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

webrtc/audio/audio_state.cc

 /*
  *  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.
  */
 
 #include "webrtc/audio/audio_state.h"
 
-#include "webrtc/base/atomicops.h"
-#include "webrtc/base/checks.h"
-#include "webrtc/base/logging.h"
 #include "webrtc/modules/audio_device/include/audio_device.h"
+#include "webrtc/rtc_base/atomicops.h"
+#include "webrtc/rtc_base/checks.h"
+#include "webrtc/rtc_base/logging.h"
 #include "webrtc/voice_engine/include/voe_errors.h"
 
 namespace webrtc {
 namespace internal {
 
 // TODO(peah): Remove the conditional in the audio_transport_proxy_ constructor
 // call when upstream dependencies have properly been resolved.
 AudioState::AudioState(const AudioState::Config& config)
     : config_(config),
       voe_base_(config.voice_engine),
@@ skipping 66 matching lines @@
     typing_noise_detected_ = false;
   }
 }
 }  // namespace internal
 
 rtc::scoped_refptr<AudioState> AudioState::Create(
     const AudioState::Config& config) {
   return rtc::scoped_refptr<AudioState>(new internal::AudioState(config));
 }
 }  // namespace webrtc