                                 " Receiver Reports",
+                             LineStyle::kLine, PointStyle::kHighlight));
       }
       rr_report_it->second.points.emplace_back(x, y);
     }
@@ -2038,7 +2031,7 @@
 
   for (const auto& stream : parsed_log_.incoming_rtp_packets_by_ssrc()) {
     const uint32_t ssrc = stream.ssrc;
-    if (!IsAudioSsrc(kIncomingPacket, ssrc))
+    if (!IsAudioSsrc(parsed_log_, kIncomingPacket, ssrc))
       continue;
     const std::vector<LoggedRtpPacketIncoming>* audio_packets =
         &stream.incoming_packets;
@@ -2058,9 +2051,10 @@
     }
 
     absl::optional<int64_t> end_time_ms =
-        log_segments_.empty()
+        parsed_log_.log_segments().empty()
             ? absl::nullopt
-            : absl::optional<int64_t>(log_segments_.front().second / 1000);
+            : absl::optional<int64_t>(
+                  parsed_log_.log_segments().front().stop_time_ms());
 
     neteq_stats[ssrc] = CreateNetEqTestAndRun(
         audio_packets, &output_events_it->second, end_time_ms,
@@ -2124,7 +2118,8 @@
                  "Time (s)", kLeftMargin, kRightMargin);
   plot->SetSuggestedYAxis(0, 1, "Relative delay (ms)", kBottomMargin,
                           kTopMargin);
-  plot->SetTitle("NetEq timing for " + GetStreamName(kIncomingPacket, ssrc));
+  plot->SetTitle("NetEq timing for " +
+                 GetStreamName(parsed_log_, kIncomingPacket, ssrc));
 }
 
 template <typename NetEqStatsType>
@@ -2150,7 +2145,8 @@
   }
 
   for (auto& series : time_series) {
-    series.second.label = GetStreamName(kIncomingPacket, series.first);
+    series.second.label =
+        GetStreamName(parsed_log_, kIncomingPacket, series.first);
     series.second.line_style = LineStyle::kLine;
     plot->AppendTimeSeries(std::move(series.second));
   }
@@ -2326,181 +2322,4 @@
   plot->SetTitle("DTLS Writable State");
 }
 
