There is an ever-growing and increasing amount of technology options that use speech recognition software. Currently, the market includes smartphones, computers, and individual smart home personal assistants that allow for hands-free access to this technology. Research studies have explored the utility of these assistive devices for the completion of activities of daily living; however, there is limited research looking at the accuracy of voice recognition software within smart home personal assistants in populations with disordered speech. In persons with amyotrophic lateral sclerosis (ALS), symptoms include changes to motor functions, speech in particular, and it is unknown how some of these devices may respond to their disordered speech. The present study aimed to examine the accuracy of the Amazon Echo to respond appropriately to commands given by dysarthric patients with ALS. Participants were asked to read a variety of commands to an Amazon Echo. The sentences and responses by the Amazon Echo were audio-recorded for transcription and intelligibility ratings, which were then analyzed to look for relationships between intelligibility, auditory-perceptual features of speech, and sentence type. Results revealed there was no significant relationship between command intelligibility and accuracy of response by the Amazon Echo, nor was there a significant relationship between any of the auditory-perceptual ratings and accuracy of response. There was, however, a significant and positive association between conversational intelligibility and accuracy of responses by the Amazon Echo. This study provides support for use of hands-free assistive technology in patients with ALS to aid in the maintenance of quality of life and activities of daily living.

In Sweden, about 10% of the adult population experienceshearing problems that cause them difficulties in everydaycommunication, and approximately 60 000 people are providedwith hearing aids each year. Despite the fact that modernhearing aids can facilitate speech communication in a widerange of listening environments, many hearing-aid users aredissatisfied with their hearing aids. It is likely that theclinical methods used for individual fitting of the hearingaids are not optimal. The current study investigates prescriptive methods fornonlinear, wide dynamic range compression (WDRC) hearinginstruments. The goal is to draw general conclusions about thepreferences of hearing aid users. Therefore, the prescriptionsare evaluated using well-established models of loudness andspeech intelligibility. Current methods differed considerably in prescribed gain.Evaluations in a laboratory test, with 20 hearing-impairedlisteners, showed that these differences led to largedifferences in perceived and calculated loudness, but only tominor differences in measured and predicted speech recognitionscores. The difference in loudness was explored in a studywhere 21 first-time hearing-aid users compared twoprescriptions. One method led to normal and the other toless-than-normal overall calculated loudness (according to theloudness model of Moore and Glasberg (1997)). The prescriptionthat led to less-than-normal overall loudness was clearlypreferred in field and in laboratory tests. Preferred overall loudness was then quantified.Hearing-impaired participants with mild to moderate hearingloss preferred considerably less-than-normal overall calculatedloudness in both eld and laboratory tests. There were nosignificant differences between inexperienced and experiencedhearing aid users. Normal-hearing participants, on the otherhand, preferred close-to-normal overall calculated loudness. Inaddition, a potential problem with the loudness model wasencountered: despite the fact that the hearing-impairedlisteners were provided with less than normal overallcalculated loudness, they rated loudness higher than thenormal-hearing listeners. The results refute the most commonly adopted rationale forprescriptive methods for WDRC hearing aids - that overallloudness should be restored to normal. Hearing-impairedlisteners with mild to moderate hearing loss preferredconsiderably less than normal overall loudness. This should betaken into account when deriving new prescriptive methods, andwhen providing clients with hearing aids. Key words:hearing impairment, hearing aid, nonlinear,WDRC, hearing aid experience, prescription, loudness, loudnessmodel, speech intelligibility, preference.

