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

webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.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 "webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h"
 
 #include <algorithm>
 #include <iterator>
 #include <utility>
 
-#include "webrtc/base/arraysize.h"
-#include "webrtc/base/checks.h"
-#include "webrtc/base/logging.h"
-#include "webrtc/base/numerics/exp_filter.h"
-#include "webrtc/base/protobuf_utils.h"
-#include "webrtc/base/ptr_util.h"
-#include "webrtc/base/safe_conversions.h"
-#include "webrtc/base/safe_minmax.h"
-#include "webrtc/base/string_to_number.h"
-#include "webrtc/base/timeutils.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h"
 #include "webrtc/modules/audio_coding/audio_network_adaptor/controller_manager.h"
 #include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
+#include "webrtc/rtc_base/arraysize.h"
+#include "webrtc/rtc_base/checks.h"
+#include "webrtc/rtc_base/logging.h"
+#include "webrtc/rtc_base/numerics/exp_filter.h"
+#include "webrtc/rtc_base/protobuf_utils.h"
+#include "webrtc/rtc_base/ptr_util.h"
+#include "webrtc/rtc_base/safe_conversions.h"
+#include "webrtc/rtc_base/safe_minmax.h"
+#include "webrtc/rtc_base/string_to_number.h"
+#include "webrtc/rtc_base/timeutils.h"
 #include "webrtc/system_wrappers/include/field_trial.h"
 
 namespace webrtc {
 
 namespace {
 
 // Codec parameters for Opus.
 // draft-spittka-payload-rtp-opus-03
 
 // Recommended bitrates:
@@ skipping 727 matching lines @@
             config_.uplink_bandwidth_update_interval_ms) {
       rtc::Optional<float> smoothed_bitrate = bitrate_smoother_->GetAverage();
       if (smoothed_bitrate)
         audio_network_adaptor_->SetUplinkBandwidth(*smoothed_bitrate);
       bitrate_smoother_last_update_time_ = rtc::Optional<int64_t>(now_ms);
     }
   }
 }
 
 }  // namespace webrtc