-void EventLogAnalyzer::PrintNotifications(FILE* file) {
-  fprintf(file, "========== TRIAGE NOTIFICATIONS ==========\n");
-  for (const auto& alert : incoming_rtp_recv_time_gaps_) {
-    fprintf(file, "%3.3lf s : %s\n", alert.Time(), alert.ToString().c_str());
-  }
-  for (const auto& alert : incoming_rtcp_recv_time_gaps_) {
-    fprintf(file, "%3.3lf s : %s\n", alert.Time(), alert.ToString().c_str());
-  }
-  for (const auto& alert : outgoing_rtp_send_time_gaps_) {
-    fprintf(file, "%3.3lf s : %s\n", alert.Time(), alert.ToString().c_str());
-  }
-  for (const auto& alert : outgoing_rtcp_send_time_gaps_) {
-    fprintf(file, "%3.3lf s : %s\n", alert.Time(), alert.ToString().c_str());
-  }
-  for (const auto& alert : incoming_seq_num_jumps_) {
-    fprintf(file, "%3.3lf s : %s\n", alert.Time(), alert.ToString().c_str());
-  }
-  for (const auto& alert : incoming_capture_time_jumps_) {
-    fprintf(file, "%3.3lf s : %s\n", alert.Time(), alert.ToString().c_str());
-  }
-  for (const auto& alert : outgoing_seq_num_jumps_) {
-    fprintf(file, "%3.3lf s : %s\n", alert.Time(), alert.ToString().c_str());
-  }
-  for (const auto& alert : outgoing_capture_time_jumps_) {
-    fprintf(file, "%3.3lf s : %s\n", alert.Time(), alert.ToString().c_str());
-  }
-  for (const auto& alert : outgoing_high_loss_alerts_) {
-    fprintf(file, "          : %s\n", alert.ToString().c_str());
-  }
-  fprintf(file, "========== END TRIAGE NOTIFICATIONS ==========\n");
-}
-
-void EventLogAnalyzer::CreateStreamGapAlerts(PacketDirection direction) {
-  // With 100 packets/s (~800kbps), false positives would require 10 s without
-  // data.
-  constexpr int64_t kMaxSeqNumJump = 1000;
-  // With a 90 kHz clock, false positives would require 10 s without data.
-  constexpr int64_t kMaxCaptureTimeJump = 900000;
-
-  int64_t end_time_us = log_segments_.empty()
-                            ? std::numeric_limits<int64_t>::max()
-                            : log_segments_.front().second;
-
-  SeqNumUnwrapper<uint16_t> seq_num_unwrapper;
-  absl::optional<int64_t> last_seq_num;
-  SeqNumUnwrapper<uint32_t> capture_time_unwrapper;
-  absl::optional<int64_t> last_capture_time;
-  // Check for gaps in sequence numbers and capture timestamps.
-  for (const auto& stream : parsed_log_.rtp_packets_by_ssrc(direction)) {
-    for (const auto& packet : stream.packet_view) {
-      if (packet.log_time_us() > end_time_us) {
-        // Only process the first (LOG_START, LOG_END) segment.
-        break;
-      }
-
-      int64_t seq_num = seq_num_unwrapper.Unwrap(packet.header.sequenceNumber);
-      if (last_seq_num.has_value() &&
-          std::abs(seq_num - last_seq_num.value()) > kMaxSeqNumJump) {
-        Alert_SeqNumJump(direction,
-                         config_.GetCallTimeSec(packet.log_time_us()),
-                         packet.header.ssrc);
-      }
-      last_seq_num.emplace(seq_num);
-
-      int64_t capture_time =
-          capture_time_unwrapper.Unwrap(packet.header.timestamp);
-      if (last_capture_time.has_value() &&
-          std::abs(capture_time - last_capture_time.value()) >
-              kMaxCaptureTimeJump) {
-        Alert_CaptureTimeJump(direction,
-                              config_.GetCallTimeSec(packet.log_time_us()),
-                              packet.header.ssrc);
-      }
-      last_capture_time.emplace(capture_time);
-    }
-  }
-}
-
-void EventLogAnalyzer::CreateTransmissionGapAlerts(PacketDirection direction) {
-  constexpr int64_t kMaxRtpTransmissionGap = 500000;
-  constexpr int64_t kMaxRtcpTransmissionGap = 2000000;
-  int64_t end_time_us = log_segments_.empty()
-                            ? std::numeric_limits<int64_t>::max()
-                            : log_segments_.front().second;
-
-  // TODO(terelius): The parser could provide a list of all packets, ordered
-  // by time, for each direction.
-  std::multimap<int64_t, const LoggedRtpPacket*> rtp_in_direction;
-  for (const auto& stream : parsed_log_.rtp_packets_by_ssrc(direction)) {
-    for (const LoggedRtpPacket& rtp_packet : stream.packet_view)
-      rtp_in_direction.emplace(rtp_packet.log_time_us(), &rtp_packet);
-  }
-  absl::optional<int64_t> last_rtp_time;
-  for (const auto& kv : rtp_in_direction) {
-    int64_t timestamp = kv.first;
-    if (timestamp > end_time_us) {
-      // Only process the first (LOG_START, LOG_END) segment.
-      break;
-    }
-    int64_t duration = timestamp - last_rtp_time.value_or(0);
-    if (last_rtp_time.has_value() && duration > kMaxRtpTransmissionGap) {
-      // No packet sent/received for more than 500 ms.
-      Alert_RtpLogTimeGap(direction, config_.GetCallTimeSec(timestamp),
-                          duration / 1000);
-    }
-    last_rtp_time.emplace(timestamp);
-  }
-
-  absl::optional<int64_t> last_rtcp_time;
-  if (direction == kIncomingPacket) {
-    for (const auto& rtcp : parsed_log_.incoming_rtcp_packets()) {
-      if (rtcp.log_time_us() > end_time_us) {
-        // Only process the first (LOG_START, LOG_END) segment.
-        break;
-      }
-      int64_t duration = rtcp.log_time_us() - last_rtcp_time.value_or(0);
-      if (last_rtcp_time.has_value() && duration > kMaxRtcpTransmissionGap) {
-        // No feedback sent/received for more than 2000 ms.
-        Alert_RtcpLogTimeGap(direction,
-                             config_.GetCallTimeSec(rtcp.log_time_us()),
-                             duration / 1000);
-      }
-      last_rtcp_time.emplace(rtcp.log_time_us());
-    }
-  } else {
-    for (const auto& rtcp : parsed_log_.outgoing_rtcp_packets()) {
-      if (rtcp.log_time_us() > end_time_us) {
-        // Only process the first (LOG_START, LOG_END) segment.
-        break;
-      }
-      int64_t duration = rtcp.log_time_us() - last_rtcp_time.value_or(0);
-      if (last_rtcp_time.has_value() && duration > kMaxRtcpTransmissionGap) {
-        // No feedback sent/received for more than 2000 ms.
-        Alert_RtcpLogTimeGap(direction,
-                             config_.GetCallTimeSec(rtcp.log_time_us()),
-                             duration / 1000);
-      }
-      last_rtcp_time.emplace(rtcp.log_time_us());
-    }
-  }
-}
-
-// TODO(terelius): Notifications could possibly be generated by the same code
-// that produces the graphs. There is some code duplication that could be
-// avoided, but that might be solved anyway when we move functionality from the
-// analyzer to the parser.
-void EventLogAnalyzer::CreateTriageNotifications() {
-  CreateStreamGapAlerts(kIncomingPacket);
-  CreateStreamGapAlerts(kOutgoingPacket);
-  CreateTransmissionGapAlerts(kIncomingPacket);
-  CreateTransmissionGapAlerts(kOutgoingPacket);
-
-  int64_t end_time_us = log_segments_.empty()
-                            ? std::numeric_limits<int64_t>::max()
-                            : log_segments_.front().second;
-
-  constexpr double kMaxLossFraction = 0.05;
-  // Loss feedback
-  int64_t total_lost_packets = 0;
-  int64_t total_expected_packets = 0;
-  for (auto& bwe_update : parsed_log_.bwe_loss_updates()) {
-    if (bwe_update.log_time_us() > end_time_us) {
-      // Only process the first (LOG_START, LOG_END) segment.
-      break;
-    }
-    int64_t lost_packets = static_cast<double>(bwe_update.fraction_lost) / 255 *
-                           bwe_update.expected_packets;
-    total_lost_packets += lost_packets;
-    total_expected_packets += bwe_update.expected_packets;
-  }
-  double avg_outgoing_loss =
-      static_cast<double>(total_lost_packets) / total_expected_packets;
-  if (avg_outgoing_loss > kMaxLossFraction) {
-    Alert_OutgoingHighLoss(avg_outgoing_loss);
-  }
-}
-
 }  // namespace webrtc
diff --git a/rtc_tools/rtc_event_log_visualizer/analyzer.h b/rtc_tools/rtc_event_log_visualizer/analyzer.h
index 1e09109..ebdfdcc 100644
--- a/rtc_tools/rtc_event_log_visualizer/analyzer.h
+++ b/rtc_tools/rtc_event_log_visualizer/analyzer.h
@@ -21,41 +21,18 @@
 #include "logging/rtc_event_log/rtc_event_log_parser.h"
 #include "modules/audio_coding/neteq/tools/neteq_stats_getter.h"
 #include "rtc_base/strings/string_builder.h"
+#include "rtc_tools/rtc_event_log_visualizer/analyzer_common.h"
 #include "rtc_tools/rtc_event_log_visualizer/plot_base.h"
-#include "rtc_tools/rtc_event_log_visualizer/triage_notifications.h"
 
