Update includes for webrtc/{base => rtc_base} rename (2/3)

webrtc/common_audio/audio_converter.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/common_audio/audio_converter.h"
 
 #include <cstring>
 #include <memory>
 #include <utility>
 #include <vector>
 
-#include "webrtc/base/checks.h"
-#include "webrtc/base/safe_conversions.h"
 #include "webrtc/common_audio/channel_buffer.h"
 #include "webrtc/common_audio/resampler/push_sinc_resampler.h"
+#include "webrtc/rtc_base/checks.h"
+#include "webrtc/rtc_base/safe_conversions.h"
 
 using rtc::checked_cast;
 
 namespace webrtc {
 
 class CopyConverter : public AudioConverter {
  public:
   CopyConverter(size_t src_channels, size_t src_frames, size_t dst_channels,
                 size_t dst_frames)
       : AudioConverter(src_channels, src_frames, dst_channels, dst_frames) {}
@@ skipping 166 matching lines @@
   RTC_CHECK(dst_channels == src_channels || dst_channels == 1 ||
             src_channels == 1);
 }
 
 void AudioConverter::CheckSizes(size_t src_size, size_t dst_capacity) const {
   RTC_CHECK_EQ(src_size, src_channels() * src_frames());
   RTC_CHECK_GE(dst_capacity, dst_channels() * dst_frames());
 }
 
 }  // namespace webrtc