| Index: webrtc/api/ortc/rtptransportcontrollerinterface.h
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| diff --git a/webrtc/api/ortc/rtptransportcontrollerinterface.h b/webrtc/api/ortc/rtptransportcontrollerinterface.h
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| new file mode 100644
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| index 0000000000000000000000000000000000000000..32aedb9fa90968fe33618a4c9ba64d5193d45499
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| --- /dev/null
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| +++ b/webrtc/api/ortc/rtptransportcontrollerinterface.h
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| @@ -0,0 +1,54 @@
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| +/*
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| + *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
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| + *
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| + *  Use of this source code is governed by a BSD-style license
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| + *  that can be found in the LICENSE file in the root of the source
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| + *  tree. An additional intellectual property rights grant can be found
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| + *  in the file PATENTS.  All contributing project authors may
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| + *  be found in the AUTHORS file in the root of the source tree.
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| + */
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| +
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| +#ifndef WEBRTC_API_ORTC_RTPTRANSPORTCONTROLLERINTERFACE_H_
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| +#define WEBRTC_API_ORTC_RTPTRANSPORTCONTROLLERINTERFACE_H_
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| +
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| +#include <vector>
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| +
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| +#include "webrtc/api/ortc/rtptransportinterface.h"
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| +
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| +namespace webrtc {
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| +
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| +class RtpTransportControllerAdapter;
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| +
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| +// Used to group RTP transports to the same remote endpoint, for the purpose of
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| +// sharing bandwidth estimation and other things. Comparing this to the
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| +// PeerConnection model, non-budled audio/video would use two RtpTransports
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| +// with a single RtpTransportController, whereas bundled media would use a
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| +// single RtpTransport.
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| +//
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| +// RtpTransports are associated with this controller when they're created, by
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| +// passing the controller into OrtcFactory's relevant "CreateRtpTransport"
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| +// method. When a transport is destroyed, it's automatically disassociated.
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| +// GetTransports returns all currently associated transports.
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| +//
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| +// This is the RTP equivalent of "IceTransportController" in ORTC; RtpTransport
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| +// is to RtpTransportController as IceTransport is to IceTransportController.
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| +class RtpTransportControllerInterface {
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| + public:
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| +  virtual ~RtpTransportControllerInterface() {}
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| +
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| +  // Returns all transports that are controlled by this controller and
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| +  // haven't yet been destroyed.
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| +  virtual std::vector<RtpTransportInterface*> GetTransports() const = 0;
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| +
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| + protected:
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| +  // Only for internal use.
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| +  // Returns a pointer to the internal (non-public) interface.
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| +  virtual RtpTransportControllerAdapter* GetInternal() = 0;
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| +
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| +  // Classes that can use this internal interface.
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| +  friend class RtpTransportAdapter;
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| +};
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| +
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| +}  // namespace webrtc
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| +
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| +#endif  // WEBRTC_API_ORTC_RTPTRANSPORTCONTROLLERINTERFACE_H_
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| 
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