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							- /*
 
-  *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
 
-  *
 
-  *  Use of this source code is governed by a BSD-style license
 
-  *  that can be found in the LICENSE file in the root of the source
 
-  *  tree. An additional intellectual property rights grant can be found
 
-  *  in the file PATENTS.  All contributing project authors may
 
-  *  be found in the AUTHORS file in the root of the source tree.
 
-  */
 
- #ifndef CALL_RTX_RECEIVE_STREAM_H_
 
- #define CALL_RTX_RECEIVE_STREAM_H_
 
- #include <cstdint>
 
- #include <map>
 
- #include "call/rtp_packet_sink_interface.h"
 
- namespace webrtc {
 
- class ReceiveStatistics;
 
- // This class is responsible for RTX decapsulation. The resulting media packets
 
- // are passed on to a sink representing the associated media stream.
 
- class RtxReceiveStream : public RtpPacketSinkInterface {
 
-  public:
 
-   RtxReceiveStream(RtpPacketSinkInterface* media_sink,
 
-                    std::map<int, int> associated_payload_types,
 
-                    uint32_t media_ssrc,
 
-                    // TODO(nisse): Delete this argument, and
 
-                    // corresponding member variable, by moving the
 
-                    // responsibility for rtcp feedback to
 
-                    // RtpStreamReceiverController.
 
-                    ReceiveStatistics* rtp_receive_statistics = nullptr);
 
-   ~RtxReceiveStream() override;
 
-   // RtpPacketSinkInterface.
 
-   void OnRtpPacket(const RtpPacketReceived& packet) override;
 
-  private:
 
-   RtpPacketSinkInterface* const media_sink_;
 
-   // Map from rtx payload type -> media payload type.
 
-   const std::map<int, int> associated_payload_types_;
 
-   // TODO(nisse): Ultimately, the media receive stream shouldn't care about the
 
-   // ssrc, and we should delete this.
 
-   const uint32_t media_ssrc_;
 
-   ReceiveStatistics* const rtp_receive_statistics_;
 
- };
 
- }  // namespace webrtc
 
- #endif  // CALL_RTX_RECEIVE_STREAM_H_
 
 
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