blob: 8997bce0d2fd6a8128db13f90b233132bcdf42f2 [file] [log] [blame]
* Copyright (c) 2016 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 <vector>
#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/video/video_timing.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
#include "modules/rtp_rtcp/source/rtp_packet.h"
namespace webrtc {
// Class to hold rtp packet with metadata for sender side.
class RtpPacketToSend : public RtpPacket {
// RtpPacketToSend::Type is deprecated. Use RtpPacketMediaType directly.
using Type = RtpPacketMediaType;
explicit RtpPacketToSend(const ExtensionManager* extensions);
RtpPacketToSend(const ExtensionManager* extensions, size_t capacity);
RtpPacketToSend(const RtpPacketToSend& packet);
RtpPacketToSend(RtpPacketToSend&& packet);
RtpPacketToSend& operator=(const RtpPacketToSend& packet);
RtpPacketToSend& operator=(RtpPacketToSend&& packet);
// Time in local time base as close as it can to frame capture time.
int64_t capture_time_ms() const { return capture_time_ms_; }
void set_capture_time_ms(int64_t time) { capture_time_ms_ = time; }
void set_packet_type(RtpPacketMediaType type) { packet_type_ = type; }
absl::optional<RtpPacketMediaType> packet_type() const {
return packet_type_;
// If this is a retransmission, indicates the sequence number of the original
// media packet that this packet represents. If RTX is used this will likely
// be different from SequenceNumber().
void set_retransmitted_sequence_number(uint16_t sequence_number) {
retransmitted_sequence_number_ = sequence_number;
absl::optional<uint16_t> retransmitted_sequence_number() {
return retransmitted_sequence_number_;
void set_allow_retransmission(bool allow_retransmission) {
allow_retransmission_ = allow_retransmission;
bool allow_retransmission() { return allow_retransmission_; }
// Additional data bound to the RTP packet for use in application code,
// outside of WebRTC.
rtc::ArrayView<const uint8_t> application_data() const {
return application_data_;
void set_application_data(rtc::ArrayView<const uint8_t> data) {
application_data_.assign(data.begin(), data.end());
void set_packetization_finish_time_ms(int64_t time) {
VideoSendTiming::GetDeltaCappedMs(capture_time_ms_, time),
void set_pacer_exit_time_ms(int64_t time) {
VideoSendTiming::GetDeltaCappedMs(capture_time_ms_, time),
void set_network_time_ms(int64_t time) {
VideoSendTiming::GetDeltaCappedMs(capture_time_ms_, time),
void set_network2_time_ms(int64_t time) {
VideoSendTiming::GetDeltaCappedMs(capture_time_ms_, time),
// Indicates if packet is the first packet of a video frame.
void set_first_packet_of_frame(bool is_first_packet) {
is_first_packet_of_frame_ = is_first_packet;
bool is_first_packet_of_frame() const { return is_first_packet_of_frame_; }
// Indicates if packet contains payload for a video key-frame.
void set_is_key_frame(bool is_key_frame) { is_key_frame_ = is_key_frame; }
bool is_key_frame() const { return is_key_frame_; }
int64_t capture_time_ms_ = 0;
absl::optional<RtpPacketMediaType> packet_type_;
bool allow_retransmission_ = false;
absl::optional<uint16_t> retransmitted_sequence_number_;
std::vector<uint8_t> application_data_;
bool is_first_packet_of_frame_ = false;
bool is_key_frame_ = false;
} // namespace webrtc