3d7db263b9
VoENetwork is kept for now, but is not really used anylonger. webrtcvoiceengine is changed to have the same behavior for unsignaled ssrc as video has, which is reflected by disabling one test case and this will be discussed and followed up. BUG=webrtc:5079 TBR=tommi Review-Url: https://codereview.webrtc.org/1909333002 Cr-Commit-Position: refs/heads/master@{#12555}
58 lines
2.7 KiB
C++
58 lines
2.7 KiB
C++
/*
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* Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_TEST_MOCK_VOE_CHANNEL_PROXY_H_
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#define WEBRTC_TEST_MOCK_VOE_CHANNEL_PROXY_H_
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#include <string>
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#include "testing/gmock/include/gmock/gmock.h"
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#include "webrtc/voice_engine/channel_proxy.h"
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namespace webrtc {
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namespace test {
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class MockVoEChannelProxy : public voe::ChannelProxy {
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public:
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MOCK_METHOD1(SetRTCPStatus, void(bool enable));
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MOCK_METHOD1(SetLocalSSRC, void(uint32_t ssrc));
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MOCK_METHOD1(SetRTCP_CNAME, void(const std::string& c_name));
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MOCK_METHOD2(SetSendAbsoluteSenderTimeStatus, void(bool enable, int id));
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MOCK_METHOD2(SetSendAudioLevelIndicationStatus, void(bool enable, int id));
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MOCK_METHOD2(SetReceiveAbsoluteSenderTimeStatus, void(bool enable, int id));
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MOCK_METHOD2(SetReceiveAudioLevelIndicationStatus, void(bool enable, int id));
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MOCK_METHOD1(EnableSendTransportSequenceNumber, void(int id));
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MOCK_METHOD1(EnableReceiveTransportSequenceNumber, void(int id));
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MOCK_METHOD3(RegisterSenderCongestionControlObjects,
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void(RtpPacketSender* rtp_packet_sender,
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TransportFeedbackObserver* transport_feedback_observer,
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PacketRouter* packet_router));
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MOCK_METHOD1(RegisterReceiverCongestionControlObjects,
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void(PacketRouter* packet_router));
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MOCK_METHOD0(ResetCongestionControlObjects, void());
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MOCK_CONST_METHOD0(GetRTCPStatistics, CallStatistics());
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MOCK_CONST_METHOD0(GetRemoteRTCPReportBlocks, std::vector<ReportBlock>());
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MOCK_CONST_METHOD0(GetNetworkStatistics, NetworkStatistics());
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MOCK_CONST_METHOD0(GetDecodingCallStatistics, AudioDecodingCallStats());
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MOCK_CONST_METHOD0(GetSpeechOutputLevelFullRange, int32_t());
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MOCK_CONST_METHOD0(GetDelayEstimate, uint32_t());
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MOCK_METHOD1(SetSendTelephoneEventPayloadType, bool(int payload_type));
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MOCK_METHOD2(SendTelephoneEventOutband, bool(int event, int duration_ms));
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MOCK_METHOD1(RegisterExternalTransport, void(Transport* transport));
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MOCK_METHOD0(DeRegisterExternalTransport, void());
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MOCK_METHOD3(ReceivedRTPPacket, bool(const uint8_t* packet,
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size_t length,
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const PacketTime& packet_time));
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MOCK_METHOD2(ReceivedRTCPPacket, bool(const uint8_t* packet, size_t length));
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};
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} // namespace test
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} // namespace webrtc
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#endif // WEBRTC_TEST_MOCK_VOE_CHANNEL_PROXY_H_
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