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1 /* | 1 /* |
2 * Copyright (C) 2012 Google Inc. All rights reserved. | 2 * Copyright (C) 2012 Google Inc. All rights reserved. |
3 * | 3 * |
4 * Redistribution and use in source and binary forms, with or without | 4 * Redistribution and use in source and binary forms, with or without |
5 * modification, are permitted provided that the following conditions are | 5 * modification, are permitted provided that the following conditions are |
6 * met: | 6 * met: |
7 * | 7 * |
8 * * Redistributions of source code must retain the above copyright | 8 * * Redistributions of source code must retain the above copyright |
9 * notice, this list of conditions and the following disclaimer. | 9 * notice, this list of conditions and the following disclaimer. |
10 * * Redistributions in binary form must reproduce the above | 10 * * Redistributions in binary form must reproduce the above |
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44 class WebRTCICECandidate; | 44 class WebRTCICECandidate; |
45 class WebRTCOfferOptions; | 45 class WebRTCOfferOptions; |
46 class WebRTCSessionDescription; | 46 class WebRTCSessionDescription; |
47 class WebRTCSessionDescriptionRequest; | 47 class WebRTCSessionDescriptionRequest; |
48 class WebRTCStatsRequest; | 48 class WebRTCStatsRequest; |
49 class WebRTCVoidRequest; | 49 class WebRTCVoidRequest; |
50 class WebString; | 50 class WebString; |
51 struct WebRTCConfiguration; | 51 struct WebRTCConfiguration; |
52 struct WebRTCDataChannelInit; | 52 struct WebRTCDataChannelInit; |
53 | 53 |
54 // Used to back histogram value of | |
55 // "WebRTC.PeerConnection.SelectedRtcpMuxPolicy", so treat as append-only. | |
56 enum RtcpMuxPolicy { | |
57 RtcpMuxPolicyRequire, | |
58 RtcpMuxPolicyNegotiate, | |
59 RtcpMuxPolicyDefault, | |
60 RtcpMuxPolicyMax | |
61 }; | |
62 | |
63 class WebRTCPeerConnectionHandler { | 54 class WebRTCPeerConnectionHandler { |
64 public: | 55 public: |
65 virtual ~WebRTCPeerConnectionHandler() {} | 56 virtual ~WebRTCPeerConnectionHandler() {} |
66 | 57 |
67 virtual bool initialize(const WebRTCConfiguration&, | 58 virtual bool initialize(const WebRTCConfiguration&, |
68 const WebMediaConstraints&) = 0; | 59 const WebMediaConstraints&) = 0; |
69 | 60 |
70 virtual void createOffer(const WebRTCSessionDescriptionRequest&, | 61 virtual void createOffer(const WebRTCSessionDescriptionRequest&, |
71 const WebMediaConstraints&) = 0; | 62 const WebMediaConstraints&) = 0; |
72 virtual void createOffer(const WebRTCSessionDescriptionRequest&, | 63 virtual void createOffer(const WebRTCSessionDescriptionRequest&, |
73 const WebRTCOfferOptions&) = 0; | 64 const WebRTCOfferOptions&) = 0; |
74 virtual void createAnswer(const WebRTCSessionDescriptionRequest&, | 65 virtual void createAnswer(const WebRTCSessionDescriptionRequest&, |
75 const WebMediaConstraints&) = 0; | 66 const WebMediaConstraints&) = 0; |
76 virtual void createAnswer(const WebRTCSessionDescriptionRequest&, | 67 virtual void createAnswer(const WebRTCSessionDescriptionRequest&, |
77 const WebRTCAnswerOptions&) = 0; | 68 const WebRTCAnswerOptions&) = 0; |
78 virtual void setLocalDescription(const WebRTCVoidRequest&, | 69 virtual void setLocalDescription(const WebRTCVoidRequest&, |
79 const WebRTCSessionDescription&) = 0; | 70 const WebRTCSessionDescription&) = 0; |
80 virtual void setRemoteDescription(const WebRTCVoidRequest&, | 71 virtual void setRemoteDescription(const WebRTCVoidRequest&, |
81 const WebRTCSessionDescription&) = 0; | 72 const WebRTCSessionDescription&) = 0; |
82 virtual WebRTCSessionDescription localDescription() = 0; | 73 virtual WebRTCSessionDescription localDescription() = 0; |
83 virtual WebRTCSessionDescription remoteDescription() = 0; | 74 virtual WebRTCSessionDescription remoteDescription() = 0; |
84 virtual bool setConfiguration(const WebRTCConfiguration&) = 0; | 75 virtual bool setConfiguration(const WebRTCConfiguration&) = 0; |
85 virtual void logSelectedRtcpMuxPolicy(RtcpMuxPolicy) = 0; | |
86 | 76 |
87 // DEPRECATED | 77 // DEPRECATED |
88 virtual bool addICECandidate(const WebRTCICECandidate&) { return false; } | 78 virtual bool addICECandidate(const WebRTCICECandidate&) { return false; } |
89 | 79 |
90 virtual bool addICECandidate(const WebRTCVoidRequest&, | 80 virtual bool addICECandidate(const WebRTCVoidRequest&, |
91 const WebRTCICECandidate&) { | 81 const WebRTCICECandidate&) { |
92 return false; | 82 return false; |
93 } | 83 } |
94 virtual bool addStream(const WebMediaStream&, const WebMediaConstraints&) = 0; | 84 virtual bool addStream(const WebMediaStream&, const WebMediaConstraints&) = 0; |
95 virtual void removeStream(const WebMediaStream&) = 0; | 85 virtual void removeStream(const WebMediaStream&) = 0; |
96 virtual void getStats(const WebRTCStatsRequest&) = 0; | 86 virtual void getStats(const WebRTCStatsRequest&) = 0; |
97 // Gets stats using the new stats collection API, see | 87 // Gets stats using the new stats collection API, see |
98 // third_party/webrtc/api/stats/. These will replace the old stats collection | 88 // third_party/webrtc/api/stats/. These will replace the old stats collection |
99 // API when the new API has matured enough. | 89 // API when the new API has matured enough. |
100 virtual void getStats(std::unique_ptr<WebRTCStatsReportCallback>) = 0; | 90 virtual void getStats(std::unique_ptr<WebRTCStatsReportCallback>) = 0; |
101 virtual WebRTCDataChannelHandler* createDataChannel( | 91 virtual WebRTCDataChannelHandler* createDataChannel( |
102 const WebString& label, | 92 const WebString& label, |
103 const WebRTCDataChannelInit&) = 0; | 93 const WebRTCDataChannelInit&) = 0; |
104 virtual WebRTCDTMFSenderHandler* createDTMFSender( | 94 virtual WebRTCDTMFSenderHandler* createDTMFSender( |
105 const WebMediaStreamTrack&) = 0; | 95 const WebMediaStreamTrack&) = 0; |
106 virtual void stop() = 0; | 96 virtual void stop() = 0; |
107 }; | 97 }; |
108 | 98 |
109 } // namespace blink | 99 } // namespace blink |
110 | 100 |
111 #endif // WebRTCPeerConnectionHandler_h | 101 #endif // WebRTCPeerConnectionHandler_h |
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