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1e2542f593
This CL makes possible to choose the level estimation for the adaptive digital GC of AGC2. The options are RMS (default and currently used estimator) and peak-based (already computed, but not used). Besides adding the new AGC2 config param for the level estimator, this CL also refactors the config class by making it more structured. Bug: webrtc:7494 Change-Id: I20eb558ca50f13536aa7bdea08d21de3b630f8bc Reviewed-on: https://webrtc-review.googlesource.com/c/110144 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25620}
55 lines
1.9 KiB
C++
55 lines
1.9 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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#ifndef MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_MODE_LEVEL_ESTIMATOR_H_
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#define MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_MODE_LEVEL_ESTIMATOR_H_
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#include <stddef.h>
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#include "modules/audio_processing/agc2/agc2_common.h" // kFullBufferSizeMs...
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#include "modules/audio_processing/agc2/saturation_protector.h"
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#include "modules/audio_processing/agc2/vad_with_level.h"
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#include "modules/audio_processing/include/audio_processing.h"
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namespace webrtc {
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class ApmDataDumper;
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class AdaptiveModeLevelEstimator {
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public:
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explicit AdaptiveModeLevelEstimator(ApmDataDumper* apm_data_dumper);
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AdaptiveModeLevelEstimator(
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ApmDataDumper* apm_data_dumper,
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AudioProcessing::Config::GainController2::LevelEstimator level_estimator,
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bool use_saturation_protector,
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float extra_saturation_margin_db);
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void UpdateEstimation(const VadWithLevel::LevelAndProbability& vad_data);
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float LatestLevelEstimate() const;
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void Reset();
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bool LevelEstimationIsConfident() const {
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return buffer_size_ms_ >= kFullBufferSizeMs;
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}
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private:
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void DebugDumpEstimate();
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const AudioProcessing::Config::GainController2::LevelEstimator
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level_estimator_;
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const bool use_saturation_protector_;
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size_t buffer_size_ms_ = 0;
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float last_estimate_with_offset_dbfs_ = kInitialSpeechLevelEstimateDbfs;
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float estimate_numerator_ = 0.f;
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float estimate_denominator_ = 0.f;
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SaturationProtector saturation_protector_;
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ApmDataDumper* const apm_data_dumper_;
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_MODE_LEVEL_ESTIMATOR_H_
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