Relevant bibliographies by topics / Perceptual speech filtering

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  1. Journal articles
  2. Dissertations / Theses
  3. Book chapters
  4. Conference papers

Academic literature on the topic 'Perceptual speech filtering'

Author: Grafiati
Published: 3 June 2025
Last updated: 20 July 2025

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Journal articles on the topic "Perceptual speech filtering"

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Zoghlami, Novlene, and Zied Lachiri. "Application of Perceptual Filtering Models to Noisy Speech Signals Enhancement." Journal of Electrical and Computer Engineering 2012 (2012): 1–12. http://dx.doi.org/10.1155/2012/282019.

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This paper describes a new speech enhancement approach using perceptually based noise reduction. The proposed approach is based on the application of two perceptual filtering models to noisy speech signals: the gammatone and the gammachirp filter banks with nonlinear resolution according to the equivalent rectangular bandwidth (ERB) scale. The perceptual filtering gives a number of subbands that are individually spectral weighted and modified according to two different noise suppression rules. The importance of an accurate noise estimate is related to the reduction of the musical noise artifac
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Lin, L., W. H. Holmes, and E. Ambikairajah. "Speech denoising using perceptual modification of Wiener filtering." Electronics Letters 38, no. 23 (2002): 1486. http://dx.doi.org/10.1049/el:20020965.

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Espy-Wilson, Carol Y., Venkatesh R. Chari, Joel M. MacAuslan, Caroline B. Huang, and Michael J. Walsh. "Enhancement of Electrolaryngeal Speech by Adaptive Filtering." Journal of Speech, Language, and Hearing Research 41, no. 6 (1998): 1253–64. http://dx.doi.org/10.1044/jslhr.4106.1253.

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Artificial larynges provide a means of verbal communication for people who have either lost or are otherwise unable to use their larynges. Although they enable adequate communication, the resulting speech has an unnatural quality and is significantly less intelligible than normal speech. One of the major problems with the widely used Transcutaneous Artificial Larynx (TAL) is the presence of a steady background noise caused by the leakage of acoustic energy from the TAL, its interface with the neck, and the surrounding neck tissue. The severity of the problem varies from speaker to speaker, par
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Boubakir, Chabane, and Daoud Berkani. "An Improved MMSE Amplitude Estimator under Generalized Gamma Distribution Based on Auditory Perception." Mathematical Problems in Engineering 2013 (2013): 1–7. http://dx.doi.org/10.1155/2013/821760.

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This paper describes a new speech enhancement approach which employs the minimum mean square error (MMSE) estimator based on the generalized gamma distribution of the short-time spectral amplitude (STSA) of a speech signal. In the proposed approach, the human perceptual auditory masking effect is incorporated into the speech enhancement system. The algorithm is based on a criterion by which the audible noise may be masked rather than being attenuated, thereby reducing the chance of speech distortion. Performance assessment is given to show that our proposal can achieve a more significant noise
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Oren, Liran, Ann W. Kummer, and Suzanne Boyce. "Secretion Bubbling as the Sound Mechanism for Nasal Rustle: A Perceptual Study." Journal of Speech, Language, and Hearing Research 65, no. 3 (2022): 869–77. http://dx.doi.org/10.1044/2021_jslhr-21-00137.

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Purpose: Secretion bubbling on the superior aspect of the velopharyngeal (VP) valve typically occurs with a small VP opening during production of oral pressure consonants. The use of high-speed nasopharyngoscopy has shown correlation between the bubbling frequency and the acoustics captured with the nasal microphone of the nasometer. The purpose of this study was to investigate if the sound generated by the bubbling process is perceived as nasal rustle (also known as nasal turbulence). Method: Speech samples were extracted from the data of patients who were diagnosed with nasal rustle (five bo
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Wang, Jie, Linhuang Yan, Jiayi Tian, and Minmin Yuan. "Speech enhancement algorithm of improved OMLSA based on bilateral spectrogram filtering." Journal of Intelligent & Fuzzy Systems 39, no. 5 (2020): 6881–89. http://dx.doi.org/10.3233/jifs-192088.

