/* * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. * * Use of this source code is governed by a BSD-style license * that can be found in the LICENSE file in the root of the source * tree. An additional intellectual property rights grant can be found * in the file PATENTS. All contributing project authors may * be found in the AUTHORS file in the root of the source tree. */ #ifndef WEBRTC_LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_H_ #define WEBRTC_LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_H_ #include #include #include #include "webrtc/base/platform_file.h" #include "webrtc/config.h" namespace webrtc { // Forward declaration of storage class that is automatically generated from // the protobuf file. namespace rtclog { class EventStream; struct StreamConfig { uint32_t local_ssrc = 0; uint32_t remote_ssrc = 0; uint32_t rtx_ssrc = 0; std::string rsid; bool remb = false; std::vector rtp_extensions; RtcpMode rtcp_mode = RtcpMode::kReducedSize; struct Codec { Codec(const std::string& payload_name, int payload_type, int rtx_payload_type) : payload_name(payload_name), payload_type(payload_type), rtx_payload_type(rtx_payload_type) {} std::string payload_name; int payload_type; int rtx_payload_type; }; std::vector codecs; }; } // namespace rtclog class Clock; class RtcEventLogImpl; struct AudioEncoderRuntimeConfig; enum class MediaType; enum class BandwidthUsage; enum PacketDirection { kIncomingPacket = 0, kOutgoingPacket }; enum ProbeFailureReason { kInvalidSendReceiveInterval, kInvalidSendReceiveRatio, kTimeout }; class RtcEventLog { public: virtual ~RtcEventLog() {} // Factory method to create an RtcEventLog object. static std::unique_ptr Create(); // TODO(nisse): webrtc::Clock is deprecated. Delete this method and // above forward declaration of Clock when // webrtc/system_wrappers/include/clock.h is deleted. static std::unique_ptr Create(const Clock* clock) { return Create(); } // Create an RtcEventLog object that does nothing. static std::unique_ptr CreateNull(); // Starts logging a maximum of max_size_bytes bytes to the specified file. // If the file already exists it will be overwritten. // If max_size_bytes <= 0, logging will be active until StopLogging is called. // The function has no effect and returns false if we can't start a new log // e.g. because we are already logging or the file cannot be opened. virtual bool StartLogging(const std::string& file_name, int64_t max_size_bytes) = 0; // Same as above. The RtcEventLog takes ownership of the file if the call // is successful, i.e. if it returns true. virtual bool StartLogging(rtc::PlatformFile platform_file, int64_t max_size_bytes) = 0; // Deprecated. Pass an explicit file size limit. bool StartLogging(const std::string& file_name) { return StartLogging(file_name, 10000000); } // Deprecated. Pass an explicit file size limit. bool StartLogging(rtc::PlatformFile platform_file) { return StartLogging(platform_file, 10000000); } // Stops logging to file and waits until the thread has finished. virtual void StopLogging() = 0; // Logs configuration information for a video receive stream. virtual void LogVideoReceiveStreamConfig( const rtclog::StreamConfig& config) = 0; // Logs configuration information for a video send stream. virtual void LogVideoSendStreamConfig(const rtclog::StreamConfig& config) = 0; // Logs configuration information for an audio receive stream. virtual void LogAudioReceiveStreamConfig( const rtclog::StreamConfig& config) = 0; // Logs configuration information for an audio send stream. virtual void LogAudioSendStreamConfig(const rtclog::StreamConfig& config) = 0; // Logs the header of an incoming or outgoing RTP packet. packet_length // is the total length of the packet, including both header and payload. virtual void LogRtpHeader(PacketDirection direction, const uint8_t* header, size_t packet_length) = 0; // Same as above but used on the sender side to log packets that are part of // a probe cluster. virtual void LogRtpHeader(PacketDirection direction, const uint8_t* header, size_t packet_length, int probe_cluster_id) = 0; // Logs an incoming or outgoing RTCP packet. virtual void LogRtcpPacket(PacketDirection direction, const uint8_t* packet, size_t length) = 0; // Logs an audio playout event. virtual void LogAudioPlayout(uint32_t ssrc) = 0; // Logs a bitrate update from the bandwidth estimator based on packet loss. virtual void LogLossBasedBweUpdate(int32_t bitrate_bps, uint8_t fraction_loss, int32_t total_packets) = 0; // Logs a bitrate update from the bandwidth estimator based on delay changes. virtual void LogDelayBasedBweUpdate(int32_t bitrate_bps, BandwidthUsage detector_state) = 0; // Logs audio encoder re-configuration driven by audio network adaptor. virtual void LogAudioNetworkAdaptation( const AudioEncoderRuntimeConfig& config) = 0; // Logs when a probe cluster is created. virtual void LogProbeClusterCreated(int id, int bitrate_bps, int min_probes, int min_bytes) = 0; // Logs the result of a successful probing attempt. virtual void LogProbeResultSuccess(int id, int bitrate_bps) = 0; // Logs the result of an unsuccessful probing attempt. virtual void LogProbeResultFailure(int id, ProbeFailureReason failure_reason) = 0; // Reads an RtcEventLog file and returns true when reading was successful. // The result is stored in the given EventStream object. // The order of the events in the EventStream is implementation defined. // The current implementation writes a LOG_START event, then the old // configurations, then the remaining events in timestamp order and finally // a LOG_END event. However, this might change without further notice. // TODO(terelius): Change result type to a vector? static bool ParseRtcEventLog(const std::string& file_name, rtclog::EventStream* result); }; // No-op implementation is used if flag is not set, or in tests. class RtcEventLogNullImpl : public RtcEventLog { public: bool StartLogging(const std::string& file_name, int64_t max_size_bytes) override { return false; } bool StartLogging(rtc::PlatformFile platform_file, int64_t max_size_bytes) override; void StopLogging() override {} void LogVideoReceiveStreamConfig( const rtclog::StreamConfig& config) override {} void LogVideoSendStreamConfig(const rtclog::StreamConfig& config) override {} void LogAudioReceiveStreamConfig( const rtclog::StreamConfig& config) override {} void LogAudioSendStreamConfig(const rtclog::StreamConfig& config) override {} void LogRtpHeader(PacketDirection direction, const uint8_t* header, size_t packet_length) override {} void LogRtpHeader(PacketDirection direction, const uint8_t* header, size_t packet_length, int probe_cluster_id) override {} void LogRtcpPacket(PacketDirection direction, const uint8_t* packet, size_t length) override {} void LogAudioPlayout(uint32_t ssrc) override {} void LogLossBasedBweUpdate(int32_t bitrate_bps, uint8_t fraction_loss, int32_t total_packets) override {} void LogDelayBasedBweUpdate(int32_t bitrate_bps, BandwidthUsage detector_state) override {} void LogAudioNetworkAdaptation( const AudioEncoderRuntimeConfig& config) override {} void LogProbeClusterCreated(int id, int bitrate_bps, int min_probes, int min_bytes) override{}; void LogProbeResultSuccess(int id, int bitrate_bps) override{}; void LogProbeResultFailure(int id, ProbeFailureReason failure_reason) override{}; }; } // namespace webrtc #endif // WEBRTC_LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_H_