1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
222
223
224
225
226
227
228
229
|
/*
* Copyright (C) 2017 Apple Inc.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
* are met:
* 1. Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
* 2. Redistributions in binary form must reproduce the above copyright
* notice, this list of conditions and the following disclaimer in the
* documentation and/or other materials provided with the distribution.
*
* THIS SOFTWARE IS PROVIDED BY APPLE INC. AND ITS CONTRIBUTORS ``AS IS'' AND ANY
* EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
* WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
* DISCLAIMED. IN NO EVENT SHALL APPLE INC. OR ITS CONTRIBUTORS BE LIABLE FOR ANY
* DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
* (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
* LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON
* ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
* (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
* SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#pragma once
#if USE(LIBWEBRTC)
#include "LibWebRTCMacros.h"
#include <webrtc/api/mediastreaminterface.h>
#include <webrtc/api/peerconnectioninterface.h>
#include <wtf/text/WTFString.h>
namespace WebCore {
class LibWebRTCProvider;
void useMockRTCPeerConnectionFactory(LibWebRTCProvider*, const String&);
class MockLibWebRTCPeerConnection : public webrtc::PeerConnectionInterface {
public:
virtual ~MockLibWebRTCPeerConnection() { }
protected:
explicit MockLibWebRTCPeerConnection(webrtc::PeerConnectionObserver& observer) : m_observer(observer) { }
private:
rtc::scoped_refptr<webrtc::StreamCollectionInterface> local_streams() override { return nullptr; }
rtc::scoped_refptr<webrtc::StreamCollectionInterface> remote_streams() override { return nullptr; }
rtc::scoped_refptr<webrtc::DtmfSenderInterface> CreateDtmfSender(webrtc::AudioTrackInterface*) override { return nullptr; }
const webrtc::SessionDescriptionInterface* local_description() const override { return nullptr; }
const webrtc::SessionDescriptionInterface* remote_description() const override { return nullptr; }
bool AddIceCandidate(const webrtc::IceCandidateInterface*) override { return true; }
void RegisterUMAObserver(webrtc::UMAObserver*) override { }
SignalingState signaling_state() override { return kStable; }
IceConnectionState ice_connection_state() override { return kIceConnectionNew; }
IceGatheringState ice_gathering_state() override { return kIceGatheringNew; }
void StopRtcEventLog() override { }
void Close() override { }
protected:
void SetRemoteDescription(webrtc::SetSessionDescriptionObserver*, webrtc::SessionDescriptionInterface*) final;
void CreateAnswer(webrtc::CreateSessionDescriptionObserver*, const webrtc::MediaConstraintsInterface*) final;
rtc::scoped_refptr<webrtc::DataChannelInterface> CreateDataChannel(const std::string&, const webrtc::DataChannelInit*) final;
bool AddStream(webrtc::MediaStreamInterface*) final;
void RemoveStream(webrtc::MediaStreamInterface*) final;
void SetLocalDescription(webrtc::SetSessionDescriptionObserver*, webrtc::SessionDescriptionInterface*) override;
bool GetStats(webrtc::StatsObserver*, webrtc::MediaStreamTrackInterface*, StatsOutputLevel) override { return false; }
void CreateOffer(webrtc::CreateSessionDescriptionObserver*, const webrtc::MediaConstraintsInterface*) override;
virtual void gotLocalDescription() { }
webrtc::PeerConnectionObserver& m_observer;
unsigned m_counter { 0 };
rtc::scoped_refptr<webrtc::MediaStreamInterface> m_stream;
bool m_isInitiator { true };
bool m_isReceivingAudio { false };
bool m_isReceivingVideo { false };
};
class MockLibWebRTCSessionDescription: public webrtc::SessionDescriptionInterface {
public:
explicit MockLibWebRTCSessionDescription(std::string&& sdp) : m_sdp(WTFMove(sdp)) { }
private:
bool ToString(std::string* out) const final { *out = m_sdp; return true; }
cricket::SessionDescription* description() final { return nullptr; }
const cricket::SessionDescription* description() const final { return nullptr; }
std::string session_id() const final { return ""; }
std::string session_version() const final { return ""; }
std::string type() const final { return ""; }
bool AddCandidate(const webrtc::IceCandidateInterface*) final { return true; }
size_t number_of_mediasections() const final { return 0; }
const webrtc::IceCandidateCollection* candidates(size_t) const final { return nullptr; }
std::string m_sdp;
};
class MockLibWebRTCIceCandidate : public webrtc::IceCandidateInterface {
public:
MockLibWebRTCIceCandidate(const char* sdp, const char* sdpMid)
: m_sdp(sdp)
, m_sdpMid(sdpMid) { }
private:
std::string sdp_mid() const final { return m_sdpMid; }
int sdp_mline_index() const final { return 0; }
const cricket::Candidate& candidate() const final { return m_candidate; }
bool ToString(std::string* out) const final { *out = m_sdp; return true; }
protected:
const char* m_sdp;
const char* m_sdpMid;
