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

webrtc/audio/audio_send_stream.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_AUDIO_AUDIO_SEND_STREAM_H_
 #define WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_
 
 #include <memory>
 #include <vector>
 
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/base/thread_checker.h"
 #include "webrtc/call/audio_send_stream.h"
 #include "webrtc/call/audio_state.h"
 #include "webrtc/call/bitrate_allocator.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
-#include "webrtc/rtc_base/constructormagic.h"
-#include "webrtc/rtc_base/thread_checker.h"
 #include "webrtc/voice_engine/transport_feedback_packet_loss_tracker.h"
 
 namespace webrtc {
 class VoiceEngine;
 class RtcEventLog;
 class RtcpBandwidthObserver;
 class RtcpRttStats;
 class RtpTransportControllerSendInterface;
 
 namespace voe {
@@ skipping 83 matching lines @@
 
   RtpRtcp* rtp_rtcp_module_;
   rtc::Optional<RtpState> const suspended_rtp_state_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioSendStream);
 };
 }  // namespace internal
 }  // namespace webrtc
 
 #endif  // WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_