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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" |
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123 const std::string& type, | 123 const std::string& type, |
124 const std::string& sdp, | 124 const std::string& sdp, |
125 webrtc::SdpParseError* error); | 125 webrtc::SdpParseError* error); |
126 | 126 |
127 // Creates a libjingle representation of an ice candidate. | 127 // Creates a libjingle representation of an ice candidate. |
128 virtual webrtc::IceCandidateInterface* CreateIceCandidate( | 128 virtual webrtc::IceCandidateInterface* CreateIceCandidate( |
129 const std::string& sdp_mid, | 129 const std::string& sdp_mid, |
130 int sdp_mline_index, | 130 int sdp_mline_index, |
131 const std::string& sdp); | 131 const std::string& sdp); |
132 | 132 |
133 // Starts recording an RTC event log. | |
134 virtual bool StartRtcEventLog(base::PlatformFile file); | |
135 | |
136 // Starts recording an RTC event log. | |
137 virtual void StopRtcEventLog(); | |
138 | |
139 WebRtcAudioDeviceImpl* GetWebRtcAudioDevice(); | 133 WebRtcAudioDeviceImpl* GetWebRtcAudioDevice(); |
140 | 134 |
141 void EnsureInitialized(); | 135 void EnsureInitialized(); |
142 scoped_refptr<base::SingleThreadTaskRunner> GetWebRtcWorkerThread() const; | 136 scoped_refptr<base::SingleThreadTaskRunner> GetWebRtcWorkerThread() const; |
143 scoped_refptr<base::SingleThreadTaskRunner> GetWebRtcSignalingThread() const; | 137 scoped_refptr<base::SingleThreadTaskRunner> GetWebRtcSignalingThread() const; |
144 | 138 |
145 protected: | 139 protected: |
146 // Asks the PeerConnection factory to create a Local Audio Source. | 140 // Asks the PeerConnection factory to create a Local Audio Source. |
147 virtual scoped_refptr<webrtc::AudioSourceInterface> CreateLocalAudioSource( | 141 virtual scoped_refptr<webrtc::AudioSourceInterface> CreateLocalAudioSource( |
148 const cricket::AudioOptions& options); | 142 const cricket::AudioOptions& options); |
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220 rtc::Thread* worker_thread_; | 214 rtc::Thread* worker_thread_; |
221 base::Thread chrome_signaling_thread_; | 215 base::Thread chrome_signaling_thread_; |
222 base::Thread chrome_worker_thread_; | 216 base::Thread chrome_worker_thread_; |
223 | 217 |
224 DISALLOW_COPY_AND_ASSIGN(PeerConnectionDependencyFactory); | 218 DISALLOW_COPY_AND_ASSIGN(PeerConnectionDependencyFactory); |
225 }; | 219 }; |
226 | 220 |
227 } // namespace content | 221 } // namespace content |
228 | 222 |
229 #endif // CONTENT_RENDERER_MEDIA_WEBRTC_PEER_CONNECTION_DEPENDENCY_FACTORY_H_ | 223 #endif // CONTENT_RENDERER_MEDIA_WEBRTC_PEER_CONNECTION_DEPENDENCY_FACTORY_H_ |
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