xref: /aosp_15_r20/external/webrtc/modules/rtp_rtcp/source/rtp_packet_received.h (revision d9f758449e529ab9291ac668be2861e7a55c2422)
1 /*
2  *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
3  *
4  *  Use of this source code is governed by a BSD-style license
5  *  that can be found in the LICENSE file in the root of the source
6  *  tree. An additional intellectual property rights grant can be found
7  *  in the file PATENTS.  All contributing project authors may
8  *  be found in the AUTHORS file in the root of the source tree.
9  */
10 #ifndef MODULES_RTP_RTCP_SOURCE_RTP_PACKET_RECEIVED_H_
11 #define MODULES_RTP_RTCP_SOURCE_RTP_PACKET_RECEIVED_H_
12 
13 #include <stdint.h>
14 
15 #include <utility>
16 
17 #include "api/array_view.h"
18 #include "api/ref_counted_base.h"
19 #include "api/rtp_headers.h"
20 #include "api/scoped_refptr.h"
21 #include "api/units/timestamp.h"
22 #include "modules/rtp_rtcp/source/rtp_packet.h"
23 
24 namespace webrtc {
25 // Class to hold rtp packet with metadata for receiver side.
26 // The metadata is not parsed from the rtp packet, but may be derived from the
27 // data that is parsed from the rtp packet.
28 class RtpPacketReceived : public RtpPacket {
29  public:
30   RtpPacketReceived();
31   explicit RtpPacketReceived(
32       const ExtensionManager* extensions,
33       webrtc::Timestamp arrival_time = webrtc::Timestamp::MinusInfinity());
34   RtpPacketReceived(const RtpPacketReceived& packet);
35   RtpPacketReceived(RtpPacketReceived&& packet);
36 
37   RtpPacketReceived& operator=(const RtpPacketReceived& packet);
38   RtpPacketReceived& operator=(RtpPacketReceived&& packet);
39 
40   ~RtpPacketReceived();
41 
42   // TODO(danilchap): Remove this function when all code update to use RtpPacket
43   // directly. Function is there just for easier backward compatibilty.
44   void GetHeader(RTPHeader* header) const;
45 
46   // Time in local time base as close as it can to packet arrived on the
47   // network.
arrival_time()48   webrtc::Timestamp arrival_time() const { return arrival_time_; }
set_arrival_time(webrtc::Timestamp time)49   void set_arrival_time(webrtc::Timestamp time) { arrival_time_ = time; }
50 
51   // Flag if packet was recovered via RTX or FEC.
recovered()52   bool recovered() const { return recovered_; }
set_recovered(bool value)53   void set_recovered(bool value) { recovered_ = value; }
54 
payload_type_frequency()55   int payload_type_frequency() const { return payload_type_frequency_; }
set_payload_type_frequency(int value)56   void set_payload_type_frequency(int value) {
57     payload_type_frequency_ = value;
58   }
59 
60   // An application can attach arbitrary data to an RTP packet using
61   // `additional_data`. The additional data does not affect WebRTC processing.
additional_data()62   rtc::scoped_refptr<rtc::RefCountedBase> additional_data() const {
63     return additional_data_;
64   }
set_additional_data(rtc::scoped_refptr<rtc::RefCountedBase> data)65   void set_additional_data(rtc::scoped_refptr<rtc::RefCountedBase> data) {
66     additional_data_ = std::move(data);
67   }
68 
69  private:
70   webrtc::Timestamp arrival_time_ = Timestamp::MinusInfinity();
71   int payload_type_frequency_ = 0;
72   bool recovered_ = false;
73   rtc::scoped_refptr<rtc::RefCountedBase> additional_data_;
74 };
75 
76 }  // namespace webrtc
77 #endif  // MODULES_RTP_RTCP_SOURCE_RTP_PACKET_RECEIVED_H_
78