mirror of
https://github.com/danog/libtgvoip.git
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82 lines
3.0 KiB
C++
82 lines
3.0 KiB
C++
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/*
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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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namespace webrtc {
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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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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.use_saturation_protector,
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config.adaptive_digital.extra_saturation_margin_db),
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gain_applier_(apm_data_dumper),
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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() = default;
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void AdaptiveAgc::Process(AudioFrameView<float> float_frame,
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float last_audio_level) {
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auto signal_with_levels = SignalWithLevels(float_frame);
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signal_with_levels.vad_result = vad_.AnalyzeFrame(float_frame);
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apm_data_dumper_->DumpRaw("agc2_vad_probability",
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signal_with_levels.vad_result.speech_probability);
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apm_data_dumper_->DumpRaw("agc2_vad_rms_dbfs",
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signal_with_levels.vad_result.speech_rms_dbfs);
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apm_data_dumper_->DumpRaw("agc2_vad_peak_dbfs",
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signal_with_levels.vad_result.speech_peak_dbfs);
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speech_level_estimator_.UpdateEstimation(signal_with_levels.vad_result);
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signal_with_levels.input_level_dbfs =
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speech_level_estimator_.LatestLevelEstimate();
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signal_with_levels.input_noise_level_dbfs =
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noise_level_estimator_.Analyze(float_frame);
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apm_data_dumper_->DumpRaw("agc2_noise_estimate_dbfs",
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signal_with_levels.input_noise_level_dbfs);
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signal_with_levels.limiter_audio_level_dbfs =
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last_audio_level > 0 ? FloatS16ToDbfs(last_audio_level) : -90.f;
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apm_data_dumper_->DumpRaw("agc2_last_limiter_audio_level",
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signal_with_levels.limiter_audio_level_dbfs);
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signal_with_levels.estimate_is_confident =
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speech_level_estimator_.LevelEstimationIsConfident();
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// The gain applier applies the gain.
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gain_applier_.Process(signal_with_levels);
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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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