SYNTHESIZING DYSARTHRIC SPEECH USING MULTI-SPEAKER TTS FOR DSYARTHRIC SPEECH RECOGNITION

Dysarthria is a motor speech disorder often characterized by reduced speech intelligibility through slow, uncoordinated control of speech production muscles. Automatic Speech recognition (ASR) systems may help dysarthric talkers communicate more effectively. However, robust dysarthria-specific ASR requires a significant amount of training speech is required, which is not readily available for dysarthric talkers. In this dissertation, we investigate dysarthric speech augmentation and synthesis methods. To better understand differences in prosodic and acoustic characteristics of dysarthric spontaneous speech at varying severity levels, a comparative study between typical and dysarthric speech was conducted. These characteristics are important components for dysarthric speech modeling, synthesis, and augmentation. For augmentation, prosodic transformation and time-feature masking have been proposed. For dysarthric speech synthesis, this dissertation has introduced a modified neural multi-talker TTS by adding a dysarthria severity level coefficient and a pause insertion model to synthesize dysarthric speech for varying severity levels. In addition, we have extended this work by using a label propagation technique to create more meaningful control variables such as a continuous Respiration, Laryngeal and Tongue (RLT) parameter, even for datasets that only provide discrete dysarthria severity level information. This approach increases the controllability of the system, so we are able to generate more dysarthric speech with a broader range. To evaluate their effectiveness for synthesis of training data, dysarthria-specific speech recognition was used. Results show that a DNN-HMM model trained on additional synthetic dysarthric speech achieves WER improvement of 12.2% compared to the baseline, and that the addition of the severity level and pause insertion controls decrease WER by 6.5%, showing the effectiveness of adding these parameters. Overall results on the TORGO database demonstrate that using dysarthric synthetic speech to increase the amount of dysarthric-patterned speech for training has a significant impact on the dysarthric ASR systems

Deep Learning-Based Speech Emotion Recognition Using Librosa

Speech Emotion Recognition is a challenge of computational paralinguistic and speech processing that tries to identify and classify the emotions expressed in spoken language. The objective is to infer from a speaker's speech patterns, such as prosody, pitch, and rhythm, their emotional state, such as happiness, rage, sadness, or frustration. In the modern world, one of the most crucial marketing tactics is emotion detection. For a person, you might tailor several things in order to best fit their interests. Due to this, we made the decision to work on a project where we could identify a person's emotions based just on their speech, allowing us to handle a variety of AI-related applications. Examples include the ability of call centers to play music during tense exchanges. Another example might be a smart automobile that slows down when someone is scared or furious. In Python, we processed and extracted features from the audio files using the Librosa module. A Python library for audio and music analysis is called Librosa. It offers the fundamental components required to develop systems for retrieving music-related information. Because of this, there is a lot of potential for this kind of application in the market that would help businesses and ensure customer safety

Improving Automatic Speech Recognition on Endangered Languages

As the world moves towards a more globalized scenario, it has brought along with it the extinction of several languages. It has been estimated that over the next century, over half of the world\u27s languages will be extinct, and an alarming 43% of the world\u27s languages are at different levels of endangerment or extinction already. The survival of many of these languages depends on the pressure imposed on the dwindling speakers of these languages. Often there is a strong correlation between endangered languages and the number and quality of recordings and documentations of each. But why do we care about preserving these less prevalent languages? The behavior of cultures is often expressed in the form of speech via one\u27s native language. The memories, ideas, major events, practices, cultures and lessons learnt, both individual as well as the community\u27s, are all communicated to the outside world via language. So, language preservation is crucial to understanding the behavior of these communities. Deep learning models have been shown to dramatically improve speech recognition accuracy but require large amounts of labelled data. Unfortunately, resource constrained languages typically fall short of the necessary data for successful training. To help alleviate the problem, data augmentation techniques fabricate many new samples from each sample. The aim of this master\u27s thesis is to examine the effect of different augmentation techniques on speech recognition of resource constrained languages. The augmentation methods being experimented with are noise augmentation, pitch augmentation, speed augmentation as well as voice transformation augmentation using Generative Adversarial Networks (GANs). This thesis also examines the effectiveness of GANs in voice transformation and its limitations. The information gained from this study will further augment the collection of data, specifically, in understanding the conditions required for the data to be collected in, so that GANs can effectively perform voice transformation. Training of the original data on the Deep Speech model resulted in 95.03% WER. Training the Seneca data on a Deep Speech model that was pretrained on an English dataset, reduced the WER to 70.43%. On adding 15 augmented samples per sample, the WER reduced to 68.33%. Finally, adding 25 augmented samples per sample, the WER reduced to 48.23%. Experiments to find the best augmentation method among noise addition, pitch variation, speed variation augmentation and GAN augmentation revealed that GAN augmentation performed the best, with a WER reduction to 60.03%

