| /* |
| * 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 LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_H_ |
| #define LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_H_ |
| |
| #include <memory> |
| #include <string> |
| |
| // TODO(eladalon): Remove this include once LogIncomingRtcpPacket(), etc., have |
| // been removed (they are currently deprecated). |
| #include "api/array_view.h" |
| #include "common_types.h" // NOLINT(build/include) |
| #include "logging/rtc_event_log/output/rtc_event_log_output.h" |
| // TODO(eladalon): This is here because of ProbeFailureReason; remove this |
| // dependency along with the deprecated LogProbeResultFailure(). |
| #include "logging/rtc_event_log/events/rtc_event_probe_result_failure.h" |
| // TODO(eladalon): Remove this #include once the deprecated versions of |
| // StartLogging() have been removed. |
| #include "rtc_base/platform_file.h" |
| |
| namespace webrtc { |
| |
| namespace rtclog { |
| class EventStream; // Storage class automatically generated from protobuf. |
| // TODO(eladalon): Get rid of this when deprecated methods are removed. |
| struct StreamConfig; |
| } // namespace rtclog |
| |
| class Clock; |
| // TODO(eladalon): The following may be removed when the deprecated methods |
| // are removed. |
| struct AudioEncoderRuntimeConfig; |
| class RtpPacketReceived; |
| class RtpPacketToSend; |
| enum class BandwidthUsage; |
| enum PacketDirection { kIncomingPacket = 0, kOutgoingPacket }; |
| |
| class RtcEventLog { |
| public: |
| enum : size_t { kUnlimitedOutput = 0 }; |
| |
| // TODO(eladalon): Two stages are upcoming. |
| // 1. Extend this to actually support the new encoding. |
| // 2. Get rid of the legacy encoding, allowing us to get rid of this enum. |
| enum class EncodingType { Legacy }; |
| |
| virtual ~RtcEventLog() {} |
| |
| // Factory method to create an RtcEventLog object. |
| // TODO(eladalon): Get rid of the default value after internal projects fixed. |
| static std::unique_ptr<RtcEventLog> Create( |
| EncodingType encoding_type = EncodingType::Legacy); |
| // TODO(nisse): webrtc::Clock is deprecated. Delete this method and |
| // above forward declaration of Clock when |
| // webrtc/system_wrappers/include/clock.h is deleted. |
| // TODO(eladalon): Get rid of the default value after internal projects fixed. |
| static std::unique_ptr<RtcEventLog> Create( |
| const Clock* clock, |
| EncodingType encoding_type = EncodingType::Legacy) { |
| return Create(encoding_type); |
| } |
| |
| // Create an RtcEventLog object that does nothing. |
| static std::unique_ptr<RtcEventLog> CreateNull(); |
| |
| // Starts logging to a given output. The output might be limited in size, |
| // and may close itself once it has reached the maximum size. |
| virtual bool StartLogging(std::unique_ptr<RtcEventLogOutput> output) = 0; |
| |
| // 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. |
| RTC_DEPRECATED 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. |
| RTC_DEPRECATED virtual bool StartLogging(rtc::PlatformFile platform_file, |
| int64_t max_size_bytes) = 0; |
| |
| // Deprecated. Pass an explicit file size limit. |
| RTC_DEPRECATED bool StartLogging(const std::string& file_name) { |
| return StartLogging(file_name, 10000000); |
| } |
| |
| // Deprecated. Pass an explicit file size limit. |
| RTC_DEPRECATED bool StartLogging(rtc::PlatformFile platform_file) { |
| return StartLogging(platform_file, 10000000); |
| } |
| |
| // Stops logging to file and waits until the file has been closed, after |
| // which it would be permissible to read and/or modify it. |
| virtual void StopLogging() = 0; |
| |
| // Log an RTC event (the type of event is determined by the subclass). |
| virtual void Log(std::unique_ptr<RtcEvent> event) = 0; |
| |
| // Logs configuration information for a video receive stream. |
