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|    1 /* |    1 /* | 
|    2  * libjingle |    2  * libjingle | 
|    3  * Copyright 2012 Google Inc. |    3  * Copyright 2012 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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|  303  |  303  | 
|  304   bool IceRestartPending() const; |  304   bool IceRestartPending() const; | 
|  305  |  305  | 
|  306   void ResetIceRestartLatch(); |  306   void ResetIceRestartLatch(); | 
|  307  |  307  | 
|  308   // Called when an RTCCertificate is generated or retrieved by |  308   // Called when an RTCCertificate is generated or retrieved by | 
|  309   // WebRTCSessionDescriptionFactory. Should happen before setLocalDescription. |  309   // WebRTCSessionDescriptionFactory. Should happen before setLocalDescription. | 
|  310   void OnCertificateReady( |  310   void OnCertificateReady( | 
|  311       const rtc::scoped_refptr<rtc::RTCCertificate>& certificate); |  311       const rtc::scoped_refptr<rtc::RTCCertificate>& certificate); | 
|  312   void OnDtlsSetupFailure(cricket::BaseChannel*, bool rtcp); |  312   void OnDtlsSetupFailure(cricket::BaseChannel*, bool rtcp); | 
|  313   void OnChannelFirstPacketReceived(cricket::BaseChannel*); |  | 
|  314  |  313  | 
|  315   // For unit test. |  314   // For unit test. | 
|  316   bool waiting_for_certificate_for_testing() const; |  315   bool waiting_for_certificate_for_testing() const; | 
|  317   const rtc::scoped_refptr<rtc::RTCCertificate>& certificate_for_testing(); |  316   const rtc::scoped_refptr<rtc::RTCCertificate>& certificate_for_testing(); | 
|  318  |  317  | 
|  319   void set_metrics_observer( |  318   void set_metrics_observer( | 
|  320       webrtc::MetricsObserverInterface* metrics_observer) { |  319       webrtc::MetricsObserverInterface* metrics_observer) { | 
|  321     metrics_observer_ = metrics_observer; |  320     metrics_observer_ = metrics_observer; | 
|  322   } |  321   } | 
|  323  |  322  | 
|  324   // Called when voice_channel_, video_channel_ and data_channel_ are created |  323   // Called when voice_channel_, video_channel_ and data_channel_ are created | 
|  325   // and destroyed. As a result of, for example, setting a new description. |  324   // and destroyed. As a result of, for example, setting a new description. | 
|  326   sigslot::signal0<> SignalVoiceChannelCreated; |  325   sigslot::signal0<> SignalVoiceChannelCreated; | 
|  327   sigslot::signal0<> SignalVoiceChannelDestroyed; |  326   sigslot::signal0<> SignalVoiceChannelDestroyed; | 
|  328   sigslot::signal0<> SignalVideoChannelCreated; |  327   sigslot::signal0<> SignalVideoChannelCreated; | 
|  329   sigslot::signal0<> SignalVideoChannelDestroyed; |  328   sigslot::signal0<> SignalVideoChannelDestroyed; | 
|  330   sigslot::signal0<> SignalDataChannelCreated; |  329   sigslot::signal0<> SignalDataChannelCreated; | 
|  331   sigslot::signal0<> SignalDataChannelDestroyed; |  330   sigslot::signal0<> SignalDataChannelDestroyed; | 
|  332  |  331  | 
|  333   // Called when a valid data channel OPEN message is received. |  332   // Called when a valid data channel OPEN message is received. | 
|  334   // std::string represents the data channel label. |  333   // std::string represents the data channel label. | 
|  335   sigslot::signal2<const std::string&, const InternalDataChannelInit&> |  334   sigslot::signal2<const std::string&, const InternalDataChannelInit&> | 
|  336       SignalDataChannelOpenMessage; |  335       SignalDataChannelOpenMessage; | 
|  337  |  336  | 
|  338   // Called when the first RTP packet is received. |  | 
|  339   sigslot::signal0<> SignalFirstMediaPacketReceived; |  | 
|  340  |  | 
|  341  private: |  337  private: | 
|  342   // Indicates the type of SessionDescription in a call to SetLocalDescription |  338   // Indicates the type of SessionDescription in a call to SetLocalDescription | 
|  343   // and SetRemoteDescription. |  339   // and SetRemoteDescription. | 
|  344   enum Action { |  340   enum Action { | 
|  345     kOffer, |  341     kOffer, | 
|  346     kPrAnswer, |  342     kPrAnswer, | 
|  347     kAnswer, |  343     kAnswer, | 
|  348   }; |  344   }; | 
|  349  |  345  | 
|  350   // Log session state. |  346   // Log session state. | 
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|  514   cricket::AudioOptions audio_options_; |  510   cricket::AudioOptions audio_options_; | 
|  515   cricket::VideoOptions video_options_; |  511   cricket::VideoOptions video_options_; | 
|  516   MetricsObserverInterface* metrics_observer_; |  512   MetricsObserverInterface* metrics_observer_; | 
|  517  |  513  | 
|  518   // Declares the bundle policy for the WebRTCSession. |  514   // Declares the bundle policy for the WebRTCSession. | 
|  519   PeerConnectionInterface::BundlePolicy bundle_policy_; |  515   PeerConnectionInterface::BundlePolicy bundle_policy_; | 
|  520  |  516  | 
|  521   // Declares the RTCP mux policy for the WebRTCSession. |  517   // Declares the RTCP mux policy for the WebRTCSession. | 
|  522   PeerConnectionInterface::RtcpMuxPolicy rtcp_mux_policy_; |  518   PeerConnectionInterface::RtcpMuxPolicy rtcp_mux_policy_; | 
|  523  |  519  | 
|  524   bool has_received_media_packet_ = false; |  | 
|  525  |  | 
|  526   RTC_DISALLOW_COPY_AND_ASSIGN(WebRtcSession); |  520   RTC_DISALLOW_COPY_AND_ASSIGN(WebRtcSession); | 
|  527 }; |  521 }; | 
|  528 }  // namespace webrtc |  522 }  // namespace webrtc | 
|  529  |  523  | 
|  530 #endif  // TALK_APP_WEBRTC_WEBRTCSESSION_H_ |  524 #endif  // TALK_APP_WEBRTC_WEBRTCSESSION_H_ | 
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