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

webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc

 /*
  *  Copyright (c) 2012 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/modules/rtp_rtcp/source/rtp_receiver_audio.h"
 
 #include <assert.h>  // assert
 #include <math.h>   // pow()
 #include <string.h>  // memcpy()
 
 #include "webrtc/common_types.h"
-#include "webrtc/rtc_base/logging.h"
-#include "webrtc/rtc_base/trace_event.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/base/trace_event.h"
 
 namespace webrtc {
 RTPReceiverStrategy* RTPReceiverStrategy::CreateAudioStrategy(
     RtpData* data_callback) {
   return new RTPReceiverAudio(data_callback);
 }
 
 RTPReceiverAudio::RTPReceiverAudio(RtpData* data_callback)
     : RTPReceiverStrategy(data_callback),
       TelephoneEventHandler(),
@@ skipping 274 matching lines @@
     // only one frame in the RED strip the one byte to help NetEq
     return data_callback_->OnReceivedPayloadData(
         payload_data + 1, payload_data_length - 1, rtp_header);
   }
 
   rtp_header->type.Audio.channel = audio_specific.channels;
   return data_callback_->OnReceivedPayloadData(payload_data,
                                                payload_data_length, rtp_header);
 }
 }  // namespace webrtc