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92ea95e34a
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
50 lines
1.8 KiB
C++
50 lines
1.8 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 CALL_RTX_RECEIVE_STREAM_H_
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#define CALL_RTX_RECEIVE_STREAM_H_
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#include <map>
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#include "call/rtp_packet_sink_interface.h"
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namespace webrtc {
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class ReceiveStatistics;
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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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// TODO(nisse): Delete this argument, and
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// corresponding member variable, by moving the
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// responsibility for rtcp feedback to
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// RtpStreamReceiverController.
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ReceiveStatistics* rtp_receive_statistics = nullptr);
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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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ReceiveStatistics* const rtp_receive_statistics_;
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};
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} // namespace webrtc
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#endif // CALL_RTX_RECEIVE_STREAM_H_
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