28e2075280
trunk/talk git-svn-id: http://webrtc.googlecode.com/svn/trunk@4318 4adac7df-926f-26a2-2b94-8c16560cd09d
104 lines
3.8 KiB
Plaintext
104 lines
3.8 KiB
Plaintext
/*
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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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#if !defined(__has_feature) || !__has_feature(objc_arc)
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#error "This file requires ARC support."
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#endif
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#import "RTCPeerConnectionObserver.h"
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#import "RTCICECandidate+internal.h"
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#import "RTCMediaStream+internal.h"
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#import "RTCEnumConverter.h"
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namespace webrtc {
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RTCPeerConnectionObserver::RTCPeerConnectionObserver(
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id<RTCPeerConnectionDelegate> delegate) {
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_delegate = delegate;
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}
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void RTCPeerConnectionObserver::SetPeerConnection(
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RTCPeerConnection *peerConnection) {
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_peerConnection = peerConnection;
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}
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void RTCPeerConnectionObserver::OnError() {
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[_delegate peerConnectionOnError:_peerConnection];
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}
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void RTCPeerConnectionObserver::OnSignalingChange(
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PeerConnectionInterface::SignalingState new_state) {
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[_delegate peerConnection:_peerConnection
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signalingStateChanged:
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[RTCEnumConverter convertSignalingStateToObjC:new_state]];
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}
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void RTCPeerConnectionObserver::OnAddStream(MediaStreamInterface* stream) {
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RTCMediaStream* mediaStream =
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[[RTCMediaStream alloc] initWithMediaStream:stream];
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[_delegate peerConnection:_peerConnection addedStream:mediaStream];
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}
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void RTCPeerConnectionObserver::OnRemoveStream(MediaStreamInterface* stream) {
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RTCMediaStream* mediaStream =
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[[RTCMediaStream alloc] initWithMediaStream:stream];
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[_delegate peerConnection:_peerConnection removedStream:mediaStream];
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}
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void RTCPeerConnectionObserver::OnDataChannel(
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DataChannelInterface* data_channel) {
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// TODO(hughv): Implement for future version.
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}
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void RTCPeerConnectionObserver::OnRenegotiationNeeded() {
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[_delegate peerConnectionOnRenegotiationNeeded:_peerConnection];
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}
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void RTCPeerConnectionObserver::OnIceConnectionChange(
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PeerConnectionInterface::IceConnectionState new_state) {
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[_delegate peerConnection:_peerConnection
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iceConnectionChanged:
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[RTCEnumConverter convertIceConnectionStateToObjC:new_state]];
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}
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void RTCPeerConnectionObserver::OnIceGatheringChange(
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PeerConnectionInterface::IceGatheringState new_state) {
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[_delegate peerConnection:_peerConnection
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iceGatheringChanged:
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[RTCEnumConverter convertIceGatheringStateToObjC:new_state]];
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}
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void RTCPeerConnectionObserver::OnIceCandidate(
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const IceCandidateInterface* candidate) {
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RTCICECandidate* iceCandidate =
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[[RTCICECandidate alloc] initWithCandidate:candidate];
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[_delegate peerConnection:_peerConnection gotICECandidate:iceCandidate];
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}
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} // namespace webrtc
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