Update includes for webrtc/{base => rtc_base} rename (1/3)

webrtc/modules/audio_mixer/frame_combiner_unittest.cc

 /*
  *  Copyright (c) 2017 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/modules/audio_mixer/frame_combiner.h"
 
 #include <numeric>
 #include <sstream>
 #include <string>
 
 #include "webrtc/audio/utility/audio_frame_operations.h"
-#include "webrtc/base/checks.h"
 #include "webrtc/modules/audio_mixer/gain_change_calculator.h"
 #include "webrtc/modules/audio_mixer/sine_wave_generator.h"
+#include "webrtc/rtc_base/checks.h"
 #include "webrtc/test/gtest.h"
 
 namespace webrtc {
 
 namespace {
 std::string ProduceDebugText(int sample_rate_hz,
                              int number_of_channels,
                              int number_of_sources) {
   std::ostringstream ss;
   ss << "Sample rate: " << sample_rate_hz << " ,";
@@ skipping 166 matching lines @@
               rtc::ArrayView<const int16_t>(frame1.data(), number_of_samples),
               rtc::ArrayView<const int16_t>(audio_frame_for_mixing.data(),
                                             number_of_samples));
         }
         RTC_DCHECK_LT(cumulative_change, 10);
       }
     }
   }
 }
 }  // namespace webrtc