| /* |
| * 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 |