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92ea95e34a
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
29 lines
788 B
C++
29 lines
788 B
C++
/*
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* Copyright 2015 The WebRTC Project Authors. All rights reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "rtc_base/asyncpacketsocket.h"
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namespace rtc {
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PacketTimeUpdateParams::PacketTimeUpdateParams()
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: rtp_sendtime_extension_id(-1),
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srtp_auth_tag_len(-1),
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srtp_packet_index(-1) {
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}
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PacketTimeUpdateParams::~PacketTimeUpdateParams() = default;
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AsyncPacketSocket::AsyncPacketSocket() {
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}
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AsyncPacketSocket::~AsyncPacketSocket() {
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}
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}; // namespace rtc
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