| /* |
| * 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 "LibWebRTCSocketClient.h" |
| |
| #if USE(LIBWEBRTC) |
| |
| #include "Connection.h" |
| #include "DataReference.h" |
| #include "LibWebRTCNetworkMessages.h" |
| #include "Logging.h" |
| #include "NetworkRTCProvider.h" |
| #include <WebCore/SharedBuffer.h> |
| #include <wtf/Function.h> |
| |
| namespace WebKit { |
| |
| LibWebRTCSocketClient::LibWebRTCSocketClient(WebCore::LibWebRTCSocketIdentifier identifier, NetworkRTCProvider& rtcProvider, std::unique_ptr<rtc::AsyncPacketSocket>&& socket, Type type, Ref<IPC::Connection>&& connection) |
| : m_identifier(identifier) |
| , m_type(type) |
| , m_rtcProvider(rtcProvider) |
| , m_socket(WTFMove(socket)) |
| , m_connection(WTFMove(connection)) |
| { |
| ASSERT(m_socket); |
| |
| m_socket->SignalReadPacket.connect(this, &LibWebRTCSocketClient::signalReadPacket); |
| m_socket->SignalSentPacket.connect(this, &LibWebRTCSocketClient::signalSentPacket); |
| m_socket->SignalClose.connect(this, &LibWebRTCSocketClient::signalClose); |
| |
| switch (type) { |
| case Type::ServerConnectionTCP: |
| return; |
| case Type::ClientTCP: |
| m_socket->SignalConnect.connect(this, &LibWebRTCSocketClient::signalConnect); |
| m_socket->SignalAddressReady.connect(this, &LibWebRTCSocketClient::signalAddressReady); |
| return; |
| case Type::ServerTCP: |
| m_socket->SignalConnect.connect(this, &LibWebRTCSocketClient::signalConnect); |
| m_socket->SignalNewConnection.connect(this, &LibWebRTCSocketClient::signalNewConnection); |
| signalAddressReady(); |
| return; |
| case Type::UDP: |
| m_socket->SignalConnect.connect(this, &LibWebRTCSocketClient::signalConnect); |
| signalAddressReady(); |
| return; |
| } |
| } |
| |
| void LibWebRTCSocketClient::sendTo(const uint8_t* data, size_t size, const rtc::SocketAddress& socketAddress, const rtc::PacketOptions& options) |
| { |
| m_socket->SendTo(data, size, socketAddress, options); |
| auto error = m_socket->GetError(); |
| RELEASE_LOG_ERROR_IF(error && m_sendError != error, Network, "LibWebRTCSocketClient::sendTo (ID=%" PRIu64 ") failed with error %d", m_identifier.toUInt64(), error); |
| m_sendError = error; |
| } |
| |
| void LibWebRTCSocketClient::close() |
| { |
| ASSERT(m_socket); |
| auto result = m_socket->Close(); |
| UNUSED_PARAM(result); |
| RELEASE_LOG_ERROR_IF(result, Network, "LibWebRTCSocketClient::close (ID=%" PRIu64 ") failed with error %d", m_identifier.toUInt64(), m_socket->GetError()); |
| |
| m_rtcProvider.takeSocket(m_identifier); |
| } |
| |
| void LibWebRTCSocketClient::setOption(int option, int value) |
| { |
| ASSERT(m_socket); |
| auto result = m_socket->SetOption(static_cast<rtc::Socket::Option>(option), value); |
| UNUSED_PARAM(result); |
| RELEASE_LOG_ERROR_IF(result, Network, "LibWebRTCSocketClient::setOption(%d, %d) (ID=%" PRIu64 ") failed with error %d", option, value, m_identifier.toUInt64(), m_socket->GetError()); |
| } |
| |
| void LibWebRTCSocketClient::signalReadPacket(rtc::AsyncPacketSocket* socket, const char* value, size_t length, const rtc::SocketAddress& address, const rtc::PacketTime& packetTime) |
| { |
| ASSERT_UNUSED(socket, m_socket.get() == socket); |
| IPC::DataReference data(reinterpret_cast<const uint8_t*>(value), length); |
| m_connection->send(Messages::LibWebRTCNetwork::SignalReadPacket(m_identifier, data, RTCNetwork::IPAddress(address.ipaddr()), address.port(), packetTime), 0); |
| } |
| |
| void LibWebRTCSocketClient::signalSentPacket(rtc::AsyncPacketSocket* socket, const rtc::SentPacket& sentPacket) |
| { |
| ASSERT_UNUSED(socket, m_socket.get() == socket); |
| m_connection->send(Messages::LibWebRTCNetwork::SignalSentPacket(m_identifier, sentPacket.packet_id, sentPacket.send_time_ms), 0); |
| } |
| |
| void LibWebRTCSocketClient::signalNewConnection(rtc::AsyncPacketSocket* socket, rtc::AsyncPacketSocket* newSocket) |
| { |
| ASSERT_UNUSED(socket, m_socket.get() == socket); |
| m_rtcProvider.newConnection(*this, std::unique_ptr<rtc::AsyncPacketSocket>(newSocket)); |
| } |
| |
| void LibWebRTCSocketClient::signalAddressReady(rtc::AsyncPacketSocket* socket, const rtc::SocketAddress& address) |
| { |
| ASSERT_UNUSED(socket, m_socket.get() == socket); |
| m_connection->send(Messages::LibWebRTCNetwork::SignalAddressReady(m_identifier, RTCNetwork::SocketAddress(address)), 0); |
| } |
| |
| void LibWebRTCSocketClient::signalAddressReady() |
| { |
| signalAddressReady(m_socket.get(), m_socket->GetLocalAddress()); |
| } |
| |
| void LibWebRTCSocketClient::signalConnect(rtc::AsyncPacketSocket* socket) |
| { |
| ASSERT_UNUSED(socket, m_socket.get() == socket); |
| m_connection->send(Messages::LibWebRTCNetwork::SignalConnect(m_identifier), 0); |
| } |
| |
| void LibWebRTCSocketClient::signalClose(rtc::AsyncPacketSocket* socket, int error) |
| { |
| ASSERT_UNUSED(socket, m_socket.get() == socket); |
| m_connection->send(Messages::LibWebRTCNetwork::SignalClose(m_identifier, error), 0); |
| |
| // We want to remove 'this' from the socket map now but we will destroy it asynchronously |
| // so that the socket parameter of signalClose remains alive as the caller of signalClose may actually being using it afterwards. |
| m_rtcProvider.callOnRTCNetworkThread([socket = m_rtcProvider.takeSocket(m_identifier)] { }); |
| } |
| |
| } // namespace WebKit |
| |
| #endif // USE(LIBWEBRTC) |