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libtgvoip/webrtc_dsp/modules/audio_processing/aec3/skew_estimator.h

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/*
* Copyright (c) 2018 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_SKEW_ESTIMATOR_H_
#define MODULES_AUDIO_PROCESSING_AEC3_SKEW_ESTIMATOR_H_
#include <stddef.h>
#include <vector>
#include "absl/types/optional.h"
#include "rtc_base/constructormagic.h"
namespace webrtc {
// Estimator of API call skew between render and capture.
class SkewEstimator {
public:
explicit SkewEstimator(size_t skew_history_size_log2);
~SkewEstimator();
// Resets the estimation.
void Reset();
// Updates the skew data for a render call.
void LogRenderCall() { ++skew_; }
// Updates and computes the skew at a capture call. Returns an optional which
// is non-null if a reliable skew has been found.
absl::optional<int> GetSkewFromCapture();
private:
const int skew_history_size_log2_;
std::vector<float> skew_history_;
int skew_ = 0;
int skew_sum_ = 0;
size_t next_index_ = 0;
bool sufficient_skew_stored_ = false;
RTC_DISALLOW_COPY_AND_ASSIGN(SkewEstimator);
};
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_AEC3_SKEW_ESTIMATOR_H_