1adce14c87
Time to keep old events in buffer is now changeable at runtime. Added unit test for removing old events from buffer. Review URL: https://codereview.webrtc.org/1303713002 Cr-Commit-Position: refs/heads/master@{#10302}
89 lines
3.1 KiB
C++
89 lines
3.1 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 WEBRTC_CALL_RTC_EVENT_LOG_H_
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#define WEBRTC_CALL_RTC_EVENT_LOG_H_
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#include <string>
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#include "webrtc/base/platform_file.h"
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#include "webrtc/base/scoped_ptr.h"
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#include "webrtc/video_receive_stream.h"
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#include "webrtc/video_send_stream.h"
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namespace webrtc {
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// Forward declaration of storage class that is automatically generated from
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// the protobuf file.
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namespace rtclog {
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class EventStream;
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} // namespace rtclog
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class RtcEventLogImpl;
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enum class MediaType;
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class RtcEventLog {
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public:
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virtual ~RtcEventLog() {}
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static rtc::scoped_ptr<RtcEventLog> Create();
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// Sets the time that events are stored in the internal event buffer
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// before the user calls StartLogging. The default is 10 000 000 us = 10 s
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virtual void SetBufferDuration(int64_t buffer_duration_us) = 0;
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// Starts logging for the specified duration to the specified file.
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// The logging will stop automatically after the specified duration.
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// If the file already exists it will be overwritten.
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// If the file cannot be opened, the RtcEventLog will not start logging.
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virtual void StartLogging(const std::string& file_name, int duration_ms) = 0;
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// Starts logging until either the 10 minute timer runs out or the StopLogging
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// function is called. The RtcEventLog takes ownership of the supplied
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// rtc::PlatformFile.
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virtual bool StartLogging(rtc::PlatformFile log_file) = 0;
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virtual void StopLogging() = 0;
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// Logs configuration information for webrtc::VideoReceiveStream
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virtual void LogVideoReceiveStreamConfig(
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const webrtc::VideoReceiveStream::Config& config) = 0;
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// Logs configuration information for webrtc::VideoSendStream
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virtual void LogVideoSendStreamConfig(
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const webrtc::VideoSendStream::Config& config) = 0;
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// Logs the header of an incoming or outgoing RTP packet. packet_length
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// is the total length of the packet, including both header and payload.
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virtual void LogRtpHeader(bool incoming,
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MediaType media_type,
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const uint8_t* header,
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size_t packet_length) = 0;
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// Logs an incoming or outgoing RTCP packet.
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virtual void LogRtcpPacket(bool incoming,
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MediaType media_type,
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const uint8_t* packet,
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size_t length) = 0;
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// Logs an audio playout event
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virtual void LogAudioPlayout(uint32_t ssrc) = 0;
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// Reads an RtcEventLog file and returns true when reading was successful.
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// The result is stored in the given EventStream object.
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static bool ParseRtcEventLog(const std::string& file_name,
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rtclog::EventStream* result);
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};
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} // namespace webrtc
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#endif // WEBRTC_CALL_RTC_EVENT_LOG_H_
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