Fixes a bug which incorrectly logs incoming RTCP as outgoing.

webrtc/call/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 12 matching lines @@
 // Forward declaration of storage class that is automatically generated from
 // the protobuf file.
 namespace rtclog {
 class EventStream;
 }  // namespace rtclog
 
 class RtcEventLogImpl;
 
 enum class MediaType;
 
+enum PacketDirection { kIncomingPacket = 0, kOutgoingPacket };
+
 class RtcEventLog {
  public:
   virtual ~RtcEventLog() {}
 
   static rtc::scoped_ptr<RtcEventLog> Create();
 
   // Sets the time that events are stored in the internal event buffer
   // before the user calls StartLogging.  The default is 10 000 000 us = 10 s
   virtual void SetBufferDuration(int64_t buffer_duration_us) = 0;
 
@@ skipping 13 matching lines @@
   // 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 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(bool incoming,
+  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(bool incoming,
+  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;
 
   // Reads an RtcEventLog file and returns true when reading was successful.
   // The result is stored in the given EventStream object.
   static bool ParseRtcEventLog(const std::string& file_name,
                                rtclog::EventStream* result);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_CALL_RTC_EVENT_LOG_H_