Speech quality prediction for voice over Internet protocol networks

Merged with duplicate record 10026.1/878 on 03.01.2017 by CS (TIS). Merged with duplicate record 10026.1/1657 on 15.03.2017 by CS (TIS)This is a digitised version of a thesis that was deposited in the University Library. If you are the author please contact PEARL Admin ([email protected]) to discuss options.IP networks are on a steep slope of innovation that will make them the long-term carrier of all types of traffic, including voice. However, such networks are not designed to support real-time voice communication because their variable characteristics (e.g. due to delay, delay variation and packet loss) lead to a deterioration in voice quality. A major challenge in such networks is how to measure or predict voice quality accurately and efficiently for QoS monitoring and/or control purposes to ensure that technical and commercial requirements are met. Voice quality can be measured using either subjective or objective methods. Subjective measurement (e.g. MOS) is the benchmark for objective methods, but it is slow, time consuming and expensive. Objective measurement can be intrusive or non-intrusive. Intrusive methods (e.g. ITU PESQ) are more accurate, but normally are unsuitable for monitoring live traffic because of the need for a reference data and to utilise the network. This makes non-intrusive methods(e.g. ITU E-model) more attractive for monitoring voice quality from IP network impairments. However, current non-intrusive methods rely on subjective tests to derive model parameters and as a result are limited and do not meet new and emerging applications. The main goal of the project is to develop novel and efficient models for non-intrusive speech quality prediction to overcome the disadvantages of current subjective-based methods and to demonstrate their usefulness in new and emerging VoIP applications. The main contributions of the thesis are fourfold: (1) a detailed understanding of the relationships between voice quality, IP network impairments (e.g. packet loss, jitter and delay) and relevant parameters associated with speech (e.g. codec type, gender and language) is provided. An understanding of the perceptual effects of these key parameters on voice quality is important as it provides a basis for the development of non-intrusive voice quality prediction models. A fundamental investigation of the impact of the parameters on perceived voice quality was carried out using the latest ITU algorithm for perceptual evaluation of speech quality, PESQ, and by exploiting the ITU E-model to obtain an objective measure of voice quality. (2) a new methodology to predict voice quality non-intrusively was developed. The method exploits the intrusive algorithm, PESQ, and a combined PESQ/E-model structure to provide a perceptually accurate prediction of both listening and conversational voice quality non-intrusively. This avoids time-consuming subjective tests and so removes one of the major obstacles in the development of models for voice quality prediction. The method is generic and as such has wide applicability in multimedia applications. Efficient regression-based models and robust artificial neural network-based learning models were developed for predicting voice quality non-intrusively for VoIP applications. (3) three applications of the new models were investigated: voice quality monitoring/prediction for real Internet VoIP traces, perceived quality driven playout buffer optimization and perceived quality driven QoS control. The neural network and regression models were both used to predict voice quality for real Internet VoIP traces based on international links. A new adaptive playout buffer and a perceptual optimization playout buffer algorithms are presented. A QoS control scheme that combines the strengths of rate-adaptive and priority marking control schemes to provide a superior QoS control in terms of measured perceived voice quality is also provided. (4) a new methodology for Internet-based subjective speech quality measurement which allows rapid assessment of voice quality for VoIP applications is proposed and assessed using both objective and traditional MOS test methods

A configurable vector processor for accelerating speech coding algorithms

The growing demand for voice-over-packer (VoIP) services and multimedia-rich applications has made increasingly important the efficient, real-time implementation of low-bit rates speech coders on embedded VLSI platforms. Such speech coders are designed to substantially reduce the bandwidth requirements thus enabling dense multichannel gateways in small form factor. This however comes at a high computational cost which mandates the use of very high performance embedded processors. This thesis investigates the potential acceleration of two major ITU-T speech coding algorithms, namely G.729A and G.723.1, through their efficient implementation on a configurable extensible vector embedded CPU architecture. New scalar and vector ISAs were introduced which resulted in up to 80% reduction in the dynamic instruction count of both workloads. These instructions were subsequently encapsulated into a parametric, hybrid SISD (scalar processor)–SIMD (vector) processor. This work presents the research and implementation of the vector datapath of this vector coprocessor which is tightly-coupled to a Sparc-V8 compliant CPU, the optimization and simulation methodologies employed and the use of Electronic System Level (ESL) techniques to rapidly design SIMD datapaths

