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7a9a092708
MediaTransport is deprecated and the code is unused. No-Try: True Bug: webrtc:9719 Change-Id: I5b864c1e74bf04df16c15f51b8fac3d407331dcd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160620 Commit-Queue: Bjorn Mellem <mellem@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org> Reviewed-by: Åsa Persson <asapersson@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29923}
38 lines
1.2 KiB
C++
38 lines
1.2 KiB
C++
/* Copyright 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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#ifndef API_TRANSPORT_MEDIA_MEDIA_TRANSPORT_CONFIG_H_
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#define API_TRANSPORT_MEDIA_MEDIA_TRANSPORT_CONFIG_H_
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#include <memory>
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#include <string>
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#include <utility>
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#include "absl/types/optional.h"
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namespace webrtc {
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// Media transport config is made available to both transport and audio / video
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// layers, but access to individual interfaces should not be open without
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// necessity.
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struct MediaTransportConfig {
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// Default constructor for no-media transport scenarios.
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MediaTransportConfig() = default;
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// Constructor for datagram transport scenarios.
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explicit MediaTransportConfig(size_t rtp_max_packet_size);
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std::string DebugString() const;
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// If provided, limits RTP packet size (excludes ICE, IP or network overhead).
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absl::optional<size_t> rtp_max_packet_size;
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};
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} // namespace webrtc
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#endif // API_TRANSPORT_MEDIA_MEDIA_TRANSPORT_CONFIG_H_
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