blob: a9f9622d00c236bdf8daaf96f5c22911e94a3d4a [file] [log] [blame]
/*
* Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "modules/audio_processing/agc2/adaptive_agc.h"
#include "common_audio/include/audio_util.h"
#include "modules/audio_processing/agc2/cpu_features.h"
#include "modules/audio_processing/agc2/vad_with_level.h"
#include "modules/audio_processing/logging/apm_data_dumper.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
namespace webrtc {
namespace {
using AdaptiveDigitalConfig =
AudioProcessing::Config::GainController2::AdaptiveDigital;
using NoiseEstimatorType =
AudioProcessing::Config::GainController2::NoiseEstimator;
// Detects the available CPU features and applies any kill-switches.
AvailableCpuFeatures GetAllowedCpuFeatures(
const AdaptiveDigitalConfig& config) {
AvailableCpuFeatures features = GetAvailableCpuFeatures();
if (!config.sse2_allowed) {
features.sse2 = false;
}
if (!config.avx2_allowed) {
features.avx2 = false;
}
if (!config.neon_allowed) {
features.neon = false;
}
return features;
}
} // namespace
AdaptiveAgc::AdaptiveAgc(ApmDataDumper* apm_data_dumper,
const AdaptiveDigitalConfig& config)
: speech_level_estimator_(apm_data_dumper,
config.adjacent_speech_frames_threshold),
vad_(config.vad_reset_period_ms, GetAllowedCpuFeatures(config)),
gain_controller_(apm_data_dumper,
config.adjacent_speech_frames_threshold,
config.max_gain_change_db_per_second,
config.max_output_noise_level_dbfs,
config.dry_run),
apm_data_dumper_(apm_data_dumper),
noise_level_estimator_(CreateNoiseFloorEstimator(apm_data_dumper)),
saturation_protector_(
CreateSaturationProtector(kSaturationProtectorInitialHeadroomDb,
kSaturationProtectorExtraHeadroomDb,
config.adjacent_speech_frames_threshold,
apm_data_dumper)) {
RTC_DCHECK(apm_data_dumper);
RTC_DCHECK(noise_level_estimator_);
RTC_DCHECK(saturation_protector_);
if (!config.use_saturation_protector) {
RTC_LOG(LS_WARNING) << "The saturation protector cannot be disabled.";
}
}
AdaptiveAgc::~AdaptiveAgc() = default;
void AdaptiveAgc::Initialize(int sample_rate_hz, int num_channels) {
gain_controller_.Initialize(sample_rate_hz, num_channels);
}
void AdaptiveAgc::Process(AudioFrameView<float> frame, float limiter_envelope) {
AdaptiveDigitalGainApplier::FrameInfo info;
VadLevelAnalyzer::Result vad_result = vad_.AnalyzeFrame(frame);
info.speech_probability = vad_result.speech_probability;
apm_data_dumper_->DumpRaw("agc2_speech_probability",
vad_result.speech_probability);
apm_data_dumper_->DumpRaw("agc2_input_rms_dbfs", vad_result.rms_dbfs);
apm_data_dumper_->DumpRaw("agc2_input_peak_dbfs", vad_result.peak_dbfs);
speech_level_estimator_.Update(vad_result);
info.speech_level_dbfs = speech_level_estimator_.level_dbfs();
info.speech_level_reliable = speech_level_estimator_.IsConfident();
apm_data_dumper_->DumpRaw("agc2_speech_level_dbfs", info.speech_level_dbfs);
apm_data_dumper_->DumpRaw("agc2_speech_level_reliable",
info.speech_level_reliable);
info.noise_rms_dbfs = noise_level_estimator_->Analyze(frame);
apm_data_dumper_->DumpRaw("agc2_noise_rms_dbfs", info.noise_rms_dbfs);
saturation_protector_->Analyze(info.speech_probability, vad_result.peak_dbfs,
info.speech_level_dbfs);
info.headroom_db = saturation_protector_->HeadroomDb();
apm_data_dumper_->DumpRaw("agc2_headroom_db", info.headroom_db);
info.limiter_envelope_dbfs = FloatS16ToDbfs(limiter_envelope);
apm_data_dumper_->DumpRaw("agc2_limiter_envelope_dbfs",
info.limiter_envelope_dbfs);
gain_controller_.Process(info, frame);
}
void AdaptiveAgc::HandleInputGainChange() {
speech_level_estimator_.Reset();
saturation_protector_->Reset();
}
} // namespace webrtc