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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.
*/
#ifndef API_RTP_HEADERS_H_
#define API_RTP_HEADERS_H_
#include <stddef.h>
#include <stdint.h>
#include <string>
#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/video/color_space.h"
#include "api/video/video_content_type.h"
#include "api/video/video_frame_marking.h"
#include "api/video/video_rotation.h"
#include "api/video/video_timing.h"
#include "common_types.h" // NOLINT(build/include)
namespace webrtc {
struct FeedbackRequest {
// Determines whether the recv delta as specified in
// https://tools.ietf.org/html/draft-holmer-rmcat-transport-wide-cc-extensions-01
// 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;
};
struct RTPHeaderExtension {
RTPHeaderExtension();
RTPHeaderExtension(const RTPHeaderExtension& other);
RTPHeaderExtension& operator=(const RTPHeaderExtension& other);
bool hasTransmissionTimeOffset;
int32_t transmissionTimeOffset;
bool hasAbsoluteSendTime;
uint32_t absoluteSendTime;
bool hasTransportSequenceNumber;
uint16_t transportSequenceNumber;
absl::optional<FeedbackRequest> feedback_request;
// Audio Level includes both level in dBov and voiced/unvoiced bit. See:
// https://datatracker.ietf.org/doc/draft-lennox-avt-rtp-audio-level-exthdr/
bool hasAudioLevel;
bool voiceActivity;
uint8_t audioLevel;
// For Coordination of Video Orientation. See
// http://www.etsi.org/deliver/etsi_ts/126100_126199/126114/12.07.00_60/
// 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;
bool has_frame_marking;
FrameMarking frame_marking;
PlayoutDelay playout_delay = {-1, -1};
// For identification of a stream when ssrc is not signaled. See
// https://tools.ietf.org/html/draft-ietf-avtext-rid-09
// TODO(danilchap): Update url from draft to release version.
std::string stream_id;
std::string repaired_stream_id;
// For identifying the media section used to interpret this RTP packet. See
// https://tools.ietf.org/html/draft-ietf-mmusic-sdp-bundle-negotiation-38
std::string mid;
absl::optional<ColorSpace> color_space;
};
struct RTPHeader {
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;
int payload_type_frequency;
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 {
kNetworkUp,
kNetworkDown,
};
} // namespace webrtc
#endif // API_RTP_HEADERS_H_