BEGIN:VCALENDAR VERSION:2.0 PRODID:-//128.220.36.25//NONSGML kigkonsult.se iCalcreator 2.26.9// CALSCALE:GREGORIAN METHOD:PUBLISH X-FROM-URL:https://www.clsp.jhu.edu X-WR-TIMEZONE:America/New_York BEGIN:VTIMEZONE TZID:America/New_York X-LIC-LOCATION:America/New_York BEGIN:STANDARD DTSTART:20231105T020000 TZOFFSETFROM:-0400 TZOFFSETTO:-0500 RDATE:20241103T020000 TZNAME:EST END:STANDARD BEGIN:DAYLIGHT DTSTART:20240310T020000 TZOFFSETFROM:-0500 TZOFFSETTO:-0400 RDATE:20250309T020000 TZNAME:EDT END:DAYLIGHT END:VTIMEZONE BEGIN:VEVENT UID:ai1ec-21031@www.clsp.jhu.edu DTSTAMP:20240329T155724Z CATEGORIES;LANGUAGE=en-US:Seminars CONTACT: DESCRIPTION:
Abstract
\nMost p eople take for granted that when they speak\, they will be heard and under stood. But for the millions who live with speech impairments caused by phy sical or neurological conditions\, trying to communicate with others can b e difficult and lead to frustration. While there have been a great number of recent advances in Automatic Speech Recognition (ASR) technologies\, th ese interfaces can be inaccessible for those with speech impairments.
\nIn this talk\, we will present Parrotron\, an end -to-end-trained speech-to-speech conversion model that maps an input spect rogram directly to another spectrogram\, without utilizing any intermediat e discrete representation. The system is also trained to emit words in add ition to a spectrogram\, in parallel. We demonstrate that this model can be trained to normalize speech from any speaker regardless of accent\, pr osody\, and background noise\, into the voice of a single canonical target speaker with a fixed accent and consistent articulation and prosody. We f urther show that this normalization model can be adapted to normalize high ly atypical speech from speakers with a variety of speech impairments (due to\, ALS\, Cerebral-Palsy\, Deafness\, Stroke\, Brain Injury\, etc.) \, resulting in significant improvements in intelligibility and naturalness\, measured via a speech recognizer and listening tests. Finally\, demonstra ting the utility of this model on other speech tasks\, we show that the sa me model architecture can be trained to perform a speech separation task.< /p>\n
Dimitri will give a brief description of some key moments in development of speech recognition algorithms that he was in volved in and their applications to YouTube closed captions\, Live Transc ribe and wearable subtitles.
\nFadi will then sp eak about the development of Parrotron.
\nBiographies
\nDimitri Kanevsky started his career at Google working on speech recognition algorithms. Prior to joining Google\, Dimitr i was a Research staff member in the Speech Algorithms Department at IBM . Prior to IBM\, he worked at a number of centers for higher mathematics\, including Max Planck Institute in Germany and the Institute for Advanced Studies in Princeton. He currently holds 295 US patents and was Master Inv entor at IBM. MIT Technology Review recognized Dimitri conversational biom etrics based security patent as one of five most influential patents for 2 003. In 2012 Dimitri was honored at the White House as a Champion of Chang e for his efforts to advance access to science\, technology\, engineering\ , and math.
\nFadi Biadsy is a senior staff researc h scientist at Google NY for the past ten years. He has been exploring and leading multiple projects at Google\, including speech recognition\, spee ch conversion\, language modeling\, and semantic understanding. He receiv ed his PhD from Columbia University in 2011. At Columbia\, he researched a variety of speech and language processing projects including\, dialect an d accent recognition\, speech recognition\, charismatic speech and questio n answering. He holds a BSc and MSc in mathematics and computer science. He worked on handwriting recognition during his masters degree and he work ed as a senior software developer for five years at Dalet digital media sy stems building multimedia broadcasting systems.
