Seminars

Dec
9
Fri
Mark Hasegawa-Johnson (University of Illinois Urbana-Champaign) “Zipf’s Law Suggests a Three-Pronged Approach to Inclusive Speech Recognition” @ Hackerman Hall B17
Dec 9 @ 12:00 pm – 1:15 pm

Abstract

Zipf’s law is commonly glossed by the aphorism “infrequent words are frequent,” but in practice, it has often meant that there are three types of words: frequent, infrequent, and out-of-vocabulary (OOV). Speech recognition solved the problem of frequent words in 1970 (with dynamic time warping).  Hidden Markov models worked well for moderately infrequent words, but the problem of OOV words was not solved until sequence-to-sequence neural nets de-reified the concept of a word.  Many other social phenomena follow power-law distributions.  The number of native speakers of the N’th most spoken language, for example, is 1.44 billion over N to the 1.09.  In languages with sufficient data, we have shown that monolingual pre-training outperforms multilingual pre-training.  In less-frequent languages, multilingual knowledge transfer can significantly reduce phone error rates.  In languages with no training data, unsupervised ASR methods can be proven to converge, as long as the eigenvalues of the language model are sufficiently well separated to be measurable. Other systems of social categorization may follow similar power-law distributions.  Disability, for example, can cause speech patterns that were never seen in the training database, but not all disabilities need do so.  The inability of speech technology to work for people with even common disabilities is probably caused by a lack of data, and can probably be solved by finding better modes of interaction between technology researchers and the communities served by technology.

Biography

Mark Hasegawa-Johnson is a William L. Everitt Faculty Fellow of Electrical and Computer Engineering at the University of Illinois in Urbana-Champaign.  He has published research in speech production and perception, source separation, voice conversion, and low-resource automatic speech recognition.

Sep
8
Fri
Daniel Khashabi (Johns Hopkins University) “Building More Helpful Language Models” @ Hackerman Hall B17
Sep 8 @ 12:00 pm – 1:15 pm

Abstract

The arms race to build increasingly larger, powerful language models (LMs) in the past year has been remarkable. Yet incorporating LMs effectively into practical applications that facilitate manual workflows remains challenging. I will discuss LMs’ limiting factors and our efforts to overcome them. I will start with challenges surrounding efficient and robust LM alignment. I will share insights from our recent paper “Self-Instruct” (ACL 2023), where we used vanilla (unaligned) LMs for aligning itself, an approach that has yielded some success. Then, I will move on to the challenge of tracing the output of LMs to reliable sources, a weakness that makes them prone to hallucinations. I will discuss our recent approach of ‘according-to’ prompting, which steers LMs to quote directly from sources observed in its pre-training. If time permits, I will discuss our ongoing project to adapt LMs to interact with web pages. Throughout the presentation, I will highlight our progress, and end with questions about our future progress.

Biography

Daniel Khashabi is an assistant professor in computer science at Johns Hopkins University and the Center for Language and Speech Processing (CLSP) member. He is interested in building reasoning-driven modular NLP systems that are robust, transparent, and communicative, particularly those that use natural language as the communication medium. Khashabi has published over 40 papers on natural language processing and AI in top-tier venues. His work touches upon developing. His research has won the ACL 2023 Outstanding Paper Award, NAACL 2022 Best Paper Award, research gifts from the Allen Institute for AI, and an Amazon Research Award 2023. Before joining Hopkins, he was a postdoctoral fellow at the Allen Institute for AI (2019-2022) and obtained a Ph.D. from the University of Pennsylvania in 2019.

Center for Language and Speech Processing