Added histogram for the output level int the audio processing module

webrtc/modules/audio_processing/audio_processing_impl.cc

Index: webrtc/modules/audio_processing/audio_processing_impl.cc
diff --git a/webrtc/modules/audio_processing/audio_processing_impl.cc b/webrtc/modules/audio_processing/audio_processing_impl.cc
index 2379cd1be8ec06fd8a2626a32f14c82d2a46631e..b1e72e61993119183da5c1ed385b72a52a30a172 100644
--- a/webrtc/modules/audio_processing/audio_processing_impl.cc
+++ b/webrtc/modules/audio_processing/audio_processing_impl.cc
@@ -1095,12 +1095,11 @@ int AudioProcessingImpl::ProcessCaptureStreamLocked() {
 
   AudioBuffer* capture_buffer = capture_.capture_audio.get();  // For brevity.
 
-  rms_.Analyze(rtc::ArrayView<const int16_t>(
+  capture_input_rms_.Analyze(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();
+  if (capture_rms_interval_counter_ >= 1000) {
+    RmsLevel::Levels levels = capture_input_rms_.AverageAndPeak();
     RTC_HISTOGRAM_COUNTS("WebRTC.Audio.ApmCaptureInputLevelAverage",
                          levels.average, 1, RmsLevel::kMinLevelDb, 100);
     RTC_HISTOGRAM_COUNTS("WebRTC.Audio.ApmCaptureInputLevelPeak", levels.peak,
@@ -1228,6 +1227,18 @@ int AudioProcessingImpl::ProcessCaptureStreamLocked() {
   // The level estimator operates on the recombined data.
   public_submodules_->level_estimator->ProcessStream(capture_buffer);
 
+  capture_output_rms_.Analyze(rtc::ArrayView<const int16_t>(
+      capture_buffer->channels_const()[0],
+      capture_nonlocked_.capture_processing_format.num_frames()));
+  if (++capture_rms_interval_counter_ >= 1000) {
+    capture_rms_interval_counter_ = 0;
+    RmsLevel::Levels levels = capture_output_rms_.AverageAndPeak();
+    RTC_HISTOGRAM_COUNTS("WebRTC.Audio.ApmCaptureOutputLevelAverage",

[hlundin-webrtc, 2016/12/06 16:07:10]

Did you not want this to be RTC_HISTOGRAM_COUNTS_LINEAR?

[peah-webrtc, 2016/12/20 14:02:39]

Fully true! Thanks! I also changed the ranges to match what is done for the
input.

Done.

+                         levels.average, 1, RmsLevel::kMinLevelDb, 100);
+    RTC_HISTOGRAM_COUNTS("WebRTC.Audio.ApmCaptureOutputLevelPeak", levels.peak,

[hlundin-webrtc, 2016/12/06 16:07:10]

And this?

[peah-webrtc, 2016/12/20 14:02:39]

Done.

+                         1, RmsLevel::kMinLevelDb, 100);
+  }
+
   capture_.was_stream_delay_set = false;
   return kNoError;
 }