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* Copyright (c) 2017 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.
#include <stddef.h>
#include <stdint.h>
#include <string>
#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/units/timestamp.h"
#include "api/video/color_space.h"
#include "api/video/video_content_type.h"
#include "api/video/video_rotation.h"
#include "api/video/video_timing.h"
namespace webrtc {
struct FeedbackRequest {
// Determines whether the recv delta as specified in
// should be included.
bool include_timestamps;
// Include feedback of received packets in the range [sequence_number -
// sequence_count + 1, sequence_number]. That is, no feedback will be sent if
// sequence_count is zero.
int sequence_count;
// The Absolute Capture Time extension is used to stamp RTP packets with a NTP
// timestamp showing when the first audio or video frame in a packet was
// originally captured. The intent of this extension is to provide a way to
// accomplish audio-to-video synchronization when RTCP-terminating intermediate
// systems (e.g. mixers) are involved. See:
struct AbsoluteCaptureTime {
// Absolute capture timestamp is the NTP timestamp of when the first frame in
// a packet was originally captured. This timestamp MUST be based on the same
// clock as the clock used to generate NTP timestamps for RTCP sender reports
// on the capture system.
// It’s not always possible to do an NTP clock readout at the exact moment of
// when a media frame is captured. A capture system MAY postpone the readout
// until a more convenient time. A capture system SHOULD have known delays
// (e.g. from hardware buffers) subtracted from the readout to make the final
// timestamp as close to the actual capture time as possible.
// This field is encoded as a 64-bit unsigned fixed-point number with the high
// 32 bits for the timestamp in seconds and low 32 bits for the fractional
// part. This is also known as the UQ32.32 format and is what the RTP
// specification defines as the canonical format to represent NTP timestamps.
uint64_t absolute_capture_timestamp;
// Estimated capture clock offset is the sender’s estimate of the offset
// between its own NTP clock and the capture system’s NTP clock. The sender is
// here defined as the system that owns the NTP clock used to generate the NTP
// timestamps for the RTCP sender reports on this stream. The sender system is
// typically either the capture system or a mixer.
// This field is encoded as a 64-bit two’s complement signed fixed-point
// number with the high 32 bits for the seconds and low 32 bits for the
// fractional part. It’s intended to make it easy for a receiver, that knows
// how to estimate the sender system’s NTP clock, to also estimate the capture
// system’s NTP clock:
// Capture NTP Clock = Sender NTP Clock + Capture Clock Offset
absl::optional<int64_t> estimated_capture_clock_offset;
inline bool operator==(const AbsoluteCaptureTime& lhs,
const AbsoluteCaptureTime& rhs) {
return (lhs.absolute_capture_timestamp == rhs.absolute_capture_timestamp) &&
(lhs.estimated_capture_clock_offset ==
inline bool operator!=(const AbsoluteCaptureTime& lhs,
const AbsoluteCaptureTime& rhs) {
return !(lhs == rhs);
struct RTPHeaderExtension {
RTPHeaderExtension(const RTPHeaderExtension& other);
RTPHeaderExtension& operator=(const RTPHeaderExtension& other);
static constexpr int kAbsSendTimeFraction = 18;
Timestamp GetAbsoluteSendTimestamp() const {
RTC_DCHECK(absoluteSendTime < (1ul << 24));
return Timestamp::Micros((absoluteSendTime * 1000000ll) /
(1 << kAbsSendTimeFraction));
bool hasTransmissionTimeOffset;
int32_t transmissionTimeOffset;
bool hasAbsoluteSendTime;
uint32_t absoluteSendTime;
absl::optional<AbsoluteCaptureTime> absolute_capture_time;
bool hasTransportSequenceNumber;
uint16_t transportSequenceNumber;
absl::optional<FeedbackRequest> feedback_request;
// Audio Level includes both level in dBov and voiced/unvoiced bit. See:
bool hasAudioLevel;
bool voiceActivity;
uint8_t audioLevel;
// For Coordination of Video Orientation. See
// ts_126114v120700p.pdf
bool hasVideoRotation;
VideoRotation videoRotation;
// TODO(ilnik): Refactor this and one above to be absl::optional() and remove
// a corresponding bool flag.
bool hasVideoContentType;
VideoContentType videoContentType;
bool has_video_timing;
VideoSendTiming video_timing;
VideoPlayoutDelay playout_delay;
// For identification of a stream when ssrc is not signaled. See
std::string stream_id;
std::string repaired_stream_id;
// For identifying the media section used to interpret this RTP packet. See
std::string mid;
absl::optional<ColorSpace> color_space;
enum { kRtpCsrcSize = 15 }; // RFC 3550 page 13
struct RTC_EXPORT RTPHeader {
RTPHeader(const RTPHeader& other);
RTPHeader& operator=(const RTPHeader& other);
bool markerBit;
uint8_t payloadType;
uint16_t sequenceNumber;
uint32_t timestamp;
uint32_t ssrc;
uint8_t numCSRCs;
uint32_t arrOfCSRCs[kRtpCsrcSize];
size_t paddingLength;
size_t headerLength;
RTPHeaderExtension extension;
// RTCP mode to use. Compound mode is described by RFC 4585 and reduced-size
// RTCP mode is described by RFC 5506.
enum class RtcpMode { kOff, kCompound, kReducedSize };
enum NetworkState {
} // namespace webrtc
#endif // API_RTP_HEADERS_H_