One use of text-to-speech synthesis (TTS) is as a component of speech-to-speech translation systems. The output of automatic machine translation (MT) can vary widely in quality, however. A synthetic voice that is extremely intelligible on naturally-occurring text may be far less intelligible when asked to render text that is automatically generated. In this paper, we compare the quality of synthesis of naturally-occurring text and its MT counterpart. We find that intelligibility of TTS on MT output is significantly lower than on either naturally-occurring text or semantically unpredictable sentences, and explore the reasons why.

This paper addresses the generation and evaluation of foreign-accented speech in concatenative text-to-speech (TTS) synthesis. We describe three possible methods of building a Spanish-accented English voice, and evaluate and compare them with respect to preference, intelligibility, and smoothness. Effects of speaking rate and content are also examined. It is found that although using an unmodified Spanish voice to read English text is possible, the result is not highly intelligible. With some modifications to the linguistic model, a relatively high level of comprehensibility and smoothness can be achieved, not differing widely from ratings given to a native voice at a comparable stage of development. Listeners in perceptual experiments were very consistent in their preference rankings of the three voices, showing that differences in voicebuilding method are both detectable and contribute to synthesis quality. 1.

The paper presents a framework for building multilingual text-to-speech systems. It addresses the issue at three levels. First it discusses the necessary steps required to build a synthetic voice from scratch in a new language. The second concerns the building of a new voice without recording any new acoustic data, and the restrictions that imposes. The third more speculative part discusses the steps that would be necessary to allow high quality synthesis of new languages by recording only minimal amounts in that language.

... of synthesis of Arabic, a language that has shot to prominence in the past few years, and synthesis on a handheld device, realization of which presents difficult software engineering problems. Our system was developed in conjunction with the DARPA BABYLON project, and has been integrated with English synthesis, English and Arabic ASR, and machine translation on a single off-the-shelf PDA. We present

This paper describes a working two-way speech-to-speech translation system that runs in near real-time on a consumer handheld computer. It can translate from English to Arabic and Arabic to English in the domain of medical interviews. We describe the general architecture and frameworks within which we developed each of the components: HMM-based recognition, interlingua translation (both rule and statistically based), and unit selection synthesis.

Apple's next-generation text-to-speech (TTS) system in MacOS X uses a superpositional pitch model, comprising a relatively smooth underlying F0 contour and a separate contribution from the in uence of the phonetic segments. This paper focuses on the data-driven modelling of the underlying contour, based on electroglottographic signals obtained from a corpus of reiterant speech. F0 extraction from such signals leads to more accurate characteristic shapes, as objectively illustrated by a typically low mean absolute frequency deviation (between 2 and 3 Hz) between original and synthetic F0 contours. This in turn supports a better (both more complete and more realistic) model of F0 behavior. Experimental results illustrate the improved prosodic representation resulting from this F0 model.

The increasing availability of carefully designed and collected speech corpora opens up new possibilities for the statistical estimation of formal multivariate prosodic models. At Apple Computer, statistical prosodic modeling exploits the Victoria corpus, created to broadly support ongoing speech synthesis research and development. This corpus is composed of five constituent parts, each designed to cover a specific aspect of speech synthesis: polyphones, prosodic contexts, reiterant speech, function word sequences, and continuous speech. This paper focuses on the use of the Victoria corpus in the statistical estimation of duration and pitch models for Apple's next-generation text-to-speech system in Macintosh OS X. Duration modeling relies primarily on the subcorpus of prosodic contexts, which is instrumental to uncover empirical evidence in favor of a piecewise linear transformation in the well-known sums-of-products approach. Pitch modeling relies primarily on the subcorpus of reiterant speech, which makes possible the optimization of superpositional pitch models with more accurate underlying smooth contours. Experimental results illustrate the improved prosodic representation resulting from these new duration and pitch models.

In this paper, we describe the design and collection of corpora for diphone synthesis, the voice building process, and our experience in the creation of a new, publically available database of ten diphone sets of one American English speaker for the Festival Speech Synthesis System [3], using the FestVox document and tools [1]. In support of our goal to make the tools and techniques available for anyone to build their own synthetic voices, we have generalized and streamlined the tasks involved from what were once arcane anecdotes, half-written one-off scripts, and partial descriptions, to detailed, complete instructions that others have followed with good results. 1. INTRODUCTION The FestVox [1] document is a growing, publically available resource that contains tools, data, and text about building complete synthetic voices in English and other languages. That work covers everything from building text analyzers, lexicons, prosodic models as well as various waveform synthesis technique...

