| /* | |
| * 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> | |
| #include "api/units/timestamp.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 : uint32_t { | |
| AlrStateEvent, | |
| RouteChangeEvent, | |
| RemoteEstimateEvent, | |
| AudioNetworkAdaptation, | |
| AudioPlayout, | |
| AudioReceiveStreamConfig, | |
| AudioSendStreamConfig, | |
| BweUpdateDelayBased, | |
| BweUpdateLossBased, | |
| BweUpdateScream, | |
| DtlsTransportState, | |
| DtlsWritableState, | |
| IceCandidatePairConfig, | |
| IceCandidatePairEvent, | |
| ProbeClusterCreated, | |
| ProbeResultFailure, | |
| ProbeResultSuccess, | |
| RtcpPacketIncoming, | |
| RtcpPacketOutgoing, | |
| RtpPacketIncoming, | |
| RtpPacketOutgoing, | |
| VideoReceiveStreamConfig, | |
| VideoSendStreamConfig, | |
| FrameDecoded, | |
| NetEqSetMinimumDelay, | |
| BeginV3Log = 0x2501580, | |
| EndV3Log = 0x2501581, | |
| FakeEvent, // For unit testing. | |
| }; | |
| RtcEvent() = default; | |
| RtcEvent(const RtcEvent&) = default; | |
| RtcEvent& operator=(const RtcEvent&) = delete; | |
| virtual ~RtcEvent() = default; | |
| virtual Type GetType() const = 0; | |
| virtual bool IsConfigEvent() const = 0; | |
| // Events are grouped by Type before being encoded. | |
| // Optionally, `GetGroupKey` can be overloaded to group the | |
| // events by a secondary key (in addition to the event type.) | |
| // This can, in some cases, improve compression efficiency | |
| // e.g. by grouping events by SSRC. | |
| virtual uint32_t GetGroupKey() const { return 0; } | |
| int64_t timestamp_ms() const { return timestamp_us_ / 1000; } | |
| int64_t timestamp_us() const { return timestamp_us_; } | |
| // Time when the event was logged. | |
| Timestamp timestamp() const { return Timestamp::Micros(timestamp_us_); } | |
| void SetTimestamp(Timestamp timestamp) { timestamp_us_ = timestamp.us(); } | |
| protected: | |
| explicit RtcEvent(int64_t timestamp_us) : timestamp_us_(timestamp_us) {} | |
| private: | |
| int64_t timestamp_us_ = 0; | |
| }; | |
| } // namespace webrtc | |
| #endif // API_RTC_EVENT_LOG_RTC_EVENT_H_ |