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00c12f6779
This CL adds the possibility to log metainformation about decoded frames in RTC event log, including encoding parsing and tests. It will be wired up in a followup CL. Bug: webrtc:8802 Change-Id: Ied598b266513d0f63fce0484d741af1782607e74 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181061 Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Commit-Queue: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31873}
77 lines
2.3 KiB
C++
77 lines
2.3 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 {
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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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};
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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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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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