Codec Detection from Speech

Tato práce se zabývá detekcí kodeků z komprimovaného řečového signálu. Cílem bylo zjistit, jaké charakteristiky rozlišují jednotlivé kodeky a následně vytvořit prostředí vhodné pro experimenty s různými typy a konfiguracemi klasifikátorů. Použity byly Support vector machines a především neuronové sítě, které byly vytvořeny pomocí nástroje Keras. Hlavním přínosem této práce je experimentální část, ve které je analyzován vliv různých parametrů neuronové sítě. Po nalezení nejvhodnější kombinace parametrů dosáhla síť přesnosti klasifikace přes 98% na testovací sadě obsahující data z 6 kodeků.This thesis deals with codec detection from compressed speech signal. The primary goal was to identify which features distinguish selected codecs, and then create an environment facilitating experiments with various types of classifiers and their configurations. Support vector machines and neural networks, modeled using the Keras library, were used. The main contribution of this work is the experimental part, in which the effects of the neural networks parameters are discussed. After tuning the parameters and finding their optimal values, the network achieved accuracy over 98% on a test set comprising data from six different codecs.

Parallelism and the software-hardware interface in embedded systems

This thesis by publications addresses issues in the architecture and microarchitecture of next generation, high performance streaming Systems-on-Chip through quantifying the most important forms of parallelism in current and emerging embedded system workloads. The work consists of three major research tracks, relating to data level parallelism, thread level parallelism and the software-hardware interface which together reflect the research interests of the author as they have been formed in the last nine years. Published works confirm that parallelism at the data level is widely accepted as the most important performance leverage for the efficient execution of embedded media and telecom applications and has been exploited via a number of approaches the most efficient being vectorlSIMD architectures. A further, complementary and substantial form of parallelism exists at the thread level but this has not been researched to the same extent in the context of embedded workloads. For the efficient execution of such applications, exploitation of both forms of parallelism is of paramount importance. This calls for a new architectural approach in the software-hardware interface as its rigidity, manifested in all desktop-based and the majority of embedded CPU's, directly affects the performance ofvectorized, threaded codes. The author advocates a holistic, mature approach where parallelism is extracted via automatic means while at the same time, the traditionally rigid hardware-software interface is optimized to match the temporal and spatial behaviour of the embedded workload. This ultimate goal calls for the precise study of these forms of parallelism for a number of applications executing on theoretical models such as instruction set simulators and parallel RAM machines as well as the development of highly parametric microarchitectural frameworks to encapSUlate that functionality.EThOS - Electronic Theses Online ServiceGBUnited Kingdo

Měření Triple play služeb v hybridní síti

The master's thesis deals with a project regarding the implementation, design and the quality of IPTV, VoIP and Data services within the Triple Play services. In heterostructural networks made up of GEPON and xDSL technologies. Different lengths of the optical and metallic paths were used for the measurements. The first part of the thesis is theoretically analyzed the development and trend of optical and metallic networks. The second part deals with the measurement of typical optical and metallic parameters on the constructed experimental network, where its integrity was tested. Another part of the thesis is the evaluation of Triple play results, regarding the test where the network was variously tasked/burdened with data traffic and evaluated according to defined standards. The last part is concerned with the Optiwave Software simulation environment.Diplomová práce se zabývá návrhem, realizací a kvalitou služeb IPTV, VoIP a Data v rámci Triple play služeb v heterostrukturní sítí tvořené GEPON a xDSL technologiemi. Pro měření byli využity různé délky optické a metalické trasy. První části diplomové práce je teoreticky rozebrán vývoj a trend optických a metalických sítí. Druhá část se zaměřuje na měření typických optických a metalických parametrů na vybudované experimentální síti, kde byla následně testována její integrita. Dalším bodem práce je vyhodnocení výsledků Triple play, kde síť je různě zatěžována datovým provozem a následně vyhodnocována podle definovaných norem. Závěr práce je věnovaný simulačnímu prostředí Optiwave.440 - Katedra telekomunikační technikyvýborn