DTSTART;TZID=America/New_York:20211105T120000 DTEND;TZID=America/New_York:20211105T131500 LOCATION:Hackerman Hall B17 @ 3400 N. Charles Street\, Baltimore\, MD 21218 SEQUENCE:0 SUMMARY:Fadi Biadsy and Dimitri Kanevsky (Google) “Speech Recognition: From Speaker Dependent to Speaker Independent to Full Personalization” “Parrot ron: A Unified E2E Speech-to Speech Conversion and ASR Model for Atypical Speech” URL:https://www.clsp.jhu.edu/events/fadi-biadsy-and-dimitri-kanevsky-google / X-COST-TYPE:free X-TAGS;LANGUAGE=en-US:2021\,Biadsy and Kanevsky\,November END:VEVENT BEGIN:VEVENT UID:ai1ec-21041@www.clsp.jhu.edu DTSTAMP:20240329T155724Z CATEGORIES;LANGUAGE=en-US:Seminars CONTACT: DESCRIPTION:Abstract
\nNarration is a universal h uman practice that serves as a key site of education\, collective memory\, fostering social belief systems\, and furthering human creativity. Recent studies in economics (Shiller\, 2020)\, climate science (Bushell et al.\, 2017)\, political polarization (Kubin et al.\, 2021)\, and mental health (Adler et al.\, 2016) suggest an emerging interdisciplinary consensus that narrative is a central concept for understanding human behavior and belie fs. For close to half a century\, the field of narratology has developed a rich set of theoretical frameworks for understanding narrative. And yet t hese theories have largely gone untested on large\, heterogenous collectio ns of texts. Scholars continue to generate schemas by extrapolating from s mall numbers of manually observed documents. In this talk\, I will discuss how we can use machine learning to develop data-driven theories of narrat ion to better understand what Labov and Waletzky called “the simplest and most fundamental narrative structures.” How can machine learning help us a pproach what we might call a minimal theory of narrativity?
\nAndrew Piper is Professor and William Dawson Scholar in the Department of Languages\, Literatures\, and Cultures at McGill University. He is the director of _.t xtlab
\n\na laboratory for cultural analytics\, and editor of the /Journal of Cultural Analytics/\, an open-access journal dedicated to the computational study of culture. He is the author of numerous books and articles on the relatio nship of technology and reading\, including /Book Was There: Reading in El ectronic Times/(Chicago 2012)\, /Enumerations: Data and Literary Study/(Ch icago 2018)\, and most recently\, /Can We Be Wrong? The Problem of Textual Evidence in a Time of Data/(Cambridge 2020).
DTSTART;TZID=America/New_York:20211112T120000 DTEND;TZID=America/New_York:20211112T131500 LOCATION:Hackerman Hall B17 @ 3400 N. Charles Street\, Baltimore\, MD 21218 SEQUENCE:0 SUMMARY:Andrew Piper (McGill University) ” How can we use machine learning to understand narration?” URL:https://www.clsp.jhu.edu/events/andrew-piper-mcgill-university-how-can- we-use-machine-learning-to-understand-narration/ X-COST-TYPE:free X-TAGS;LANGUAGE=en-US:2021\,November\,Piper END:VEVENT BEGIN:VEVENT UID:ai1ec-21057@www.clsp.jhu.edu DTSTAMP:20240329T155724Z CATEGORIES;LANGUAGE=en-US:Seminars CONTACT: DESCRIPTION:Abstract
\nThis talk will outline the major challenging in porting mainstream speech technology to the domain o f clinical applications\; in particular\, the need for personalised system s\, the challenge of working in an inherently sparse data domain and devel oping meaningful collaborations with all stakeholders. The talk will give an overview of recent state-of-the-art research from current projects incl uding in the areas of recognition of disordered speech\, automatic process ing of conversations and the automatic detection and tracking of paralingu istic information at the University of Sheffield (UK)’s Speech and Hearing (SPandH) & Healthcare lab.
\nBiography
\nHei di is a Senior Lecturer (associate professor) in Computer Science at the U niversity of Sheffield\, United Kingdom. Her research interests are on the application of AI-based voice technologies to healthcare. In particular\, the detection and monitoring of people’s physical and mental health inclu ding verbal and non-verbal traits for expressions of emotion\, anxiety\, d epression and neurodegenerative conditions in e.g.\, therapeutic or diagno stic settings.
