Automatic speech recognition with deep neural networks for impaired speech

The final publication is available at https://link.springer.com/chapter/10.1007%2F978-3-319-49169-1_10Automatic Speech Recognition has reached almost human performance in some controlled scenarios. However, recognition of impaired speech is a difficult task for two main reasons: data is (i) scarce and (ii) heterogeneous. In this work we train different architectures on a database of dysarthric speech. A comparison between architectures shows that, even with a small database, hybrid DNN-HMM models outperform classical GMM-HMM according to word error rate measures. A DNN is able to improve the recognition word error rate a 13% for subjects with dysarthria with respect to the best classical architecture. This improvement is higher than the one given by other deep neural networks such as CNNs, TDNNs and LSTMs. All the experiments have been done with the Kaldi toolkit for speech recognition for which we have adapted several recipes to deal with dysarthric speech and work on the TORGO database. These recipes are publicly available.Peer ReviewedPostprint (author's final draft

A study on the relationship between the intelligibility and quality of algorithmically-modified speech for normal hearing listeners

This study investigates the relationship between the intelligibility and quality of modified speech in noise and in quiet. Speech signals were processed by seven algorithms designed to increase speech intelligibility in noise without altering speech intensity. In three noise maskers, including both stationary and fluctuating noise at two signal-to-noise ratios (SNR), listeners identified keywords from unmodified or modified sentences. The intelligibility performance of each type of speech was measured as the listeners’ word recognition rate in each condition, while the quality was rated as a mean opinion score. In quiet, only the perceptual quality of each type of speech was assessed. The results suggest that when listening in noise, modification performance on improving intelligibility is more important than its potential negative impact on speech quality. However, when listening in quiet or at SNRs in which intelligibility is no longer an issue to listeners, the impact to speech quality due to modification becomes a concern

An analysis of the short utterance problem for speaker characterization

Speaker characterization has always been conditioned by the length of the evaluated utterances. Despite performing well with large amounts of audio, significant degradations in performance are obtained when short utterances are considered. In this work we present an analysis of the short utterance problem providing an alternative point of view. From our perspective the performance in the evaluation of short utterances is highly influenced by the phonetic similarity between enrollment and test utterances. Both enrollment and test should contain similar phonemes to properly discriminate, being degraded otherwise. In this study we also interpret short utterances as incomplete long utterances where some acoustic units are either unbalanced or just missing. These missing units are responsible for the speaker representations to be unreliable. These unreliable representations are biased with respect to the reference counterparts, obtained from long utterances. These undesired shifts increase the intra-speaker variability, causing a significant loss of performance. According to our experiments, short utterances (3-60 s) can perform as accurate as if long utterances were involved by just reassuring the phonetic distributions. This analysis is determined by the current embedding extraction approach, based on the accumulation of local short-time information. Thus it is applicable to most of the state-of-the-art embeddings, including traditional i-vectors and Deep Neural Network (DNN) xvectors

Linguistically-constrained formant-based i-vectors for automatic speaker recognition

This is the author’s version of a work that was accepted for publication in Speech Communication. Changes resulting from the publishing process, such as peer review, editing, corrections, structural formatting, and other quality control mechanisms may not be reflected in this document. Changes may have been made to this work since it was submitted for publication. A definitive version was subsequently published in Speech Communication, VOL 76 (2016) DOI 10.1016/j.specom.2015.11.002This paper presents a large-scale study of the discriminative abilities of formant frequencies for automatic speaker recognition. Exploiting both the static and dynamic information in formant frequencies, we present linguistically-constrained formant-based i-vector systems providing well calibrated likelihood ratios per comparison of the occurrences of the same isolated linguistic units in two given utterances. As a first result, the reported analysis on the discriminative and calibration properties of the different linguistic units provide useful insights, for instance, to forensic phonetic practitioners. Furthermore, it is shown that the set of units which are more discriminative for every speaker vary from speaker to speaker. Secondly, linguistically-constrained systems are combined at score-level through average and logistic regression speaker-independent fusion rules exploiting the different speaker-distinguishing information spread among the different linguistic units. Testing on the English-only trials of the core condition of the NIST 2006 SRE (24,000 voice comparisons of 5 minutes telephone conversations from 517 speakers -219 male and 298 female-), we report equal error rates of 9.57 and 12.89% for male and female speakers respectively, using only formant frequencies as speaker discriminative information. Additionally, when the formant-based system is fused with a cepstral i-vector system, we obtain relative improvements of ∼6% in EER (from 6.54 to 6.13%) and ∼15% in minDCF (from 0.0327 to 0.0279), compared to the cepstral system alone.This work has been supported by the Spanish Ministry of Economy and Competitiveness (project CMC-V2: Caracterizacion, Modelado y Compensacion de Variabilidad en la Señal de Voz, TEC2012-37585-C02-01). Also, the authors would like to thank SRI for providing the Decipher phonetic transcriptions of the NIST 2004, 2005 and 2006 SREs that have allowed to carry out this work

