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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" |
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120 virtual webrtc::IceCandidateInterface* CreateIceCandidate( | 120 virtual webrtc::IceCandidateInterface* CreateIceCandidate( |
121 const std::string& sdp_mid, | 121 const std::string& sdp_mid, |
122 int sdp_mline_index, | 122 int sdp_mline_index, |
123 const std::string& sdp); | 123 const std::string& sdp); |
124 | 124 |
125 WebRtcAudioDeviceImpl* GetWebRtcAudioDevice(); | 125 WebRtcAudioDeviceImpl* GetWebRtcAudioDevice(); |
126 | 126 |
127 scoped_refptr<base::MessageLoopProxy> GetWebRtcWorkerThread() const; | 127 scoped_refptr<base::MessageLoopProxy> GetWebRtcWorkerThread() const; |
128 scoped_refptr<base::MessageLoopProxy> GetWebRtcSignalingThread() const; | 128 scoped_refptr<base::MessageLoopProxy> GetWebRtcSignalingThread() const; |
129 | 129 |
| 130 void set_disable_multiple_routes(bool disable_multiple_routes) { |
| 131 disable_multiple_routes_ = disable_multiple_routes; |
| 132 } |
| 133 |
130 protected: | 134 protected: |
131 // Asks the PeerConnection factory to create a Local Audio Source. | 135 // Asks the PeerConnection factory to create a Local Audio Source. |
132 virtual scoped_refptr<webrtc::AudioSourceInterface> | 136 virtual scoped_refptr<webrtc::AudioSourceInterface> |
133 CreateLocalAudioSource( | 137 CreateLocalAudioSource( |
134 const webrtc::MediaConstraintsInterface* constraints); | 138 const webrtc::MediaConstraintsInterface* constraints); |
135 | 139 |
136 // Creates a media::AudioCapturerSource with an implementation that is | 140 // Creates a media::AudioCapturerSource with an implementation that is |
137 // specific for a WebAudio source. The created WebAudioCapturerSource | 141 // specific for a WebAudio source. The created WebAudioCapturerSource |
138 // instance will function as audio source instead of the default | 142 // instance will function as audio source instead of the default |
139 // WebRtcAudioCapturer. | 143 // WebRtcAudioCapturer. |
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197 scoped_refptr<P2PSocketDispatcher> p2p_socket_dispatcher_; | 201 scoped_refptr<P2PSocketDispatcher> p2p_socket_dispatcher_; |
198 scoped_refptr<WebRtcAudioDeviceImpl> audio_device_; | 202 scoped_refptr<WebRtcAudioDeviceImpl> audio_device_; |
199 | 203 |
200 // PeerConnection threads. signaling_thread_ is created from the | 204 // PeerConnection threads. signaling_thread_ is created from the |
201 // "current" chrome thread. | 205 // "current" chrome thread. |
202 rtc::Thread* signaling_thread_; | 206 rtc::Thread* signaling_thread_; |
203 rtc::Thread* worker_thread_; | 207 rtc::Thread* worker_thread_; |
204 base::Thread chrome_signaling_thread_; | 208 base::Thread chrome_signaling_thread_; |
205 base::Thread chrome_worker_thread_; | 209 base::Thread chrome_worker_thread_; |
206 | 210 |
| 211 bool disable_multiple_routes_; |
| 212 |
207 DISALLOW_COPY_AND_ASSIGN(PeerConnectionDependencyFactory); | 213 DISALLOW_COPY_AND_ASSIGN(PeerConnectionDependencyFactory); |
208 }; | 214 }; |
209 | 215 |
210 } // namespace content | 216 } // namespace content |
211 | 217 |
212 #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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