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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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