Novel Pitch Detection Algorithm With Application to Speech Coding

This thesis introduces a novel method for accurate pitch detection and speech segmentation, named Multi-feature, Autocorrelation (ACR) and Wavelet Technique (MAWT). MAWT uses feature extraction, and ACR applied on Linear Predictive Coding (LPC) residuals, with a wavelet-based refinement step. MAWT opens the way for a unique approach to modeling: although speech is divided into segments, the success of voicing decisions is not crucial. Experiments demonstrate the superiority of MAWT in pitch period detection accuracy over existing methods, and illustrate its advantages for speech segmentation. These advantages are more pronounced for gain-varying and transitional speech, and under noisy conditions

The voice activity detection (VAD) recorder and VAD network recorder : a thesis presented in partial fulfilment of the requirements for the degree of Master of Science in Computer Science at Massey University

The project is to provide a feasibility study for the AudioGraph tool, focusing on two application areas: the VAD (voice activity detector) recorder and the VAD network recorder. The first one achieves a low bit-rate speech recording on the fly, using a GSM compression coder with a simple VAD algorithm; and the second one provides two-way speech over IP, fulfilling echo cancellation with a simplex channel. The latter is required for implementing a synchronous AudioGraph. In the first chapter we introduce the background of this project, specifically, the VoIP technology, the AudioGraph tool, and the VAD algorithms. We also discuss the problems set for this project. The second chapter presents all the relevant techniques in detail, including sound representation, speech-coding schemes, sound file formats, PowerPlant and Macintosh programming issues, and the simple VAD algorithm we have developed. The third chapter discusses the implementation issues, including the systems' objective, architecture, the problems encountered and solutions used. The fourth chapter illustrates the results of the two applications. The user documentations for the applications are given, and after that, we analyse the parameters based on the results. We also present the default settings of the parameters, which could be used in the AudioGraph system. The last chapter provides conclusions and future work

Improved compactly computable objective measures for predicting the acceptiability of speech communications systems

Issued as Monthly status reports [1-7], and Final report, Project no. E-21-61

Secure mobile radio communication over narrowband RF channel.

by Wong Chun Kau, Jolly.Thesis (M.Phil.)--Chinese University of Hong Kong, 1992.Includes bibliographical references (leaves 84-88).ABSTRACT --- p.1ACKNOWLEDGEMENT --- p.3Chapter 1. --- INTRODUCTION --- p.7Chapter 1.1 --- Land Mobile Radio (LMR) CommunicationsChapter 1.2 --- Paramilitary Communications SecurityChapter 1.3 --- Voice Scrambling MethodsChapter 1.4 --- Digital Voice EncryptionChapter 1.5 --- Digital Secure LMRChapter 2. --- DESIGN GOALS --- p.20Chapter 2.1 --- System Concept and ConfigurationChapter 2.2 --- Operational RequirementsChapter 2.2.1 --- Operating conditionsChapter 2.2.2 --- Intelligibility and speech qualityChapter 2.2.3 --- Field coverage and transmission delayChapter 2.2.4 --- Reliability and maintenanceChapter 2.3 --- Functional RequirementsChapter 2.3.1 --- Major system featuresChapter 2.3.2 --- Cryptographic featuresChapter 2.3.3 --- Phone patch facilityChapter 2.3.4 --- Mobile data capabilityChapter 2.4 --- Bandwidth RequirementsChapter 2.5 --- Bit Error Rate RequirementsChapter 3. --- VOICE CODERS --- p.38Chapter 3.1 --- Digital Speech Coding MethodsChapter 3.1.1 --- Waveform codingChapter 3.1.2 --- Linear predictive codingChapter 3.1.3 --- Sub-band codingChapter 3.1.4 --- VocodersChapter 3.2 --- Performance EvaluationChapter 4. --- CRYPTOGRAPHIC CONCERNS --- p.52Chapter 4.1 --- Basic Concepts and CryptoanalysisChapter 4.2 --- Digital Encryption TechniquesChapter 4.3 --- Crypto SynchronizationChapter 4.3.1 --- Auto synchronizationChapter 4.3.2 --- Initial synchronizationChapter 4.3.3 --- Continuous synchronizationChapter 4.3.4 --- Hybrid synchronizationChapter 5. --- DIGITAL MODULATION --- p.63Chapter 5.1 --- Narrowband Channel RequirementsChapter 5.2 --- Narrowband Digital FMChapter 5.3 --- Performance EvaluationChapter 6. --- SYSTEM IMPLEMENTATION --- p.71Chapter 6.1 --- Potential EMC ProblemsChapter 6.2 --- Frequency PlanningChapter 6.3 --- Key ManagementChapter 6.4 --- Potential Electromagnetic Compatibility (EMC) ProblemsChapter 7. --- CONCLUSION --- p.80LIST OF ILLUSTRATIONS --- p.81REFERENCES --- p.82APPENDICES --- p.89Chapter I. --- Path Propagation Loss(L) Vs Distance (d)Chapter II. --- Speech Quality Assessment Tests performedby Special Duties Unit (SDU

