[NOT FOR REVIEW] Convert channel counts to size_t.

webrtc/modules/audio_coding/main/test/EncodeDecodeTest.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 30 matching lines @@
                             int16_t seqNo, uint32_t timeStamp, uint32_t ssrc);
   RTPStream* _rtpStream;
   int32_t _frequency;
   int16_t _seqNo;
 };
 
 class Sender {
  public:
   Sender();
   void Setup(AudioCodingModule *acm, RTPStream *rtpStream,
-             std::string in_file_name, int sample_rate, int channels);
+             std::string in_file_name, int sample_rate, size_t channels);
   void Teardown();
   void Run();
   bool Add10MsData();
 
   //for auto_test and logging
   uint8_t testMode;
   uint8_t codeId;
 
  protected:
   AudioCodingModule* _acm;
 
  private:
   PCMFile _pcmFile;
   AudioFrame _audioFrame;
   TestPacketization* _packetization;
 };
 
 class Receiver {
  public:
   Receiver();
   virtual ~Receiver() {};
   void Setup(AudioCodingModule *acm, RTPStream *rtpStream,
-             std::string out_file_name, int channels);
+             std::string out_file_name, size_t channels);
   void Teardown();
   void Run();
   virtual bool IncomingPacket();
   bool PlayoutData();
 
   //for auto_test and logging
   uint8_t codeId;
   uint8_t testMode;
 
  private:
@@ skipping 29 matching lines @@
                            int testMode);
 
  protected:
   Sender _sender;
   Receiver _receiver;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_ENCODEDECODETEST_H_