Misc. small cleanups

webrtc/modules/audio_processing/gain_control_impl.cc

 /*
  *  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 93 matching lines @@
 // Read chunks of data that were received and queued on the render side from
 // a queue. All the data chunks are buffered into the farend signal of the AGC.
 void GainControlImpl::ReadQueuedRenderData() {
   rtc::CritScope cs(crit_capture_);
 
   if (!is_component_enabled()) {
     return;
   }
 
   while (render_signal_queue_->Remove(&capture_queue_buffer_)) {
-    int buffer_index = 0;
-    const int num_frames_per_band =
+    size_t buffer_index = 0;
+    const size_t num_frames_per_band =
         capture_queue_buffer_.size() / num_handles();
     for (int i = 0; i < num_handles(); i++) {
       Handle* my_handle = static_cast<Handle*>(handle(i));
       WebRtcAgc_AddFarend(my_handle, &capture_queue_buffer_[buffer_index],
                           num_frames_per_band);
 
       buffer_index += num_frames_per_band;
     }
   }
 }
@@ skipping 312 matching lines @@
   return apm_->num_output_channels();
 }
 
 int GainControlImpl::GetHandleError(void* handle) const {
   // The AGC has no get_error() function.
   // (Despite listing errors in its interface...)
   assert(handle != NULL);
   return AudioProcessing::kUnspecifiedError;
 }
 }  // namespace webrtc