| /* |
| * 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/modules/rtp_rtcp/source/rtp_packet_received.h" |
| #include "webrtc/test/gmock.h" |
| #include "webrtc/voice_engine/channel_proxy.h" |
| |
| namespace webrtc { |
| namespace test { |
| |
| class MockVoEChannelProxy : public voe::ChannelProxy { |
| public: |
| // GTest doesn't like move-only types, like std::unique_ptr |
| bool SetEncoder(int payload_type, |
| std::unique_ptr<AudioEncoder> encoder) { |
| return SetEncoderForMock(payload_type, &encoder); |
| } |
| MOCK_METHOD2(SetEncoderForMock, |
| bool(int payload_type, |
| std::unique_ptr<AudioEncoder>* encoder)); |
| MOCK_METHOD1( |
| ModifyEncoder, |
| void(rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)> modifier)); |
| 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(SetSendAudioLevelIndicationStatus, 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_METHOD2(RegisterSenderCongestionControlObjects, |
| void(RtpTransportControllerSendInterface* transport, |
| RtcpBandwidthObserver* bandwidth_observer)); |
| MOCK_METHOD1(RegisterReceiverCongestionControlObjects, |
| void(PacketRouter* packet_router)); |
| MOCK_METHOD0(ResetSenderCongestionControlObjects, void()); |
| MOCK_METHOD0(ResetReceiverCongestionControlObjects, 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(GetANAStatistics, ANAStats()); |
| MOCK_CONST_METHOD0(GetSpeechOutputLevel, int()); |
| MOCK_CONST_METHOD0(GetSpeechOutputLevelFullRange, int()); |
| MOCK_CONST_METHOD0(GetTotalOutputEnergy, double()); |
| MOCK_CONST_METHOD0(GetTotalOutputDuration, double()); |
| MOCK_CONST_METHOD0(GetDelayEstimate, uint32_t()); |
| MOCK_METHOD2(SetSendTelephoneEventPayloadType, bool(int payload_type, |
| int payload_frequency)); |
| MOCK_METHOD2(SendTelephoneEventOutband, bool(int event, int duration_ms)); |
| MOCK_METHOD2(SetBitrate, void(int bitrate_bps, int64_t probing_interval_ms)); |
| // TODO(solenberg): Talk the compiler into accepting this mock method: |
| // MOCK_METHOD1(SetSink, void(std::unique_ptr<AudioSinkInterface> sink)); |
| MOCK_METHOD1(SetInputMute, void(bool muted)); |
| MOCK_METHOD1(RegisterExternalTransport, void(Transport* transport)); |
| MOCK_METHOD0(DeRegisterExternalTransport, void()); |
| MOCK_METHOD1(OnRtpPacket, void(const RtpPacketReceived& packet)); |
| 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(SetRtcpRttStats, void(RtcpRttStats* rtcp_rtt_stats)); |
| MOCK_METHOD2(GetAudioFrameWithInfo, |
| AudioMixer::Source::AudioFrameInfo(int sample_rate_hz, |
| AudioFrame* audio_frame)); |
| MOCK_CONST_METHOD0(NeededFrequency, int()); |
| MOCK_METHOD1(SetTransportOverhead, void(int transport_overhead_per_packet)); |
| MOCK_METHOD1(AssociateSendChannel, |
| void(const ChannelProxy& send_channel_proxy)); |
| MOCK_METHOD0(DisassociateSendChannel, void()); |
| MOCK_CONST_METHOD2(GetRtpRtcp, void(RtpRtcp** rtp_rtcp, |
| RtpReceiver** rtp_receiver)); |
| MOCK_CONST_METHOD0(GetPlayoutTimestamp, uint32_t()); |
| MOCK_METHOD1(SetMinimumPlayoutDelay, void(int delay_ms)); |
| MOCK_CONST_METHOD1(GetRecCodec, bool(CodecInst* codec_inst)); |
| MOCK_METHOD1(SetReceiveCodecs, |
| void(const std::map<int, SdpAudioFormat>& codecs)); |
| MOCK_METHOD1(OnTwccBasedUplinkPacketLossRate, void(float packet_loss_rate)); |
| MOCK_METHOD1(OnRecoverableUplinkPacketLossRate, |
| void(float recoverable_packet_loss_rate)); |
| MOCK_CONST_METHOD0(GetSources, std::vector<RtpSource>()); |
| }; |
| } // namespace test |
| } // namespace webrtc |
| |
| #endif // WEBRTC_TEST_MOCK_VOE_CHANNEL_PROXY_H_ |