Fixes a bug which incorrectly logs incoming RTCP as outgoing.

webrtc/call/mock/mock_rtc_event_log.h

Index: webrtc/call/mock/mock_rtc_event_log.h
diff --git a/webrtc/call/mock/mock_rtc_event_log.h b/webrtc/call/mock/mock_rtc_event_log.h
new file mode 100644
index 0000000000000000000000000000000000000000..f523105d0e40d666c033b0f24b8dc4953f608b17
--- /dev/null
+++ b/webrtc/call/mock/mock_rtc_event_log.h
@@ -0,0 +1,61 @@
+/*
+ *  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_MOCK_MOCK_RTC_EVENT_LOG_H_
+#define WEBRTC_CALL_MOCK_MOCK_RTC_EVENT_LOG_H_
+
+#include <string>
+
+#include "testing/gmock/include/gmock/gmock.h"
+
+#include "webrtc/call/rtc_event_log.h"
+
+namespace webrtc {
+
+class MockRtcEventLog : public RtcEventLog {
+ public:
+  MOCK_METHOD1(SetBufferDuration, void(int64_t buffer_duration_us));
+
+  MOCK_METHOD2(StartLogging,
+               void(const std::string& file_name, int duration_ms));
+
+  MOCK_METHOD1(StartLogging, bool(rtc::PlatformFile log_file));
+
+  MOCK_METHOD0(StopLogging, void());
+
+  MOCK_METHOD1(LogVideoReceiveStreamConfig,
+               void(const webrtc::VideoReceiveStream::Config& config));
+
+  MOCK_METHOD1(LogVideoSendStreamConfig,
+               void(const webrtc::VideoSendStream::Config& config));
+
+  MOCK_METHOD4(LogRtpHeader,
+               void(PacketDirection direction,
+                    MediaType media_type,
+                    const uint8_t* header,
+                    size_t packet_length));
+
+  MOCK_METHOD4(LogRtcpPacket,
+               void(PacketDirection direction,
+                    MediaType media_type,
+                    const uint8_t* packet,
+                    size_t length));
+
+  MOCK_METHOD1(LogAudioPlayout, void(uint32_t ssrc));
+
+  MOCK_METHOD3(LogBwePacketLossEvent,
+               void(int32_t bitrate,
+                    uint8_t fraction_loss,
+                    int32_t total_packets));
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_CALL_MOCK_MOCK_RTC_EVENT_LOG_H_