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AGC2 new data dumps

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Bug: webrtc:7494
Change-Id: Id288dd426e1c2754805bc548fbffe0eaeaacf3da
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/213420
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33605}
This commit is contained in:
Alessio Bazzica 2021-03-31 15:16:05 +02:00 committed by Commit Bot
parent 841d74ea80
commit 8aaa604375
6 changed files with 24 additions and 15 deletions
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2
modules/audio_processing/agc2/adaptive_digital_gain_applier.cc
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@ -136,6 +136,8 @@ void AdaptiveDigitalGainApplier::Process(const FrameInfo& info,
} else if (frames_to_gain_increase_allowed_ > 0) {
frames_to_gain_increase_allowed_--;
}
apm_data_dumper_->DumpRaw("agc2_frames_to_gain_increase_allowed",
frames_to_gain_increase_allowed_);
const float gain_change_this_frame_db = ComputeGainChangeThisFrameDb(
target_gain_db, last_gain_db_,

6
modules/audio_processing/agc2/adaptive_mode_level_estimator.cc
View File

@ -186,7 +186,7 @@ void AdaptiveModeLevelEstimator::ResetLevelEstimatorState(
void AdaptiveModeLevelEstimator::DumpDebugData() const {
apm_data_dumper_->DumpRaw("agc2_adaptive_level_estimate_dbfs", level_dbfs_);
apm_data_dumper_->DumpRaw("agc2_adaptive_num_adjacent_speech_frames_",
apm_data_dumper_->DumpRaw("agc2_adaptive_num_adjacent_speech_frames",
num_adjacent_speech_frames_);
apm_data_dumper_->DumpRaw("agc2_adaptive_preliminary_level_estimate_num",
preliminary_state_.level_dbfs.numerator);
@ -194,6 +194,10 @@ void AdaptiveModeLevelEstimator::DumpDebugData() const {
preliminary_state_.level_dbfs.denominator);
apm_data_dumper_->DumpRaw("agc2_adaptive_preliminary_saturation_margin_db",
preliminary_state_.saturation_protector.margin_db);
apm_data_dumper_->DumpRaw("agc2_adaptive_preliminary_time_to_full_buffer_ms",
preliminary_state_.time_to_full_buffer_ms);
apm_data_dumper_->DumpRaw("agc2_adaptive_reliable_time_to_full_buffer_ms",
reliable_state_.time_to_full_buffer_ms);
}
} // namespace webrtc

1
modules/audio_processing/agc2/agc2_common.h
View File

@ -45,6 +45,7 @@ constexpr float kInitialSpeechLevelEstimateDbfs = -30.0f;
// Robust VAD probability and speech decisions.
constexpr int kDefaultVadRnnResetPeriodMs = 1500;
static_assert(kDefaultVadRnnResetPeriodMs % kFrameDurationMs == 0, "");
constexpr float kDefaultSmoothedVadProbabilityAttack = 1.0f;
constexpr int kDefaultLevelEstimatorAdjacentSpeechFramesThreshold = 1;

8
modules/audio_processing/agc2/limiter.cc
View File

@ -125,9 +125,11 @@ void Limiter::Process(AudioFrameView<float> signal) {
last_scaling_factor_ = scaling_factors_.back();
// Dump data for debug.
apm_data_dumper_->DumpRaw("agc2_gain_curve_applier_scaling_factors",
samples_per_channel,
per_sample_scaling_factors_.data());
apm_data_dumper_->DumpRaw("agc2_limiter_last_scaling_factor",
last_scaling_factor_);
apm_data_dumper_->DumpRaw(
"agc2_limiter_region",
static_cast<int>(interp_gain_curve_.get_stats().region));
}
InterpolatedGainCurve::Stats Limiter::GetGainCurveStats() const {

14
modules/audio_processing/gain_controller2.cc
View File

@ -24,14 +24,13 @@ namespace webrtc {
int GainController2::instance_count_ = 0;
GainController2::GainController2()
: data_dumper_(
new ApmDataDumper(rtc::AtomicOps::Increment(&instance_count_))),
: data_dumper_(rtc::AtomicOps::Increment(&instance_count_)),
gain_applier_(/*hard_clip_samples=*/false,
/*initial_gain_factor=*/0.f),
limiter_(static_cast<size_t>(48000), data_dumper_.get(), "Agc2"),
limiter_(static_cast<size_t>(48000), &data_dumper_, "Agc2"),
calls_since_last_limiter_log_(0) {
if (config_.adaptive_digital.enabled) {
adaptive_agc_.reset(new AdaptiveAgc(data_dumper_.get()));
adaptive_agc_ = std::make_unique<AdaptiveAgc>(&data_dumper_);
}
}
@ -43,12 +42,13 @@ void GainController2::Initialize(int sample_rate_hz) {
sample_rate_hz == AudioProcessing::kSampleRate32kHz ||
sample_rate_hz == AudioProcessing::kSampleRate48kHz);
limiter_.SetSampleRate(sample_rate_hz);
data_dumper_->InitiateNewSetOfRecordings();
data_dumper_->DumpRaw("sample_rate_hz", sample_rate_hz);
data_dumper_.InitiateNewSetOfRecordings();
data_dumper_.DumpRaw("sample_rate_hz", sample_rate_hz);
calls_since_last_limiter_log_ = 0;
}
void GainController2::Process(AudioBuffer* audio) {
data_dumper_.DumpRaw("agc2_notified_analog_level", analog_level_);
AudioFrameView<float> float_frame(audio->channels(), audio->num_channels(),
audio->num_frames());
// Apply fixed gain first, then the adaptive one.
@ -90,7 +90,7 @@ void GainController2::ApplyConfig(
}
gain_applier_.SetGainFactor(DbToRatio(config_.fixed_digital.gain_db));
if (config_.adaptive_digital.enabled) {
adaptive_agc_.reset(new AdaptiveAgc(data_dumper_.get(), config_));
adaptive_agc_ = std::make_unique<AdaptiveAgc>(&data_dumper_, config_);
} else {
adaptive_agc_.reset();
}

8
modules/audio_processing/gain_controller2.h
View File

@ -18,11 +18,11 @@
#include "modules/audio_processing/agc2/gain_applier.h"
#include "modules/audio_processing/agc2/limiter.h"
#include "modules/audio_processing/include/audio_processing.h"
#include "modules/audio_processing/logging/apm_data_dumper.h"
#include "rtc_base/constructor_magic.h"
namespace webrtc {
class ApmDataDumper;
class AudioBuffer;
// Gain Controller 2 aims to automatically adjust levels by acting on the
@ -30,6 +30,8 @@ class AudioBuffer;
class GainController2 {
public:
GainController2();
GainController2(const GainController2&) = delete;
GainController2& operator=(const GainController2&) = delete;
~GainController2();
void Initialize(int sample_rate_hz);
@ -41,15 +43,13 @@ class GainController2 {
private:
static int instance_count_;
std::unique_ptr<ApmDataDumper> data_dumper_;
ApmDataDumper data_dumper_;
AudioProcessing::Config::GainController2 config_;
GainApplier gain_applier_;
std::unique_ptr<AdaptiveAgc> adaptive_agc_;
Limiter limiter_;
int calls_since_last_limiter_log_;
int analog_level_ = -1;
RTC_DISALLOW_COPY_AND_ASSIGN(GainController2);
};
} // namespace webrtc