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Diffstat (limited to 'webrtc/test/mock_voe_channel_proxy.h')
-rw-r--r-- | webrtc/test/mock_voe_channel_proxy.h | 48 |
1 files changed, 48 insertions, 0 deletions
diff --git a/webrtc/test/mock_voe_channel_proxy.h b/webrtc/test/mock_voe_channel_proxy.h new file mode 100644 index 0000000000..b5d79c18ea --- /dev/null +++ b/webrtc/test/mock_voe_channel_proxy.h @@ -0,0 +1,48 @@ +/* + * 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_ |