음성언어 이해에서의 중의성 해소

학위논문(박사) -- 서울대학교대학원 : 공과대학 전기·정보공학부, 2022. 8. 김남수.언어의 중의성은 필연적이다. 그것은 언어가 의사 소통의 수단이지만, 모든 사람이 생각하는 어떤 개념이 완벽히 동일하게 전달될 수 없는 것에 기인한다. 이는 필연적인 요소이기도 하지만, 언어 이해에서 중의성은 종종 의사 소통의 단절이나 실패를 가져오기도 한다. 언어의 중의성에는 다양한 층위가 존재한다. 하지만, 모든 상황에서 중의성이 해소될 필요는 없다. 태스크마다, 도메인마다 다른 양상의 중의성이 존재하며, 이를 잘 정의하고 해소될 수 있는 중의성임을 파악한 후 중의적인 부분 간의 경계를 잘 정하는 것이 중요하다. 본고에서는 음성 언어 처리, 특히 의도 이해에 있어 어떤 양상의 중의성이 발생할 수 있는지 알아보고, 이를 해소하기 위한 연구를 진행한다. 이러한 현상은 다양한 언어에서 발생하지만, 그 정도 및 양상은 언어에 따라서 다르게 나타나는 경우가 많다. 우리의 연구에서 주목하는 부분은, 음성 언어에 담긴 정보량과 문자 언어의 정보량 차이로 인해 중의성이 발생하는 경우들이다. 본 연구는 운율(prosody)에 따라 문장 형식 및 의도가 다르게 표현되는 경우가 많은 한국어를 대상으로 진행된다. 한국어에서는 다양한 기능이 있는(multi-functional한) 종결어미(sentence ender), 빈번한 탈락 현상(pro-drop), 의문사 간섭(wh-intervention) 등으로 인해, 같은 텍스트가 여러 의도로 읽히는 현상이 발생하곤 한다. 이것이 의도 이해에 혼선을 가져올 수 있다는 데에 착안하여, 본 연구에서는 이러한 중의성을 먼저 정의하고, 중의적인 문장들을 감지할 수 있도록 말뭉치를 구축한다. 의도 이해를 위한 말뭉치를 구축하는 과정에서 문장의 지향성(directivity)과 수사성(rhetoricalness)이 고려된다. 이것은 음성 언어의 의도를 서술, 질문, 명령, 수사의문문, 그리고 수사명령문으로 구분하게 하는 기준이 된다. 본 연구에서는 기록된 음성 언어(spoken language)를 충분히 높은 일치도(kappa = 0.85)로 주석한 말뭉치를 이용해, 음성이 주어지지 않은 상황에서 중의적인 텍스트를 감지하는 데에 어떤 전략 혹은 언어 모델이 효과적인가를 보이고, 해당 태스크의 특징을 정성적으로 분석한다. 또한, 우리는 텍스트 층위에서만 중의성에 접근하지 않고, 실제로 음성이 주어진 상황에서 중의성 해소(disambiguation)가 가능한지를 알아보기 위해, 텍스트가 중의적인 발화들만으로 구성된 인공적인 음성 말뭉치를 설계하고 다양한 집중(attention) 기반 신경망(neural network) 모델들을 이용해 중의성을 해소한다. 이 과정에서 모델 기반 통사적/의미적 중의성 해소가 어떠한 경우에 가장 효과적인지 관찰하고, 인간의 언어 처리와 어떤 연관이 있는지에 대한 관점을 제시한다. 본 연구에서는 마지막으로, 위와 같은 절차로 의도 이해 과정에서의 중의성이 해소되었을 경우, 이를 어떻게 산업계 혹은 연구 단에서 활용할 수 있는가에 대한 간략한 로드맵을 제시한다. 텍스트에 기반한 중의성 파악과 음성 기반의 의도 이해 모듈을 통합한다면, 오류의 전파를 줄이면서도 효율적으로 중의성을 다룰 수 있는 시스템을 만들 수 있을 것이다. 이러한 시스템은 대화 매니저(dialogue manager)와 통합되어 간단한 대화(chit-chat)가 가능한 목적 지향 대화 시스템(task-oriented dialogue system)을 구축할 수도 있고, 단일 언어 조건(monolingual condition)을 넘어 음성 번역에서의 에러를 줄이는 데에 활용될 수도 있다. 우리는 본고를 통해, 운율에 민감한(prosody-sensitive) 언어에서 의도 이해를 위한 중의성 해소가 가능하며, 이를 산업 및 연구 단에서 활용할 수 있음을 보이고자 한다. 본 연구가 다른 언어 및 도메인에서도 고질적인 중의성 문제를 해소하는 데에 도움이 되길 바라며, 이를 위해 연구를 진행하는 데에 활용된 리소스, 결과물 및 코드들을 공유함으로써 학계의 발전에 이바지하고자 한다.Ambiguity in the language is inevitable. It is because, albeit language is a means of communication, a particular concept that everyone thinks of cannot be conveyed in a perfectly identical manner. As this is an inevitable factor, ambiguity in language understanding often leads to breakdown or failure of communication. There are various hierarchies of language ambiguity. However, not all ambiguity needs to be resolved. Different aspects of ambiguity exist for each domain and task, and it is crucial to define the boundary after recognizing the ambiguity that can be well-defined and resolved. In this dissertation, we investigate the types of ambiguity that appear in spoken language processing, especially in intention understanding, and conduct research to define and resolve it. Although this phenomenon occurs in various languages, its degree and aspect depend on the language investigated. The factor we focus on is cases where the ambiguity comes from the gap between the amount of information in the spoken language and the text. Here, we study the Korean language, which often shows different sentence structures and intentions depending on the prosody. In the Korean language, a text is often read with multiple intentions due to multi-functional sentence enders, frequent pro-drop, wh-intervention, etc. We first define this type of ambiguity and construct a corpus that helps detect ambiguous sentences, given that such utterances can be problematic for intention understanding. In constructing a corpus for intention understanding, we consider the directivity and rhetoricalness of a sentence. They make up a criterion for