[NOT FOR REVIEW] Convert channel counts to size_t.

webrtc/modules/audio_processing/audio_buffer.h

 /*
  *  Copyright (c) 2011 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 16 matching lines @@
 enum Band {
   kBand0To8kHz = 0,
   kBand8To16kHz = 1,
   kBand16To24kHz = 2
 };
 
 class AudioBuffer {
  public:
   // TODO(ajm): Switch to take ChannelLayouts.
   AudioBuffer(size_t input_num_frames,
-              int num_input_channels,
+              size_t num_input_channels,
               size_t process_num_frames,
-              int num_process_channels,
+              size_t num_process_channels,
               size_t output_num_frames);
   virtual ~AudioBuffer();
 
-  int num_channels() const;
-  void set_num_channels(int num_channels);
+  size_t num_channels() const;
+  void set_num_channels(size_t num_channels);
   size_t num_frames() const;
   size_t num_frames_per_band() const;
   size_t num_keyboard_frames() const;
   size_t num_bands() const;
 
   // Returns a pointer array to the full-band channels.
   // Usage:
   // channels()[channel][sample].
   // Where:
   // 0 <= channel < |num_proc_channels_|
   // 0 <= sample < |proc_num_frames_|
   int16_t* const* channels();
   const int16_t* const* channels_const() const;
   float* const* channels_f();
   const float* const* channels_const_f() const;
 
   // Returns a pointer array to the bands for a specific channel.
   // Usage:
   // split_bands(channel)[band][sample].
   // Where:
   // 0 <= channel < |num_proc_channels_|
   // 0 <= band < |num_bands_|
   // 0 <= sample < |num_split_frames_|
-  int16_t* const* split_bands(int channel);
-  const int16_t* const* split_bands_const(int channel) const;
-  float* const* split_bands_f(int channel);
-  const float* const* split_bands_const_f(int channel) const;
+  int16_t* const* split_bands(size_t channel);
+  const int16_t* const* split_bands_const(size_t channel) const;
+  float* const* split_bands_f(size_t channel);
+  const float* const* split_bands_const_f(size_t channel) const;
 
   // Returns a pointer array to the channels for a specific band.
   // Usage:
   // split_channels(band)[channel][sample].
   // Where:
   // 0 <= band < |num_bands_|
   // 0 <= channel < |num_proc_channels_|
   // 0 <= sample < |num_split_frames_|
   int16_t* const* split_channels(Band band);
   const int16_t* const* split_channels_const(Band band) const;
@@ skipping 39 matching lines @@
   // Recombine the different bands into one signal.
   void MergeFrequencyBands();
 
  private:
   // Called from DeinterleaveFrom() and CopyFrom().
   void InitForNewData();
 
   // The audio is passed into DeinterleaveFrom() or CopyFrom() with input
   // format (samples per channel and number of channels).
   const size_t input_num_frames_;
-  const int num_input_channels_;
+  const size_t num_input_channels_;
   // The audio is stored by DeinterleaveFrom() or CopyFrom() with processing
   // format.
   const size_t proc_num_frames_;
-  const int num_proc_channels_;
+  const size_t num_proc_channels_;
   // The audio is returned by InterleaveTo() and CopyTo() with output samples
   // per channels and the current number of channels. This last one can be
   // changed at any time using set_num_channels().
   const size_t output_num_frames_;
-  int num_channels_;
+  size_t num_channels_;
 
   size_t num_bands_;
   size_t num_split_frames_;
   bool mixed_low_pass_valid_;
   bool reference_copied_;
   AudioFrame::VADActivity activity_;
 
   const float* keyboard_data_;
   rtc::scoped_ptr<IFChannelBuffer> data_;
   rtc::scoped_ptr<IFChannelBuffer> split_data_;
   rtc::scoped_ptr<SplittingFilter> splitting_filter_;
   rtc::scoped_ptr<ChannelBuffer<int16_t> > mixed_low_pass_channels_;
   rtc::scoped_ptr<ChannelBuffer<int16_t> > low_pass_reference_channels_;
   rtc::scoped_ptr<IFChannelBuffer> input_buffer_;
   rtc::scoped_ptr<ChannelBuffer<float> > process_buffer_;
   ScopedVector<PushSincResampler> input_resamplers_;
   ScopedVector<PushSincResampler> output_resamplers_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_BUFFER_H_