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

webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h

 /*
  *  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.
  */
 
 #ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_H_
 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_H_
 
 #include <vector>
 
 #include "webrtc/api/audio_codecs/audio_encoder.h"
 #include "webrtc/api/audio_codecs/audio_format.h"
-#include "webrtc/base/constructormagic.h"
-#include "webrtc/base/scoped_ref_ptr.h"
 #include "webrtc/modules/audio_coding/codecs/isac/locked_bandwidth_info.h"
+#include "webrtc/rtc_base/constructormagic.h"
+#include "webrtc/rtc_base/scoped_ref_ptr.h"
 
 namespace webrtc {
 
 struct CodecInst;
 
 template <typename T>
 class AudioEncoderIsacT final : public AudioEncoder {
  public:
   // Allowed combinations of sample rate, frame size, and bit rate are
   //  - 16000 Hz, 30 ms, 10000-32000 bps
@@ skipping 64 matching lines @@
 
   // Timestamp of the previously encoded packet.
   uint32_t last_encoded_timestamp_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderIsacT);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_H_