Update a ton of audio code to use size_t more correctly and in general reduce

webrtc/voice_engine/include/voe_external_media.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.
  */
 #ifndef WEBRTC_VOICE_ENGINE_VOE_EXTERNAL_MEDIA_H
@@ skipping 13 matching lines @@
   // be processed. The audio can be accessed in several different modes
   // given by the |type| parameter. The function should modify the
   // original data and ensure that it is copied back to the |audio10ms|
   // array. The number of samples in the frame cannot be changed.
   // The sampling frequency will depend upon the codec used.
   // If |isStereo| is true, audio10ms will contain 16-bit PCM data
   // samples in interleaved stereo format (L0,R0,L1,R1,...).
   virtual void Process(int channel,
                        ProcessingTypes type,
                        int16_t audio10ms[],
-                       int length,
+                       size_t length,
                        int samplingFreq,
                        bool isStereo) = 0;
 
  protected:
   virtual ~VoEMediaProcess() {}
 };
 
 class WEBRTC_DLLEXPORT VoEExternalMedia {
  public:
   // Factory for the VoEExternalMedia sub-API. Increases an internal
@@ skipping 31 matching lines @@
   virtual int SetExternalMixing(int channel, bool enable) = 0;
 
  protected:
   VoEExternalMedia() {}
   virtual ~VoEExternalMedia() {}
 };
 
 }  // namespace webrtc
 
 #endif  //  WEBRTC_VOICE_ENGINE_VOE_EXTERNAL_MEDIA_H