| 1 | /* |
| 2 | * Copyright (C) 2017 Apple Inc. All rights reserved. |
| 3 | * |
| 4 | * Redistribution and use in source and binary forms, with or without |
| 5 | * modification, are permitted provided that the following conditions |
| 6 | * are met: |
| 7 | * 1. Redistributions of source code must retain the above copyright |
| 8 | * notice, this list of conditions and the following disclaimer. |
| 9 | * 2. Redistributions in binary form must reproduce the above copyright |
| 10 | * notice, this list of conditions and the following disclaimer in the |
| 11 | * documentation and/or other materials provided with the distribution. |
| 12 | * |
| 13 | * THIS SOFTWARE IS PROVIDED BY APPLE INC. AND ITS CONTRIBUTORS ``AS IS'' |
| 14 | * AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, |
| 15 | * THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR |
| 16 | * PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL APPLE INC. OR ITS CONTRIBUTORS |
| 17 | * BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR |
| 18 | * CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF |
| 19 | * SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS |
| 20 | * INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN |
| 21 | * CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) |
| 22 | * ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF |
| 23 | * THE POSSIBILITY OF SUCH DAMAGE. |
| 24 | */ |
| 25 | |
| 26 | #include "config.h" |
| 27 | #include "LibWebRTCSocketClient.h" |
| 28 | |
| 29 | #if USE(LIBWEBRTC) |
| 30 | |
| 31 | #include "Connection.h" |
| 32 | #include "DataReference.h" |
| 33 | #include "NetworkRTCProvider.h" |
| 34 | #include "WebRTCSocketMessages.h" |
| 35 | #include <WebCore/SharedBuffer.h> |
| 36 | #include <wtf/Function.h> |
| 37 | |
| 38 | namespace WebKit { |
| 39 | |
| 40 | LibWebRTCSocketClient::LibWebRTCSocketClient(uint64_t identifier, NetworkRTCProvider& rtcProvider, std::unique_ptr<rtc::AsyncPacketSocket>&& socket, Type type) |
| 41 | : m_identifier(identifier) |
| 42 | , m_type(type) |
| 43 | , m_rtcProvider(rtcProvider) |
| 44 | , m_socket(WTFMove(socket)) |
| 45 | { |
| 46 | ASSERT(m_socket); |
| 47 | |
| 48 | m_socket->SignalReadPacket.connect(this, &LibWebRTCSocketClient::signalReadPacket); |
| 49 | m_socket->SignalSentPacket.connect(this, &LibWebRTCSocketClient::signalSentPacket); |
| 50 | m_socket->SignalClose.connect(this, &LibWebRTCSocketClient::signalClose); |
| 51 | |
| 52 | switch (type) { |
| 53 | case Type::ServerConnectionTCP: |
| 54 | return; |
| 55 | case Type::ClientTCP: |
| 56 | m_socket->SignalConnect.connect(this, &LibWebRTCSocketClient::signalConnect); |
| 57 | m_socket->SignalAddressReady.connect(this, &LibWebRTCSocketClient::signalAddressReady); |
| 58 | return; |
| 59 | case Type::ServerTCP: |
| 60 | m_socket->SignalConnect.connect(this, &LibWebRTCSocketClient::signalConnect); |
| 61 | m_socket->SignalNewConnection.connect(this, &LibWebRTCSocketClient::signalNewConnection); |
| 62 | signalAddressReady(); |
| 63 | return; |
| 64 | case Type::UDP: |
| 65 | m_socket->SignalConnect.connect(this, &LibWebRTCSocketClient::signalConnect); |
| 66 | signalAddressReady(); |
| 67 | return; |
| 68 | } |
| 69 | } |
| 70 | |
| 71 | void LibWebRTCSocketClient::sendTo(const WebCore::SharedBuffer& buffer, const rtc::SocketAddress& socketAddress, const rtc::PacketOptions& options) |
| 72 | { |
| 73 | m_socket->SendTo(reinterpret_cast<const uint8_t*>(buffer.data()), buffer.size(), socketAddress, options); |
| 74 | } |
| 75 | |
| 76 | void LibWebRTCSocketClient::close() |
| 77 | { |
| 78 | ASSERT(m_socket); |
| 79 | m_socket->Close(); |
| 80 | m_rtcProvider.takeSocket(m_identifier); |
