/* * 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 #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(delay_num_samples) / frame_size; weights_[2] = static_cast(frame_size - delay_num_samples) / frame_size; } else { delay_num_samples -= frame_size; weights_[0] = static_cast(delay_num_samples) / frame_size; weights_[1] = static_cast(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