activity
20192026
most citedTransformer-based Acoustic Modeling for Hybrid Speech Recognition

261 citations · 426 across the 33 of their papers we have counts for

collaborators

36 papers

cs.SD2026

HEAR: Real Voices, Real Bias: A Large-Scale Human-Recorded, Demographically Diverse Benchmark for Audio Language Models

Shen Yan, Duc Le, Irina-Elena Veliche

We introduce HEAR (Human-recorded Evaluation of Audio-LLM bias by Real speakers), a large-scale, ecologically valid benchmark comprising 87k real human audio samples from 843 demog…

eess.AS2026

MELD: Mel-Spectrogram-Based Speech Language Modeling with Discrete Latent Variables

Sung-Lin Yeh, Wei Zhou, Gil Keren +6

Recent speech language models rely on encoders that are optimized separately from autoregressive models. Since these encoders are unaware of the downstream objectives, the extracte…

eess.AS2024

Textless Streaming Speech-to-Speech Translation using Semantic Speech Tokens

Jinzheng Zhao, Niko Moritz, Egor Lakomkin +7

Cascaded speech-to-speech translation systems often suffer from the error accumulation problem and high latency, which is a result of cascaded modules whose inference delays accumu…

cs.AI2024

The Llama 3 Herd of Models

Aaron Grattafiori, Abhimanyu Dubey, Abhinav Jauhri +556

Modern artificial intelligence (AI) systems are powered by foundation models. This paper presents a new set of foundation models, called Llama 3. It is a herd of language models th…

cs.CL2024★ 1 cited

IndicVoices: Towards building an Inclusive Multilingual Speech Dataset for Indian Languages

Tahir Javed, Janki Atul Nawale, Eldho Ittan George +18

We present INDICVOICES, a dataset of natural and spontaneous speech containing a total of 7348 hours of read (9%), extempore (74%) and conversational (17%) audio from 16237 speaker…

cs.SD2023

Seq2seq for Automatic Paraphasia Detection in Aphasic Speech

Matthew Perez, Duc Le, Amrit Romana +3

Paraphasias are speech errors that are often characteristic of aphasia and they represent an important signal in assessing disease severity and subtype. Traditionally, clinicians m…

cs.CL2023

Modality Confidence Aware Training for Robust End-to-End Spoken Language Understanding

Suyoun Kim, Akshat Shrivastava, Duc Le +3

End-to-end (E2E) spoken language understanding (SLU) systems that generate a semantic parse from speech have become more promising recently. This approach uses a single model that…

cs.CL2023★ 1 cited

Text Generation with Speech Synthesis for ASR Data Augmentation

Zhuangqun Huang, Gil Keren, Ziran Jiang +11

Aiming at reducing the reliance on expensive human annotations, data synthesis for Automatic Speech Recognition (ASR) has remained an active area of research. While prior work main…

eess.AS2022

Improving Fast-slow Encoder based Transducer with Streaming Deliberation

Ke Li, Jay Mahadeokar, Jinxi Guo +5

This paper introduces a fast-slow encoder based transducer with streaming deliberation for end-to-end automatic speech recognition. We aim to improve the recognition accuracy of th…

cs.CL2022★ 1 cited

Massively Multilingual ASR on 70 Languages: Tokenization, Architecture, and Generalization Capabilities

Andros Tjandra, Nayan Singhal, David Zhang +4

End-to-end multilingual ASR has become more appealing because of several reasons such as simplifying the training and deployment process and positive performance transfer from high…

eess.AS2022

Factorized Blank Thresholding for Improved Runtime Efficiency of Neural Transducers

Duc Le, Frank Seide, Yuhao Wang +4

We show how factoring the RNN-T's output distribution can significantly reduce the computation cost and power consumption for on-device ASR inference with no loss in accuracy. With…

cs.CL2022

Joint Audio/Text Training for Transformer Rescorer of Streaming Speech Recognition

