c6abf6cd3f
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.
57 lines
2.0 KiB
C++
57 lines
2.0 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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#include "modules/audio_processing/transient/voice_probability_delay_unit.h"
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#include <array>
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#include "rtc_base/checks.h"
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namespace webrtc {
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VoiceProbabilityDelayUnit::VoiceProbabilityDelayUnit(int delay_num_samples,
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int sample_rate_hz) {
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Initialize(delay_num_samples, sample_rate_hz);
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}
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void VoiceProbabilityDelayUnit::Initialize(int delay_num_samples,
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int sample_rate_hz) {
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RTC_DCHECK_GE(delay_num_samples, 0);
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RTC_DCHECK_LE(delay_num_samples, sample_rate_hz / 50)
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<< "The implementation does not support delays greater than 20 ms.";
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int frame_size = rtc::CheckedDivExact(sample_rate_hz, 100); // 10 ms.
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if (delay_num_samples <= frame_size) {
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weights_[0] = 0.0f;
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weights_[1] = static_cast<float>(delay_num_samples) / frame_size;
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weights_[2] =
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static_cast<float>(frame_size - delay_num_samples) / frame_size;
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} else {
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delay_num_samples -= frame_size;
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weights_[0] = static_cast<float>(delay_num_samples) / frame_size;
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weights_[1] =
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static_cast<float>(frame_size - delay_num_samples) / frame_size;
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weights_[2] = 0.0f;
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}
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// Resets the delay unit.
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last_probabilities_.fill(0.0f);
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}
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float VoiceProbabilityDelayUnit::Delay(float voice_probability) {
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float weighted_probability = weights_[0] * last_probabilities_[0] +
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weights_[1] * last_probabilities_[1] +
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weights_[2] * voice_probability;
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last_probabilities_[0] = last_probabilities_[1];
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last_probabilities_[1] = voice_probability;
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return weighted_probability;
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}
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} // namespace webrtc
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