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The results of noise-resistant speech signal processing, with the use of linear adaptive filters. To enhance the filtration efficiency, the filtration is proposed to use a cascade signal. This approach improves the noise resistance of the signal. A comparative analysis of the performance of the developed algorithm with known methods of filtration. The effectiveness of the proposed filter is demonstrated...
The growing demand for robust speech processing applications able to operate in adverse scenarios calls for new evaluation protocols and datasets beyond artificial laboratory conditions. The characteristics of real data for a given scenario are rarely discussed in the literature. As a result, methods are often tested based on the author expertise and not always in scenarios with actual practical value...
An automatic, text-independent speaker verification (SV) system is proposed using Line Spectral Frequency (LSF) features. The state-of-the-art Gaussian Mixture Model with Universal Background Model (GMM-UBM) framework is used for speaker modeling and verification. A score-level fusion based technique is employed in order to extract complementary information from static and dynamic LSF features and...
The information extraction and recognition of speech emotional information under the noise environment has been the research hotspot and difficulty, which limits the application of speech emotion recognition technology. An improved spectral subtraction algorithm is presented based on the analysis on the basic principle of spectral subtraction. According to the distribution of noise, the improved algorithm...
Information to be exchanged between two parties needs compression for achieving its efficient transmission. Encoded information gets distorted during its transmission over a channel due to noise. For monitoring and analysis of such noisy traffic of an adversary over communication networks, it is required to find the type of information, whether it is text or speech, then to restore it for further...
Objective measures are favored and widely used by many researchers in evaluating the quality of noise-suppressed speech. A good and reliable objective measure should have property that it could evaluate speech quality in consistent and well correlated with subjective ratings. In this paper, several widely used objective measures are applied to the speech signals with the Chinese languages including...
Use of modern technological advances in real-time biomedical analysis is very crucial. Current work focuses on glottal pathology discrimination based on non-invasive speech analysis techniques. Primary set back in developing such method is irregular performance depreciation of several state of the art acoustic features. To excuse such problems, we have used glottal to noise excitation ratio, which...
The subjective Chinese speech intelligibility evaluation experiments and objective speech transmission index STIPA measurements were respectively carried out under a total of 234 simulated transmission conditions. The relationship curve between Chinese intelligibility scores and STIPA was established and compared with the standard reference curves given in IEC 60268-16 2011. It is found that remarkable...
Speech enhancement and speech separation are important frontends of many speech processing systems. In real tasks, the background noises are often mixed with some human voice interferences. In this paper, we explore a framework to unify speech enhancement and speech separation for a speaker-dependent scenario based on deep neural networks (DNNs). Using a supervised method, DNN is adopted to directly...
The performance of speaker verification system (SVS) declines dramatically in noisy environments. To suppress the adverse impact of the noise on SVS, this paper investigates employing the nonnegative matrix factorization (NMF) technique to reconstruct the speech based on the pre-trained speech basis matrix (SBM) and noise basis matrix (NBM). The contribution of this research lies in utilizing the...
Non-negative spectrogram decomposition and its variants have been extensively investigated for speech enhancement due to their efficiency in extracting perceptually meaningful components from mixtures. Usually, these approaches are implemented on the condition that training samples for one or more sources are available beforehand. However, in many real-world scenarios, it is always impossible for...
Objective evaluation of speech quality is an important part of the quality of communications service, and the speech in high-noise environments affect the people's auditory perception, which is an important factor for people to determine the speech quality. In this paper, a non-intrusive evaluation method for speech quality in high-noise environments is proposed, and a noise tracking and subtraction...
The speaker verification (SV) task has been an active area of research in the last thirty years. One of the recent research topics is on improving the robustness of SV system in challenging environments. This paper examines the robustness of current state of the art SV system against background noise corruptions. Specifically, we consider the scenario where the SV system is trained from noise free...
In this paper, we propose a linear pre- and post-filter for improving the perceptual speech quality of a vocoder based on amplitude spectrum of residual signal. The goal of the pre-filter in analyzer is to make the noise of spectrum inaudible. It attempts to hide the noise under the signal spectrum by exploiting human auditory masking. The post-filter in synthesizer is to improve speech quality. Experimental...
Ever increasing volumes of media content and the desire to extract information from media archives motivate the studies into semantic audio information mining. Much research in this filed concerns development of bespoke systems, in which soundtracks are exclusively classified and segmented, and a specific type of sound is recognized and analyzed. This approach however is detrimental to the complete...
This paper presents a way of improving the recognition rate of a typical Hidden Markov Model (HMM) -based Automatic Speech Recognition (ASR) system by integrating the l1 - least absolute deviation (LAD) algorithm and the l0 - least square (LS) algorithm in a framework designed to selectively use them based on the level of impulse noise present in speech signal. We present the overall architecture...
Efficient digital transmission of continuous-amplitude signals requires source coding which comes at the price of unavoidable quantization errors. Thus, even in clear channel conditions, the quality of the decoded signal is limited due to these source coding errors (quantization). Hybrid Digital-Analog (HDA) codes circumvent this limitation by additionally transmitting the source coding error with...
Dysphonia is a qualitative and quantitative alteration of the voice due to a structural or functional modification of one or more organs involved in voice production. Voice disorders are prevalent in certain working categories, particularly those of teachers, singers and actors. It is possible to evaluate the state of health of a voice through the acoustic analysis of the speech signal. This provides...
This Manuscript probe delinquent of classification of uninterrupted of broad-spectrum aural data for content based recovery. This paper is dealing with scheme for classifying aural data & segmentation is also done on same data so that processing rate is faster. Aural data is able to classify into eight categories Simple speech, noise, silence, music single speech with music, double speech with...
Reverberated speech signals in noisy acoustical environments cause some problems such as reducing speech intelligibility, distinguishing speakers, locating source, quality for hands-free telephony, hearing aid, etc. Adaptive filters can be applied to suppress the interfering signals and reduce the reverberation effects or to dereverberate the received speech signals at microphone. In this paper, Bayesian...
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