Some API deprecation -- ExperimentalAgc and ExperimentalNs are gone. We're continuing to carry iSAC even though it's gone upstream, but maybe we'll want to drop that soon.
44 lines
1.7 KiB
C++
44 lines
1.7 KiB
C++
/*
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* Copyright (c) 2022 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_TRANSIENT_VOICE_PROBABILITY_DELAY_UNIT_H_
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#define MODULES_AUDIO_PROCESSING_TRANSIENT_VOICE_PROBABILITY_DELAY_UNIT_H_
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#include <array>
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namespace webrtc {
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// Iteratively produces a sequence of delayed voice probability values given a
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// fixed delay between 0 and 20 ms and given a sequence of voice probability
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// values observed every 10 ms. Supports fractional delays, that are delays
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// which are not a multiple integer of 10 ms. Applies interpolation with
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// fractional delays; otherwise, returns a previously observed value according
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// to the given fixed delay.
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class VoiceProbabilityDelayUnit {
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public:
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// Ctor. `delay_num_samples` is the delay in number of samples and it must be
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// non-negative and less than 20 ms.
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VoiceProbabilityDelayUnit(int delay_num_samples, int sample_rate_hz);
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// Handles delay and sample rate changes and resets the delay unit.
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void Initialize(int delay_num_samples, int sample_rate_hz);
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// Observes `voice_probability` and returns a delayed voice probability.
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float Delay(float voice_probability);
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private:
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std::array<float, 3> weights_;
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std::array<float, 2> last_probabilities_;
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_TRANSIENT_VOICE_PROBABILITY_DELAY_UNIT_H_
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