Hospital noise has been an area of concern for medical professionals and researchers for the last century. Researchers have attempted to characterize the soundscape of hospital wards and have made some preliminary links between noise and human outcomes. In the past, most of the research has used traditional acoustic metrics. These traditional metrics, such as average sound level, are readily measured using sound level meters and have been the primary results reported in previous studies. However, it has been shown that these traditional metrics may be insufficient in fully characterizing the wards. The two studies presented here use traditional metrics and nontraditional metrics to define the soundscape of hospital wards. The uncovered links, between both sound level metrics and psychoacoustic metrics and patient physiological measurements, are discussed. Correlations and risk ratios demonstrate the presence and the strength of these relationships. These results demonstrate the relationships between hospital acoustics and patient physiological arousal. Additionally, the effects of adding absorption in a hospital ward are presented. Sound level, sound power, reverberation time and other acoustic metrics are directly affected. The speech intelligibility in these wards is evaluated in order to highlight the temporal nature of speech intelligibility. With both studies combined, both traditional and nontraditional acoustic measures are shown to have statistically significant relationships to both patient and staff outcomes.

<p>In Sweden, about 10% of the adult population experienceshearing problems that cause them difficulties in everydaycommunication, and approximately 60 000 people are providedwith hearing aids each year. Despite the fact that modernhearing aids can facilitate speech communication in a widerange of listening environments, many hearing-aid users aredissatisfied with their hearing aids. It is likely that theclinical methods used for individual fitting of the hearingaids are not optimal.</p><p>The current study investigates prescriptive methods fornonlinear, wide dynamic range compression (WDRC) hearinginstruments. The goal is to draw general conclusions about thepreferences of hearing aid users. Therefore, the prescriptionsare evaluated using well-established models of loudness andspeech intelligibility.</p><p>Current methods differed considerably in prescribed gain.Evaluations in a laboratory test, with 20 hearing-impairedlisteners, showed that these differences led to largedifferences in perceived and calculated loudness, but only tominor differences in measured and predicted speech recognitionscores. The difference in loudness was explored in a studywhere 21 first-time hearing-aid users compared twoprescriptions. One method led to normal and the other toless-than-normal overall calculated loudness (according to theloudness model of Moore and Glasberg (1997)). The prescriptionthat led to less-than-normal overall loudness was clearlypreferred in field and in laboratory tests.</p><p>Preferred overall loudness was then quantified.Hearing-impaired participants with mild to moderate hearingloss preferred considerably less-than-normal overall calculatedloudness in both eld and laboratory tests. There were nosignificant differences between inexperienced and experiencedhearing aid users. Normal-hearing participants, on the otherhand, preferred close-to-normal overall calculated loudness. Inaddition, a potential problem with the loudness model wasencountered: despite the fact that the hearing-impairedlisteners were provided with less than normal overallcalculated loudness, they rated loudness higher than thenormal-hearing listeners.</p><p>The results refute the most commonly adopted rationale forprescriptive methods for WDRC hearing aids - that overallloudness should be restored to normal. Hearing-impairedlisteners with mild to moderate hearing loss preferredconsiderably less than normal overall loudness. This should betaken into account when deriving new prescriptive methods, andwhen providing clients with hearing aids.</p><p><b>Key words:</b>hearing impairment, hearing aid, nonlinear,WDRC, hearing aid experience, prescription, loudness, loudnessmodel, speech intelligibility, preference.</p>

published_or_final_version / Speech and Hearing Sciences / Master / Master of Science in Audiology

