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612872b29d
To be able to simulate offline some scenario in which the javascript layer set the minimum base buffer size of neteq, it is required to record those API calls. This change introduces this. Bug: webrtc:14763 Change-Id: Ic817913eda60978d6fca3f8e12229aeec505ca25 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287122 Auto-Submit: Lionel Koenig <lionelk@webrtc.org> Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Lionel Koenig <lionelk@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39104}
88 lines
2.7 KiB
C++
88 lines
2.7 KiB
C++
/*
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* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef API_RTC_EVENT_LOG_RTC_EVENT_H_
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#define API_RTC_EVENT_LOG_RTC_EVENT_H_
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#include <cstdint>
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namespace webrtc {
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// This class allows us to store unencoded RTC events. Subclasses of this class
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// store the actual information. This allows us to keep all unencoded events,
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// even when their type and associated information differ, in the same buffer.
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// Additionally, it prevents dependency leaking - a module that only logs
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// events of type RtcEvent_A doesn't need to know about anything associated
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// with events of type RtcEvent_B.
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class RtcEvent {
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public:
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// Subclasses of this class have to associate themselves with a unique value
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// of Type. This leaks the information of existing subclasses into the
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// superclass, but the *actual* information - rtclog::StreamConfig, etc. -
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// is kept separate.
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enum class Type : uint32_t {
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AlrStateEvent,
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RouteChangeEvent,
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RemoteEstimateEvent,
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AudioNetworkAdaptation,
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AudioPlayout,
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AudioReceiveStreamConfig,
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AudioSendStreamConfig,
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BweUpdateDelayBased,
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BweUpdateLossBased,
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DtlsTransportState,
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DtlsWritableState,
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IceCandidatePairConfig,
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IceCandidatePairEvent,
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ProbeClusterCreated,
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ProbeResultFailure,
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ProbeResultSuccess,
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RtcpPacketIncoming,
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RtcpPacketOutgoing,
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RtpPacketIncoming,
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RtpPacketOutgoing,
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VideoReceiveStreamConfig,
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VideoSendStreamConfig,
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GenericPacketSent,
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GenericPacketReceived,
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GenericAckReceived,
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FrameDecoded,
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NetEqSetMinimumDelay,
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BeginV3Log = 0x2501580,
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EndV3Log = 0x2501581,
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FakeEvent, // For unit testing.
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};
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RtcEvent();
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virtual ~RtcEvent() = default;
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virtual Type GetType() const = 0;
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virtual bool IsConfigEvent() const = 0;
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// Events are grouped by Type before being encoded.
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// Optionally, `GetGroupKey` can be overloaded to group the
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// events by a secondary key (in addition to the event type.)
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// This can, in some cases, improve compression efficiency
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// e.g. by grouping events by SSRC.
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virtual uint32_t GetGroupKey() const { return 0; }
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int64_t timestamp_ms() const { return timestamp_us_ / 1000; }
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int64_t timestamp_us() const { return timestamp_us_; }
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protected:
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explicit RtcEvent(int64_t timestamp_us) : timestamp_us_(timestamp_us) {}
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const int64_t timestamp_us_;
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};
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} // namespace webrtc
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#endif // API_RTC_EVENT_LOG_RTC_EVENT_H_
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