Speech Activity and Speaker Change Point Detection for Online Streams

Disertační práce je věnována dvěma si blízkým řečovým úlohám a následně jejich použití v online prostředí. Konkrétně se jedná o úlohy detekce řeči a detekce změny mluvčího. Ty jsou často nedílnou součástí systémů pro zpracování řeči (např. pro diarizaci mluvčích nebo rozpoznávání řeči), kde slouží pro předzpracování akustického signálu. Obě úlohy jsou v literatuře velmi aktivním tématem, ale většina existujících prací je směřována primárně na offline využití. Nicméně právě online nasazení je nezbytné pro některé řečové aplikace, které musí fungovat v reálném čase (např. monitorovací systémy).Úvodní část disertační práce je tvořena třemi kapitolami. V té první jsou vysvětleny základní pojmy a následně je nastíněno využití obou úloh. Druhá kapitola je věnována současnému poznání a je doplněna o přehled existujících nástrojů. Poslední kapitola se skládá z motivace a z praktického použití zmíněných úloh v monitorovacích systémech. V závěru úvodní části jsou stanoveny cíle práce.Následující dvě kapitoly jsou věnovány teoretickým základům obou úloh. Představují vybrané přístupy, které jsou buď relevantní pro disertační práci (porovnání výsledků), nebo jsou zaměřené na použití v online prostředí.V další kapitole je předložen finální přístup pro detekci řeči. Postupný návrh tohoto přístupu, společně s experimentálním vyhodnocením, je zde detailně rozebrán. Přístup dosahuje nejlepších výsledků na korpusu QUT-NOISE-TIMIT v podmínkách s nízkým a středním zašuměním. Přístup je také začleněn do monitorovacího systému, kde doplňuje svojí funkcionalitou rozpoznávač řeči.Následující kapitola detailně představuje finální přístup pro detekci změny mluvčího. Ten byl navržen v rámci několika po sobě jdoucích experimentů, které tato kapitola také přibližuje. Výsledky získané na databázi COST278 se blíží výsledkům, kterých dosáhl referenční offline systém, ale předložený přístup jich docílil v online módu a to s nízkou latencí.Výstupy disertační práce jsou shrnuty v závěrečné kapitole.The main focus of this thesis lies on two closely interrelated tasks, speech activity detection and speaker change point detection, and their applications in online processing. These tasks commonly play a crucial role of speech preprocessors utilized in speech-processing applications, such as automatic speech recognition or speaker diarization. While their use in offline systems is extensively covered in literature, the number of published works focusing on online use is limited.This is unfortunate, as many speech-processing applications (e.g., monitoring systems) are required to be run in real time.The thesis begins with a three-chapter opening part, where the first introductory chapter explains the basic concepts and outlines the practical use of both tasks. It is followed by a chapter, which reviews the current state of the art and lists the existing toolkits. That part is concluded by a chapter explaining the motivation behind this work and the practical use in monitoring systems; ultimately, this chapter sets the main goals of this thesis.The next two chapters cover the theoretical background of both tasks. They present selected approaches relevant to this work (e.g., used for result comparisons) or focused on online processing.The following chapter proposes the final speech activity detection approach for online use. Within this chapter, a detailed description of the development of this approach is available as well as its thorough experimental evaluation. This approach yields state-of-the-art results under low- and medium-noise conditions on the standardized QUT-NOISE-TIMIT corpus. It is also integrated into a monitoring system, where it supplements a speech recognition system.The final speaker change point detection approach is proposed in the following chapter. It was designed in a series of consecutive experiments, which are extensively detailed in this chapter. An experimental evaluation of this approach on the COST278 database shows the performance of approaching the offline reference system while operating in online mode with low latency.Finally, the last chapter summarizes all the results of this thesis

