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

webrtc/voice_engine/mock/fake_voe_external_media.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 24 matching lines @@
     callback_map_.erase(type);
     return 0;
   }
   WEBRTC_STUB(GetAudioFrame, (int channel, int desired_sample_rate_hz,
                               AudioFrame* frame));
   WEBRTC_STUB(SetExternalMixing, (int channel, bool enable));
 
   // Use this to trigger the Process() callback to a registered media processor.
   // If |audio| is NULL, a zero array of the correct length will be forwarded.
   void CallProcess(ProcessingTypes type, int16_t* audio,
-                   int samples_per_channel, int sample_rate_hz,
+                   size_t samples_per_channel, int sample_rate_hz,
                    int num_channels) {
-    const int length = samples_per_channel * num_channels;
+    const size_t length = samples_per_channel * num_channels;
     rtc::scoped_ptr<int16_t[]> data;
     if (!audio) {
       data.reset(new int16_t[length]);
       memset(data.get(), 0, length * sizeof(data[0]));
       audio = data.get();
     }
 
     std::map<ProcessingTypes, VoEMediaProcess*>::const_iterator it =
         callback_map_.find(type);
     if (it != callback_map_.end()) {
       it->second->Process(0, type, audio, samples_per_channel, sample_rate_hz,
                           num_channels == 2 ? true : false);
     }
   }
 
  private:
   std::map<ProcessingTypes, VoEMediaProcess*> callback_map_;
 };
 
 }  // namespace webrtc
 
 #endif  //  WEBRTC_VOICE_ENGINE_INCLUDE_MOCK_FAKE_VOE_EXTERNAL_MEDIA_H_