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https://github.com/danog/libtgvoip.git
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5caaaafa42
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.
52 lines
1.4 KiB
C++
52 lines
1.4 KiB
C++
/*
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* Copyright (c) 2018 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_AEC3_SKEW_ESTIMATOR_H_
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#define MODULES_AUDIO_PROCESSING_AEC3_SKEW_ESTIMATOR_H_
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#include <stddef.h>
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#include <vector>
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#include "absl/types/optional.h"
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#include "rtc_base/constructormagic.h"
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namespace webrtc {
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// Estimator of API call skew between render and capture.
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class SkewEstimator {
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public:
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explicit SkewEstimator(size_t skew_history_size_log2);
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~SkewEstimator();
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// Resets the estimation.
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void Reset();
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// Updates the skew data for a render call.
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void LogRenderCall() { ++skew_; }
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// Updates and computes the skew at a capture call. Returns an optional which
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// is non-null if a reliable skew has been found.
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absl::optional<int> GetSkewFromCapture();
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private:
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const int skew_history_size_log2_;
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std::vector<float> skew_history_;
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int skew_ = 0;
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int skew_sum_ = 0;
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size_t next_index_ = 0;
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bool sufficient_skew_stored_ = false;
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RTC_DISALLOW_COPY_AND_ASSIGN(SkewEstimator);
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_AEC3_SKEW_ESTIMATOR_H_
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