| /* |
| * Copyright (c) 2017 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 "modules/rtp_rtcp/include/rtp_rtcp_defines.h" |
| |
| namespace webrtc { |
| |
| PayloadUnion::PayloadUnion(const AudioPayload& payload) |
| : audio_payload_(payload) {} |
| PayloadUnion::PayloadUnion(const VideoPayload& payload) |
| : video_payload_(payload) {} |
| PayloadUnion::PayloadUnion(const PayloadUnion&) = default; |
| PayloadUnion::PayloadUnion(PayloadUnion&&) = default; |
| PayloadUnion::~PayloadUnion() = default; |
| |
| PayloadUnion& PayloadUnion::operator=(const PayloadUnion&) = default; |
| PayloadUnion& PayloadUnion::operator=(PayloadUnion&&) = default; |
| |
| } // namespace webrtc |
