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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 "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<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(uint8_t event,
uint32_t duration_ms));
};
} // namespace test
} // namespace webrtc
#endif // WEBRTC_TEST_MOCK_VOE_CHANNEL_PROXY_H_
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