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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/basictypes.h" | 10 #include "base/basictypes.h" |
| 11 #include "base/files/file.h" | 11 #include "base/files/file.h" |
| 12 #include "base/threading/thread.h" | 12 #include "base/threading/thread.h" |
| 13 #include "content/common/content_export.h" | 13 #include "content/common/content_export.h" |
| 14 #include "content/public/renderer/render_process_observer.h" | 14 #include "content/public/renderer/render_process_observer.h" |
| 15 #include "content/renderer/media/aec_dump_message_filter.h" |
| 15 #include "content/renderer/p2p/socket_dispatcher.h" | 16 #include "content/renderer/p2p/socket_dispatcher.h" |
| 16 #include "ipc/ipc_platform_file.h" | 17 #include "ipc/ipc_platform_file.h" |
| 17 #include "third_party/libjingle/source/talk/app/webrtc/peerconnectioninterface.h
" | 18 #include "third_party/libjingle/source/talk/app/webrtc/peerconnectioninterface.h
" |
| 18 #include "third_party/libjingle/source/talk/app/webrtc/videosourceinterface.h" | 19 #include "third_party/libjingle/source/talk/app/webrtc/videosourceinterface.h" |
| 19 | 20 |
| 20 namespace base { | 21 namespace base { |
| 21 class WaitableEvent; | 22 class WaitableEvent; |
| 22 } | 23 } |
| 23 | 24 |
| 24 namespace talk_base { | 25 namespace talk_base { |
| (...skipping 23 matching lines...) Expand all Loading... |
| 48 class WebRtcAudioDeviceImpl; | 49 class WebRtcAudioDeviceImpl; |
| 49 class WebRtcLocalAudioTrack; | 50 class WebRtcLocalAudioTrack; |
| 50 class WebRtcLoggingHandlerImpl; | 51 class WebRtcLoggingHandlerImpl; |
| 51 class WebRtcLoggingMessageFilter; | 52 class WebRtcLoggingMessageFilter; |
| 52 class WebRtcVideoCapturerAdapter; | 53 class WebRtcVideoCapturerAdapter; |
| 53 struct StreamDeviceInfo; | 54 struct StreamDeviceInfo; |
| 54 | 55 |
| 55 // Object factory for RTC PeerConnections. | 56 // Object factory for RTC PeerConnections. |
| 56 class CONTENT_EXPORT PeerConnectionDependencyFactory | 57 class CONTENT_EXPORT PeerConnectionDependencyFactory |
| 57 : NON_EXPORTED_BASE(public base::NonThreadSafe), | 58 : NON_EXPORTED_BASE(public base::NonThreadSafe), |
| 58 public RenderProcessObserver { | 59 NON_EXPORTED_BASE(public AecDumpMessageFilter::AecDumpDelegate) { |
| 59 public: | 60 public: |
| 60 PeerConnectionDependencyFactory( | 61 PeerConnectionDependencyFactory( |
| 61 P2PSocketDispatcher* p2p_socket_dispatcher); | 62 P2PSocketDispatcher* p2p_socket_dispatcher); |
| 62 virtual ~PeerConnectionDependencyFactory(); | 63 virtual ~PeerConnectionDependencyFactory(); |
| 63 | 64 |
| 64 // Create a RTCPeerConnectionHandler object that implements the | 65 // Create a RTCPeerConnectionHandler object that implements the |
| 65 // WebKit WebRTCPeerConnectionHandler interface. | 66 // WebKit WebRTCPeerConnectionHandler interface. |
| 66 blink::WebRTCPeerConnectionHandler* CreateRTCPeerConnectionHandler( | 67 blink::WebRTCPeerConnectionHandler* CreateRTCPeerConnectionHandler( |
| 67 blink::WebRTCPeerConnectionHandlerClient* client); | 68 blink::WebRTCPeerConnectionHandlerClient* client); |
| 68 | 69 |
| (...skipping 53 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
| 122 | 123 |
| 123 WebRtcAudioDeviceImpl* GetWebRtcAudioDevice(); | 124 WebRtcAudioDeviceImpl* GetWebRtcAudioDevice(); |
| 124 | 125 |
| 125 static void AddNativeAudioTrackToBlinkTrack( | 126 static void AddNativeAudioTrackToBlinkTrack( |
| 126 webrtc::MediaStreamTrackInterface* native_track, | 127 webrtc::MediaStreamTrackInterface* native_track, |
| 127 const blink::WebMediaStreamTrack& webkit_track, | 128 const blink::WebMediaStreamTrack& webkit_track, |
| 128 bool is_local_track); | 129 bool is_local_track); |
| 129 | 130 |
| 130 scoped_refptr<base::MessageLoopProxy> GetWebRtcWorkerThread() const; | 131 scoped_refptr<base::MessageLoopProxy> GetWebRtcWorkerThread() const; |
| 131 | 132 |
| 133 // AecDumpMessageFilter::AecDumpDelegate implementation. |
| 134 // TODO(xians): Remove when option to disable audio track processing is |
| 135 // removed. |
| 136 virtual void OnAecDumpFile( |
| 137 const IPC::PlatformFileForTransit& file_handle) OVERRIDE; |
| 138 virtual void OnDisableAecDump() OVERRIDE; |
| 139 virtual void OnIpcClosing() OVERRIDE; |
| 140 |
| 132 protected: | 141 protected: |
| 133 // Asks the PeerConnection factory to create a Local Audio Source. | 142 // Asks the PeerConnection factory to create a Local Audio Source. |
