commit 1b08dc33ebcdb48f951258cf75d97dff0e60b4f1
author peah <peah@webrtc.org> Tue Dec 20 13:45:58 2016 -0800
committer Commit bot <commit-bot@chromium.org> Tue Dec 20 21:45:58 2016 +0000
tree 555de1f34c9dca9b60da6b6d733ab226a0a7c459
parent 0838327ec96993a7ff3459bb3588203b6ccfc260

To verify the upcoming code changes it is required
that the level of the output in the audio processing
module is monitored. This CL adds that.

BUG=webrtc:6181, webrtc:6183, webrtc:6220

Review-Url: https://codereview.webrtc.org/2549143004
Cr-Commit-Position: refs/heads/master@{#15718}

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 ce71461..3a8f7f5 100644
--- a/webrtc/modules/audio_processing/audio_processing_impl.cc
+++ b/webrtc/modules/audio_processing/audio_processing_impl.cc
@@ -1132,12 +1132,13 @@
 
   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();
+  const bool log_rms = ++capture_rms_interval_counter_ >= 1000;
+  if (log_rms) {
+    capture_rms_interval_counter_ = 0;
+    RmsLevel::Levels levels = capture_input_rms_.AverageAndPeak();
     RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.ApmCaptureInputLevelAverageRms",
                                 levels.average, 1, RmsLevel::kMinLevelDb, 64);
     RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.ApmCaptureInputLevelPeakRms",
@@ -1275,6 +1276,17 @@
   // 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 (log_rms) {
+    RmsLevel::Levels levels = capture_output_rms_.AverageAndPeak();
+    RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.ApmCaptureOutputLevelAverageRms",
+                                levels.average, 1, RmsLevel::kMinLevelDb, 64);
+    RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.ApmCaptureOutputLevelPeakRms",
+                                levels.peak, 1, RmsLevel::kMinLevelDb, 64);
+  }
+
   capture_.was_stream_delay_set = false;
   return kNoError;
 }