| /* |
| * Copyright (c) 2016 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_CALL_RTC_EVENT_LOG_PARSER_H_ |
| #define WEBRTC_CALL_RTC_EVENT_LOG_PARSER_H_ |
| |
| #include <string> |
| #include <vector> |
| |
| #include "webrtc/base/ignore_wundef.h" |
| #include "webrtc/call/rtc_event_log.h" |
| #include "webrtc/video_receive_stream.h" |
| #include "webrtc/video_send_stream.h" |
| |
| // Files generated at build-time by the protobuf compiler. |
| RTC_PUSH_IGNORING_WUNDEF() |
| #ifdef WEBRTC_ANDROID_PLATFORM_BUILD |
| #include "external/webrtc/webrtc/call/rtc_event_log.pb.h" |
| #else |
| #include "webrtc/call/rtc_event_log.pb.h" |
| #endif |
| RTC_POP_IGNORING_WUNDEF() |
| |
| namespace webrtc { |
| |
| enum class MediaType; |
| |
| class ParsedRtcEventLog { |
| friend class RtcEventLogTestHelper; |
| |
| public: |
| enum EventType { |
| UNKNOWN_EVENT = 0, |
| LOG_START = 1, |
| LOG_END = 2, |
| RTP_EVENT = 3, |
| RTCP_EVENT = 4, |
| AUDIO_PLAYOUT_EVENT = 5, |
| BWE_PACKET_LOSS_EVENT = 6, |
| BWE_PACKET_DELAY_EVENT = 7, |
| VIDEO_RECEIVER_CONFIG_EVENT = 8, |
| VIDEO_SENDER_CONFIG_EVENT = 9, |
| AUDIO_RECEIVER_CONFIG_EVENT = 10, |
| AUDIO_SENDER_CONFIG_EVENT = 11 |
| }; |
| |
| // Reads an RtcEventLog file and returns true if parsing was successful. |
| bool ParseFile(const std::string& file_name); |
| |
| // Reads an RtcEventLog from a string and returns true if successful. |
| bool ParseString(const std::string& s); |
| |
| // Reads an RtcEventLog from an istream and returns true if successful. |
| bool ParseStream(std::istream& stream); |
| |
| // Returns the number of events in an EventStream. |
| size_t GetNumberOfEvents() const; |
| |
| // Reads the arrival timestamp (in microseconds) from a rtclog::Event. |
| int64_t GetTimestamp(size_t index) const; |
| |
| // Reads the event type of the rtclog::Event at |index|. |
| EventType GetEventType(size_t index) const; |
| |
| // Reads the header, direction, media type, header length and packet length |
| // from the RTP event at |index|, and stores the values in the corresponding |
| // output parameters. The output parameters can be set to nullptr if those |
| // values aren't needed. |
| // NB: The header must have space for at least IP_PACKET_SIZE bytes. |
| void GetRtpHeader(size_t index, |
| PacketDirection* incoming, |
| MediaType* media_type, |
| uint8_t* header, |
| size_t* header_length, |
| size_t* total_length) const; |
| |
| // Reads packet, direction, media type and packet length from the RTCP event |
| // at |index|, and stores the values in the corresponding output parameters. |
| // The output parameters can be set to nullptr if those values aren't needed. |
| // NB: The packet must have space for at least IP_PACKET_SIZE bytes. |
| void GetRtcpPacket(size_t index, |
| PacketDirection* incoming, |
| MediaType* media_type, |
| uint8_t* packet, |
| size_t* length) const; |
| |
| // Reads a config event to a (non-NULL) VideoReceiveStream::Config struct. |
| // Only the fields that are stored in the protobuf will be written. |
| void GetVideoReceiveConfig(size_t index, |
| VideoReceiveStream::Config* config) const; |
| |
| // Reads a config event to a (non-NULL) VideoSendStream::Config struct. |
| // Only the fields that are stored in the protobuf will be written. |
| void GetVideoSendConfig(size_t index, VideoSendStream::Config* config) const; |
| |
| // Reads the SSRC from the audio playout event at |index|. The SSRC is stored |
| // in the output parameter ssrc. The output parameter can be set to nullptr |
| // and in that case the function only asserts that the event is well formed. |
| void GetAudioPlayout(size_t index, uint32_t* ssrc) const; |
| |
| // Reads bitrate, fraction loss (as defined in RFC 1889) and total number of |
| // expected packets from the BWE event at |index| and stores the values in |
| // the corresponding output parameters. The output parameters can be set to |
| // nullptr if those values aren't needed. |
| // NB: The packet must have space for at least IP_PACKET_SIZE bytes. |
| void GetBwePacketLossEvent(size_t index, |
| int32_t* bitrate, |
| uint8_t* fraction_loss, |
| int32_t* total_packets) const; |
| |
| private: |
| std::vector<rtclog::Event> events_; |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // WEBRTC_CALL_RTC_EVENT_LOG_PARSER_H_ |