| /* |
| * 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 LOGGING_RTC_EVENT_LOG_EVENTS_RTC_EVENT_H_ |
| #define LOGGING_RTC_EVENT_LOG_EVENTS_RTC_EVENT_H_ |
| |
| #include <typedefs.h> |
| |
| #include "rtc_base/timeutils.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 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 { |
| AudioNetworkAdaptation, |
| AudioPlayout, |
| AudioReceiveStreamConfig, |
| AudioSendStreamConfig, |
| BweUpdateDelayBased, |
| BweUpdateLossBased, |
| LoggingStarted, |
| LoggingStopped, |
| ProbeClusterCreated, |
| ProbeResultFailure, |
| ProbeResultSuccess, |
| RtcpPacketIncoming, |
| RtcpPacketOutgoing, |
| RtpPacketIncoming, |
| RtpPacketOutgoing, |
| VideoReceiveStreamConfig, |
| VideoSendStreamConfig |
| }; |
| |
| RtcEvent() : timestamp_us_(rtc::TimeMicros()) {} |
| virtual ~RtcEvent() = default; |
| |
| virtual Type GetType() const = 0; |
| |
| virtual bool IsConfigEvent() const = 0; |
| |
| const int64_t timestamp_us_; |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // LOGGING_RTC_EVENT_LOG_EVENTS_RTC_EVENT_H_ |