Add a new UMA metric in APM to track incoming capture-side audio level

webrtc/modules/audio_processing/rms_level.h

 /*
  *  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.
  */
 
@@ skipping 14 matching lines @@
 // The expected approach is to provide constant-sized chunks of audio to
 // Analyze(). When enough chunks have been accumulated to form a packet, call
 // Average() to get the audio level indicator for the RTP header.
 class RmsLevel {
  public:
   struct Levels {
     int average;
     int peak;
   };
 
+  static constexpr int kMinLevelDb = 127;

[hlundin-webrtc, 2016/11/29 11:03:55]

Had to move this back out to the public part of the class.

+
   RmsLevel();
   ~RmsLevel();
 
   // Can be called to reset internal states, but is not required during normal
   // operation.
   void Reset();
 
   // Pass each chunk of audio to Analyze() to accumulate the level.
   void Analyze(rtc::ArrayView<const int16_t> data);
 
@@ skipping 19 matching lines @@
   float sum_square_;
   size_t sample_count_;
   float max_sum_square_;
   rtc::Optional<size_t> block_size_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_RMS_LEVEL_H_