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/*
* 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 <vector>
#include "audio/channel_proxy.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
#include "test/gmock.h"
namespace webrtc {
namespace test {
class MockVoEChannelProxy : public voe::ChannelProxy {
public:
// GMock doesn't like move-only types, like std::unique_ptr.
virtual 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_METHOD1(EnableSendTransportSequenceNumber, 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(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));
MOCK_METHOD1(SetSink, void(AudioSinkInterface* sink));
MOCK_METHOD1(SetInputMute, void(bool muted));
MOCK_METHOD1(RegisterTransport, void(Transport* transport));
MOCK_METHOD1(OnRtpPacket, void(const RtpPacketReceived& packet));
MOCK_METHOD2(ReceivedRTCPPacket, bool(const uint8_t* packet, size_t length));
MOCK_METHOD1(SetChannelOutputVolumeScaling, void(float scaling));
MOCK_METHOD1(SetRtcEventLog, void(RtcEventLog* event_log));
MOCK_METHOD2(GetAudioFrameWithInfo,
AudioMixer::Source::AudioFrameInfo(int sample_rate_hz,
AudioFrame* audio_frame));
MOCK_CONST_METHOD0(PreferredSampleRate, int());
// 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(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>());
MOCK_METHOD0(StartSend, void());
MOCK_METHOD0(StopSend, void());
MOCK_METHOD0(StartPlayout, void());
MOCK_METHOD0(StopPlayout, void());
};
} // namespace test
} // namespace webrtc
#endif // AUDIO_MOCK_VOE_CHANNEL_PROXY_H_
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