 namespace webrtc {
 
-class AnalyzerConfig {
- public:
-  float GetCallTimeSec(int64_t timestamp_us) const {
-    int64_t offset = normalize_time_ ? begin_time_ : 0;
-    return static_cast<float>(timestamp_us - offset) / 1000000;
-  }
-
-  float CallBeginTimeSec() const { return GetCallTimeSec(begin_time_); }
-
-  float CallEndTimeSec() const { return GetCallTimeSec(end_time_); }
-
-  // Window and step size used for calculating moving averages, e.g. bitrate.
-  // The generated data points will be |step_| microseconds apart.
-  // Only events occurring at most |window_duration_| microseconds before the
-  // current data point will be part of the average.
-  int64_t window_duration_;
-  int64_t step_;
-
-  // First and last events of the log.
-  int64_t begin_time_;
-  int64_t end_time_;
-  bool normalize_time_;
-};
-
 class EventLogAnalyzer {
  public:
   // The EventLogAnalyzer keeps a reference to the ParsedRtcEventLogNew for the
   // duration of its lifetime. The ParsedRtcEventLogNew must not be destroyed or
   // modified while the EventLogAnalyzer is being used.
   EventLogAnalyzer(const ParsedRtcEventLog& log, bool normalize_time);
+  EventLogAnalyzer(const ParsedRtcEventLog& log, const AnalyzerConfig& config);
 
   void CreatePacketGraph(PacketDirection direction, Plot* plot);
 