classifying the intention of spoken language into a statement, question, command, rhetorical question, and rhetorical command. Using the corpus annotated with sufficiently high agreement on a spoken language corpus, we show that colloquial corpus-based language models are effective in classifying ambiguous text given only textual data, and qualitatively analyze the characteristics of the task. We do not handle ambiguity only at the text level. To find out whether actual disambiguation is possible given a speech input, we design an artificial spoken language corpus composed only of ambiguous sentences, and resolve ambiguity with various attention-based neural network architectures. In this process, we observe that the ambiguity resolution is most effective when both textual and acoustic input co-attends each feature, especially when the audio processing module conveys attention information to the text module in a multi-hop manner. Finally, assuming the case that the ambiguity of intention understanding is resolved by proposed strategies, we present a brief roadmap of how the results can be utilized at the industry or research level. By integrating text-based ambiguity detection and speech-based intention understanding module, we can build a system that handles ambiguity efficiently while reducing error propagation. Such a system can be integrated with dialogue managers to make up a task-oriented dialogue system capable of chit-chat, or it can be used for error reduction in multilingual circumstances such as speech translation, beyond merely monolingual conditions. Throughout the dissertation, we want to show that ambiguity resolution for intention understanding in prosody-sensitive language can be achieved and can be utilized at the industry or research level. We hope that this study helps tackle chronic ambiguity issues in other languages ​​or other domains, linking linguistic science and engineering approaches.1 Introduction 1 1.1 Motivation 2 1.2 Research Goal 4 1.3 Outline of the Dissertation 5 2 Related Work 6 2.1 Spoken Language Understanding 6 2.2 Speech Act and Intention 8 2.2.1 Performatives and statements 8 2.2.2 Illocutionary act and speech act 9 2.2.3 Formal semantic approaches 11 2.3 Ambiguity of Intention Understanding in Korean 14 2.3.1 Ambiguities in language 14 2.3.2 Speech act and intention understanding in Korean 16 3 Ambiguity in Intention Understanding of Spoken Language 20 3.1 Intention Understanding and Ambiguity 20 3.2 Annotation Protocol 23 3.2.1 Fragments 24 3.2.2 Clear-cut cases 26 3.2.3 Intonation-dependent utterances 28 3.3 Data Construction . 32 3.3.1 Source scripts 32 3.3.2 Agreement 32 3.3.3 Augmentation 33 3.3.4 Train split 33 3.4 Experiments and Results 34 3.4.1 Models 34 3.4.2 Implementation 36 3.4.3 Results 37 3.5 Findings and Summary 44 3.5.1 Findings 44 3.5.2 Summary 45 4 Disambiguation of Speech Intention 47 4.1 Ambiguity Resolution 47 4.1.1 Prosody and syntax 48 4.1.2 Disambiguation with prosody 50 4.1.3 Approaches in SLU 50 4.2 Dataset Construction 51 4.2.1 Script generation 52 4.2.2 Label tagging 54 4.2.3 Recording 56 4.3 Experiments and Results 57 4.3.1 Models 57 4.3.2 Results 60 4.4 Summary 63 5 System Integration and Application 65 5.1 System Integration for Intention Identification 65 5.1.1 Proof of concept 65 5.1.2 Preliminary study 69 5.2 Application to Spoken Dialogue System 75 5.2.1 What is 'Free-running' 76 5.2.2 Omakase chatbot 76 5.3 Beyond Monolingual Approaches 84 5.3.1 Spoken language translation 85 5.3.2 Dataset 87 5.3.3 Analysis 94 5.3.4 Discussion 95 5.4 Summary 100 6 Conclusion and Future Work 103 Bibliography 105 Abstract (In Korean) 124 Acknowledgment 126박

