| /* |
| * 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 <string> |
| #include "webrtc/test/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(SetNACKStatus, void(bool enable, int max_packets)); |
| MOCK_METHOD2(SetSendAbsoluteSenderTimeStatus, void(bool enable, int id)); |
| MOCK_METHOD2(SetSendAudioLevelIndicationStatus, void(bool enable, int id)); |
| MOCK_METHOD2(SetReceiveAbsoluteSenderTimeStatus, void(bool enable, int id)); |
| MOCK_METHOD2(SetReceiveAudioLevelIndicationStatus, void(bool enable, int id)); |
| MOCK_METHOD1(EnableSendTransportSequenceNumber, void(int id)); |
| MOCK_METHOD1(EnableReceiveTransportSequenceNumber, void(int id)); |
| MOCK_METHOD3(RegisterSenderCongestionControlObjects, |
| void(RtpPacketSender* rtp_packet_sender, |
| TransportFeedbackObserver* transport_feedback_observer, |
| PacketRouter* packet_router)); |
| MOCK_METHOD1(RegisterReceiverCongestionControlObjects, |
| void(PacketRouter* packet_router)); |
| MOCK_METHOD0(ResetCongestionControlObjects, void()); |
| MOCK_CONST_METHOD0(GetRTCPStatistics, CallStatistics()); |
| MOCK_CONST_METHOD0(GetRemoteRTCPReportBlocks, std::vector<ReportBlock>()); |
| 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(int event, int duration_ms)); |
| MOCK_METHOD1(SetInputMute, void(bool muted)); |
| // TODO(solenberg): Talk the compiler into accepting this mock method: |
| // MOCK_METHOD1(SetSink, void(std::unique_ptr<AudioSinkInterface> sink)); |
| MOCK_METHOD1(RegisterExternalTransport, void(Transport* transport)); |
| MOCK_METHOD0(DeRegisterExternalTransport, void()); |
| MOCK_METHOD3(ReceivedRTPPacket, bool(const uint8_t* packet, |
| size_t length, |
| const PacketTime& packet_time)); |
| MOCK_METHOD2(ReceivedRTCPPacket, bool(const uint8_t* packet, size_t length)); |
| MOCK_CONST_METHOD0(GetAudioDecoderFactory, |
| const rtc::scoped_refptr<AudioDecoderFactory>&()); |
| MOCK_METHOD1(SetChannelOutputVolumeScaling, void(float scaling)); |
| MOCK_METHOD1(SetRtcEventLog, void(RtcEventLog* event_log)); |
| MOCK_METHOD1(SetBitrate, void(int bitrate_bps)); |
| }; |
| } // namespace test |
| } // namespace webrtc |
| |
| #endif // WEBRTC_TEST_MOCK_VOE_CHANNEL_PROXY_H_ |