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| 1 /* | 1 /* |
| 2 * libjingle | 2 * libjingle |
| 3 * Copyright 2011 Google Inc. | 3 * Copyright 2011 Google Inc. |
| 4 * | 4 * |
| 5 * Redistribution and use in source and binary forms, with or without | 5 * Redistribution and use in source and binary forms, with or without |
| 6 * modification, are permitted provided that the following conditions are met: | 6 * modification, are permitted provided that the following conditions are met: |
| 7 * | 7 * |
| 8 * 1. Redistributions of source code must retain the above copyright notice, | 8 * 1. Redistributions of source code must retain the above copyright notice, |
| 9 * this list of conditions and the following disclaimer. | 9 * this list of conditions and the following disclaimer. |
| 10 * 2. Redistributions in binary form must reproduce the above copyright notice, | 10 * 2. Redistributions in binary form must reproduce the above copyright notice, |
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| 80 | 80 |
| 81 rtc::scoped_refptr<AudioTrackInterface> | 81 rtc::scoped_refptr<AudioTrackInterface> |
| 82 CreateAudioTrack(const std::string& id, | 82 CreateAudioTrack(const std::string& id, |
| 83 AudioSourceInterface* audio_source) override; | 83 AudioSourceInterface* audio_source) override; |
| 84 | 84 |
| 85 bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes) override; | 85 bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes) override; |
| 86 void StopAecDump() override; | 86 void StopAecDump() override; |
| 87 bool StartRtcEventLog(rtc::PlatformFile file) override; | 87 bool StartRtcEventLog(rtc::PlatformFile file) override; |
| 88 void StopRtcEventLog() override; | 88 void StopRtcEventLog() override; |
| 89 | 89 |
| 90 virtual webrtc::MediaControllerInterface* CreateMediaController() const; | 90 virtual webrtc::MediaControllerInterface* CreateMediaController( |
| 91 const cricket::MediaConfig &config) const; |
| 91 virtual rtc::Thread* signaling_thread(); | 92 virtual rtc::Thread* signaling_thread(); |
| 92 virtual rtc::Thread* worker_thread(); | 93 virtual rtc::Thread* worker_thread(); |
| 93 const Options& options() const { return options_; } | 94 const Options& options() const { return options_; } |
| 94 | 95 |
| 95 protected: | 96 protected: |
| 96 PeerConnectionFactory(); | 97 PeerConnectionFactory(); |
| 97 PeerConnectionFactory( | 98 PeerConnectionFactory( |
| 98 rtc::Thread* worker_thread, | 99 rtc::Thread* worker_thread, |
| 99 rtc::Thread* signaling_thread, | 100 rtc::Thread* signaling_thread, |
| 100 AudioDeviceModule* default_adm, | 101 AudioDeviceModule* default_adm, |
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| 123 video_decoder_factory_; | 124 video_decoder_factory_; |
| 124 rtc::scoped_ptr<rtc::BasicNetworkManager> default_network_manager_; | 125 rtc::scoped_ptr<rtc::BasicNetworkManager> default_network_manager_; |
| 125 rtc::scoped_ptr<rtc::BasicPacketSocketFactory> default_socket_factory_; | 126 rtc::scoped_ptr<rtc::BasicPacketSocketFactory> default_socket_factory_; |
| 126 | 127 |
| 127 rtc::scoped_refptr<RefCountedDtlsIdentityStore> dtls_identity_store_; | 128 rtc::scoped_refptr<RefCountedDtlsIdentityStore> dtls_identity_store_; |
| 128 }; | 129 }; |
| 129 | 130 |
| 130 } // namespace webrtc | 131 } // namespace webrtc |
| 131 | 132 |
| 132 #endif // TALK_APP_WEBRTC_PEERCONNECTIONFACTORY_H_ | 133 #endif // TALK_APP_WEBRTC_PEERCONNECTIONFACTORY_H_ |
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