A MODEL FOR PREDICTING THE PERFORMANCE OF IP VIDEOCONFERENCING

With the incorporation of free desktop videoconferencing (DVC) software on the majority of the world's PCs, over the recent years, there has, inevitably, been considerable interest in using DVC over the Internet. The growing popularity of DVC increases the need for multimedia quality assessment. However, the task of predicting the perceived multimedia quality over the Internet Protocol (IP) networks is complicated by the fact that the audio and video streams are susceptible to unique impairments due to the unpredictable nature of IP networks, different types of task scenarios, different levels of complexity, and other related factors. To date, a standard consensus to define the IP media Quality of Service (QoS) has yet to be implemented. The thesis addresses this problem by investigating a new approach to assess the quality of audio, video, and audiovisual overall as perceived in low cost DVC systems. The main aim of the thesis is to investigate current methods used to assess the perceived IP media quality, and then propose a model which will predict the quality of audiovisual experience from prevailing network parameters. This thesis investigates the effects of various traffic conditions, such as, packet loss, jitter, and delay and other factors that may influence end user acceptance, when low cost DVC is used over the Internet. It also investigates the interaction effects between the audio and video media, and the issues involving the lip sychronisation error. The thesis provides the empirical evidence that the subjective mean opinion score (MOS) of the perceived multimedia quality is unaffected by lip synchronisation error in low cost DVC systems. The data-gathering approach that is advocated in this thesis involves both field and laboratory trials to enable the comparisons of results between classroom-based experiments and real-world environments to be made, and to provide actual real-world confirmation of the bench tests. The subjective test method was employed since it has been proven to be more robust and suitable for the research studies, as compared to objective testing techniques. The MOS results, and the number of observations obtained, have enabled a set of criteria to be established that can be used to determine the acceptable QoS for given network conditions and task scenarios. Based upon these comprehensive findings, the final contribution of the thesis is the proposal of a new adaptive architecture method that is intended to enable the performance of IP based DVC of a particular session to be predicted for a given network condition

Comparison of Wideband Earpiece Integrations in Mobile Phone

Perinteisesti puhelinverkoissa välitettävä puhe on ollut kapeakaistaista, kaistan ollessa 300 - 3400 Hz. Voidaan kuitenkin olettaa, että laajakaistaiset puhepalvelut tulevat saamaan markkinoilla enemmän jalansijaa tulevina vuosina. Tässä lopputyössä esitellään puheenkoodauksen perusteet laajakaistaisen adaptiivisen moninopeuspuhekoodekin (AMR-WB) kanssa. Laajakaistainen puhekoodekki laajentaa puhekaistan 50-7000 Hz käyttäen 16 kHz näytetaajuutta. Käytännössä laajempi kaista tarkoittaa parannuksia puheen ymmärrettävyyteen ja tekee siitä luonnollisemman ja mukavamman kuuloista. Tämän lopputyön päätavoite on vertailla kahden eri laajakaistaisen matkapuhelinkuulokkeen integrointia. Kysymys kuuluu, kuinka paljon käyttäjä hyötyy isommasta kuulokkeesta matkapuhelimessa? Kuulokkeiden suorituskyvyn selvittämiseksi niille tehtiin objektiivisia mittauksia vapaakentässä. Mittauksia tehtiin myös puhelimelle pää- ja torsosimulaattorissa (HATS) johdottamalla kuuloke suoraan vahvistimelle, sekä lisäksi puhelun ollessa aktiivisena GSM ja WCDMA verkoissa. Objektiiviset mittaukset osoittivat kahden eri integroinnin väliset erot kuulokkeiden taajuusvasteessa ja särössä erityisesti matalilla taajuuksilla. Lopuksi tehtiin kuuntelukoe tarkoituksena selvittää erottaako loppukäyttäjä pienemmän ja isomman kuulokkeen välistä eroa käyttäen kapeakaistaisia ja laajakaistaisia puhelinääninäytteitä. Kuuntelukokeen tuloksien pohjalta voidaan sanoa, että käyttäjä erottaa kahden eri integroinnin erot ja miespuhuja hyötyy naispuhujaa enemmän isommasta kuulokkeesta laajakaistaisella puhekoodekilla.The speech in telecommunication networks has been traditionally narrowband ranging from 300 Hz to 3400 Hz. It can be expected that wideband speech call services will increase their foothold in the markets during the coming years. In this thesis speech coding basics with adaptive multirate wideband (AMR-WB) are introduced. The wideband codec widens the speech band to new range from 50 Hz to 7000 Hz using 16 kHz sampling frequency. In practice the wider band means improvements to speech intelligibility and makes it more natural and comfortable to listen to. The main focus of this thesis work is to compare two different wideband earpiece integrations. The question is how much the end-user will benefit from using a larger earpiece in a mobile phone? To find out speaker performance, objective measurements in free field were done for the earpiece modules. Measurements were performed also for the phone on head and torso simulator (HATS) by wiring the earpieces directly to a power amplifier and with over the air on GSM and WCDMA networks. The results of objective measurements showed differences between the earpiece integrations especially on low frequencies in frequency response and distortion. Finally the subjective listening test is done for comparison to see if the end-user notices the difference between smaller and larger earpiece integrations using narrowband and wideband speech samples. Based on these subjective test results it can be said that the user can differentiate between two different integrations and that a male speaker benefits more from a larger earpiece than a female speaker

