Removed RTPHeader from NetEq's Packet struct.

webrtc/modules/audio_coding/neteq/packet_buffer.h

 /*
  *  Copyright (c) 2012 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 62 matching lines @@
   virtual int NextTimestamp(uint32_t* next_timestamp) const;
 
   // Gets the timestamp for the first packet in the buffer with a timestamp no
   // lower than the input limit |timestamp|. The result is written to the output
   // variable |next_timestamp|.
   // Returns PacketBuffer::kBufferEmpty if the buffer is empty,
   // PacketBuffer::kOK otherwise.
   virtual int NextHigherTimestamp(uint32_t timestamp,
                                   uint32_t* next_timestamp) const;
 
-  // Returns a (constant) pointer the RTP header of the first packet in the
-  // buffer. Returns NULL if the buffer is empty.
-  virtual const RTPHeader* NextRtpHeader() const;
+  // Returns a (constant) pointer to the first packet in the buffer. Returns
+  // NULL if the buffer is empty.
+  virtual const Packet* PeekNextPacket() const;
 
   // Extracts the first packet in the buffer and returns a pointer to it.
   // Returns NULL if the buffer is empty. The caller is responsible for deleting
   // the packet.
   // Subsequent packets with the same timestamp as the one extracted will be
   // discarded and properly deleted. The number of discarded packets will be
   // written to the output variable |discard_count|.
   virtual Packet* GetNextPacket(size_t* discard_count);
 
   // Discards the first packet in the buffer. The packet is deleted.
@@ skipping 51 matching lines @@
 
  private:
   size_t max_number_of_packets_;
   PacketList buffer_;
   const TickTimer* tick_timer_;
   RTC_DISALLOW_COPY_AND_ASSIGN(PacketBuffer);
 };
 
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_PACKET_BUFFER_H_