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