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In this paper, a bilateral spectrogram filtering (BSF)-based optimally modified log-spectral amplitude (OMLSA) estimator for single-channel speech enhancement is proposed, which can significantly improve the performance of OMLSA, especially in highly non-stationary noise environments, by taking advantage of bilateral filtering (BF), a widely used technology in image and visual processing, to preprocess the spectrogram of the noisy speech. BSF is capable of not only sharpening details, removing unwanted textures or background noise from the noisy speech spectrogram, but also preserving edges wh
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Renza, Diego, Jaisson Vargas, and Dora M. Ballesteros. "Robust Speech Hashing for Digital Audio Forensics." Applied Sciences 10, no. 1 (2019): 249. http://dx.doi.org/10.3390/app10010249.

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The verification of the integrity and authenticity of multimedia content is an essential task in the forensic field, in order to make digital evidence admissible. The main objective is to establish whether the multimedia content has been manipulated with significant changes to its content, such as the removal of noise (e.g., a gunshot) that could clarify the facts of a crime. In this project we propose a method to generate a summary value for audio recordings, known as hash. Our method is robust, which means that if the audio has been modified slightly (without changing its significant content
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Charitha, Mali. "Deep Causal Speech Enhancement and Recognition Using Efficient Long-Short Term Memory Recurrent Neural Network." INTERNATIONAL JOURNAL OF SCIENTIFIC RESEARCH IN ENGINEERING AND MANAGEMENT 09, no. 06 (2025): 1–9. https://doi.org/10.55041/ijsrem49326.

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Abstract - In this work, we propose an attention-based beamforming framework for multi-channel speech enhancement that dynamically adapts spatial filtering to complex acoustic environments. Traditional beamforming methods rely on fixed or heuristically derived spatial filters, limiting their robustness in the presence of non-stationary noise and reverberation. Our approach leverages a self-attention mechanism to learn context-aware representations of spatial cues across multiple microphone channels, enabling the model to emphasize target speech while suppressing interfering sources. By integra
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Li, Ying-Yi, Pravin Ramadas, and Jerry Gibson. "Multimode Tree-Coding of Speech with Pre-/Post-Weighting." Applied Sciences 12, no. 4 (2022): 2026. http://dx.doi.org/10.3390/app12042026.

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As speech-coding standards have improved over the years, so complexity has increased, and less emphasis been placed on low encoding/decoding delay. We present a low-complexity, low-delay speech codec based on tree-coding with sample-by-sample adaptive long- and short-code generators that incorporates pre- and post-filtering for perceptual weighting and multimode speech classification with comfort noise generation (CNG). The pre-/post-weighting filters adapt based on the code generator parameters available at both the encoder and decoder rather than the usual method that uses the input speech.
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KHALDI, KAIS, MONIA TURKI-HADJ ALOUANE, and ABDEL-OUAHAB BOUDRAA. "VOICED SPEECH ENHANCEMENT BASED ON ADAPTIVE FILTERING OF SELECTED INTRINSIC MODE FUNCTIONS." Advances in Adaptive Data Analysis 02, no. 01 (2010): 65–80. http://dx.doi.org/10.1142/s1793536910000409.

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In this paper a new method for voiced speech enhancement combining the Empirical Mode Decomposition (EMD) and the Adaptive Center Weighted Average (ACWA) filter is introduced. Noisy signal is decomposed adaptively into intrinsic oscillatory components called Intrinsic Mode Functions (IMFs). Since voiced speech structure is mostly distributed on both medium and low frequencies, the shorter scale IMFs of the noisy signal are beneath noise, however the longer scale ones are less noisy. Therefore, the main idea of the proposed approach is to only filter the shorter scale IMFs, and to keep the long
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Dissertations / Theses on the topic "Perceptual speech filtering"

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Klein, Mark 1977. "Signal subspace speech enhancement with perceptual post-filtering." Thesis, McGill University, 2002. http://digitool.Library.McGill.CA:80/R/?func=dbin-jump-full&object_id=33975.

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Speech enhancement blocks form a critical part of voice communications systems. Unfortunately, most enhancement schemes have difficulty eliminating noise from speech without introducing distortion or artefacts. Many of the disturbances originate from poor parameter estimation and interframe fluctuations.<br>This thesis introduces the Enhanced Signal Subspace (ESS) system to mitigate the above problems. Based on a signal subspace framework, ESS has been designed to attenuate disturbances while minimizing audible distortion.<br>Artefacts are reduced by employing an auditory post-filter to smooth
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Wang, Yao Electrical Engineering &amp Telecommunications Faculty of Engineering UNSW. "Single channel speech enhancement based on perceptual temporal masking model." Awarded by:University of New South Wales. Electrical Engineering & Telecommunications, 2007. http://handle.unsw.edu.au/1959.4/40454.

