Fix the standard deviation calculation in the level controller perf tests.

webrtc/modules/audio_processing/level_controller/level_controller_complexity_unittest.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 117 matching lines @@
     RTC_DCHECK(!timestamps_us_.empty());
     return static_cast<double>(std::accumulate(timestamps_us_.begin(),
                                                timestamps_us_.end(), 0)) /
            timestamps_us_.size();
   }
 
   double GetDurationStandardDeviation() const {
     RTC_DCHECK(!timestamps_us_.empty());
     double average_duration = GetDurationAverage();
 
-    int64_t variance =
-        std::accumulate(timestamps_us_.begin(), timestamps_us_.end(), 0,
-                        [average_duration](const int64_t& a, const int64_t& b) {
-                          return a + (b - average_duration);
-                        });
+    double variance = std::accumulate(
+        timestamps_us_.begin(), timestamps_us_.end(), 0.0,
+        [average_duration](const double& a, const int64_t& b) {
+          return a + (b - average_duration) * (b - average_duration);
+        });
 
     return sqrt(variance / timestamps_us_.size());
   }
 
  private:
   webrtc::Clock* clock_;
   rtc::Optional<int64_t> start_timestamp_us_;
   std::vector<int64_t> timestamps_us_;
 };
 
@@ skipping 193 matching lines @@
         RunTogetherWithApm("LevelControlAndDefaultMobileApm", 48000, 48000,
                            capture_input_sample_rate_hz,
                            capture_output_sample_rate_hz, num_channels, true,
                            true);
       }
     }
   }
 }
 
 }  // namespace webrtc