/* * 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 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, RemoteEstimateEvent, AudioNetworkAdaptation, AudioPlayout, AudioReceiveStreamConfig, AudioSendStreamConfig, BweUpdateDelayBased, BweUpdateLossBased, DtlsTransportState, DtlsWritableState, IceCandidatePairConfig, IceCandidatePairEvent, ProbeClusterCreated, ProbeResultFailure, ProbeResultSuccess, RtcpPacketIncoming, RtcpPacketOutgoing, RtpPacketIncoming, RtpPacketOutgoing, VideoReceiveStreamConfig, VideoSendStreamConfig, GenericPacketSent, GenericPacketReceived, GenericAckReceived, FrameDecoded }; 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_