Clear speech is a form of communication that talkers naturally use when speaking in difficult listening conditions or with a person who has a hearing loss. Clear speech, on average, provides listeners with hearing impairments an intelligibility benefit of 17 percentage points (Picheny, Durlach, & Braida, 1985) over conversational speech. In addition, it provides increased intelligibility in various listening conditions (Krause & Braida, 2003, among others), with different stimuli (Bradlow & Bent, 2002; Gagne, Rochette, & Charest, 2002; Helfer, 1997, among others) and across listener populations (Bradlow, Kraus, & Hayes, 2003, among others). Recently, researchers have attempted to compare their findings with clear and conversational speech, at slow and normal rates, with results from other investigators' studies in an effort to determine the relative benefits of clear speech across populations and environments. However, relative intelligibility benefits are difficult to determine unless baseline performance levels can be equated, suggesting that listener psychometric functions with clear speech are needed. The purpose of this study was to determine how speech intelligibility, as measured by percentage key words correct in nonsense sentences by young adults, varies with changes in speaking condition, talker and signal-to-noise ratio (SNR). Forty young, normal hearing adults were presented with grammatically correct nonsense sentences at five SNRs. Each listener heard a total of 800 sentences in four speaking conditions: clear and conversational styles, at slow and normal rates (i.e., clear/slow, clear/normal, conversational/slow, and conversational/normal). Overall results indicate clear/slow and conversational/slow were the most intelligible conditions, followed by clear/normal and then conversational/normal conditions. Moreover, the average intelligibility benefit for clear/slow, clear/normal and conversational/slow conditions (relative to conversational/normal) was maintained across an SNR range of -4 to 0 dB in the middle, or linear, portion of the psychometric function. However, when results are examined by talker, differences are observed in the benefit provided by each condition and in how the benefit varies across noise levels. In order to counteract talker variability, research with a larger number of talkers is recommended for future studies.

Clear speech refers to a speaking style that is more intelligible than typical, conversational speaking styles. It is usually produced at a slower rate compared to conversational speech. Clear speech has been shown to be more intelligible than conversational speech for a large variety of populations, including both hearing impaired (Schum, 1996; Picheny, Durlach, & Braida, 1985; and Payton, Uchanski, & Braida, 1994) and normal hearing individuals (e.g. Uchanski, Choi, Braida, Reed, & Durlach, 1996) under a variety of conditions, including those in which presentation level, speaker, and environment are varied. Although clear speech is typically slower than normally produced conversational speech, recent studies have shown that it can be produced at normal rates with training (Krause & Braida, 2002). If clear speech at normal rates is shown to be as effective for individuals with hearing loss as clear speech at slow rates, it would have both clinical and research implications. The purpose of this study was to determine the effectiveness of clear speech at normal rates for older individuals with hearing loss. It examined the way in which intelligibility, measured as percent correct keyword scores on nonsense sentences, varied as a result of speaking mode (clear versus conversational speech) and speaking rate (slow versus normal) in six adults aged 55-75 years old with moderate, sloping, hearing loss. Each listener was presented with nonsense sentences in four speech conditions: clear speech at slow rates (clear/slow), clear speech at normal rates (clear/normal), conversational speech at slow rates (conv/slow), and conversational speech at normal rates (conv/normal) read by four different talkers. Sentences were presented monaurally in quiet to the listeners via headphones. Results indicated that clear/slow speech was the most intelligible condition overall. Neither conv/slow nor clear/normal provided an intelligibility benefit relative to conv/normal speech on average, suggesting that for older adults with moderate, sloping hearing loss, the combination of using clear speech and a slower speaking rate is more beneficial to intelligibility than the additive effects of altering either speaking rate or speaking mode alone. It has been suggested previously (Krause, 2001) that audiological characteristics may contribute to the lack of clear/normal benefit for certain listeners with hearing loss. Although clear/normal speech was not beneficial on average to listeners in this study, there were cases in which the clear/normal speech of a particular talker provided a benefit to a particular listener. Thus, severity and configuration of hearing loss alone cannot fully explain the degree to which listeners from hearing loss do (or do not) benefit from clear/normal speech. More studies are needed to investigate the benefits of clear/normal speech for different audiological configurations, including individuals with flat losses. In addition, the listening tasks should include more difficult conditions in order to compensate for potential ceiling effects.

published_or_final_version / Speech and Hearing Sciences / Master / Master of Philosophy

