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The algebraic codebook search methods adopted by ITU-T G.729 and its annexes have high computational load and search time. To reduce the code-search time and computational load of Conjugate Structure-Algebraic Code-Excited Linear Prediction (CS-ACELP) coders, we propose a new algebraic codebook structure. In this proposed technique the structure of G.729 algebraic codebook is modified such that it...
The goal of this paper is to compare the performance of two time-frequency decomposition in a context of speech coding. These decompositions are based on wavelet and wavelet frame theory. The main advantages of wavelet frame compared to wavelet are perfect reconstruction, resilience to quantization noise, nearly shift-invariant, symmetry and good time-frequency localization. The evaluation tests reveal...
Prioritization of voice packets is useful to cope with packet losses in VoIP services by providing a mechanism to priority-enabled networks for dropping first the least important content, in order to reconstruct the best quality signal for user perception. A contribution to an efficient solution for voice packet prioritization is proposed in this paper, using an Arduino platform to implement a dynamic...
The major contribution of frames theory, in particular wavelet frames, is to ensure a perfect and stable signals reconstruction. This paper focuses on the speech reconstruction stability, in distortion sense, of the tight framelet packet transform which is new time-frequency decomposition derived from wavelet frame theory. A comparison with classical wavelet transform has been conducted. The experimental...
The quality of speech is essential in communication systems. Enhanced Voice Services (EVS) is one of the speech Coders that standardized by the 3GPP for LTE. EVS supports audio bandwidth, offers high quality for mixed content (i.e. speech and music). Many coders that depend on the concept of ACELP may be affected by the change in the spoken language or accent. EVS Coder working with both ACELP and...
The goal of this paper is to compare the performance of two time-frequency decomposition in a context of speech coding. These decompositions are based on wavelet and wavelet frame theory. The main advantages of wavelet frame compared to wavelet are perfect reconstruction, resilience to quantization noise, nearly shift-invariant, symmetry and good time-frequency localization. The evaluation tests reveal...
This paper provides a detailed comparison study between three different vehicles' Bluetooth built-in noise cancellation filter with two widely used techniques in speech enhancement, Spectral Subtraction (SS) and Wiener filtering (WF). The main purpose is to determine if any of these two filters provide superior audio quality over the built-in filter. In literature, several authors have compared the...
The modelling of the network effects on the quality of speech in the Voice over IP networks is the main focus of this paper. The main purpose of the ideas presented here is to achieve high-precision estimation of the speech quality in the environments where the classical approaches of speech quality determination fail. To achieve this high precision a modular neural network model is used to map the...
The objective of this paper is to implement SPEEX decoding on ARM microprocessor. SPEEX [1] is based on the voice compression algorithm technology of Code Excited Linear Prediction (CELP) [2], which can effectively compress voice and retain the integrity of speech. For hardware part, we give up the high-cost, high-power consumption digital signal processor, and select the STM32 series ARM microprocessor...
The perceived quality of speech captured in the presence of background noise is an important performance metric for communication devices, including portable computers and mobile phones. For a realistic evaluation of speech quality, a device under test (DUT) needs to be exposed to a variety of noise conditions either in real noise environments or via noise recordings, typically delivered over a loudspeaker...
This paper discusses on the enhancement of noisy speech in highly nonstationary noise environment. The concept of Minimum Mean Square Error (MMSE) inspired Data driven noise power (DDNP) estimation and geometrical approach to Spectral Subtraction (GASS) is used in cascade in order to obtain a better Segmental Signal to Noise Ratio (SSNR) and good Perceptual Evaluation of Speech Quality (PESQ) score...
Wireless Communications is used in various applications and the number of wireless devices has also increased over the past several decades. However, wireless communication systems may be significantly affected by interference signal. In this paper, the carrier to noise ratio (CNR) or the bit error rate (BER) performance of Amplitude Modulation (AM), Frequency Modulation (FM), MPSK (M-array Phase...
The paper deals with the modelling of the network effects on the quality of speech in the Voice over IP networks. The main purpose of the ideas presented here is to achieve highprecision estimation of the speech quality in the environment in which the classical approaches of speech quality determination fail. To achieve such a high precision a modular neural network model is used to map the effects...
Underwater speech communication plays an important role in many marine missions, such as oceanographic investigation, environmental monitoring, underwater rescuing and sightseeing. However, the difficulties of underwater acoustic channels such as multipath, time varying and Doppler shifting pose significant challenges to the design of high performance underwater acoustic speech communication. In this...
The real-time measurement of quality of the underwater acoustic voice communications is the important link to safeguard the quality of communication. Using real-time measurements, voice modulation parameters can be adjusted in a timely manner and the adaptive ability of the link can be improved. This paper proposes a kind of objective assessment model of voice quality based on parameter extraction...
In this paper, we present the design of a new packet loss concealment (PLC) algorithm based on Dynamic Forward Error Correction (FEC) to improve speech quality degradation caused by packet losses. It is to be implemented for the standard ITU-T AMR-WD G.722.2 codec. The performance proposed scheme is compared with the original method embedded in the standard ITU-T G.772.2. To evaluate its performance,...
Recently, standard of voice quality on “Voice Call” - one of GSM service, is measured by using standard Quality of Service (QoS). However, according to ministerial of Communication and Information regulation (Permenkominfo) 2008, the guaranteed quality of Voice Call service was Endpoint Service Availability, and not the quality delivered to end-users. In this paper, we proposed a subjective method...
Speech quality evaluation is a very complex task with important applications. With the deployment of 3G and 4G networks, the end-user is requiring more quality of service. Currently, the assessment of speech quality over mobile phones is done in a controlled environment with a reference clean signal and its distorted version. However, this does not evaluate the quality of speech in a real-time situation...
This paper presents comparison of VoIP quality from four VoIP applications, Kakao, Line, Skype and Viber. This study was conducted with Thai and English speech files over a broadband network in one province near Bangkok based on the best efforts before gathering the degraded speech files. Then, those files have been processed using Perceptual Evaluation Speech Quality Technique (PESQ) in order to...
This paper presents a study of VoIP quality from social networking applications over 3G with Fair Usage Policy (FUP) using stationary tests. It has been found that VoIP quality over 3G with FUP from the same VoIP applications tend to provide lower VoIP quality than over 3G with Access Unlimited Internet Service (AUIS), particularly Skype provides lower VoIP quality obviously. Also it has been found...
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