Update audio code to use size_t more correctly,

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 70 matching lines @@
   // 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;
 
   // 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(int* discard_count);
+  virtual Packet* GetNextPacket(size_t* discard_count);
 
   // Discards the first packet in the buffer. The packet is deleted.
   // Returns PacketBuffer::kBufferEmpty if the buffer is empty,
   // PacketBuffer::kOK otherwise.
   virtual int DiscardNextPacket();
 
   // Discards all packets that are (strictly) older than timestamp_limit,
   // but newer than timestamp_limit - horizon_samples. Setting horizon_samples
   // to zero implies that the horizon is set to half the timestamp range. That
   // is, if a packet is more than 2^31 timestamps into the future compared with
   // timestamp_limit (including wrap-around), it is considered old.
   // Returns number of packets discarded.
   virtual int DiscardOldPackets(uint32_t timestamp_limit,
                                 uint32_t horizon_samples);
 
   // Discards all packets that are (strictly) older than timestamp_limit.
   virtual int DiscardAllOldPackets(uint32_t timestamp_limit);
 
   // Returns the number of packets in the buffer, including duplicates and
   // redundant packets.
-  virtual int NumPacketsInBuffer() const;
+  virtual size_t NumPacketsInBuffer() const;
 
   // Returns the number of samples in the buffer, including samples carried in
   // duplicate and redundant packets.
-  virtual int NumSamplesInBuffer(DecoderDatabase* decoder_database,
-                                 int last_decoded_length) const;
+  virtual size_t NumSamplesInBuffer(DecoderDatabase* decoder_database,
+                                    size_t last_decoded_length) const;
 
   // Increase the waiting time counter for every packet in the buffer by |inc|.
   // The default value for |inc| is 1.
   virtual void IncrementWaitingTimes(int inc = 1);
 
   virtual void BufferStat(int* num_packets, int* max_num_packets) const;
 
   // Static method that properly deletes the first packet, and its payload
   // array, in |packet_list|. Returns false if |packet_list| already was empty,
   // otherwise true.
@@ skipping 18 matching lines @@
   }
 
  private:
   size_t max_number_of_packets_;
   PacketList buffer_;
   DISALLOW_COPY_AND_ASSIGN(PacketBuffer);
 };
 
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_PACKET_BUFFER_H_