| 81 | } |
| 82 | |
| 83 | void LibWebRTCSocketClient::setOption(int option, int value) |
| 84 | { |
| 85 | ASSERT(m_socket); |
| 86 | m_socket->SetOption(static_cast<rtc::Socket::Option>(option), value); |
| 87 | } |
| 88 | |
| 89 | void LibWebRTCSocketClient::signalReadPacket(rtc::AsyncPacketSocket* socket, const char* value, size_t length, const rtc::SocketAddress& address, const rtc::PacketTime& packetTime) |
| 90 | { |
| 91 | ASSERT_UNUSED(socket, m_socket.get() == socket); |
| 92 | auto buffer = WebCore::SharedBuffer::create(value, length); |
| 93 | m_rtcProvider.sendFromMainThread([identifier = m_identifier, buffer = WTFMove(buffer), address = RTCNetwork::isolatedCopy(address), packetTime](IPC::Connection& connection) { |
| 94 | IPC::DataReference data(reinterpret_cast<const uint8_t*>(buffer->data()), buffer->size()); |
| 95 | connection.send(Messages::WebRTCSocket::SignalReadPacket(data, RTCNetwork::IPAddress(address.ipaddr()), address.port(), packetTime), identifier); |
| 96 | }); |
| 97 | } |
| 98 | |
| 99 | void LibWebRTCSocketClient::signalSentPacket(rtc::AsyncPacketSocket* socket, const rtc::SentPacket& sentPacket) |
| 100 | { |
| 101 | ASSERT_UNUSED(socket, m_socket.get() == socket); |
| 102 | m_rtcProvider.sendFromMainThread([identifier = m_identifier, sentPacket](IPC::Connection& connection) { |
| 103 | connection.send(Messages::WebRTCSocket::SignalSentPacket(sentPacket.packet_id, sentPacket.send_time_ms), identifier); |
| 104 | }); |
| 105 | } |
| 106 | |
| 107 | void LibWebRTCSocketClient::signalNewConnection(rtc::AsyncPacketSocket* socket, rtc::AsyncPacketSocket* newSocket) |
| 108 | { |
| 109 | ASSERT_UNUSED(socket, m_socket.get() == socket); |
| 110 | m_rtcProvider.newConnection(*this, std::unique_ptr<rtc::AsyncPacketSocket>(newSocket)); |
| 111 | } |
| 112 | |
| 113 | void LibWebRTCSocketClient::signalAddressReady(rtc::AsyncPacketSocket* socket, const rtc::SocketAddress& address) |
| 114 | { |
| 115 | ASSERT_UNUSED(socket, m_socket.get() == socket); |
| 116 | m_rtcProvider.sendFromMainThread([identifier = m_identifier, address = RTCNetwork::isolatedCopy(address)](IPC::Connection& connection) { |
| 117 | connection.send(Messages::WebRTCSocket::SignalAddressReady(RTCNetwork::SocketAddress(address)), identifier); |
| 118 | }); |
| 119 | } |
| 120 | |
| 121 | void LibWebRTCSocketClient::signalAddressReady() |
| 122 | { |
| 123 | signalAddressReady(m_socket.get(), m_socket->GetLocalAddress()); |
| 124 | } |
| 125 | |
| 126 | void LibWebRTCSocketClient::signalConnect(rtc::AsyncPacketSocket* socket) |
| 127 | { |
| 128 | ASSERT_UNUSED(socket, m_socket.get() == socket); |
| 129 | m_rtcProvider.sendFromMainThread([identifier = m_identifier](IPC::Connection& connection) { |
| 130 | connection.send(Messages::WebRTCSocket::SignalConnect(), identifier); |
| 131 | }); |
| 132 | } |
| 133 | |
| 134 | void LibWebRTCSocketClient::signalClose(rtc::AsyncPacketSocket* socket, int error) |
| 135 | { |
| 136 | ASSERT_UNUSED(socket, m_socket.get() == socket); |
| 137 | m_rtcProvider.sendFromMainThread([identifier = m_identifier, error](IPC::Connection& connection) { |
| 138 | connection.send(Messages::WebRTCSocket::SignalClose(error), identifier); |
| 139 | }); |
| 140 | // We want to remove 'this' from the socket map now but we will destroy it asynchronously |
| 141 | // so that the socket parameter of signalClose remains alive as the caller of signalClose may actually being using it afterwards. |
| 142 | m_rtcProvider.callOnRTCNetworkThread([socket = m_rtcProvider.takeSocket(m_identifier)] { }); |
| 143 | } |
| 144 | |
| 145 | } // namespace WebKit |
| 146 | |
| 147 | #endif // USE(LIBWEBRTC) |
| 148 | |