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media_transport_config.cc
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media_transport_config.cc
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/*
* 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/media_transport_config.h"
#include "rtc_base/checks.h"
#include "rtc_base/strings/string_builder.h"
namespace webrtc {
MediaTransportConfig::MediaTransportConfig(
MediaTransportInterface* media_transport)
: media_transport(media_transport) {
RTC_DCHECK(media_transport != nullptr);
}
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 { // TODO(sukhanov): Add rtp_max_packet_size (requires fixing
// audio_send/receive_stream_unittest.cc).
rtc::StringBuilder result;
result << "{media_transport: "
<< (media_transport != nullptr ? "(Transport)" : "null") << "}";
return result.Release();
}
} // namespace webrtc