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1 /* | |
2 * Copyright 2017 The WebRTC project authors. All Rights Reserved. | |
3 * | |
4 * Use of this source code is governed by a BSD-style license | |
5 * that can be found in the LICENSE file in the root of the source | |
6 * tree. An additional intellectual property rights grant can be found | |
7 * in the file PATENTS. All contributing project authors may | |
8 * be found in the AUTHORS file in the root of the source tree. | |
9 */ | |
10 | |
11 #ifndef WEBRTC_API_ORTC_RTPTRANSPORTCONTROLLERINTERFACE_H_ | |
12 #define WEBRTC_API_ORTC_RTPTRANSPORTCONTROLLERINTERFACE_H_ | |
13 | |
14 #include <vector> | |
15 | |
16 #include "webrtc/api/ortc/rtptransportinterface.h" | |
17 | |
18 namespace webrtc { | |
19 | |
20 class RtpTransportControllerShim; | |
pthatcher1
2017/02/10 22:36:52
Can you leave a comment somewhere about what "shim
Taylor Brandstetter
2017/02/14 06:55:04
This is just a forward declaration. I have a comme
| |
21 | |
22 // Used to group RTP transports to the same remote endpoint, for the purpose of | |
23 // sharing bandwidth estimation and other things. Comparing this to the | |
24 // PeerConnection model, non-budled audio/video would use two RtpTransports | |
25 // with a single RtpTransportController, whereas bundled media would use a | |
26 // single RtpTransport. | |
27 // | |
28 // RtpTransports are associated with this controller when they're created, by | |
29 // passing the controller into OrtcFactory's relevant "CreateRtpTransport" | |
30 // method. When a transport is destroyed, it's automatically disassociated. | |
31 // GetTransports returns all currently associated transports. | |
32 // | |
33 // This is the RTP equivalent of "IceTransportController" in ORTC; RtpTransport | |
34 // is to RtpTransportController as IceTransport is to IceTransportController. | |
35 class RtpTransportControllerInterface { | |
36 public: | |
37 virtual ~RtpTransportControllerInterface() {} | |
38 | |
39 // Returns all transports that are controlled by this controller and | |
40 // haven't yet been destroyed. | |
41 virtual std::vector<RtpTransportInterface*> GetTransports() const = 0; | |
42 | |
43 protected: | |
44 // Only for internal use. | |
45 // Returns a pointer to the internal (non-public) interface. | |
46 virtual RtpTransportControllerShim* GetInternal() = 0; | |
47 | |
48 // Classes that can use this internal interface. | |
49 friend class RtpTransportShim; | |
50 }; | |
51 | |
52 } // namespace webrtc | |
53 | |
54 #endif // WEBRTC_API_ORTC_RTPTRANSPORTCONTROLLERINTERFACE_H_ | |
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