Merge changes I7bbf776e,I1b827825

am: fe8b4a6579

* commit 'fe8b4a657979b49e1701bd92f6d5814a99e0b2be': (7237 commits)
  WIP: Changes after merge commit 'cb3f9bd'
  Make the nonlinear beamformer steerable
  Utilize bitrate above codec max to protect video.
  Enable VP9 internal resize by default.
  Filter overlapping RTP header extensions.
  Make VCMEncodedFrameCallback const.
  MediaCodecVideoEncoder: Add number of quality resolution downscales to Encoded callback.
  Remove redudant encoder rate calls.
  Create isolate files for nonparallel tests.
  Register header extensions in RtpRtcpObserver to avoid log spam.
  Make an enum class out of NetEqDecoder, and hide the neteq_decoders_ table
  ACM: Move NACK functionality inside NetEq
  Fix chromium-style warnings in webrtc/sound/.
  Create a 'webrtc_nonparallel_tests' target.
  Update scalability structure data according to updates in the RTP payload profile.
  audio_coding: rename interface -> include
  Rewrote perform_action_on_all_files to be parallell.
  Update reference indices according to updates in the RTP payload profile.
  Disable P2PTransport...TestFailoverControlledSide on Memcheck
  pass clangcl compile options to ignore warnings in gflags.cc
  ...

1 files changed, 128 insertions, 0 deletions

diff --git a/webrtc/p2p/base/stunrequest.h b/webrtc/p2p/base/stunrequest.h
new file mode 100644
index 0000000000..267b4a1959
--- /dev/null
+++ b/webrtc/p2p/base/stunrequest.h
@@ -0,0 +1,128 @@
+/*
+ *  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_P2P_BASE_STUNREQUEST_H_
+#define WEBRTC_P2P_BASE_STUNREQUEST_H_
+
+#include <map>
+#include <string>
+#include "webrtc/p2p/base/stun.h"
+#include "webrtc/base/sigslot.h"
+#include "webrtc/base/thread.h"
+
+namespace cricket {
+
+class StunRequest;
+
+// Manages a set of STUN requests, sending and resending until we receive a
+// response or determine that the request has timed out.
+class StunRequestManager {
+ public:
+  StunRequestManager(rtc::Thread* thread);
+  ~StunRequestManager();
+
+  // Starts sending the given request (perhaps after a delay).
+  void Send(StunRequest* request);
+  void SendDelayed(StunRequest* request, int delay);
+
+  // Removes a stun request that was added previously.  This will happen
+  // automatically when a request succeeds, fails, or times out.
+  void Remove(StunRequest* request);
+
+  // Removes all stun requests that were added previously.
+  void Clear();
+
+  // Determines whether the given message is a response to one of the
+  // outstanding requests, and if so, processes it appropriately.
+  bool CheckResponse(StunMessage* msg);
+  bool CheckResponse(const char* data, size_t size);
+
+  bool empty() { return requests_.empty(); }
+
+  // Set the Origin header for outgoing stun messages.
+  void set_origin(const std::string& origin) { origin_ = origin; }
+
+  // Raised when there are bytes to be sent.
+  sigslot::signal3<const void*, size_t, StunRequest*> SignalSendPacket;
+
+ private:
+  typedef std::map<std::string, StunRequest*> RequestMap;
+
+  rtc::Thread* thread_;
+  RequestMap requests_;
+  std::string origin_;
+
+  friend class StunRequest;
+};
+
+// Represents an individual request to be sent.  The STUN message can either be
+// constructed beforehand or built on demand.
+class StunRequest : public rtc::MessageHandler {
+ public:
+  StunRequest();
+  StunRequest(StunMessage* request);
+  virtual ~StunRequest();
+
+  // Causes our wrapped StunMessage to be Prepared
+  void Construct();
+
+  // The manager handling this request (if it has been scheduled for sending).
+  StunRequestManager* manager() { return manager_; }
+
+  // Returns the transaction ID of this request.
+  const std::string& id() { return msg_->transaction_id(); }
+
+  // the origin value
+  const std::string& origin() const { return origin_; }
+  void set_origin(const std::string& origin) { origin_ = origin; }
+
+  // Returns the STUN type of the request message.
+  int type();
+
+  // Returns a const pointer to |msg_|.
+  const StunMessage* msg() const;
+
+  // Time elapsed since last send (in ms)
+  uint32_t Elapsed() const;
+
+ protected:
+  int count_;
+  bool timeout_;
+  std::string origin_;
+
+  // Fills in a request object to be sent.  Note that request's transaction ID
+  // will already be set and cannot be changed.
+  virtual void Prepare(StunMessage* request) {}
+
+  // Called when the message receives a response or times out.
+  virtual void OnResponse(StunMessage* response) {}
+  virtual void OnErrorResponse(StunMessage* response) {}
+  virtual void OnTimeout() {}
+  // Called when the message is sent.
+  virtual void OnSent();
+  // Returns the next delay for resends.
+  virtual int resend_delay();
+
+ private:
+  void set_manager(StunRequestManager* manager);
+
+  // Handles messages for sending and timeout.
+  void OnMessage(rtc::Message* pmsg);
+
+  StunRequestManager* manager_;
+  StunMessage* msg_;
+  uint32_t tstamp_;
+
+  friend class StunRequestManager;
+};
+
+}  // namespace cricket
+
+#endif  // WEBRTC_P2P_BASE_STUNREQUEST_H_