Move GetFileSize, NormalizeFilePath to base namespace

content/renderer/media/webrtc_audio_device_unittest.cc

 // Copyright (c) 2012 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.
 
 #include <vector>
 
 #include "base/environment.h"
 #include "base/file_util.h"
 #include "base/files/file_path.h"
 #include "base/path_service.h"
@@ skipping 276 matching lines @@
 void ReadDataFromSpeechFile(char* data, int length) {
   base::FilePath data_file;
   CHECK(PathService::Get(base::DIR_SOURCE_ROOT, &data_file));
   data_file =
       data_file.Append(FILE_PATH_LITERAL("media"))
                .Append(FILE_PATH_LITERAL("test"))
                .Append(FILE_PATH_LITERAL("data"))
                .Append(FILE_PATH_LITERAL("speech_16b_stereo_48kHz.raw"));
   DCHECK(base::PathExists(data_file));
   int64 data_file_size64 = 0;
-  DCHECK(file_util::GetFileSize(data_file, &data_file_size64));
+  DCHECK(base::GetFileSize(data_file, &data_file_size64));
   EXPECT_EQ(length, file_util::ReadFile(data_file, data, length));
   DCHECK(data_file_size64 > length);
 }
 
 void SetChannelCodec(webrtc::VoiceEngine* engine, int channel) {
   // TODO(xians): move the codec as an input param to this function, and add
   // tests for different codecs, also add support to Android and IOS.
 #if !defined(OS_ANDROID) && !defined(OS_IOS)
   webrtc::CodecInst isac;
   strcpy(isac.plname, "ISAC");
@@ skipping 660 matching lines @@
     LOG(WARNING) << "Test disabled due to the test hangs on WinXP.";
     return;
   }
 #endif
   int latency = RunWebRtcLoopbackTimeTest(audio_manager_.get(), true);
   PrintPerfResultMs("webrtc_loopback_with_signal_processing (100 packets)",
                     "t", latency);
 }
 
 }  // namespace content