Postfiltering techniques in low bit-rate speech coders

Thesis (M.Eng.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 1999.Includes bibliographical references (leaves 78-80).by Azhar K. Mustapha.M.Eng

MSAT-X: A technical introduction and status report

A technical introduction and status report for the Mobile Satellite Experiment (MSAT-X) program is presented. The concepts of a Mobile Satellite System (MSS) and its unique challenges are introduced. MSAT-X's role and objectives are delineated with focus on its achievements. An outline of MSS design philosophy is followed by a presentation and analysis of the MSAT-X results, which are cast in a broader context of an MSS. The current phase of MSAT-X has focused notably on the ground segment of MSS. The accomplishments in the four critical technology areas of vehicle antennas, modem and mobile terminal design, speech coding, and networking are presented. A concise evolutionary trace is incorporated in each area to elucidate the rationale leading to the current design choices. The findings in the area of propagation channel modeling are also summarized and their impact on system design discussed. To facilitate the assessment of the MSAT-X results, technology and subsystem recommendations are also included and integrated with a quantitative first-generation MSS design

Quality aspects of Internet telephony

Internet telephony has had a tremendous impact on how people communicate. Many now maintain contact using some form of Internet telephony. Therefore the motivation for this work has been to address the quality aspects of real-world Internet telephony for both fixed and wireless telecommunication. The focus has been on the quality aspects of voice communication, since poor quality leads often to user dissatisfaction. The scope of the work has been broad in order to address the main factors within IP-based voice communication. The first four chapters of this dissertation constitute the background material. The first chapter outlines where Internet telephony is deployed today. It also motivates the topics and techniques used in this research. The second chapter provides the background on Internet telephony including signalling, speech coding and voice Internetworking. The third chapter focuses solely on quality measures for packetised voice systems and finally the fourth chapter is devoted to the history of voice research. The appendix of this dissertation constitutes the research contributions. It includes an examination of the access network, focusing on how calls are multiplexed in wired and wireless systems. Subsequently in the wireless case, we consider how to handover calls from 802.11 networks to the cellular infrastructure. We then consider the Internet backbone where most of our work is devoted to measurements specifically for Internet telephony. The applications of these measurements have been estimating telephony arrival processes, measuring call quality, and quantifying the trend in Internet telephony quality over several years. We also consider the end systems, since they are responsible for reconstructing a voice stream given loss and delay constraints. Finally we estimate voice quality using the ITU proposal PESQ and the packet loss process. The main contribution of this work is a systematic examination of Internet telephony. We describe several methods to enable adaptable solutions for maintaining consistent voice quality. We have also found that relatively small technical changes can lead to substantial user quality improvements. A second contribution of this work is a suite of software tools designed to ascertain voice quality in IP networks. Some of these tools are in use within commercial systems today

Speech, time-frequency representations

This paper presents a review on the use of time frequency representations in the fields of speech analysis and automatic speech processing . Three main groups of methods are considered : speech production based methods, general signal analysis methods, auditory-based methods . After this review, some short conclusions on their carrent use, and on some possible future evolutions are proposed .Le propos de cet article est de présenter une bibliographie récente sur l'utilisation des méthodes de représentation temps-fréquence en analyse et en traitement automatique de la parole. Les méthodes sont classées en trois grandes familles: méthodes dérivées de la production, méthodes d'analyse du signal, méthodes modélisant la perception. Après ce panorama, quelques rapides conclusions sur l'état actuel de l'utilisation de ces méthodes, et quelques perspectives sont tentée