@@ -138,55 +115,6 @@
   void PrintNotifications(FILE* file);
 
  private:
-  struct LayerDescription {
-    LayerDescription(uint32_t ssrc,
-                     uint8_t spatial_layer,
-                     uint8_t temporal_layer)
-        : ssrc(ssrc),
-          spatial_layer(spatial_layer),
-          temporal_layer(temporal_layer) {}
-    bool operator<(const LayerDescription& other) const {
-      if (ssrc != other.ssrc)
-        return ssrc < other.ssrc;
-      if (spatial_layer != other.spatial_layer)
-        return spatial_layer < other.spatial_layer;
-      return temporal_layer < other.temporal_layer;
-    }
-    uint32_t ssrc;
-    uint8_t spatial_layer;
-    uint8_t temporal_layer;
-  };
-
-  bool IsRtxSsrc(PacketDirection direction, uint32_t ssrc) const {
-    if (direction == kIncomingPacket) {
-      return parsed_log_.incoming_rtx_ssrcs().find(ssrc) !=
-             parsed_log_.incoming_rtx_ssrcs().end();
-    } else {
-      return parsed_log_.outgoing_rtx_ssrcs().find(ssrc) !=
-             parsed_log_.outgoing_rtx_ssrcs().end();
-    }
-  }
-
-  bool IsVideoSsrc(PacketDirection direction, uint32_t ssrc) const {
-    if (direction == kIncomingPacket) {
-      return parsed_log_.incoming_video_ssrcs().find(ssrc) !=
-             parsed_log_.incoming_video_ssrcs().end();
-    } else {
-      return parsed_log_.outgoing_video_ssrcs().find(ssrc) !=
-             parsed_log_.outgoing_video_ssrcs().end();
-    }
-  }
-
-  bool IsAudioSsrc(PacketDirection direction, uint32_t ssrc) const {
-    if (direction == kIncomingPacket) {
-      return parsed_log_.incoming_audio_ssrcs().find(ssrc) !=
-             parsed_log_.incoming_audio_ssrcs().end();
-    } else {
-      return parsed_log_.outgoing_audio_ssrcs().find(ssrc) !=
-             parsed_log_.outgoing_audio_ssrcs().end();
-    }
-  }
-
   template <typename NetEqStatsType>
   void CreateNetEqStatsGraphInternal(
       const NetEqStatsGetterMap& neteq_stats,
@@ -201,82 +129,6 @@
                                           const IterableType& packets,
                                           const std::string& label);
 
-  void CreateStreamGapAlerts(PacketDirection direction);
-  void CreateTransmissionGapAlerts(PacketDirection direction);
-
-  std::string GetStreamName(PacketDirection direction, uint32_t ssrc) const {
-    char buffer[200];
-    rtc::SimpleStringBuilder name(buffer);
-    if (IsAudioSsrc(direction, ssrc)) {
-      name << "Audio ";
-    } else if (IsVideoSsrc(direction, ssrc)) {
-      name << "Video ";
-    } else {
-      name << "Unknown ";
-    }
-    if (IsRtxSsrc(direction, ssrc)) {
-      name << "RTX ";
-    }
-    if (direction == kIncomingPacket)
-      name << "(In) ";
-    else
-      name << "(Out) ";
-    name << "SSRC " << ssrc;
-    return name.str();
-  }
-
-  std::string GetLayerName(LayerDescription layer) const {
-    char buffer[100];
-    rtc::SimpleStringBuilder name(buffer);
-    name << "SSRC " << layer.ssrc << " sl " << layer.spatial_layer << ", tl "
-         << layer.temporal_layer;
-    return name.str();
-  }
-
-  void Alert_RtpLogTimeGap(PacketDirection direction,
-                           float time_seconds,
-                           int64_t duration) {
-    if (direction == kIncomingPacket) {
-      incoming_rtp_recv_time_gaps_.emplace_back(time_seconds, duration);
-    } else {
-      outgoing_rtp_send_time_gaps_.emplace_back(time_seconds, duration);
-    }
-  }
-
-  void Alert_RtcpLogTimeGap(PacketDirection direction,
-                            float time_seconds,
-                            int64_t duration) {
-    if (direction == kIncomingPacket) {
-      incoming_rtcp_recv_time_gaps_.emplace_back(time_seconds, duration);
-    } else {
-      outgoing_rtcp_send_time_gaps_.emplace_back(time_seconds, duration);
-    }
-  }
-
-  void Alert_SeqNumJump(PacketDirection direction,
-                        float time_seconds,
-                        uint32_t ssrc) {
-    if (direction == kIncomingPacket) {
-      incoming_seq_num_jumps_.emplace_back(time_seconds, ssrc);
-    } else {
-      outgoing_seq_num_jumps_.emplace_back(time_seconds, ssrc);
-    }
-  }
-
-  void Alert_CaptureTimeJump(PacketDirection direction,
-                             float time_seconds,
-                             uint32_t ssrc) {
-    if (direction == kIncomingPacket) {
-      incoming_capture_time_jumps_.emplace_back(time_seconds, ssrc);
-    } else {
-      outgoing_capture_time_jumps_.emplace_back(time_seconds, ssrc);
-    }
-  }
-
-  void Alert_OutgoingHighLoss(double avg_loss_fraction) {
-    outgoing_high_loss_alerts_.emplace_back(avg_loss_fraction);
-  }
-
   std::string GetCandidatePairLogDescriptionFromId(uint32_t candidate_pair_id);
 
