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92ea95e34a
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
94 lines
3.4 KiB
C++
94 lines
3.4 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 MODULES_AUDIO_PROCESSING_RESIDUAL_ECHO_DETECTOR_H_
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#define MODULES_AUDIO_PROCESSING_RESIDUAL_ECHO_DETECTOR_H_
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#include <vector>
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#include "api/array_view.h"
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#include "modules/audio_processing/echo_detector/circular_buffer.h"
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#include "modules/audio_processing/echo_detector/mean_variance_estimator.h"
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#include "modules/audio_processing/echo_detector/moving_max.h"
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#include "modules/audio_processing/echo_detector/normalized_covariance_estimator.h"
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namespace webrtc {
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class ApmDataDumper;
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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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// This function is for testing purposes only.
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void SetReliabilityForTest(float value) { reliability_ = value; }
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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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float echo_likelihood_recent_max() const {
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return recent_likelihood_max_.max();
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}
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private:
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static int instance_count_;
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std::unique_ptr<ApmDataDumper> data_dumper_;
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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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// Reliability of the current likelihood.
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float reliability_ = 0.f;
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MovingMax recent_likelihood_max_;
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int log_counter_ = 0;
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_RESIDUAL_ECHO_DETECTOR_H_
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