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libtgvoip/webrtc_dsp/modules/audio_processing/aec3/erl_estimator.h
Grishka 5caaaafa42 Updated WebRTC APM
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.
2018-11-23 04:02:53 +03:00

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1.6 KiB
C++

/*
* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef MODULES_AUDIO_PROCESSING_AEC3_ERL_ESTIMATOR_H_
#define MODULES_AUDIO_PROCESSING_AEC3_ERL_ESTIMATOR_H_
#include <stddef.h>
#include <array>
#include "api/array_view.h"
#include "modules/audio_processing/aec3/aec3_common.h"
#include "rtc_base/constructormagic.h"
namespace webrtc {
// Estimates the echo return loss based on the signal spectra.
class ErlEstimator {
public:
explicit ErlEstimator(size_t startup_phase_length_blocks_);
~ErlEstimator();
// Resets the ERL estimation.
void Reset();
// Updates the ERL estimate.
void Update(bool converged_filter,
rtc::ArrayView<const float> render_spectrum,
rtc::ArrayView<const float> capture_spectrum);
// Returns the most recent ERL estimate.
const std::array<float, kFftLengthBy2Plus1>& Erl() const { return erl_; }
float ErlTimeDomain() const { return erl_time_domain_; }
private:
const size_t startup_phase_length_blocks__;
std::array<float, kFftLengthBy2Plus1> erl_;
std::array<int, kFftLengthBy2Minus1> hold_counters_;
float erl_time_domain_;
int hold_counter_time_domain_;
size_t blocks_since_reset_ = 0;
RTC_DISALLOW_COPY_AND_ASSIGN(ErlEstimator);
};
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_AEC3_ERL_ESTIMATOR_H_