Update audio code to use size_t more correctly, webrtc/voice_engine/ portion.

webrtc/voice_engine/test/auto_test/fakes/fake_media_process.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.
  */
 #ifndef VOICE_ENGINE_MAIN_TEST_AUTO_TEST_FAKE_MEDIA_PROCESS_H_
 #define VOICE_ENGINE_MAIN_TEST_AUTO_TEST_FAKE_MEDIA_PROCESS_H_
 
 #include <math.h>
 
 class FakeMediaProcess : public webrtc::VoEMediaProcess {
  public:
   FakeMediaProcess() : frequency(0) {}
   virtual void Process(int channel,
                        const webrtc::ProcessingTypes type,
                        int16_t audio_10ms[],
-                       int length,
+                       size_t length,
                        int sampling_freq_hz,
                        bool stereo) {
-    for (int i = 0; i < length; i++) {
+    for (size_t i = 0; i < length; i++) {
       if (!stereo) {
         audio_10ms[i] = static_cast<int16_t>(audio_10ms[i] *
             sin(2.0 * 3.14 * frequency * 400.0 / sampling_freq_hz));
       } else {
         // Interleaved stereo.
         audio_10ms[2 * i] = static_cast<int16_t> (
             audio_10ms[2 * i] * sin(2.0 * 3.14 *
                 frequency * 400.0 / sampling_freq_hz));
         audio_10ms[2 * i + 1] = static_cast<int16_t> (
             audio_10ms[2 * i + 1] * sin(2.0 * 3.14 *
                 frequency * 400.0 / sampling_freq_hz));
       }
       frequency++;
     }
   }
 
  private:
   int frequency;
 };
 
 #endif  // VOICE_ENGINE_MAIN_TEST_AUTO_TEST_FAKE_MEDIA_PROCESS_H_