28e2075280
trunk/talk git-svn-id: http://webrtc.googlecode.com/svn/trunk@4318 4adac7df-926f-26a2-2b94-8c16560cd09d
127 lines
5.0 KiB
Plaintext
127 lines
5.0 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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#import "RTCEnumConverter.h"
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#include "talk/app/webrtc/peerconnectioninterface.h"
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@implementation RTCEnumConverter
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+ (RTCICEConnectionState)convertIceConnectionStateToObjC:
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(webrtc::PeerConnectionInterface::IceConnectionState)nativeState {
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switch (nativeState) {
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case webrtc::PeerConnectionInterface::kIceConnectionNew:
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return RTCICEConnectionNew;
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case webrtc::PeerConnectionInterface::kIceConnectionChecking:
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return RTCICEConnectionChecking;
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case webrtc::PeerConnectionInterface::kIceConnectionConnected:
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return RTCICEConnectionConnected;
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case webrtc::PeerConnectionInterface::kIceConnectionCompleted:
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return RTCICEConnectionCompleted;
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case webrtc::PeerConnectionInterface::kIceConnectionFailed:
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return RTCICEConnectionFailed;
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case webrtc::PeerConnectionInterface::kIceConnectionDisconnected:
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return RTCICEConnectionDisconnected;
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case webrtc::PeerConnectionInterface::kIceConnectionClosed:
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return RTCICEConnectionClosed;
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}
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}
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+ (RTCICEGatheringState)convertIceGatheringStateToObjC:
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(webrtc::PeerConnectionInterface::IceGatheringState)nativeState {
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switch (nativeState) {
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case webrtc::PeerConnectionInterface::kIceGatheringNew:
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return RTCICEGatheringNew;
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case webrtc::PeerConnectionInterface::kIceGatheringGathering:
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return RTCICEGatheringGathering;
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case webrtc::PeerConnectionInterface::kIceGatheringComplete:
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return RTCICEGatheringComplete;
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}
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}
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+ (RTCSignalingState)convertSignalingStateToObjC:
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(webrtc::PeerConnectionInterface::SignalingState)nativeState {
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switch (nativeState) {
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case webrtc::PeerConnectionInterface::kStable:
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return RTCSignalingStable;
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case webrtc::PeerConnectionInterface::kHaveLocalOffer:
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return RTCSignalingHaveLocalOffer;
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case webrtc::PeerConnectionInterface::kHaveLocalPrAnswer:
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return RTCSignalingHaveLocalPrAnswer;
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case webrtc::PeerConnectionInterface::kHaveRemoteOffer:
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return RTCSignalingHaveRemoteOffer;
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case webrtc::PeerConnectionInterface::kHaveRemotePrAnswer:
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return RTCSignalingHaveRemotePrAnswer;
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case webrtc::PeerConnectionInterface::kClosed:
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return RTCSignalingClosed;
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}
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}
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+ (RTCSourceState)convertSourceStateToObjC:
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(webrtc::MediaSourceInterface::SourceState)nativeState {
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switch (nativeState) {
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case webrtc::MediaSourceInterface::kInitializing:
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return RTCSourceStateInitializing;
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case webrtc::MediaSourceInterface::kLive:
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return RTCSourceStateLive;
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case webrtc::MediaSourceInterface::kEnded:
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return RTCSourceStateEnded;
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case webrtc::MediaSourceInterface::kMuted:
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return RTCSourceStateMuted;
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}
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}
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+ (webrtc::MediaStreamTrackInterface::TrackState)
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convertTrackStateToNative:(RTCTrackState)state {
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switch (state) {
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case RTCTrackStateInitializing:
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return webrtc::MediaStreamTrackInterface::kInitializing;
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case RTCTrackStateLive:
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return webrtc::MediaStreamTrackInterface::kLive;
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case RTCTrackStateEnded:
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return webrtc::MediaStreamTrackInterface::kEnded;
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case RTCTrackStateFailed:
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return webrtc::MediaStreamTrackInterface::kFailed;
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}
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}
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+ (RTCTrackState)convertTrackStateToObjC:
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(webrtc::MediaStreamTrackInterface::TrackState)nativeState {
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switch (nativeState) {
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case webrtc::MediaStreamTrackInterface::kInitializing:
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return RTCTrackStateInitializing;
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case webrtc::MediaStreamTrackInterface::kLive:
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return RTCTrackStateLive;
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case webrtc::MediaStreamTrackInterface::kEnded:
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return RTCTrackStateEnded;
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case webrtc::MediaStreamTrackInterface::kFailed:
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return RTCTrackStateFailed;
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}
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}
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@end
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