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https://github.com/danog/libtgvoip.git
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5caaaafa42
I'm now using the entire audio processing module from WebRTC as opposed to individual DSP algorithms pulled from there before. Seems to work better this way.
45 lines
1.5 KiB
C++
45 lines
1.5 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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#ifndef MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_AGC_H_
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#define MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_AGC_H_
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#include "modules/audio_processing/agc2/adaptive_digital_gain_applier.h"
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#include "modules/audio_processing/agc2/adaptive_mode_level_estimator.h"
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#include "modules/audio_processing/agc2/noise_level_estimator.h"
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#include "modules/audio_processing/agc2/vad_with_level.h"
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#include "modules/audio_processing/include/audio_frame_view.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 AdaptiveAgc {
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public:
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explicit AdaptiveAgc(ApmDataDumper* apm_data_dumper);
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AdaptiveAgc(ApmDataDumper* apm_data_dumper,
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const AudioProcessing::Config::GainController2& config);
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~AdaptiveAgc();
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void Process(AudioFrameView<float> float_frame, float last_audio_level);
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void Reset();
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private:
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AdaptiveModeLevelEstimator speech_level_estimator_;
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VadWithLevel vad_;
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AdaptiveDigitalGainApplier gain_applier_;
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ApmDataDumper* const apm_data_dumper_;
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NoiseLevelEstimator noise_level_estimator_;
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_AGC_H_
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