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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 WebRTCDataChannelHandler; | 44 class WebRTCDataChannelHandler; |
| 45 class WebRTCICECandidate; | 45 class WebRTCICECandidate; |
| 46 class WebRTCOfferOptions; | 46 class WebRTCOfferOptions; |
| 47 class WebRTCSessionDescription; | 47 class WebRTCSessionDescription; |
| 48 class WebRTCSessionDescriptionRequest; | 48 class WebRTCSessionDescriptionRequest; |
| 49 class WebRTCStatsRequest; | 49 class WebRTCStatsRequest; |
| 50 class WebRTCVoidRequest; | 50 class WebRTCVoidRequest; |
| 51 class WebString; | 51 class WebString; |
| 52 struct WebRTCDataChannelInit; | 52 struct WebRTCDataChannelInit; |
| 53 | 53 |
| 54 // Used to back histogram value of "WebRTC.PeerConnection.SelectedRtcpMuxPolicy"
, so treat as append-only. | 54 // Used to back histogram value of |
| 55 // "WebRTC.PeerConnection.SelectedRtcpMuxPolicy", so treat as append-only. |
| 55 enum RtcpMuxPolicy { | 56 enum RtcpMuxPolicy { |
| 56 RtcpMuxPolicyRequire, | 57 RtcpMuxPolicyRequire, |
| 57 RtcpMuxPolicyNegotiate, | 58 RtcpMuxPolicyNegotiate, |
| 58 RtcpMuxPolicyDefault, | 59 RtcpMuxPolicyDefault, |
| 59 RtcpMuxPolicyMax | 60 RtcpMuxPolicyMax |
| 60 }; | 61 }; |
| 61 | 62 |
| 62 class WebRTCPeerConnectionHandler { | 63 class WebRTCPeerConnectionHandler { |
| 63 public: | 64 public: |
| 64 virtual ~WebRTCPeerConnectionHandler() {} | 65 virtual ~WebRTCPeerConnectionHandler() {} |
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| 86 // DEPRECATED | 87 // DEPRECATED |
| 87 virtual bool addICECandidate(const WebRTCICECandidate&) { return false; } | 88 virtual bool addICECandidate(const WebRTCICECandidate&) { return false; } |
| 88 | 89 |
| 89 virtual bool addICECandidate(const WebRTCVoidRequest&, | 90 virtual bool addICECandidate(const WebRTCVoidRequest&, |
| 90 const WebRTCICECandidate&) { | 91 const WebRTCICECandidate&) { |
| 91 return false; | 92 return false; |
| 92 } | 93 } |
| 93 virtual bool addStream(const WebMediaStream&, const WebMediaConstraints&) = 0; | 94 virtual bool addStream(const WebMediaStream&, const WebMediaConstraints&) = 0; |
| 94 virtual void removeStream(const WebMediaStream&) = 0; | 95 virtual void removeStream(const WebMediaStream&) = 0; |
| 95 virtual void getStats(const WebRTCStatsRequest&) = 0; | 96 virtual void getStats(const WebRTCStatsRequest&) = 0; |
| 96 // Gets stats using the new stats collection API, see third_party/webrtc/api/s
tats/. | 97 // Gets stats using the new stats collection API, see |
| 97 // These will replace the old stats collection API when the new API has mature
d enough. | 98 // third_party/webrtc/api/stats/. These will replace the old stats collection |
| 99 // API when the new API has matured enough. |
| 98 virtual void getStats(std::unique_ptr<WebRTCStatsReportCallback>) = 0; | 100 virtual void getStats(std::unique_ptr<WebRTCStatsReportCallback>) = 0; |
| 99 virtual WebRTCDataChannelHandler* createDataChannel( | 101 virtual WebRTCDataChannelHandler* createDataChannel( |
| 100 const WebString& label, | 102 const WebString& label, |
| 101 const WebRTCDataChannelInit&) = 0; | 103 const WebRTCDataChannelInit&) = 0; |
| 102 virtual WebRTCDTMFSenderHandler* createDTMFSender( | 104 virtual WebRTCDTMFSenderHandler* createDTMFSender( |
| 103 const WebMediaStreamTrack&) = 0; | 105 const WebMediaStreamTrack&) = 0; |
| 104 virtual void stop() = 0; | 106 virtual void stop() = 0; |
| 105 }; | 107 }; |
| 106 | 108 |
| 107 } // namespace blink | 109 } // namespace blink |
| 108 | 110 |
| 109 #endif // WebRTCPeerConnectionHandler_h | 111 #endif // WebRTCPeerConnectionHandler_h |
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