| 134 virtual scoped_refptr<webrtc::AudioSourceInterface> | 143 virtual scoped_refptr<webrtc::AudioSourceInterface> |
| 135 CreateLocalAudioSource( | 144 CreateLocalAudioSource( |
| 136 const webrtc::MediaConstraintsInterface* constraints); | 145 const webrtc::MediaConstraintsInterface* constraints); |
| 137 | 146 |
| 138 // Creates a media::AudioCapturerSource with an implementation that is | 147 // Creates a media::AudioCapturerSource with an implementation that is |
| 139 // specific for a WebAudio source. The created WebAudioCapturerSource | 148 // specific for a WebAudio source. The created WebAudioCapturerSource |
| 140 // instance will function as audio source instead of the default | 149 // instance will function as audio source instead of the default |
| 141 // WebRtcAudioCapturer. | 150 // WebRtcAudioCapturer. |
| (...skipping 28 matching lines...) Expand all Loading... |
| 170 // creating PeerConnection objects. | 179 // creating PeerConnection objects. |
| 171 void CreatePeerConnectionFactory(); | 180 void CreatePeerConnectionFactory(); |
| 172 | 181 |
| 173 void InitializeWorkerThread(talk_base::Thread** thread, | 182 void InitializeWorkerThread(talk_base::Thread** thread, |
| 174 base::WaitableEvent* event); | 183 base::WaitableEvent* event); |
| 175 | 184 |
| 176 void CreateIpcNetworkManagerOnWorkerThread(base::WaitableEvent* event); | 185 void CreateIpcNetworkManagerOnWorkerThread(base::WaitableEvent* event); |
| 177 void DeleteIpcNetworkManager(); | 186 void DeleteIpcNetworkManager(); |
| 178 void CleanupPeerConnectionFactory(); | 187 void CleanupPeerConnectionFactory(); |
| 179 | 188 |
| 180 // RenderProcessObserver implementation. | |
| 181 virtual bool OnControlMessageReceived(const IPC::Message& message) OVERRIDE; | |
| 182 | |
| 183 void OnAecDumpFile(IPC::PlatformFileForTransit file_handle); | |
| 184 void OnDisableAecDump(); | |
| 185 | |
| 186 void StartAecDump(base::File aec_dump_file); | |
| 187 | |
| 188 // Helper method to create a WebRtcAudioDeviceImpl. | 189 // Helper method to create a WebRtcAudioDeviceImpl. |
| 189 void EnsureWebRtcAudioDeviceImpl(); | 190 void EnsureWebRtcAudioDeviceImpl(); |
| 190 | 191 |
| 191 // We own network_manager_, must be deleted on the worker thread. | 192 // We own network_manager_, must be deleted on the worker thread. |
| 192 // The network manager uses |p2p_socket_dispatcher_|. | 193 // The network manager uses |p2p_socket_dispatcher_|. |
| 193 IpcNetworkManager* network_manager_; | 194 IpcNetworkManager* network_manager_; |
| 194 scoped_ptr<IpcPacketSocketFactory> socket_factory_; | 195 scoped_ptr<IpcPacketSocketFactory> socket_factory_; |
| 195 | 196 |
| 196 scoped_refptr<webrtc::PeerConnectionFactoryInterface> pc_factory_; | 197 scoped_refptr<webrtc::PeerConnectionFactoryInterface> pc_factory_; |
| 197 | 198 |
| 198 scoped_refptr<P2PSocketDispatcher> p2p_socket_dispatcher_; | 199 scoped_refptr<P2PSocketDispatcher> p2p_socket_dispatcher_; |
| 199 scoped_refptr<WebRtcAudioDeviceImpl> audio_device_; | 200 scoped_refptr<WebRtcAudioDeviceImpl> audio_device_; |
| 200 | 201 |
| 202 // This is only used if audio track processing is disabled. |
| 203 // TODO(xians): Remove when option to disable audio track processing is |
| 204 // removed. |
| 205 scoped_refptr<AecDumpMessageFilter> aec_dump_message_filter_; |
| 206 |
| 201 // PeerConnection threads. signaling_thread_ is created from the | 207 // PeerConnection threads. signaling_thread_ is created from the |
| 202 // "current" chrome thread. | 208 // "current" chrome thread. |
| 203 talk_base::Thread* signaling_thread_; | 209 talk_base::Thread* signaling_thread_; |
| 204 talk_base::Thread* worker_thread_; | 210 talk_base::Thread* worker_thread_; |
| 205 base::Thread chrome_worker_thread_; | 211 base::Thread chrome_worker_thread_; |
| 206 | 212 |
| 207 base::File aec_dump_file_; | |
| 208 | |
| 209 DISALLOW_COPY_AND_ASSIGN(PeerConnectionDependencyFactory); | 213 DISALLOW_COPY_AND_ASSIGN(PeerConnectionDependencyFactory); |
| 210 }; | 214 }; |
| 211 | 215 |
| 212 } // namespace content | 216 } // namespace content |
| 213 | 217 |
| 214 #endif // CONTENT_RENDERER_MEDIA_WEBRTC_PEER_CONNECTION_DEPENDENCY_FACTORY_H_ | 218 #endif // CONTENT_RENDERER_MEDIA_WEBRTC_PEER_CONNECTION_DEPENDENCY_FACTORY_H_ |
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