blob: b9b19cb6f0f1656ad3b566d720dfc1ccf6974159 [file] [log] [blame]
/*
* Copyright 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 "api/transport/media/media_transport_config.h"
#include "rtc_base/checks.h"
#include "rtc_base/strings/string_builder.h"
namespace webrtc {
MediaTransportConfig::MediaTransportConfig(size_t rtp_max_packet_size)
: rtp_max_packet_size(rtp_max_packet_size) {
RTC_DCHECK_GT(rtp_max_packet_size, 0);
}
std::string MediaTransportConfig::DebugString() const {
rtc::StringBuilder result;
result << "{rtp_max_packet_size: " << rtp_max_packet_size.value_or(0) << "}";
return result.Release();
}
} // namespace webrtc