Fixing bug where Connection drops packets when presumed writable.

webrtc/p2p/base/port.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.
  */
 
@@ skipping 443 matching lines @@
   bool writable() const { return write_state_ == STATE_WRITABLE; }
   bool receiving() const { return receiving_; }
 
   // Determines whether the connection has finished connecting.  This can only
   // be false for TCP connections.
   bool connected() const { return connected_; }
   bool weak() const { return !(writable() && receiving() && connected()); }
   bool active() const {
     return write_state_ != STATE_WRITE_TIMEOUT;
   }
-  virtual bool ReadyToSendMedia() const {
-    return write_state_ == STATE_WRITABLE ||
-           write_state_ == STATE_WRITE_UNRELIABLE;
-  }
 
   // A connection is dead if it can be safely deleted.
   bool dead(int64_t now) const;
 
   // Estimate of the round-trip time over this connection.
   int rtt() const { return rtt_; }
 
   ConnectionInfo stats();
 
   sigslot::signal1<Connection*> SignalStateChange;
@@ skipping 207 matching lines @@
            const rtc::PacketOptions& options) override;
   int GetError() override { return error_; }
 
  private:
   int error_ = 0;
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_P2P_BASE_PORT_H_