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

webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.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_G722_AUDIO_ENCODER_G722_H_
 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_AUDIO_ENCODER_G722_H_
 
 #include <memory>
 
 #include "webrtc/api/audio_codecs/audio_encoder.h"
 #include "webrtc/api/audio_codecs/audio_format.h"
 #include "webrtc/api/audio_codecs/g722/audio_encoder_g722_config.h"
-#include "webrtc/base/buffer.h"
-#include "webrtc/base/constructormagic.h"
 #include "webrtc/modules/audio_coding/codecs/g722/g722_interface.h"
+#include "webrtc/rtc_base/buffer.h"
+#include "webrtc/rtc_base/constructormagic.h"
 
 namespace webrtc {
 
 struct CodecInst;
 
 class AudioEncoderG722Impl final : public AudioEncoder {
  public:
   static rtc::Optional<AudioEncoderG722Config> SdpToConfig(
       const SdpAudioFormat& format);
 
@@ skipping 36 matching lines @@
   const size_t num_10ms_frames_per_packet_;
   size_t num_10ms_frames_buffered_;
   uint32_t first_timestamp_in_buffer_;
   const std::unique_ptr<EncoderState[]> encoders_;
   rtc::Buffer interleave_buffer_;
   RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderG722Impl);
 };
 
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_AUDIO_ENCODER_G722_H_