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

webrtc/common_audio/audio_converter_unittest.cc

 /*
  *  Copyright (c) 2014 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 <cmath>
 #include <algorithm>
 #include <memory>
 #include <vector>
 
-#include "webrtc/base/arraysize.h"
-#include "webrtc/base/format_macros.h"
 #include "webrtc/common_audio/audio_converter.h"
 #include "webrtc/common_audio/channel_buffer.h"
 #include "webrtc/common_audio/resampler/push_sinc_resampler.h"
+#include "webrtc/rtc_base/arraysize.h"
+#include "webrtc/rtc_base/format_macros.h"
 #include "webrtc/test/gtest.h"
 
 namespace webrtc {
 
 typedef std::unique_ptr<ChannelBuffer<float>> ScopedBuffer;
 
 // Sets the signal value to increase by |data| with every sample.
 ScopedBuffer CreateBuffer(const std::vector<float>& data, size_t frames) {
   const size_t num_channels = data.size();
   ScopedBuffer sb(new ChannelBuffer<float>(frames, num_channels));
@@ skipping 121 matching lines @@
              ++dst_channel) {
           RunAudioConverterTest(kChannels[src_channel], kSampleRates[src_rate],
                                 kChannels[dst_channel], kSampleRates[dst_rate]);
         }
       }
     }
   }
 }
 
 }  // namespace webrtc