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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 "base/macros.h" | 8 #include "base/macros.h" |
9 #include "base/memory/ref_counted.h" | 9 #include "base/memory/ref_counted.h" |
10 #include "base/memory/scoped_ptr.h" | 10 #include "base/memory/scoped_ptr.h" |
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38 // 1. When the first data channel is created, if it wasn't created by this | 38 // 1. When the first data channel is created, if it wasn't created by this |
39 // event handler. | 39 // event handler. |
40 // 2. Whenever a media stream is added or removed. | 40 // 2. Whenever a media stream is added or removed. |
41 virtual void OnWebrtcTransportConnecting() = 0; | 41 virtual void OnWebrtcTransportConnecting() = 0; |
42 | 42 |
43 // Called when the transport is connected. | 43 // Called when the transport is connected. |
44 virtual void OnWebrtcTransportConnected() = 0; | 44 virtual void OnWebrtcTransportConnected() = 0; |
45 | 45 |
46 // Called when there is an error connecting the session. | 46 // Called when there is an error connecting the session. |
47 virtual void OnWebrtcTransportError(ErrorCode error) = 0; | 47 virtual void OnWebrtcTransportError(ErrorCode error) = 0; |
| 48 |
| 49 // Called when an incoming media stream is added or removed. |
| 50 virtual void OnWebrtcTransportMediaStreamAdded( |
| 51 scoped_refptr<webrtc::MediaStreamInterface> stream) = 0; |
| 52 virtual void OnWebrtcTransportMediaStreamRemoved( |
| 53 scoped_refptr<webrtc::MediaStreamInterface> stream) = 0; |
48 }; | 54 }; |
49 | 55 |
50 WebrtcTransport(rtc::Thread* worker_thread, | 56 WebrtcTransport(rtc::Thread* worker_thread, |
51 scoped_refptr<TransportContext> transport_context, | 57 scoped_refptr<TransportContext> transport_context, |
52 EventHandler* event_handler); | 58 EventHandler* event_handler); |
53 ~WebrtcTransport() override; | 59 ~WebrtcTransport() override; |
54 | 60 |
55 webrtc::PeerConnectionInterface* peer_connection() { | 61 webrtc::PeerConnectionInterface* peer_connection() { |
56 return peer_connection_; | 62 return peer_connection_; |
57 } | 63 } |
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118 | 124 |
119 bool negotiation_pending_ = false; | 125 bool negotiation_pending_ = false; |
120 | 126 |
121 bool connected_ = false; | 127 bool connected_ = false; |
122 | 128 |
123 scoped_ptr<buzz::XmlElement> pending_transport_info_message_; | 129 scoped_ptr<buzz::XmlElement> pending_transport_info_message_; |
124 base::OneShotTimer transport_info_timer_; | 130 base::OneShotTimer transport_info_timer_; |
125 | 131 |
126 ScopedVector<webrtc::IceCandidateInterface> pending_incoming_candidates_; | 132 ScopedVector<webrtc::IceCandidateInterface> pending_incoming_candidates_; |
127 | 133 |
128 std::list<rtc::scoped_refptr<webrtc::MediaStreamInterface>> | |
129 unclaimed_streams_; | |
130 | |
131 WebrtcDataStreamAdapter outgoing_data_stream_adapter_; | 134 WebrtcDataStreamAdapter outgoing_data_stream_adapter_; |
132 WebrtcDataStreamAdapter incoming_data_stream_adapter_; | 135 WebrtcDataStreamAdapter incoming_data_stream_adapter_; |
133 | 136 |
134 base::WeakPtrFactory<WebrtcTransport> weak_factory_; | 137 base::WeakPtrFactory<WebrtcTransport> weak_factory_; |
135 | 138 |
136 DISALLOW_COPY_AND_ASSIGN(WebrtcTransport); | 139 DISALLOW_COPY_AND_ASSIGN(WebrtcTransport); |
137 }; | 140 }; |
138 | 141 |
139 } // namespace protocol | 142 } // namespace protocol |
140 } // namespace remoting | 143 } // namespace remoting |
141 | 144 |
142 #endif // REMOTING_PROTOCOL_WEBRTC_TRANSPORT_H_ | 145 #endif // REMOTING_PROTOCOL_WEBRTC_TRANSPORT_H_ |
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