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/*
* 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_