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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...
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.
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...
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...
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 focuses on problems of the end to end voice quality assessment. Up to now there have been developed lot of measuring methods, which objectively determine final quality of transmission. This paper deals with the PESQ (Perceptual Evaluation of Speech) algorithm described in the ITU-T P.862 and its use. The basic intent of this work was to develop a device which is able to measure voice transmission...
New methods of hiding data in the pitch delay data of the adaptive multi-rate (AMR) speech codec are proposed. The proposed methods and the pitch quantization (PQ) method are compared in terms of embedding bit rate and objective speech quality obtained by the perceptual evaluation for speech quality (PESQ) algorithm. PQ, the previous method, quantizes the pitch delay data by embedding data. The proposed...
ITU-T P.862 - ldquoPerceptual Evaluation of Speech Quality (PESQ)rdquo is well known as an intrusive objective speech quality assessment method. Some reports have found that the PESQ time alignment mechanism fails to estimate delay where signals with high packet loss rate and dynamic time processing are present. A new time-alignment algorithm to improve the PESQ accuracy for time-scale modified voice...
As the service requirements of network applications shift from high throughput to high media quality, interactivity, and responsiveness, the definition of QoE (Quality of Experience) has become multidimensional. Although it may not be difficult to measure individual dimensions of the QoE, how to capture users' overall perceptions when they are using network applications remains an open question. In...
Voice over Internet protocol (VoIP) is a technology that transports voice data packets across packet switched networks using the Internet protocol (IP). However, the current Internet, which was originally designed for data communications, provides best-effort service only, and does not guarantee to transmission quality. As known, packet delay dramatically degrades the quality of VoIP calls. It is...
Error resilient coding is strongly demanded by voice communication over unreliable packet networks. In this paper, we propose a new multiple description scheme named distributed subframe interleaving, and employ this new scheme to create two novel multiple description codecs based on CELP. With introducing distributed subframe interleaving operated on two adjacent speech packets, parameters of LSP...
PESQ is the international standard for objective speech quality measurement and has been widely used in industry for voice quality assessment for VoIP products and systems. However, current research efforts involving PESQ are mainly focused on simulated IP networks. The need remains to establish how accurate PESQ is when used in real wireless systems, especially in new wireless VoIP environment. The...
A key feature of wireless mesh networks is that multiple independent paths through the network are available. Multiple descriptions coding is often suggested as a source coding scheme to take advantage of this path diversity. We compare multiple description (MD) coding with path diversity (PD) against a full-rate single description (SD) coder without PD, and two simple PD methods of 1) repeating a...
Delay, jitter and packet loss significantly degrade the speech quality in VoIP applications, especially in VoWLAN. In order to solve this problem, different jitter buffer and error concealment techniques are introduced. In this paper, we propose a new simple but effective error concealment method for VoIP, which uses the contour tendency of the last two packets to predict and recover the current packet...
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