Roll chromium_revision 33a7a547b9..0e44c5e141 (452838:453130)

webrtc/modules/audio_mixer/default_output_rate_calculator.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.
  */
 
@@ skipping 13 matching lines @@
   using NativeRate = AudioProcessing::NativeRate;
   const int maximal_frequency = *std::max_element(
       preferred_sample_rates.begin(), preferred_sample_rates.end());
 
   RTC_DCHECK_LE(NativeRate::kSampleRate8kHz, maximal_frequency);
   RTC_DCHECK_GE(NativeRate::kSampleRate48kHz, maximal_frequency);
 
   static constexpr NativeRate native_rates[] = {
       NativeRate::kSampleRate8kHz, NativeRate::kSampleRate16kHz,
       NativeRate::kSampleRate32kHz, NativeRate::kSampleRate48kHz};
-  const auto rounded_up_index = std::lower_bound(
+  const auto* rounded_up_index = std::lower_bound(
       std::begin(native_rates), std::end(native_rates), maximal_frequency);
   RTC_DCHECK(rounded_up_index != std::end(native_rates));
   return *rounded_up_index;
 }
 }  // namespace webrtc