Publish rtc event log api and default factory for it in api/

Bug: webrtc:10206
Change-Id: I34194ddb6fd2b0a3d7c553fadc9ddc1ea9740da0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137500
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28023}

api/rtc_event_log/rtc_event.h

@@ -0,0 +1,75 @@
+/*
+ *  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 API_RTC_EVENT_LOG_RTC_EVENT_H_
+#define API_RTC_EVENT_LOG_RTC_EVENT_H_
+
+#include <cstdint>
+
+namespace webrtc {
+
+// This class allows us to store unencoded RTC events. Subclasses of this class
+// store the actual information. This allows us to keep all unencoded events,
+// even when their type and associated information differ, in the same buffer.
+// Additionally, it prevents dependency leaking - a module that only logs
+// events of type RtcEvent_A doesn't need to know about anything associated
+// with events of type RtcEvent_B.
+class RtcEvent {
+ public:
+  // Subclasses of this class have to associate themselves with a unique value
+  // of Type. This leaks the information of existing subclasses into the
+  // superclass, but the *actual* information - rtclog::StreamConfig, etc. -
+  // is kept separate.
+  enum class Type {
+    AlrStateEvent,
+    RouteChangeEvent,
+    AudioNetworkAdaptation,
+    AudioPlayout,
+    AudioReceiveStreamConfig,
+    AudioSendStreamConfig,
+    BweUpdateDelayBased,
+    BweUpdateLossBased,
+    DtlsTransportState,
+    DtlsWritableState,
+    IceCandidatePairConfig,
+    IceCandidatePairEvent,
+    ProbeClusterCreated,
+    ProbeResultFailure,
+    ProbeResultSuccess,
+    RtcpPacketIncoming,
+    RtcpPacketOutgoing,
+    RtpPacketIncoming,
+    RtpPacketOutgoing,
+    VideoReceiveStreamConfig,
+    VideoSendStreamConfig,
+    GenericPacketSent,
+    GenericPacketReceived,
+    GenericAckReceived
+  };
+
+  RtcEvent();
+  virtual ~RtcEvent() = default;
+
+  virtual Type GetType() const = 0;
+
+  virtual bool IsConfigEvent() const = 0;
+
+  int64_t timestamp_ms() const { return timestamp_us_ / 1000; }
+  int64_t timestamp_us() const { return timestamp_us_; }
+
+ protected:
+  explicit RtcEvent(int64_t timestamp_us) : timestamp_us_(timestamp_us) {}
+
+  const int64_t timestamp_us_;
+};
+
+}  // namespace webrtc
+
+#endif  // API_RTC_EVENT_LOG_RTC_EVENT_H_