Convert channel counts to size_t.

webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_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.
  */
 
 #ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_TOOLS_AUDIO_CODEC_SPEED_TEST_H_
 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_TOOLS_AUDIO_CODEC_SPEED_TEST_H_
 
 #include <string>
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
 // Define coding parameter as
 // <channels, bit_rate, file_name, extension, if_save_output>.
-typedef std::tr1::tuple<int, int, std::string, std::string, bool> coding_param;
+typedef std::tr1::tuple<size_t, int, std::string, std::string, bool>
+    coding_param;
 
 class AudioCodecSpeedTest : public testing::TestWithParam<coding_param> {
  protected:
   AudioCodecSpeedTest(int block_duration_ms,
                       int input_sampling_khz,
                       int output_sampling_khz);
   virtual void SetUp();
   virtual void TearDown();
 
   // EncodeABlock(...) does the following:
@@ skipping 33 matching lines @@
   rtc::scoped_ptr<uint8_t[]> bit_stream_;
 
   // Maximum number of bytes in output bitstream for a frame of audio.
   size_t max_bytes_;
 
   size_t encoded_bytes_;
   float encoding_time_ms_;
   float decoding_time_ms_;
   FILE* out_file_;
 
-  int channels_;
+  size_t channels_;
 
   // Bit rate is in bit-per-second.
   int bit_rate_;
 
   std::string in_filename_;
 
   // Determines whether to save the output to file.
   bool save_out_data_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_TOOLS_AUDIO_CODEC_SPEED_TEST_H_