Modeling Spoken Information Queries for Virtual Assistants: Open Problems, Challenges and Opportunities

Virtual assistants are becoming increasingly important speech-driven Information Retrieval platforms that assist users with various tasks. We discuss open problems and challenges with respect to modeling spoken information queries for virtual assistants, and list opportunities where Information Retrieval methods and research can be applied to improve the quality of virtual assistant speech recognition. We discuss how query domain classification, knowledge graphs and user interaction data, and query personalization can be helpful to improve the accurate recognition of spoken information domain queries. Finally, we also provide a brief overview of current problems and challenges in speech recognition.Comment: SIGIR '23. The 46th International ACM SIGIR Conference on Research & Development in Information Retrieva

Challenges of developing a digital scribe to reduce clinical documentation burden.

Clinicians spend a large amount of time on clinical documentation of patient encounters, often impacting quality of care and clinician satisfaction, and causing physician burnout. Advances in artificial intelligence (AI) and machine learning (ML) open the possibility of automating clinical documentation with digital scribes, using speech recognition to eliminate manual documentation by clinicians or medical scribes. However, developing a digital scribe is fraught with problems due to the complex nature of clinical environments and clinical conversations. This paper identifies and discusses major challenges associated with developing automated speech-based documentation in clinical settings: recording high-quality audio, converting audio to transcripts using speech recognition, inducing topic structure from conversation data, extracting medical concepts, generating clinically meaningful summaries of conversations, and obtaining clinical data for AI and ML algorithms

El-WOZ: a client-server wizard-of-oz open-source interface

International audienceWizard of Oz (WOZ) prototyping employs a human wizard to simulate anticipated functions of a future system. In Natural Language Processing this method is usually used to obtain early feedback on dialogue designs, to collect language corpora, or to explore interaction strategies. Yet, existing tools often require complex client-server configurations and setup routines, or suffer from compatibility problems with different platforms. Integrated solutions, which may also be used by designers and researchers without technical background, are missing. In this paper we present a framework for multi-lingual dialog research, which combines speech recognition and synthesis with WOZ. All components are open source and adaptable toIn this paper, we present a speech recording interface developed in the context of a project on automatic speech recognition for elderly native speakers of European Portuguese. In order to collect spontaneous speech in a situation of interaction with a machine, this interface was designed as a Wizard-of-Oz (WOZ) plateform. In this setup, users interact with a fake automated dialog system controled by a human wizard. It was implemented as a client-server application and the subjects interact with a talking head. The human wizard chooses pre-defined questions or sentences in a graphical user interface, which are then synthesized and spoken aloud by the avatar on the client side. A small spontaneous speech corpus was collected in a daily center. Eight speakers between 75 and 90 years old were recorded. They appreciated the interface and felt at ease with the avatar. Manual orthographic transcriptions were created for the total of about 45 minutes of speech. different application scenario

Keskusteluavustimen kehittäminen kuulovammaisia varten automaattista puheentunnistusta käyttäen

Understanding and participating in conversations has been reported as one of the biggest challenges hearing impaired people face in their daily lives. These communication problems have been shown to have wide-ranging negative consequences, affecting their quality of life and the opportunities available to them in education and employment. A conversational assistance application was investigated to alleviate these problems. The application uses automatic speech recognition technology to provide real-time speech-to-text transcriptions to the user, with the goal of helping deaf and hard of hearing persons in conversational situations. To validate the method and investigate its usefulness, a prototype application was developed for testing purposes using open-source software. A user test was designed and performed with test participants representing the target user group. The results indicate that the Conversation Assistant method is valid, meaning it can help the hearing impaired to follow and participate in conversational situations. Speech recognition accuracy, especially in noisy environments, was identified as the primary target for further development for increased usefulness of the application. Conversely, recognition speed was deemed to be sufficient and already surpass the transcription speed of human transcribers.Keskustelupuheen ymmärtäminen ja keskusteluihin osallistuminen on raportoitu yhdeksi suurimmista haasteista, joita kuulovammaiset kohtaavat jokapäiväisessä elämässään. Näillä viestintäongelmilla on osoitettu olevan laaja-alaisia negatiivisia vaikutuksia, jotka heijastuvat elämänlaatuun ja heikentävät kuulovammaisten yhdenvertaisia osallistumismahdollisuuksia opiskeluun ja työelämään. Työssä kehitettiin ja arvioitiin apusovellusta keskustelupuheen ymmärtämisen ja keskusteluihin osallistumisen helpottamiseksi. Sovellus käyttää automaattista puheentunnistusta reaaliaikaiseen puheen tekstittämiseen kuuroja ja huonokuuloisia varten. Menetelmän toimivuuden vahvistamiseksi ja sen hyödyllisyyden tutkimiseksi siitä kehitettiin prototyyppisovellus käyttäjätestausta varten avointa lähdekoodia hyödyntäen. Testaamista varten suunniteltiin ja toteutettiin käyttäjäkoe sovelluksen kohderyhmää edustavilla koekäyttäjillä. Saadut tulokset viittaavat siihen, että työssä esitetty Keskusteluavustin on toimiva ja hyödyllinen apuväline huonokuuloisille ja kuuroille. Puheentunnistustarkkuus erityisesti meluisissa olosuhteissa osoittautui ensisijaiseksi kehityskohteeksi apusovelluksen hyödyllisyyden lisäämiseksi. Puheentunnistuksen nopeus arvioitiin puolestaan jo riittävän nopeaksi, ylittäen selkeästi kirjoitustulkkien kirjoitusnopeuden

