Update audio code to use size_t more correctly,

webrtc/modules/audio_coding/neteq/mock/mock_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 23 matching lines @@
           const DecoderDatabase& decoder_database,
           uint8_t* current_rtp_payload_type,
           uint8_t* current_cng_rtp_payload_type));
   MOCK_CONST_METHOD1(NextTimestamp,
       int(uint32_t* next_timestamp));
   MOCK_CONST_METHOD2(NextHigherTimestamp,
       int(uint32_t timestamp, uint32_t* next_timestamp));
   MOCK_CONST_METHOD0(NextRtpHeader,
       const RTPHeader*());
   MOCK_METHOD1(GetNextPacket,
-      Packet*(int* discard_count));
+      Packet*(size_t* discard_count));
   MOCK_METHOD0(DiscardNextPacket,
       int());
   MOCK_METHOD2(DiscardOldPackets,
       int(uint32_t timestamp_limit, uint32_t horizon_samples));
   MOCK_METHOD1(DiscardAllOldPackets,
       int(uint32_t timestamp_limit));
   MOCK_CONST_METHOD0(NumPacketsInBuffer,
-      int());
+      size_t());
   MOCK_METHOD1(IncrementWaitingTimes,
       void(int));
   MOCK_CONST_METHOD0(current_memory_bytes,
       int());
 };
 
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_PACKET_BUFFER_H_