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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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25 * DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY | 25 * DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY |
26 * THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT | 26 * THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT |
27 * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE | 27 * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE |
28 * OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. | 28 * OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. |
29 */ | 29 */ |
30 | 30 |
31 #ifndef WebRTCPeerConnectionHandler_h | 31 #ifndef WebRTCPeerConnectionHandler_h |
32 #define WebRTCPeerConnectionHandler_h | 32 #define WebRTCPeerConnectionHandler_h |
33 | 33 |
34 #include "WebRTCStats.h" | 34 #include "WebRTCStats.h" |
| 35 #include "WebVector.h" |
35 | 36 |
36 namespace blink { | 37 namespace blink { |
37 | 38 |
38 class WebMediaConstraints; | 39 class WebMediaConstraints; |
39 class WebMediaStream; | 40 class WebMediaStream; |
40 class WebMediaStreamTrack; | 41 class WebMediaStreamTrack; |
41 class WebRTCAnswerOptions; | 42 class WebRTCAnswerOptions; |
42 class WebRTCDTMFSenderHandler; | 43 class WebRTCDTMFSenderHandler; |
43 class WebRTCDataChannelHandler; | 44 class WebRTCDataChannelHandler; |
44 enum class WebRTCErrorType; | 45 enum class WebRTCErrorType; |
45 class WebRTCICECandidate; | 46 class WebRTCICECandidate; |
46 class WebRTCOfferOptions; | 47 class WebRTCOfferOptions; |
| 48 class WebRTCRtpReceiver; |
47 class WebRTCSessionDescription; | 49 class WebRTCSessionDescription; |
48 class WebRTCSessionDescriptionRequest; | 50 class WebRTCSessionDescriptionRequest; |
49 class WebRTCStatsRequest; | 51 class WebRTCStatsRequest; |
50 class WebRTCVoidRequest; | 52 class WebRTCVoidRequest; |
51 class WebString; | 53 class WebString; |
52 struct WebRTCConfiguration; | 54 struct WebRTCConfiguration; |
53 struct WebRTCDataChannelInit; | 55 struct WebRTCDataChannelInit; |
54 | 56 |
55 class WebRTCPeerConnectionHandler { | 57 class WebRTCPeerConnectionHandler { |
56 public: | 58 public: |
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85 virtual bool addStream(const WebMediaStream&, const WebMediaConstraints&) = 0; | 87 virtual bool addStream(const WebMediaStream&, const WebMediaConstraints&) = 0; |
86 virtual void removeStream(const WebMediaStream&) = 0; | 88 virtual void removeStream(const WebMediaStream&) = 0; |
87 virtual void getStats(const WebRTCStatsRequest&) = 0; | 89 virtual void getStats(const WebRTCStatsRequest&) = 0; |
88 // Gets stats using the new stats collection API, see | 90 // Gets stats using the new stats collection API, see |
89 // third_party/webrtc/api/stats/. These will replace the old stats collection | 91 // third_party/webrtc/api/stats/. These will replace the old stats collection |
90 // API when the new API has matured enough. | 92 // API when the new API has matured enough. |
91 virtual void getStats(std::unique_ptr<WebRTCStatsReportCallback>) = 0; | 93 virtual void getStats(std::unique_ptr<WebRTCStatsReportCallback>) = 0; |
92 virtual WebRTCDataChannelHandler* createDataChannel( | 94 virtual WebRTCDataChannelHandler* createDataChannel( |
93 const WebString& label, | 95 const WebString& label, |
94 const WebRTCDataChannelInit&) = 0; | 96 const WebRTCDataChannelInit&) = 0; |
| 97 // Gets receivers used by the peer connection. These are wrappers referencing |
| 98 // webrtc-layer receivers, multiple |WebRTCRtpReceiver| objects referencing |
| 99 // the same webrtc-layer receiver have the same |id|. |
| 100 virtual WebVector<std::unique_ptr<WebRTCRtpReceiver>> getReceivers() = 0; |
95 virtual WebRTCDTMFSenderHandler* createDTMFSender( | 101 virtual WebRTCDTMFSenderHandler* createDTMFSender( |
96 const WebMediaStreamTrack&) = 0; | 102 const WebMediaStreamTrack&) = 0; |
97 virtual void stop() = 0; | 103 virtual void stop() = 0; |
98 }; | 104 }; |
99 | 105 |
100 } // namespace blink | 106 } // namespace blink |
101 | 107 |
102 #endif // WebRTCPeerConnectionHandler_h | 108 #endif // WebRTCPeerConnectionHandler_h |
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