/* * 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. */ #ifndef WEBRTC_TEST_MOCK_VOE_CHANNEL_PROXY_H_ #define WEBRTC_TEST_MOCK_VOE_CHANNEL_PROXY_H_ #include #include "testing/gmock/include/gmock/gmock.h" #include "webrtc/voice_engine/channel_proxy.h" namespace webrtc { namespace test { class MockVoEChannelProxy : public voe::ChannelProxy { public: MOCK_METHOD1(SetRTCPStatus, void(bool enable)); MOCK_METHOD1(SetLocalSSRC, void(uint32_t ssrc)); MOCK_METHOD1(SetRTCP_CNAME, void(const std::string& c_name)); MOCK_METHOD2(SetSendAbsoluteSenderTimeStatus, void(bool enable, int id)); MOCK_METHOD2(SetSendAudioLevelIndicationStatus, void(bool enable, int id)); MOCK_METHOD1(EnableSendTransportSequenceNumber, void(int id)); MOCK_METHOD2(SetReceiveAbsoluteSenderTimeStatus, void(bool enable, int id)); MOCK_METHOD2(SetReceiveAudioLevelIndicationStatus, void(bool enable, int id)); MOCK_METHOD3(SetCongestionControlObjects, void(RtpPacketSender* rtp_packet_sender, TransportFeedbackObserver* transport_feedback_observer, PacketRouter* seq_num_allocator)); MOCK_CONST_METHOD0(GetRTCPStatistics, CallStatistics()); MOCK_CONST_METHOD0(GetRemoteRTCPReportBlocks, std::vector()); MOCK_CONST_METHOD0(GetNetworkStatistics, NetworkStatistics()); MOCK_CONST_METHOD0(GetDecodingCallStatistics, AudioDecodingCallStats()); MOCK_CONST_METHOD0(GetSpeechOutputLevelFullRange, int32_t()); MOCK_CONST_METHOD0(GetDelayEstimate, uint32_t()); MOCK_METHOD1(SetSendTelephoneEventPayloadType, bool(int payload_type)); MOCK_METHOD2(SendTelephoneEventOutband, bool(uint8_t event, uint32_t duration_ms)); }; } // namespace test } // namespace webrtc #endif // WEBRTC_TEST_MOCK_VOE_CHANNEL_PROXY_H_