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/*
* Copyright (c) 2018 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/source/rtp_packet_to_send.h"
#include <cstddef>
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
namespace webrtc {
RtpPacketToSend::RtpPacketToSend(const ExtensionManager* extensions)
: RtpPacket(extensions) {}
RtpPacketToSend::RtpPacketToSend(const ExtensionManager* extensions,
size_t capacity)
: RtpPacket(extensions, capacity) {}
RtpPacketToSend::RtpPacketToSend(const RtpPacketToSend& packet) = default;
RtpPacketToSend::RtpPacketToSend(RtpPacketToSend&& packet) = default;
RtpPacketToSend& RtpPacketToSend::operator=(const RtpPacketToSend& packet) =
default;
RtpPacketToSend& RtpPacketToSend::operator=(RtpPacketToSend&& packet) = default;
RtpPacketToSend::~RtpPacketToSend() = default;
void RtpPacketToSend::set_packet_type(RtpPacketMediaType type) {
if (packet_type_ == RtpPacketMediaType::kAudio) {
original_packet_type_ = OriginalType::kAudio;
} else if (packet_type_ == RtpPacketMediaType::kVideo) {
original_packet_type_ = OriginalType::kVideo;
}
packet_type_ = type;
}
} // namespace webrtc
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