| RTC_DEPRECATED virtual void LogVideoReceiveStreamConfig( |
| const rtclog::StreamConfig& config) = 0; |
| |
| // Logs configuration information for a video send stream. |
| RTC_DEPRECATED virtual void LogVideoSendStreamConfig( |
| const rtclog::StreamConfig& config) = 0; |
| |
| // Logs configuration information for an audio receive stream. |
| RTC_DEPRECATED virtual void LogAudioReceiveStreamConfig( |
| const rtclog::StreamConfig& config) = 0; |
| |
| // Logs configuration information for an audio send stream. |
| RTC_DEPRECATED virtual void LogAudioSendStreamConfig( |
| const rtclog::StreamConfig& config) = 0; |
| |
| RTC_DEPRECATED virtual void LogRtpHeader(PacketDirection direction, |
| const uint8_t* header, |
| size_t packet_length) {} |
| |
| RTC_DEPRECATED virtual void LogRtpHeader(PacketDirection direction, |
| const uint8_t* header, |
| size_t packet_length, |
| int probe_cluster_id) {} |
| |
| // Logs the header of an incoming RTP packet. |packet_length| |
| // is the total length of the packet, including both header and payload. |
| RTC_DEPRECATED virtual void LogIncomingRtpHeader( |
| const RtpPacketReceived& packet) = 0; |
| |
| // Logs the header of an incoming RTP packet. |packet_length| |
| // is the total length of the packet, including both header and payload. |
| RTC_DEPRECATED virtual void LogOutgoingRtpHeader( |
| const RtpPacketToSend& packet, |
| int probe_cluster_id) = 0; |
| |
| RTC_DEPRECATED virtual void LogRtcpPacket(PacketDirection direction, |
| const uint8_t* header, |
| size_t packet_length) {} |
| |
| // Logs an incoming RTCP packet. |
| RTC_DEPRECATED virtual void LogIncomingRtcpPacket( |
| rtc::ArrayView<const uint8_t> packet) = 0; |
| |
| // Logs an outgoing RTCP packet. |
| RTC_DEPRECATED virtual void LogOutgoingRtcpPacket( |
| rtc::ArrayView<const uint8_t> packet) = 0; |
| |
| // Logs an audio playout event. |
| RTC_DEPRECATED virtual void LogAudioPlayout(uint32_t ssrc) = 0; |
| |
| // Logs a bitrate update from the bandwidth estimator based on packet loss. |
| RTC_DEPRECATED 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. |
| RTC_DEPRECATED virtual void LogDelayBasedBweUpdate( |
| int32_t bitrate_bps, |
| BandwidthUsage detector_state) = 0; |
| |
| // Logs audio encoder re-configuration driven by audio network adaptor. |
| RTC_DEPRECATED virtual void LogAudioNetworkAdaptation( |
| const AudioEncoderRuntimeConfig& config) = 0; |
| |
| // Logs when a probe cluster is created. |
| RTC_DEPRECATED virtual void LogProbeClusterCreated(int id, |
| int bitrate_bps, |
| int min_probes, |
| int min_bytes) = 0; |
| |
| // Logs the result of a successful probing attempt. |
| RTC_DEPRECATED virtual void LogProbeResultSuccess(int id, |
| int bitrate_bps) = 0; |
| |
| // Logs the result of an unsuccessful probing attempt. |
| RTC_DEPRECATED virtual void LogProbeResultFailure( |
| int id, |
| ProbeFailureReason failure_reason) = 0; |
| }; |
| |
| // No-op implementation is used if flag is not set, or in tests. |
| class RtcEventLogNullImpl : public RtcEventLog { |
| public: |
| bool StartLogging(std::unique_ptr<RtcEventLogOutput> output) override { |
| return false; |
| } |
| 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 { |
| return false; |
| } |
| void StopLogging() override {} |
| void Log(std::unique_ptr<RtcEvent> event) 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 LogIncomingRtpHeader(const RtpPacketReceived& packet) override {} |
| void LogOutgoingRtpHeader(const RtpPacketToSend& packet, |
| int probe_cluster_id) override {} |
| void LogIncomingRtcpPacket(rtc::ArrayView<const uint8_t> packet) override {} |
| void LogOutgoingRtcpPacket(rtc::ArrayView<const uint8_t> packet) 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 // LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_H_ |