DTSTART;TZID=America/New_York:20211119T120000 DTEND;TZID=America/New_York:20211119T131500 LOCATION:Hackerman Hall B17 @ 3400 N. Charles Street\, Baltimore\, MD 21218 SEQUENCE:0 SUMMARY:Heidi Christensen (University of Sheffield\, UK) Virtual Seminar “A utomated Processing of Pathological Speech: Recent Work and Ongoing Challe nges” URL:https://www.clsp.jhu.edu/events/heidi-christensen-university-of-sheffie ld-uk-virtual-seminar-automated-processing-of-pathological-speech-recent-w ork-and-ongoing-challenges/ X-COST-TYPE:free X-TAGS;LANGUAGE=en-US:2021\,Christensen\,November END:VEVENT BEGIN:VEVENT UID:ai1ec-21275@www.clsp.jhu.edu DTSTAMP:20240329T155724Z CATEGORIES;LANGUAGE=en-US:Student Seminars CONTACT: DESCRIPTION:Abstract
\n\n\n\n\nAutomatic discovery of phon e or word-like units is one of the core objectives in zero-resource speech processing. Recent attempts employ contrastive predictive coding (CPC)\, where the model learns representations by predicting the next frame given past context. However\, CPC only looks at the audio signal’s structure at the frame level. The speech structure exists beyond frame-level\, i.e.\, a t phone level or even higher. We propose a segmental contrastive predictiv e coding (SCPC) framework to learn from the signal structure at both the f rame and phone levels.\n\n\nSCPC is a hierarchical model with three stages trained in an end-to-end m anner. In the first stage\, the model predicts future feature frames and e xtracts frame-level representation from the raw waveform. In the second st age\, a differentiable boundary detector finds variable-length segments. I n the last stage\, the model predicts future segments to learn segment rep resentations. Experiments show that our model outperforms existing phone a nd word segmentation methods on TIMIT and Buckeye datasets.
Abstract
\nVoice conversion (VC) is a significant aspect of artificial intelligence. It is the study of how to convert one’s voice to sound like that of another without changing the lin guistic content. Voice conversion belongs to a general technical field of speech synthesis\, which converts text to speech or changes the properties of speech\, for example\, voice identity\, emotion\, and accents. Voice c onversion involves multiple speech processing techniques\, such as speech analysis\, spectral conversion\, prosody conversion\, speaker characteriza tion\, and vocoding. With the recent advances in theory and practice\, we are now able to produce human-like voice quality with high speaker similar ity. In this talk\, Dr. Sisman will present the recent advances in voice c onversion and discuss their promise and limitations. Dr. Sisman will also provide a summary of the available resources for expressive voice conversi on research.
\nBiography
\nDr. Berrak Sisman (Member\, IEEE) received the Ph.D. degree in electrical and computer engin eering from National University of Singapore in 2020\, fully funded by A*S TAR Graduate Academy under Singapore International Graduate Award (SINGA). She is currently working as a tenure-track Assistant Professor at the Eri k Jonsson School Department of Electrical and Computer Engineering at Univ ersity of Texas at Dallas\, United States. Prior to joining UT Dallas\, sh e was a faculty member at Singapore University of Technology and Design (2 020-2022). She was a Postdoctoral Research Fellow at the National Universi ty of Singapore (2019-2020). She was an exchange doctoral student at the U niversity of Edinburgh and a visiting scholar at The Centre for Speech Tec hnology Research (CSTR)\, University of Edinburgh (2019). She was a visiti ng researcher at RIKEN Advanced Intelligence Project in Japan (2018). Her research is focused on machine learning\, signal processing\, emotion\, sp eech synthesis and voice conversion.
\nDr. Sisman has served as the Area Chair at INTERSPEECH 2021\, INTERSPEECH 2022\, IEEE SLT 2022 and as t he Publication Chair at ICASSP 2022. She has been elected as a member of t he IEEE Speech and Language Processing Technical Committee (SLTC) in the a rea of Speech Synthesis for the term from January 2022 to December 2024. S he plays leadership roles in conference organizations and active in techni cal committees. She has served as the General Coordinator of the Student A dvisory Committee (SAC) of International Speech Communication Association (ISCA).