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In most speech communication systems, the presence of background noise causes the quality and intelligibility of speech to degrade, especially when the Signal-to-Noise Ratio (SNR) is low. Numerous speech enhancement techniques have been employed successfully in many applications. However, at low signal-to-noise ratios most of these speech enhancement techniques tend to introduce a perceptually annoying residual noise known as "musical noise". The research presented in this thesis aims to minimize this musical noise and maximize the noise reduction ability of speech enhancement algorithms to im
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Book chapters on the topic "Perceptual speech filtering"

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Han, Wei, Xiongwei Zhang, Jibin Yang, Meng Sun, and Gang Min. "Joint Optimization of a Perceptual Modified Wiener Filtering Mask and Deep Neural Networks for Monaural Speech Separation." In Lecture Notes in Computer Science. Springer International Publishing, 2016. http://dx.doi.org/10.1007/978-3-319-48896-7_46.

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Conference papers on the topic "Perceptual speech filtering"

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Tuan Van Pham, Michael Stark, and Gernot Kubin. "Perceptual wavelet filtering for robust speech recognition." In ICASSP 2008 - 2008 IEEE International Conference on Acoustics, Speech and Signal Processing. IEEE, 2008. http://dx.doi.org/10.1109/icassp.2008.4518627.

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Amehraye, A., D. Pastor, and A. Tamtaoui. "Perceptual improvement of Wiener filtering." In ICASSP 2008 - 2008 IEEE International Conference on Acoustics, Speech and Signal Processing. IEEE, 2008. http://dx.doi.org/10.1109/icassp.2008.4518051.

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Klein and Kabal. "Signal subspace speech enhancement with perceptual post-filtering." In IEEE International Conference on Acoustics Speech and Signal Processing ICASSP-02. IEEE, 2002. http://dx.doi.org/10.1109/icassp.2002.1005795.

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Klein, Mark, and Peter Kabal. "Signal subspace speech enhancement with perceptual post-filtering." In Proceedings of ICASSP '02. IEEE, 2002. http://dx.doi.org/10.1109/icassp.2002.5743773.

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Ma, N., M. Bouchard, and R. A. Goubran. "A perceptual kalman filtering-based approach for speech enhancement." In Seventh International Symposium on Signal Processing and Its Applications, 2003. Proceedings. IEEE, 2003. http://dx.doi.org/10.1109/isspa.2003.1224718.

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Lin, L., W. H. Holmes, and E. Ambikairajah. "Speech enhancement based on a perceptual modification of wiener filtering." In 7th International Conference on Spoken Language Processing (ICSLP 2002). ISCA, 2002. http://dx.doi.org/10.21437/icslp.2002-259.

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Odugu, Kishore, and B. M. S. Sreenivasa Rao. "New speech enhancement using Gamma tone filters and Perceptual Wiener filtering based on sub banding." In 2013 International Conference on Signal Processing and Communication (ICSC). IEEE, 2013. http://dx.doi.org/10.1109/icspcom.2013.6719789.

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Shujau, M., C. H. Ritz, and I. S. Burnett. "Linear Predictive perceptual filtering for Acoustic Vector Sensors: Exploiting directional recordings for high quality speech enhancement." In ICASSP 2011 - 2011 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP). IEEE, 2011. http://dx.doi.org/10.1109/icassp.2011.5947496.

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Biswas, Astik, P. K. Sahu, Anirban Bhowmick, and Mahesh Chandra. "Acoustic feature extraction using ERB like wavelet sub-band perceptual Wiener filtering for noisy speech recognition." In 2014 Annual IEEE India Conference (INDICON). IEEE, 2014. http://dx.doi.org/10.1109/indicon.2014.7030474.

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Yao Wang, Jiong An, Vidhyasaharan Sethu, and Eliathamby Ambikairajah. "Perceptually motivated pre-filter for speech enhancement using Kalman filtering." In 2007 6th International Conference on Information, Communications & Signal Processing. IEEE, 2007. http://dx.doi.org/10.1109/icics.2007.4449758.

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