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

webrtc/call/audio_state.h

 /*
  *  Copyright (c) 2015 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_CALL_AUDIO_STATE_H_
 #define WEBRTC_CALL_AUDIO_STATE_H_
 
 #include "webrtc/api/audio/audio_mixer.h"
-#include "webrtc/base/refcount.h"
-#include "webrtc/base/scoped_ref_ptr.h"
+#include "webrtc/rtc_base/refcount.h"
+#include "webrtc/rtc_base/scoped_ref_ptr.h"
 
 namespace webrtc {
 
 class AudioProcessing;
 class VoiceEngine;
 
 // WORK IN PROGRESS
 // This class is under development and is not yet intended for for use outside
 // of WebRtc/Libjingle. Please use the VoiceEngine API instead.
 // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4690
@@ skipping 20 matching lines @@
 
   // TODO(solenberg): Replace scoped_refptr with shared_ptr once we can use it.
   static rtc::scoped_refptr<AudioState> Create(
       const AudioState::Config& config);
 
   virtual ~AudioState() {}
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_CALL_AUDIO_STATE_H_