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1 // Copyright 2015 The Chromium Authors. All rights reserved. | 1 // Copyright 2015 The Chromium Authors. All rights reserved. |
2 // Use of this source code is governed by a BSD-style license that can be | 2 // Use of this source code is governed by a BSD-style license that can be |
3 // found in the LICENSE file. | 3 // found in the LICENSE file. |
4 | 4 |
5 #ifndef REMOTING_PROTOCOL_WEBRTC_TRANSPORT_H_ | 5 #ifndef REMOTING_PROTOCOL_WEBRTC_TRANSPORT_H_ |
6 #define REMOTING_PROTOCOL_WEBRTC_TRANSPORT_H_ | 6 #define REMOTING_PROTOCOL_WEBRTC_TRANSPORT_H_ |
7 | 7 |
8 #include <memory> | 8 #include <memory> |
9 #include <string> | 9 #include <string> |
10 | 10 |
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23 | 23 |
24 namespace webrtc { | 24 namespace webrtc { |
25 class FakeAudioDeviceModule; | 25 class FakeAudioDeviceModule; |
26 } // namespace webrtc | 26 } // namespace webrtc |
27 | 27 |
28 namespace remoting { | 28 namespace remoting { |
29 namespace protocol { | 29 namespace protocol { |
30 | 30 |
31 class TransportContext; | 31 class TransportContext; |
32 class MessageChannelFactory; | 32 class MessageChannelFactory; |
| 33 class MessagePipe; |
33 | 34 |
34 class WebrtcTransport : public Transport, | 35 class WebrtcTransport : public Transport, |
35 public webrtc::PeerConnectionObserver { | 36 public webrtc::PeerConnectionObserver { |
36 public: | 37 public: |
37 class EventHandler { | 38 class EventHandler { |
38 public: | 39 public: |
39 // Called after |peer_connection| has been created but before handshake. The | 40 // Called after |peer_connection| has been created but before handshake. The |
40 // handler should create data channels and media streams. Renegotiation will | 41 // handler should create data channels and media streams. Renegotiation will |
41 // be required in two cases after this method returns: | 42 // be required in two cases after this method returns: |
42 // 1. When the first data channel is created, if it wasn't created by this | 43 // 1. When the first data channel is created, if it wasn't created by this |
43 // event handler. | 44 // event handler. |
44 // 2. Whenever a media stream is added or removed. | 45 // 2. Whenever a media stream is added or removed. |
45 virtual void OnWebrtcTransportConnecting() = 0; | 46 virtual void OnWebrtcTransportConnecting() = 0; |
46 | 47 |
47 // Called when the transport is connected. | 48 // Called when the transport is connected. |
48 virtual void OnWebrtcTransportConnected() = 0; | 49 virtual void OnWebrtcTransportConnected() = 0; |
49 | 50 |
50 // Called when there is an error connecting the session. | 51 // Called when there is an error connecting the session. |
51 virtual void OnWebrtcTransportError(ErrorCode error) = 0; | 52 virtual void OnWebrtcTransportError(ErrorCode error) = 0; |
52 | 53 |
| 54 // Called when a new data channel is created by the peer. |
| 55 virtual void OnWebrtcTransportIncomingDataChannel( |
| 56 const std::string& name, |
| 57 std::unique_ptr<MessagePipe> pipe) = 0; |
| 58 |
53 // Called when an incoming media stream is added or removed. | 59 // Called when an incoming media stream is added or removed. |
54 virtual void OnWebrtcTransportMediaStreamAdded( | 60 virtual void OnWebrtcTransportMediaStreamAdded( |
55 scoped_refptr<webrtc::MediaStreamInterface> stream) = 0; | 61 scoped_refptr<webrtc::MediaStreamInterface> stream) = 0; |
56 virtual void OnWebrtcTransportMediaStreamRemoved( | 62 virtual void OnWebrtcTransportMediaStreamRemoved( |
57 scoped_refptr<webrtc::MediaStreamInterface> stream) = 0; | 63 scoped_refptr<webrtc::MediaStreamInterface> stream) = 0; |
| 64 |