IberSPEECH 2020: XI Jornadas en Tecnología del Habla and VII Iberian SLTech

IberSPEECH2020 is a two-day event, bringing together the best researchers and practitioners in speech and language technologies in Iberian languages to promote interaction and discussion. The organizing committee has planned a wide variety of scientific and social activities, including technical paper presentations, keynote lectures, presentation of projects, laboratories activities, recent PhD thesis, discussion panels, a round table, and awards to the best thesis and papers. The program of IberSPEECH2020 includes a total of 32 contributions that will be presented distributed among 5 oral sessions, a PhD session, and a projects session. To ensure the quality of all the contributions, each submitted paper was reviewed by three members of the scientific review committee. All the papers in the conference will be accessible through the International Speech Communication Association (ISCA) Online Archive. Paper selection was based on the scores and comments provided by the scientific review committee, which includes 73 researchers from different institutions (mainly from Spain and Portugal, but also from France, Germany, Brazil, Iran, Greece, Hungary, Czech Republic, Ucrania, Slovenia). Furthermore, it is confirmed to publish an extension of selected papers as a special issue of the Journal of Applied Sciences, “IberSPEECH 2020: Speech and Language Technologies for Iberian Languages”, published by MDPI with fully open access. In addition to regular paper sessions, the IberSPEECH2020 scientific program features the following activities: the ALBAYZIN evaluation challenge session.Red Española de Tecnologías del Habla. Universidad de Valladoli

Producing Acoustic-Prosodic Entrainment in a Robotic Learning Companion to Build Learner Rapport

