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/*
* Copyright (c) 2016 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 WEBRTC_MODULES_AUDIO_PROCESSING_RESIDUAL_ECHO_DETECTOR_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_RESIDUAL_ECHO_DETECTOR_H_
#include <vector>
#include "webrtc/base/array_view.h"
#include "webrtc/modules/audio_processing/echo_detector/circular_buffer.h"
#include "webrtc/modules/audio_processing/echo_detector/mean_variance_estimator.h"
#include "webrtc/modules/audio_processing/echo_detector/moving_max.h"
#include "webrtc/modules/audio_processing/echo_detector/normalized_covariance_estimator.h"
namespace webrtc {
class AudioBuffer;
class EchoDetector;
class ResidualEchoDetector {
public:
ResidualEchoDetector();
~ResidualEchoDetector();
// This function should be called while holding the render lock.
void AnalyzeRenderAudio(rtc::ArrayView<const float> render_audio);
// This function should be called while holding the capture lock.
void AnalyzeCaptureAudio(rtc::ArrayView<const float> capture_audio);
// This function should be called while holding the capture lock.
void Initialize();
// This function is for testing purposes only.
void SetReliabilityForTest(float value) { reliability_ = value; }
static void PackRenderAudioBuffer(AudioBuffer* audio,
std::vector<float>* packed_buffer);
// This function should be called while holding the capture lock.
float echo_likelihood() const { return echo_likelihood_; }
float echo_likelihood_recent_max() const {
return recent_likelihood_max_.max();
}
private:
// Keep track if the |Process| function has been previously called.
bool first_process_call_ = true;
// Buffer for storing the power of incoming farend buffers. This is needed for
// cases where calls to BufferFarend and Process are jittery.
CircularBuffer render_buffer_;
// Count how long ago it was that the size of |render_buffer_| was zero. This
// value is also reset to zero when clock drift is detected and a value from
// the renderbuffer is discarded, even though the buffer is not actually zero
// at that point. This is done to avoid repeatedly removing elements in this
// situation.
size_t frames_since_zero_buffer_size_ = 0;
// Circular buffers containing delayed versions of the power, mean and
// standard deviation, for calculating the delayed covariance values.
std::vector<float> render_power_;
std::vector<float> render_power_mean_;
std::vector<float> render_power_std_dev_;
// Covariance estimates for different delay values.
std::vector<NormalizedCovarianceEstimator> covariances_;
// Index where next element should be inserted in all of the above circular
// buffers.
size_t next_insertion_index_ = 0;
MeanVarianceEstimator render_statistics_;
MeanVarianceEstimator capture_statistics_;
// Current echo likelihood.
float echo_likelihood_ = 0.f;
// Reliability of the current likelihood.
float reliability_ = 0.f;
MovingMax recent_likelihood_max_;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_RESIDUAL_ECHO_DETECTOR_H_
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