112 lines
4.1 KiB
C
112 lines
4.1 KiB
C
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/*
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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 <memory>
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#include <string>
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#include "webrtc/base/platform_file.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 Clock;
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class RtcEventLogImpl;
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enum class MediaType;
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enum PacketDirection { kIncomingPacket = 0, kOutgoingPacket };
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class RtcEventLog {
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public:
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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(const Clock* clock);
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// Starts logging a maximum of max_size_bytes bytes to the specified file.
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// If the file already exists it will be overwritten.
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// If max_size_bytes <= 0, logging will be active until StopLogging is called.
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// The function has no effect and returns false if we can't start a new log
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// e.g. because we are already logging or the file cannot be opened.
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virtual bool StartLogging(const std::string& file_name,
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int64_t max_size_bytes) = 0;
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// Same as above. The RtcEventLog takes ownership of the file if the call
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// is successful, i.e. if it returns true.
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virtual bool StartLogging(rtc::PlatformFile platform_file,
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int64_t max_size_bytes) = 0;
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// Deprecated. Pass an explicit file size limit.
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bool StartLogging(const std::string& file_name) {
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return StartLogging(file_name, 10000000);
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}
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// Deprecated. Pass an explicit file size limit.
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bool StartLogging(rtc::PlatformFile platform_file) {
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return StartLogging(platform_file, 10000000);
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}
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// Stops logging to file and waits until the thread has finished.
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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(PacketDirection direction,
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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(PacketDirection direction,
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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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// Logs a bitrate update from the bandwidth estimator based on packet loss.
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virtual void LogBwePacketLossEvent(int32_t bitrate,
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uint8_t fraction_loss,
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int32_t total_packets) = 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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// The order of the events in the EventStream is implementation defined.
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// The current implementation writes a LOG_START event, then the old
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// configurations, then the remaining events in timestamp order and finally
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// a LOG_END event. However, this might change without further notice.
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// TODO(terelius): Change result type to a vector?
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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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