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

webrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.cc

 /*
  *  Copyright (c) 2016 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/neteq/tools/fake_decode_from_file.h"
 
+#include "webrtc/base/checks.h"
+#include "webrtc/base/safe_conversions.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
-#include "webrtc/rtc_base/checks.h"
-#include "webrtc/rtc_base/safe_conversions.h"
 
 namespace webrtc {
 namespace test {
 
 int FakeDecodeFromFile::DecodeInternal(const uint8_t* encoded,
                                        size_t encoded_len,
                                        int sample_rate_hz,
                                        int16_t* decoded,
                                        SpeechType* speech_type) {
   if (encoded_len == 0) {
@@ skipping 68 matching lines @@
   RTC_CHECK_GE(encoded.size(), 12);
   ByteWriter<uint32_t>::WriteLittleEndian(&encoded[0], timestamp);
   ByteWriter<uint32_t>::WriteLittleEndian(&encoded[4],
                                           rtc::checked_cast<uint32_t>(samples));
   ByteWriter<uint32_t>::WriteLittleEndian(
       &encoded[8], rtc::checked_cast<uint32_t>(original_payload_size_bytes));
 }
 
 }  // namespace test
 }  // namespace webrtc