DTSTART;TZID=America/New_York:20221104T120000 DTEND;TZID=America/New_York:20221104T131500 LOCATION:Hackerman Hall B17 @ 3400 N. Charles Street\, Baltimore\, MD 21218 SEQUENCE:0 SUMMARY:Berrak Sisman (University of Texas at Dallas) “Speech Synthesis and Voice Conversion: Machine Learning can Mimic Anyone’s Voice” URL:https://www.clsp.jhu.edu/events/berrak-sisman-university-of-texas-at-da llas/ X-COST-TYPE:free X-TAGS;LANGUAGE=en-US:2022\,November\,Sisman END:VEVENT BEGIN:VEVENT UID:ai1ec-22408@www.clsp.jhu.edu DTSTAMP:20240329T155724Z CATEGORIES;LANGUAGE=en-US:Seminars CONTACT: DESCRIPTION:Abstract
\nAbstract
\nDriven by the goal of erad icating language barriers on a global scale\, machine translation has soli dified itself as a key focus of artificial intelligence research today. Ho wever\, such efforts have coalesced around a small subset of languages\, l eaving behind the vast majority of mostly low-resource languages. What doe s it take to break the 200 language barrier while ensuring safe\, high-qua lity results\, all while keeping ethical considerations in mind? In this t alk\, I introduce No Language Left Behind\, an initiative to break languag e barriers for low-resource languages. In No Language Left Behind\, we too k on the low-resource language translation challenge by first contextualiz ing the need for translation support through exploratory interviews with n ative speakers. Then\, we created datasets and models aimed at narrowing t he performance gap between low and high-resource languages. We proposed mu ltiple architectural and training improvements to counteract overfitting w hile training on thousands of tasks. Critically\, we evaluated the perform ance of over 40\,000 different translation directions using a human-transl ated benchmark\, Flores-200\, and combined human evaluation with a novel t oxicity benchmark covering all languages in Flores-200 to assess translati on safety. Our model achieves an improvement of 44% BLEU relative to the p revious state-of-the-art\, laying important groundwork towards realizing a universal translation system in an open-source manner.
\nBi ography
\nAngela is a research scientis t at Meta AI Research in New York\, focusing on supporting efforts in spee ch and language research. Recent projects include No Language Left Behind (https://ai.facebook.com/r esearch/no-language-left-behind/) and Universal Speech Translation for Unwritten Languages (https://ai.faceb ook.com/blog/ai-translation-hokkien/). Before translation\, Angela pre viously focused on research in on-device models for NLP and computer visio n and text generation.
\nDTSTART;TZID=America/New_York:20221118T120000 DTEND;TZID=America/New_York:20221118T131500 LOCATION:Hackerman Hall B17 @ 3400 N. Charles Street\, Baltimore\, MD 21218 SEQUENCE:0 SUMMARY:Angela Fan (Meta AI Research) “No Language Left Behind: Scaling Hu man-Centered Machine Translation” URL:https://www.clsp.jhu.edu/events/angela-fan-facebook/ X-COST-TYPE:free X-TAGS;LANGUAGE=en-US:2022\,Fan\,November END:VEVENT BEGIN:VEVENT UID:ai1ec-23304@www.clsp.jhu.edu DTSTAMP:20240329T155724Z CATEGORIES;LANGUAGE=en-US:Seminars CONTACT: DESCRIPTION:
Abstract
\nTransformers are essential to pretraining. As we approach 5 years of BERT\, the connection between a ttention as architecture and transfer learning remains key to this central thread in NLP. Other architectures such as CNNs and RNNs have been used t o replicate pretraining results\, but these either fail to reach the same accuracy or require supplemental attention layers. This work revisits the semanal BERT result and considers pretraining without attention. We consid er replacing self-attention layers with recently developed approach for lo ng-range sequence modeling and transformer architecture variants. Specific ally\, inspired by recent papers like the structured space space sequence model (S4)\, we use simple routing layers based on state-space models (SSM ) and a bidirectional model architecture based on multiplicative gating. W e discuss the results of the proposed Bidirectional Gated SSM (BiGS) and p resent a range of analysis into its properties. Results show that architec ture does seem to have a notable impact on downstream performance and a di fferent inductive bias that is worth exploring further.
\nBi ography
\nAbstract
\nAbstract
\nMultil ingual machine translation has proven immensely useful for both parameter efficiency and overall performance for many language pairs via complete pa rameter sharing. However\, some language pairs in multilingual models can see worse performance than in bilingual models\, especially in the one-to- many translation setting. Motivated by their empirical differences\, we ex amine the geometric differences in representations from bilingual models v ersus those from one-to-many multilingual models. Specifically\, we measur e the isotropy of these representations using intrinsic dimensionality and IsoScore\, in order to measure how these representations utilize the dime nsions in their underlying vector space. We find that for a given language pair\, its multilingual model decoder representations are consistently le ss isotropic than comparable bilingual model decoder representations. Addi tionally\, we show that much of this anisotropy in multilingual decoder re presentations can be attributed to modeling language-specific information\ , therefore limiting remaining representational capacity.
