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Side by Side Diff: content/renderer/media/webrtc/peer_connection_dependency_factory.h

Issue 1855193002: Move the call to enable the WebRTC event log from PeerConnectionFactory to PeerConnection. (Closed) Base URL: https://chromium.googlesource.com/chromium/src.git@master
Patch Set: Added limit to number of log files and the size of the log files. Created 4 years, 8 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/files/file.h" 10 #include "base/files/file.h"
(...skipping 109 matching lines...) Expand 10 before | Expand all | Expand 10 after
120 const std::string& type, 120 const std::string& type,
121 const std::string& sdp, 121 const std::string& sdp,
122 webrtc::SdpParseError* error); 122 webrtc::SdpParseError* error);
123 123
124 // Creates a libjingle representation of an ice candidate. 124 // Creates a libjingle representation of an ice candidate.
125 virtual webrtc::IceCandidateInterface* CreateIceCandidate( 125 virtual webrtc::IceCandidateInterface* CreateIceCandidate(
126 const std::string& sdp_mid, 126 const std::string& sdp_mid,
127 int sdp_mline_index, 127 int sdp_mline_index,
128 const std::string& sdp); 128 const std::string& sdp);
129 129
130 // Starts recording an RTC event log.
131 virtual bool StartRtcEventLog(base::PlatformFile file);
132
133 // Starts recording an RTC event log.
134 virtual void StopRtcEventLog();
135
136 WebRtcAudioDeviceImpl* GetWebRtcAudioDevice(); 130 WebRtcAudioDeviceImpl* GetWebRtcAudioDevice();
137 131
138 void EnsureInitialized(); 132 void EnsureInitialized();
139 scoped_refptr<base::SingleThreadTaskRunner> GetWebRtcWorkerThread() const; 133 scoped_refptr<base::SingleThreadTaskRunner> GetWebRtcWorkerThread() const;
140 scoped_refptr<base::SingleThreadTaskRunner> GetWebRtcSignalingThread() const; 134 scoped_refptr<base::SingleThreadTaskRunner> GetWebRtcSignalingThread() const;
141 135
142 protected: 136 protected:
143 // Asks the PeerConnection factory to create a Local Audio Source. 137 // Asks the PeerConnection factory to create a Local Audio Source.
144 virtual scoped_refptr<webrtc::AudioSourceInterface> CreateLocalAudioSource( 138 virtual scoped_refptr<webrtc::AudioSourceInterface> CreateLocalAudioSource(
145 const cricket::AudioOptions& options); 139 const cricket::AudioOptions& options);
(...skipping 71 matching lines...) Expand 10 before | Expand all | Expand 10 after
217 rtc::Thread* worker_thread_; 211 rtc::Thread* worker_thread_;
218 base::Thread chrome_signaling_thread_; 212 base::Thread chrome_signaling_thread_;
219 base::Thread chrome_worker_thread_; 213 base::Thread chrome_worker_thread_;
220 214
221 DISALLOW_COPY_AND_ASSIGN(PeerConnectionDependencyFactory); 215 DISALLOW_COPY_AND_ASSIGN(PeerConnectionDependencyFactory);
222 }; 216 };
223 217
224 } // namespace content 218 } // namespace content
225 219
226 #endif // CONTENT_RENDERER_MEDIA_WEBRTC_PEER_CONNECTION_DEPENDENCY_FACTORY_H_ 220 #endif // CONTENT_RENDERER_MEDIA_WEBRTC_PEER_CONNECTION_DEPENDENCY_FACTORY_H_
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