Convert channel counts to size_t.

webrtc/modules/audio_coding/codecs/opus/opus_inst.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_OPUS_OPUS_INST_H_
 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_OPUS_INST_H_
 
 #include <stddef.h>
 
 #include "opus.h"
 
 struct WebRtcOpusEncInst {
   OpusEncoder* encoder;
-  int channels;
+  size_t channels;
   int in_dtx_mode;
   // When Opus is in DTX mode, we use |zero_counts| to count consecutive zeros
   // to break long zero segment so as to prevent DTX from going wrong. We use
   // one counter for each channel. After each encoding, |zero_counts| contain
   // the remaining zeros from the last frame.
   // TODO(minyue): remove this when Opus gets an internal fix to DTX.
   size_t* zero_counts;
 };
 
 struct WebRtcOpusDecInst {
   OpusDecoder* decoder;
   int prev_decoded_samples;
-  int channels;
+  size_t channels;
   int in_dtx_mode;
 };
 
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_OPUS_INST_H_