78 lines
2.3 KiB
C
78 lines
2.3 KiB
C
|
/*
|
||
|
|
* 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>
|
||
|
|
|
||
|
|
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_
|