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Gain metrics for digital adaptive AGC.

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We add 2 metrics for measuring applied digital gain to
AgcManagerDirect. We also add an applied gain and an estimated noise
metric to Agc2.

Chromium histogram CL is
https://chromium-review.googlesource.com/c/chromium/src/+/1170833

Bug: webrtc:7494
Change-Id: Ie40873f9e43bc7d34d8f5473cd73bd47eb84e855
Reviewed-on: https://webrtc-review.googlesource.com/93468
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24290}
This commit is contained in:
Alex Loiko
2018-08-10 14:43:51 +02:00
committed by Commit Bot
parent 847a9c70c2
commit f3122e0efe
5 changed files with 23 additions and 0 deletions
Download Patch File Download Diff File Expand all files Collapse all files

10
modules/audio_processing/agc/agc_manager_direct.cc
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@ -454,6 +454,13 @@ void AgcManagerDirect::UpdateGain() {
}
void AgcManagerDirect::UpdateCompressor() {
calls_since_last_gain_log_++;
if (calls_since_last_gain_log_ == 100) {
calls_since_last_gain_log_ = 0;
RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.Agc.DigitalGainApplied",
compression_, 0, kMaxCompressionGain,
kMaxCompressionGain + 1);
}
if (compression_ == target_compression_) {
return;
}
@ -478,6 +485,9 @@ void AgcManagerDirect::UpdateCompressor() {
// Set the new compression gain.
if (new_compression != compression_) {
RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.Agc.DigitalGainUpdated",
new_compression, 0, kMaxCompressionGain,
kMaxCompressionGain + 1);
compression_ = new_compression;
compression_accumulator_ = new_compression;
if (gctrl_->set_compression_gain_db(compression_) != 0) {

1
modules/audio_processing/agc/agc_manager_direct.h
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@ -126,6 +126,7 @@ class AgcManagerDirect final {
const bool disable_digital_adaptive_;
int startup_min_level_;
const int clipped_level_min_;
int calls_since_last_gain_log_ = 0;
std::unique_ptr<DebugFile> file_preproc_;
std::unique_ptr<DebugFile> file_postproc_;

1
modules/audio_processing/agc2/BUILD.gn
View File

@ -65,6 +65,7 @@ rtc_source_set("adaptive_digital") {
"../../../rtc_base:checks",
"../../../rtc_base:rtc_base_approved",
"../../../rtc_base:safe_minmax",
"../../../system_wrappers:metrics_api",
]
}

10
modules/audio_processing/agc2/adaptive_digital_gain_applier.cc
View File

@ -16,6 +16,7 @@
#include "modules/audio_processing/agc2/agc2_common.h"
#include "modules/audio_processing/logging/apm_data_dumper.h"
#include "rtc_base/numerics/safe_minmax.h"
#include "system_wrappers/include/metrics.h"
namespace webrtc {
namespace {
@ -76,6 +77,15 @@ void AdaptiveDigitalGainApplier::Process(
float input_noise_level_dbfs,
const VadWithLevel::LevelAndProbability vad_result,
AudioFrameView<float> float_frame) {
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",
input_noise_level_dbfs, 0, 100, 101);
}
input_level_dbfs = std::min(input_level_dbfs, 0.f);
RTC_DCHECK_GE(input_level_dbfs, -150.f);

1
modules/audio_processing/agc2/adaptive_digital_gain_applier.h
View File

@ -32,6 +32,7 @@ class AdaptiveDigitalGainApplier {
private:
float last_gain_db_ = kInitialAdaptiveDigitalGainDb;
GainApplier gain_applier_;
int calls_since_last_gain_log_ = 0;
// For some combinations of noise and speech probability, increasing
// the level is not allowed. Since we may get VAD results in bursts,