af27ed01d7
This algorithm calculates an estimate of the Pearson product-moment correlation coefficient between the power of 10ms audio buffers taken from the render and capture sides, for various different delay values. BUG=webrtc:6525 Review-Url: https://codereview.webrtc.org/2419563003 Cr-Commit-Position: refs/heads/master@{#14824}
79 lines
3.0 KiB
C++
79 lines
3.0 KiB
C++
/*
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* Copyright (c) 2016 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 WEBRTC_MODULES_AUDIO_PROCESSING_RESIDUAL_ECHO_DETECTOR_H_
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#define WEBRTC_MODULES_AUDIO_PROCESSING_RESIDUAL_ECHO_DETECTOR_H_
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#include <vector>
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#include "webrtc/base/array_view.h"
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#include "webrtc/modules/audio_processing/echo_detector/circular_buffer.h"
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#include "webrtc/modules/audio_processing/echo_detector/mean_variance_estimator.h"
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#include "webrtc/modules/audio_processing/echo_detector/normalized_covariance_estimator.h"
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namespace webrtc {
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class AudioBuffer;
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class EchoDetector;
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class ResidualEchoDetector {
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public:
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ResidualEchoDetector();
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~ResidualEchoDetector();
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// This function should be called while holding the render lock.
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void AnalyzeRenderAudio(rtc::ArrayView<const float> render_audio);
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// This function should be called while holding the capture lock.
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void AnalyzeCaptureAudio(rtc::ArrayView<const float> capture_audio);
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// This function should be called while holding the capture lock.
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void Initialize();
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static void PackRenderAudioBuffer(AudioBuffer* audio,
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std::vector<float>* packed_buffer);
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// This function should be called while holding the capture lock.
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float echo_likelihood() const { return echo_likelihood_; }
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private:
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// Keep track if the |Process| function has been previously called.
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bool first_process_call_ = true;
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// Buffer for storing the power of incoming farend buffers. This is needed for
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// cases where calls to BufferFarend and Process are jittery.
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CircularBuffer render_buffer_;
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// Count how long ago it was that the size of |render_buffer_| was zero. This
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// value is also reset to zero when clock drift is detected and a value from
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// the renderbuffer is discarded, even though the buffer is not actually zero
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// at that point. This is done to avoid repeatedly removing elements in this
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// situation.
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size_t frames_since_zero_buffer_size_ = 0;
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// Circular buffers containing delayed versions of the power, mean and
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// standard deviation, for calculating the delayed covariance values.
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std::vector<float> render_power_;
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std::vector<float> render_power_mean_;
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std::vector<float> render_power_std_dev_;
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// Covariance estimates for different delay values.
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std::vector<NormalizedCovarianceEstimator> covariances_;
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// Index where next element should be inserted in all of the above circular
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// buffers.
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size_t next_insertion_index_ = 0;
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MeanVarianceEstimator render_statistics_;
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MeanVarianceEstimator capture_statistics_;
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// Current echo likelihood.
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float echo_likelihood_ = 0.f;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_PROCESSING_RESIDUAL_ECHO_DETECTOR_H_
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