Marcel Wältermann

Speech Quality Estimation: Models and Trends

This article presents a tutorial overview of models for estimating the quality experienced by users of speech transmission and communication services. Such models can be classified as either parametric or signal based. Signal-based models use input speech signals measured at the electrical or acoustic interfaces of the transmission channel. Parametric models, on the other hand, depend on signal and system parameters estimated during network planning or at run time. This tutorial describes the underlying principles as well as advantages and limitations of existing models. It also presents new developments, thus serving as a guide to an appropriate usage of the multitude of current and emerging speech quality models.

Impairment Factor Framework for Wide-Band Speech Codecs

A new method is described for quantifying the quality degradation introduced by wide-band speech codecs via a one-dimensional impairment factor. The method is based on auditory listening-only tests, but the resulting impairment factors may be used for predicting speech quality in an instrumental way, e.g., for network planning purposes. Following the method, auditory test results are first transformed to an overall quality rating scale, and then adjusted to rule out test-specific effects. The derived impairment factors fit into the common framework which is defined by the E-model for narrow-band telephone networks, and which is hereby extended towards wide-band speech transmission. This paper presents the necessary auditory test data, describes the derivation and adjustment methodology, and provides numerical values for a range of wide-band speech codecs. The values are tested for their robustness in case of codec tandems and adjusted to represent the effects of packet loss

Video quality in next generation mobile networks — Perception of time-varying transmission

Next Generation Mobile Networks (NGMNs) provide an all-IP wireless platform for multimedia service delivery. The integrated communication system creates new perspectives for wireless video distribution and quality provisioning. Depending on the mobility patterns of the nomadic users, the link layer characteristics may rapidly change. Furthermore, mobility and service adaptation events like network handovers, video bit rate switching, or codec changeovers, may affect the user perception. Provisioning “always best connected” video services requires a thorough knowledge of video quality perception in NGMNs. In this paper, we address this problem. A subjective test including 50 NGMN conditions has been carried out. They were arranged in groups to assess the impact on perceived quality of 1) the access technology and network handover, 2) video codecs and codec changeover, 3) video bit rate and bit rate switching, and 4) to provide guidelines for packet loss adaptation. We could identify the perceptual bottlenecks of the future wireless communication and are able to propose perceptual guidelines for mobility management. In this way, this paper contributes to the service quality improvement of future wireless communications.

Speech Quality While Roaming in Next Generation Networks

In NGNs, handovers between different wireless access technologies provide seamless roaming during voice calls. The resulting speech quality depends on the audio bandwidth of the speech codecs used in the respective networks, as well as on degradations resulting from the handover, coding, and packet loss. We present the results of four listening experiments where speech quality is quantified as a function of network and codec characteristics, and compare them to estimations obtained from instrumental models. The results show when and under which circumstances a network handover and/or codec changeover should be scheduled in order to obtain better speech quality. This is important for the development of high-quality roaming strategies.

A Technique for Seamless VoIP-Codec Switching in Next Generation Networks

Vertical handovers in Next Generation Networks enable a new experience of mobility, since application layer sessions are maintained while roaming across different access networks. For real-time media services like Voice-over-IP, a change in the underlying network technology is particulary challenging. Ongoing calls are not suspended during the handover, however, the handover may go along with an audible gap during the transition time due to lost or delayed packets and an adaptation in call parameters such as those of the employed speech codec. This, in turn, may translate in an unfamiliar speech quality perception. This contribution introduces an ergonomic technique that aims at seamlessly switching the speech codec in Voice-over-IP calls during vertical handovers, based on SIP/SDP session renegotiation, the establishment of a parallel media stream and RTP packet filtering. Evaluation results are presented, showing that the proposed approach does not cause any interruption of the audio stream in about 90% of the test cases, clearly outperforming simple re-negotiation of session parameters that does not take a seamless transition into account (interruptions in all test cases). PESQ speech quality estimates reveal a quality advantage of 5.4% on average for the considered scenario.

Extension of the E-model towards super-wideband speech transmission

In this paper, the quality gain of super-wideband (SWB) speech, transmitted in the much wider frequency range of 50-14000 Hz compared to the standard 300-3400 Hz narrowband, is quantified employing the E-model framework, a parametric tool for speech quality prediction. Based on two listening experiments, a linear extrapolation of the E-model transmission rating scale was found that leads to a maximum quality advantage of 39% relative to wideband (50-7000 Hz) transmission, and 79% relative to narrowband. Furthermore, narrowband, wideband, and super-wideband conditions can be quantified on this universal quality scale. Equipment Impairment Factors were derived and discussed for several SWB codecs. It will further be shown that a model quantifying the quality impact of linear distortions, reflected by the Bandwidth Impairment Factor, can successfully applied to SWB conditions. The correlation between the overall impairment and the model predictions amounts to r = 0.977 for linearly distorted speech samples.

E-model supported switching between narrowband and wideband speech quality

Forthcoming Next Generation Networks (NGNs) introduce a new trend of integration into one All-IP platform. This process will decouple VoIP services from the network providers, introducing new service distribution aspects. VoIP devices will be enabled to seamlessly switch between wideband and narrowband speech operators and roam across diverse networks on demand, even during active calls. To design user-friendly speech quality switching posed by the NGNs, a reliable non-intrusive quality prediction model for quality monitoring will be required. Until now, none of the existing quality prediction models has been fully validated for use in NGNs. This prohibits quality prediction, when switching between wideband and narrowband speech codecs is necessary. In this paper, we validate the applicability of the E-Model for NGNs, targeting the situations when the speech bandwidth must be changed. We propose an additive codec switching impairment factor Iesw that improves the E-Model estimation for the new quality phenomenon. In this way, the achieved E-Model correlation r = 0.93 enables VoIP quality monitoring of dynamic VoIP quality profiles, which is required to provide maximal user satisfaction in NGNs.

Parameter-based prediction of speech quality in listening context—Towards a WB E-model

The paper presents a first version of a wideband E-model applicable to planning future telephone networks in a listening-only context. It allows to predict wideband speech quality under bandwidth restrictions and wideband and narrowband speech coding with and without transmission errors, including the quality impact due to noise. The paper presents the different model components and discusses their prediction performance for a large set of listening tests. Based on a discussion of the proposed listening model, an outlook on required future research including the conversational situation is provided.

A new dimension-based framework model for the quality of speech communication services

In this paper, we identify quality dimensions which are relevant for speech communication services, such as mobile telephony or Voice-over-IP. These include dimensions perceived when listening to degraded speech, talking against echoes, double-talk capabilities, interacting with delay, conversing over channels with time-varying characteristics, and service-related dimensions experienced during speech connection set-up and maintenance. For each dimension, we review subjective evaluation metrics and instrumental quality prediction models. We group these dimensions in a framework model which is able to diagnostically assess speech communication services, and may be used for monitoring and maintenance.

Towards a universal scale for perceptual value

This paper addresses the problem of context effects in psychophysical measurement, and how these effects are reflected by MOS values obtained in auditory tests. Based on a simple model of the perception and judgment processes which are relevant in a subjective quality-judgment situation, it advocates for using a universal scale for the value which is related to the perceptual event, and which has a meaning beyond the specific test context. The E-models transmission rating scale [1] is considered to be a candidate for such a scale.