| Index: webrtc/call/rtc_event_log_parser.h
|
| diff --git a/webrtc/call/rtc_event_log_parser.h b/webrtc/call/rtc_event_log_parser.h
|
| deleted file mode 100644
|
| index a50ec20391de1b53bf76187b6f993f69c8cde9e7..0000000000000000000000000000000000000000
|
| --- a/webrtc/call/rtc_event_log_parser.h
|
| +++ /dev/null
|
| @@ -1,123 +0,0 @@
|
| -/*
|
| - * 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_
|
|
|