abstract: With advances in automatic speech recognition, spoken dialogue systems are assuming increasingly social roles. There is a growing need for these systems to be socially responsive, capable of building rapport with users. In human-human interactions, rapport is critical to patient-doctor communication, conflict resolution, educational interactions, and social engagement. Rapport between people promotes successful collaboration, motivation, and task success. Dialogue systems which can build rapport with their user may produce similar effects, personalizing interactions to create better outcomes. This dissertation focuses on how dialogue systems can build rapport utilizing acoustic-prosodic entrainment. Acoustic-prosodic entrainment occurs when individuals adapt their acoustic-prosodic features of speech, such as tone of voice or loudness, to one another over the course of a conversation. Correlated with liking and task success, a dialogue system which entrains may enhance rapport. Entrainment, however, is very challenging to model. People entrain on different features in many ways and how to design entrainment to build rapport is unclear. The first goal of this dissertation is to explore how acoustic-prosodic entrainment can be modeled to build rapport. Towards this goal, this work presents a series of studies comparing, evaluating, and iterating on the design of entrainment, motivated and informed by human-human dialogue. These models of entrainment are implemented in the dialogue system of a robotic learning companion. Learning companions are educational agents that engage students socially to increase motivation and facilitate learning. As a learning companion’s ability to be socially responsive increases, so do vital learning outcomes. A second goal of this dissertation is to explore the effects of entrainment on concrete outcomes such as learning in interactions with robotic learning companions. This dissertation results in contributions both technical and theoretical. Technical contributions include a robust and modular dialogue system capable of producing prosodic entrainment and other socially-responsive behavior. One of the first systems of its kind, the results demonstrate that an entraining, social learning companion can positively build rapport and increase learning. This dissertation provides support for exploring phenomena like entrainment to enhance factors such as rapport and learning and provides a platform with which to explore these phenomena in future work.Dissertation/ThesisDoctoral Dissertation Computer Science 201

Dysarthric Speech Recognition and Offline Handwriting Recognition using Deep Neural Networks

Millions of people around the world are diagnosed with neurological disorders like Parkinson’s, Cerebral Palsy or Amyotrophic Lateral Sclerosis. Due to the neurological damage as the disease progresses, the person suffering from the disease loses control of muscles, along with speech deterioration. Speech deterioration is due to neuro motor condition that limits manipulation of the articulators of the vocal tract, the condition collectively called as dysarthria. Even though dysarthric speech is grammatically and syntactically correct, it is difficult for humans to understand and for Automatic Speech Recognition (ASR) systems to decipher. With the emergence of deep learning, speech recognition systems have improved a lot compared to traditional speech recognition systems, which use sophisticated preprocessing techniques to extract speech features. In this digital era there are still many documents that are handwritten many of which need to be digitized. Offline handwriting recognition involves recognizing handwritten characters from images of handwritten text (i.e. scanned documents). This is an interesting task as it involves sequence learning with computer vision. The task is more difficult than Optical Character Recognition (OCR), because handwritten letters can be written in virtually infinite different styles. This thesis proposes exploiting deep learning techniques like Convolutional Neural Networks (CNN) and Recurrent Neural Networks (RNN) for offline handwriting recognition. For speech recognition, we compare traditional methods for speech recognition with recent deep learning methods. Also, we apply speaker adaptation methods both at feature level and at parameter level to improve recognition of dysarthric speech

Dealing with linguistic mismatches for automatic speech recognition

Recent breakthroughs in automatic speech recognition (ASR) have resulted in a word error rate (WER) on par with human transcribers on the English Switchboard benchmark. However, dealing with linguistic mismatches between the training and testing data is still a significant challenge that remains unsolved. Under the monolingual environment, it is well-known that the performance of ASR systems degrades significantly when presented with the speech from speakers with different accents, dialects, and speaking styles than those encountered during system training. Under the multi-lingual environment, ASR systems trained on a source language achieve even worse performance when tested on another target language because of mismatches in terms of the number of phonemes, lexical ambiguity, and power of phonotactic constraints provided by phone-level n-grams. In order to address the issues of linguistic mismatches for current ASR systems, my dissertation investigates both knowledge-gnostic and knowledge-agnostic solutions. In the first part, classic theories relevant to acoustics and articulatory phonetics that present capability of being transferred across a dialect continuum from local dialects to another standardized language are re-visited. Experiments demonstrate the potentials that acoustic correlates in the vicinity of landmarks could help to build a bridge for dealing with mismatches across difference local or global varieties in a dialect continuum. In the second part, we design an end-to-end acoustic modeling approach based on connectionist temporal classification loss and propose to link the training of acoustics and accent altogether in a manner similar to the learning process in human speech perception. This joint model not only performed well on ASR with multiple accents but also boosted accuracies of accent identification task in comparison to separately-trained models