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

webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.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/g711/audio_encoder_pcm.h"
 
 #include <algorithm>
 #include <limits>
 
-#include "webrtc/base/checks.h"
-#include "webrtc/base/string_to_number.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/codecs/g711/g711_interface.h"
+#include "webrtc/rtc_base/checks.h"
+#include "webrtc/rtc_base/string_to_number.h"
 
 namespace webrtc {
 
 namespace {
 
 template <typename T>
 typename T::Config CreateConfig(const CodecInst& codec_inst) {
   typename T::Config config;
   config.frame_size_ms = codec_inst.pacsize / 8;
   config.num_channels = codec_inst.channels;
@@ skipping 151 matching lines @@
 
 size_t AudioEncoderPcmU::BytesPerSample() const {
   return 1;
 }
 
 AudioEncoder::CodecType AudioEncoderPcmU::GetCodecType() const {
   return AudioEncoder::CodecType::kPcmU;
 }
 
 }  // namespace webrtc