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https://github.com/mollyim/webrtc.git
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316f3ac13b
- Implement datagram transport adaptor, which wraps datagram transport in DtlsTransportInternal. Datagram adaptor owns both ICE and Datagram Transports. - Implement setup of datagram transport based on RTCConfiguration flag use_datagram_transport. This is very similar to MediaTransport setup with the exception that we create DTLS datagram adaptor. - Propagate maximum datagram size to video encoder via MediaTransportConfig. TODO: Currently this CL can only be tested in downstream projects. Once we add fake datagram transport, we will be able to implement unit tests similar to loopback media transport. Bug: webrtc:9719 Change-Id: I4fa4a5725598dfee5da4f0f374269a7e289d48ed Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138100 Commit-Queue: Anton Sukhanov <sukhanov@webrtc.org> Reviewed-by: Bjorn Mellem <mellem@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28047}
39 lines
1.3 KiB
C++
39 lines
1.3 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/string_utils.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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