webrtc/rtc_base/asyncpacketsocket.cc
Qingsi Wang 6e641e64b2 Signal detailed packet info for each packet sent.
Per-packet info is now signaled in SentPacket to provide useful stats
for bandwidth consumption and overhead analysis in the network stack.

Bug: webrtc:9103
Change-Id: I2b8f6491567d0fa54cc559fc5a96d7aac7d9565e
Reviewed-on: https://webrtc-review.googlesource.com/66281
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22834}
2018-04-12 04:46:06 +00:00

39 lines
1.4 KiB
C++

/*
* Copyright 2015 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 "rtc_base/asyncpacketsocket.h"
namespace rtc {
PacketTimeUpdateParams::PacketTimeUpdateParams() = default;
PacketTimeUpdateParams::PacketTimeUpdateParams(
const PacketTimeUpdateParams& other) = default;
PacketTimeUpdateParams::~PacketTimeUpdateParams() = default;
PacketOptions::PacketOptions() = default;
PacketOptions::PacketOptions(DiffServCodePoint dscp) : dscp(dscp) {}
PacketOptions::PacketOptions(const PacketOptions& other) = default;
PacketOptions::~PacketOptions() = default;
AsyncPacketSocket::AsyncPacketSocket() = default;
AsyncPacketSocket::~AsyncPacketSocket() = default;
void CopySocketInformationToPacketInfo(size_t packet_size_bytes,
const AsyncPacketSocket& socket_from,
rtc::PacketInfo* info) {
info->packet_size_bytes = packet_size_bytes;
info->local_socket_address = socket_from.GetLocalAddress();
info->remote_socket_address = socket_from.GetRemoteAddress();
}
}; // namespace rtc