[NOT FOR REVIEW] Convert channel counts to size_t.

webrtc/modules/audio_processing/include/audio_processing.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 248 matching lines @@
   // TODO(ajm): Remove after voice engine no longer requires it to resample
   // the reverse stream to the forward rate.
   virtual int input_sample_rate_hz() const = 0;
   // TODO(ajm): Remove after Chromium no longer depends on it.
   virtual int sample_rate_hz() const = 0;
 
   // TODO(ajm): Only intended for internal use. Make private and friend the
   // necessary classes?
   virtual int proc_sample_rate_hz() const = 0;
   virtual int proc_split_sample_rate_hz() const = 0;
-  virtual int num_input_channels() const = 0;
-  virtual int num_output_channels() const = 0;
-  virtual int num_reverse_channels() const = 0;
+  virtual size_t num_input_channels() const = 0;
+  virtual size_t num_output_channels() const = 0;
+  virtual size_t num_reverse_channels() const = 0;
 
   // Set to true when the output of AudioProcessing will be muted or in some
   // other way not used. Ideally, the captured audio would still be processed,
   // but some components may change behavior based on this information.
   // Default false.
   virtual void set_output_will_be_muted(bool muted) = 0;
   virtual bool output_will_be_muted() const = 0;
 
   // Processes a 10 ms |frame| of the primary audio stream. On the client-side,
   // this is the near-end (or captured) audio.
@@ skipping 181 matching lines @@
   //               keyboard channel if it is present. When passing a
   //               StreamConfig with an array of arrays T*[N],
   //
   //                N == {num_channels + 1  if  has_keyboard
   //                     {num_channels      if  !has_keyboard
   //
   // has_keyboard: True if the stream has a keyboard channel. When has_keyboard
   //               is true, the last channel in any corresponding list of
   //               channels is the keyboard channel.
   StreamConfig(int sample_rate_hz = 0,
-               int num_channels = 0,
+               size_t num_channels = 0,
                bool has_keyboard = false)
       : sample_rate_hz_(sample_rate_hz),
         num_channels_(num_channels),
         has_keyboard_(has_keyboard),
         num_frames_(calculate_frames(sample_rate_hz)) {}
 
   void set_sample_rate_hz(int value) {
     sample_rate_hz_ = value;
     num_frames_ = calculate_frames(value);
   }
-  void set_num_channels(int value) { num_channels_ = value; }
+  void set_num_channels(size_t value) { num_channels_ = value; }
   void set_has_keyboard(bool value) { has_keyboard_ = value; }
 
   int sample_rate_hz() const { return sample_rate_hz_; }
 
   // The number of channels in the stream, not including the keyboard channel if
   // present.
-  int num_channels() const { return num_channels_; }
+  size_t num_channels() const { return num_channels_; }
 
   bool has_keyboard() const { return has_keyboard_; }
   size_t num_frames() const { return num_frames_; }
 
   bool operator==(const StreamConfig& other) const {
     return sample_rate_hz_ == other.sample_rate_hz_ &&
            num_channels_ == other.num_channels_ &&
            has_keyboard_ == other.has_keyboard_;
   }
 
   bool operator!=(const StreamConfig& other) const { return !(*this == other); }
 
  private:
   static size_t calculate_frames(int sample_rate_hz) {
     return static_cast<size_t>(
         AudioProcessing::kChunkSizeMs * sample_rate_hz / 1000);
   }
 
   int sample_rate_hz_;
-  int num_channels_;
+  size_t num_channels_;
   bool has_keyboard_;
   size_t num_frames_;
 };
 
 class ProcessingConfig {
  public:
   enum StreamName {
     kInputStream,
     kOutputStream,
     kReverseStream,
@@ skipping 385 matching lines @@
   // This does not impact the size of frames passed to |ProcessStream()|.
   virtual int set_frame_size_ms(int size) = 0;
   virtual int frame_size_ms() const = 0;
 
  protected:
   virtual ~VoiceDetection() {}
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_PROCESSING_H_