   const ParsedRtcEventLog& parsed_log_;
@@ -285,20 +137,6 @@
   // If left empty, all SSRCs will be considered relevant.
   std::vector<uint32_t> desired_ssrc_;
 
-  // Stores the timestamps for all log segments, in the form of associated start
-  // and end events.
-  std::vector<std::pair<int64_t, int64_t>> log_segments_;
-
-  std::vector<IncomingRtpReceiveTimeGap> incoming_rtp_recv_time_gaps_;
-  std::vector<IncomingRtcpReceiveTimeGap> incoming_rtcp_recv_time_gaps_;
-  std::vector<OutgoingRtpSendTimeGap> outgoing_rtp_send_time_gaps_;
-  std::vector<OutgoingRtcpSendTimeGap> outgoing_rtcp_send_time_gaps_;
-  std::vector<IncomingSeqNumJump> incoming_seq_num_jumps_;
-  std::vector<IncomingCaptureTimeJump> incoming_capture_time_jumps_;
-  std::vector<OutgoingSeqNoJump> outgoing_seq_num_jumps_;
-  std::vector<OutgoingCaptureTimeJump> outgoing_capture_time_jumps_;
-  std::vector<OutgoingHighLoss> outgoing_high_loss_alerts_;
-
   std::map<uint32_t, std::string> candidate_pair_desc_by_id_;
 
