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Issue 916873004: Add a Preference to allow WebRTC only bind to "any address" (all 0s) (Closed) Base URL: https://chromium.googlesource.com/chromium/src.git@master
Patch Set: Created 5 years, 10 months ago
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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"
(...skipping 109 matching lines...)
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.
(...skipping 57 matching lines...)
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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