blob: 0f3a7f8ffd31fd94dd978da496eeab738d25cea1 [file] [log] [blame]
/*
* Copyright (c) 2019 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 "api/test/network_emulation/network_emulation_interfaces.h"
#include "rtc_base/net_helper.h"
namespace webrtc {
EmulatedIpPacket::EmulatedIpPacket(const rtc::SocketAddress& from,
const rtc::SocketAddress& to,
rtc::CopyOnWriteBuffer data,
Timestamp arrival_time,
uint16_t application_overhead)
: from(from),
to(to),
data(data),
headers_size(to.ipaddr().overhead() + application_overhead +
cricket::kUdpHeaderSize),
arrival_time(arrival_time) {
RTC_DCHECK(to.family() == AF_INET || to.family() == AF_INET6);
}
DataRate EmulatedNetworkOutgoingStats::AverageSendRate() const {
RTC_DCHECK_GE(packets_sent, 2);
RTC_DCHECK(first_packet_sent_time.IsFinite());
RTC_DCHECK(last_packet_sent_time.IsFinite());
return (bytes_sent - first_sent_packet_size) /
(last_packet_sent_time - first_packet_sent_time);
}
DataRate EmulatedNetworkIncomingStats::AverageReceiveRate() const {
RTC_DCHECK_GE(packets_received, 2);
RTC_DCHECK(first_packet_received_time.IsFinite());
RTC_DCHECK(last_packet_received_time.IsFinite());
return (bytes_received - first_received_packet_size) /
(last_packet_received_time - first_packet_received_time);
}
} // namespace webrtc