| /* |
| * 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 <memory> |
| #include <string> |
| |
| #include "webrtc/base/platform_file.h" |
| #include "webrtc/call/audio_receive_stream.h" |
| #include "webrtc/call/audio_send_stream.h" |
| #include "webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h" |
| #include "webrtc/video_receive_stream.h" |
| #include "webrtc/video_send_stream.h" |
| |
| namespace webrtc { |
| |
| // Forward declaration of storage class that is automatically generated from |
| // the protobuf file. |
| namespace rtclog { |
| class EventStream; |
| } // namespace rtclog |
| |
| class Clock; |
| class RtcEventLogImpl; |
| |
| enum class MediaType; |
| |
| enum PacketDirection { kIncomingPacket = 0, kOutgoingPacket }; |
| |
| class RtcEventLog { |
| public: |
| virtual ~RtcEventLog() {} |
| |
| // Factory method to create an RtcEventLog object. |
| static std::unique_ptr<RtcEventLog> 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<RtcEventLog> Create(const Clock* clock) { |
| return Create(); |
| } |
| |
| // Create an RtcEventLog object that does nothing. |
| static std::unique_ptr<RtcEventLog> 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 webrtc::VideoReceiveStream. |
| virtual void LogVideoReceiveStreamConfig( |
| const webrtc::VideoReceiveStream::Config& config) = 0; |
| |
| // Logs configuration information for webrtc::VideoSendStream. |
| virtual void LogVideoSendStreamConfig( |
| const webrtc::VideoSendStream::Config& config) = 0; |
| |
| // Logs configuration information for webrtc::AudioReceiveStream. |
| virtual void LogAudioReceiveStreamConfig( |
| const webrtc::AudioReceiveStream::Config& config) = 0; |
| |
| // Logs configuration information for webrtc::AudioSendStream. |
| virtual void LogAudioSendStreamConfig( |
| const webrtc::AudioSendStream::Config& 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, |
| MediaType media_type, |
| const uint8_t* header, |
| size_t packet_length) = 0; |
| |
| // Logs an incoming or outgoing RTCP packet. |
| virtual void LogRtcpPacket(PacketDirection direction, |
| MediaType media_type, |
| 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 LogBwePacketLossEvent(int32_t bitrate, |
| uint8_t fraction_loss, |
| int32_t total_packets) = 0; |
| |
| // Logs audio encoder re-configuration driven by audio network adaptor. |
| virtual void LogAudioNetworkAdaptation( |
| const AudioNetworkAdaptor::EncoderRuntimeConfig& config) = 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 final : 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 VideoReceiveStream::Config& config) override {} |
| void LogVideoSendStreamConfig( |
| const VideoSendStream::Config& config) override {} |
| void LogAudioReceiveStreamConfig( |
| const AudioReceiveStream::Config& config) override {} |
| void LogAudioSendStreamConfig( |
| const AudioSendStream::Config& config) override {} |
| void LogRtpHeader(PacketDirection direction, |
| MediaType media_type, |
| const uint8_t* header, |
| size_t packet_length) override {} |
| void LogRtcpPacket(PacketDirection direction, |
| MediaType media_type, |
| const uint8_t* packet, |
| size_t length) override {} |
| void LogAudioPlayout(uint32_t ssrc) override {} |
| void LogBwePacketLossEvent(int32_t bitrate, |
| uint8_t fraction_loss, |
| int32_t total_packets) override {} |
| void LogAudioNetworkAdaptation( |
| const AudioNetworkAdaptor::EncoderRuntimeConfig& config) override{}; |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // WEBRTC_LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_H_ |