| Chen Xing | d2a6686 | 2019-06-03 14:53:42 +0200 | [diff] [blame] | 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 | |
| 18 | RtpPacketInfo::RtpPacketInfo() |
| 19 | : ssrc_(0), sequence_number_(0), rtp_timestamp_(0), receive_time_ms_(-1) {} |
| 20 | |
| 21 | RtpPacketInfo::RtpPacketInfo(uint32_t ssrc, |
| 22 | std::vector<uint32_t> csrcs, |
| 23 | uint16_t sequence_number, |
| 24 | uint32_t rtp_timestamp, |
| 25 | absl::optional<uint8_t> audio_level, |
| 26 | int64_t receive_time_ms) |
| 27 | : ssrc_(ssrc), |
| 28 | csrcs_(std::move(csrcs)), |
| 29 | sequence_number_(sequence_number), |
| 30 | rtp_timestamp_(rtp_timestamp), |
| 31 | audio_level_(audio_level), |
| 32 | receive_time_ms_(receive_time_ms) {} |
| 33 | |
| 34 | RtpPacketInfo::RtpPacketInfo(const RTPHeader& rtp_header, |
| 35 | int64_t receive_time_ms) |
| 36 | : ssrc_(rtp_header.ssrc), |
| 37 | sequence_number_(rtp_header.sequenceNumber), |
| 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 | |
| 50 | bool operator==(const RtpPacketInfo& lhs, const RtpPacketInfo& rhs) { |
| 51 | return (lhs.ssrc() == rhs.ssrc()) && (lhs.csrcs() == rhs.csrcs()) && |
| 52 | (lhs.sequence_number() == rhs.sequence_number()) && |
| 53 | (lhs.rtp_timestamp() == rhs.rtp_timestamp()) && |
| 54 | (lhs.audio_level() == rhs.audio_level()) && |
| 55 | (lhs.receive_time_ms() == rhs.receive_time_ms()); |
| 56 | } |
| 57 | |
| 58 | } // namespace webrtc |