xref: /aosp_15_r20/external/webrtc/api/rtp_packet_info.cc (revision d9f758449e529ab9291ac668be2861e7a55c2422)
1 /*
2  *  Copyright (c) 2019 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 
11 #include "api/rtp_packet_info.h"
12 
13 #include <algorithm>
14 #include <utility>
15 
16 namespace webrtc {
17 
RtpPacketInfo()18 RtpPacketInfo::RtpPacketInfo()
19     : ssrc_(0), rtp_timestamp_(0), receive_time_(Timestamp::MinusInfinity()) {}
20 
RtpPacketInfo(uint32_t ssrc,std::vector<uint32_t> csrcs,uint32_t rtp_timestamp,Timestamp receive_time)21 RtpPacketInfo::RtpPacketInfo(uint32_t ssrc,
22                              std::vector<uint32_t> csrcs,
23                              uint32_t rtp_timestamp,
24                              Timestamp receive_time)
25     : ssrc_(ssrc),
26       csrcs_(std::move(csrcs)),
27       rtp_timestamp_(rtp_timestamp),
28       receive_time_(receive_time) {}
29 
RtpPacketInfo(const RTPHeader & rtp_header,Timestamp receive_time)30 RtpPacketInfo::RtpPacketInfo(const RTPHeader& rtp_header,
31                              Timestamp receive_time)
32     : ssrc_(rtp_header.ssrc),
33       rtp_timestamp_(rtp_header.timestamp),
34       receive_time_(receive_time) {
35   const auto& extension = rtp_header.extension;
36   const auto csrcs_count = std::min<size_t>(rtp_header.numCSRCs, kRtpCsrcSize);
37 
38   csrcs_.assign(&rtp_header.arrOfCSRCs[0], &rtp_header.arrOfCSRCs[csrcs_count]);
39 
40   if (extension.hasAudioLevel) {
41     audio_level_ = extension.audioLevel;
42   }
43 
44   absolute_capture_time_ = extension.absolute_capture_time;
45 }
46 
operator ==(const RtpPacketInfo & lhs,const RtpPacketInfo & rhs)47 bool operator==(const RtpPacketInfo& lhs, const RtpPacketInfo& rhs) {
48   return (lhs.ssrc() == rhs.ssrc()) && (lhs.csrcs() == rhs.csrcs()) &&
49          (lhs.rtp_timestamp() == rhs.rtp_timestamp()) &&
50          (lhs.receive_time() == rhs.receive_time()) &&
51          (lhs.audio_level() == rhs.audio_level()) &&
52          (lhs.absolute_capture_time() == rhs.absolute_capture_time()) &&
53          (lhs.local_capture_clock_offset() == rhs.local_capture_clock_offset());
54 }
55 
56 }  // namespace webrtc
57