Address some violations of chromium-style.

webrtc/modules/rtp_rtcp/source/rtp_packet.h

 /*
  *  Copyright (c) 2016 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_MODULES_RTP_RTCP_SOURCE_RTP_PACKET_H_
@@ skipping 107 matching lines @@
   uint8_t* AllocatePayload(size_t size_bytes);
   bool SetPadding(uint8_t size_bytes, Random* random);
 
  protected:
   // |extensions| required for SetExtension/ReserveExtension functions during
   // packet creating and used if available in Parse function.
   // Adding and getting extensions will fail until |extensions| is
   // provided via constructor or IdentifyExtensions function.
   Packet();
   explicit Packet(const ExtensionManager* extensions);
-  Packet(const Packet&) = default;
+  Packet(const Packet&);
   Packet(const ExtensionManager* extensions, size_t capacity);
   virtual ~Packet();
 
   Packet& operator=(const Packet&) = default;
 
  private:
   struct ExtensionInfo {
     ExtensionType type;
     uint16_t offset;
     uint8_t length;
@@ skipping 58 matching lines @@
   auto buffer = AllocateExtension(Extension::kId, Extension::kValueSizeBytes);
   if (buffer.empty())
     return false;
   memset(buffer.data(), 0, Extension::kValueSizeBytes);
   return true;
 }
 }  // namespace rtp
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_PACKET_H_