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/*
* 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.
*/
#ifndef API_RTP_PACKET_INFO_H_
#define API_RTP_PACKET_INFO_H_
#include <cstdint>
#include <utility>
#include <vector>
#include "absl/types/optional.h"
#include "api/rtp_headers.h"
#include "api/units/timestamp.h"
#include "rtc_base/system/rtc_export.h"
namespace webrtc {
//
// Structure to hold information about a received |RtpPacket|. It is primarily
// used to carry per-packet information from when a packet is received until
// the information is passed to |SourceTracker|.
//
class RTC_EXPORT RtpPacketInfo {
public:
RtpPacketInfo();
RtpPacketInfo(uint32_t ssrc,
std::vector<uint32_t> csrcs,
uint32_t rtp_timestamp,
absl::optional<uint8_t> audio_level,
absl::optional<AbsoluteCaptureTime> absolute_capture_time,
Timestamp receive_time);
RtpPacketInfo(const RTPHeader& rtp_header, Timestamp receive_time);
// TODO(bugs.webrtc.org/12722): Deprecated, remove once downstream projects
// are updated.
RtpPacketInfo(uint32_t ssrc,
std::vector<uint32_t> csrcs,
uint32_t rtp_timestamp,
absl::optional<uint8_t> audio_level,
absl::optional<AbsoluteCaptureTime> absolute_capture_time,
int64_t receive_time_ms);
RtpPacketInfo(const RTPHeader& rtp_header, int64_t receive_time_ms);
RtpPacketInfo(const RtpPacketInfo& other) = default;
RtpPacketInfo(RtpPacketInfo&& other) = default;
RtpPacketInfo& operator=(const RtpPacketInfo& other) = default;
RtpPacketInfo& operator=(RtpPacketInfo&& other) = default;
uint32_t ssrc() const { return ssrc_; }
void set_ssrc(uint32_t value) { ssrc_ = value; }
const std::vector<uint32_t>& csrcs() const { return csrcs_; }
void set_csrcs(std::vector<uint32_t> value) { csrcs_ = std::move(value); }
uint32_t rtp_timestamp() const { return rtp_timestamp_; }
void set_rtp_timestamp(uint32_t value) { rtp_timestamp_ = value; }
absl::optional<uint8_t> audio_level() const { return audio_level_; }
void set_audio_level(absl::optional<uint8_t> value) { audio_level_ = value; }
const absl::optional<AbsoluteCaptureTime>& absolute_capture_time() const {
return absolute_capture_time_;
}
void set_absolute_capture_time(
const absl::optional<AbsoluteCaptureTime>& value) {
absolute_capture_time_ = value;
}
const absl::optional<int64_t>& local_capture_clock_offset() const {
return local_capture_clock_offset_;
}
void set_local_capture_clock_offset(const absl::optional<int64_t>& value) {
local_capture_clock_offset_ = value;
}
Timestamp receive_time() const { return receive_time_; }
void set_receive_time(Timestamp value) { receive_time_ = value; }
// TODO(bugs.webrtc.org/12722): Deprecated, remove once downstream projects
// are updated.
int64_t receive_time_ms() const { return receive_time_.ms(); }
private:
// Fields from the RTP header:
// https://tools.ietf.org/html/rfc3550#section-5.1
uint32_t ssrc_;
std::vector<uint32_t> csrcs_;
uint32_t rtp_timestamp_;
// Fields from the Audio Level header extension:
// https://tools.ietf.org/html/rfc6464#section-3
absl::optional<uint8_t> audio_level_;
// Fields from the Absolute Capture Time header extension:
// http://www.webrtc.org/experiments/rtp-hdrext/abs-capture-time
// To not be confused with |local_capture_clock_offset_|, the
// |estimated_capture_clock_offset| in |absolute_capture_time_| should
// represent the clock offset between a remote sender and the capturer, and
// thus equals to the corresponding values in the received RTP packets,
// subjected to possible interpolations.
absl::optional<AbsoluteCaptureTime> absolute_capture_time_;
// Clock offset against capturer's clock. Should be derived from the estimated
// capture clock offset defined in the Absolute Capture Time header extension.
absl::optional<int64_t> local_capture_clock_offset_;
// Local |webrtc::Clock|-based timestamp of when the packet was received.
Timestamp receive_time_;
};
bool operator==(const RtpPacketInfo& lhs, const RtpPacketInfo& rhs);
inline bool operator!=(const RtpPacketInfo& lhs, const RtpPacketInfo& rhs) {
return !(lhs == rhs);
}
} // namespace webrtc
#endif // API_RTP_PACKET_INFO_H_
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