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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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