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/*
* Copyright (C) 2017 Apple Inc. All rights reserved.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
* are met:
* 1. Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
* 2. Redistributions in binary form must reproduce the above copyright
* notice, this list of conditions and the following disclaimer in the
* documentation and/or other materials provided with the distribution.
*
* THIS SOFTWARE IS PROVIDED BY APPLE INC. AND ITS CONTRIBUTORS ``AS IS''
* AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO,
* THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR
* PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL APPLE INC. OR ITS CONTRIBUTORS
* BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
* CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
* SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
* INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
* CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
* ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF
* THE POSSIBILITY OF SUCH DAMAGE.
*/
#include "config.h"
#include "NetworkRTCSocket.h"
#if USE(LIBWEBRTC)
#include "DataReference.h"
#include "LibWebRTCSocketClient.h"
#include "NetworkRTCProvider.h"
#include <WebCore/SharedBuffer.h>
#include <wtf/Function.h>
namespace WebKit {
NetworkRTCSocket::NetworkRTCSocket(uint64_t identifier, NetworkRTCProvider& rtcProvider)
: m_identifier(identifier)
, m_rtcProvider(rtcProvider)
{
}
void NetworkRTCSocket::sendTo(const IPC::DataReference& data, const RTCNetwork::SocketAddress& socketAddress, int packetID, int rtpSendtimeExtensionID, String srtpAuth, int64_t srtpPacketIndex, int dscp)
{
auto buffer = WebCore::SharedBuffer::create(data.data(), data.size());
rtc::PacketOptions options;
options.packet_id = packetID;
options.packet_time_params.rtp_sendtime_extension_id = rtpSendtimeExtensionID;
options.packet_time_params.srtp_packet_index = srtpPacketIndex;
options.dscp = static_cast<rtc::DiffServCodePoint>(dscp);
if (srtpAuth.utf8().length()) {
options.packet_time_params.srtp_auth_key = std::vector<char>(srtpAuth.utf8().data(), srtpAuth.utf8().data() + srtpAuth.utf8().length());
options.packet_time_params.srtp_auth_tag_len = srtpAuth.utf8().length();
} else
options.packet_time_params.srtp_auth_tag_len = -1;
m_rtcProvider.callSocket(m_identifier, [buffer = WTFMove(buffer), socketAddress, options](LibWebRTCSocketClient& client) {
client.sendTo(buffer.get(), socketAddress.value, options);
});
}
void NetworkRTCSocket::close()
{
m_rtcProvider.callSocket(m_identifier, [](LibWebRTCSocketClient& socket) {
socket.close();
});
}
void NetworkRTCSocket::setOption(int option, int value)
{
m_rtcProvider.callSocket(m_identifier, [option, value](LibWebRTCSocketClient& socket) {
socket.setOption(option, value);
});
}
} // namespace WebKit
#endif // USE(LIBWEBRTC)
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