
- Removes a strong-reference cycle between RTCPeerConnection and RTCPeerConnectionObserver - Gives RTCPeerConnectionObserver a virtual dtor - Ensures RTCPeerConnectionTest tears down correctly - Ensures AppRTCDemo tears down correctly This is the talk/ half; the webrtc/ half is in https://webrtc-codereview.appspot.com/10539005 BUG=3054,3055,3100 R=noahric@google.com Review URL: https://webrtc-codereview.appspot.com/10499005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5771 4adac7df-926f-26a2-2b94-8c16560cd09d
84 lines
3.4 KiB
Objective-C
84 lines
3.4 KiB
Objective-C
/*
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* libjingle
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* Copyright 2013, Google Inc.
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*
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* Redistribution and use in source and binary forms, with or without
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* modification, are permitted provided that the following conditions are met:
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*
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* 1. Redistributions of source code must retain the above copyright notice,
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* this list of conditions and the following disclaimer.
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* 2. Redistributions in binary form must reproduce the above copyright notice,
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* this list of conditions and the following disclaimer in the documentation
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* and/or other materials provided with the distribution.
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* 3. The name of the author may not be used to endorse or promote products
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* derived from this software without specific prior written permission.
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*
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* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
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* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
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* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
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* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
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* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
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* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
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* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
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* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
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* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
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* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
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*/
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#include "talk/app/webrtc/peerconnectioninterface.h"
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#import "RTCPeerConnection.h"
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#import "RTCPeerConnectionDelegate.h"
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// These objects are created by RTCPeerConnectionFactory to wrap an
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// id<RTCPeerConnectionDelegate> and call methods on that interface.
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namespace webrtc {
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class RTCPeerConnectionObserver : public PeerConnectionObserver {
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public:
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explicit RTCPeerConnectionObserver(id<RTCPeerConnectionDelegate> delegate);
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virtual ~RTCPeerConnectionObserver();
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// |peerConnection| owns |this|, so outlives it.
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void SetPeerConnection(RTCPeerConnection *peerConnection);
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virtual void OnError() OVERRIDE;
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// Triggered when the SignalingState changed.
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virtual void OnSignalingChange(
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PeerConnectionInterface::SignalingState new_state) OVERRIDE;
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// Triggered when media is received on a new stream from remote peer.
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virtual void OnAddStream(MediaStreamInterface* stream) OVERRIDE;
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// Triggered when a remote peer close a stream.
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virtual void OnRemoveStream(MediaStreamInterface* stream) OVERRIDE;
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// Triggered when a remote peer open a data channel.
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virtual void OnDataChannel(DataChannelInterface* data_channel) OVERRIDE;
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// Triggered when renegotiation is needed, for example the ICE has restarted.
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virtual void OnRenegotiationNeeded() OVERRIDE;
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// Called any time the ICEConnectionState changes
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virtual void OnIceConnectionChange(
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PeerConnectionInterface::IceConnectionState new_state) OVERRIDE;
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// Called any time the ICEGatheringState changes
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virtual void OnIceGatheringChange(
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PeerConnectionInterface::IceGatheringState new_state) OVERRIDE;
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// New Ice candidate have been found.
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virtual void OnIceCandidate(const IceCandidateInterface* candidate) OVERRIDE;
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private:
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id<RTCPeerConnectionDelegate> _delegate;
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// __unsafe_unretained is in fact safe because |_peerConnection| owns |this|;
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// see comment on SetPeerConnection() above.
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__unsafe_unretained RTCPeerConnection *_peerConnection;
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};
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} // namespace webrtc
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