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

webrtc/modules/audio_processing/audio_processing_impl.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.
  */
 
@@ skipping 1077 matching lines @@
         public_submodules_->echo_cancellation->stream_drift_samples());
     msg->set_level(gain_control()->stream_analog_level());
     msg->set_keypress(capture_.key_pressed);
   }
 #endif
 
   MaybeUpdateHistograms();
 
   AudioBuffer* capture_buffer = capture_.capture_audio.get();  // For brevity.
 
+  rms_.Process(rtc::ArrayView<const int16_t>(
+      capture_buffer->channels_const()[0],
+      capture_nonlocked_.capture_processing_format.num_frames()));
+  if (++rms_interval_counter_ >= 1000) {
+    rms_interval_counter_ = 0;
+    RMSLevel::Levels levels = rms_.AverageAndPeak();
+    RTC_HISTOGRAM_COUNTS("WebRTC.Audio.ApmCaptureInputLevelAverage",
+                         levels.average, 1, RMSLevel::kMinLevelDb, 100);

[peah-webrtc, 2016/11/29 09:02:27]

Why having a minimum value of 0? For a signal of maximum level the maximum value
is 0 dBFS.

[hlundin-webrtc, 2016/11/29 11:02:39]

The value we get is between 0 and 127, which corresponds to 0 to -127 dBFS. And
since UMA histograms always create a zero-bucket anyway, the range [1, 127]
makes sense imo.

+    RTC_HISTOGRAM_COUNTS("WebRTC.Audio.ApmCaptureInputLevelPeak", levels.peak,
+                         1, RMSLevel::kMinLevelDb, 100);

[peah-webrtc, 2016/11/29 09:02:27]

Why having a minimum value of 0? For a signal of maximum level the maximum value
is 0 dBFS.

[hlundin-webrtc, 2016/11/29 11:02:39]

See above.

+  }
+
   if (constants_.use_experimental_agc &&
       public_submodules_->gain_control->is_enabled()) {
     private_submodules_->agc_manager->AnalyzePreProcess(
         capture_buffer->channels()[0], capture_buffer->num_channels(),
         capture_nonlocked_.capture_processing_format.num_frames());
   }
 
   if (submodule_states_.CaptureMultiBandSubModulesActive() &&
       SampleRateSupportsMultiBand(
           capture_nonlocked_.capture_processing_format.sample_rate_hz())) {
@@ skipping 771 matching lines @@
       capture_processing_format(kSampleRate16kHz),
       split_rate(kSampleRate16kHz) {}
 
 AudioProcessingImpl::ApmCaptureState::~ApmCaptureState() = default;
 
 AudioProcessingImpl::ApmRenderState::ApmRenderState() = default;
 
 AudioProcessingImpl::ApmRenderState::~ApmRenderState() = default;
 
 }  // namespace webrtc