Remove <iostream> include from file_audio_device.cc

webrtc/modules/audio_device/dummy/file_audio_device.cc

 /*
  *  Copyright (c) 2014 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 <iostream>
 #include "webrtc/modules/audio_device/dummy/file_audio_device.h"
 #include "webrtc/system_wrappers/include/sleep.h"
 #include "webrtc/system_wrappers/include/thread_wrapper.h"
 
 namespace webrtc {
 
-int kRecordingFixedSampleRate = 48000;
-int kRecordingNumChannels = 2;
-int kPlayoutFixedSampleRate = 48000;
-int kPlayoutNumChannels = 2;
-int kPlayoutBufferSize = kPlayoutFixedSampleRate / 100
-                         * kPlayoutNumChannels * 2;
-int kRecordingBufferSize = kRecordingFixedSampleRate / 100
-                           * kRecordingNumChannels * 2;
+const int kRecordingFixedSampleRate = 48000;
+const int kRecordingNumChannels = 2;
+const int kPlayoutFixedSampleRate = 48000;
+const int kPlayoutNumChannels = 2;
+const int kPlayoutBufferSize = kPlayoutFixedSampleRate / 100
+                               * kPlayoutNumChannels * 2;
+const int kRecordingBufferSize = kRecordingFixedSampleRate / 100
+                                 * kRecordingNumChannels * 2;
 
 FileAudioDevice::FileAudioDevice(const int32_t id,
                                  const char* inputFilename,
                                  const char* outputFilename):
     _ptrAudioBuffer(NULL),
     _recordingBuffer(NULL),
     _playoutBuffer(NULL),
     _recordingFramesLeft(0),
     _playoutFramesLeft(0),
     _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
@@ skipping 507 matching lines @@
         _critSect.Enter();
       }
     }
 
     _critSect.Leave();
     SleepMs(10 - (_clock->CurrentNtpInMilliseconds() - currentTime));
     return true;
 }
 
 }  // namespace webrtc