1 files changed, 81 insertions, 0 deletions
diff --git a/Source/WebKit2/NetworkProcess/webrtc/NetworkRTCSocket.cpp b/Source/WebKit2/NetworkProcess/webrtc/NetworkRTCSocket.cpp
new file mode 100644
index 000000000..fcd12498f
--- /dev/null
+++ b/Source/WebKit2/NetworkProcess/webrtc/NetworkRTCSocket.cpp
@@ -0,0 +1,81 @@
+/*
+ * 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)