[NOT FOR REVIEW] Convert channel counts to size_t.

webrtc/modules/audio_coding/main/test/EncodeDecodeTest.cc

 /*
  *  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 34 matching lines @@
 }
 
 Sender::Sender()
     : _acm(NULL),
       _pcmFile(),
       _audioFrame(),
       _packetization(NULL) {
 }
 
 void Sender::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) {
   struct CodecInst sendCodec;
   int noOfCodecs = acm->NumberOfCodecs();
   int codecNo;
 
   // Open input file
   const std::string file_name = webrtc::test::ResourcePath(in_file_name, "pcm");
   _pcmFile.Open(file_name, sample_rate, "rb");
   if (channels == 2) {
     _pcmFile.ReadStereo(true);
   }
@@ skipping 46 matching lines @@
     }
   }
 }
 
 Receiver::Receiver()
     : _playoutLengthSmpls(WEBRTC_10MS_PCM_AUDIO),
       _payloadSizeBytes(MAX_INCOMING_PAYLOAD) {
 }
 
 void Receiver::Setup(AudioCodingModule *acm, RTPStream *rtpStream,
-                     std::string out_file_name, int channels) {
+                     std::string out_file_name, size_t channels) {
   struct CodecInst recvCodec = CodecInst();
   int noOfCodecs;
   EXPECT_EQ(0, acm->InitializeReceiver());
 
   noOfCodecs = acm->NumberOfCodecs();
   for (int i = 0; i < noOfCodecs; i++) {
     EXPECT_EQ(0, acm->Codec(i, &recvCodec));
     if (recvCodec.channels == channels)
       EXPECT_EQ(0, acm->RegisterReceiveCodec(recvCodec));
     // Forces mono/stereo for Opus.
@@ skipping 210 matching lines @@
   if (acm->SendCodec(&sendCodecInst) >= 0) {
     _sender.Run();
   }
   _sender.Teardown();
   rtpFile.Close();
 
   return fileName;
 }
 
 }  // namespace webrtc