1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
|
/*
* libjingle
* Copyright 2013, Google Inc.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions are met:
*
* 1. Redistributions of source code must retain the above copyright notice,
* this list of conditions and the following disclaimer.
* 2. Redistributions in binary form must reproduce the above copyright notice,
* this list of conditions and the following disclaimer in the documentation
* and/or other materials provided with the distribution.
* 3. The name of the author may not be used to endorse or promote products
* derived from this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#import "RTCEnumConverter.h"
#include "talk/app/webrtc/peerconnectioninterface.h"
@implementation RTCEnumConverter
+ (RTCICEConnectionState)convertIceConnectionStateToObjC:
(webrtc::PeerConnectionInterface::IceConnectionState)nativeState {
switch (nativeState) {
case webrtc::PeerConnectionInterface::kIceConnectionNew:
return RTCICEConnectionNew;
case webrtc::PeerConnectionInterface::kIceConnectionChecking:
return RTCICEConnectionChecking;
case webrtc::PeerConnectionInterface::kIceConnectionConnected:
return RTCICEConnectionConnected;
case webrtc::PeerConnectionInterface::kIceConnectionCompleted:
return RTCICEConnectionCompleted;
case webrtc::PeerConnectionInterface::kIceConnectionFailed:
return RTCICEConnectionFailed;
case webrtc::PeerConnectionInterface::kIceConnectionDisconnected:
return RTCICEConnectionDisconnected;
case webrtc::PeerConnectionInterface::kIceConnectionClosed:
return RTCICEConnectionClosed;
}
}
+ (RTCICEGatheringState)convertIceGatheringStateToObjC:
(webrtc::PeerConnectionInterface::IceGatheringState)nativeState {
switch (nativeState) {
case webrtc::PeerConnectionInterface::kIceGatheringNew:
return RTCICEGatheringNew;
case webrtc::PeerConnectionInterface::kIceGatheringGathering:
return RTCICEGatheringGathering;
case webrtc::PeerConnectionInterface::kIceGatheringComplete:
return RTCICEGatheringComplete;
}
}
+ (RTCSignalingState)convertSignalingStateToObjC:
(webrtc::PeerConnectionInterface::SignalingState)nativeState {
switch (nativeState) {
case webrtc::PeerConnectionInterface::kStable:
return RTCSignalingStable;
case webrtc::PeerConnectionInterface::kHaveLocalOffer:
return RTCSignalingHaveLocalOffer;
case webrtc::PeerConnectionInterface::kHaveLocalPrAnswer:
return RTCSignalingHaveLocalPrAnswer;
case webrtc::PeerConnectionInterface::kHaveRemoteOffer:
return RTCSignalingHaveRemoteOffer;
case webrtc::PeerConnectionInterface::kHaveRemotePrAnswer:
return RTCSignalingHaveRemotePrAnswer;
case webrtc::PeerConnectionInterface::kClosed:
return RTCSignalingClosed;
}
}
+ (webrtc::PeerConnectionInterface::StatsOutputLevel)
convertStatsOutputLevelToNative:(RTCStatsOutputLevel)statsOutputLevel {
switch (statsOutputLevel) {
case RTCStatsOutputLevelStandard:
return webrtc::PeerConnectionInterface::kStatsOutputLevelStandard;
case RTCStatsOutputLevelDebug:
return webrtc::PeerConnectionInterface::kStatsOutputLevelDebug;
}
}
+ (RTCSourceState)convertSourceStateToObjC:
(webrtc::MediaSourceInterface::SourceState)nativeState {
switch (nativeState) {
case webrtc::MediaSourceInterface::kInitializing:
return RTCSourceStateInitializing;
case webrtc::MediaSourceInterface::kLive:
return RTCSourceStateLive;
case webrtc::MediaSourceInterface::kEnded:
return RTCSourceStateEnded;
case webrtc::MediaSourceInterface::kMuted:
return RTCSourceStateMuted;
}
}
+ (webrtc::MediaStreamTrackInterface::TrackState)
convertTrackStateToNative:(RTCTrackState)state {
switch (state) {
case RTCTrackStateInitializing:
return webrtc::MediaStreamTrackInterface::kInitializing;
case RTCTrackStateLive:
return webrtc::MediaStreamTrackInterface::kLive;
case RTCTrackStateEnded:
return webrtc::MediaStreamTrackInterface::kEnded;
case RTCTrackStateFailed:
return webrtc::MediaStreamTrackInterface::kFailed;
}
}
+ (RTCTrackState)convertTrackStateToObjC:
(webrtc::MediaStreamTrackInterface::TrackState)nativeState {
switch (nativeState) {
case webrtc::MediaStreamTrackInterface::kInitializing:
return RTCTrackStateInitializing;
case webrtc::MediaStreamTrackInterface::kLive:
return RTCTrackStateLive;
case webrtc::MediaStreamTrackInterface::kEnded:
return RTCTrackStateEnded;
case webrtc::MediaStreamTrackInterface::kFailed:
return RTCTrackStateFailed;
}
}
@end
|
