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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.
49 lines
1.5 KiB
C++
49 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_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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