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

webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc

 /*
  *  Copyright (c) 2014 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/codecs/g722/audio_encoder_g722.h"
 
 #include <algorithm>
 
 #include <limits>
+#include "webrtc/base/checks.h"
+#include "webrtc/base/safe_conversions.h"
+#include "webrtc/base/string_to_number.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/codecs/g722/g722_interface.h"
-#include "webrtc/rtc_base/checks.h"
-#include "webrtc/rtc_base/safe_conversions.h"
-#include "webrtc/rtc_base/string_to_number.h"
 
 namespace webrtc {
 
 namespace {
 
 const size_t kSampleRateHz = 16000;
 
 AudioEncoderG722Config CreateConfig(const CodecInst& codec_inst) {
   AudioEncoderG722Config config;
   config.num_channels = rtc::dchecked_cast<int>(codec_inst.channels);
@@ skipping 162 matching lines @@
 
 AudioEncoderG722Impl::EncoderState::~EncoderState() {
   RTC_CHECK_EQ(0, WebRtcG722_FreeEncoder(encoder));
 }
 
 size_t AudioEncoderG722Impl::SamplesPerChannel() const {
   return kSampleRateHz / 100 * num_10ms_frames_per_packet_;
 }
 
 }  // namespace webrtc