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

webrtc/audio/audio_receive_stream.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_receive_stream.h"
 
 #include <string>
 #include <utility>
 
 #include "webrtc/api/call/audio_sink.h"
 #include "webrtc/audio/audio_send_stream.h"
 #include "webrtc/audio/audio_state.h"
 #include "webrtc/audio/conversion.h"
-#include "webrtc/base/checks.h"
-#include "webrtc/base/logging.h"
-#include "webrtc/base/timeutils.h"
 #include "webrtc/call/rtp_stream_receiver_controller_interface.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
+#include "webrtc/rtc_base/checks.h"
+#include "webrtc/rtc_base/logging.h"
+#include "webrtc/rtc_base/timeutils.h"
 #include "webrtc/voice_engine/channel_proxy.h"
 #include "webrtc/voice_engine/include/voe_base.h"
 #include "webrtc/voice_engine/voice_engine_impl.h"
 
 namespace webrtc {
 
 std::string AudioReceiveStream::Config::Rtp::ToString() const {
   std::stringstream ss;
   ss << "{remote_ssrc: " << remote_ssrc;
   ss << ", local_ssrc: " << local_ssrc;
@@ skipping 302 matching lines @@
 int AudioReceiveStream::SetVoiceEnginePlayout(bool playout) {
   ScopedVoEInterface<VoEBase> base(voice_engine());
   if (playout) {
     return base->StartPlayout(config_.voe_channel_id);
   } else {
     return base->StopPlayout(config_.voe_channel_id);
   }
 }
 }  // namespace internal
 }  // namespace webrtc