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| 1 // Copyright 2014 The Chromium Authors. All rights reserved. | 1 // Copyright 2014 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 CONTENT_RENDERER_MEDIA_WEBRTC_PEER_CONNECTION_DEPENDENCY_FACTORY_H_ | 5 #ifndef CONTENT_RENDERER_MEDIA_WEBRTC_PEER_CONNECTION_DEPENDENCY_FACTORY_H_ |
| 6 #define CONTENT_RENDERER_MEDIA_WEBRTC_PEER_CONNECTION_DEPENDENCY_FACTORY_H_ | 6 #define CONTENT_RENDERER_MEDIA_WEBRTC_PEER_CONNECTION_DEPENDENCY_FACTORY_H_ |
| 7 | 7 |
| 8 #include <string> | 8 #include <string> |
| 9 | 9 |
| 10 #include "base/files/file.h" | 10 #include "base/files/file.h" |
| 11 #include "base/macros.h" | 11 #include "base/macros.h" |
| 12 #include "base/message_loop/message_loop.h" | |
| 13 #include "base/single_thread_task_runner.h" | 12 #include "base/single_thread_task_runner.h" |
| 14 #include "base/threading/thread.h" | 13 #include "base/threading/thread.h" |
| 15 #include "content/common/content_export.h" | 14 #include "content/common/content_export.h" |
| 16 #include "content/renderer/media/webrtc/stun_field_trial.h" | 15 #include "content/renderer/media/webrtc/stun_field_trial.h" |
| 17 #include "content/renderer/p2p/socket_dispatcher.h" | 16 #include "content/renderer/p2p/socket_dispatcher.h" |
| 18 #include "ipc/ipc_platform_file.h" | 17 #include "ipc/ipc_platform_file.h" |
| 19 #include "third_party/webrtc/api/peerconnectioninterface.h" | 18 #include "third_party/webrtc/api/peerconnectioninterface.h" |
| 20 #include "third_party/webrtc/p2p/stunprober/stunprober.h" | 19 #include "third_party/webrtc/p2p/stunprober/stunprober.h" |
| 21 | 20 |
| 22 namespace base { | 21 namespace base { |
| (...skipping 25 matching lines...) Expand all Loading... |
| 48 class IpcNetworkManager; | 47 class IpcNetworkManager; |
| 49 class IpcPacketSocketFactory; | 48 class IpcPacketSocketFactory; |
| 50 class WebRtcAudioDeviceImpl; | 49 class WebRtcAudioDeviceImpl; |
| 51 class WebRtcLoggingHandlerImpl; | 50 class WebRtcLoggingHandlerImpl; |
| 52 class WebRtcLoggingMessageFilter; | 51 class WebRtcLoggingMessageFilter; |
| 53 class WebRtcVideoCapturerAdapter; | 52 class WebRtcVideoCapturerAdapter; |
| 54 struct StreamDeviceInfo; | 53 struct StreamDeviceInfo; |
| 55 | 54 |
| 56 // Object factory for RTC PeerConnections. | 55 // Object factory for RTC PeerConnections. |
| 57 class CONTENT_EXPORT PeerConnectionDependencyFactory | 56 class CONTENT_EXPORT PeerConnectionDependencyFactory |
| 58 : NON_EXPORTED_BASE(base::MessageLoop::DestructionObserver), | 57 : NON_EXPORTED_BASE(public base::NonThreadSafe) { |
| 59 NON_EXPORTED_BASE(public base::NonThreadSafe) { | |
| 60 public: | 58 public: |
| 61 PeerConnectionDependencyFactory( | 59 PeerConnectionDependencyFactory( |
| 62 P2PSocketDispatcher* p2p_socket_dispatcher); | 60 P2PSocketDispatcher* p2p_socket_dispatcher); |
| 63 ~PeerConnectionDependencyFactory() override; | 61 virtual ~PeerConnectionDependencyFactory(); |
| 64 | 62 |
| 65 // Create a RTCPeerConnectionHandler object that implements the | 63 // Create a RTCPeerConnectionHandler object that implements the |
| 66 // WebKit WebRTCPeerConnectionHandler interface. | 64 // WebKit WebRTCPeerConnectionHandler interface. |
| 67 blink::WebRTCPeerConnectionHandler* CreateRTCPeerConnectionHandler( | 65 blink::WebRTCPeerConnectionHandler* CreateRTCPeerConnectionHandler( |
| 68 blink::WebRTCPeerConnectionHandlerClient* client); | 66 blink::WebRTCPeerConnectionHandlerClient* client); |
| 69 | 67 |
| 70 // Asks the PeerConnection factory to create a Local MediaStream object. | 68 // Asks the PeerConnection factory to create a Local MediaStream object. |
| 71 virtual scoped_refptr<webrtc::MediaStreamInterface> | 69 virtual scoped_refptr<webrtc::MediaStreamInterface> |
| 72 CreateLocalMediaStream(const std::string& label); | 70 CreateLocalMediaStream(const std::string& label); |
| 73 | 71 |
| (...skipping 33 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
