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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.
55 lines
1.7 KiB
C++
55 lines
1.7 KiB
C++
/*
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* Copyright (c) 2017 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_AEC3_REVERB_MODEL_FALLBACK_H_
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#define MODULES_AUDIO_PROCESSING_AEC3_REVERB_MODEL_FALLBACK_H_
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#include <stddef.h>
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#include <array>
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#include <vector>
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#include "modules/audio_processing/aec3/aec3_common.h"
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namespace webrtc {
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// The ReverbModelFallback class describes an exponential reverberant model.
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// This model is expected to be applied over the echo power spectrum that
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// is estimated by the linear filter.
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class ReverbModelFallback {
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public:
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explicit ReverbModelFallback(size_t length_blocks);
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~ReverbModelFallback();
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// Resets the state
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void Reset();
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// Adds the estimated unmodelled echo power to the residual echo power
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// estimate.
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void AddEchoReverb(const std::array<float, kFftLengthBy2Plus1>& S2,
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size_t delay,
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float reverb_decay_factor,
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std::array<float, kFftLengthBy2Plus1>* R2);
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// Returns the current power spectrum reverberation contributions.
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const std::array<float, kFftLengthBy2Plus1>& GetPowerSpectrum() const {
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return R2_reverb_;
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}
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private:
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std::array<float, kFftLengthBy2Plus1> R2_reverb_;
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int S2_old_index_ = 0;
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std::vector<std::array<float, kFftLengthBy2Plus1>> S2_old_;
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_AEC3_REVERB_MODEL_FALLBACK_H_
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