Spoken command recognition for robotics

In this thesis, I investigate spoken command recognition technology for robotics. While high robustness is expected, the distant and noisy conditions in which the system has to operate make the task very challenging. Unlike commercial systems which all rely on a "wake-up" word to initiate the interaction, the pipeline proposed here directly detect and recognizes commands from the continuous audio stream. In order to keep the task manageable despite low-resource conditions, I propose to focus on a limited set of commands, thus trading off flexibility of the system against robustness. Domain and speaker adaptation strategies based on a multi-task regularization paradigm are first explored. More precisely, two different methods are proposed which rely on a tied loss function which penalizes the distance between the output of several networks. The first method considers each speaker or domain as a task. A canonical task-independent network is jointly trained with task-dependent models, allowing both types of networks to improve by learning from one another. While an improvement of 3.2% on the frame error rate (FER) of the task-independent network is obtained, this only partially carried over to the phone error rate (PER), with 1.5% of improvement. Similarly, a second method explored the parallel training of the canonical network with a privileged model having access to i-vectors. This method proved less effective with only 1.2% of improvement on the FER. In order to make the developed technology more accessible, I also investigated the use of a sequence-to-sequence (S2S) architecture for command classification. The use of an attention-based encoder-decoder model reduced the classification error by 40% relative to a strong convolutional neural network (CNN)-hidden Markov model (HMM) baseline, showing the relevance of S2S architectures in such context. In order to improve the flexibility of the trained system, I also explored strategies for few-shot learning, which allow to extend the set of commands with minimum requirements in terms of data. Retraining a model on the combination of original and new commands, I managed to achieve 40.5% of accuracy on the new commands with only 10 examples for each of them. This scores goes up to 81.5% of accuracy with a larger set of 100 examples per new command. An alternative strategy, based on model adaptation achieved even better scores, with 68.8% and 88.4% of accuracy with 10 and 100 examples respectively, while being faster to train. This high performance is obtained at the expense of the original categories though, on which the accuracy deteriorated. Those results are very promising as the methods allow to easily extend an existing S2S model with minimal resources. Finally, a full spoken command recognition system (named iCubrec) has been developed for the iCub platform. The pipeline relies on a voice activity detection (VAD) system to propose a fully hand-free experience. By segmenting only regions that are likely to contain commands, the VAD module also allows to reduce greatly the computational cost of the pipeline. Command candidates are then passed to the deep neural network (DNN)-HMM command recognition system for transcription. The VoCub dataset has been specifically gathered to train a DNN-based acoustic model for our task. Through multi-condition training with the CHiME4 dataset, an accuracy of 94.5% is reached on VoCub test set. A filler model, complemented by a rejection mechanism based on a confidence score, is finally added to the system to reject non-command speech in a live demonstration of the system

Deep Spoken Keyword Spotting:An Overview

Spoken keyword spotting (KWS) deals with the identification of keywords in audio streams and has become a fast-growing technology thanks to the paradigm shift introduced by deep learning a few years ago. This has allowed the rapid embedding of deep KWS in a myriad of small electronic devices with different purposes like the activation of voice assistants. Prospects suggest a sustained growth in terms of social use of this technology. Thus, it is not surprising that deep KWS has become a hot research topic among speech scientists, who constantly look for KWS performance improvement and computational complexity reduction. This context motivates this paper, in which we conduct a literature review into deep spoken KWS to assist practitioners and researchers who are interested in this technology. Specifically, this overview has a comprehensive nature by covering a thorough analysis of deep KWS systems (which includes speech features, acoustic modeling and posterior handling), robustness methods, applications, datasets, evaluation metrics, performance of deep KWS systems and audio-visual KWS. The analysis performed in this paper allows us to identify a number of directions for future research, including directions adopted from automatic speech recognition research and directions that are unique to the problem of spoken KWS

