blob: 75b09c4646817fdaa0c3cf8d4281ef6311922cfa [file] [log] [blame] [edit]
/*
* Copyright 2018 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 RTC_BASE_NETWORK_SENT_PACKET_H_
#define RTC_BASE_NETWORK_SENT_PACKET_H_
#include <stddef.h>
#include <stdint.h>
#include <optional>
#include "rtc_base/system/rtc_export.h"
namespace rtc {
enum class PacketType {
kUnknown,
kData,
kIceConnectivityCheck,
kIceConnectivityCheckResponse,
kStunMessage,
kTurnMessage,
};
enum class PacketInfoProtocolType {
kUnknown,
kUdp,
kTcp,
kSsltcp,
kTls,
};
struct RTC_EXPORT PacketInfo {
PacketInfo();
PacketInfo(const PacketInfo& info);
~PacketInfo();
bool included_in_feedback = false;
bool included_in_allocation = false;
// `is_media` is true if this is an audio or video packet, excluding
// retransmissions.
bool is_media = false;
PacketType packet_type = PacketType::kUnknown;
PacketInfoProtocolType protocol = PacketInfoProtocolType::kUnknown;
// A unique id assigned by the network manager, and std::nullopt if not set.
std::optional<uint16_t> network_id;
size_t packet_size_bytes = 0;
size_t turn_overhead_bytes = 0;
size_t ip_overhead_bytes = 0;
};
struct RTC_EXPORT SentPacket {
SentPacket();
SentPacket(int64_t packet_id, int64_t send_time_ms);
SentPacket(int64_t packet_id,
int64_t send_time_ms,
const rtc::PacketInfo& info);
int64_t packet_id = -1;
int64_t send_time_ms = -1;
rtc::PacketInfo info;
};
} // namespace rtc
#endif // RTC_BASE_NETWORK_SENT_PACKET_H_