Bandwidth reservation in mobile ad hoc networks for providing QoS : adaptation for voice support

Le support de qualité de service (QoS) dans les réseaux MANETs (Mobile Ad-Hoc NETworks) a attiré une grande attention ces dernières années. Bien que beaucoup de travaux de recherche ont été consacré pour offrir la QoS dans les réseaux filaires et cellulaires, les solutions de QoS pour le support du trafic temps réel dans les MANET reste l'un des domaines de recherche les plus difficiles et les moins explorés. En fait, les applications temps réel telles que la voix et la vidéo ne pourrait pas fonctionner correctement dans les MANET sans l'utilisation d'un protocole de contrôle d'accès au support (MAC) orienté QoS. En effet, les trafics temps réel demandent des exigences strictes en termes de délai de transmission et de taux de perte de paquets qui peuvent être remplies uniquement si la sous-couche MAC fournit un délai d'accès au canal borné, et un faible taux de collision. Le but de cette thèse est la proposition et l'analyse d'un protocole MAC basé sur la réservation pour garantir la QoS dans les MANETs. Tout d'abord, nous étudions un problème majeur dans la réservation de ressources dans les MANETs qui est la cohérence des réservations. Notre analyse des protocoles de réservation existant pour les MANETs révèle que de nombreux conflits de réservations entre les nœuds voisins se produisent pendant la phase d'établissement de réservation. Ces conflits, qui sont principalement dues à la collision des messages de contrôle de réservation, ont un impact important sur les performances du protocole de réservation, et conduisent à un taux de collision et de perte de paquet importants pendant la durée de vie de la connexion, ce qui n'est pas acceptable pour les trafics temps réels. Nous proposons un nouveau protocole MAC basé sur la réservation qui résout ces conflits. Le principe de notre protocole est d'établir une meilleure coordination entre les nœuds voisins afin d'assurer la cohérence des réservations. Ainsi, avant de considérer qu'une réservation est réussite, le protocole s'assure que chaque message de contrôle envoyé par un nœud pour établir une réservation est bien reçu par tous ses nœuds voisins. Dans la deuxième partie de cette thèse, nous appliquons le protocole de réservation proposé au trafic de type voix. Ainsi, nous étendons ce protocole afin de prendre en compte les caractéristiques du trafic voix, tout en permettant le transport de trafic de données. Nous nous focalisons sur l'utilisation efficace de la bande passante et les mécanismes pour réduire le gaspillage de bande passante. La dernière partie de cette thèse concerne l'extension du protocole proposé en vue de réserver la bande passante pour une connexion temps réel sur un chemin. Ainsi, le protocole MAC de réservation proposé est couplé avec un protocole de routage réactif. En outre, le protocole est étendu avec des mécanismes de gestion de à mobilité afin de faire face à la dégradation des performances due à la mobilité des nœuds. Nous évaluons les performances du protocole proposé dans plusieurs scénarios dans lesquels nous montrons sa supériorité par rapport aux standards existants.QoS provisioning over Mobile Ad-Hoc Networks (MANETs) has attracted a great attention in recent years. While much research effort has been devoted to provide QoS over wired and cellular networks, QoS solutions for the support of real-time traffic over MANETs remains one of the most challenging and least explored areas. In fact, real-time applications such as voice and video could not function properly on MANETs without a QoS oriented medium access control (MAC) scheme. Indeed, real-time traffics claim strict requirements in terms of transmission delay and packet dropping that can be fulfilled only if the MAC sub-layer provides bounded channel access delay, and low collision rate. The purpose of this thesis is the proposal and analysis of an efficient reservation MAC protocol to provide QoS support over MANETs. Firstly, we study one major issue in resource reservation for MANETs which is reservation consistency. Our analysis of existing reservation MAC protocols for MANETs reveals that many reservation conflicts between neighbor nodes occur during the reservation establishment phase. These conflicts which are mainly due to collisions of reservation control messages, have an important impact on the performance of the reservation protocol, and lead to a significant collision and loss of packets during the life-time of the connection, which is not acceptable for real-time traffics. We design a new reservation MAC protocol that resolves these conflicts. The main principle of our protocol is to achieve better coordination between neighbor nodes in order to ensure consistency of reservations. Thus, before considering a reservation as successful, the protocol tries to ensure that each reservation control message transmitted by a node is successfully received by all its neighbors. In the second part of this thesis, we apply the proposed reservation protocol to voice traffic. Thus, we extend this protocol in order to take into account the characteristics of voice traffic, while enabling data traffic. We focus on efficient bandwidth utilization and mechanisms to reduce the waste of bandwidth. The last part of this thesis relates to the extension of the proposed protocol in order to reserve resources for a real-time connection along a path. Thus, the proposed reservation MAC protocol is coupled with a reactive routing protocol. In addition, the protocol is extended with mobility handling mechanisms in order to cope with performance degradation due to mobility of nodes. We evaluate the performance of the proposed scheme in several scenarios where we show its superiority compared to existing standards

