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_ms_(-1) {}
20
RtpPacketInfo(uint32_t ssrc,std::vector<uint32_t> csrcs,uint32_t rtp_timestamp,absl::optional<uint8_t> audio_level,absl::optional<AbsoluteCaptureTime> absolute_capture_time,int64_t receive_time_ms)21 RtpPacketInfo::RtpPacketInfo(
22 uint32_t ssrc,
23 std::vector<uint32_t> csrcs,
24 uint32_t rtp_timestamp,
25 absl::optional<uint8_t> audio_level,
26 absl::optional<AbsoluteCaptureTime> absolute_capture_time,
27 int64_t receive_time_ms)
28 : ssrc_(ssrc),
29 csrcs_(std::move(csrcs)),
30 rtp_timestamp_(rtp_timestamp),
31 audio_level_(audio_level),
32 absolute_capture_time_(absolute_capture_time),
33 receive_time_ms_(receive_time_ms) {}
34
RtpPacketInfo(const RTPHeader & rtp_header,int64_t receive_time_ms)35 RtpPacketInfo::RtpPacketInfo(const RTPHeader& rtp_header,
36 int64_t receive_time_ms)
37 : ssrc_(rtp_header.ssrc),
38 rtp_timestamp_(rtp_header.timestamp),
39 receive_time_ms_(receive_time_ms) {
40 const auto& extension = rtp_header.extension;
41 const auto csrcs_count = std::min<size_t>(rtp_header.numCSRCs, kRtpCsrcSize);
42
43 csrcs_.assign(&rtp_header.arrOfCSRCs[0], &rtp_header.arrOfCSRCs[csrcs_count]);
44
45 if (extension.hasAudioLevel) {
46 audio_level_ = extension.audioLevel;
47 }
48
49 absolute_capture_time_ = extension.absolute_capture_time;
50 }
51
operator ==(const RtpPacketInfo & lhs,const RtpPacketInfo & rhs)52 bool operator==(const RtpPacketInfo& lhs, const RtpPacketInfo& rhs) {
53 return (lhs.ssrc() == rhs.ssrc()) && (lhs.csrcs() == rhs.csrcs()) &&
54 (lhs.rtp_timestamp() == rhs.rtp_timestamp()) &&
55 (lhs.audio_level() == rhs.audio_level()) &&
56 (lhs.absolute_capture_time() == rhs.absolute_capture_time()) &&
57 (lhs.receive_time_ms() == rhs.receive_time_ms());
58 }
59
60 } // namespace webrtc
61