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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
commit | 6bf0a5cb5034a7e684dcc3500e841785237ce2dd (patch) | |
tree | a68f146d7fa01f0134297619fbe7e33db084e0aa /third_party/libwebrtc/api/rtp_packet_info.h | |
parent | Initial commit. (diff) | |
download | thunderbird-upstream.tar.xz thunderbird-upstream.zip |
Adding upstream version 1:115.7.0.upstream/1%115.7.0upstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to '')
-rw-r--r-- | third_party/libwebrtc/api/rtp_packet_info.h | 117 |
1 files changed, 117 insertions, 0 deletions
diff --git a/third_party/libwebrtc/api/rtp_packet_info.h b/third_party/libwebrtc/api/rtp_packet_info.h new file mode 100644 index 0000000000..8df12a36cf --- /dev/null +++ b/third_party/libwebrtc/api/rtp_packet_info.h @@ -0,0 +1,117 @@ +/* + * 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/time_delta.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, + Timestamp receive_time); + + RtpPacketInfo(const RTPHeader& rtp_header, Timestamp receive_time); + + 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; } + + Timestamp receive_time() const { return receive_time_; } + void set_receive_time(Timestamp value) { receive_time_ = value; } + + absl::optional<uint8_t> audio_level() const { return audio_level_; } + RtpPacketInfo& set_audio_level(absl::optional<uint8_t> value) { + audio_level_ = value; + return *this; + } + + const absl::optional<AbsoluteCaptureTime>& absolute_capture_time() const { + return absolute_capture_time_; + } + RtpPacketInfo& set_absolute_capture_time( + const absl::optional<AbsoluteCaptureTime>& value) { + absolute_capture_time_ = value; + return *this; + } + + const absl::optional<TimeDelta>& local_capture_clock_offset() const { + return local_capture_clock_offset_; + } + RtpPacketInfo& set_local_capture_clock_offset( + absl::optional<TimeDelta> value) { + local_capture_clock_offset_ = value; + return *this; + } + + 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_; + + // Local `webrtc::Clock`-based timestamp of when the packet was received. + Timestamp receive_time_; + + // 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 + absl::optional<AbsoluteCaptureTime> absolute_capture_time_; + + // Clock offset between the local clock and the capturer's clock. + // Do not confuse with `AbsoluteCaptureTime::estimated_capture_clock_offset` + // which instead represents the clock offset between a remote sender and the + // capturer. The following holds: + // Capture's NTP Clock = Local NTP Clock + Local-Capture Clock Offset + absl::optional<TimeDelta> local_capture_clock_offset_; +}; + +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_ |