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

webrtc/modules/rtp_rtcp/source/rtp_sender_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_sender_audio.h"
 
 #include <string.h>
 
 #include <memory>
 #include <utility>
 
+#include "webrtc/base/logging.h"
+#include "webrtc/base/timeutils.h"
+#include "webrtc/base/trace_event.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h"
-#include "webrtc/rtc_base/logging.h"
-#include "webrtc/rtc_base/timeutils.h"
-#include "webrtc/rtc_base/trace_event.h"
 
 namespace webrtc {
 
 RTPSenderAudio::RTPSenderAudio(Clock* clock, RTPSender* rtp_sender)
     : clock_(clock),
       rtp_sender_(rtp_sender) {}
 
 RTPSenderAudio::~RTPSenderAudio() {}
 
 int32_t RTPSenderAudio::RegisterAudioPayload(
@@ skipping 305 matching lines @@
         TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), "Audio::SendTelephoneEvent",
         "timestamp", packet->Timestamp(), "seqnum", packet->SequenceNumber());
     result = rtp_sender_->SendToNetwork(std::move(packet), kAllowRetransmission,
                                         RtpPacketSender::kHighPriority);
     send_count--;
   } while (send_count > 0 && result);
 
   return result;
 }
 }  // namespace webrtc