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d2a6686a10
This change adds classes so that we later can plumb information about received packets to each audio and video frame. It's not wired up to do anything yet. Bug: webrtc:10668 Change-Id: I962df493a76692f668314f78d6792d7636c5a31b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138203 Commit-Queue: Chen Xing <chxg@google.com> Reviewed-by: Minyue Li <minyue@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28138}
84 lines
2.7 KiB
C++
84 lines
2.7 KiB
C++
/*
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* Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef API_RTP_PACKET_INFO_H_
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#define API_RTP_PACKET_INFO_H_
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#include <cstdint>
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#include <utility>
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#include <vector>
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#include "absl/types/optional.h"
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#include "api/rtp_headers.h"
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namespace webrtc {
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// Structure to hold information about a received |RtpPacket|.
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class RtpPacketInfo {
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public:
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RtpPacketInfo();
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RtpPacketInfo(uint32_t ssrc,
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std::vector<uint32_t> csrcs,
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uint16_t sequence_number,
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uint32_t rtp_timestamp,
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absl::optional<uint8_t> audio_level,
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int64_t receive_time_ms);
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RtpPacketInfo(const RTPHeader& rtp_header, int64_t receive_time_ms);
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RtpPacketInfo(const RtpPacketInfo& other) = default;
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RtpPacketInfo(RtpPacketInfo&& other) = default;
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RtpPacketInfo& operator=(const RtpPacketInfo& other) = default;
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RtpPacketInfo& operator=(RtpPacketInfo&& other) = default;
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uint32_t ssrc() const { return ssrc_; }
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void set_ssrc(uint32_t value) { ssrc_ = value; }
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const std::vector<uint32_t>& csrcs() const { return csrcs_; }
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void set_csrcs(std::vector<uint32_t> value) { csrcs_ = std::move(value); }
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uint16_t sequence_number() const { return sequence_number_; }
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void set_sequence_number(uint16_t value) { sequence_number_ = value; }
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uint32_t rtp_timestamp() const { return rtp_timestamp_; }
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void set_rtp_timestamp(uint32_t value) { rtp_timestamp_ = value; }
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absl::optional<uint8_t> audio_level() const { return audio_level_; }
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void set_audio_level(absl::optional<uint8_t> value) { audio_level_ = value; }
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int64_t receive_time_ms() const { return receive_time_ms_; }
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void set_receive_time_ms(int64_t value) { receive_time_ms_ = value; }
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private:
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// Fields from the RTP header:
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// https://tools.ietf.org/html/rfc3550#section-5.1
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uint32_t ssrc_;
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std::vector<uint32_t> csrcs_;
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uint16_t sequence_number_;
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uint32_t rtp_timestamp_;
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// Fields from the Audio Level header extension:
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// https://tools.ietf.org/html/rfc6464#section-3
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absl::optional<uint8_t> audio_level_;
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// Local |webrtc::Clock|-based timestamp of when the packet was received.
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int64_t receive_time_ms_;
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};
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bool operator==(const RtpPacketInfo& lhs, const RtpPacketInfo& rhs);
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inline bool operator!=(const RtpPacketInfo& lhs, const RtpPacketInfo& rhs) {
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return !(lhs == rhs);
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}
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} // namespace webrtc
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#endif // API_RTP_PACKET_INFO_H_
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