henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 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 | #import "RTCEnumConverter.h" |
| 29 | |
| 30 | #include "talk/app/webrtc/peerconnectioninterface.h" |
| 31 | |
| 32 | @implementation RTCEnumConverter |
| 33 | |
| 34 | + (RTCICEConnectionState)convertIceConnectionStateToObjC: |
| 35 | (webrtc::PeerConnectionInterface::IceConnectionState)nativeState { |
| 36 | switch (nativeState) { |
| 37 | case webrtc::PeerConnectionInterface::kIceConnectionNew: |
| 38 | return RTCICEConnectionNew; |
| 39 | case webrtc::PeerConnectionInterface::kIceConnectionChecking: |
| 40 | return RTCICEConnectionChecking; |
| 41 | case webrtc::PeerConnectionInterface::kIceConnectionConnected: |
| 42 | return RTCICEConnectionConnected; |
| 43 | case webrtc::PeerConnectionInterface::kIceConnectionCompleted: |
| 44 | return RTCICEConnectionCompleted; |
| 45 | case webrtc::PeerConnectionInterface::kIceConnectionFailed: |
| 46 | return RTCICEConnectionFailed; |
| 47 | case webrtc::PeerConnectionInterface::kIceConnectionDisconnected: |
| 48 | return RTCICEConnectionDisconnected; |
| 49 | case webrtc::PeerConnectionInterface::kIceConnectionClosed: |
| 50 | return RTCICEConnectionClosed; |
| 51 | } |
| 52 | } |
| 53 | |
| 54 | + (RTCICEGatheringState)convertIceGatheringStateToObjC: |
| 55 | (webrtc::PeerConnectionInterface::IceGatheringState)nativeState { |
| 56 | switch (nativeState) { |
| 57 | case webrtc::PeerConnectionInterface::kIceGatheringNew: |
| 58 | return RTCICEGatheringNew; |
| 59 | case webrtc::PeerConnectionInterface::kIceGatheringGathering: |
| 60 | return RTCICEGatheringGathering; |
| 61 | case webrtc::PeerConnectionInterface::kIceGatheringComplete: |
| 62 | return RTCICEGatheringComplete; |
| 63 | } |
| 64 | } |
| 65 | |
| 66 | + (RTCSignalingState)convertSignalingStateToObjC: |
| 67 | (webrtc::PeerConnectionInterface::SignalingState)nativeState { |
| 68 | switch (nativeState) { |
| 69 | case webrtc::PeerConnectionInterface::kStable: |
| 70 | return RTCSignalingStable; |
| 71 | case webrtc::PeerConnectionInterface::kHaveLocalOffer: |
| 72 | return RTCSignalingHaveLocalOffer; |
| 73 | case webrtc::PeerConnectionInterface::kHaveLocalPrAnswer: |
| 74 | return RTCSignalingHaveLocalPrAnswer; |
| 75 | case webrtc::PeerConnectionInterface::kHaveRemoteOffer: |
| 76 | return RTCSignalingHaveRemoteOffer; |
| 77 | case webrtc::PeerConnectionInterface::kHaveRemotePrAnswer: |
| 78 | return RTCSignalingHaveRemotePrAnswer; |
| 79 | case webrtc::PeerConnectionInterface::kClosed: |
| 80 | return RTCSignalingClosed; |
| 81 | } |
| 82 | } |
| 83 | |
tkchin@webrtc.org | 19b1be1 | 2014-04-22 21:05:38 +0000 | [diff] [blame^] | 84 | + (webrtc::PeerConnectionInterface::StatsOutputLevel) |
| 85 | convertStatsOutputLevelToNative:(RTCStatsOutputLevel)statsOutputLevel { |
| 86 | switch (statsOutputLevel) { |
| 87 | case RTCStatsOutputLevelStandard: |
| 88 | return webrtc::PeerConnectionInterface::kStatsOutputLevelStandard; |
| 89 | case RTCStatsOutputLevelDebug: |
| 90 | return webrtc::PeerConnectionInterface::kStatsOutputLevelDebug; |
| 91 | } |
| 92 | } |
| 93 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 94 | + (RTCSourceState)convertSourceStateToObjC: |
| 95 | (webrtc::MediaSourceInterface::SourceState)nativeState { |
| 96 | switch (nativeState) { |
| 97 | case webrtc::MediaSourceInterface::kInitializing: |
| 98 | return RTCSourceStateInitializing; |
| 99 | case webrtc::MediaSourceInterface::kLive: |
| 100 | return RTCSourceStateLive; |
| 101 | case webrtc::MediaSourceInterface::kEnded: |
| 102 | return RTCSourceStateEnded; |
| 103 | case webrtc::MediaSourceInterface::kMuted: |
| 104 | return RTCSourceStateMuted; |
| 105 | } |
| 106 | } |
| 107 | |
| 108 | + (webrtc::MediaStreamTrackInterface::TrackState) |
| 109 | convertTrackStateToNative:(RTCTrackState)state { |
| 110 | switch (state) { |
| 111 | case RTCTrackStateInitializing: |
| 112 | return webrtc::MediaStreamTrackInterface::kInitializing; |
| 113 | case RTCTrackStateLive: |
| 114 | return webrtc::MediaStreamTrackInterface::kLive; |
| 115 | case RTCTrackStateEnded: |
| 116 | return webrtc::MediaStreamTrackInterface::kEnded; |
| 117 | case RTCTrackStateFailed: |
| 118 | return webrtc::MediaStreamTrackInterface::kFailed; |
| 119 | } |
| 120 | } |
| 121 | |
| 122 | + (RTCTrackState)convertTrackStateToObjC: |
| 123 | (webrtc::MediaStreamTrackInterface::TrackState)nativeState { |
| 124 | switch (nativeState) { |
| 125 | case webrtc::MediaStreamTrackInterface::kInitializing: |
| 126 | return RTCTrackStateInitializing; |
| 127 | case webrtc::MediaStreamTrackInterface::kLive: |
| 128 | return RTCTrackStateLive; |
| 129 | case webrtc::MediaStreamTrackInterface::kEnded: |
| 130 | return RTCTrackStateEnded; |
| 131 | case webrtc::MediaStreamTrackInterface::kFailed: |
| 132 | return RTCTrackStateFailed; |
| 133 | } |
| 134 | } |
| 135 | |
| 136 | @end |