/* * 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 #include #include #include #include #include "api/rtp_headers.h" #include "api/units/timestamp.h" namespace webrtc { RtpPacketInfo::RtpPacketInfo() : ssrc_(0), rtp_timestamp_(0), receive_time_(Timestamp::MinusInfinity()) {} RtpPacketInfo::RtpPacketInfo(uint32_t ssrc, std::vector 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(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