Match existing type usage better.

talk/app/webrtc/test/fakeaudiocapturemodule.cc

 /*
  * libjingle
  * Copyright 2012 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
@@ skipping 27 matching lines @@
 // Even simpler buffers would likely just contain audio sample values of 0.
 static const int kHighSampleValue = 10000;
 
 // Same value as src/modules/audio_device/main/source/audio_device_config.h in
 // https://code.google.com/p/webrtc/
 static const uint32 kAdmMaxIdleTimeProcess = 1000;
 
 // Constants here are derived by running VoE using a real ADM.
 // The constants correspond to 10ms of mono audio at 44kHz.
 static const int kTimePerFrameMs = 10;
-static const int kNumberOfChannels = 1;
+static const uint8_t kNumberOfChannels = 1;
 static const int kSamplesPerSecond = 44000;
 static const int kTotalDelayMs = 0;
 static const int kClockDriftMs = 0;
 static const uint32_t kMaxVolume = 14392;
 
 enum {
   MSG_START_PROCESS,
   MSG_RUN_PROCESS,
   MSG_STOP_PROCESS,
 };
@@ skipping 686 matching lines @@
     ASSERT(false);
   }
   SetMicrophoneVolume(current_mic_level);
 }
 
 void FakeAudioCaptureModule::StopProcessP() {
   ASSERT(rtc::Thread::Current() == process_thread_);
   started_ = false;
   process_thread_->Clear(this);
 }