/* * 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 #include "testing/gmock/include/gmock/gmock.h" #include "webrtc/call/rtc_event_log.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 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_