English Conversational Telephone Speech Recognition by Humans and Machines

One of the most difficult speech recognition tasks is accurate recognition of human to human communication. Advances in deep learning over the last few years have produced major speech recognition improvements on the representative Switchboard conversational corpus. Word error rates that just a few years ago were 14% have dropped to 8.0%, then 6.6% and most recently 5.8%, and are now believed to be within striking range of human performance. This then raises two issues - what IS human performance, and how far down can we still drive speech recognition error rates? A recent paper by Microsoft suggests that we have already achieved human performance. In trying to verify this statement, we performed an independent set of human performance measurements on two conversational tasks and found that human performance may be considerably better than what was earlier reported, giving the community a significantly harder goal to achieve. We also report on our own efforts in this area, presenting a set of acoustic and language modeling techniques that lowered the word error rate of our own English conversational telephone LVCSR system to the level of 5.5%/10.3% on the Switchboard/CallHome subsets of the Hub5 2000 evaluation, which - at least at the writing of this paper - is a new performance milestone (albeit not at what we measure to be human performance!). On the acoustic side, we use a score fusion of three models: one LSTM with multiple feature inputs, a second LSTM trained with speaker-adversarial multi-task learning and a third residual net (ResNet) with 25 convolutional layers and time-dilated convolutions. On the language modeling side, we use word and character LSTMs and convolutional WaveNet-style language models

Transfer learning of language-independent end-to-end ASR with language model fusion

This work explores better adaptation methods to low-resource languages using an external language model (LM) under the framework of transfer learning. We first build a language-independent ASR system in a unified sequence-to-sequence (S2S) architecture with a shared vocabulary among all languages. During adaptation, we perform LM fusion transfer, where an external LM is integrated into the decoder network of the attention-based S2S model in the whole adaptation stage, to effectively incorporate linguistic context of the target language. We also investigate various seed models for transfer learning. Experimental evaluations using the IARPA BABEL data set show that LM fusion transfer improves performances on all target five languages compared with simple transfer learning when the external text data is available. Our final system drastically reduces the performance gap from the hybrid systems.Comment: Accepted at ICASSP201

The Microsoft 2017 Conversational Speech Recognition System

We describe the 2017 version of Microsoft's conversational speech recognition system, in which we update our 2016 system with recent developments in neural-network-based acoustic and language modeling to further advance the state of the art on the Switchboard speech recognition task. The system adds a CNN-BLSTM acoustic model to the set of model architectures we combined previously, and includes character-based and dialog session aware LSTM language models in rescoring. For system combination we adopt a two-stage approach, whereby subsets of acoustic models are first combined at the senone/frame level, followed by a word-level voting via confusion networks. We also added a confusion network rescoring step after system combination. The resulting system yields a 5.1\% word error rate on the 2000 Switchboard evaluation set

Regularization and Compression of Deep Neural Networks

Deep neural networks (DNN) are the state-of-the-art machine learning models outperforming traditional machine learning methods in a number of domains from vision and speech to natural language understanding and autonomous control. With large amounts of data becoming available, the task performance of DNNs in these domains predictably scales with the size of the DNNs. However, in data-scarce scenarios, large DNNs overfit to the training dataset resulting in inferior performance. Additionally, in scenarios where enormous amounts of data is available, large DNNs incur large inference latencies and memory costs. Thus, while imperative for achieving state-of-the-art performances, large DNNs require large amounts of data for training and large computational resources during inference. These two problems could be mitigated by sparsely training large DNNs. Imposing sparsity constraints during training limits the capacity of the model to overfit to the training set while still being able to obtain good generalization. Sparse DNNs have most of their weights close to zero after training. Therefore, most of the weights could be removed resulting in smaller inference costs. To effectively train sparse DNNs, this thesis proposes two new sparse stochastic regularization techniques called Bridgeout and Sparseout. Furthermore, Bridgeout is used to prune convolutional neural networks for low-cost inference. Bridgeout randomly perturbs the weights of a parametric model such as a DNN. It is theoretically shown that Bridgeout constrains the weights of linear models to a sparse subspace. Empirically, Bridgeout has been shown to perform better, on image classification tasks, than state-of-the-art DNNs when the data is limited. Sparseout is an activations counter-part of Bridgeout, operating on the outputs of the neurons instead of the weights of the neurons. Theoretically, Sparseout has been shown to be a general case of the commonly used Dropout regularization method. Empirical evidence suggests that Sparseout is capable of controlling the level of activations sparsity in neural networks. This flexibility allows Sparseout to perform better than Dropout on image classification and language modelling tasks. Furthermore, using Sparseout, it is found that activation sparsity is beneficial to recurrent neural networks for language modeling but densification of activations favors convolutional neural networks for image classification. To address the problem of high computational cost during inference, this thesis evaluates Bridgeout for pruning convolutional neural networks (CNN). It is shown that recent CNN architectures such as VGG, ResNet and Wide-ResNet trained with Bridgeout are more robust to one-shot filter pruning compared to non-sparse stochastic regularization

A Review of Deep Learning Techniques for Speech Processing

The field of speech processing has undergone a transformative shift with the advent of deep learning. The use of multiple processing layers has enabled the creation of models capable of extracting intricate features from speech data. This development has paved the way for unparalleled advancements in speech recognition, text-to-speech synthesis, automatic speech recognition, and emotion recognition, propelling the performance of these tasks to unprecedented heights. The power of deep learning techniques has opened up new avenues for research and innovation in the field of speech processing, with far-reaching implications for a range of industries and applications. This review paper provides a comprehensive overview of the key deep learning models and their applications in speech-processing tasks. We begin by tracing the evolution of speech processing research, from early approaches, such as MFCC and HMM, to more recent advances in deep learning architectures, such as CNNs, RNNs, transformers, conformers, and diffusion models. We categorize the approaches and compare their strengths and weaknesses for solving speech-processing tasks. Furthermore, we extensively cover various speech-processing tasks, datasets, and benchmarks used in the literature and describe how different deep-learning networks have been utilized to tackle these tasks. Additionally, we discuss the challenges and future directions of deep learning in speech processing, including the need for more parameter-efficient, interpretable models and the potential of deep learning for multimodal speech processing. By examining the field's evolution, comparing and contrasting different approaches, and highlighting future directions and challenges, we hope to inspire further research in this exciting and rapidly advancing field

Computational Intelligence and Human- Computer Interaction: Modern Methods and Applications

The present book contains all of the articles that were accepted and published in the Special Issue of MDPI’s journal Mathematics titled "Computational Intelligence and Human–Computer Interaction: Modern Methods and Applications". This Special Issue covered a wide range of topics connected to the theory and application of different computational intelligence techniques to the domain of human–computer interaction, such as automatic speech recognition, speech processing and analysis, virtual reality, emotion-aware applications, digital storytelling, natural language processing, smart cars and devices, and online learning. We hope that this book will be interesting and useful for those working in various areas of artificial intelligence, human–computer interaction, and software engineering as well as for those who are interested in how these domains are connected in real-life situations