Update audio code to use size_t more correctly, webrtc/modules/audio_processing/

webrtc/modules/audio_processing/agc/agc_manager_direct.h

 /*
  *  Copyright (c) 2013 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 39 matching lines @@
   // by the manager.
   AgcManagerDirect(Agc* agc,
                    GainControl* gctrl,
                    VolumeCallbacks* volume_callbacks,
                    int startup_min_level);
   ~AgcManagerDirect();
 
   int Initialize();
   void AnalyzePreProcess(int16_t* audio,
                          int num_channels,
-                         int samples_per_channel);
-  void Process(const int16_t* audio, int length, int sample_rate_hz);
+                         size_t samples_per_channel);
+  void Process(const int16_t* audio, size_t length, int sample_rate_hz);
 
   // Sets a new microphone level, after first checking that it hasn't been
   // updated by the user, in which case no action is taken.
   void SetLevel(int new_level);
 
   // Set the maximum level the AGC is allowed to apply. Also updates the
   // maximum compression gain to compensate. The level must be at least
   // |kClippedLevelMin|.
   void SetMaxLevel(int level);
 
@@ skipping 23 matching lines @@
   bool startup_;
   int startup_min_level_;
 
   rtc::scoped_ptr<DebugFile> file_preproc_;
   rtc::scoped_ptr<DebugFile> file_postproc_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AGC_AGC_MANAGER_DIRECT_H_