AnotherFoxGuy/webrtc-audio-processing
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Update to current webrtc library

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This is from the upstream library commit id
3326535126e435f1ba647885ce43a8f0f3d317eb, corresponding to Chromium
88.0.4290.1.
This commit is contained in:
Arun Raghavan
2020-10-12 18:08:02 -04:00
parent b1b02581d3
commit bcec8b0b21
859 changed files with 76187 additions and 49580 deletions
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90
webrtc/modules/audio_processing/agc2/adaptive_agc.cc Normal file
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/*
* 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/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 {
void DumpDebugData(const AdaptiveDigitalGainApplier::FrameInfo& info,
ApmDataDumper& dumper) {
dumper.DumpRaw("agc2_vad_probability", info.vad_result.speech_probability);
dumper.DumpRaw("agc2_vad_rms_dbfs", info.vad_result.rms_dbfs);
dumper.DumpRaw("agc2_vad_peak_dbfs", info.vad_result.peak_dbfs);
dumper.DumpRaw("agc2_noise_estimate_dbfs", info.input_noise_level_dbfs);
dumper.DumpRaw("agc2_last_limiter_audio_level", info.limiter_envelope_dbfs);
}
constexpr int kGainApplierAdjacentSpeechFramesThreshold = 1;
constexpr float kMaxGainChangePerSecondDb = 3.f;
constexpr float kMaxOutputNoiseLevelDbfs = -50.f;
} // namespace
AdaptiveAgc::AdaptiveAgc(ApmDataDumper* apm_data_dumper)
: speech_level_estimator_(apm_data_dumper),
gain_applier_(apm_data_dumper,
kGainApplierAdjacentSpeechFramesThreshold,
kMaxGainChangePerSecondDb,
kMaxOutputNoiseLevelDbfs),
apm_data_dumper_(apm_data_dumper),
noise_level_estimator_(apm_data_dumper) {
RTC_DCHECK(apm_data_dumper);
}
AdaptiveAgc::AdaptiveAgc(ApmDataDumper* apm_data_dumper,
const AudioProcessing::Config::GainController2& config)
: speech_level_estimator_(
apm_data_dumper,
config.adaptive_digital.level_estimator,
config.adaptive_digital
.level_estimator_adjacent_speech_frames_threshold,
config.adaptive_digital.initial_saturation_margin_db,
config.adaptive_digital.extra_saturation_margin_db),
vad_(config.adaptive_digital.vad_probability_attack),
gain_applier_(
apm_data_dumper,
config.adaptive_digital.gain_applier_adjacent_speech_frames_threshold,
config.adaptive_digital.max_gain_change_db_per_second,
config.adaptive_digital.max_output_noise_level_dbfs),
apm_data_dumper_(apm_data_dumper),
noise_level_estimator_(apm_data_dumper) {
RTC_DCHECK(apm_data_dumper);
if (!config.adaptive_digital.use_saturation_protector) {
RTC_LOG(LS_WARNING) << "The saturation protector cannot be disabled.";
}
}
AdaptiveAgc::~AdaptiveAgc() = default;
void AdaptiveAgc::Process(AudioFrameView<float> frame, float limiter_envelope) {
AdaptiveDigitalGainApplier::FrameInfo info;
info.vad_result = vad_.AnalyzeFrame(frame);
speech_level_estimator_.Update(info.vad_result);
info.input_level_dbfs = speech_level_estimator_.level_dbfs();
info.input_noise_level_dbfs = noise_level_estimator_.Analyze(frame);
info.limiter_envelope_dbfs =
limiter_envelope > 0 ? FloatS16ToDbfs(limiter_envelope) : -90.f;
info.estimate_is_confident = speech_level_estimator_.IsConfident();
DumpDebugData(info, *apm_data_dumper_);
gain_applier_.Process(info, frame);
}
void AdaptiveAgc::Reset() {
speech_level_estimator_.Reset();
}
} // namespace webrtc