Convert channel counts to size_t.

webrtc/modules/audio_coding/test/opus_test.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 14 matching lines @@
 
 class OpusTest : public ACMTest {
  public:
   OpusTest();
   ~OpusTest();
 
   void Perform();
 
  private:
   void Run(TestPackStereo* channel,
-           int channels,
+           size_t channels,
            int bitrate,
            size_t frame_length,
            int percent_loss = 0);
 
   void OpenOutFile(int test_number);
 
   rtc::scoped_ptr<AudioCodingModule> acm_receiver_;
   TestPackStereo* channel_a2b_;
   PCMFile in_file_stereo_;
   PCMFile in_file_mono_;
   PCMFile out_file_;
   PCMFile out_file_standalone_;
   int counter_;
   uint8_t payload_type_;
   uint32_t rtp_timestamp_;
   acm2::ACMResampler resampler_;
   WebRtcOpusEncInst* opus_mono_encoder_;
   WebRtcOpusEncInst* opus_stereo_encoder_;
   WebRtcOpusDecInst* opus_mono_decoder_;
   WebRtcOpusDecInst* opus_stereo_decoder_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_TEST_OPUS_TEST_H_