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1 /* | 1 /* |
2 * Copyright 2011 The WebRTC project authors. All Rights Reserved. | 2 * Copyright 2011 The WebRTC project authors. All Rights Reserved. |
3 * | 3 * |
4 * Use of this source code is governed by a BSD-style license | 4 * Use of this source code is governed by a BSD-style license |
5 * that can be found in the LICENSE file in the root of the source | 5 * that can be found in the LICENSE file in the root of the source |
6 * tree. An additional intellectual property rights grant can be found | 6 * tree. An additional intellectual property rights grant can be found |
7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
9 */ | 9 */ |
10 | 10 |
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60 rtc::scoped_refptr<VideoTrackInterface> | 60 rtc::scoped_refptr<VideoTrackInterface> |
61 CreateVideoTrack(const std::string& id, | 61 CreateVideoTrack(const std::string& id, |
62 VideoSourceInterface* video_source) override; | 62 VideoSourceInterface* video_source) override; |
63 | 63 |
64 rtc::scoped_refptr<AudioTrackInterface> | 64 rtc::scoped_refptr<AudioTrackInterface> |
65 CreateAudioTrack(const std::string& id, | 65 CreateAudioTrack(const std::string& id, |
66 AudioSourceInterface* audio_source) override; | 66 AudioSourceInterface* audio_source) override; |
67 | 67 |
68 bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes) override; | 68 bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes) override; |
69 void StopAecDump() override; | 69 void StopAecDump() override; |
70 bool StartRtcEventLog(rtc::PlatformFile file) override; | 70 // TODO(ivoc) Remove after Chrome is updated. |
71 void StopRtcEventLog() override; | 71 bool StartRtcEventLog(rtc::PlatformFile file) override { return false; } |
| 72 // TODO(ivoc) Remove after Chrome is updated. |
| 73 void StopRtcEventLog() override {} |
72 | 74 |
73 virtual webrtc::MediaControllerInterface* CreateMediaController( | 75 virtual webrtc::MediaControllerInterface* CreateMediaController( |
74 const cricket::MediaConfig& config) const; | 76 const cricket::MediaConfig& config) const; |
75 virtual rtc::Thread* signaling_thread(); | 77 virtual rtc::Thread* signaling_thread(); |
76 virtual rtc::Thread* worker_thread(); | 78 virtual rtc::Thread* worker_thread(); |
77 const Options& options() const { return options_; } | 79 const Options& options() const { return options_; } |
78 | 80 |
79 protected: | 81 protected: |
80 PeerConnectionFactory(); | 82 PeerConnectionFactory(); |
81 PeerConnectionFactory( | 83 PeerConnectionFactory( |
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107 video_decoder_factory_; | 109 video_decoder_factory_; |
108 rtc::scoped_ptr<rtc::BasicNetworkManager> default_network_manager_; | 110 rtc::scoped_ptr<rtc::BasicNetworkManager> default_network_manager_; |
109 rtc::scoped_ptr<rtc::BasicPacketSocketFactory> default_socket_factory_; | 111 rtc::scoped_ptr<rtc::BasicPacketSocketFactory> default_socket_factory_; |
110 | 112 |
111 rtc::scoped_refptr<RefCountedDtlsIdentityStore> dtls_identity_store_; | 113 rtc::scoped_refptr<RefCountedDtlsIdentityStore> dtls_identity_store_; |
112 }; | 114 }; |
113 | 115 |
114 } // namespace webrtc | 116 } // namespace webrtc |
115 | 117 |
116 #endif // WEBRTC_API_PEERCONNECTIONFACTORY_H_ | 118 #endif // WEBRTC_API_PEERCONNECTIONFACTORY_H_ |
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