cricket::Candidate m_candidate;
};
class MockLibWebRTCAudioTrack : public webrtc::AudioTrackInterface {
public:
explicit MockLibWebRTCAudioTrack(const std::string& id, webrtc::AudioSourceInterface* source)
: m_id(id)
, m_source(source) { }
private:
webrtc::AudioSourceInterface* GetSource() const final { return m_source; }
void AddSink(webrtc::AudioTrackSinkInterface*) final { }
void RemoveSink(webrtc::AudioTrackSinkInterface*) final { }
void RegisterObserver(webrtc::ObserverInterface*) final { }
void UnregisterObserver(webrtc::ObserverInterface*) final { }
std::string kind() const final { return "audio"; }
std::string id() const final { return m_id; }
bool enabled() const final { return m_enabled; }
TrackState state() const final { return kLive; }
bool set_enabled(bool enabled) final { m_enabled = enabled; return true; }
bool m_enabled;
std::string m_id;
webrtc::AudioSourceInterface* m_source { nullptr };
};
class MockLibWebRTCVideoTrack : public webrtc::VideoTrackInterface {
public:
explicit MockLibWebRTCVideoTrack(const std::string& id, webrtc::VideoTrackSourceInterface* source)
: m_id(id)
, m_source(source) { }
private:
webrtc::VideoTrackSourceInterface* GetSource() const final { return m_source; }
void RegisterObserver(webrtc::ObserverInterface*) final { }
void UnregisterObserver(webrtc::ObserverInterface*) final { }
std::string kind() const final { return "video"; }
std::string id() const final { return m_id; }
bool enabled() const final { return m_enabled; }
TrackState state() const final { return kLive; }
bool set_enabled(bool enabled) final { m_enabled = enabled; return true; }
bool m_enabled;
std::string m_id;
webrtc::VideoTrackSourceInterface* m_source { nullptr };
};
class MockLibWebRTCDataChannel : public webrtc::DataChannelInterface {
public:
MockLibWebRTCDataChannel(std::string&& label, bool ordered, bool reliable, int id)
: m_label(WTFMove(label))
, m_ordered(ordered)
, m_reliable(reliable)
, m_id(id) { }
private:
void RegisterObserver(webrtc::DataChannelObserver*) final { }
void UnregisterObserver() final { }
std::string label() const final { return m_label; }
bool reliable() const final { return m_reliable; }
bool ordered() const final { return m_ordered; }
int id() const final { return m_id; }
DataState state() const final { return kConnecting; }
uint64_t buffered_amount() const final { return 0; }
void Close() final { }
bool Send(const webrtc::DataBuffer&) final { return true; }
std::string m_label;
bool m_ordered { true };
bool m_reliable { false };
int m_id { -1 };
};
class MockLibWebRTCPeerConnectionFactory : public webrtc::PeerConnectionFactoryInterface {
public:
static rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> create(String&& testCase) { return new rtc::RefCountedObject<MockLibWebRTCPeerConnectionFactory>(WTFMove(testCase)); }
protected:
MockLibWebRTCPeerConnectionFactory(String&&);
private:
rtc::scoped_refptr<webrtc::PeerConnectionInterface> CreatePeerConnection(const webrtc::PeerConnectionInterface::RTCConfiguration&, const webrtc::MediaConstraintsInterface*, std::unique_ptr<cricket::PortAllocator>, std::unique_ptr<rtc::RTCCertificateGeneratorInterface>, webrtc::PeerConnectionObserver*) final { return nullptr; }
rtc::scoped_refptr<webrtc::PeerConnectionInterface> CreatePeerConnection(const webrtc::PeerConnectionInterface::RTCConfiguration&, std::unique_ptr<cricket::PortAllocator>, std::unique_ptr<rtc::RTCCertificateGeneratorInterface>, webrtc::PeerConnectionObserver*) final;
rtc::scoped_refptr<webrtc::MediaStreamInterface> CreateLocalMediaStream(const std::string&) final;
void SetOptions(const Options&) final { }
rtc::scoped_refptr<webrtc::AudioSourceInterface> CreateAudioSource(const cricket::AudioOptions&) final { return nullptr; }
rtc::scoped_refptr<webrtc::AudioSourceInterface> CreateAudioSource(const webrtc::MediaConstraintsInterface*) final { return nullptr; }
rtc::scoped_refptr<webrtc::VideoTrackSourceInterface> CreateVideoSource(cricket::VideoCapturer*) final { return nullptr; }
rtc::scoped_refptr<webrtc::VideoTrackSourceInterface> CreateVideoSource(cricket::VideoCapturer*, const webrtc::MediaConstraintsInterface*) final { return nullptr; }
rtc::scoped_refptr<webrtc::VideoTrackInterface> CreateVideoTrack(const std::string& id, webrtc::VideoTrackSourceInterface* source) final { return new rtc::RefCountedObject<MockLibWebRTCVideoTrack>(id, source); }
rtc::scoped_refptr<webrtc::AudioTrackInterface> CreateAudioTrack(const std::string& id, webrtc::AudioSourceInterface* source) final { return new rtc::RefCountedObject<MockLibWebRTCAudioTrack>(id, source); }
bool StartAecDump(rtc::PlatformFile, int64_t) final { return false; }
void StopAecDump() final { }
bool StartRtcEventLog(rtc::PlatformFile, int64_t) final { return false; }
bool StartRtcEventLog(rtc::PlatformFile) final { return false; }
void StopRtcEventLog() final { }
private:
String m_testCase;
unsigned m_numberOfRealPeerConnections { 0 };
};
} // namespace WebCore
#endif // USE(LIBWEBRTC)
|