Quality of media traffic over Lossy internet protocol networks: Measurement and improvement.

Voice over Internet Protocol (VoIP) is an active area of research in the world of communication. The high revenue made by the telecommunication companies is a motivation to develop solutions that transmit voice over other media rather than the traditional, circuit switching network. However, while IP networks can carry data traffic very well due to their besteffort nature, they are not designed to carry real-time applications such as voice. As such several degradations can happen to the speech signal before it reaches its destination. Therefore, it is important for legal, commercial, and technical reasons to measure the quality of VoIP applications accurately and non-intrusively. Several methods were proposed to measure the speech quality: some of these methods are subjective, others are intrusive-based while others are non-intrusive. One of the non-intrusive methods for measuring the speech quality is the E-model standardised by the International Telecommunication Union-Telecommunication Standardisation Sector (ITU-T). Although the E-model is a non-intrusive method for measuring the speech quality, but it depends on the time-consuming, expensive and hard to conduct subjective tests to calibrate its parameters, consequently it is applicable to a limited number of conditions and speech coders. Also, it is less accurate than the intrusive methods such as Perceptual Evaluation of Speech Quality (PESQ) because it does not consider the contents of the received signal. In this thesis an approach to extend the E-model based on PESQ is proposed. Using this method the E-model can be extended to new network conditions and applied to new speech coders without the need for the subjective tests. The modified E-model calibrated using PESQ is compared with the E-model calibrated using i ii subjective tests to prove its effectiveness. During the above extension the relation between quality estimation using the E-model and PESQ is investigated and a correction formula is proposed to correct the deviation in speech quality estimation. Another extension to the E-model to improve its accuracy in comparison with the PESQ looks into the content of the degraded signal and classifies packet loss into either Voiced or Unvoiced based on the received surrounding packets. The accuracy of the proposed method is evaluated by comparing the estimation of the new method that takes packet class into consideration with the measurement provided by PESQ as a more accurate, intrusive method for measuring the speech quality. The above two extensions for quality estimation of the E-model are combined to offer a method for estimating the quality of VoIP applications accurately, nonintrusively without the need for the time-consuming, expensive, and hard to conduct subjective tests. Finally, the applicability of the E-model or the modified E-model in measuring the quality of services in Service Oriented Computing (SOC) is illustrated