Move and rename FdopenPlatformFile to file_util

content/renderer/media/media_stream_audio_processor_options.h

 // Copyright 2013 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #ifndef CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_PROCESSOR_OPTIONS_H_
 #define CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_PROCESSOR_OPTIONS_H_
 
 #include <string>
 
-#include "base/platform_file.h"
+#include "base/files/file.h"
 #include "content/common/content_export.h"
 #include "third_party/WebKit/public/platform/WebMediaConstraints.h"
 #include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h"
 
 namespace webrtc {
 
 class AudioFrame;
 class AudioProcessing;
 class MediaConstraintsInterface;
 class TypingDetection;
@@ skipping 81 matching lines @@
 
 // Enables the typing detection in |audio_processing|.
 void EnableTypingDetection(AudioProcessing* audio_processing,
                            webrtc::TypingDetection* typing_detector);
 
 // Enables the experimental echo cancellation in |audio_processing|.
 void EnableExperimentalEchoCancellation(AudioProcessing* audio_processing);
 
 // Starts the echo cancellation dump in |audio_processing|.
 void StartEchoCancellationDump(AudioProcessing* audio_processing,
-                               const base::PlatformFile& aec_dump_file);
+                               base::File aec_dump_file);
 
 // Stops the echo cancellation dump in |audio_processing|.
 // This method has no impact if echo cancellation dump has not been started on
 // |audio_processing|.
 void StopEchoCancellationDump(AudioProcessing* audio_processing);
 
 void EnableAutomaticGainControl(AudioProcessing* audio_processing);
 
 void GetAecStats(AudioProcessing* audio_processing,
                  webrtc::AudioProcessorInterface::AudioProcessorStats* stats);
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_PROCESSOR_OPTIONS_H_