OLD | NEW |
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_ |
OLD | NEW |