DTSTART;TZID=America/New_York:20231106T120000 DTEND;TZID=America/New_York:20231106T131500 LOCATION:Hackerman Hall B17 @ 3400 N. Charles Street\, Baltimore\, MD 21218 SEQUENCE:0 SUMMARY:Student Seminar – Neha Verma “Exploring Geometric Representational Disparities Between Multilingual and Bilingual Translation Models” URL:https://www.clsp.jhu.edu/events/student-seminar-neha-verma-exploring-ge ometric-representational-disparities-between-multilingual-and-bilingual-tr anslation-models/ X-COST-TYPE:free X-TAGS;LANGUAGE=en-US:2023\,November\,Verma END:VEVENT BEGIN:VEVENT UID:ai1ec-24157@www.clsp.jhu.edu DTSTAMP:20240329T155724Z CATEGORIES;LANGUAGE=en-US:Seminars CONTACT: DESCRIPTION:Abstract
\nIn this talk\, I will pres ent a simple extension of image-based Masked Autoencoders (MAE) to self-su pervised representation learning from audio spectrograms. Following the Tr ansformer encoder-decoder design in MAE\, our Audio-MAE first encodes audi o spectrogram patches with a high masking ratio\, feeding only the non-mas ked tokens through encoder layers. The decoder then re-orders and decodes the encoded context padded with mask tokens\, in order to reconstruct the input spectrogram. We find it beneficial to incorporate local window atten tion in the decoder\, as audio spectrograms are highly correlated in local time and frequency bands. We then fine-tune the encoder with a lower mask ing ratio on target datasets. Empirically\, Audio-MAE sets new state-of-th e-art performance on six audio and speech classification tasks\, outperfor ming other recent models that use external supervised pre-training.
\n< p>Bio\nFlorian Metze is a Research Scientist Manag er at Meta AI in New York\, supporting a team of researchers and engineers working on multi-modal (image\, video\, audio\, text) content understandi ng for Meta’s Family of Apps (Instagram\, Threads\, Facebook\, WhatsApp). He used to be an Associate Research Professor at Carnegie Mellon Universit y\, in the School of Computer Science’s Language Technologies Institute\, where he still is an Adjunct Professor. He is also a co-founder of Abridge \, a company working on extracting information from doctor patient convers ations. His work covers many areas of speech recognition and multi-media a nalysis with a focus on end-to-end deep learning. Currently\, he focuses o n multi-modal processing of videos\, and using that information to recomme nd unconnected content. In the past\, he has worked on low resource and mu lti-lingual speech processing\, speech recognition with articulatory featu res\, large-scale multi-media retrieval and summarization\, information ex traction from medical interviews\, and recognition of personality or simil ar meta-data from speech.
\nFor more information\, please see http://www.cs.cmu.edu/directory /fmetze
\nDTSTART;TZID=America/New_York:20231110T120000 DTEND;TZID=America/New_York:20231110T131500 LOCATION:Hackerman Hall B17 @ 3400 N. Charles Street\, Baltimore\, MD 21218 SEQUENCE:0 SUMMARY:Florian Metze (CMU) “Masked Autoencoders that Listen” URL:https://www.clsp.jhu.edu/events/florian-metze-cmu/ X-COST-TYPE:free X-TAGS;LANGUAGE=en-US:2023\,Metze\,November END:VEVENT BEGIN:VEVENT UID:ai1ec-24159@www.clsp.jhu.edu DTSTAMP:20240329T155724Z CATEGORIES;LANGUAGE=en-US:Student Seminars CONTACT: DESCRIPTION: DTSTART;TZID=America/New_York:20231113T120000 DTEND;TZID=America/New_York:20231113T131500 LOCATION:Hackerman Hall B17 @ 3400 N. Charles Street\, Baltimore\, MD 21218 SEQUENCE:0 SUMMARY:Student Seminar – Kate Sanders URL:https://www.clsp.jhu.edu/events/student-seminar-kate-sanders/ X-COST-TYPE:free X-TAGS;LANGUAGE=en-US:2023\,November\,Sanders END:VEVENT BEGIN:VEVENT UID:ai1ec-24163@www.clsp.jhu.edu DTSTAMP:20240329T155724Z CATEGORIES;LANGUAGE=en-US:Seminars CONTACT: DESCRIPTION:
Abstract
