| /* |
| * Copyright (c) 2019 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #include "api/rtp_packet_info.h" |
| |
| #include <algorithm> |
| #include <utility> |
| |
| namespace webrtc { |
| |
| RtpPacketInfo::RtpPacketInfo() |
| : ssrc_(0), rtp_timestamp_(0), receive_time_(Timestamp::MinusInfinity()) {} |
| |
| RtpPacketInfo::RtpPacketInfo(uint32_t ssrc, |
| std::vector<uint32_t> csrcs, |
| uint32_t rtp_timestamp, |
| Timestamp receive_time) |
| : ssrc_(ssrc), |
| csrcs_(std::move(csrcs)), |
| rtp_timestamp_(rtp_timestamp), |
| receive_time_(receive_time) {} |
| |
| RtpPacketInfo::RtpPacketInfo(const RTPHeader& rtp_header, |
| Timestamp receive_time) |
| : ssrc_(rtp_header.ssrc), |
| rtp_timestamp_(rtp_header.timestamp), |
| receive_time_(receive_time) { |
| const auto& extension = rtp_header.extension; |
| const auto csrcs_count = std::min<size_t>(rtp_header.numCSRCs, kRtpCsrcSize); |
| |
| csrcs_.assign(&rtp_header.arrOfCSRCs[0], &rtp_header.arrOfCSRCs[csrcs_count]); |
| |
| if (extension.audio_level()) { |
| audio_level_ = extension.audio_level()->level(); |
| } |
| |
| absolute_capture_time_ = extension.absolute_capture_time; |
| } |
| |
| bool operator==(const RtpPacketInfo& lhs, const RtpPacketInfo& rhs) { |
| return (lhs.ssrc() == rhs.ssrc()) && (lhs.csrcs() == rhs.csrcs()) && |
| (lhs.rtp_timestamp() == rhs.rtp_timestamp()) && |
| (lhs.receive_time() == rhs.receive_time()) && |
| (lhs.audio_level() == rhs.audio_level()) && |
| (lhs.absolute_capture_time() == rhs.absolute_capture_time()) && |
| (lhs.local_capture_clock_offset() == rhs.local_capture_clock_offset()); |
| } |
| |
| } // namespace webrtc |