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Diffstat (limited to 'api/transport/media/media_transport_config.h')
| -rw-r--r-- | api/transport/media/media_transport_config.h | 38 |
1 files changed, 0 insertions, 38 deletions
diff --git a/api/transport/media/media_transport_config.h b/api/transport/media/media_transport_config.h deleted file mode 100644 index 7ef65453ae..0000000000 --- a/api/transport/media/media_transport_config.h +++ /dev/null @@ -1,38 +0,0 @@ -/* 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. - */ -#ifndef API_TRANSPORT_MEDIA_MEDIA_TRANSPORT_CONFIG_H_ -#define API_TRANSPORT_MEDIA_MEDIA_TRANSPORT_CONFIG_H_ - -#include <memory> -#include <string> -#include <utility> - -#include "absl/types/optional.h" - -namespace webrtc { - -// Media transport config is made available to both transport and audio / video -// layers, but access to individual interfaces should not be open without -// necessity. -struct MediaTransportConfig { - // Default constructor for no-media transport scenarios. - MediaTransportConfig() = default; - - // Constructor for datagram transport scenarios. - explicit MediaTransportConfig(size_t rtp_max_packet_size); - - std::string DebugString() const; - - // If provided, limits RTP packet size (excludes ICE, IP or network overhead). - absl::optional<size_t> rtp_max_packet_size; -}; - -} // namespace webrtc - -#endif // API_TRANSPORT_MEDIA_MEDIA_TRANSPORT_CONFIG_H_ |