Convert channel counts to size_t.

webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h

 /*
  *  Copyright (c) 2014 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 81 matching lines @@
   // |neteq_|.
   int Transmit();
 
   // Runs encoding / transmitting / decoding.
   void Simulate();
 
   // Write to log file. Usage Log() << ...
   std::ofstream& Log();
 
   NetEqDecoder decoder_type_;
-  const int channels_;
+  const size_t channels_;
 
  private:
   int decoded_time_ms_;
   int decodable_time_ms_;
   double drift_factor_;
   int packet_loss_rate_;
   const int block_duration_ms_;
   const int in_sampling_khz_;
   const int out_sampling_khz_;
 
@@ skipping 19 matching lines @@
   rtc::scoped_ptr<int16_t[]> out_data_;
   WebRtcRTPHeader rtp_header_;
 
   size_t total_payload_size_bytes_;
 };
 
 }  // namespace test
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_QUALITY_TEST_H_