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

webrtc/voice_engine/transmit_mixer_unittest.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.
  */
 
 #include "webrtc/voice_engine/transmit_mixer.h"
 
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/voice_engine/include/voe_external_media.h"
 
 namespace webrtc {
 namespace voe {
 namespace {
 
 class MediaCallback : public VoEMediaProcess {
  public:
   virtual void Process(int channel, ProcessingTypes type,
-                       int16_t audio[], int samples_per_channel,
+                       int16_t audio[], size_t samples_per_channel,
                        int sample_rate_hz, bool is_stereo) {
   }
 };
 
 // TODO(andrew): Mock VoEMediaProcess, and verify the behavior when calling
 // PrepareDemux().
 TEST(TransmitMixerTest, RegisterExternalMediaCallback) {
   TransmitMixer* tm = NULL;
   ASSERT_EQ(0, TransmitMixer::Create(tm, 0));
   ASSERT_TRUE(tm != NULL);
@@ skipping 16 matching lines @@
   EXPECT_EQ(-1, tm->DeRegisterExternalMediaProcessing(kRecordingPerChannel));
   EXPECT_EQ(0, tm->DeRegisterExternalMediaProcessing(
                    kRecordingAllChannelsMixed));
   EXPECT_EQ(0, tm->DeRegisterExternalMediaProcessing(kRecordingPreprocessing));
   TransmitMixer::Destroy(tm);
 }
 
 }  // namespace
 }  // namespace voe
 }  // namespace webrtc