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30b4839d10
Analogous to https://webrtc-review.googlesource.com/c/src/+/92398, for RtpVideoStreamReceiver. Bug: webrtc:7135 Change-Id: I0639f9982da2ed80edbcf900cf14f8ae982ef80c Reviewed-on: https://webrtc-review.googlesource.com/93820 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24309}
102 lines
4.7 KiB
C++
102 lines
4.7 KiB
C++
/*
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* Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef AUDIO_MOCK_VOE_CHANNEL_PROXY_H_
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#define AUDIO_MOCK_VOE_CHANNEL_PROXY_H_
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#include <map>
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#include <memory>
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#include <string>
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#include <vector>
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#include "audio/channel_proxy.h"
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#include "modules/rtp_rtcp/source/rtp_packet_received.h"
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#include "test/gmock.h"
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namespace webrtc {
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namespace test {
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class MockVoEChannelProxy : public voe::ChannelProxy {
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public:
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// GMock doesn't like move-only types, like std::unique_ptr.
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virtual bool SetEncoder(int payload_type,
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std::unique_ptr<AudioEncoder> encoder) {
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return SetEncoderForMock(payload_type, &encoder);
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}
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MOCK_METHOD2(SetEncoderForMock,
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bool(int payload_type, std::unique_ptr<AudioEncoder>* encoder));
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MOCK_METHOD1(
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ModifyEncoder,
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void(rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)> modifier));
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MOCK_METHOD1(SetRTCPStatus, void(bool enable));
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MOCK_METHOD1(SetLocalSSRC, void(uint32_t ssrc));
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MOCK_METHOD1(SetRTCP_CNAME, void(const std::string& c_name));
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MOCK_METHOD2(SetNACKStatus, void(bool enable, int max_packets));
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MOCK_METHOD2(SetSendAudioLevelIndicationStatus, void(bool enable, int id));
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MOCK_METHOD1(EnableSendTransportSequenceNumber, void(int id));
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MOCK_METHOD2(RegisterSenderCongestionControlObjects,
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void(RtpTransportControllerSendInterface* transport,
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RtcpBandwidthObserver* bandwidth_observer));
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MOCK_METHOD1(RegisterReceiverCongestionControlObjects,
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void(PacketRouter* packet_router));
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MOCK_METHOD0(ResetSenderCongestionControlObjects, void());
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MOCK_METHOD0(ResetReceiverCongestionControlObjects, void());
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MOCK_CONST_METHOD0(GetRTCPStatistics, CallStatistics());
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MOCK_CONST_METHOD0(GetRemoteRTCPReportBlocks, std::vector<ReportBlock>());
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MOCK_CONST_METHOD0(GetNetworkStatistics, NetworkStatistics());
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MOCK_CONST_METHOD0(GetDecodingCallStatistics, AudioDecodingCallStats());
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MOCK_CONST_METHOD0(GetANAStatistics, ANAStats());
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MOCK_CONST_METHOD0(GetSpeechOutputLevelFullRange, int());
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MOCK_CONST_METHOD0(GetTotalOutputEnergy, double());
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MOCK_CONST_METHOD0(GetTotalOutputDuration, double());
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MOCK_CONST_METHOD0(GetDelayEstimate, uint32_t());
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MOCK_METHOD2(SetSendTelephoneEventPayloadType,
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bool(int payload_type, int payload_frequency));
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MOCK_METHOD2(SendTelephoneEventOutband, bool(int event, int duration_ms));
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MOCK_METHOD2(SetBitrate, void(int bitrate_bps, int64_t probing_interval_ms));
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MOCK_METHOD1(SetSink, void(AudioSinkInterface* sink));
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MOCK_METHOD1(SetInputMute, void(bool muted));
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MOCK_METHOD1(RegisterTransport, void(Transport* transport));
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MOCK_METHOD1(OnRtpPacket, void(const RtpPacketReceived& packet));
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MOCK_METHOD2(ReceivedRTCPPacket, bool(const uint8_t* packet, size_t length));
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MOCK_METHOD1(SetChannelOutputVolumeScaling, void(float scaling));
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MOCK_METHOD2(GetAudioFrameWithInfo,
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AudioMixer::Source::AudioFrameInfo(int sample_rate_hz,
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AudioFrame* audio_frame));
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MOCK_CONST_METHOD0(PreferredSampleRate, int());
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// GMock doesn't like move-only types, like std::unique_ptr.
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virtual void ProcessAndEncodeAudio(std::unique_ptr<AudioFrame> audio_frame) {
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ProcessAndEncodeAudioForMock(&audio_frame);
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}
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MOCK_METHOD1(ProcessAndEncodeAudioForMock,
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void(std::unique_ptr<AudioFrame>* audio_frame));
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MOCK_METHOD1(SetTransportOverhead, void(int transport_overhead_per_packet));
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MOCK_METHOD1(AssociateSendChannel,
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void(const ChannelProxy& send_channel_proxy));
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MOCK_METHOD0(DisassociateSendChannel, void());
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MOCK_CONST_METHOD0(GetRtpRtcp, RtpRtcp*());
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MOCK_CONST_METHOD0(GetPlayoutTimestamp, uint32_t());
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MOCK_METHOD1(SetMinimumPlayoutDelay, void(int delay_ms));
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MOCK_CONST_METHOD1(GetRecCodec, bool(CodecInst* codec_inst));
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MOCK_METHOD1(SetReceiveCodecs,
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void(const std::map<int, SdpAudioFormat>& codecs));
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MOCK_METHOD1(OnTwccBasedUplinkPacketLossRate, void(float packet_loss_rate));
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MOCK_METHOD1(OnRecoverableUplinkPacketLossRate,
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void(float recoverable_packet_loss_rate));
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MOCK_CONST_METHOD0(GetSources, std::vector<RtpSource>());
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MOCK_METHOD0(StartSend, void());
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MOCK_METHOD0(StopSend, void());
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MOCK_METHOD0(StartPlayout, void());
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MOCK_METHOD0(StopPlayout, void());
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};
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} // namespace test
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} // namespace webrtc
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#endif // AUDIO_MOCK_VOE_CHANNEL_PROXY_H_
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