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 <utility> 16 #include <vector> 17 18 #include "absl/types/optional.h" 19 #include "api/rtp_headers.h" 20 #include "rtc_base/system/rtc_export.h" 21 22 namespace webrtc { 23 24 // 25 // Structure to hold information about a received |RtpPacket|. It is primarily 26 // used to carry per-packet information from when a packet is received until 27 // the information is passed to |SourceTracker|. 28 // 29 class RTC_EXPORT RtpPacketInfo { 30 public: 31 RtpPacketInfo(); 32 33 RtpPacketInfo(uint32_t ssrc, 34 std::vector<uint32_t> csrcs, 35 uint32_t rtp_timestamp, 36 absl::optional<uint8_t> audio_level, 37 absl::optional<AbsoluteCaptureTime> absolute_capture_time, 38 int64_t receive_time_ms); 39 40 RtpPacketInfo(const RTPHeader& rtp_header, int64_t receive_time_ms); 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 ssrc()47 uint32_t ssrc() const { return ssrc_; } set_ssrc(uint32_t value)48 void set_ssrc(uint32_t value) { ssrc_ = value; } 49 csrcs()50 const std::vector<uint32_t>& csrcs() const { return csrcs_; } set_csrcs(std::vector<uint32_t> value)51 void set_csrcs(std::vector<uint32_t> value) { csrcs_ = std::move(value); } 52 rtp_timestamp()53 uint32_t rtp_timestamp() const { return rtp_timestamp_; } set_rtp_timestamp(uint32_t value)54 void set_rtp_timestamp(uint32_t value) { rtp_timestamp_ = value; } 55 audio_level()56 absl::optional<uint8_t> audio_level() const { return audio_level_; } set_audio_level(absl::optional<uint8_t> value)57 void set_audio_level(absl::optional<uint8_t> value) { audio_level_ = value; } 58 absolute_capture_time()59 const absl::optional<AbsoluteCaptureTime>& absolute_capture_time() const { 60 return absolute_capture_time_; 61 } set_absolute_capture_time(const absl::optional<AbsoluteCaptureTime> & value)62 void set_absolute_capture_time( 63 const absl::optional<AbsoluteCaptureTime>& value) { 64 absolute_capture_time_ = value; 65 } 66 receive_time_ms()67 int64_t receive_time_ms() const { return receive_time_ms_; } set_receive_time_ms(int64_t value)68 void set_receive_time_ms(int64_t value) { receive_time_ms_ = value; } 69 70 private: 71 // Fields from the RTP header: 72 // https://tools.ietf.org/html/rfc3550#section-5.1 73 uint32_t ssrc_; 74 std::vector<uint32_t> csrcs_; 75 uint32_t rtp_timestamp_; 76 77 // Fields from the Audio Level header extension: 78 // https://tools.ietf.org/html/rfc6464#section-3 79 absl::optional<uint8_t> audio_level_; 80 81 // Fields from the Absolute Capture Time header extension: 82 // http://www.webrtc.org/experiments/rtp-hdrext/abs-capture-time 83 absl::optional<AbsoluteCaptureTime> absolute_capture_time_; 84 85 // Local |webrtc::Clock|-based timestamp of when the packet was received. 86 int64_t receive_time_ms_; 87 }; 88 89 bool operator==(const RtpPacketInfo& lhs, const RtpPacketInfo& rhs); 90 91 inline bool operator!=(const RtpPacketInfo& lhs, const RtpPacketInfo& rhs) { 92 return !(lhs == rhs); 93 } 94 95 } // namespace webrtc 96 97 #endif // API_RTP_PACKET_INFO_H_ 98