1 files changed, 178 insertions, 0 deletions
diff --git a/Source/WebKit2/WebProcess/Network/webrtc/LibWebRTCSocket.cpp b/Source/WebKit2/WebProcess/Network/webrtc/LibWebRTCSocket.cpp
new file mode 100644
index 000000000..20277d577
--- /dev/null
+++ b/Source/WebKit2/WebProcess/Network/webrtc/LibWebRTCSocket.cpp
@@ -0,0 +1,178 @@
+/*
+ * 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 "LibWebRTCSocket.h"
+
+#if USE(LIBWEBRTC)
+
+#include "DataReference.h"
+#include "LibWebRTCSocketFactory.h"
+#include "NetworkProcessConnection.h"
+#include "NetworkRTCSocketMessages.h"
+#include "WebProcess.h"
+#include <WebCore/SharedBuffer.h>
+#include <wtf/Function.h>
+#include <wtf/MainThread.h>
+
+namespace WebKit {
+
+static inline void sendOnMainThread(Function<void(IPC::Connection&)>&& callback)
+{
+    callOnMainThread([callback = WTFMove(callback)]() {
+        callback(WebProcess::singleton().networkConnection().connection());
+    });
+}
+
+LibWebRTCSocket::LibWebRTCSocket(LibWebRTCSocketFactory& factory, uint64_t identifier, Type type, const rtc::SocketAddress& localAddress, const rtc::SocketAddress& remoteAddress)
+    : m_factory(factory)
+    , m_identifier(identifier)
+    , m_type(type)
+    , m_localAddress(localAddress)
+    , m_remoteAddress(remoteAddress)
+{
+    memset(&m_options, 1, MAX_SOCKET_OPTION);
+}
+
+LibWebRTCSocket::~LibWebRTCSocket()
+{
+    m_factory.detach(*this);
+}
+
+rtc::SocketAddress LibWebRTCSocket::GetLocalAddress() const
+{
+    return m_localAddress;
+}
+
+rtc::SocketAddress LibWebRTCSocket::GetRemoteAddress() const
+{
+    return m_remoteAddress;
+}
+
+void LibWebRTCSocket::signalAddressReady(const rtc::SocketAddress& address)
+{
+    m_localAddress = address;
+    m_state = (m_type == Type::ClientTCP) ? STATE_CONNECTED : STATE_BOUND;
+    SignalAddressReady(this, m_localAddress);
+}
+
+void LibWebRTCSocket::signalReadPacket(const WebCore::SharedBuffer& buffer, rtc::SocketAddress&& address, int64_t timestamp)
+{
+    m_remoteAddress = WTFMove(address);
+    SignalReadPacket(this, buffer.data(), buffer.size(), m_remoteAddress, rtc::PacketTime(timestamp, 0));
+}
+
+void LibWebRTCSocket::signalSentPacket(int rtcPacketID, int64_t sendTimeMs)
+{
+    m_availableSendingBytes += m_beingSentPacketSizes.takeFirst();
+    SignalSentPacket(this, rtc::SentPacket(rtcPacketID, sendTimeMs));
+    if (m_shouldSignalReadyToSend) {
+        m_shouldSignalReadyToSend = false;
+        SignalReadyToSend(this);
+    }
+}
+
+void LibWebRTCSocket::signalConnect()
+{
+    m_state = STATE_CONNECTED;
+    SignalConnect(this);
+}
+
+void LibWebRTCSocket::signalClose(int error)
+{
+    m_state = STATE_CLOSED;
+    SignalClose(this, error);
+}
+
+static inline String authKey(const rtc::PacketOptions& options)
+{
+    if (options.packet_time_params.srtp_auth_key.size() <= 0)
+        return { };
+    return String(options.packet_time_params.srtp_auth_key.data(), options.packet_time_params.srtp_auth_key.size());
+}
+
+bool LibWebRTCSocket::willSend(size_t size)
+{
+    if (size > m_availableSendingBytes) {
+        m_shouldSignalReadyToSend = true;
+        setError(EWOULDBLOCK);
+        return false;
+    }
+    m_availableSendingBytes -= size;
+    m_beingSentPacketSizes.append(size);
+    return true;
+}
+
+int LibWebRTCSocket::SendTo(const void *value, size_t size, const rtc::SocketAddress& address, const rtc::PacketOptions& options)
+{
+    if (!willSend(size))
+        return -1;
+
+    auto buffer = WebCore::SharedBuffer::create(static_cast<const uint8_t*>(value), size);
+    auto identifier = this->identifier();
+
+    sendOnMainThread([identifier, buffer = WTFMove(buffer), address, options](IPC::Connection& connection) {
+        IPC::DataReference data(reinterpret_cast<const uint8_t*>(buffer->data()), buffer->size());
+        String srtpAuthKey = authKey(options);
+        Messages::NetworkRTCSocket::SendTo message(data, RTCNetwork::SocketAddress(address), options.packet_id, options.packet_time_params.rtp_sendtime_extension_id, srtpAuthKey, options.packet_time_params.srtp_packet_index, options.dscp);
+        connection.send(WTFMove(message), identifier);
+    });
+    return size;
+}
+
+int LibWebRTCSocket::Close()
+{
+    auto identifier = this->identifier();
+    sendOnMainThread([identifier](IPC::Connection& connection) {
+        connection.send(Messages::NetworkRTCSocket::Close(), identifier);
+    });
+    return 0;
+}
+
+int LibWebRTCSocket::GetOption(rtc::Socket::Option option, int* value)
+{
+    ASSERT(option < MAX_SOCKET_OPTION);
+    int storedValue = m_options[option];
+    if (storedValue != -1)
+        *value = m_options[option];
+    return 0;
+}
+
+int LibWebRTCSocket::SetOption(rtc::Socket::Option option, int value)
+{
+    ASSERT(option < MAX_SOCKET_OPTION);
+
+    m_options[option] = value;
+
+    auto identifier = this->identifier();
+    sendOnMainThread([identifier, option, value](IPC::Connection& connection) {
+        connection.send(Messages::NetworkRTCSocket::SetOption(option, value), identifier);
+    });
+    return 0;
+}
+
+} // namespace WebKit
+
+#endif // USE(LIBWEBRTC)