mirror of
https://github.com/mollyim/webrtc.git
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3ed1dbb56e
Bug: None Change-Id: I02a43a16e8d9bf3a1e2c9f6442a1c119620e1288 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252286 Auto-Submit: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36067}
77 lines
2.8 KiB
C++
77 lines
2.8 KiB
C++
/*
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* Copyright (c) 2016 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 MODULES_RTP_RTCP_SOURCE_RTP_PACKET_RECEIVED_H_
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#define MODULES_RTP_RTCP_SOURCE_RTP_PACKET_RECEIVED_H_
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#include <stdint.h>
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#include <utility>
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#include "api/array_view.h"
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#include "api/ref_counted_base.h"
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#include "api/rtp_headers.h"
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#include "api/scoped_refptr.h"
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#include "api/units/timestamp.h"
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#include "modules/rtp_rtcp/source/rtp_packet.h"
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namespace webrtc {
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// Class to hold rtp packet with metadata for receiver side.
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// The metadata is not parsed from the rtp packet, but may be derived from the
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// data that is parsed from the rtp packet.
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class RtpPacketReceived : public RtpPacket {
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public:
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RtpPacketReceived();
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explicit RtpPacketReceived(
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const ExtensionManager* extensions,
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webrtc::Timestamp arrival_time = webrtc::Timestamp::MinusInfinity());
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RtpPacketReceived(const RtpPacketReceived& packet);
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RtpPacketReceived(RtpPacketReceived&& packet);
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RtpPacketReceived& operator=(const RtpPacketReceived& packet);
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RtpPacketReceived& operator=(RtpPacketReceived&& packet);
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~RtpPacketReceived();
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// TODO(danilchap): Remove this function when all code update to use RtpPacket
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// directly. Function is there just for easier backward compatibilty.
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void GetHeader(RTPHeader* header) const;
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// Time in local time base as close as it can to packet arrived on the
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// network.
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webrtc::Timestamp arrival_time() const { return arrival_time_; }
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void set_arrival_time(webrtc::Timestamp time) { arrival_time_ = time; }
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// Flag if packet was recovered via RTX or FEC.
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bool recovered() const { return recovered_; }
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void set_recovered(bool value) { recovered_ = value; }
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int payload_type_frequency() const { return payload_type_frequency_; }
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void set_payload_type_frequency(int value) {
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payload_type_frequency_ = value;
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}
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// An application can attach arbitrary data to an RTP packet using
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// `additional_data`. The additional data does not affect WebRTC processing.
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rtc::scoped_refptr<rtc::RefCountedBase> additional_data() const {
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return additional_data_;
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}
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void set_additional_data(rtc::scoped_refptr<rtc::RefCountedBase> data) {
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additional_data_ = std::move(data);
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}
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private:
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webrtc::Timestamp arrival_time_ = Timestamp::MinusInfinity();
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int payload_type_frequency_ = 0;
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bool recovered_ = false;
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rtc::scoped_refptr<rtc::RefCountedBase> additional_data_;
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};
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} // namespace webrtc
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#endif // MODULES_RTP_RTCP_SOURCE_RTP_PACKET_RECEIVED_H_
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