| 107 const std::string& sdp_mid, | 105 const std::string& sdp_mid, |
| 108 int sdp_mline_index, | 106 int sdp_mline_index, |
| 109 const std::string& sdp); | 107 const std::string& sdp); |
| 110 | 108 |
| 111 WebRtcAudioDeviceImpl* GetWebRtcAudioDevice(); | 109 WebRtcAudioDeviceImpl* GetWebRtcAudioDevice(); |
| 112 | 110 |
| 113 void EnsureInitialized(); | 111 void EnsureInitialized(); |
| 114 scoped_refptr<base::SingleThreadTaskRunner> GetWebRtcWorkerThread() const; | 112 scoped_refptr<base::SingleThreadTaskRunner> GetWebRtcWorkerThread() const; |
| 115 virtual scoped_refptr<base::SingleThreadTaskRunner> GetWebRtcSignalingThread() | 113 virtual scoped_refptr<base::SingleThreadTaskRunner> GetWebRtcSignalingThread() |
| 116 const; | 114 const; |
| 115 void ShutDown(); |
| 117 | 116 |
| 118 protected: | 117 protected: |
| 119 // Asks the PeerConnection factory to create a Local VideoTrack object with | 118 // Asks the PeerConnection factory to create a Local VideoTrack object with |
| 120 // the video source using |capturer|. | 119 // the video source using |capturer|. |
| 121 virtual scoped_refptr<webrtc::VideoTrackInterface> | 120 virtual scoped_refptr<webrtc::VideoTrackInterface> |
| 122 CreateLocalVideoTrack(const std::string& id, | 121 CreateLocalVideoTrack(const std::string& id, |
| 123 cricket::VideoCapturer* capturer); | 122 cricket::VideoCapturer* capturer); |
| 124 | 123 |
| 125 virtual const scoped_refptr<webrtc::PeerConnectionFactoryInterface>& | 124 virtual const scoped_refptr<webrtc::PeerConnectionFactoryInterface>& |
| 126 GetPcFactory(); | 125 GetPcFactory(); |
| 127 virtual bool PeerConnectionFactoryCreated(); | 126 virtual bool PeerConnectionFactoryCreated(); |
| 128 | 127 |
| 129 // Helper method to create a WebRtcAudioDeviceImpl. | 128 // Helper method to create a WebRtcAudioDeviceImpl. |
| 130 void EnsureWebRtcAudioDeviceImpl(); | 129 void EnsureWebRtcAudioDeviceImpl(); |
| 131 | 130 |
| 132 private: | 131 private: |
| 133 // Implement base::MessageLoop::DestructionObserver. | |
| 134 // This makes sure the libjingle PeerConnectionFactory is released before | |
| 135 // the renderer message loop is destroyed. | |
| 136 void WillDestroyCurrentMessageLoop() override; | |
| 137 | |
| 138 // Functions related to Stun probing trial to determine how fast we could send | 132 // Functions related to Stun probing trial to determine how fast we could send |
| 139 // Stun request without being dropped by NAT. | 133 // Stun request without being dropped by NAT. |
| 140 void TryScheduleStunProbeTrial(); | 134 void TryScheduleStunProbeTrial(); |
| 141 void StartStunProbeTrialOnWorkerThread(const std::string& params); | 135 void StartStunProbeTrialOnWorkerThread(const std::string& params); |
| 142 | 136 |
| 143 // Creates |pc_factory_|, which in turn is used for | 137 // Creates |pc_factory_|, which in turn is used for |
| 144 // creating PeerConnection objects. | 138 // creating PeerConnection objects. |
| 145 void CreatePeerConnectionFactory(); | 139 void CreatePeerConnectionFactory(); |
| 146 | 140 |
| 147 void InitializeSignalingThread( | 141 void InitializeSignalingThread( |
| (...skipping 25 matching lines...) Expand all Loading... |
| 173 rtc::Thread* worker_thread_; | 167 rtc::Thread* worker_thread_; |
| 174 base::Thread chrome_signaling_thread_; | 168 base::Thread chrome_signaling_thread_; |
| 175 base::Thread chrome_worker_thread_; | 169 base::Thread chrome_worker_thread_; |
| 176 | 170 |
| 177 DISALLOW_COPY_AND_ASSIGN(PeerConnectionDependencyFactory); | 171 DISALLOW_COPY_AND_ASSIGN(PeerConnectionDependencyFactory); |
| 178 }; | 172 }; |
| 179 | 173 |
| 180 } // namespace content | 174 } // namespace content |
| 181 | 175 |
| 182 #endif // CONTENT_RENDERER_MEDIA_WEBRTC_PEER_CONNECTION_DEPENDENCY_FACTORY_H_ | 176 #endif // CONTENT_RENDERER_MEDIA_WEBRTC_PEER_CONNECTION_DEPENDENCY_FACTORY_H_ |
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