Danil Chapovalov | b32f2c7 | 2019-05-22 13:39:25 +0200 | [diff] [blame] | 1 | /* |
| 2 | * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. |
| 3 | * |
| 4 | * Use of this source code is governed by a BSD-style license |
| 5 | * that can be found in the LICENSE file in the root of the source |
| 6 | * tree. An additional intellectual property rights grant can be found |
| 7 | * in the file PATENTS. All contributing project authors may |
| 8 | * be found in the AUTHORS file in the root of the source tree. |
| 9 | */ |
| 10 | |
| 11 | #ifndef API_RTC_EVENT_LOG_RTC_EVENT_H_ |
| 12 | #define API_RTC_EVENT_LOG_RTC_EVENT_H_ |
| 13 | |
| 14 | #include <cstdint> |
| 15 | |
| 16 | namespace webrtc { |
| 17 | |
| 18 | // This class allows us to store unencoded RTC events. Subclasses of this class |
| 19 | // store the actual information. This allows us to keep all unencoded events, |
| 20 | // even when their type and associated information differ, in the same buffer. |
| 21 | // Additionally, it prevents dependency leaking - a module that only logs |
| 22 | // events of type RtcEvent_A doesn't need to know about anything associated |
| 23 | // with events of type RtcEvent_B. |
| 24 | class RtcEvent { |
| 25 | public: |
| 26 | // Subclasses of this class have to associate themselves with a unique value |
| 27 | // of Type. This leaks the information of existing subclasses into the |
| 28 | // superclass, but the *actual* information - rtclog::StreamConfig, etc. - |
| 29 | // is kept separate. |
| 30 | enum class Type { |
| 31 | AlrStateEvent, |
| 32 | RouteChangeEvent, |
Sebastian Jansson | 0a5ed89 | 2019-09-18 15:37:31 +0200 | [diff] [blame] | 33 | RemoteEstimateEvent, |
Danil Chapovalov | b32f2c7 | 2019-05-22 13:39:25 +0200 | [diff] [blame] | 34 | AudioNetworkAdaptation, |
| 35 | AudioPlayout, |
| 36 | AudioReceiveStreamConfig, |
| 37 | AudioSendStreamConfig, |
| 38 | BweUpdateDelayBased, |
| 39 | BweUpdateLossBased, |
| 40 | DtlsTransportState, |
| 41 | DtlsWritableState, |
| 42 | IceCandidatePairConfig, |
| 43 | IceCandidatePairEvent, |
| 44 | ProbeClusterCreated, |
| 45 | ProbeResultFailure, |
| 46 | ProbeResultSuccess, |
| 47 | RtcpPacketIncoming, |
| 48 | RtcpPacketOutgoing, |
| 49 | RtpPacketIncoming, |
| 50 | RtpPacketOutgoing, |
| 51 | VideoReceiveStreamConfig, |
| 52 | VideoSendStreamConfig, |
| 53 | GenericPacketSent, |
| 54 | GenericPacketReceived, |
| 55 | GenericAckReceived |
| 56 | }; |
| 57 | |
| 58 | RtcEvent(); |
| 59 | virtual ~RtcEvent() = default; |
| 60 | |
| 61 | virtual Type GetType() const = 0; |
| 62 | |
| 63 | virtual bool IsConfigEvent() const = 0; |
| 64 | |
| 65 | int64_t timestamp_ms() const { return timestamp_us_ / 1000; } |
| 66 | int64_t timestamp_us() const { return timestamp_us_; } |
| 67 | |
| 68 | protected: |
| 69 | explicit RtcEvent(int64_t timestamp_us) : timestamp_us_(timestamp_us) {} |
| 70 | |
| 71 | const int64_t timestamp_us_; |
| 72 | }; |
| 73 | |
| 74 | } // namespace webrtc |
| 75 | |
| 76 | #endif // API_RTC_EVENT_LOG_RTC_EVENT_H_ |