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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.
72 lines
2.0 KiB
C++
72 lines
2.0 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_SATURATION_PROTECTOR_H_
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#define MODULES_AUDIO_PROCESSING_AGC2_SATURATION_PROTECTOR_H_
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#include <array>
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#include "modules/audio_processing/agc2/agc2_common.h"
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#include "modules/audio_processing/agc2/vad_with_level.h"
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namespace webrtc {
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class ApmDataDumper;
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class SaturationProtector {
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public:
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explicit SaturationProtector(ApmDataDumper* apm_data_dumper);
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SaturationProtector(ApmDataDumper* apm_data_dumper,
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float extra_saturation_margin_db);
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// Update and return margin estimate. This method should be called
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// whenever a frame is reliably classified as 'speech'.
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//
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// Returned value is in DB scale.
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void UpdateMargin(const VadWithLevel::LevelAndProbability& vad_data,
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float last_speech_level_estimate_dbfs);
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// Returns latest computed margin. Used in cases when speech is not
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// detected.
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float LastMargin() const;
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// Resets the internal memory.
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void Reset();
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void DebugDumpEstimate() const;
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private:
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// Computes a delayed envelope of peaks.
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class PeakEnveloper {
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public:
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PeakEnveloper();
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void Process(float frame_peak_dbfs);
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float Query() const;
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private:
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size_t speech_time_in_estimate_ms_ = 0;
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float current_superframe_peak_dbfs_ = -90.f;
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size_t elements_in_buffer_ = 0;
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std::array<float, kPeakEnveloperBufferSize> peak_delay_buffer_ = {};
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};
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ApmDataDumper* apm_data_dumper_;
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float last_margin_;
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PeakEnveloper peak_enveloper_;
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const float extra_saturation_margin_db_;
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_AGC2_SATURATION_PROTECTOR_H_
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