Convert channel counts to size_t.

webrtc/modules/media_file/media_file_utility.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 162 matching lines @@
     // Update codecInst according to the current audio codec being used for
     // reading or writing.
     int32_t codec_info(CodecInst& codecInst);
 
 private:
     // Biggest WAV frame supported is 10 ms at 48kHz of 2 channel, 16 bit audio.
     static const size_t WAV_MAX_BUFFER_SIZE = 480 * 2 * 2;
 
 
     int32_t InitWavCodec(uint32_t samplesPerSec,
-                         uint32_t channels,
+                         size_t channels,
                          uint32_t bitsPerSample,
                          uint32_t formatTag);
 
     // Parse the WAV header in stream.
     int32_t ReadWavHeader(InStream& stream);
 
     // Update the WAV header. freqInHz, bytesPerSample, channels, format,
     // lengthInBytes specify characterists of the audio data.
     // freqInHz is the sampling frequency. bytesPerSample is the sample size in
     // bytes. channels is the number of channels, e.g. 1 is mono and 2 is
     // stereo. format is the encode format (e.g. PCMU, PCMA, PCM etc).
     // lengthInBytes is the number of bytes the audio samples are using up.
     int32_t WriteWavHeader(OutStream& stream,
                            uint32_t freqInHz,
                            size_t bytesPerSample,
-                           uint32_t channels,
+                           size_t channels,
                            uint32_t format,
                            size_t lengthInBytes);
 
     // Put dataLengthInBytes of audio data from stream into the audioBuffer.
     // The return value is the number of bytes written to audioBuffer.
     int32_t ReadWavData(InStream& stream, uint8_t* audioBuffer,
                         size_t dataLengthInBytes);
 
     // Update the current audio codec being used for reading or writing
     // according to codecInst.
@@ skipping 65 matching lines @@
 
     // Only reading or writing can be enabled, not both.
     bool _reading;
     bool _writing;
 
     // Scratch buffer used for turning stereo audio to mono.
     uint8_t _tempData[WAV_MAX_BUFFER_SIZE];
 };
 }  // namespace webrtc
 #endif // WEBRTC_MODULES_MEDIA_FILE_MEDIA_FILE_UTILITY_H_