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diff --git a/chromium/third_party/webrtc/base/asyncpacketsocket.h b/chromium/third_party/webrtc/base/asyncpacketsocket.h
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+++ b/chromium/third_party/webrtc/base/asyncpacketsocket.h
@@ -0,0 +1,140 @@
+/*
+ *  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.
+ */
+
+#ifndef WEBRTC_BASE_ASYNCPACKETSOCKET_H_
+#define WEBRTC_BASE_ASYNCPACKETSOCKET_H_
+
+#include "webrtc/base/dscp.h"
+#include "webrtc/base/sigslot.h"
+#include "webrtc/base/socket.h"
+#include "webrtc/base/timeutils.h"
+
+namespace rtc {
+
+// This structure holds the info needed to update the packet send time header
+// extension, including the information needed to update the authentication tag
+// after changing the value.
+struct PacketTimeUpdateParams {
+  PacketTimeUpdateParams()
+      : rtp_sendtime_extension_id(-1), srtp_auth_tag_len(-1),
+        srtp_packet_index(-1) {
+  }
+
+  int rtp_sendtime_extension_id;    // extension header id present in packet.
+  std::vector<char> srtp_auth_key;  // Authentication key.
+  int srtp_auth_tag_len;            // Authentication tag length.
+  int64 srtp_packet_index;          // Required for Rtp Packet authentication.
+};
+
+// This structure holds meta information for the packet which is about to send
+// over network.
+struct PacketOptions {
+  PacketOptions() : dscp(DSCP_NO_CHANGE) {}
+  explicit PacketOptions(DiffServCodePoint dscp) : dscp(dscp) {}
+
+  DiffServCodePoint dscp;
+  PacketTimeUpdateParams packet_time_params;
+};
+
+// This structure will have the information about when packet is actually
+// received by socket.
+struct PacketTime {
+  PacketTime() : timestamp(-1), not_before(-1) {}
+  PacketTime(int64 timestamp, int64 not_before)
+      : timestamp(timestamp), not_before(not_before) {
+  }
+
+  int64 timestamp;  // Receive time after socket delivers the data.
+  int64 not_before; // Earliest possible time the data could have arrived,
+                    // indicating the potential error in the |timestamp| value,
+                    // in case the system, is busy. For example, the time of
+                    // the last select() call.
+                    // If unknown, this value will be set to zero.
+};
+
+inline PacketTime CreatePacketTime(int64 not_before) {
+  return PacketTime(TimeMicros(), not_before);
+}
+
+// Provides the ability to receive packets asynchronously. Sends are not
+// buffered since it is acceptable to drop packets under high load.
+class AsyncPacketSocket : public sigslot::has_slots<> {
+ public:
+  enum State {
+    STATE_CLOSED,
+    STATE_BINDING,
+    STATE_BOUND,
+    STATE_CONNECTING,
+    STATE_CONNECTED
+  };
+
+  AsyncPacketSocket() { }
+  virtual ~AsyncPacketSocket() { }
+
+  // Returns current local address. Address may be set to NULL if the
+  // socket is not bound yet (GetState() returns STATE_BINDING).
+  virtual SocketAddress GetLocalAddress() const = 0;
+
+  // Returns remote address. Returns zeroes if this is not a client TCP socket.
+  virtual SocketAddress GetRemoteAddress() const = 0;
+
+  // Send a packet.
+  virtual int Send(const void *pv, size_t cb, const PacketOptions& options) = 0;
+  virtual int SendTo(const void *pv, size_t cb, const SocketAddress& addr,
+                     const PacketOptions& options) = 0;
+
+  // Close the socket.
+  virtual int Close() = 0;
+
+  // Returns current state of the socket.
+  virtual State GetState() const = 0;
+
+  // Get/set options.
+  virtual int GetOption(Socket::Option opt, int* value) = 0;
+  virtual int SetOption(Socket::Option opt, int value) = 0;
+
+  // Get/Set current error.
+  // TODO: Remove SetError().
+  virtual int GetError() const = 0;
+  virtual void SetError(int error) = 0;
+
+  // Emitted each time a packet is read. Used only for UDP and
+  // connected TCP sockets.
+  sigslot::signal5<AsyncPacketSocket*, const char*, size_t,
+                   const SocketAddress&,
+                   const PacketTime&> SignalReadPacket;
+
+  // Emitted when the socket is currently able to send.
+  sigslot::signal1<AsyncPacketSocket*> SignalReadyToSend;
+
+  // Emitted after address for the socket is allocated, i.e. binding
+  // is finished. State of the socket is changed from BINDING to BOUND
+  // (for UDP and server TCP sockets) or CONNECTING (for client TCP
+  // sockets).
+  sigslot::signal2<AsyncPacketSocket*, const SocketAddress&> SignalAddressReady;
+
+  // Emitted for client TCP sockets when state is changed from
+  // CONNECTING to CONNECTED.
+  sigslot::signal1<AsyncPacketSocket*> SignalConnect;
+
+  // Emitted for client TCP sockets when state is changed from
+  // CONNECTED to CLOSED.
+  sigslot::signal2<AsyncPacketSocket*, int> SignalClose;
+
+  // Used only for listening TCP sockets.
+  sigslot::signal2<AsyncPacketSocket*, AsyncPacketSocket*> SignalNewConnection;
+
+ private:
+  DISALLOW_EVIL_CONSTRUCTORS(AsyncPacketSocket);
+};
+
+}  // namespace rtc
+
+#endif  // WEBRTC_BASE_ASYNCPACKETSOCKET_H_