Rename some variables and methods in RTC event log.

webrtc/logging/rtc_event_log/rtc_event_log.h

 /*
  *  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.
  */
 
@@ skipping 94 matching lines @@
   // 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,
+  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.
-  virtual void LogBwePacketDelayEvent(int32_t bitrate,
+  virtual void LogDelayBasedBweUpdate(int32_t bitrate_bps,
                                       BandwidthUsage detector_state) = 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
@@ skipping 24 matching lines @@
       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,
+  void LogLossBasedBweUpdate(int32_t bitrate_bps,
                              uint8_t fraction_loss,
                              int32_t total_packets) override {}
-  void LogBwePacketDelayEvent(int32_t bitrate,
+  void LogDelayBasedBweUpdate(int32_t bitrate_bps,
                               BandwidthUsage detector_state) override {}
   void LogAudioNetworkAdaptation(
       const AudioNetworkAdaptor::EncoderRuntimeConfig& config) override {}
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_H_