We discuss our ongoing attempt to design and evaluate universal human-machine speech-based interfaces. We describe one such initial design suitable for database retrieval applications, and discuss its implementation in a movie information application prototype. Initial user studies provided encouraging results regarding the usability of the design, as well as suggest some questions for further investigation. 1. INTRODUCTION Speech recognition technology has made spoken interaction with machines feasible. However, no suitable universal interaction paradigm has yet been proposed for humans to communicate effectively, efficiently and effortlessly by voice with machines. On one hand, natural language applications have been demonstrated in narrow domains, but building such systems is data-, labor- and expertise-intensive. Perhaps more importantly, unconstrained natural language severely strains recognition technology, and fails to delineate the functional limitations of the machine. On th...

The Carnegie Mellon Communicator is a telephone-based dialog system that supports planning in a travel domain. The implementation of such a system requires two complimentary components, an architecture capable of managing interaction and the task, as well as a knowledge base that captures the speech, language and task characteristics specific to the domain. Given a suitable architecture, the principal effort in development in taken up in the acquisition and processing of a domain knowledge base. This paper describes a variety of techniques we have applied to modeling in acoustic, language, task, generation and synthesis components of the system. 1. INTRODUCTION System development involves a great deal of knowledge engineering, which is both time-consuming and requires a variety of experts to participate in the process. Therefore methods that seek to minimize this resource, for example through training based on domain-specific corpora are preferred. Effective use of corpora, however, ...

Some Single-Microphone Speech Enhancement algorithms based on the iterative Wiener filtering Method due to Lim-Oppenheim [2] are evaluated. In the original Lim-Oppenheim algorithm, AR spectral estimation of speech is carried out using a second-order analysis, but our algorithms consider an AR estimation from a cumulant analysis. This work extends some preceding papers due to the authors [4], [5]. Third- and fourth-order cumulant-based algorithms are compared to classical second-order one. This comparison is evaluated by considering three different noisy environments. A detailed study based on a frame-by-frame analysis leads to an optimum iteration of each algorithm as a trade-off between noise reduction and distortion effects. Voiced and unvoiced sounds are separately discussed. We conclude that third-order cumulant-based algorithm offers a more valuable performance than the others.

The DIPLOMAT project at Carnegie Mellon University instantiates a program of rapid-deployment speech-to-speech machine translation; we have developed techniques for quickly producing text-to-speech (TTS) systems for new target languages to support this work. While the resulting systems are not immediately of comparable quality to commercial systems on unrestricted tasks in well-developed languages, they are more than adequate for limited-domain scenarios and rapid prototyping -- they generalize to unseen data with some degradation, while quality in-domain can be quite good. Voices and engines for synthesizing new target languages may be developed in a period as short as two weeks after text corpus collection. We have successfully used these techniques to build a TTS module for English, Croatian, Spanish, Haitian Creole and Korean. 1. The DIPLOMAT Project The DIPLOMAT project is an experiment in rapid-deployment, wearable, bi-directional speech translation systems. An implementation of the complete synthesis system should be available at an acceptable level of quality within a few weeks after initial recording corpus design, with continual, graceful improvement to a good level of quality over a period of months. For a small domain, the concatenative synthesis method used here can be quite adequate; however, the current synthesis solution, called phonebox, is a rapid prototyping tool rather than a substitute for a full-fledged speech synthesis system in the target language. The synthesizer receives text-only input through a socket or pipe, and can easily be replaced by an arbitrary synthesis engine and a small wrapper. We must often trade off the quality synthesis in the short term in favor of development time, memory limitations, and runtime efficiency, in order to provide basic communication technologies in contexts where communication barriers are significant, and to do so on commonly available hardware. So far, DIPLOMAT has worked with English, Croatian, Haitian Creole, Spanish and Korean.

This paper presents trainable methods for generating letter to sound rules from a given lexicon for use in pronouncing out-of-vocabulary words and as a method for lexicon compression. As the relationship between a string of letters and a string of phonemes representing its pronunciation for many languages is not trivial, we discuss two alignment procedures, one fully automatic and one hand-seeded which produce reasonable alignments of letters to phones. Top Down Induction Tree models are trained on the aligned entries. We show how combined phoneme/stress prediction is better than separate prediction processes, and still better when including in the model the last phonemes transcribed and part of speech information. For the lexicons we have tested, our models have a word accuracy (including stress) of 78% for OALD, 62% for CMU and 94% for BRULEX. The extremely high scores on the training sets allow substantial size reductions (more than 1/20). WWW site: http://tcts.fpms.ac.be/synthesis/mbrdico

Currently, speech synthesizers are controlled by a multitude of proprietary tag sets. These tag sets vary substantially across synthesizers and are an inhibitor to the adoption of speech synthesis technology by developers. SABLE is an XML/SGML-based markup scheme for text-to-speech synthesis, developed to address the need for a common TTS control paradigm. This paper presents an overview of the SABLE specification, and provides links to sites where further information on SABLE can be accessed.