Convert channel counts to size_t.

webrtc/modules/audio_processing/gain_control_impl.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 50 matching lines @@
   int set_analog_level_limits(int minimum, int maximum) override;
   int analog_level_minimum() const override;
   int analog_level_maximum() const override;
   bool stream_is_saturated() const override;
 
   // ProcessingComponent implementation.
   void* CreateHandle() const override;
   int InitializeHandle(void* handle) const override;
   int ConfigureHandle(void* handle) const override;
   void DestroyHandle(void* handle) const override;
-  int num_handles_required() const override;
+  size_t num_handles_required() const override;
   int GetHandleError(void* handle) const override;
 
   void AllocateRenderQueue();
 
   // Not guarded as its public API is thread safe.
   const AudioProcessing* apm_;
 
   rtc::CriticalSection* const crit_render_ ACQUIRED_BEFORE(crit_capture_);
   rtc::CriticalSection* const crit_capture_;
 
@@ skipping 14 matching lines @@
   std::vector<int16_t> capture_queue_buffer_ GUARDED_BY(crit_capture_);
 
   // Lock protection not needed.
   rtc::scoped_ptr<
       SwapQueue<std::vector<int16_t>, RenderQueueItemVerifier<int16_t>>>
       render_signal_queue_;
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_GAIN_CONTROL_IMPL_H_