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+/*
+ * Copyright (c) 2022 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_TRANSIENT_VOICE_PROBABILITY_DELAY_UNIT_H_
+#define MODULES_AUDIO_PROCESSING_TRANSIENT_VOICE_PROBABILITY_DELAY_UNIT_H_
+
+#include <array>
+
+namespace webrtc {
+
+// Iteratively produces a sequence of delayed voice probability values given a
+// fixed delay between 0 and 20 ms and given a sequence of voice probability
+// values observed every 10 ms. Supports fractional delays, that are delays
+// which are not a multiple integer of 10 ms. Applies interpolation with
+// fractional delays; otherwise, returns a previously observed value according
+// to the given fixed delay.
+class VoiceProbabilityDelayUnit {
+ public:
+ // Ctor. `delay_num_samples` is the delay in number of samples and it must be
+ // non-negative and less than 20 ms.
+ VoiceProbabilityDelayUnit(int delay_num_samples, int sample_rate_hz);
+
+ // Handles delay and sample rate changes and resets the delay unit.
+ void Initialize(int delay_num_samples, int sample_rate_hz);
+
+ // Observes `voice_probability` and returns a delayed voice probability.
+ float Delay(float voice_probability);
+
+ private:
+ std::array<float, 3> weights_;
+ std::array<float, 2> last_probabilities_;
+};
+
+} // namespace webrtc
+
+#endif // MODULES_AUDIO_PROCESSING_TRANSIENT_VOICE_PROBABILITY_DELAY_UNIT_H_