| 65 protected: |
| 66 virtual ~EventHandler() {} |
58 }; | 67 }; |
59 | 68 |
60 WebrtcTransport(rtc::Thread* worker_thread, | 69 WebrtcTransport(rtc::Thread* worker_thread, |
61 scoped_refptr<TransportContext> transport_context, | 70 scoped_refptr<TransportContext> transport_context, |
62 EventHandler* event_handler); | 71 EventHandler* event_handler); |
63 ~WebrtcTransport() override; | 72 ~WebrtcTransport() override; |
64 | 73 |
65 webrtc::PeerConnectionInterface* peer_connection() { | 74 webrtc::PeerConnectionInterface* peer_connection() { |
66 return peer_connection_; | 75 return peer_connection_; |
67 } | 76 } |
68 webrtc::PeerConnectionFactoryInterface* peer_connection_factory() { | 77 webrtc::PeerConnectionFactoryInterface* peer_connection_factory() { |
69 return peer_connection_factory_; | 78 return peer_connection_factory_; |
70 } | 79 } |
71 remoting::WebrtcVideoEncoderFactory* video_encoder_factory() { | 80 remoting::WebrtcVideoEncoderFactory* video_encoder_factory() { |
72 return video_encoder_factory_; | 81 return video_encoder_factory_; |
73 } | 82 } |
74 | 83 |
75 // Factories for outgoing and incoming data channels. Must be used only after | 84 // Factory for outgoing data channels. Must be used only after the transport |
76 // the transport is connected. | 85 // is connected. |
77 MessageChannelFactory* outgoing_channel_factory() { | 86 MessageChannelFactory* outgoing_channel_factory() { |
78 return &outgoing_data_stream_adapter_; | 87 return data_stream_adapter_.get(); |
79 } | |
80 MessageChannelFactory* incoming_channel_factory() { | |
81 return &incoming_data_stream_adapter_; | |
82 } | 88 } |
83 | 89 |
84 // Transport interface. | 90 // Transport interface. |
85 void Start(Authenticator* authenticator, | 91 void Start(Authenticator* authenticator, |
86 SendTransportInfoCallback send_transport_info_callback) override; | 92 SendTransportInfoCallback send_transport_info_callback) override; |
87 bool ProcessTransportInfo(buzz::XmlElement* transport_info) override; | 93 bool ProcessTransportInfo(buzz::XmlElement* transport_info) override; |
88 void Close(ErrorCode error); | 94 void Close(ErrorCode error); |
89 | 95 |
90 private: | 96 private: |
91 void OnLocalSessionDescriptionCreated( | 97 void OnLocalSessionDescriptionCreated( |
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137 | 143 |
138 bool negotiation_pending_ = false; | 144 bool negotiation_pending_ = false; |
139 | 145 |
140 bool connected_ = false; | 146 bool connected_ = false; |
141 | 147 |
142 std::unique_ptr<buzz::XmlElement> pending_transport_info_message_; | 148 std::unique_ptr<buzz::XmlElement> pending_transport_info_message_; |
143 base::OneShotTimer transport_info_timer_; | 149 base::OneShotTimer transport_info_timer_; |
144 | 150 |
145 ScopedVector<webrtc::IceCandidateInterface> pending_incoming_candidates_; | 151 ScopedVector<webrtc::IceCandidateInterface> pending_incoming_candidates_; |
146 | 152 |
147 WebrtcDataStreamAdapter outgoing_data_stream_adapter_; | 153 std::unique_ptr<WebrtcDataStreamAdapter> data_stream_adapter_; |
148 WebrtcDataStreamAdapter incoming_data_stream_adapter_; | |
149 | 154 |
150 base::WeakPtrFactory<WebrtcTransport> weak_factory_; | 155 base::WeakPtrFactory<WebrtcTransport> weak_factory_; |
151 | 156 |
152 DISALLOW_COPY_AND_ASSIGN(WebrtcTransport); | 157 DISALLOW_COPY_AND_ASSIGN(WebrtcTransport); |
153 }; | 158 }; |
154 | 159 |
155 } // namespace protocol | 160 } // namespace protocol |
156 } // namespace remoting | 161 } // namespace remoting |
157 | 162 |
158 #endif // REMOTING_PROTOCOL_WEBRTC_TRANSPORT_H_ | 163 #endif // REMOTING_PROTOCOL_WEBRTC_TRANSPORT_H_ |
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