Index: webrtc/modules/audio_processing/agc/agc_manager_direct.cc |
diff --git a/webrtc/modules/audio_processing/agc/agc_manager_direct.cc b/webrtc/modules/audio_processing/agc/agc_manager_direct.cc |
index 92715dc61e1693fa8a4422753824ea0c2bc688a7..dc9ba42d5123eac6f4f1c9276965bb37d09fdf9b 100644 |
--- a/webrtc/modules/audio_processing/agc/agc_manager_direct.cc |
+++ b/webrtc/modules/audio_processing/agc/agc_manager_direct.cc |
@@ -21,6 +21,7 @@ |
#include "webrtc/modules/audio_processing/gain_control_impl.h" |
#include "webrtc/modules/include/module_common_types.h" |
#include "webrtc/system_wrappers/include/logging.h" |
+#include "webrtc/system_wrappers/include/metrics.h" |
namespace webrtc { |
@@ -218,6 +219,8 @@ void AgcManagerDirect::AnalyzePreProcess(int16_t* audio, |
// Always decrease the maximum level, even if the current level is below |
// threshold. |
SetMaxLevel(std::max(kClippedLevelMin, max_level_ - kClippedLevelStep)); |
+ RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.AgcClippingAdjustmentAllowed", |
+ level_ - kClippedLevelStep >= kClippedLevelMin); |
if (level_ > kClippedLevelMin) { |
// Don't try to adjust the level if we're already below the limit. As |
// a consequence, if the user has brought the level above the limit, we |