\nThe almost un limited multimedia content available on video-sharing websites has opened new challenges and opportunities for building robust multimodal solutions. This seminar will describe our novel multimodal architectures that (1) ar e robust to missing modalities\, (2) can identify noisy or less discrimina tive features\, and (3) can leverage unlabeled data. First\, we present a strategy that effectively combines auxiliary networks\, a transformer arch itecture\, and an optimized training mechanism for handling missing featur es. This problem is relevant since it is expected that during inference th e multimodal system will face cases with missing features due to noise or occlusion. We implement this approach for audiovisual emotion recognition achieving state-of-the-art performance. Second\, we present a multimodal f ramework for dealing with scenarios characterized by noisy or less discrim inative features. This situation is commonly observed in audiovisual autom atic speech recognition (AV-ASR) with clean speech\, where the performance often drops compared to a speech-only solution due to the variability of visual features. The proposed approach is a deep learning solution with a gating layer that diminishes the effect of noisy or uninformative visual f eatures\, keeping only useful information. The approach improves\, or at l east\, maintains performance when visual features are used. Third\, we dis cuss alternative strategies to leverage unlabeled multimodal data. A promi sing approach is to use multimodal pretext tasks that are carefully design ed to learn better representations for predicting a given task\, leveragin g the relationship between acoustic and facial features. Another approach is using multimodal ladder networks where intermediate representations are predicted across modalities using lateral connections. These models offer principled solutions to increase the generalization and robustness of com mon speech-processing tasks when using multimodal architectures. p>\n
Bio
\nCarlos Busso is a Profess or at the University of Texas at Dallas’s Electrical and Computer Engineer ing Department\, where he is also the director of the Multimodal Signal Pr ocessing (MSP) Laboratory. His research interest is in human-centered mult imodal machine intelligence and application\, with a focus on the broad ar eas of affective computing\, multimodal human-machine interfaces\, in-vehi cle active safety systems\, and machine learning methods for multimodal pr ocessing. He has worked on audio-visual emotion recognition\, analysis of emotional modulation in gestures and speech\, designing realistic human-li ke virtual characters\, and detection of driver distractions. He is a reci pient of an NSF CAREER Award. In 2014\, he received the ICMI Ten-Year Tech nical Impact Award. In 2015\, his student received the third prize IEEE IT SS Best Dissertation Award (N. Li). He also received the Hewlett Packard B est Paper Award at the IEEE ICME 2011 (with J. Jain)\, and the Best Paper Award at the AAAC ACII 2017 (with Yannakakis and Cowie). He received the B est of IEEE Transactions on Affective Computing Paper Collection in 2021 ( with R. Lotfian) and the Best Paper Award from IEEE Transactions on Affect ive Computing in 2022 (with Yannakakis and Cowie). He received the ACM ICM I Community Service Award in 2023. In 2023\, he received the Distinguished Alumni Award in the Mid-Career/Academia category by the Signal and Image Processing Institute (SIPI) at the University of Southern California. He i s currently serving as an associate editor of the IEEE Transactions on Aff ective Computing. He is an IEEE Fellow. He is a member of the ISCA\, and A AAC and a senior member of ACM.
DTSTART;TZID=America/New_York:20231117T120000 DTEND;TZID=America/New_York:20231117T131500 LOCATION:Hackerman Hall B17 @ 3400 N. Charles Street\, Baltimore\, MD 21218 SEQUENCE:0 SUMMARY:Carlos Busso (University of Texas at Dallas) “Multimodal Machine Le arning for Human-Centric Tasks” URL:https://www.clsp.jhu.edu/events/carl-busso-university-of-texas-at-dalla s-multimodal-machine-learning-for-human-centric-tasks/ X-COST-TYPE:free X-TAGS;LANGUAGE=en-US:2023\,Busso\,November END:VEVENT BEGIN:VEVENT UID:ai1ec-24165@www.clsp.jhu.edu DTSTAMP:20240329T155724Z CATEGORIES;LANGUAGE=en-US:Student Seminars CONTACT: DESCRIPTION: DTSTART;TZID=America/New_York:20231127T120000 DTEND;TZID=America/New_York:20231127T131500 LOCATION:Hackerman Hall B17 @ 3400 N. Charles Street\, Baltimore\, MD 21218 SEQUENCE:0 SUMMARY:Student Seminar – Aleem Khan URL:https://www.clsp.jhu.edu/events/student-seminar-aleem-khan/ X-COST-TYPE:free X-TAGS;LANGUAGE=en-US:2023\,Khan\,November END:VEVENT END:VCALENDAR