Update includes for webrtc/{base => rtc_base} rename (2/3)

webrtc/audio/utility/audio_frame_operations.cc

 /*
  *  Copyright (c) 2012 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 "webrtc/audio/utility/audio_frame_operations.h"
 
 #include <algorithm>
 
-#include "webrtc/base/checks.h"
-#include "webrtc/base/safe_conversions.h"
 #include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/rtc_base/checks.h"
+#include "webrtc/rtc_base/safe_conversions.h"
 
 namespace webrtc {
 namespace {
 
 // 2.7ms @ 48kHz, 4ms @ 32kHz, 8ms @ 16kHz.
 const size_t kMuteFadeFrames = 128;
 const float kMuteFadeInc = 1.0f / kMuteFadeFrames;
 
 }  // namespace
 
@@ skipping 293 matching lines @@
   }
 
   int16_t* frame_data = frame->mutable_data();
   for (size_t i = 0; i < frame->samples_per_channel_ * frame->num_channels_;
        i++) {
     frame_data[i] = rtc::saturated_cast<int16_t>(scale * frame_data[i]);
   }
   return 0;
 }
 }  // namespace webrtc