   AnalyzerConfig config_;
diff --git a/rtc_tools/rtc_event_log_visualizer/analyzer_common.cc b/rtc_tools/rtc_event_log_visualizer/analyzer_common.cc
new file mode 100644
index 0000000..3d3ce5a
--- /dev/null
+++ b/rtc_tools/rtc_event_log_visualizer/analyzer_common.cc
@@ -0,0 +1,83 @@
+
+/*
+ *  Copyright (c) 2020 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 "rtc_tools/rtc_event_log_visualizer/analyzer_common.h"
+
+namespace webrtc {
+
+bool IsRtxSsrc(const ParsedRtcEventLog& parsed_log,
+               PacketDirection direction,
+               uint32_t ssrc) {
+  if (direction == kIncomingPacket) {
+    return parsed_log.incoming_rtx_ssrcs().find(ssrc) !=
+           parsed_log.incoming_rtx_ssrcs().end();
+  } else {
+    return parsed_log.outgoing_rtx_ssrcs().find(ssrc) !=
+           parsed_log.outgoing_rtx_ssrcs().end();
+  }
+}
+
+bool IsVideoSsrc(const ParsedRtcEventLog& parsed_log,
+                 PacketDirection direction,
+                 uint32_t ssrc) {
+  if (direction == kIncomingPacket) {
+    return parsed_log.incoming_video_ssrcs().find(ssrc) !=
+           parsed_log.incoming_video_ssrcs().end();
+  } else {
+    return parsed_log.outgoing_video_ssrcs().find(ssrc) !=
+           parsed_log.outgoing_video_ssrcs().end();
+  }
+}
+
+bool IsAudioSsrc(const ParsedRtcEventLog& parsed_log,
+                 PacketDirection direction,
+                 uint32_t ssrc) {
+  if (direction == kIncomingPacket) {
+    return parsed_log.incoming_audio_ssrcs().find(ssrc) !=
+           parsed_log.incoming_audio_ssrcs().end();
+  } else {
+    return parsed_log.outgoing_audio_ssrcs().find(ssrc) !=
+           parsed_log.outgoing_audio_ssrcs().end();
+  }
+}
+
+std::string GetStreamName(const ParsedRtcEventLog& parsed_log,
+                          PacketDirection direction,
+                          uint32_t ssrc) {
+  char buffer[200];
+  rtc::SimpleStringBuilder name(buffer);
+  if (IsAudioSsrc(parsed_log, direction, ssrc)) {
+    name << "Audio ";
+  } else if (IsVideoSsrc(parsed_log, direction, ssrc)) {
+    name << "Video ";
+  } else {
+    name << "Unknown ";
+  }
+  if (IsRtxSsrc(parsed_log, direction, ssrc)) {
+    name << "RTX ";
+  }
+  if (direction == kIncomingPacket)
+    name << "(In) ";
+  else
+    name << "(Out) ";
+  name << "SSRC " << ssrc;
+  return name.str();
+}
+
+std::string GetLayerName(LayerDescription layer) {
+  char buffer[100];
+  rtc::SimpleStringBuilder name(buffer);
+  name << "SSRC " << layer.ssrc << " sl " << layer.spatial_layer << ", tl "
+       << layer.temporal_layer;
+  return name.str();
+}
+
+}  // namespace webrtc
diff --git a/rtc_tools/rtc_event_log_visualizer/analyzer_common.h b/rtc_tools/rtc_event_log_visualizer/analyzer_common.h
new file mode 100644
index 0000000..3ac651e
--- /dev/null
+++ b/rtc_tools/rtc_event_log_visualizer/analyzer_common.h
@@ -0,0 +1,79 @@
+/*
+ *  Copyright (c) 2020 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.
+ */
+
+#ifndef RTC_TOOLS_RTC_EVENT_LOG_VISUALIZER_ANALYZER_COMMON_H_
+#define RTC_TOOLS_RTC_EVENT_LOG_VISUALIZER_ANALYZER_COMMON_H_
+
+#include <cstdint>
+#include <string>
+
+#include "logging/rtc_event_log/rtc_event_log_parser.h"
+
+namespace webrtc {
+
+class AnalyzerConfig {
+ public:
+  float GetCallTimeSec(int64_t timestamp_us) const {
+    int64_t offset = normalize_time_ ? begin_time_ : 0;
+    return static_cast<float>(timestamp_us - offset) / 1000000;
+  }
+
+  float CallBeginTimeSec() const { return GetCallTimeSec(begin_time_); }
+
+  float CallEndTimeSec() const { return GetCallTimeSec(end_time_); }
+
+  // Window and step size used for calculating moving averages, e.g. bitrate.
+  // The generated data points will be |step_| microseconds apart.
+  // Only events occurring at most |window_duration_| microseconds before the
+  // current data point will be part of the average.
+  int64_t window_duration_;
+  int64_t step_;
+
+  // First and last events of the log.
+  int64_t begin_time_;
+  int64_t end_time_;
+  bool normalize_time_;
+};
+
+struct LayerDescription {
+  LayerDescription(uint32_t ssrc, uint8_t spatial_layer, uint8_t temporal_layer)
+      : ssrc(ssrc),
+        spatial_layer(spatial_layer),
+        temporal_layer(temporal_layer) {}
+  bool operator<(const LayerDescription& other) const {
+    if (ssrc != other.ssrc)
+      return ssrc < other.ssrc;
+    if (spatial_layer != other.spatial_layer)
+      return spatial_layer < other.spatial_layer;
+    return temporal_layer < other.temporal_layer;
+  }
+  uint32_t ssrc;
+  uint8_t spatial_layer;
+  uint8_t temporal_layer;
+};
+
+bool IsRtxSsrc(const ParsedRtcEventLog& parsed_log,
+               PacketDirection direction,
+               uint32_t ssrc);
+bool IsVideoSsrc(const ParsedRtcEventLog& parsed_log,
+                 PacketDirection direction,
+                 uint32_t ssrc);
+bool IsAudioSsrc(const ParsedRtcEventLog& parsed_log,
+                 PacketDirection direction,
+                 uint32_t ssrc);
+
+std::string GetStreamName(const ParsedRtcEventLog& parsed_log,
+                          PacketDirection direction,
+                          uint32_t ssrc);
+std::string GetLayerName(LayerDescription layer);
+
+}  // namespace webrtc
+
+#endif  // RTC_TOOLS_RTC_EVENT_LOG_VISUALIZER_ANALYZER_COMMON_H_
diff --git a/rtc_tools/rtc_event_log_visualizer/main.cc b/rtc_tools/rtc_event_log_visualizer/main.cc
index eb36b26..21768c9 100644
--- a/rtc_tools/rtc_event_log_visualizer/main.cc
+++ b/rtc_tools/rtc_event_log_visualizer/main.cc
@@ -30,6 +30,7 @@
 #include "modules/rtp_rtcp/source/rtcp_packet/report_block.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/logging.h"
+#include "rtc_tools/rtc_event_log_visualizer/alerts.h"
 #include "rtc_tools/rtc_event_log_visualizer/analyzer.h"
 #include "rtc_tools/rtc_event_log_visualizer/plot_base.h"
 #include "rtc_tools/rtc_event_log_visualizer/plot_protobuf.h"
@@ -194,9 +195,9 @@
       "A tool for visualizing WebRTC event logs.\n"
       "Example usage:\n"
       "./event_log_visualizer <logfile> | python\n");
-  absl::FlagsUsageConfig config;
-  config.contains_help_flags = &ContainsHelppackageFlags;
-  absl::SetFlagsUsageConfig(config);
+  absl::FlagsUsageConfig flag_config;
+  flag_config.contains_help_flags = &ContainsHelppackageFlags;
+  absl::SetFlagsUsageConfig(flag_config);
   std::vector<char*> args = absl::ParseCommandLine(argc, argv);
 
