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

webrtc/modules/audio_coding/codecs/opus/opus_complexity_unittest.cc

 /*
  *  Copyright (c) 2016 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/base/format_macros.h"
-#include "webrtc/base/timeutils.h"
 #include "webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h"
 #include "webrtc/modules/audio_coding/neteq/tools/audio_loop.h"
+#include "webrtc/rtc_base/format_macros.h"
+#include "webrtc/rtc_base/timeutils.h"
 #include "webrtc/test/gtest.h"
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/test/testsupport/perf_test.h"
 
 namespace webrtc {
 
 namespace {
 int64_t RunComplexityTest(const AudioEncoderOpusConfig& config) {
   // Create encoder.
   constexpr int payload_type = 17;
@@ skipping 64 matching lines @@
   int64_t runtime_10999bps = RunComplexityTest(config);
 
   config.bitrate_bps = rtc::Optional<int>(15500);
   int64_t runtime_15500bps = RunComplexityTest(config);
 
   test::PrintResult("opus_encoding_complexity_ratio", "", "adaptation_off",
                     100.0 * runtime_10999bps / runtime_15500bps, "percent",
                     true);
 }
 }  // namespace webrtc