Fast Speech in Unit Selection Speech Synthesis

Moers-Prinz D. Fast Speech in Unit Selection Speech Synthesis. Bielefeld: Universität Bielefeld; 2020.Speech synthesis is part of the everyday life of many people with severe visual disabilities. For those who are reliant on assistive speech technology the possibility to choose a fast speaking rate is reported to be essential. But also expressive speech synthesis and other spoken language interfaces may require an integration of fast speech. Architectures like formant or diphone synthesis are able to produce synthetic speech at fast speech rates, but the generated speech does not sound very natural. Unit selection synthesis systems, however, are capable of delivering more natural output. Nevertheless, fast speech has not been adequately implemented into such systems to date. Thus, the goal of the work presented here was to determine an optimal strategy for modeling fast speech in unit selection speech synthesis to provide potential users with a more natural sounding alternative for fast speech output

Evaluating cognitive load of text-to-speech synthesis

This thesis addresses the vital topic of evaluating synthetic speech and its impact on the end-user, taking into consideration potential negative implications on cognitive load. While conventional methods like transcription tests and Mean Opinion Scores (MOS) tests offer a valuable overall understanding of system performance, they fail to provide deeper insights into the reasons behind the performance. As text-to-speech (TTS) systems are increasingly used in real-world applications, it becomes crucial to explore whether synthetic speech imposes a greater cognitive load on listeners compared to human speech, as excessive cognitive effort could lead to fatigue over time. The study focuses on assessing the cognitive load of synthetic speech by presenting two methodologies: the dual-task paradigm and pupillometry. The dual-task paradigm initially seemed promising but was eventually deemed unreliable and unsuitable due to uncertainties in experimental setups which requires further investigation. However, pupillometry emerged as a viable approach, demonstrating its efficacy in detecting differences in cognitive load among various speech synthesizers. Notably, the research confirmed that accurate measurement of listening difficulty requires imposing sufficient cognitive load on listeners. To achieve this, the most viable experimental setup involved measuring the pupil response while listening to speech in the presence of noise. Through these experiments, intriguing contrasts between human and synthetic speech were revealed. Human speech consistently demanded the least cognitive load. On the other hand, state-of-the-art TTS systems showed promising results, indicating a significant improvement in their cognitive load performance compared to rule-based synthesizers of the past. Pupillometry offers a deeper understanding of the contributing factors to increased cognitive load in synthetic speech processing. Particularly, an experiment highlighted that the separate modeling of spectral feature prediction and duration in TTS systems led to heightened cognitive load. However, encouragingly, many modern end-to-end TTS systems have addressed these issues by predicting acoustic features within a unified framework, and thus effectively reducing the overall cognitive load imposed by synthetic speech. As the gap between human and synthetic speech diminishes with advancements in TTS technology, continuous evaluation using pupillometry remains essential for optimizing TTS systems for low cognitive load. Although pupillometry demands advanced analysis techniques and is time-consuming, the meaningful insights it provides into the cognitive load of synthetic speech contribute to an enhanced user experience and better TTS system development. Overall, this work successfully establishes pupillometry as a viable and effective method for measuring cognitive load of synthetic speech, propelling synthetic speech evaluation beyond traditional metrics. By gaining a deeper understanding of synthetic speech's interaction with the human cognitive processing system, researchers and developers can work towards creating TTS systems that offer improved user experiences with reduced cognitive load, ultimately enhancing the overall usability and acceptance of such technologies. Note: There was a 2-year break in the work reported in this thesis where an initial pilot was performed in early 2020 and was then suspended due to the covid-19 pandemic. Experiments were therefore rerun in 2022/23 with the most recent state-of-the-art models so that we could determine whether the increased cognitive load result is still applicable. This thesis was thus concluded by answering whether such cognitive load methods developed in this thesis are still useful, practical and/or relevant for current state-of-the-art text-to-speech systems

Realistic and expressive talking head : implementation and evaluation

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Mobile Audiovisual Terminal: System Design and Subjective Testing in DECT and UMTS networks

