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dfd7f3074e
Bug: None Change-Id: I3618903ebd6d6888c2a9548dc97761d0d2e74d0a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140040 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28207}
38 lines
1.2 KiB
C++
38 lines
1.2 KiB
C++
/*
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* 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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#include "api/media_transport_config.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/strings/string_builder.h"
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namespace webrtc {
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MediaTransportConfig::MediaTransportConfig(
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MediaTransportInterface* media_transport)
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: media_transport(media_transport) {
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RTC_DCHECK(media_transport != nullptr);
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}
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MediaTransportConfig::MediaTransportConfig(size_t rtp_max_packet_size)
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: rtp_max_packet_size(rtp_max_packet_size) {
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RTC_DCHECK_GT(rtp_max_packet_size, 0);
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}
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std::string MediaTransportConfig::DebugString()
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const { // TODO(sukhanov): Add rtp_max_packet_size (requires fixing
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// audio_send/receive_stream_unittest.cc).
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rtc::StringBuilder result;
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result << "{media_transport: "
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<< (media_transport != nullptr ? "(Transport)" : "null") << "}";
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return result.Release();
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}
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} // namespace webrtc
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