| 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
|
|
|