rtp_packet_info.h (3897B)
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 #ifndef API_RTP_PACKET_INFO_H_ 12 #define API_RTP_PACKET_INFO_H_ 13 14 #include <cstdint> 15 #include <optional> 16 #include <utility> 17 #include <vector> 18 19 #include "api/rtp_headers.h" 20 #include "api/units/time_delta.h" 21 #include "api/units/timestamp.h" 22 #include "rtc_base/system/rtc_export.h" 23 24 namespace webrtc { 25 26 // 27 // Structure to hold information about a received `RtpPacket`. It is primarily 28 // used to carry per-packet information from when a packet is received until 29 // the information is passed to `SourceTracker`. 30 // 31 class RTC_EXPORT RtpPacketInfo { 32 public: 33 RtpPacketInfo(); 34 35 RtpPacketInfo(uint32_t ssrc, 36 std::vector<uint32_t> csrcs, 37 uint32_t rtp_timestamp, 38 Timestamp receive_time); 39 40 RtpPacketInfo(const RTPHeader& rtp_header, Timestamp receive_time); 41 42 RtpPacketInfo(const RtpPacketInfo& other) = default; 43 RtpPacketInfo(RtpPacketInfo&& other) = default; 44 RtpPacketInfo& operator=(const RtpPacketInfo& other) = default; 45 RtpPacketInfo& operator=(RtpPacketInfo&& other) = default; 46 47 uint32_t ssrc() const { return ssrc_; } 48 void set_ssrc(uint32_t value) { ssrc_ = value; } 49 50 const std::vector<uint32_t>& csrcs() const { return csrcs_; } 51 void set_csrcs(std::vector<uint32_t> value) { csrcs_ = std::move(value); } 52 53 uint32_t rtp_timestamp() const { return rtp_timestamp_; } 54 void set_rtp_timestamp(uint32_t value) { rtp_timestamp_ = value; } 55 56 Timestamp receive_time() const { return receive_time_; } 57 void set_receive_time(Timestamp value) { receive_time_ = value; } 58 59 std::optional<uint8_t> audio_level() const { return audio_level_; } 60 RtpPacketInfo& set_audio_level(std::optional<uint8_t> value) { 61 audio_level_ = value; 62 return *this; 63 } 64 65 const std::optional<AbsoluteCaptureTime>& absolute_capture_time() const { 66 return absolute_capture_time_; 67 } 68 RtpPacketInfo& set_absolute_capture_time( 69 const std::optional<AbsoluteCaptureTime>& value) { 70 absolute_capture_time_ = value; 71 return *this; 72 } 73 74 const std::optional<TimeDelta>& local_capture_clock_offset() const { 75 return local_capture_clock_offset_; 76 } 77 RtpPacketInfo& set_local_capture_clock_offset( 78 std::optional<TimeDelta> value) { 79 local_capture_clock_offset_ = value; 80 return *this; 81 } 82 83 private: 84 // Fields from the RTP header: 85 // https://tools.ietf.org/html/rfc3550#section-5.1 86 uint32_t ssrc_; 87 std::vector<uint32_t> csrcs_; 88 uint32_t rtp_timestamp_; 89 90 // Local `Clock`-based timestamp of when the packet was received. 91 Timestamp receive_time_; 92 93 // Fields from the Audio Level header extension: 94 // https://tools.ietf.org/html/rfc6464#section-3 95 std::optional<uint8_t> audio_level_; 96 97 // Fields from the Absolute Capture Time header extension: 98 // http://www.webrtc.org/experiments/rtp-hdrext/abs-capture-time 99 std::optional<AbsoluteCaptureTime> absolute_capture_time_; 100 101 // Clock offset between the local clock and the capturer's clock. 102 // Do not confuse with `AbsoluteCaptureTime::estimated_capture_clock_offset` 103 // which instead represents the clock offset between a remote sender and the 104 // capturer. The following holds: 105 // Capture's NTP Clock = Local NTP Clock + Local-Capture Clock Offset 106 std::optional<TimeDelta> local_capture_clock_offset_; 107 }; 108 109 bool operator==(const RtpPacketInfo& lhs, const RtpPacketInfo& rhs); 110 111 inline bool operator!=(const RtpPacketInfo& lhs, const RtpPacketInfo& rhs) { 112 return !(lhs == rhs); 113 } 114 115 } // namespace webrtc 116 117 #endif // API_RTP_PACKET_INFO_H_