Suyoun Kim, Ke Li, Lucas Kabela +4

Recently, there has been an increasing interest in two-pass streaming end-to-end speech recognition (ASR) that incorporates a 2nd-pass rescoring model on top of the conventional 1s…

eess.AS2022★ 1 cited

Learning ASR pathways: A sparse multilingual ASR model

Mu Yang, Andros Tjandra, Chunxi Liu +3

Neural network pruning compresses automatic speech recognition (ASR) models effectively. However, in multilingual ASR, language-agnostic pruning may lead to severe performance drop…

cs.CL2022

STOP: A dataset for Spoken Task Oriented Semantic Parsing

Paden Tomasello, Akshat Shrivastava, Daniel Lazar +12

End-to-end spoken language understanding (SLU) predicts intent directly from audio using a single model. It promises to improve the performance of assistant systems by leveraging a…

eess.AS2022

An Investigation of Monotonic Transducers for Large-Scale Automatic Speech Recognition

Niko Moritz, Frank Seide, Duc Le +2

The two most popular loss functions for streaming end-to-end automatic speech recognition (ASR) are RNN-Transducer (RNN-T) and connectionist temporal classification (CTC). Between…

cs.CL2022

Deliberation Model for On-Device Spoken Language Understanding

Duc Le, Akshat Shrivastava, Paden Tomasello +4

We propose a novel deliberation-based approach to end-to-end (E2E) spoken language understanding (SLU), where a streaming automatic speech recognition (ASR) model produces the firs…

cs.CL2022

Streaming parallel transducer beam search with fast-slow cascaded encoders

Jay Mahadeokar, Yangyang Shi, Ke Li +5

Streaming ASR with strict latency constraints is required in many speech recognition applications. In order to achieve the required latency, streaming ASR models sacrifice accuracy…

cs.CL2022

Neural-FST Class Language Model for End-to-End Speech Recognition

Antoine Bruguier, Duc Le, Rohit Prabhavalkar +7

We propose Neural-FST Class Language Model (NFCLM) for end-to-end speech recognition, a novel method that combines neural network language models (NNLMs) and finite state transduce…

cs.CL2021

Scaling ASR Improves Zero and Few Shot Learning

Alex Xiao, Weiyi Zheng, Gil Keren +6

With 4.5 million hours of English speech from 10 different sources across 120 countries and models of up to 10 billion parameters, we explore the frontiers of scale for automatic s…

cs.CL2021★ 5 cited

Evaluating User Perception of Speech Recognition System Quality with Semantic Distance Metric

Suyoun Kim, Duc Le, Weiyi Zheng +6

Measuring automatic speech recognition (ASR) system quality is critical for creating user-satisfying voice-driven applications. Word Error Rate (WER) has been traditionally used to…

cs.SD2021★ 1 cited

Flexi-Transducer: Optimizing Latency, Accuracy and Compute forMulti-Domain On-Device Scenarios

Jay Mahadeokar, Yangyang Shi, Yuan Shangguan +7

Often, the storage and computational constraints of embeddeddevices demand that a single on-device ASR model serve multiple use-cases / domains. In this paper, we propose aFlexible…

cs.CL2021

Dynamic Encoder Transducer: A Flexible Solution For Trading Off Accuracy For Latency

Yangyang Shi, Varun Nagaraja, Chunyang Wu +9

We propose a dynamic encoder transducer (DET) for on-device speech recognition. One DET model scales to multiple devices with different computation capacities without retraining or…

cs.CL2021★ 5 cited

Semantic Distance: A New Metric for ASR Performance Analysis Towards Spoken Language Understanding

Suyoun Kim, Abhinav Arora, Duc Le +4

Word Error Rate (WER) has been the predominant metric used to evaluate the performance of automatic speech recognition (ASR) systems. However, WER is sometimes not a good indicator…

cs.SD2021

Dissecting User-Perceived Latency of On-Device E2E Speech Recognition

Yuan Shangguan, Rohit Prabhavalkar, Hang Su +8

As speech-enabled devices such as smartphones and smart speakers become increasingly ubiquitous, there is growing interest in building automatic speech recognition (ASR) systems th…