Safe numeric library: base/numerics (copied from Chromium)

webrtc/modules/audio_coding/neteq/expand_unittest.cc

 /*
  *  Copyright (c) 2012 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.
  */
 
 // Unit tests for Expand class.
 
 #include "webrtc/modules/audio_coding/neteq/expand.h"
 
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/base/safe_conversions.h"
+#include "webrtc/base/numerics/safe_conversions.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 #include "webrtc/modules/audio_coding/neteq/background_noise.h"
 #include "webrtc/modules/audio_coding/neteq/random_vector.h"
 #include "webrtc/modules/audio_coding/neteq/statistics_calculator.h"
 #include "webrtc/modules/audio_coding/neteq/sync_buffer.h"
 #include "webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
 namespace webrtc {
 
@@ skipping 138 matching lines @@
   expand_.SetParametersForNormalAfterExpand();
   // Convert |sum_output_len_samples| to milliseconds.
   EXPECT_EQ(rtc::checked_cast<int>(sum_output_len_samples /
                                    (test_sample_rate_hz_ / 1000)),
             statistics_.last_outage_duration_ms());
 }
 
 // TODO(hlundin): Write more tests.
 
 }  // namespace webrtc