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Currently, there are more than 500 Quranic recitations available freely on the internet. There is also a growing trends on the use of smart phone compare to traditional desktop PCs for accessing the internet. On such limited device, a high quality speech compression for Quranic recitation is favorable. In this paper, we developed a high quality speech compression for Quranic recitation by modifying...
This paper presents a comparative analysis for enhancement of noisy single channel Hindi speech patterns', using a binary mask threshold function in mother wavelet transforms. In this wavelet transform a three level of wavelet decomposition is used and all three levels are given individually to binary mask threshold for removing noise and enhancing the speech patterns. The suitability of the binary...
This paper addresses a packet loss concealment method (PLC) based on piggybacking to improve speech quality degradation caused by packet losses for code excited linear predictive (CELP) type coders. We applied our proposed scheme to the standard ITU-T G.729 Conjugate-Structure Algebraic CELP (CS-CELP) speech coder to evaluate its performance. The average spectral distortion (Avg. SD), the perceptual...
This paper undertakes a detailed comparative analysis of both PESQ and VISQOL model behaviour, when tested against speech samples modified through playout delay adjustments. The adjustments are typical (in extent and magnitude) to those introduced by VoIP jitter buffer algorithms. Furthermore, the analysis examines the impact of adjustment location as well as speaker factors on MOS scores predicted...
Intercom headsets are mandatory communication apparatus in high noise environments (HNE). The headset selection in HNE, such as combat vehicles, is crucial for achieving the objectives of communication, as it serves the needs for both noise reduction and voice reproduction. Although military-grade intercom headsets are typically used under extreme environmental conditions, a standard performance evaluation...
In all modern communications, speech coding plays a very vital role. It is a basic element in the efficient use of BW and QoS issues. Many standard bodies generate a series of speech coders with different rate, quality and delay combinations. Most of these CODECs have been built for 7 languages not including the Arabic or its accents. In this paper an extensive performance testing is done on the G...
Voice over IP (VoIP) now has tremendous influence on the telecommunication market with its flexibility and price advantage. Users of VoIP expect call quality to be as good as, if not better than the traditional Public Switched Telephone Network (PSTN). However in VoIP, factors that are related to the IP transport network such as packet loss, delay, bandwidth, jitter, and voice encoding (codec) all...
In this paper, we present a novel algorithm to accurately estimate time-variant noises without signal activity detection for acoustic signal enhancement. This is the first algorithm that does not require nor assume noise-only at the beginnings of recordings. This is the first algorithm that can directly estimate noises during the signal activity periods instead of by smoothing noises from neighbouring...
We describe a testing framework that can provide online estimates of audio and video call quality on network paths, without requiring either end-user involvement or prior availability of audio/video sequences or network traces. The framework includes a tool that emulates the audio and video traffic of IP calls and employs an extended E-Model to measure the audio quality and VQM to estimate video quality...
This paper presents a packet loss concealment (PLC) method based on interpolation by separation odd and even frames to improve speech quality deterioration caused by packet losses for CELP based coders. We applied our scheme to the standard ITU-T G.729 speech coder to evaluate the proposed method. The perceptual evaluation of speech quality (PESQ) and enhanced modified bark spectral distortion (EMBSD)...
ITU-T recommendation G.107 introduced the E-model, a repeatable way to assess if a network is prepared to carry a VoIP call or not. Various studies show that the E-model is complex with many factors to be used in monitoring purposes. Consequently, simplified versions of the E-model have been proposed to simplify the calculations and focus on the most important factors required for monitoring the call...
The main goal of this experiment is to evaluate performance of ITU-T P.863 POLQA in case of low-bitrate coders, namely MELPe, and Chinese language. Performance of POLQA is compared to its predecessor ITU-T P.862 PESQ.
This paper deals with analysis of relation between packet delay variations (jitter), length of jitter buffer and final voice transmission quality. For adjusting of IP channel network emulator NISTNet is used. For adjusting of buffer length VoIP client Linphone is used. Criterion of transmission quality is a MOS parameter investigated with an algorithms ITU-T P.862 PESQ and P.863 POLQA.
In order to accomplish speech quality evaluation tasks in an efficient and economical way, objective computational models which simulate human hearing characteristics of speech perception have been intensively researched in the past decades. Several models have already been developed and standardized in telecommunication industry to assess the listening quality of speech signals transmitted through...
This paper presents a packet loss concealment method based on media specific forward error correction (MS-FEC) to improve speech quality deterioration caused by packet losses for CELP based coders. Based on the ITU G.729 CS-ACELP codec operating at 8 Kbps, we applied our metho to evaluate its performance. The enhanced modified bark spectral distortion (EMBSD) and perceptual evaluation of speech quality...
Perceived speech quality, or Quality of Experience (QoE), is the key criteria for evaluating VoIP service. Most of the existing solutions are intrusive, in a sense where they require both the original and the transmitted audio sequences. These solutions give good estimation of the QoE, but they cannot be used in real-time. In fact, Service Provider and Network Provider are highly interested on automatic...
Excitation signal has essential role in speech synthesis filters to produce natural speech. In this study, a new method is proposed for modeling the glottal pulse shape of a speaker. A search is done on the glottal pulse shape space using simulated annealing method. The PESQ measure and Cepstral distance between the original signal and the synthesized signal are used as the cost function. An LPC filter...
In real-time multi-media services, that uses internet infrastructure for transferring data traffics, the quality of service and consequently the level of user satisfaction are significant parameters. Our objective in this paper is to investigate the capability of Bayesian classifiers for estimating the quality of perceived voice in VoIP (Voice over IP) system. In this study, some quality parameters...
This paper deals with analysis of relation between IP channel characteristics and final voice transmission quality. For adjusting of IP channel network emulator NISTNet is used. Criterion of transmission quality is a MOS parameter investigated with an algorithms ITU-T P.862 PESQ, future P.863 POLQA and P.563 3SQM. The following characteristics have been explored: codec PCM, codec Speex, jitter and...
In this paper, we proposed a modified version of K-means clustering algorithm for single channel separation of speech and music from mixed signal. K-means method fails for high dimensional data processing due to computational complexity and curse of dimensionality issues. To improve the performance of clustering algorithm, we used PCA technique and suggested a novel schema to increase the quality...
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