Comparing word and syllable prominence rated by naïve listeners

Prominence has been widely studied on the word level and the syllable level. An extensive study comparing the two approaches is missing in the literature. This study investigates how word and syllable prominence relate to each other in German. We find that perceptual ratings based on the word level are more extreme than those based on the syllable level. The correlations between word prominence and acoustic features are greater than the correlations between syllable prominence and acoustic features

Robust speech recognition with spectrogram factorisation

Communication by speech is intrinsic for humans. Since the breakthrough of mobile devices and wireless communication, digital transmission of speech has become ubiquitous. Similarly distribution and storage of audio and video data has increased rapidly. However, despite being technically capable to record and process audio signals, only a fraction of digital systems and services are actually able to work with spoken input, that is, to operate on the lexical content of speech. One persistent obstacle for practical deployment of automatic speech recognition systems is inadequate robustness against noise and other interferences, which regularly corrupt signals recorded in real-world environments. Speech and diverse noises are both complex signals, which are not trivially separable. Despite decades of research and a multitude of different approaches, the problem has not been solved to a sufficient extent. Especially the mathematically ill-posed problem of separating multiple sources from a single-channel input requires advanced models and algorithms to be solvable. One promising path is using a composite model of long-context atoms to represent a mixture of non-stationary sources based on their spectro-temporal behaviour. Algorithms derived from the family of non-negative matrix factorisations have been applied to such problems to separate and recognise individual sources like speech. This thesis describes a set of tools developed for non-negative modelling of audio spectrograms, especially involving speech and real-world noise sources. An overview is provided to the complete framework starting from model and feature definitions, advancing to factorisation algorithms, and finally describing different routes for separation, enhancement, and recognition tasks. Current issues and their potential solutions are discussed both theoretically and from a practical point of view. The included publications describe factorisation-based recognition systems, which have been evaluated on publicly available speech corpora in order to determine the efficiency of various separation and recognition algorithms. Several variants and system combinations that have been proposed in literature are also discussed. The work covers a broad span of factorisation-based system components, which together aim at providing a practically viable solution to robust processing and recognition of speech in everyday situations

Multi-Sensory Emotion Recognition with Speech and Facial Expression

Emotion plays an important role in human beings’ daily lives. Understanding emotions and recognizing how to react to others’ feelings are fundamental to engaging in successful social interactions. Currently, emotion recognition is not only significant in human beings’ daily lives, but also a hot topic in academic research, as new techniques such as emotion recognition from speech context inspires us as to how emotions are related to the content we are uttering. The demand and importance of emotion recognition have highly increased in many applications in recent years, such as video games, human computer interaction, cognitive computing, and affective computing. Emotion recognition can be done from many sources including text, speech, hand, and body gesture as well as facial expression. Presently, most of the emotion recognition methods only use one of these sources. The emotion of human beings changes every second and using a single way to process the emotion recognition may not reflect the emotion correctly. This research is motivated by the desire to understand and evaluate human beings’ emotion from multiple ways such as speech and facial expressions. In this dissertation, multi-sensory emotion recognition has been exploited. The proposed framework can recognize emotion from speech, facial expression, and both of them. There are three important parts in the design of the system: the facial emotion recognizer, the speech emotion recognizer, and the information fusion. The information fusion part uses the results from the speech emotion recognition and facial emotion recognition. Then, a novel weighted method is used to integrate the results, and a final decision of the emotion is given after the fusion. The experiments show that with the weighted fusion methods, the accuracy can be improved to an average of 3.66% compared to fusion without adding weight. The improvement of the recognition rate can reach 18.27% and 5.66% compared to the speech emotion recognition and facial expression recognition, respectively. By improving the emotion recognition accuracy, the proposed multi-sensory emotion recognition system can help to improve the naturalness of human computer interaction