Split LogRtpHeader and LogRtcpPacket into separate versions for incoming and outgoing packets.

webrtc/logging/rtc_event_log/mock/mock_rtc_event_log.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_LOGGING_RTC_EVENT_LOG_MOCK_MOCK_RTC_EVENT_LOG_H_
 #define WEBRTC_LOGGING_RTC_EVENT_LOG_MOCK_MOCK_RTC_EVENT_LOG_H_
 
 #include <string>
 
 #include "webrtc/logging/rtc_event_log/rtc_event_log.h"
 #include "webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h"
 #include "webrtc/test/gmock.h"
 
 namespace webrtc {
 
 class MockRtcEventLog : public RtcEventLog {
  public:
   MOCK_METHOD2(StartLogging,
                bool(const std::string& file_name, int64_t max_size_bytes));
 
   MOCK_METHOD2(StartLogging,
                bool(rtc::PlatformFile log_file, int64_t max_size_bytes));
 
   MOCK_METHOD0(StopLogging, void());
 
   MOCK_METHOD1(LogVideoReceiveStreamConfig,
                void(const rtclog::StreamConfig& config));
 
   MOCK_METHOD1(LogVideoSendStreamConfig,
                void(const rtclog::StreamConfig& config));
 
   MOCK_METHOD1(LogAudioReceiveStreamConfig,
                void(const rtclog::StreamConfig& config));
 
   MOCK_METHOD1(LogAudioSendStreamConfig,
                void(const rtclog::StreamConfig& config));
 
-  MOCK_METHOD3(LogRtpHeader,
-               void(PacketDirection direction,
-                    const uint8_t* header,
-                    size_t packet_length));
+  MOCK_METHOD1(LogIncomingRtpHeader, void(const RtpPacketReceived& packet));
 
-  MOCK_METHOD4(LogRtpHeader,
-               void(PacketDirection direction,
-                    const uint8_t* header,
-                    size_t packet_length,
-                    int probe_cluster_id));
+  MOCK_METHOD2(LogOutgoingRtpHeader,
+               void(const RtpPacketToSend& packet, int probe_cluster_id));
 
-  MOCK_METHOD3(LogRtcpPacket,
-               void(PacketDirection direction,
-                    const uint8_t* packet,
-                    size_t length));
+  MOCK_METHOD1(LogIncomingRtcpPacket,
+               void(rtc::ArrayView<const uint8_t> packet));
+
+  MOCK_METHOD1(LogOutgoingRtcpPacket,
+               void(rtc::ArrayView<const uint8_t> packet));
 
   MOCK_METHOD1(LogAudioPlayout, void(uint32_t ssrc));
 
   MOCK_METHOD3(LogLossBasedBweUpdate,
                void(int32_t bitrate_bps,
                     uint8_t fraction_loss,
                     int32_t total_packets));
 
   MOCK_METHOD2(LogDelayBasedBweUpdate,
                void(int32_t bitrate_bps, BandwidthUsage detector_state));
 
   MOCK_METHOD1(LogAudioNetworkAdaptation,
                void(const AudioEncoderRuntimeConfig& config));
 
   MOCK_METHOD4(LogProbeClusterCreated,
                void(int id, int bitrate_bps, int min_probes, int min_bytes));
 
   MOCK_METHOD2(LogProbeResultSuccess, void(int id, int bitrate_bps));
   MOCK_METHOD2(LogProbeResultFailure,
                void(int id, ProbeFailureReason failure_reason));
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_LOGGING_RTC_EVENT_LOG_MOCK_MOCK_RTC_EVENT_LOG_H_