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/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
/* vim:set ts=2 sw=2 sts=2 et cindent: */
/* This Source Code Form is subject to the terms of the Mozilla Public
* License, v. 2.0. If a copy of the MPL was not distributed with this file,
* You can obtain one at http://mozilla.org/MPL/2.0/. */
#ifndef MEDIA_WEBRTC_SIGNALING_GTEST_MOCKCONDUIT_H_
#define MEDIA_WEBRTC_SIGNALING_GTEST_MOCKCONDUIT_H_
#include "gmock/gmock.h"
#include "MediaConduitInterface.h"
#include "libwebrtcglue/FrameTransformer.h"
namespace webrtc {
std::ostream& operator<<(std::ostream& aStream,
const webrtc::Call::Stats& aObj) {
aStream << aObj.ToString(0);
return aStream;
}
} // namespace webrtc
namespace mozilla {
class MockConduit : public MediaSessionConduit {
public:
MockConduit() = default;
MOCK_CONST_METHOD0(type, Type());
MOCK_CONST_METHOD0(ActiveSendPayloadType, Maybe<int>());
MOCK_CONST_METHOD0(ActiveRecvPayloadType, Maybe<int>());
MOCK_METHOD1(SetTransportActive, void(bool));
MOCK_METHOD0(SenderRtpSendEvent, MediaEventSourceExc<MediaPacket>&());
MOCK_METHOD0(SenderRtcpSendEvent, MediaEventSourceExc<MediaPacket>&());
MOCK_METHOD0(ReceiverRtcpSendEvent, MediaEventSourceExc<MediaPacket>&());
MOCK_METHOD1(
ConnectReceiverRtpEvent,
void(MediaEventSourceExc<webrtc::RtpPacketReceived, webrtc::RTPHeader>&));
MOCK_METHOD1(ConnectReceiverRtcpEvent,
void(MediaEventSourceExc<MediaPacket>&));
MOCK_METHOD1(ConnectSenderRtcpEvent, void(MediaEventSourceExc<MediaPacket>&));
MOCK_CONST_METHOD0(LastRtcpReceived, Maybe<DOMHighResTimeStamp>());
MOCK_CONST_METHOD1(RtpSendBaseSeqFor, Maybe<uint16_t>(uint32_t));
MOCK_CONST_METHOD0(GetNow, DOMHighResTimeStamp());
MOCK_CONST_METHOD0(GetTimestampMaker, dom::RTCStatsTimestampMaker&());
MOCK_CONST_METHOD0(GetLocalSSRCs, Ssrcs());
MOCK_CONST_METHOD0(GetRemoteSSRC, Maybe<Ssrc>());
MOCK_METHOD1(UnsetRemoteSSRC, void(Ssrc));
MOCK_METHOD0(DisableSsrcChanges, void());
MOCK_CONST_METHOD1(HasCodecPluginID, bool(uint64_t));
MOCK_METHOD0(RtcpByeEvent, MediaEventSource<void>&());
MOCK_METHOD0(RtcpTimeoutEvent, MediaEventSource<void>&());
MOCK_METHOD0(RtpPacketEvent, MediaEventSource<void>&());
MOCK_METHOD3(SendRtp,
bool(const uint8_t*, size_t, const webrtc::PacketOptions&));
MOCK_METHOD2(SendSenderRtcp, bool(const uint8_t*, size_t));
MOCK_METHOD2(SendReceiverRtcp, bool(const uint8_t*, size_t));
MOCK_METHOD2(DeliverPacket, void(rtc::CopyOnWriteBuffer, PacketType));
MOCK_METHOD0(Shutdown, RefPtr<GenericPromise>());
MOCK_METHOD0(AsAudioSessionConduit, Maybe<RefPtr<AudioSessionConduit>>());
MOCK_METHOD0(AsVideoSessionConduit, Maybe<RefPtr<VideoSessionConduit>>());
MOCK_CONST_METHOD0(GetCallStats, Maybe<webrtc::Call::Stats>());
MOCK_METHOD1(SetJitterBufferTarget, void(DOMHighResTimeStamp));
MOCK_CONST_METHOD0(GetUpstreamRtpSources, std::vector<webrtc::RtpSource>());
};
} // namespace mozilla
#endif
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