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      1 /*
      2  * libjingle
      3  * Copyright 2013, Google Inc.
      4  *
      5  * Redistribution and use in source and binary forms, with or without
      6  * modification, are permitted provided that the following conditions are met:
      7  *
      8  *  1. Redistributions of source code must retain the above copyright notice,
      9  *     this list of conditions and the following disclaimer.
     10  *  2. Redistributions in binary form must reproduce the above copyright notice,
     11  *     this list of conditions and the following disclaimer in the documentation
     12  *     and/or other materials provided with the distribution.
     13  *  3. The name of the author may not be used to endorse or promote products
     14  *     derived from this software without specific prior written permission.
     15  *
     16  * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
     17  * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
     18  * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
     19  * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
     20  * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
     21  * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
     22  * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
     23  * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
     24  * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
     25  * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
     26  */
     27 
     28 #include "talk/app/webrtc/peerconnectioninterface.h"
     29 
     30 #import "RTCPeerConnection.h"
     31 #import "RTCPeerConnectionDelegate.h"
     32 
     33 // These objects are created by RTCPeerConnectionFactory to wrap an
     34 // id<RTCPeerConnectionDelegate> and call methods on that interface.
     35 
     36 namespace webrtc {
     37 
     38 class RTCPeerConnectionObserver : public PeerConnectionObserver {
     39 
     40  public:
     41   explicit RTCPeerConnectionObserver(id<RTCPeerConnectionDelegate> delegate);
     42 
     43   void SetPeerConnection(RTCPeerConnection *peerConnection);
     44 
     45   virtual void OnError() OVERRIDE;
     46 
     47   // Triggered when the SignalingState changed.
     48   virtual void OnSignalingChange(
     49       PeerConnectionInterface::SignalingState new_state) OVERRIDE;
     50 
     51   // Triggered when media is received on a new stream from remote peer.
     52   virtual void OnAddStream(MediaStreamInterface* stream) OVERRIDE;
     53 
     54   // Triggered when a remote peer close a stream.
     55   virtual void OnRemoveStream(MediaStreamInterface* stream) OVERRIDE;
     56 
     57   // Triggered when a remote peer open a data channel.
     58   virtual void OnDataChannel(DataChannelInterface* data_channel) OVERRIDE;
     59 
     60   // Triggered when renegotation is needed, for example the ICE has restarted.
     61   virtual void OnRenegotiationNeeded() OVERRIDE;
     62 
     63   // Called any time the ICEConnectionState changes
     64   virtual void OnIceConnectionChange(
     65       PeerConnectionInterface::IceConnectionState new_state) OVERRIDE;
     66 
     67   // Called any time the ICEGatheringState changes
     68   virtual void OnIceGatheringChange(
     69       PeerConnectionInterface::IceGatheringState new_state) OVERRIDE;
     70 
     71   // New Ice candidate have been found.
     72   virtual void OnIceCandidate(const IceCandidateInterface* candidate) OVERRIDE;
     73 
     74  private:
     75   id<RTCPeerConnectionDelegate> _delegate;
     76   RTCPeerConnection *_peerConnection;
     77 };
     78 
     79 } // namespace webrtc
     80