Browsing by Author "Samy El-Hennawey, Mohamed"

Now showing 1 - 7 of 7

Measurement of the effects of temporal clipping on speech quality
(IEEE, 2006) Ding, Lijing; Radwan, Ayman; Samy El-Hennawey, Mohamed; A Goubran, Rafik
This paper investigates the effects of temporal clipping on perceived speech quality. Temporal clipping usually results from voice activity detection (VAD), or line echo canceller's nonlinear processor, and the clipped speech portions are replaced by comfort noise. A nonintrusive algorithm is proposed to predict speech quality based on the clipping statistics. Mean opinion score (MOS) is used as a metric for speech quality and is measured by perceptual evaluation of speech quality (PESQ). The impacts of speech frame size and noise spectrum on the algorithm are also investigated. The results show that the proposed algorithm can efficiently predict the speech quality. The correlation coefficient between the prediction and the measurement is about 0.975, and the root mean square error for the prediction is 0.20 MOS. The algorithm can be used as an integral part of a general speech quality assessment scheme in voice over Internet protocol (VoIP)
Non-intrusive single-ended speech quality assessment in VoIP
(Elsevier, 2007) Ding, Lijing; Lin, Zhong; Radwan, Ayman; Samy El-Hennawey, Mohamed; A Goubran, Rafik
Evaluating speech quality in voice over Internet protocol (VoIP) in a non-intrusive manner is challenging, because it relies on a degraded speech signal only. In this paper, a parametric, non-intrusive VoIP speech quality assessment algorithm is proposed, which adopts a three-step strategy, impairment detection, individual effect modeling and an overall model. Mainly based on voice payload analysis, the algorithm also combines Internet protocol analysis approach and the ITU-T E-model. It quantifies the individual contributions to speech quality from several major VoIP impairments, including packet loss, temporal clipping and noise. Also, an overall assessment model is developed. The performance is evaluated through intensive simulations, and the results show that the algorithm is effective and accurate. For the overall model, the correlation between prediction and measurement is 0.90; the root mean square error (RMSE) is 0.27 mean opinion score (MOS). The algorithm aims to be implemented at the receive-end media gateway or IP terminal, for identifying the root causes of speech quality degradation as well as quality assessment in VoIP.
Nonintrusive Measurement of Echo-Path Parameters in VoIP Environments
(IEEE, 2006) Ding, Lijing; Samy El-Hennawey, Mohamed; A Goubran, Rafik
This paper proposes two echo-path delay measurement methods suitable for voice-over-Internet-protocol environments, where the echo suffers from excessive delay and nonlinear distortion. The proposed methods aim at greatly reducing the computational requirements while maintaining good measurement accuracy. The delay measurement is based on the cross correlation; the computation reduction is achieved by using either downsampled speeches or sparse speeches for the two methods, respectively. The echo-path loss is also measured by using the obtained delay information. The performance under codec distortion, packet loss, noise, and double talk conditions is examined through simulations and real field measurements. The results show that the proposed methods are effective and accurate
Performance of the /spl pi//4-DQPSK, GMSK, and QAM modulation schemes in mobile radio with multipath fading and nonlinearities
(IEEE, 2017) Hashem, Bassam; Samy El-Hennawey, Mohamed
As mobile communications have become so indispensable, every possible effort should be spent to achieve the optimum operating conditions. In addition to the normal impairments associated with wireless communications, in general, the mobile channel suffers from particular limitations that confine the performance of a mobile radio system. Among those impairments are the bandwidth limitation, interference, and multipath fading. With the strong motivation toward portable radio and personal communication systems, power limitation has manifest itself in the picture, and, consequently, nonlinear operation of the amplifiers involved (hence, the channel) will have to be dealt with. Constant envelope modulation schemes have been used in digital mobile radio systems recently installed. The Gaussian minimum shift keying (GMSK) is employed in the Global System for Mobile (GSM) communications installed in Europe, while in the North American IS-54 system, the modulation scheme used is the /spl pi//4-DQPSK. As the quest for higher data rates has kept on increasing, multilevel modulation schemes have been proposed with their performance over nonlinear channels having been overlooked. The paper provides a comparative study, based on simulation, and tests the performance of various modulation schemes operating over a wide variety of mobile radio channel conditions. The effective throughput of all systems is also considered
Performance Study of Objective Voice Quality Measures in VoIP
(IEEE, 2007) Ding, Lijing; Radwan, Ayman; Samy El-Hennawey, Mohamed; A Goubran, Rafik
With the advent of voice over internet protocol (VoIP) service, assessing its voice quality is an area of intense research interest. Due to time-consuming and expensive natures of widely accepted subjective mean opinion score (MOS) test, objective methods are often used as an alternative. This paper presents applicability and accuracy analysis of several leading objective methods in quality testing. Particularly, the paper focuses on packet loss, which is one of major impairments in VoIP. A speech database covering typical packet loss conditions is designed. The subjective MOS test is conducted and the results are evaluated with objective MOS. The database is also used to verify a non-intrusive VoIP speech quality assessor the authors developed (El-Hennawey et al., 2006). The results show that perceptual evaluation of speech quality (PESQ) based algorithms are generally acceptable for quantifying the effects of packet loss. In addition, we also find that their performance is limited for codec G.711 without packet loss concealment (PLC).
Transcoders and mixers for voice-over-IP conferencing
(Carleton University; RPX Clearinghouse LLC, 2009) Samy El-Hennawey, Mohamed; A. Goubran, Rafik; Qian, Zhihong
Transcoders and mixers having reduced algorithmic delay and processing complexity. An improved mixer for signals having encoded speech parameters wherein the parameters obtained through decoding are used by a parameter estimator to improve the encoding by providing a parameter estimate for the mixed signal. In the case of pitch parameters, the mixer uses the principle of strong-pitch-domination. The mixing of wideband signals is simplified by performing mixing of individual lower and upper sub-bands. A transcoder and a mixer that converts a wideband signal into a narrowband signal relies upon high frequency suppression. A transcoder and a mixer that converts a narrowband signal into a wideband signal relies upon filter combination.
Voice call quality using 802.11e on a wireless mesh network
(IEEE, 2009) van Geyn, David; Hassanein, Hossam; Samy El-Hennawey, Mohamed
Wireless local area networks (WLANs) provide an affordable solution for last mile network access. They also allow for extension of a network by configuring a wireless mesh network (WMN) where it may otherwise be physically infeasible or cost prohibitive to do so. With the increasing use of real-time applications such as video conferencing and Voice over IP (VoIP), networks are stressed to guarantee QoS requirements for these applications. Examples of key requirements include bounded delay and packet loss ratios. Addressing this issue in WLANs, the IEEE 802.11e amendment was proposed to provide a QoS mechanism. However, the performance of 802.11e in meshed environments is yet to be studied. In this work, we study VoIP call quality in a meshed environment with provisions for QoS. We study the call quality and throughput of background traffic in an experimental WMN testbed in order to test how well the IEEE 802.11e QoS provisions support voice calls. Call quality is tested in different configurations and scenarios. We study the effect of the number of wireless hops on VoIP call quality. In addition, we investigate the number of VoIP calls that can be supported simultaneously for different numbers of wireless hops. We also study how fairly the network treats different calls in different configurations. Then, we look at how much effective bandwidth a VoIP call uses on the network. Finally, we examine the VoIP call quality of different calls when calls have different QoS parameters and study the effect that a busy central node has on traffic passing through it. We provide suggestions to improve call quality on a WMN and hint at possible future work.