It is anticipated that there will shortly be a requirement for multimedia terminals that operate via mobile communications systems. This paper presents a functional specification for such a terminal operating at 32 kb/s in a digital European cordless telecommunications (DECT) and universal mobile telecommunications system (UMTS) radio network. A terminal has been built, based on a PC with digital signal processor (DSP) boards for audio and video coding and decoding. Speech coding is by a phonetically driven code-excited linear prediction (CELP) speech coder and video coding by a block-oriented hybrid discrete cosine transform (DCT) coder. Separate channel coding is provided for the audio and video data. The paper describes the techniques used for audio and video coding, channel coding, and synchronization. Methods of subjective testing in a DECT network and in a UMTS network are also described. These consisted of subjective tests of first impressions of the mobile audio–visual terminal (MAVT) quality, interactive tests, and the completion of an exit questionnaire. The test results showed that the quality of the audio was sufficiently good for comprehension and the video was sufficiently good for following and repeating simple mechanical tasks. However, the quality of the MAVT was not good enough for general use where high-quality audio and video was needed, especially when transmission was in a noisy radio environment

Methods for speaking style conversion from normal speech to high vocal effort speech

This thesis deals with vocal-effort-focused speaking style conversion (SSC). Specifically, we studied two topics on conversion of normal speech to high vocal effort. The first topic involves the conversion of normal speech to shouted speech. We employed this conversion in a speaker recognition system with vocal effort mismatch between test and enrollment utterances (shouted speech vs. normal speech). The mismatch causes a degradation of the system's speaker identification performance. As solution, we proposed a SSC system that included a novel spectral mapping, used along a statistical mapping technique, to transform the mel-frequency spectral energies of normal speech enrollment utterances towards their counterparts in shouted speech. We evaluated the proposed solution by comparing speaker identification rates for a state-of-the-art i-vector-based speaker recognition system, with and without applying SSC to the enrollment utterances. Our results showed that applying the proposed SSC pre-processing to the enrollment data improves considerably the speaker identification rates. The second topic involves a normal-to-Lombard speech conversion. We proposed a vocoder-based parametric SSC system to perform the conversion. This system first extracts speech features using the vocoder. Next, a mapping technique, robust to data scarcity, maps the features. Finally, the vocoder synthesizes the mapped features into speech. We used two vocoders in the conversion system, for comparison: a glottal vocoder and the widely used STRAIGHT. We assessed the converted speech from the two vocoder cases with two subjective listening tests that measured similarity to Lombard speech and naturalness. The similarity subjective test showed that, for both vocoder cases, our proposed SSC system was able to convert normal speech to Lombard speech. The naturalness subjective test showed that the converted samples using the glottal vocoder were clearly more natural than those obtained with STRAIGHT

A Study of Accomodation of Prosodic and Temporal Features in Spoken Dialogues in View of Speech Technology Applications

Inter-speaker accommodation is a well-known property of human speech and human interaction in general. Broadly it refers to the behavioural patterns of two (or more) interactants and the effect of the (verbal and non-verbal) behaviour of each to that of the other(s). Implementation of thisbehavior in spoken dialogue systems is desirable as an improvement on the naturalness of humanmachine interaction. However, traditional qualitative descriptions of accommodation phenomena do not provide sufficient information for such an implementation. Therefore, a quantitativedescription of inter-speaker accommodation is required. This thesis proposes a methodology of monitoring accommodation during a human or humancomputer dialogue, which utilizes a moving average filter over sequential frames for each speaker. These frames are time-aligned across the speakers, hence the name Time Aligned Moving Average (TAMA). Analysis of spontaneous human dialogue recordings by means of the TAMA methodology reveals ubiquitous accommodation of prosodic features (pitch, intensity and speech rate) across interlocutors, and allows for statistical (time series) modeling of the behaviour, in a way which is meaningful for implementation in spoken dialogue system (SDS) environments.In addition, a novel dialogue representation is proposed that provides an additional point of view to that of TAMA in monitoring accommodation of temporal features (inter-speaker pause length and overlap frequency). This representation is a percentage turn distribution of individual speakercontributions in a dialogue frame which circumvents strict attribution of speaker-turns, by considering both interlocutors as synchronously active. Both TAMA and turn distribution metrics indicate that correlation of average pause length and overlap frequency between speakers can be attributed to accommodation (a debated issue), and point to possible improvements in SDS “turntaking” behaviour. Although the findings of the prosodic and temporal analyses can directly inform SDS implementations, further work is required in order to describe inter-speaker accommodation sufficiently, as well as to develop an adequate testing platform for evaluating the magnitude ofperceived improvement in human-machine interaction. Therefore, this thesis constitutes a first step towards a convincingly useful implementation of accommodation in spoken dialogue systems

