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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
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| 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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