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The standard ITU-T G729 coder uses an interframe quantization of the LSF parameters which causes error propagation to the next frames. We propose the use of intraframe quantization schemes to overcome this problem. We give a performance comparison between intraframe and interframe quantization methods of LSF parameters for ITU-T G729. Simulations results show that our intraframe quantization method...
One of the most significant applications of the speech processing is speech enhancement. Several classical algorithms have been proposed and they suffer from one serious drawback of enhancing speech signal under very low signal to noise ratio. In this paper, this is accomplished by exploiting the harmonic structure of voiced segments. This method improves speech quality by suppressing the noise in...
In speech production systems, the vocal tract is modeled by a filter and glottal pulse by an excitation signal. In most traditional systems impulse train or noise is used as the excitation. In this paper the effects of different excitation signals on Mel-generalized cepstral filters (LPC, warped LPC, mel cepstral and ML cepstral) are studied. Excitation signals with different pulse shapes are used...
Compressive sensing (CS) is a new approach to simultaneous sensing and compression of sparse and compressible signals, i.e. speech signal. Compressive sensing is a new paradigm of acquiring signals, fundamentally different from uniform rate digitization followed by compression, often used for transmission or storage. In this paper, a novel algorithm for speech coding utilizing CS principle is developed...
The evolution of telecommunications technology from circuit to packet switching, in which the Internet is the great paradigm, offers nowadays more flexibility in services and higher efficiency in the transport infrastructure that was not experienced before. As consequence, several new services have emerged. Among them, Voice over IP (VoIP) is, indubitably, one of the most popular, mainly because it...
The aim of this paper is to investigate the importance of phase and magnitude spectra in speech enhancement at different conditions with emphasizing on the role of phase spectrum. The speech signal is exposed to additive noise in different SNRs. Then, it is decomposed into different frame lengths from 32 to 1024 ms. In synthesis stage we have used clean phase spectrum along with noisy magnitude spectrum...
The aim of this paper is to investigate the importance of phase and magnitude spectra in speech enhancement at different conditions with emphasizing on the role of phase spectrum. The speech signal is exposed to additive noise in different SNRs. Then, it is decomposed into different frame lengths from 32 to 1024 ms. In synthesis stage we have used clean phase spectrum along with noisy magnitude spectrum...
VoIP (Voice over IP) is a modern service with enormous potential for yet further growth. It uses the already available and universally implemented IP transport platform. One significant problem, however, is ensuring the Quality of Service, abbreviated QoS. This paper addresses exactly that issue. In an extensive investigation the influence of jitter buffers on QoS is being examined in depth. Two implementations,...
This paper describes a double-ended quality assessment system for speech with a bandwidth of up to 14 kHz (so-called super-wideband speech). The quality assessment system is based on a combination of local and global features, where the local features are dependent on a time alignment procedure and the global features are not. The system is evaluated over a large set of subjectively scored narrowband,...
The Singular Value Decomposition (SVD) is a powerful tool used for subspace division. In this paper a novel approach for speech signal enhancement is presented which is based on SVD and Genetic Algorithm (GA). The method is derived from the effects of environmental noises on the singular vectors as well as the singular values of a clean speech. This article reviews the existing approaches for subspace...
This paper presents a dual-mode switching method between time-domain codec and transform-domain codec of audio coding. It is a key technique of unified speech and audio (music) coding, since the replaying audio quality corresponds to the suitable codec selection and smooth switching between them. The proposed method consists of two steps, codec mode selection and switching. The binary decision trees...
A well known and used method of intrusive objective quality assessment is ITU-T P.862 - “Perceptual Evaluation of Speech Quality (PESQ) suffers from certain weaknesses. The standard is falling behind the latest coding technologies and as some papers have shown lacks the sufficient accuracy in performing reliable and accurate estimation. The proposed method of Enhanced Voice Quality Estimation (EVQE)...
Different from traditional firms, online retailers do not often produce common goods on their own. On the contrary, they sell merchandises that traditional firms produce. Furthermore, online retailers are also different from traditional retailers because the former do not often have frontline employees, physical outlets, or displaying layouts. Therefore, when building a strong online retailer brand,...
Speech enhancement is the process of de-noising a speech signal for improved quality and better intelligibility. Several speech enhancement methods have been proposed including: DWFM filtering, Donoho, Massart, and Kalman. To measure the performance of these filters, a speech evaluation method is needed. SNR is one of the most common methods for speech evaluation. The problem of SNR as a waveform...
IP networks are indispensable nowadays. They are among the most efficient platforms. The constantly growing number of users and new services in these nets – the largest representative being the Internet – requires a good quality of any application used. The determination of the QoS in the real-time services is particularly important. This paper is dedicated to exactly this aspect of the real-time...
Voice over Internet Protocol (VoIP) is a technology that transports voice data packets across packet switched networks using the Internet Protocol (IP). Perceived speech quality is the key metric for QoS for VoIP applications. The most popular objective measurement method for VoIP voice is E-Model. The primary aim of this work is to investigate the how the delay affect the VoIP quality and how to...
This paper proposes a data hiding approach to embed data on compressed speech bit stream in order to transmit two simultaneous speeches instead of one. The host and embedded signals are Enhanced Full Rate (EFR) and Mixed-Excitation Linear Predictive enhanced (MELPe) encoded speech bit streams. Host and hidden speech quality is determined by Perceptual Evaluation of Speech Quality (PESQ) which is an...
The growth of Internet has led to the development of many new applications and technologies. Voice over Internet Protocol (VoIP) is one of the fastest growing applications. Calculating the quality of calls has been a complex task. The ITU E-Model gives a framework to measure quality of VoIP calls but the MOS element is a subjective measure. In this paper, we discuss a novel method using Random Neural...
Most speech enhancement algorithms heavily depend on the noise power spectral density (PSD). Because this quantity is unknown in practice, estimation from the noisy data is necessary. We present a low complexity method for noise PSD estimation. The algorithm is based on a minimum mean-squared error estimator of the noise magnitude-squared DFT coefficients. Compared to minimum statistics based noise...
In this paper, we propose a closed loop system to improve the performance of single-channel speech separation in a speaker independent scenario. The system is composed of two interconnected blocks: a separation block and a speaker identification block. The improvement is accomplished by incorporating the speaker identities found by the speaker identification block as additional information for the...
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