commit 290d43aa1406790132db794f81103922442c24c4
author henrik.lundin <henrik.lundin@webrtc.org> Tue Nov 29 08:09:09 2016 -0800
committer Commit bot <commit-bot@chromium.org> Tue Nov 29 16:09:17 2016 +0000
tree ad3c2a99f3c5a3003c4dca4dd27a1f6df5b29d00
parent 939e08f5f4b167451253d3acd5663dd9ead25875

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

This CL adds WebRTC.Audio.ApmCaptureInputLevelAverage and
WebRTC.Audio.ApmCaptureInputLevelPeak. The metrics are updated once
every 10 seconds.

BUG=webrtc:6622

Review-Url: https://codereview.webrtc.org/2534473004
Cr-Commit-Position: refs/heads/master@{#15300}

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 061495d..2379cd1 100644
--- a/webrtc/modules/audio_processing/audio_processing_impl.cc
+++ b/webrtc/modules/audio_processing/audio_processing_impl.cc
@@ -1095,6 +1095,18 @@
 
   AudioBuffer* capture_buffer = capture_.capture_audio.get();  // For brevity.
 
+  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();
+    RTC_HISTOGRAM_COUNTS("WebRTC.Audio.ApmCaptureInputLevelAverage",
+                         levels.average, 1, RmsLevel::kMinLevelDb, 100);
+    RTC_HISTOGRAM_COUNTS("WebRTC.Audio.ApmCaptureInputLevelPeak", levels.peak,
+                         1, RmsLevel::kMinLevelDb, 100);
+  }
+
   if (constants_.use_experimental_agc &&
       public_submodules_->gain_control->is_enabled()) {
     private_submodules_->agc_manager->AnalyzePreProcess(