Repo created

TMessagesProj/jni/voip/webrtc/rtc_base/server_socket_adapters.cc

@@ -0,0 +1,58 @@
+/*
+ *  Copyright 2004 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "rtc_base/server_socket_adapters.h"
+
+#include <string>
+
+#include "rtc_base/byte_buffer.h"
+
+namespace rtc {
+
+AsyncProxyServerSocket::AsyncProxyServerSocket(Socket* socket,
+                                               size_t buffer_size)
+    : BufferedReadAdapter(socket, buffer_size) {}
+
+AsyncProxyServerSocket::~AsyncProxyServerSocket() = default;
+
+AsyncSSLServerSocket::AsyncSSLServerSocket(Socket* socket)
+    : BufferedReadAdapter(socket, 1024) {
+  BufferInput(true);
+}
+
+void AsyncSSLServerSocket::ProcessInput(char* data, size_t* len) {
+  // We only accept client hello messages.
+  const ArrayView<const uint8_t> client_hello =
+      AsyncSSLSocket::SslClientHello();
+  if (*len < client_hello.size()) {
+    return;
+  }
+
+  if (memcmp(client_hello.data(), data, client_hello.size()) != 0) {
+    Close();
+    SignalCloseEvent(this, 0);
+    return;
+  }
+
+  *len -= client_hello.size();
+
+  // Clients should not send more data until the handshake is completed.
+  RTC_DCHECK(*len == 0);
+
+  const ArrayView<const uint8_t> server_hello =
+      AsyncSSLSocket::SslServerHello();
+  // Send a server hello back to the client.
+  DirectSend(server_hello.data(), server_hello.size());
+
+  // Handshake completed for us, redirect input to our parent.
+  BufferInput(false);
+}
+
+}  // namespace rtc