Fix Chromium clang plugin warnings

webrtc/modules/audio_coding/neteq/tools/packet.h

 /*
  *  Copyright (c) 2014 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 44 matching lines @@
   // parser. Note that when the object is constructed using any of these
   // methods, the header will be parsed using a default RtpHeaderParser object.
   // In particular, RTP header extensions won't be parsed.
   Packet(uint8_t* packet_memory, size_t allocated_bytes, double time_ms);
 
   Packet(uint8_t* packet_memory,
          size_t allocated_bytes,
          size_t virtual_packet_length_bytes,
          double time_ms);
 
-  virtual ~Packet() {}
+  virtual ~Packet();
 
   // Parses the first bytes of the RTP payload, interpreting them as RED headers
   // according to RFC 2198. The headers will be inserted into |headers|. The
   // caller of the method assumes ownership of the objects in the list, and
   // must delete them properly.
   bool ExtractRedHeaders(std::list<RTPHeader*>* headers) const;
 
   // Deletes all RTPHeader objects in |headers|, but does not delete |headers|
   // itself.
   static void DeleteRedHeaders(std::list<RTPHeader*>* headers);
@@ skipping 37 matching lines @@
   size_t virtual_payload_length_bytes_;
   double time_ms_;     // Used to denote a packet's arrival time.
   bool valid_header_;  // Set by the RtpHeaderParser.
 
   RTC_DISALLOW_COPY_AND_ASSIGN(Packet);
 };
 
 }  // namespace test
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_PACKET_H_