Refactor/reimplement RTC event log triage alerts.

- Moves AnalyzerConfig and helper functions IsAudioSsrc, IsVideoSsrc, IsRtxSsrc, GetStreamNam and GetLayerName to analyzer_common.h
- Moves log_segments() code to rtc_event_log_parser.h
- Moves TriageAlert/Notification code to a new file with a couple of minor fixes to make it less spammy.

Bug: webrtc:11566
Change-Id: Ib33941d8185f7382fc72ed65768e46015e0320de
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174824
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31318}

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_