| /* | 
 |  *  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(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_ |