[NOT FOR REVIEW] Convert channel counts to size_t.

webrtc/common_audio/audio_converter.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_COMMON_AUDIO_AUDIO_CONVERTER_H_
 #define WEBRTC_COMMON_AUDIO_AUDIO_CONVERTER_H_
 
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/scoped_ptr.h"
 
 namespace webrtc {
 
 // Format conversion (remixing and resampling) for audio. Only simple remixing
 // conversions are supported: downmix to mono (i.e. |dst_channels| == 1) or
 // upmix from mono (i.e. |src_channels == 1|).
 //
 // The source and destination chunks have the same duration in time; specifying
 // the number of frames is equivalent to specifying the sample rates.
 class AudioConverter {
  public:
   // Returns a new AudioConverter, which will use the supplied format for its
   // lifetime. Caller is responsible for the memory.
-  static rtc::scoped_ptr<AudioConverter> Create(int src_channels,
+  static rtc::scoped_ptr<AudioConverter> Create(size_t src_channels,
                                                 size_t src_frames,
-                                                int dst_channels,
+                                                size_t dst_channels,
                                                 size_t dst_frames);
   virtual ~AudioConverter() {};
 
   // Convert |src|, containing |src_size| samples, to |dst|, having a sample
   // capacity of |dst_capacity|. Both point to a series of buffers containing
   // the samples for each channel. The sizes must correspond to the format
   // passed to Create().
   virtual void Convert(const float* const* src, size_t src_size,
                        float* const* dst, size_t dst_capacity) = 0;
 
-  int src_channels() const { return src_channels_; }
+  size_t src_channels() const { return src_channels_; }
   size_t src_frames() const { return src_frames_; }
-  int dst_channels() const { return dst_channels_; }
+  size_t dst_channels() const { return dst_channels_; }
   size_t dst_frames() const { return dst_frames_; }
 
  protected:
   AudioConverter();
-  AudioConverter(int src_channels, size_t src_frames, int dst_channels,
+  AudioConverter(size_t src_channels, size_t src_frames, size_t dst_channels,
                  size_t dst_frames);
 
   // Helper to CHECK that inputs are correctly sized.
   void CheckSizes(size_t src_size, size_t dst_capacity) const;
 
  private:
-  const int src_channels_;
+  const size_t src_channels_;
   const size_t src_frames_;
-  const int dst_channels_;
+  const size_t dst_channels_;
   const size_t dst_frames_;
 
   DISALLOW_COPY_AND_ASSIGN(AudioConverter);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_COMMON_AUDIO_AUDIO_CONVERTER_H_