This thesis studies disfluency in spontaneous Swedish speech, i.e., the occurrence of hesitation phenomena like eh, öh, truncated words, repetitions and repairs, mispronunciations, truncated words and so on. The thesis is divided into three parts: PART I provides the background, both concerning scientific, personal and industrial–academic aspects in the Tuning in quotes, and the Preamble and Introduction (chapter 1). PART II consists of one chapter only, chapter 2, which dives into the etiology of disfluency. Consequently it describes previous research on disfluencies, also including areas that are not the main focus of the present tome, like stuttering, psychotherapy, philosophy, neurology, discourse perspectives, speech production, application-driven perspectives, cognitive aspects, and so on. A discussion on terminology and definitions is also provided. The goal of this chapter is to provide as broad a picture as possible of the phenomenon of disfluency, and how all those different and varying perspectives are related to each other. PART III describes the linguistic data studied and analyzed in this thesis, with the following structure: Chapter 3 describes how the speech data were collected, and for what reason. Sum totals of the data and the post-processing method are also described. Chapter 4 describes how the data were transcribed, annotated and analyzed. The labeling method is described in detail, as is the method employed to do frequency counts. Chapter 5 presents the analysis and results for all different categories of disfluencies. Besides general frequency and distribution of the different types of disfluencies, both inter- and intra-corpus results are presented, as are co-occurrences of different types of disfluencies. Also, inter- and intra-speaker differences are discussed. Chapter 6 discusses the results, mainly in light of previous research. Reasons for the observed frequencies and distribution are proposed, as are their relation to language typology, as well as syntactic, morphological and phonetic reasons for the observed phenomena. Future work is also envisaged, both work that is possible on the present data set, work that is possible on the present data set given extended labeling and work that I think should be carried out, but where the present data set fails, in one way or another, to meet the requirements of such studies. Appendices 1–4 list the sum total of all data analyzed in this thesis (apart from Tok Pisin data). Appendix 5 provides an example of a full human–computer dialogue. / The electronic version of the printed dissertation is a corrected version where typos as well as phrases have been corrected. A list with the corrections is presented in the errata list above.

Speech signals radiated in confined spaces are subject to reverberation due to reflections of surrounding walls and obstacles. Reverberation leads to severe degradation of speech intelligibility and can be prohibitive for applications where speech is digitally recorded, such as audio conferencing or hearing aids. Dereverberation of speech is therefore an important field in speech enhancement. Driven by consumer demand, blind speech dereverberation has become a popular field in the research community and has led to many interesting approaches in the literature. However, most existing methods are dictated by their underlying models and hence suffer from assumptions that constrain the approaches to specific subproblems of blind speech dereverberation. For example, many approaches limit the dereverberation to voiced speech sounds, leading to poor results for unvoiced speech. Few approaches tackle single-sensor blind speech dereverberation, and only a very limited subset allows for dereverberation of speech from moving speakers. Therefore, the aim of this dissertation is the development of a flexible and extendible framework for blind speech dereverberation accommodating different speech sound types, single- or multiple sensor as well as stationary and moving speakers. Bayesian methods benefit from – rather than being dictated by – appropriate model choices. Therefore, the problem of blind speech dereverberation is considered from a Bayesian perspective in this thesis. A generic sequential Monte Carlo approach accommodating a multitude of models for the speech production mechanism and room transfer function is consequently derived. In this approach both the anechoic source signal and reverberant channel are estimated using their optimal estimators by means of Rao-Blackwellisation of the state-space of unknown variables. The remaining model parameters are estimated using sequential importance resampling. The proposed approach is implemented for two different speech production models for stationary speakers, demonstrating substantial reduction in reverberation for both unvoiced and voiced speech sounds. Furthermore, the channel model is extended to facilitate blind dereverberation of speech from moving speakers. Due to the structure of measurement model, single- as well as multi-microphone processing is facilitated, accommodating physically constrained scenarios where only a single sensor can be used as well as allowing for the exploitation of spatial diversity in scenarios where the physical size of microphone arrays is of no concern. This dissertation is concluded with a survey of possible directions for future research, including the use of switching Markov source models, joint target tracking and enhancement, as well as an extension to subband processing for improved computational efficiency.