Evidence map›Paper›PMID 37167227›Full record

ArticlePloS one2023

Causal speech enhancement using dynamical-weighted loss and attention encoder-decoder recurrent neural network.

Fahad Khalil Peracha, Muhammad Irfan Khattak, Nema Salem, Nasir Saleem

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Article in PloS one, 2023. The graph could read no effect estimate from its abstract, so it casts no vote on the map. Cited by 3 papers.

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3citing papers in PubMed
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1 · What the graph read from it

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2 · The registry

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3 · Its place in the literature

Who cites it

3 citing papers in PubMed.

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4 · The record

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5 · Who and what money

Authors and funding

4 authors.

Fahad Khalil PerachaDepartment of Electrical Engineering, University of Engineering and Technology, Peshawar, KPK, Pakistan.
Muhammad Irfan KhattakDepartment of Electrical Engineering, University of Engineering and Technology, Peshawar, KPK, Pakistan.
Nema SalemElectrical and Computer Engineering Department, Effat College of Engineering, Effat University, Jeddah, KSA.ORCID 0000-0002-8440-4111
Nasir SaleemDepartment of Electrical Engineering, University of Engineering and Technology, Peshawar, KPK, Pakistan.ORCID 0000-0003-0010-0629

Funding

No grant is acknowledged in the PubMed record.

6 · The paper itself

Abstract

Speech enhancement (SE) reduces background noise signals in target speech and is applied at the front end in various real-world applications, including robust ASRs and real-time processing in mobile phone communications. SE systems are commonly integrated into mobile phones to increase quality and intelligibility. As a result, a low-latency system is required to operate in real-world applications. On the other hand, these systems need efficient optimization. This research focuses on the single-microphone SE operating in real-time systems with better optimization. We propose a causal data-driven model that uses attention encoder-decoder long short-term memory (LSTM) to estimate the time-frequency mask from a noisy speech in order to make a clean speech for real-time applications that need low-latency causal processing. The encoder-decoder LSTM and a causal attention mechanism are used in the proposed model. Furthermore, a dynamical-weighted (DW) loss function is proposed to improve model learning by varying the weight loss values. Experiments demonstrated that the proposed model consistently improves voice quality, intelligibility, and noise suppression. In the causal processing mode, the LSTM-based estimated suppression time-frequency mask outperforms the baseline model for unseen noise types. The proposed SE improved the STOI by 2.64% (baseline LSTM-IRM), 6.6% (LSTM-KF), 4.18% (DeepXi-KF), and 3.58% (DeepResGRU-KF). In addition, we examine word error rates (WERs) using Google's Automatic Speech Recognition (ASR). The ASR results show that error rates decreased from 46.33% (noisy signals) to 13.11% (proposed) 15.73% (LSTM), and 14.97% (LSTM-KF).

Indexed as

SpeechSpeech PerceptionNeural Networks, ComputerNoiseSpeech Intelligibility

Identifiers

PMID37167227
PMCPMC10174555

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Read under generation 80e0d062 · epoch 390. Bibliography from PubMed, PubMed Central and OpenAlex; grants from NIH RePORTER; trial links from ClinicalTrials.gov; estimates, votes and beliefs from the OpenQuestion graph.