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988cc0870b
This CL is the result of running include-what-you-use tool on part of the code base (audio target and dependencies) plus manual fixes. bug: webrtc:8311 Change-Id: I277d281ce943c3ecc1bd45fd8d83055931743604 Reviewed-on: https://webrtc-review.googlesource.com/c/106280 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25311}
31 lines
1.1 KiB
C++
31 lines
1.1 KiB
C++
/*
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* Copyright (c) 2018 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 "modules/rtp_rtcp/source/rtp_packet_to_send.h"
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#include <cstdint>
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namespace webrtc {
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RtpPacketToSend::RtpPacketToSend(const ExtensionManager* extensions)
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: RtpPacket(extensions) {}
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RtpPacketToSend::RtpPacketToSend(const ExtensionManager* extensions,
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size_t capacity)
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: RtpPacket(extensions, capacity) {}
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RtpPacketToSend::RtpPacketToSend(const RtpPacketToSend& packet) = default;
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RtpPacketToSend::RtpPacketToSend(RtpPacketToSend&& packet) = default;
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RtpPacketToSend& RtpPacketToSend::operator=(const RtpPacketToSend& packet) =
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default;
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RtpPacketToSend& RtpPacketToSend::operator=(RtpPacketToSend&& packet) = default;
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RtpPacketToSend::~RtpPacketToSend() = default;
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} // namespace webrtc
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