Move gtest_prod_util.h out of webrtc/test tree.

webrtc/common_audio/resampler/sinc_resampler.h

 /*
  *  Copyright (c) 2013 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.
  */
 
 // Modified from the Chromium original here:
 // src/media/base/sinc_resampler.h
 
 #ifndef WEBRTC_COMMON_AUDIO_RESAMPLER_SINC_RESAMPLER_H_
 #define WEBRTC_COMMON_AUDIO_RESAMPLER_SINC_RESAMPLER_H_
 
 #include "webrtc/base/constructormagic.h"
+#include "webrtc/base/gtest_prod_util.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/system_wrappers/include/aligned_malloc.h"
-#include "webrtc/test/testsupport/gtest_prod_util.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
 // Callback class for providing more data into the resampler.  Expects |frames|
 // of data to be rendered into |destination|; zero padded if not enough frames
 // are available to satisfy the request.
 class SincResamplerCallback {
  public:
   virtual ~SincResamplerCallback() {}
@@ skipping 130 matching lines @@
   float* const r2_;
   float* r3_;
   float* r4_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(SincResampler);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_COMMON_AUDIO_RESAMPLER_SINC_RESAMPLER_H_