| Index: webrtc/call/rtp_stream_receiver_controller_interface.h
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| diff --git a/webrtc/call/rtp_stream_receiver_controller_interface.h b/webrtc/call/rtp_stream_receiver_controller_interface.h
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| new file mode 100644
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| index 0000000000000000000000000000000000000000..51d25a525e3c26ad514d9dd0fda8d7f66c46c6f7
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| --- /dev/null
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| +++ b/webrtc/call/rtp_stream_receiver_controller_interface.h
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| @@ -0,0 +1,47 @@
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| +/*
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| + *  Copyright (c) 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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| +#ifndef WEBRTC_CALL_RTP_STREAM_RECEIVER_CONTROLLER_INTERFACE_H_
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| +#define WEBRTC_CALL_RTP_STREAM_RECEIVER_CONTROLLER_INTERFACE_H_
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| +
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| +#include <memory>
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| +
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| +#include "webrtc/call/rtp_packet_sink_interface.h"
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| +
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| +namespace webrtc {
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| +
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| +// An RtpStreamReceiver is responsible for the rtp-specific but
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| +// media-independent state needed for receiving an RTP stream.
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| +// TODO(nisse): Currently, only owns the association between ssrc and
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| +// the stream's RtpPacketSinkInterface. Ownership of corresponding
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| +// objects from modules/rtp_rtcp/ should move to this class (or
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| +// rather, the corresponding implementation class). We should add
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| +// methods for getting rtp receive stats, and for sending RTCP
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| +// messages related to the receive stream.
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| +class RtpStreamReceiverInterface {
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| + public:
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| +  virtual ~RtpStreamReceiverInterface() {}
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| +};
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| +
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| +// This class acts as a factory for RtpStreamReceiver objects.
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| +class RtpStreamReceiverControllerInterface {
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| + public:
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| +  virtual ~RtpStreamReceiverControllerInterface() {}
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| +
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| +  virtual std::unique_ptr<RtpStreamReceiverInterface> CreateReceiver(
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| +      uint32_t ssrc,
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| +      RtpPacketSinkInterface* sink) = 0;
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| +  // For registering additional sinks, needed for FlexFEC.
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| +  virtual void AddSink(uint32_t ssrc, RtpPacketSinkInterface* sink) = 0;
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| +  virtual size_t RemoveSink(const RtpPacketSinkInterface* sink) = 0;
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| +};
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| +
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| +}  // namespace webrtc
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| +
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| +#endif  // WEBRTC_CALL_RTP_STREAM_RECEIVER_CONTROLLER_INTERFACE_H_
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| 
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