Miscellaneous changes split from https://codereview.webrtc.org/1230503003 .

webrtc/modules/audio_device/android/audio_device_unittest.cc

 /*
  *  Copyright (c) 2015 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 332 matching lines @@
     PRINT("] ");
     for (auto it = latencies_.begin(); it != latencies_.end(); ++it) {
       PRINT("%d ", *it);
     }
     PRINT("\n");
     PRINT("%s[min, max, avg]=[%d, %d, %d] ms\n", kTag,
         min_latency(), max_latency(), average_latency());
   }
 
   int IndexToMilliseconds(double index) const {
-    return 10.0 * (index / frames_per_buffer_) + 0.5;
+    return static_cast<int>(10.0 * (index / frames_per_buffer_) + 0.5);
   }
 
  private:
   Clock* clock_;
   const int frames_per_buffer_;
   const int bytes_per_buffer_;
   int play_count_;
   int rec_count_;
   int64_t pulse_time_;
   std::vector<int> latencies_;
@@ skipping 628 matching lines @@
                                1000 * kMeasureLatencyTimeInSec));
   StopPlayout();
   StopRecording();
   // Verify that the correct number of transmitted impulses are detected.
   EXPECT_EQ(latency_audio_stream->num_latency_values(),
             kImpulseFrequencyInHz * kMeasureLatencyTimeInSec - 1);
   latency_audio_stream->PrintResults();
 }
 
 }  // namespace webrtc