   // Print RTC_LOG warnings and errors even in release builds.
@@ -261,8 +262,20 @@
     }
   }
 
-  webrtc::EventLogAnalyzer analyzer(parsed_log,
-                                    absl::GetFlag(FLAGS_normalize_time));
+  webrtc::AnalyzerConfig config;
+  config.window_duration_ = 250000;
+  config.step_ = 10000;
+  config.normalize_time_ = absl::GetFlag(FLAGS_normalize_time);
+  config.begin_time_ = parsed_log.first_timestamp();
+  config.end_time_ = parsed_log.last_timestamp();
+  if (config.end_time_ < config.begin_time_) {
+    RTC_LOG(LS_WARNING) << "Log end time " << config.end_time_
+                        << " not after begin time " << config.begin_time_
+                        << ". Nothing to analyze. Is the log broken?";
+    return -1;
+  }
+
+  webrtc::EventLogAnalyzer analyzer(parsed_log, config);
   std::unique_ptr<webrtc::PlotCollection> collection;
   if (absl::GetFlag(FLAGS_protobuf_output)) {
     collection.reset(new webrtc::ProtobufPlotCollection());
@@ -614,8 +627,9 @@
   collection->Draw();
 
   if (absl::GetFlag(FLAGS_print_triage_alerts)) {
-    analyzer.CreateTriageNotifications();
-    analyzer.PrintNotifications(stderr);
+    webrtc::TriageHelper triage_alerts(config);
+    triage_alerts.AnalyzeLog(parsed_log);
+    triage_alerts.Print(stderr);
   }
 
