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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> |
| 9 |
| 8 #include "base/macros.h" | 10 #include "base/macros.h" |
| 9 #include "base/memory/ref_counted.h" | 11 #include "base/memory/ref_counted.h" |
| 10 #include "base/memory/scoped_ptr.h" | |
| 11 #include "base/memory/scoped_vector.h" | 12 #include "base/memory/scoped_vector.h" |
| 12 #include "base/memory/weak_ptr.h" | 13 #include "base/memory/weak_ptr.h" |
| 13 #include "base/threading/thread_checker.h" | 14 #include "base/threading/thread_checker.h" |
| 14 #include "base/timer/timer.h" | 15 #include "base/timer/timer.h" |
| 15 #include "crypto/hmac.h" | 16 #include "crypto/hmac.h" |
| 16 #include "remoting/protocol/transport.h" | 17 #include "remoting/protocol/transport.h" |
| 17 #include "remoting/protocol/webrtc_data_stream_adapter.h" | 18 #include "remoting/protocol/webrtc_data_stream_adapter.h" |
| 18 #include "remoting/signaling/signal_strategy.h" | 19 #include "remoting/signaling/signal_strategy.h" |
| 19 #include "third_party/webrtc/api/peerconnectioninterface.h" | 20 #include "third_party/webrtc/api/peerconnectioninterface.h" |
| 20 | 21 |
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| 75 return &incoming_data_stream_adapter_; | 76 return &incoming_data_stream_adapter_; |
| 76 } | 77 } |
| 77 | 78 |
| 78 // Transport interface. | 79 // Transport interface. |
| 79 void Start(Authenticator* authenticator, | 80 void Start(Authenticator* authenticator, |
| 80 SendTransportInfoCallback send_transport_info_callback) override; | 81 SendTransportInfoCallback send_transport_info_callback) override; |
| 81 bool ProcessTransportInfo(buzz::XmlElement* transport_info) override; | 82 bool ProcessTransportInfo(buzz::XmlElement* transport_info) override; |
| 82 | 83 |
| 83 private: | 84 private: |
| 84 void OnLocalSessionDescriptionCreated( | 85 void OnLocalSessionDescriptionCreated( |
| 85 scoped_ptr<webrtc::SessionDescriptionInterface> description, | 86 std::unique_ptr<webrtc::SessionDescriptionInterface> description, |
| 86 const std::string& error); | 87 const std::string& error); |
| 87 void OnLocalDescriptionSet(bool success, const std::string& error); | 88 void OnLocalDescriptionSet(bool success, const std::string& error); |
| 88 void OnRemoteDescriptionSet(bool send_answer, | 89 void OnRemoteDescriptionSet(bool send_answer, |
| 89 bool success, | 90 bool success, |
| 90 const std::string& error); | 91 const std::string& error); |
| 91 | 92 |
| 92 // webrtc::PeerConnectionObserver interface. | 93 // webrtc::PeerConnectionObserver interface. |
| 93 void OnSignalingChange( | 94 void OnSignalingChange( |
| 94 webrtc::PeerConnectionInterface::SignalingState new_state) override; | 95 webrtc::PeerConnectionInterface::SignalingState new_state) override; |
| 95 void OnAddStream(webrtc::MediaStreamInterface* stream) override; | 96 void OnAddStream(webrtc::MediaStreamInterface* stream) override; |
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| 112 | 113 |
| 113 base::ThreadChecker thread_checker_; | 114 base::ThreadChecker thread_checker_; |
| 114 | 115 |
| 115 rtc::Thread* worker_thread_; | 116 rtc::Thread* worker_thread_; |
| 116 scoped_refptr<TransportContext> transport_context_; | 117 scoped_refptr<TransportContext> transport_context_; |
| 117 EventHandler* event_handler_ = nullptr; | 118 EventHandler* event_handler_ = nullptr; |
| 118 SendTransportInfoCallback send_transport_info_callback_; | 119 SendTransportInfoCallback send_transport_info_callback_; |
| 119 | 120 |
| 120 crypto::HMAC handshake_hmac_; | 121 crypto::HMAC handshake_hmac_; |
| 121 | 122 |
| 122 scoped_ptr<webrtc::FakeAudioDeviceModule> fake_audio_device_module_; | 123 std::unique_ptr<webrtc::FakeAudioDeviceModule> fake_audio_device_module_; |
| 123 | 124 |
| 124 rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> | 125 rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> |
| 125 peer_connection_factory_; | 126 peer_connection_factory_; |
| 126 rtc::scoped_refptr<webrtc::PeerConnectionInterface> peer_connection_; | 127 rtc::scoped_refptr<webrtc::PeerConnectionInterface> peer_connection_; |
| 127 | 128 |
| 128 bool negotiation_pending_ = false; | 129 bool negotiation_pending_ = false; |
| 129 | 130 |
| 130 bool connected_ = false; | 131 bool connected_ = false; |
| 131 | 132 |
| 132 scoped_ptr<buzz::XmlElement> pending_transport_info_message_; | 133 std::unique_ptr<buzz::XmlElement> pending_transport_info_message_; |
| 133 base::OneShotTimer transport_info_timer_; | 134 base::OneShotTimer transport_info_timer_; |
| 134 | 135 |
| 135 ScopedVector<webrtc::IceCandidateInterface> pending_incoming_candidates_; | 136 ScopedVector<webrtc::IceCandidateInterface> pending_incoming_candidates_; |
| 136 | 137 |
| 137 WebrtcDataStreamAdapter outgoing_data_stream_adapter_; | 138 WebrtcDataStreamAdapter outgoing_data_stream_adapter_; |
| 138 WebrtcDataStreamAdapter incoming_data_stream_adapter_; | 139 WebrtcDataStreamAdapter incoming_data_stream_adapter_; |
| 139 | 140 |
| 140 base::WeakPtrFactory<WebrtcTransport> weak_factory_; | 141 base::WeakPtrFactory<WebrtcTransport> weak_factory_; |
| 141 | 142 |
| 142 DISALLOW_COPY_AND_ASSIGN(WebrtcTransport); | 143 DISALLOW_COPY_AND_ASSIGN(WebrtcTransport); |
| 143 }; | 144 }; |
| 144 | 145 |
| 145 } // namespace protocol | 146 } // namespace protocol |
| 146 } // namespace remoting | 147 } // namespace remoting |
| 147 | 148 |
| 148 #endif // REMOTING_PROTOCOL_WEBRTC_TRANSPORT_H_ | 149 #endif // REMOTING_PROTOCOL_WEBRTC_TRANSPORT_H_ |
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