Index: webrtc/logging/rtc_event_log/events/rtc_event.h |
diff --git a/webrtc/logging/rtc_event_log/events/rtc_event.h b/webrtc/logging/rtc_event_log/events/rtc_event.h |
new file mode 100644 |
index 0000000000000000000000000000000000000000..79a39351070bf06fc4f41e2949e09e991831878c |
--- /dev/null |
+++ b/webrtc/logging/rtc_event_log/events/rtc_event.h |
@@ -0,0 +1,56 @@ |
+/* |
+ * 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 WEBRTC_LOGGING_RTC_EVENT_LOG_EVENTS_RTC_EVENT_H_ |
+#define WEBRTC_LOGGING_RTC_EVENT_LOG_EVENTS_RTC_EVENT_H_ |
+ |
+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 |
+ // 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 { |
+ AudioNetworkAdaptation, |
+ AudioPlayout, |
+ AudioSendStreamConfig, |
+ BweUpdateDelayBased, |
+ BweUpdateLossBased, |
+ LoggingStarted, |
+ LoggingStopped, |
+ ProbeClusterCreated, |
+ ProbeResultFailure, |
+ ProbeResultSuccess, |
+ RtcpHeader, |
+ RtcpHeaderIncoming, |
+ RtcpHeaderOutgoing, |
+ RtpHeader, |
+ RtpHeaderIncoming, |
+ RtpHeaderOutgoing, |
+ VideoReceiveStreamConfig, |
+ VideoSendStreamConfig |
+ }; |
+ |
+ virtual ~RtcEvent() = default; |
+ |
+ virtual Type GetType() const = 0; |
+}; |
+ |
+} // namespace webrtc |
+ |
+#endif // WEBRTC_LOGGING_RTC_EVENT_LOG_EVENTS_RTC_EVENT_H_ |