   return 0;
diff --git a/rtc_tools/rtc_event_log_visualizer/triage_notifications.h b/rtc_tools/rtc_event_log_visualizer/triage_notifications.h
deleted file mode 100644
index 23b31ece..0000000
--- a/rtc_tools/rtc_event_log_visualizer/triage_notifications.h
+++ /dev/null
@@ -1,158 +0,0 @@
-/*
- *  Copyright (c) 2017 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.
- */
-
-#ifndef RTC_TOOLS_RTC_EVENT_LOG_VISUALIZER_TRIAGE_NOTIFICATIONS_H_
-#define RTC_TOOLS_RTC_EVENT_LOG_VISUALIZER_TRIAGE_NOTIFICATIONS_H_
-
-#include <string>
-
-namespace webrtc {
-
-class IncomingRtpReceiveTimeGap {
- public:
-  IncomingRtpReceiveTimeGap(float time_seconds, int64_t duration)
-      : time_seconds_(time_seconds), duration_(duration) {}
-  float Time() const { return time_seconds_; }
-  std::string ToString() const {
-    return std::string("No RTP packets received for ") +
-           std::to_string(duration_) + std::string(" ms");
-  }
-
- private:
-  float time_seconds_;
-  int64_t duration_;
-};
-
-class IncomingRtcpReceiveTimeGap {
- public:
-  IncomingRtcpReceiveTimeGap(float time_seconds, int64_t duration)
-      : time_seconds_(time_seconds), duration_(duration) {}
-  float Time() const { return time_seconds_; }
-  std::string ToString() const {
-    return std::string("No RTCP packets received for ") +
-           std::to_string(duration_) + std::string(" ms");
-  }
-
- private:
-  float time_seconds_;
-  int64_t duration_;
-};
-
-class OutgoingRtpSendTimeGap {
- public:
-  OutgoingRtpSendTimeGap(float time_seconds, int64_t duration)
-      : time_seconds_(time_seconds), duration_(duration) {}
-  float Time() const { return time_seconds_; }
-  std::string ToString() const {
-    return std::string("No RTP packets sent for ") + std::to_string(duration_) +
-           std::string(" ms");
-  }
-
- private:
-  float time_seconds_;
-  int64_t duration_;
-};
-
-class OutgoingRtcpSendTimeGap {
- public:
-  OutgoingRtcpSendTimeGap(float time_seconds, int64_t duration)
-      : time_seconds_(time_seconds), duration_(duration) {}
-  float Time() const { return time_seconds_; }
-  std::string ToString() const {
-    return std::string("No RTCP packets sent for ") +
-           std::to_string(duration_) + std::string(" ms");
-  }
-
- private:
-  float time_seconds_;
-  int64_t duration_;
-};
-
-class IncomingSeqNumJump {
- public:
-  IncomingSeqNumJump(float time_seconds, uint32_t ssrc)
-      : time_seconds_(time_seconds), ssrc_(ssrc) {}
-  float Time() const { return time_seconds_; }
-  std::string ToString() const {
-    return std::string("Sequence number jumps on incoming SSRC ") +
-           std::to_string(ssrc_);
-  }
-
- private:
-  float time_seconds_;
-
-  uint32_t ssrc_;
-};
-
-class IncomingCaptureTimeJump {
- public:
-  IncomingCaptureTimeJump(float time_seconds, uint32_t ssrc)
-      : time_seconds_(time_seconds), ssrc_(ssrc) {}
-  float Time() const { return time_seconds_; }
-  std::string ToString() const {
-    return std::string("Capture timestamp jumps on incoming SSRC ") +
-           std::to_string(ssrc_);
-  }
-
- private:
-  float time_seconds_;
-
-  uint32_t ssrc_;
-};
-
-class OutgoingSeqNoJump {
- public:
-  OutgoingSeqNoJump(float time_seconds, uint32_t ssrc)
-      : time_seconds_(time_seconds), ssrc_(ssrc) {}
-  float Time() const { return time_seconds_; }
-  std::string ToString() const {
-    return std::string("Sequence number jumps on outgoing SSRC ") +
-           std::to_string(ssrc_);
-  }
-
- private:
-  float time_seconds_;
-
-  uint32_t ssrc_;
-};
-
-class OutgoingCaptureTimeJump {
- public:
-  OutgoingCaptureTimeJump(float time_seconds, uint32_t ssrc)
-      : time_seconds_(time_seconds), ssrc_(ssrc) {}
-  float Time() const { return time_seconds_; }
-  std::string ToString() const {
-    return std::string("Capture timestamp jumps on outgoing SSRC ") +
-           std::to_string(ssrc_);
-  }
-
- private:
-  float time_seconds_;
-
-  uint32_t ssrc_;
-};
-
-class OutgoingHighLoss {
- public:
-  explicit OutgoingHighLoss(double avg_loss_fraction)
-      : avg_loss_fraction_(avg_loss_fraction) {}
-  std::string ToString() const {
-    return std::string("High average loss (") +
-           std::to_string(avg_loss_fraction_ * 100) +
-           std::string("%) across the call.");
-  }
-
- private:
-  double avg_loss_fraction_;
-};
-
-}  // namespace webrtc
-
-#endif  // RTC_TOOLS_RTC_EVENT_LOG_VISUALIZER_TRIAGE_NOTIFICATIONS_H_