webkitgtk-2.16.5HEADwebkitgtk-2.16.5master

1 files changed, 127 insertions, 0 deletions

diff --git a/Source/WebKit2/NetworkProcess/webrtc/LibWebRTCSocketClient.cpp b/Source/WebKit2/NetworkProcess/webrtc/LibWebRTCSocketClient.cpp
new file mode 100644
index 000000000..33c6be505
--- /dev/null
+++ b/Source/WebKit2/NetworkProcess/webrtc/LibWebRTCSocketClient.cpp
@@ -0,0 +1,127 @@
+/*
+ * 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 "NetworkRTCProvider.h"
+#include "WebRTCSocketMessages.h"
+#include <WebCore/SharedBuffer.h>
+#include <wtf/Function.h>
+
+namespace WebKit {
+
+LibWebRTCSocketClient::LibWebRTCSocketClient(uint64_t identifier, NetworkRTCProvider& rtcProvider, std::unique_ptr<rtc::AsyncPacketSocket>&& socket, Type type)
+    : m_identifier(identifier)
+    , m_rtcProvider(rtcProvider)
+    , m_socket(WTFMove(socket))
+{
+    if (!m_socket) {
+        rtcProvider.sendFromMainThread([identifier](IPC::Connection& connection) {
+            connection.send(Messages::WebRTCSocket::SignalClose(1), identifier);
+        });
+        return;
+    }
+
+    m_socket->SignalReadPacket.connect(this, &LibWebRTCSocketClient::signalReadPacket);
+    m_socket->SignalSentPacket.connect(this, &LibWebRTCSocketClient::signalSentPacket);
+    m_socket->SignalConnect.connect(this, &LibWebRTCSocketClient::signalConnect);
+    m_socket->SignalClose.connect(this, &LibWebRTCSocketClient::signalClose);
+
+    if (type == Type::ClientTCP) {
+        m_socket->SignalAddressReady.connect(this, &LibWebRTCSocketClient::signalAddressReady);
+        return;
+    }
+    signalAddressReady();
+}
+
+void LibWebRTCSocketClient::sendTo(const WebCore::SharedBuffer& buffer, const rtc::SocketAddress& socketAddress, const rtc::PacketOptions& options)
+{
+    m_socket->SendTo(reinterpret_cast<const uint8_t*>(buffer.data()), buffer.size(), socketAddress, options);
+}
+
+void LibWebRTCSocketClient::close()
+{
+    ASSERT(m_socket);
+    m_socket->Close();
+    m_rtcProvider.takeSocket(m_identifier);
+}
+
+void LibWebRTCSocketClient::setOption(int option, int value)
+{
+    ASSERT(m_socket);
+    m_socket->SetOption(static_cast<rtc::Socket::Option>(option), value);
+}
+
+void LibWebRTCSocketClient::signalReadPacket(rtc::AsyncPacketSocket*, const char* value, size_t length, const rtc::SocketAddress& address, const rtc::PacketTime& packetTime)
+{
+    auto buffer = WebCore::SharedBuffer::create(value, length);
+    m_rtcProvider.sendFromMainThread([identifier = m_identifier, buffer = WTFMove(buffer), address = RTCNetwork::isolatedCopy(address), packetTime](IPC::Connection& connection) {
+        IPC::DataReference data(reinterpret_cast<const uint8_t*>(buffer->data()), buffer->size());
+        connection.send(Messages::WebRTCSocket::SignalReadPacket(data, RTCNetwork::IPAddress(address.ipaddr()), address.port(), packetTime.timestamp), identifier);
+    });
+}
+
+void LibWebRTCSocketClient::signalSentPacket(rtc::AsyncPacketSocket*, const rtc::SentPacket& sentPacket)
+{
+    m_rtcProvider.sendFromMainThread([identifier = m_identifier, sentPacket](IPC::Connection& connection) {
+        connection.send(Messages::WebRTCSocket::SignalSentPacket(sentPacket.packet_id, sentPacket.send_time_ms), identifier);
+    });
+}
+
+void LibWebRTCSocketClient::signalAddressReady(rtc::AsyncPacketSocket*, const rtc::SocketAddress& address)
+{
+    m_rtcProvider.sendFromMainThread([identifier = m_identifier, address = RTCNetwork::isolatedCopy(address)](IPC::Connection& connection) {
+        connection.send(Messages::WebRTCSocket::SignalAddressReady(RTCNetwork::SocketAddress(address)), identifier);
+    });
+}
+
+void LibWebRTCSocketClient::signalAddressReady()
+{
+    signalAddressReady(m_socket.get(), m_socket->GetLocalAddress());
+}
+
+void LibWebRTCSocketClient::signalConnect(rtc::AsyncPacketSocket*)
+{
+    m_rtcProvider.sendFromMainThread([identifier = m_identifier](IPC::Connection& connection) {
+        connection.send(Messages::WebRTCSocket::SignalConnect(), identifier);
+    });
+}
+
+void LibWebRTCSocketClient::signalClose(rtc::AsyncPacketSocket*, int error)
+{
+    m_rtcProvider.sendFromMainThread([identifier = m_identifier, error](IPC::Connection& connection) {
+        connection.send(Messages::WebRTCSocket::SignalClose(error), identifier);
+    });
+    m_rtcProvider.takeSocket(m_identifier);
+}
+
+} // namespace WebKit
+
+#endif // USE(LIBWEBRTC)