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AGC2 logs

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- Now every 10s
- Also logging estimated speech+noise level
- Also logging via RTC_LOG

Bug: webrtc:7494
Change-Id: Ib60a74d319d29c8f6ae4ea6dae8f2bca687c4c25
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186780
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32316}
This commit is contained in:
Alessio Bazzica 2020-10-05 16:54:41 +02:00 committed by Commit Bot
parent 9f0c89bd56
commit eacbd972ab
1 changed files with 18 additions and 10 deletions
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28
modules/audio_processing/agc2/adaptive_digital_gain_applier.cc
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@ -16,6 +16,7 @@
#include "modules/audio_processing/agc2/agc2_common.h"
#include "modules/audio_processing/logging/apm_data_dumper.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/numerics/safe_minmax.h"
#include "system_wrappers/include/metrics.h"
@ -122,16 +123,6 @@ void AdaptiveDigitalGainApplier::Process(const FrameInfo& info,
<< "`frame` does not look like a 10 ms frame for an APM supported sample "
"rate";
// Log every second.
calls_since_last_gain_log_++;
if (calls_since_last_gain_log_ == 100) {
calls_since_last_gain_log_ = 0;
RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.Agc2.DigitalGainApplied",
last_gain_db_, 0, kMaxGainDb, kMaxGainDb + 1);
RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.Agc2.EstimatedNoiseLevel",
-info.input_noise_level_dbfs, 0, 100, 101);
}
const float target_gain_db = LimitGainByLowConfidence(
LimitGainByNoise(ComputeGainDb(std::min(info.input_level_dbfs, 0.f)),
info.input_noise_level_dbfs,
@ -167,5 +158,22 @@ void AdaptiveDigitalGainApplier::Process(const FrameInfo& info,
// Remember that the gain has changed for the next iteration.
last_gain_db_ = last_gain_db_ + gain_change_this_frame_db;
apm_data_dumper_->DumpRaw("agc2_applied_gain_db", last_gain_db_);
// Log every 10 seconds.
calls_since_last_gain_log_++;
if (calls_since_last_gain_log_ == 1000) {
calls_since_last_gain_log_ = 0;
RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.Agc2.DigitalGainApplied",
last_gain_db_, 0, kMaxGainDb, kMaxGainDb + 1);
RTC_HISTOGRAM_COUNTS_LINEAR(
"WebRTC.Audio.Agc2.EstimatedSpeechPlusNoiseLevel",
-info.input_level_dbfs, 0, 100, 101);
RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.Agc2.EstimatedNoiseLevel",
-info.input_noise_level_dbfs, 0, 100, 101);
RTC_LOG(LS_INFO) << "AGC2 adaptive digital"
<< " | speech_plus_noise_dbfs: " << info.input_level_dbfs
<< " | noise_dbfs: " << info.input_noise_level_dbfs
<< " | gain_db: " << last_gain_db_;
}
}
} // namespace webrtc