Automatic Diagnosis of Distortion Type of Arabic /r/ Phoneme Using Feed Forward Neural Network

The paper is not for recognizing normal formed speech but for distorted speech via examining the ability of feed forward Artificial Neural Networks (ANN) to recognize speech flaws. In this paper we take the Arabic /r/ phoneme distortion that is somewhat common among native speakers as a case study.To do this, r-Distype program is developed as a script written using Praat speech processing software tool. r-Distype program automatically develops a feed forward ANN that tests the spoken word (which includes /r/ phoneme) to detect any possible type of distortion. Multiple feed forward ANNs of different architectures have been developed and their achievements reported. Training data and testing data of the developed ANNs are sets of spoken Arabic words that contain /r/ phoneme in different positions so they cover all distortion types of Arabic /r/ phoneme. These sets of words were produced by different genders and different ages.The results obtained from developed ANNs were used to draw a conclusion about automating the detection of pronunciation problems in general.Such computerised system would be a good tool for diagnosing speech flaws and gives a great help in speech therapy. Also, the idea itself may open a new research subarea of speech recognition that is automatic speech therapy. Keywords: Distortion, Arabic /r/ phoneme, articulation disorders, Artificial Neural Network, Praa

Deep spiking neural networks with applications to human gesture recognition

The spiking neural networks (SNNs), as the 3rd generation of Artificial Neural Networks (ANNs), are a class of event-driven neuromorphic algorithms that potentially have a wide range of application domains and are applicable to a variety of extremely low power neuromorphic hardware. The work presented in this thesis addresses the challenges of human gesture recognition using novel SNN algorithms. It discusses the design of these algorithms for both visual and auditory domain human gesture recognition as well as event-based pre-processing toolkits for audio signals. From the visual gesture recognition aspect, a novel SNN-based event-driven hand gesture recognition system is proposed. This system is shown to be effective in an experiment on hand gesture recognition with its spiking recurrent convolutional neural network (SCRNN) design, which combines both designed convolution operation and recurrent connectivity to maintain spatial and temporal relations with address-event-representation (AER) data. The proposed SCRNN architecture can achieve arbitrary temporal resolution, which means it can exploit temporal correlations between event collections. This design utilises a backpropagation-based training algorithm and does not suffer from gradient vanishing/explosion problems. From the audio perspective, a novel end-to-end spiking speech emotion recognition system (SER) is proposed. This system employs the MFCC as its main speech feature extractor as well as a self-designed latency coding algorithm to effciently convert the raw signal to AER input that can be used for SNN. A two-layer spiking recurrent architecture is proposed to address temporal correlations between spike trains. The robustness of this system is supported by several open public datasets, which demonstrate state of the arts recognition accuracy and a significant reduction in network size, computational costs, and training speed. In addition to directly contributing to neuromorphic SER, this thesis proposes a novel speech-coding algorithm based on the working mechanism of humans auditory organ system. The algorithm mimics the functionality of the cochlea and successfully provides an alternative method of event-data acquisition for audio-based data. The algorithm is then further simplified and extended into an application of speech enhancement which is jointly used in the proposed SER system. This speech-enhancement method uses the lateral inhibition mechanism as a frequency coincidence detector to remove uncorrelated noise in the time-frequency spectrum. The method is shown to be effective by experiments for up to six types of noise.The spiking neural networks (SNNs), as the 3rd generation of Artificial Neural Networks (ANNs), are a class of event-driven neuromorphic algorithms that potentially have a wide range of application domains and are applicable to a variety of extremely low power neuromorphic hardware. The work presented in this thesis addresses the challenges of human gesture recognition using novel SNN algorithms. It discusses the design of these algorithms for both visual and auditory domain human gesture recognition as well as event-based pre-processing toolkits for audio signals. From the visual gesture recognition aspect, a novel SNN-based event-driven hand gesture recognition system is proposed. This system is shown to be effective in an experiment on hand gesture recognition with its spiking recurrent convolutional neural network (SCRNN) design, which combines both designed convolution operation and recurrent connectivity to maintain spatial and temporal relations with address-event-representation (AER) data. The proposed SCRNN architecture can achieve arbitrary temporal resolution, which means it can exploit temporal correlations between event collections. This design utilises a backpropagation-based training algorithm and does not suffer from gradient vanishing/explosion problems. From the audio perspective, a novel end-to-end spiking speech emotion recognition system (SER) is proposed. This system employs the MFCC as its main speech feature extractor as well as a self-designed latency coding algorithm to effciently convert the raw signal to AER input that can be used for SNN. A two-layer spiking recurrent architecture is proposed to address temporal correlations between spike trains. The robustness of this system is supported by several open public datasets, which demonstrate state of the arts recognition accuracy and a significant reduction in network size, computational costs, and training speed. In addition to directly contributing to neuromorphic SER, this thesis proposes a novel speech-coding algorithm based on the working mechanism of humans auditory organ system. The algorithm mimics the functionality of the cochlea and successfully provides an alternative method of event-data acquisition for audio-based data. The algorithm is then further simplified and extended into an application of speech enhancement which is jointly used in the proposed SER system. This speech-enhancement method uses the lateral inhibition mechanism as a frequency coincidence detector to remove uncorrelated noise in the time-frequency spectrum. The method is shown to be effective by experiments for up to six types of noise