1 files changed, 46 insertions, 0 deletions
diff --git a/third_party/libwebrtc/api/test/network_emulation/network_emulation_interfaces.cc b/third_party/libwebrtc/api/test/network_emulation/network_emulation_interfaces.cc
new file mode 100644
index 0000000000..0f3a7f8ffd
--- /dev/null
+++ b/third_party/libwebrtc/api/test/network_emulation/network_emulation_interfaces.cc
@@ -0,0 +1,46 @@
+/*
+ *  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