xref: /aosp_15_r20/external/webrtc/api/test/network_emulation/network_emulation_interfaces.cc (revision d9f758449e529ab9291ac668be2861e7a55c2422)
1 /*
2  *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
3  *
4  *  Use of this source code is governed by a BSD-style license
5  *  that can be found in the LICENSE file in the root of the source
6  *  tree. An additional intellectual property rights grant can be found
7  *  in the file PATENTS.  All contributing project authors may
8  *  be found in the AUTHORS file in the root of the source tree.
9  */
10 #include "api/test/network_emulation/network_emulation_interfaces.h"
11 
12 #include "rtc_base/net_helper.h"
13 
14 namespace webrtc {
15 
EmulatedIpPacket(const rtc::SocketAddress & from,const rtc::SocketAddress & to,rtc::CopyOnWriteBuffer data,Timestamp arrival_time,uint16_t application_overhead)16 EmulatedIpPacket::EmulatedIpPacket(const rtc::SocketAddress& from,
17                                    const rtc::SocketAddress& to,
18                                    rtc::CopyOnWriteBuffer data,
19                                    Timestamp arrival_time,
20                                    uint16_t application_overhead)
21     : from(from),
22       to(to),
23       data(data),
24       headers_size(to.ipaddr().overhead() + application_overhead +
25                    cricket::kUdpHeaderSize),
26       arrival_time(arrival_time) {
27   RTC_DCHECK(to.family() == AF_INET || to.family() == AF_INET6);
28 }
29 
AverageSendRate() const30 DataRate EmulatedNetworkOutgoingStats::AverageSendRate() const {
31   RTC_DCHECK_GE(packets_sent, 2);
32   RTC_DCHECK(first_packet_sent_time.IsFinite());
33   RTC_DCHECK(last_packet_sent_time.IsFinite());
34   return (bytes_sent - first_sent_packet_size) /
35          (last_packet_sent_time - first_packet_sent_time);
36 }
37 
AverageReceiveRate() const38 DataRate EmulatedNetworkIncomingStats::AverageReceiveRate() const {
39   RTC_DCHECK_GE(packets_received, 2);
40   RTC_DCHECK(first_packet_received_time.IsFinite());
41   RTC_DCHECK(last_packet_received_time.IsFinite());
42   return (bytes_received - first_received_packet_size) /
43          (last_packet_received_time - first_packet_received_time);
44 }
45 
46 }  // namespace webrtc
47