Further Investigation of MDS as a Tool for Evaluation of Speech Quality of Synthesized Speech

The dissertation investigates MDS as a tool for the evaluation of the quality of synthesized speech. More specifically, it investigates the relations between Weighted Euclidean Distance Scaling and Simple Euclidean Distance Scaling, and how aggregating data affects the MDS configuration. It is investigated to what extent a subset of experimental participants and/or experimental stimuli are representative of a larger test set. For that purpose an experiment was conducted on the basis of a subset of stimuli used in the Blizzard Challenge 2008. Issues in the evaluation of Speech Synthesis are discussed and an overview of the basics of multi-dimensional scaling is given to an extent that allows comprehension of methods used in the application of Multi-dimensional scaling to speech synthesis evaluation. Based on the experimental findings, further experiments are suggested with the goal in mind that testing procedures can be optimized to such an extent that the number of experimental participants can be drastically reduced

Intelligibility of synthetic speech in noise and reverberation

Synthetic speech is a valuable means of output, in a range of application contexts, for people with visual, cognitive, or other impairments or for situations were other means are not practicable. Noise and reverberation occur in many of these application contexts and are known to have devastating effects on the intelligibility of natural speech, yet very little was known about the effects on synthetic speech based on unit selection or hidden Markov models. In this thesis, we put forward an approach for assessing the intelligibility of synthetic and natural speech in noise, reverberation, or a combination of the two. The approach uses an experimental methodology consisting of Amazon Mechanical Turk, Matrix sentences, and noises that approximate the real-world, evaluated with generalized linear mixed models. The experimental methodologies were assessed against their traditional counterparts and were found to provide a number of additional benefits, whilst maintaining equivalent measures of relative performance. Subsequent experiments were carried out to establish the efficacy of the approach in measuring intelligibility in noise and then reverberation. Finally, the approach was applied to natural speech and the two synthetic speech systems in combinations of noise and reverberation. We have examine and report on the intelligibility of current synthesis systems in real-life noises and reverberation using techniques that bridge the gap between the audiology and speech synthesis communities and using Amazon Mechanical Turk. In the process, we establish Amazon Mechanical Turk and Matrix sentences as valuable tools in the assessment of synthetic speech intelligibility

Corpus-based unit selection for natural-sounding speech synthesis

Thesis (Ph. D.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 2003.Includes bibliographical references (p. 179-196).This electronic version was submitted by the student author. The certified thesis is available in the Institute Archives and Special Collections.Speech synthesis is an automatic encoding process carried out by machine through which symbols conveying linguistic information are converted into an acoustic waveform. In the past decade or so, a recent trend toward a non-parametric, corpus-based approach has focused on using real human speech as source material for producing novel natural-sounding speech. This work proposes a communication-theoretic formulation in which unit selection is a noisy channel through which an input sequence of symbols passes and an output sequence, possibly corrupted due to the coverage limits of the corpus, emerges. The penalty of approximation is quantified by substitution and concatenation costs which grade what unit contexts are interchangeable and where concatenations are not perceivable. These costs are semi-automatically derived from data and are found to agree with acoustic-phonetic knowledge. The implementation is based on a finite-state transducer (FST) representation that has been successfully used in speech and language processing applications including speech recognition. A proposed constraint kernel topology connects all units in the corpus with associated substitution and concatenation costs and enables an efficient Viterbi search that operates with low latency and scales to large corpora. An A* search can be applied in a second, rescoring pass to incorporate finer acoustic modelling. Extensions to this FST-based search include hierarchical and paralinguistic modelling. The search can also be used in an iterative feedback loop to record new utterances to enhance corpus coverage. This speech synthesis framework has been deployed across various domains and languages in many voices, a testament to its flexibility and rapid prototyping capability.(cont.) Experimental subjects completing tasks in a given air travel planning scenario by interacting in real time with a spoken dialogue system over the telephone have found the system "easiest to understand" out of eight competing systems. In more detailed listening evaluations, subjective opinions garnered from human participants are found to be correlated with objective measures calculable by machine.by Jon Rong-Wei Yi.Ph.D