Moved RtcEventLog files from call/ to logging/ (new top level dir)

webrtc/call/rtc_event_log_parser.h

-/*
- *  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_