386449971a
BUG=webrtc:7135 Review-Url: https://codereview.webrtc.org/3005793002 Cr-Commit-Position: refs/heads/master@{#19599}
43 lines
1.4 KiB
C++
43 lines
1.4 KiB
C++
/*
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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_RTX_RECEIVE_STREAM_H_
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#define WEBRTC_CALL_RTX_RECEIVE_STREAM_H_
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#include <map>
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#include "webrtc/call/rtp_packet_sink_interface.h"
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namespace webrtc {
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// This class is responsible for RTX decapsulation. The resulting media packets
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// are passed on to a sink representing the associated media stream.
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class RtxReceiveStream : public RtpPacketSinkInterface {
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public:
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RtxReceiveStream(RtpPacketSinkInterface* media_sink,
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std::map<int, int> associated_payload_types,
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uint32_t media_ssrc);
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~RtxReceiveStream() override;
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// RtpPacketSinkInterface.
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void OnRtpPacket(const RtpPacketReceived& packet) override;
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private:
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RtpPacketSinkInterface* const media_sink_;
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// Map from rtx payload type -> media payload type.
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const std::map<int, int> associated_payload_types_;
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// TODO(nisse): Ultimately, the media receive stream shouldn't care about the
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// ssrc, and we should delete this.
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const uint32_t media_ssrc_;
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};
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} // namespace webrtc
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#endif // WEBRTC_CALL_RTX_RECEIVE_STREAM_H_
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