LOW RESOURCE HIGH ACCURACY KEYWORD SPOTTING

Keyword spotting (KWS) is a task to automatically detect keywords of interest in continuous speech, which has been an active research topic for over 40 years. Recently there is a rising demand for KWS techniques in resource constrained conditions. For example, as for the year of 2016, USC Shoah Foundation covers audio-visual testimonies from survivors and other witnesses of the Holocaust in 63 countries and 39 languages, and providing search capability for those testimonies requires substantial KWS technologies in low language resource conditions, as for most languages, resources for developing KWS systems are not as rich as that for English. Despite the fact that KWS has been in the literature for a long time, KWS techniques in resource constrained conditions have not been researched extensively. In this dissertation, we improve KWS performance in two low resource conditions: low language resource condition where language specific data is inadequate, and low computation resource condition where KWS runs on computation constrained devices. For low language resource KWS, we focus on applications for speech data mining, where large vocabulary continuous speech recognition (LVCSR)-based KWS techniques are widely used. Keyword spotting for those applications are also known as keyword search (KWS) or spoken term detection (STD). A key issue for this type of KWS technique is the out-of-vocabulary (OOV) keyword problem. LVCSR-based KWS can only search for words that are defined in the LVCSR's lexicon, which is typically very small in a low language resource condition. To alleviate the OOV keyword problem, we propose a technique named "proxy keyword search" that enables us to search for OOV keywords with regular LVCSR-based KWS systems. We also develop a technique that expands LVCSR's lexicon automatically by adding hallucinated words, which increases keyword coverage and therefore improves KWS performance. Finally we explore the possibility of building LVCSR-based KWS systems with limited lexicon, or even without an expert pronunciation lexicon. For low computation resource KWS, we focus on wake-word applications, which usually run on computation constrained devices such as mobile phones or tablets. We first develop a deep neural network (DNN)-based keyword spotter, which is lightweight and accurate enough that we are able to run it on devices continuously. This keyword spotter typically requires a pre-defined keyword, such as "Okay Google". We then propose a long short-term memory (LSTM)-based feature extractor for query-by-example KWS, which enables the users to define their own keywords

Leveraging Spatiotemporal Relationships of High-frequency Activation in Human Electrocorticographic Recordings for Speech Brain-Computer-Interface

Speech production is one of the most intricate yet natural human behaviors and is most keenly appreciated when it becomes difficult or impossible; as is the case for patients suffering from locked-in syndrome. Burgeoning understanding of the various cortical representations of language has brought into question the viability of a speech neuroprosthesis using implanted electrodes. The temporal resolution of intracranial electrophysiological recordings, frequently billed as a great asset of electrocorticography (ECoG), has actually been a hindrance as speech decoders have struggled to take advantage of this timing information. There have been few demonstrations of how well a speech neuroprosthesis will realistically generalize across contexts when constructed using causal feature extraction and language models that can be applied and adapted in real-time. The research detailed in this dissertation aims primarily to characterize the spatiotemporal relationships of high frequency activity across ECoG arrays during word production. Once identified, these relationships map to motor and semantic representations of speech through the use of algorithms and classifiers that rapidly quantify these relationships in single-trials. The primary hypothesis put forward by this dissertation is that the onset, duration and temporal profile of high frequency activity in ECoG recordings is a useful feature for speech decoding. These features have rarely been used in state-of-the-art speech decoders, which tend to produce output from instantaneous high frequency power across cortical sites, or rely upon precise behavioral time-locking to take advantage of high frequency activity at several time-points relative to behavioral onset times. This hypothesis was examined in three separate studies. First, software was created that rapidly characterizes spatiotemporal relationships of neural features. Second, semantic representations of speech were examined using these spatiotemporal features. Finally, utterances were discriminated in single-trials with low latency and high accuracy using spatiotemporal matched filters in a neural keyword-spotting paradigm. Outcomes from this dissertation inform implant placement for a human speech prosthesis and provide the scientific and methodological basis to motivate further research of an implant specifically for speech-based brain-computer-interfaces

Learning from Very Few Samples: A Survey

Few sample learning (FSL) is significant and challenging in the field of machine learning. The capability of learning and generalizing from very few samples successfully is a noticeable demarcation separating artificial intelligence and human intelligence since humans can readily establish their cognition to novelty from just a single or a handful of examples whereas machine learning algorithms typically entail hundreds or thousands of supervised samples to guarantee generalization ability. Despite the long history dated back to the early 2000s and the widespread attention in recent years with booming deep learning technologies, little surveys or reviews for FSL are available until now. In this context, we extensively review 300+ papers of FSL spanning from the 2000s to 2019 and provide a timely and comprehensive survey for FSL. In this survey, we review the evolution history as well as the current progress on FSL, categorize FSL approaches into the generative model based and discriminative model based kinds in principle, and emphasize particularly on the meta learning based FSL approaches. We also summarize several recently emerging extensional topics of FSL and review the latest advances on these topics. Furthermore, we highlight the important FSL applications covering many research hotspots in computer vision, natural language processing, audio and speech, reinforcement learning and robotic, data analysis, etc. Finally, we conclude the survey with a discussion on promising trends in the hope of providing guidance and insights to follow-up researches.Comment: 30 page