Multi-networking with Android L.

webrtc/base/asyncpacketsocket.h

 /*
  *  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 {
 
+class Network;
+
 // 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();
   ~PacketTimeUpdateParams();
 
   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.
@@ skipping 66 matching lines @@
 
   // 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;
 
+  virtual void BindToNetwork(const Network* network) {}
+
   // 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 each time a packet is sent.
   sigslot::signal2<AsyncPacketSocket*, const SentPacket&> SignalSentPacket;
 
   // Emitted when the socket is currently able to send.
@@ skipping 16 matching lines @@
   // Used only for listening TCP sockets.
   sigslot::signal2<AsyncPacketSocket*, AsyncPacketSocket*> SignalNewConnection;
 
  private:
   RTC_DISALLOW_COPY_AND_ASSIGN(AsyncPacketSocket);
 };
 
 }  // namespace rtc
 
 #endif  // WEBRTC_BASE_ASYNCPACKETSOCKET_H_