bcec8b0b21
This is from the upstream library commit id 3326535126e435f1ba647885ce43a8f0f3d317eb, corresponding to Chromium 88.0.4290.1.
91 lines
3.5 KiB
C++
91 lines
3.5 KiB
C++
/*
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* Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "modules/audio_processing/agc2/adaptive_agc.h"
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#include "common_audio/include/audio_util.h"
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#include "modules/audio_processing/agc2/vad_with_level.h"
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#include "modules/audio_processing/logging/apm_data_dumper.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/logging.h"
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namespace webrtc {
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namespace {
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void DumpDebugData(const AdaptiveDigitalGainApplier::FrameInfo& info,
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ApmDataDumper& dumper) {
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dumper.DumpRaw("agc2_vad_probability", info.vad_result.speech_probability);
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dumper.DumpRaw("agc2_vad_rms_dbfs", info.vad_result.rms_dbfs);
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dumper.DumpRaw("agc2_vad_peak_dbfs", info.vad_result.peak_dbfs);
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dumper.DumpRaw("agc2_noise_estimate_dbfs", info.input_noise_level_dbfs);
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dumper.DumpRaw("agc2_last_limiter_audio_level", info.limiter_envelope_dbfs);
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}
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constexpr int kGainApplierAdjacentSpeechFramesThreshold = 1;
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constexpr float kMaxGainChangePerSecondDb = 3.f;
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constexpr float kMaxOutputNoiseLevelDbfs = -50.f;
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} // namespace
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AdaptiveAgc::AdaptiveAgc(ApmDataDumper* apm_data_dumper)
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: speech_level_estimator_(apm_data_dumper),
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gain_applier_(apm_data_dumper,
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kGainApplierAdjacentSpeechFramesThreshold,
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kMaxGainChangePerSecondDb,
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kMaxOutputNoiseLevelDbfs),
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apm_data_dumper_(apm_data_dumper),
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noise_level_estimator_(apm_data_dumper) {
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RTC_DCHECK(apm_data_dumper);
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}
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AdaptiveAgc::AdaptiveAgc(ApmDataDumper* apm_data_dumper,
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const AudioProcessing::Config::GainController2& config)
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: speech_level_estimator_(
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apm_data_dumper,
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config.adaptive_digital.level_estimator,
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config.adaptive_digital
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.level_estimator_adjacent_speech_frames_threshold,
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config.adaptive_digital.initial_saturation_margin_db,
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config.adaptive_digital.extra_saturation_margin_db),
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vad_(config.adaptive_digital.vad_probability_attack),
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gain_applier_(
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apm_data_dumper,
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config.adaptive_digital.gain_applier_adjacent_speech_frames_threshold,
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config.adaptive_digital.max_gain_change_db_per_second,
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config.adaptive_digital.max_output_noise_level_dbfs),
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apm_data_dumper_(apm_data_dumper),
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noise_level_estimator_(apm_data_dumper) {
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RTC_DCHECK(apm_data_dumper);
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if (!config.adaptive_digital.use_saturation_protector) {
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RTC_LOG(LS_WARNING) << "The saturation protector cannot be disabled.";
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}
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}
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AdaptiveAgc::~AdaptiveAgc() = default;
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void AdaptiveAgc::Process(AudioFrameView<float> frame, float limiter_envelope) {
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AdaptiveDigitalGainApplier::FrameInfo info;
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info.vad_result = vad_.AnalyzeFrame(frame);
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speech_level_estimator_.Update(info.vad_result);
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info.input_level_dbfs = speech_level_estimator_.level_dbfs();
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info.input_noise_level_dbfs = noise_level_estimator_.Analyze(frame);
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info.limiter_envelope_dbfs =
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limiter_envelope > 0 ? FloatS16ToDbfs(limiter_envelope) : -90.f;
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info.estimate_is_confident = speech_level_estimator_.IsConfident();
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DumpDebugData(info, *apm_data_dumper_);
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gain_applier_.Process(info, frame);
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}
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void AdaptiveAgc::Reset() {
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speech_level_estimator_.Reset();
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}
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} // namespace webrtc
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