| /* |
| * 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 AUDIO_MOCK_VOE_CHANNEL_PROXY_H_ |
| #define AUDIO_MOCK_VOE_CHANNEL_PROXY_H_ |
| |
| #include <map> |
| #include <memory> |
| #include <string> |
| #include <utility> |
| #include <vector> |
| |
| #include "api/test/mock_frame_encryptor.h" |
| #include "audio/channel_receive.h" |
| #include "audio/channel_send.h" |
| #include "modules/rtp_rtcp/source/rtp_packet_received.h" |
| #include "test/gmock.h" |
| |
| namespace webrtc { |
| namespace test { |
| |
| class MockChannelReceive : public voe::ChannelReceiveInterface { |
| public: |
| MOCK_METHOD2(SetNACKStatus, void(bool enable, int max_packets)); |
| MOCK_METHOD1(RegisterReceiverCongestionControlObjects, |
| void(PacketRouter* packet_router)); |
| MOCK_METHOD0(ResetReceiverCongestionControlObjects, void()); |
| MOCK_CONST_METHOD0(GetRTCPStatistics, CallReceiveStatistics()); |
| MOCK_CONST_METHOD0(GetNetworkStatistics, NetworkStatistics()); |
| MOCK_CONST_METHOD0(GetDecodingCallStatistics, AudioDecodingCallStats()); |
| MOCK_CONST_METHOD0(GetSpeechOutputLevelFullRange, int()); |
| MOCK_CONST_METHOD0(GetTotalOutputEnergy, double()); |
| MOCK_CONST_METHOD0(GetTotalOutputDuration, double()); |
| MOCK_CONST_METHOD0(GetDelayEstimate, uint32_t()); |
| MOCK_METHOD1(SetSink, void(AudioSinkInterface* sink)); |
| MOCK_METHOD1(OnRtpPacket, void(const RtpPacketReceived& packet)); |
| MOCK_METHOD2(ReceivedRTCPPacket, void(const uint8_t* packet, size_t length)); |
| MOCK_METHOD1(SetChannelOutputVolumeScaling, void(float scaling)); |
| MOCK_METHOD2(GetAudioFrameWithInfo, |
| AudioMixer::Source::AudioFrameInfo(int sample_rate_hz, |
| AudioFrame* audio_frame)); |
| MOCK_CONST_METHOD0(PreferredSampleRate, int()); |
| MOCK_METHOD1(SetAssociatedSendChannel, |
| void(const voe::ChannelSendInterface* send_channel)); |
| MOCK_CONST_METHOD2(GetPlayoutRtpTimestamp, |
| bool(uint32_t* rtp_timestamp, int64_t* time_ms)); |
| MOCK_METHOD2(SetEstimatedPlayoutNtpTimestampMs, |
| void(int64_t ntp_timestamp_ms, int64_t time_ms)); |
| MOCK_CONST_METHOD1(GetCurrentEstimatedPlayoutNtpTimestampMs, |
| absl::optional<int64_t>(int64_t now_ms)); |
| MOCK_CONST_METHOD0(GetSyncInfo, absl::optional<Syncable::Info>()); |
| MOCK_METHOD1(SetMinimumPlayoutDelay, void(int delay_ms)); |
| MOCK_METHOD1(SetBaseMinimumPlayoutDelayMs, bool(int delay_ms)); |
| MOCK_CONST_METHOD0(GetBaseMinimumPlayoutDelayMs, int()); |
| MOCK_CONST_METHOD0(GetReceiveCodec, |
| absl::optional<std::pair<int, SdpAudioFormat>>()); |
| MOCK_METHOD1(SetReceiveCodecs, |
| void(const std::map<int, SdpAudioFormat>& codecs)); |
| MOCK_CONST_METHOD0(GetSources, std::vector<RtpSource>()); |
| MOCK_METHOD0(StartPlayout, void()); |
| MOCK_METHOD0(StopPlayout, void()); |
| }; |
| |
| class MockChannelSend : public voe::ChannelSendInterface { |
| public: |
| // GMock doesn't like move-only types, like std::unique_ptr. |
| virtual void SetEncoder(int payload_type, |
| std::unique_ptr<AudioEncoder> encoder) { |
| return SetEncoderForMock(payload_type, &encoder); |
| } |
| MOCK_METHOD2(SetEncoderForMock, |
| void(int payload_type, std::unique_ptr<AudioEncoder>* encoder)); |
| MOCK_METHOD1( |
| ModifyEncoder, |
| void(rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)> modifier)); |
| MOCK_METHOD1(CallEncoder, |
| void(rtc::FunctionView<void(AudioEncoder*)> modifier)); |
| MOCK_METHOD1(SetRTCP_CNAME, void(absl::string_view c_name)); |
| MOCK_METHOD2(SetSendAudioLevelIndicationStatus, void(bool enable, int id)); |
| MOCK_METHOD2(RegisterSenderCongestionControlObjects, |
| void(RtpTransportControllerSendInterface* transport, |
| RtcpBandwidthObserver* bandwidth_observer)); |
| MOCK_METHOD0(ResetSenderCongestionControlObjects, void()); |
| MOCK_CONST_METHOD0(GetRTCPStatistics, CallSendStatistics()); |
| MOCK_CONST_METHOD0(GetRemoteRTCPReportBlocks, std::vector<ReportBlock>()); |
| MOCK_CONST_METHOD0(GetANAStatistics, ANAStats()); |
| MOCK_METHOD2(RegisterCngPayloadType, |
| void(int payload_type, int payload_frequency)); |
| MOCK_METHOD2(SetSendTelephoneEventPayloadType, |
| void(int payload_type, int payload_frequency)); |
| MOCK_METHOD2(SendTelephoneEventOutband, bool(int event, int duration_ms)); |
| MOCK_METHOD1(OnBitrateAllocation, void(BitrateAllocationUpdate update)); |
| MOCK_METHOD1(SetInputMute, void(bool muted)); |
| MOCK_METHOD2(ReceivedRTCPPacket, void(const uint8_t* packet, size_t length)); |
| // GMock doesn't like move-only types, like std::unique_ptr. |
| virtual void ProcessAndEncodeAudio(std::unique_ptr<AudioFrame> audio_frame) { |
| ProcessAndEncodeAudioForMock(&audio_frame); |
| } |
| MOCK_METHOD1(ProcessAndEncodeAudioForMock, |
| void(std::unique_ptr<AudioFrame>* audio_frame)); |
| MOCK_METHOD1(SetTransportOverhead, |
| void(size_t transport_overhead_per_packet)); |
| MOCK_CONST_METHOD0(GetRtpRtcp, RtpRtcp*()); |
| MOCK_CONST_METHOD0(GetBitrate, int()); |
| MOCK_METHOD1(OnTwccBasedUplinkPacketLossRate, void(float packet_loss_rate)); |
| MOCK_METHOD1(OnRecoverableUplinkPacketLossRate, |
| void(float recoverable_packet_loss_rate)); |
| MOCK_CONST_METHOD0(GetRTT, int64_t()); |
| MOCK_METHOD0(StartSend, void()); |
| MOCK_METHOD0(StopSend, void()); |
| MOCK_METHOD1( |
| SetFrameEncryptor, |
| void(rtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor)); |
| }; |
| } // namespace test |
| } // namespace webrtc |
| |
| #endif // AUDIO_MOCK_VOE_CHANNEL_PROXY_H_ |