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The new API stores events gathered by event type. For example, it is possible to ask for a list of all incoming RTCP messages or all audio playout events. The new API is experimental and may change over next few weeks. Once it has stabilized and all unit tests and existing tools have been ported to the new API, the old one will be removed. This CL also updates the event_log_visualizer tool to use the new parser API. This is not a funcional change except for: - Incoming and outgoing audio level are now drawn in two separate plots. - Incoming and outgoing timstamps are now drawn in two separate plots. - RTCP count is no longer split into Video and Audio. It also counts all RTCP packets rather than only specific message types. - Slight timing difference in sendside BWE simulation due to only iterating over transport feedbacks and not over all RTCP packets. This timing changes are not visible in the plots. Media type for RTCP messages might not be identified correctly by rtc_event_log2text anymore. On the other hand, assigning a specific media type to an RTCP packet was a bit hacky to begin with. Bug: webrtc:8111 Change-Id: Ib244338c86a2c1a010c668a7aba440482023b512 Reviewed-on: https://webrtc-review.googlesource.com/73140 Reviewed-by: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Minyue Li <minyue@webrtc.org> Commit-Queue: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23056}
75 lines
2.6 KiB
C++
75 lines
2.6 KiB
C++
/*
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* Copyright (c) 2015 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 LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_H_
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#define LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_H_
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#include <memory>
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#include <string>
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#include <utility>
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#include "api/rtceventlogoutput.h"
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#include "logging/rtc_event_log/events/rtc_event.h"
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#include "rtc_base/task_queue.h"
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namespace webrtc {
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// TODO(terelius): Move this to the parser.
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enum PacketDirection { kIncomingPacket = 0, kOutgoingPacket };
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class RtcEventLog {
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public:
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enum : size_t { kUnlimitedOutput = 0 };
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enum : int64_t { kImmediateOutput = 0 };
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// TODO(eladalon): Two stages are upcoming.
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// 1. Extend this to actually support the new encoding.
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// 2. Get rid of the legacy encoding, allowing us to get rid of this enum.
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enum class EncodingType { Legacy };
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virtual ~RtcEventLog() {}
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// Factory method to create an RtcEventLog object.
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static std::unique_ptr<RtcEventLog> Create(EncodingType encoding_type);
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// Factory method to create an RtcEventLog object which uses the given
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// rtc::TaskQueue for emitting output.
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static std::unique_ptr<RtcEventLog> Create(
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EncodingType encoding_type,
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std::unique_ptr<rtc::TaskQueue> task_queue);
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// Create an RtcEventLog object that does nothing.
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static std::unique_ptr<RtcEventLog> CreateNull();
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// Starts logging to a given output. The output might be limited in size,
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// and may close itself once it has reached the maximum size.
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virtual bool StartLogging(std::unique_ptr<RtcEventLogOutput> output,
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int64_t output_period_ms) = 0;
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// Stops logging to file and waits until the file has been closed, after
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// which it would be permissible to read and/or modify it.
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virtual void StopLogging() = 0;
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// Log an RTC event (the type of event is determined by the subclass).
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virtual void Log(std::unique_ptr<RtcEvent> event) = 0;
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};
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// No-op implementation is used if flag is not set, or in tests.
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class RtcEventLogNullImpl : public RtcEventLog {
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public:
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bool StartLogging(std::unique_ptr<RtcEventLogOutput> output,
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int64_t output_period_ms) override;
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void StopLogging() override {}
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void Log(std::unique_ptr<RtcEvent> event) override {}
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};
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} // namespace webrtc
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#endif // LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_H_
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