blob: f3d2cefa52bd7911a426dc0a2505c29684d00355 (
plain)
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
|
// Copyright 2015 The Chromium Authors. All rights reserved.
// Use of this source code is governed by a BSD-style license that can be
// found in the LICENSE file.
#ifndef REMOTING_PROTOCOL_WEBRTC_TRANSPORT_H_
#define REMOTING_PROTOCOL_WEBRTC_TRANSPORT_H_
#include "base/macros.h"
#include "base/memory/ref_counted.h"
#include "base/memory/scoped_ptr.h"
#include "base/memory/scoped_vector.h"
#include "base/memory/weak_ptr.h"
#include "base/threading/thread_checker.h"
#include "base/timer/timer.h"
#include "crypto/hmac.h"
#include "remoting/protocol/transport.h"
#include "remoting/protocol/webrtc_data_stream_adapter.h"
#include "remoting/signaling/signal_strategy.h"
#include "third_party/webrtc/api/peerconnectioninterface.h"
namespace webrtc {
class FakeAudioDeviceModule;
} // namespace webrtc
namespace remoting {
namespace protocol {
class TransportContext;
class MessageChannelFactory;
class WebrtcTransport : public Transport,
public webrtc::PeerConnectionObserver {
public:
class EventHandler {
public:
// Called after |peer_connection| has been created but before handshake. The
// handler should create data channels and media streams. Renegotiation will
// be required in two cases after this method returns:
// 1. When the first data channel is created, if it wasn't created by this
// event handler.
// 2. Whenever a media stream is added or removed.
virtual void OnWebrtcTransportConnecting() = 0;
// Called when the transport is connected.
virtual void OnWebrtcTransportConnected() = 0;
// Called when there is an error connecting the session.
virtual void OnWebrtcTransportError(ErrorCode error) = 0;
// Called when an incoming media stream is added or removed.
virtual void OnWebrtcTransportMediaStreamAdded(
scoped_refptr<webrtc::MediaStreamInterface> stream) = 0;
virtual void OnWebrtcTransportMediaStreamRemoved(
scoped_refptr<webrtc::MediaStreamInterface> stream) = 0;
};
WebrtcTransport(rtc::Thread* worker_thread,
scoped_refptr<TransportContext> transport_context,
EventHandler* event_handler);
~WebrtcTransport() override;
webrtc::PeerConnectionInterface* peer_connection() {
return peer_connection_;
}
webrtc::PeerConnectionFactoryInterface* peer_connection_factory() {
return peer_connection_factory_;
}
// Factories for outgoing and incoming data channels. Must be used only after
// the transport is connected.
MessageChannelFactory* outgoing_channel_factory() {
return &outgoing_data_stream_adapter_;
}
MessageChannelFactory* incoming_channel_factory() {
return &incoming_data_stream_adapter_;
}
// Transport interface.
void Start(Authenticator* authenticator,
SendTransportInfoCallback send_transport_info_callback) override;
bool ProcessTransportInfo(buzz::XmlElement* transport_info) override;
private:
void OnLocalSessionDescriptionCreated(
scoped_ptr<webrtc::SessionDescriptionInterface> description,
const std::string& error);
void OnLocalDescriptionSet(bool success, const std::string& error);
void OnRemoteDescriptionSet(bool send_answer,
bool success,
const std::string& error);
// webrtc::PeerConnectionObserver interface.
void OnSignalingChange(
webrtc::PeerConnectionInterface::SignalingState new_state) override;
void OnAddStream(webrtc::MediaStreamInterface* stream) override;
void OnRemoveStream(webrtc::MediaStreamInterface* stream) override;
void OnDataChannel(webrtc::DataChannelInterface* data_channel) override;
void OnRenegotiationNeeded() override;
void OnIceConnectionChange(
webrtc::PeerConnectionInterface::IceConnectionState new_state) override;
void OnIceGatheringChange(
webrtc::PeerConnectionInterface::IceGatheringState new_state) override;
void OnIceCandidate(const webrtc::IceCandidateInterface* candidate) override;
void RequestNegotiation();
void SendOffer();
void EnsurePendingTransportInfoMessage();
void SendTransportInfo();
void AddPendingCandidatesIfPossible();
void Close(ErrorCode error);
base::ThreadChecker thread_checker_;
rtc::Thread* worker_thread_;
scoped_refptr<TransportContext> transport_context_;
EventHandler* event_handler_ = nullptr;
SendTransportInfoCallback send_transport_info_callback_;
crypto::HMAC handshake_hmac_;
scoped_ptr<webrtc::FakeAudioDeviceModule> fake_audio_device_module_;
rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface>
peer_connection_factory_;
rtc::scoped_refptr<webrtc::PeerConnectionInterface> peer_connection_;
bool negotiation_pending_ = false;
bool connected_ = false;
scoped_ptr<buzz::XmlElement> pending_transport_info_message_;
base::OneShotTimer transport_info_timer_;
ScopedVector<webrtc::IceCandidateInterface> pending_incoming_candidates_;
WebrtcDataStreamAdapter outgoing_data_stream_adapter_;
WebrtcDataStreamAdapter incoming_data_stream_adapter_;
base::WeakPtrFactory<WebrtcTransport> weak_factory_;
DISALLOW_COPY_AND_ASSIGN(WebrtcTransport);
};
} // namespace protocol
} // namespace remoting
#endif // REMOTING_PROTOCOL_WEBRTC_TRANSPORT_H_
|
