blob: eff6223686a4a2a1b2b21b41bd701989f0f9e20c [file] [log] [blame]
* 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.h>
#include "api/array_view.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 {
// Class to represent the value of RTP header extensions that are
// variable-length strings (e.g., RtpStreamId and RtpMid).
// Unlike std::string, it can be copied with memcpy and cleared with memset.
// Empty value represents unset header extension (use empty() to query).
class StringRtpHeaderExtension {
// String RTP header extensions are limited to 16 bytes because it is the
// maximum length that can be encoded with one-byte header extensions.
static constexpr size_t kMaxSize = 16;
static bool IsLegalMidName(rtc::ArrayView<const char> name);
static bool IsLegalRsidName(rtc::ArrayView<const char> name);
// TODO( Deprecate and remove when third parties have
// migrated to "IsLegalRsidName".
static bool IsLegalName(rtc::ArrayView<const char> name) {
return IsLegalRsidName(name);
StringRtpHeaderExtension() { value_[0] = 0; }
explicit StringRtpHeaderExtension(rtc::ArrayView<const char> value) {
Set(, value.size());
StringRtpHeaderExtension(const StringRtpHeaderExtension&) = default;
StringRtpHeaderExtension& operator=(const StringRtpHeaderExtension&) =
bool empty() const { return value_[0] == 0; }
const char* data() const { return value_; }
size_t size() const { return strnlen(value_, kMaxSize); }
void Set(rtc::ArrayView<const uint8_t> value) {
Set(reinterpret_cast<const char*>(, value.size());
void Set(const char* data, size_t size);
friend bool operator==(const StringRtpHeaderExtension& lhs,
const StringRtpHeaderExtension& rhs) {
return strncmp(lhs.value_, rhs.value_, kMaxSize) == 0;
friend bool operator!=(const StringRtpHeaderExtension& lhs,
const StringRtpHeaderExtension& rhs) {
return !(lhs == rhs);
char value_[kMaxSize];
// StreamId represents RtpStreamId which is a string.
typedef StringRtpHeaderExtension StreamId;
// Mid represents RtpMid which is a string.
typedef StringRtpHeaderExtension Mid;
struct 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;
// 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;
bool has_frame_marking;
FrameMarking frame_marking;
PlayoutDelay playout_delay = {-1, -1};
// For identification of a stream when ssrc is not signaled. See
// TODO(danilchap): Update url from draft to release version.
StreamId stream_id;
StreamId repaired_stream_id;
// For identifying the media section used to interpret this RTP packet. See
Mid mid;
struct 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 {
struct RtpKeepAliveConfig final {
// If no packet has been sent for |timeout_interval_ms|, send a keep-alive
// packet. The keep-alive packet is an empty (no payload) RTP packet with a
// payload type of 20 as long as the other end has not negotiated the use of
// this value. If this value has already been negotiated, then some other
// unused static payload type from table 5 of RFC 3551 shall be used and set
// in |payload_type|.
int64_t timeout_interval_ms = -1;
uint8_t payload_type = 20;
bool operator==(const RtpKeepAliveConfig& o) const {
return timeout_interval_ms == o.timeout_interval_ms &&
payload_type == o.payload_type;
bool operator!=(const RtpKeepAliveConfig& o) const { return !(*this == o); }
} // namespace webrtc
#endif // API_RTP_HEADERS_H_