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https://github.com/danog/libtgvoip.git
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49 lines
1.5 KiB
C
49 lines
1.5 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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#ifndef MODULES_AUDIO_PROCESSING_AGC2_VAD_WITH_LEVEL_H_
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#define MODULES_AUDIO_PROCESSING_AGC2_VAD_WITH_LEVEL_H_
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#include "common_audio/resampler/include/push_resampler.h"
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#include "modules/audio_processing/agc2/rnn_vad/features_extraction.h"
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#include "modules/audio_processing/agc2/rnn_vad/rnn.h"
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#include "modules/audio_processing/include/audio_frame_view.h"
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namespace webrtc {
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class VadWithLevel {
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public:
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struct LevelAndProbability {
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constexpr LevelAndProbability(float prob, float rms, float peak)
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: speech_probability(prob),
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speech_rms_dbfs(rms),
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speech_peak_dbfs(peak) {}
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LevelAndProbability() = default;
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float speech_probability = 0;
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float speech_rms_dbfs = 0; // Root mean square in decibels to full-scale.
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float speech_peak_dbfs = 0;
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};
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VadWithLevel();
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~VadWithLevel();
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LevelAndProbability AnalyzeFrame(AudioFrameView<const float> frame);
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private:
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void SetSampleRate(int sample_rate_hz);
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rnn_vad::RnnBasedVad rnn_vad_;
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rnn_vad::FeaturesExtractor features_extractor_;
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PushResampler<float> resampler_;
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_AGC2_VAD_WITH_LEVEL_H_
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