/* -*- 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