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

webrtc/modules/audio_coding/acm2/acm_receiver.cc

 /*
  *  Copyright (c) 2013 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/audio_coding/acm2/acm_receiver.h"
 
 #include <stdlib.h>  // malloc
 
 #include <algorithm>  // sort
 #include <vector>
 
 #include "webrtc/api/audio_codecs/audio_decoder.h"
+#include "webrtc/base/checks.h"
+#include "webrtc/base/format_macros.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/base/safe_conversions.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/acm2/acm_resampler.h"
 #include "webrtc/modules/audio_coding/acm2/call_statistics.h"
+#include "webrtc/modules/audio_coding/neteq/include/neteq.h"
+#include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
-#include "webrtc/modules/audio_coding/neteq/include/neteq.h"
-#include "webrtc/rtc_base/checks.h"
-#include "webrtc/rtc_base/format_macros.h"
-#include "webrtc/rtc_base/logging.h"
-#include "webrtc/rtc_base/safe_conversions.h"
-#include "webrtc/system_wrappers/include/clock.h"
 
 namespace webrtc {
 
 namespace acm2 {
 
 AcmReceiver::AcmReceiver(const AudioCodingModule::Config& config)
     : last_audio_buffer_(new int16_t[AudioFrame::kMaxDataSizeSamples]),
       neteq_(NetEq::Create(config.neteq_config, config.decoder_factory)),
       clock_(config.clock),
       resampled_last_output_frame_(true) {
@@ skipping 353 matching lines @@
 
 void AcmReceiver::GetDecodingCallStatistics(
     AudioDecodingCallStats* stats) const {
   rtc::CritScope lock(&crit_sect_);
   *stats = call_stats_.GetDecodingStatistics();
 }
 
 }  // namespace acm2
 
 }  // namespace webrtc